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-rw-r--r--sound/soc/fsl/Kconfig299
-rw-r--r--sound/soc/fsl/Makefile69
-rw-r--r--sound/soc/fsl/efika-audio-fabric.c91
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c235
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c597
-rw-r--r--sound/soc/fsl/fsl_asrc.c1016
-rw-r--r--sound/soc/fsl/fsl_asrc.h458
-rw-r--r--sound/soc/fsl/fsl_asrc_dma.c391
-rw-r--r--sound/soc/fsl/fsl_dma.c977
-rw-r--r--sound/soc/fsl/fsl_dma.h129
-rw-r--r--sound/soc/fsl/fsl_esai.c869
-rw-r--r--sound/soc/fsl/fsl_esai.h354
-rw-r--r--sound/soc/fsl/fsl_sai.c689
-rw-r--r--sound/soc/fsl/fsl_sai.h147
-rw-r--r--sound/soc/fsl/fsl_spdif.c1287
-rw-r--r--sound/soc/fsl/fsl_spdif.h199
-rw-r--r--sound/soc/fsl/fsl_ssi.c1485
-rw-r--r--sound/soc/fsl/fsl_ssi.h268
-rw-r--r--sound/soc/fsl/fsl_ssi_dbg.c163
-rw-r--r--sound/soc/fsl/fsl_utils.c91
-rw-r--r--sound/soc/fsl/fsl_utils.h25
-rw-r--r--sound/soc/fsl/imx-audmux.c378
-rw-r--r--sound/soc/fsl/imx-audmux.h11
-rw-r--r--sound/soc/fsl/imx-es8328.c233
-rw-r--r--sound/soc/fsl/imx-mc13783.c172
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c66
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c393
-rw-r--r--sound/soc/fsl/imx-pcm.h66
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c214
-rw-r--r--sound/soc/fsl/imx-spdif.c102
-rw-r--r--sound/soc/fsl/imx-ssi.c658
-rw-r--r--sound/soc/fsl/imx-ssi.h218
-rw-r--r--sound/soc/fsl/imx-wm8962.c322
-rw-r--r--sound/soc/fsl/mpc5200_dma.c511
-rw-r--r--sound/soc/fsl/mpc5200_dma.h87
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c350
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.h13
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c241
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c433
-rw-r--r--sound/soc/fsl/mx27vis-aic32x4.c234
-rw-r--r--sound/soc/fsl/p1022_ds.c442
-rw-r--r--sound/soc/fsl/p1022_rdk.c392
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c137
-rw-r--r--sound/soc/fsl/phycore-ac97.c125
-rw-r--r--sound/soc/fsl/wm1133-ev1.c292
45 files changed, 15929 insertions, 0 deletions
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
new file mode 100644
index 000000000..19c302b0d
--- /dev/null
+++ b/sound/soc/fsl/Kconfig
@@ -0,0 +1,299 @@
+menu "SoC Audio for Freescale CPUs"
+
+comment "Common SoC Audio options for Freescale CPUs:"
+
+config SND_SOC_FSL_ASRC
+ tristate "Asynchronous Sample Rate Converter (ASRC) module support"
+ select REGMAP_MMIO
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ help
+ Say Y if you want to add Asynchronous Sample Rate Converter (ASRC)
+ support for the Freescale CPUs.
+ This option is only useful for out-of-tree drivers since
+ in-tree drivers select it automatically.
+
+config SND_SOC_FSL_SAI
+ tristate "Synchronous Audio Interface (SAI) module support"
+ select REGMAP_MMIO
+ select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ help
+ Say Y if you want to add Synchronous Audio Interface (SAI)
+ support for the Freescale CPUs.
+ This option is only useful for out-of-tree drivers since
+ in-tree drivers select it automatically.
+
+config SND_SOC_FSL_SSI
+ tristate "Synchronous Serial Interface module (SSI) support"
+ select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
+ select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
+ select REGMAP_MMIO
+ help
+ Say Y if you want to add Synchronous Serial Interface (SSI)
+ support for the Freescale CPUs.
+ This option is only useful for out-of-tree drivers since
+ in-tree drivers select it automatically.
+
+config SND_SOC_FSL_SPDIF
+ tristate "Sony/Philips Digital Interface (S/PDIF) module support"
+ select REGMAP_MMIO
+ select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
+ select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
+ help
+ Say Y if you want to add Sony/Philips Digital Interface (SPDIF)
+ support for the Freescale CPUs.
+ This option is only useful for out-of-tree drivers since
+ in-tree drivers select it automatically.
+
+config SND_SOC_FSL_ESAI
+ tristate "Enhanced Serial Audio Interface (ESAI) module support"
+ select REGMAP_MMIO
+ select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
+ help
+ Say Y if you want to add Enhanced Synchronous Audio Interface
+ (ESAI) support for the Freescale CPUs.
+ This option is only useful for out-of-tree drivers since
+ in-tree drivers select it automatically.
+
+config SND_SOC_FSL_UTILS
+ tristate
+
+config SND_SOC_IMX_PCM_DMA
+ tristate
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+
+config SND_SOC_IMX_AUDMUX
+ tristate "Digital Audio Mux module support"
+ help
+ Say Y if you want to add Digital Audio Mux (AUDMUX) support
+ for the ARM i.MX CPUs.
+ This option is only useful for out-of-tree drivers since
+ in-tree drivers select it automatically.
+
+config SND_POWERPC_SOC
+ tristate "SoC Audio for Freescale PowerPC CPUs"
+ depends on FSL_SOC || PPC_MPC52xx
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the PowerPC CPUs.
+
+config SND_IMX_SOC
+ tristate "SoC Audio for Freescale i.MX CPUs"
+ depends on ARCH_MXC || COMPILE_TEST
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the i.MX CPUs.
+
+if SND_POWERPC_SOC
+
+config SND_MPC52xx_DMA
+ tristate
+
+config SND_SOC_POWERPC_DMA
+ tristate
+
+comment "SoC Audio support for Freescale PPC boards:"
+
+config SND_SOC_MPC8610_HPCD
+ tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
+ # I2C is necessary for the CS4270 driver
+ depends on MPC8610_HPCD && I2C
+ select SND_SOC_FSL_SSI
+ select SND_SOC_FSL_UTILS
+ select SND_SOC_POWERPC_DMA
+ select SND_SOC_CS4270
+ select SND_SOC_CS4270_VD33_ERRATA
+ default y if MPC8610_HPCD
+ help
+ Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
+
+config SND_SOC_P1022_DS
+ tristate "ALSA SoC support for the Freescale P1022 DS board"
+ # I2C is necessary for the WM8776 driver
+ depends on P1022_DS && I2C
+ select SND_SOC_FSL_SSI
+ select SND_SOC_FSL_UTILS
+ select SND_SOC_POWERPC_DMA
+ select SND_SOC_WM8776
+ default y if P1022_DS
+ help
+ Say Y if you want to enable audio on the Freescale P1022 DS board.
+ This will also include the Wolfson Microelectronics WM8776 codec
+ driver.
+
+config SND_SOC_P1022_RDK
+ tristate "ALSA SoC support for the Freescale / iVeia P1022 RDK board"
+ # I2C is necessary for the WM8960 driver
+ depends on P1022_RDK && I2C
+ select SND_SOC_FSL_SSI
+ select SND_SOC_FSL_UTILS
+ select SND_SOC_POWERPC_DMA
+ select SND_SOC_WM8960
+ default y if P1022_RDK
+ help
+ Say Y if you want to enable audio on the Freescale / iVeia
+ P1022 RDK board. This will also include the Wolfson
+ Microelectronics WM8960 codec driver.
+
+config SND_SOC_MPC5200_I2S
+ tristate "Freescale MPC5200 PSC in I2S mode driver"
+ depends on PPC_MPC52xx && PPC_BESTCOMM
+ select SND_MPC52xx_DMA
+ select PPC_BESTCOMM_GEN_BD
+ help
+ Say Y here to support the MPC5200 PSCs in I2S mode.
+
+config SND_SOC_MPC5200_AC97
+ tristate "Freescale MPC5200 PSC in AC97 mode driver"
+ depends on PPC_MPC52xx && PPC_BESTCOMM
+ select SND_SOC_AC97_BUS
+ select SND_MPC52xx_DMA
+ select PPC_BESTCOMM_GEN_BD
+ help
+ Say Y here to support the MPC5200 PSCs in AC97 mode.
+
+config SND_MPC52xx_SOC_PCM030
+ tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
+ depends on PPC_MPC5200_SIMPLE
+ select SND_SOC_MPC5200_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for sound on the Phytec pcm030
+ baseboard.
+
+config SND_MPC52xx_SOC_EFIKA
+ tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
+ depends on PPC_EFIKA
+ select SND_SOC_MPC5200_AC97
+ select SND_SOC_STAC9766
+ help
+ Say Y if you want to add support for sound on the Efika.
+
+endif # SND_POWERPC_SOC
+
+if SND_IMX_SOC
+
+config SND_SOC_IMX_SSI
+ tristate
+ select SND_SOC_FSL_UTILS
+
+config SND_SOC_IMX_PCM_FIQ
+ tristate
+ select FIQ
+
+comment "SoC Audio support for Freescale i.MX boards:"
+
+config SND_MXC_SOC_WM1133_EV1
+ tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted"
+ depends on MACH_MX31ADS_WM1133_EV1
+ select SND_SOC_WM8350
+ select SND_SOC_IMX_PCM_FIQ
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_SSI
+ help
+ Enable support for audio on the i.MX31ADS with the WM1133-EV1
+ PMIC board with WM8835x fitted.
+
+config SND_SOC_MX27VIS_AIC32X4
+ tristate "SoC audio support for Visstrim M10 boards"
+ depends on MACH_IMX27_VISSTRIM_M10 && I2C
+ select SND_SOC_TLV320AIC32X4
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_SSI
+ help
+ Say Y if you want to add support for SoC audio on Visstrim SM10
+ board with TLV320AIC32X4 codec.
+
+config SND_SOC_PHYCORE_AC97
+ tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
+ depends on MACH_PCM043 || MACH_PCA100
+ select SND_SOC_AC97_BUS
+ select SND_SOC_WM9712
+ select SND_SOC_IMX_PCM_FIQ
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_SSI
+ help
+ Say Y if you want to add support for SoC audio on Phytec phyCORE
+ and phyCARD boards in AC97 mode
+
+config SND_SOC_EUKREA_TLV320
+ tristate "Eukrea TLV320"
+ depends on ARCH_MXC && I2C
+ select SND_SOC_TLV320AIC23_I2C
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_SSI
+ select SND_SOC_FSL_SSI
+ select SND_SOC_IMX_PCM_DMA
+ help
+ Enable I2S based access to the TLV320AIC23B codec attached
+ to the SSI interface
+
+config SND_SOC_IMX_WM8962
+ tristate "SoC Audio support for i.MX boards with wm8962"
+ depends on OF && I2C && INPUT
+ select SND_SOC_WM8962
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_FSL_SSI
+ help
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a wm8962 codec.
+
+config SND_SOC_IMX_ES8328
+ tristate "SoC Audio support for i.MX boards with the ES8328 codec"
+ depends on OF && (I2C || SPI)
+ select SND_SOC_ES8328_I2C if I2C
+ select SND_SOC_ES8328_SPI if SPI_MASTER
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_FSL_SSI
+ help
+ Say Y if you want to add support for the ES8328 audio codec connected
+ via SSI/I2S over either SPI or I2C.
+
+config SND_SOC_IMX_SGTL5000
+ tristate "SoC Audio support for i.MX boards with sgtl5000"
+ depends on OF && I2C
+ select SND_SOC_SGTL5000
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_FSL_SSI
+ help
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a sgtl5000 codec.
+
+config SND_SOC_IMX_SPDIF
+ tristate "SoC Audio support for i.MX boards with S/PDIF"
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_FSL_SPDIF
+ help
+ SoC Audio support for i.MX boards with S/PDIF
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a S/DPDIF.
+
+config SND_SOC_IMX_MC13783
+ tristate "SoC Audio support for I.MX boards with mc13783"
+ depends on MFD_MC13XXX && ARM
+ select SND_SOC_IMX_SSI
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_MC13783
+ select SND_SOC_IMX_PCM_DMA
+
+config SND_SOC_FSL_ASOC_CARD
+ tristate "Generic ASoC Sound Card with ASRC support"
+ depends on OF && I2C
+ select SND_SOC_IMX_AUDMUX
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_FSL_ESAI
+ select SND_SOC_FSL_SAI
+ select SND_SOC_FSL_SSI
+ help
+ ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
+ ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
+ and SGTL5000.
+ Say Y if you want to add support for Freescale Generic ASoC Sound Card.
+
+endif # SND_IMX_SOC
+
+endmenu
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
new file mode 100644
index 000000000..d28dc25c9
--- /dev/null
+++ b/sound/soc/fsl/Makefile
@@ -0,0 +1,69 @@
+# MPC8610 HPCD Machine Support
+snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o
+obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o
+
+# P1022 DS Machine Support
+snd-soc-p1022-ds-objs := p1022_ds.o
+obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
+
+# P1022 RDK Machine Support
+snd-soc-p1022-rdk-objs := p1022_rdk.o
+obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
+
+# Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o
+snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
+snd-soc-fsl-sai-objs := fsl_sai.o
+snd-soc-fsl-ssi-y := fsl_ssi.o
+snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o
+snd-soc-fsl-spdif-objs := fsl_spdif.o
+snd-soc-fsl-esai-objs := fsl_esai.o
+snd-soc-fsl-utils-objs := fsl_utils.o
+snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o
+obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o
+obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
+obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
+obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
+obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o
+obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
+obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
+
+# MPC5200 Platform Support
+obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
+obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
+obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
+
+# MPC5200 Machine Support
+obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
+obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
+
+# i.MX Platform Support
+snd-soc-imx-ssi-objs := imx-ssi.o
+snd-soc-imx-audmux-objs := imx-audmux.o
+obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o
+obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
+
+obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o
+obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o
+
+# i.MX Machine Support
+snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
+snd-soc-phycore-ac97-objs := phycore-ac97.o
+snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
+snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+snd-soc-imx-es8328-objs := imx-es8328.o
+snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
+snd-soc-imx-wm8962-objs := imx-wm8962.o
+snd-soc-imx-spdif-objs := imx-spdif.o
+snd-soc-imx-mc13783-objs := imx-mc13783.o
+
+obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
+obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
+obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
+obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
+obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
+obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
+obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
+obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
+obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
new file mode 100644
index 000000000..b2acd3293
--- /dev/null
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -0,0 +1,91 @@
+/*
+ * Efika driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/stac9766.h"
+
+#define DRV_NAME "efika-audio-fabric"
+
+static struct snd_soc_dai_link efika_fabric_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 Analog",
+ .codec_dai_name = "stac9766-hifi-analog",
+ .cpu_dai_name = "mpc5200-psc-ac97.0",
+ .platform_name = "mpc5200-pcm-audio",
+ .codec_name = "stac9766-codec",
+},
+{
+ .name = "AC97",
+ .stream_name = "AC97 IEC958",
+ .codec_dai_name = "stac9766-hifi-IEC958",
+ .cpu_dai_name = "mpc5200-psc-ac97.1",
+ .platform_name = "mpc5200-pcm-audio",
+ .codec_name = "stac9766-codec",
+},
+};
+
+static struct snd_soc_card card = {
+ .name = "Efika",
+ .owner = THIS_MODULE,
+ .dai_link = efika_fabric_dai,
+ .num_links = ARRAY_SIZE(efika_fabric_dai),
+};
+
+static __init int efika_fabric_init(void)
+{
+ struct platform_device *pdev;
+ int rc;
+
+ if (!of_machine_is_compatible("bplan,efika"))
+ return -ENODEV;
+
+ pdev = platform_device_alloc("soc-audio", 1);
+ if (!pdev) {
+ pr_err("efika_fabric_init: platform_device_alloc() failed\n");
+ return -ENODEV;
+ }
+
+ platform_set_drvdata(pdev, &card);
+
+ rc = platform_device_add(pdev);
+ if (rc) {
+ pr_err("efika_fabric_init: platform_device_add() failed\n");
+ platform_device_put(pdev);
+ return -ENODEV;
+ }
+ return 0;
+}
+
+module_init(efika_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
new file mode 100644
index 000000000..e1aa3834b
--- /dev/null
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -0,0 +1,235 @@
+/*
+ * eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode
+ *
+ * Copyright 2010 Eric Bénard, Eukréa Electromatique <eric@eukrea.com>
+ *
+ * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+ * which is Copyright 2009 Simtec Electronics
+ * and on sound/soc/imx/phycore-ac97.c which is
+ * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/errno.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "imx-ssi.h"
+#include "fsl_ssi.h"
+#include "imx-audmux.h"
+
+#define CODEC_CLOCK 12000000
+
+static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+ CODEC_CLOCK, SND_SOC_CLOCK_OUT);
+ if (ret) {
+ dev_err(cpu_dai->dev,
+ "Failed to set the codec sysclk.\n");
+ return ret;
+ }
+
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
+ SND_SOC_CLOCK_IN);
+ /* fsl_ssi lacks the set_sysclk ops */
+ if (ret && ret != -EINVAL) {
+ dev_err(cpu_dai->dev,
+ "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops eukrea_tlv320_snd_ops = {
+ .hw_params = eukrea_tlv320_hw_params,
+};
+
+static struct snd_soc_dai_link eukrea_tlv320_dai = {
+ .name = "tlv320aic23",
+ .stream_name = "TLV320AIC23",
+ .codec_dai_name = "tlv320aic23-hifi",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .ops = &eukrea_tlv320_snd_ops,
+};
+
+static struct snd_soc_card eukrea_tlv320 = {
+ .owner = THIS_MODULE,
+ .dai_link = &eukrea_tlv320_dai,
+ .num_links = 1,
+};
+
+static int eukrea_tlv320_probe(struct platform_device *pdev)
+{
+ int ret;
+ int int_port = 0, ext_port;
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np = NULL, *codec_np = NULL;
+
+ eukrea_tlv320.dev = &pdev->dev;
+ if (np) {
+ ret = snd_soc_of_parse_card_name(&eukrea_tlv320,
+ "eukrea,model");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "eukrea,model node missing or invalid.\n");
+ goto err;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node,
+ "ssi-controller", 0);
+ if (!ssi_np) {
+ dev_err(&pdev->dev,
+ "ssi-controller missing or invalid.\n");
+ ret = -ENODEV;
+ goto err;
+ }
+
+ codec_np = of_parse_phandle(ssi_np, "codec-handle", 0);
+ if (codec_np)
+ eukrea_tlv320_dai.codec_of_node = codec_np;
+ else
+ dev_err(&pdev->dev, "codec-handle node missing or invalid.\n");
+
+ ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "fsl,mux-int-port node missing or invalid.\n");
+ return ret;
+ }
+ ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "fsl,mux-ext-port node missing or invalid.\n");
+ return ret;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the audmux API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+
+ eukrea_tlv320_dai.cpu_of_node = ssi_np;
+ eukrea_tlv320_dai.platform_of_node = ssi_np;
+ } else {
+ eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0";
+ eukrea_tlv320_dai.platform_name = "imx-ssi.0";
+ eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a";
+ eukrea_tlv320.name = "cpuimx-audio";
+ }
+
+ if (machine_is_eukrea_cpuimx27() ||
+ of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) {
+ imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+ IMX_AUDMUX_V1_PCR_SYN |
+ IMX_AUDMUX_V1_PCR_TFSDIR |
+ IMX_AUDMUX_V1_PCR_TCLKDIR |
+ IMX_AUDMUX_V1_PCR_RFSDIR |
+ IMX_AUDMUX_V1_PCR_RCLKDIR |
+ IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+ IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+ IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4)
+ );
+ imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4,
+ IMX_AUDMUX_V1_PCR_SYN |
+ IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+ );
+ } else if (machine_is_eukrea_cpuimx25sd() ||
+ machine_is_eukrea_cpuimx35sd() ||
+ machine_is_eukrea_cpuimx51sd() ||
+ of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) {
+ if (!np)
+ ext_port = machine_is_eukrea_cpuimx25sd() ?
+ 4 : 3;
+
+ imx_audmux_v2_configure_port(int_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port),
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)
+ );
+ imx_audmux_v2_configure_port(ext_port,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)
+ );
+ } else {
+ if (np) {
+ /* The eukrea,asoc-tlv320 driver was explicitely
+ * requested (through the device tree).
+ */
+ dev_err(&pdev->dev,
+ "Missing or invalid audmux DT node.\n");
+ return -ENODEV;
+ } else {
+ /* Return happy.
+ * We might run on a totally different machine.
+ */
+ return 0;
+ }
+ }
+
+ ret = snd_soc_register_card(&eukrea_tlv320);
+err:
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ of_node_put(ssi_np);
+
+ return ret;
+}
+
+static int eukrea_tlv320_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&eukrea_tlv320);
+
+ return 0;
+}
+
+static const struct of_device_id imx_tlv320_dt_ids[] = {
+ { .compatible = "eukrea,asoc-tlv320"},
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids);
+
+static struct platform_driver eukrea_tlv320_driver = {
+ .driver = {
+ .name = "eukrea_tlv320",
+ .of_match_table = imx_tlv320_dt_ids,
+ },
+ .probe = eukrea_tlv320_probe,
+ .remove = eukrea_tlv320_remove,
+};
+
+module_platform_driver(eukrea_tlv320_driver);
+
+MODULE_AUTHOR("Eric Bénard <eric@eukrea.com>");
+MODULE_DESCRIPTION("CPUIMX ALSA SoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:eukrea_tlv320");
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
new file mode 100644
index 000000000..de4388710
--- /dev/null
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -0,0 +1,597 @@
+/*
+ * Freescale Generic ASoC Sound Card driver with ASRC
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "fsl_esai.h"
+#include "fsl_sai.h"
+#include "imx-audmux.h"
+
+#include "../codecs/sgtl5000.h"
+#include "../codecs/wm8962.h"
+
+#define RX 0
+#define TX 1
+
+/* Default DAI format without Master and Slave flag */
+#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
+
+/**
+ * CODEC private data
+ *
+ * @mclk_freq: Clock rate of MCLK
+ * @mclk_id: MCLK (or main clock) id for set_sysclk()
+ * @fll_id: FLL (or secordary clock) id for set_sysclk()
+ * @pll_id: PLL id for set_pll()
+ */
+struct codec_priv {
+ unsigned long mclk_freq;
+ u32 mclk_id;
+ u32 fll_id;
+ u32 pll_id;
+};
+
+/**
+ * CPU private data
+ *
+ * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
+ * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
+ * @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ * @slot_width: Slot width of each frame
+ *
+ * Note: [1] for tx and [0] for rx
+ */
+struct cpu_priv {
+ unsigned long sysclk_freq[2];
+ u32 sysclk_dir[2];
+ u32 sysclk_id[2];
+ u32 slot_width;
+};
+
+/**
+ * Freescale Generic ASOC card private data
+ *
+ * @dai_link[3]: DAI link structure including normal one and DPCM link
+ * @pdev: platform device pointer
+ * @codec_priv: CODEC private data
+ * @cpu_priv: CPU private data
+ * @card: ASoC card structure
+ * @sample_rate: Current sample rate
+ * @sample_format: Current sample format
+ * @asrc_rate: ASRC sample rate used by Back-Ends
+ * @asrc_format: ASRC sample format used by Back-Ends
+ * @dai_fmt: DAI format between CPU and CODEC
+ * @name: Card name
+ */
+
+struct fsl_asoc_card_priv {
+ struct snd_soc_dai_link dai_link[3];
+ struct platform_device *pdev;
+ struct codec_priv codec_priv;
+ struct cpu_priv cpu_priv;
+ struct snd_soc_card card;
+ u32 sample_rate;
+ u32 sample_format;
+ u32 asrc_rate;
+ u32 asrc_format;
+ u32 dai_fmt;
+ char name[32];
+};
+
+/**
+ * This dapm route map exsits for DPCM link only.
+ * The other routes shall go through Device Tree.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"CPU-Playback", NULL, "ASRC-Playback"},
+ {"Playback", NULL, "CPU-Playback"},
+ {"ASRC-Capture", NULL, "CPU-Capture"},
+ {"CPU-Capture", NULL, "Capture"},
+};
+
+/* Add all possible widgets into here without being redundant */
+static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("AMIC", NULL),
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct cpu_priv *cpu_priv = &priv->cpu_priv;
+ struct device *dev = rtd->card->dev;
+ int ret;
+
+ priv->sample_rate = params_rate(params);
+ priv->sample_format = params_format(params);
+
+ /*
+ * If codec-dai is DAI Master and all configurations are already in the
+ * set_bias_level(), bypass the remaining settings in hw_params().
+ * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
+ */
+ if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM)
+ return 0;
+
+ /* Specific configurations of DAIs starts from here */
+ ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
+ cpu_priv->sysclk_freq[tx],
+ cpu_priv->sysclk_dir[tx]);
+ if (ret) {
+ dev_err(dev, "failed to set sysclk for cpu dai\n");
+ return ret;
+ }
+
+ if (cpu_priv->slot_width) {
+ ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
+ cpu_priv->slot_width);
+ if (ret) {
+ dev_err(dev, "failed to set TDM slot for cpu dai\n");
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops fsl_asoc_card_ops = {
+ .hw_params = fsl_asoc_card_hw_params,
+};
+
+static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_interval *rate;
+ struct snd_mask *mask;
+
+ rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ rate->max = rate->min = priv->asrc_rate;
+
+ mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ snd_mask_none(mask);
+ snd_mask_set(mask, priv->asrc_format);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
+ /* Default ASoC DAI Link*/
+ {
+ .name = "HiFi",
+ .stream_name = "HiFi",
+ .ops = &fsl_asoc_card_ops,
+ },
+ /* DPCM Link between Front-End and Back-End (Optional) */
+ {
+ .name = "HiFi-ASRC-FE",
+ .stream_name = "HiFi-ASRC-FE",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .dynamic = 1,
+ },
+ {
+ .name = "HiFi-ASRC-BE",
+ .stream_name = "HiFi-ASRC-BE",
+ .platform_name = "snd-soc-dummy",
+ .be_hw_params_fixup = be_hw_params_fixup,
+ .ops = &fsl_asoc_card_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .no_pcm = 1,
+ },
+};
+
+static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct codec_priv *codec_priv = &priv->codec_priv;
+ struct device *dev = card->dev;
+ unsigned int pll_out;
+ int ret;
+
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
+ break;
+
+ if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+ pll_out = priv->sample_rate * 384;
+ else
+ pll_out = priv->sample_rate * 256;
+
+ ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
+ codec_priv->mclk_id,
+ codec_priv->mclk_freq, pll_out);
+ if (ret) {
+ dev_err(dev, "failed to start FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
+ pll_out, SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
+ break;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+ codec_priv->mclk_freq,
+ SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "failed to switch away from FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
+ if (ret) {
+ dev_err(dev, "failed to stop FLL: %d\n", ret);
+ return ret;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int fsl_asoc_card_audmux_init(struct device_node *np,
+ struct fsl_asoc_card_priv *priv)
+{
+ struct device *dev = &priv->pdev->dev;
+ u32 int_ptcr = 0, ext_ptcr = 0;
+ int int_port, ext_port;
+ int ret;
+
+ ret = of_property_read_u32(np, "mux-int-port", &int_port);
+ if (ret) {
+ dev_err(dev, "mux-int-port missing or invalid\n");
+ return ret;
+ }
+ ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(dev, "mux-ext-port missing or invalid\n");
+ return ret;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the AUDMUX API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+
+ /*
+ * Use asynchronous mode (6 wires) for all cases.
+ * If only 4 wires are needed, just set SSI into
+ * synchronous mode and enable 4 PADs in IOMUX.
+ */
+ switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+ IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_RFSDIR |
+ IMX_AUDMUX_V2_PTCR_RCLKDIR |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_RCLKDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR;
+ ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+ IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+ IMX_AUDMUX_V2_PTCR_RFSDIR |
+ IMX_AUDMUX_V2_PTCR_TFSDIR;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_RFSDIR |
+ IMX_AUDMUX_V2_PTCR_TFSDIR;
+ ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+ IMX_AUDMUX_V2_PTCR_RCLKDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+ IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+ IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+ IMX_AUDMUX_V2_PTCR_RFSDIR |
+ IMX_AUDMUX_V2_PTCR_RCLKDIR |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Asynchronous mode can not be set along with RCLKDIR */
+ ret = imx_audmux_v2_configure_port(int_port, 0,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+ if (ret) {
+ dev_err(dev, "audmux internal port setup failed\n");
+ return ret;
+ }
+
+ ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+ if (ret) {
+ dev_err(dev, "audmux internal port setup failed\n");
+ return ret;
+ }
+
+ ret = imx_audmux_v2_configure_port(ext_port, 0,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(dev, "audmux external port setup failed\n");
+ return ret;
+ }
+
+ ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(dev, "audmux external port setup failed\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
+{
+ struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct codec_priv *codec_priv = &priv->codec_priv;
+ struct device *dev = card->dev;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+ codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "failed to set sysclk in %s\n", __func__);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int fsl_asoc_card_probe(struct platform_device *pdev)
+{
+ struct device_node *cpu_np, *codec_np, *asrc_np;
+ struct device_node *np = pdev->dev.of_node;
+ struct platform_device *asrc_pdev = NULL;
+ struct platform_device *cpu_pdev;
+ struct fsl_asoc_card_priv *priv;
+ struct i2c_client *codec_dev;
+ struct clk *codec_clk;
+ u32 width;
+ int ret;
+
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ cpu_np = of_parse_phandle(np, "audio-cpu", 0);
+ /* Give a chance to old DT binding */
+ if (!cpu_np)
+ cpu_np = of_parse_phandle(np, "ssi-controller", 0);
+ codec_np = of_parse_phandle(np, "audio-codec", 0);
+ if (!cpu_np || !codec_np) {
+ dev_err(&pdev->dev, "phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ cpu_pdev = of_find_device_by_node(cpu_np);
+ if (!cpu_pdev) {
+ dev_err(&pdev->dev, "failed to find CPU DAI device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ codec_dev = of_find_i2c_device_by_node(codec_np);
+ if (!codec_dev) {
+ dev_err(&pdev->dev, "failed to find codec platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ asrc_np = of_parse_phandle(np, "audio-asrc", 0);
+ if (asrc_np)
+ asrc_pdev = of_find_device_by_node(asrc_np);
+
+ /* Get the MCLK rate only, and leave it controlled by CODEC drivers */
+ codec_clk = clk_get(&codec_dev->dev, NULL);
+ if (!IS_ERR(codec_clk)) {
+ priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
+ clk_put(codec_clk);
+ }
+
+ /* Default sample rate and format, will be updated in hw_params() */
+ priv->sample_rate = 44100;
+ priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
+
+ /* Assign a default DAI format, and allow each card to overwrite it */
+ priv->dai_fmt = DAI_FMT_BASE;
+
+ /* Diversify the card configurations */
+ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
+ priv->card.set_bias_level = NULL;
+ priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
+ priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
+ priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
+ priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
+ priv->cpu_priv.slot_width = 32;
+ priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+ } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
+ priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
+ priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+ } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
+ priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
+ priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
+ priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
+ priv->codec_priv.pll_id = WM8962_FLL;
+ priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+ } else {
+ dev_err(&pdev->dev, "unknown Device Tree compatible\n");
+ return -EINVAL;
+ }
+
+ /* Common settings for corresponding Freescale CPU DAI driver */
+ if (strstr(cpu_np->name, "ssi")) {
+ /* Only SSI needs to configure AUDMUX */
+ ret = fsl_asoc_card_audmux_init(np, priv);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to init audmux\n");
+ goto asrc_fail;
+ }
+ } else if (strstr(cpu_np->name, "esai")) {
+ priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
+ priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
+ } else if (strstr(cpu_np->name, "sai")) {
+ priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
+ priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
+ }
+
+ sprintf(priv->name, "%s-audio", codec_dev->name);
+
+ /* Initialize sound card */
+ priv->pdev = pdev;
+ priv->card.dev = &pdev->dev;
+ priv->card.name = priv->name;
+ priv->card.dai_link = priv->dai_link;
+ priv->card.dapm_routes = audio_map;
+ priv->card.late_probe = fsl_asoc_card_late_probe;
+ priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
+ priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
+ priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
+
+ memcpy(priv->dai_link, fsl_asoc_card_dai,
+ sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+ ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
+ if (ret) {
+ dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
+ goto asrc_fail;
+ }
+
+ /* Normal DAI Link */
+ priv->dai_link[0].cpu_of_node = cpu_np;
+ priv->dai_link[0].codec_of_node = codec_np;
+ priv->dai_link[0].codec_dai_name = codec_dev->name;
+ priv->dai_link[0].platform_of_node = cpu_np;
+ priv->dai_link[0].dai_fmt = priv->dai_fmt;
+ priv->card.num_links = 1;
+
+ if (asrc_pdev) {
+ /* DPCM DAI Links only if ASRC exsits */
+ priv->dai_link[1].cpu_of_node = asrc_np;
+ priv->dai_link[1].platform_of_node = asrc_np;
+ priv->dai_link[2].codec_dai_name = codec_dev->name;
+ priv->dai_link[2].codec_of_node = codec_np;
+ priv->dai_link[2].cpu_of_node = cpu_np;
+ priv->dai_link[2].dai_fmt = priv->dai_fmt;
+ priv->card.num_links = 3;
+
+ ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
+ &priv->asrc_rate);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get output rate\n");
+ ret = -EINVAL;
+ goto asrc_fail;
+ }
+
+ ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get output rate\n");
+ ret = -EINVAL;
+ goto asrc_fail;
+ }
+
+ if (width == 24)
+ priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
+ else
+ priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
+ }
+
+ /* Finish card registering */
+ platform_set_drvdata(pdev, priv);
+ snd_soc_card_set_drvdata(&priv->card, priv);
+
+ ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+
+asrc_fail:
+ of_node_put(asrc_np);
+fail:
+ of_node_put(codec_np);
+ of_node_put(cpu_np);
+
+ return ret;
+}
+
+static const struct of_device_id fsl_asoc_card_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-cs42888", },
+ { .compatible = "fsl,imx-audio-sgtl5000", },
+ { .compatible = "fsl,imx-audio-wm8962", },
+ {}
+};
+
+static struct platform_driver fsl_asoc_card_driver = {
+ .probe = fsl_asoc_card_probe,
+ .driver = {
+ .name = "fsl-asoc-card",
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = fsl_asoc_card_dt_ids,
+ },
+};
+module_platform_driver(fsl_asoc_card_driver);
+
+MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
+MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
+MODULE_ALIAS("platform:fsl-asoc-card");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
new file mode 100644
index 000000000..c068494ba
--- /dev/null
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -0,0 +1,1016 @@
+/*
+ * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/dma-mapping.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/platform_data/dma-imx.h>
+#include <linux/pm_runtime.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_asrc.h"
+
+#define IDEAL_RATIO_DECIMAL_DEPTH 26
+
+#define pair_err(fmt, ...) \
+ dev_err(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
+
+#define pair_dbg(fmt, ...) \
+ dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
+
+/* Sample rates are aligned with that defined in pcm.h file */
+static const u8 process_option[][8][2] = {
+ /* 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */
+ {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */
+ {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */
+ {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */
+ {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */
+ {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */
+ {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */
+ {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */
+ {{0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */
+ {{1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */
+ {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */
+ {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */
+ {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */
+ {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */
+};
+
+/* Corresponding to process_option */
+static int supported_input_rate[] = {
+ 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200,
+ 96000, 176400, 192000,
+};
+
+static int supported_asrc_rate[] = {
+ 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000,
+};
+
+/**
+ * The following tables map the relationship between asrc_inclk/asrc_outclk in
+ * fsl_asrc.h and the registers of ASRCSR
+ */
+static unsigned char input_clk_map_imx35[] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf,
+};
+
+static unsigned char output_clk_map_imx35[] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf,
+};
+
+/* i.MX53 uses the same map for input and output */
+static unsigned char input_clk_map_imx53[] = {
+/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */
+ 0x0, 0x1, 0x2, 0x7, 0x4, 0x5, 0x6, 0x3, 0x8, 0x9, 0xa, 0xb, 0xc, 0xf, 0xe, 0xd,
+};
+
+static unsigned char output_clk_map_imx53[] = {
+/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */
+ 0x8, 0x9, 0xa, 0x7, 0xc, 0x5, 0x6, 0xb, 0x0, 0x1, 0x2, 0x3, 0x4, 0xf, 0xe, 0xd,
+};
+
+static unsigned char *clk_map[2];
+
+/**
+ * Request ASRC pair
+ *
+ * It assigns pair by the order of A->C->B because allocation of pair B,
+ * within range [ANCA, ANCA+ANCB-1], depends on the channels of pair A
+ * while pair A and pair C are comparatively independent.
+ */
+static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair)
+{
+ enum asrc_pair_index index = ASRC_INVALID_PAIR;
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ struct device *dev = &asrc_priv->pdev->dev;
+ unsigned long lock_flags;
+ int i, ret = 0;
+
+ spin_lock_irqsave(&asrc_priv->lock, lock_flags);
+
+ for (i = ASRC_PAIR_A; i < ASRC_PAIR_MAX_NUM; i++) {
+ if (asrc_priv->pair[i] != NULL)
+ continue;
+
+ index = i;
+
+ if (i != ASRC_PAIR_B)
+ break;
+ }
+
+ if (index == ASRC_INVALID_PAIR) {
+ dev_err(dev, "all pairs are busy now\n");
+ ret = -EBUSY;
+ } else if (asrc_priv->channel_avail < channels) {
+ dev_err(dev, "can't afford required channels: %d\n", channels);
+ ret = -EINVAL;
+ } else {
+ asrc_priv->channel_avail -= channels;
+ asrc_priv->pair[index] = pair;
+ pair->channels = channels;
+ pair->index = index;
+ }
+
+ spin_unlock_irqrestore(&asrc_priv->lock, lock_flags);
+
+ return ret;
+}
+
+/**
+ * Release ASRC pair
+ *
+ * It clears the resource from asrc_priv and releases the occupied channels.
+ */
+static void fsl_asrc_release_pair(struct fsl_asrc_pair *pair)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ unsigned long lock_flags;
+
+ /* Make sure the pair is disabled */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_MASK(index), 0);
+
+ spin_lock_irqsave(&asrc_priv->lock, lock_flags);
+
+ asrc_priv->channel_avail += pair->channels;
+ asrc_priv->pair[index] = NULL;
+ pair->error = 0;
+
+ spin_unlock_irqrestore(&asrc_priv->lock, lock_flags);
+}
+
+/**
+ * Configure input and output thresholds
+ */
+static void fsl_asrc_set_watermarks(struct fsl_asrc_pair *pair, u32 in, u32 out)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+
+ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index),
+ ASRMCRi_EXTTHRSHi_MASK |
+ ASRMCRi_INFIFO_THRESHOLD_MASK |
+ ASRMCRi_OUTFIFO_THRESHOLD_MASK,
+ ASRMCRi_EXTTHRSHi |
+ ASRMCRi_INFIFO_THRESHOLD(in) |
+ ASRMCRi_OUTFIFO_THRESHOLD(out));
+}
+
+/**
+ * Calculate the total divisor between asrck clock rate and sample rate
+ *
+ * It follows the formula clk_rate = samplerate * (2 ^ prescaler) * divider
+ */
+static u32 fsl_asrc_cal_asrck_divisor(struct fsl_asrc_pair *pair, u32 div)
+{
+ u32 ps;
+
+ /* Calculate the divisors: prescaler [2^0, 2^7], divder [1, 8] */
+ for (ps = 0; div > 8; ps++)
+ div >>= 1;
+
+ return ((div - 1) << ASRCDRi_AxCPi_WIDTH) | ps;
+}
+
+/**
+ * Calculate and set the ratio for Ideal Ratio mode only
+ *
+ * The ratio is a 32-bit fixed point value with 26 fractional bits.
+ */
+static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair,
+ int inrate, int outrate)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ unsigned long ratio;
+ int i;
+
+ if (!outrate) {
+ pair_err("output rate should not be zero\n");
+ return -EINVAL;
+ }
+
+ /* Calculate the intergal part of the ratio */
+ ratio = (inrate / outrate) << IDEAL_RATIO_DECIMAL_DEPTH;
+
+ /* ... and then the 26 depth decimal part */
+ inrate %= outrate;
+
+ for (i = 1; i <= IDEAL_RATIO_DECIMAL_DEPTH; i++) {
+ inrate <<= 1;
+
+ if (inrate < outrate)
+ continue;
+
+ ratio |= 1 << (IDEAL_RATIO_DECIMAL_DEPTH - i);
+ inrate -= outrate;
+
+ if (!inrate)
+ break;
+ }
+
+ regmap_write(asrc_priv->regmap, REG_ASRIDRL(index), ratio);
+ regmap_write(asrc_priv->regmap, REG_ASRIDRH(index), ratio >> 24);
+
+ return 0;
+}
+
+/**
+ * Configure the assigned ASRC pair
+ *
+ * It configures those ASRC registers according to a configuration instance
+ * of struct asrc_config which includes in/output sample rate, width, channel
+ * and clock settings.
+ */
+static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
+{
+ struct asrc_config *config = pair->config;
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ u32 inrate, outrate, indiv, outdiv;
+ u32 clk_index[2], div[2];
+ int in, out, channels;
+ struct clk *clk;
+ bool ideal;
+
+ if (!config) {
+ pair_err("invalid pair config\n");
+ return -EINVAL;
+ }
+
+ /* Validate channels */
+ if (config->channel_num < 1 || config->channel_num > 10) {
+ pair_err("does not support %d channels\n", config->channel_num);
+ return -EINVAL;
+ }
+
+ /* Validate output width */
+ if (config->output_word_width == ASRC_WIDTH_8_BIT) {
+ pair_err("does not support 8bit width output\n");
+ return -EINVAL;
+ }
+
+ inrate = config->input_sample_rate;
+ outrate = config->output_sample_rate;
+ ideal = config->inclk == INCLK_NONE;
+
+ /* Validate input and output sample rates */
+ for (in = 0; in < ARRAY_SIZE(supported_input_rate); in++)
+ if (inrate == supported_input_rate[in])
+ break;
+
+ if (in == ARRAY_SIZE(supported_input_rate)) {
+ pair_err("unsupported input sample rate: %dHz\n", inrate);
+ return -EINVAL;
+ }
+
+ for (out = 0; out < ARRAY_SIZE(supported_asrc_rate); out++)
+ if (outrate == supported_asrc_rate[out])
+ break;
+
+ if (out == ARRAY_SIZE(supported_asrc_rate)) {
+ pair_err("unsupported output sample rate: %dHz\n", outrate);
+ return -EINVAL;
+ }
+
+ /* Validate input and output clock sources */
+ clk_index[IN] = clk_map[IN][config->inclk];
+ clk_index[OUT] = clk_map[OUT][config->outclk];
+
+ /* We only have output clock for ideal ratio mode */
+ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]];
+
+ div[IN] = clk_get_rate(clk) / inrate;
+ if (div[IN] == 0) {
+ pair_err("failed to support input sample rate %dHz by asrck_%x\n",
+ inrate, clk_index[ideal ? OUT : IN]);
+ return -EINVAL;
+ }
+
+ clk = asrc_priv->asrck_clk[clk_index[OUT]];
+
+ /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */
+ if (ideal)
+ div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE;
+ else
+ div[OUT] = clk_get_rate(clk) / outrate;
+
+ if (div[OUT] == 0) {
+ pair_err("failed to support output sample rate %dHz by asrck_%x\n",
+ outrate, clk_index[OUT]);
+ return -EINVAL;
+ }
+
+ /* Set the channel number */
+ channels = config->channel_num;
+
+ if (asrc_priv->channel_bits < 4)
+ channels /= 2;
+
+ /* Update channels for current pair */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCNCR,
+ ASRCNCR_ANCi_MASK(index, asrc_priv->channel_bits),
+ ASRCNCR_ANCi(index, channels, asrc_priv->channel_bits));
+
+ /* Default setting: Automatic selection for processing mode */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ATSi_MASK(index), ASRCTR_ATS(index));
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_USRi_MASK(index), 0);
+
+ /* Set the input and output clock sources */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCSR,
+ ASRCSR_AICSi_MASK(index) | ASRCSR_AOCSi_MASK(index),
+ ASRCSR_AICS(index, clk_index[IN]) |
+ ASRCSR_AOCS(index, clk_index[OUT]));
+
+ /* Calculate the input clock divisors */
+ indiv = fsl_asrc_cal_asrck_divisor(pair, div[IN]);
+ outdiv = fsl_asrc_cal_asrck_divisor(pair, div[OUT]);
+
+ /* Suppose indiv and outdiv includes prescaler, so add its MASK too */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCDR(index),
+ ASRCDRi_AOCPi_MASK(index) | ASRCDRi_AICPi_MASK(index) |
+ ASRCDRi_AOCDi_MASK(index) | ASRCDRi_AICDi_MASK(index),
+ ASRCDRi_AOCP(index, outdiv) | ASRCDRi_AICP(index, indiv));
+
+ /* Implement word_width configurations */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR1(index),
+ ASRMCR1i_OW16_MASK | ASRMCR1i_IWD_MASK,
+ ASRMCR1i_OW16(config->output_word_width) |
+ ASRMCR1i_IWD(config->input_word_width));
+
+ /* Enable BUFFER STALL */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index),
+ ASRMCRi_BUFSTALLi_MASK, ASRMCRi_BUFSTALLi);
+
+ /* Set default thresholds for input and output FIFO */
+ fsl_asrc_set_watermarks(pair, ASRC_INPUTFIFO_THRESHOLD,
+ ASRC_INPUTFIFO_THRESHOLD);
+
+ /* Configure the followings only for Ideal Ratio mode */
+ if (!ideal)
+ return 0;
+
+ /* Clear ASTSx bit to use Ideal Ratio mode */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ATSi_MASK(index), 0);
+
+ /* Enable Ideal Ratio mode */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_IDRi_MASK(index) | ASRCTR_USRi_MASK(index),
+ ASRCTR_IDR(index) | ASRCTR_USR(index));
+
+ /* Apply configurations for pre- and post-processing */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCFG,
+ ASRCFG_PREMODi_MASK(index) | ASRCFG_POSTMODi_MASK(index),
+ ASRCFG_PREMOD(index, process_option[in][out][0]) |
+ ASRCFG_POSTMOD(index, process_option[in][out][1]));
+
+ return fsl_asrc_set_ideal_ratio(pair, inrate, outrate);
+}
+
+/**
+ * Start the assigned ASRC pair
+ *
+ * It enables the assigned pair and makes it stopped at the stall level.
+ */
+static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ int reg, retry = 10, i;
+
+ /* Enable the current pair */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_MASK(index), ASRCTR_ASRCE(index));
+
+ /* Wait for status of initialization */
+ do {
+ udelay(5);
+ regmap_read(asrc_priv->regmap, REG_ASRCFG, &reg);
+ reg &= ASRCFG_INIRQi_MASK(index);
+ } while (!reg && --retry);
+
+ /* Make the input fifo to ASRC STALL level */
+ regmap_read(asrc_priv->regmap, REG_ASRCNCR, &reg);
+ for (i = 0; i < pair->channels * 4; i++)
+ regmap_write(asrc_priv->regmap, REG_ASRDI(index), 0);
+
+ /* Enable overload interrupt */
+ regmap_write(asrc_priv->regmap, REG_ASRIER, ASRIER_AOLIE);
+}
+
+/**
+ * Stop the assigned ASRC pair
+ */
+static void fsl_asrc_stop_pair(struct fsl_asrc_pair *pair)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+
+ /* Stop the current pair */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_MASK(index), 0);
+}
+
+/**
+ * Get DMA channel according to the pair and direction.
+ */
+struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir)
+{
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ enum asrc_pair_index index = pair->index;
+ char name[4];
+
+ sprintf(name, "%cx%c", dir == IN ? 'r' : 't', index + 'a');
+
+ return dma_request_slave_channel(&asrc_priv->pdev->dev, name);
+}
+EXPORT_SYMBOL_GPL(fsl_asrc_get_dma_channel);
+
+static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai);
+ int width = snd_pcm_format_width(params_format(params));
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ unsigned int channels = params_channels(params);
+ unsigned int rate = params_rate(params);
+ struct asrc_config config;
+ int word_width, ret;
+
+ ret = fsl_asrc_request_pair(channels, pair);
+ if (ret) {
+ dev_err(dai->dev, "fail to request asrc pair\n");
+ return ret;
+ }
+
+ pair->config = &config;
+
+ if (width == 16)
+ width = ASRC_WIDTH_16_BIT;
+ else
+ width = ASRC_WIDTH_24_BIT;
+
+ if (asrc_priv->asrc_width == 16)
+ word_width = ASRC_WIDTH_16_BIT;
+ else
+ word_width = ASRC_WIDTH_24_BIT;
+
+ config.pair = pair->index;
+ config.channel_num = channels;
+ config.inclk = INCLK_NONE;
+ config.outclk = OUTCLK_ASRCK1_CLK;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ config.input_word_width = width;
+ config.output_word_width = word_width;
+ config.input_sample_rate = rate;
+ config.output_sample_rate = asrc_priv->asrc_rate;
+ } else {
+ config.input_word_width = word_width;
+ config.output_word_width = width;
+ config.input_sample_rate = asrc_priv->asrc_rate;
+ config.output_sample_rate = rate;
+ }
+
+ ret = fsl_asrc_config_pair(pair);
+ if (ret) {
+ dev_err(dai->dev, "fail to config asrc pair\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int fsl_asrc_dai_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ if (pair)
+ fsl_asrc_release_pair(pair);
+
+ return 0;
+}
+
+static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ fsl_asrc_start_pair(pair);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ fsl_asrc_stop_pair(pair);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops fsl_asrc_dai_ops = {
+ .hw_params = fsl_asrc_dai_hw_params,
+ .hw_free = fsl_asrc_dai_hw_free,
+ .trigger = fsl_asrc_dai_trigger,
+};
+
+static int fsl_asrc_dai_probe(struct snd_soc_dai *dai)
+{
+ struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_init_dma_data(dai, &asrc_priv->dma_params_tx,
+ &asrc_priv->dma_params_rx);
+
+ return 0;
+}
+
+#define FSL_ASRC_RATES SNDRV_PCM_RATE_8000_192000
+#define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE)
+
+static struct snd_soc_dai_driver fsl_asrc_dai = {
+ .probe = fsl_asrc_dai_probe,
+ .playback = {
+ .stream_name = "ASRC-Playback",
+ .channels_min = 1,
+ .channels_max = 10,
+ .rates = FSL_ASRC_RATES,
+ .formats = FSL_ASRC_FORMATS,
+ },
+ .capture = {
+ .stream_name = "ASRC-Capture",
+ .channels_min = 1,
+ .channels_max = 10,
+ .rates = FSL_ASRC_RATES,
+ .formats = FSL_ASRC_FORMATS,
+ },
+ .ops = &fsl_asrc_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_asrc_component = {
+ .name = "fsl-asrc-dai",
+};
+
+static bool fsl_asrc_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ASRCTR:
+ case REG_ASRIER:
+ case REG_ASRCNCR:
+ case REG_ASRCFG:
+ case REG_ASRCSR:
+ case REG_ASRCDR1:
+ case REG_ASRCDR2:
+ case REG_ASRSTR:
+ case REG_ASRPM1:
+ case REG_ASRPM2:
+ case REG_ASRPM3:
+ case REG_ASRPM4:
+ case REG_ASRPM5:
+ case REG_ASRTFR1:
+ case REG_ASRCCR:
+ case REG_ASRDOA:
+ case REG_ASRDOB:
+ case REG_ASRDOC:
+ case REG_ASRIDRHA:
+ case REG_ASRIDRLA:
+ case REG_ASRIDRHB:
+ case REG_ASRIDRLB:
+ case REG_ASRIDRHC:
+ case REG_ASRIDRLC:
+ case REG_ASR76K:
+ case REG_ASR56K:
+ case REG_ASRMCRA:
+ case REG_ASRFSTA:
+ case REG_ASRMCRB:
+ case REG_ASRFSTB:
+ case REG_ASRMCRC:
+ case REG_ASRFSTC:
+ case REG_ASRMCR1A:
+ case REG_ASRMCR1B:
+ case REG_ASRMCR1C:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_asrc_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ASRSTR:
+ case REG_ASRDIA:
+ case REG_ASRDIB:
+ case REG_ASRDIC:
+ case REG_ASRDOA:
+ case REG_ASRDOB:
+ case REG_ASRDOC:
+ case REG_ASRFSTA:
+ case REG_ASRFSTB:
+ case REG_ASRFSTC:
+ case REG_ASRCFG:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ASRCTR:
+ case REG_ASRIER:
+ case REG_ASRCNCR:
+ case REG_ASRCFG:
+ case REG_ASRCSR:
+ case REG_ASRCDR1:
+ case REG_ASRCDR2:
+ case REG_ASRSTR:
+ case REG_ASRPM1:
+ case REG_ASRPM2:
+ case REG_ASRPM3:
+ case REG_ASRPM4:
+ case REG_ASRPM5:
+ case REG_ASRTFR1:
+ case REG_ASRCCR:
+ case REG_ASRDIA:
+ case REG_ASRDIB:
+ case REG_ASRDIC:
+ case REG_ASRIDRHA:
+ case REG_ASRIDRLA:
+ case REG_ASRIDRHB:
+ case REG_ASRIDRLB:
+ case REG_ASRIDRHC:
+ case REG_ASRIDRLC:
+ case REG_ASR76K:
+ case REG_ASR56K:
+ case REG_ASRMCRA:
+ case REG_ASRMCRB:
+ case REG_ASRMCRC:
+ case REG_ASRMCR1A:
+ case REG_ASRMCR1B:
+ case REG_ASRMCR1C:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static struct reg_default fsl_asrc_reg[] = {
+ { REG_ASRCTR, 0x0000 }, { REG_ASRIER, 0x0000 },
+ { REG_ASRCNCR, 0x0000 }, { REG_ASRCFG, 0x0000 },
+ { REG_ASRCSR, 0x0000 }, { REG_ASRCDR1, 0x0000 },
+ { REG_ASRCDR2, 0x0000 }, { REG_ASRSTR, 0x0000 },
+ { REG_ASRRA, 0x0000 }, { REG_ASRRB, 0x0000 },
+ { REG_ASRRC, 0x0000 }, { REG_ASRPM1, 0x0000 },
+ { REG_ASRPM2, 0x0000 }, { REG_ASRPM3, 0x0000 },
+ { REG_ASRPM4, 0x0000 }, { REG_ASRPM5, 0x0000 },
+ { REG_ASRTFR1, 0x0000 }, { REG_ASRCCR, 0x0000 },
+ { REG_ASRDIA, 0x0000 }, { REG_ASRDOA, 0x0000 },
+ { REG_ASRDIB, 0x0000 }, { REG_ASRDOB, 0x0000 },
+ { REG_ASRDIC, 0x0000 }, { REG_ASRDOC, 0x0000 },
+ { REG_ASRIDRHA, 0x0000 }, { REG_ASRIDRLA, 0x0000 },
+ { REG_ASRIDRHB, 0x0000 }, { REG_ASRIDRLB, 0x0000 },
+ { REG_ASRIDRHC, 0x0000 }, { REG_ASRIDRLC, 0x0000 },
+ { REG_ASR76K, 0x0A47 }, { REG_ASR56K, 0x0DF3 },
+ { REG_ASRMCRA, 0x0000 }, { REG_ASRFSTA, 0x0000 },
+ { REG_ASRMCRB, 0x0000 }, { REG_ASRFSTB, 0x0000 },
+ { REG_ASRMCRC, 0x0000 }, { REG_ASRFSTC, 0x0000 },
+ { REG_ASRMCR1A, 0x0000 }, { REG_ASRMCR1B, 0x0000 },
+ { REG_ASRMCR1C, 0x0000 },
+};
+
+static const struct regmap_config fsl_asrc_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+
+ .max_register = REG_ASRMCR1C,
+ .reg_defaults = fsl_asrc_reg,
+ .num_reg_defaults = ARRAY_SIZE(fsl_asrc_reg),
+ .readable_reg = fsl_asrc_readable_reg,
+ .volatile_reg = fsl_asrc_volatile_reg,
+ .writeable_reg = fsl_asrc_writeable_reg,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+/**
+ * Initialize ASRC registers with a default configurations
+ */
+static int fsl_asrc_init(struct fsl_asrc *asrc_priv)
+{
+ /* Halt ASRC internal FP when input FIFO needs data for pair A, B, C */
+ regmap_write(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEN);
+
+ /* Disable interrupt by default */
+ regmap_write(asrc_priv->regmap, REG_ASRIER, 0x0);
+
+ /* Apply recommended settings for parameters from Reference Manual */
+ regmap_write(asrc_priv->regmap, REG_ASRPM1, 0x7fffff);
+ regmap_write(asrc_priv->regmap, REG_ASRPM2, 0x255555);
+ regmap_write(asrc_priv->regmap, REG_ASRPM3, 0xff7280);
+ regmap_write(asrc_priv->regmap, REG_ASRPM4, 0xff7280);
+ regmap_write(asrc_priv->regmap, REG_ASRPM5, 0xff7280);
+
+ /* Base address for task queue FIFO. Set to 0x7C */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRTFR1,
+ ASRTFR1_TF_BASE_MASK, ASRTFR1_TF_BASE(0xfc));
+
+ /* Set the processing clock for 76KHz to 133M */
+ regmap_write(asrc_priv->regmap, REG_ASR76K, 0x06D6);
+
+ /* Set the processing clock for 56KHz to 133M */
+ return regmap_write(asrc_priv->regmap, REG_ASR56K, 0x0947);
+}
+
+/**
+ * Interrupt handler for ASRC
+ */
+static irqreturn_t fsl_asrc_isr(int irq, void *dev_id)
+{
+ struct fsl_asrc *asrc_priv = (struct fsl_asrc *)dev_id;
+ struct device *dev = &asrc_priv->pdev->dev;
+ enum asrc_pair_index index;
+ u32 status;
+
+ regmap_read(asrc_priv->regmap, REG_ASRSTR, &status);
+
+ /* Clean overload error */
+ regmap_write(asrc_priv->regmap, REG_ASRSTR, ASRSTR_AOLE);
+
+ /*
+ * We here use dev_dbg() for all exceptions because ASRC itself does
+ * not care if FIFO overflowed or underrun while a warning in the
+ * interrupt would result a ridged conversion.
+ */
+ for (index = ASRC_PAIR_A; index < ASRC_PAIR_MAX_NUM; index++) {
+ if (!asrc_priv->pair[index])
+ continue;
+
+ if (status & ASRSTR_ATQOL) {
+ asrc_priv->pair[index]->error |= ASRC_TASK_Q_OVERLOAD;
+ dev_dbg(dev, "ASRC Task Queue FIFO overload\n");
+ }
+
+ if (status & ASRSTR_AOOL(index)) {
+ asrc_priv->pair[index]->error |= ASRC_OUTPUT_TASK_OVERLOAD;
+ pair_dbg("Output Task Overload\n");
+ }
+
+ if (status & ASRSTR_AIOL(index)) {
+ asrc_priv->pair[index]->error |= ASRC_INPUT_TASK_OVERLOAD;
+ pair_dbg("Input Task Overload\n");
+ }
+
+ if (status & ASRSTR_AODO(index)) {
+ asrc_priv->pair[index]->error |= ASRC_OUTPUT_BUFFER_OVERFLOW;
+ pair_dbg("Output Data Buffer has overflowed\n");
+ }
+
+ if (status & ASRSTR_AIDU(index)) {
+ asrc_priv->pair[index]->error |= ASRC_INPUT_BUFFER_UNDERRUN;
+ pair_dbg("Input Data Buffer has underflowed\n");
+ }
+ }
+
+ return IRQ_HANDLED;
+}
+
+static int fsl_asrc_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct fsl_asrc *asrc_priv;
+ struct resource *res;
+ void __iomem *regs;
+ int irq, ret, i;
+ char tmp[16];
+
+ asrc_priv = devm_kzalloc(&pdev->dev, sizeof(*asrc_priv), GFP_KERNEL);
+ if (!asrc_priv)
+ return -ENOMEM;
+
+ asrc_priv->pdev = pdev;
+
+ /* Get the addresses and IRQ */
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ asrc_priv->paddr = res->start;
+
+ asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs,
+ &fsl_asrc_regmap_config);
+ if (IS_ERR(asrc_priv->regmap)) {
+ dev_err(&pdev->dev, "failed to init regmap\n");
+ return PTR_ERR(asrc_priv->regmap);
+ }
+
+ irq = platform_get_irq(pdev, 0);
+ if (irq < 0) {
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+ return irq;
+ }
+
+ ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0,
+ dev_name(&pdev->dev), asrc_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret);
+ return ret;
+ }
+
+ asrc_priv->mem_clk = devm_clk_get(&pdev->dev, "mem");
+ if (IS_ERR(asrc_priv->mem_clk)) {
+ dev_err(&pdev->dev, "failed to get mem clock\n");
+ return PTR_ERR(asrc_priv->mem_clk);
+ }
+
+ asrc_priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg");
+ if (IS_ERR(asrc_priv->ipg_clk)) {
+ dev_err(&pdev->dev, "failed to get ipg clock\n");
+ return PTR_ERR(asrc_priv->ipg_clk);
+ }
+
+ for (i = 0; i < ASRC_CLK_MAX_NUM; i++) {
+ sprintf(tmp, "asrck_%x", i);
+ asrc_priv->asrck_clk[i] = devm_clk_get(&pdev->dev, tmp);
+ if (IS_ERR(asrc_priv->asrck_clk[i])) {
+ dev_err(&pdev->dev, "failed to get %s clock\n", tmp);
+ return PTR_ERR(asrc_priv->asrck_clk[i]);
+ }
+ }
+
+ if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx35-asrc")) {
+ asrc_priv->channel_bits = 3;
+ clk_map[IN] = input_clk_map_imx35;
+ clk_map[OUT] = output_clk_map_imx35;
+ } else {
+ asrc_priv->channel_bits = 4;
+ clk_map[IN] = input_clk_map_imx53;
+ clk_map[OUT] = output_clk_map_imx53;
+ }
+
+ ret = fsl_asrc_init(asrc_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to init asrc %d\n", ret);
+ return -EINVAL;
+ }
+
+ asrc_priv->channel_avail = 10;
+
+ ret = of_property_read_u32(np, "fsl,asrc-rate",
+ &asrc_priv->asrc_rate);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get output rate\n");
+ return -EINVAL;
+ }
+
+ ret = of_property_read_u32(np, "fsl,asrc-width",
+ &asrc_priv->asrc_width);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to get output width\n");
+ return -EINVAL;
+ }
+
+ if (asrc_priv->asrc_width != 16 && asrc_priv->asrc_width != 24) {
+ dev_warn(&pdev->dev, "unsupported width, switching to 24bit\n");
+ asrc_priv->asrc_width = 24;
+ }
+
+ platform_set_drvdata(pdev, asrc_priv);
+ pm_runtime_enable(&pdev->dev);
+ spin_lock_init(&asrc_priv->lock);
+
+ ret = devm_snd_soc_register_component(&pdev->dev, &fsl_asrc_component,
+ &fsl_asrc_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register ASoC DAI\n");
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_platform(&pdev->dev, &fsl_asrc_platform);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register ASoC platform\n");
+ return ret;
+ }
+
+ dev_info(&pdev->dev, "driver registered\n");
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int fsl_asrc_runtime_resume(struct device *dev)
+{
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+ int i;
+
+ clk_prepare_enable(asrc_priv->mem_clk);
+ clk_prepare_enable(asrc_priv->ipg_clk);
+ for (i = 0; i < ASRC_CLK_MAX_NUM; i++)
+ clk_prepare_enable(asrc_priv->asrck_clk[i]);
+
+ return 0;
+}
+
+static int fsl_asrc_runtime_suspend(struct device *dev)
+{
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+ int i;
+
+ for (i = 0; i < ASRC_CLK_MAX_NUM; i++)
+ clk_disable_unprepare(asrc_priv->asrck_clk[i]);
+ clk_disable_unprepare(asrc_priv->ipg_clk);
+ clk_disable_unprepare(asrc_priv->mem_clk);
+
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+#ifdef CONFIG_PM_SLEEP
+static int fsl_asrc_suspend(struct device *dev)
+{
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+
+ regcache_cache_only(asrc_priv->regmap, true);
+ regcache_mark_dirty(asrc_priv->regmap);
+
+ return 0;
+}
+
+static int fsl_asrc_resume(struct device *dev)
+{
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+ u32 asrctr;
+
+ /* Stop all pairs provisionally */
+ regmap_read(asrc_priv->regmap, REG_ASRCTR, &asrctr);
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_ALL_MASK, 0);
+
+ /* Restore all registers */
+ regcache_cache_only(asrc_priv->regmap, false);
+ regcache_sync(asrc_priv->regmap);
+
+ /* Restart enabled pairs */
+ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR,
+ ASRCTR_ASRCEi_ALL_MASK, asrctr);
+
+ return 0;
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops fsl_asrc_pm = {
+ SET_RUNTIME_PM_OPS(fsl_asrc_runtime_suspend, fsl_asrc_runtime_resume, NULL)
+ SET_SYSTEM_SLEEP_PM_OPS(fsl_asrc_suspend, fsl_asrc_resume)
+};
+
+static const struct of_device_id fsl_asrc_ids[] = {
+ { .compatible = "fsl,imx35-asrc", },
+ { .compatible = "fsl,imx53-asrc", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, fsl_asrc_ids);
+
+static struct platform_driver fsl_asrc_driver = {
+ .probe = fsl_asrc_probe,
+ .driver = {
+ .name = "fsl-asrc",
+ .of_match_table = fsl_asrc_ids,
+ .pm = &fsl_asrc_pm,
+ },
+};
+module_platform_driver(fsl_asrc_driver);
+
+MODULE_DESCRIPTION("Freescale ASRC ASoC driver");
+MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
+MODULE_ALIAS("platform:fsl-asrc");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h
new file mode 100644
index 000000000..4aed63c4b
--- /dev/null
+++ b/sound/soc/fsl/fsl_asrc.h
@@ -0,0 +1,458 @@
+/*
+ * fsl_asrc.h - Freescale ASRC ALSA SoC header file
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _FSL_ASRC_H
+#define _FSL_ASRC_H
+
+#define IN 0
+#define OUT 1
+
+#define ASRC_DMA_BUFFER_NUM 2
+#define ASRC_INPUTFIFO_THRESHOLD 32
+#define ASRC_OUTPUTFIFO_THRESHOLD 32
+#define ASRC_FIFO_THRESHOLD_MIN 0
+#define ASRC_FIFO_THRESHOLD_MAX 63
+#define ASRC_DMA_BUFFER_SIZE (1024 * 48 * 4)
+#define ASRC_MAX_BUFFER_SIZE (1024 * 48)
+#define ASRC_OUTPUT_LAST_SAMPLE 8
+
+#define IDEAL_RATIO_RATE 1000000
+
+#define REG_ASRCTR 0x00
+#define REG_ASRIER 0x04
+#define REG_ASRCNCR 0x0C
+#define REG_ASRCFG 0x10
+#define REG_ASRCSR 0x14
+
+#define REG_ASRCDR1 0x18
+#define REG_ASRCDR2 0x1C
+#define REG_ASRCDR(i) ((i < 2) ? REG_ASRCDR1 : REG_ASRCDR2)
+
+#define REG_ASRSTR 0x20
+#define REG_ASRRA 0x24
+#define REG_ASRRB 0x28
+#define REG_ASRRC 0x2C
+#define REG_ASRPM1 0x40
+#define REG_ASRPM2 0x44
+#define REG_ASRPM3 0x48
+#define REG_ASRPM4 0x4C
+#define REG_ASRPM5 0x50
+#define REG_ASRTFR1 0x54
+#define REG_ASRCCR 0x5C
+
+#define REG_ASRDIA 0x60
+#define REG_ASRDOA 0x64
+#define REG_ASRDIB 0x68
+#define REG_ASRDOB 0x6C
+#define REG_ASRDIC 0x70
+#define REG_ASRDOC 0x74
+#define REG_ASRDI(i) (REG_ASRDIA + (i << 3))
+#define REG_ASRDO(i) (REG_ASRDOA + (i << 3))
+#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i))
+
+#define REG_ASRIDRHA 0x80
+#define REG_ASRIDRLA 0x84
+#define REG_ASRIDRHB 0x88
+#define REG_ASRIDRLB 0x8C
+#define REG_ASRIDRHC 0x90
+#define REG_ASRIDRLC 0x94
+#define REG_ASRIDRH(i) (REG_ASRIDRHA + (i << 3))
+#define REG_ASRIDRL(i) (REG_ASRIDRLA + (i << 3))
+
+#define REG_ASR76K 0x98
+#define REG_ASR56K 0x9C
+
+#define REG_ASRMCRA 0xA0
+#define REG_ASRFSTA 0xA4
+#define REG_ASRMCRB 0xA8
+#define REG_ASRFSTB 0xAC
+#define REG_ASRMCRC 0xB0
+#define REG_ASRFSTC 0xB4
+#define REG_ASRMCR(i) (REG_ASRMCRA + (i << 3))
+#define REG_ASRFST(i) (REG_ASRFSTA + (i << 3))
+
+#define REG_ASRMCR1A 0xC0
+#define REG_ASRMCR1B 0xC4
+#define REG_ASRMCR1C 0xC8
+#define REG_ASRMCR1(i) (REG_ASRMCR1A + (i << 2))
+
+
+/* REG0 0x00 REG_ASRCTR */
+#define ASRCTR_ATSi_SHIFT(i) (20 + i)
+#define ASRCTR_ATSi_MASK(i) (1 << ASRCTR_ATSi_SHIFT(i))
+#define ASRCTR_ATS(i) (1 << ASRCTR_ATSi_SHIFT(i))
+#define ASRCTR_USRi_SHIFT(i) (14 + (i << 1))
+#define ASRCTR_USRi_MASK(i) (1 << ASRCTR_USRi_SHIFT(i))
+#define ASRCTR_USR(i) (1 << ASRCTR_USRi_SHIFT(i))
+#define ASRCTR_IDRi_SHIFT(i) (13 + (i << 1))
+#define ASRCTR_IDRi_MASK(i) (1 << ASRCTR_IDRi_SHIFT(i))
+#define ASRCTR_IDR(i) (1 << ASRCTR_IDRi_SHIFT(i))
+#define ASRCTR_SRST_SHIFT 4
+#define ASRCTR_SRST_MASK (1 << ASRCTR_SRST_SHIFT)
+#define ASRCTR_SRST (1 << ASRCTR_SRST_SHIFT)
+#define ASRCTR_ASRCEi_SHIFT(i) (1 + i)
+#define ASRCTR_ASRCEi_MASK(i) (1 << ASRCTR_ASRCEi_SHIFT(i))
+#define ASRCTR_ASRCE(i) (1 << ASRCTR_ASRCEi_SHIFT(i))
+#define ASRCTR_ASRCEi_ALL_MASK (0x7 << ASRCTR_ASRCEi_SHIFT(0))
+#define ASRCTR_ASRCEN_SHIFT 0
+#define ASRCTR_ASRCEN_MASK (1 << ASRCTR_ASRCEN_SHIFT)
+#define ASRCTR_ASRCEN (1 << ASRCTR_ASRCEN_SHIFT)
+
+/* REG1 0x04 REG_ASRIER */
+#define ASRIER_AFPWE_SHIFT 7
+#define ASRIER_AFPWE_MASK (1 << ASRIER_AFPWE_SHIFT)
+#define ASRIER_AFPWE (1 << ASRIER_AFPWE_SHIFT)
+#define ASRIER_AOLIE_SHIFT 6
+#define ASRIER_AOLIE_MASK (1 << ASRIER_AOLIE_SHIFT)
+#define ASRIER_AOLIE (1 << ASRIER_AOLIE_SHIFT)
+#define ASRIER_ADOEi_SHIFT(i) (3 + i)
+#define ASRIER_ADOEi_MASK(i) (1 << ASRIER_ADOEi_SHIFT(i))
+#define ASRIER_ADOE(i) (1 << ASRIER_ADOEi_SHIFT(i))
+#define ASRIER_ADIEi_SHIFT(i) (0 + i)
+#define ASRIER_ADIEi_MASK(i) (1 << ASRIER_ADIEi_SHIFT(i))
+#define ASRIER_ADIE(i) (1 << ASRIER_ADIEi_SHIFT(i))
+
+/* REG2 0x0C REG_ASRCNCR */
+#define ASRCNCR_ANCi_SHIFT(i, b) (b * i)
+#define ASRCNCR_ANCi_MASK(i, b) (((1 << b) - 1) << ASRCNCR_ANCi_SHIFT(i, b))
+#define ASRCNCR_ANCi(i, v, b) ((v << ASRCNCR_ANCi_SHIFT(i, b)) & ASRCNCR_ANCi_MASK(i, b))
+
+/* REG3 0x10 REG_ASRCFG */
+#define ASRCFG_INIRQi_SHIFT(i) (21 + i)
+#define ASRCFG_INIRQi_MASK(i) (1 << ASRCFG_INIRQi_SHIFT(i))
+#define ASRCFG_INIRQi (1 << ASRCFG_INIRQi_SHIFT(i))
+#define ASRCFG_NDPRi_SHIFT(i) (18 + i)
+#define ASRCFG_NDPRi_MASK(i) (1 << ASRCFG_NDPRi_SHIFT(i))
+#define ASRCFG_NDPRi (1 << ASRCFG_NDPRi_SHIFT(i))
+#define ASRCFG_POSTMODi_SHIFT(i) (8 + (i << 2))
+#define ASRCFG_POSTMODi_WIDTH 2
+#define ASRCFG_POSTMODi_MASK(i) (((1 << ASRCFG_POSTMODi_WIDTH) - 1) << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_POSTMOD(i, v) ((v) << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_POSTMODi_UP(i) (0 << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_POSTMODi_DCON(i) (1 << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_POSTMODi_DOWN(i) (2 << ASRCFG_POSTMODi_SHIFT(i))
+#define ASRCFG_PREMODi_SHIFT(i) (6 + (i << 2))
+#define ASRCFG_PREMODi_WIDTH 2
+#define ASRCFG_PREMODi_MASK(i) (((1 << ASRCFG_PREMODi_WIDTH) - 1) << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMOD(i, v) ((v) << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMODi_UP(i) (0 << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMODi_DCON(i) (1 << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMODi_DOWN(i) (2 << ASRCFG_PREMODi_SHIFT(i))
+#define ASRCFG_PREMODi_BYPASS(i) (3 << ASRCFG_PREMODi_SHIFT(i))
+
+/* REG4 0x14 REG_ASRCSR */
+#define ASRCSR_AxCSi_WIDTH 4
+#define ASRCSR_AxCSi_MASK ((1 << ASRCSR_AxCSi_WIDTH) - 1)
+#define ASRCSR_AOCSi_SHIFT(i) (12 + (i << 2))
+#define ASRCSR_AOCSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AOCSi_SHIFT(i))
+#define ASRCSR_AOCS(i, v) ((v) << ASRCSR_AOCSi_SHIFT(i))
+#define ASRCSR_AICSi_SHIFT(i) (i << 2)
+#define ASRCSR_AICSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AICSi_SHIFT(i))
+#define ASRCSR_AICS(i, v) ((v) << ASRCSR_AICSi_SHIFT(i))
+
+/* REG5&6 0x18 & 0x1C REG_ASRCDR1 & ASRCDR2 */
+#define ASRCDRi_AxCPi_WIDTH 3
+#define ASRCDRi_AICPi_SHIFT(i) (0 + (i % 2) * 6)
+#define ASRCDRi_AICPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICPi_SHIFT(i))
+#define ASRCDRi_AICP(i, v) ((v) << ASRCDRi_AICPi_SHIFT(i))
+#define ASRCDRi_AICDi_SHIFT(i) (3 + (i % 2) * 6)
+#define ASRCDRi_AICDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICDi_SHIFT(i))
+#define ASRCDRi_AICD(i, v) ((v) << ASRCDRi_AICDi_SHIFT(i))
+#define ASRCDRi_AOCPi_SHIFT(i) ((i < 2) ? 12 + i * 6 : 6)
+#define ASRCDRi_AOCPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCPi_SHIFT(i))
+#define ASRCDRi_AOCP(i, v) ((v) << ASRCDRi_AOCPi_SHIFT(i))
+#define ASRCDRi_AOCDi_SHIFT(i) ((i < 2) ? 15 + i * 6 : 9)
+#define ASRCDRi_AOCDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCDi_SHIFT(i))
+#define ASRCDRi_AOCD(i, v) ((v) << ASRCDRi_AOCDi_SHIFT(i))
+
+/* REG7 0x20 REG_ASRSTR */
+#define ASRSTR_DSLCNT_SHIFT 21
+#define ASRSTR_DSLCNT_MASK (1 << ASRSTR_DSLCNT_SHIFT)
+#define ASRSTR_DSLCNT (1 << ASRSTR_DSLCNT_SHIFT)
+#define ASRSTR_ATQOL_SHIFT 20
+#define ASRSTR_ATQOL_MASK (1 << ASRSTR_ATQOL_SHIFT)
+#define ASRSTR_ATQOL (1 << ASRSTR_ATQOL_SHIFT)
+#define ASRSTR_AOOLi_SHIFT(i) (17 + i)
+#define ASRSTR_AOOLi_MASK(i) (1 << ASRSTR_AOOLi_SHIFT(i))
+#define ASRSTR_AOOL(i) (1 << ASRSTR_AOOLi_SHIFT(i))
+#define ASRSTR_AIOLi_SHIFT(i) (14 + i)
+#define ASRSTR_AIOLi_MASK(i) (1 << ASRSTR_AIOLi_SHIFT(i))
+#define ASRSTR_AIOL(i) (1 << ASRSTR_AIOLi_SHIFT(i))
+#define ASRSTR_AODOi_SHIFT(i) (11 + i)
+#define ASRSTR_AODOi_MASK(i) (1 << ASRSTR_AODOi_SHIFT(i))
+#define ASRSTR_AODO(i) (1 << ASRSTR_AODOi_SHIFT(i))
+#define ASRSTR_AIDUi_SHIFT(i) (8 + i)
+#define ASRSTR_AIDUi_MASK(i) (1 << ASRSTR_AIDUi_SHIFT(i))
+#define ASRSTR_AIDU(i) (1 << ASRSTR_AIDUi_SHIFT(i))
+#define ASRSTR_FPWT_SHIFT 7
+#define ASRSTR_FPWT_MASK (1 << ASRSTR_FPWT_SHIFT)
+#define ASRSTR_FPWT (1 << ASRSTR_FPWT_SHIFT)
+#define ASRSTR_AOLE_SHIFT 6
+#define ASRSTR_AOLE_MASK (1 << ASRSTR_AOLE_SHIFT)
+#define ASRSTR_AOLE (1 << ASRSTR_AOLE_SHIFT)
+#define ASRSTR_AODEi_SHIFT(i) (3 + i)
+#define ASRSTR_AODFi_MASK(i) (1 << ASRSTR_AODEi_SHIFT(i))
+#define ASRSTR_AODF(i) (1 << ASRSTR_AODEi_SHIFT(i))
+#define ASRSTR_AIDEi_SHIFT(i) (0 + i)
+#define ASRSTR_AIDEi_MASK(i) (1 << ASRSTR_AIDEi_SHIFT(i))
+#define ASRSTR_AIDE(i) (1 << ASRSTR_AIDEi_SHIFT(i))
+
+/* REG10 0x54 REG_ASRTFR1 */
+#define ASRTFR1_TF_BASE_WIDTH 7
+#define ASRTFR1_TF_BASE_SHIFT 6
+#define ASRTFR1_TF_BASE_MASK (((1 << ASRTFR1_TF_BASE_WIDTH) - 1) << ASRTFR1_TF_BASE_SHIFT)
+#define ASRTFR1_TF_BASE(i) ((i) << ASRTFR1_TF_BASE_SHIFT)
+
+/*
+ * REG22 0xA0 REG_ASRMCRA
+ * REG24 0xA8 REG_ASRMCRB
+ * REG26 0xB0 REG_ASRMCRC
+ */
+#define ASRMCRi_ZEROBUFi_SHIFT 23
+#define ASRMCRi_ZEROBUFi_MASK (1 << ASRMCRi_ZEROBUFi_SHIFT)
+#define ASRMCRi_ZEROBUFi (1 << ASRMCRi_ZEROBUFi_SHIFT)
+#define ASRMCRi_EXTTHRSHi_SHIFT 22
+#define ASRMCRi_EXTTHRSHi_MASK (1 << ASRMCRi_EXTTHRSHi_SHIFT)
+#define ASRMCRi_EXTTHRSHi (1 << ASRMCRi_EXTTHRSHi_SHIFT)
+#define ASRMCRi_BUFSTALLi_SHIFT 21
+#define ASRMCRi_BUFSTALLi_MASK (1 << ASRMCRi_BUFSTALLi_SHIFT)
+#define ASRMCRi_BUFSTALLi (1 << ASRMCRi_BUFSTALLi_SHIFT)
+#define ASRMCRi_BYPASSPOLYi_SHIFT 20
+#define ASRMCRi_BYPASSPOLYi_MASK (1 << ASRMCRi_BYPASSPOLYi_SHIFT)
+#define ASRMCRi_BYPASSPOLYi (1 << ASRMCRi_BYPASSPOLYi_SHIFT)
+#define ASRMCRi_OUTFIFO_THRESHOLD_WIDTH 6
+#define ASRMCRi_OUTFIFO_THRESHOLD_SHIFT 12
+#define ASRMCRi_OUTFIFO_THRESHOLD_MASK (((1 << ASRMCRi_OUTFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT)
+#define ASRMCRi_OUTFIFO_THRESHOLD(v) (((v) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT) & ASRMCRi_OUTFIFO_THRESHOLD_MASK)
+#define ASRMCRi_RSYNIFi_SHIFT 11
+#define ASRMCRi_RSYNIFi_MASK (1 << ASRMCRi_RSYNIFi_SHIFT)
+#define ASRMCRi_RSYNIFi (1 << ASRMCRi_RSYNIFi_SHIFT)
+#define ASRMCRi_RSYNOFi_SHIFT 10
+#define ASRMCRi_RSYNOFi_MASK (1 << ASRMCRi_RSYNOFi_SHIFT)
+#define ASRMCRi_RSYNOFi (1 << ASRMCRi_RSYNOFi_SHIFT)
+#define ASRMCRi_INFIFO_THRESHOLD_WIDTH 6
+#define ASRMCRi_INFIFO_THRESHOLD_SHIFT 0
+#define ASRMCRi_INFIFO_THRESHOLD_MASK (((1 << ASRMCRi_INFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_INFIFO_THRESHOLD_SHIFT)
+#define ASRMCRi_INFIFO_THRESHOLD(v) (((v) << ASRMCRi_INFIFO_THRESHOLD_SHIFT) & ASRMCRi_INFIFO_THRESHOLD_MASK)
+
+/*
+ * REG23 0xA4 REG_ASRFSTA
+ * REG25 0xAC REG_ASRFSTB
+ * REG27 0xB4 REG_ASRFSTC
+ */
+#define ASRFSTi_OAFi_SHIFT 23
+#define ASRFSTi_OAFi_MASK (1 << ASRFSTi_OAFi_SHIFT)
+#define ASRFSTi_OAFi (1 << ASRFSTi_OAFi_SHIFT)
+#define ASRFSTi_OUTPUT_FIFO_WIDTH 7
+#define ASRFSTi_OUTPUT_FIFO_SHIFT 12
+#define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT)
+#define ASRFSTi_IAEi_SHIFT 11
+#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT)
+#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT)
+#define ASRFSTi_INPUT_FIFO_WIDTH 7
+#define ASRFSTi_INPUT_FIFO_SHIFT 0
+#define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1)
+
+/* REG28 0xC0 & 0xC4 & 0xC8 REG_ASRMCR1i */
+#define ASRMCR1i_IWD_WIDTH 3
+#define ASRMCR1i_IWD_SHIFT 9
+#define ASRMCR1i_IWD_MASK (((1 << ASRMCR1i_IWD_WIDTH) - 1) << ASRMCR1i_IWD_SHIFT)
+#define ASRMCR1i_IWD(v) ((v) << ASRMCR1i_IWD_SHIFT)
+#define ASRMCR1i_IMSB_SHIFT 8
+#define ASRMCR1i_IMSB_MASK (1 << ASRMCR1i_IMSB_SHIFT)
+#define ASRMCR1i_IMSB_MSB (1 << ASRMCR1i_IMSB_SHIFT)
+#define ASRMCR1i_IMSB_LSB (0 << ASRMCR1i_IMSB_SHIFT)
+#define ASRMCR1i_OMSB_SHIFT 2
+#define ASRMCR1i_OMSB_MASK (1 << ASRMCR1i_OMSB_SHIFT)
+#define ASRMCR1i_OMSB_MSB (1 << ASRMCR1i_OMSB_SHIFT)
+#define ASRMCR1i_OMSB_LSB (0 << ASRMCR1i_OMSB_SHIFT)
+#define ASRMCR1i_OSGN_SHIFT 1
+#define ASRMCR1i_OSGN_MASK (1 << ASRMCR1i_OSGN_SHIFT)
+#define ASRMCR1i_OSGN (1 << ASRMCR1i_OSGN_SHIFT)
+#define ASRMCR1i_OW16_SHIFT 0
+#define ASRMCR1i_OW16_MASK (1 << ASRMCR1i_OW16_SHIFT)
+#define ASRMCR1i_OW16(v) ((v) << ASRMCR1i_OW16_SHIFT)
+
+
+enum asrc_pair_index {
+ ASRC_INVALID_PAIR = -1,
+ ASRC_PAIR_A = 0,
+ ASRC_PAIR_B = 1,
+ ASRC_PAIR_C = 2,
+};
+
+#define ASRC_PAIR_MAX_NUM (ASRC_PAIR_C + 1)
+
+enum asrc_inclk {
+ INCLK_NONE = 0x03,
+ INCLK_ESAI_RX = 0x00,
+ INCLK_SSI1_RX = 0x01,
+ INCLK_SSI2_RX = 0x02,
+ INCLK_SSI3_RX = 0x07,
+ INCLK_SPDIF_RX = 0x04,
+ INCLK_MLB_CLK = 0x05,
+ INCLK_PAD = 0x06,
+ INCLK_ESAI_TX = 0x08,
+ INCLK_SSI1_TX = 0x09,
+ INCLK_SSI2_TX = 0x0a,
+ INCLK_SSI3_TX = 0x0b,
+ INCLK_SPDIF_TX = 0x0c,
+ INCLK_ASRCK1_CLK = 0x0f,
+};
+
+enum asrc_outclk {
+ OUTCLK_NONE = 0x03,
+ OUTCLK_ESAI_TX = 0x00,
+ OUTCLK_SSI1_TX = 0x01,
+ OUTCLK_SSI2_TX = 0x02,
+ OUTCLK_SSI3_TX = 0x07,
+ OUTCLK_SPDIF_TX = 0x04,
+ OUTCLK_MLB_CLK = 0x05,
+ OUTCLK_PAD = 0x06,
+ OUTCLK_ESAI_RX = 0x08,
+ OUTCLK_SSI1_RX = 0x09,
+ OUTCLK_SSI2_RX = 0x0a,
+ OUTCLK_SSI3_RX = 0x0b,
+ OUTCLK_SPDIF_RX = 0x0c,
+ OUTCLK_ASRCK1_CLK = 0x0f,
+};
+
+#define ASRC_CLK_MAX_NUM 16
+
+enum asrc_word_width {
+ ASRC_WIDTH_24_BIT = 0,
+ ASRC_WIDTH_16_BIT = 1,
+ ASRC_WIDTH_8_BIT = 2,
+};
+
+struct asrc_config {
+ enum asrc_pair_index pair;
+ unsigned int channel_num;
+ unsigned int buffer_num;
+ unsigned int dma_buffer_size;
+ unsigned int input_sample_rate;
+ unsigned int output_sample_rate;
+ enum asrc_word_width input_word_width;
+ enum asrc_word_width output_word_width;
+ enum asrc_inclk inclk;
+ enum asrc_outclk outclk;
+};
+
+struct asrc_req {
+ unsigned int chn_num;
+ enum asrc_pair_index index;
+};
+
+struct asrc_querybuf {
+ unsigned int buffer_index;
+ unsigned int input_length;
+ unsigned int output_length;
+ unsigned long input_offset;
+ unsigned long output_offset;
+};
+
+struct asrc_convert_buffer {
+ void *input_buffer_vaddr;
+ void *output_buffer_vaddr;
+ unsigned int input_buffer_length;
+ unsigned int output_buffer_length;
+};
+
+struct asrc_status_flags {
+ enum asrc_pair_index index;
+ unsigned int overload_error;
+};
+
+enum asrc_error_status {
+ ASRC_TASK_Q_OVERLOAD = 0x01,
+ ASRC_OUTPUT_TASK_OVERLOAD = 0x02,
+ ASRC_INPUT_TASK_OVERLOAD = 0x04,
+ ASRC_OUTPUT_BUFFER_OVERFLOW = 0x08,
+ ASRC_INPUT_BUFFER_UNDERRUN = 0x10,
+};
+
+struct dma_block {
+ dma_addr_t dma_paddr;
+ void *dma_vaddr;
+ unsigned int length;
+};
+
+/**
+ * fsl_asrc_pair: ASRC Pair private data
+ *
+ * @asrc_priv: pointer to its parent module
+ * @config: configuration profile
+ * @error: error record
+ * @index: pair index (ASRC_PAIR_A, ASRC_PAIR_B, ASRC_PAIR_C)
+ * @channels: occupied channel number
+ * @desc: input and output dma descriptors
+ * @dma_chan: inputer and output DMA channels
+ * @dma_data: private dma data
+ * @pos: hardware pointer position
+ * @private: pair private area
+ */
+struct fsl_asrc_pair {
+ struct fsl_asrc *asrc_priv;
+ struct asrc_config *config;
+ unsigned int error;
+
+ enum asrc_pair_index index;
+ unsigned int channels;
+
+ struct dma_async_tx_descriptor *desc[2];
+ struct dma_chan *dma_chan[2];
+ struct imx_dma_data dma_data;
+ unsigned int pos;
+
+ void *private;
+};
+
+/**
+ * fsl_asrc_pair: ASRC private data
+ *
+ * @dma_params_rx: DMA parameters for receive channel
+ * @dma_params_tx: DMA parameters for transmit channel
+ * @pdev: platform device pointer
+ * @regmap: regmap handler
+ * @paddr: physical address to the base address of registers
+ * @mem_clk: clock source to access register
+ * @ipg_clk: clock source to drive peripheral
+ * @asrck_clk: clock sources to driver ASRC internal logic
+ * @lock: spin lock for resource protection
+ * @pair: pair pointers
+ * @channel_bits: width of ASRCNCR register for each pair
+ * @channel_avail: non-occupied channel numbers
+ * @asrc_rate: default sample rate for ASoC Back-Ends
+ * @asrc_width: default sample width for ASoC Back-Ends
+ */
+struct fsl_asrc {
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct platform_device *pdev;
+ struct regmap *regmap;
+ unsigned long paddr;
+ struct clk *mem_clk;
+ struct clk *ipg_clk;
+ struct clk *asrck_clk[ASRC_CLK_MAX_NUM];
+ spinlock_t lock;
+
+ struct fsl_asrc_pair *pair[ASRC_PAIR_MAX_NUM];
+ unsigned int channel_bits;
+ unsigned int channel_avail;
+
+ int asrc_rate;
+ int asrc_width;
+};
+
+extern struct snd_soc_platform_driver fsl_asrc_platform;
+struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir);
+#endif /* _FSL_ASRC_H */
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
new file mode 100644
index 000000000..ffc000bc1
--- /dev/null
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -0,0 +1,391 @@
+/*
+ * Freescale ASRC ALSA SoC Platform (DMA) driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/dma-mapping.h>
+#include <linux/module.h>
+#include <linux/platform_data/dma-imx.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_asrc.h"
+
+#define FSL_ASRC_DMABUF_SIZE (256 * 1024)
+
+static struct snd_pcm_hardware snd_imx_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .buffer_bytes_max = FSL_ASRC_DMABUF_SIZE,
+ .period_bytes_min = 128,
+ .period_bytes_max = 65535, /* Limited by SDMA engine */
+ .periods_min = 2,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ if (!imx_dma_is_general_purpose(chan))
+ return false;
+
+ chan->private = param;
+
+ return true;
+}
+
+static void fsl_asrc_dma_complete(void *arg)
+{
+ struct snd_pcm_substream *substream = arg;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ pair->pos += snd_pcm_lib_period_bytes(substream);
+ if (pair->pos >= snd_pcm_lib_buffer_bytes(substream))
+ pair->pos = 0;
+
+ snd_pcm_period_elapsed(substream);
+}
+
+static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream)
+{
+ u8 dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? OUT : IN;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ struct device *dev = rtd->platform->dev;
+ unsigned long flags = DMA_CTRL_ACK;
+
+ /* Prepare and submit Front-End DMA channel */
+ if (!substream->runtime->no_period_wakeup)
+ flags |= DMA_PREP_INTERRUPT;
+
+ pair->pos = 0;
+ pair->desc[!dir] = dmaengine_prep_dma_cyclic(
+ pair->dma_chan[!dir], runtime->dma_addr,
+ snd_pcm_lib_buffer_bytes(substream),
+ snd_pcm_lib_period_bytes(substream),
+ dir == OUT ? DMA_TO_DEVICE : DMA_FROM_DEVICE, flags);
+ if (!pair->desc[!dir]) {
+ dev_err(dev, "failed to prepare slave DMA for Front-End\n");
+ return -ENOMEM;
+ }
+
+ pair->desc[!dir]->callback = fsl_asrc_dma_complete;
+ pair->desc[!dir]->callback_param = substream;
+
+ dmaengine_submit(pair->desc[!dir]);
+
+ /* Prepare and submit Back-End DMA channel */
+ pair->desc[dir] = dmaengine_prep_dma_cyclic(
+ pair->dma_chan[dir], 0xffff, 64, 64, DMA_DEV_TO_DEV, 0);
+ if (!pair->desc[dir]) {
+ dev_err(dev, "failed to prepare slave DMA for Back-End\n");
+ return -ENOMEM;
+ }
+
+ dmaengine_submit(pair->desc[dir]);
+
+ return 0;
+}
+
+static int fsl_asrc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ int ret;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = fsl_asrc_dma_prepare_and_submit(substream);
+ if (ret)
+ return ret;
+ dma_async_issue_pending(pair->dma_chan[IN]);
+ dma_async_issue_pending(pair->dma_chan[OUT]);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dmaengine_terminate_all(pair->dma_chan[OUT]);
+ dmaengine_terminate_all(pair->dma_chan[IN]);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL;
+ struct snd_dmaengine_dai_dma_data *dma_params_be = NULL;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ struct fsl_asrc *asrc_priv = pair->asrc_priv;
+ struct dma_slave_config config_fe, config_be;
+ enum asrc_pair_index index = pair->index;
+ struct device *dev = rtd->platform->dev;
+ int stream = substream->stream;
+ struct imx_dma_data *tmp_data;
+ struct snd_soc_dpcm *dpcm;
+ struct dma_chan *tmp_chan;
+ struct device *dev_be;
+ u8 dir = tx ? OUT : IN;
+ dma_cap_mask_t mask;
+ int ret;
+
+ /* Fetch the Back-End dma_data from DPCM */
+ list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dpcm->be;
+ struct snd_pcm_substream *substream_be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+
+ if (dpcm->fe != rtd)
+ continue;
+
+ substream_be = snd_soc_dpcm_get_substream(be, stream);
+ dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be);
+ dev_be = dai->dev;
+ break;
+ }
+
+ if (!dma_params_be) {
+ dev_err(dev, "failed to get the substream of Back-End\n");
+ return -EINVAL;
+ }
+
+ /* Override dma_data of the Front-End and config its dmaengine */
+ dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index);
+ dma_params_fe->maxburst = dma_params_be->maxburst;
+
+ pair->dma_chan[!dir] = fsl_asrc_get_dma_channel(pair, !dir);
+ if (!pair->dma_chan[!dir]) {
+ dev_err(dev, "failed to request DMA channel\n");
+ return -EINVAL;
+ }
+
+ memset(&config_fe, 0, sizeof(config_fe));
+ ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe);
+ if (ret) {
+ dev_err(dev, "failed to prepare DMA config for Front-End\n");
+ return ret;
+ }
+
+ ret = dmaengine_slave_config(pair->dma_chan[!dir], &config_fe);
+ if (ret) {
+ dev_err(dev, "failed to config DMA channel for Front-End\n");
+ return ret;
+ }
+
+ /* Request and config DMA channel for Back-End */
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+ dma_cap_set(DMA_CYCLIC, mask);
+
+ /* Get DMA request of Back-End */
+ tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx");
+ tmp_data = tmp_chan->private;
+ pair->dma_data.dma_request = tmp_data->dma_request;
+ dma_release_channel(tmp_chan);
+
+ /* Get DMA request of Front-End */
+ tmp_chan = fsl_asrc_get_dma_channel(pair, dir);
+ tmp_data = tmp_chan->private;
+ pair->dma_data.dma_request2 = tmp_data->dma_request;
+ pair->dma_data.peripheral_type = tmp_data->peripheral_type;
+ pair->dma_data.priority = tmp_data->priority;
+ dma_release_channel(tmp_chan);
+
+ pair->dma_chan[dir] = dma_request_channel(mask, filter, &pair->dma_data);
+ if (!pair->dma_chan[dir]) {
+ dev_err(dev, "failed to request DMA channel for Back-End\n");
+ return -EINVAL;
+ }
+
+ if (asrc_priv->asrc_width == 16)
+ buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ else
+ buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
+
+ config_be.direction = DMA_DEV_TO_DEV;
+ config_be.src_addr_width = buswidth;
+ config_be.src_maxburst = dma_params_be->maxburst;
+ config_be.dst_addr_width = buswidth;
+ config_be.dst_maxburst = dma_params_be->maxburst;
+
+ if (tx) {
+ config_be.src_addr = asrc_priv->paddr + REG_ASRDO(index);
+ config_be.dst_addr = dma_params_be->addr;
+ } else {
+ config_be.dst_addr = asrc_priv->paddr + REG_ASRDI(index);
+ config_be.src_addr = dma_params_be->addr;
+ }
+
+ ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be);
+ if (ret) {
+ dev_err(dev, "failed to config DMA channel for Back-End\n");
+ return ret;
+ }
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int fsl_asrc_dma_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ if (pair->dma_chan[IN])
+ dma_release_channel(pair->dma_chan[IN]);
+
+ if (pair->dma_chan[OUT])
+ dma_release_channel(pair->dma_chan[OUT]);
+
+ pair->dma_chan[IN] = NULL;
+ pair->dma_chan[OUT] = NULL;
+
+ return 0;
+}
+
+static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct device *dev = rtd->platform->dev;
+ struct fsl_asrc *asrc_priv = dev_get_drvdata(dev);
+ struct fsl_asrc_pair *pair;
+
+ pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL);
+ if (!pair) {
+ dev_err(dev, "failed to allocate pair\n");
+ return -ENOMEM;
+ }
+
+ pair->asrc_priv = asrc_priv;
+
+ runtime->private_data = pair;
+
+ snd_pcm_hw_constraint_integer(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
+
+ return 0;
+}
+
+static int fsl_asrc_dma_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+ struct fsl_asrc *asrc_priv;
+
+ if (!pair)
+ return 0;
+
+ asrc_priv = pair->asrc_priv;
+
+ if (asrc_priv->pair[pair->index] == pair)
+ asrc_priv->pair[pair->index] = NULL;
+
+ kfree(pair);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t fsl_asrc_dma_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_asrc_pair *pair = runtime->private_data;
+
+ return bytes_to_frames(substream->runtime, pair->pos);
+}
+
+static struct snd_pcm_ops fsl_asrc_dma_pcm_ops = {
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = fsl_asrc_dma_hw_params,
+ .hw_free = fsl_asrc_dma_hw_free,
+ .trigger = fsl_asrc_dma_trigger,
+ .open = fsl_asrc_dma_startup,
+ .close = fsl_asrc_dma_shutdown,
+ .pointer = fsl_asrc_dma_pcm_pointer,
+};
+
+static int fsl_asrc_dma_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret, i;
+
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ if (ret) {
+ dev_err(card->dev, "failed to set DMA mask\n");
+ return ret;
+ }
+
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+ substream = pcm->streams[i].substream;
+ if (!substream)
+ continue;
+
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+ FSL_ASRC_DMABUF_SIZE, &substream->dma_buffer);
+ if (ret) {
+ dev_err(card->dev, "failed to allocate DMA buffer\n");
+ goto err;
+ }
+ }
+
+ return 0;
+
+err:
+ if (--i == 0 && pcm->streams[i].substream)
+ snd_dma_free_pages(&pcm->streams[i].substream->dma_buffer);
+
+ return ret;
+}
+
+static void fsl_asrc_dma_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ int i;
+
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+ substream = pcm->streams[i].substream;
+ if (!substream)
+ continue;
+
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+}
+
+struct snd_soc_platform_driver fsl_asrc_platform = {
+ .ops = &fsl_asrc_dma_pcm_ops,
+ .pcm_new = fsl_asrc_dma_pcm_new,
+ .pcm_free = fsl_asrc_dma_pcm_free,
+};
+EXPORT_SYMBOL_GPL(fsl_asrc_platform);
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
new file mode 100644
index 000000000..93d7e56c6
--- /dev/null
+++ b/sound/soc/fsl/fsl_dma.c
@@ -0,0 +1,977 @@
+/*
+ * Freescale DMA ALSA SoC PCM driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007-2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ *
+ * This driver implements ASoC support for the Elo DMA controller, which is
+ * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms,
+ * the PCM driver is what handles the DMA buffer.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <linux/gfp.h>
+#include <linux/of_address.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <linux/list.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/io.h>
+
+#include "fsl_dma.h"
+#include "fsl_ssi.h" /* For the offset of stx0 and srx0 */
+
+/*
+ * The formats that the DMA controller supports, which is anything
+ * that is 8, 16, or 32 bits.
+ */
+#define FSLDMA_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_U16_BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_U24_LE | \
+ SNDRV_PCM_FMTBIT_U24_BE | \
+ SNDRV_PCM_FMTBIT_S32_LE | \
+ SNDRV_PCM_FMTBIT_S32_BE | \
+ SNDRV_PCM_FMTBIT_U32_LE | \
+ SNDRV_PCM_FMTBIT_U32_BE)
+struct dma_object {
+ struct snd_soc_platform_driver dai;
+ dma_addr_t ssi_stx_phys;
+ dma_addr_t ssi_srx_phys;
+ unsigned int ssi_fifo_depth;
+ struct ccsr_dma_channel __iomem *channel;
+ unsigned int irq;
+ bool assigned;
+ char path[1];
+};
+
+/*
+ * The number of DMA links to use. Two is the bare minimum, but if you
+ * have really small links you might need more.
+ */
+#define NUM_DMA_LINKS 2
+
+/** fsl_dma_private: p-substream DMA data
+ *
+ * Each substream has a 1-to-1 association with a DMA channel.
+ *
+ * The link[] array is first because it needs to be aligned on a 32-byte
+ * boundary, so putting it first will ensure alignment without padding the
+ * structure.
+ *
+ * @link[]: array of link descriptors
+ * @dma_channel: pointer to the DMA channel's registers
+ * @irq: IRQ for this DMA channel
+ * @substream: pointer to the substream object, needed by the ISR
+ * @ssi_sxx_phys: bus address of the STX or SRX register to use
+ * @ld_buf_phys: physical address of the LD buffer
+ * @current_link: index into link[] of the link currently being processed
+ * @dma_buf_phys: physical address of the DMA buffer
+ * @dma_buf_next: physical address of the next period to process
+ * @dma_buf_end: physical address of the byte after the end of the DMA
+ * @buffer period_size: the size of a single period
+ * @num_periods: the number of periods in the DMA buffer
+ */
+struct fsl_dma_private {
+ struct fsl_dma_link_descriptor link[NUM_DMA_LINKS];
+ struct ccsr_dma_channel __iomem *dma_channel;
+ unsigned int irq;
+ struct snd_pcm_substream *substream;
+ dma_addr_t ssi_sxx_phys;
+ unsigned int ssi_fifo_depth;
+ dma_addr_t ld_buf_phys;
+ unsigned int current_link;
+ dma_addr_t dma_buf_phys;
+ dma_addr_t dma_buf_next;
+ dma_addr_t dma_buf_end;
+ size_t period_size;
+ unsigned int num_periods;
+};
+
+/**
+ * fsl_dma_hardare: define characteristics of the PCM hardware.
+ *
+ * The PCM hardware is the Freescale DMA controller. This structure defines
+ * the capabilities of that hardware.
+ *
+ * Since the sampling rate and data format are not controlled by the DMA
+ * controller, we specify no limits for those values. The only exception is
+ * period_bytes_min, which is set to a reasonably low value to prevent the
+ * DMA controller from generating too many interrupts per second.
+ *
+ * Since each link descriptor has a 32-bit byte count field, we set
+ * period_bytes_max to the largest 32-bit number. We also have no maximum
+ * number of periods.
+ *
+ * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a
+ * limitation in the SSI driver requires the sample rates for playback and
+ * capture to be the same.
+ */
+static const struct snd_pcm_hardware fsl_dma_hardware = {
+
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_JOINT_DUPLEX |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = FSLDMA_PCM_FORMATS,
+ .period_bytes_min = 512, /* A reasonable limit */
+ .period_bytes_max = (u32) -1,
+ .periods_min = NUM_DMA_LINKS,
+ .periods_max = (unsigned int) -1,
+ .buffer_bytes_max = 128 * 1024, /* A reasonable limit */
+};
+
+/**
+ * fsl_dma_abort_stream: tell ALSA that the DMA transfer has aborted
+ *
+ * This function should be called by the ISR whenever the DMA controller
+ * halts data transfer.
+ */
+static void fsl_dma_abort_stream(struct snd_pcm_substream *substream)
+{
+ snd_pcm_stop_xrun(substream);
+}
+
+/**
+ * fsl_dma_update_pointers - update LD pointers to point to the next period
+ *
+ * As each period is completed, this function changes the the link
+ * descriptor pointers for that period to point to the next period.
+ */
+static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private)
+{
+ struct fsl_dma_link_descriptor *link =
+ &dma_private->link[dma_private->current_link];
+
+ /* Update our link descriptors to point to the next period. On a 36-bit
+ * system, we also need to update the ESAD bits. We also set (keep) the
+ * snoop bits. See the comments in fsl_dma_hw_params() about snooping.
+ */
+ if (dma_private->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ link->source_addr = cpu_to_be32(dma_private->dma_buf_next);
+#ifdef CONFIG_PHYS_64BIT
+ link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP |
+ upper_32_bits(dma_private->dma_buf_next));
+#endif
+ } else {
+ link->dest_addr = cpu_to_be32(dma_private->dma_buf_next);
+#ifdef CONFIG_PHYS_64BIT
+ link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP |
+ upper_32_bits(dma_private->dma_buf_next));
+#endif
+ }
+
+ /* Update our variables for next time */
+ dma_private->dma_buf_next += dma_private->period_size;
+
+ if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
+ dma_private->dma_buf_next = dma_private->dma_buf_phys;
+
+ if (++dma_private->current_link >= NUM_DMA_LINKS)
+ dma_private->current_link = 0;
+}
+
+/**
+ * fsl_dma_isr: interrupt handler for the DMA controller
+ *
+ * @irq: IRQ of the DMA channel
+ * @dev_id: pointer to the dma_private structure for this DMA channel
+ */
+static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
+{
+ struct fsl_dma_private *dma_private = dev_id;
+ struct snd_pcm_substream *substream = dma_private->substream;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->platform->dev;
+ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
+ irqreturn_t ret = IRQ_NONE;
+ u32 sr, sr2 = 0;
+
+ /* We got an interrupt, so read the status register to see what we
+ were interrupted for.
+ */
+ sr = in_be32(&dma_channel->sr);
+
+ if (sr & CCSR_DMA_SR_TE) {
+ dev_err(dev, "dma transmit error\n");
+ fsl_dma_abort_stream(substream);
+ sr2 |= CCSR_DMA_SR_TE;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sr & CCSR_DMA_SR_CH)
+ ret = IRQ_HANDLED;
+
+ if (sr & CCSR_DMA_SR_PE) {
+ dev_err(dev, "dma programming error\n");
+ fsl_dma_abort_stream(substream);
+ sr2 |= CCSR_DMA_SR_PE;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sr & CCSR_DMA_SR_EOLNI) {
+ sr2 |= CCSR_DMA_SR_EOLNI;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sr & CCSR_DMA_SR_CB)
+ ret = IRQ_HANDLED;
+
+ if (sr & CCSR_DMA_SR_EOSI) {
+ /* Tell ALSA we completed a period. */
+ snd_pcm_period_elapsed(substream);
+
+ /*
+ * Update our link descriptors to point to the next period. We
+ * only need to do this if the number of periods is not equal to
+ * the number of links.
+ */
+ if (dma_private->num_periods != NUM_DMA_LINKS)
+ fsl_dma_update_pointers(dma_private);
+
+ sr2 |= CCSR_DMA_SR_EOSI;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sr & CCSR_DMA_SR_EOLSI) {
+ sr2 |= CCSR_DMA_SR_EOLSI;
+ ret = IRQ_HANDLED;
+ }
+
+ /* Clear the bits that we set */
+ if (sr2)
+ out_be32(&dma_channel->sr, sr2);
+
+ return ret;
+}
+
+/**
+ * fsl_dma_new: initialize this PCM driver.
+ *
+ * This function is called when the codec driver calls snd_soc_new_pcms(),
+ * once for each .dai_link in the machine driver's snd_soc_card
+ * structure.
+ *
+ * snd_dma_alloc_pages() is just a front-end to dma_alloc_coherent(), which
+ * (currently) always allocates the DMA buffer in lowmem, even if GFP_HIGHMEM
+ * is specified. Therefore, any DMA buffers we allocate will always be in low
+ * memory, but we support for 36-bit physical addresses anyway.
+ *
+ * Regardless of where the memory is actually allocated, since the device can
+ * technically DMA to any 36-bit address, we do need to set the DMA mask to 36.
+ */
+static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret;
+
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(36));
+ if (ret)
+ return ret;
+
+ /* Some codecs have separate DAIs for playback and capture, so we
+ * should allocate a DMA buffer only for the streams that are valid.
+ */
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
+ fsl_dma_hardware.buffer_bytes_max,
+ &pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer);
+ if (ret) {
+ dev_err(card->dev, "can't alloc playback dma buffer\n");
+ return ret;
+ }
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
+ fsl_dma_hardware.buffer_bytes_max,
+ &pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->dma_buffer);
+ if (ret) {
+ dev_err(card->dev, "can't alloc capture dma buffer\n");
+ snd_dma_free_pages(&pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_dma_open: open a new substream.
+ *
+ * Each substream has its own DMA buffer.
+ *
+ * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link
+ * descriptors that ping-pong from one period to the next. For example, if
+ * there are six periods and two link descriptors, this is how they look
+ * before playback starts:
+ *
+ * The last link descriptor
+ * ____________ points back to the first
+ * | |
+ * V |
+ * ___ ___ |
+ * | |->| |->|
+ * |___| |___|
+ * | |
+ * | |
+ * V V
+ * _________________________________________
+ * | | | | | | | The DMA buffer is
+ * | | | | | | | divided into 6 parts
+ * |______|______|______|______|______|______|
+ *
+ * and here's how they look after the first period is finished playing:
+ *
+ * ____________
+ * | |
+ * V |
+ * ___ ___ |
+ * | |->| |->|
+ * |___| |___|
+ * | |
+ * |______________
+ * | |
+ * V V
+ * _________________________________________
+ * | | | | | | |
+ * | | | | | | |
+ * |______|______|______|______|______|______|
+ *
+ * The first link descriptor now points to the third period. The DMA
+ * controller is currently playing the second period. When it finishes, it
+ * will jump back to the first descriptor and play the third period.
+ *
+ * There are four reasons we do this:
+ *
+ * 1. The only way to get the DMA controller to automatically restart the
+ * transfer when it gets to the end of the buffer is to use chaining
+ * mode. Basic direct mode doesn't offer that feature.
+ * 2. We need to receive an interrupt at the end of every period. The DMA
+ * controller can generate an interrupt at the end of every link transfer
+ * (aka segment). Making each period into a DMA segment will give us the
+ * interrupts we need.
+ * 3. By creating only two link descriptors, regardless of the number of
+ * periods, we do not need to reallocate the link descriptors if the
+ * number of periods changes.
+ * 4. All of the audio data is still stored in a single, contiguous DMA
+ * buffer, which is what ALSA expects. We're just dividing it into
+ * contiguous parts, and creating a link descriptor for each one.
+ */
+static int fsl_dma_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->platform->dev;
+ struct dma_object *dma =
+ container_of(rtd->platform->driver, struct dma_object, dai);
+ struct fsl_dma_private *dma_private;
+ struct ccsr_dma_channel __iomem *dma_channel;
+ dma_addr_t ld_buf_phys;
+ u64 temp_link; /* Pointer to next link descriptor */
+ u32 mr;
+ unsigned int channel;
+ int ret = 0;
+ unsigned int i;
+
+ /*
+ * Reject any DMA buffer whose size is not a multiple of the period
+ * size. We need to make sure that the DMA buffer can be evenly divided
+ * into periods.
+ */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ dev_err(dev, "invalid buffer size\n");
+ return ret;
+ }
+
+ channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ if (dma->assigned) {
+ dev_err(dev, "dma channel already assigned\n");
+ return -EBUSY;
+ }
+
+ dma_private = dma_alloc_coherent(dev, sizeof(struct fsl_dma_private),
+ &ld_buf_phys, GFP_KERNEL);
+ if (!dma_private) {
+ dev_err(dev, "can't allocate dma private data\n");
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dma_private->ssi_sxx_phys = dma->ssi_stx_phys;
+ else
+ dma_private->ssi_sxx_phys = dma->ssi_srx_phys;
+
+ dma_private->ssi_fifo_depth = dma->ssi_fifo_depth;
+ dma_private->dma_channel = dma->channel;
+ dma_private->irq = dma->irq;
+ dma_private->substream = substream;
+ dma_private->ld_buf_phys = ld_buf_phys;
+ dma_private->dma_buf_phys = substream->dma_buffer.addr;
+
+ ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio",
+ dma_private);
+ if (ret) {
+ dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n",
+ dma_private->irq, ret);
+ dma_free_coherent(dev, sizeof(struct fsl_dma_private),
+ dma_private, dma_private->ld_buf_phys);
+ return ret;
+ }
+
+ dma->assigned = 1;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
+ runtime->private_data = dma_private;
+
+ /* Program the fixed DMA controller parameters */
+
+ dma_channel = dma_private->dma_channel;
+
+ temp_link = dma_private->ld_buf_phys +
+ sizeof(struct fsl_dma_link_descriptor);
+
+ for (i = 0; i < NUM_DMA_LINKS; i++) {
+ dma_private->link[i].next = cpu_to_be64(temp_link);
+
+ temp_link += sizeof(struct fsl_dma_link_descriptor);
+ }
+ /* The last link descriptor points to the first */
+ dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
+
+ /* Tell the DMA controller where the first link descriptor is */
+ out_be32(&dma_channel->clndar,
+ CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
+ out_be32(&dma_channel->eclndar,
+ CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
+
+ /* The manual says the BCR must be clear before enabling EMP */
+ out_be32(&dma_channel->bcr, 0);
+
+ /*
+ * Program the mode register for interrupts, external master control,
+ * and source/destination hold. Also clear the Channel Abort bit.
+ */
+ mr = in_be32(&dma_channel->mr) &
+ ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
+
+ /*
+ * We want External Master Start and External Master Pause enabled,
+ * because the SSI is controlling the DMA controller. We want the DMA
+ * controller to be set up in advance, and then we signal only the SSI
+ * to start transferring.
+ *
+ * We want End-Of-Segment Interrupts enabled, because this will generate
+ * an interrupt at the end of each segment (each link descriptor
+ * represents one segment). Each DMA segment is the same thing as an
+ * ALSA period, so this is how we get an interrupt at the end of every
+ * period.
+ *
+ * We want Error Interrupt enabled, so that we can get an error if
+ * the DMA controller is mis-programmed somehow.
+ */
+ mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
+ CCSR_DMA_MR_EMS_EN;
+
+ /* For playback, we want the destination address to be held. For
+ capture, set the source address to be held. */
+ mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
+
+ out_be32(&dma_channel->mr, mr);
+
+ return 0;
+}
+
+/**
+ * fsl_dma_hw_params: continue initializing the DMA links
+ *
+ * This function obtains hardware parameters about the opened stream and
+ * programs the DMA controller accordingly.
+ *
+ * One drawback of big-endian is that when copying integers of different
+ * sizes to a fixed-sized register, the address to which the integer must be
+ * copied is dependent on the size of the integer.
+ *
+ * For example, if P is the address of a 32-bit register, and X is a 32-bit
+ * integer, then X should be copied to address P. However, if X is a 16-bit
+ * integer, then it should be copied to P+2. If X is an 8-bit register,
+ * then it should be copied to P+3.
+ *
+ * So for playback of 8-bit samples, the DMA controller must transfer single
+ * bytes from the DMA buffer to the last byte of the STX0 register, i.e.
+ * offset by 3 bytes. For 16-bit samples, the offset is two bytes.
+ *
+ * For 24-bit samples, the offset is 1 byte. However, the DMA controller
+ * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4,
+ * and 8 bytes at a time). So we do not support packed 24-bit samples.
+ * 24-bit data must be padded to 32 bits.
+ */
+static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->platform->dev;
+
+ /* Number of bits per sample */
+ unsigned int sample_bits =
+ snd_pcm_format_physical_width(params_format(hw_params));
+
+ /* Number of bytes per frame */
+ unsigned int sample_bytes = sample_bits / 8;
+
+ /* Bus address of SSI STX register */
+ dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys;
+
+ /* Size of the DMA buffer, in bytes */
+ size_t buffer_size = params_buffer_bytes(hw_params);
+
+ /* Number of bytes per period */
+ size_t period_size = params_period_bytes(hw_params);
+
+ /* Pointer to next period */
+ dma_addr_t temp_addr = substream->dma_buffer.addr;
+
+ /* Pointer to DMA controller */
+ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
+
+ u32 mr; /* DMA Mode Register */
+
+ unsigned int i;
+
+ /* Initialize our DMA tracking variables */
+ dma_private->period_size = period_size;
+ dma_private->num_periods = params_periods(hw_params);
+ dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
+ dma_private->dma_buf_next = dma_private->dma_buf_phys +
+ (NUM_DMA_LINKS * period_size);
+
+ if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
+ /* This happens if the number of periods == NUM_DMA_LINKS */
+ dma_private->dma_buf_next = dma_private->dma_buf_phys;
+
+ mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK |
+ CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK);
+
+ /* Due to a quirk of the SSI's STX register, the target address
+ * for the DMA operations depends on the sample size. So we calculate
+ * that offset here. While we're at it, also tell the DMA controller
+ * how much data to transfer per sample.
+ */
+ switch (sample_bits) {
+ case 8:
+ mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
+ ssi_sxx_phys += 3;
+ break;
+ case 16:
+ mr |= CCSR_DMA_MR_DAHTS_2 | CCSR_DMA_MR_SAHTS_2;
+ ssi_sxx_phys += 2;
+ break;
+ case 32:
+ mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4;
+ break;
+ default:
+ /* We should never get here */
+ dev_err(dev, "unsupported sample size %u\n", sample_bits);
+ return -EINVAL;
+ }
+
+ /*
+ * BWC determines how many bytes are sent/received before the DMA
+ * controller checks the SSI to see if it needs to stop. BWC should
+ * always be a multiple of the frame size, so that we always transmit
+ * whole frames. Each frame occupies two slots in the FIFO. The
+ * parameter for CCSR_DMA_MR_BWC() is rounded down the next power of two
+ * (MR[BWC] can only represent even powers of two).
+ *
+ * To simplify the process, we set BWC to the largest value that is
+ * less than or equal to the FIFO watermark. For playback, this ensures
+ * that we transfer the maximum amount without overrunning the FIFO.
+ * For capture, this ensures that we transfer the maximum amount without
+ * underrunning the FIFO.
+ *
+ * f = SSI FIFO depth
+ * w = SSI watermark value (which equals f - 2)
+ * b = DMA bandwidth count (in bytes)
+ * s = sample size (in bytes, which equals frame_size * 2)
+ *
+ * For playback, we never transmit more than the transmit FIFO
+ * watermark, otherwise we might write more data than the FIFO can hold.
+ * The watermark is equal to the FIFO depth minus two.
+ *
+ * For capture, two equations must hold:
+ * w > f - (b / s)
+ * w >= b / s
+ *
+ * So, b > 2 * s, but b must also be <= s * w. To simplify, we set
+ * b = s * w, which is equal to
+ * (dma_private->ssi_fifo_depth - 2) * sample_bytes.
+ */
+ mr |= CCSR_DMA_MR_BWC((dma_private->ssi_fifo_depth - 2) * sample_bytes);
+
+ out_be32(&dma_channel->mr, mr);
+
+ for (i = 0; i < NUM_DMA_LINKS; i++) {
+ struct fsl_dma_link_descriptor *link = &dma_private->link[i];
+
+ link->count = cpu_to_be32(period_size);
+
+ /* The snoop bit tells the DMA controller whether it should tell
+ * the ECM to snoop during a read or write to an address. For
+ * audio, we use DMA to transfer data between memory and an I/O
+ * device (the SSI's STX0 or SRX0 register). Snooping is only
+ * needed if there is a cache, so we need to snoop memory
+ * addresses only. For playback, that means we snoop the source
+ * but not the destination. For capture, we snoop the
+ * destination but not the source.
+ *
+ * Note that failing to snoop properly is unlikely to cause
+ * cache incoherency if the period size is larger than the
+ * size of L1 cache. This is because filling in one period will
+ * flush out the data for the previous period. So if you
+ * increased period_bytes_min to a large enough size, you might
+ * get more performance by not snooping, and you'll still be
+ * okay. You'll need to update fsl_dma_update_pointers() also.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ link->source_addr = cpu_to_be32(temp_addr);
+ link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP |
+ upper_32_bits(temp_addr));
+
+ link->dest_addr = cpu_to_be32(ssi_sxx_phys);
+ link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP |
+ upper_32_bits(ssi_sxx_phys));
+ } else {
+ link->source_addr = cpu_to_be32(ssi_sxx_phys);
+ link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP |
+ upper_32_bits(ssi_sxx_phys));
+
+ link->dest_addr = cpu_to_be32(temp_addr);
+ link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP |
+ upper_32_bits(temp_addr));
+ }
+
+ temp_addr += period_size;
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_dma_pointer: determine the current position of the DMA transfer
+ *
+ * This function is called by ALSA when ALSA wants to know where in the
+ * stream buffer the hardware currently is.
+ *
+ * For playback, the SAR register contains the physical address of the most
+ * recent DMA transfer. For capture, the value is in the DAR register.
+ *
+ * The base address of the buffer is stored in the source_addr field of the
+ * first link descriptor.
+ */
+static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->platform->dev;
+ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
+ dma_addr_t position;
+ snd_pcm_uframes_t frames;
+
+ /* Obtain the current DMA pointer, but don't read the ESAD bits if we
+ * only have 32-bit DMA addresses. This function is typically called
+ * in interrupt context, so we need to optimize it.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ position = in_be32(&dma_channel->sar);
+#ifdef CONFIG_PHYS_64BIT
+ position |= (u64)(in_be32(&dma_channel->satr) &
+ CCSR_DMA_ATR_ESAD_MASK) << 32;
+#endif
+ } else {
+ position = in_be32(&dma_channel->dar);
+#ifdef CONFIG_PHYS_64BIT
+ position |= (u64)(in_be32(&dma_channel->datr) &
+ CCSR_DMA_ATR_ESAD_MASK) << 32;
+#endif
+ }
+
+ /*
+ * When capture is started, the SSI immediately starts to fill its FIFO.
+ * This means that the DMA controller is not started until the FIFO is
+ * full. However, ALSA calls this function before that happens, when
+ * MR.DAR is still zero. In this case, just return zero to indicate
+ * that nothing has been received yet.
+ */
+ if (!position)
+ return 0;
+
+ if ((position < dma_private->dma_buf_phys) ||
+ (position > dma_private->dma_buf_end)) {
+ dev_err(dev, "dma pointer is out of range, halting stream\n");
+ return SNDRV_PCM_POS_XRUN;
+ }
+
+ frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
+
+ /*
+ * If the current address is just past the end of the buffer, wrap it
+ * around.
+ */
+ if (frames == runtime->buffer_size)
+ frames = 0;
+
+ return frames;
+}
+
+/**
+ * fsl_dma_hw_free: release resources allocated in fsl_dma_hw_params()
+ *
+ * Release the resources allocated in fsl_dma_hw_params() and de-program the
+ * registers.
+ *
+ * This function can be called multiple times.
+ */
+static int fsl_dma_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+
+ if (dma_private) {
+ struct ccsr_dma_channel __iomem *dma_channel;
+
+ dma_channel = dma_private->dma_channel;
+
+ /* Stop the DMA */
+ out_be32(&dma_channel->mr, CCSR_DMA_MR_CA);
+ out_be32(&dma_channel->mr, 0);
+
+ /* Reset all the other registers */
+ out_be32(&dma_channel->sr, -1);
+ out_be32(&dma_channel->clndar, 0);
+ out_be32(&dma_channel->eclndar, 0);
+ out_be32(&dma_channel->satr, 0);
+ out_be32(&dma_channel->sar, 0);
+ out_be32(&dma_channel->datr, 0);
+ out_be32(&dma_channel->dar, 0);
+ out_be32(&dma_channel->bcr, 0);
+ out_be32(&dma_channel->nlndar, 0);
+ out_be32(&dma_channel->enlndar, 0);
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_dma_close: close the stream.
+ */
+static int fsl_dma_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct device *dev = rtd->platform->dev;
+ struct dma_object *dma =
+ container_of(rtd->platform->driver, struct dma_object, dai);
+
+ if (dma_private) {
+ if (dma_private->irq)
+ free_irq(dma_private->irq, dma_private);
+
+ /* Deallocate the fsl_dma_private structure */
+ dma_free_coherent(dev, sizeof(struct fsl_dma_private),
+ dma_private, dma_private->ld_buf_phys);
+ substream->runtime->private_data = NULL;
+ }
+
+ dma->assigned = 0;
+
+ return 0;
+}
+
+/*
+ * Remove this PCM driver.
+ */
+static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ unsigned int i;
+
+ for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+ substream = pcm->streams[i].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+/**
+ * find_ssi_node -- returns the SSI node that points to its DMA channel node
+ *
+ * Although this DMA driver attempts to operate independently of the other
+ * devices, it still needs to determine some information about the SSI device
+ * that it's working with. Unfortunately, the device tree does not contain
+ * a pointer from the DMA channel node to the SSI node -- the pointer goes the
+ * other way. So we need to scan the device tree for SSI nodes until we find
+ * the one that points to the given DMA channel node. It's ugly, but at least
+ * it's contained in this one function.
+ */
+static struct device_node *find_ssi_node(struct device_node *dma_channel_np)
+{
+ struct device_node *ssi_np, *np;
+
+ for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") {
+ /* Check each DMA phandle to see if it points to us. We
+ * assume that device_node pointers are a valid comparison.
+ */
+ np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0);
+ of_node_put(np);
+ if (np == dma_channel_np)
+ return ssi_np;
+
+ np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0);
+ of_node_put(np);
+ if (np == dma_channel_np)
+ return ssi_np;
+ }
+
+ return NULL;
+}
+
+static struct snd_pcm_ops fsl_dma_ops = {
+ .open = fsl_dma_open,
+ .close = fsl_dma_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = fsl_dma_hw_params,
+ .hw_free = fsl_dma_hw_free,
+ .pointer = fsl_dma_pointer,
+};
+
+static int fsl_soc_dma_probe(struct platform_device *pdev)
+ {
+ struct dma_object *dma;
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np;
+ struct resource res;
+ const uint32_t *iprop;
+ int ret;
+
+ /* Find the SSI node that points to us. */
+ ssi_np = find_ssi_node(np);
+ if (!ssi_np) {
+ dev_err(&pdev->dev, "cannot find parent SSI node\n");
+ return -ENODEV;
+ }
+
+ ret = of_address_to_resource(ssi_np, 0, &res);
+ if (ret) {
+ dev_err(&pdev->dev, "could not determine resources for %s\n",
+ ssi_np->full_name);
+ of_node_put(ssi_np);
+ return ret;
+ }
+
+ dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL);
+ if (!dma) {
+ dev_err(&pdev->dev, "could not allocate dma object\n");
+ of_node_put(ssi_np);
+ return -ENOMEM;
+ }
+
+ strcpy(dma->path, np->full_name);
+ dma->dai.ops = &fsl_dma_ops;
+ dma->dai.pcm_new = fsl_dma_new;
+ dma->dai.pcm_free = fsl_dma_free_dma_buffers;
+
+ /* Store the SSI-specific information that we need */
+ dma->ssi_stx_phys = res.start + CCSR_SSI_STX0;
+ dma->ssi_srx_phys = res.start + CCSR_SSI_SRX0;
+
+ iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
+ if (iprop)
+ dma->ssi_fifo_depth = be32_to_cpup(iprop);
+ else
+ /* Older 8610 DTs didn't have the fifo-depth property */
+ dma->ssi_fifo_depth = 8;
+
+ of_node_put(ssi_np);
+
+ ret = snd_soc_register_platform(&pdev->dev, &dma->dai);
+ if (ret) {
+ dev_err(&pdev->dev, "could not register platform\n");
+ kfree(dma);
+ return ret;
+ }
+
+ dma->channel = of_iomap(np, 0);
+ dma->irq = irq_of_parse_and_map(np, 0);
+
+ dev_set_drvdata(&pdev->dev, dma);
+
+ return 0;
+}
+
+static int fsl_soc_dma_remove(struct platform_device *pdev)
+{
+ struct dma_object *dma = dev_get_drvdata(&pdev->dev);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ iounmap(dma->channel);
+ irq_dispose_mapping(dma->irq);
+ kfree(dma);
+
+ return 0;
+}
+
+static const struct of_device_id fsl_soc_dma_ids[] = {
+ { .compatible = "fsl,ssi-dma-channel", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids);
+
+static struct platform_driver fsl_soc_dma_driver = {
+ .driver = {
+ .name = "fsl-pcm-audio",
+ .of_match_table = fsl_soc_dma_ids,
+ },
+ .probe = fsl_soc_dma_probe,
+ .remove = fsl_soc_dma_remove,
+};
+
+module_platform_driver(fsl_soc_dma_driver);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM Driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h
new file mode 100644
index 000000000..78fee97e8
--- /dev/null
+++ b/sound/soc/fsl/fsl_dma.h
@@ -0,0 +1,129 @@
+/*
+ * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _MPC8610_PCM_H
+#define _MPC8610_PCM_H
+
+struct ccsr_dma {
+ u8 res0[0x100];
+ struct ccsr_dma_channel {
+ __be32 mr; /* Mode register */
+ __be32 sr; /* Status register */
+ __be32 eclndar; /* Current link descriptor extended addr reg */
+ __be32 clndar; /* Current link descriptor address register */
+ __be32 satr; /* Source attributes register */
+ __be32 sar; /* Source address register */
+ __be32 datr; /* Destination attributes register */
+ __be32 dar; /* Destination address register */
+ __be32 bcr; /* Byte count register */
+ __be32 enlndar; /* Next link descriptor extended address reg */
+ __be32 nlndar; /* Next link descriptor address register */
+ u8 res1[4];
+ __be32 eclsdar; /* Current list descriptor extended addr reg */
+ __be32 clsdar; /* Current list descriptor address register */
+ __be32 enlsdar; /* Next list descriptor extended address reg */
+ __be32 nlsdar; /* Next list descriptor address register */
+ __be32 ssr; /* Source stride register */
+ __be32 dsr; /* Destination stride register */
+ u8 res2[0x38];
+ } channel[4];
+ __be32 dgsr;
+};
+
+#define CCSR_DMA_MR_BWC_DISABLED 0x0F000000
+#define CCSR_DMA_MR_BWC_SHIFT 24
+#define CCSR_DMA_MR_BWC_MASK 0x0F000000
+#define CCSR_DMA_MR_BWC(x) \
+ ((ilog2(x) << CCSR_DMA_MR_BWC_SHIFT) & CCSR_DMA_MR_BWC_MASK)
+#define CCSR_DMA_MR_EMP_EN 0x00200000
+#define CCSR_DMA_MR_EMS_EN 0x00040000
+#define CCSR_DMA_MR_DAHTS_MASK 0x00030000
+#define CCSR_DMA_MR_DAHTS_1 0x00000000
+#define CCSR_DMA_MR_DAHTS_2 0x00010000
+#define CCSR_DMA_MR_DAHTS_4 0x00020000
+#define CCSR_DMA_MR_DAHTS_8 0x00030000
+#define CCSR_DMA_MR_SAHTS_MASK 0x0000C000
+#define CCSR_DMA_MR_SAHTS_1 0x00000000
+#define CCSR_DMA_MR_SAHTS_2 0x00004000
+#define CCSR_DMA_MR_SAHTS_4 0x00008000
+#define CCSR_DMA_MR_SAHTS_8 0x0000C000
+#define CCSR_DMA_MR_DAHE 0x00002000
+#define CCSR_DMA_MR_SAHE 0x00001000
+#define CCSR_DMA_MR_SRW 0x00000400
+#define CCSR_DMA_MR_EOSIE 0x00000200
+#define CCSR_DMA_MR_EOLNIE 0x00000100
+#define CCSR_DMA_MR_EOLSIE 0x00000080
+#define CCSR_DMA_MR_EIE 0x00000040
+#define CCSR_DMA_MR_XFE 0x00000020
+#define CCSR_DMA_MR_CDSM_SWSM 0x00000010
+#define CCSR_DMA_MR_CA 0x00000008
+#define CCSR_DMA_MR_CTM 0x00000004
+#define CCSR_DMA_MR_CC 0x00000002
+#define CCSR_DMA_MR_CS 0x00000001
+
+#define CCSR_DMA_SR_TE 0x00000080
+#define CCSR_DMA_SR_CH 0x00000020
+#define CCSR_DMA_SR_PE 0x00000010
+#define CCSR_DMA_SR_EOLNI 0x00000008
+#define CCSR_DMA_SR_CB 0x00000004
+#define CCSR_DMA_SR_EOSI 0x00000002
+#define CCSR_DMA_SR_EOLSI 0x00000001
+
+/* ECLNDAR takes bits 32-36 of the CLNDAR register */
+static inline u32 CCSR_DMA_ECLNDAR_ADDR(u64 x)
+{
+ return (x >> 32) & 0xf;
+}
+
+#define CCSR_DMA_CLNDAR_ADDR(x) ((x) & 0xFFFFFFFE)
+#define CCSR_DMA_CLNDAR_EOSIE 0x00000008
+
+/* SATR and DATR, combined */
+#define CCSR_DMA_ATR_PBATMU 0x20000000
+#define CCSR_DMA_ATR_TFLOWLVL_0 0x00000000
+#define CCSR_DMA_ATR_TFLOWLVL_1 0x06000000
+#define CCSR_DMA_ATR_TFLOWLVL_2 0x08000000
+#define CCSR_DMA_ATR_TFLOWLVL_3 0x0C000000
+#define CCSR_DMA_ATR_PCIORDER 0x02000000
+#define CCSR_DMA_ATR_SME 0x01000000
+#define CCSR_DMA_ATR_NOSNOOP 0x00040000
+#define CCSR_DMA_ATR_SNOOP 0x00050000
+#define CCSR_DMA_ATR_ESAD_MASK 0x0000000F
+
+/**
+ * List Descriptor for extended chaining mode DMA operations.
+ *
+ * The CLSDAR register points to the first (in a linked-list) List
+ * Descriptor. Each object must be aligned on a 32-byte boundary. Each
+ * list descriptor points to a linked-list of link Descriptors.
+ */
+struct fsl_dma_list_descriptor {
+ __be64 next; /* Address of next list descriptor */
+ __be64 first_link; /* Address of first link descriptor */
+ __be32 source; /* Source stride */
+ __be32 dest; /* Destination stride */
+ u8 res[8]; /* Reserved */
+} __attribute__ ((aligned(32), packed));
+
+/**
+ * Link Descriptor for basic and extended chaining mode DMA operations.
+ *
+ * A Link Descriptor points to a single DMA buffer. Each link descriptor
+ * must be aligned on a 32-byte boundary.
+ */
+struct fsl_dma_link_descriptor {
+ __be32 source_attr; /* Programmed into SATR register */
+ __be32 source_addr; /* Programmed into SAR register */
+ __be32 dest_attr; /* Programmed into DATR register */
+ __be32 dest_addr; /* Programmed into DAR register */
+ __be64 next; /* Address of next link descriptor */
+ __be32 count; /* Byte count */
+ u8 res[4]; /* Reserved */
+} __attribute__ ((aligned(32), packed));
+
+#endif
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
new file mode 100644
index 000000000..5c7597191
--- /dev/null
+++ b/sound/soc/fsl/fsl_esai.c
@@ -0,0 +1,869 @@
+/*
+ * Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/dmaengine.h>
+#include <linux/module.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_esai.h"
+#include "imx-pcm.h"
+
+#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000
+#define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+/**
+ * fsl_esai: ESAI private data
+ *
+ * @dma_params_rx: DMA parameters for receive channel
+ * @dma_params_tx: DMA parameters for transmit channel
+ * @pdev: platform device pointer
+ * @regmap: regmap handler
+ * @coreclk: clock source to access register
+ * @extalclk: esai clock source to derive HCK, SCK and FS
+ * @fsysclk: system clock source to derive HCK, SCK and FS
+ * @fifo_depth: depth of tx/rx FIFO
+ * @slot_width: width of each DAI slot
+ * @slots: number of slots
+ * @hck_rate: clock rate of desired HCKx clock
+ * @sck_rate: clock rate of desired SCKx clock
+ * @hck_dir: the direction of HCKx pads
+ * @sck_div: if using PSR/PM dividers for SCKx clock
+ * @slave_mode: if fully using DAI slave mode
+ * @synchronous: if using tx/rx synchronous mode
+ * @name: driver name
+ */
+struct fsl_esai {
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct platform_device *pdev;
+ struct regmap *regmap;
+ struct clk *coreclk;
+ struct clk *extalclk;
+ struct clk *fsysclk;
+ u32 fifo_depth;
+ u32 slot_width;
+ u32 slots;
+ u32 hck_rate[2];
+ u32 sck_rate[2];
+ bool hck_dir[2];
+ bool sck_div[2];
+ bool slave_mode;
+ bool synchronous;
+ char name[32];
+};
+
+static irqreturn_t esai_isr(int irq, void *devid)
+{
+ struct fsl_esai *esai_priv = (struct fsl_esai *)devid;
+ struct platform_device *pdev = esai_priv->pdev;
+ u32 esr;
+
+ regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr);
+
+ if (esr & ESAI_ESR_TINIT_MASK)
+ dev_dbg(&pdev->dev, "isr: Transmition Initialized\n");
+
+ if (esr & ESAI_ESR_RFF_MASK)
+ dev_warn(&pdev->dev, "isr: Receiving overrun\n");
+
+ if (esr & ESAI_ESR_TFE_MASK)
+ dev_warn(&pdev->dev, "isr: Transmition underrun\n");
+
+ if (esr & ESAI_ESR_TLS_MASK)
+ dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n");
+
+ if (esr & ESAI_ESR_TDE_MASK)
+ dev_dbg(&pdev->dev, "isr: Transmition data exception\n");
+
+ if (esr & ESAI_ESR_TED_MASK)
+ dev_dbg(&pdev->dev, "isr: Transmitting even slots\n");
+
+ if (esr & ESAI_ESR_TD_MASK)
+ dev_dbg(&pdev->dev, "isr: Transmitting data\n");
+
+ if (esr & ESAI_ESR_RLS_MASK)
+ dev_dbg(&pdev->dev, "isr: Just received the last slot\n");
+
+ if (esr & ESAI_ESR_RDE_MASK)
+ dev_dbg(&pdev->dev, "isr: Receiving data exception\n");
+
+ if (esr & ESAI_ESR_RED_MASK)
+ dev_dbg(&pdev->dev, "isr: Receiving even slots\n");
+
+ if (esr & ESAI_ESR_RD_MASK)
+ dev_dbg(&pdev->dev, "isr: Receiving data\n");
+
+ return IRQ_HANDLED;
+}
+
+/**
+ * This function is used to calculate the divisors of psr, pm, fp and it is
+ * supposed to be called in set_dai_sysclk() and set_bclk().
+ *
+ * @ratio: desired overall ratio for the paticipating dividers
+ * @usefp: for HCK setting, there is no need to set fp divider
+ * @fp: bypass other dividers by setting fp directly if fp != 0
+ * @tx: current setting is for playback or capture
+ */
+static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio,
+ bool usefp, u32 fp)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+ u32 psr, pm = 999, maxfp, prod, sub, savesub, i, j;
+
+ maxfp = usefp ? 16 : 1;
+
+ if (usefp && fp)
+ goto out_fp;
+
+ if (ratio > 2 * 8 * 256 * maxfp || ratio < 2) {
+ dev_err(dai->dev, "the ratio is out of range (2 ~ %d)\n",
+ 2 * 8 * 256 * maxfp);
+ return -EINVAL;
+ } else if (ratio % 2) {
+ dev_err(dai->dev, "the raio must be even if using upper divider\n");
+ return -EINVAL;
+ }
+
+ ratio /= 2;
+
+ psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8;
+
+ /* Set the max fluctuation -- 0.1% of the max devisor */
+ savesub = (psr ? 1 : 8) * 256 * maxfp / 1000;
+
+ /* Find the best value for PM */
+ for (i = 1; i <= 256; i++) {
+ for (j = 1; j <= maxfp; j++) {
+ /* PSR (1 or 8) * PM (1 ~ 256) * FP (1 ~ 16) */
+ prod = (psr ? 1 : 8) * i * j;
+
+ if (prod == ratio)
+ sub = 0;
+ else if (prod / ratio == 1)
+ sub = prod - ratio;
+ else if (ratio / prod == 1)
+ sub = ratio - prod;
+ else
+ continue;
+
+ /* Calculate the fraction */
+ sub = sub * 1000 / ratio;
+ if (sub < savesub) {
+ savesub = sub;
+ pm = i;
+ fp = j;
+ }
+
+ /* We are lucky */
+ if (savesub == 0)
+ goto out;
+ }
+ }
+
+ if (pm == 999) {
+ dev_err(dai->dev, "failed to calculate proper divisors\n");
+ return -EINVAL;
+ }
+
+out:
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx),
+ ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK,
+ psr | ESAI_xCCR_xPM(pm));
+
+out_fp:
+ /* Bypass fp if not being required */
+ if (maxfp <= 1)
+ return 0;
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx),
+ ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp));
+
+ return 0;
+}
+
+/**
+ * This function mainly configures the clock frequency of MCLK (HCKT/HCKR)
+ *
+ * @Parameters:
+ * clk_id: The clock source of HCKT/HCKR
+ * (Input from outside; output from inside, FSYS or EXTAL)
+ * freq: The required clock rate of HCKT/HCKR
+ * dir: The clock direction of HCKT/HCKR
+ *
+ * Note: If the direction is input, we do not care about clk_id.
+ */
+static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+ struct clk *clksrc = esai_priv->extalclk;
+ bool tx = clk_id <= ESAI_HCKT_EXTAL;
+ bool in = dir == SND_SOC_CLOCK_IN;
+ u32 ratio, ecr = 0;
+ unsigned long clk_rate;
+ int ret;
+
+ /* Bypass divider settings if the requirement doesn't change */
+ if (freq == esai_priv->hck_rate[tx] && dir == esai_priv->hck_dir[tx])
+ return 0;
+
+ /* sck_div can be only bypassed if ETO/ERO=0 and SNC_SOC_CLOCK_OUT */
+ esai_priv->sck_div[tx] = true;
+
+ /* Set the direction of HCKT/HCKR pins */
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx),
+ ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD);
+
+ if (in)
+ goto out;
+
+ switch (clk_id) {
+ case ESAI_HCKT_FSYS:
+ case ESAI_HCKR_FSYS:
+ clksrc = esai_priv->fsysclk;
+ break;
+ case ESAI_HCKT_EXTAL:
+ ecr |= ESAI_ECR_ETI;
+ case ESAI_HCKR_EXTAL:
+ ecr |= ESAI_ECR_ERI;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (IS_ERR(clksrc)) {
+ dev_err(dai->dev, "no assigned %s clock\n",
+ clk_id % 2 ? "extal" : "fsys");
+ return PTR_ERR(clksrc);
+ }
+ clk_rate = clk_get_rate(clksrc);
+
+ ratio = clk_rate / freq;
+ if (ratio * freq > clk_rate)
+ ret = ratio * freq - clk_rate;
+ else if (ratio * freq < clk_rate)
+ ret = clk_rate - ratio * freq;
+ else
+ ret = 0;
+
+ /* Block if clock source can not be divided into the required rate */
+ if (ret != 0 && clk_rate / ret < 1000) {
+ dev_err(dai->dev, "failed to derive required HCK%c rate\n",
+ tx ? 'T' : 'R');
+ return -EINVAL;
+ }
+
+ /* Only EXTAL source can be output directly without using PSR and PM */
+ if (ratio == 1 && clksrc == esai_priv->extalclk) {
+ /* Bypass all the dividers if not being needed */
+ ecr |= tx ? ESAI_ECR_ETO : ESAI_ECR_ERO;
+ goto out;
+ } else if (ratio < 2) {
+ /* The ratio should be no less than 2 if using other sources */
+ dev_err(dai->dev, "failed to derive required HCK%c rate\n",
+ tx ? 'T' : 'R');
+ return -EINVAL;
+ }
+
+ ret = fsl_esai_divisor_cal(dai, tx, ratio, false, 0);
+ if (ret)
+ return ret;
+
+ esai_priv->sck_div[tx] = false;
+
+out:
+ esai_priv->hck_dir[tx] = dir;
+ esai_priv->hck_rate[tx] = freq;
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR,
+ tx ? ESAI_ECR_ETI | ESAI_ECR_ETO :
+ ESAI_ECR_ERI | ESAI_ECR_ERO, ecr);
+
+ return 0;
+}
+
+/**
+ * This function configures the related dividers according to the bclk rate
+ */
+static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+ u32 hck_rate = esai_priv->hck_rate[tx];
+ u32 sub, ratio = hck_rate / freq;
+ int ret;
+
+ /* Don't apply for fully slave mode or unchanged bclk */
+ if (esai_priv->slave_mode || esai_priv->sck_rate[tx] == freq)
+ return 0;
+
+ if (ratio * freq > hck_rate)
+ sub = ratio * freq - hck_rate;
+ else if (ratio * freq < hck_rate)
+ sub = hck_rate - ratio * freq;
+ else
+ sub = 0;
+
+ /* Block if clock source can not be divided into the required rate */
+ if (sub != 0 && hck_rate / sub < 1000) {
+ dev_err(dai->dev, "failed to derive required SCK%c rate\n",
+ tx ? 'T' : 'R');
+ return -EINVAL;
+ }
+
+ /* The ratio should be contented by FP alone if bypassing PM and PSR */
+ if (!esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) {
+ dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n");
+ return -EINVAL;
+ }
+
+ ret = fsl_esai_divisor_cal(dai, tx, ratio, true,
+ esai_priv->sck_div[tx] ? 0 : ratio);
+ if (ret)
+ return ret;
+
+ /* Save current bclk rate */
+ esai_priv->sck_rate[tx] = freq;
+
+ return 0;
+}
+
+static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
+ u32 rx_mask, int slots, int slot_width)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
+ ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA,
+ ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask));
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB,
+ ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask));
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
+ ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA,
+ ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask));
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB,
+ ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
+
+ esai_priv->slot_width = slot_width;
+ esai_priv->slots = slots;
+
+ return 0;
+}
+
+static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+ u32 xcr = 0, xccr = 0, mask;
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* Data on rising edge of bclk, frame low, 1clk before data */
+ xcr |= ESAI_xCR_xFSR;
+ xccr |= ESAI_xCCR_xFSP | ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /* Data on rising edge of bclk, frame high */
+ xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ /* Data on rising edge of bclk, frame high, right aligned */
+ xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* Data on rising edge of bclk, frame high, 1clk before data */
+ xcr |= ESAI_xCR_xFSL | ESAI_xCR_xFSR;
+ xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ /* Data on rising edge of bclk, frame high */
+ xcr |= ESAI_xCR_xFSL;
+ xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ /* Nothing to do for both normal cases */
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ /* Invert bit clock */
+ xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ /* Invert frame clock */
+ xccr ^= ESAI_xCCR_xFSP;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ /* Invert both clocks */
+ xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ esai_priv->slave_mode = false;
+
+ /* DAI clock master masks */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ esai_priv->slave_mode = true;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ xccr |= ESAI_xCCR_xCKD;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ xccr |= ESAI_xCCR_xFSD;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR;
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr);
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr);
+
+ mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP |
+ ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA;
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr);
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr);
+
+ return 0;
+}
+
+static int fsl_esai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+ int ret;
+
+ /*
+ * Some platforms might use the same bit to gate all three or two of
+ * clocks, so keep all clocks open/close at the same time for safety
+ */
+ ret = clk_prepare_enable(esai_priv->coreclk);
+ if (ret)
+ return ret;
+ if (!IS_ERR(esai_priv->extalclk)) {
+ ret = clk_prepare_enable(esai_priv->extalclk);
+ if (ret)
+ goto err_extalck;
+ }
+ if (!IS_ERR(esai_priv->fsysclk)) {
+ ret = clk_prepare_enable(esai_priv->fsysclk);
+ if (ret)
+ goto err_fsysclk;
+ }
+
+ if (!dai->active) {
+ /* Set synchronous mode */
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR,
+ ESAI_SAICR_SYNC, esai_priv->synchronous ?
+ ESAI_SAICR_SYNC : 0);
+
+ /* Set a default slot number -- 2 */
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
+ ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
+ ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+ }
+
+ return 0;
+
+err_fsysclk:
+ if (!IS_ERR(esai_priv->extalclk))
+ clk_disable_unprepare(esai_priv->extalclk);
+err_extalck:
+ clk_disable_unprepare(esai_priv->coreclk);
+
+ return ret;
+}
+
+static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ u32 width = snd_pcm_format_width(params_format(params));
+ u32 channels = params_channels(params);
+ u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
+ u32 slot_width = width;
+ u32 bclk, mask, val;
+ int ret;
+
+ /* Override slot_width if being specifially set */
+ if (esai_priv->slot_width)
+ slot_width = esai_priv->slot_width;
+
+ bclk = params_rate(params) * slot_width * esai_priv->slots;
+
+ ret = fsl_esai_set_bclk(dai, tx, bclk);
+ if (ret)
+ return ret;
+
+ /* Use Normal mode to support monaural audio */
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+ ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ?
+ ESAI_xCR_xMOD_NETWORK : 0);
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+ ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR);
+
+ mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK |
+ (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK);
+ val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) |
+ (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins));
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
+
+ mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0);
+ val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0);
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
+
+ /* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
+ ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO));
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
+ ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO));
+ return 0;
+}
+
+static void fsl_esai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+
+ if (!IS_ERR(esai_priv->fsysclk))
+ clk_disable_unprepare(esai_priv->fsysclk);
+ if (!IS_ERR(esai_priv->extalclk))
+ clk_disable_unprepare(esai_priv->extalclk);
+ clk_disable_unprepare(esai_priv->coreclk);
+}
+
+static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ u8 i, channels = substream->runtime->channels;
+ u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+ ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN);
+
+ /* Write initial words reqiured by ESAI as normal procedure */
+ for (i = 0; tx && i < channels; i++)
+ regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0);
+
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+ tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
+ tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+ tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0);
+
+ /* Disable and reset FIFO */
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+ ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR);
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+ ESAI_xFCR_xFR, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops fsl_esai_dai_ops = {
+ .startup = fsl_esai_startup,
+ .shutdown = fsl_esai_shutdown,
+ .trigger = fsl_esai_trigger,
+ .hw_params = fsl_esai_hw_params,
+ .set_sysclk = fsl_esai_set_dai_sysclk,
+ .set_fmt = fsl_esai_set_dai_fmt,
+ .set_tdm_slot = fsl_esai_set_dai_tdm_slot,
+};
+
+static int fsl_esai_dai_probe(struct snd_soc_dai *dai)
+{
+ struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_init_dma_data(dai, &esai_priv->dma_params_tx,
+ &esai_priv->dma_params_rx);
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver fsl_esai_dai = {
+ .probe = fsl_esai_dai_probe,
+ .playback = {
+ .stream_name = "CPU-Playback",
+ .channels_min = 1,
+ .channels_max = 12,
+ .rates = FSL_ESAI_RATES,
+ .formats = FSL_ESAI_FORMATS,
+ },
+ .capture = {
+ .stream_name = "CPU-Capture",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = FSL_ESAI_RATES,
+ .formats = FSL_ESAI_FORMATS,
+ },
+ .ops = &fsl_esai_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_esai_component = {
+ .name = "fsl-esai",
+};
+
+static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ESAI_ERDR:
+ case REG_ESAI_ECR:
+ case REG_ESAI_ESR:
+ case REG_ESAI_TFCR:
+ case REG_ESAI_TFSR:
+ case REG_ESAI_RFCR:
+ case REG_ESAI_RFSR:
+ case REG_ESAI_RX0:
+ case REG_ESAI_RX1:
+ case REG_ESAI_RX2:
+ case REG_ESAI_RX3:
+ case REG_ESAI_SAISR:
+ case REG_ESAI_SAICR:
+ case REG_ESAI_TCR:
+ case REG_ESAI_TCCR:
+ case REG_ESAI_RCR:
+ case REG_ESAI_RCCR:
+ case REG_ESAI_TSMA:
+ case REG_ESAI_TSMB:
+ case REG_ESAI_RSMA:
+ case REG_ESAI_RSMB:
+ case REG_ESAI_PRRC:
+ case REG_ESAI_PCRC:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_ESAI_ETDR:
+ case REG_ESAI_ECR:
+ case REG_ESAI_TFCR:
+ case REG_ESAI_RFCR:
+ case REG_ESAI_TX0:
+ case REG_ESAI_TX1:
+ case REG_ESAI_TX2:
+ case REG_ESAI_TX3:
+ case REG_ESAI_TX4:
+ case REG_ESAI_TX5:
+ case REG_ESAI_TSR:
+ case REG_ESAI_SAICR:
+ case REG_ESAI_TCR:
+ case REG_ESAI_TCCR:
+ case REG_ESAI_RCR:
+ case REG_ESAI_RCCR:
+ case REG_ESAI_TSMA:
+ case REG_ESAI_TSMB:
+ case REG_ESAI_RSMA:
+ case REG_ESAI_RSMB:
+ case REG_ESAI_PRRC:
+ case REG_ESAI_PCRC:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config fsl_esai_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+
+ .max_register = REG_ESAI_PCRC,
+ .readable_reg = fsl_esai_readable_reg,
+ .writeable_reg = fsl_esai_writeable_reg,
+};
+
+static int fsl_esai_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct fsl_esai *esai_priv;
+ struct resource *res;
+ const uint32_t *iprop;
+ void __iomem *regs;
+ int irq, ret;
+
+ esai_priv = devm_kzalloc(&pdev->dev, sizeof(*esai_priv), GFP_KERNEL);
+ if (!esai_priv)
+ return -ENOMEM;
+
+ esai_priv->pdev = pdev;
+ strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1);
+
+ /* Get the addresses and IRQ */
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ esai_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+ "core", regs, &fsl_esai_regmap_config);
+ if (IS_ERR(esai_priv->regmap)) {
+ dev_err(&pdev->dev, "failed to init regmap: %ld\n",
+ PTR_ERR(esai_priv->regmap));
+ return PTR_ERR(esai_priv->regmap);
+ }
+
+ esai_priv->coreclk = devm_clk_get(&pdev->dev, "core");
+ if (IS_ERR(esai_priv->coreclk)) {
+ dev_err(&pdev->dev, "failed to get core clock: %ld\n",
+ PTR_ERR(esai_priv->coreclk));
+ return PTR_ERR(esai_priv->coreclk);
+ }
+
+ esai_priv->extalclk = devm_clk_get(&pdev->dev, "extal");
+ if (IS_ERR(esai_priv->extalclk))
+ dev_warn(&pdev->dev, "failed to get extal clock: %ld\n",
+ PTR_ERR(esai_priv->extalclk));
+
+ esai_priv->fsysclk = devm_clk_get(&pdev->dev, "fsys");
+ if (IS_ERR(esai_priv->fsysclk))
+ dev_warn(&pdev->dev, "failed to get fsys clock: %ld\n",
+ PTR_ERR(esai_priv->fsysclk));
+
+ irq = platform_get_irq(pdev, 0);
+ if (irq < 0) {
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+ return irq;
+ }
+
+ ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0,
+ esai_priv->name, esai_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to claim irq %u\n", irq);
+ return ret;
+ }
+
+ /* Set a default slot number */
+ esai_priv->slots = 2;
+
+ /* Set a default master/slave state */
+ esai_priv->slave_mode = true;
+
+ /* Determine the FIFO depth */
+ iprop = of_get_property(np, "fsl,fifo-depth", NULL);
+ if (iprop)
+ esai_priv->fifo_depth = be32_to_cpup(iprop);
+ else
+ esai_priv->fifo_depth = 64;
+
+ esai_priv->dma_params_tx.maxburst = 16;
+ esai_priv->dma_params_rx.maxburst = 16;
+ esai_priv->dma_params_tx.addr = res->start + REG_ESAI_ETDR;
+ esai_priv->dma_params_rx.addr = res->start + REG_ESAI_ERDR;
+
+ esai_priv->synchronous =
+ of_property_read_bool(np, "fsl,esai-synchronous");
+
+ /* Implement full symmetry for synchronous mode */
+ if (esai_priv->synchronous) {
+ fsl_esai_dai.symmetric_rates = 1;
+ fsl_esai_dai.symmetric_channels = 1;
+ fsl_esai_dai.symmetric_samplebits = 1;
+ }
+
+ dev_set_drvdata(&pdev->dev, esai_priv);
+
+ /* Reset ESAI unit */
+ ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to reset ESAI: %d\n", ret);
+ return ret;
+ }
+
+ /*
+ * We need to enable ESAI so as to access some of its registers.
+ * Otherwise, we would fail to dump regmap from user space.
+ */
+ ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to enable ESAI: %d\n", ret);
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component,
+ &fsl_esai_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+ return ret;
+ }
+
+ ret = imx_pcm_dma_init(pdev);
+ if (ret)
+ dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
+
+ return ret;
+}
+
+static const struct of_device_id fsl_esai_dt_ids[] = {
+ { .compatible = "fsl,imx35-esai", },
+ { .compatible = "fsl,vf610-esai", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids);
+
+static struct platform_driver fsl_esai_driver = {
+ .probe = fsl_esai_probe,
+ .driver = {
+ .name = "fsl-esai-dai",
+ .of_match_table = fsl_esai_dt_ids,
+ },
+};
+
+module_platform_driver(fsl_esai_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale ESAI CPU DAI driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:fsl-esai-dai");
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
new file mode 100644
index 000000000..5e793bbb6
--- /dev/null
+++ b/sound/soc/fsl/fsl_esai.h
@@ -0,0 +1,354 @@
+/*
+ * fsl_esai.h - ALSA ESAI interface for the Freescale i.MX SoC
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <Guangyu.Chen@freescale.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _FSL_ESAI_DAI_H
+#define _FSL_ESAI_DAI_H
+
+/* ESAI Register Map */
+#define REG_ESAI_ETDR 0x00
+#define REG_ESAI_ERDR 0x04
+#define REG_ESAI_ECR 0x08
+#define REG_ESAI_ESR 0x0C
+#define REG_ESAI_TFCR 0x10
+#define REG_ESAI_TFSR 0x14
+#define REG_ESAI_RFCR 0x18
+#define REG_ESAI_RFSR 0x1C
+#define REG_ESAI_xFCR(tx) (tx ? REG_ESAI_TFCR : REG_ESAI_RFCR)
+#define REG_ESAI_xFSR(tx) (tx ? REG_ESAI_TFSR : REG_ESAI_RFSR)
+#define REG_ESAI_TX0 0x80
+#define REG_ESAI_TX1 0x84
+#define REG_ESAI_TX2 0x88
+#define REG_ESAI_TX3 0x8C
+#define REG_ESAI_TX4 0x90
+#define REG_ESAI_TX5 0x94
+#define REG_ESAI_TSR 0x98
+#define REG_ESAI_RX0 0xA0
+#define REG_ESAI_RX1 0xA4
+#define REG_ESAI_RX2 0xA8
+#define REG_ESAI_RX3 0xAC
+#define REG_ESAI_SAISR 0xCC
+#define REG_ESAI_SAICR 0xD0
+#define REG_ESAI_TCR 0xD4
+#define REG_ESAI_TCCR 0xD8
+#define REG_ESAI_RCR 0xDC
+#define REG_ESAI_RCCR 0xE0
+#define REG_ESAI_xCR(tx) (tx ? REG_ESAI_TCR : REG_ESAI_RCR)
+#define REG_ESAI_xCCR(tx) (tx ? REG_ESAI_TCCR : REG_ESAI_RCCR)
+#define REG_ESAI_TSMA 0xE4
+#define REG_ESAI_TSMB 0xE8
+#define REG_ESAI_RSMA 0xEC
+#define REG_ESAI_RSMB 0xF0
+#define REG_ESAI_xSMA(tx) (tx ? REG_ESAI_TSMA : REG_ESAI_RSMA)
+#define REG_ESAI_xSMB(tx) (tx ? REG_ESAI_TSMB : REG_ESAI_RSMB)
+#define REG_ESAI_PRRC 0xF8
+#define REG_ESAI_PCRC 0xFC
+
+/* ESAI Control Register -- REG_ESAI_ECR 0x8 */
+#define ESAI_ECR_ETI_SHIFT 19
+#define ESAI_ECR_ETI_MASK (1 << ESAI_ECR_ETI_SHIFT)
+#define ESAI_ECR_ETI (1 << ESAI_ECR_ETI_SHIFT)
+#define ESAI_ECR_ETO_SHIFT 18
+#define ESAI_ECR_ETO_MASK (1 << ESAI_ECR_ETO_SHIFT)
+#define ESAI_ECR_ETO (1 << ESAI_ECR_ETO_SHIFT)
+#define ESAI_ECR_ERI_SHIFT 17
+#define ESAI_ECR_ERI_MASK (1 << ESAI_ECR_ERI_SHIFT)
+#define ESAI_ECR_ERI (1 << ESAI_ECR_ERI_SHIFT)
+#define ESAI_ECR_ERO_SHIFT 16
+#define ESAI_ECR_ERO_MASK (1 << ESAI_ECR_ERO_SHIFT)
+#define ESAI_ECR_ERO (1 << ESAI_ECR_ERO_SHIFT)
+#define ESAI_ECR_ERST_SHIFT 1
+#define ESAI_ECR_ERST_MASK (1 << ESAI_ECR_ERST_SHIFT)
+#define ESAI_ECR_ERST (1 << ESAI_ECR_ERST_SHIFT)
+#define ESAI_ECR_ESAIEN_SHIFT 0
+#define ESAI_ECR_ESAIEN_MASK (1 << ESAI_ECR_ESAIEN_SHIFT)
+#define ESAI_ECR_ESAIEN (1 << ESAI_ECR_ESAIEN_SHIFT)
+
+/* ESAI Status Register -- REG_ESAI_ESR 0xC */
+#define ESAI_ESR_TINIT_SHIFT 10
+#define ESAI_ESR_TINIT_MASK (1 << ESAI_ESR_TINIT_SHIFT)
+#define ESAI_ESR_TINIT (1 << ESAI_ESR_TINIT_SHIFT)
+#define ESAI_ESR_RFF_SHIFT 9
+#define ESAI_ESR_RFF_MASK (1 << ESAI_ESR_RFF_SHIFT)
+#define ESAI_ESR_RFF (1 << ESAI_ESR_RFF_SHIFT)
+#define ESAI_ESR_TFE_SHIFT 8
+#define ESAI_ESR_TFE_MASK (1 << ESAI_ESR_TFE_SHIFT)
+#define ESAI_ESR_TFE (1 << ESAI_ESR_TFE_SHIFT)
+#define ESAI_ESR_TLS_SHIFT 7
+#define ESAI_ESR_TLS_MASK (1 << ESAI_ESR_TLS_SHIFT)
+#define ESAI_ESR_TLS (1 << ESAI_ESR_TLS_SHIFT)
+#define ESAI_ESR_TDE_SHIFT 6
+#define ESAI_ESR_TDE_MASK (1 << ESAI_ESR_TDE_SHIFT)
+#define ESAI_ESR_TDE (1 << ESAI_ESR_TDE_SHIFT)
+#define ESAI_ESR_TED_SHIFT 5
+#define ESAI_ESR_TED_MASK (1 << ESAI_ESR_TED_SHIFT)
+#define ESAI_ESR_TED (1 << ESAI_ESR_TED_SHIFT)
+#define ESAI_ESR_TD_SHIFT 4
+#define ESAI_ESR_TD_MASK (1 << ESAI_ESR_TD_SHIFT)
+#define ESAI_ESR_TD (1 << ESAI_ESR_TD_SHIFT)
+#define ESAI_ESR_RLS_SHIFT 3
+#define ESAI_ESR_RLS_MASK (1 << ESAI_ESR_RLS_SHIFT)
+#define ESAI_ESR_RLS (1 << ESAI_ESR_RLS_SHIFT)
+#define ESAI_ESR_RDE_SHIFT 2
+#define ESAI_ESR_RDE_MASK (1 << ESAI_ESR_RDE_SHIFT)
+#define ESAI_ESR_RDE (1 << ESAI_ESR_RDE_SHIFT)
+#define ESAI_ESR_RED_SHIFT 1
+#define ESAI_ESR_RED_MASK (1 << ESAI_ESR_RED_SHIFT)
+#define ESAI_ESR_RED (1 << ESAI_ESR_RED_SHIFT)
+#define ESAI_ESR_RD_SHIFT 0
+#define ESAI_ESR_RD_MASK (1 << ESAI_ESR_RD_SHIFT)
+#define ESAI_ESR_RD (1 << ESAI_ESR_RD_SHIFT)
+
+/*
+ * Transmit FIFO Configuration Register -- REG_ESAI_TFCR 0x10
+ * Receive FIFO Configuration Register -- REG_ESAI_RFCR 0x18
+ */
+#define ESAI_xFCR_TIEN_SHIFT 19
+#define ESAI_xFCR_TIEN_MASK (1 << ESAI_xFCR_TIEN_SHIFT)
+#define ESAI_xFCR_TIEN (1 << ESAI_xFCR_TIEN_SHIFT)
+#define ESAI_xFCR_REXT_SHIFT 19
+#define ESAI_xFCR_REXT_MASK (1 << ESAI_xFCR_REXT_SHIFT)
+#define ESAI_xFCR_REXT (1 << ESAI_xFCR_REXT_SHIFT)
+#define ESAI_xFCR_xWA_SHIFT 16
+#define ESAI_xFCR_xWA_WIDTH 3
+#define ESAI_xFCR_xWA_MASK (((1 << ESAI_xFCR_xWA_WIDTH) - 1) << ESAI_xFCR_xWA_SHIFT)
+#define ESAI_xFCR_xWA(v) (((8 - ((v) >> 2)) << ESAI_xFCR_xWA_SHIFT) & ESAI_xFCR_xWA_MASK)
+#define ESAI_xFCR_xFWM_SHIFT 8
+#define ESAI_xFCR_xFWM_WIDTH 8
+#define ESAI_xFCR_xFWM_MASK (((1 << ESAI_xFCR_xFWM_WIDTH) - 1) << ESAI_xFCR_xFWM_SHIFT)
+#define ESAI_xFCR_xFWM(v) ((((v) - 1) << ESAI_xFCR_xFWM_SHIFT) & ESAI_xFCR_xFWM_MASK)
+#define ESAI_xFCR_xE_SHIFT 2
+#define ESAI_xFCR_TE_WIDTH 6
+#define ESAI_xFCR_RE_WIDTH 4
+#define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
+#define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
+#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK)
+#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK)
+#define ESAI_xFCR_xFR_SHIFT 1
+#define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT)
+#define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT)
+#define ESAI_xFCR_xFEN_SHIFT 0
+#define ESAI_xFCR_xFEN_MASK (1 << ESAI_xFCR_xFEN_SHIFT)
+#define ESAI_xFCR_xFEN (1 << ESAI_xFCR_xFEN_SHIFT)
+
+/*
+ * Transmit FIFO Status Register -- REG_ESAI_TFSR 0x14
+ * Receive FIFO Status Register --REG_ESAI_RFSR 0x1C
+ */
+#define ESAI_xFSR_NTFO_SHIFT 12
+#define ESAI_xFSR_NRFI_SHIFT 12
+#define ESAI_xFSR_NTFI_SHIFT 8
+#define ESAI_xFSR_NRFO_SHIFT 8
+#define ESAI_xFSR_NTFx_WIDTH 3
+#define ESAI_xFSR_NRFx_WIDTH 2
+#define ESAI_xFSR_NTFO_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFO_SHIFT)
+#define ESAI_xFSR_NTFI_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFI_SHIFT)
+#define ESAI_xFSR_NRFO_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFO_SHIFT)
+#define ESAI_xFSR_NRFI_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFI_SHIFT)
+#define ESAI_xFSR_xFCNT_SHIFT 0
+#define ESAI_xFSR_xFCNT_WIDTH 8
+#define ESAI_xFSR_xFCNT_MASK (((1 << ESAI_xFSR_xFCNT_WIDTH) - 1) << ESAI_xFSR_xFCNT_SHIFT)
+
+/* ESAI Transmit Slot Register -- REG_ESAI_TSR 0x98 */
+#define ESAI_TSR_SHIFT 0
+#define ESAI_TSR_WIDTH 24
+#define ESAI_TSR_MASK (((1 << ESAI_TSR_WIDTH) - 1) << ESAI_TSR_SHIFT)
+
+/* Serial Audio Interface Status Register -- REG_ESAI_SAISR 0xCC */
+#define ESAI_SAISR_TODFE_SHIFT 17
+#define ESAI_SAISR_TODFE_MASK (1 << ESAI_SAISR_TODFE_SHIFT)
+#define ESAI_SAISR_TODFE (1 << ESAI_SAISR_TODFE_SHIFT)
+#define ESAI_SAISR_TEDE_SHIFT 16
+#define ESAI_SAISR_TEDE_MASK (1 << ESAI_SAISR_TEDE_SHIFT)
+#define ESAI_SAISR_TEDE (1 << ESAI_SAISR_TEDE_SHIFT)
+#define ESAI_SAISR_TDE_SHIFT 15
+#define ESAI_SAISR_TDE_MASK (1 << ESAI_SAISR_TDE_SHIFT)
+#define ESAI_SAISR_TDE (1 << ESAI_SAISR_TDE_SHIFT)
+#define ESAI_SAISR_TUE_SHIFT 14
+#define ESAI_SAISR_TUE_MASK (1 << ESAI_SAISR_TUE_SHIFT)
+#define ESAI_SAISR_TUE (1 << ESAI_SAISR_TUE_SHIFT)
+#define ESAI_SAISR_TFS_SHIFT 13
+#define ESAI_SAISR_TFS_MASK (1 << ESAI_SAISR_TFS_SHIFT)
+#define ESAI_SAISR_TFS (1 << ESAI_SAISR_TFS_SHIFT)
+#define ESAI_SAISR_RODF_SHIFT 10
+#define ESAI_SAISR_RODF_MASK (1 << ESAI_SAISR_RODF_SHIFT)
+#define ESAI_SAISR_RODF (1 << ESAI_SAISR_RODF_SHIFT)
+#define ESAI_SAISR_REDF_SHIFT 9
+#define ESAI_SAISR_REDF_MASK (1 << ESAI_SAISR_REDF_SHIFT)
+#define ESAI_SAISR_REDF (1 << ESAI_SAISR_REDF_SHIFT)
+#define ESAI_SAISR_RDF_SHIFT 8
+#define ESAI_SAISR_RDF_MASK (1 << ESAI_SAISR_RDF_SHIFT)
+#define ESAI_SAISR_RDF (1 << ESAI_SAISR_RDF_SHIFT)
+#define ESAI_SAISR_ROE_SHIFT 7
+#define ESAI_SAISR_ROE_MASK (1 << ESAI_SAISR_ROE_SHIFT)
+#define ESAI_SAISR_ROE (1 << ESAI_SAISR_ROE_SHIFT)
+#define ESAI_SAISR_RFS_SHIFT 6
+#define ESAI_SAISR_RFS_MASK (1 << ESAI_SAISR_RFS_SHIFT)
+#define ESAI_SAISR_RFS (1 << ESAI_SAISR_RFS_SHIFT)
+#define ESAI_SAISR_IF2_SHIFT 2
+#define ESAI_SAISR_IF2_MASK (1 << ESAI_SAISR_IF2_SHIFT)
+#define ESAI_SAISR_IF2 (1 << ESAI_SAISR_IF2_SHIFT)
+#define ESAI_SAISR_IF1_SHIFT 1
+#define ESAI_SAISR_IF1_MASK (1 << ESAI_SAISR_IF1_SHIFT)
+#define ESAI_SAISR_IF1 (1 << ESAI_SAISR_IF1_SHIFT)
+#define ESAI_SAISR_IF0_SHIFT 0
+#define ESAI_SAISR_IF0_MASK (1 << ESAI_SAISR_IF0_SHIFT)
+#define ESAI_SAISR_IF0 (1 << ESAI_SAISR_IF0_SHIFT)
+
+/* Serial Audio Interface Control Register -- REG_ESAI_SAICR 0xD0 */
+#define ESAI_SAICR_ALC_SHIFT 8
+#define ESAI_SAICR_ALC_MASK (1 << ESAI_SAICR_ALC_SHIFT)
+#define ESAI_SAICR_ALC (1 << ESAI_SAICR_ALC_SHIFT)
+#define ESAI_SAICR_TEBE_SHIFT 7
+#define ESAI_SAICR_TEBE_MASK (1 << ESAI_SAICR_TEBE_SHIFT)
+#define ESAI_SAICR_TEBE (1 << ESAI_SAICR_TEBE_SHIFT)
+#define ESAI_SAICR_SYNC_SHIFT 6
+#define ESAI_SAICR_SYNC_MASK (1 << ESAI_SAICR_SYNC_SHIFT)
+#define ESAI_SAICR_SYNC (1 << ESAI_SAICR_SYNC_SHIFT)
+#define ESAI_SAICR_OF2_SHIFT 2
+#define ESAI_SAICR_OF2_MASK (1 << ESAI_SAICR_OF2_SHIFT)
+#define ESAI_SAICR_OF2 (1 << ESAI_SAICR_OF2_SHIFT)
+#define ESAI_SAICR_OF1_SHIFT 1
+#define ESAI_SAICR_OF1_MASK (1 << ESAI_SAICR_OF1_SHIFT)
+#define ESAI_SAICR_OF1 (1 << ESAI_SAICR_OF1_SHIFT)
+#define ESAI_SAICR_OF0_SHIFT 0
+#define ESAI_SAICR_OF0_MASK (1 << ESAI_SAICR_OF0_SHIFT)
+#define ESAI_SAICR_OF0 (1 << ESAI_SAICR_OF0_SHIFT)
+
+/*
+ * Transmit Control Register -- REG_ESAI_TCR 0xD4
+ * Receive Control Register -- REG_ESAI_RCR 0xDC
+ */
+#define ESAI_xCR_xLIE_SHIFT 23
+#define ESAI_xCR_xLIE_MASK (1 << ESAI_xCR_xLIE_SHIFT)
+#define ESAI_xCR_xLIE (1 << ESAI_xCR_xLIE_SHIFT)
+#define ESAI_xCR_xIE_SHIFT 22
+#define ESAI_xCR_xIE_MASK (1 << ESAI_xCR_xIE_SHIFT)
+#define ESAI_xCR_xIE (1 << ESAI_xCR_xIE_SHIFT)
+#define ESAI_xCR_xEDIE_SHIFT 21
+#define ESAI_xCR_xEDIE_MASK (1 << ESAI_xCR_xEDIE_SHIFT)
+#define ESAI_xCR_xEDIE (1 << ESAI_xCR_xEDIE_SHIFT)
+#define ESAI_xCR_xEIE_SHIFT 20
+#define ESAI_xCR_xEIE_MASK (1 << ESAI_xCR_xEIE_SHIFT)
+#define ESAI_xCR_xEIE (1 << ESAI_xCR_xEIE_SHIFT)
+#define ESAI_xCR_xPR_SHIFT 19
+#define ESAI_xCR_xPR_MASK (1 << ESAI_xCR_xPR_SHIFT)
+#define ESAI_xCR_xPR (1 << ESAI_xCR_xPR_SHIFT)
+#define ESAI_xCR_PADC_SHIFT 17
+#define ESAI_xCR_PADC_MASK (1 << ESAI_xCR_PADC_SHIFT)
+#define ESAI_xCR_PADC (1 << ESAI_xCR_PADC_SHIFT)
+#define ESAI_xCR_xFSR_SHIFT 16
+#define ESAI_xCR_xFSR_MASK (1 << ESAI_xCR_xFSR_SHIFT)
+#define ESAI_xCR_xFSR (1 << ESAI_xCR_xFSR_SHIFT)
+#define ESAI_xCR_xFSL_SHIFT 15
+#define ESAI_xCR_xFSL_MASK (1 << ESAI_xCR_xFSL_SHIFT)
+#define ESAI_xCR_xFSL (1 << ESAI_xCR_xFSL_SHIFT)
+#define ESAI_xCR_xSWS_SHIFT 10
+#define ESAI_xCR_xSWS_WIDTH 5
+#define ESAI_xCR_xSWS_MASK (((1 << ESAI_xCR_xSWS_WIDTH) - 1) << ESAI_xCR_xSWS_SHIFT)
+#define ESAI_xCR_xSWS(s, w) ((w < 24 ? (s - w + ((w - 8) >> 2)) : (s < 32 ? 0x1e : 0x1f)) << ESAI_xCR_xSWS_SHIFT)
+#define ESAI_xCR_xMOD_SHIFT 8
+#define ESAI_xCR_xMOD_WIDTH 2
+#define ESAI_xCR_xMOD_MASK (((1 << ESAI_xCR_xMOD_WIDTH) - 1) << ESAI_xCR_xMOD_SHIFT)
+#define ESAI_xCR_xMOD_ONDEMAND (0x1 << ESAI_xCR_xMOD_SHIFT)
+#define ESAI_xCR_xMOD_NETWORK (0x1 << ESAI_xCR_xMOD_SHIFT)
+#define ESAI_xCR_xMOD_AC97 (0x3 << ESAI_xCR_xMOD_SHIFT)
+#define ESAI_xCR_xWA_SHIFT 7
+#define ESAI_xCR_xWA_MASK (1 << ESAI_xCR_xWA_SHIFT)
+#define ESAI_xCR_xWA (1 << ESAI_xCR_xWA_SHIFT)
+#define ESAI_xCR_xSHFD_SHIFT 6
+#define ESAI_xCR_xSHFD_MASK (1 << ESAI_xCR_xSHFD_SHIFT)
+#define ESAI_xCR_xSHFD (1 << ESAI_xCR_xSHFD_SHIFT)
+#define ESAI_xCR_xE_SHIFT 0
+#define ESAI_xCR_TE_WIDTH 6
+#define ESAI_xCR_RE_WIDTH 4
+#define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
+#define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
+#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK)
+#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK)
+
+/*
+ * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8
+ * Receive Clock Control Register -- REG_ESAI_RCCR 0xE0
+ */
+#define ESAI_xCCR_xHCKD_SHIFT 23
+#define ESAI_xCCR_xHCKD_MASK (1 << ESAI_xCCR_xHCKD_SHIFT)
+#define ESAI_xCCR_xHCKD (1 << ESAI_xCCR_xHCKD_SHIFT)
+#define ESAI_xCCR_xFSD_SHIFT 22
+#define ESAI_xCCR_xFSD_MASK (1 << ESAI_xCCR_xFSD_SHIFT)
+#define ESAI_xCCR_xFSD (1 << ESAI_xCCR_xFSD_SHIFT)
+#define ESAI_xCCR_xCKD_SHIFT 21
+#define ESAI_xCCR_xCKD_MASK (1 << ESAI_xCCR_xCKD_SHIFT)
+#define ESAI_xCCR_xCKD (1 << ESAI_xCCR_xCKD_SHIFT)
+#define ESAI_xCCR_xHCKP_SHIFT 20
+#define ESAI_xCCR_xHCKP_MASK (1 << ESAI_xCCR_xHCKP_SHIFT)
+#define ESAI_xCCR_xHCKP (1 << ESAI_xCCR_xHCKP_SHIFT)
+#define ESAI_xCCR_xFSP_SHIFT 19
+#define ESAI_xCCR_xFSP_MASK (1 << ESAI_xCCR_xFSP_SHIFT)
+#define ESAI_xCCR_xFSP (1 << ESAI_xCCR_xFSP_SHIFT)
+#define ESAI_xCCR_xCKP_SHIFT 18
+#define ESAI_xCCR_xCKP_MASK (1 << ESAI_xCCR_xCKP_SHIFT)
+#define ESAI_xCCR_xCKP (1 << ESAI_xCCR_xCKP_SHIFT)
+#define ESAI_xCCR_xFP_SHIFT 14
+#define ESAI_xCCR_xFP_WIDTH 4
+#define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT)
+#define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK)
+#define ESAI_xCCR_xDC_SHIFT 9
+#define ESAI_xCCR_xDC_WIDTH 5
+#define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT)
+#define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK)
+#define ESAI_xCCR_xPSR_SHIFT 8
+#define ESAI_xCCR_xPSR_MASK (1 << ESAI_xCCR_xPSR_SHIFT)
+#define ESAI_xCCR_xPSR_BYPASS (1 << ESAI_xCCR_xPSR_SHIFT)
+#define ESAI_xCCR_xPSR_DIV8 (0 << ESAI_xCCR_xPSR_SHIFT)
+#define ESAI_xCCR_xPM_SHIFT 0
+#define ESAI_xCCR_xPM_WIDTH 8
+#define ESAI_xCCR_xPM_MASK (((1 << ESAI_xCCR_xPM_WIDTH) - 1) << ESAI_xCCR_xPM_SHIFT)
+#define ESAI_xCCR_xPM(v) ((((v) - 1) << ESAI_xCCR_xPM_SHIFT) & ESAI_xCCR_xPM_MASK)
+
+/* Transmit Slot Mask Register A/B -- REG_ESAI_TSMA/B 0xE4 ~ 0xF0 */
+#define ESAI_xSMA_xS_SHIFT 0
+#define ESAI_xSMA_xS_WIDTH 16
+#define ESAI_xSMA_xS_MASK (((1 << ESAI_xSMA_xS_WIDTH) - 1) << ESAI_xSMA_xS_SHIFT)
+#define ESAI_xSMA_xS(v) ((v) & ESAI_xSMA_xS_MASK)
+#define ESAI_xSMB_xS_SHIFT 0
+#define ESAI_xSMB_xS_WIDTH 16
+#define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT)
+#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK)
+
+/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */
+#define ESAI_PRRC_PDC_SHIFT 0
+#define ESAI_PRRC_PDC_WIDTH 12
+#define ESAI_PRRC_PDC_MASK (((1 << ESAI_PRRC_PDC_WIDTH) - 1) << ESAI_PRRC_PDC_SHIFT)
+#define ESAI_PRRC_PDC(v) ((v) & ESAI_PRRC_PDC_MASK)
+
+/* Port C Control Register -- REG_ESAI_PCRC 0xFC */
+#define ESAI_PCRC_PC_SHIFT 0
+#define ESAI_PCRC_PC_WIDTH 12
+#define ESAI_PCRC_PC_MASK (((1 << ESAI_PCRC_PC_WIDTH) - 1) << ESAI_PCRC_PC_SHIFT)
+#define ESAI_PCRC_PC(v) ((v) & ESAI_PCRC_PC_MASK)
+
+#define ESAI_GPIO 0xfff
+
+/* ESAI clock source */
+#define ESAI_HCKT_FSYS 0
+#define ESAI_HCKT_EXTAL 1
+#define ESAI_HCKR_FSYS 2
+#define ESAI_HCKR_EXTAL 3
+
+/* ESAI clock divider */
+#define ESAI_TX_DIV_PSR 0
+#define ESAI_TX_DIV_PM 1
+#define ESAI_TX_DIV_FP 2
+#define ESAI_RX_DIV_PSR 3
+#define ESAI_RX_DIV_PM 4
+#define ESAI_RX_DIV_FP 5
+#endif /* _FSL_ESAI_DAI_H */
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
new file mode 100644
index 000000000..ec79c3d5e
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.c
@@ -0,0 +1,689 @@
+/*
+ * Freescale ALSA SoC Digital Audio Interface (SAI) driver.
+ *
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * This program is free software, you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation, either version 2 of the License, or(at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/dmaengine.h>
+#include <linux/module.h>
+#include <linux/of_address.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_sai.h"
+#include "imx-pcm.h"
+
+#define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\
+ FSL_SAI_CSR_FEIE)
+
+static irqreturn_t fsl_sai_isr(int irq, void *devid)
+{
+ struct fsl_sai *sai = (struct fsl_sai *)devid;
+ struct device *dev = &sai->pdev->dev;
+ u32 flags, xcsr, mask;
+ bool irq_none = true;
+
+ /*
+ * Both IRQ status bits and IRQ mask bits are in the xCSR but
+ * different shifts. And we here create a mask only for those
+ * IRQs that we activated.
+ */
+ mask = (FSL_SAI_FLAGS >> FSL_SAI_CSR_xIE_SHIFT) << FSL_SAI_CSR_xF_SHIFT;
+
+ /* Tx IRQ */
+ regmap_read(sai->regmap, FSL_SAI_TCSR, &xcsr);
+ flags = xcsr & mask;
+
+ if (flags)
+ irq_none = false;
+ else
+ goto irq_rx;
+
+ if (flags & FSL_SAI_CSR_WSF)
+ dev_dbg(dev, "isr: Start of Tx word detected\n");
+
+ if (flags & FSL_SAI_CSR_SEF)
+ dev_warn(dev, "isr: Tx Frame sync error detected\n");
+
+ if (flags & FSL_SAI_CSR_FEF) {
+ dev_warn(dev, "isr: Transmit underrun detected\n");
+ /* FIFO reset for safety */
+ xcsr |= FSL_SAI_CSR_FR;
+ }
+
+ if (flags & FSL_SAI_CSR_FWF)
+ dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n");
+
+ if (flags & FSL_SAI_CSR_FRF)
+ dev_dbg(dev, "isr: Transmit FIFO watermark has been reached\n");
+
+ flags &= FSL_SAI_CSR_xF_W_MASK;
+ xcsr &= ~FSL_SAI_CSR_xF_MASK;
+
+ if (flags)
+ regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr);
+
+irq_rx:
+ /* Rx IRQ */
+ regmap_read(sai->regmap, FSL_SAI_RCSR, &xcsr);
+ flags = xcsr & mask;
+
+ if (flags)
+ irq_none = false;
+ else
+ goto out;
+
+ if (flags & FSL_SAI_CSR_WSF)
+ dev_dbg(dev, "isr: Start of Rx word detected\n");
+
+ if (flags & FSL_SAI_CSR_SEF)
+ dev_warn(dev, "isr: Rx Frame sync error detected\n");
+
+ if (flags & FSL_SAI_CSR_FEF) {
+ dev_warn(dev, "isr: Receive overflow detected\n");
+ /* FIFO reset for safety */
+ xcsr |= FSL_SAI_CSR_FR;
+ }
+
+ if (flags & FSL_SAI_CSR_FWF)
+ dev_dbg(dev, "isr: Enabled receive FIFO is full\n");
+
+ if (flags & FSL_SAI_CSR_FRF)
+ dev_dbg(dev, "isr: Receive FIFO watermark has been reached\n");
+
+ flags &= FSL_SAI_CSR_xF_W_MASK;
+ xcsr &= ~FSL_SAI_CSR_xF_MASK;
+
+ if (flags)
+ regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr);
+
+out:
+ if (irq_none)
+ return IRQ_NONE;
+ else
+ return IRQ_HANDLED;
+}
+
+static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int fsl_dir)
+{
+ struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ bool tx = fsl_dir == FSL_FMT_TRANSMITTER;
+ u32 val_cr2 = 0;
+
+ switch (clk_id) {
+ case FSL_SAI_CLK_BUS:
+ val_cr2 |= FSL_SAI_CR2_MSEL_BUS;
+ break;
+ case FSL_SAI_CLK_MAST1:
+ val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1;
+ break;
+ case FSL_SAI_CLK_MAST2:
+ val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2;
+ break;
+ case FSL_SAI_CLK_MAST3:
+ val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx),
+ FSL_SAI_CR2_MSEL_MASK, val_cr2);
+
+ return 0;
+}
+
+static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ int ret;
+
+ if (dir == SND_SOC_CLOCK_IN)
+ return 0;
+
+ ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
+ FSL_FMT_TRANSMITTER);
+ if (ret) {
+ dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret);
+ return ret;
+ }
+
+ ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
+ FSL_FMT_RECEIVER);
+ if (ret)
+ dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret);
+
+ return ret;
+}
+
+static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt, int fsl_dir)
+{
+ struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ bool tx = fsl_dir == FSL_FMT_TRANSMITTER;
+ u32 val_cr2 = 0, val_cr4 = 0;
+
+ if (!sai->is_lsb_first)
+ val_cr4 |= FSL_SAI_CR4_MF;
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /*
+ * Frame low, 1clk before data, one word length for frame sync,
+ * frame sync starts one serial clock cycle earlier,
+ * that is, together with the last bit of the previous
+ * data word.
+ */
+ val_cr2 |= FSL_SAI_CR2_BCP;
+ val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /*
+ * Frame high, one word length for frame sync,
+ * frame sync asserts with the first bit of the frame.
+ */
+ val_cr2 |= FSL_SAI_CR2_BCP;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /*
+ * Frame high, 1clk before data, one bit for frame sync,
+ * frame sync starts one serial clock cycle earlier,
+ * that is, together with the last bit of the previous
+ * data word.
+ */
+ val_cr2 |= FSL_SAI_CR2_BCP;
+ val_cr4 |= FSL_SAI_CR4_FSE;
+ sai->is_dsp_mode = true;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ /*
+ * Frame high, one bit for frame sync,
+ * frame sync asserts with the first bit of the frame.
+ */
+ val_cr2 |= FSL_SAI_CR2_BCP;
+ sai->is_dsp_mode = true;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ /* To be done */
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ /* Invert both clocks */
+ val_cr2 ^= FSL_SAI_CR2_BCP;
+ val_cr4 ^= FSL_SAI_CR4_FSP;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ /* Invert bit clock */
+ val_cr2 ^= FSL_SAI_CR2_BCP;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ /* Invert frame clock */
+ val_cr4 ^= FSL_SAI_CR4_FSP;
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ /* Nothing to do for both normal cases */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI clock master masks */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+ val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx),
+ FSL_SAI_CR2_BCP | FSL_SAI_CR2_BCD_MSTR, val_cr2);
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx),
+ FSL_SAI_CR4_MF | FSL_SAI_CR4_FSE |
+ FSL_SAI_CR4_FSP | FSL_SAI_CR4_FSD_MSTR, val_cr4);
+
+ return 0;
+}
+
+static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ int ret;
+
+ ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER);
+ if (ret) {
+ dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret);
+ return ret;
+ }
+
+ ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER);
+ if (ret)
+ dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret);
+
+ return ret;
+}
+
+static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned int channels = params_channels(params);
+ u32 word_width = snd_pcm_format_width(params_format(params));
+ u32 val_cr4 = 0, val_cr5 = 0;
+
+ if (!sai->is_dsp_mode)
+ val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
+
+ val_cr5 |= FSL_SAI_CR5_WNW(word_width);
+ val_cr5 |= FSL_SAI_CR5_W0W(word_width);
+
+ if (sai->is_lsb_first)
+ val_cr5 |= FSL_SAI_CR5_FBT(0);
+ else
+ val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
+
+ val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
+
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx),
+ FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK,
+ val_cr4);
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx),
+ FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK |
+ FSL_SAI_CR5_FBT_MASK, val_cr5);
+ regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << channels) - 1));
+
+ return 0;
+}
+
+static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ u32 xcsr, count = 100;
+
+ /*
+ * Asynchronous mode: Clear SYNC for both Tx and Rx.
+ * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx.
+ * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx.
+ */
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
+ sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0);
+
+ /*
+ * It is recommended that the transmitter is the last enabled
+ * and the first disabled.
+ */
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+ FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE);
+
+ regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
+ FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE);
+ regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
+ FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE);
+
+ regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+ FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+ FSL_SAI_CSR_FRDE, 0);
+ regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx),
+ FSL_SAI_CSR_xIE_MASK, 0);
+
+ /* Check if the opposite FRDE is also disabled */
+ regmap_read(sai->regmap, FSL_SAI_xCSR(!tx), &xcsr);
+ if (!(xcsr & FSL_SAI_CSR_FRDE)) {
+ /* Disable both directions and reset their FIFOs */
+ regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
+ FSL_SAI_CSR_TERE, 0);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
+ FSL_SAI_CSR_TERE, 0);
+
+ /* TERE will remain set till the end of current frame */
+ do {
+ udelay(10);
+ regmap_read(sai->regmap, FSL_SAI_xCSR(tx), &xcsr);
+ } while (--count && xcsr & FSL_SAI_CSR_TERE);
+
+ regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
+ FSL_SAI_CSR_FR, FSL_SAI_CSR_FR);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
+ FSL_SAI_CSR_FR, FSL_SAI_CSR_FR);
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int fsl_sai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct device *dev = &sai->pdev->dev;
+ int ret;
+
+ ret = clk_prepare_enable(sai->bus_clk);
+ if (ret) {
+ dev_err(dev, "failed to enable bus clock: %d\n", ret);
+ return ret;
+ }
+
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE,
+ FSL_SAI_CR3_TRCE);
+
+ return 0;
+}
+
+static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+ regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, 0);
+
+ clk_disable_unprepare(sai->bus_clk);
+}
+
+static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
+ .set_sysclk = fsl_sai_set_dai_sysclk,
+ .set_fmt = fsl_sai_set_dai_fmt,
+ .hw_params = fsl_sai_hw_params,
+ .trigger = fsl_sai_trigger,
+ .startup = fsl_sai_startup,
+ .shutdown = fsl_sai_shutdown,
+};
+
+static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
+{
+ struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
+
+ /* Software Reset for both Tx and Rx */
+ regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+ /* Clear SR bit to finish the reset */
+ regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_MAXBURST_TX * 2);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_MAXBURST_RX - 1);
+
+ snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
+ &sai->dma_params_rx);
+
+ snd_soc_dai_set_drvdata(cpu_dai, sai);
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver fsl_sai_dai = {
+ .probe = fsl_sai_dai_probe,
+ .playback = {
+ .stream_name = "CPU-Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = FSL_SAI_FORMATS,
+ },
+ .capture = {
+ .stream_name = "CPU-Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = FSL_SAI_FORMATS,
+ },
+ .ops = &fsl_sai_pcm_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_component = {
+ .name = "fsl-sai",
+};
+
+static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TCSR:
+ case FSL_SAI_TCR1:
+ case FSL_SAI_TCR2:
+ case FSL_SAI_TCR3:
+ case FSL_SAI_TCR4:
+ case FSL_SAI_TCR5:
+ case FSL_SAI_TFR:
+ case FSL_SAI_TMR:
+ case FSL_SAI_RCSR:
+ case FSL_SAI_RCR1:
+ case FSL_SAI_RCR2:
+ case FSL_SAI_RCR3:
+ case FSL_SAI_RCR4:
+ case FSL_SAI_RCR5:
+ case FSL_SAI_RDR:
+ case FSL_SAI_RFR:
+ case FSL_SAI_RMR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TFR:
+ case FSL_SAI_RFR:
+ case FSL_SAI_TDR:
+ case FSL_SAI_RDR:
+ return true;
+ default:
+ return false;
+ }
+
+}
+
+static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TCSR:
+ case FSL_SAI_TCR1:
+ case FSL_SAI_TCR2:
+ case FSL_SAI_TCR3:
+ case FSL_SAI_TCR4:
+ case FSL_SAI_TCR5:
+ case FSL_SAI_TDR:
+ case FSL_SAI_TMR:
+ case FSL_SAI_RCSR:
+ case FSL_SAI_RCR1:
+ case FSL_SAI_RCR2:
+ case FSL_SAI_RCR3:
+ case FSL_SAI_RCR4:
+ case FSL_SAI_RCR5:
+ case FSL_SAI_RMR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config fsl_sai_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+
+ .max_register = FSL_SAI_RMR,
+ .readable_reg = fsl_sai_readable_reg,
+ .volatile_reg = fsl_sai_volatile_reg,
+ .writeable_reg = fsl_sai_writeable_reg,
+};
+
+static int fsl_sai_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct fsl_sai *sai;
+ struct resource *res;
+ void __iomem *base;
+ char tmp[8];
+ int irq, ret, i;
+
+ sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
+ if (!sai)
+ return -ENOMEM;
+
+ sai->pdev = pdev;
+
+ if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
+ sai->sai_on_imx = true;
+
+ sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(base))
+ return PTR_ERR(base);
+
+ sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+ "bus", base, &fsl_sai_regmap_config);
+
+ /* Compatible with old DTB cases */
+ if (IS_ERR(sai->regmap))
+ sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+ "sai", base, &fsl_sai_regmap_config);
+ if (IS_ERR(sai->regmap)) {
+ dev_err(&pdev->dev, "regmap init failed\n");
+ return PTR_ERR(sai->regmap);
+ }
+
+ /* No error out for old DTB cases but only mark the clock NULL */
+ sai->bus_clk = devm_clk_get(&pdev->dev, "bus");
+ if (IS_ERR(sai->bus_clk)) {
+ dev_err(&pdev->dev, "failed to get bus clock: %ld\n",
+ PTR_ERR(sai->bus_clk));
+ sai->bus_clk = NULL;
+ }
+
+ for (i = 0; i < FSL_SAI_MCLK_MAX; i++) {
+ sprintf(tmp, "mclk%d", i + 1);
+ sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp);
+ if (IS_ERR(sai->mclk_clk[i])) {
+ dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n",
+ i + 1, PTR_ERR(sai->mclk_clk[i]));
+ sai->mclk_clk[i] = NULL;
+ }
+ }
+
+ irq = platform_get_irq(pdev, 0);
+ if (irq < 0) {
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+ return irq;
+ }
+
+ ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, 0, np->name, sai);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to claim irq %u\n", irq);
+ return ret;
+ }
+
+ /* Sync Tx with Rx as default by following old DT binding */
+ sai->synchronous[RX] = true;
+ sai->synchronous[TX] = false;
+ fsl_sai_dai.symmetric_rates = 1;
+ fsl_sai_dai.symmetric_channels = 1;
+ fsl_sai_dai.symmetric_samplebits = 1;
+
+ if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) &&
+ of_find_property(np, "fsl,sai-asynchronous", NULL)) {
+ /* error out if both synchronous and asynchronous are present */
+ dev_err(&pdev->dev, "invalid binding for synchronous mode\n");
+ return -EINVAL;
+ }
+
+ if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) {
+ /* Sync Rx with Tx */
+ sai->synchronous[RX] = false;
+ sai->synchronous[TX] = true;
+ } else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) {
+ /* Discard all settings for asynchronous mode */
+ sai->synchronous[RX] = false;
+ sai->synchronous[TX] = false;
+ fsl_sai_dai.symmetric_rates = 0;
+ fsl_sai_dai.symmetric_channels = 0;
+ fsl_sai_dai.symmetric_samplebits = 0;
+ }
+
+ sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
+ sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
+ sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
+ sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX;
+
+ platform_set_drvdata(pdev, sai);
+
+ ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
+ &fsl_sai_dai, 1);
+ if (ret)
+ return ret;
+
+ if (sai->sai_on_imx)
+ return imx_pcm_dma_init(pdev);
+ else
+ return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
+ SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+}
+
+static const struct of_device_id fsl_sai_ids[] = {
+ { .compatible = "fsl,vf610-sai", },
+ { .compatible = "fsl,imx6sx-sai", },
+ { /* sentinel */ }
+};
+
+static struct platform_driver fsl_sai_driver = {
+ .probe = fsl_sai_probe,
+ .driver = {
+ .name = "fsl-sai",
+ .of_match_table = fsl_sai_ids,
+ },
+};
+module_platform_driver(fsl_sai_driver);
+
+MODULE_DESCRIPTION("Freescale Soc SAI Interface");
+MODULE_AUTHOR("Xiubo Li, <Li.Xiubo@freescale.com>");
+MODULE_ALIAS("platform:fsl-sai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
new file mode 100644
index 000000000..34667209b
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.h
@@ -0,0 +1,147 @@
+/*
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __FSL_SAI_H
+#define __FSL_SAI_H
+
+#include <sound/dmaengine_pcm.h>
+
+#define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+/* SAI Register Map Register */
+#define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */
+#define FSL_SAI_TCR1 0x04 /* SAI Transmit Configuration 1 */
+#define FSL_SAI_TCR2 0x08 /* SAI Transmit Configuration 2 */
+#define FSL_SAI_TCR3 0x0c /* SAI Transmit Configuration 3 */
+#define FSL_SAI_TCR4 0x10 /* SAI Transmit Configuration 4 */
+#define FSL_SAI_TCR5 0x14 /* SAI Transmit Configuration 5 */
+#define FSL_SAI_TDR 0x20 /* SAI Transmit Data */
+#define FSL_SAI_TFR 0x40 /* SAI Transmit FIFO */
+#define FSL_SAI_TMR 0x60 /* SAI Transmit Mask */
+#define FSL_SAI_RCSR 0x80 /* SAI Receive Control */
+#define FSL_SAI_RCR1 0x84 /* SAI Receive Configuration 1 */
+#define FSL_SAI_RCR2 0x88 /* SAI Receive Configuration 2 */
+#define FSL_SAI_RCR3 0x8c /* SAI Receive Configuration 3 */
+#define FSL_SAI_RCR4 0x90 /* SAI Receive Configuration 4 */
+#define FSL_SAI_RCR5 0x94 /* SAI Receive Configuration 5 */
+#define FSL_SAI_RDR 0xa0 /* SAI Receive Data */
+#define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */
+#define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */
+
+#define FSL_SAI_xCSR(tx) (tx ? FSL_SAI_TCSR : FSL_SAI_RCSR)
+#define FSL_SAI_xCR1(tx) (tx ? FSL_SAI_TCR1 : FSL_SAI_RCR1)
+#define FSL_SAI_xCR2(tx) (tx ? FSL_SAI_TCR2 : FSL_SAI_RCR2)
+#define FSL_SAI_xCR3(tx) (tx ? FSL_SAI_TCR3 : FSL_SAI_RCR3)
+#define FSL_SAI_xCR4(tx) (tx ? FSL_SAI_TCR4 : FSL_SAI_RCR4)
+#define FSL_SAI_xCR5(tx) (tx ? FSL_SAI_TCR5 : FSL_SAI_RCR5)
+#define FSL_SAI_xDR(tx) (tx ? FSL_SAI_TDR : FSL_SAI_RDR)
+#define FSL_SAI_xFR(tx) (tx ? FSL_SAI_TFR : FSL_SAI_RFR)
+#define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR)
+
+/* SAI Transmit/Recieve Control Register */
+#define FSL_SAI_CSR_TERE BIT(31)
+#define FSL_SAI_CSR_FR BIT(25)
+#define FSL_SAI_CSR_SR BIT(24)
+#define FSL_SAI_CSR_xF_SHIFT 16
+#define FSL_SAI_CSR_xF_W_SHIFT 18
+#define FSL_SAI_CSR_xF_MASK (0x1f << FSL_SAI_CSR_xF_SHIFT)
+#define FSL_SAI_CSR_xF_W_MASK (0x7 << FSL_SAI_CSR_xF_W_SHIFT)
+#define FSL_SAI_CSR_WSF BIT(20)
+#define FSL_SAI_CSR_SEF BIT(19)
+#define FSL_SAI_CSR_FEF BIT(18)
+#define FSL_SAI_CSR_FWF BIT(17)
+#define FSL_SAI_CSR_FRF BIT(16)
+#define FSL_SAI_CSR_xIE_SHIFT 8
+#define FSL_SAI_CSR_xIE_MASK (0x1f << FSL_SAI_CSR_xIE_SHIFT)
+#define FSL_SAI_CSR_WSIE BIT(12)
+#define FSL_SAI_CSR_SEIE BIT(11)
+#define FSL_SAI_CSR_FEIE BIT(10)
+#define FSL_SAI_CSR_FWIE BIT(9)
+#define FSL_SAI_CSR_FRIE BIT(8)
+#define FSL_SAI_CSR_FRDE BIT(0)
+
+/* SAI Transmit and Recieve Configuration 1 Register */
+#define FSL_SAI_CR1_RFW_MASK 0x1f
+
+/* SAI Transmit and Recieve Configuration 2 Register */
+#define FSL_SAI_CR2_SYNC BIT(30)
+#define FSL_SAI_CR2_MSEL_MASK (0xff << 26)
+#define FSL_SAI_CR2_MSEL_BUS 0
+#define FSL_SAI_CR2_MSEL_MCLK1 BIT(26)
+#define FSL_SAI_CR2_MSEL_MCLK2 BIT(27)
+#define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27))
+#define FSL_SAI_CR2_BCP BIT(25)
+#define FSL_SAI_CR2_BCD_MSTR BIT(24)
+
+/* SAI Transmit and Recieve Configuration 3 Register */
+#define FSL_SAI_CR3_TRCE BIT(16)
+#define FSL_SAI_CR3_WDFL(x) (x)
+#define FSL_SAI_CR3_WDFL_MASK 0x1f
+
+/* SAI Transmit and Recieve Configuration 4 Register */
+#define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16)
+#define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16)
+#define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8)
+#define FSL_SAI_CR4_SYWD_MASK (0x1f << 8)
+#define FSL_SAI_CR4_MF BIT(4)
+#define FSL_SAI_CR4_FSE BIT(3)
+#define FSL_SAI_CR4_FSP BIT(1)
+#define FSL_SAI_CR4_FSD_MSTR BIT(0)
+
+/* SAI Transmit and Recieve Configuration 5 Register */
+#define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24)
+#define FSL_SAI_CR5_WNW_MASK (0x1f << 24)
+#define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16)
+#define FSL_SAI_CR5_W0W_MASK (0x1f << 16)
+#define FSL_SAI_CR5_FBT(x) ((x) << 8)
+#define FSL_SAI_CR5_FBT_MASK (0x1f << 8)
+
+/* SAI type */
+#define FSL_SAI_DMA BIT(0)
+#define FSL_SAI_USE_AC97 BIT(1)
+#define FSL_SAI_NET BIT(2)
+#define FSL_SAI_TRA_SYN BIT(3)
+#define FSL_SAI_REC_SYN BIT(4)
+#define FSL_SAI_USE_I2S_SLAVE BIT(5)
+
+#define FSL_FMT_TRANSMITTER 0
+#define FSL_FMT_RECEIVER 1
+
+/* SAI clock sources */
+#define FSL_SAI_CLK_BUS 0
+#define FSL_SAI_CLK_MAST1 1
+#define FSL_SAI_CLK_MAST2 2
+#define FSL_SAI_CLK_MAST3 3
+
+#define FSL_SAI_MCLK_MAX 3
+
+/* SAI data transfer numbers per DMA request */
+#define FSL_SAI_MAXBURST_TX 6
+#define FSL_SAI_MAXBURST_RX 6
+
+struct fsl_sai {
+ struct platform_device *pdev;
+ struct regmap *regmap;
+ struct clk *bus_clk;
+ struct clk *mclk_clk[FSL_SAI_MCLK_MAX];
+
+ bool is_lsb_first;
+ bool is_dsp_mode;
+ bool sai_on_imx;
+ bool synchronous[2];
+
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+};
+
+#define TX 1
+#define RX 0
+
+#endif /* __FSL_SAI_H */
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
new file mode 100644
index 000000000..91eb3aef7
--- /dev/null
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -0,0 +1,1287 @@
+/*
+ * Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * Based on stmp3xxx_spdif_dai.c
+ * Vladimir Barinov <vbarinov@embeddedalley.com>
+ * Copyright 2008 SigmaTel, Inc
+ * Copyright 2008 Embedded Alley Solutions, Inc
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/bitrev.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_address.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
+#include <linux/regmap.h>
+
+#include <sound/asoundef.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/soc.h>
+
+#include "fsl_spdif.h"
+#include "imx-pcm.h"
+
+#define FSL_SPDIF_TXFIFO_WML 0x8
+#define FSL_SPDIF_RXFIFO_WML 0x8
+
+#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC)
+#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL |\
+ INT_URX_OV | INT_QRX_FUL | INT_QRX_OV |\
+ INT_UQ_SYNC | INT_UQ_ERR | INT_RXFIFO_RESYNC |\
+ INT_LOSS_LOCK | INT_DPLL_LOCKED)
+
+#define SIE_INTR_FOR(tx) (tx ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE)
+
+/* Index list for the values that has if (DPLL Locked) condition */
+static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb };
+#define SRPC_NODPLL_START1 0x5
+#define SRPC_NODPLL_START2 0xc
+
+#define DEFAULT_RXCLK_SRC 1
+
+/*
+ * SPDIF control structure
+ * Defines channel status, subcode and Q sub
+ */
+struct spdif_mixer_control {
+ /* spinlock to access control data */
+ spinlock_t ctl_lock;
+
+ /* IEC958 channel tx status bit */
+ unsigned char ch_status[4];
+
+ /* User bits */
+ unsigned char subcode[2 * SPDIF_UBITS_SIZE];
+
+ /* Q subcode part of user bits */
+ unsigned char qsub[2 * SPDIF_QSUB_SIZE];
+
+ /* Buffer offset for U/Q */
+ u32 upos;
+ u32 qpos;
+
+ /* Ready buffer index of the two buffers */
+ u32 ready_buf;
+};
+
+/**
+ * fsl_spdif_priv: Freescale SPDIF private data
+ *
+ * @fsl_spdif_control: SPDIF control data
+ * @cpu_dai_drv: cpu dai driver
+ * @pdev: platform device pointer
+ * @regmap: regmap handler
+ * @dpll_locked: dpll lock flag
+ * @txrate: the best rates for playback
+ * @txclk_df: STC_TXCLK_DF dividers value for playback
+ * @sysclk_df: STC_SYSCLK_DF dividers value for playback
+ * @txclk_src: STC_TXCLK_SRC values for playback
+ * @rxclk_src: SRPC_CLKSRC_SEL values for capture
+ * @txclk: tx clock sources for playback
+ * @rxclk: rx clock sources for capture
+ * @coreclk: core clock for register access via DMA
+ * @sysclk: system clock for rx clock rate measurement
+ * @dma_params_tx: DMA parameters for transmit channel
+ * @dma_params_rx: DMA parameters for receive channel
+ */
+struct fsl_spdif_priv {
+ struct spdif_mixer_control fsl_spdif_control;
+ struct snd_soc_dai_driver cpu_dai_drv;
+ struct platform_device *pdev;
+ struct regmap *regmap;
+ bool dpll_locked;
+ u32 txrate[SPDIF_TXRATE_MAX];
+ u8 txclk_df[SPDIF_TXRATE_MAX];
+ u8 sysclk_df[SPDIF_TXRATE_MAX];
+ u8 txclk_src[SPDIF_TXRATE_MAX];
+ u8 rxclk_src;
+ struct clk *txclk[SPDIF_TXRATE_MAX];
+ struct clk *rxclk;
+ struct clk *coreclk;
+ struct clk *sysclk;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+};
+
+/* DPLL locked and lock loss interrupt handler */
+static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 locked;
+
+ regmap_read(regmap, REG_SPDIF_SRPC, &locked);
+ locked &= SRPC_DPLL_LOCKED;
+
+ dev_dbg(&pdev->dev, "isr: Rx dpll %s \n",
+ locked ? "locked" : "loss lock");
+
+ spdif_priv->dpll_locked = locked ? true : false;
+}
+
+/* Receiver found illegal symbol interrupt handler */
+static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+
+ dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n");
+
+ /* Clear illegal symbol if DPLL unlocked since no audio stream */
+ if (!spdif_priv->dpll_locked)
+ regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0);
+}
+
+/* U/Q Channel receive register full */
+static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name)
+{
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 *pos, size, val, reg;
+
+ switch (name) {
+ case 'U':
+ pos = &ctrl->upos;
+ size = SPDIF_UBITS_SIZE;
+ reg = REG_SPDIF_SRU;
+ break;
+ case 'Q':
+ pos = &ctrl->qpos;
+ size = SPDIF_QSUB_SIZE;
+ reg = REG_SPDIF_SRQ;
+ break;
+ default:
+ dev_err(&pdev->dev, "unsupported channel name\n");
+ return;
+ }
+
+ dev_dbg(&pdev->dev, "isr: %c Channel receive register full\n", name);
+
+ if (*pos >= size * 2) {
+ *pos = 0;
+ } else if (unlikely((*pos % size) + 3 > size)) {
+ dev_err(&pdev->dev, "User bit receivce buffer overflow\n");
+ return;
+ }
+
+ regmap_read(regmap, reg, &val);
+ ctrl->subcode[*pos++] = val >> 16;
+ ctrl->subcode[*pos++] = val >> 8;
+ ctrl->subcode[*pos++] = val;
+}
+
+/* U/Q Channel sync found */
+static void spdif_irq_uq_sync(struct fsl_spdif_priv *spdif_priv)
+{
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct platform_device *pdev = spdif_priv->pdev;
+
+ dev_dbg(&pdev->dev, "isr: U/Q Channel sync found\n");
+
+ /* U/Q buffer reset */
+ if (ctrl->qpos == 0)
+ return;
+
+ /* Set ready to this buffer */
+ ctrl->ready_buf = (ctrl->qpos - 1) / SPDIF_QSUB_SIZE + 1;
+}
+
+/* U/Q Channel framing error */
+static void spdif_irq_uq_err(struct fsl_spdif_priv *spdif_priv)
+{
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 val;
+
+ dev_dbg(&pdev->dev, "isr: U/Q Channel framing error\n");
+
+ /* Read U/Q data to clear the irq and do buffer reset */
+ regmap_read(regmap, REG_SPDIF_SRU, &val);
+ regmap_read(regmap, REG_SPDIF_SRQ, &val);
+
+ /* Drop this U/Q buffer */
+ ctrl->ready_buf = 0;
+ ctrl->upos = 0;
+ ctrl->qpos = 0;
+}
+
+/* Get spdif interrupt status and clear the interrupt */
+static u32 spdif_intr_status_clear(struct fsl_spdif_priv *spdif_priv)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val, val2;
+
+ regmap_read(regmap, REG_SPDIF_SIS, &val);
+ regmap_read(regmap, REG_SPDIF_SIE, &val2);
+
+ regmap_write(regmap, REG_SPDIF_SIC, val & val2);
+
+ return val;
+}
+
+static irqreturn_t spdif_isr(int irq, void *devid)
+{
+ struct fsl_spdif_priv *spdif_priv = (struct fsl_spdif_priv *)devid;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 sis;
+
+ sis = spdif_intr_status_clear(spdif_priv);
+
+ if (sis & INT_DPLL_LOCKED)
+ spdif_irq_dpll_lock(spdif_priv);
+
+ if (sis & INT_TXFIFO_UNOV)
+ dev_dbg(&pdev->dev, "isr: Tx FIFO under/overrun\n");
+
+ if (sis & INT_TXFIFO_RESYNC)
+ dev_dbg(&pdev->dev, "isr: Tx FIFO resync\n");
+
+ if (sis & INT_CNEW)
+ dev_dbg(&pdev->dev, "isr: cstatus new\n");
+
+ if (sis & INT_VAL_NOGOOD)
+ dev_dbg(&pdev->dev, "isr: validity flag no good\n");
+
+ if (sis & INT_SYM_ERR)
+ spdif_irq_sym_error(spdif_priv);
+
+ if (sis & INT_BIT_ERR)
+ dev_dbg(&pdev->dev, "isr: receiver found parity bit error\n");
+
+ if (sis & INT_URX_FUL)
+ spdif_irq_uqrx_full(spdif_priv, 'U');
+
+ if (sis & INT_URX_OV)
+ dev_dbg(&pdev->dev, "isr: U Channel receive register overrun\n");
+
+ if (sis & INT_QRX_FUL)
+ spdif_irq_uqrx_full(spdif_priv, 'Q');
+
+ if (sis & INT_QRX_OV)
+ dev_dbg(&pdev->dev, "isr: Q Channel receive register overrun\n");
+
+ if (sis & INT_UQ_SYNC)
+ spdif_irq_uq_sync(spdif_priv);
+
+ if (sis & INT_UQ_ERR)
+ spdif_irq_uq_err(spdif_priv);
+
+ if (sis & INT_RXFIFO_UNOV)
+ dev_dbg(&pdev->dev, "isr: Rx FIFO under/overrun\n");
+
+ if (sis & INT_RXFIFO_RESYNC)
+ dev_dbg(&pdev->dev, "isr: Rx FIFO resync\n");
+
+ if (sis & INT_LOSS_LOCK)
+ spdif_irq_dpll_lock(spdif_priv);
+
+ /* FIXME: Write Tx FIFO to clear TxEm */
+ if (sis & INT_TX_EM)
+ dev_dbg(&pdev->dev, "isr: Tx FIFO empty\n");
+
+ /* FIXME: Read Rx FIFO to clear RxFIFOFul */
+ if (sis & INT_RXFIFO_FUL)
+ dev_dbg(&pdev->dev, "isr: Rx FIFO full\n");
+
+ return IRQ_HANDLED;
+}
+
+static int spdif_softreset(struct fsl_spdif_priv *spdif_priv)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val, cycle = 1000;
+
+ regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET);
+
+ /*
+ * RESET bit would be cleared after finishing its reset procedure,
+ * which typically lasts 8 cycles. 1000 cycles will keep it safe.
+ */
+ do {
+ regmap_read(regmap, REG_SPDIF_SCR, &val);
+ } while ((val & SCR_SOFT_RESET) && cycle--);
+
+ if (cycle)
+ return 0;
+ else
+ return -EBUSY;
+}
+
+static void spdif_set_cstatus(struct spdif_mixer_control *ctrl,
+ u8 mask, u8 cstatus)
+{
+ ctrl->ch_status[3] &= ~mask;
+ ctrl->ch_status[3] |= cstatus & mask;
+}
+
+static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv)
+{
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 ch_status;
+
+ ch_status = (bitrev8(ctrl->ch_status[0]) << 16) |
+ (bitrev8(ctrl->ch_status[1]) << 8) |
+ bitrev8(ctrl->ch_status[2]);
+ regmap_write(regmap, REG_SPDIF_STCSCH, ch_status);
+
+ dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status);
+
+ ch_status = bitrev8(ctrl->ch_status[3]) << 16;
+ regmap_write(regmap, REG_SPDIF_STCSCL, ch_status);
+
+ dev_dbg(&pdev->dev, "STCSCL: 0x%06x\n", ch_status);
+}
+
+/* Set SPDIF PhaseConfig register for rx clock */
+static int spdif_set_rx_clksrc(struct fsl_spdif_priv *spdif_priv,
+ enum spdif_gainsel gainsel, int dpll_locked)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ u8 clksrc = spdif_priv->rxclk_src;
+
+ if (clksrc >= SRPC_CLKSRC_MAX || gainsel >= GAINSEL_MULTI_MAX)
+ return -EINVAL;
+
+ regmap_update_bits(regmap, REG_SPDIF_SRPC,
+ SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK,
+ SRPC_CLKSRC_SEL_SET(clksrc) | SRPC_GAINSEL_SET(gainsel));
+
+ return 0;
+}
+
+static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
+ int sample_rate)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ unsigned long csfs = 0;
+ u32 stc, mask, rate;
+ u8 clk, txclk_df, sysclk_df;
+ int ret;
+
+ switch (sample_rate) {
+ case 32000:
+ rate = SPDIF_TXRATE_32000;
+ csfs = IEC958_AES3_CON_FS_32000;
+ break;
+ case 44100:
+ rate = SPDIF_TXRATE_44100;
+ csfs = IEC958_AES3_CON_FS_44100;
+ break;
+ case 48000:
+ rate = SPDIF_TXRATE_48000;
+ csfs = IEC958_AES3_CON_FS_48000;
+ break;
+ case 96000:
+ rate = SPDIF_TXRATE_96000;
+ csfs = IEC958_AES3_CON_FS_96000;
+ break;
+ case 192000:
+ rate = SPDIF_TXRATE_192000;
+ csfs = IEC958_AES3_CON_FS_192000;
+ break;
+ default:
+ dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate);
+ return -EINVAL;
+ }
+
+ clk = spdif_priv->txclk_src[rate];
+ if (clk >= STC_TXCLK_SRC_MAX) {
+ dev_err(&pdev->dev, "tx clock source is out of range\n");
+ return -EINVAL;
+ }
+
+ txclk_df = spdif_priv->txclk_df[rate];
+ if (txclk_df == 0) {
+ dev_err(&pdev->dev, "the txclk_df can't be zero\n");
+ return -EINVAL;
+ }
+
+ sysclk_df = spdif_priv->sysclk_df[rate];
+
+ /* Don't mess up the clocks from other modules */
+ if (clk != STC_TXCLK_SPDIF_ROOT)
+ goto clk_set_bypass;
+
+ /*
+ * The S/PDIF block needs a clock of 64 * fs * txclk_df.
+ * So request 64 * fs * (txclk_df + 1) to get rounded.
+ */
+ ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (txclk_df + 1));
+ if (ret) {
+ dev_err(&pdev->dev, "failed to set tx clock rate\n");
+ return ret;
+ }
+
+clk_set_bypass:
+ dev_dbg(&pdev->dev, "expected clock rate = %d\n",
+ (64 * sample_rate * txclk_df * sysclk_df));
+ dev_dbg(&pdev->dev, "actual clock rate = %ld\n",
+ clk_get_rate(spdif_priv->txclk[rate]));
+
+ /* set fs field in consumer channel status */
+ spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs);
+
+ /* select clock source and divisor */
+ stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) |
+ STC_TXCLK_DF(txclk_df) | STC_SYSCLK_DF(sysclk_df);
+ mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK |
+ STC_TXCLK_DF_MASK | STC_SYSCLK_DF_MASK;
+ regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc);
+
+ dev_dbg(&pdev->dev, "set sample rate to %dHz for %dHz playback\n",
+ spdif_priv->txrate[rate], sample_rate);
+
+ return 0;
+}
+
+static int fsl_spdif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct platform_device *pdev = spdif_priv->pdev;
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 scr, mask, i;
+ int ret;
+
+ /* Reset module and interrupts only for first initialization */
+ if (!cpu_dai->active) {
+ ret = clk_prepare_enable(spdif_priv->coreclk);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to enable core clock\n");
+ return ret;
+ }
+
+ ret = spdif_softreset(spdif_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to soft reset\n");
+ goto err;
+ }
+
+ /* Disable all the interrupts */
+ regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ scr = SCR_TXFIFO_AUTOSYNC | SCR_TXFIFO_CTRL_NORMAL |
+ SCR_TXSEL_NORMAL | SCR_USRC_SEL_CHIP |
+ SCR_TXFIFO_FSEL_IF8;
+ mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK |
+ SCR_TXSEL_MASK | SCR_USRC_SEL_MASK |
+ SCR_TXFIFO_FSEL_MASK;
+ for (i = 0; i < SPDIF_TXRATE_MAX; i++)
+ clk_prepare_enable(spdif_priv->txclk[i]);
+ } else {
+ scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC;
+ mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK|
+ SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK;
+ clk_prepare_enable(spdif_priv->rxclk);
+ }
+ regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr);
+
+ /* Power up SPDIF module */
+ regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0);
+
+ return 0;
+
+err:
+ clk_disable_unprepare(spdif_priv->coreclk);
+
+ return ret;
+}
+
+static void fsl_spdif_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 scr, mask, i;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ scr = 0;
+ mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK |
+ SCR_TXSEL_MASK | SCR_USRC_SEL_MASK |
+ SCR_TXFIFO_FSEL_MASK;
+ for (i = 0; i < SPDIF_TXRATE_MAX; i++)
+ clk_disable_unprepare(spdif_priv->txclk[i]);
+ } else {
+ scr = SCR_RXFIFO_OFF | SCR_RXFIFO_CTL_ZERO;
+ mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK|
+ SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK;
+ clk_disable_unprepare(spdif_priv->rxclk);
+ }
+ regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr);
+
+ /* Power down SPDIF module only if tx&rx are both inactive */
+ if (!cpu_dai->active) {
+ spdif_intr_status_clear(spdif_priv);
+ regmap_update_bits(regmap, REG_SPDIF_SCR,
+ SCR_LOW_POWER, SCR_LOW_POWER);
+ clk_disable_unprepare(spdif_priv->coreclk);
+ }
+}
+
+static int fsl_spdif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u32 sample_rate = params_rate(params);
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = spdif_set_sample_rate(substream, sample_rate);
+ if (ret) {
+ dev_err(&pdev->dev, "%s: set sample rate failed: %d\n",
+ __func__, sample_rate);
+ return ret;
+ }
+ spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK,
+ IEC958_AES3_CON_CLOCK_1000PPM);
+ spdif_write_channel_status(spdif_priv);
+ } else {
+ /* Setup rx clock source */
+ ret = spdif_set_rx_clksrc(spdif_priv, SPDIF_DEFAULT_GAINSEL, 1);
+ }
+
+ return ret;
+}
+
+static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ u32 intr = SIE_INTR_FOR(tx);
+ u32 dmaen = SCR_DMA_xX_EN(tx);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ regmap_update_bits(regmap, REG_SPDIF_SIE, intr, intr);
+ regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, dmaen);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0);
+ regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+ .startup = fsl_spdif_startup,
+ .hw_params = fsl_spdif_hw_params,
+ .trigger = fsl_spdif_trigger,
+ .shutdown = fsl_spdif_shutdown,
+};
+
+
+/*
+ * FSL SPDIF IEC958 controller(mixer) functions
+ *
+ * Channel status get/put control
+ * User bit value get/put control
+ * Valid bit value get control
+ * DPLL lock status get control
+ * User bit sync mode selection control
+ */
+
+static int fsl_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+
+ return 0;
+}
+
+static int fsl_spdif_pb_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uvalue)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+
+ uvalue->value.iec958.status[0] = ctrl->ch_status[0];
+ uvalue->value.iec958.status[1] = ctrl->ch_status[1];
+ uvalue->value.iec958.status[2] = ctrl->ch_status[2];
+ uvalue->value.iec958.status[3] = ctrl->ch_status[3];
+
+ return 0;
+}
+
+static int fsl_spdif_pb_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uvalue)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+
+ ctrl->ch_status[0] = uvalue->value.iec958.status[0];
+ ctrl->ch_status[1] = uvalue->value.iec958.status[1];
+ ctrl->ch_status[2] = uvalue->value.iec958.status[2];
+ ctrl->ch_status[3] = uvalue->value.iec958.status[3];
+
+ spdif_write_channel_status(spdif_priv);
+
+ return 0;
+}
+
+/* Get channel status from SPDIF_RX_CCHAN register */
+static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 cstatus, val;
+
+ regmap_read(regmap, REG_SPDIF_SIS, &val);
+ if (!(val & INT_CNEW))
+ return -EAGAIN;
+
+ regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus);
+ ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF;
+ ucontrol->value.iec958.status[1] = (cstatus >> 8) & 0xFF;
+ ucontrol->value.iec958.status[2] = cstatus & 0xFF;
+
+ regmap_read(regmap, REG_SPDIF_SRCSL, &cstatus);
+ ucontrol->value.iec958.status[3] = (cstatus >> 16) & 0xFF;
+ ucontrol->value.iec958.status[4] = (cstatus >> 8) & 0xFF;
+ ucontrol->value.iec958.status[5] = cstatus & 0xFF;
+
+ /* Clear intr */
+ regmap_write(regmap, REG_SPDIF_SIC, INT_CNEW);
+
+ return 0;
+}
+
+/*
+ * Get User bits (subcode) from chip value which readed out
+ * in UChannel register.
+ */
+static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ unsigned long flags;
+ int ret = -EAGAIN;
+
+ spin_lock_irqsave(&ctrl->ctl_lock, flags);
+ if (ctrl->ready_buf) {
+ int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE;
+ memcpy(&ucontrol->value.iec958.subcode[0],
+ &ctrl->subcode[idx], SPDIF_UBITS_SIZE);
+ ret = 0;
+ }
+ spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
+
+ return ret;
+}
+
+/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */
+static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = SPDIF_QSUB_SIZE;
+
+ return 0;
+}
+
+/* Get Q subcode from chip value which readed out in QChannel register */
+static int fsl_spdif_qget(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
+ unsigned long flags;
+ int ret = -EAGAIN;
+
+ spin_lock_irqsave(&ctrl->ctl_lock, flags);
+ if (ctrl->ready_buf) {
+ int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE;
+ memcpy(&ucontrol->value.bytes.data[0],
+ &ctrl->qsub[idx], SPDIF_QSUB_SIZE);
+ ret = 0;
+ }
+ spin_unlock_irqrestore(&ctrl->ctl_lock, flags);
+
+ return ret;
+}
+
+/* Valid bit infomation */
+static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+
+ return 0;
+}
+
+/* Get valid good bit from interrupt status register */
+static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val;
+
+ regmap_read(regmap, REG_SPDIF_SIS, &val);
+ ucontrol->value.integer.value[0] = (val & INT_VAL_NOGOOD) != 0;
+ regmap_write(regmap, REG_SPDIF_SIC, INT_VAL_NOGOOD);
+
+ return 0;
+}
+
+/* DPLL lock infomation */
+static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 16000;
+ uinfo->value.integer.max = 96000;
+
+ return 0;
+}
+
+static u32 gainsel_multi[GAINSEL_MULTI_MAX] = {
+ 24, 16, 12, 8, 6, 4, 3,
+};
+
+/* Get RX data clock rate given the SPDIF bus_clk */
+static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv,
+ enum spdif_gainsel gainsel)
+{
+ struct regmap *regmap = spdif_priv->regmap;
+ struct platform_device *pdev = spdif_priv->pdev;
+ u64 tmpval64, busclk_freq = 0;
+ u32 freqmeas, phaseconf;
+ u8 clksrc;
+
+ regmap_read(regmap, REG_SPDIF_SRFM, &freqmeas);
+ regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf);
+
+ clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf;
+
+ /* Get bus clock from system */
+ if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED))
+ busclk_freq = clk_get_rate(spdif_priv->sysclk);
+
+ /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */
+ tmpval64 = (u64) busclk_freq * freqmeas;
+ do_div(tmpval64, gainsel_multi[gainsel] * 1024);
+ do_div(tmpval64, 128 * 1024);
+
+ dev_dbg(&pdev->dev, "FreqMeas: %d\n", freqmeas);
+ dev_dbg(&pdev->dev, "BusclkFreq: %lld\n", busclk_freq);
+ dev_dbg(&pdev->dev, "RxRate: %lld\n", tmpval64);
+
+ return (int)tmpval64;
+}
+
+/*
+ * Get DPLL lock or not info from stable interrupt status register.
+ * User application must use this control to get locked,
+ * then can do next PCM operation
+ */
+static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int rate = 0;
+
+ if (spdif_priv->dpll_locked)
+ rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL);
+
+ ucontrol->value.integer.value[0] = rate;
+
+ return 0;
+}
+
+/* User bit sync mode info */
+static int fsl_spdif_usync_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+
+ return 0;
+}
+
+/*
+ * User bit sync mode:
+ * 1 CD User channel subcode
+ * 0 Non-CD data
+ */
+static int fsl_spdif_usync_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val;
+
+ regmap_read(regmap, REG_SPDIF_SRCD, &val);
+ ucontrol->value.integer.value[0] = (val & SRCD_CD_USER) != 0;
+
+ return 0;
+}
+
+/*
+ * User bit sync mode:
+ * 1 CD User channel subcode
+ * 0 Non-CD data
+ */
+static int fsl_spdif_usync_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regmap = spdif_priv->regmap;
+ u32 val = ucontrol->value.integer.value[0] << SRCD_CD_USER_OFFSET;
+
+ regmap_update_bits(regmap, REG_SPDIF_SRCD, SRCD_CD_USER, val);
+
+ return 0;
+}
+
+/* FSL SPDIF IEC958 controller defines */
+static struct snd_kcontrol_new fsl_spdif_ctrls[] = {
+ /* Status cchanel controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_WRITE |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_info,
+ .get = fsl_spdif_pb_get,
+ .put = fsl_spdif_pb_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_info,
+ .get = fsl_spdif_capture_get,
+ },
+ /* User bits controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Subcode Capture Default",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_info,
+ .get = fsl_spdif_subcode_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Q-subcode Capture Default",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_qinfo,
+ .get = fsl_spdif_qget,
+ },
+ /* Valid bit error controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 V-Bit Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_vbit_info,
+ .get = fsl_spdif_vbit_get,
+ },
+ /* DPLL lock info get controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "RX Sample Rate",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_rxrate_info,
+ .get = fsl_spdif_rxrate_get,
+ },
+ /* User bit sync mode set/get controller */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 USyncMode CDText",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_WRITE |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = fsl_spdif_usync_info,
+ .get = fsl_spdif_usync_get,
+ .put = fsl_spdif_usync_put,
+ },
+};
+
+static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
+{
+ struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai);
+
+ snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx,
+ &spdif_private->dma_params_rx);
+
+ snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls));
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver fsl_spdif_dai = {
+ .probe = &fsl_spdif_dai_probe,
+ .playback = {
+ .stream_name = "CPU-Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = FSL_SPDIF_RATES_PLAYBACK,
+ .formats = FSL_SPDIF_FORMATS_PLAYBACK,
+ },
+ .capture = {
+ .stream_name = "CPU-Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = FSL_SPDIF_RATES_CAPTURE,
+ .formats = FSL_SPDIF_FORMATS_CAPTURE,
+ },
+ .ops = &fsl_spdif_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_spdif_component = {
+ .name = "fsl-spdif",
+};
+
+/* FSL SPDIF REGMAP */
+
+static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_SPDIF_SCR:
+ case REG_SPDIF_SRCD:
+ case REG_SPDIF_SRPC:
+ case REG_SPDIF_SIE:
+ case REG_SPDIF_SIS:
+ case REG_SPDIF_SRL:
+ case REG_SPDIF_SRR:
+ case REG_SPDIF_SRCSH:
+ case REG_SPDIF_SRCSL:
+ case REG_SPDIF_SRU:
+ case REG_SPDIF_SRQ:
+ case REG_SPDIF_STCSCH:
+ case REG_SPDIF_STCSCL:
+ case REG_SPDIF_SRFM:
+ case REG_SPDIF_STC:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case REG_SPDIF_SCR:
+ case REG_SPDIF_SRCD:
+ case REG_SPDIF_SRPC:
+ case REG_SPDIF_SIE:
+ case REG_SPDIF_SIC:
+ case REG_SPDIF_STL:
+ case REG_SPDIF_STR:
+ case REG_SPDIF_STCSCH:
+ case REG_SPDIF_STCSCL:
+ case REG_SPDIF_STC:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config fsl_spdif_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+
+ .max_register = REG_SPDIF_STC,
+ .readable_reg = fsl_spdif_readable_reg,
+ .writeable_reg = fsl_spdif_writeable_reg,
+};
+
+static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
+ struct clk *clk, u64 savesub,
+ enum spdif_txrate index, bool round)
+{
+ const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
+ bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk);
+ u64 rate_ideal, rate_actual, sub;
+ u32 sysclk_dfmin, sysclk_dfmax;
+ u32 txclk_df, sysclk_df, arate;
+
+ /* The sysclk has an extra divisor [2, 512] */
+ sysclk_dfmin = is_sysclk ? 2 : 1;
+ sysclk_dfmax = is_sysclk ? 512 : 1;
+
+ for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) {
+ for (txclk_df = 1; txclk_df <= 128; txclk_df++) {
+ rate_ideal = rate[index] * (txclk_df + 1) * 64;
+ if (round)
+ rate_actual = clk_round_rate(clk, rate_ideal);
+ else
+ rate_actual = clk_get_rate(clk);
+
+ arate = rate_actual / 64;
+ arate /= txclk_df * sysclk_df;
+
+ if (arate == rate[index]) {
+ /* We are lucky */
+ savesub = 0;
+ spdif_priv->txclk_df[index] = txclk_df;
+ spdif_priv->sysclk_df[index] = sysclk_df;
+ spdif_priv->txrate[index] = arate;
+ goto out;
+ } else if (arate / rate[index] == 1) {
+ /* A little bigger than expect */
+ sub = (u64)(arate - rate[index]) * 100000;
+ do_div(sub, rate[index]);
+ if (sub >= savesub)
+ continue;
+ savesub = sub;
+ spdif_priv->txclk_df[index] = txclk_df;
+ spdif_priv->sysclk_df[index] = sysclk_df;
+ spdif_priv->txrate[index] = arate;
+ } else if (rate[index] / arate == 1) {
+ /* A little smaller than expect */
+ sub = (u64)(rate[index] - arate) * 100000;
+ do_div(sub, rate[index]);
+ if (sub >= savesub)
+ continue;
+ savesub = sub;
+ spdif_priv->txclk_df[index] = txclk_df;
+ spdif_priv->sysclk_df[index] = sysclk_df;
+ spdif_priv->txrate[index] = arate;
+ }
+ }
+ }
+
+out:
+ return savesub;
+}
+
+static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
+ enum spdif_txrate index)
+{
+ const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
+ struct platform_device *pdev = spdif_priv->pdev;
+ struct device *dev = &pdev->dev;
+ u64 savesub = 100000, ret;
+ struct clk *clk;
+ char tmp[16];
+ int i;
+
+ for (i = 0; i < STC_TXCLK_SRC_MAX; i++) {
+ sprintf(tmp, "rxtx%d", i);
+ clk = devm_clk_get(&pdev->dev, tmp);
+ if (IS_ERR(clk)) {
+ dev_err(dev, "no rxtx%d clock in devicetree\n", i);
+ return PTR_ERR(clk);
+ }
+ if (!clk_get_rate(clk))
+ continue;
+
+ ret = fsl_spdif_txclk_caldiv(spdif_priv, clk, savesub, index,
+ i == STC_TXCLK_SPDIF_ROOT);
+ if (savesub == ret)
+ continue;
+
+ savesub = ret;
+ spdif_priv->txclk[index] = clk;
+ spdif_priv->txclk_src[index] = i;
+
+ /* To quick catch a divisor, we allow a 0.1% deviation */
+ if (savesub < 100)
+ break;
+ }
+
+ dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n",
+ spdif_priv->txclk_src[index], rate[index]);
+ dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n",
+ spdif_priv->txclk_df[index], rate[index]);
+ if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk))
+ dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n",
+ spdif_priv->sysclk_df[index], rate[index]);
+ dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n",
+ rate[index], spdif_priv->txrate[index]);
+
+ return 0;
+}
+
+static int fsl_spdif_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct fsl_spdif_priv *spdif_priv;
+ struct spdif_mixer_control *ctrl;
+ struct resource *res;
+ void __iomem *regs;
+ int irq, ret, i;
+
+ if (!np)
+ return -ENODEV;
+
+ spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL);
+ if (!spdif_priv)
+ return -ENOMEM;
+
+ spdif_priv->pdev = pdev;
+
+ /* Initialize this copy of the CPU DAI driver structure */
+ memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
+ spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev);
+
+ /* Get the addresses and IRQ */
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ regs = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+ "core", regs, &fsl_spdif_regmap_config);
+ if (IS_ERR(spdif_priv->regmap)) {
+ dev_err(&pdev->dev, "regmap init failed\n");
+ return PTR_ERR(spdif_priv->regmap);
+ }
+
+ irq = platform_get_irq(pdev, 0);
+ if (irq < 0) {
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+ return irq;
+ }
+
+ ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0,
+ dev_name(&pdev->dev), spdif_priv);
+ if (ret) {
+ dev_err(&pdev->dev, "could not claim irq %u\n", irq);
+ return ret;
+ }
+
+ /* Get system clock for rx clock rate calculation */
+ spdif_priv->sysclk = devm_clk_get(&pdev->dev, "rxtx5");
+ if (IS_ERR(spdif_priv->sysclk)) {
+ dev_err(&pdev->dev, "no sys clock (rxtx5) in devicetree\n");
+ return PTR_ERR(spdif_priv->sysclk);
+ }
+
+ /* Get core clock for data register access via DMA */
+ spdif_priv->coreclk = devm_clk_get(&pdev->dev, "core");
+ if (IS_ERR(spdif_priv->coreclk)) {
+ dev_err(&pdev->dev, "no core clock in devicetree\n");
+ return PTR_ERR(spdif_priv->coreclk);
+ }
+
+ /* Select clock source for rx/tx clock */
+ spdif_priv->rxclk = devm_clk_get(&pdev->dev, "rxtx1");
+ if (IS_ERR(spdif_priv->rxclk)) {
+ dev_err(&pdev->dev, "no rxtx1 clock in devicetree\n");
+ return PTR_ERR(spdif_priv->rxclk);
+ }
+ spdif_priv->rxclk_src = DEFAULT_RXCLK_SRC;
+
+ for (i = 0; i < SPDIF_TXRATE_MAX; i++) {
+ ret = fsl_spdif_probe_txclk(spdif_priv, i);
+ if (ret)
+ return ret;
+ }
+
+ /* Initial spinlock for control data */
+ ctrl = &spdif_priv->fsl_spdif_control;
+ spin_lock_init(&ctrl->ctl_lock);
+
+ /* Init tx channel status default value */
+ ctrl->ch_status[0] = IEC958_AES0_CON_NOT_COPYRIGHT |
+ IEC958_AES0_CON_EMPHASIS_5015;
+ ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID;
+ ctrl->ch_status[2] = 0x00;
+ ctrl->ch_status[3] = IEC958_AES3_CON_FS_44100 |
+ IEC958_AES3_CON_CLOCK_1000PPM;
+
+ spdif_priv->dpll_locked = false;
+
+ spdif_priv->dma_params_tx.maxburst = FSL_SPDIF_TXFIFO_WML;
+ spdif_priv->dma_params_rx.maxburst = FSL_SPDIF_RXFIFO_WML;
+ spdif_priv->dma_params_tx.addr = res->start + REG_SPDIF_STL;
+ spdif_priv->dma_params_rx.addr = res->start + REG_SPDIF_SRL;
+
+ /* Register with ASoC */
+ dev_set_drvdata(&pdev->dev, spdif_priv);
+
+ ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
+ &spdif_priv->cpu_dai_drv, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+ return ret;
+ }
+
+ ret = imx_pcm_dma_init(pdev);
+ if (ret)
+ dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret);
+
+ return ret;
+}
+
+static const struct of_device_id fsl_spdif_dt_ids[] = {
+ { .compatible = "fsl,imx35-spdif", },
+ { .compatible = "fsl,vf610-spdif", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids);
+
+static struct platform_driver fsl_spdif_driver = {
+ .driver = {
+ .name = "fsl-spdif-dai",
+ .of_match_table = fsl_spdif_dt_ids,
+ },
+ .probe = fsl_spdif_probe,
+};
+
+module_platform_driver(fsl_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale S/PDIF CPU DAI Driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:fsl-spdif-dai");
diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h
new file mode 100644
index 000000000..00bd3514c
--- /dev/null
+++ b/sound/soc/fsl/fsl_spdif.h
@@ -0,0 +1,199 @@
+/*
+ * fsl_spdif.h - ALSA S/PDIF interface for the Freescale i.MX SoC
+ *
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <b42378@freescale.com>
+ *
+ * Based on fsl_ssi.h
+ * Author: Timur Tabi <timur@freescale.com>
+ * Copyright 2007-2008 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _FSL_SPDIF_DAI_H
+#define _FSL_SPDIF_DAI_H
+
+/* S/PDIF Register Map */
+#define REG_SPDIF_SCR 0x0 /* SPDIF Configuration Register */
+#define REG_SPDIF_SRCD 0x4 /* CDText Control Register */
+#define REG_SPDIF_SRPC 0x8 /* PhaseConfig Register */
+#define REG_SPDIF_SIE 0xc /* InterruptEn Register */
+#define REG_SPDIF_SIS 0x10 /* InterruptStat Register */
+#define REG_SPDIF_SIC 0x10 /* InterruptClear Register */
+#define REG_SPDIF_SRL 0x14 /* SPDIFRxLeft Register */
+#define REG_SPDIF_SRR 0x18 /* SPDIFRxRight Register */
+#define REG_SPDIF_SRCSH 0x1c /* SPDIFRxCChannel_h Register */
+#define REG_SPDIF_SRCSL 0x20 /* SPDIFRxCChannel_l Register */
+#define REG_SPDIF_SRU 0x24 /* UchannelRx Register */
+#define REG_SPDIF_SRQ 0x28 /* QchannelRx Register */
+#define REG_SPDIF_STL 0x2C /* SPDIFTxLeft Register */
+#define REG_SPDIF_STR 0x30 /* SPDIFTxRight Register */
+#define REG_SPDIF_STCSCH 0x34 /* SPDIFTxCChannelCons_h Register */
+#define REG_SPDIF_STCSCL 0x38 /* SPDIFTxCChannelCons_l Register */
+#define REG_SPDIF_SRFM 0x44 /* FreqMeas Register */
+#define REG_SPDIF_STC 0x50 /* SPDIFTxClk Register */
+
+
+/* SPDIF Configuration register */
+#define SCR_RXFIFO_CTL_OFFSET 23
+#define SCR_RXFIFO_CTL_MASK (1 << SCR_RXFIFO_CTL_OFFSET)
+#define SCR_RXFIFO_CTL_ZERO (1 << SCR_RXFIFO_CTL_OFFSET)
+#define SCR_RXFIFO_OFF_OFFSET 22
+#define SCR_RXFIFO_OFF_MASK (1 << SCR_RXFIFO_OFF_OFFSET)
+#define SCR_RXFIFO_OFF (1 << SCR_RXFIFO_OFF_OFFSET)
+#define SCR_RXFIFO_RST_OFFSET 21
+#define SCR_RXFIFO_RST_MASK (1 << SCR_RXFIFO_RST_OFFSET)
+#define SCR_RXFIFO_RST (1 << SCR_RXFIFO_RST_OFFSET)
+#define SCR_RXFIFO_FSEL_OFFSET 19
+#define SCR_RXFIFO_FSEL_MASK (0x3 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF0 (0x0 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF4 (0x1 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF8 (0x2 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_FSEL_IF12 (0x3 << SCR_RXFIFO_FSEL_OFFSET)
+#define SCR_RXFIFO_AUTOSYNC_OFFSET 18
+#define SCR_RXFIFO_AUTOSYNC_MASK (1 << SCR_RXFIFO_AUTOSYNC_OFFSET)
+#define SCR_RXFIFO_AUTOSYNC (1 << SCR_RXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_AUTOSYNC_OFFSET 17
+#define SCR_TXFIFO_AUTOSYNC_MASK (1 << SCR_TXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_AUTOSYNC (1 << SCR_TXFIFO_AUTOSYNC_OFFSET)
+#define SCR_TXFIFO_FSEL_OFFSET 15
+#define SCR_TXFIFO_FSEL_MASK (0x3 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF0 (0x0 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF4 (0x1 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF8 (0x2 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_TXFIFO_FSEL_IF12 (0x3 << SCR_TXFIFO_FSEL_OFFSET)
+#define SCR_LOW_POWER (1 << 13)
+#define SCR_SOFT_RESET (1 << 12)
+#define SCR_TXFIFO_CTRL_OFFSET 10
+#define SCR_TXFIFO_CTRL_MASK (0x3 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_ZERO (0x0 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_NORMAL (0x1 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_TXFIFO_CTRL_ONESAMPLE (0x2 << SCR_TXFIFO_CTRL_OFFSET)
+#define SCR_DMA_RX_EN_OFFSET 9
+#define SCR_DMA_RX_EN_MASK (1 << SCR_DMA_RX_EN_OFFSET)
+#define SCR_DMA_RX_EN (1 << SCR_DMA_RX_EN_OFFSET)
+#define SCR_DMA_TX_EN_OFFSET 8
+#define SCR_DMA_TX_EN_MASK (1 << SCR_DMA_TX_EN_OFFSET)
+#define SCR_DMA_TX_EN (1 << SCR_DMA_TX_EN_OFFSET)
+#define SCR_VAL_OFFSET 5
+#define SCR_VAL_MASK (1 << SCR_VAL_OFFSET)
+#define SCR_VAL_CLEAR (1 << SCR_VAL_OFFSET)
+#define SCR_TXSEL_OFFSET 2
+#define SCR_TXSEL_MASK (0x7 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_OFF (0 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_RX (1 << SCR_TXSEL_OFFSET)
+#define SCR_TXSEL_NORMAL (0x5 << SCR_TXSEL_OFFSET)
+#define SCR_USRC_SEL_OFFSET 0x0
+#define SCR_USRC_SEL_MASK (0x3 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_NONE (0x0 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET)
+#define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET)
+
+#define SCR_DMA_xX_EN(tx) (tx ? SCR_DMA_TX_EN : SCR_DMA_RX_EN)
+
+/* SPDIF CDText control */
+#define SRCD_CD_USER_OFFSET 1
+#define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET)
+
+/* SPDIF Phase Configuration register */
+#define SRPC_DPLL_LOCKED (1 << 6)
+#define SRPC_CLKSRC_SEL_OFFSET 7
+#define SRPC_CLKSRC_SEL_MASK (0xf << SRPC_CLKSRC_SEL_OFFSET)
+#define SRPC_CLKSRC_SEL_SET(x) ((x << SRPC_CLKSRC_SEL_OFFSET) & SRPC_CLKSRC_SEL_MASK)
+#define SRPC_CLKSRC_SEL_LOCKED_OFFSET1 5
+#define SRPC_CLKSRC_SEL_LOCKED_OFFSET2 2
+#define SRPC_GAINSEL_OFFSET 3
+#define SRPC_GAINSEL_MASK (0x7 << SRPC_GAINSEL_OFFSET)
+#define SRPC_GAINSEL_SET(x) ((x << SRPC_GAINSEL_OFFSET) & SRPC_GAINSEL_MASK)
+
+#define SRPC_CLKSRC_MAX 16
+
+enum spdif_gainsel {
+ GAINSEL_MULTI_24 = 0,
+ GAINSEL_MULTI_16,
+ GAINSEL_MULTI_12,
+ GAINSEL_MULTI_8,
+ GAINSEL_MULTI_6,
+ GAINSEL_MULTI_4,
+ GAINSEL_MULTI_3,
+};
+#define GAINSEL_MULTI_MAX (GAINSEL_MULTI_3 + 1)
+#define SPDIF_DEFAULT_GAINSEL GAINSEL_MULTI_8
+
+/* SPDIF interrupt mask define */
+#define INT_DPLL_LOCKED (1 << 20)
+#define INT_TXFIFO_UNOV (1 << 19)
+#define INT_TXFIFO_RESYNC (1 << 18)
+#define INT_CNEW (1 << 17)
+#define INT_VAL_NOGOOD (1 << 16)
+#define INT_SYM_ERR (1 << 15)
+#define INT_BIT_ERR (1 << 14)
+#define INT_URX_FUL (1 << 10)
+#define INT_URX_OV (1 << 9)
+#define INT_QRX_FUL (1 << 8)
+#define INT_QRX_OV (1 << 7)
+#define INT_UQ_SYNC (1 << 6)
+#define INT_UQ_ERR (1 << 5)
+#define INT_RXFIFO_UNOV (1 << 4)
+#define INT_RXFIFO_RESYNC (1 << 3)
+#define INT_LOSS_LOCK (1 << 2)
+#define INT_TX_EM (1 << 1)
+#define INT_RXFIFO_FUL (1 << 0)
+
+/* SPDIF Clock register */
+#define STC_SYSCLK_DF_OFFSET 11
+#define STC_SYSCLK_DF_MASK (0x1ff << STC_SYSCLK_DF_OFFSET)
+#define STC_SYSCLK_DF(x) ((((x) - 1) << STC_SYSCLK_DF_OFFSET) & STC_SYSCLK_DF_MASK)
+#define STC_TXCLK_SRC_OFFSET 8
+#define STC_TXCLK_SRC_MASK (0x7 << STC_TXCLK_SRC_OFFSET)
+#define STC_TXCLK_SRC_SET(x) ((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK)
+#define STC_TXCLK_ALL_EN_OFFSET 7
+#define STC_TXCLK_ALL_EN_MASK (1 << STC_TXCLK_ALL_EN_OFFSET)
+#define STC_TXCLK_ALL_EN (1 << STC_TXCLK_ALL_EN_OFFSET)
+#define STC_TXCLK_DF_OFFSET 0
+#define STC_TXCLK_DF_MASK (0x7ff << STC_TXCLK_DF_OFFSET)
+#define STC_TXCLK_DF(x) ((((x) - 1) << STC_TXCLK_DF_OFFSET) & STC_TXCLK_DF_MASK)
+#define STC_TXCLK_SRC_MAX 8
+
+#define STC_TXCLK_SPDIF_ROOT 1
+
+/* SPDIF tx rate */
+enum spdif_txrate {
+ SPDIF_TXRATE_32000 = 0,
+ SPDIF_TXRATE_44100,
+ SPDIF_TXRATE_48000,
+ SPDIF_TXRATE_96000,
+ SPDIF_TXRATE_192000,
+};
+#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_192000 + 1)
+
+
+#define SPDIF_CSTATUS_BYTE 6
+#define SPDIF_UBITS_SIZE 96
+#define SPDIF_QSUB_SIZE (SPDIF_UBITS_SIZE / 8)
+
+
+#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_192000)
+
+#define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_96000)
+
+#define FSL_SPDIF_FORMATS_PLAYBACK (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+#define FSL_SPDIF_FORMATS_CAPTURE (SNDRV_PCM_FMTBIT_S24_LE)
+
+#endif /* _FSL_SPDIF_DAI_H */
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
new file mode 100644
index 000000000..0d4880421
--- /dev/null
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -0,0 +1,1485 @@
+/*
+ * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007-2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ *
+ *
+ * Some notes why imx-pcm-fiq is used instead of DMA on some boards:
+ *
+ * The i.MX SSI core has some nasty limitations in AC97 mode. While most
+ * sane processor vendors have a FIFO per AC97 slot, the i.MX has only
+ * one FIFO which combines all valid receive slots. We cannot even select
+ * which slots we want to receive. The WM9712 with which this driver
+ * was developed with always sends GPIO status data in slot 12 which
+ * we receive in our (PCM-) data stream. The only chance we have is to
+ * manually skip this data in the FIQ handler. With sampling rates different
+ * from 48000Hz not every frame has valid receive data, so the ratio
+ * between pcm data and GPIO status data changes. Our FIQ handler is not
+ * able to handle this, hence this driver only works with 48000Hz sampling
+ * rate.
+ * Reading and writing AC97 registers is another challenge. The core
+ * provides us status bits when the read register is updated with *another*
+ * value. When we read the same register two times (and the register still
+ * contains the same value) these status bits are not set. We work
+ * around this by not polling these bits but only wait a fixed delay.
+ */
+
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/spinlock.h>
+#include <linux/of.h>
+#include <linux/of_address.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "fsl_ssi.h"
+#include "imx-pcm.h"
+
+/**
+ * FSLSSI_I2S_RATES: sample rates supported by the I2S
+ *
+ * This driver currently only supports the SSI running in I2S slave mode,
+ * which means the codec determines the sample rate. Therefore, we tell
+ * ALSA that we support all rates and let the codec driver decide what rates
+ * are really supported.
+ */
+#define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS
+
+/**
+ * FSLSSI_I2S_FORMATS: audio formats supported by the SSI
+ *
+ * The SSI has a limitation in that the samples must be in the same byte
+ * order as the host CPU. This is because when multiple bytes are written
+ * to the STX register, the bytes and bits must be written in the same
+ * order. The STX is a shift register, so all the bits need to be aligned
+ * (bit-endianness must match byte-endianness). Processors typically write
+ * the bits within a byte in the same order that the bytes of a word are
+ * written in. So if the host CPU is big-endian, then only big-endian
+ * samples will be written to STX properly.
+ */
+#ifdef __BIG_ENDIAN
+#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_S20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S24_BE)
+#else
+#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+#endif
+
+#define FSLSSI_SIER_DBG_RX_FLAGS (CCSR_SSI_SIER_RFF0_EN | \
+ CCSR_SSI_SIER_RLS_EN | CCSR_SSI_SIER_RFS_EN | \
+ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_RFRC_EN)
+#define FSLSSI_SIER_DBG_TX_FLAGS (CCSR_SSI_SIER_TFE0_EN | \
+ CCSR_SSI_SIER_TLS_EN | CCSR_SSI_SIER_TFS_EN | \
+ CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TFRC_EN)
+
+enum fsl_ssi_type {
+ FSL_SSI_MCP8610,
+ FSL_SSI_MX21,
+ FSL_SSI_MX35,
+ FSL_SSI_MX51,
+};
+
+struct fsl_ssi_reg_val {
+ u32 sier;
+ u32 srcr;
+ u32 stcr;
+ u32 scr;
+};
+
+struct fsl_ssi_rxtx_reg_val {
+ struct fsl_ssi_reg_val rx;
+ struct fsl_ssi_reg_val tx;
+};
+static const struct regmap_config fsl_ssi_regconfig = {
+ .max_register = CCSR_SSI_SACCDIS,
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+ .val_format_endian = REGMAP_ENDIAN_NATIVE,
+};
+
+struct fsl_ssi_soc_data {
+ bool imx;
+ bool offline_config;
+ u32 sisr_write_mask;
+};
+
+/**
+ * fsl_ssi_private: per-SSI private data
+ *
+ * @reg: Pointer to the regmap registers
+ * @irq: IRQ of this SSI
+ * @cpu_dai_drv: CPU DAI driver for this device
+ *
+ * @dai_fmt: DAI configuration this device is currently used with
+ * @i2s_mode: i2s and network mode configuration of the device. Is used to
+ * switch between normal and i2s/network mode
+ * mode depending on the number of channels
+ * @use_dma: DMA is used or FIQ with stream filter
+ * @use_dual_fifo: DMA with support for both FIFOs used
+ * @fifo_deph: Depth of the SSI FIFOs
+ * @rxtx_reg_val: Specific register settings for receive/transmit configuration
+ *
+ * @clk: SSI clock
+ * @baudclk: SSI baud clock for master mode
+ * @baudclk_streams: Active streams that are using baudclk
+ * @bitclk_freq: bitclock frequency set by .set_dai_sysclk
+ *
+ * @dma_params_tx: DMA transmit parameters
+ * @dma_params_rx: DMA receive parameters
+ * @ssi_phys: physical address of the SSI registers
+ *
+ * @fiq_params: FIQ stream filtering parameters
+ *
+ * @pdev: Pointer to pdev used for deprecated fsl-ssi sound card
+ *
+ * @dbg_stats: Debugging statistics
+ *
+ * @soc: SoC specifc data
+ */
+struct fsl_ssi_private {
+ struct regmap *regs;
+ int irq;
+ struct snd_soc_dai_driver cpu_dai_drv;
+
+ unsigned int dai_fmt;
+ u8 i2s_mode;
+ bool use_dma;
+ bool use_dual_fifo;
+ bool has_ipg_clk_name;
+ unsigned int fifo_depth;
+ struct fsl_ssi_rxtx_reg_val rxtx_reg_val;
+
+ struct clk *clk;
+ struct clk *baudclk;
+ unsigned int baudclk_streams;
+ unsigned int bitclk_freq;
+
+ /* DMA params */
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+ dma_addr_t ssi_phys;
+
+ /* params for non-dma FIQ stream filtered mode */
+ struct imx_pcm_fiq_params fiq_params;
+
+ /* Used when using fsl-ssi as sound-card. This is only used by ppc and
+ * should be replaced with simple-sound-card. */
+ struct platform_device *pdev;
+
+ struct fsl_ssi_dbg dbg_stats;
+
+ const struct fsl_ssi_soc_data *soc;
+};
+
+/*
+ * imx51 and later SoCs have a slightly different IP that allows the
+ * SSI configuration while the SSI unit is running.
+ *
+ * More important, it is necessary on those SoCs to configure the
+ * sperate TX/RX DMA bits just before starting the stream
+ * (fsl_ssi_trigger). The SDMA unit has to be configured before fsl_ssi
+ * sends any DMA requests to the SDMA unit, otherwise it is not defined
+ * how the SDMA unit handles the DMA request.
+ *
+ * SDMA units are present on devices starting at imx35 but the imx35
+ * reference manual states that the DMA bits should not be changed
+ * while the SSI unit is running (SSIEN). So we support the necessary
+ * online configuration of fsl-ssi starting at imx51.
+ */
+
+static struct fsl_ssi_soc_data fsl_ssi_mpc8610 = {
+ .imx = false,
+ .offline_config = true,
+ .sisr_write_mask = CCSR_SSI_SISR_RFRC | CCSR_SSI_SISR_TFRC |
+ CCSR_SSI_SISR_ROE0 | CCSR_SSI_SISR_ROE1 |
+ CCSR_SSI_SISR_TUE0 | CCSR_SSI_SISR_TUE1,
+};
+
+static struct fsl_ssi_soc_data fsl_ssi_imx21 = {
+ .imx = true,
+ .offline_config = true,
+ .sisr_write_mask = 0,
+};
+
+static struct fsl_ssi_soc_data fsl_ssi_imx35 = {
+ .imx = true,
+ .offline_config = true,
+ .sisr_write_mask = CCSR_SSI_SISR_RFRC | CCSR_SSI_SISR_TFRC |
+ CCSR_SSI_SISR_ROE0 | CCSR_SSI_SISR_ROE1 |
+ CCSR_SSI_SISR_TUE0 | CCSR_SSI_SISR_TUE1,
+};
+
+static struct fsl_ssi_soc_data fsl_ssi_imx51 = {
+ .imx = true,
+ .offline_config = false,
+ .sisr_write_mask = CCSR_SSI_SISR_ROE0 | CCSR_SSI_SISR_ROE1 |
+ CCSR_SSI_SISR_TUE0 | CCSR_SSI_SISR_TUE1,
+};
+
+static const struct of_device_id fsl_ssi_ids[] = {
+ { .compatible = "fsl,mpc8610-ssi", .data = &fsl_ssi_mpc8610 },
+ { .compatible = "fsl,imx51-ssi", .data = &fsl_ssi_imx51 },
+ { .compatible = "fsl,imx35-ssi", .data = &fsl_ssi_imx35 },
+ { .compatible = "fsl,imx21-ssi", .data = &fsl_ssi_imx21 },
+ {}
+};
+MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
+
+static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private)
+{
+ return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97);
+}
+
+static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
+{
+ return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+ SND_SOC_DAIFMT_CBS_CFS;
+}
+
+static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private)
+{
+ return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+ SND_SOC_DAIFMT_CBM_CFS;
+}
+/**
+ * fsl_ssi_isr: SSI interrupt handler
+ *
+ * Although it's possible to use the interrupt handler to send and receive
+ * data to/from the SSI, we use the DMA instead. Programming is more
+ * complicated, but the performance is much better.
+ *
+ * This interrupt handler is used only to gather statistics.
+ *
+ * @irq: IRQ of the SSI device
+ * @dev_id: pointer to the ssi_private structure for this SSI device
+ */
+static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
+{
+ struct fsl_ssi_private *ssi_private = dev_id;
+ struct regmap *regs = ssi_private->regs;
+ __be32 sisr;
+ __be32 sisr2;
+
+ /* We got an interrupt, so read the status register to see what we
+ were interrupted for. We mask it with the Interrupt Enable register
+ so that we only check for events that we're interested in.
+ */
+ regmap_read(regs, CCSR_SSI_SISR, &sisr);
+
+ sisr2 = sisr & ssi_private->soc->sisr_write_mask;
+ /* Clear the bits that we set */
+ if (sisr2)
+ regmap_write(regs, CCSR_SSI_SISR, sisr2);
+
+ fsl_ssi_dbg_isr(&ssi_private->dbg_stats, sisr);
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * Enable/Disable all rx/tx config flags at once.
+ */
+static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private,
+ bool enable)
+{
+ struct regmap *regs = ssi_private->regs;
+ struct fsl_ssi_rxtx_reg_val *vals = &ssi_private->rxtx_reg_val;
+
+ if (enable) {
+ regmap_update_bits(regs, CCSR_SSI_SIER,
+ vals->rx.sier | vals->tx.sier,
+ vals->rx.sier | vals->tx.sier);
+ regmap_update_bits(regs, CCSR_SSI_SRCR,
+ vals->rx.srcr | vals->tx.srcr,
+ vals->rx.srcr | vals->tx.srcr);
+ regmap_update_bits(regs, CCSR_SSI_STCR,
+ vals->rx.stcr | vals->tx.stcr,
+ vals->rx.stcr | vals->tx.stcr);
+ } else {
+ regmap_update_bits(regs, CCSR_SSI_SRCR,
+ vals->rx.srcr | vals->tx.srcr, 0);
+ regmap_update_bits(regs, CCSR_SSI_STCR,
+ vals->rx.stcr | vals->tx.stcr, 0);
+ regmap_update_bits(regs, CCSR_SSI_SIER,
+ vals->rx.sier | vals->tx.sier, 0);
+ }
+}
+
+/*
+ * Calculate the bits that have to be disabled for the current stream that is
+ * getting disabled. This keeps the bits enabled that are necessary for the
+ * second stream to work if 'stream_active' is true.
+ *
+ * Detailed calculation:
+ * These are the values that need to be active after disabling. For non-active
+ * second stream, this is 0:
+ * vals_stream * !!stream_active
+ *
+ * The following computes the overall differences between the setup for the
+ * to-disable stream and the active stream, a simple XOR:
+ * vals_disable ^ (vals_stream * !!(stream_active))
+ *
+ * The full expression adds a mask on all values we care about
+ */
+#define fsl_ssi_disable_val(vals_disable, vals_stream, stream_active) \
+ ((vals_disable) & \
+ ((vals_disable) ^ ((vals_stream) * (u32)!!(stream_active))))
+
+/*
+ * Enable/Disable a ssi configuration. You have to pass either
+ * ssi_private->rxtx_reg_val.rx or tx as vals parameter.
+ */
+static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
+ struct fsl_ssi_reg_val *vals)
+{
+ struct regmap *regs = ssi_private->regs;
+ struct fsl_ssi_reg_val *avals;
+ int nr_active_streams;
+ u32 scr_val;
+ int keep_active;
+
+ regmap_read(regs, CCSR_SSI_SCR, &scr_val);
+
+ nr_active_streams = !!(scr_val & CCSR_SSI_SCR_TE) +
+ !!(scr_val & CCSR_SSI_SCR_RE);
+
+ if (nr_active_streams - 1 > 0)
+ keep_active = 1;
+ else
+ keep_active = 0;
+
+ /* Find the other direction values rx or tx which we do not want to
+ * modify */
+ if (&ssi_private->rxtx_reg_val.rx == vals)
+ avals = &ssi_private->rxtx_reg_val.tx;
+ else
+ avals = &ssi_private->rxtx_reg_val.rx;
+
+ /* If vals should be disabled, start with disabling the unit */
+ if (!enable) {
+ u32 scr = fsl_ssi_disable_val(vals->scr, avals->scr,
+ keep_active);
+ regmap_update_bits(regs, CCSR_SSI_SCR, scr, 0);
+ }
+
+ /*
+ * We are running on a SoC which does not support online SSI
+ * reconfiguration, so we have to enable all necessary flags at once
+ * even if we do not use them later (capture and playback configuration)
+ */
+ if (ssi_private->soc->offline_config) {
+ if ((enable && !nr_active_streams) ||
+ (!enable && !keep_active))
+ fsl_ssi_rxtx_config(ssi_private, enable);
+
+ goto config_done;
+ }
+
+ /*
+ * Configure single direction units while the SSI unit is running
+ * (online configuration)
+ */
+ if (enable) {
+ regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
+ regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr);
+ regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr);
+ } else {
+ u32 sier;
+ u32 srcr;
+ u32 stcr;
+
+ /*
+ * Disabling the necessary flags for one of rx/tx while the
+ * other stream is active is a little bit more difficult. We
+ * have to disable only those flags that differ between both
+ * streams (rx XOR tx) and that are set in the stream that is
+ * disabled now. Otherwise we could alter flags of the other
+ * stream
+ */
+
+ /* These assignments are simply vals without bits set in avals*/
+ sier = fsl_ssi_disable_val(vals->sier, avals->sier,
+ keep_active);
+ srcr = fsl_ssi_disable_val(vals->srcr, avals->srcr,
+ keep_active);
+ stcr = fsl_ssi_disable_val(vals->stcr, avals->stcr,
+ keep_active);
+
+ regmap_update_bits(regs, CCSR_SSI_SRCR, srcr, 0);
+ regmap_update_bits(regs, CCSR_SSI_STCR, stcr, 0);
+ regmap_update_bits(regs, CCSR_SSI_SIER, sier, 0);
+ }
+
+config_done:
+ /* Enabling of subunits is done after configuration */
+ if (enable)
+ regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr);
+}
+
+
+static void fsl_ssi_rx_config(struct fsl_ssi_private *ssi_private, bool enable)
+{
+ fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.rx);
+}
+
+static void fsl_ssi_tx_config(struct fsl_ssi_private *ssi_private, bool enable)
+{
+ fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.tx);
+}
+
+/*
+ * Setup rx/tx register values used to enable/disable the streams. These will
+ * be used later in fsl_ssi_config to setup the streams without the need to
+ * check for all different SSI modes.
+ */
+static void fsl_ssi_setup_reg_vals(struct fsl_ssi_private *ssi_private)
+{
+ struct fsl_ssi_rxtx_reg_val *reg = &ssi_private->rxtx_reg_val;
+
+ reg->rx.sier = CCSR_SSI_SIER_RFF0_EN;
+ reg->rx.srcr = CCSR_SSI_SRCR_RFEN0;
+ reg->rx.scr = 0;
+ reg->tx.sier = CCSR_SSI_SIER_TFE0_EN;
+ reg->tx.stcr = CCSR_SSI_STCR_TFEN0;
+ reg->tx.scr = 0;
+
+ if (!fsl_ssi_is_ac97(ssi_private)) {
+ reg->rx.scr = CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE;
+ reg->rx.sier |= CCSR_SSI_SIER_RFF0_EN;
+ reg->tx.scr = CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE;
+ reg->tx.sier |= CCSR_SSI_SIER_TFE0_EN;
+ }
+
+ if (ssi_private->use_dma) {
+ reg->rx.sier |= CCSR_SSI_SIER_RDMAE;
+ reg->tx.sier |= CCSR_SSI_SIER_TDMAE;
+ } else {
+ reg->rx.sier |= CCSR_SSI_SIER_RIE;
+ reg->tx.sier |= CCSR_SSI_SIER_TIE;
+ }
+
+ reg->rx.sier |= FSLSSI_SIER_DBG_RX_FLAGS;
+ reg->tx.sier |= FSLSSI_SIER_DBG_TX_FLAGS;
+}
+
+static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private)
+{
+ struct regmap *regs = ssi_private->regs;
+
+ /*
+ * Setup the clock control register
+ */
+ regmap_write(regs, CCSR_SSI_STCCR,
+ CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13));
+ regmap_write(regs, CCSR_SSI_SRCCR,
+ CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13));
+
+ /*
+ * Enable AC97 mode and startup the SSI
+ */
+ regmap_write(regs, CCSR_SSI_SACNT,
+ CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV);
+ regmap_write(regs, CCSR_SSI_SACCDIS, 0xff);
+ regmap_write(regs, CCSR_SSI_SACCEN, 0x300);
+
+ /*
+ * Enable SSI, Transmit and Receive. AC97 has to communicate with the
+ * codec before a stream is started.
+ */
+ regmap_update_bits(regs, CCSR_SSI_SCR,
+ CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE,
+ CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE);
+
+ regmap_write(regs, CCSR_SSI_SOR, CCSR_SSI_SOR_WAIT(3));
+}
+
+/**
+ * fsl_ssi_startup: create a new substream
+ *
+ * This is the first function called when a stream is opened.
+ *
+ * If this is the first stream open, then grab the IRQ and program most of
+ * the SSI registers.
+ */
+static int fsl_ssi_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private =
+ snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ int ret;
+
+ ret = clk_prepare_enable(ssi_private->clk);
+ if (ret)
+ return ret;
+
+ /* When using dual fifo mode, it is safer to ensure an even period
+ * size. If appearing to an odd number while DMA always starts its
+ * task from fifo0, fifo1 would be neglected at the end of each
+ * period. But SSI would still access fifo1 with an invalid data.
+ */
+ if (ssi_private->use_dual_fifo)
+ snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
+
+ return 0;
+}
+
+/**
+ * fsl_ssi_shutdown: shutdown the SSI
+ *
+ */
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private =
+ snd_soc_dai_get_drvdata(rtd->cpu_dai);
+
+ clk_disable_unprepare(ssi_private->clk);
+
+}
+
+/**
+ * fsl_ssi_set_bclk - configure Digital Audio Interface bit clock
+ *
+ * Note: This function can be only called when using SSI as DAI master
+ *
+ * Quick instruction for parameters:
+ * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels
+ * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK.
+ */
+static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regs = ssi_private->regs;
+ int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret;
+ u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i;
+ unsigned long clkrate, baudrate, tmprate;
+ u64 sub, savesub = 100000;
+ unsigned int freq;
+ bool baudclk_is_used;
+
+ /* Prefer the explicitly set bitclock frequency */
+ if (ssi_private->bitclk_freq)
+ freq = ssi_private->bitclk_freq;
+ else
+ freq = params_channels(hw_params) * 32 * params_rate(hw_params);
+
+ /* Don't apply it to any non-baudclk circumstance */
+ if (IS_ERR(ssi_private->baudclk))
+ return -EINVAL;
+
+ baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream));
+
+ /* It should be already enough to divide clock by setting pm alone */
+ psr = 0;
+ div2 = 0;
+
+ factor = (div2 + 1) * (7 * psr + 1) * 2;
+
+ for (i = 0; i < 255; i++) {
+ tmprate = freq * factor * (i + 1);
+
+ if (baudclk_is_used)
+ clkrate = clk_get_rate(ssi_private->baudclk);
+ else
+ clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
+
+ /*
+ * Hardware limitation: The bclk rate must be
+ * never greater than 1/5 IPG clock rate
+ */
+ if (clkrate * 5 > clk_get_rate(ssi_private->clk))
+ continue;
+
+ clkrate /= factor;
+ afreq = clkrate / (i + 1);
+
+ if (freq == afreq)
+ sub = 0;
+ else if (freq / afreq == 1)
+ sub = freq - afreq;
+ else if (afreq / freq == 1)
+ sub = afreq - freq;
+ else
+ continue;
+
+ /* Calculate the fraction */
+ sub *= 100000;
+ do_div(sub, freq);
+
+ if (sub < savesub) {
+ baudrate = tmprate;
+ savesub = sub;
+ pm = i;
+ }
+
+ /* We are lucky */
+ if (savesub == 0)
+ break;
+ }
+
+ /* No proper pm found if it is still remaining the initial value */
+ if (pm == 999) {
+ dev_err(cpu_dai->dev, "failed to handle the required sysclk\n");
+ return -EINVAL;
+ }
+
+ stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) |
+ (psr ? CCSR_SSI_SxCCR_PSR : 0);
+ mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 |
+ CCSR_SSI_SxCCR_PSR;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || synchronous)
+ regmap_update_bits(regs, CCSR_SSI_STCCR, mask, stccr);
+ else
+ regmap_update_bits(regs, CCSR_SSI_SRCCR, mask, stccr);
+
+ if (!baudclk_is_used) {
+ ret = clk_set_rate(ssi_private->baudclk, baudrate);
+ if (ret) {
+ dev_err(cpu_dai->dev, "failed to set baudclk rate\n");
+ return -EINVAL;
+ }
+ }
+
+ return 0;
+}
+
+static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+
+ ssi_private->bitclk_freq = freq;
+
+ return 0;
+}
+
+/**
+ * fsl_ssi_hw_params - program the sample size
+ *
+ * Most of the SSI registers have been programmed in the startup function,
+ * but the word length must be programmed here. Unfortunately, programming
+ * the SxCCR.WL bits requires the SSI to be temporarily disabled. This can
+ * cause a problem with supporting simultaneous playback and capture. If
+ * the SSI is already playing a stream, then that stream may be temporarily
+ * stopped when you start capture.
+ *
+ * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
+ * clock master.
+ */
+static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai)
+{
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regs = ssi_private->regs;
+ unsigned int channels = params_channels(hw_params);
+ unsigned int sample_size =
+ snd_pcm_format_width(params_format(hw_params));
+ u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
+ int ret;
+ u32 scr_val;
+ int enabled;
+
+ regmap_read(regs, CCSR_SSI_SCR, &scr_val);
+ enabled = scr_val & CCSR_SSI_SCR_SSIEN;
+
+ /*
+ * If we're in synchronous mode, and the SSI is already enabled,
+ * then STCCR is already set properly.
+ */
+ if (enabled && ssi_private->cpu_dai_drv.symmetric_rates)
+ return 0;
+
+ if (fsl_ssi_is_i2s_master(ssi_private)) {
+ ret = fsl_ssi_set_bclk(substream, cpu_dai, hw_params);
+ if (ret)
+ return ret;
+
+ /* Do not enable the clock if it is already enabled */
+ if (!(ssi_private->baudclk_streams & BIT(substream->stream))) {
+ ret = clk_prepare_enable(ssi_private->baudclk);
+ if (ret)
+ return ret;
+
+ ssi_private->baudclk_streams |= BIT(substream->stream);
+ }
+ }
+
+ if (!fsl_ssi_is_ac97(ssi_private)) {
+ u8 i2smode;
+ /*
+ * Switch to normal net mode in order to have a frame sync
+ * signal every 32 bits instead of 16 bits
+ */
+ if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16)
+ i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL |
+ CCSR_SSI_SCR_NET;
+ else
+ i2smode = ssi_private->i2s_mode;
+
+ regmap_update_bits(regs, CCSR_SSI_SCR,
+ CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
+ channels == 1 ? 0 : i2smode);
+ }
+
+ /*
+ * FIXME: The documentation says that SxCCR[WL] should not be
+ * modified while the SSI is enabled. The only time this can
+ * happen is if we're trying to do simultaneous playback and
+ * capture in asynchronous mode. Unfortunately, I have been enable
+ * to get that to work at all on the P1022DS. Therefore, we don't
+ * bother to disable/enable the SSI when setting SxCCR[WL], because
+ * the SSI will stop anyway. Maybe one day, this will get fixed.
+ */
+
+ /* In synchronous mode, the SSI uses STCCR for capture */
+ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
+ ssi_private->cpu_dai_drv.symmetric_rates)
+ regmap_update_bits(regs, CCSR_SSI_STCCR, CCSR_SSI_SxCCR_WL_MASK,
+ wl);
+ else
+ regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK,
+ wl);
+
+ return 0;
+}
+
+static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private =
+ snd_soc_dai_get_drvdata(rtd->cpu_dai);
+
+ if (fsl_ssi_is_i2s_master(ssi_private) &&
+ ssi_private->baudclk_streams & BIT(substream->stream)) {
+ clk_disable_unprepare(ssi_private->baudclk);
+ ssi_private->baudclk_streams &= ~BIT(substream->stream);
+ }
+
+ return 0;
+}
+
+static int _fsl_ssi_set_dai_fmt(struct device *dev,
+ struct fsl_ssi_private *ssi_private,
+ unsigned int fmt)
+{
+ struct regmap *regs = ssi_private->regs;
+ u32 strcr = 0, stcr, srcr, scr, mask;
+ u8 wm;
+
+ ssi_private->dai_fmt = fmt;
+
+ if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
+ dev_err(dev, "baudclk is missing which is necessary for master mode\n");
+ return -EINVAL;
+ }
+
+ fsl_ssi_setup_reg_vals(ssi_private);
+
+ regmap_read(regs, CCSR_SSI_SCR, &scr);
+ scr &= ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK);
+ scr |= CCSR_SSI_SCR_SYNC_TX_FS;
+
+ mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR |
+ CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL |
+ CCSR_SSI_STCR_TEFS;
+ regmap_read(regs, CCSR_SSI_STCR, &stcr);
+ regmap_read(regs, CCSR_SSI_SRCR, &srcr);
+ stcr &= ~mask;
+ srcr &= ~mask;
+
+ ssi_private->i2s_mode = CCSR_SSI_SCR_NET;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFS:
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER;
+ regmap_update_bits(regs, CCSR_SSI_STCCR,
+ CCSR_SSI_SxCCR_DC_MASK,
+ CCSR_SSI_SxCCR_DC(2));
+ regmap_update_bits(regs, CCSR_SSI_SRCCR,
+ CCSR_SSI_SxCCR_DC_MASK,
+ CCSR_SSI_SxCCR_DC(2));
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Data on rising edge of bclk, frame low, 1clk before data */
+ strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP |
+ CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /* Data on rising edge of bclk, frame high */
+ strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* Data on rising edge of bclk, frame high, 1clk before data */
+ strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP |
+ CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ /* Data on rising edge of bclk, frame high */
+ strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP |
+ CCSR_SSI_STCR_TXBIT0;
+ break;
+ case SND_SOC_DAIFMT_AC97:
+ ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_NORMAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ scr |= ssi_private->i2s_mode;
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ /* Nothing to do for both normal cases */
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ /* Invert bit clock */
+ strcr ^= CCSR_SSI_STCR_TSCKP;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ /* Invert frame clock */
+ strcr ^= CCSR_SSI_STCR_TFSI;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ /* Invert both clocks */
+ strcr ^= CCSR_SSI_STCR_TSCKP;
+ strcr ^= CCSR_SSI_STCR_TFSI;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* DAI clock master masks */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR;
+ scr |= CCSR_SSI_SCR_SYS_CLK_EN;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ strcr &= ~CCSR_SSI_STCR_TXDIR;
+ strcr |= CCSR_SSI_STCR_TFDIR;
+ scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ stcr |= strcr;
+ srcr |= strcr;
+
+ if (ssi_private->cpu_dai_drv.symmetric_rates) {
+ /* Need to clear RXDIR when using SYNC mode */
+ srcr &= ~CCSR_SSI_SRCR_RXDIR;
+ scr |= CCSR_SSI_SCR_SYN;
+ }
+
+ regmap_write(regs, CCSR_SSI_STCR, stcr);
+ regmap_write(regs, CCSR_SSI_SRCR, srcr);
+ regmap_write(regs, CCSR_SSI_SCR, scr);
+
+ /*
+ * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't
+ * use FIFO 1. We program the transmit water to signal a DMA transfer
+ * if there are only two (or fewer) elements left in the FIFO. Two
+ * elements equals one frame (left channel, right channel). This value,
+ * however, depends on the depth of the transmit buffer.
+ *
+ * We set the watermark on the same level as the DMA burstsize. For
+ * fiq it is probably better to use the biggest possible watermark
+ * size.
+ */
+ if (ssi_private->use_dma)
+ wm = ssi_private->fifo_depth - 2;
+ else
+ wm = ssi_private->fifo_depth;
+
+ regmap_write(regs, CCSR_SSI_SFCSR,
+ CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) |
+ CCSR_SSI_SFCSR_TFWM1(wm) | CCSR_SSI_SFCSR_RFWM1(wm));
+
+ if (ssi_private->use_dual_fifo) {
+ regmap_update_bits(regs, CCSR_SSI_SRCR, CCSR_SSI_SRCR_RFEN1,
+ CCSR_SSI_SRCR_RFEN1);
+ regmap_update_bits(regs, CCSR_SSI_STCR, CCSR_SSI_STCR_TFEN1,
+ CCSR_SSI_STCR_TFEN1);
+ regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_TCH_EN,
+ CCSR_SSI_SCR_TCH_EN);
+ }
+
+ if (fmt & SND_SOC_DAIFMT_AC97)
+ fsl_ssi_setup_ac97(ssi_private);
+
+ return 0;
+
+}
+
+/**
+ * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format.
+ */
+static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+
+ return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt);
+}
+
+/**
+ * fsl_ssi_set_dai_tdm_slot - set TDM slot number
+ *
+ * Note: This function can be only called when using SSI as DAI master
+ */
+static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask,
+ u32 rx_mask, int slots, int slot_width)
+{
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+ struct regmap *regs = ssi_private->regs;
+ u32 val;
+
+ /* The slot number should be >= 2 if using Network mode or I2S mode */
+ regmap_read(regs, CCSR_SSI_SCR, &val);
+ val &= CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET;
+ if (val && slots < 2) {
+ dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n");
+ return -EINVAL;
+ }
+
+ regmap_update_bits(regs, CCSR_SSI_STCCR, CCSR_SSI_SxCCR_DC_MASK,
+ CCSR_SSI_SxCCR_DC(slots));
+ regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_DC_MASK,
+ CCSR_SSI_SxCCR_DC(slots));
+
+ /* The register SxMSKs needs SSI to provide essential clock due to
+ * hardware design. So we here temporarily enable SSI to set them.
+ */
+ regmap_read(regs, CCSR_SSI_SCR, &val);
+ val &= CCSR_SSI_SCR_SSIEN;
+ regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN,
+ CCSR_SSI_SCR_SSIEN);
+
+ regmap_write(regs, CCSR_SSI_STMSK, ~tx_mask);
+ regmap_write(regs, CCSR_SSI_SRMSK, ~rx_mask);
+
+ regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val);
+
+ return 0;
+}
+
+/**
+ * fsl_ssi_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ *
+ * The DMA channel is in external master start and pause mode, which
+ * means the SSI completely controls the flow of data.
+ */
+static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct regmap *regs = ssi_private->regs;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ fsl_ssi_tx_config(ssi_private, true);
+ else
+ fsl_ssi_rx_config(ssi_private, true);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ fsl_ssi_tx_config(ssi_private, false);
+ else
+ fsl_ssi_rx_config(ssi_private, false);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ if (fsl_ssi_is_ac97(ssi_private)) {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ regmap_write(regs, CCSR_SSI_SOR, CCSR_SSI_SOR_TX_CLR);
+ else
+ regmap_write(regs, CCSR_SSI_SOR, CCSR_SSI_SOR_RX_CLR);
+ }
+
+ return 0;
+}
+
+static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
+{
+ struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai);
+
+ if (ssi_private->soc->imx && ssi_private->use_dma) {
+ dai->playback_dma_data = &ssi_private->dma_params_tx;
+ dai->capture_dma_data = &ssi_private->dma_params_rx;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
+ .startup = fsl_ssi_startup,
+ .shutdown = fsl_ssi_shutdown,
+ .hw_params = fsl_ssi_hw_params,
+ .hw_free = fsl_ssi_hw_free,
+ .set_fmt = fsl_ssi_set_dai_fmt,
+ .set_sysclk = fsl_ssi_set_dai_sysclk,
+ .set_tdm_slot = fsl_ssi_set_dai_tdm_slot,
+ .trigger = fsl_ssi_trigger,
+};
+
+/* Template for the CPU dai driver structure */
+static struct snd_soc_dai_driver fsl_ssi_dai_template = {
+ .probe = fsl_ssi_dai_probe,
+ .playback = {
+ .stream_name = "CPU-Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = FSLSSI_I2S_RATES,
+ .formats = FSLSSI_I2S_FORMATS,
+ },
+ .capture = {
+ .stream_name = "CPU-Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = FSLSSI_I2S_RATES,
+ .formats = FSLSSI_I2S_FORMATS,
+ },
+ .ops = &fsl_ssi_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_ssi_component = {
+ .name = "fsl-ssi",
+};
+
+static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
+ .bus_control = true,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &fsl_ssi_dai_ops,
+};
+
+
+static struct fsl_ssi_private *fsl_ac97_data;
+
+static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct regmap *regs = fsl_ac97_data->regs;
+ unsigned int lreg;
+ unsigned int lval;
+
+ if (reg > 0x7f)
+ return;
+
+
+ lreg = reg << 12;
+ regmap_write(regs, CCSR_SSI_SACADD, lreg);
+
+ lval = val << 4;
+ regmap_write(regs, CCSR_SSI_SACDAT, lval);
+
+ regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK,
+ CCSR_SSI_SACNT_WR);
+ udelay(100);
+}
+
+static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct regmap *regs = fsl_ac97_data->regs;
+
+ unsigned short val = -1;
+ u32 reg_val;
+ unsigned int lreg;
+
+ lreg = (reg & 0x7f) << 12;
+ regmap_write(regs, CCSR_SSI_SACADD, lreg);
+ regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK,
+ CCSR_SSI_SACNT_RD);
+
+ udelay(100);
+
+ regmap_read(regs, CCSR_SSI_SACDAT, &reg_val);
+ val = (reg_val >> 4) & 0xffff;
+
+ return val;
+}
+
+static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = {
+ .read = fsl_ssi_ac97_read,
+ .write = fsl_ssi_ac97_write,
+};
+
+/**
+ * Make every character in a string lower-case
+ */
+static void make_lowercase(char *s)
+{
+ char *p = s;
+ char c;
+
+ while ((c = *p)) {
+ if ((c >= 'A') && (c <= 'Z'))
+ *p = c + ('a' - 'A');
+ p++;
+ }
+}
+
+static int fsl_ssi_imx_probe(struct platform_device *pdev,
+ struct fsl_ssi_private *ssi_private, void __iomem *iomem)
+{
+ struct device_node *np = pdev->dev.of_node;
+ u32 dmas[4];
+ int ret;
+
+ if (ssi_private->has_ipg_clk_name)
+ ssi_private->clk = devm_clk_get(&pdev->dev, "ipg");
+ else
+ ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(ssi_private->clk)) {
+ ret = PTR_ERR(ssi_private->clk);
+ dev_err(&pdev->dev, "could not get clock: %d\n", ret);
+ return ret;
+ }
+
+ if (!ssi_private->has_ipg_clk_name) {
+ ret = clk_prepare_enable(ssi_private->clk);
+ if (ret) {
+ dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret);
+ return ret;
+ }
+ }
+
+ /* For those SLAVE implementations, we ingore non-baudclk cases
+ * and, instead, abandon MASTER mode that needs baud clock.
+ */
+ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud");
+ if (IS_ERR(ssi_private->baudclk))
+ dev_dbg(&pdev->dev, "could not get baud clock: %ld\n",
+ PTR_ERR(ssi_private->baudclk));
+
+ /*
+ * We have burstsize be "fifo_depth - 2" to match the SSI
+ * watermark setting in fsl_ssi_startup().
+ */
+ ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2;
+ ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2;
+ ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0;
+ ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0;
+
+ ret = of_property_read_u32_array(np, "dmas", dmas, 4);
+ if (ssi_private->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) {
+ ssi_private->use_dual_fifo = true;
+ /* When using dual fifo mode, we need to keep watermark
+ * as even numbers due to dma script limitation.
+ */
+ ssi_private->dma_params_tx.maxburst &= ~0x1;
+ ssi_private->dma_params_rx.maxburst &= ~0x1;
+ }
+
+ if (!ssi_private->use_dma) {
+
+ /*
+ * Some boards use an incompatible codec. To get it
+ * working, we are using imx-fiq-pcm-audio, that
+ * can handle those codecs. DMA is not possible in this
+ * situation.
+ */
+
+ ssi_private->fiq_params.irq = ssi_private->irq;
+ ssi_private->fiq_params.base = iomem;
+ ssi_private->fiq_params.dma_params_rx =
+ &ssi_private->dma_params_rx;
+ ssi_private->fiq_params.dma_params_tx =
+ &ssi_private->dma_params_tx;
+
+ ret = imx_pcm_fiq_init(pdev, &ssi_private->fiq_params);
+ if (ret)
+ goto error_pcm;
+ } else {
+ ret = imx_pcm_dma_init(pdev);
+ if (ret)
+ goto error_pcm;
+ }
+
+ return 0;
+
+error_pcm:
+
+ if (!ssi_private->has_ipg_clk_name)
+ clk_disable_unprepare(ssi_private->clk);
+ return ret;
+}
+
+static void fsl_ssi_imx_clean(struct platform_device *pdev,
+ struct fsl_ssi_private *ssi_private)
+{
+ if (!ssi_private->use_dma)
+ imx_pcm_fiq_exit(pdev);
+ if (!ssi_private->has_ipg_clk_name)
+ clk_disable_unprepare(ssi_private->clk);
+}
+
+static int fsl_ssi_probe(struct platform_device *pdev)
+{
+ struct fsl_ssi_private *ssi_private;
+ int ret = 0;
+ struct device_node *np = pdev->dev.of_node;
+ const struct of_device_id *of_id;
+ const char *p, *sprop;
+ const uint32_t *iprop;
+ struct resource *res;
+ void __iomem *iomem;
+ char name[64];
+
+ /* SSIs that are not connected on the board should have a
+ * status = "disabled"
+ * property in their device tree nodes.
+ */
+ if (!of_device_is_available(np))
+ return -ENODEV;
+
+ of_id = of_match_device(fsl_ssi_ids, &pdev->dev);
+ if (!of_id || !of_id->data)
+ return -EINVAL;
+
+ ssi_private = devm_kzalloc(&pdev->dev, sizeof(*ssi_private),
+ GFP_KERNEL);
+ if (!ssi_private) {
+ dev_err(&pdev->dev, "could not allocate DAI object\n");
+ return -ENOMEM;
+ }
+
+ ssi_private->soc = of_id->data;
+
+ sprop = of_get_property(np, "fsl,mode", NULL);
+ if (sprop) {
+ if (!strcmp(sprop, "ac97-slave"))
+ ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97;
+ }
+
+ ssi_private->use_dma = !of_property_read_bool(np,
+ "fsl,fiq-stream-filter");
+
+ if (fsl_ssi_is_ac97(ssi_private)) {
+ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_ac97_dai,
+ sizeof(fsl_ssi_ac97_dai));
+
+ fsl_ac97_data = ssi_private;
+
+ snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev);
+ } else {
+ /* Initialize this copy of the CPU DAI driver structure */
+ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template,
+ sizeof(fsl_ssi_dai_template));
+ }
+ ssi_private->cpu_dai_drv.name = dev_name(&pdev->dev);
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ iomem = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(iomem))
+ return PTR_ERR(iomem);
+ ssi_private->ssi_phys = res->start;
+
+ ret = of_property_match_string(np, "clock-names", "ipg");
+ if (ret < 0) {
+ ssi_private->has_ipg_clk_name = false;
+ ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
+ &fsl_ssi_regconfig);
+ } else {
+ ssi_private->has_ipg_clk_name = true;
+ ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev,
+ "ipg", iomem, &fsl_ssi_regconfig);
+ }
+ if (IS_ERR(ssi_private->regs)) {
+ dev_err(&pdev->dev, "Failed to init register map\n");
+ return PTR_ERR(ssi_private->regs);
+ }
+
+ ssi_private->irq = platform_get_irq(pdev, 0);
+ if (ssi_private->irq < 0) {
+ dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
+ return ssi_private->irq;
+ }
+
+ /* Are the RX and the TX clocks locked? */
+ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) {
+ ssi_private->cpu_dai_drv.symmetric_rates = 1;
+ ssi_private->cpu_dai_drv.symmetric_channels = 1;
+ ssi_private->cpu_dai_drv.symmetric_samplebits = 1;
+ }
+
+ /* Determine the FIFO depth. */
+ iprop = of_get_property(np, "fsl,fifo-depth", NULL);
+ if (iprop)
+ ssi_private->fifo_depth = be32_to_cpup(iprop);
+ else
+ /* Older 8610 DTs didn't have the fifo-depth property */
+ ssi_private->fifo_depth = 8;
+
+ dev_set_drvdata(&pdev->dev, ssi_private);
+
+ if (ssi_private->soc->imx) {
+ ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem);
+ if (ret)
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component,
+ &ssi_private->cpu_dai_drv, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+ goto error_asoc_register;
+ }
+
+ if (ssi_private->use_dma) {
+ ret = devm_request_irq(&pdev->dev, ssi_private->irq,
+ fsl_ssi_isr, 0, dev_name(&pdev->dev),
+ ssi_private);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "could not claim irq %u\n",
+ ssi_private->irq);
+ goto error_asoc_register;
+ }
+ }
+
+ ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev);
+ if (ret)
+ goto error_asoc_register;
+
+ /*
+ * If codec-handle property is missing from SSI node, we assume
+ * that the machine driver uses new binding which does not require
+ * SSI driver to trigger machine driver's probe.
+ */
+ if (!of_get_property(np, "codec-handle", NULL))
+ goto done;
+
+ /* Trigger the machine driver's probe function. The platform driver
+ * name of the machine driver is taken from /compatible property of the
+ * device tree. We also pass the address of the CPU DAI driver
+ * structure.
+ */
+ sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL);
+ /* Sometimes the compatible name has a "fsl," prefix, so we strip it. */
+ p = strrchr(sprop, ',');
+ if (p)
+ sprop = p + 1;
+ snprintf(name, sizeof(name), "snd-soc-%s", sprop);
+ make_lowercase(name);
+
+ ssi_private->pdev =
+ platform_device_register_data(&pdev->dev, name, 0, NULL, 0);
+ if (IS_ERR(ssi_private->pdev)) {
+ ret = PTR_ERR(ssi_private->pdev);
+ dev_err(&pdev->dev, "failed to register platform: %d\n", ret);
+ goto error_sound_card;
+ }
+
+done:
+ if (ssi_private->dai_fmt)
+ _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private,
+ ssi_private->dai_fmt);
+
+ return 0;
+
+error_sound_card:
+ fsl_ssi_debugfs_remove(&ssi_private->dbg_stats);
+
+error_asoc_register:
+ if (ssi_private->soc->imx)
+ fsl_ssi_imx_clean(pdev, ssi_private);
+
+ return ret;
+}
+
+static int fsl_ssi_remove(struct platform_device *pdev)
+{
+ struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev);
+
+ fsl_ssi_debugfs_remove(&ssi_private->dbg_stats);
+
+ if (ssi_private->pdev)
+ platform_device_unregister(ssi_private->pdev);
+
+ if (ssi_private->soc->imx)
+ fsl_ssi_imx_clean(pdev, ssi_private);
+
+ return 0;
+}
+
+static struct platform_driver fsl_ssi_driver = {
+ .driver = {
+ .name = "fsl-ssi-dai",
+ .of_match_table = fsl_ssi_ids,
+ },
+ .probe = fsl_ssi_probe,
+ .remove = fsl_ssi_remove,
+};
+
+module_platform_driver(fsl_ssi_driver);
+
+MODULE_ALIAS("platform:fsl-ssi-dai");
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
new file mode 100644
index 000000000..506510540
--- /dev/null
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -0,0 +1,268 @@
+/*
+ * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 SoC
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
+ * under the terms of the GNU General Public License version 2. This
+ * program is licensed "as is" without any warranty of any kind, whether
+ * express or implied.
+ */
+
+#ifndef _MPC8610_I2S_H
+#define _MPC8610_I2S_H
+
+/* SSI registers */
+#define CCSR_SSI_STX0 0x00
+#define CCSR_SSI_STX1 0x04
+#define CCSR_SSI_SRX0 0x08
+#define CCSR_SSI_SRX1 0x0c
+#define CCSR_SSI_SCR 0x10
+#define CCSR_SSI_SISR 0x14
+#define CCSR_SSI_SIER 0x18
+#define CCSR_SSI_STCR 0x1c
+#define CCSR_SSI_SRCR 0x20
+#define CCSR_SSI_STCCR 0x24
+#define CCSR_SSI_SRCCR 0x28
+#define CCSR_SSI_SFCSR 0x2c
+#define CCSR_SSI_STR 0x30
+#define CCSR_SSI_SOR 0x34
+#define CCSR_SSI_SACNT 0x38
+#define CCSR_SSI_SACADD 0x3c
+#define CCSR_SSI_SACDAT 0x40
+#define CCSR_SSI_SATAG 0x44
+#define CCSR_SSI_STMSK 0x48
+#define CCSR_SSI_SRMSK 0x4c
+#define CCSR_SSI_SACCST 0x50
+#define CCSR_SSI_SACCEN 0x54
+#define CCSR_SSI_SACCDIS 0x58
+
+#define CCSR_SSI_SCR_SYNC_TX_FS 0x00001000
+#define CCSR_SSI_SCR_RFR_CLK_DIS 0x00000800
+#define CCSR_SSI_SCR_TFR_CLK_DIS 0x00000400
+#define CCSR_SSI_SCR_TCH_EN 0x00000100
+#define CCSR_SSI_SCR_SYS_CLK_EN 0x00000080
+#define CCSR_SSI_SCR_I2S_MODE_MASK 0x00000060
+#define CCSR_SSI_SCR_I2S_MODE_NORMAL 0x00000000
+#define CCSR_SSI_SCR_I2S_MODE_MASTER 0x00000020
+#define CCSR_SSI_SCR_I2S_MODE_SLAVE 0x00000040
+#define CCSR_SSI_SCR_SYN 0x00000010
+#define CCSR_SSI_SCR_NET 0x00000008
+#define CCSR_SSI_SCR_RE 0x00000004
+#define CCSR_SSI_SCR_TE 0x00000002
+#define CCSR_SSI_SCR_SSIEN 0x00000001
+
+#define CCSR_SSI_SISR_RFRC 0x01000000
+#define CCSR_SSI_SISR_TFRC 0x00800000
+#define CCSR_SSI_SISR_CMDAU 0x00040000
+#define CCSR_SSI_SISR_CMDDU 0x00020000
+#define CCSR_SSI_SISR_RXT 0x00010000
+#define CCSR_SSI_SISR_RDR1 0x00008000
+#define CCSR_SSI_SISR_RDR0 0x00004000
+#define CCSR_SSI_SISR_TDE1 0x00002000
+#define CCSR_SSI_SISR_TDE0 0x00001000
+#define CCSR_SSI_SISR_ROE1 0x00000800
+#define CCSR_SSI_SISR_ROE0 0x00000400
+#define CCSR_SSI_SISR_TUE1 0x00000200
+#define CCSR_SSI_SISR_TUE0 0x00000100
+#define CCSR_SSI_SISR_TFS 0x00000080
+#define CCSR_SSI_SISR_RFS 0x00000040
+#define CCSR_SSI_SISR_TLS 0x00000020
+#define CCSR_SSI_SISR_RLS 0x00000010
+#define CCSR_SSI_SISR_RFF1 0x00000008
+#define CCSR_SSI_SISR_RFF0 0x00000004
+#define CCSR_SSI_SISR_TFE1 0x00000002
+#define CCSR_SSI_SISR_TFE0 0x00000001
+
+#define CCSR_SSI_SIER_RFRC_EN 0x01000000
+#define CCSR_SSI_SIER_TFRC_EN 0x00800000
+#define CCSR_SSI_SIER_RDMAE 0x00400000
+#define CCSR_SSI_SIER_RIE 0x00200000
+#define CCSR_SSI_SIER_TDMAE 0x00100000
+#define CCSR_SSI_SIER_TIE 0x00080000
+#define CCSR_SSI_SIER_CMDAU_EN 0x00040000
+#define CCSR_SSI_SIER_CMDDU_EN 0x00020000
+#define CCSR_SSI_SIER_RXT_EN 0x00010000
+#define CCSR_SSI_SIER_RDR1_EN 0x00008000
+#define CCSR_SSI_SIER_RDR0_EN 0x00004000
+#define CCSR_SSI_SIER_TDE1_EN 0x00002000
+#define CCSR_SSI_SIER_TDE0_EN 0x00001000
+#define CCSR_SSI_SIER_ROE1_EN 0x00000800
+#define CCSR_SSI_SIER_ROE0_EN 0x00000400
+#define CCSR_SSI_SIER_TUE1_EN 0x00000200
+#define CCSR_SSI_SIER_TUE0_EN 0x00000100
+#define CCSR_SSI_SIER_TFS_EN 0x00000080
+#define CCSR_SSI_SIER_RFS_EN 0x00000040
+#define CCSR_SSI_SIER_TLS_EN 0x00000020
+#define CCSR_SSI_SIER_RLS_EN 0x00000010
+#define CCSR_SSI_SIER_RFF1_EN 0x00000008
+#define CCSR_SSI_SIER_RFF0_EN 0x00000004
+#define CCSR_SSI_SIER_TFE1_EN 0x00000002
+#define CCSR_SSI_SIER_TFE0_EN 0x00000001
+
+#define CCSR_SSI_STCR_TXBIT0 0x00000200
+#define CCSR_SSI_STCR_TFEN1 0x00000100
+#define CCSR_SSI_STCR_TFEN0 0x00000080
+#define CCSR_SSI_STCR_TFDIR 0x00000040
+#define CCSR_SSI_STCR_TXDIR 0x00000020
+#define CCSR_SSI_STCR_TSHFD 0x00000010
+#define CCSR_SSI_STCR_TSCKP 0x00000008
+#define CCSR_SSI_STCR_TFSI 0x00000004
+#define CCSR_SSI_STCR_TFSL 0x00000002
+#define CCSR_SSI_STCR_TEFS 0x00000001
+
+#define CCSR_SSI_SRCR_RXEXT 0x00000400
+#define CCSR_SSI_SRCR_RXBIT0 0x00000200
+#define CCSR_SSI_SRCR_RFEN1 0x00000100
+#define CCSR_SSI_SRCR_RFEN0 0x00000080
+#define CCSR_SSI_SRCR_RFDIR 0x00000040
+#define CCSR_SSI_SRCR_RXDIR 0x00000020
+#define CCSR_SSI_SRCR_RSHFD 0x00000010
+#define CCSR_SSI_SRCR_RSCKP 0x00000008
+#define CCSR_SSI_SRCR_RFSI 0x00000004
+#define CCSR_SSI_SRCR_RFSL 0x00000002
+#define CCSR_SSI_SRCR_REFS 0x00000001
+
+/* STCCR and SRCCR */
+#define CCSR_SSI_SxCCR_DIV2_SHIFT 18
+#define CCSR_SSI_SxCCR_DIV2 0x00040000
+#define CCSR_SSI_SxCCR_PSR_SHIFT 17
+#define CCSR_SSI_SxCCR_PSR 0x00020000
+#define CCSR_SSI_SxCCR_WL_SHIFT 13
+#define CCSR_SSI_SxCCR_WL_MASK 0x0001E000
+#define CCSR_SSI_SxCCR_WL(x) \
+ (((((x) / 2) - 1) << CCSR_SSI_SxCCR_WL_SHIFT) & CCSR_SSI_SxCCR_WL_MASK)
+#define CCSR_SSI_SxCCR_DC_SHIFT 8
+#define CCSR_SSI_SxCCR_DC_MASK 0x00001F00
+#define CCSR_SSI_SxCCR_DC(x) \
+ ((((x) - 1) << CCSR_SSI_SxCCR_DC_SHIFT) & CCSR_SSI_SxCCR_DC_MASK)
+#define CCSR_SSI_SxCCR_PM_SHIFT 0
+#define CCSR_SSI_SxCCR_PM_MASK 0x000000FF
+#define CCSR_SSI_SxCCR_PM(x) \
+ ((((x) - 1) << CCSR_SSI_SxCCR_PM_SHIFT) & CCSR_SSI_SxCCR_PM_MASK)
+
+/*
+ * The xFCNT bits are read-only, and the xFWM bits are read/write. Use the
+ * CCSR_SSI_SFCSR_xFCNTy() macros to read the FIFO counters, and use the
+ * CCSR_SSI_SFCSR_xFWMy() macros to set the watermarks.
+ */
+#define CCSR_SSI_SFCSR_RFCNT1_SHIFT 28
+#define CCSR_SSI_SFCSR_RFCNT1_MASK 0xF0000000
+#define CCSR_SSI_SFCSR_RFCNT1(x) \
+ (((x) & CCSR_SSI_SFCSR_RFCNT1_MASK) >> CCSR_SSI_SFCSR_RFCNT1_SHIFT)
+#define CCSR_SSI_SFCSR_TFCNT1_SHIFT 24
+#define CCSR_SSI_SFCSR_TFCNT1_MASK 0x0F000000
+#define CCSR_SSI_SFCSR_TFCNT1(x) \
+ (((x) & CCSR_SSI_SFCSR_TFCNT1_MASK) >> CCSR_SSI_SFCSR_TFCNT1_SHIFT)
+#define CCSR_SSI_SFCSR_RFWM1_SHIFT 20
+#define CCSR_SSI_SFCSR_RFWM1_MASK 0x00F00000
+#define CCSR_SSI_SFCSR_RFWM1(x) \
+ (((x) << CCSR_SSI_SFCSR_RFWM1_SHIFT) & CCSR_SSI_SFCSR_RFWM1_MASK)
+#define CCSR_SSI_SFCSR_TFWM1_SHIFT 16
+#define CCSR_SSI_SFCSR_TFWM1_MASK 0x000F0000
+#define CCSR_SSI_SFCSR_TFWM1(x) \
+ (((x) << CCSR_SSI_SFCSR_TFWM1_SHIFT) & CCSR_SSI_SFCSR_TFWM1_MASK)
+#define CCSR_SSI_SFCSR_RFCNT0_SHIFT 12
+#define CCSR_SSI_SFCSR_RFCNT0_MASK 0x0000F000
+#define CCSR_SSI_SFCSR_RFCNT0(x) \
+ (((x) & CCSR_SSI_SFCSR_RFCNT0_MASK) >> CCSR_SSI_SFCSR_RFCNT0_SHIFT)
+#define CCSR_SSI_SFCSR_TFCNT0_SHIFT 8
+#define CCSR_SSI_SFCSR_TFCNT0_MASK 0x00000F00
+#define CCSR_SSI_SFCSR_TFCNT0(x) \
+ (((x) & CCSR_SSI_SFCSR_TFCNT0_MASK) >> CCSR_SSI_SFCSR_TFCNT0_SHIFT)
+#define CCSR_SSI_SFCSR_RFWM0_SHIFT 4
+#define CCSR_SSI_SFCSR_RFWM0_MASK 0x000000F0
+#define CCSR_SSI_SFCSR_RFWM0(x) \
+ (((x) << CCSR_SSI_SFCSR_RFWM0_SHIFT) & CCSR_SSI_SFCSR_RFWM0_MASK)
+#define CCSR_SSI_SFCSR_TFWM0_SHIFT 0
+#define CCSR_SSI_SFCSR_TFWM0_MASK 0x0000000F
+#define CCSR_SSI_SFCSR_TFWM0(x) \
+ (((x) << CCSR_SSI_SFCSR_TFWM0_SHIFT) & CCSR_SSI_SFCSR_TFWM0_MASK)
+
+#define CCSR_SSI_STR_TEST 0x00008000
+#define CCSR_SSI_STR_RCK2TCK 0x00004000
+#define CCSR_SSI_STR_RFS2TFS 0x00002000
+#define CCSR_SSI_STR_RXSTATE(x) (((x) >> 8) & 0x1F)
+#define CCSR_SSI_STR_TXD2RXD 0x00000080
+#define CCSR_SSI_STR_TCK2RCK 0x00000040
+#define CCSR_SSI_STR_TFS2RFS 0x00000020
+#define CCSR_SSI_STR_TXSTATE(x) ((x) & 0x1F)
+
+#define CCSR_SSI_SOR_CLKOFF 0x00000040
+#define CCSR_SSI_SOR_RX_CLR 0x00000020
+#define CCSR_SSI_SOR_TX_CLR 0x00000010
+#define CCSR_SSI_SOR_INIT 0x00000008
+#define CCSR_SSI_SOR_WAIT_SHIFT 1
+#define CCSR_SSI_SOR_WAIT_MASK 0x00000006
+#define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT)
+#define CCSR_SSI_SOR_SYNRST 0x00000001
+
+#define CCSR_SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5)
+#define CCSR_SSI_SACNT_WR 0x00000010
+#define CCSR_SSI_SACNT_RD 0x00000008
+#define CCSR_SSI_SACNT_RDWR_MASK 0x00000018
+#define CCSR_SSI_SACNT_TIF 0x00000004
+#define CCSR_SSI_SACNT_FV 0x00000002
+#define CCSR_SSI_SACNT_AC97EN 0x00000001
+
+
+struct device;
+
+#if IS_ENABLED(CONFIG_DEBUG_FS)
+
+struct fsl_ssi_dbg {
+ struct dentry *dbg_dir;
+ struct dentry *dbg_stats;
+
+ struct {
+ unsigned int rfrc;
+ unsigned int tfrc;
+ unsigned int cmdau;
+ unsigned int cmddu;
+ unsigned int rxt;
+ unsigned int rdr1;
+ unsigned int rdr0;
+ unsigned int tde1;
+ unsigned int tde0;
+ unsigned int roe1;
+ unsigned int roe0;
+ unsigned int tue1;
+ unsigned int tue0;
+ unsigned int tfs;
+ unsigned int rfs;
+ unsigned int tls;
+ unsigned int rls;
+ unsigned int rff1;
+ unsigned int rff0;
+ unsigned int tfe1;
+ unsigned int tfe0;
+ } stats;
+};
+
+void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *ssi_dbg, u32 sisr);
+
+int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev);
+
+void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg);
+
+#else
+
+struct fsl_ssi_dbg {
+};
+
+static inline void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *stats, u32 sisr)
+{
+}
+
+static inline int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg,
+ struct device *dev)
+{
+ return 0;
+}
+
+static inline void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg)
+{
+}
+#endif /* ! IS_ENABLED(CONFIG_DEBUG_FS) */
+
+#endif
diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c
new file mode 100644
index 000000000..5469ffbc0
--- /dev/null
+++ b/sound/soc/fsl/fsl_ssi_dbg.c
@@ -0,0 +1,163 @@
+/*
+ * Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions
+ *
+ * Copyright 2014 Markus Pargmann <mpa@pengutronix.de>, Pengutronix
+ *
+ * Splitted from fsl_ssi.c
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/debugfs.h>
+#include <linux/device.h>
+#include <linux/kernel.h>
+
+#include "fsl_ssi.h"
+
+void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *dbg, u32 sisr)
+{
+ if (sisr & CCSR_SSI_SISR_RFRC)
+ dbg->stats.rfrc++;
+
+ if (sisr & CCSR_SSI_SISR_TFRC)
+ dbg->stats.tfrc++;
+
+ if (sisr & CCSR_SSI_SISR_CMDAU)
+ dbg->stats.cmdau++;
+
+ if (sisr & CCSR_SSI_SISR_CMDDU)
+ dbg->stats.cmddu++;
+
+ if (sisr & CCSR_SSI_SISR_RXT)
+ dbg->stats.rxt++;
+
+ if (sisr & CCSR_SSI_SISR_RDR1)
+ dbg->stats.rdr1++;
+
+ if (sisr & CCSR_SSI_SISR_RDR0)
+ dbg->stats.rdr0++;
+
+ if (sisr & CCSR_SSI_SISR_TDE1)
+ dbg->stats.tde1++;
+
+ if (sisr & CCSR_SSI_SISR_TDE0)
+ dbg->stats.tde0++;
+
+ if (sisr & CCSR_SSI_SISR_ROE1)
+ dbg->stats.roe1++;
+
+ if (sisr & CCSR_SSI_SISR_ROE0)
+ dbg->stats.roe0++;
+
+ if (sisr & CCSR_SSI_SISR_TUE1)
+ dbg->stats.tue1++;
+
+ if (sisr & CCSR_SSI_SISR_TUE0)
+ dbg->stats.tue0++;
+
+ if (sisr & CCSR_SSI_SISR_TFS)
+ dbg->stats.tfs++;
+
+ if (sisr & CCSR_SSI_SISR_RFS)
+ dbg->stats.rfs++;
+
+ if (sisr & CCSR_SSI_SISR_TLS)
+ dbg->stats.tls++;
+
+ if (sisr & CCSR_SSI_SISR_RLS)
+ dbg->stats.rls++;
+
+ if (sisr & CCSR_SSI_SISR_RFF1)
+ dbg->stats.rff1++;
+
+ if (sisr & CCSR_SSI_SISR_RFF0)
+ dbg->stats.rff0++;
+
+ if (sisr & CCSR_SSI_SISR_TFE1)
+ dbg->stats.tfe1++;
+
+ if (sisr & CCSR_SSI_SISR_TFE0)
+ dbg->stats.tfe0++;
+}
+
+/* Show the statistics of a flag only if its interrupt is enabled. The
+ * compiler will optimze this code to a no-op if the interrupt is not
+ * enabled.
+ */
+#define SIER_SHOW(flag, name) \
+ do { \
+ if (CCSR_SSI_SIER_##flag) \
+ seq_printf(s, #name "=%u\n", ssi_dbg->stats.name); \
+ } while (0)
+
+
+/**
+ * fsl_sysfs_ssi_show: display SSI statistics
+ *
+ * Display the statistics for the current SSI device. To avoid confusion,
+ * we only show those counts that are enabled.
+ */
+static int fsl_ssi_stats_show(struct seq_file *s, void *unused)
+{
+ struct fsl_ssi_dbg *ssi_dbg = s->private;
+
+ SIER_SHOW(RFRC_EN, rfrc);
+ SIER_SHOW(TFRC_EN, tfrc);
+ SIER_SHOW(CMDAU_EN, cmdau);
+ SIER_SHOW(CMDDU_EN, cmddu);
+ SIER_SHOW(RXT_EN, rxt);
+ SIER_SHOW(RDR1_EN, rdr1);
+ SIER_SHOW(RDR0_EN, rdr0);
+ SIER_SHOW(TDE1_EN, tde1);
+ SIER_SHOW(TDE0_EN, tde0);
+ SIER_SHOW(ROE1_EN, roe1);
+ SIER_SHOW(ROE0_EN, roe0);
+ SIER_SHOW(TUE1_EN, tue1);
+ SIER_SHOW(TUE0_EN, tue0);
+ SIER_SHOW(TFS_EN, tfs);
+ SIER_SHOW(RFS_EN, rfs);
+ SIER_SHOW(TLS_EN, tls);
+ SIER_SHOW(RLS_EN, rls);
+ SIER_SHOW(RFF1_EN, rff1);
+ SIER_SHOW(RFF0_EN, rff0);
+ SIER_SHOW(TFE1_EN, tfe1);
+ SIER_SHOW(TFE0_EN, tfe0);
+
+ return 0;
+}
+
+static int fsl_ssi_stats_open(struct inode *inode, struct file *file)
+{
+ return single_open(file, fsl_ssi_stats_show, inode->i_private);
+}
+
+static const struct file_operations fsl_ssi_stats_ops = {
+ .open = fsl_ssi_stats_open,
+ .read = seq_read,
+ .llseek = seq_lseek,
+ .release = single_release,
+};
+
+int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev)
+{
+ ssi_dbg->dbg_dir = debugfs_create_dir(dev_name(dev), NULL);
+ if (!ssi_dbg->dbg_dir)
+ return -ENOMEM;
+
+ ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO,
+ ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops);
+ if (!ssi_dbg->dbg_stats) {
+ debugfs_remove(ssi_dbg->dbg_dir);
+ return -ENOMEM;
+ }
+
+ return 0;
+}
+
+void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg)
+{
+ debugfs_remove(ssi_dbg->dbg_stats);
+ debugfs_remove(ssi_dbg->dbg_dir);
+}
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
new file mode 100644
index 000000000..b9e42b503
--- /dev/null
+++ b/sound/soc/fsl/fsl_utils.c
@@ -0,0 +1,91 @@
+/**
+ * Freescale ALSA SoC Machine driver utility
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/of_address.h>
+#include <sound/soc.h>
+
+#include "fsl_utils.h"
+
+/**
+ * fsl_asoc_get_dma_channel - determine the dma channel for a SSI node
+ *
+ * @ssi_np: pointer to the SSI device tree node
+ * @name: name of the phandle pointing to the dma channel
+ * @dai: ASoC DAI link pointer to be filled with platform_name
+ * @dma_channel_id: dma channel id to be returned
+ * @dma_id: dma id to be returned
+ *
+ * This function determines the dma and channel id for given SSI node. It
+ * also discovers the platform_name for the ASoC DAI link.
+ */
+int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
+ const char *name,
+ struct snd_soc_dai_link *dai,
+ unsigned int *dma_channel_id,
+ unsigned int *dma_id)
+{
+ struct resource res;
+ struct device_node *dma_channel_np, *dma_np;
+ const u32 *iprop;
+ int ret;
+
+ dma_channel_np = of_parse_phandle(ssi_np, name, 0);
+ if (!dma_channel_np)
+ return -EINVAL;
+
+ if (!of_device_is_compatible(dma_channel_np, "fsl,ssi-dma-channel")) {
+ of_node_put(dma_channel_np);
+ return -EINVAL;
+ }
+
+ /* Determine the dev_name for the device_node. This code mimics the
+ * behavior of of_device_make_bus_id(). We need this because ASoC uses
+ * the dev_name() of the device to match the platform (DMA) device with
+ * the CPU (SSI) device. It's all ugly and hackish, but it works (for
+ * now).
+ *
+ * dai->platform name should already point to an allocated buffer.
+ */
+ ret = of_address_to_resource(dma_channel_np, 0, &res);
+ if (ret) {
+ of_node_put(dma_channel_np);
+ return ret;
+ }
+ snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
+ (unsigned long long) res.start, dma_channel_np->name);
+
+ iprop = of_get_property(dma_channel_np, "cell-index", NULL);
+ if (!iprop) {
+ of_node_put(dma_channel_np);
+ return -EINVAL;
+ }
+ *dma_channel_id = be32_to_cpup(iprop);
+
+ dma_np = of_get_parent(dma_channel_np);
+ iprop = of_get_property(dma_np, "cell-index", NULL);
+ if (!iprop) {
+ of_node_put(dma_np);
+ return -EINVAL;
+ }
+ *dma_id = be32_to_cpup(iprop);
+
+ of_node_put(dma_np);
+ of_node_put(dma_channel_np);
+
+ return 0;
+}
+EXPORT_SYMBOL(fsl_asoc_get_dma_channel);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale ASoC utility code");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h
new file mode 100644
index 000000000..1687b66ef
--- /dev/null
+++ b/sound/soc/fsl/fsl_utils.h
@@ -0,0 +1,25 @@
+/**
+ * Freescale ALSA SoC Machine driver utility
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _FSL_UTILS_H
+#define _FSL_UTILS_H
+
+#define DAI_NAME_SIZE 32
+
+struct snd_soc_dai_link;
+struct device_node;
+
+int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name,
+ struct snd_soc_dai_link *dai,
+ unsigned int *dma_channel_id,
+ unsigned int *dma_id);
+#endif /* _FSL_UTILS_H */
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
new file mode 100644
index 000000000..d9050d946
--- /dev/null
+++ b/sound/soc/fsl/imx-audmux.c
@@ -0,0 +1,378 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ * Copyright 2009 Pengutronix, Sascha Hauer <s.hauer@pengutronix.de>
+ *
+ * Initial development of this code was funded by
+ * Phytec Messtechnik GmbH, http://www.phytec.de
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/debugfs.h>
+#include <linux/err.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include "imx-audmux.h"
+
+#define DRIVER_NAME "imx-audmux"
+
+static struct clk *audmux_clk;
+static void __iomem *audmux_base;
+
+#define IMX_AUDMUX_V2_PTCR(x) ((x) * 8)
+#define IMX_AUDMUX_V2_PDCR(x) ((x) * 8 + 4)
+
+#ifdef CONFIG_DEBUG_FS
+static struct dentry *audmux_debugfs_root;
+
+/* There is an annoying discontinuity in the SSI numbering with regard
+ * to the Linux number of the devices */
+static const char *audmux_port_string(int port)
+{
+ switch (port) {
+ case MX31_AUDMUX_PORT1_SSI0:
+ return "imx-ssi.0";
+ case MX31_AUDMUX_PORT2_SSI1:
+ return "imx-ssi.1";
+ case MX31_AUDMUX_PORT3_SSI_PINS_3:
+ return "SSI3";
+ case MX31_AUDMUX_PORT4_SSI_PINS_4:
+ return "SSI4";
+ case MX31_AUDMUX_PORT5_SSI_PINS_5:
+ return "SSI5";
+ case MX31_AUDMUX_PORT6_SSI_PINS_6:
+ return "SSI6";
+ default:
+ return "UNKNOWN";
+ }
+}
+
+static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
+ size_t count, loff_t *ppos)
+{
+ ssize_t ret;
+ char *buf;
+ uintptr_t port = (uintptr_t)file->private_data;
+ u32 pdcr, ptcr;
+
+ if (audmux_clk) {
+ ret = clk_prepare_enable(audmux_clk);
+ if (ret)
+ return ret;
+ }
+
+ ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port));
+ pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port));
+
+ if (audmux_clk)
+ clk_disable_unprepare(audmux_clk);
+
+ buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
+ pdcr, ptcr);
+
+ if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "TxFS output from %s, ",
+ audmux_port_string((ptcr >> 27) & 0x7));
+ else
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "TxFS input, ");
+
+ if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "TxClk output from %s",
+ audmux_port_string((ptcr >> 22) & 0x7));
+ else
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "TxClk input");
+
+ ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+
+ if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) {
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "Port is symmetric");
+ } else {
+ if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "RxFS output from %s, ",
+ audmux_port_string((ptcr >> 17) & 0x7));
+ else
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "RxFS input, ");
+
+ if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "RxClk output from %s",
+ audmux_port_string((ptcr >> 12) & 0x7));
+ else
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "RxClk input");
+ }
+
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ "\nData received from %s\n",
+ audmux_port_string((pdcr >> 13) & 0x7));
+
+ ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+
+ kfree(buf);
+
+ return ret;
+}
+
+static const struct file_operations audmux_debugfs_fops = {
+ .open = simple_open,
+ .read = audmux_read_file,
+ .llseek = default_llseek,
+};
+
+static void audmux_debugfs_init(void)
+{
+ uintptr_t i;
+ char buf[20];
+
+ audmux_debugfs_root = debugfs_create_dir("audmux", NULL);
+ if (!audmux_debugfs_root) {
+ pr_warning("Failed to create AUDMUX debugfs root\n");
+ return;
+ }
+
+ for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) {
+ snprintf(buf, sizeof(buf), "ssi%lu", i);
+ if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
+ (void *)i, &audmux_debugfs_fops))
+ pr_warning("Failed to create AUDMUX port %lu debugfs file\n",
+ i);
+ }
+}
+
+static void audmux_debugfs_remove(void)
+{
+ debugfs_remove_recursive(audmux_debugfs_root);
+}
+#else
+static inline void audmux_debugfs_init(void)
+{
+}
+
+static inline void audmux_debugfs_remove(void)
+{
+}
+#endif
+
+static enum imx_audmux_type {
+ IMX21_AUDMUX,
+ IMX31_AUDMUX,
+} audmux_type;
+
+static struct platform_device_id imx_audmux_ids[] = {
+ {
+ .name = "imx21-audmux",
+ .driver_data = IMX21_AUDMUX,
+ }, {
+ .name = "imx31-audmux",
+ .driver_data = IMX31_AUDMUX,
+ }, {
+ /* sentinel */
+ }
+};
+MODULE_DEVICE_TABLE(platform, imx_audmux_ids);
+
+static const struct of_device_id imx_audmux_dt_ids[] = {
+ { .compatible = "fsl,imx21-audmux", .data = &imx_audmux_ids[0], },
+ { .compatible = "fsl,imx31-audmux", .data = &imx_audmux_ids[1], },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_audmux_dt_ids);
+
+static const uint8_t port_mapping[] = {
+ 0x0, 0x4, 0x8, 0x10, 0x14, 0x1c,
+};
+
+int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr)
+{
+ if (audmux_type != IMX21_AUDMUX)
+ return -EINVAL;
+
+ if (!audmux_base)
+ return -ENOSYS;
+
+ if (port >= ARRAY_SIZE(port_mapping))
+ return -EINVAL;
+
+ writel(pcr, audmux_base + port_mapping[port]);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port);
+
+int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr,
+ unsigned int pdcr)
+{
+ int ret;
+
+ if (audmux_type != IMX31_AUDMUX)
+ return -EINVAL;
+
+ if (!audmux_base)
+ return -ENOSYS;
+
+ if (audmux_clk) {
+ ret = clk_prepare_enable(audmux_clk);
+ if (ret)
+ return ret;
+ }
+
+ writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port));
+ writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port));
+
+ if (audmux_clk)
+ clk_disable_unprepare(audmux_clk);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port);
+
+static int imx_audmux_parse_dt_defaults(struct platform_device *pdev,
+ struct device_node *of_node)
+{
+ struct device_node *child;
+
+ for_each_available_child_of_node(of_node, child) {
+ unsigned int port;
+ unsigned int ptcr = 0;
+ unsigned int pdcr = 0;
+ unsigned int pcr = 0;
+ unsigned int val;
+ int ret;
+ int i = 0;
+
+ ret = of_property_read_u32(child, "fsl,audmux-port", &port);
+ if (ret) {
+ dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n",
+ child->full_name);
+ continue;
+ }
+ if (!of_property_read_bool(child, "fsl,port-config")) {
+ dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n",
+ child->full_name);
+ continue;
+ }
+
+ for (i = 0; (ret = of_property_read_u32_index(child,
+ "fsl,port-config", i, &val)) == 0;
+ ++i) {
+ if (audmux_type == IMX31_AUDMUX) {
+ if (i % 2)
+ pdcr |= val;
+ else
+ ptcr |= val;
+ } else {
+ pcr |= val;
+ }
+ }
+
+ if (ret != -EOVERFLOW) {
+ dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n",
+ i, child->full_name);
+ continue;
+ }
+
+ if (audmux_type == IMX31_AUDMUX) {
+ if (i % 2) {
+ dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n",
+ child->full_name);
+ continue;
+ }
+ imx_audmux_v2_configure_port(port, ptcr, pdcr);
+ } else {
+ imx_audmux_v1_configure_port(port, pcr);
+ }
+ }
+
+ return 0;
+}
+
+static int imx_audmux_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ const struct of_device_id *of_id =
+ of_match_device(imx_audmux_dt_ids, &pdev->dev);
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ audmux_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(audmux_base))
+ return PTR_ERR(audmux_base);
+
+ audmux_clk = devm_clk_get(&pdev->dev, "audmux");
+ if (IS_ERR(audmux_clk)) {
+ dev_dbg(&pdev->dev, "cannot get clock: %ld\n",
+ PTR_ERR(audmux_clk));
+ audmux_clk = NULL;
+ }
+
+ if (of_id)
+ pdev->id_entry = of_id->data;
+ audmux_type = pdev->id_entry->driver_data;
+ if (audmux_type == IMX31_AUDMUX)
+ audmux_debugfs_init();
+
+ if (of_id)
+ imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
+
+ return 0;
+}
+
+static int imx_audmux_remove(struct platform_device *pdev)
+{
+ if (audmux_type == IMX31_AUDMUX)
+ audmux_debugfs_remove();
+
+ return 0;
+}
+
+static struct platform_driver imx_audmux_driver = {
+ .probe = imx_audmux_probe,
+ .remove = imx_audmux_remove,
+ .id_table = imx_audmux_ids,
+ .driver = {
+ .name = DRIVER_NAME,
+ .of_match_table = imx_audmux_dt_ids,
+ }
+};
+
+static int __init imx_audmux_init(void)
+{
+ return platform_driver_register(&imx_audmux_driver);
+}
+subsys_initcall(imx_audmux_init);
+
+static void __exit imx_audmux_exit(void)
+{
+ platform_driver_unregister(&imx_audmux_driver);
+}
+module_exit(imx_audmux_exit);
+
+MODULE_DESCRIPTION("Freescale i.MX AUDMUX driver");
+MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRIVER_NAME);
diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h
new file mode 100644
index 000000000..38a4209af
--- /dev/null
+++ b/sound/soc/fsl/imx-audmux.h
@@ -0,0 +1,11 @@
+#ifndef __IMX_AUDMUX_H
+#define __IMX_AUDMUX_H
+
+#include <dt-bindings/sound/fsl-imx-audmux.h>
+
+int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr);
+
+int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr,
+ unsigned int pdcr);
+
+#endif /* __IMX_AUDMUX_H */
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
new file mode 100644
index 000000000..20e7400e2
--- /dev/null
+++ b/sound/soc/fsl/imx-es8328.c
@@ -0,0 +1,233 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/of_gpio.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE 32
+#define MUX_PORT_MAX 7
+
+struct imx_es8328_data {
+ struct device *dev;
+ struct snd_soc_dai_link dai;
+ struct snd_soc_card card;
+ char codec_dai_name[DAI_NAME_SIZE];
+ char platform_name[DAI_NAME_SIZE];
+ int jack_gpio;
+};
+
+static struct snd_soc_jack_gpio headset_jack_gpios[] = {
+ {
+ .gpio = -1,
+ .name = "headset-gpio",
+ .report = SND_JACK_HEADSET,
+ .invert = 0,
+ .debounce_time = 200,
+ },
+};
+
+static struct snd_soc_jack headset_jack;
+
+static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct imx_es8328_data *data = container_of(rtd->card,
+ struct imx_es8328_data, card);
+ int ret = 0;
+
+ /* Headphone jack detection */
+ if (gpio_is_valid(data->jack_gpio)) {
+ ret = snd_soc_card_jack_new(rtd->card, "Headphone",
+ SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+ &headset_jack, NULL, 0);
+ if (ret)
+ return ret;
+
+ headset_jack_gpios[0].gpio = data->jack_gpio;
+ ret = snd_soc_jack_add_gpios(&headset_jack,
+ ARRAY_SIZE(headset_jack_gpios),
+ headset_jack_gpios);
+ }
+
+ return ret;
+}
+
+static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0),
+};
+
+static int imx_es8328_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np = NULL, *codec_np = NULL;
+ struct platform_device *ssi_pdev;
+ struct imx_es8328_data *data;
+ u32 int_port, ext_port;
+ int ret;
+ struct device *dev = &pdev->dev;
+
+ ret = of_property_read_u32(np, "mux-int-port", &int_port);
+ if (ret) {
+ dev_err(dev, "mux-int-port missing or invalid\n");
+ goto fail;
+ }
+ if (int_port > MUX_PORT_MAX || int_port == 0) {
+ dev_err(dev, "mux-int-port: hardware only has %d mux ports\n",
+ MUX_PORT_MAX);
+ goto fail;
+ }
+
+ ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(dev, "mux-ext-port missing or invalid\n");
+ goto fail;
+ }
+ if (ext_port > MUX_PORT_MAX || ext_port == 0) {
+ dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n",
+ MUX_PORT_MAX);
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the audmux API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+ ret = imx_audmux_v2_configure_port(int_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+ if (ret) {
+ dev_err(dev, "audmux internal port setup failed\n");
+ return ret;
+ }
+ ret = imx_audmux_v2_configure_port(ext_port,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(dev, "audmux external port setup failed\n");
+ return ret;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+ codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+ if (!ssi_np || !codec_np) {
+ dev_err(dev, "phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ssi_pdev = of_find_device_by_node(ssi_np);
+ if (!ssi_pdev) {
+ dev_err(dev, "failed to find SSI platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ data->dev = dev;
+
+ data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0);
+
+ data->dai.name = "hifi";
+ data->dai.stream_name = "hifi";
+ data->dai.codec_dai_name = "es8328-hifi-analog";
+ data->dai.codec_of_node = codec_np;
+ data->dai.cpu_of_node = ssi_np;
+ data->dai.platform_of_node = ssi_np;
+ data->dai.init = &imx_es8328_dai_init;
+ data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ data->card.dev = dev;
+ data->card.dapm_widgets = imx_es8328_dapm_widgets;
+ data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets);
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret) {
+ dev_err(dev, "Unable to parse card name\n");
+ goto fail;
+ }
+ ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+ if (ret) {
+ dev_err(dev, "Unable to parse routing: %d\n", ret);
+ goto fail;
+ }
+ data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
+ data->card.dai_link = &data->dai;
+
+ ret = snd_soc_register_card(&data->card);
+ if (ret) {
+ dev_err(dev, "Unable to register: %d\n", ret);
+ goto fail;
+ }
+
+ platform_set_drvdata(pdev, data);
+fail:
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
+
+ return ret;
+}
+
+static int imx_es8328_remove(struct platform_device *pdev)
+{
+ struct imx_es8328_data *data = platform_get_drvdata(pdev);
+
+ snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios),
+ headset_jack_gpios);
+
+ snd_soc_unregister_card(&data->card);
+
+ return 0;
+}
+
+static const struct of_device_id imx_es8328_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-es8328", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids);
+
+static struct platform_driver imx_es8328_driver = {
+ .driver = {
+ .name = "imx-es8328",
+ .of_match_table = imx_es8328_dt_ids,
+ },
+ .probe = imx_es8328_probe,
+ .remove = imx_es8328_remove,
+};
+module_platform_driver(imx_es8328_driver);
+
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-audio-es8328");
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
new file mode 100644
index 000000000..9e6493d4e
--- /dev/null
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -0,0 +1,172 @@
+/*
+ * imx-mc13783.c -- SoC audio for imx based boards with mc13783 codec
+ *
+ * Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch>
+ *
+ * Heavly based on phycore-mc13783:
+ * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/mc13783.h"
+#include "imx-ssi.h"
+#include "imx-audmux.h"
+
+#define FMT_SSI (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBM_CFM)
+
+static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 4, 16);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, MC13783_CLK_CLIA, 26000000, 0);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16);
+ if (ret)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops imx_mc13783_hifi_ops = {
+ .hw_params = imx_mc13783_hifi_hw_params,
+};
+
+static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = {
+ {
+ .name = "MC13783",
+ .stream_name = "Sound",
+ .codec_dai_name = "mc13783-hifi",
+ .codec_name = "mc13783-codec",
+ .cpu_dai_name = "imx-ssi.0",
+ .platform_name = "imx-ssi.0",
+ .ops = &imx_mc13783_hifi_ops,
+ .symmetric_rates = 1,
+ .dai_fmt = FMT_SSI,
+ },
+};
+
+static const struct snd_soc_dapm_widget imx_mc13783_widget[] = {
+ SND_SOC_DAPM_MIC("Mic", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route imx_mc13783_routes[] = {
+ {"Speaker", NULL, "LSP"},
+ {"Headphone", NULL, "HSL"},
+ {"Headphone", NULL, "HSR"},
+
+ {"MC1LIN", NULL, "MC1 Bias"},
+ {"MC2IN", NULL, "MC2 Bias"},
+ {"MC1 Bias", NULL, "Mic"},
+ {"MC2 Bias", NULL, "Mic"},
+};
+
+static struct snd_soc_card imx_mc13783 = {
+ .name = "imx_mc13783",
+ .owner = THIS_MODULE,
+ .dai_link = imx_mc13783_dai_mc13783,
+ .num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783),
+ .dapm_widgets = imx_mc13783_widget,
+ .num_dapm_widgets = ARRAY_SIZE(imx_mc13783_widget),
+ .dapm_routes = imx_mc13783_routes,
+ .num_dapm_routes = ARRAY_SIZE(imx_mc13783_routes),
+};
+
+static int imx_mc13783_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ imx_mc13783.dev = &pdev->dev;
+
+ ret = snd_soc_register_card(&imx_mc13783);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+ ret);
+ return ret;
+ }
+
+ if (machine_is_mx31_3ds() || machine_is_mx31moboard()) {
+ imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
+ IMX_AUDMUX_V2_PDCR_MODE(1) |
+ IMX_AUDMUX_V2_PDCR_INMMASK(0xfc));
+ imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR |
+ IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+ IMX_AUDMUX_V2_PTCR_RFSDIR |
+ IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+ IMX_AUDMUX_V2_PTCR_RCLKDIR |
+ IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4),
+ IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4));
+ } else if (machine_is_mx27_3ds()) {
+ imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+ IMX_AUDMUX_V1_PCR_SYN |
+ IMX_AUDMUX_V1_PCR_TFSDIR |
+ IMX_AUDMUX_V1_PCR_TCLKDIR |
+ IMX_AUDMUX_V1_PCR_RFSDIR |
+ IMX_AUDMUX_V1_PCR_RCLKDIR |
+ IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+ IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+ IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4)
+ );
+ imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4,
+ IMX_AUDMUX_V1_PCR_SYN |
+ IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+ );
+ }
+
+ return ret;
+}
+
+static int imx_mc13783_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&imx_mc13783);
+
+ return 0;
+}
+
+static struct platform_driver imx_mc13783_audio_driver = {
+ .driver = {
+ .name = "imx_mc13783",
+ },
+ .probe = imx_mc13783_probe,
+ .remove = imx_mc13783_remove
+};
+
+module_platform_driver(imx_mc13783_audio_driver);
+
+MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>");
+MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch");
+MODULE_DESCRIPTION("imx with mc13783 codec ALSA SoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx_mc13783");
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
new file mode 100644
index 000000000..0db94f492
--- /dev/null
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -0,0 +1,66 @@
+/*
+ * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
+ *
+ * This code is based on code copyrighted by Freescale,
+ * Liam Girdwood, Javier Martin and probably others.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+#include <linux/platform_device.h>
+#include <linux/dmaengine.h>
+#include <linux/types.h>
+#include <linux/module.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "imx-pcm.h"
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ if (!imx_dma_is_general_purpose(chan))
+ return false;
+
+ chan->private = param;
+
+ return true;
+}
+
+static const struct snd_pcm_hardware imx_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
+ .period_bytes_min = 128,
+ .period_bytes_max = 65535, /* Limited by SDMA engine */
+ .periods_min = 2,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = {
+ .pcm_hardware = &imx_pcm_hardware,
+ .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+ .compat_filter_fn = filter,
+ .prealloc_buffer_size = IMX_SSI_DMABUF_SIZE,
+};
+
+int imx_pcm_dma_init(struct platform_device *pdev)
+{
+ return devm_snd_dmaengine_pcm_register(&pdev->dev,
+ &imx_dmaengine_pcm_config,
+ SND_DMAENGINE_PCM_FLAG_COMPAT);
+}
+EXPORT_SYMBOL_GPL(imx_pcm_dma_init);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
new file mode 100644
index 000000000..7abf6a079
--- /dev/null
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -0,0 +1,393 @@
+/*
+ * imx-pcm-fiq.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
+ *
+ * This code is based on code copyrighted by Freescale,
+ * Liam Girdwood, Javier Martin and probably others.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/fiq.h>
+
+#include <linux/platform_data/asoc-imx-ssi.h>
+
+#include "imx-ssi.h"
+#include "imx-pcm.h"
+
+struct imx_pcm_runtime_data {
+ unsigned int period;
+ int periods;
+ unsigned long offset;
+ struct hrtimer hrt;
+ int poll_time_ns;
+ struct snd_pcm_substream *substream;
+ atomic_t playing;
+ atomic_t capturing;
+};
+
+static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
+{
+ struct imx_pcm_runtime_data *iprtd =
+ container_of(hrt, struct imx_pcm_runtime_data, hrt);
+ struct snd_pcm_substream *substream = iprtd->substream;
+ struct pt_regs regs;
+
+ if (!atomic_read(&iprtd->playing) && !atomic_read(&iprtd->capturing))
+ return HRTIMER_NORESTART;
+
+ get_fiq_regs(&regs);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ iprtd->offset = regs.ARM_r8 & 0xffff;
+ else
+ iprtd->offset = regs.ARM_r9 & 0xffff;
+
+ snd_pcm_period_elapsed(substream);
+
+ hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns));
+
+ return HRTIMER_RESTART;
+}
+
+static struct fiq_handler fh = {
+ .name = DRV_NAME,
+};
+
+static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+
+ iprtd->periods = params_periods(params);
+ iprtd->period = params_period_bytes(params);
+ iprtd->offset = 0;
+ iprtd->poll_time_ns = 1000000000 / params_rate(params) *
+ params_period_size(params);
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+ struct pt_regs regs;
+
+ get_fiq_regs(&regs);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ regs.ARM_r8 = (iprtd->period * iprtd->periods - 1) << 16;
+ else
+ regs.ARM_r9 = (iprtd->period * iprtd->periods - 1) << 16;
+
+ set_fiq_regs(&regs);
+
+ return 0;
+}
+
+static int imx_pcm_fiq;
+
+static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ atomic_set(&iprtd->playing, 1);
+ else
+ atomic_set(&iprtd->capturing, 1);
+ hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns),
+ HRTIMER_MODE_REL);
+ enable_fiq(imx_pcm_fiq);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ atomic_set(&iprtd->playing, 0);
+ else
+ atomic_set(&iprtd->capturing, 0);
+ if (!atomic_read(&iprtd->playing) &&
+ !atomic_read(&iprtd->capturing))
+ disable_fiq(imx_pcm_fiq);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+
+ return bytes_to_frames(substream->runtime, iprtd->offset);
+}
+
+static struct snd_pcm_hardware snd_imx_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
+ .period_bytes_min = 128,
+ .period_bytes_max = 16 * 1024,
+ .periods_min = 4,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static int snd_imx_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd;
+ int ret;
+
+ iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
+ if (iprtd == NULL)
+ return -ENOMEM;
+ runtime->private_data = iprtd;
+
+ iprtd->substream = substream;
+
+ atomic_set(&iprtd->playing, 0);
+ atomic_set(&iprtd->capturing, 0);
+ hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+ iprtd->hrt.function = snd_hrtimer_callback;
+
+ ret = snd_pcm_hw_constraint_integer(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ kfree(iprtd);
+ return ret;
+ }
+
+ snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
+ return 0;
+}
+
+static int snd_imx_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+
+ hrtimer_cancel(&iprtd->hrt);
+
+ kfree(iprtd);
+
+ return 0;
+}
+
+static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+
+ ret = dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area, runtime->dma_addr, runtime->dma_bytes);
+
+ pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+ return ret;
+}
+
+static struct snd_pcm_ops imx_pcm_ops = {
+ .open = snd_imx_open,
+ .close = snd_imx_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_imx_pcm_hw_params,
+ .prepare = snd_imx_pcm_prepare,
+ .trigger = snd_imx_pcm_trigger,
+ .pointer = snd_imx_pcm_pointer,
+ .mmap = snd_imx_pcm_mmap,
+};
+
+static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = IMX_SSI_DMABUF_SIZE;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+
+ return 0;
+}
+
+static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret;
+
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = imx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ return ret;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = imx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+static int ssi_irq = 0;
+
+static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ struct snd_pcm_substream *substream;
+ int ret;
+
+ ret = imx_pcm_new(rtd);
+ if (ret)
+ return ret;
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (substream) {
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+ imx_ssi_fiq_tx_buffer = (unsigned long)buf->area;
+ }
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ if (substream) {
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+ imx_ssi_fiq_rx_buffer = (unsigned long)buf->area;
+ }
+
+ set_fiq_handler(&imx_ssi_fiq_start,
+ &imx_ssi_fiq_end - &imx_ssi_fiq_start);
+
+ return 0;
+}
+
+static void imx_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static void imx_pcm_fiq_free(struct snd_pcm *pcm)
+{
+ mxc_set_irq_fiq(ssi_irq, 0);
+ release_fiq(&fh);
+ imx_pcm_free(pcm);
+}
+
+static struct snd_soc_platform_driver imx_soc_platform_fiq = {
+ .ops = &imx_pcm_ops,
+ .pcm_new = imx_pcm_fiq_new,
+ .pcm_free = imx_pcm_fiq_free,
+};
+
+int imx_pcm_fiq_init(struct platform_device *pdev,
+ struct imx_pcm_fiq_params *params)
+{
+ int ret;
+
+ ret = claim_fiq(&fh);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to claim fiq: %d", ret);
+ return ret;
+ }
+
+ mxc_set_irq_fiq(params->irq, 1);
+ ssi_irq = params->irq;
+
+ imx_pcm_fiq = params->irq;
+
+ imx_ssi_fiq_base = (unsigned long)params->base;
+
+ params->dma_params_tx->maxburst = 4;
+ params->dma_params_rx->maxburst = 6;
+
+ ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq);
+ if (ret)
+ goto failed_register;
+
+ return 0;
+
+failed_register:
+ mxc_set_irq_fiq(ssi_irq, 0);
+ release_fiq(&fh);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(imx_pcm_fiq_init);
+
+void imx_pcm_fiq_exit(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+}
+EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
new file mode 100644
index 000000000..c79cb2747
--- /dev/null
+++ b/sound/soc/fsl/imx-pcm.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
+ *
+ * This code is based on code copyrighted by Freescale,
+ * Liam Girdwood, Javier Martin and probably others.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef _IMX_PCM_H
+#define _IMX_PCM_H
+
+#include <linux/platform_data/dma-imx.h>
+
+/*
+ * Do not change this as the FIQ handler depends on this size
+ */
+#define IMX_SSI_DMABUF_SIZE (64 * 1024)
+
+static inline void
+imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data,
+ int dma, enum sdma_peripheral_type peripheral_type)
+{
+ dma_data->dma_request = dma;
+ dma_data->priority = DMA_PRIO_HIGH;
+ dma_data->peripheral_type = peripheral_type;
+}
+
+struct imx_pcm_fiq_params {
+ int irq;
+ void __iomem *base;
+
+ /* Pointer to original ssi driver to setup tx rx sizes */
+ struct snd_dmaengine_dai_dma_data *dma_params_rx;
+ struct snd_dmaengine_dai_dma_data *dma_params_tx;
+};
+
+#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA)
+int imx_pcm_dma_init(struct platform_device *pdev);
+#else
+static inline int imx_pcm_dma_init(struct platform_device *pdev)
+{
+ return -ENODEV;
+}
+#endif
+
+#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ)
+int imx_pcm_fiq_init(struct platform_device *pdev,
+ struct imx_pcm_fiq_params *params);
+void imx_pcm_fiq_exit(struct platform_device *pdev);
+#else
+static inline int imx_pcm_fiq_init(struct platform_device *pdev,
+ struct imx_pcm_fiq_params *params)
+{
+ return -ENODEV;
+}
+
+static inline void imx_pcm_fiq_exit(struct platform_device *pdev)
+{
+}
+#endif
+
+#endif /* _IMX_PCM_H */
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
new file mode 100644
index 000000000..b99e0b5e0
--- /dev/null
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -0,0 +1,214 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+
+#include "../codecs/sgtl5000.h"
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE 32
+
+struct imx_sgtl5000_data {
+ struct snd_soc_dai_link dai;
+ struct snd_soc_card card;
+ char codec_dai_name[DAI_NAME_SIZE];
+ char platform_name[DAI_NAME_SIZE];
+ struct clk *codec_clk;
+ unsigned int clk_frequency;
+};
+
+static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card);
+ struct device *dev = rtd->card->dev;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
+ data->clk_frequency, SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "could not set codec driver clock params\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget imx_sgtl5000_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Line Out Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static int imx_sgtl5000_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np, *codec_np;
+ struct platform_device *ssi_pdev;
+ struct i2c_client *codec_dev;
+ struct imx_sgtl5000_data *data = NULL;
+ int int_port, ext_port;
+ int ret;
+
+ ret = of_property_read_u32(np, "mux-int-port", &int_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-int-port missing or invalid\n");
+ return ret;
+ }
+ ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-ext-port missing or invalid\n");
+ return ret;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the audmux API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+ ret = imx_audmux_v2_configure_port(int_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux internal port setup failed\n");
+ return ret;
+ }
+ ret = imx_audmux_v2_configure_port(ext_port,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux external port setup failed\n");
+ return ret;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+ codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+ if (!ssi_np || !codec_np) {
+ dev_err(&pdev->dev, "phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ssi_pdev = of_find_device_by_node(ssi_np);
+ if (!ssi_pdev) {
+ dev_err(&pdev->dev, "failed to find SSI platform device\n");
+ ret = -EPROBE_DEFER;
+ goto fail;
+ }
+ codec_dev = of_find_i2c_device_by_node(codec_np);
+ if (!codec_dev) {
+ dev_err(&pdev->dev, "failed to find codec platform device\n");
+ return -EPROBE_DEFER;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ data->codec_clk = clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(data->codec_clk)) {
+ ret = PTR_ERR(data->codec_clk);
+ goto fail;
+ }
+
+ data->clk_frequency = clk_get_rate(data->codec_clk);
+
+ data->dai.name = "HiFi";
+ data->dai.stream_name = "HiFi";
+ data->dai.codec_dai_name = "sgtl5000";
+ data->dai.codec_of_node = codec_np;
+ data->dai.cpu_of_node = ssi_np;
+ data->dai.platform_of_node = ssi_np;
+ data->dai.init = &imx_sgtl5000_dai_init;
+ data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ data->card.dev = &pdev->dev;
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto fail;
+ ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+ if (ret)
+ goto fail;
+ data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
+ data->card.dai_link = &data->dai;
+ data->card.dapm_widgets = imx_sgtl5000_dapm_widgets;
+ data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets);
+
+ platform_set_drvdata(pdev, &data->card);
+ snd_soc_card_set_drvdata(&data->card, data);
+
+ ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ goto fail;
+ }
+
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
+
+ return 0;
+
+fail:
+ if (data && !IS_ERR(data->codec_clk))
+ clk_put(data->codec_clk);
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
+
+ return ret;
+}
+
+static int imx_sgtl5000_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card);
+
+ clk_put(data->codec_clk);
+
+ return 0;
+}
+
+static const struct of_device_id imx_sgtl5000_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-sgtl5000", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_sgtl5000_dt_ids);
+
+static struct platform_driver imx_sgtl5000_driver = {
+ .driver = {
+ .name = "imx-sgtl5000",
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = imx_sgtl5000_dt_ids,
+ },
+ .probe = imx_sgtl5000_probe,
+ .remove = imx_sgtl5000_remove,
+};
+module_platform_driver(imx_sgtl5000_driver);
+
+MODULE_AUTHOR("Shawn Guo <shawn.guo@linaro.org>");
+MODULE_DESCRIPTION("Freescale i.MX SGTL5000 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-sgtl5000");
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
new file mode 100644
index 000000000..33da26a12
--- /dev/null
+++ b/sound/soc/fsl/imx-spdif.c
@@ -0,0 +1,102 @@
+/*
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+struct imx_spdif_data {
+ struct snd_soc_dai_link dai;
+ struct snd_soc_card card;
+};
+
+static int imx_spdif_audio_probe(struct platform_device *pdev)
+{
+ struct device_node *spdif_np, *np = pdev->dev.of_node;
+ struct imx_spdif_data *data;
+ int ret = 0;
+
+ spdif_np = of_parse_phandle(np, "spdif-controller", 0);
+ if (!spdif_np) {
+ dev_err(&pdev->dev, "failed to find spdif-controller\n");
+ ret = -EINVAL;
+ goto end;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto end;
+ }
+
+ data->dai.name = "S/PDIF PCM";
+ data->dai.stream_name = "S/PDIF PCM";
+ data->dai.codec_dai_name = "snd-soc-dummy-dai";
+ data->dai.codec_name = "snd-soc-dummy";
+ data->dai.cpu_of_node = spdif_np;
+ data->dai.platform_of_node = spdif_np;
+ data->dai.playback_only = true;
+ data->dai.capture_only = true;
+
+ if (of_property_read_bool(np, "spdif-out"))
+ data->dai.capture_only = false;
+
+ if (of_property_read_bool(np, "spdif-in"))
+ data->dai.playback_only = false;
+
+ if (data->dai.playback_only && data->dai.capture_only) {
+ dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
+ goto end;
+ }
+
+ data->card.dev = &pdev->dev;
+ data->card.dai_link = &data->dai;
+ data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
+
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto end;
+
+ ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+ goto end;
+ }
+
+ platform_set_drvdata(pdev, data);
+
+end:
+ of_node_put(spdif_np);
+
+ return ret;
+}
+
+static const struct of_device_id imx_spdif_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-spdif", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
+
+static struct platform_driver imx_spdif_driver = {
+ .driver = {
+ .name = "imx-spdif",
+ .of_match_table = imx_spdif_dt_ids,
+ },
+ .probe = imx_spdif_audio_probe,
+};
+
+module_platform_driver(imx_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-spdif");
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
new file mode 100644
index 000000000..461ce27b8
--- /dev/null
+++ b/sound/soc/fsl/imx-ssi.c
@@ -0,0 +1,658 @@
+/*
+ * imx-ssi.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de>
+ *
+ * This code is based on code copyrighted by Freescale,
+ * Liam Girdwood, Javier Martin and probably others.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ *
+ * The i.MX SSI core has some nasty limitations in AC97 mode. While most
+ * sane processor vendors have a FIFO per AC97 slot, the i.MX has only
+ * one FIFO which combines all valid receive slots. We cannot even select
+ * which slots we want to receive. The WM9712 with which this driver
+ * was developed with always sends GPIO status data in slot 12 which
+ * we receive in our (PCM-) data stream. The only chance we have is to
+ * manually skip this data in the FIQ handler. With sampling rates different
+ * from 48000Hz not every frame has valid receive data, so the ratio
+ * between pcm data and GPIO status data changes. Our FIQ handler is not
+ * able to handle this, hence this driver only works with 48000Hz sampling
+ * rate.
+ * Reading and writing AC97 registers is another challenge. The core
+ * provides us status bits when the read register is updated with *another*
+ * value. When we read the same register two times (and the register still
+ * contains the same value) these status bits are not set. We work
+ * around this by not polling these bits but only wait a fixed delay.
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <linux/platform_data/asoc-imx-ssi.h>
+
+#include "imx-ssi.h"
+#include "fsl_utils.h"
+
+#define SSI_SACNT_DEFAULT (SSI_SACNT_AC97EN | SSI_SACNT_FV)
+
+/*
+ * SSI Network Mode or TDM slots configuration.
+ * Should only be called when port is inactive (i.e. SSIEN = 0).
+ */
+static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
+{
+ struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 sccr;
+
+ sccr = readl(ssi->base + SSI_STCCR);
+ sccr &= ~SSI_STCCR_DC_MASK;
+ sccr |= SSI_STCCR_DC(slots - 1);
+ writel(sccr, ssi->base + SSI_STCCR);
+
+ sccr = readl(ssi->base + SSI_SRCCR);
+ sccr &= ~SSI_STCCR_DC_MASK;
+ sccr |= SSI_STCCR_DC(slots - 1);
+ writel(sccr, ssi->base + SSI_SRCCR);
+
+ writel(~tx_mask, ssi->base + SSI_STMSK);
+ writel(~rx_mask, ssi->base + SSI_SRMSK);
+
+ return 0;
+}
+
+/*
+ * SSI DAI format configuration.
+ * Should only be called when port is inactive (i.e. SSIEN = 0).
+ */
+static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 strcr = 0, scr;
+
+ scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET);
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* data on rising edge of bclk, frame low 1clk before data */
+ strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0;
+ scr |= SSI_SCR_NET;
+ if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) {
+ scr &= ~SSI_I2S_MODE_MASK;
+ scr |= SSI_SCR_I2S_MODE_SLAVE;
+ }
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /* data on rising edge of bclk, frame high with data */
+ strcr |= SSI_STCR_TXBIT0;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ /* data on rising edge of bclk, frame high with data */
+ strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* data on rising edge of bclk, frame high 1clk before data */
+ strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
+ break;
+ }
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ strcr |= SSI_STCR_TFSI;
+ strcr &= ~SSI_STCR_TSCKP;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP;
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ strcr &= ~SSI_STCR_TFSI;
+ strcr |= SSI_STCR_TSCKP;
+ break;
+ }
+
+ /* DAI clock master masks */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ default:
+ /* Master mode not implemented, needs handling of clocks. */
+ return -EINVAL;
+ }
+
+ strcr |= SSI_STCR_TFEN0;
+
+ if (ssi->flags & IMX_SSI_NET)
+ scr |= SSI_SCR_NET;
+ if (ssi->flags & IMX_SSI_SYN)
+ scr |= SSI_SCR_SYN;
+
+ writel(strcr, ssi->base + SSI_STCR);
+ writel(strcr, ssi->base + SSI_SRCR);
+ writel(scr, ssi->base + SSI_SCR);
+
+ return 0;
+}
+
+/*
+ * SSI system clock configuration.
+ * Should only be called when port is inactive (i.e. SSIEN = 0).
+ */
+static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 scr;
+
+ scr = readl(ssi->base + SSI_SCR);
+
+ switch (clk_id) {
+ case IMX_SSP_SYS_CLK:
+ if (dir == SND_SOC_CLOCK_OUT)
+ scr |= SSI_SCR_SYS_CLK_EN;
+ else
+ scr &= ~SSI_SCR_SYS_CLK_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ writel(scr, ssi->base + SSI_SCR);
+
+ return 0;
+}
+
+/*
+ * SSI Clock dividers
+ * Should only be called when port is inactive (i.e. SSIEN = 0).
+ */
+static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 stccr, srccr;
+
+ stccr = readl(ssi->base + SSI_STCCR);
+ srccr = readl(ssi->base + SSI_SRCCR);
+
+ switch (div_id) {
+ case IMX_SSI_TX_DIV_2:
+ stccr &= ~SSI_STCCR_DIV2;
+ stccr |= div;
+ break;
+ case IMX_SSI_TX_DIV_PSR:
+ stccr &= ~SSI_STCCR_PSR;
+ stccr |= div;
+ break;
+ case IMX_SSI_TX_DIV_PM:
+ stccr &= ~0xff;
+ stccr |= SSI_STCCR_PM(div);
+ break;
+ case IMX_SSI_RX_DIV_2:
+ stccr &= ~SSI_STCCR_DIV2;
+ stccr |= div;
+ break;
+ case IMX_SSI_RX_DIV_PSR:
+ stccr &= ~SSI_STCCR_PSR;
+ stccr |= div;
+ break;
+ case IMX_SSI_RX_DIV_PM:
+ stccr &= ~0xff;
+ stccr |= SSI_STCCR_PM(div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ writel(stccr, ssi->base + SSI_STCCR);
+ writel(srccr, ssi->base + SSI_SRCCR);
+
+ return 0;
+}
+
+/*
+ * Should only be called when port is inactive (i.e. SSIEN = 0),
+ * although can be called multiple times by upper layers.
+ */
+static int imx_ssi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 reg, sccr;
+
+ /* Tx/Rx config */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = SSI_STCCR;
+ else
+ reg = SSI_SRCCR;
+
+ if (ssi->flags & IMX_SSI_SYN)
+ reg = SSI_STCCR;
+
+ sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK;
+
+ /* DAI data (word) size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ sccr |= SSI_SRCCR_WL(16);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ sccr |= SSI_SRCCR_WL(20);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ sccr |= SSI_SRCCR_WL(24);
+ break;
+ }
+
+ writel(sccr, ssi->base + reg);
+
+ return 0;
+}
+
+static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct imx_ssi *ssi = snd_soc_dai_get_drvdata(dai);
+ unsigned int sier_bits, sier;
+ unsigned int scr;
+
+ scr = readl(ssi->base + SSI_SCR);
+ sier = readl(ssi->base + SSI_SIER);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (ssi->flags & IMX_SSI_DMA)
+ sier_bits = SSI_SIER_TDMAE;
+ else
+ sier_bits = SSI_SIER_TIE | SSI_SIER_TFE0_EN;
+ } else {
+ if (ssi->flags & IMX_SSI_DMA)
+ sier_bits = SSI_SIER_RDMAE;
+ else
+ sier_bits = SSI_SIER_RIE | SSI_SIER_RFF0_EN;
+ }
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ scr |= SSI_SCR_TE;
+ else
+ scr |= SSI_SCR_RE;
+ sier |= sier_bits;
+
+ scr |= SSI_SCR_SSIEN;
+
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ scr &= ~SSI_SCR_TE;
+ else
+ scr &= ~SSI_SCR_RE;
+ sier &= ~sier_bits;
+
+ if (!(scr & (SSI_SCR_TE | SSI_SCR_RE)))
+ scr &= ~SSI_SCR_SSIEN;
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (!(ssi->flags & IMX_SSI_USE_AC97))
+ /* rx/tx are always enabled to access ac97 registers */
+ writel(scr, ssi->base + SSI_SCR);
+
+ writel(sier, ssi->base + SSI_SIER);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = {
+ .hw_params = imx_ssi_hw_params,
+ .set_fmt = imx_ssi_set_dai_fmt,
+ .set_clkdiv = imx_ssi_set_dai_clkdiv,
+ .set_sysclk = imx_ssi_set_dai_sysclk,
+ .set_tdm_slot = imx_ssi_set_dai_tdm_slot,
+ .trigger = imx_ssi_trigger,
+};
+
+static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
+{
+ struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
+ uint32_t val;
+
+ snd_soc_dai_set_drvdata(dai, ssi);
+
+ val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.maxburst) |
+ SSI_SFCSR_RFWM0(ssi->dma_params_rx.maxburst);
+ writel(val, ssi->base + SSI_SFCSR);
+
+ /* Tx/Rx config */
+ dai->playback_dma_data = &ssi->dma_params_tx;
+ dai->capture_dma_data = &ssi->dma_params_rx;
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver imx_ssi_dai = {
+ .probe = imx_ssi_dai_probe,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &imx_ssi_pcm_dai_ops,
+};
+
+static struct snd_soc_dai_driver imx_ac97_dai = {
+ .probe = imx_ssi_dai_probe,
+ .bus_control = true,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &imx_ssi_pcm_dai_ops,
+};
+
+static const struct snd_soc_component_driver imx_component = {
+ .name = DRV_NAME,
+};
+
+static void setup_channel_to_ac97(struct imx_ssi *imx_ssi)
+{
+ void __iomem *base = imx_ssi->base;
+
+ writel(0x0, base + SSI_SCR);
+ writel(0x0, base + SSI_STCR);
+ writel(0x0, base + SSI_SRCR);
+
+ writel(SSI_SCR_SYN | SSI_SCR_NET, base + SSI_SCR);
+
+ writel(SSI_SFCSR_RFWM0(8) |
+ SSI_SFCSR_TFWM0(8) |
+ SSI_SFCSR_RFWM1(8) |
+ SSI_SFCSR_TFWM1(8), base + SSI_SFCSR);
+
+ writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_STCCR);
+ writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_SRCCR);
+
+ writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN, base + SSI_SCR);
+ writel(SSI_SOR_WAIT(3), base + SSI_SOR);
+
+ writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN |
+ SSI_SCR_TE | SSI_SCR_RE,
+ base + SSI_SCR);
+
+ writel(SSI_SACNT_DEFAULT, base + SSI_SACNT);
+ writel(0xff, base + SSI_SACCDIS);
+ writel(0x300, base + SSI_SACCEN);
+}
+
+static struct imx_ssi *ac97_ssi;
+
+static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct imx_ssi *imx_ssi = ac97_ssi;
+ void __iomem *base = imx_ssi->base;
+ unsigned int lreg;
+ unsigned int lval;
+
+ if (reg > 0x7f)
+ return;
+
+ pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val);
+
+ lreg = reg << 12;
+ writel(lreg, base + SSI_SACADD);
+
+ lval = val << 4;
+ writel(lval , base + SSI_SACDAT);
+
+ writel(SSI_SACNT_DEFAULT | SSI_SACNT_WR, base + SSI_SACNT);
+ udelay(100);
+}
+
+static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct imx_ssi *imx_ssi = ac97_ssi;
+ void __iomem *base = imx_ssi->base;
+
+ unsigned short val = -1;
+ unsigned int lreg;
+
+ lreg = (reg & 0x7f) << 12 ;
+ writel(lreg, base + SSI_SACADD);
+ writel(SSI_SACNT_DEFAULT | SSI_SACNT_RD, base + SSI_SACNT);
+
+ udelay(100);
+
+ val = (readl(base + SSI_SACDAT) >> 4) & 0xffff;
+
+ pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val);
+
+ return val;
+}
+
+static void imx_ssi_ac97_reset(struct snd_ac97 *ac97)
+{
+ struct imx_ssi *imx_ssi = ac97_ssi;
+
+ if (imx_ssi->ac97_reset)
+ imx_ssi->ac97_reset(ac97);
+ /* First read sometimes fails, do a dummy read */
+ imx_ssi_ac97_read(ac97, 0);
+}
+
+static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct imx_ssi *imx_ssi = ac97_ssi;
+
+ if (imx_ssi->ac97_warm_reset)
+ imx_ssi->ac97_warm_reset(ac97);
+
+ /* First read sometimes fails, do a dummy read */
+ imx_ssi_ac97_read(ac97, 0);
+}
+
+static struct snd_ac97_bus_ops imx_ssi_ac97_ops = {
+ .read = imx_ssi_ac97_read,
+ .write = imx_ssi_ac97_write,
+ .reset = imx_ssi_ac97_reset,
+ .warm_reset = imx_ssi_ac97_warm_reset
+};
+
+static int imx_ssi_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ struct imx_ssi *ssi;
+ struct imx_ssi_platform_data *pdata = pdev->dev.platform_data;
+ int ret = 0;
+ struct snd_soc_dai_driver *dai;
+
+ ssi = devm_kzalloc(&pdev->dev, sizeof(*ssi), GFP_KERNEL);
+ if (!ssi)
+ return -ENOMEM;
+ dev_set_drvdata(&pdev->dev, ssi);
+
+ if (pdata) {
+ ssi->ac97_reset = pdata->ac97_reset;
+ ssi->ac97_warm_reset = pdata->ac97_warm_reset;
+ ssi->flags = pdata->flags;
+ }
+
+ ssi->irq = platform_get_irq(pdev, 0);
+
+ ssi->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(ssi->clk)) {
+ ret = PTR_ERR(ssi->clk);
+ dev_err(&pdev->dev, "Cannot get the clock: %d\n",
+ ret);
+ goto failed_clk;
+ }
+ ret = clk_prepare_enable(ssi->clk);
+ if (ret)
+ goto failed_clk;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ ssi->base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(ssi->base)) {
+ ret = PTR_ERR(ssi->base);
+ goto failed_register;
+ }
+
+ if (ssi->flags & IMX_SSI_USE_AC97) {
+ if (ac97_ssi) {
+ dev_err(&pdev->dev, "AC'97 SSI already registered\n");
+ ret = -EBUSY;
+ goto failed_register;
+ }
+ ac97_ssi = ssi;
+ setup_channel_to_ac97(ssi);
+ dai = &imx_ac97_dai;
+ } else
+ dai = &imx_ssi_dai;
+
+ writel(0x0, ssi->base + SSI_SIER);
+
+ ssi->dma_params_rx.addr = res->start + SSI_SRX0;
+ ssi->dma_params_tx.addr = res->start + SSI_STX0;
+
+ ssi->dma_params_tx.maxburst = 6;
+ ssi->dma_params_rx.maxburst = 4;
+
+ ssi->dma_params_tx.filter_data = &ssi->filter_data_tx;
+ ssi->dma_params_rx.filter_data = &ssi->filter_data_rx;
+
+ res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0");
+ if (res) {
+ imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start,
+ IMX_DMATYPE_SSI);
+ }
+
+ res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0");
+ if (res) {
+ imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start,
+ IMX_DMATYPE_SSI);
+ }
+
+ platform_set_drvdata(pdev, ssi);
+
+ ret = snd_soc_set_ac97_ops(&imx_ssi_ac97_ops);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret);
+ goto failed_register;
+ }
+
+ ret = snd_soc_register_component(&pdev->dev, &imx_component,
+ dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "register DAI failed\n");
+ goto failed_register;
+ }
+
+ ssi->fiq_params.irq = ssi->irq;
+ ssi->fiq_params.base = ssi->base;
+ ssi->fiq_params.dma_params_rx = &ssi->dma_params_rx;
+ ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx;
+
+ ssi->fiq_init = imx_pcm_fiq_init(pdev, &ssi->fiq_params);
+ ssi->dma_init = imx_pcm_dma_init(pdev);
+
+ if (ssi->fiq_init && ssi->dma_init) {
+ ret = ssi->fiq_init;
+ goto failed_pcm;
+ }
+
+ return 0;
+
+failed_pcm:
+ snd_soc_unregister_component(&pdev->dev);
+failed_register:
+ clk_disable_unprepare(ssi->clk);
+failed_clk:
+ snd_soc_set_ac97_ops(NULL);
+
+ return ret;
+}
+
+static int imx_ssi_remove(struct platform_device *pdev)
+{
+ struct imx_ssi *ssi = platform_get_drvdata(pdev);
+
+ if (!ssi->fiq_init)
+ imx_pcm_fiq_exit(pdev);
+
+ snd_soc_unregister_component(&pdev->dev);
+
+ if (ssi->flags & IMX_SSI_USE_AC97)
+ ac97_ssi = NULL;
+
+ clk_disable_unprepare(ssi->clk);
+ snd_soc_set_ac97_ops(NULL);
+
+ return 0;
+}
+
+static struct platform_driver imx_ssi_driver = {
+ .probe = imx_ssi_probe,
+ .remove = imx_ssi_remove,
+
+ .driver = {
+ .name = "imx-ssi",
+ },
+};
+
+module_platform_driver(imx_ssi_driver);
+
+/* Module information */
+MODULE_AUTHOR("Sascha Hauer, <s.hauer@pengutronix.de>");
+MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx-ssi");
diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
new file mode 100644
index 000000000..be6562365
--- /dev/null
+++ b/sound/soc/fsl/imx-ssi.h
@@ -0,0 +1,218 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _IMX_SSI_H
+#define _IMX_SSI_H
+
+#define SSI_STX0 0x00
+#define SSI_STX1 0x04
+#define SSI_SRX0 0x08
+#define SSI_SRX1 0x0c
+
+#define SSI_SCR 0x10
+#define SSI_SCR_CLK_IST (1 << 9)
+#define SSI_SCR_CLK_IST_SHIFT 9
+#define SSI_SCR_TCH_EN (1 << 8)
+#define SSI_SCR_SYS_CLK_EN (1 << 7)
+#define SSI_SCR_I2S_MODE_NORM (0 << 5)
+#define SSI_SCR_I2S_MODE_MSTR (1 << 5)
+#define SSI_SCR_I2S_MODE_SLAVE (2 << 5)
+#define SSI_I2S_MODE_MASK (3 << 5)
+#define SSI_SCR_SYN (1 << 4)
+#define SSI_SCR_NET (1 << 3)
+#define SSI_SCR_RE (1 << 2)
+#define SSI_SCR_TE (1 << 1)
+#define SSI_SCR_SSIEN (1 << 0)
+
+#define SSI_SISR 0x14
+#define SSI_SISR_MASK ((1 << 19) - 1)
+#define SSI_SISR_CMDAU (1 << 18)
+#define SSI_SISR_CMDDU (1 << 17)
+#define SSI_SISR_RXT (1 << 16)
+#define SSI_SISR_RDR1 (1 << 15)
+#define SSI_SISR_RDR0 (1 << 14)
+#define SSI_SISR_TDE1 (1 << 13)
+#define SSI_SISR_TDE0 (1 << 12)
+#define SSI_SISR_ROE1 (1 << 11)
+#define SSI_SISR_ROE0 (1 << 10)
+#define SSI_SISR_TUE1 (1 << 9)
+#define SSI_SISR_TUE0 (1 << 8)
+#define SSI_SISR_TFS (1 << 7)
+#define SSI_SISR_RFS (1 << 6)
+#define SSI_SISR_TLS (1 << 5)
+#define SSI_SISR_RLS (1 << 4)
+#define SSI_SISR_RFF1 (1 << 3)
+#define SSI_SISR_RFF0 (1 << 2)
+#define SSI_SISR_TFE1 (1 << 1)
+#define SSI_SISR_TFE0 (1 << 0)
+
+#define SSI_SIER 0x18
+#define SSI_SIER_RDMAE (1 << 22)
+#define SSI_SIER_RIE (1 << 21)
+#define SSI_SIER_TDMAE (1 << 20)
+#define SSI_SIER_TIE (1 << 19)
+#define SSI_SIER_CMDAU_EN (1 << 18)
+#define SSI_SIER_CMDDU_EN (1 << 17)
+#define SSI_SIER_RXT_EN (1 << 16)
+#define SSI_SIER_RDR1_EN (1 << 15)
+#define SSI_SIER_RDR0_EN (1 << 14)
+#define SSI_SIER_TDE1_EN (1 << 13)
+#define SSI_SIER_TDE0_EN (1 << 12)
+#define SSI_SIER_ROE1_EN (1 << 11)
+#define SSI_SIER_ROE0_EN (1 << 10)
+#define SSI_SIER_TUE1_EN (1 << 9)
+#define SSI_SIER_TUE0_EN (1 << 8)
+#define SSI_SIER_TFS_EN (1 << 7)
+#define SSI_SIER_RFS_EN (1 << 6)
+#define SSI_SIER_TLS_EN (1 << 5)
+#define SSI_SIER_RLS_EN (1 << 4)
+#define SSI_SIER_RFF1_EN (1 << 3)
+#define SSI_SIER_RFF0_EN (1 << 2)
+#define SSI_SIER_TFE1_EN (1 << 1)
+#define SSI_SIER_TFE0_EN (1 << 0)
+
+#define SSI_STCR 0x1c
+#define SSI_STCR_TXBIT0 (1 << 9)
+#define SSI_STCR_TFEN1 (1 << 8)
+#define SSI_STCR_TFEN0 (1 << 7)
+#define SSI_FIFO_ENABLE_0_SHIFT 7
+#define SSI_STCR_TFDIR (1 << 6)
+#define SSI_STCR_TXDIR (1 << 5)
+#define SSI_STCR_TSHFD (1 << 4)
+#define SSI_STCR_TSCKP (1 << 3)
+#define SSI_STCR_TFSI (1 << 2)
+#define SSI_STCR_TFSL (1 << 1)
+#define SSI_STCR_TEFS (1 << 0)
+
+#define SSI_SRCR 0x20
+#define SSI_SRCR_RXBIT0 (1 << 9)
+#define SSI_SRCR_RFEN1 (1 << 8)
+#define SSI_SRCR_RFEN0 (1 << 7)
+#define SSI_FIFO_ENABLE_0_SHIFT 7
+#define SSI_SRCR_RFDIR (1 << 6)
+#define SSI_SRCR_RXDIR (1 << 5)
+#define SSI_SRCR_RSHFD (1 << 4)
+#define SSI_SRCR_RSCKP (1 << 3)
+#define SSI_SRCR_RFSI (1 << 2)
+#define SSI_SRCR_RFSL (1 << 1)
+#define SSI_SRCR_REFS (1 << 0)
+
+#define SSI_SRCCR 0x28
+#define SSI_SRCCR_DIV2 (1 << 18)
+#define SSI_SRCCR_PSR (1 << 17)
+#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13)
+#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8)
+#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0)
+#define SSI_SRCCR_WL_MASK (0xf << 13)
+#define SSI_SRCCR_DC_MASK (0x1f << 8)
+#define SSI_SRCCR_PM_MASK (0xff << 0)
+
+#define SSI_STCCR 0x24
+#define SSI_STCCR_DIV2 (1 << 18)
+#define SSI_STCCR_PSR (1 << 17)
+#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13)
+#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8)
+#define SSI_STCCR_PM(x) (((x) & 0xff) << 0)
+#define SSI_STCCR_WL_MASK (0xf << 13)
+#define SSI_STCCR_DC_MASK (0x1f << 8)
+#define SSI_STCCR_PM_MASK (0xff << 0)
+
+#define SSI_SFCSR 0x2c
+#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28)
+#define SSI_RX_FIFO_1_COUNT_SHIFT 28
+#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24)
+#define SSI_TX_FIFO_1_COUNT_SHIFT 24
+#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20)
+#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16)
+#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12)
+#define SSI_RX_FIFO_0_COUNT_SHIFT 12
+#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8)
+#define SSI_TX_FIFO_0_COUNT_SHIFT 8
+#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4)
+#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0)
+#define SSI_SFCSR_RFWM0_MASK (0xf << 4)
+#define SSI_SFCSR_TFWM0_MASK (0xf << 0)
+
+#define SSI_STR 0x30
+#define SSI_STR_TEST (1 << 15)
+#define SSI_STR_RCK2TCK (1 << 14)
+#define SSI_STR_RFS2TFS (1 << 13)
+#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8)
+#define SSI_STR_TXD2RXD (1 << 7)
+#define SSI_STR_TCK2RCK (1 << 6)
+#define SSI_STR_TFS2RFS (1 << 5)
+#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0)
+
+#define SSI_SOR 0x34
+#define SSI_SOR_CLKOFF (1 << 6)
+#define SSI_SOR_RX_CLR (1 << 5)
+#define SSI_SOR_TX_CLR (1 << 4)
+#define SSI_SOR_INIT (1 << 3)
+#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1)
+#define SSI_SOR_WAIT_MASK (0x3 << 1)
+#define SSI_SOR_SYNRST (1 << 0)
+
+#define SSI_SACNT 0x38
+#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5)
+#define SSI_SACNT_WR (1 << 4)
+#define SSI_SACNT_RD (1 << 3)
+#define SSI_SACNT_TIF (1 << 2)
+#define SSI_SACNT_FV (1 << 1)
+#define SSI_SACNT_AC97EN (1 << 0)
+
+#define SSI_SACADD 0x3c
+#define SSI_SACDAT 0x40
+#define SSI_SATAG 0x44
+#define SSI_STMSK 0x48
+#define SSI_SRMSK 0x4c
+#define SSI_SACCST 0x50
+#define SSI_SACCEN 0x54
+#define SSI_SACCDIS 0x58
+
+/* SSI clock sources */
+#define IMX_SSP_SYS_CLK 0
+
+/* SSI audio dividers */
+#define IMX_SSI_TX_DIV_2 0
+#define IMX_SSI_TX_DIV_PSR 1
+#define IMX_SSI_TX_DIV_PM 2
+#define IMX_SSI_RX_DIV_2 3
+#define IMX_SSI_RX_DIV_PSR 4
+#define IMX_SSI_RX_DIV_PM 5
+
+#define DRV_NAME "imx-ssi"
+
+#include <linux/dmaengine.h>
+#include <linux/platform_data/dma-imx.h>
+#include <sound/dmaengine_pcm.h>
+#include "imx-pcm.h"
+
+struct imx_ssi {
+ struct platform_device *ac97_dev;
+
+ struct snd_soc_dai *imx_ac97;
+ struct clk *clk;
+ void __iomem *base;
+ int irq;
+ int fiq_enable;
+ unsigned int offset;
+
+ unsigned int flags;
+
+ void (*ac97_reset) (struct snd_ac97 *ac97);
+ void (*ac97_warm_reset)(struct snd_ac97 *ac97);
+
+ struct snd_dmaengine_dai_dma_data dma_params_rx;
+ struct snd_dmaengine_dai_dma_data dma_params_tx;
+ struct imx_dma_data filter_data_tx;
+ struct imx_dma_data filter_data_rx;
+ struct imx_pcm_fiq_params fiq_params;
+
+ int fiq_init;
+ int dma_init;
+};
+
+#endif /* _IMX_SSI_H */
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
new file mode 100644
index 000000000..b38b98cae
--- /dev/null
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -0,0 +1,322 @@
+/*
+ * Copyright 2013 Freescale Semiconductor, Inc.
+ *
+ * Based on imx-sgtl5000.c
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <linux/pinctrl/consumer.h>
+
+#include "../codecs/wm8962.h"
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE 32
+
+struct imx_wm8962_data {
+ struct snd_soc_dai_link dai;
+ struct snd_soc_card card;
+ char codec_dai_name[DAI_NAME_SIZE];
+ char platform_name[DAI_NAME_SIZE];
+ struct clk *codec_clk;
+ unsigned int clk_frequency;
+};
+
+struct imx_priv {
+ struct platform_device *pdev;
+};
+static struct imx_priv card_priv;
+
+static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_MIC("AMIC", NULL),
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static int sample_rate = 44100;
+static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE;
+
+static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ sample_rate = params_rate(params);
+ sample_format = params_format(params);
+
+ return 0;
+}
+
+static struct snd_soc_ops imx_hifi_ops = {
+ .hw_params = imx_hifi_hw_params,
+};
+
+static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
+ struct device *dev = &priv->pdev->dev;
+ unsigned int pll_out;
+ int ret;
+
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+ if (sample_format == SNDRV_PCM_FORMAT_S24_LE)
+ pll_out = sample_rate * 384;
+ else
+ pll_out = sample_rate * 256;
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ WM8962_FLL_MCLK, data->clk_frequency,
+ pll_out);
+ if (ret < 0) {
+ dev_err(dev, "failed to start FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8962_SYSCLK_FLL, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE) {
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8962_SYSCLK_MCLK, data->clk_frequency,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(dev,
+ "failed to switch away from FLL: %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ 0, 0, 0);
+ if (ret < 0) {
+ dev_err(dev, "failed to stop FLL: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int imx_wm8962_late_probe(struct snd_soc_card *card)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
+ struct device *dev = &priv->pdev->dev;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+ data->clk_frequency, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ dev_err(dev, "failed to set sysclk in %s\n", __func__);
+
+ return ret;
+}
+
+static int imx_wm8962_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np, *codec_np;
+ struct platform_device *ssi_pdev;
+ struct imx_priv *priv = &card_priv;
+ struct i2c_client *codec_dev;
+ struct imx_wm8962_data *data;
+ int int_port, ext_port;
+ int ret;
+
+ priv->pdev = pdev;
+
+ ret = of_property_read_u32(np, "mux-int-port", &int_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-int-port missing or invalid\n");
+ return ret;
+ }
+ ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-ext-port missing or invalid\n");
+ return ret;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the audmux API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+ ret = imx_audmux_v2_configure_port(int_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux internal port setup failed\n");
+ return ret;
+ }
+ ret = imx_audmux_v2_configure_port(ext_port,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux external port setup failed\n");
+ return ret;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+ codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+ if (!ssi_np || !codec_np) {
+ dev_err(&pdev->dev, "phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ssi_pdev = of_find_device_by_node(ssi_np);
+ if (!ssi_pdev) {
+ dev_err(&pdev->dev, "failed to find SSI platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+ codec_dev = of_find_i2c_device_by_node(codec_np);
+ if (!codec_dev || !codec_dev->dev.driver) {
+ dev_err(&pdev->dev, "failed to find codec platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(data->codec_clk)) {
+ ret = PTR_ERR(data->codec_clk);
+ dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
+ goto fail;
+ }
+
+ data->clk_frequency = clk_get_rate(data->codec_clk);
+ ret = clk_prepare_enable(data->codec_clk);
+ if (ret) {
+ dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret);
+ goto fail;
+ }
+
+ data->dai.name = "HiFi";
+ data->dai.stream_name = "HiFi";
+ data->dai.codec_dai_name = "wm8962";
+ data->dai.codec_of_node = codec_np;
+ data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev);
+ data->dai.platform_of_node = ssi_np;
+ data->dai.ops = &imx_hifi_ops;
+ data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ data->card.dev = &pdev->dev;
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto clk_fail;
+ ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+ if (ret)
+ goto clk_fail;
+ data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
+ data->card.dai_link = &data->dai;
+ data->card.dapm_widgets = imx_wm8962_dapm_widgets;
+ data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets);
+
+ data->card.late_probe = imx_wm8962_late_probe;
+ data->card.set_bias_level = imx_wm8962_set_bias_level;
+
+ platform_set_drvdata(pdev, &data->card);
+ snd_soc_card_set_drvdata(&data->card, data);
+
+ ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ goto clk_fail;
+ }
+
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
+
+ return 0;
+
+clk_fail:
+ clk_disable_unprepare(data->codec_clk);
+fail:
+ of_node_put(ssi_np);
+ of_node_put(codec_np);
+
+ return ret;
+}
+
+static int imx_wm8962_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
+
+ if (!IS_ERR(data->codec_clk))
+ clk_disable_unprepare(data->codec_clk);
+
+ return 0;
+}
+
+static const struct of_device_id imx_wm8962_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-wm8962", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids);
+
+static struct platform_driver imx_wm8962_driver = {
+ .driver = {
+ .name = "imx-wm8962",
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = imx_wm8962_dt_ids,
+ },
+ .probe = imx_wm8962_probe,
+ .remove = imx_wm8962_remove,
+};
+module_platform_driver(imx_wm8962_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-wm8962");
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
new file mode 100644
index 000000000..0b82e209b
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -0,0 +1,511 @@
+/*
+ * Freescale MPC5200 PSC DMA
+ * ALSA SoC Platform driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/of_address.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+
+#include <sound/soc.h>
+
+#include <linux/fsl/bestcomm/bestcomm.h>
+#include <linux/fsl/bestcomm/gen_bd.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+
+/*
+ * Interrupt handlers
+ */
+static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma)
+{
+ struct psc_dma *psc_dma = _psc_dma;
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+ u16 isr;
+
+ isr = in_be16(&regs->mpc52xx_psc_isr);
+
+ /* Playback underrun error */
+ if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
+ psc_dma->stats.underrun_count++;
+
+ /* Capture overrun error */
+ if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
+ psc_dma->stats.overrun_count++;
+
+ out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+
+ return IRQ_HANDLED;
+}
+
+/**
+ * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * @s: pointer to stream private data structure
+ *
+ * Enqueues another audio period buffer into the bestcomm queue.
+ *
+ * Note: The routine must only be called when there is space available in
+ * the queue. Otherwise the enqueue will fail and the audio ring buffer
+ * will get out of sync
+ */
+static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
+{
+ struct bcom_bd *bd;
+
+ /* Prepare and enqueue the next buffer descriptor */
+ bd = bcom_prepare_next_buffer(s->bcom_task);
+ bd->status = s->period_bytes;
+ bd->data[0] = s->runtime->dma_addr + (s->period_next * s->period_bytes);
+ bcom_submit_next_buffer(s->bcom_task, NULL);
+
+ /* Update for next period */
+ s->period_next = (s->period_next + 1) % s->runtime->periods;
+}
+
+/* Bestcomm DMA irq handler */
+static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream)
+{
+ struct psc_dma_stream *s = _psc_dma_stream;
+
+ spin_lock(&s->psc_dma->lock);
+ /* For each finished period, dequeue the completed period buffer
+ * and enqueue a new one in it's place. */
+ while (bcom_buffer_done(s->bcom_task)) {
+ bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+ s->period_current = (s->period_current+1) % s->runtime->periods;
+ s->period_count++;
+
+ psc_dma_bcom_enqueue_next_buffer(s);
+ }
+ spin_unlock(&s->psc_dma->lock);
+
+ /* If the stream is active, then also inform the PCM middle layer
+ * of the period finished event. */
+ if (s->active)
+ snd_pcm_period_elapsed(s->stream);
+
+ return IRQ_HANDLED;
+}
+
+static int psc_dma_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+/**
+ * psc_dma_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ */
+static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+ u16 imr;
+ unsigned long flags;
+ int i;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ dev_dbg(psc_dma->dev, "START: stream=%i fbits=%u ps=%u #p=%u\n",
+ substream->pstr->stream, runtime->frame_bits,
+ (int)runtime->period_size, runtime->periods);
+ s->period_bytes = frames_to_bytes(runtime,
+ runtime->period_size);
+ s->period_next = 0;
+ s->period_current = 0;
+ s->active = 1;
+ s->period_count = 0;
+ s->runtime = runtime;
+
+ /* Fill up the bestcomm bd queue and enable DMA.
+ * This will begin filling the PSC's fifo.
+ */
+ spin_lock_irqsave(&psc_dma->lock, flags);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ bcom_gen_bd_rx_reset(s->bcom_task);
+ else
+ bcom_gen_bd_tx_reset(s->bcom_task);
+
+ for (i = 0; i < runtime->periods; i++)
+ if (!bcom_queue_full(s->bcom_task))
+ psc_dma_bcom_enqueue_next_buffer(s);
+
+ bcom_enable(s->bcom_task);
+ spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+ out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ dev_dbg(psc_dma->dev, "STOP: stream=%i periods_count=%i\n",
+ substream->pstr->stream, s->period_count);
+ s->active = 0;
+
+ spin_lock_irqsave(&psc_dma->lock, flags);
+ bcom_disable(s->bcom_task);
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ bcom_gen_bd_rx_reset(s->bcom_task);
+ else
+ bcom_gen_bd_tx_reset(s->bcom_task);
+ spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+ break;
+
+ default:
+ dev_dbg(psc_dma->dev, "unhandled trigger: stream=%i cmd=%i\n",
+ substream->pstr->stream, cmd);
+ return -EINVAL;
+ }
+
+ /* Update interrupt enable settings */
+ imr = 0;
+ if (psc_dma->playback.active)
+ imr |= MPC52xx_PSC_IMR_TXEMP;
+ if (psc_dma->capture.active)
+ imr |= MPC52xx_PSC_IMR_ORERR;
+ out_be16(&regs->isr_imr.imr, psc_dma->imr | imr);
+
+ return 0;
+}
+
+
+/* ---------------------------------------------------------------------
+ * The PSC DMA 'ASoC platform' driver
+ *
+ * Can be referenced by an 'ASoC machine' driver
+ * This driver only deals with the audio bus; it doesn't have any
+ * interaction with the attached codec
+ */
+
+static const struct snd_pcm_hardware psc_dma_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_BATCH,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
+ .period_bytes_max = 1024 * 1024,
+ .period_bytes_min = 32,
+ .periods_min = 2,
+ .periods_max = 256,
+ .buffer_bytes_max = 2 * 1024 * 1024,
+ .fifo_size = 512,
+};
+
+static int psc_dma_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct psc_dma_stream *s;
+ int rc;
+
+ dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware);
+
+ rc = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (rc < 0) {
+ dev_err(substream->pcm->card->dev, "invalid buffer size\n");
+ return rc;
+ }
+
+ s->stream = substream;
+ return 0;
+}
+
+static int psc_dma_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct psc_dma_stream *s;
+
+ dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ if (!psc_dma->playback.active &&
+ !psc_dma->capture.active) {
+
+ /* Disable all interrupts and reset the PSC */
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+ out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */
+ }
+ s->stream = NULL;
+ return 0;
+}
+
+static snd_pcm_uframes_t
+psc_dma_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct psc_dma_stream *s;
+ dma_addr_t count;
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ count = s->period_current * s->period_bytes;
+
+ return bytes_to_frames(substream->runtime, count);
+}
+
+static int
+psc_dma_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static struct snd_pcm_ops psc_dma_ops = {
+ .open = psc_dma_open,
+ .close = psc_dma_close,
+ .hw_free = psc_dma_hw_free,
+ .ioctl = snd_pcm_lib_ioctl,
+ .pointer = psc_dma_pointer,
+ .trigger = psc_dma_trigger,
+ .hw_params = psc_dma_hw_params,
+};
+
+static int psc_dma_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ size_t size = psc_dma_hardware.buffer_bytes_max;
+ int rc;
+
+ dev_dbg(rtd->platform->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n",
+ card, dai, pcm);
+
+ rc = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ if (rc)
+ return rc;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+ size, &pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer);
+ if (rc)
+ goto playback_alloc_err;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+ size, &pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->dma_buffer);
+ if (rc)
+ goto capture_alloc_err;
+ }
+
+ return 0;
+
+ capture_alloc_err:
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream)
+ snd_dma_free_pages(&pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer);
+
+ playback_alloc_err:
+ dev_err(card->dev, "Cannot allocate buffer(s)\n");
+
+ return -ENOMEM;
+}
+
+static void psc_dma_free(struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ struct snd_pcm_substream *substream;
+ int stream;
+
+ dev_dbg(rtd->platform->dev, "psc_dma_free(pcm=%p)\n", pcm);
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+static struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
+ .ops = &psc_dma_ops,
+ .pcm_new = &psc_dma_new,
+ .pcm_free = &psc_dma_free,
+};
+
+int mpc5200_audio_dma_create(struct platform_device *op)
+{
+ phys_addr_t fifo;
+ struct psc_dma *psc_dma;
+ struct resource res;
+ int size, irq, rc;
+ const __be32 *prop;
+ void __iomem *regs;
+ int ret;
+
+ /* Fetch the registers and IRQ of the PSC */
+ irq = irq_of_parse_and_map(op->dev.of_node, 0);
+ if (of_address_to_resource(op->dev.of_node, 0, &res)) {
+ dev_err(&op->dev, "Missing reg property\n");
+ return -ENODEV;
+ }
+ regs = ioremap(res.start, resource_size(&res));
+ if (!regs) {
+ dev_err(&op->dev, "Could not map registers\n");
+ return -ENODEV;
+ }
+
+ /* Allocate and initialize the driver private data */
+ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
+ if (!psc_dma) {
+ ret = -ENOMEM;
+ goto out_unmap;
+ }
+
+ /* Get the PSC ID */
+ prop = of_get_property(op->dev.of_node, "cell-index", &size);
+ if (!prop || size < sizeof *prop) {
+ ret = -ENODEV;
+ goto out_free;
+ }
+
+ spin_lock_init(&psc_dma->lock);
+ mutex_init(&psc_dma->mutex);
+ psc_dma->id = be32_to_cpu(*prop);
+ psc_dma->irq = irq;
+ psc_dma->psc_regs = regs;
+ psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs;
+ psc_dma->dev = &op->dev;
+ psc_dma->playback.psc_dma = psc_dma;
+ psc_dma->capture.psc_dma = psc_dma;
+ snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id);
+
+ /* Find the address of the fifo data registers and setup the
+ * DMA tasks */
+ fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
+ psc_dma->capture.bcom_task =
+ bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512);
+ psc_dma->playback.bcom_task =
+ bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo);
+ if (!psc_dma->capture.bcom_task ||
+ !psc_dma->playback.bcom_task) {
+ dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
+ ret = -ENODEV;
+ goto out_free;
+ }
+
+ /* Disable all interrupts and reset the PSC */
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+ /* reset receiver */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX);
+ /* reset transmitter */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX);
+ /* reset error */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT);
+ /* reset mode */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1);
+
+ /* Set up mode register;
+ * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
+ * Second write: register Normal mode for non loopback
+ */
+ out_8(&psc_dma->psc_regs->mode, 0);
+ out_8(&psc_dma->psc_regs->mode, 0);
+
+ /* Set the TX and RX fifo alarm thresholds */
+ out_be16(&psc_dma->fifo_regs->rfalarm, 0x100);
+ out_8(&psc_dma->fifo_regs->rfcntl, 0x4);
+ out_be16(&psc_dma->fifo_regs->tfalarm, 0x100);
+ out_8(&psc_dma->fifo_regs->tfcntl, 0x7);
+
+ /* Lookup the IRQ numbers */
+ psc_dma->playback.irq =
+ bcom_get_task_irq(psc_dma->playback.bcom_task);
+ psc_dma->capture.irq =
+ bcom_get_task_irq(psc_dma->capture.bcom_task);
+
+ rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED,
+ "psc-dma-status", psc_dma);
+ rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED,
+ "psc-dma-capture", &psc_dma->capture);
+ rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED,
+ "psc-dma-playback", &psc_dma->playback);
+ if (rc) {
+ ret = -ENODEV;
+ goto out_irq;
+ }
+
+ /* Save what we've done so it can be found again later */
+ dev_set_drvdata(&op->dev, psc_dma);
+
+ /* Tell the ASoC OF helpers about it */
+ return snd_soc_register_platform(&op->dev, &mpc5200_audio_dma_platform);
+out_irq:
+ free_irq(psc_dma->irq, psc_dma);
+ free_irq(psc_dma->capture.irq, &psc_dma->capture);
+ free_irq(psc_dma->playback.irq, &psc_dma->playback);
+out_free:
+ kfree(psc_dma);
+out_unmap:
+ iounmap(regs);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
+
+int mpc5200_audio_dma_destroy(struct platform_device *op)
+{
+ struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
+
+ dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n");
+
+ snd_soc_unregister_platform(&op->dev);
+
+ bcom_gen_bd_rx_release(psc_dma->capture.bcom_task);
+ bcom_gen_bd_tx_release(psc_dma->playback.bcom_task);
+
+ /* Release irqs */
+ free_irq(psc_dma->irq, psc_dma);
+ free_irq(psc_dma->capture.irq, &psc_dma->capture);
+ free_irq(psc_dma->playback.irq, &psc_dma->playback);
+
+ iounmap(psc_dma->psc_regs);
+ kfree(psc_dma);
+ dev_set_drvdata(&op->dev, NULL);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy);
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
new file mode 100644
index 000000000..dff253fde
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -0,0 +1,87 @@
+/*
+ * Freescale MPC5200 Audio DMA driver
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__
+#define __SOUND_SOC_FSL_MPC5200_DMA_H__
+
+#define PSC_STREAM_NAME_LEN 32
+
+/**
+ * psc_ac97_stream - Data specific to a single stream (playback or capture)
+ * @active: flag indicating if the stream is active
+ * @psc_dma: pointer back to parent psc_dma data structure
+ * @bcom_task: bestcomm task structure
+ * @irq: irq number for bestcomm task
+ * @period_end: physical address of end of DMA region
+ * @period_next_pt: physical address of next DMA buffer to enqueue
+ * @period_bytes: size of DMA period in bytes
+ * @ac97_slot_bits: Enable bits for turning on the correct AC97 slot
+ */
+struct psc_dma_stream {
+ struct snd_pcm_runtime *runtime;
+ int active;
+ struct psc_dma *psc_dma;
+ struct bcom_task *bcom_task;
+ int irq;
+ struct snd_pcm_substream *stream;
+ int period_next;
+ int period_current;
+ int period_bytes;
+ int period_count;
+
+ /* AC97 state */
+ u32 ac97_slot_bits;
+};
+
+/**
+ * psc_dma - Private driver data
+ * @name: short name for this device ("PSC0", "PSC1", etc)
+ * @psc_regs: pointer to the PSC's registers
+ * @fifo_regs: pointer to the PSC's FIFO registers
+ * @irq: IRQ of this PSC
+ * @dev: struct device pointer
+ * @dai: the CPU DAI for this device
+ * @sicr: Base value used in serial interface control register; mode is ORed
+ * with this value.
+ * @playback: Playback stream context data
+ * @capture: Capture stream context data
+ */
+struct psc_dma {
+ char name[32];
+ struct mpc52xx_psc __iomem *psc_regs;
+ struct mpc52xx_psc_fifo __iomem *fifo_regs;
+ unsigned int irq;
+ struct device *dev;
+ spinlock_t lock;
+ struct mutex mutex;
+ u32 sicr;
+ uint sysclk;
+ int imr;
+ int id;
+ unsigned int slots;
+
+ /* per-stream data */
+ struct psc_dma_stream playback;
+ struct psc_dma_stream capture;
+
+ /* Statistics */
+ struct {
+ unsigned long overrun_count;
+ unsigned long underrun_count;
+ } stats;
+};
+
+/* Utility for retrieving psc_dma_stream structure from a substream */
+static inline struct psc_dma_stream *
+to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma)
+{
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return &psc_dma->capture;
+ return &psc_dma->playback;
+}
+
+int mpc5200_audio_dma_create(struct platform_device *op);
+int mpc5200_audio_dma_destroy(struct platform_device *op);
+
+#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
new file mode 100644
index 000000000..0bab76051
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -0,0 +1,350 @@
+/*
+ * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
+ *
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/delay.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/time.h>
+#include <asm/delay.h>
+#include <asm/mpc52xx.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+
+#define DRV_NAME "mpc5200-psc-ac97"
+
+/* ALSA only supports a single AC97 device so static is recommend here */
+static struct psc_dma *psc_dma;
+
+static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ int status;
+ unsigned int val;
+
+ mutex_lock(&psc_dma->mutex);
+
+ /* Wait for command send status zero = ready */
+ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_CMDSEND), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 bus (rdy)\n");
+ mutex_unlock(&psc_dma->mutex);
+ return -ENODEV;
+ }
+
+ /* Force clear the data valid bit */
+ in_be32(&psc_dma->psc_regs->ac97_data);
+
+ /* Send the read */
+ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));
+
+ /* Wait for the answer */
+ status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_DATA_VAL), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 read (val) %x\n",
+ in_be16(&psc_dma->psc_regs->sr_csr.status));
+ mutex_unlock(&psc_dma->mutex);
+ return -ENODEV;
+ }
+ /* Get the data */
+ val = in_be32(&psc_dma->psc_regs->ac97_data);
+ if (((val >> 24) & 0x7f) != reg) {
+ pr_err("reg echo error on ac97 read\n");
+ mutex_unlock(&psc_dma->mutex);
+ return -ENODEV;
+ }
+ val = (val >> 8) & 0xffff;
+
+ mutex_unlock(&psc_dma->mutex);
+ return (unsigned short) val;
+}
+
+static void psc_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ int status;
+
+ mutex_lock(&psc_dma->mutex);
+
+ /* Wait for command status zero = ready */
+ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_CMDSEND), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 bus (write)\n");
+ goto out;
+ }
+ /* Write data */
+ out_be32(&psc_dma->psc_regs->ac97_cmd,
+ ((reg & 0x7f) << 24) | (val << 8));
+
+ out:
+ mutex_unlock(&psc_dma->mutex);
+}
+
+static void psc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ mutex_lock(&psc_dma->mutex);
+
+ out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR);
+ udelay(3);
+ out_be32(&regs->sicr, psc_dma->sicr);
+
+ mutex_unlock(&psc_dma->mutex);
+}
+
+static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ mutex_lock(&psc_dma->mutex);
+ dev_dbg(psc_dma->dev, "cold reset\n");
+
+ mpc5200_psc_ac97_gpio_reset(psc_dma->id);
+
+ /* Notify the PSC that a reset has occurred */
+ out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_ACRB);
+
+ /* Re-enable RX and TX */
+ out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
+
+ mutex_unlock(&psc_dma->mutex);
+
+ msleep(1);
+ psc_ac97_warm_reset(ac97);
+}
+
+static struct snd_ac97_bus_ops psc_ac97_ops = {
+ .read = psc_ac97_read,
+ .write = psc_ac97_write,
+ .reset = psc_ac97_cold_reset,
+ .warm_reset = psc_ac97_warm_reset,
+};
+
+static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai);
+ struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
+
+ dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+ " periods=%i buffer_size=%i buffer_bytes=%i channels=%i"
+ " rate=%i format=%i\n",
+ __func__, substream, params_period_size(params),
+ params_period_bytes(params), params_periods(params),
+ params_buffer_size(params), params_buffer_bytes(params),
+ params_channels(params), params_rate(params),
+ params_format(params));
+
+ /* Determine the set of enable bits to turn on */
+ s->ac97_slot_bits = (params_channels(params) == 1) ? 0x100 : 0x300;
+ if (substream->pstr->stream != SNDRV_PCM_STREAM_CAPTURE)
+ s->ac97_slot_bits <<= 16;
+ return 0;
+}
+
+static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai);
+
+ dev_dbg(psc_dma->dev, "%s(substream=%p)\n", __func__, substream);
+
+ if (params_channels(params) == 1)
+ out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000);
+ else
+ out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000);
+
+ return 0;
+}
+
+static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(dai);
+ struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ dev_dbg(psc_dma->dev, "AC97 START: stream=%i\n",
+ substream->pstr->stream);
+
+ /* Set the slot enable bits */
+ psc_dma->slots |= s->ac97_slot_bits;
+ out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ dev_dbg(psc_dma->dev, "AC97 STOP: stream=%i\n",
+ substream->pstr->stream);
+
+ /* Clear the slot enable bits */
+ psc_dma->slots &= ~(s->ac97_slot_bits);
+ out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+ break;
+ }
+ return 0;
+}
+
+static int psc_ac97_probe(struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai);
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ /* Go */
+ out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
+ return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_ac97_dai_template: template CPU Digital Audio Interface
+ */
+static const struct snd_soc_dai_ops psc_ac97_analog_ops = {
+ .hw_params = psc_ac97_hw_analog_params,
+ .trigger = psc_ac97_trigger,
+};
+
+static const struct snd_soc_dai_ops psc_ac97_digital_ops = {
+ .hw_params = psc_ac97_hw_digital_params,
+};
+
+static struct snd_soc_dai_driver psc_ac97_dai[] = {
+{
+ .name = "mpc5200-psc-ac97.0",
+ .bus_control = true,
+ .probe = psc_ac97_probe,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+ },
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+ },
+ .ops = &psc_ac97_analog_ops,
+},
+{
+ .name = "mpc5200-psc-ac97.1",
+ .bus_control = true,
+ .playback = {
+ .stream_name = "AC97 SPDIF",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+ },
+ .ops = &psc_ac97_digital_ops,
+} };
+
+static const struct snd_soc_component_driver psc_ac97_component = {
+ .name = DRV_NAME,
+};
+
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int psc_ac97_of_probe(struct platform_device *op)
+{
+ int rc;
+ struct mpc52xx_psc __iomem *regs;
+
+ rc = mpc5200_audio_dma_create(op);
+ if (rc != 0)
+ return rc;
+
+ rc = snd_soc_set_ac97_ops(&psc_ac97_ops);
+ if (rc != 0) {
+ dev_err(&op->dev, "Failed to set AC'97 ops: %d\n", rc);
+ return rc;
+ }
+
+ rc = snd_soc_register_component(&op->dev, &psc_ac97_component,
+ psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
+ if (rc != 0) {
+ dev_err(&op->dev, "Failed to register DAI\n");
+ return rc;
+ }
+
+ psc_dma = dev_get_drvdata(&op->dev);
+ regs = psc_dma->psc_regs;
+
+ psc_dma->imr = 0;
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+
+ /* Configure the serial interface mode to AC97 */
+ psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97;
+ out_be32(&regs->sicr, psc_dma->sicr);
+
+ /* No slots active */
+ out_be32(&regs->ac97_slots, 0x00000000);
+
+ return 0;
+}
+
+static int psc_ac97_of_remove(struct platform_device *op)
+{
+ mpc5200_audio_dma_destroy(op);
+ snd_soc_unregister_component(&op->dev);
+ snd_soc_set_ac97_ops(NULL);
+ return 0;
+}
+
+/* Match table for of_platform binding */
+static const struct of_device_id psc_ac97_match[] = {
+ { .compatible = "fsl,mpc5200-psc-ac97", },
+ { .compatible = "fsl,mpc5200b-psc-ac97", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, psc_ac97_match);
+
+static struct platform_driver psc_ac97_driver = {
+ .probe = psc_ac97_of_probe,
+ .remove = psc_ac97_of_remove,
+ .driver = {
+ .name = "mpc5200-psc-ac97",
+ .of_match_table = psc_ac97_match,
+ },
+};
+
+module_platform_driver(psc_ac97_driver);
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION("mpc5200 AC97 module");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h
new file mode 100644
index 000000000..e881e784b
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.h
@@ -0,0 +1,13 @@
+/*
+ * Freescale MPC5200 PSC in AC97 mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+
+#define MPC5200_AC97_NORMAL 0
+#define MPC5200_AC97_SPDIF 1
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
new file mode 100644
index 000000000..d8232943c
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -0,0 +1,241 @@
+/*
+ * Freescale MPC5200 PSC in I2S mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+
+/**
+ * PSC_I2S_RATES: sample rates supported by the I2S
+ *
+ * This driver currently only supports the PSC running in I2S slave mode,
+ * which means the codec determines the sample rate. Therefore, we tell
+ * ALSA that we support all rates and let the codec driver decide what rates
+ * are really supported.
+ */
+#define PSC_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS
+
+/**
+ * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
+ */
+#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
+
+static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ u32 mode;
+
+ dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+ " periods=%i buffer_size=%i buffer_bytes=%i\n",
+ __func__, substream, params_period_size(params),
+ params_period_bytes(params), params_periods(params),
+ params_buffer_size(params), params_buffer_bytes(params));
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ mode = MPC52xx_PSC_SICR_SIM_CODEC_8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_BE:
+ mode = MPC52xx_PSC_SICR_SIM_CODEC_16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_BE:
+ mode = MPC52xx_PSC_SICR_SIM_CODEC_24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_BE:
+ mode = MPC52xx_PSC_SICR_SIM_CODEC_32;
+ break;
+ default:
+ dev_dbg(psc_dma->dev, "invalid format\n");
+ return -EINVAL;
+ }
+ out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode);
+
+ return 0;
+}
+
+/**
+ * psc_i2s_set_sysclk: set the clock frequency and direction
+ *
+ * This function is called by the machine driver to tell us what the clock
+ * frequency and direction are.
+ *
+ * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
+ * and we don't care about the frequency. Return an error if the direction
+ * is not SND_SOC_CLOCK_IN.
+ *
+ * @clk_id: reserved, should be zero
+ * @freq: the frequency of the given clock ID, currently ignored
+ * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
+ */
+static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai);
+ dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
+ cpu_dai, dir);
+ return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
+}
+
+/**
+ * psc_i2s_set_fmt: set the serial format.
+ *
+ * This function is called by the machine driver to tell us what serial
+ * format to use.
+ *
+ * This driver only supports I2S mode. Return an error if the format is
+ * not SND_SOC_DAIFMT_I2S.
+ *
+ * @format: one of SND_SOC_DAIFMT_xxx
+ */
+static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
+{
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai);
+ dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
+ cpu_dai, format);
+ return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_i2s_dai_template: template CPU Digital Audio Interface
+ */
+static const struct snd_soc_dai_ops psc_i2s_dai_ops = {
+ .hw_params = psc_i2s_hw_params,
+ .set_sysclk = psc_i2s_set_sysclk,
+ .set_fmt = psc_i2s_set_fmt,
+};
+
+static struct snd_soc_dai_driver psc_i2s_dai[] = {{
+ .name = "mpc5200-psc-i2s.0",
+ .playback = {
+ .stream_name = "I2S Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PSC_I2S_RATES,
+ .formats = PSC_I2S_FORMATS,
+ },
+ .capture = {
+ .stream_name = "I2S Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PSC_I2S_RATES,
+ .formats = PSC_I2S_FORMATS,
+ },
+ .ops = &psc_i2s_dai_ops,
+} };
+
+static const struct snd_soc_component_driver psc_i2s_component = {
+ .name = "mpc5200-i2s",
+};
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int psc_i2s_of_probe(struct platform_device *op)
+{
+ int rc;
+ struct psc_dma *psc_dma;
+ struct mpc52xx_psc __iomem *regs;
+
+ rc = mpc5200_audio_dma_create(op);
+ if (rc != 0)
+ return rc;
+
+ rc = snd_soc_register_component(&op->dev, &psc_i2s_component,
+ psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
+ if (rc != 0) {
+ pr_err("Failed to register DAI\n");
+ return rc;
+ }
+
+ psc_dma = dev_get_drvdata(&op->dev);
+ regs = psc_dma->psc_regs;
+
+ /* Configure the serial interface mode; defaulting to CODEC8 mode */
+ psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
+ MPC52xx_PSC_SICR_CLKPOL;
+ out_be32(&psc_dma->psc_regs->sicr,
+ psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
+
+ /* Check for the codec handle. If it is not present then we
+ * are done */
+ if (!of_get_property(op->dev.of_node, "codec-handle", NULL))
+ return 0;
+
+ /* Due to errata in the dma mode; need to line up enabling
+ * the transmitter with a transition on the frame sync
+ * line */
+
+ /* first make sure it is low */
+ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
+ ;
+ /* then wait for the transition to high */
+ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
+ ;
+ /* Finally, enable the PSC.
+ * Receiver must always be enabled; even when we only want
+ * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */
+
+ /* Go */
+ out_8(&psc_dma->psc_regs->command,
+ MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
+
+ return 0;
+
+}
+
+static int psc_i2s_of_remove(struct platform_device *op)
+{
+ mpc5200_audio_dma_destroy(op);
+ snd_soc_unregister_component(&op->dev);
+ return 0;
+}
+
+/* Match table for of_platform binding */
+static const struct of_device_id psc_i2s_match[] = {
+ { .compatible = "fsl,mpc5200-psc-i2s", },
+ { .compatible = "fsl,mpc5200b-psc-i2s", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, psc_i2s_match);
+
+static struct platform_driver psc_i2s_driver = {
+ .probe = psc_i2s_of_probe,
+ .remove = psc_i2s_of_remove,
+ .driver = {
+ .name = "mpc5200-psc-i2s",
+ .of_match_table = psc_i2s_match,
+ },
+};
+
+module_platform_driver(psc_i2s_driver);
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
new file mode 100644
index 000000000..9621b9140
--- /dev/null
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -0,0 +1,433 @@
+/**
+ * Freescale MPC8610HPCD ALSA SoC Machine driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007-2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/of_address.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <asm/fsl_guts.h>
+
+#include "fsl_dma.h"
+#include "fsl_ssi.h"
+#include "fsl_utils.h"
+
+/* There's only one global utilities register */
+static phys_addr_t guts_phys;
+
+/**
+ * mpc8610_hpcd_data: machine-specific ASoC device data
+ *
+ * This structure contains data for a single sound platform device on an
+ * MPC8610 HPCD. Some of the data is taken from the device tree.
+ */
+struct mpc8610_hpcd_data {
+ struct snd_soc_dai_link dai[2];
+ struct snd_soc_card card;
+ unsigned int dai_format;
+ unsigned int codec_clk_direction;
+ unsigned int cpu_clk_direction;
+ unsigned int clk_frequency;
+ unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */
+ unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */
+ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
+ char codec_dai_name[DAI_NAME_SIZE];
+ char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
+};
+
+/**
+ * mpc8610_hpcd_machine_probe: initialize the board
+ *
+ * This function is used to initialize the board-specific hardware.
+ *
+ * Here we program the DMACR and PMUXCR registers.
+ */
+static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card)
+{
+ struct mpc8610_hpcd_data *machine_data =
+ container_of(card, struct mpc8610_hpcd_data, card);
+ struct ccsr_guts __iomem *guts;
+
+ guts = ioremap(guts_phys, sizeof(struct ccsr_guts));
+ if (!guts) {
+ dev_err(card->dev, "could not map global utilities\n");
+ return -ENOMEM;
+ }
+
+ /* Program the signal routing between the SSI and the DMA */
+ guts_set_dmacr(guts, machine_data->dma_id[0],
+ machine_data->dma_channel_id[0],
+ CCSR_GUTS_DMACR_DEV_SSI);
+ guts_set_dmacr(guts, machine_data->dma_id[1],
+ machine_data->dma_channel_id[1],
+ CCSR_GUTS_DMACR_DEV_SSI);
+
+ guts_set_pmuxcr_dma(guts, machine_data->dma_id[0],
+ machine_data->dma_channel_id[0], 0);
+ guts_set_pmuxcr_dma(guts, machine_data->dma_id[1],
+ machine_data->dma_channel_id[1], 0);
+
+ switch (machine_data->ssi_id) {
+ case 0:
+ clrsetbits_be32(&guts->pmuxcr,
+ CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI);
+ break;
+ case 1:
+ clrsetbits_be32(&guts->pmuxcr,
+ CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI);
+ break;
+ }
+
+ iounmap(guts);
+
+ return 0;
+}
+
+/**
+ * mpc8610_hpcd_startup: program the board with various hardware parameters
+ *
+ * This function takes board-specific information, like clock frequencies
+ * and serial data formats, and passes that information to the codec and
+ * transport drivers.
+ */
+static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct mpc8610_hpcd_data *machine_data =
+ container_of(rtd->card, struct mpc8610_hpcd_data, card);
+ struct device *dev = rtd->card->dev;
+ int ret = 0;
+
+ /* Tell the codec driver what the serial protocol is. */
+ ret = snd_soc_dai_set_fmt(rtd->codec_dai, machine_data->dai_format);
+ if (ret < 0) {
+ dev_err(dev, "could not set codec driver audio format\n");
+ return ret;
+ }
+
+ /*
+ * Tell the codec driver what the MCLK frequency is, and whether it's
+ * a slave or master.
+ */
+ ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0,
+ machine_data->clk_frequency,
+ machine_data->codec_clk_direction);
+ if (ret < 0) {
+ dev_err(dev, "could not set codec driver clock params\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+/**
+ * mpc8610_hpcd_machine_remove: Remove the sound device
+ *
+ * This function is called to remove the sound device for one SSI. We
+ * de-program the DMACR and PMUXCR register.
+ */
+static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card)
+{
+ struct mpc8610_hpcd_data *machine_data =
+ container_of(card, struct mpc8610_hpcd_data, card);
+ struct ccsr_guts __iomem *guts;
+
+ guts = ioremap(guts_phys, sizeof(struct ccsr_guts));
+ if (!guts) {
+ dev_err(card->dev, "could not map global utilities\n");
+ return -ENOMEM;
+ }
+
+ /* Restore the signal routing */
+
+ guts_set_dmacr(guts, machine_data->dma_id[0],
+ machine_data->dma_channel_id[0], 0);
+ guts_set_dmacr(guts, machine_data->dma_id[1],
+ machine_data->dma_channel_id[1], 0);
+
+ switch (machine_data->ssi_id) {
+ case 0:
+ clrsetbits_be32(&guts->pmuxcr,
+ CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA);
+ break;
+ case 1:
+ clrsetbits_be32(&guts->pmuxcr,
+ CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA);
+ break;
+ }
+
+ iounmap(guts);
+
+ return 0;
+}
+
+/**
+ * mpc8610_hpcd_ops: ASoC machine driver operations
+ */
+static struct snd_soc_ops mpc8610_hpcd_ops = {
+ .startup = mpc8610_hpcd_startup,
+};
+
+/**
+ * mpc8610_hpcd_probe: platform probe function for the machine driver
+ *
+ * Although this is a machine driver, the SSI node is the "master" node with
+ * respect to audio hardware connections. Therefore, we create a new ASoC
+ * device for each new SSI node that has a codec attached.
+ */
+static int mpc8610_hpcd_probe(struct platform_device *pdev)
+{
+ struct device *dev = pdev->dev.parent;
+ /* ssi_pdev is the platform device for the SSI node that probed us */
+ struct platform_device *ssi_pdev =
+ container_of(dev, struct platform_device, dev);
+ struct device_node *np = ssi_pdev->dev.of_node;
+ struct device_node *codec_np = NULL;
+ struct mpc8610_hpcd_data *machine_data;
+ int ret = -ENODEV;
+ const char *sprop;
+ const u32 *iprop;
+
+ /* Find the codec node for this SSI. */
+ codec_np = of_parse_phandle(np, "codec-handle", 0);
+ if (!codec_np) {
+ dev_err(dev, "invalid codec node\n");
+ return -EINVAL;
+ }
+
+ machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
+ if (!machine_data) {
+ ret = -ENOMEM;
+ goto error_alloc;
+ }
+
+ machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
+ machine_data->dai[0].ops = &mpc8610_hpcd_ops;
+
+ /* ASoC core can match codec with device node */
+ machine_data->dai[0].codec_of_node = codec_np;
+
+ /* The DAI name from the codec (snd_soc_dai_driver.name) */
+ machine_data->dai[0].codec_dai_name = "cs4270-hifi";
+
+ /* We register two DAIs per SSI, one for playback and the other for
+ * capture. Currently, we only support codecs that have one DAI for
+ * both playback and capture.
+ */
+ memcpy(&machine_data->dai[1], &machine_data->dai[0],
+ sizeof(struct snd_soc_dai_link));
+
+ /* Get the device ID */
+ iprop = of_get_property(np, "cell-index", NULL);
+ if (!iprop) {
+ dev_err(&pdev->dev, "cell-index property not found\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ machine_data->ssi_id = be32_to_cpup(iprop);
+
+ /* Get the serial format and clock direction. */
+ sprop = of_get_property(np, "fsl,mode", NULL);
+ if (!sprop) {
+ dev_err(&pdev->dev, "fsl,mode property not found\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ if (strcasecmp(sprop, "i2s-slave") == 0) {
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
+
+ /* In i2s-slave mode, the codec has its own clock source, so we
+ * need to get the frequency from the device tree and pass it to
+ * the codec driver.
+ */
+ iprop = of_get_property(codec_np, "clock-frequency", NULL);
+ if (!iprop || !*iprop) {
+ dev_err(&pdev->dev, "codec bus-frequency "
+ "property is missing or invalid\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ machine_data->clk_frequency = be32_to_cpup(iprop);
+ } else if (strcasecmp(sprop, "i2s-master") == 0) {
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "lj-slave") == 0) {
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "lj-master") == 0) {
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "rj-slave") == 0) {
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "rj-master") == 0) {
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "ac97-slave") == 0) {
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "ac97-master") == 0) {
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else {
+ dev_err(&pdev->dev,
+ "unrecognized fsl,mode property '%s'\n", sprop);
+ ret = -EINVAL;
+ goto error;
+ }
+
+ if (!machine_data->clk_frequency) {
+ dev_err(&pdev->dev, "unknown clock frequency\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ /* Find the playback DMA channel to use. */
+ machine_data->dai[0].platform_name = machine_data->platform_name[0];
+ ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma",
+ &machine_data->dai[0],
+ &machine_data->dma_channel_id[0],
+ &machine_data->dma_id[0]);
+ if (ret) {
+ dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n");
+ goto error;
+ }
+
+ /* Find the capture DMA channel to use. */
+ machine_data->dai[1].platform_name = machine_data->platform_name[1];
+ ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma",
+ &machine_data->dai[1],
+ &machine_data->dma_channel_id[1],
+ &machine_data->dma_id[1]);
+ if (ret) {
+ dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n");
+ goto error;
+ }
+
+ /* Initialize our DAI data structure. */
+ machine_data->dai[0].stream_name = "playback";
+ machine_data->dai[1].stream_name = "capture";
+ machine_data->dai[0].name = machine_data->dai[0].stream_name;
+ machine_data->dai[1].name = machine_data->dai[1].stream_name;
+
+ machine_data->card.probe = mpc8610_hpcd_machine_probe;
+ machine_data->card.remove = mpc8610_hpcd_machine_remove;
+ machine_data->card.name = pdev->name; /* The platform driver name */
+ machine_data->card.owner = THIS_MODULE;
+ machine_data->card.dev = &pdev->dev;
+ machine_data->card.num_links = 2;
+ machine_data->card.dai_link = machine_data->dai;
+
+ /* Register with ASoC */
+ ret = snd_soc_register_card(&machine_data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "could not register card\n");
+ goto error;
+ }
+
+ of_node_put(codec_np);
+
+ return 0;
+
+error:
+ kfree(machine_data);
+error_alloc:
+ of_node_put(codec_np);
+ return ret;
+}
+
+/**
+ * mpc8610_hpcd_remove: remove the platform device
+ *
+ * This function is called when the platform device is removed.
+ */
+static int mpc8610_hpcd_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct mpc8610_hpcd_data *machine_data =
+ container_of(card, struct mpc8610_hpcd_data, card);
+
+ snd_soc_unregister_card(card);
+ kfree(machine_data);
+
+ return 0;
+}
+
+static struct platform_driver mpc8610_hpcd_driver = {
+ .probe = mpc8610_hpcd_probe,
+ .remove = mpc8610_hpcd_remove,
+ .driver = {
+ /* The name must match 'compatible' property in the device tree,
+ * in lowercase letters.
+ */
+ .name = "snd-soc-mpc8610hpcd",
+ },
+};
+
+/**
+ * mpc8610_hpcd_init: machine driver initialization.
+ *
+ * This function is called when this module is loaded.
+ */
+static int __init mpc8610_hpcd_init(void)
+{
+ struct device_node *guts_np;
+ struct resource res;
+
+ pr_info("Freescale MPC8610 HPCD ALSA SoC machine driver\n");
+
+ /* Get the physical address of the global utilities registers */
+ guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts");
+ if (of_address_to_resource(guts_np, 0, &res)) {
+ pr_err("mpc8610-hpcd: missing/invalid global utilities node\n");
+ return -EINVAL;
+ }
+ guts_phys = res.start;
+
+ return platform_driver_register(&mpc8610_hpcd_driver);
+}
+
+/**
+ * mpc8610_hpcd_exit: machine driver exit
+ *
+ * This function is called when this driver is unloaded.
+ */
+static void __exit mpc8610_hpcd_exit(void)
+{
+ platform_driver_unregister(&mpc8610_hpcd_driver);
+}
+
+module_init(mpc8610_hpcd_init);
+module_exit(mpc8610_hpcd_exit);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC machine driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
new file mode 100644
index 000000000..198eeb3f3
--- /dev/null
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -0,0 +1,234 @@
+/*
+ * mx27vis-aic32x4.c
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/platform_data/asoc-mx27vis.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/tlv320aic32x4.h"
+#include "imx-ssi.h"
+#include "imx-audmux.h"
+
+#define MX27VIS_AMP_GAIN 0
+#define MX27VIS_AMP_MUTE 1
+
+static int mx27vis_amp_gain;
+static int mx27vis_amp_mute;
+static int mx27vis_amp_gain0_gpio;
+static int mx27vis_amp_gain1_gpio;
+static int mx27vis_amp_mutel_gpio;
+static int mx27vis_amp_muter_gpio;
+
+static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+ 25000000, SND_SOC_CLOCK_OUT);
+ if (ret) {
+ pr_err("%s: failed setting codec sysclk\n", __func__);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret) {
+ pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops mx27vis_aic32x4_snd_ops = {
+ .hw_params = mx27vis_aic32x4_hw_params,
+};
+
+static int mx27vis_amp_set(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int value = ucontrol->value.integer.value[0];
+ unsigned int reg = mc->reg;
+ int max = mc->max;
+
+ if (value > max)
+ return -EINVAL;
+
+ switch (reg) {
+ case MX27VIS_AMP_GAIN:
+ gpio_set_value(mx27vis_amp_gain0_gpio, value & 1);
+ gpio_set_value(mx27vis_amp_gain1_gpio, value >> 1);
+ mx27vis_amp_gain = value;
+ break;
+ case MX27VIS_AMP_MUTE:
+ gpio_set_value(mx27vis_amp_mutel_gpio, value & 1);
+ gpio_set_value(mx27vis_amp_muter_gpio, value >> 1);
+ mx27vis_amp_mute = value;
+ break;
+ }
+ return 0;
+}
+
+static int mx27vis_amp_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+
+ switch (reg) {
+ case MX27VIS_AMP_GAIN:
+ ucontrol->value.integer.value[0] = mx27vis_amp_gain;
+ break;
+ case MX27VIS_AMP_MUTE:
+ ucontrol->value.integer.value[0] = mx27vis_amp_mute;
+ break;
+ }
+ return 0;
+}
+
+/* From 6dB to 24dB in steps of 6dB */
+static const DECLARE_TLV_DB_SCALE(mx27vis_amp_tlv, 600, 600, 0);
+
+static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = {
+ SOC_DAPM_PIN_SWITCH("External Mic"),
+ SOC_SINGLE_EXT_TLV("LO Ext Boost", MX27VIS_AMP_GAIN, 0, 3, 0,
+ mx27vis_amp_get, mx27vis_amp_set, mx27vis_amp_tlv),
+ SOC_DOUBLE_EXT("LO Ext Mute Switch", MX27VIS_AMP_MUTE, 0, 1, 1, 0,
+ mx27vis_amp_get, mx27vis_amp_set),
+};
+
+static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("External Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
+ {"Mic Bias", NULL, "External Mic"},
+ {"IN1_R", NULL, "Mic Bias"},
+ {"IN2_R", NULL, "Mic Bias"},
+ {"IN3_R", NULL, "Mic Bias"},
+ {"IN1_L", NULL, "Mic Bias"},
+ {"IN2_L", NULL, "Mic Bias"},
+ {"IN3_L", NULL, "Mic Bias"},
+};
+
+static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
+ .name = "tlv320aic32x4",
+ .stream_name = "TLV320AIC32X4",
+ .codec_dai_name = "tlv320aic32x4-hifi",
+ .platform_name = "imx-ssi.0",
+ .codec_name = "tlv320aic32x4.0-0018",
+ .cpu_dai_name = "imx-ssi.0",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .ops = &mx27vis_aic32x4_snd_ops,
+};
+
+static struct snd_soc_card mx27vis_aic32x4 = {
+ .name = "visstrim_m10-audio",
+ .owner = THIS_MODULE,
+ .dai_link = &mx27vis_aic32x4_dai,
+ .num_links = 1,
+ .controls = mx27vis_aic32x4_controls,
+ .num_controls = ARRAY_SIZE(mx27vis_aic32x4_controls),
+ .dapm_widgets = aic32x4_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets),
+ .dapm_routes = aic32x4_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
+};
+
+static int mx27vis_aic32x4_probe(struct platform_device *pdev)
+{
+ struct snd_mx27vis_platform_data *pdata = pdev->dev.platform_data;
+ int ret;
+
+ if (!pdata) {
+ dev_err(&pdev->dev, "No platform data supplied\n");
+ return -EINVAL;
+ }
+
+ mx27vis_amp_gain0_gpio = pdata->amp_gain0_gpio;
+ mx27vis_amp_gain1_gpio = pdata->amp_gain1_gpio;
+ mx27vis_amp_mutel_gpio = pdata->amp_mutel_gpio;
+ mx27vis_amp_muter_gpio = pdata->amp_muter_gpio;
+
+ mx27vis_aic32x4.dev = &pdev->dev;
+ ret = snd_soc_register_card(&mx27vis_aic32x4);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+ ret);
+ return ret;
+ }
+
+ /* Connect SSI0 as clock slave to SSI1 external pins */
+ imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+ IMX_AUDMUX_V1_PCR_SYN |
+ IMX_AUDMUX_V1_PCR_TFSDIR |
+ IMX_AUDMUX_V1_PCR_TCLKDIR |
+ IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) |
+ IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1)
+ );
+ imx_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1,
+ IMX_AUDMUX_V1_PCR_SYN |
+ IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+ );
+
+ return ret;
+}
+
+static int mx27vis_aic32x4_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&mx27vis_aic32x4);
+
+ return 0;
+}
+
+static struct platform_driver mx27vis_aic32x4_audio_driver = {
+ .driver = {
+ .name = "mx27vis",
+ },
+ .probe = mx27vis_aic32x4_probe,
+ .remove = mx27vis_aic32x4_remove,
+};
+
+module_platform_driver(mx27vis_aic32x4_audio_driver);
+
+MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
+MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mx27vis");
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
new file mode 100644
index 000000000..71c1a7dc3
--- /dev/null
+++ b/sound/soc/fsl/p1022_ds.c
@@ -0,0 +1,442 @@
+/**
+ * Freescale P1022DS ALSA SoC Machine driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/of_address.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <asm/fsl_guts.h>
+
+#include "fsl_dma.h"
+#include "fsl_ssi.h"
+#include "fsl_utils.h"
+
+/* P1022-specific PMUXCR and DMUXCR bit definitions */
+
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_MASK 0x0001c000
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI 0x00010000
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_SSI 0x00018000
+
+#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK 0x00000c00
+#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI 0x00000000
+
+#define CCSR_GUTS_DMUXCR_PAD 1 /* DMA controller/channel set to pad */
+#define CCSR_GUTS_DMUXCR_SSI 2 /* DMA controller/channel set to SSI */
+
+/*
+ * Set the DMACR register in the GUTS
+ *
+ * The DMACR register determines the source of initiated transfers for each
+ * channel on each DMA controller. Rather than have a bunch of repetitive
+ * macros for the bit patterns, we just have a function that calculates
+ * them.
+ *
+ * guts: Pointer to GUTS structure
+ * co: The DMA controller (0 or 1)
+ * ch: The channel on the DMA controller (0, 1, 2, or 3)
+ * device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx)
+ */
+static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts,
+ unsigned int co, unsigned int ch, unsigned int device)
+{
+ unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch));
+
+ clrsetbits_be32(&guts->dmuxcr, 3 << shift, device << shift);
+}
+
+/* There's only one global utilities register */
+static phys_addr_t guts_phys;
+
+/**
+ * machine_data: machine-specific ASoC device data
+ *
+ * This structure contains data for a single sound platform device on an
+ * P1022 DS. Some of the data is taken from the device tree.
+ */
+struct machine_data {
+ struct snd_soc_dai_link dai[2];
+ struct snd_soc_card card;
+ unsigned int dai_format;
+ unsigned int codec_clk_direction;
+ unsigned int cpu_clk_direction;
+ unsigned int clk_frequency;
+ unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */
+ unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */
+ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
+ char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
+};
+
+/**
+ * p1022_ds_machine_probe: initialize the board
+ *
+ * This function is used to initialize the board-specific hardware.
+ *
+ * Here we program the DMACR and PMUXCR registers.
+ */
+static int p1022_ds_machine_probe(struct snd_soc_card *card)
+{
+ struct machine_data *mdata =
+ container_of(card, struct machine_data, card);
+ struct ccsr_guts __iomem *guts;
+
+ guts = ioremap(guts_phys, sizeof(struct ccsr_guts));
+ if (!guts) {
+ dev_err(card->dev, "could not map global utilities\n");
+ return -ENOMEM;
+ }
+
+ /* Enable SSI Tx signal */
+ clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK,
+ CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI);
+
+ /* Enable SSI Rx signal */
+ clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK,
+ CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI);
+
+ /* Enable DMA Channel for SSI */
+ guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0],
+ CCSR_GUTS_DMUXCR_SSI);
+
+ guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1],
+ CCSR_GUTS_DMUXCR_SSI);
+
+ iounmap(guts);
+
+ return 0;
+}
+
+/**
+ * p1022_ds_startup: program the board with various hardware parameters
+ *
+ * This function takes board-specific information, like clock frequencies
+ * and serial data formats, and passes that information to the codec and
+ * transport drivers.
+ */
+static int p1022_ds_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct machine_data *mdata =
+ container_of(rtd->card, struct machine_data, card);
+ struct device *dev = rtd->card->dev;
+ int ret = 0;
+
+ /* Tell the codec driver what the serial protocol is. */
+ ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format);
+ if (ret < 0) {
+ dev_err(dev, "could not set codec driver audio format\n");
+ return ret;
+ }
+
+ /*
+ * Tell the codec driver what the MCLK frequency is, and whether it's
+ * a slave or master.
+ */
+ ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency,
+ mdata->codec_clk_direction);
+ if (ret < 0) {
+ dev_err(dev, "could not set codec driver clock params\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+/**
+ * p1022_ds_machine_remove: Remove the sound device
+ *
+ * This function is called to remove the sound device for one SSI. We
+ * de-program the DMACR and PMUXCR register.
+ */
+static int p1022_ds_machine_remove(struct snd_soc_card *card)
+{
+ struct machine_data *mdata =
+ container_of(card, struct machine_data, card);
+ struct ccsr_guts __iomem *guts;
+
+ guts = ioremap(guts_phys, sizeof(struct ccsr_guts));
+ if (!guts) {
+ dev_err(card->dev, "could not map global utilities\n");
+ return -ENOMEM;
+ }
+
+ /* Restore the signal routing */
+ clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK);
+ clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK);
+ guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], 0);
+ guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], 0);
+
+ iounmap(guts);
+
+ return 0;
+}
+
+/**
+ * p1022_ds_ops: ASoC machine driver operations
+ */
+static struct snd_soc_ops p1022_ds_ops = {
+ .startup = p1022_ds_startup,
+};
+
+/**
+ * p1022_ds_probe: platform probe function for the machine driver
+ *
+ * Although this is a machine driver, the SSI node is the "master" node with
+ * respect to audio hardware connections. Therefore, we create a new ASoC
+ * device for each new SSI node that has a codec attached.
+ */
+static int p1022_ds_probe(struct platform_device *pdev)
+{
+ struct device *dev = pdev->dev.parent;
+ /* ssi_pdev is the platform device for the SSI node that probed us */
+ struct platform_device *ssi_pdev =
+ container_of(dev, struct platform_device, dev);
+ struct device_node *np = ssi_pdev->dev.of_node;
+ struct device_node *codec_np = NULL;
+ struct machine_data *mdata;
+ int ret = -ENODEV;
+ const char *sprop;
+ const u32 *iprop;
+
+ /* Find the codec node for this SSI. */
+ codec_np = of_parse_phandle(np, "codec-handle", 0);
+ if (!codec_np) {
+ dev_err(dev, "could not find codec node\n");
+ return -EINVAL;
+ }
+
+ mdata = kzalloc(sizeof(struct machine_data), GFP_KERNEL);
+ if (!mdata) {
+ ret = -ENOMEM;
+ goto error_put;
+ }
+
+ mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
+ mdata->dai[0].ops = &p1022_ds_ops;
+
+ /* ASoC core can match codec with device node */
+ mdata->dai[0].codec_of_node = codec_np;
+
+ /* We register two DAIs per SSI, one for playback and the other for
+ * capture. We support codecs that have separate DAIs for both playback
+ * and capture.
+ */
+ memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link));
+
+ /* The DAI names from the codec (snd_soc_dai_driver.name) */
+ mdata->dai[0].codec_dai_name = "wm8776-hifi-playback";
+ mdata->dai[1].codec_dai_name = "wm8776-hifi-capture";
+
+ /* Get the device ID */
+ iprop = of_get_property(np, "cell-index", NULL);
+ if (!iprop) {
+ dev_err(&pdev->dev, "cell-index property not found\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ mdata->ssi_id = be32_to_cpup(iprop);
+
+ /* Get the serial format and clock direction. */
+ sprop = of_get_property(np, "fsl,mode", NULL);
+ if (!sprop) {
+ dev_err(&pdev->dev, "fsl,mode property not found\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ if (strcasecmp(sprop, "i2s-slave") == 0) {
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+
+ /* In i2s-slave mode, the codec has its own clock source, so we
+ * need to get the frequency from the device tree and pass it to
+ * the codec driver.
+ */
+ iprop = of_get_property(codec_np, "clock-frequency", NULL);
+ if (!iprop || !*iprop) {
+ dev_err(&pdev->dev, "codec bus-frequency "
+ "property is missing or invalid\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ mdata->clk_frequency = be32_to_cpup(iprop);
+ } else if (strcasecmp(sprop, "i2s-master") == 0) {
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "lj-slave") == 0) {
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "lj-master") == 0) {
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "rj-slave") == 0) {
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "rj-master") == 0) {
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "ac97-slave") == 0) {
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "ac97-master") == 0) {
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else {
+ dev_err(&pdev->dev,
+ "unrecognized fsl,mode property '%s'\n", sprop);
+ ret = -EINVAL;
+ goto error;
+ }
+
+ if (!mdata->clk_frequency) {
+ dev_err(&pdev->dev, "unknown clock frequency\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ /* Find the playback DMA channel to use. */
+ mdata->dai[0].platform_name = mdata->platform_name[0];
+ ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
+ &mdata->dma_channel_id[0],
+ &mdata->dma_id[0]);
+ if (ret) {
+ dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n");
+ goto error;
+ }
+
+ /* Find the capture DMA channel to use. */
+ mdata->dai[1].platform_name = mdata->platform_name[1];
+ ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
+ &mdata->dma_channel_id[1],
+ &mdata->dma_id[1]);
+ if (ret) {
+ dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n");
+ goto error;
+ }
+
+ /* Initialize our DAI data structure. */
+ mdata->dai[0].stream_name = "playback";
+ mdata->dai[1].stream_name = "capture";
+ mdata->dai[0].name = mdata->dai[0].stream_name;
+ mdata->dai[1].name = mdata->dai[1].stream_name;
+
+ mdata->card.probe = p1022_ds_machine_probe;
+ mdata->card.remove = p1022_ds_machine_remove;
+ mdata->card.name = pdev->name; /* The platform driver name */
+ mdata->card.owner = THIS_MODULE;
+ mdata->card.dev = &pdev->dev;
+ mdata->card.num_links = 2;
+ mdata->card.dai_link = mdata->dai;
+
+ /* Register with ASoC */
+ ret = snd_soc_register_card(&mdata->card);
+ if (ret) {
+ dev_err(&pdev->dev, "could not register card\n");
+ goto error;
+ }
+
+ of_node_put(codec_np);
+
+ return 0;
+
+error:
+ kfree(mdata);
+error_put:
+ of_node_put(codec_np);
+ return ret;
+}
+
+/**
+ * p1022_ds_remove: remove the platform device
+ *
+ * This function is called when the platform device is removed.
+ */
+static int p1022_ds_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct machine_data *mdata =
+ container_of(card, struct machine_data, card);
+
+ snd_soc_unregister_card(card);
+ kfree(mdata);
+
+ return 0;
+}
+
+static struct platform_driver p1022_ds_driver = {
+ .probe = p1022_ds_probe,
+ .remove = p1022_ds_remove,
+ .driver = {
+ /*
+ * The name must match 'compatible' property in the device tree,
+ * in lowercase letters.
+ */
+ .name = "snd-soc-p1022ds",
+ },
+};
+
+/**
+ * p1022_ds_init: machine driver initialization.
+ *
+ * This function is called when this module is loaded.
+ */
+static int __init p1022_ds_init(void)
+{
+ struct device_node *guts_np;
+ struct resource res;
+
+ /* Get the physical address of the global utilities registers */
+ guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts");
+ if (of_address_to_resource(guts_np, 0, &res)) {
+ pr_err("snd-soc-p1022ds: missing/invalid global utils node\n");
+ of_node_put(guts_np);
+ return -EINVAL;
+ }
+ guts_phys = res.start;
+ of_node_put(guts_np);
+
+ return platform_driver_register(&p1022_ds_driver);
+}
+
+/**
+ * p1022_ds_exit: machine driver exit
+ *
+ * This function is called when this driver is unloaded.
+ */
+static void __exit p1022_ds_exit(void)
+{
+ platform_driver_unregister(&p1022_ds_driver);
+}
+
+module_init(p1022_ds_init);
+module_exit(p1022_ds_exit);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale P1022 DS ALSA SoC machine driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c
new file mode 100644
index 000000000..ee2904842
--- /dev/null
+++ b/sound/soc/fsl/p1022_rdk.c
@@ -0,0 +1,392 @@
+/**
+ * Freescale P1022RDK ALSA SoC Machine driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ *
+ * Note: in order for audio to work correctly, the output controls need
+ * to be enabled, because they control the clock. So for playback, for
+ * example:
+ *
+ * amixer sset 'Left Output Mixer PCM' on
+ * amixer sset 'Right Output Mixer PCM' on
+ */
+
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/of_address.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <asm/fsl_guts.h>
+
+#include "fsl_dma.h"
+#include "fsl_ssi.h"
+#include "fsl_utils.h"
+
+/* P1022-specific PMUXCR and DMUXCR bit definitions */
+
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_MASK 0x0001c000
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI 0x00010000
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_SSI 0x00018000
+
+#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK 0x00000c00
+#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI 0x00000000
+
+#define CCSR_GUTS_DMUXCR_PAD 1 /* DMA controller/channel set to pad */
+#define CCSR_GUTS_DMUXCR_SSI 2 /* DMA controller/channel set to SSI */
+
+/*
+ * Set the DMACR register in the GUTS
+ *
+ * The DMACR register determines the source of initiated transfers for each
+ * channel on each DMA controller. Rather than have a bunch of repetitive
+ * macros for the bit patterns, we just have a function that calculates
+ * them.
+ *
+ * guts: Pointer to GUTS structure
+ * co: The DMA controller (0 or 1)
+ * ch: The channel on the DMA controller (0, 1, 2, or 3)
+ * device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx)
+ */
+static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts,
+ unsigned int co, unsigned int ch, unsigned int device)
+{
+ unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch));
+
+ clrsetbits_be32(&guts->dmuxcr, 3 << shift, device << shift);
+}
+
+/* There's only one global utilities register */
+static phys_addr_t guts_phys;
+
+/**
+ * machine_data: machine-specific ASoC device data
+ *
+ * This structure contains data for a single sound platform device on an
+ * P1022 RDK. Some of the data is taken from the device tree.
+ */
+struct machine_data {
+ struct snd_soc_dai_link dai[2];
+ struct snd_soc_card card;
+ unsigned int dai_format;
+ unsigned int codec_clk_direction;
+ unsigned int cpu_clk_direction;
+ unsigned int clk_frequency;
+ unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */
+ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
+ char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
+};
+
+/**
+ * p1022_rdk_machine_probe: initialize the board
+ *
+ * This function is used to initialize the board-specific hardware.
+ *
+ * Here we program the DMACR and PMUXCR registers.
+ */
+static int p1022_rdk_machine_probe(struct snd_soc_card *card)
+{
+ struct machine_data *mdata =
+ container_of(card, struct machine_data, card);
+ struct ccsr_guts __iomem *guts;
+
+ guts = ioremap(guts_phys, sizeof(struct ccsr_guts));
+ if (!guts) {
+ dev_err(card->dev, "could not map global utilities\n");
+ return -ENOMEM;
+ }
+
+ /* Enable SSI Tx signal */
+ clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK,
+ CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI);
+
+ /* Enable SSI Rx signal */
+ clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK,
+ CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI);
+
+ /* Enable DMA Channel for SSI */
+ guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0],
+ CCSR_GUTS_DMUXCR_SSI);
+
+ guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1],
+ CCSR_GUTS_DMUXCR_SSI);
+
+ iounmap(guts);
+
+ return 0;
+}
+
+/**
+ * p1022_rdk_startup: program the board with various hardware parameters
+ *
+ * This function takes board-specific information, like clock frequencies
+ * and serial data formats, and passes that information to the codec and
+ * transport drivers.
+ */
+static int p1022_rdk_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct machine_data *mdata =
+ container_of(rtd->card, struct machine_data, card);
+ struct device *dev = rtd->card->dev;
+ int ret = 0;
+
+ /* Tell the codec driver what the serial protocol is. */
+ ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format);
+ if (ret < 0) {
+ dev_err(dev, "could not set codec driver audio format (ret=%i)\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, mdata->clk_frequency,
+ mdata->clk_frequency);
+ if (ret < 0) {
+ dev_err(dev, "could not set codec PLL frequency (ret=%i)\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+/**
+ * p1022_rdk_machine_remove: Remove the sound device
+ *
+ * This function is called to remove the sound device for one SSI. We
+ * de-program the DMACR and PMUXCR register.
+ */
+static int p1022_rdk_machine_remove(struct snd_soc_card *card)
+{
+ struct machine_data *mdata =
+ container_of(card, struct machine_data, card);
+ struct ccsr_guts __iomem *guts;
+
+ guts = ioremap(guts_phys, sizeof(struct ccsr_guts));
+ if (!guts) {
+ dev_err(card->dev, "could not map global utilities\n");
+ return -ENOMEM;
+ }
+
+ /* Restore the signal routing */
+ clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK);
+ clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK);
+ guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], 0);
+ guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], 0);
+
+ iounmap(guts);
+
+ return 0;
+}
+
+/**
+ * p1022_rdk_ops: ASoC machine driver operations
+ */
+static struct snd_soc_ops p1022_rdk_ops = {
+ .startup = p1022_rdk_startup,
+};
+
+/**
+ * p1022_rdk_probe: platform probe function for the machine driver
+ *
+ * Although this is a machine driver, the SSI node is the "master" node with
+ * respect to audio hardware connections. Therefore, we create a new ASoC
+ * device for each new SSI node that has a codec attached.
+ */
+static int p1022_rdk_probe(struct platform_device *pdev)
+{
+ struct device *dev = pdev->dev.parent;
+ /* ssi_pdev is the platform device for the SSI node that probed us */
+ struct platform_device *ssi_pdev =
+ container_of(dev, struct platform_device, dev);
+ struct device_node *np = ssi_pdev->dev.of_node;
+ struct device_node *codec_np = NULL;
+ struct machine_data *mdata;
+ const u32 *iprop;
+ int ret;
+
+ /* Find the codec node for this SSI. */
+ codec_np = of_parse_phandle(np, "codec-handle", 0);
+ if (!codec_np) {
+ dev_err(dev, "could not find codec node\n");
+ return -EINVAL;
+ }
+
+ mdata = kzalloc(sizeof(struct machine_data), GFP_KERNEL);
+ if (!mdata) {
+ ret = -ENOMEM;
+ goto error_put;
+ }
+
+ mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
+ mdata->dai[0].ops = &p1022_rdk_ops;
+
+ /* ASoC core can match codec with device node */
+ mdata->dai[0].codec_of_node = codec_np;
+
+ /*
+ * We register two DAIs per SSI, one for playback and the other for
+ * capture. We support codecs that have separate DAIs for both playback
+ * and capture.
+ */
+ memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link));
+
+ /* The DAI names from the codec (snd_soc_dai_driver.name) */
+ mdata->dai[0].codec_dai_name = "wm8960-hifi";
+ mdata->dai[1].codec_dai_name = mdata->dai[0].codec_dai_name;
+
+ /*
+ * Configure the SSI for I2S slave mode. Older device trees have
+ * an fsl,mode property, but we ignore that since there's really
+ * only one way to configure the SSI.
+ */
+ mdata->dai_format = SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
+ mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+
+ /*
+ * In i2s-slave mode, the codec has its own clock source, so we
+ * need to get the frequency from the device tree and pass it to
+ * the codec driver.
+ */
+ iprop = of_get_property(codec_np, "clock-frequency", NULL);
+ if (!iprop || !*iprop) {
+ dev_err(&pdev->dev, "codec bus-frequency property is missing or invalid\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ mdata->clk_frequency = be32_to_cpup(iprop);
+
+ if (!mdata->clk_frequency) {
+ dev_err(&pdev->dev, "unknown clock frequency\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ /* Find the playback DMA channel to use. */
+ mdata->dai[0].platform_name = mdata->platform_name[0];
+ ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
+ &mdata->dma_channel_id[0],
+ &mdata->dma_id[0]);
+ if (ret) {
+ dev_err(&pdev->dev, "missing/invalid playback DMA phandle (ret=%i)\n",
+ ret);
+ goto error;
+ }
+
+ /* Find the capture DMA channel to use. */
+ mdata->dai[1].platform_name = mdata->platform_name[1];
+ ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
+ &mdata->dma_channel_id[1],
+ &mdata->dma_id[1]);
+ if (ret) {
+ dev_err(&pdev->dev, "missing/invalid capture DMA phandle (ret=%i)\n",
+ ret);
+ goto error;
+ }
+
+ /* Initialize our DAI data structure. */
+ mdata->dai[0].stream_name = "playback";
+ mdata->dai[1].stream_name = "capture";
+ mdata->dai[0].name = mdata->dai[0].stream_name;
+ mdata->dai[1].name = mdata->dai[1].stream_name;
+
+ mdata->card.probe = p1022_rdk_machine_probe;
+ mdata->card.remove = p1022_rdk_machine_remove;
+ mdata->card.name = pdev->name; /* The platform driver name */
+ mdata->card.owner = THIS_MODULE;
+ mdata->card.dev = &pdev->dev;
+ mdata->card.num_links = 2;
+ mdata->card.dai_link = mdata->dai;
+
+ /* Register with ASoC */
+ ret = snd_soc_register_card(&mdata->card);
+ if (ret) {
+ dev_err(&pdev->dev, "could not register card (ret=%i)\n", ret);
+ goto error;
+ }
+
+ return 0;
+
+error:
+ kfree(mdata);
+error_put:
+ of_node_put(codec_np);
+ return ret;
+}
+
+/**
+ * p1022_rdk_remove: remove the platform device
+ *
+ * This function is called when the platform device is removed.
+ */
+static int p1022_rdk_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct machine_data *mdata =
+ container_of(card, struct machine_data, card);
+
+ snd_soc_unregister_card(card);
+ kfree(mdata);
+
+ return 0;
+}
+
+static struct platform_driver p1022_rdk_driver = {
+ .probe = p1022_rdk_probe,
+ .remove = p1022_rdk_remove,
+ .driver = {
+ /*
+ * The name must match 'compatible' property in the device tree,
+ * in lowercase letters.
+ */
+ .name = "snd-soc-p1022rdk",
+ },
+};
+
+/**
+ * p1022_rdk_init: machine driver initialization.
+ *
+ * This function is called when this module is loaded.
+ */
+static int __init p1022_rdk_init(void)
+{
+ struct device_node *guts_np;
+ struct resource res;
+
+ /* Get the physical address of the global utilities registers */
+ guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts");
+ if (of_address_to_resource(guts_np, 0, &res)) {
+ pr_err("snd-soc-p1022rdk: missing/invalid global utils node\n");
+ of_node_put(guts_np);
+ return -EINVAL;
+ }
+ guts_phys = res.start;
+ of_node_put(guts_np);
+
+ return platform_driver_register(&p1022_rdk_driver);
+}
+
+/**
+ * p1022_rdk_exit: machine driver exit
+ *
+ * This function is called when this driver is unloaded.
+ */
+static void __exit p1022_rdk_exit(void)
+{
+ platform_driver_unregister(&p1022_rdk_driver);
+}
+
+late_initcall(p1022_rdk_init);
+module_exit(p1022_rdk_exit);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale / iVeia P1022 RDK ALSA SoC machine driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
new file mode 100644
index 000000000..ec731223c
--- /dev/null
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -0,0 +1,137 @@
+/*
+ * Phytec pcm030 driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+
+#include <sound/soc.h>
+
+#include "mpc5200_dma.h"
+
+#define DRV_NAME "pcm030-audio-fabric"
+
+struct pcm030_audio_data {
+ struct snd_soc_card *card;
+ struct platform_device *codec_device;
+};
+
+static struct snd_soc_dai_link pcm030_fabric_dai[] = {
+{
+ .name = "AC97.0",
+ .stream_name = "AC97 Analog",
+ .codec_dai_name = "wm9712-hifi",
+ .cpu_dai_name = "mpc5200-psc-ac97.0",
+ .codec_name = "wm9712-codec",
+},
+{
+ .name = "AC97.1",
+ .stream_name = "AC97 IEC958",
+ .codec_dai_name = "wm9712-aux",
+ .cpu_dai_name = "mpc5200-psc-ac97.1",
+ .codec_name = "wm9712-codec",
+},
+};
+
+static struct snd_soc_card pcm030_card = {
+ .name = "pcm030",
+ .owner = THIS_MODULE,
+ .dai_link = pcm030_fabric_dai,
+ .num_links = ARRAY_SIZE(pcm030_fabric_dai),
+};
+
+static int pcm030_fabric_probe(struct platform_device *op)
+{
+ struct device_node *np = op->dev.of_node;
+ struct device_node *platform_np;
+ struct snd_soc_card *card = &pcm030_card;
+ struct pcm030_audio_data *pdata;
+ int ret;
+ int i;
+
+ if (!of_machine_is_compatible("phytec,pcm030"))
+ return -ENODEV;
+
+ pdata = devm_kzalloc(&op->dev, sizeof(struct pcm030_audio_data),
+ GFP_KERNEL);
+ if (!pdata)
+ return -ENOMEM;
+
+ card->dev = &op->dev;
+
+ pdata->card = card;
+
+ platform_np = of_parse_phandle(np, "asoc-platform", 0);
+ if (!platform_np) {
+ dev_err(&op->dev, "ac97 not registered\n");
+ return -ENODEV;
+ }
+
+ for (i = 0; i < card->num_links; i++)
+ card->dai_link[i].platform_of_node = platform_np;
+
+ ret = request_module("snd-soc-wm9712");
+ if (ret)
+ dev_err(&op->dev, "request_module returned: %d\n", ret);
+
+ pdata->codec_device = platform_device_alloc("wm9712-codec", -1);
+ if (!pdata->codec_device)
+ dev_err(&op->dev, "platform_device_alloc() failed\n");
+
+ ret = platform_device_add(pdata->codec_device);
+ if (ret)
+ dev_err(&op->dev, "platform_device_add() failed: %d\n", ret);
+
+ ret = snd_soc_register_card(card);
+ if (ret)
+ dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret);
+
+ platform_set_drvdata(op, pdata);
+
+ return ret;
+}
+
+static int pcm030_fabric_remove(struct platform_device *op)
+{
+ struct pcm030_audio_data *pdata = platform_get_drvdata(op);
+ int ret;
+
+ ret = snd_soc_unregister_card(pdata->card);
+ platform_device_unregister(pdata->codec_device);
+
+ return ret;
+}
+
+static const struct of_device_id pcm030_audio_match[] = {
+ { .compatible = "phytec,pcm030-audio-fabric", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, pcm030_audio_match);
+
+static struct platform_driver pcm030_fabric_driver = {
+ .probe = pcm030_fabric_probe,
+ .remove = pcm030_fabric_remove,
+ .driver = {
+ .name = DRV_NAME,
+ .of_match_table = pcm030_audio_match,
+ },
+};
+
+module_platform_driver(pcm030_fabric_driver);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c
new file mode 100644
index 000000000..ae403c296
--- /dev/null
+++ b/sound/soc/fsl/phycore-ac97.c
@@ -0,0 +1,125 @@
+/*
+ * phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode
+ *
+ * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+
+#include "imx-audmux.h"
+
+static struct snd_soc_card imx_phycore;
+
+static struct snd_soc_ops imx_phycore_hifi_ops = {
+};
+
+static struct snd_soc_dai_link imx_phycore_dai_ac97[] = {
+ {
+ .name = "HiFi",
+ .stream_name = "HiFi",
+ .codec_dai_name = "wm9712-hifi",
+ .codec_name = "wm9712-codec",
+ .cpu_dai_name = "imx-ssi.0",
+ .platform_name = "imx-ssi.0",
+ .ops = &imx_phycore_hifi_ops,
+ },
+};
+
+static struct snd_soc_card imx_phycore = {
+ .name = "PhyCORE-ac97-audio",
+ .owner = THIS_MODULE,
+ .dai_link = imx_phycore_dai_ac97,
+ .num_links = ARRAY_SIZE(imx_phycore_dai_ac97),
+};
+
+static struct platform_device *imx_phycore_snd_ac97_device;
+static struct platform_device *imx_phycore_snd_device;
+
+static int __init imx_phycore_init(void)
+{
+ int ret;
+
+ if (machine_is_pca100()) {
+ imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+ IMX_AUDMUX_V1_PCR_SYN | /* 4wire mode */
+ IMX_AUDMUX_V1_PCR_TFCSEL(3) |
+ IMX_AUDMUX_V1_PCR_TCLKDIR | /* clock is output */
+ IMX_AUDMUX_V1_PCR_RXDSEL(3));
+ imx_audmux_v1_configure_port(3,
+ IMX_AUDMUX_V1_PCR_SYN | /* 4wire mode */
+ IMX_AUDMUX_V1_PCR_TFCSEL(0) |
+ IMX_AUDMUX_V1_PCR_TFSDIR |
+ IMX_AUDMUX_V1_PCR_RXDSEL(0));
+ } else if (machine_is_pcm043()) {
+ imx_audmux_v2_configure_port(3,
+ IMX_AUDMUX_V2_PTCR_SYN | /* 4wire mode */
+ IMX_AUDMUX_V2_PTCR_TFSEL(0) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(0));
+ imx_audmux_v2_configure_port(0,
+ IMX_AUDMUX_V2_PTCR_SYN | /* 4wire mode */
+ IMX_AUDMUX_V2_PTCR_TCSEL(3) |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR, /* clock is output */
+ IMX_AUDMUX_V2_PDCR_RXDSEL(3));
+ } else {
+ /* return happy. We might run on a totally different machine */
+ return 0;
+ }
+
+ imx_phycore_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+ if (!imx_phycore_snd_ac97_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(imx_phycore_snd_ac97_device, &imx_phycore);
+ ret = platform_device_add(imx_phycore_snd_ac97_device);
+ if (ret)
+ goto fail1;
+
+ imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1);
+ if (!imx_phycore_snd_device) {
+ ret = -ENOMEM;
+ goto fail2;
+ }
+ ret = platform_device_add(imx_phycore_snd_device);
+
+ if (ret) {
+ printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+ goto fail3;
+ }
+
+ return 0;
+
+fail3:
+ platform_device_put(imx_phycore_snd_device);
+fail2:
+ platform_device_del(imx_phycore_snd_ac97_device);
+fail1:
+ platform_device_put(imx_phycore_snd_ac97_device);
+ return ret;
+}
+
+static void __exit imx_phycore_exit(void)
+{
+ platform_device_unregister(imx_phycore_snd_device);
+ platform_device_unregister(imx_phycore_snd_ac97_device);
+}
+
+late_initcall(imx_phycore_init);
+module_exit(imx_phycore_exit);
+
+MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>");
+MODULE_DESCRIPTION("PhyCORE ALSA SoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
new file mode 100644
index 000000000..b454972dc
--- /dev/null
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -0,0 +1,292 @@
+/*
+ * wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
+ *
+ * Copyright (c) 2010 Wolfson Microelectronics plc
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * Based on an earlier driver for the same hardware by Liam Girdwood.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "imx-ssi.h"
+#include "../codecs/wm8350.h"
+#include "imx-audmux.h"
+
+/* There is a silicon mic on the board optionally connected via a solder pad
+ * SP1. Define this to enable it.
+ */
+#undef USE_SIMIC
+
+struct _wm8350_audio {
+ unsigned int channels;
+ snd_pcm_format_t format;
+ unsigned int rate;
+ unsigned int sysclk;
+ unsigned int bclkdiv;
+ unsigned int clkdiv;
+ unsigned int lr_rate;
+};
+
+/* in order of power consumption per rate (lowest first) */
+static const struct _wm8350_audio wm8350_audio[] = {
+ /* 16bit mono modes */
+ {1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
+ WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},
+
+ /* 16 bit stereo modes */
+ {2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
+ WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
+ {2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
+ WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
+ {2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
+ WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
+ {2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
+ WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+ {2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
+ WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+ {2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
+ WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
+ {2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
+ WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
+ {2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
+ WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+ {2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
+ WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+
+ /* 24bit stereo modes */
+ {2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
+ WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+ {2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
+ WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+ {2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
+ WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+ {2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
+ WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+};
+
+static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int i, found = 0;
+ snd_pcm_format_t format = params_format(params);
+ unsigned int rate = params_rate(params);
+ unsigned int channels = params_channels(params);
+
+ /* find the correct audio parameters */
+ for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
+ if (rate == wm8350_audio[i].rate &&
+ format == wm8350_audio[i].format &&
+ channels == wm8350_audio[i].channels) {
+ found = 1;
+ break;
+ }
+ }
+ if (!found)
+ return -EINVAL;
+
+ /* codec FLL input is 14.75 MHz from MCLK */
+ snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
+
+ /* TODO: The SSI driver should figure this out for us */
+ switch (channels) {
+ case 2:
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
+ break;
+ case 1:
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 1, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set MCLK as the codec system clock for DAC and ADC */
+ snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
+ wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);
+
+ /* set codec BCLK division for sample rate */
+ snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
+ wm8350_audio[i].bclkdiv);
+
+ /* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
+ snd_soc_dai_set_clkdiv(codec_dai,
+ WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
+ snd_soc_dai_set_clkdiv(codec_dai,
+ WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);
+
+ /* now configure DAC and ADC clocks */
+ snd_soc_dai_set_clkdiv(codec_dai,
+ WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);
+
+ snd_soc_dai_set_clkdiv(codec_dai,
+ WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);
+
+ return 0;
+}
+
+static struct snd_soc_ops wm1133_ev1_ops = {
+ .hw_params = wm1133_ev1_hw_params,
+};
+
+static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
+#ifdef USE_SIMIC
+ SND_SOC_DAPM_MIC("SiMIC", NULL),
+#endif
+ SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+ SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+/* imx32ads soc_card audio map */
+static const struct snd_soc_dapm_route wm1133_ev1_map[] = {
+
+#ifdef USE_SIMIC
+ /* SiMIC --> IN1LN (with automatic bias) via SP1 */
+ { "IN1LN", NULL, "Mic Bias" },
+ { "Mic Bias", NULL, "SiMIC" },
+#endif
+
+ /* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
+ { "IN1LN", NULL, "Mic Bias" },
+ { "IN1LP", NULL, "Mic1 Jack" },
+ { "Mic Bias", NULL, "Mic1 Jack" },
+
+ /* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
+ { "IN1RN", NULL, "Mic Bias" },
+ { "IN1RP", NULL, "Mic2 Jack" },
+ { "Mic Bias", NULL, "Mic2 Jack" },
+
+ /* Line in Jack --> AUX (L+R) */
+ { "IN3R", NULL, "Line In Jack" },
+ { "IN3L", NULL, "Line In Jack" },
+
+ /* Out1 --> Headphone Jack */
+ { "Headphone Jack", NULL, "OUT1R" },
+ { "Headphone Jack", NULL, "OUT1L" },
+
+ /* Out1 --> Line Out Jack */
+ { "Line Out Jack", NULL, "OUT2R" },
+ { "Line Out Jack", NULL, "OUT2L" },
+};
+
+static struct snd_soc_jack hp_jack;
+
+static struct snd_soc_jack_pin hp_jack_pins[] = {
+ { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
+};
+
+static struct snd_soc_jack mic_jack;
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+ { .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
+ { .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
+};
+
+static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+
+ /* Headphone jack detection */
+ snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
+ &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
+ wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
+
+ /* Microphone jack detection */
+ snd_soc_card_jack_new(rtd->card, "Microphone",
+ SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack,
+ mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
+ wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
+ SND_JACK_BTN_0);
+
+ snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias");
+
+ return 0;
+}
+
+
+static struct snd_soc_dai_link wm1133_ev1_dai = {
+ .name = "WM1133-EV1",
+ .stream_name = "Audio",
+ .cpu_dai_name = "imx-ssi.0",
+ .codec_dai_name = "wm8350-hifi",
+ .platform_name = "imx-ssi.0",
+ .codec_name = "wm8350-codec.0-0x1a",
+ .init = wm1133_ev1_init,
+ .ops = &wm1133_ev1_ops,
+ .symmetric_rates = 1,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+};
+
+static struct snd_soc_card wm1133_ev1 = {
+ .name = "WM1133-EV1",
+ .owner = THIS_MODULE,
+ .dai_link = &wm1133_ev1_dai,
+ .num_links = 1,
+
+ .dapm_widgets = wm1133_ev1_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets),
+ .dapm_routes = wm1133_ev1_map,
+ .num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map),
+};
+
+static struct platform_device *wm1133_ev1_snd_device;
+
+static int __init wm1133_ev1_audio_init(void)
+{
+ int ret;
+ unsigned int ptcr, pdcr;
+
+ /* SSI0 mastered by port 5 */
+ ptcr = IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR |
+ IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
+ pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
+ imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);
+
+ ptcr = IMX_AUDMUX_V2_PTCR_SYN;
+ pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
+ imx_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);
+
+ wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!wm1133_ev1_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1);
+ ret = platform_device_add(wm1133_ev1_snd_device);
+
+ if (ret)
+ platform_device_put(wm1133_ev1_snd_device);
+
+ return ret;
+}
+module_init(wm1133_ev1_audio_init);
+
+static void __exit wm1133_ev1_audio_exit(void)
+{
+ platform_device_unregister(wm1133_ev1_snd_device);
+}
+module_exit(wm1133_ev1_audio_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
+MODULE_LICENSE("GPL");