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authorAndré Fabian Silva Delgado <emulatorman@parabola.nu>2015-09-08 01:01:14 -0300
committerAndré Fabian Silva Delgado <emulatorman@parabola.nu>2015-09-08 01:01:14 -0300
commite5fd91f1ef340da553f7a79da9540c3db711c937 (patch)
treeb11842027dc6641da63f4bcc524f8678263304a3 /sound/hda/hdac_stream.c
parent2a9b0348e685a63d97486f6749622b61e9e3292f (diff)
Linux-libre 4.2-gnu
Diffstat (limited to 'sound/hda/hdac_stream.c')
-rw-r--r--sound/hda/hdac_stream.c697
1 files changed, 697 insertions, 0 deletions
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
new file mode 100644
index 000000000..4c15d0acc
--- /dev/null
+++ b/sound/hda/hdac_stream.c
@@ -0,0 +1,697 @@
+/*
+ * HD-audio stream operations
+ */
+
+#include <linux/kernel.h>
+#include <linux/delay.h>
+#include <linux/export.h>
+#include <linux/clocksource.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/hdaudio.h>
+#include <sound/hda_register.h>
+#include "trace.h"
+
+/**
+ * snd_hdac_stream_init - initialize each stream (aka device)
+ * @bus: HD-audio core bus
+ * @azx_dev: HD-audio core stream object to initialize
+ * @idx: stream index number
+ * @direction: stream direction (SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE)
+ * @tag: the tag id to assign
+ *
+ * Assign the starting bdl address to each stream (device) and initialize.
+ */
+void snd_hdac_stream_init(struct hdac_bus *bus, struct hdac_stream *azx_dev,
+ int idx, int direction, int tag)
+{
+ azx_dev->bus = bus;
+ /* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
+ azx_dev->sd_addr = bus->remap_addr + (0x20 * idx + 0x80);
+ /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */
+ azx_dev->sd_int_sta_mask = 1 << idx;
+ azx_dev->index = idx;
+ azx_dev->direction = direction;
+ azx_dev->stream_tag = tag;
+ snd_hdac_dsp_lock_init(azx_dev);
+ list_add_tail(&azx_dev->list, &bus->stream_list);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_init);
+
+/**
+ * snd_hdac_stream_start - start a stream
+ * @azx_dev: HD-audio core stream to start
+ * @fresh_start: false = wallclock timestamp relative to period wallclock
+ *
+ * Start a stream, set start_wallclk and set the running flag.
+ */
+void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+
+ trace_snd_hdac_stream_start(bus, azx_dev);
+
+ azx_dev->start_wallclk = snd_hdac_chip_readl(bus, WALLCLK);
+ if (!fresh_start)
+ azx_dev->start_wallclk -= azx_dev->period_wallclk;
+
+ /* enable SIE */
+ snd_hdac_chip_updatel(bus, INTCTL, 0, 1 << azx_dev->index);
+ /* set DMA start and interrupt mask */
+ snd_hdac_stream_updateb(azx_dev, SD_CTL,
+ 0, SD_CTL_DMA_START | SD_INT_MASK);
+ azx_dev->running = true;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_start);
+
+/**
+ * snd_hdac_stream_clear - stop a stream DMA
+ * @azx_dev: HD-audio core stream to stop
+ */
+void snd_hdac_stream_clear(struct hdac_stream *azx_dev)
+{
+ snd_hdac_stream_updateb(azx_dev, SD_CTL,
+ SD_CTL_DMA_START | SD_INT_MASK, 0);
+ snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
+ azx_dev->running = false;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_clear);
+
+/**
+ * snd_hdac_stream_stop - stop a stream
+ * @azx_dev: HD-audio core stream to stop
+ *
+ * Stop a stream DMA and disable stream interrupt
+ */
+void snd_hdac_stream_stop(struct hdac_stream *azx_dev)
+{
+ trace_snd_hdac_stream_stop(azx_dev->bus, azx_dev);
+
+ snd_hdac_stream_clear(azx_dev);
+ /* disable SIE */
+ snd_hdac_chip_updatel(azx_dev->bus, INTCTL, 1 << azx_dev->index, 0);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_stop);
+
+/**
+ * snd_hdac_stream_reset - reset a stream
+ * @azx_dev: HD-audio core stream to reset
+ */
+void snd_hdac_stream_reset(struct hdac_stream *azx_dev)
+{
+ unsigned char val;
+ int timeout;
+
+ snd_hdac_stream_clear(azx_dev);
+
+ snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_STREAM_RESET);
+ udelay(3);
+ timeout = 300;
+ do {
+ val = snd_hdac_stream_readb(azx_dev, SD_CTL) &
+ SD_CTL_STREAM_RESET;
+ if (val)
+ break;
+ } while (--timeout);
+ val &= ~SD_CTL_STREAM_RESET;
+ snd_hdac_stream_writeb(azx_dev, SD_CTL, val);
+ udelay(3);
+
+ timeout = 300;
+ /* waiting for hardware to report that the stream is out of reset */
+ do {
+ val = snd_hdac_stream_readb(azx_dev, SD_CTL) &
+ SD_CTL_STREAM_RESET;
+ if (!val)
+ break;
+ } while (--timeout);
+
+ /* reset first position - may not be synced with hw at this time */
+ if (azx_dev->posbuf)
+ *azx_dev->posbuf = 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_reset);
+
+/**
+ * snd_hdac_stream_setup - set up the SD for streaming
+ * @azx_dev: HD-audio core stream to set up
+ */
+int snd_hdac_stream_setup(struct hdac_stream *azx_dev)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+ struct snd_pcm_runtime *runtime;
+ unsigned int val;
+
+ if (azx_dev->substream)
+ runtime = azx_dev->substream->runtime;
+ else
+ runtime = NULL;
+ /* make sure the run bit is zero for SD */
+ snd_hdac_stream_clear(azx_dev);
+ /* program the stream_tag */
+ val = snd_hdac_stream_readl(azx_dev, SD_CTL);
+ val = (val & ~SD_CTL_STREAM_TAG_MASK) |
+ (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT);
+ if (!bus->snoop)
+ val |= SD_CTL_TRAFFIC_PRIO;
+ snd_hdac_stream_writel(azx_dev, SD_CTL, val);
+
+ /* program the length of samples in cyclic buffer */
+ snd_hdac_stream_writel(azx_dev, SD_CBL, azx_dev->bufsize);
+
+ /* program the stream format */
+ /* this value needs to be the same as the one programmed */
+ snd_hdac_stream_writew(azx_dev, SD_FORMAT, azx_dev->format_val);
+
+ /* program the stream LVI (last valid index) of the BDL */
+ snd_hdac_stream_writew(azx_dev, SD_LVI, azx_dev->frags - 1);
+
+ /* program the BDL address */
+ /* lower BDL address */
+ snd_hdac_stream_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr);
+ /* upper BDL address */
+ snd_hdac_stream_writel(azx_dev, SD_BDLPU,
+ upper_32_bits(azx_dev->bdl.addr));
+
+ /* enable the position buffer */
+ if (bus->use_posbuf && bus->posbuf.addr) {
+ if (!(snd_hdac_chip_readl(bus, DPLBASE) & AZX_DPLBASE_ENABLE))
+ snd_hdac_chip_writel(bus, DPLBASE,
+ (u32)bus->posbuf.addr | AZX_DPLBASE_ENABLE);
+ }
+
+ /* set the interrupt enable bits in the descriptor control register */
+ snd_hdac_stream_updatel(azx_dev, SD_CTL, 0, SD_INT_MASK);
+
+ if (azx_dev->direction == SNDRV_PCM_STREAM_PLAYBACK)
+ azx_dev->fifo_size =
+ snd_hdac_stream_readw(azx_dev, SD_FIFOSIZE) + 1;
+ else
+ azx_dev->fifo_size = 0;
+
+ /* when LPIB delay correction gives a small negative value,
+ * we ignore it; currently set the threshold statically to
+ * 64 frames
+ */
+ if (runtime && runtime->period_size > 64)
+ azx_dev->delay_negative_threshold =
+ -frames_to_bytes(runtime, 64);
+ else
+ azx_dev->delay_negative_threshold = 0;
+
+ /* wallclk has 24Mhz clock source */
+ if (runtime)
+ azx_dev->period_wallclk = (((runtime->period_size * 24000) /
+ runtime->rate) * 1000);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_setup);
+
+/**
+ * snd_hdac_stream_cleanup - cleanup a stream
+ * @azx_dev: HD-audio core stream to clean up
+ */
+void snd_hdac_stream_cleanup(struct hdac_stream *azx_dev)
+{
+ snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0);
+ snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0);
+ snd_hdac_stream_writel(azx_dev, SD_CTL, 0);
+ azx_dev->bufsize = 0;
+ azx_dev->period_bytes = 0;
+ azx_dev->format_val = 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_cleanup);
+
+/**
+ * snd_hdac_stream_assign - assign a stream for the PCM
+ * @bus: HD-audio core bus
+ * @substream: PCM substream to assign
+ *
+ * Look for an unused stream for the given PCM substream, assign it
+ * and return the stream object. If no stream is free, returns NULL.
+ * The function tries to keep using the same stream object when it's used
+ * beforehand. Also, when bus->reverse_assign flag is set, the last free
+ * or matching entry is returned. This is needed for some strange codecs.
+ */
+struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus,
+ struct snd_pcm_substream *substream)
+{
+ struct hdac_stream *azx_dev;
+ struct hdac_stream *res = NULL;
+
+ /* make a non-zero unique key for the substream */
+ int key = (substream->pcm->device << 16) | (substream->number << 2) |
+ (substream->stream + 1);
+
+ list_for_each_entry(azx_dev, &bus->stream_list, list) {
+ if (azx_dev->direction != substream->stream)
+ continue;
+ if (azx_dev->opened)
+ continue;
+ if (azx_dev->assigned_key == key) {
+ res = azx_dev;
+ break;
+ }
+ if (!res || bus->reverse_assign)
+ res = azx_dev;
+ }
+ if (res) {
+ spin_lock_irq(&bus->reg_lock);
+ res->opened = 1;
+ res->running = 0;
+ res->assigned_key = key;
+ res->substream = substream;
+ spin_unlock_irq(&bus->reg_lock);
+ }
+ return res;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_assign);
+
+/**
+ * snd_hdac_stream_release - release the assigned stream
+ * @azx_dev: HD-audio core stream to release
+ *
+ * Release the stream that has been assigned by snd_hdac_stream_assign().
+ */
+void snd_hdac_stream_release(struct hdac_stream *azx_dev)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+
+ spin_lock_irq(&bus->reg_lock);
+ azx_dev->opened = 0;
+ azx_dev->running = 0;
+ azx_dev->substream = NULL;
+ spin_unlock_irq(&bus->reg_lock);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_release);
+
+/*
+ * set up a BDL entry
+ */
+static int setup_bdle(struct hdac_bus *bus,
+ struct snd_dma_buffer *dmab,
+ struct hdac_stream *azx_dev, __le32 **bdlp,
+ int ofs, int size, int with_ioc)
+{
+ __le32 *bdl = *bdlp;
+
+ while (size > 0) {
+ dma_addr_t addr;
+ int chunk;
+
+ if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES)
+ return -EINVAL;
+
+ addr = snd_sgbuf_get_addr(dmab, ofs);
+ /* program the address field of the BDL entry */
+ bdl[0] = cpu_to_le32((u32)addr);
+ bdl[1] = cpu_to_le32(upper_32_bits(addr));
+ /* program the size field of the BDL entry */
+ chunk = snd_sgbuf_get_chunk_size(dmab, ofs, size);
+ /* one BDLE cannot cross 4K boundary on CTHDA chips */
+ if (bus->align_bdle_4k) {
+ u32 remain = 0x1000 - (ofs & 0xfff);
+
+ if (chunk > remain)
+ chunk = remain;
+ }
+ bdl[2] = cpu_to_le32(chunk);
+ /* program the IOC to enable interrupt
+ * only when the whole fragment is processed
+ */
+ size -= chunk;
+ bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01);
+ bdl += 4;
+ azx_dev->frags++;
+ ofs += chunk;
+ }
+ *bdlp = bdl;
+ return ofs;
+}
+
+/**
+ * snd_hdac_stream_setup_periods - set up BDL entries
+ * @azx_dev: HD-audio core stream to set up
+ *
+ * Set up the buffer descriptor table of the given stream based on the
+ * period and buffer sizes of the assigned PCM substream.
+ */
+int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+ struct snd_pcm_substream *substream = azx_dev->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ __le32 *bdl;
+ int i, ofs, periods, period_bytes;
+ int pos_adj, pos_align;
+
+ /* reset BDL address */
+ snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0);
+ snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0);
+
+ period_bytes = azx_dev->period_bytes;
+ periods = azx_dev->bufsize / period_bytes;
+
+ /* program the initial BDL entries */
+ bdl = (__le32 *)azx_dev->bdl.area;
+ ofs = 0;
+ azx_dev->frags = 0;
+
+ pos_adj = bus->bdl_pos_adj;
+ if (!azx_dev->no_period_wakeup && pos_adj > 0) {
+ pos_align = pos_adj;
+ pos_adj = (pos_adj * runtime->rate + 47999) / 48000;
+ if (!pos_adj)
+ pos_adj = pos_align;
+ else
+ pos_adj = ((pos_adj + pos_align - 1) / pos_align) *
+ pos_align;
+ pos_adj = frames_to_bytes(runtime, pos_adj);
+ if (pos_adj >= period_bytes) {
+ dev_warn(bus->dev, "Too big adjustment %d\n",
+ pos_adj);
+ pos_adj = 0;
+ } else {
+ ofs = setup_bdle(bus, snd_pcm_get_dma_buf(substream),
+ azx_dev,
+ &bdl, ofs, pos_adj, true);
+ if (ofs < 0)
+ goto error;
+ }
+ } else
+ pos_adj = 0;
+
+ for (i = 0; i < periods; i++) {
+ if (i == periods - 1 && pos_adj)
+ ofs = setup_bdle(bus, snd_pcm_get_dma_buf(substream),
+ azx_dev, &bdl, ofs,
+ period_bytes - pos_adj, 0);
+ else
+ ofs = setup_bdle(bus, snd_pcm_get_dma_buf(substream),
+ azx_dev, &bdl, ofs,
+ period_bytes,
+ !azx_dev->no_period_wakeup);
+ if (ofs < 0)
+ goto error;
+ }
+ return 0;
+
+ error:
+ dev_err(bus->dev, "Too many BDL entries: buffer=%d, period=%d\n",
+ azx_dev->bufsize, period_bytes);
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_setup_periods);
+
+/* snd_hdac_stream_set_params - set stream parameters
+ * @azx_dev: HD-audio core stream for which parameters are to be set
+ * @format_val: format value parameter
+ *
+ * Setup the HD-audio core stream parameters from substream of the stream
+ * and passed format value
+ */
+int snd_hdac_stream_set_params(struct hdac_stream *azx_dev,
+ unsigned int format_val)
+{
+
+ unsigned int bufsize, period_bytes;
+ struct snd_pcm_substream *substream = azx_dev->substream;
+ struct snd_pcm_runtime *runtime;
+ int err;
+
+ if (!substream)
+ return -EINVAL;
+ runtime = substream->runtime;
+ bufsize = snd_pcm_lib_buffer_bytes(substream);
+ period_bytes = snd_pcm_lib_period_bytes(substream);
+
+ if (bufsize != azx_dev->bufsize ||
+ period_bytes != azx_dev->period_bytes ||
+ format_val != azx_dev->format_val ||
+ runtime->no_period_wakeup != azx_dev->no_period_wakeup) {
+ azx_dev->bufsize = bufsize;
+ azx_dev->period_bytes = period_bytes;
+ azx_dev->format_val = format_val;
+ azx_dev->no_period_wakeup = runtime->no_period_wakeup;
+ err = snd_hdac_stream_setup_periods(azx_dev);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_set_params);
+
+static cycle_t azx_cc_read(const struct cyclecounter *cc)
+{
+ struct hdac_stream *azx_dev = container_of(cc, struct hdac_stream, cc);
+
+ return snd_hdac_chip_readl(azx_dev->bus, WALLCLK);
+}
+
+static void azx_timecounter_init(struct hdac_stream *azx_dev,
+ bool force, cycle_t last)
+{
+ struct timecounter *tc = &azx_dev->tc;
+ struct cyclecounter *cc = &azx_dev->cc;
+ u64 nsec;
+
+ cc->read = azx_cc_read;
+ cc->mask = CLOCKSOURCE_MASK(32);
+
+ /*
+ * Converting from 24 MHz to ns means applying a 125/3 factor.
+ * To avoid any saturation issues in intermediate operations,
+ * the 125 factor is applied first. The division is applied
+ * last after reading the timecounter value.
+ * Applying the 1/3 factor as part of the multiplication
+ * requires at least 20 bits for a decent precision, however
+ * overflows occur after about 4 hours or less, not a option.
+ */
+
+ cc->mult = 125; /* saturation after 195 years */
+ cc->shift = 0;
+
+ nsec = 0; /* audio time is elapsed time since trigger */
+ timecounter_init(tc, cc, nsec);
+ if (force) {
+ /*
+ * force timecounter to use predefined value,
+ * used for synchronized starts
+ */
+ tc->cycle_last = last;
+ }
+}
+
+/**
+ * snd_hdac_stream_timecounter_init - initialize time counter
+ * @azx_dev: HD-audio core stream (master stream)
+ * @streams: bit flags of streams to set up
+ *
+ * Initializes the time counter of streams marked by the bit flags (each
+ * bit corresponds to the stream index).
+ * The trigger timestamp of PCM substream assigned to the given stream is
+ * updated accordingly, too.
+ */
+void snd_hdac_stream_timecounter_init(struct hdac_stream *azx_dev,
+ unsigned int streams)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+ struct snd_pcm_runtime *runtime = azx_dev->substream->runtime;
+ struct hdac_stream *s;
+ bool inited = false;
+ cycle_t cycle_last = 0;
+ int i = 0;
+
+ list_for_each_entry(s, &bus->stream_list, list) {
+ if (streams & (1 << i)) {
+ azx_timecounter_init(s, inited, cycle_last);
+ if (!inited) {
+ inited = true;
+ cycle_last = s->tc.cycle_last;
+ }
+ }
+ i++;
+ }
+
+ snd_pcm_gettime(runtime, &runtime->trigger_tstamp);
+ runtime->trigger_tstamp_latched = true;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_timecounter_init);
+
+/**
+ * snd_hdac_stream_sync_trigger - turn on/off stream sync register
+ * @azx_dev: HD-audio core stream (master stream)
+ * @streams: bit flags of streams to sync
+ */
+void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set,
+ unsigned int streams, unsigned int reg)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+ unsigned int val;
+
+ if (!reg)
+ reg = AZX_REG_SSYNC;
+ val = _snd_hdac_chip_read(l, bus, reg);
+ if (set)
+ val |= streams;
+ else
+ val &= ~streams;
+ _snd_hdac_chip_write(l, bus, reg, val);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_sync_trigger);
+
+/**
+ * snd_hdac_stream_sync - sync with start/strop trigger operation
+ * @azx_dev: HD-audio core stream (master stream)
+ * @start: true = start, false = stop
+ * @streams: bit flags of streams to sync
+ *
+ * For @start = true, wait until all FIFOs get ready.
+ * For @start = false, wait until all RUN bits are cleared.
+ */
+void snd_hdac_stream_sync(struct hdac_stream *azx_dev, bool start,
+ unsigned int streams)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+ int i, nwait, timeout;
+ struct hdac_stream *s;
+
+ for (timeout = 5000; timeout; timeout--) {
+ nwait = 0;
+ i = 0;
+ list_for_each_entry(s, &bus->stream_list, list) {
+ if (streams & (1 << i)) {
+ if (start) {
+ /* check FIFO gets ready */
+ if (!(snd_hdac_stream_readb(s, SD_STS) &
+ SD_STS_FIFO_READY))
+ nwait++;
+ } else {
+ /* check RUN bit is cleared */
+ if (snd_hdac_stream_readb(s, SD_CTL) &
+ SD_CTL_DMA_START)
+ nwait++;
+ }
+ }
+ i++;
+ }
+ if (!nwait)
+ break;
+ cpu_relax();
+ }
+}
+EXPORT_SYMBOL_GPL(snd_hdac_stream_sync);
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+/**
+ * snd_hdac_dsp_prepare - prepare for DSP loading
+ * @azx_dev: HD-audio core stream used for DSP loading
+ * @format: HD-audio stream format
+ * @byte_size: data chunk byte size
+ * @bufp: allocated buffer
+ *
+ * Allocate the buffer for the given size and set up the given stream for
+ * DSP loading. Returns the stream tag (>= 0), or a negative error code.
+ */
+int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format,
+ unsigned int byte_size, struct snd_dma_buffer *bufp)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+ u32 *bdl;
+ int err;
+
+ snd_hdac_dsp_lock(azx_dev);
+ spin_lock_irq(&bus->reg_lock);
+ if (azx_dev->running || azx_dev->locked) {
+ spin_unlock_irq(&bus->reg_lock);
+ err = -EBUSY;
+ goto unlock;
+ }
+ azx_dev->locked = true;
+ spin_unlock_irq(&bus->reg_lock);
+
+ err = bus->io_ops->dma_alloc_pages(bus, SNDRV_DMA_TYPE_DEV_SG,
+ byte_size, bufp);
+ if (err < 0)
+ goto err_alloc;
+
+ azx_dev->substream = NULL;
+ azx_dev->bufsize = byte_size;
+ azx_dev->period_bytes = byte_size;
+ azx_dev->format_val = format;
+
+ snd_hdac_stream_reset(azx_dev);
+
+ /* reset BDL address */
+ snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0);
+ snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0);
+
+ azx_dev->frags = 0;
+ bdl = (u32 *)azx_dev->bdl.area;
+ err = setup_bdle(bus, bufp, azx_dev, &bdl, 0, byte_size, 0);
+ if (err < 0)
+ goto error;
+
+ snd_hdac_stream_setup(azx_dev);
+ snd_hdac_dsp_unlock(azx_dev);
+ return azx_dev->stream_tag;
+
+ error:
+ bus->io_ops->dma_free_pages(bus, bufp);
+ err_alloc:
+ spin_lock_irq(&bus->reg_lock);
+ azx_dev->locked = false;
+ spin_unlock_irq(&bus->reg_lock);
+ unlock:
+ snd_hdac_dsp_unlock(azx_dev);
+ return err;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_dsp_prepare);
+
+/**
+ * snd_hdac_dsp_trigger - start / stop DSP loading
+ * @azx_dev: HD-audio core stream used for DSP loading
+ * @start: trigger start or stop
+ */
+void snd_hdac_dsp_trigger(struct hdac_stream *azx_dev, bool start)
+{
+ if (start)
+ snd_hdac_stream_start(azx_dev, true);
+ else
+ snd_hdac_stream_stop(azx_dev);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_dsp_trigger);
+
+/**
+ * snd_hdac_dsp_cleanup - clean up the stream from DSP loading to normal
+ * @azx_dev: HD-audio core stream used for DSP loading
+ * @dmab: buffer used by DSP loading
+ */
+void snd_hdac_dsp_cleanup(struct hdac_stream *azx_dev,
+ struct snd_dma_buffer *dmab)
+{
+ struct hdac_bus *bus = azx_dev->bus;
+
+ if (!dmab->area || !azx_dev->locked)
+ return;
+
+ snd_hdac_dsp_lock(azx_dev);
+ /* reset BDL address */
+ snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0);
+ snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0);
+ snd_hdac_stream_writel(azx_dev, SD_CTL, 0);
+ azx_dev->bufsize = 0;
+ azx_dev->period_bytes = 0;
+ azx_dev->format_val = 0;
+
+ bus->io_ops->dma_free_pages(bus, dmab);
+ dmab->area = NULL;
+
+ spin_lock_irq(&bus->reg_lock);
+ azx_dev->locked = false;
+ spin_unlock_irq(&bus->reg_lock);
+ snd_hdac_dsp_unlock(azx_dev);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_dsp_cleanup);
+#endif /* CONFIG_SND_HDA_DSP_LOADER */