linux/sound/hda/hdac_stream.c
Cezary Rojewski 5eb4ff884f ALSA: hda: Add code_loading parameter to stream setup
AudioDSP firmware is the one who kicks SDxFIFOS calculation when a
stream is decoupled mode. During firmware bring up procedure, there is
no firmware running and the code-loading stream is always a decoupled
one. So, there is none to trigger the calculation and we end up with
false-positive timeout (-110) messages.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20231006102857.749143-4-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2023-10-06 14:30:39 +02:00

1021 lines
27 KiB
C

// SPDX-License-Identifier: GPL-2.0-only
/*
* HD-audio stream operations
*/
#include <linux/kernel.h>
#include <linux/delay.h>
#include <linux/export.h>
#include <linux/clocksource.h>
#include <sound/compress_driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/hdaudio.h>
#include <sound/hda_register.h>
#include "trace.h"
/*
* the hdac_stream library is intended to be used with the following
* transitions. The states are not formally defined in the code but loosely
* inspired by boolean variables. Note that the 'prepared' field is not used
* in this library but by the callers during the hw_params/prepare transitions
*
* |
* stream_init() |
* v
* +--+-------+
* | unused |
* +--+----+--+
* | ^
* stream_assign() | | stream_release()
* v |
* +--+----+--+
* | opened |
* +--+----+--+
* | ^
* stream_reset() | |
* stream_setup() | | stream_cleanup()
* v |
* +--+----+--+
* | prepared |
* +--+----+--+
* | ^
* stream_start() | | stream_stop()
* v |
* +--+----+--+
* | running |
* +----------+
*/
/**
* snd_hdac_get_stream_stripe_ctl - get stripe control value
* @bus: HD-audio core bus
* @substream: PCM substream
*/
int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned int channels = runtime->channels,
rate = runtime->rate,
bits_per_sample = runtime->sample_bits,
max_sdo_lines, value, sdo_line;
/* T_AZA_GCAP_NSDO is 1:2 bitfields in GCAP */
max_sdo_lines = snd_hdac_chip_readl(bus, GCAP) & AZX_GCAP_NSDO;
/* following is from HD audio spec */
for (sdo_line = max_sdo_lines; sdo_line > 0; sdo_line >>= 1) {
if (rate > 48000)
value = (channels * bits_per_sample *
(rate / 48000)) / sdo_line;
else
value = (channels * bits_per_sample) / sdo_line;
if (value >= bus->sdo_limit)
break;
}
/* stripe value: 0 for 1SDO, 1 for 2SDO, 2 for 4SDO lines */
return sdo_line >> 1;
}
EXPORT_SYMBOL_GPL(snd_hdac_get_stream_stripe_ctl);
/**
* 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);
if (bus->spbcap) {
azx_dev->spib_addr = bus->spbcap + AZX_SPB_BASE +
AZX_SPB_INTERVAL * idx +
AZX_SPB_SPIB;
azx_dev->fifo_addr = bus->spbcap + AZX_SPB_BASE +
AZX_SPB_INTERVAL * idx +
AZX_SPB_MAXFIFO;
}
if (bus->drsmcap)
azx_dev->dpibr_addr = bus->drsmcap + AZX_DRSM_BASE +
AZX_DRSM_INTERVAL * idx;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_init);
/**
* snd_hdac_stream_start - start a stream
* @azx_dev: HD-audio core stream to start
*
* Start a stream, set start_wallclk and set the running flag.
*/
void snd_hdac_stream_start(struct hdac_stream *azx_dev)
{
struct hdac_bus *bus = azx_dev->bus;
int stripe_ctl;
trace_snd_hdac_stream_start(bus, azx_dev);
azx_dev->start_wallclk = snd_hdac_chip_readl(bus, WALLCLK);
/* enable SIE */
snd_hdac_chip_updatel(bus, INTCTL,
1 << azx_dev->index,
1 << azx_dev->index);
/* set stripe control */
if (azx_dev->stripe) {
if (azx_dev->substream)
stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream);
else
stripe_ctl = 0;
snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK,
stripe_ctl);
}
/* set DMA start and interrupt mask */
if (bus->access_sdnctl_in_dword)
snd_hdac_stream_updatel(azx_dev, SD_CTL,
0, SD_CTL_DMA_START | SD_INT_MASK);
else
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 - helper to clear stream registers and stop DMA transfers
* @azx_dev: HD-audio core stream to stop
*/
static 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 */
if (azx_dev->stripe)
snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0);
azx_dev->running = false;
}
/**
* 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_stop_streams - stop all streams
* @bus: HD-audio core bus
*/
void snd_hdac_stop_streams(struct hdac_bus *bus)
{
struct hdac_stream *stream;
list_for_each_entry(stream, &bus->stream_list, list)
snd_hdac_stream_stop(stream);
}
EXPORT_SYMBOL_GPL(snd_hdac_stop_streams);
/**
* snd_hdac_stop_streams_and_chip - stop all streams and chip if running
* @bus: HD-audio core bus
*/
void snd_hdac_stop_streams_and_chip(struct hdac_bus *bus)
{
if (bus->chip_init) {
snd_hdac_stop_streams(bus);
snd_hdac_bus_stop_chip(bus);
}
}
EXPORT_SYMBOL_GPL(snd_hdac_stop_streams_and_chip);
/**
* 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 dma_run_state;
snd_hdac_stream_clear(azx_dev);
dma_run_state = snd_hdac_stream_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START;
snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_STREAM_RESET);
/* wait for hardware to report that the stream entered reset */
snd_hdac_stream_readb_poll(azx_dev, SD_CTL, val, (val & SD_CTL_STREAM_RESET), 3, 300);
if (azx_dev->bus->dma_stop_delay && dma_run_state)
udelay(azx_dev->bus->dma_stop_delay);
snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_STREAM_RESET, 0);
/* wait for hardware to report that the stream is out of reset */
snd_hdac_stream_readb_poll(azx_dev, SD_CTL, val, !(val & SD_CTL_STREAM_RESET), 3, 300);
/* 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
* @code_loading: Whether the stream is for PCM or code-loading.
*/
int snd_hdac_stream_setup(struct hdac_stream *azx_dev, bool code_loading)
{
struct hdac_bus *bus = azx_dev->bus;
struct snd_pcm_runtime *runtime;
unsigned int val;
u16 reg;
int ret;
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 (!code_loading) {
/* Once SDxFMT is set, the controller programs SDxFIFOS to non-zero value. */
ret = snd_hdac_stream_readw_poll(azx_dev, SD_FIFOSIZE, reg,
reg & AZX_SD_FIFOSIZE_MASK, 3, 300);
if (ret)
dev_dbg(bus->dev, "polling SD_FIFOSIZE 0x%04x failed: %d\n",
AZX_REG_SD_FIFOSIZE, ret);
azx_dev->fifo_size = reg;
}
/* 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->number << 2) | (substream->stream + 1);
if (substream->pcm)
key |= (substream->pcm->device << 16);
spin_lock_irq(&bus->reg_lock);
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) {
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_locked - release the assigned stream
* @azx_dev: HD-audio core stream to release
*
* Release the stream that has been assigned by snd_hdac_stream_assign().
* The bus->reg_lock needs to be taken at a higher level
*/
void snd_hdac_stream_release_locked(struct hdac_stream *azx_dev)
{
azx_dev->opened = 0;
azx_dev->running = 0;
azx_dev->substream = NULL;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_release_locked);
/**
* 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);
snd_hdac_stream_release_locked(azx_dev);
spin_unlock_irq(&bus->reg_lock);
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_release);
/**
* snd_hdac_get_stream - return hdac_stream based on stream_tag and
* direction
*
* @bus: HD-audio core bus
* @dir: direction for the stream to be found
* @stream_tag: stream tag for stream to be found
*/
struct hdac_stream *snd_hdac_get_stream(struct hdac_bus *bus,
int dir, int stream_tag)
{
struct hdac_stream *s;
list_for_each_entry(s, &bus->stream_list, list) {
if (s->direction == dir && s->stream_tag == stream_tag)
return s;
}
return NULL;
}
EXPORT_SYMBOL_GPL(snd_hdac_get_stream);
/*
* 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_compr_stream *cstream = azx_dev->cstream;
struct snd_pcm_runtime *runtime = NULL;
struct snd_dma_buffer *dmab;
__le32 *bdl;
int i, ofs, periods, period_bytes;
int pos_adj, pos_align;
if (substream) {
runtime = substream->runtime;
dmab = snd_pcm_get_dma_buf(substream);
} else if (cstream) {
dmab = snd_pcm_get_dma_buf(cstream);
} else {
WARN(1, "No substream or cstream assigned\n");
return -EINVAL;
}
/* 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 (runtime && !azx_dev->no_period_wakeup && pos_adj > 0) {
pos_align = pos_adj;
pos_adj = DIV_ROUND_UP(pos_adj * runtime->rate, 48000);
if (!pos_adj)
pos_adj = pos_align;
else
pos_adj = roundup(pos_adj, 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, dmab, 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, dmab, azx_dev,
&bdl, ofs, period_bytes - pos_adj, 0);
else
ofs = setup_bdle(bus, dmab, 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)
{
struct snd_pcm_substream *substream = azx_dev->substream;
struct snd_compr_stream *cstream = azx_dev->cstream;
unsigned int bufsize, period_bytes;
unsigned int no_period_wakeup;
int err;
if (substream) {
bufsize = snd_pcm_lib_buffer_bytes(substream);
period_bytes = snd_pcm_lib_period_bytes(substream);
no_period_wakeup = substream->runtime->no_period_wakeup;
} else if (cstream) {
bufsize = cstream->runtime->buffer_size;
period_bytes = cstream->runtime->fragment_size;
no_period_wakeup = 0;
} else {
return -EINVAL;
}
if (bufsize != azx_dev->bufsize ||
period_bytes != azx_dev->period_bytes ||
format_val != azx_dev->format_val ||
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 = 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 u64 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, u64 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);
/*
* Calculate the optimal mult/shift values. The counter wraps
* around after ~178.9 seconds.
*/
clocks_calc_mult_shift(&cc->mult, &cc->shift, 24000000,
NSEC_PER_SEC, 178);
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;
u64 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)
* @set: true = set, false = clear
* @streams: bit flags of streams to sync
* @reg: the stream sync register address
*/
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_readl(bus, reg);
if (set)
val |= streams;
else
val &= ~streams;
_snd_hdac_chip_writel(bus, reg, val);
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_sync_trigger);
/**
* snd_hdac_stream_sync - sync with start/stop 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++)))
continue;
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++;
/*
* Perform stream reset if DMA RUN
* bit not cleared within given timeout
*/
if (timeout == 1)
snd_hdac_stream_reset(s);
}
}
}
if (!nwait)
break;
cpu_relax();
}
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_sync);
/**
* snd_hdac_stream_spbcap_enable - enable SPIB for a stream
* @bus: HD-audio core bus
* @enable: flag to enable/disable SPIB
* @index: stream index for which SPIB need to be enabled
*/
void snd_hdac_stream_spbcap_enable(struct hdac_bus *bus,
bool enable, int index)
{
u32 mask = 0;
if (!bus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL\n");
return;
}
mask |= (1 << index);
if (enable)
snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, mask);
else
snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, 0);
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_spbcap_enable);
/**
* snd_hdac_stream_set_spib - sets the spib value of a stream
* @bus: HD-audio core bus
* @azx_dev: hdac_stream
* @value: spib value to set
*/
int snd_hdac_stream_set_spib(struct hdac_bus *bus,
struct hdac_stream *azx_dev, u32 value)
{
if (!bus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL\n");
return -EINVAL;
}
writel(value, azx_dev->spib_addr);
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_set_spib);
/**
* snd_hdac_stream_get_spbmaxfifo - gets the spib value of a stream
* @bus: HD-audio core bus
* @azx_dev: hdac_stream
*
* Return maxfifo for the stream
*/
int snd_hdac_stream_get_spbmaxfifo(struct hdac_bus *bus,
struct hdac_stream *azx_dev)
{
if (!bus->spbcap) {
dev_err(bus->dev, "Address of SPB capability is NULL\n");
return -EINVAL;
}
return readl(azx_dev->fifo_addr);
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_get_spbmaxfifo);
/**
* snd_hdac_stream_drsm_enable - enable DMA resume for a stream
* @bus: HD-audio core bus
* @enable: flag to enable/disable DRSM
* @index: stream index for which DRSM need to be enabled
*/
void snd_hdac_stream_drsm_enable(struct hdac_bus *bus,
bool enable, int index)
{
u32 mask = 0;
if (!bus->drsmcap) {
dev_err(bus->dev, "Address of DRSM capability is NULL\n");
return;
}
mask |= (1 << index);
if (enable)
snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, mask);
else
snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, 0);
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_drsm_enable);
/*
* snd_hdac_stream_wait_drsm - wait for HW to clear RSM for a stream
* @azx_dev: HD-audio core stream to await RSM for
*
* Returns 0 on success and -ETIMEDOUT upon a timeout.
*/
int snd_hdac_stream_wait_drsm(struct hdac_stream *azx_dev)
{
struct hdac_bus *bus = azx_dev->bus;
u32 mask, reg;
int ret;
mask = 1 << azx_dev->index;
ret = read_poll_timeout(snd_hdac_reg_readl, reg, !(reg & mask), 250, 2000, false, bus,
bus->drsmcap + AZX_REG_DRSM_CTL);
if (ret)
dev_dbg(bus->dev, "polling RSM 0x%08x failed: %d\n", mask, ret);
return ret;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_wait_drsm);
/**
* snd_hdac_stream_set_dpibr - sets the dpibr value of a stream
* @bus: HD-audio core bus
* @azx_dev: hdac_stream
* @value: dpib value to set
*/
int snd_hdac_stream_set_dpibr(struct hdac_bus *bus,
struct hdac_stream *azx_dev, u32 value)
{
if (!bus->drsmcap) {
dev_err(bus->dev, "Address of DRSM capability is NULL\n");
return -EINVAL;
}
writel(value, azx_dev->dpibr_addr);
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_set_dpibr);
/**
* snd_hdac_stream_set_lpib - sets the lpib value of a stream
* @azx_dev: hdac_stream
* @value: lpib value to set
*/
int snd_hdac_stream_set_lpib(struct hdac_stream *azx_dev, u32 value)
{
snd_hdac_stream_writel(azx_dev, SD_LPIB, value);
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_set_lpib);
#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;
__le32 *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 = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, bus->dev,
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 = (__le32 *)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, true);
snd_hdac_dsp_unlock(azx_dev);
return azx_dev->stream_tag;
error:
snd_dma_free_pages(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);
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;
snd_dma_free_pages(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 */