summaryrefslogtreecommitdiff
path: root/sound/pci/ca0106
diff options
context:
space:
mode:
authorAndré Fabian Silva Delgado <emulatorman@parabola.nu>2015-08-05 17:04:01 -0300
committerAndré Fabian Silva Delgado <emulatorman@parabola.nu>2015-08-05 17:04:01 -0300
commit57f0f512b273f60d52568b8c6b77e17f5636edc0 (patch)
tree5e910f0e82173f4ef4f51111366a3f1299037a7b /sound/pci/ca0106
Initial import
Diffstat (limited to 'sound/pci/ca0106')
-rw-r--r--sound/pci/ca0106/Makefile3
-rw-r--r--sound/pci/ca0106/ca0106.h742
-rw-r--r--sound/pci/ca0106/ca0106_main.c1972
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c932
-rw-r--r--sound/pci/ca0106/ca0106_proc.c457
-rw-r--r--sound/pci/ca0106/ca_midi.c316
-rw-r--r--sound/pci/ca0106/ca_midi.h66
7 files changed, 4488 insertions, 0 deletions
diff --git a/sound/pci/ca0106/Makefile b/sound/pci/ca0106/Makefile
new file mode 100644
index 000000000..dcbae7b31
--- /dev/null
+++ b/sound/pci/ca0106/Makefile
@@ -0,0 +1,3 @@
+snd-ca0106-objs := ca0106_main.o ca0106_proc.o ca0106_mixer.o ca_midi.o
+
+obj-$(CONFIG_SND_CA0106) += snd-ca0106.o
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
new file mode 100644
index 000000000..04402c14c
--- /dev/null
+++ b/sound/pci/ca0106/ca0106.h
@@ -0,0 +1,742 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.22
+ *
+ * FEATURES currently supported:
+ * See ca0106_main.c for features.
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
+ * 0.0.9
+ * Fix AC3 output.
+ * Enable S32_LE format support.
+ * 0.0.10
+ * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ * 0.0.11
+ * Add Model name recognition.
+ * 0.0.12
+ * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ * Remove redundent "voice" handling.
+ * 0.0.13
+ * Single trigger call for multi channels.
+ * 0.0.14
+ * Set limits based on what the sound card hardware can do.
+ * playback periods_min=2, periods_max=8
+ * capture hw constraints require period_size = n * 64 bytes.
+ * playback hw constraints require period_size = n * 64 bytes.
+ * 0.0.15
+ * Separated ca0106.c into separate functional .c files.
+ * 0.0.16
+ * Implement 192000 sample rate.
+ * 0.0.17
+ * Add support for SB0410 and SB0413.
+ * 0.0.18
+ * Modified Copyright message.
+ * 0.0.19
+ * Added I2C and SPI registers. Filled in interrupt enable.
+ * 0.0.20
+ * Added GPIO info for SB Live 24bit.
+ * 0.0.21
+ * Implement support for Line-in capture on SB Live 24bit.
+ * 0.0.22
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ *
+ *
+ * This code was initially based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/************************************************************************************************/
+/* PCI function 0 registers, address = <val> + PCIBASE0 */
+/************************************************************************************************/
+
+#define PTR 0x00 /* Indexed register set pointer register */
+ /* NOTE: The CHANNELNUM and ADDRESS words can */
+ /* be modified independently of each other. */
+ /* CNL[1:0], ADDR[27:16] */
+
+#define DATA 0x04 /* Indexed register set data register */
+ /* DATA[31:0] */
+
+#define IPR 0x08 /* Global interrupt pending register */
+ /* Clear pending interrupts by writing a 1 to */
+ /* the relevant bits and zero to the other bits */
+#define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */
+#define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */
+#define IPR_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */
+#define IPR_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */
+#define IPR_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */
+#define IPR_SPI 0x00000800 /* SPI transaction completed */
+#define IPR_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */
+#define IPR_I2C_DAC 0x00000200 /* I2C DAC transaction completed */
+#define IPR_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x76 */
+#define IPR_GPI 0x00000080 /* General Purpose input changed */
+#define IPR_SRC_LOCKED 0x00000040 /* SRC lock status changed */
+#define IPR_SPDIF_STATUS 0x00000020 /* SPDIF status changed */
+#define IPR_TIMER2 0x00000010 /* 192000Hz Timer */
+#define IPR_TIMER1 0x00000008 /* 44100Hz Timer */
+#define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */
+#define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */
+#define IPR_PCI 0x00000001 /* PCI Bus error */
+
+#define INTE 0x0c /* Interrupt enable register */
+
+#define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */
+#define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */
+#define INTE_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */
+#define INTE_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */
+#define INTE_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */
+#define INTE_SPI 0x00000800 /* SPI transaction completed */
+#define INTE_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */
+#define INTE_I2C_DAC 0x00000200 /* I2C DAC transaction completed */
+#define INTE_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x75 */
+#define INTE_GPI 0x00000080 /* General Purpose input changed */
+#define INTE_SRC_LOCKED 0x00000040 /* SRC lock status changed */
+#define INTE_SPDIF_STATUS 0x00000020 /* SPDIF status changed */
+#define INTE_TIMER2 0x00000010 /* 192000Hz Timer */
+#define INTE_TIMER1 0x00000008 /* 44100Hz Timer */
+#define INTE_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */
+#define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */
+#define INTE_PCI 0x00000001 /* PCI Bus error */
+
+#define UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */
+#define HCFG 0x14 /* Hardware config register */
+ /* 0x1000 causes AC3 to fails. It adds a dither bit. */
+
+#define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */
+#define HCFG_CAPTURE_I2S_BYPASS 0x08000000 /* 1 = bypass I2S input async SRC. */
+#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000 /* 1 = bypass SPDIF input async SRC. */
+#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000 /* 0 = I2S IN mixer output, 1 = I2S IN1. */
+#define HCFG_FORCE_LOCK 0x01000000 /* For test only. Force input SRC tracker to lock. */
+#define HCFG_PLAYBACK_ATTENUATION 0x00006000 /* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */
+#define HCFG_PLAYBACK_DITHER 0x00001000 /* 1 = Add dither bit to all playback channels. */
+#define HCFG_PLAYBACK_S32_LE 0x00000800 /* 1 = S32_LE, 0 = S16_LE */
+#define HCFG_CAPTURE_S32_LE 0x00000400 /* 1 = S32_LE, 0 = S16_LE (S32_LE current not working) */
+#define HCFG_8_CHANNEL_PLAY 0x00000200 /* 1 = 8 channels, 0 = 2 channels per substream.*/
+#define HCFG_8_CHANNEL_CAPTURE 0x00000100 /* 1 = 8 channels, 0 = 2 channels per substream.*/
+#define HCFG_MONO 0x00000080 /* 1 = I2S Input mono */
+#define HCFG_I2S_OUTPUT 0x00000010 /* 1 = I2S Output disabled */
+#define HCFG_AC97 0x00000008 /* 0 = AC97 1.0, 1 = AC97 2.0 */
+#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004 /* 1 = Cancel bustmaster accesses to soundcache */
+ /* NOTE: This should generally never be used. */
+#define HCFG_LOCK_CAPTURE_CACHE 0x00000002 /* 1 = Cancel bustmaster accesses to soundcache */
+ /* NOTE: This should generally never be used. */
+#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */
+ /* Should be set to 1 when the EMU10K1 is */
+ /* completely initialized. */
+#define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */
+ /* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */
+ /* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */
+ /* SB Live 24bit:
+ * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in.
+ * bit 9 0 = Mute / 1 = Analog out.
+ * bit 10 0 = Line-in / 1 = Mic-in.
+ * bit 11 0 = ? / 1 = ?
+ * bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit.
+ * bit 13 0 = ? / 1 = ?
+ * bit 14 0 = Mute / 1 = Analog out
+ * bit 15 0 = ? / 1 = ?
+ * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit.
+ */
+ /* 8 general purpose programmable In/Out pins.
+ * GPI [8:0] Read only. Default 0.
+ * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF)
+ * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin.
+ */
+#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */
+
+#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
+
+/********************************************************************************************************/
+/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */
+/********************************************************************************************************/
+
+/* Initially all registers from 0x00 to 0x3f have zero contents. */
+#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
+ /* One list entry: 4 bytes for DMA address,
+ * 4 bytes for period_size << 16.
+ * One list entry is 8 bytes long.
+ * One list entry for each period in the buffer.
+ */
+ /* ADDR[31:0], Default: 0x0 */
+#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
+ /* SIZE[21:16], Default: 0x8 */
+#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
+ /* PTR[5:0], Default: 0x0 */
+#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */
+#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */
+ /* DMA[31:0], Default: 0x0 */
+#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
+ /* SIZE[31:16], Default: 0x0 */
+#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
+ /* POINTER[15:0], Default: 0x0 */
+#define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */
+ /* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */
+#define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */
+ /* Cache size valid [5:0] */
+#define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */
+#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
+ /* DMA[31:0], Default: 0x0 */
+#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
+ /* SIZE[31:16], Default: 0x0 */
+#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
+ /* POINTER[15:0], Default: 0x0 */
+#define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */
+ /* Cache size valid [5:0] */
+#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */
+/* 0x21 - 0x3f unused */
+#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
+ /* Playback (0x1<<channel_id) */
+ /* Capture (0x100<<channel_id) */
+ /* Playback sample rate 96000 = 0x20000 */
+ /* Start Playback [3:0] (one bit per channel)
+ * Start Capture [11:8] (one bit per channel)
+ * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * Playback mixer in enable [27:24] (one bit per channel)
+ * Playback mixer out enable [31:28] (one bit per channel)
+ */
+/* The Digital out jack is shared with the Center/LFE Analogue output.
+ * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
+ * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
+ * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
+ * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red.
+ * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
+ */
+/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
+ * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS
+ * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM.
+ * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output
+ */
+/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel.
+ * A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs.
+ */
+#define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */
+#define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */
+#define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */
+#define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */
+ /* When Channel set to 0: */
+#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */
+#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */
+#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */
+#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */
+#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */
+#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */
+#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */
+#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */
+#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */
+#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */
+#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */
+#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */
+#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */
+#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */
+#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */
+#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */
+#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */
+#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */
+#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */
+#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */
+#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */
+#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */
+#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */
+
+ /* When Channel set to 1: */
+#define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */
+#define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */
+#define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */
+#define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */
+#define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */
+#define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */
+#define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */
+#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */
+#define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */
+#define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */
+#define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */
+#define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */
+#define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */
+
+#define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */
+ /* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE.
+ * But as the jack is shared, use 0xf00.
+ * The Windows2000 driver uses 0x0000000f for both digital and analog.
+ * 0xf00 introduces interesting noises onto the Center/LFE.
+ * If you turn the volume up, you hear computer noise,
+ * e.g. mouse moving, changing between app windows etc.
+ * So, I am going to set this to 0x0000000f all the time now,
+ * same as the windows driver does.
+ * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog.
+ */
+ /* When Channel = 0:
+ * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit)
+ * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate)
+ * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass)
+ */
+ /* When Channel = 1:
+ * SPDIF 0 User data [7:0]
+ * SPDIF 1 User data [15:8]
+ * SPDIF 0 User data [23:16]
+ * SPDIF 0 User data [31:24]
+ * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts.
+ */
+#define WATERMARK 0x46 /* Test bit to indicate cache usage level */
+#define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS.
+ * When Channel = 0: Bits the same as SPCS channel 0.
+ * When Channel = 1: Bits the same as SPCS channel 1.
+ * When Channel = 2:
+ * SPDIF Input User data [16:0]
+ * SPDIF Input Frame count [21:16]
+ */
+#define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */
+#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */
+#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */
+#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */
+#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */
+#define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */
+#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */
+ /* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3
+ * Record source select for channel 0 [18:16]
+ * Record source select for channel 1 [22:20]
+ * Record source select for channel 2 [26:24]
+ * Record source select for channel 3 [30:28]
+ * 0 - SPDIF mixer output.
+ * 1 - i2s mixer output.
+ * 2 - SPDIF input.
+ * 3 - i2s input.
+ * 4 - AC97 capture.
+ * 5 - SRC output.
+ */
+#define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */
+#define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */
+
+#define PLAYBACK_ROUTING1 0x63 /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */
+#define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */
+#define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */
+#define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */
+#define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */
+ /* Channel_id's handle stereo channels. Channel X is a single mono channel */
+ /* Host is input from the PCI bus. */
+ /* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
+ */
+
+#define PLAYBACK_ROUTING2 0x64 /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */
+ /* SRC is input from the capture inputs. */
+ /* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
+ */
+
+#define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */
+ /* SPDIF Mixer input control:
+ * Invert SRC to SPDIF Mixer [7-0] (One bit per channel)
+ * Invert Host to SPDIF Mixer [15:8] (One bit per channel)
+ * SRC to SPDIF Mixer disable [23:16] (One bit per channel)
+ * Host to SPDIF Mixer disable [31:24] (One bit per channel)
+ */
+#define PLAYBACK_VOLUME1 0x66 /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */
+ /* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */
+ /* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */
+ /* One register for each of the 4 stereo streams. */
+ /* SRC Right volume [7:0]
+ * SRC Left volume [15:8]
+ * Host Right volume [23:16]
+ * Host Left volume [31:24]
+ */
+#define CAPTURE_ROUTING1 0x67 /* Capture Routing. Default 0x32765410 */
+ /* Similar to register 0x63, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
+#define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */
+ /* Similar to register 0x64, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
+#define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */
+ /* Similar to register 0x65, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
+#define PLAYBACK_VOLUME2 0x6a /* Playback Analog volume per channel. Does not effect AC3 output */
+ /* Similar to register 0x66, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
+#define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */
+#define MIDI_UART_A_DATA 0x6c /* Midi Uart A Data */
+#define MIDI_UART_A_CMD 0x6d /* Midi Uart A Command/Status */
+#define MIDI_UART_B_DATA 0x6e /* Midi Uart B Data (currently unused) */
+#define MIDI_UART_B_CMD 0x6f /* Midi Uart B Command/Status (currently unused) */
+
+/* unique channel identifier for midi->channel */
+
+#define CA0106_MIDI_CHAN_A 0x1
+#define CA0106_MIDI_CHAN_B 0x2
+
+/* from mpu401 */
+
+#define CA0106_MIDI_INPUT_AVAIL 0x80
+#define CA0106_MIDI_OUTPUT_READY 0x40
+#define CA0106_MPU401_RESET 0xff
+#define CA0106_MPU401_ENTER_UART 0x3f
+#define CA0106_MPU401_ACK 0xfe
+
+#define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */
+ /* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0
+ * Rate Locked [20]
+ * SPDIF Locked [21] For SPDIF channel only.
+ * Valid Audio [22] For SPDIF channel only.
+ */
+#define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */
+ /* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */
+ /* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */
+ /* Sample rate output control register Channel=0
+ * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
+ * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source.
+ * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
+ * Record mixer output enable [12:10]
+ * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * I2S output source select [18] (0=Audio from host, 1=Audio from SRC)
+ * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0)
+ * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.)
+ * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.)
+ * I2S input mode [23] (0=Slave, 1=Master)
+ * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * SPDIF output source select [26] (0=host, 1=SRC)
+ * Not used [27]
+ * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
+ * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
+ */
+ /* Sample rate output control register Channel=1
+ * I2S Input 0 volume Right [7:0]
+ * I2S Input 0 volume Left [15:8]
+ * I2S Input 1 volume Right [23:16]
+ * I2S Input 1 volume Left [31:24]
+ */
+ /* Sample rate output control register Channel=2
+ * SPDIF Input volume Right [23:16]
+ * SPDIF Input volume Left [31:24]
+ */
+ /* Sample rate output control register Channel=3
+ * No used
+ */
+#define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */
+#define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */
+#define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */
+#define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */
+ /* Audio output control
+ * AC97 output enable [5:0]
+ * I2S output enable [19:16]
+ * SPDIF output enable [27:24]
+ */
+#define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */
+#define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */
+#define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */
+ /* Sets which Interrupts are enabled. */
+ /* 0x00000001 = Half period. Playback.
+ * 0x00000010 = Full period. Playback.
+ * 0x00000100 = Half buffer. Playback.
+ * 0x00001000 = Full buffer. Playback.
+ * 0x00010000 = Half buffer. Capture.
+ * 0x00100000 = Full buffer. Capture.
+ * Capture can only do 2 periods.
+ * 0x01000000 = End audio. Playback.
+ * 0x40000000 = Half buffer Playback,Caputre xrun.
+ * 0x80000000 = Full buffer Playback,Caputre xrun.
+ */
+#define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */
+ /* Shows which interrupts are active at the moment. */
+ /* Same bit layout as EXTENDED_INT_MASK */
+#define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */
+#define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */
+#define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */
+ /* Causes interrupts based on timer intervals. */
+#define SPI 0x7a /* SPI: Serial Interface Register */
+#define I2C_A 0x7b /* I2C Address. 32 bit */
+#define I2C_D0 0x7c /* I2C Data Port 0. 32 bit */
+#define I2C_D1 0x7d /* I2C Data Port 1. 32 bit */
+//I2C values
+#define I2C_A_ADC_ADD_MASK 0x000000fe //The address is a 7 bit address
+#define I2C_A_ADC_RW_MASK 0x00000001 //bit mask for R/W
+#define I2C_A_ADC_TRANS_MASK 0x00000010 //Bit mask for I2c address DAC value
+#define I2C_A_ADC_ABORT_MASK 0x00000020 //Bit mask for I2C transaction abort flag
+#define I2C_A_ADC_LAST_MASK 0x00000040 //Bit mask for Last word transaction
+#define I2C_A_ADC_BYTE_MASK 0x00000080 //Bit mask for Byte Mode
+
+#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC
+#define I2C_A_ADC_READ 0x00000001 //To perform a read operation
+#define I2C_A_ADC_START 0x00000100 //Start I2C transaction
+#define I2C_A_ADC_ABORT 0x00000200 //I2C transaction abort
+#define I2C_A_ADC_LAST 0x00000400 //I2C last transaction
+#define I2C_A_ADC_BYTE 0x00000800 //I2C one byte mode
+
+#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register
+#define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register
+
+#define ADC_TIMEOUT 0x00000007 //ADC Timeout Clock Disable
+#define ADC_IFC_CTRL 0x0000000b //ADC Interface Control
+#define ADC_MASTER 0x0000000c //ADC Master Mode Control
+#define ADC_POWER 0x0000000d //ADC PowerDown Control
+#define ADC_ATTEN_ADCL 0x0000000e //ADC Attenuation ADCL
+#define ADC_ATTEN_ADCR 0x0000000f //ADC Attenuation ADCR
+#define ADC_ALC_CTRL1 0x00000010 //ADC ALC Control 1
+#define ADC_ALC_CTRL2 0x00000011 //ADC ALC Control 2
+#define ADC_ALC_CTRL3 0x00000012 //ADC ALC Control 3
+#define ADC_NOISE_CTRL 0x00000013 //ADC Noise Gate Control
+#define ADC_LIMIT_CTRL 0x00000014 //ADC Limiter Control
+#define ADC_MUX 0x00000015 //ADC Mux offset
+
+#if 0
+/* FIXME: Not tested yet. */
+#define ADC_GAIN_MASK 0x000000ff //Mask for ADC Gain
+#define ADC_ZERODB 0x000000cf //Value to set ADC to 0dB
+#define ADC_MUTE_MASK 0x000000c0 //Mask for ADC mute
+#define ADC_MUTE 0x000000c0 //Value to mute ADC
+#define ADC_OSR 0x00000008 //Mask for ADC oversample rate select
+#define ADC_TIMEOUT_DISABLE 0x00000008 //Value and mask to disable Timeout clock
+#define ADC_HPF_DISABLE 0x00000100 //Value and mask to disable High pass filter
+#define ADC_TRANWIN_MASK 0x00000070 //Mask for Length of Transient Window
+#endif
+
+#define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux
+#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used)
+#define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux
+#define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux
+#define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux
+
+#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
+#define PCM_FRONT_CHANNEL 0
+#define PCM_REAR_CHANNEL 1
+#define PCM_CENTER_LFE_CHANNEL 2
+#define PCM_UNKNOWN_CHANNEL 3
+#define CONTROL_FRONT_CHANNEL 0
+#define CONTROL_REAR_CHANNEL 3
+#define CONTROL_CENTER_LFE_CHANNEL 1
+#define CONTROL_UNKNOWN_CHANNEL 2
+
+
+/* Based on WM8768 Datasheet Rev 4.2 page 32 */
+#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */
+#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */
+
+#define SPI_LDA1_REG 0 /* digital attenuation */
+#define SPI_RDA1_REG 1
+#define SPI_LDA2_REG 4
+#define SPI_RDA2_REG 5
+#define SPI_LDA3_REG 6
+#define SPI_RDA3_REG 7
+#define SPI_LDA4_REG 13
+#define SPI_RDA4_REG 14
+#define SPI_MASTDA_REG 8
+
+#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */
+#define SPI_DA_BIT_0dB 0xff /* 0 dB */
+#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */
+
+#define SPI_PL_REG 2
+#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */
+#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */
+#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */
+#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */
+#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */
+#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */
+#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */
+#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */
+#define SPI_IZD_REG 2
+#define SPI_IZD_BIT (1<<4) /* infinite zero detect */
+
+#define SPI_FMT_REG 3
+#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */
+#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */
+#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */
+#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */
+#define SPI_LRP_REG 3
+#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */
+#define SPI_BCP_REG 3
+#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */
+#define SPI_IWL_REG 3
+#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */
+#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */
+#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */
+#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */
+
+#define SPI_MS_REG 10
+#define SPI_MS_BIT (1<<5) /* master mode */
+#define SPI_RATE_REG 10 /* only applies in master mode */
+#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */
+#define SPI_RATE_BIT_192 (1<<6)
+#define SPI_RATE_BIT_256 (2<<6)
+#define SPI_RATE_BIT_384 (3<<6)
+#define SPI_RATE_BIT_512 (4<<6)
+#define SPI_RATE_BIT_768 (5<<6)
+
+/* They really do label the bit for the 4th channel "4" and not "3" */
+#define SPI_DMUTE0_REG 9
+#define SPI_DMUTE1_REG 9
+#define SPI_DMUTE2_REG 9
+#define SPI_DMUTE4_REG 15
+#define SPI_DMUTE0_BIT (1<<3)
+#define SPI_DMUTE1_BIT (1<<4)
+#define SPI_DMUTE2_BIT (1<<5)
+#define SPI_DMUTE4_BIT (1<<2)
+
+#define SPI_PHASE0_REG 3
+#define SPI_PHASE1_REG 3
+#define SPI_PHASE2_REG 3
+#define SPI_PHASE4_REG 15
+#define SPI_PHASE0_BIT (1<<6)
+#define SPI_PHASE1_BIT (1<<7)
+#define SPI_PHASE2_BIT (1<<8)
+#define SPI_PHASE4_BIT (1<<3)
+
+#define SPI_PDWN_REG 2 /* power down all DACs */
+#define SPI_PDWN_BIT (1<<2)
+#define SPI_DACD0_REG 10 /* power down individual DACs */
+#define SPI_DACD1_REG 10
+#define SPI_DACD2_REG 10
+#define SPI_DACD4_REG 15
+#define SPI_DACD0_BIT (1<<1)
+#define SPI_DACD1_BIT (1<<2)
+#define SPI_DACD2_BIT (1<<3)
+#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */
+
+#define SPI_PWRDNALL_REG 10 /* power down everything */
+#define SPI_PWRDNALL_BIT (1<<4)
+
+#include "ca_midi.h"
+
+struct snd_ca0106;
+
+struct snd_ca0106_channel {
+ struct snd_ca0106 *emu;
+ int number;
+ int use;
+ void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel);
+ struct snd_ca0106_pcm *epcm;
+};
+
+struct snd_ca0106_pcm {
+ struct snd_ca0106 *emu;
+ struct snd_pcm_substream *substream;
+ int channel_id;
+ unsigned short running;
+};
+
+struct snd_ca0106_details {
+ u32 serial;
+ char * name;
+ int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in.
+ ac97 = 1 -> Default to AC97 in. */
+ int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in
+ gpio_type = 2 -> shared side-out/line-in. */
+ int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume
+ controls, phone, mic, line-in and aux. */
+ u16 spi_dac; /* spi_dac = 0 -> no spi interface for DACs
+ spi_dac = 0x<front><rear><center-lfe><side>
+ -> specifies DAC id for each channel pair. */
+};
+
+// definition of the chip-specific record
+struct snd_ca0106 {
+ struct snd_card *card;
+ struct snd_ca0106_details *details;
+ struct pci_dev *pci;
+
+ unsigned long port;
+ struct resource *res_port;
+ int irq;
+
+ unsigned int serial; /* serial number */
+ unsigned short model; /* subsystem id */
+
+ spinlock_t emu_lock;
+
+ struct snd_ac97 *ac97;
+ struct snd_pcm *pcm[4];
+
+ struct snd_ca0106_channel playback_channels[4];
+ struct snd_ca0106_channel capture_channels[4];
+ u32 spdif_bits[4]; /* s/pdif out default setup */
+ u32 spdif_str_bits[4]; /* s/pdif out per-stream setup */
+ int spdif_enable;
+ int capture_source;
+ int i2c_capture_source;
+ u8 i2c_capture_volume[4][2];
+ int capture_mic_line_in;
+
+ struct snd_dma_buffer buffer;
+
+ struct snd_ca_midi midi;
+ struct snd_ca_midi midi2;
+
+ u16 spi_dac_reg[16];
+
+#ifdef CONFIG_PM_SLEEP
+#define NUM_SAVED_VOLUMES 9
+ unsigned int saved_vol[NUM_SAVED_VOLUMES];
+#endif
+};
+
+int snd_ca0106_mixer(struct snd_ca0106 *emu);
+int snd_ca0106_proc_init(struct snd_ca0106 * emu);
+
+unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu,
+ unsigned int reg,
+ unsigned int chn);
+
+void snd_ca0106_ptr_write(struct snd_ca0106 *emu,
+ unsigned int reg,
+ unsigned int chn,
+ unsigned int data);
+
+int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
+
+int snd_ca0106_spi_write(struct snd_ca0106 * emu,
+ unsigned int data);
+
+#ifdef CONFIG_PM_SLEEP
+void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip);
+void snd_ca0106_mixer_resume(struct snd_ca0106 *chip);
+#else
+#define snd_ca0106_mixer_suspend(chip) do { } while (0)
+#define snd_ca0106_mixer_resume(chip) do { } while (0)
+#endif
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
new file mode 100644
index 000000000..dd75b7536
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -0,0 +1,1972 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.25
+ *
+ * FEATURES currently supported:
+ * Front, Rear and Center/LFE.
+ * Surround40 and Surround51.
+ * Capture from MIC an LINE IN input.
+ * SPDIF digital playback of PCM stereo and AC3/DTS works.
+ * (One can use a standard mono mini-jack to one RCA plugs cable.
+ * or one can use a standard stereo mini-jack to two RCA plugs cable.
+ * Plug one of the RCA plugs into the Coax input of the external decoder/receiver.)
+ * ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. )
+ * Notes on how to capture sound:
+ * The AC97 is used in the PLAYBACK direction.
+ * The output from the AC97 chip, instead of reaching the speakers, is fed into the Philips 1361T ADC.
+ * So, to record from the MIC, set the MIC Playback volume to max,
+ * unmute the MIC and turn up the MASTER Playback volume.
+ * So, to prevent feedback when capturing, minimise the "Capture feedback into Playback" volume.
+ *
+ * The only playback controls that currently do anything are: -
+ * Analog Front
+ * Analog Rear
+ * Analog Center/LFE
+ * SPDIF Front
+ * SPDIF Rear
+ * SPDIF Center/LFE
+ *
+ * For capture from Mic in or Line in.
+ * Digital/Analog ( switch must be in Analog mode for CAPTURE. )
+ *
+ * CAPTURE feedback into PLAYBACK
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
+ * 0.0.9
+ * Fix AC3 output.
+ * Enable S32_LE format support.
+ * 0.0.10
+ * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ * 0.0.11
+ * Add Model name recognition.
+ * 0.0.12
+ * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ * Remove redundent "voice" handling.
+ * 0.0.13
+ * Single trigger call for multi channels.
+ * 0.0.14
+ * Set limits based on what the sound card hardware can do.
+ * playback periods_min=2, periods_max=8
+ * capture hw constraints require period_size = n * 64 bytes.
+ * playback hw constraints require period_size = n * 64 bytes.
+ * 0.0.15
+ * Minor updates.
+ * 0.0.16
+ * Implement 192000 sample rate.
+ * 0.0.17
+ * Add support for SB0410 and SB0413.
+ * 0.0.18
+ * Modified Copyright message.
+ * 0.0.19
+ * Finally fix support for SB Live 24 bit. SB0410 and SB0413.
+ * The output codec needs resetting, otherwise all output is muted.
+ * 0.0.20
+ * Merge "pci_disable_device(pci);" fixes.
+ * 0.0.21
+ * Add 4 capture channels. (SPDIF only comes in on channel 0. )
+ * Add SPDIF capture using optional digital I/O module for SB Live 24bit. (Analog capture does not yet work.)
+ * 0.0.22
+ * Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901
+ * 0.0.23
+ * Implement support for Line-in capture on SB Live 24bit.
+ * 0.0.24
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ * 0.0.25
+ * Powerdown SPI DAC channels when not in use
+ *
+ * BUGS:
+ * Some stability problems when unloading the snd-ca0106 kernel module.
+ * --
+ *
+ * TODO:
+ * 4 Capture channels, only one implemented so far.
+ * Other capture rates apart from 48khz not implemented.
+ * MIDI
+ * --
+ * GENERAL INFO:
+ * Model: SB0310
+ * P17 Chip: CA0106-DAT
+ * AC97 Codec: STAC 9721
+ * ADC: Philips 1361T (Stereo 24bit)
+ * DAC: WM8746EDS (6-channel, 24bit, 192Khz)
+ *
+ * GENERAL INFO:
+ * Model: SB0410
+ * P17 Chip: CA0106-DAT
+ * AC97 Codec: None
+ * ADC: WM8775EDS (4 Channel)
+ * DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support)
+ * SPDIF Out control switches between Mic in and SPDIF out.
+ * No sound out or mic input working yet.
+ *
+ * GENERAL INFO:
+ * Model: SB0413
+ * P17 Chip: CA0106-DAT
+ * AC97 Codec: None.
+ * ADC: Unknown
+ * DAC: Unknown
+ * Trying to handle it like the SB0410.
+ *
+ * This code was initially based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+
+MODULE_AUTHOR("James Courtier-Dutton <James@superbug.demon.co.uk>");
+MODULE_DESCRIPTION("CA0106");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}");
+
+// module parameters (see "Module Parameters")
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the CA0106 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the CA0106 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable the CA0106 soundcard.");
+module_param_array(subsystem, uint, NULL, 0444);
+MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
+
+#include "ca0106.h"
+
+static struct snd_ca0106_details ca0106_chip_details[] = {
+ /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
+ /* It is really just a normal SB Live 24bit. */
+ /* Tested:
+ * See ALSA bug#3251
+ */
+ { .serial = 0x10131102,
+ .name = "X-Fi Extreme Audio [SBxxxx]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
+ /* It is really just a normal SB Live 24bit. */
+ /*
+ * CTRL:CA0111-WTLF
+ * ADC: WM8775SEDS
+ * DAC: CS4382-KQZ
+ */
+ /* Tested:
+ * Playback on front, rear, center/lfe speakers
+ * Capture from Mic in.
+ * Not-Tested:
+ * Capture from Line in.
+ * Playback to digital out.
+ */
+ { .serial = 0x10121102,
+ .name = "X-Fi Extreme Audio [SB0790]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */
+ /* AudigyLS[SB0310] */
+ { .serial = 0x10021102,
+ .name = "AudigyLS [SB0310]",
+ .ac97 = 1 } ,
+ /* Unknown AudigyLS that also says SB0310 on it */
+ { .serial = 0x10051102,
+ .name = "AudigyLS [SB0310b]",
+ .ac97 = 1 } ,
+ /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */
+ { .serial = 0x10061102,
+ .name = "Live! 7.1 24bit [SB0410]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */
+ { .serial = 0x10071102,
+ .name = "Live! 7.1 24bit [SB0413]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* New Audigy SE. Has a different DAC. */
+ /* SB0570:
+ * CTRL:CA0106-DAT
+ * ADC: WM8775EDS
+ * DAC: WM8768GEDS
+ */
+ { .serial = 0x100a1102,
+ .name = "Audigy SE [SB0570]",
+ .gpio_type = 1,
+ .i2c_adc = 1,
+ .spi_dac = 0x4021 } ,
+ /* New Audigy LS. Has a different DAC. */
+ /* SB0570:
+ * CTRL:CA0106-DAT
+ * ADC: WM8775EDS
+ * DAC: WM8768GEDS
+ */
+ { .serial = 0x10111102,
+ .name = "Audigy SE OEM [SB0570a]",
+ .gpio_type = 1,
+ .i2c_adc = 1,
+ .spi_dac = 0x4021 } ,
+ /* Sound Blaster 5.1vx
+ * Tested: Playback on front, rear, center/lfe speakers
+ * Not-Tested: Capture
+ */
+ { .serial = 0x10041102,
+ .name = "Sound Blaster 5.1vx [SB1070]",
+ .gpio_type = 1,
+ .i2c_adc = 0,
+ .spi_dac = 0x0124
+ } ,
+ /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */
+ /* SB0438
+ * CTRL:CA0106-DAT
+ * ADC: WM8775SEDS
+ * DAC: CS4382-KQZ
+ */
+ { .serial = 0x10091462,
+ .name = "MSI K8N Diamond MB [SB0438]",
+ .gpio_type = 2,
+ .i2c_adc = 1 } ,
+ /* MSI K8N Diamond PLUS MB */
+ { .serial = 0x10091102,
+ .name = "MSI K8N Diamond MB",
+ .gpio_type = 2,
+ .i2c_adc = 1,
+ .spi_dac = 0x4021 } ,
+ /* Giga-byte GA-G1975X mobo
+ * Novell bnc#395807
+ */
+ /* FIXME: the GPIO and I2C setting aren't tested well */
+ { .serial = 0x1458a006,
+ .name = "Giga-byte GA-G1975X",
+ .gpio_type = 1,
+ .i2c_adc = 1 },
+ /* Shuttle XPC SD31P which has an onboard Creative Labs
+ * Sound Blaster Live! 24-bit EAX
+ * high-definition 7.1 audio processor".
+ * Added using info from andrewvegan in alsa bug #1298
+ */
+ { .serial = 0x30381297,
+ .name = "Shuttle XPC SD31P [SD31P]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* Shuttle XPC SD11G5 which has an onboard Creative Labs
+ * Sound Blaster Live! 24-bit EAX
+ * high-definition 7.1 audio processor".
+ * Fixes ALSA bug#1600
+ */
+ { .serial = 0x30411297,
+ .name = "Shuttle XPC SD11G5 [SD11G5]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ { .serial = 0,
+ .name = "AudigyLS [Unknown]" }
+};
+
+/* hardware definition */
+static struct snd_pcm_hardware snd_ca0106_playback_hw = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000),
+ .rate_min = 48000,
+ .rate_max = 192000,
+ .channels_min = 2, //1,
+ .channels_max = 2, //6,
+ .buffer_bytes_max = ((65536 - 64) * 8),
+ .period_bytes_min = 64,
+ .period_bytes_max = (65536 - 64),
+ .periods_min = 2,
+ .periods_max = 8,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_ca0106_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+#if 0 /* FIXME: looks like 44.1kHz capture causes noisy output on 48kHz */
+ .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
+ .rate_min = 44100,
+#else
+ .rates = (SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
+ .rate_min = 48000,
+#endif /* FIXME */
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536 - 128,
+ .period_bytes_min = 64,
+ .period_bytes_max = 32768 - 64,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu,
+ unsigned int reg,
+ unsigned int chn)
+{
+ unsigned long flags;
+ unsigned int regptr, val;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+void snd_ca0106_ptr_write(struct snd_ca0106 *emu,
+ unsigned int reg,
+ unsigned int chn,
+ unsigned int data)
+{
+ unsigned int regptr;
+ unsigned long flags;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ outl(data, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+int snd_ca0106_spi_write(struct snd_ca0106 * emu,
+ unsigned int data)
+{
+ unsigned int reset, set;
+ unsigned int reg, tmp;
+ int n, result;
+ reg = SPI;
+ if (data > 0xffff) /* Only 16bit values allowed */
+ return 1;
+ tmp = snd_ca0106_ptr_read(emu, reg, 0);
+ reset = (tmp & ~0x3ffff) | 0x20000; /* Set xxx20000 */
+ set = reset | 0x10000; /* Set xxx1xxxx */
+ snd_ca0106_ptr_write(emu, reg, 0, reset | data);
+ tmp = snd_ca0106_ptr_read(emu, reg, 0); /* write post */
+ snd_ca0106_ptr_write(emu, reg, 0, set | data);
+ result = 1;
+ /* Wait for status bit to return to 0 */
+ for (n = 0; n < 100; n++) {
+ udelay(10);
+ tmp = snd_ca0106_ptr_read(emu, reg, 0);
+ if (!(tmp & 0x10000)) {
+ result = 0;
+ break;
+ }
+ }
+ if (result) /* Timed out */
+ return 1;
+ snd_ca0106_ptr_write(emu, reg, 0, reset | data);
+ tmp = snd_ca0106_ptr_read(emu, reg, 0); /* Write post */
+ return 0;
+}
+
+/* The ADC does not support i2c read, so only write is implemented */
+int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
+ u32 reg,
+ u32 value)
+{
+ u32 tmp;
+ int timeout = 0;
+ int status;
+ int retry;
+ if ((reg > 0x7f) || (value > 0x1ff)) {
+ dev_err(emu->card->dev, "i2c_write: invalid values.\n");
+ return -EINVAL;
+ }
+
+ tmp = reg << 25 | value << 16;
+ /*
+ dev_dbg(emu->card->dev, "I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+ */
+ /* Not sure what this I2C channel controls. */
+ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */
+
+ /* This controls the I2C connected to the WM8775 ADC Codec */
+ snd_ca0106_ptr_write(emu, I2C_D1, 0, tmp);
+
+ for (retry = 0; retry < 10; retry++) {
+ /* Send the data to i2c */
+ //tmp = snd_ca0106_ptr_read(emu, I2C_A, 0);
+ //tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK);
+ tmp = 0;
+ tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD);
+ snd_ca0106_ptr_write(emu, I2C_A, 0, tmp);
+
+ /* Wait till the transaction ends */
+ while (1) {
+ status = snd_ca0106_ptr_read(emu, I2C_A, 0);
+ /*dev_dbg(emu->card->dev, "I2C:status=0x%x\n", status);*/
+ timeout++;
+ if ((status & I2C_A_ADC_START) == 0)
+ break;
+
+ if (timeout > 1000)
+ break;
+ }
+ //Read back and see if the transaction is successful
+ if ((status & I2C_A_ADC_ABORT) == 0)
+ break;
+ }
+
+ if (retry == 10) {
+ dev_err(emu->card->dev, "Writing to ADC failed!\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int intr_enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ intr_enable = inl(emu->port + INTE) | intrenb;
+ outl(intr_enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int intr_enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ intr_enable = inl(emu->port + INTE) & ~intrenb;
+ outl(intr_enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+
+static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ kfree(runtime->private_data);
+}
+
+static const int spi_dacd_reg[] = {
+ SPI_DACD0_REG,
+ SPI_DACD1_REG,
+ SPI_DACD2_REG,
+ 0,
+ SPI_DACD4_REG,
+};
+static const int spi_dacd_bit[] = {
+ SPI_DACD0_BIT,
+ SPI_DACD1_BIT,
+ SPI_DACD2_BIT,
+ 0,
+ SPI_DACD4_BIT,
+};
+
+static void restore_spdif_bits(struct snd_ca0106 *chip, int idx)
+{
+ if (chip->spdif_str_bits[idx] != chip->spdif_bits[idx]) {
+ chip->spdif_str_bits[idx] = chip->spdif_bits[idx];
+ snd_ca0106_ptr_write(chip, SPCS0 + idx, 0,
+ chip->spdif_str_bits[idx]);
+ }
+}
+
+static int snd_ca0106_channel_dac(struct snd_ca0106 *chip,
+ struct snd_ca0106_details *details,
+ int channel_id)
+{
+ switch (channel_id) {
+ case PCM_FRONT_CHANNEL:
+ return (details->spi_dac & 0xf000) >> (4 * 3);
+ case PCM_REAR_CHANNEL:
+ return (details->spi_dac & 0x0f00) >> (4 * 2);
+ case PCM_CENTER_LFE_CHANNEL:
+ return (details->spi_dac & 0x00f0) >> (4 * 1);
+ case PCM_UNKNOWN_CHANNEL:
+ return (details->spi_dac & 0x000f) >> (4 * 0);
+ default:
+ dev_dbg(chip->card->dev, "ca0106: unknown channel_id %d\n",
+ channel_id);
+ }
+ return 0;
+}
+
+static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id,
+ int power)
+{
+ if (chip->details->spi_dac) {
+ const int dac = snd_ca0106_channel_dac(chip, chip->details,
+ channel_id);
+ const int reg = spi_dacd_reg[dac];
+ const int bit = spi_dacd_bit[dac];
+
+ if (power)
+ /* Power up */
+ chip->spi_dac_reg[reg] &= ~bit;
+ else
+ /* Power down */
+ chip->spi_dac_reg[reg] |= bit;
+ return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ }
+ return 0;
+}
+
+/* open_playback callback */
+static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream,
+ int channel_id)
+{
+ struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+ struct snd_ca0106_channel *channel = &(chip->playback_channels[channel_id]);
+ struct snd_ca0106_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = chip;
+ epcm->substream = substream;
+ epcm->channel_id=channel_id;
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_ca0106_pcm_free_substream;
+
+ runtime->hw = snd_ca0106_playback_hw;
+
+ channel->emu = chip;
+ channel->number = channel_id;
+
+ channel->use = 1;
+ /*
+ dev_dbg(chip->card->dev, "open:channel_id=%d, chip=%p, channel=%p\n",
+ channel_id, chip, channel);
+ */
+ //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
+ channel->epcm = epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
+ return err;
+ snd_pcm_set_sync(substream);
+
+ /* Front channel dac should already be on */
+ if (channel_id != PCM_FRONT_CHANNEL) {
+ err = snd_ca0106_pcm_power_dac(chip, channel_id, 1);
+ if (err < 0)
+ return err;
+ }
+
+ restore_spdif_bits(chip, channel_id);
+
+ return 0;
+}
+
+/* close callback */
+static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ chip->playback_channels[epcm->channel_id].use = 0;
+
+ restore_spdif_bits(chip, epcm->channel_id);
+
+ /* Front channel dac should stay on */
+ if (epcm->channel_id != PCM_FRONT_CHANNEL) {
+ int err;
+ err = snd_ca0106_pcm_power_dac(chip, epcm->channel_id, 0);
+ if (err < 0)
+ return err;
+ }
+
+ /* FIXME: maybe zero others */
+ return 0;
+}
+
+static int snd_ca0106_pcm_open_playback_front(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_center_lfe(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_unknown(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_rear(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL);
+}
+
+/* open_capture callback */
+static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substream,
+ int channel_id)
+{
+ struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+ struct snd_ca0106_channel *channel = &(chip->capture_channels[channel_id]);
+ struct snd_ca0106_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL) {
+ dev_err(chip->card->dev,
+ "open_capture_channel: failed epcm alloc\n");
+ return -ENOMEM;
+ }
+ epcm->emu = chip;
+ epcm->substream = substream;
+ epcm->channel_id=channel_id;
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_ca0106_pcm_free_substream;
+
+ runtime->hw = snd_ca0106_capture_hw;
+
+ channel->emu = chip;
+ channel->number = channel_id;
+
+ channel->use = 1;
+ /*
+ dev_dbg(chip->card->dev, "open:channel_id=%d, chip=%p, channel=%p\n",
+ channel_id, chip, channel);
+ */
+ //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
+ channel->epcm = epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes);
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
+ return err;
+ return 0;
+}
+
+/* close callback */
+static int snd_ca0106_pcm_close_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ chip->capture_channels[epcm->channel_id].use = 0;
+ /* FIXME: maybe zero others */
+ return 0;
+}
+
+static int snd_ca0106_pcm_open_0_capture(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 0);
+}
+
+static int snd_ca0106_pcm_open_1_capture(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 1);
+}
+
+static int snd_ca0106_pcm_open_2_capture(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 2);
+}
+
+static int snd_ca0106_pcm_open_3_capture(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 3);
+}
+
+/* hw_params callback */
+static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* hw_params callback */
+static int snd_ca0106_pcm_hw_params_capture(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* prepare playback callback */
+static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ int channel = epcm->channel_id;
+ u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel));
+ u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size);
+ u32 hcfg_mask = HCFG_PLAYBACK_S32_LE;
+ u32 hcfg_set = 0x00000000;
+ u32 hcfg;
+ u32 reg40_mask = 0x30000 << (channel<<1);
+ u32 reg40_set = 0;
+ u32 reg40;
+ /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */
+ u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */
+ u32 reg71_set = 0;
+ u32 reg71;
+ int i;
+
+#if 0 /* debug */
+ dev_dbg(emu->card->dev,
+ "prepare:channel_number=%d, rate=%d, format=0x%x, "
+ "channels=%d, buffer_size=%ld, period_size=%ld, "
+ "periods=%u, frames_to_bytes=%d\n",
+ channel, runtime->rate, runtime->format,
+ runtime->channels, runtime->buffer_size,
+ runtime->period_size, runtime->periods,
+ frames_to_bytes(runtime, 1));
+ dev_dbg(emu->card->dev,
+ "dma_addr=%x, dma_area=%p, table_base=%p\n",
+ runtime->dma_addr, runtime->dma_area, table_base);
+ dev_dbg(emu->card->dev,
+ "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+ emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#endif /* debug */
+ /* Rate can be set per channel. */
+ /* reg40 control host to fifo */
+ /* reg71 controls DAC rate. */
+ switch (runtime->rate) {
+ case 44100:
+ reg40_set = 0x10000 << (channel<<1);
+ reg71_set = 0x01010000;
+ break;
+ case 48000:
+ reg40_set = 0;
+ reg71_set = 0;
+ break;
+ case 96000:
+ reg40_set = 0x20000 << (channel<<1);
+ reg71_set = 0x02020000;
+ break;
+ case 192000:
+ reg40_set = 0x30000 << (channel<<1);
+ reg71_set = 0x03030000;
+ break;
+ default:
+ reg40_set = 0;
+ reg71_set = 0;
+ break;
+ }
+ /* Format is a global setting */
+ /* FIXME: Only let the first channel accessed set this. */
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hcfg_set = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ hcfg_set = HCFG_PLAYBACK_S32_LE;
+ break;
+ default:
+ hcfg_set = 0;
+ break;
+ }
+ hcfg = inl(emu->port + HCFG) ;
+ hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
+ outl(hcfg, emu->port + HCFG);
+ reg40 = snd_ca0106_ptr_read(emu, 0x40, 0);
+ reg40 = (reg40 & ~reg40_mask) | reg40_set;
+ snd_ca0106_ptr_write(emu, 0x40, 0, reg40);
+ reg71 = snd_ca0106_ptr_read(emu, 0x71, 0);
+ reg71 = (reg71 & ~reg71_mask) | reg71_set;
+ snd_ca0106_ptr_write(emu, 0x71, 0, reg71);
+
+ /* FIXME: Check emu->buffer.size before actually writing to it. */
+ for(i=0; i < runtime->periods; i++) {
+ table_base[i*2] = runtime->dma_addr + (i * period_size_bytes);
+ table_base[i*2+1] = period_size_bytes << 16;
+ }
+
+ snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel));
+ snd_ca0106_ptr_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19);
+ snd_ca0106_ptr_write(emu, PLAYBACK_LIST_PTR, channel, 0);
+ snd_ca0106_ptr_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr);
+ snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes
+ /* FIXME test what 0 bytes does. */
+ snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes
+ snd_ca0106_ptr_write(emu, PLAYBACK_POINTER, channel, 0);
+ snd_ca0106_ptr_write(emu, 0x07, channel, 0x0);
+ snd_ca0106_ptr_write(emu, 0x08, channel, 0);
+ snd_ca0106_ptr_write(emu, PLAYBACK_MUTE, 0x0, 0x0); /* Unmute output */
+#if 0
+ snd_ca0106_ptr_write(emu, SPCS0, 0,
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT );
+#endif
+
+ return 0;
+}
+
+/* prepare capture callback */
+static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ int channel = epcm->channel_id;
+ u32 hcfg_mask = HCFG_CAPTURE_S32_LE;
+ u32 hcfg_set = 0x00000000;
+ u32 hcfg;
+ u32 over_sampling=0x2;
+ u32 reg71_mask = 0x0000c000 ; /* Global. Set ADC rate. */
+ u32 reg71_set = 0;
+ u32 reg71;
+
+#if 0 /* debug */
+ dev_dbg(emu->card->dev,
+ "prepare:channel_number=%d, rate=%d, format=0x%x, "
+ "channels=%d, buffer_size=%ld, period_size=%ld, "
+ "periods=%u, frames_to_bytes=%d\n",
+ channel, runtime->rate, runtime->format,
+ runtime->channels, runtime->buffer_size,
+ runtime->period_size, runtime->periods,
+ frames_to_bytes(runtime, 1));
+ dev_dbg(emu->card->dev,
+ "dma_addr=%x, dma_area=%p, table_base=%p\n",
+ runtime->dma_addr, runtime->dma_area, table_base);
+ dev_dbg(emu->card->dev,
+ "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+ emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#endif /* debug */
+ /* reg71 controls ADC rate. */
+ switch (runtime->rate) {
+ case 44100:
+ reg71_set = 0x00004000;
+ break;
+ case 48000:
+ reg71_set = 0;
+ break;
+ case 96000:
+ reg71_set = 0x00008000;
+ over_sampling=0xa;
+ break;
+ case 192000:
+ reg71_set = 0x0000c000;
+ over_sampling=0xa;
+ break;
+ default:
+ reg71_set = 0;
+ break;
+ }
+ /* Format is a global setting */
+ /* FIXME: Only let the first channel accessed set this. */
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hcfg_set = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ hcfg_set = HCFG_CAPTURE_S32_LE;
+ break;
+ default:
+ hcfg_set = 0;
+ break;
+ }
+ hcfg = inl(emu->port + HCFG) ;
+ hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
+ outl(hcfg, emu->port + HCFG);
+ reg71 = snd_ca0106_ptr_read(emu, 0x71, 0);
+ reg71 = (reg71 & ~reg71_mask) | reg71_set;
+ snd_ca0106_ptr_write(emu, 0x71, 0, reg71);
+ if (emu->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
+ snd_ca0106_i2c_write(emu, ADC_MASTER, over_sampling); /* Adjust the over sampler to better suit the capture rate. */
+ }
+
+
+ /*
+ dev_dbg(emu->card->dev,
+ "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, "
+ "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",
+ channel, runtime->rate, runtime->format, runtime->channels,
+ runtime->buffer_size, runtime->period_size,
+ frames_to_bytes(runtime, 1));
+ */
+ snd_ca0106_ptr_write(emu, 0x13, channel, 0);
+ snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
+ snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
+ snd_ca0106_ptr_write(emu, CAPTURE_POINTER, channel, 0);
+
+ return 0;
+}
+
+/* trigger_playback callback */
+static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime;
+ struct snd_ca0106_pcm *epcm;
+ int channel;
+ int result = 0;
+ struct snd_pcm_substream *s;
+ u32 basic = 0;
+ u32 extended = 0;
+ u32 bits;
+ int running = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ default:
+ running = 0;
+ break;
+ }
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) != emu ||
+ s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ continue;
+ runtime = s->runtime;
+ epcm = runtime->private_data;
+ channel = epcm->channel_id;
+ /* dev_dbg(emu->card->dev, "channel=%d\n", channel); */
+ epcm->running = running;
+ basic |= (0x1 << channel);
+ extended |= (0x10 << channel);
+ snd_pcm_trigger_done(s, substream);
+ }
+ /* dev_dbg(emu->card->dev, "basic=0x%x, extended=0x%x\n",basic, extended); */
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0);
+ bits |= extended;
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits);
+ bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0);
+ bits |= basic;
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0);
+ bits &= ~basic;
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits);
+ bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0);
+ bits &= ~extended;
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits);
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* trigger_capture callback */
+static int snd_ca0106_pcm_trigger_capture(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ int channel = epcm->channel_id;
+ int result = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<<channel));
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(0x100<<channel));
+ epcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<<channel));
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel));
+ epcm->running = 0;
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* pointer_playback callback */
+static snd_pcm_uframes_t
+snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ unsigned int ptr, prev_ptr;
+ int channel = epcm->channel_id;
+ int timeout = 10;
+
+ if (!epcm->running)
+ return 0;
+
+ prev_ptr = -1;
+ do {
+ ptr = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+ ptr = (ptr >> 3) * runtime->period_size;
+ ptr += bytes_to_frames(runtime,
+ snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel));
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+ if (prev_ptr == ptr)
+ return ptr;
+ prev_ptr = ptr;
+ } while (--timeout);
+ dev_warn(emu->card->dev, "ca0106: unstable DMA pointer!\n");
+ return 0;
+}
+
+/* pointer_capture callback */
+static snd_pcm_uframes_t
+snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
+ int channel = epcm->channel_id;
+
+ if (!epcm->running)
+ return 0;
+
+ ptr1 = snd_ca0106_ptr_read(emu, CAPTURE_POINTER, channel);
+ ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr=ptr2;
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+ /*
+ dev_dbg(emu->card->dev, "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+ "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+ ptr1, ptr2, ptr, (int)runtime->buffer_size,
+ (int)runtime->period_size, (int)runtime->frame_bits,
+ (int)runtime->rate);
+ */
+ return ptr;
+}
+
+/* operators */
+static struct snd_pcm_ops snd_ca0106_playback_front_ops = {
+ .open = snd_ca0106_pcm_open_playback_front,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static struct snd_pcm_ops snd_ca0106_capture_0_ops = {
+ .open = snd_ca0106_pcm_open_0_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static struct snd_pcm_ops snd_ca0106_capture_1_ops = {
+ .open = snd_ca0106_pcm_open_1_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static struct snd_pcm_ops snd_ca0106_capture_2_ops = {
+ .open = snd_ca0106_pcm_open_2_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static struct snd_pcm_ops snd_ca0106_capture_3_ops = {
+ .open = snd_ca0106_pcm_open_3_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static struct snd_pcm_ops snd_ca0106_playback_center_lfe_ops = {
+ .open = snd_ca0106_pcm_open_playback_center_lfe,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static struct snd_pcm_ops snd_ca0106_playback_unknown_ops = {
+ .open = snd_ca0106_pcm_open_playback_unknown,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static struct snd_pcm_ops snd_ca0106_playback_rear_ops = {
+ .open = snd_ca0106_pcm_open_playback_rear,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+
+static unsigned short snd_ca0106_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct snd_ca0106 *emu = ac97->private_data;
+ unsigned long flags;
+ unsigned short val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ val = inw(emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+static void snd_ca0106_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ struct snd_ca0106 *emu = ac97->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ outw(val, emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static int snd_ca0106_ac97(struct snd_ca0106 *chip)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_ca0106_ac97_write,
+ .read = snd_ca0106_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+ pbus->no_vra = 1; /* we don't need VRA */
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.scaps = AC97_SCAP_NO_SPDIF;
+ return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
+}
+
+static void ca0106_stop_chip(struct snd_ca0106 *chip);
+
+static int snd_ca0106_free(struct snd_ca0106 *chip)
+{
+ if (chip->res_port != NULL) {
+ /* avoid access to already used hardware */
+ ca0106_stop_chip(chip);
+ }
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ // release the data
+#if 1
+ if (chip->buffer.area)
+ snd_dma_free_pages(&chip->buffer);
+#endif
+
+ // release the i/o port
+ release_and_free_resource(chip->res_port);
+
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_ca0106_dev_free(struct snd_device *device)
+{
+ struct snd_ca0106 *chip = device->device_data;
+ return snd_ca0106_free(chip);
+}
+
+static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id)
+{
+ unsigned int status;
+
+ struct snd_ca0106 *chip = dev_id;
+ int i;
+ int mask;
+ unsigned int stat76;
+ struct snd_ca0106_channel *pchannel;
+
+ status = inl(chip->port + IPR);
+ if (! status)
+ return IRQ_NONE;
+
+ stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0);
+ /*
+ dev_dbg(emu->card->dev, "interrupt status = 0x%08x, stat76=0x%08x\n",
+ status, stat76);
+ dev_dbg(emu->card->dev, "ptr=0x%08x\n",
+ snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+ */
+ mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */
+ for(i = 0; i < 4; i++) {
+ pchannel = &(chip->playback_channels[i]);
+ if (stat76 & mask) {
+/* FIXME: Select the correct substream for period elapsed */
+ if(pchannel->use) {
+ snd_pcm_period_elapsed(pchannel->epcm->substream);
+ /* dev_dbg(emu->card->dev, "interrupt [%d] used\n", i); */
+ }
+ }
+ /*
+ dev_dbg(emu->card->dev, "channel=%p\n", pchannel);
+ dev_dbg(emu->card->dev, "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number);
+ */
+ mask <<= 1;
+ }
+ mask = 0x110000; /* 0x1 for one half, 0x10 for the other half period. */
+ for(i = 0; i < 4; i++) {
+ pchannel = &(chip->capture_channels[i]);
+ if (stat76 & mask) {
+/* FIXME: Select the correct substream for period elapsed */
+ if(pchannel->use) {
+ snd_pcm_period_elapsed(pchannel->epcm->substream);
+ /* dev_dbg(emu->card->dev, "interrupt [%d] used\n", i); */
+ }
+ }
+ /*
+ dev_dbg(emu->card->dev, "channel=%p\n", pchannel);
+ dev_dbg(emu->card->dev, "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number);
+ */
+ mask <<= 1;
+ }
+
+ snd_ca0106_ptr_write(chip, EXTENDED_INT, 0, stat76);
+
+ if (chip->midi.dev_id &&
+ (status & (chip->midi.ipr_tx|chip->midi.ipr_rx))) {
+ if (chip->midi.interrupt)
+ chip->midi.interrupt(&chip->midi, status);
+ else
+ chip->midi.interrupt_disable(&chip->midi, chip->midi.tx_enable | chip->midi.rx_enable);
+ }
+
+ // acknowledge the interrupt if necessary
+ outl(status, chip->port+IPR);
+
+ return IRQ_HANDLED;
+}
+
+static const struct snd_pcm_chmap_elem surround_map[] = {
+ { .channels = 2,
+ .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } },
+ { }
+};
+
+static const struct snd_pcm_chmap_elem clfe_map[] = {
+ { .channels = 2,
+ .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } },
+ { }
+};
+
+static const struct snd_pcm_chmap_elem side_map[] = {
+ { .channels = 2,
+ .map = { SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } },
+ { }
+};
+
+static int snd_ca0106_pcm(struct snd_ca0106 *emu, int device)
+{
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ const struct snd_pcm_chmap_elem *map = NULL;
+ int err;
+
+ err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = emu;
+
+ switch (device) {
+ case 0:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_front_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_0_ops);
+ map = snd_pcm_std_chmaps;
+ break;
+ case 1:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_rear_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_1_ops);
+ map = surround_map;
+ break;
+ case 2:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_center_lfe_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_2_ops);
+ map = clfe_map;
+ break;
+ case 3:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_unknown_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_3_ops);
+ map = side_map;
+ break;
+ }
+
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "CA0106");
+
+ for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ substream;
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ 64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */
+ return err;
+ }
+
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ substream;
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ 64*1024, 64*1024)) < 0)
+ return err;
+ }
+
+ err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, map, 2,
+ 1 << 2, NULL);
+ if (err < 0)
+ return err;
+
+ emu->pcm[device] = pcm;
+
+ return 0;
+}
+
+#define SPI_REG(reg, value) (((reg) << SPI_REG_SHIFT) | (value))
+static unsigned int spi_dac_init[] = {
+ SPI_REG(SPI_LDA1_REG, SPI_DA_BIT_0dB), /* 0dB dig. attenuation */
+ SPI_REG(SPI_RDA1_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_PL_REG, SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT),
+ SPI_REG(SPI_FMT_REG, SPI_FMT_BIT_I2S | SPI_IWL_BIT_24),
+ SPI_REG(SPI_LDA2_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA2_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_LDA3_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA3_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_MASTDA_REG, SPI_DA_BIT_0dB),
+ SPI_REG(9, 0x00),
+ SPI_REG(SPI_MS_REG, SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT),
+ SPI_REG(12, 0x00),
+ SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE),
+ SPI_REG(SPI_DACD4_REG, SPI_DACD4_BIT),
+};
+
+static unsigned int i2c_adc_init[][2] = {
+ { 0x17, 0x00 }, /* Reset */
+ { 0x07, 0x00 }, /* Timeout */
+ { 0x0b, 0x22 }, /* Interface control */
+ { 0x0c, 0x22 }, /* Master mode control */
+ { 0x0d, 0x08 }, /* Powerdown control */
+ { 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */
+ { 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */
+ { 0x10, 0x7b }, /* ALC Control 1 */
+ { 0x11, 0x00 }, /* ALC Control 2 */
+ { 0x12, 0x32 }, /* ALC Control 3 */
+ { 0x13, 0x00 }, /* Noise gate control */
+ { 0x14, 0xa6 }, /* Limiter control */
+ { 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */
+};
+
+static void ca0106_init_chip(struct snd_ca0106 *chip, int resume)
+{
+ int ch;
+ unsigned int def_bits;
+
+ outl(0, chip->port + INTE);
+
+ /*
+ * Init to 0x02109204 :
+ * Clock accuracy = 0 (1000ppm)
+ * Sample Rate = 2 (48kHz)
+ * Audio Channel = 1 (Left of 2)
+ * Source Number = 0 (Unspecified)
+ * Generation Status = 1 (Original for Cat Code 12)
+ * Cat Code = 12 (Digital Signal Mixer)
+ * Mode = 0 (Mode 0)
+ * Emphasis = 0 (None)
+ * CP = 1 (Copyright unasserted)
+ * AN = 0 (Audio data)
+ * P = 0 (Consumer)
+ */
+ def_bits =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT;
+ if (!resume) {
+ chip->spdif_str_bits[0] = chip->spdif_bits[0] = def_bits;
+ chip->spdif_str_bits[1] = chip->spdif_bits[1] = def_bits;
+ chip->spdif_str_bits[2] = chip->spdif_bits[2] = def_bits;
+ chip->spdif_str_bits[3] = chip->spdif_bits[3] = def_bits;
+ }
+ /* Only SPCS1 has been tested */
+ snd_ca0106_ptr_write(chip, SPCS1, 0, chip->spdif_str_bits[1]);
+ snd_ca0106_ptr_write(chip, SPCS0, 0, chip->spdif_str_bits[0]);
+ snd_ca0106_ptr_write(chip, SPCS2, 0, chip->spdif_str_bits[2]);
+ snd_ca0106_ptr_write(chip, SPCS3, 0, chip->spdif_str_bits[3]);
+
+ snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000);
+ snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000);
+
+ /* Write 0x8000 to AC97_REC_GAIN to mute it. */
+ outb(AC97_REC_GAIN, chip->port + AC97ADDRESS);
+ outw(0x8000, chip->port + AC97DATA);
+#if 0 /* FIXME: what are these? */
+ snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006);
+ snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006);
+ snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006);
+ snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006);
+#endif
+
+ /* OSS drivers set this. */
+ /* snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); */
+
+ /* Analog or Digital output */
+ snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
+ /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers.
+ * Use 0x000f0000 for surround71
+ */
+ snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000);
+
+ chip->spdif_enable = 0; /* Set digital SPDIF output off */
+ /*snd_ca0106_ptr_write(chip, 0x45, 0, 0);*/ /* Analogue out */
+ /*snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00);*/ /* Digital out */
+
+ /* goes to 0x40c80000 when doing SPDIF IN/OUT */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000);
+ /* (Mute) CAPTURE feedback into PLAYBACK volume.
+ * Only lower 16 bits matter.
+ */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff);
+ /* SPDIF IN Volume */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000);
+ /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000);
+
+ snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410);
+ snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676);
+ snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410);
+ snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676);
+
+ for (ch = 0; ch < 4; ch++) {
+ /* Only high 16 bits matter */
+ snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030);
+ snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030);
+#if 0 /* Mute */
+ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040);
+ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040);
+ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff);
+ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff);
+#endif
+ }
+ if (chip->details->i2c_adc == 1) {
+ /* Select MIC, Line in, TAD in, AUX in */
+ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
+ /* Default to CAPTURE_SOURCE to i2s in */
+ if (!resume)
+ chip->capture_source = 3;
+ } else if (chip->details->ac97 == 1) {
+ /* Default to AC97 in */
+ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x444400e4);
+ /* Default to CAPTURE_SOURCE to AC97 in */
+ if (!resume)
+ chip->capture_source = 4;
+ } else {
+ /* Select MIC, Line in, TAD in, AUX in */
+ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
+ /* Default to Set CAPTURE_SOURCE to i2s in */
+ if (!resume)
+ chip->capture_source = 3;
+ }
+
+ if (chip->details->gpio_type == 2) {
+ /* The SB0438 use GPIO differently. */
+ /* FIXME: Still need to find out what the other GPIO bits do.
+ * E.g. For digital spdif out.
+ */
+ outl(0x0, chip->port+GPIO);
+ /* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */
+ outl(0x005f5301, chip->port+GPIO); /* Analog */
+ } else if (chip->details->gpio_type == 1) {
+ /* The SB0410 and SB0413 use GPIO differently. */
+ /* FIXME: Still need to find out what the other GPIO bits do.
+ * E.g. For digital spdif out.
+ */
+ outl(0x0, chip->port+GPIO);
+ /* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */
+ outl(0x005f5301, chip->port+GPIO); /* Analog */
+ } else {
+ outl(0x0, chip->port+GPIO);
+ outl(0x005f03a3, chip->port+GPIO); /* Analog */
+ /* outl(0x005f02a2, chip->port+GPIO); */ /* SPDIF */
+ }
+ snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */
+
+ /* outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); */
+ /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
+ /* outl(0x00001409, chip->port+HCFG); */
+ /* outl(0x00000009, chip->port+HCFG); */
+ /* AC97 2.0, Enable outputs. */
+ outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG);
+
+ if (chip->details->i2c_adc == 1) {
+ /* The SB0410 and SB0413 use I2C to control ADC. */
+ int size, n;
+
+ size = ARRAY_SIZE(i2c_adc_init);
+ /* dev_dbg(emu->card->dev, "I2C:array size=0x%x\n", size); */
+ for (n = 0; n < size; n++)
+ snd_ca0106_i2c_write(chip, i2c_adc_init[n][0],
+ i2c_adc_init[n][1]);
+ for (n = 0; n < 4; n++) {
+ chip->i2c_capture_volume[n][0] = 0xcf;
+ chip->i2c_capture_volume[n][1] = 0xcf;
+ }
+ chip->i2c_capture_source = 2; /* Line in */
+ /* Enable Line-in capture. MIC in currently untested. */
+ /* snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); */
+ }
+
+ if (chip->details->spi_dac) {
+ /* The SB0570 use SPI to control DAC. */
+ int size, n;
+
+ size = ARRAY_SIZE(spi_dac_init);
+ for (n = 0; n < size; n++) {
+ int reg = spi_dac_init[n] >> SPI_REG_SHIFT;
+
+ snd_ca0106_spi_write(chip, spi_dac_init[n]);
+ if (reg < ARRAY_SIZE(chip->spi_dac_reg))
+ chip->spi_dac_reg[reg] = spi_dac_init[n];
+ }
+
+ /* Enable front dac only */
+ snd_ca0106_pcm_power_dac(chip, PCM_FRONT_CHANNEL, 1);
+ }
+}
+
+static void ca0106_stop_chip(struct snd_ca0106 *chip)
+{
+ /* disable interrupts */
+ snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
+ outl(0, chip->port + INTE);
+ snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
+ udelay(1000);
+ /* disable audio */
+ /* outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); */
+ outl(0, chip->port + HCFG);
+ /* FIXME: We need to stop and DMA transfers here.
+ * But as I am not sure how yet, we cannot from the dma pages.
+ * So we can fix: snd-malloc: Memory leak? pages not freed = 8
+ */
+}
+
+static int snd_ca0106_create(int dev, struct snd_card *card,
+ struct pci_dev *pci,
+ struct snd_ca0106 **rchip)
+{
+ struct snd_ca0106 *chip;
+ struct snd_ca0106_details *c;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ca0106_dev_free,
+ };
+
+ *rchip = NULL;
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+ if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) {
+ dev_err(card->dev, "error to set 32bit mask DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ spin_lock_init(&chip->emu_lock);
+
+ chip->port = pci_resource_start(pci, 0);
+ chip->res_port = request_region(chip->port, 0x20, "snd_ca0106");
+ if (!chip->res_port) {
+ snd_ca0106_free(chip);
+ dev_err(card->dev, "cannot allocate the port\n");
+ return -EBUSY;
+ }
+
+ if (request_irq(pci->irq, snd_ca0106_interrupt,
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
+ snd_ca0106_free(chip);
+ dev_err(card->dev, "cannot grab irq\n");
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ /* This stores the periods table. */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ 1024, &chip->buffer) < 0) {
+ snd_ca0106_free(chip);
+ return -ENOMEM;
+ }
+
+ pci_set_master(pci);
+ /* read serial */
+ pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
+ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
+ dev_info(card->dev, "Model %04x Rev %08x Serial %08x\n",
+ chip->model, pci->revision, chip->serial);
+ strcpy(card->driver, "CA0106");
+ strcpy(card->shortname, "CA0106");
+
+ for (c = ca0106_chip_details; c->serial; c++) {
+ if (subsystem[dev]) {
+ if (c->serial == subsystem[dev])
+ break;
+ } else if (c->serial == chip->serial)
+ break;
+ }
+ chip->details = c;
+ if (subsystem[dev]) {
+ dev_info(card->dev, "Sound card name=%s, "
+ "subsystem=0x%x. Forced to subsystem=0x%x\n",
+ c->name, chip->serial, subsystem[dev]);
+ }
+
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ c->name, chip->port, chip->irq);
+
+ ca0106_init_chip(chip, 0);
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ snd_ca0106_free(chip);
+ return err;
+ }
+ *rchip = chip;
+ return 0;
+}
+
+
+static void ca0106_midi_interrupt_enable(struct snd_ca_midi *midi, int intr)
+{
+ snd_ca0106_intr_enable((struct snd_ca0106 *)(midi->dev_id), intr);
+}
+
+static void ca0106_midi_interrupt_disable(struct snd_ca_midi *midi, int intr)
+{
+ snd_ca0106_intr_disable((struct snd_ca0106 *)(midi->dev_id), intr);
+}
+
+static unsigned char ca0106_midi_read(struct snd_ca_midi *midi, int idx)
+{
+ return (unsigned char)snd_ca0106_ptr_read((struct snd_ca0106 *)(midi->dev_id),
+ midi->port + idx, 0);
+}
+
+static void ca0106_midi_write(struct snd_ca_midi *midi, int data, int idx)
+{
+ snd_ca0106_ptr_write((struct snd_ca0106 *)(midi->dev_id), midi->port + idx, 0, data);
+}
+
+static struct snd_card *ca0106_dev_id_card(void *dev_id)
+{
+ return ((struct snd_ca0106 *)dev_id)->card;
+}
+
+static int ca0106_dev_id_port(void *dev_id)
+{
+ return ((struct snd_ca0106 *)dev_id)->port;
+}
+
+static int snd_ca0106_midi(struct snd_ca0106 *chip, unsigned int channel)
+{
+ struct snd_ca_midi *midi;
+ char *name;
+ int err;
+
+ if (channel == CA0106_MIDI_CHAN_B) {
+ name = "CA0106 MPU-401 (UART) B";
+ midi = &chip->midi2;
+ midi->tx_enable = INTE_MIDI_TX_B;
+ midi->rx_enable = INTE_MIDI_RX_B;
+ midi->ipr_tx = IPR_MIDI_TX_B;
+ midi->ipr_rx = IPR_MIDI_RX_B;
+ midi->port = MIDI_UART_B_DATA;
+ } else {
+ name = "CA0106 MPU-401 (UART)";
+ midi = &chip->midi;
+ midi->tx_enable = INTE_MIDI_TX_A;
+ midi->rx_enable = INTE_MIDI_TX_B;
+ midi->ipr_tx = IPR_MIDI_TX_A;
+ midi->ipr_rx = IPR_MIDI_RX_A;
+ midi->port = MIDI_UART_A_DATA;
+ }
+
+ midi->reset = CA0106_MPU401_RESET;
+ midi->enter_uart = CA0106_MPU401_ENTER_UART;
+ midi->ack = CA0106_MPU401_ACK;
+
+ midi->input_avail = CA0106_MIDI_INPUT_AVAIL;
+ midi->output_ready = CA0106_MIDI_OUTPUT_READY;
+
+ midi->channel = channel;
+
+ midi->interrupt_enable = ca0106_midi_interrupt_enable;
+ midi->interrupt_disable = ca0106_midi_interrupt_disable;
+
+ midi->read = ca0106_midi_read;
+ midi->write = ca0106_midi_write;
+
+ midi->get_dev_id_card = ca0106_dev_id_card;
+ midi->get_dev_id_port = ca0106_dev_id_port;
+
+ midi->dev_id = chip;
+
+ if ((err = ca_midi_init(chip, midi, 0, name)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+static int snd_ca0106_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_ca0106 *chip;
+ int i, err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+ 0, &card);
+ if (err < 0)
+ return err;
+
+ err = snd_ca0106_create(dev, card, pci, &chip);
+ if (err < 0)
+ goto error;
+ card->private_data = chip;
+
+ for (i = 0; i < 4; i++) {
+ err = snd_ca0106_pcm(chip, i);
+ if (err < 0)
+ goto error;
+ }
+
+ if (chip->details->ac97 == 1) {
+ /* The SB0410 and SB0413 do not have an AC97 chip. */
+ err = snd_ca0106_ac97(chip);
+ if (err < 0)
+ goto error;
+ }
+ err = snd_ca0106_mixer(chip);
+ if (err < 0)
+ goto error;
+
+ dev_dbg(card->dev, "probe for MIDI channel A ...");
+ err = snd_ca0106_midi(chip, CA0106_MIDI_CHAN_A);
+ if (err < 0)
+ goto error;
+ dev_dbg(card->dev, " done.\n");
+
+#ifdef CONFIG_PROC_FS
+ snd_ca0106_proc_init(chip);
+#endif
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+
+ error:
+ snd_card_free(card);
+ return err;
+}
+
+static void snd_ca0106_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int snd_ca0106_suspend(struct device *dev)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct snd_ca0106 *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ for (i = 0; i < 4; i++)
+ snd_pcm_suspend_all(chip->pcm[i]);
+ if (chip->details->ac97)
+ snd_ac97_suspend(chip->ac97);
+ snd_ca0106_mixer_suspend(chip);
+
+ ca0106_stop_chip(chip);
+ return 0;
+}
+
+static int snd_ca0106_resume(struct device *dev)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct snd_ca0106 *chip = card->private_data;
+ int i;
+
+ ca0106_init_chip(chip, 1);
+
+ if (chip->details->ac97)
+ snd_ac97_resume(chip->ac97);
+ snd_ca0106_mixer_resume(chip);
+ if (chip->details->spi_dac) {
+ for (i = 0; i < ARRAY_SIZE(chip->spi_dac_reg); i++)
+ snd_ca0106_spi_write(chip, chip->spi_dac_reg[i]);
+ }
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+
+static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume);
+#define SND_CA0106_PM_OPS &snd_ca0106_pm
+#else
+#define SND_CA0106_PM_OPS NULL
+#endif
+
+// PCI IDs
+static const struct pci_device_id snd_ca0106_ids[] = {
+ { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */
+ { 0, }
+};
+MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
+
+// pci_driver definition
+static struct pci_driver ca0106_driver = {
+ .name = KBUILD_MODNAME,
+ .id_table = snd_ca0106_ids,
+ .probe = snd_ca0106_probe,
+ .remove = snd_ca0106_remove,
+ .driver = {
+ .pm = SND_CA0106_PM_OPS,
+ },
+};
+
+module_pci_driver(ca0106_driver);
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
new file mode 100644
index 000000000..025805cba
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -0,0 +1,932 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.18
+ *
+ * FEATURES currently supported:
+ * See ca0106_main.c for features.
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
+ * 0.0.9
+ * Fix AC3 output.
+ * Enable S32_LE format support.
+ * 0.0.10
+ * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ * 0.0.11
+ * Add Model name recognition.
+ * 0.0.12
+ * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ * Remove redundent "voice" handling.
+ * 0.0.13
+ * Single trigger call for multi channels.
+ * 0.0.14
+ * Set limits based on what the sound card hardware can do.
+ * playback periods_min=2, periods_max=8
+ * capture hw constraints require period_size = n * 64 bytes.
+ * playback hw constraints require period_size = n * 64 bytes.
+ * 0.0.15
+ * Separated ca0106.c into separate functional .c files.
+ * 0.0.16
+ * Modified Copyright message.
+ * 0.0.17
+ * Implement Mic and Line in Capture.
+ * 0.0.18
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ *
+ * This code was initially based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+#include <linux/io.h>
+
+#include "ca0106.h"
+
+static void ca0106_spdif_enable(struct snd_ca0106 *emu)
+{
+ unsigned int val;
+
+ if (emu->spdif_enable) {
+ /* Digital */
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
+ val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000;
+ snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val);
+ val = inl(emu->port + GPIO) & ~0x101;
+ outl(val, emu->port + GPIO);
+
+ } else {
+ /* Analog */
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000);
+ val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000;
+ snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val);
+ val = inl(emu->port + GPIO) | 0x101;
+ outl(val, emu->port + GPIO);
+ }
+}
+
+static void ca0106_set_capture_source(struct snd_ca0106 *emu)
+{
+ unsigned int val = emu->capture_source;
+ unsigned int source, mask;
+ source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
+ mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
+ snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
+}
+
+static void ca0106_set_i2c_capture_source(struct snd_ca0106 *emu,
+ unsigned int val, int force)
+{
+ unsigned int ngain, ogain;
+ u32 source;
+
+ snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
+ ngain = emu->i2c_capture_volume[val][0]; /* Left */
+ ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
+ if (force || ngain != ogain)
+ snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ngain & 0xff);
+ ngain = emu->i2c_capture_volume[val][1]; /* Right */
+ ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Right */
+ if (force || ngain != ogain)
+ snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ngain & 0xff);
+ source = 1 << val;
+ snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */
+ emu->i2c_capture_source = val;
+}
+
+static void ca0106_set_capture_mic_line_in(struct snd_ca0106 *emu)
+{
+ u32 tmp;
+
+ if (emu->capture_mic_line_in) {
+ /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */
+ tmp = inl(emu->port+GPIO) & ~0x400;
+ tmp = tmp | 0x400;
+ outl(tmp, emu->port+GPIO);
+ /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); */
+ } else {
+ /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */
+ tmp = inl(emu->port+GPIO) & ~0x400;
+ outl(tmp, emu->port+GPIO);
+ /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); */
+ }
+}
+
+static void ca0106_set_spdif_bits(struct snd_ca0106 *emu, int idx)
+{
+ snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, emu->spdif_str_bits[idx]);
+}
+
+/*
+ */
+static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1);
+static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
+
+#define snd_ca0106_shared_spdif_info snd_ctl_boolean_mono_info
+
+static int snd_ca0106_shared_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = emu->spdif_enable;
+ return 0;
+}
+
+static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+
+ val = !!ucontrol->value.integer.value[0];
+ change = (emu->spdif_enable != val);
+ if (change) {
+ emu->spdif_enable = val;
+ ca0106_spdif_enable(emu);
+ }
+ return change;
+}
+
+static int snd_ca0106_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[6] = {
+ "IEC958 out", "i2s mixer out", "IEC958 in", "i2s in", "AC97 in", "SRC out"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1, 6, texts);
+}
+
+static int snd_ca0106_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->capture_source;
+ return 0;
+}
+
+static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ if (val >= 6)
+ return -EINVAL;
+ change = (emu->capture_source != val);
+ if (change) {
+ emu->capture_source = val;
+ ca0106_set_capture_source(emu);
+ }
+ return change;
+}
+
+static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[4] = {
+ "Phone", "Mic", "Line in", "Aux"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1, 4, texts);
+}
+
+static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->i2c_capture_source;
+ return 0;
+}
+
+static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int source_id;
+ int change = 0;
+ /* If the capture source has changed,
+ * update the capture volume from the cached value
+ * for the particular source.
+ */
+ source_id = ucontrol->value.enumerated.item[0] ;
+ if (source_id >= 4)
+ return -EINVAL;
+ change = (emu->i2c_capture_source != source_id);
+ if (change) {
+ ca0106_set_i2c_capture_source(emu, source_id, 0);
+ }
+ return change;
+}
+
+static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[2] = { "Side out", "Line in" };
+
+ return snd_ctl_enum_info(uinfo, 1, 2, texts);
+}
+
+static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[2] = { "Line in", "Mic in" };
+
+ return snd_ctl_enum_info(uinfo, 1, 2, texts);
+}
+
+static int snd_ca0106_capture_mic_line_in_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->capture_mic_line_in;
+ return 0;
+}
+
+static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ if (val > 1)
+ return -EINVAL;
+ change = (emu->capture_mic_line_in != val);
+ if (change) {
+ emu->capture_mic_line_in = val;
+ ca0106_set_capture_mic_line_in(emu);
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Shared Mic/Line in Capture Switch",
+ .info = snd_ca0106_capture_mic_line_in_info,
+ .get = snd_ca0106_capture_mic_line_in_get,
+ .put = snd_ca0106_capture_mic_line_in_put
+};
+
+static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Shared Line in/Side out Capture Switch",
+ .info = snd_ca0106_capture_line_in_side_out_info,
+ .get = snd_ca0106_capture_mic_line_in_get,
+ .put = snd_ca0106_capture_mic_line_in_put
+};
+
+
+static int snd_ca0106_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 void decode_spdif_bits(unsigned char *status, unsigned int bits)
+{
+ status[0] = (bits >> 0) & 0xff;
+ status[1] = (bits >> 8) & 0xff;
+ status[2] = (bits >> 16) & 0xff;
+ status[3] = (bits >> 24) & 0xff;
+}
+
+static int snd_ca0106_spdif_get_default(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ decode_spdif_bits(ucontrol->value.iec958.status,
+ emu->spdif_bits[idx]);
+ return 0;
+}
+
+static int snd_ca0106_spdif_get_stream(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ decode_spdif_bits(ucontrol->value.iec958.status,
+ emu->spdif_str_bits[idx]);
+ return 0;
+}
+
+static int snd_ca0106_spdif_get_mask(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static unsigned int encode_spdif_bits(unsigned char *status)
+{
+ return ((unsigned int)status[0] << 0) |
+ ((unsigned int)status[1] << 8) |
+ ((unsigned int)status[2] << 16) |
+ ((unsigned int)status[3] << 24);
+}
+
+static int snd_ca0106_spdif_put_default(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int val;
+
+ val = encode_spdif_bits(ucontrol->value.iec958.status);
+ if (val != emu->spdif_bits[idx]) {
+ emu->spdif_bits[idx] = val;
+ /* FIXME: this isn't safe, but needed to keep the compatibility
+ * with older alsa-lib config
+ */
+ emu->spdif_str_bits[idx] = val;
+ ca0106_set_spdif_bits(emu, idx);
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_ca0106_spdif_put_stream(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int val;
+
+ val = encode_spdif_bits(ucontrol->value.iec958.status);
+ if (val != emu->spdif_str_bits[idx]) {
+ emu->spdif_str_bits[idx] = val;
+ ca0106_set_spdif_bits(emu, idx);
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_ca0106_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_ca0106_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int value;
+ int channel_id, reg;
+
+ channel_id = (kcontrol->private_value >> 8) & 0xff;
+ reg = kcontrol->private_value & 0xff;
+
+ value = snd_ca0106_ptr_read(emu, reg, channel_id);
+ ucontrol->value.integer.value[0] = 0xff - ((value >> 24) & 0xff); /* Left */
+ ucontrol->value.integer.value[1] = 0xff - ((value >> 16) & 0xff); /* Right */
+ return 0;
+}
+
+static int snd_ca0106_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int oval, nval;
+ int channel_id, reg;
+
+ channel_id = (kcontrol->private_value >> 8) & 0xff;
+ reg = kcontrol->private_value & 0xff;
+
+ oval = snd_ca0106_ptr_read(emu, reg, channel_id);
+ nval = ((0xff - ucontrol->value.integer.value[0]) << 24) |
+ ((0xff - ucontrol->value.integer.value[1]) << 16);
+ nval |= ((0xff - ucontrol->value.integer.value[0]) << 8) |
+ ((0xff - ucontrol->value.integer.value[1]) );
+ if (oval == nval)
+ return 0;
+ snd_ca0106_ptr_write(emu, reg, channel_id, nval);
+ return 1;
+}
+
+static int snd_ca0106_i2c_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_ca0106_i2c_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ int source_id;
+
+ source_id = kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = emu->i2c_capture_volume[source_id][0];
+ ucontrol->value.integer.value[1] = emu->i2c_capture_volume[source_id][1];
+ return 0;
+}
+
+static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int ogain;
+ unsigned int ngain;
+ int source_id;
+ int change = 0;
+
+ source_id = kcontrol->private_value;
+ ogain = emu->i2c_capture_volume[source_id][0]; /* Left */
+ ngain = ucontrol->value.integer.value[0];
+ if (ngain > 0xff)
+ return -EINVAL;
+ if (ogain != ngain) {
+ if (emu->i2c_capture_source == source_id)
+ snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) );
+ emu->i2c_capture_volume[source_id][0] = ucontrol->value.integer.value[0];
+ change = 1;
+ }
+ ogain = emu->i2c_capture_volume[source_id][1]; /* Right */
+ ngain = ucontrol->value.integer.value[1];
+ if (ngain > 0xff)
+ return -EINVAL;
+ if (ogain != ngain) {
+ if (emu->i2c_capture_source == source_id)
+ snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
+ emu->i2c_capture_volume[source_id][1] = ucontrol->value.integer.value[1];
+ change = 1;
+ }
+
+ return change;
+}
+
+#define spi_mute_info snd_ctl_boolean_mono_info
+
+static int spi_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+ unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+
+ ucontrol->value.integer.value[0] = !(emu->spi_dac_reg[reg] & bit);
+ return 0;
+}
+
+static int spi_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+ unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+ int ret;
+
+ ret = emu->spi_dac_reg[reg] & bit;
+ if (ucontrol->value.integer.value[0]) {
+ if (!ret) /* bit already cleared, do nothing */
+ return 0;
+ emu->spi_dac_reg[reg] &= ~bit;
+ } else {
+ if (ret) /* bit already set, do nothing */
+ return 0;
+ emu->spi_dac_reg[reg] |= bit;
+ }
+
+ ret = snd_ca0106_spi_write(emu, emu->spi_dac_reg[reg]);
+ return ret ? -EINVAL : 1;
+}
+
+#define CA_VOLUME(xname,chid,reg) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .info = snd_ca0106_volume_info, \
+ .get = snd_ca0106_volume_get, \
+ .put = snd_ca0106_volume_put, \
+ .tlv = { .p = snd_ca0106_db_scale1 }, \
+ .private_value = ((chid) << 8) | (reg) \
+}
+
+static struct snd_kcontrol_new snd_ca0106_volume_ctls[] = {
+ CA_VOLUME("Analog Front Playback Volume",
+ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2),
+ CA_VOLUME("Analog Rear Playback Volume",
+ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2),
+ CA_VOLUME("Analog Center/LFE Playback Volume",
+ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2),
+ CA_VOLUME("Analog Side Playback Volume",
+ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2),
+
+ CA_VOLUME("IEC958 Front Playback Volume",
+ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1),
+ CA_VOLUME("IEC958 Rear Playback Volume",
+ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1),
+ CA_VOLUME("IEC958 Center/LFE Playback Volume",
+ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1),
+ CA_VOLUME("IEC958 Unknown Playback Volume",
+ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1),
+
+ CA_VOLUME("CAPTURE feedback Playback Volume",
+ 1, CAPTURE_CONTROL),
+
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .count = 4,
+ .info = snd_ca0106_spdif_info,
+ .get = snd_ca0106_spdif_get_mask
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Playback Switch",
+ .info = snd_ca0106_shared_spdif_info,
+ .get = snd_ca0106_shared_spdif_get,
+ .put = snd_ca0106_shared_spdif_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Source Capture Enum",
+ .info = snd_ca0106_capture_source_info,
+ .get = snd_ca0106_capture_source_get,
+ .put = snd_ca0106_capture_source_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Source Capture Enum",
+ .info = snd_ca0106_i2c_capture_source_info,
+ .get = snd_ca0106_i2c_capture_source_get,
+ .put = snd_ca0106_i2c_capture_source_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .count = 4,
+ .info = snd_ca0106_spdif_info,
+ .get = snd_ca0106_spdif_get_default,
+ .put = snd_ca0106_spdif_put_default
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .count = 4,
+ .info = snd_ca0106_spdif_info,
+ .get = snd_ca0106_spdif_get_stream,
+ .put = snd_ca0106_spdif_put_stream
+ },
+};
+
+#define I2C_VOLUME(xname,chid) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .info = snd_ca0106_i2c_volume_info, \
+ .get = snd_ca0106_i2c_volume_get, \
+ .put = snd_ca0106_i2c_volume_put, \
+ .tlv = { .p = snd_ca0106_db_scale2 }, \
+ .private_value = chid \
+}
+
+static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] = {
+ I2C_VOLUME("Phone Capture Volume", 0),
+ I2C_VOLUME("Mic Capture Volume", 1),
+ I2C_VOLUME("Line in Capture Volume", 2),
+ I2C_VOLUME("Aux Capture Volume", 3),
+};
+
+static const int spi_dmute_reg[] = {
+ SPI_DMUTE0_REG,
+ SPI_DMUTE1_REG,
+ SPI_DMUTE2_REG,
+ 0,
+ SPI_DMUTE4_REG,
+};
+static const int spi_dmute_bit[] = {
+ SPI_DMUTE0_BIT,
+ SPI_DMUTE1_BIT,
+ SPI_DMUTE2_BIT,
+ 0,
+ SPI_DMUTE4_BIT,
+};
+
+static struct snd_kcontrol_new
+snd_ca0106_volume_spi_dac_ctl(struct snd_ca0106_details *details,
+ int channel_id)
+{
+ struct snd_kcontrol_new spi_switch = {0};
+ int reg, bit;
+ int dac_id;
+
+ spi_switch.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ spi_switch.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ spi_switch.info = spi_mute_info;
+ spi_switch.get = spi_mute_get;
+ spi_switch.put = spi_mute_put;
+
+ switch (channel_id) {
+ case PCM_FRONT_CHANNEL:
+ spi_switch.name = "Analog Front Playback Switch";
+ dac_id = (details->spi_dac & 0xf000) >> (4 * 3);
+ break;
+ case PCM_REAR_CHANNEL:
+ spi_switch.name = "Analog Rear Playback Switch";
+ dac_id = (details->spi_dac & 0x0f00) >> (4 * 2);
+ break;
+ case PCM_CENTER_LFE_CHANNEL:
+ spi_switch.name = "Analog Center/LFE Playback Switch";
+ dac_id = (details->spi_dac & 0x00f0) >> (4 * 1);
+ break;
+ case PCM_UNKNOWN_CHANNEL:
+ spi_switch.name = "Analog Side Playback Switch";
+ dac_id = (details->spi_dac & 0x000f) >> (4 * 0);
+ break;
+ default:
+ /* Unused channel */
+ spi_switch.name = NULL;
+ dac_id = 0;
+ }
+ reg = spi_dmute_reg[dac_id];
+ bit = spi_dmute_bit[dac_id];
+
+ spi_switch.private_value = (reg << SPI_REG_SHIFT) | bit;
+
+ return spi_switch;
+}
+
+static int remove_ctl(struct snd_card *card, const char *name)
+{
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ strcpy(id.name, name);
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_remove_id(card, &id);
+}
+
+static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name)
+{
+ struct snd_ctl_elem_id sid;
+ memset(&sid, 0, sizeof(sid));
+ /* FIXME: strcpy is bad. */
+ strcpy(sid.name, name);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_find_id(card, &sid);
+}
+
+static int rename_ctl(struct snd_card *card, const char *src, const char *dst)
+{
+ struct snd_kcontrol *kctl = ctl_find(card, src);
+ if (kctl) {
+ strcpy(kctl->id.name, dst);
+ return 0;
+ }
+ return -ENOENT;
+}
+
+#define ADD_CTLS(emu, ctls) \
+ do { \
+ int i, _err; \
+ for (i = 0; i < ARRAY_SIZE(ctls); i++) { \
+ _err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \
+ if (_err < 0) \
+ return _err; \
+ } \
+ } while (0)
+
+static
+DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1);
+
+static char *slave_vols[] = {
+ "Analog Front Playback Volume",
+ "Analog Rear Playback Volume",
+ "Analog Center/LFE Playback Volume",
+ "Analog Side Playback Volume",
+ "IEC958 Front Playback Volume",
+ "IEC958 Rear Playback Volume",
+ "IEC958 Center/LFE Playback Volume",
+ "IEC958 Unknown Playback Volume",
+ "CAPTURE feedback Playback Volume",
+ NULL
+};
+
+static char *slave_sws[] = {
+ "Analog Front Playback Switch",
+ "Analog Rear Playback Switch",
+ "Analog Center/LFE Playback Switch",
+ "Analog Side Playback Switch",
+ "IEC958 Playback Switch",
+ NULL
+};
+
+static void add_slaves(struct snd_card *card,
+ struct snd_kcontrol *master, char **list)
+{
+ for (; *list; list++) {
+ struct snd_kcontrol *slave = ctl_find(card, *list);
+ if (slave)
+ snd_ctl_add_slave(master, slave);
+ }
+}
+
+int snd_ca0106_mixer(struct snd_ca0106 *emu)
+{
+ int err;
+ struct snd_card *card = emu->card;
+ char **c;
+ struct snd_kcontrol *vmaster;
+ static char *ca0106_remove_ctls[] = {
+ "Master Mono Playback Switch",
+ "Master Mono Playback Volume",
+ "3D Control - Switch",
+ "3D Control Sigmatel - Depth",
+ "PCM Playback Switch",
+ "PCM Playback Volume",
+ "CD Playback Switch",
+ "CD Playback Volume",
+ "Phone Playback Switch",
+ "Phone Playback Volume",
+ "Video Playback Switch",
+ "Video Playback Volume",
+ "Beep Playback Switch",
+ "Beep Playback Volume",
+ "Mono Output Select",
+ "Capture Source",
+ "Capture Switch",
+ "Capture Volume",
+ "External Amplifier",
+ "Sigmatel 4-Speaker Stereo Playback Switch",
+ "Surround Phase Inversion Playback Switch",
+ NULL
+ };
+ static char *ca0106_rename_ctls[] = {
+ "Master Playback Switch", "Capture Switch",
+ "Master Playback Volume", "Capture Volume",
+ "Line Playback Switch", "AC97 Line Capture Switch",
+ "Line Playback Volume", "AC97 Line Capture Volume",
+ "Aux Playback Switch", "AC97 Aux Capture Switch",
+ "Aux Playback Volume", "AC97 Aux Capture Volume",
+ "Mic Playback Switch", "AC97 Mic Capture Switch",
+ "Mic Playback Volume", "AC97 Mic Capture Volume",
+ "Mic Select", "AC97 Mic Select",
+ "Mic Boost (+20dB)", "AC97 Mic Boost (+20dB)",
+ NULL
+ };
+#if 1
+ for (c = ca0106_remove_ctls; *c; c++)
+ remove_ctl(card, *c);
+ for (c = ca0106_rename_ctls; *c; c += 2)
+ rename_ctl(card, c[0], c[1]);
+#endif
+
+ ADD_CTLS(emu, snd_ca0106_volume_ctls);
+ if (emu->details->i2c_adc == 1) {
+ ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls);
+ if (emu->details->gpio_type == 1)
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu));
+ else /* gpio_type == 2 */
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_line_in_side_out, emu));
+ if (err < 0)
+ return err;
+ }
+ if (emu->details->spi_dac) {
+ int i;
+ for (i = 0;; i++) {
+ struct snd_kcontrol_new ctl;
+ ctl = snd_ca0106_volume_spi_dac_ctl(emu->details, i);
+ if (!ctl.name)
+ break;
+ err = snd_ctl_add(card, snd_ctl_new1(&ctl, emu));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ /* Create virtual master controls */
+ vmaster = snd_ctl_make_virtual_master("Master Playback Volume",
+ snd_ca0106_master_db_scale);
+ if (!vmaster)
+ return -ENOMEM;
+ err = snd_ctl_add(card, vmaster);
+ if (err < 0)
+ return err;
+ add_slaves(card, vmaster, slave_vols);
+
+ if (emu->details->spi_dac) {
+ vmaster = snd_ctl_make_virtual_master("Master Playback Switch",
+ NULL);
+ if (!vmaster)
+ return -ENOMEM;
+ err = snd_ctl_add(card, vmaster);
+ if (err < 0)
+ return err;
+ add_slaves(card, vmaster, slave_sws);
+ }
+
+ strcpy(card->mixername, "CA0106");
+ return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+struct ca0106_vol_tbl {
+ unsigned int channel_id;
+ unsigned int reg;
+};
+
+static struct ca0106_vol_tbl saved_volumes[NUM_SAVED_VOLUMES] = {
+ { CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2 },
+ { CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2 },
+ { CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2 },
+ { CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2 },
+ { CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1 },
+ { CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1 },
+ { CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1 },
+ { CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1 },
+ { 1, CAPTURE_CONTROL },
+};
+
+void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip)
+{
+ int i;
+
+ /* save volumes */
+ for (i = 0; i < NUM_SAVED_VOLUMES; i++)
+ chip->saved_vol[i] =
+ snd_ca0106_ptr_read(chip, saved_volumes[i].reg,
+ saved_volumes[i].channel_id);
+}
+
+void snd_ca0106_mixer_resume(struct snd_ca0106 *chip)
+{
+ int i;
+
+ for (i = 0; i < NUM_SAVED_VOLUMES; i++)
+ snd_ca0106_ptr_write(chip, saved_volumes[i].reg,
+ saved_volumes[i].channel_id,
+ chip->saved_vol[i]);
+
+ ca0106_spdif_enable(chip);
+ ca0106_set_capture_source(chip);
+ ca0106_set_i2c_capture_source(chip, chip->i2c_capture_source, 1);
+ for (i = 0; i < 4; i++)
+ ca0106_set_spdif_bits(chip, i);
+ if (chip->details->i2c_adc)
+ ca0106_set_capture_mic_line_in(chip);
+}
+#endif /* CONFIG_PM_SLEEP */
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
new file mode 100644
index 000000000..2c5c28adb
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -0,0 +1,457 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.18
+ *
+ * FEATURES currently supported:
+ * See ca0106_main.c for features.
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
+ * 0.0.9
+ * Fix AC3 output.
+ * Enable S32_LE format support.
+ * 0.0.10
+ * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ * 0.0.11
+ * Add Model name recognition.
+ * 0.0.12
+ * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ * Remove redundent "voice" handling.
+ * 0.0.13
+ * Single trigger call for multi channels.
+ * 0.0.14
+ * Set limits based on what the sound card hardware can do.
+ * playback periods_min=2, periods_max=8
+ * capture hw constraints require period_size = n * 64 bytes.
+ * playback hw constraints require period_size = n * 64 bytes.
+ * 0.0.15
+ * Separate ca0106.c into separate functional .c files.
+ * 0.0.16
+ * Modified Copyright message.
+ * 0.0.17
+ * Add iec958 file in proc file system to show status of SPDIF in.
+ * 0.0.18
+ * Implement support for Line-in capture on SB Live 24bit.
+ *
+ * This code was initially based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/asoundef.h>
+
+#include "ca0106.h"
+
+
+#ifdef CONFIG_PROC_FS
+
+struct snd_ca0106_category_str {
+ int val;
+ const char *name;
+};
+
+static struct snd_ca0106_category_str snd_ca0106_con_category[] = {
+ { IEC958_AES1_CON_DAT, "DAT" },
+ { IEC958_AES1_CON_VCR, "VCR" },
+ { IEC958_AES1_CON_MICROPHONE, "microphone" },
+ { IEC958_AES1_CON_SYNTHESIZER, "synthesizer" },
+ { IEC958_AES1_CON_RATE_CONVERTER, "rate converter" },
+ { IEC958_AES1_CON_MIXER, "mixer" },
+ { IEC958_AES1_CON_SAMPLER, "sampler" },
+ { IEC958_AES1_CON_PCM_CODER, "PCM coder" },
+ { IEC958_AES1_CON_IEC908_CD, "CD" },
+ { IEC958_AES1_CON_NON_IEC908_CD, "non-IEC908 CD" },
+ { IEC958_AES1_CON_GENERAL, "general" },
+};
+
+
+static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 value)
+{
+ int i;
+ u32 status[4];
+ status[0] = value & 0xff;
+ status[1] = (value >> 8) & 0xff;
+ status[2] = (value >> 16) & 0xff;
+ status[3] = (value >> 24) & 0xff;
+
+ if (! (status[0] & IEC958_AES0_PROFESSIONAL)) {
+ /* consumer */
+ snd_iprintf(buffer, "Mode: consumer\n");
+ snd_iprintf(buffer, "Data: ");
+ if (!(status[0] & IEC958_AES0_NONAUDIO)) {
+ snd_iprintf(buffer, "audio\n");
+ } else {
+ snd_iprintf(buffer, "non-audio\n");
+ }
+ snd_iprintf(buffer, "Rate: ");
+ switch (status[3] & IEC958_AES3_CON_FS) {
+ case IEC958_AES3_CON_FS_44100:
+ snd_iprintf(buffer, "44100 Hz\n");
+ break;
+ case IEC958_AES3_CON_FS_48000:
+ snd_iprintf(buffer, "48000 Hz\n");
+ break;
+ case IEC958_AES3_CON_FS_32000:
+ snd_iprintf(buffer, "32000 Hz\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Copyright: ");
+ if (status[0] & IEC958_AES0_CON_NOT_COPYRIGHT) {
+ snd_iprintf(buffer, "permitted\n");
+ } else {
+ snd_iprintf(buffer, "protected\n");
+ }
+ snd_iprintf(buffer, "Emphasis: ");
+ if ((status[0] & IEC958_AES0_CON_EMPHASIS) != IEC958_AES0_CON_EMPHASIS_5015) {
+ snd_iprintf(buffer, "none\n");
+ } else {
+ snd_iprintf(buffer, "50/15us\n");
+ }
+ snd_iprintf(buffer, "Category: ");
+ for (i = 0; i < ARRAY_SIZE(snd_ca0106_con_category); i++) {
+ if ((status[1] & IEC958_AES1_CON_CATEGORY) == snd_ca0106_con_category[i].val) {
+ snd_iprintf(buffer, "%s\n", snd_ca0106_con_category[i].name);
+ break;
+ }
+ }
+ if (i >= ARRAY_SIZE(snd_ca0106_con_category)) {
+ snd_iprintf(buffer, "unknown 0x%x\n", status[1] & IEC958_AES1_CON_CATEGORY);
+ }
+ snd_iprintf(buffer, "Original: ");
+ if (status[1] & IEC958_AES1_CON_ORIGINAL) {
+ snd_iprintf(buffer, "original\n");
+ } else {
+ snd_iprintf(buffer, "1st generation\n");
+ }
+ snd_iprintf(buffer, "Clock: ");
+ switch (status[3] & IEC958_AES3_CON_CLOCK) {
+ case IEC958_AES3_CON_CLOCK_1000PPM:
+ snd_iprintf(buffer, "1000 ppm\n");
+ break;
+ case IEC958_AES3_CON_CLOCK_50PPM:
+ snd_iprintf(buffer, "50 ppm\n");
+ break;
+ case IEC958_AES3_CON_CLOCK_VARIABLE:
+ snd_iprintf(buffer, "variable pitch\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ } else {
+ snd_iprintf(buffer, "Mode: professional\n");
+ snd_iprintf(buffer, "Data: ");
+ if (!(status[0] & IEC958_AES0_NONAUDIO)) {
+ snd_iprintf(buffer, "audio\n");
+ } else {
+ snd_iprintf(buffer, "non-audio\n");
+ }
+ snd_iprintf(buffer, "Rate: ");
+ switch (status[0] & IEC958_AES0_PRO_FS) {
+ case IEC958_AES0_PRO_FS_44100:
+ snd_iprintf(buffer, "44100 Hz\n");
+ break;
+ case IEC958_AES0_PRO_FS_48000:
+ snd_iprintf(buffer, "48000 Hz\n");
+ break;
+ case IEC958_AES0_PRO_FS_32000:
+ snd_iprintf(buffer, "32000 Hz\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Rate Locked: ");
+ if (status[0] & IEC958_AES0_PRO_FREQ_UNLOCKED)
+ snd_iprintf(buffer, "no\n");
+ else
+ snd_iprintf(buffer, "yes\n");
+ snd_iprintf(buffer, "Emphasis: ");
+ switch (status[0] & IEC958_AES0_PRO_EMPHASIS) {
+ case IEC958_AES0_PRO_EMPHASIS_CCITT:
+ snd_iprintf(buffer, "CCITT J.17\n");
+ break;
+ case IEC958_AES0_PRO_EMPHASIS_NONE:
+ snd_iprintf(buffer, "none\n");
+ break;
+ case IEC958_AES0_PRO_EMPHASIS_5015:
+ snd_iprintf(buffer, "50/15us\n");
+ break;
+ case IEC958_AES0_PRO_EMPHASIS_NOTID:
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Stereophonic: ");
+ if ((status[1] & IEC958_AES1_PRO_MODE) == IEC958_AES1_PRO_MODE_STEREOPHONIC) {
+ snd_iprintf(buffer, "stereo\n");
+ } else {
+ snd_iprintf(buffer, "not indicated\n");
+ }
+ snd_iprintf(buffer, "Userbits: ");
+ switch (status[1] & IEC958_AES1_PRO_USERBITS) {
+ case IEC958_AES1_PRO_USERBITS_192:
+ snd_iprintf(buffer, "192bit\n");
+ break;
+ case IEC958_AES1_PRO_USERBITS_UDEF:
+ snd_iprintf(buffer, "user-defined\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Sample Bits: ");
+ switch (status[2] & IEC958_AES2_PRO_SBITS) {
+ case IEC958_AES2_PRO_SBITS_20:
+ snd_iprintf(buffer, "20 bit\n");
+ break;
+ case IEC958_AES2_PRO_SBITS_24:
+ snd_iprintf(buffer, "24 bit\n");
+ break;
+ case IEC958_AES2_PRO_SBITS_UDEF:
+ snd_iprintf(buffer, "user defined\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Word Length: ");
+ switch (status[2] & IEC958_AES2_PRO_WORDLEN) {
+ case IEC958_AES2_PRO_WORDLEN_22_18:
+ snd_iprintf(buffer, "22 bit or 18 bit\n");
+ break;
+ case IEC958_AES2_PRO_WORDLEN_23_19:
+ snd_iprintf(buffer, "23 bit or 19 bit\n");
+ break;
+ case IEC958_AES2_PRO_WORDLEN_24_20:
+ snd_iprintf(buffer, "24 bit or 20 bit\n");
+ break;
+ case IEC958_AES2_PRO_WORDLEN_20_16:
+ snd_iprintf(buffer, "20 bit or 16 bit\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ }
+}
+
+static void snd_ca0106_proc_iec958(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ u32 value;
+
+ value = snd_ca0106_ptr_read(emu, SAMPLE_RATE_TRACKER_STATUS, 0);
+ snd_iprintf(buffer, "Status: %s, %s, %s\n",
+ (value & 0x100000) ? "Rate Locked" : "Not Rate Locked",
+ (value & 0x200000) ? "SPDIF Locked" : "No SPDIF Lock",
+ (value & 0x400000) ? "Audio Valid" : "No valid audio" );
+ snd_iprintf(buffer, "Estimated sample rate: %u\n",
+ ((value & 0xfffff) * 48000) / 0x8000 );
+ if (value & 0x200000) {
+ snd_iprintf(buffer, "IEC958/SPDIF input status:\n");
+ value = snd_ca0106_ptr_read(emu, SPDIF_INPUT_STATUS, 0);
+ snd_ca0106_proc_dump_iec958(buffer, value);
+ }
+
+ snd_iprintf(buffer, "\n");
+}
+
+static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned long flags;
+ char line[64];
+ u32 reg, val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", &reg, &val) != 2)
+ continue;
+ if (reg < 0x40 && val <= 0xffffffff) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(val, emu->port + (reg & 0xfffffffc));
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ }
+ }
+}
+
+static void snd_ca0106_proc_reg_read32(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned long value;
+ unsigned long flags;
+ int i;
+ snd_iprintf(buffer, "Registers:\n\n");
+ for(i = 0; i < 0x20; i+=4) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ value = inl(emu->port + i);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "Register %02X: %08lX\n", i, value);
+ }
+}
+
+static void snd_ca0106_proc_reg_read16(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned int value;
+ unsigned long flags;
+ int i;
+ snd_iprintf(buffer, "Registers:\n\n");
+ for(i = 0; i < 0x20; i+=2) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ value = inw(emu->port + i);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "Register %02X: %04X\n", i, value);
+ }
+}
+
+static void snd_ca0106_proc_reg_read8(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned int value;
+ unsigned long flags;
+ int i;
+ snd_iprintf(buffer, "Registers:\n\n");
+ for(i = 0; i < 0x20; i+=1) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ value = inb(emu->port + i);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "Register %02X: %02X\n", i, value);
+ }
+}
+
+static void snd_ca0106_proc_reg_read1(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned long value;
+ int i,j;
+
+ snd_iprintf(buffer, "Registers\n");
+ for(i = 0; i < 0x40; i++) {
+ snd_iprintf(buffer, "%02X: ",i);
+ for (j = 0; j < 4; j++) {
+ value = snd_ca0106_ptr_read(emu, i, j);
+ snd_iprintf(buffer, "%08lX ", value);
+ }
+ snd_iprintf(buffer, "\n");
+ }
+}
+
+static void snd_ca0106_proc_reg_read2(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned long value;
+ int i,j;
+
+ snd_iprintf(buffer, "Registers\n");
+ for(i = 0x40; i < 0x80; i++) {
+ snd_iprintf(buffer, "%02X: ",i);
+ for (j = 0; j < 4; j++) {
+ value = snd_ca0106_ptr_read(emu, i, j);
+ snd_iprintf(buffer, "%08lX ", value);
+ }
+ snd_iprintf(buffer, "\n");
+ }
+}
+
+static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ char line[64];
+ unsigned int reg, channel_id , val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
+ continue;
+ if (reg < 0x80 && val <= 0xffffffff && channel_id <= 3)
+ snd_ca0106_ptr_write(emu, reg, channel_id, val);
+ }
+}
+
+static void snd_ca0106_proc_i2c_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ char line[64];
+ unsigned int reg, val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", &reg, &val) != 2)
+ continue;
+ if ((reg <= 0x7f) || (val <= 0x1ff)) {
+ snd_ca0106_i2c_write(emu, reg, val);
+ }
+ }
+}
+
+int snd_ca0106_proc_init(struct snd_ca0106 *emu)
+{
+ struct snd_info_entry *entry;
+
+ if(! snd_card_proc_new(emu->card, "iec958", &entry))
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_iec958);
+ if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32);
+ entry->c.text.write = snd_ca0106_proc_reg_write32;
+ entry->mode |= S_IWUSR;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry))
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16);
+ if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry))
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read8);
+ if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1);
+ entry->c.text.write = snd_ca0106_proc_reg_write;
+ entry->mode |= S_IWUSR;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) {
+ entry->c.text.write = snd_ca0106_proc_i2c_write;
+ entry->private_data = emu;
+ entry->mode |= S_IWUSR;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry))
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2);
+ return 0;
+}
+
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c
new file mode 100644
index 000000000..b91c7f6d1
--- /dev/null
+++ b/sound/pci/ca0106/ca_midi.c
@@ -0,0 +1,316 @@
+/*
+ * Copyright 10/16/2005 Tilman Kranz <tilde@tk-sls.de>
+ * Creative Audio MIDI, for the CA0106 Driver
+ * Version: 0.0.1
+ *
+ * Changelog:
+ * Implementation is based on mpu401 and emu10k1x and
+ * tested with ca0106.
+ * mpu401: Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * emu10k1x: Copyright (c) by Francisco Moraes <fmoraes@nc.rr.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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ */
+
+#include <linux/spinlock.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+
+#include "ca_midi.h"
+
+#define ca_midi_write_data(midi, data) midi->write(midi, data, 0)
+#define ca_midi_write_cmd(midi, data) midi->write(midi, data, 1)
+#define ca_midi_read_data(midi) midi->read(midi, 0)
+#define ca_midi_read_stat(midi) midi->read(midi, 1)
+#define ca_midi_input_avail(midi) (!(ca_midi_read_stat(midi) & midi->input_avail))
+#define ca_midi_output_ready(midi) (!(ca_midi_read_stat(midi) & midi->output_ready))
+
+static void ca_midi_clear_rx(struct snd_ca_midi *midi)
+{
+ int timeout = 100000;
+ for (; timeout > 0 && ca_midi_input_avail(midi); timeout--)
+ ca_midi_read_data(midi);
+#ifdef CONFIG_SND_DEBUG
+ if (timeout <= 0)
+ pr_err("ca_midi_clear_rx: timeout (status = 0x%x)\n",
+ ca_midi_read_stat(midi));
+#endif
+}
+
+static void ca_midi_interrupt(struct snd_ca_midi *midi, unsigned int status)
+{
+ unsigned char byte;
+
+ if (midi->rmidi == NULL) {
+ midi->interrupt_disable(midi,midi->tx_enable | midi->rx_enable);
+ return;
+ }
+
+ spin_lock(&midi->input_lock);
+ if ((status & midi->ipr_rx) && ca_midi_input_avail(midi)) {
+ if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+ ca_midi_clear_rx(midi);
+ } else {
+ byte = ca_midi_read_data(midi);
+ if(midi->substream_input)
+ snd_rawmidi_receive(midi->substream_input, &byte, 1);
+
+
+ }
+ }
+ spin_unlock(&midi->input_lock);
+
+ spin_lock(&midi->output_lock);
+ if ((status & midi->ipr_tx) && ca_midi_output_ready(midi)) {
+ if (midi->substream_output &&
+ snd_rawmidi_transmit(midi->substream_output, &byte, 1) == 1) {
+ ca_midi_write_data(midi, byte);
+ } else {
+ midi->interrupt_disable(midi,midi->tx_enable);
+ }
+ }
+ spin_unlock(&midi->output_lock);
+
+}
+
+static void ca_midi_cmd(struct snd_ca_midi *midi, unsigned char cmd, int ack)
+{
+ unsigned long flags;
+ int timeout, ok;
+
+ spin_lock_irqsave(&midi->input_lock, flags);
+ ca_midi_write_data(midi, 0x00);
+ /* ca_midi_clear_rx(midi); */
+
+ ca_midi_write_cmd(midi, cmd);
+ if (ack) {
+ ok = 0;
+ timeout = 10000;
+ while (!ok && timeout-- > 0) {
+ if (ca_midi_input_avail(midi)) {
+ if (ca_midi_read_data(midi) == midi->ack)
+ ok = 1;
+ }
+ }
+ if (!ok && ca_midi_read_data(midi) == midi->ack)
+ ok = 1;
+ } else {
+ ok = 1;
+ }
+ spin_unlock_irqrestore(&midi->input_lock, flags);
+ if (!ok)
+ pr_err("ca_midi_cmd: 0x%x failed at 0x%x (status = 0x%x, data = 0x%x)!!!\n",
+ cmd,
+ midi->get_dev_id_port(midi->dev_id),
+ ca_midi_read_stat(midi),
+ ca_midi_read_data(midi));
+}
+
+static int ca_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->midi_mode |= CA_MIDI_MODE_INPUT;
+ midi->substream_input = substream;
+ if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ ca_midi_cmd(midi, midi->reset, 1);
+ ca_midi_cmd(midi, midi->enter_uart, 1);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+}
+
+static int ca_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->midi_mode |= CA_MIDI_MODE_OUTPUT;
+ midi->substream_output = substream;
+ if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ ca_midi_cmd(midi, midi->reset, 1);
+ ca_midi_cmd(midi, midi->enter_uart, 1);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+}
+
+static int ca_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->interrupt_disable(midi,midi->rx_enable);
+ midi->midi_mode &= ~CA_MIDI_MODE_INPUT;
+ midi->substream_input = NULL;
+ if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ ca_midi_cmd(midi, midi->reset, 0);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+}
+
+static int ca_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return -ENXIO;
+
+ spin_lock_irqsave(&midi->open_lock, flags);
+
+ midi->interrupt_disable(midi,midi->tx_enable);
+ midi->midi_mode &= ~CA_MIDI_MODE_OUTPUT;
+ midi->substream_output = NULL;
+
+ if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ ca_midi_cmd(midi, midi->reset, 0);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+}
+
+static void ca_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return;
+
+ if (up) {
+ midi->interrupt_enable(midi,midi->rx_enable);
+ } else {
+ midi->interrupt_disable(midi, midi->rx_enable);
+ }
+}
+
+static void ca_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return;
+
+ if (up) {
+ int max = 4;
+ unsigned char byte;
+
+ spin_lock_irqsave(&midi->output_lock, flags);
+
+ /* try to send some amount of bytes here before interrupts */
+ while (max > 0) {
+ if (ca_midi_output_ready(midi)) {
+ if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT) ||
+ snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+ /* no more data */
+ spin_unlock_irqrestore(&midi->output_lock, flags);
+ return;
+ }
+ ca_midi_write_data(midi, byte);
+ max--;
+ } else {
+ break;
+ }
+ }
+
+ spin_unlock_irqrestore(&midi->output_lock, flags);
+ midi->interrupt_enable(midi,midi->tx_enable);
+
+ } else {
+ midi->interrupt_disable(midi,midi->tx_enable);
+ }
+}
+
+static struct snd_rawmidi_ops ca_midi_output =
+{
+ .open = ca_midi_output_open,
+ .close = ca_midi_output_close,
+ .trigger = ca_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops ca_midi_input =
+{
+ .open = ca_midi_input_open,
+ .close = ca_midi_input_close,
+ .trigger = ca_midi_input_trigger,
+};
+
+static void ca_midi_free(struct snd_ca_midi *midi)
+{
+ midi->interrupt = NULL;
+ midi->interrupt_enable = NULL;
+ midi->interrupt_disable = NULL;
+ midi->read = NULL;
+ midi->write = NULL;
+ midi->get_dev_id_card = NULL;
+ midi->get_dev_id_port = NULL;
+ midi->rmidi = NULL;
+}
+
+static void ca_rmidi_free(struct snd_rawmidi *rmidi)
+{
+ ca_midi_free(rmidi->private_data);
+}
+
+int ca_midi_init(void *dev_id, struct snd_ca_midi *midi, int device, char *name)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if ((err = snd_rawmidi_new(midi->get_dev_id_card(midi->dev_id), name, device, 1, 1, &rmidi)) < 0)
+ return err;
+
+ midi->dev_id = dev_id;
+ midi->interrupt = ca_midi_interrupt;
+
+ spin_lock_init(&midi->open_lock);
+ spin_lock_init(&midi->input_lock);
+ spin_lock_init(&midi->output_lock);
+
+ strcpy(rmidi->name, name);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &ca_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &ca_midi_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = midi;
+ rmidi->private_free = ca_rmidi_free;
+
+ midi->rmidi = rmidi;
+ return 0;
+}
+
diff --git a/sound/pci/ca0106/ca_midi.h b/sound/pci/ca0106/ca_midi.h
new file mode 100644
index 000000000..922ed3e37
--- /dev/null
+++ b/sound/pci/ca0106/ca_midi.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright 10/16/2005 Tilman Kranz <tilde@tk-sls.de>
+ * Creative Audio MIDI, for the CA0106 Driver
+ * Version: 0.0.1
+ *
+ * Changelog:
+ * See ca_midi.c
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/spinlock.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
+
+#define CA_MIDI_MODE_INPUT MPU401_MODE_INPUT
+#define CA_MIDI_MODE_OUTPUT MPU401_MODE_OUTPUT
+
+struct snd_ca_midi {
+
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *substream_input;
+ struct snd_rawmidi_substream *substream_output;
+
+ void *dev_id;
+
+ spinlock_t input_lock;
+ spinlock_t output_lock;
+ spinlock_t open_lock;
+
+ unsigned int channel;
+
+ unsigned int midi_mode;
+ int port;
+ int tx_enable, rx_enable;
+ int ipr_tx, ipr_rx;
+
+ int input_avail, output_ready;
+ int ack, reset, enter_uart;
+
+ void (*interrupt)(struct snd_ca_midi *midi, unsigned int status);
+ void (*interrupt_enable)(struct snd_ca_midi *midi, int intr);
+ void (*interrupt_disable)(struct snd_ca_midi *midi, int intr);
+
+ unsigned char (*read)(struct snd_ca_midi *midi, int idx);
+ void (*write)(struct snd_ca_midi *midi, int data, int idx);
+
+ /* get info from dev_id */
+ struct snd_card *(*get_dev_id_card)(void *dev_id);
+ int (*get_dev_id_port)(void *dev_id);
+};
+
+int ca_midi_init(void *card, struct snd_ca_midi *midi, int device, char *name);
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-12 23:32:05 +0000