Merge tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It was a busy development cycle at this time, as you can see a wide
  range of changes in diffstat.  There are no big changes but many
  refactoring and improvements.  Here we go some highlights:

  ALSA core:
   - Procfs codes were cleaned up to use seq_file
   - Procfs can be opt out via Kconfig (only for EXPERT)
   - Two types of jack API were unified finally; now both kctl and input
     jack devs are handled via a single function call.

  HD-audio:
   - Continued code restructuring for the future ASoC driver; now HDA
     controller driver is split to a core helper module.
   - Preliminary codes for Skylake audio support in HDA core.
   - Proper i915 gfx power well management for SKL & co
   - Enabled runtime PM as default for Intel HDMI/DP codecs
   - Newer Tegra chip supports
   - More quirks for Dell headsets, Alienware (with CA0132), etc.
   - A couple of DRM ELD helper API functions

  ASoC:
   - Support for loading ASoC topology maps from firmware, intended to
     be used to allow self-describing DSP firmware images to be built
     which can map controls added by the DSP to userspace without the
     kernel needing to know about individual DSP firmwares
   - Lots of refactoring to avoid direct access to snd_soc_codec where
     it's not needed supporting future refactoring
   - Big refactoring, cleanup and enhancement for the Wolfson ADSP
     driver
   - Cleanup series for TI TAS2552 and R-CAR drivers
   - Fixes and improvements on RT56xx codecs
   - Support for TI TAS571x power amplifiers
   - Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs
   - Support for x86 systems with RT5650 and Qualcomm Storm
   - Support for Mediatek AFE (Audio Front End) unit
   - Other various small fixes to ASoC codec drivers

  Firewire:
   - Enhanced to allow non-blocking streams to use timestamp
     synchronization
   - Improve support for DM1500 and BeBoBv3

  Misc:
   - Cleanup of old pci API functions over all PCI sound drivers
   - Fix long-standing regression of the old powermac i2c setup"

* tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (533 commits)
  ALSA: pcm: Fix pcm_class sysfs output
  ALSA: hda-beep: Update authors dead email address
  ASoC: wm_adsp: Move DSP Rate controls into the codec
  ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
  ALSA: hda: provide default bus io ops extended hdac
  ALSA: hda: add hda link cleanup routine
  ALSA: hda: add hdac_ext stream creation and cleanup routines
  ASoC: rsrc-card: remove unused ret
  ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
  ASoC: mediatek: Add machine driver for rt5650 rt5676 codec
  ASoC: mediatek: Add machine driver for MAX98090 codec
  ASoC: mediatek: Add AFE platform driver
  ASoC: rsnd: remove io from rsnd_mod
  ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working()
  ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr()
  ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA
  ...

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 000000000000..4c15d0accc9e
--- /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 */