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ALSA: Further rework: 

* Get rid of non-async (ie tick task) mode due to inherent brokenness
 * Get rid of nonblock mode; we never write if buffers aren't sufficient
 * Move driver init into pcm_open() instead of pcm_init()
 * Much better underrun handling
 * Better error handling in some situations
 * Add in recording functionality
 * Use smaller audio buffers to avoid glitching
 * Don't start audio buffer with silence
 * Allow device name to be overridden by target

Recording portions based on work done by Lorenzo Miori in g#633

Change-Id: I0197bdc1749c28109eb79def1e6a3e1d62d8cef3
This commit is contained in:
Solomon Peachy 2020-10-22 17:15:45 -04:00
parent 2cf75bf008
commit 46e357f1bb
2 changed files with 370 additions and 192 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

556
firmware/target/hosted/pcm-alsa.c
View File

@ -19,25 +19,16 @@
*
****************************************************************************/
/*
* Based, but heavily modified, on the example given at
* http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html
*
* This driver uses the so-called unsafe async callback method and hardcoded device
* names. It fails when the audio device is busy by other apps.
* This driver uses the so-called unsafe async callback method.
*
* To make the async callback safer, an alternative stack is installed, since
* it's run from a signal hanlder (which otherwise uses the user stack). If
* tick tasks are run from a signal handler too, please install
* an alternative stack for it too.
* it's run from a signal hanlder (which otherwise uses the user stack).
*
* TODO: Rewrite this to do it properly with multithreading
*
* Alternatively, a version using polling in a tick task is provided. While
* supposedly safer, it appears to use more CPU (however I didn't measure it
* accurately, only looked at htop). At least, in this mode the "default"
* device works which doesnt break with other apps running.
* TODO: Rewrite this to properly use multithreading and/or direct mmap()
*/
#include "autoconf.h"
@ -69,8 +60,7 @@
* default doesnt seem to work with async callback but doesn't break
* with multple applications running */
#define DEFAULT_PLAYBACK_DEVICE "plughw:0,0"
#define USE_ASYNC_CALLBACK
#define DEFAULT_CAPTURE_DEVICE "default"
static const snd_pcm_access_t access_ = SND_PCM_ACCESS_RW_INTERLEAVED; /* access mode */
#if defined(SONY_NWZ_LINUX) || defined(HAVE_FIIO_LINUX_CODEC)
@ -93,12 +83,30 @@ static sample_t *frames = NULL;
static const void *pcm_data = 0;
static size_t pcm_size = 0;
#ifdef USE_ASYNC_CALLBACK
static snd_async_handler_t *ahandler;
static snd_async_handler_t *ahandler = NULL;
static pthread_mutex_t pcm_mtx;
static char signal_stack[SIGSTKSZ];
#else
static int recursion;
static const char *playback_dev = DEFAULT_PLAYBACK_DEVICE;
#ifdef HAVE_RECORDING
static void *pcm_data_rec = DEFAULT_CAPTURE_DEVICE;
static const char *capture_dev = NULL;
static snd_pcm_stream_t current_alsa_mode; /* SND_PCM_STREAM_PLAYBACK / _CAPTURE */
#endif
static const char *current_alsa_device;
void pcm_alsa_set_playback_device(const char *device)
{
playback_dev = device;
}
#ifdef HAVE_RECORDING
void pcm_alsa_set_capture_device(const char *device)
{
capture_dev = device;
}
#endif
static int set_hwparams(snd_pcm_t *handle)
@ -108,44 +116,46 @@ static int set_hwparams(snd_pcm_t *handle)
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_malloc(&params);
/* Size playback buffers based on sample rate */
/* Size playback buffers based on sample rate.
Note these are in FRAMES, and are sized to be about 10ms
for the buffer and 2.5ms for the period */
if (pcm_sampr > SAMPR_96) {
buffer_size = MIX_FRAME_SAMPLES * 32 * 4; /* ~64k */
period_size = MIX_FRAME_SAMPLES * 4 * 4; /* ~16k */
buffer_size = MIX_FRAME_SAMPLES * 16 * 4; /* 32k */
period_size = MIX_FRAME_SAMPLES * 4 * 4; /* 4k */
} else if (pcm_sampr > SAMPR_48) {
buffer_size = MIX_FRAME_SAMPLES * 32 * 2; /* ~32k */
period_size = MIX_FRAME_SAMPLES * 4 * 2; /* ~8k */
buffer_size = MIX_FRAME_SAMPLES * 16 * 2; /* 16k */
period_size = MIX_FRAME_SAMPLES * 4 * 2; /* 2k */
} else {
buffer_size = MIX_FRAME_SAMPLES * 32; /* ~16k */
period_size = MIX_FRAME_SAMPLES * 4; /* ~4k */
buffer_size = MIX_FRAME_SAMPLES * 16; /* 4k */
period_size = MIX_FRAME_SAMPLES * 4; /* 1k */
}
/* choose all parameters */
err = snd_pcm_hw_params_any(handle, params);
if (err < 0)
{
panicf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
panicf("Broken configuration for playback: no configurations available: %s", snd_strerror(err));
goto error;
}
/* set the interleaved read/write format */
err = snd_pcm_hw_params_set_access(handle, params, access_);
if (err < 0)
{
panicf("Access type not available for playback: %s\n", snd_strerror(err));
panicf("Access type not available for playback: %s", snd_strerror(err));
goto error;
}
/* set the sample format */
err = snd_pcm_hw_params_set_format(handle, params, format);
if (err < 0)
{
logf("Sample format not available for playback: %s\n", snd_strerror(err));
logf("Sample format not available for playback: %s", snd_strerror(err));
goto error;
}
/* set the count of channels */
err = snd_pcm_hw_params_set_channels(handle, params, channels);
if (err < 0)
{
logf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err));
logf("Channels count (%i) not available for playbacks: %s", channels, snd_strerror(err));
goto error;
}
/* set the stream rate */
@ -153,13 +163,13 @@ static int set_hwparams(snd_pcm_t *handle)
err = snd_pcm_hw_params_set_rate_near(handle, params, &srate, 0);
if (err < 0)
{
logf("Rate %iHz not available for playback: %s\n", pcm_sampr, snd_strerror(err));
logf("Rate %luHz not available for playback: %s", pcm_sampr, snd_strerror(err));
goto error;
}
real_sample_rate = srate;
if (real_sample_rate != pcm_sampr)
{
logf("Rate doesn't match (requested %iHz, get %iHz)\n", pcm_sampr, real_sample_rate);
logf("Rate doesn't match (requested %luHz, get %dHz)", pcm_sampr, real_sample_rate);
err = -EINVAL;
goto error;
}
@ -168,7 +178,7 @@ static int set_hwparams(snd_pcm_t *handle)
err = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size);
if (err < 0)
{
logf("Unable to set buffer size %ld for playback: %s\n", buffer_size, snd_strerror(err));
logf("Unable to set buffer size %ld for playback: %s", buffer_size, snd_strerror(err));
goto error;
}
@ -176,7 +186,7 @@ static int set_hwparams(snd_pcm_t *handle)
err = snd_pcm_hw_params_set_period_size_near (handle, params, &period_size, NULL);
if (err < 0)
{
logf("Unable to set period size %ld for playback: %s\n", period_size, snd_strerror(err));
logf("Unable to set period size %ld for playback: %s", period_size, snd_strerror(err));
goto error;
}
@ -187,7 +197,7 @@ static int set_hwparams(snd_pcm_t *handle)
err = snd_pcm_hw_params(handle, params);
if (err < 0)
{
logf("Unable to set hw params for playback: %s\n", snd_strerror(err));
logf("Unable to set hw params for playback: %s", snd_strerror(err));
goto error;
}
@ -209,28 +219,28 @@ static int set_swparams(snd_pcm_t *handle)
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0)
{
logf("Unable to determine current swparams for playback: %s\n", snd_strerror(err));
logf("Unable to determine current swparams for playback: %s", snd_strerror(err));
goto error;
}
/* start the transfer when the buffer is half full */
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, buffer_size / 2);
if (err < 0)
{
logf("Unable to set start threshold mode for playback: %s\n", snd_strerror(err));
logf("Unable to set start threshold mode for playback: %s", snd_strerror(err));
goto error;
}
/* allow the transfer when at least period_size samples can be processed */
err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_size);
if (err < 0)
{
logf("Unable to set avail min for playback: %s\n", snd_strerror(err));
logf("Unable to set avail min for playback: %s", snd_strerror(err));
goto error;
}
/* write the parameters to the playback device */
err = snd_pcm_sw_params(handle, swparams);
if (err < 0)
{
logf("Unable to set sw params for playback: %s\n", snd_strerror(err));
logf("Unable to set sw params for playback: %s", snd_strerror(err));
goto error;
}
@ -273,20 +283,20 @@ void pcm_alsa_set_digital_volume(int vol_db_l, int vol_db_r)
dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 2);
else
dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 1);
logf("l: %d dB -> factor = %d\n", vol_db_l - 48, dig_vol_mult_l);
logf("l: %d dB -> factor = %d", vol_db_l - 48, dig_vol_mult_l);
if(r_r == 0)
dig_vol_mult_r = 1 << vol_shift_r;
else if(r_r == 1)
dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 2);
else
dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 1);
logf("r: %d dB -> factor = %d\n", vol_db_r - 48, dig_vol_mult_r);
logf("r: %d dB -> factor = %d", vol_db_r - 48, dig_vol_mult_r);
}
/* copy pcm samples to a spare buffer, suitable for snd_pcm_writei() */
static bool fill_frames(void)
static bool copy_frames(bool first)
{
ssize_t copy_n, frames_left = period_size;
ssize_t nframes, frames_left = period_size;
bool new_buffer = false;
while (frames_left > 0)
@ -294,24 +304,41 @@ static bool fill_frames(void)
if (!pcm_size)
{
new_buffer = true;
if (!pcm_play_dma_complete_callback(PCM_DMAST_OK, &pcm_data,
&pcm_size))
#ifdef HAVE_RECORDING
switch (current_alsa_mode)
{
return false;
case SND_PCM_STREAM_PLAYBACK:
#endif
if (!pcm_play_dma_complete_callback(PCM_DMAST_OK, &pcm_data, &pcm_size))
{
return false;
}
#ifdef HAVE_RECORDING
break;
case SND_PCM_STREAM_CAPTURE:
if (!pcm_play_dma_complete_callback(PCM_DMAST_OK, &pcm_data, &pcm_size))
{
return false;
}
break;
default:
break;
}
#endif
}
/* Note: This assumes stereo 16-bit */
if (pcm_size % 4)
panicf("Wrong pcm_size");
/* the compiler will optimize this test away */
copy_n = MIN((ssize_t)pcm_size/4, frames_left);
nframes = MIN((ssize_t)pcm_size/4, frames_left);
if (format == SND_PCM_FORMAT_S32_LE)
{
/* We have to convert 16-bit to 32-bit, the need to multiply the
* sample by some value so the sound is not too low */
const short *pcm_ptr = pcm_data;
sample_t *sample_ptr = &frames[2*(period_size-frames_left)];
for (int i = 0; i < copy_n; i++)
for (int i = 0; i < nframes; i++)
{
*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_l;
*sample_ptr++ = *pcm_ptr++ * dig_vol_mult_r;
@ -319,66 +346,203 @@ static bool fill_frames(void)
}
else
{
/* Rockbox and PCM have same format: memcopy */
memcpy(&frames[2*(period_size-frames_left)], pcm_data, copy_n * 4);
#ifdef HAVE_RECORDING
switch (current_alsa_mode)
{
case SND_PCM_STREAM_PLAYBACK:
#endif
/* Rockbox and PCM have same format: memcopy */
memcpy(&frames[2*(period_size-frames_left)], pcm_data, nframes * 4);
#ifdef HAVE_RECORDING
break;
case SND_PCM_STREAM_CAPTURE:
memcpy(pcm_data_rec, &frames[2*(period_size-frames_left)], nframes * 4);
break;
default:
break;
}
#endif
}
pcm_data += copy_n*4;
pcm_size -= copy_n*4;
frames_left -= copy_n;
pcm_data += nframes*4;
pcm_size -= nframes*4;
frames_left -= nframes;
if (new_buffer)
if (new_buffer && !first)
{
new_buffer = false;
pcm_play_dma_status_callback(PCM_DMAST_STARTED);
#ifdef HAVE_RECORDING
switch (current_alsa_mode)
{
case SND_PCM_STREAM_PLAYBACK:
#endif
pcm_play_dma_status_callback(PCM_DMAST_STARTED);
#ifdef HAVE_RECORDING
break;
case SND_PCM_STREAM_CAPTURE:
pcm_rec_dma_status_callback(PCM_DMAST_STARTED);
break;
default:
break;
}
#endif
}
}
return true;
}
#ifdef USE_ASYNC_CALLBACK
static void async_callback(snd_async_handler_t *ahandler)
{
int err;
if (!ahandler) return;
snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler);
if (!handle) return;
if (pthread_mutex_trylock(&pcm_mtx) != 0)
return;
#else
static void pcm_tick(void)
{
if (snd_pcm_state(handle) != SND_PCM_STATE_RUNNING)
return;
#endif
while (snd_pcm_avail_update(handle) >= period_size)
snd_pcm_state_t state = snd_pcm_state(handle);
if (state == SND_PCM_STATE_XRUN)
{
if (fill_frames())
logf("underrun!");
err = snd_pcm_recover(handle, -EPIPE, 0);
if (err < 0) {
logf("XRUN Recovery error: %s", snd_strerror(err));
goto abort;
}
}
else if (state == SND_PCM_STATE_DRAINING)
{
logf("draining...");
goto abort;
}
else if (state == SND_PCM_STATE_SETUP)
{
goto abort;
}
#ifdef HAVE_RECORDING
if (current_alsa_mode == SND_PCM_STREAM_PLAYBACK)
{
#endif
while (snd_pcm_avail_update(handle) >= period_size)
{
int err = snd_pcm_writei(handle, frames, period_size);
if (err < 0 && err != period_size && err != -EAGAIN)
if (copy_frames(false))
{
logf("Write error: written %i expected %li\n", err, period_size);
err = snd_pcm_writei(handle, frames, period_size);
if (err < 0 && err != period_size && err != -EAGAIN)
{
logf("Write error: written %i expected %li", err, period_size);
break;
}
}
else
{
logf("%s: No Data (%d).", __func__, state);
break;
}
}
else
#ifdef HAVE_RECORDING
}
else if (current_alsa_mode == SND_PCM_STREAM_CAPTURE)
{
while (snd_pcm_avail_update(handle) >= period_size)
{
logf("%s: No Data.\n", __func__);
break;
int err = snd_pcm_readi(handle, frames, period_size);
if (err < 0 && err != period_size && err != -EAGAIN)
{
logf("Read error: read %i expected %li", err, period_size);
break;
}
/* start the fake DMA transfer */
if (!copy_frames(false))
{
/* do not spam logf */
/* logf("%s: No Data.", __func__); */
break;
}
}
}
#ifdef USE_ASYNC_CALLBACK
#endif
if (snd_pcm_state(handle) == SND_PCM_STATE_PREPARED)
{
err = snd_pcm_start(handle);
if (err < 0) {
logf("cb start error: %s", snd_strerror(err));
goto abort;
}
}
abort:
pthread_mutex_unlock(&pcm_mtx);
}
static void close_hwdev(void)
{
logf("closedev (%p)", handle);
if (handle) {
snd_pcm_drain(handle);
#ifdef AUDIOHW_MUTE_ON_PAUSE
audiohw_mute(true);
#endif
if (ahandler) {
snd_async_del_handler(ahandler);
ahandler = NULL;
}
snd_pcm_close(handle);
handle = NULL;
}
current_alsa_device = NULL;
#ifdef HAVE_RECORDING
pcm_data_rec = NULL;
#endif
}
static int async_rw(snd_pcm_t *handle)
static void alsadev_cleanup(void)
{
free(frames);
frames = NULL;
close_hwdev();
}
static void open_hwdev(const char *device, snd_pcm_stream_t mode)
{
int err;
snd_pcm_sframes_t sample_size;
sample_t *samples;
#ifdef USE_ASYNC_CALLBACK
logf("opendev %s (%p)", device, handle);
if (handle && device == current_alsa_device
#ifdef HAVE_RECORDING
&& current_alsa_mode == mode
#endif
)
{
return;
}
/* Close old handle */
close_hwdev();
if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
panicf("%s(): Cannot open device %s: %s", __func__, device, snd_strerror(err));
}
last_sample_rate = 0;
pthread_mutexattr_t attr;
pthread_mutexattr_init(&attr);
pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
pthread_mutex_init(&pcm_mtx, &attr);
/* assign alternative stack for the signal handlers */
stack_t ss = {
.ss_sp = signal_stack,
@ -390,15 +554,13 @@ static int async_rw(snd_pcm_t *handle)
err = sigaltstack(&ss, NULL);
if (err < 0)
{
logf("Unable to install alternative signal stack: %s", strerror(err));
return err;
panicf("Unable to install alternative signal stack: %s", strerror(err));
}
err = snd_async_add_pcm_handler(&ahandler, handle, async_callback, NULL);
if (err < 0)
{
logf("Unable to register async handler: %s\n", snd_strerror(err));
return err;
panicf("Unable to register async handler: %s", snd_strerror(err));
}
/* only modify the stack the handler runs on */
@ -407,76 +569,17 @@ static int async_rw(snd_pcm_t *handle)
err = sigaction(SIGIO, &sa, NULL);
if (err < 0)
{
logf("Unable to install alternative signal stack: %s", strerror(err));
return err;
panicf("Unable to install alternative signal stack: %s", strerror(err));
}
#ifdef HAVE_RECORDING
current_alsa_mode = mode;
#else
(void)mode;
#endif
current_alsa_device = device;
/* fill buffer with silence to initiate playback without noisy click */
sample_size = buffer_size;
samples = calloc(1, sample_size * channels * sizeof(sample_t));
snd_pcm_format_set_silence(format, samples, sample_size);
err = snd_pcm_writei(handle, samples, sample_size);
free(samples);
if (err < 0)
{
logf("Initial write error: %s\n", snd_strerror(err));
return err;
}
if (err != (ssize_t)sample_size)
{
logf("Initial write error: written %i expected %li\n", err, sample_size);
return err;
}
snd_pcm_state_t state = snd_pcm_state(handle);
logf("PCM RW State %d", state);
if (state == SND_PCM_STATE_PREPARED)
{
err = snd_pcm_start(handle);
if (err < 0)
{
logf("Start error: %s\n", snd_strerror(err));
return err;
}
} else {
return state;
}
return 0;
}
void cleanup(void)
{
free(frames);
frames = NULL;
snd_pcm_close(handle);
handle = NULL;
}
static void open_hwdev(const char *device)
{
int err;
logf("opendev %s (%p)", device, handle);
/* Close old handle first, if needed */
if (handle) {
pcm_play_dma_stop();
snd_pcm_close(handle);
handle = NULL;
}
if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
panicf("%s(): Cannot open device %s: %s\n", __func__, device, snd_strerror(err));
}
if ((err = snd_pcm_nonblock(handle, 1)))
panicf("Could not set non-block mode: %s\n", snd_strerror(err));
last_sample_rate = 0;
atexit(alsadev_cleanup);
}
void pcm_play_dma_init(void)
@ -485,51 +588,30 @@ void pcm_play_dma_init(void)
audiohw_preinit();
open_hwdev(DEFAULT_PLAYBACK_DEVICE);
open_hwdev(playback_dev, SND_PCM_STREAM_PLAYBACK);
#ifdef USE_ASYNC_CALLBACK
pthread_mutexattr_t attr;
pthread_mutexattr_init(&attr);
pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
pthread_mutex_init(&pcm_mtx, &attr);
#else
tick_add_task(pcm_tick);
#endif
atexit(cleanup);
return;
}
void pcm_play_lock(void)
{
#ifdef USE_ASYNC_CALLBACK
pthread_mutex_lock(&pcm_mtx);
#else
if (recursion++ == 0)
tick_remove_task(pcm_tick);
#endif
}
void pcm_play_unlock(void)
{
#ifdef USE_ASYNC_CALLBACK
pthread_mutex_unlock(&pcm_mtx);
#else
if (--recursion == 0)
tick_add_task(pcm_tick);
#endif
}
static void pcm_dma_apply_settings_nolock(void)
{
logf("PCM DMA Settings %d %d", last_sample_rate, pcm_sampr);
logf("PCM DMA Settings %d %lu", last_sample_rate, pcm_sampr);
if (last_sample_rate != pcm_sampr)
{
last_sample_rate = pcm_sampr;
last_sample_rate = pcm_sampr;
#ifdef AUDIOHW_MUTE_ON_SRATE_CHANGE
// XXX AK4450 (xDuoo X3ii) needs to be muted when switching rates.
audiohw_mute(true);
#endif
snd_pcm_drop(handle);
@ -555,6 +637,8 @@ void pcm_dma_apply_settings(void)
void pcm_play_dma_pause(bool pause)
{
logf("PCM DMA pause %d", pause);
if (!handle) return;
#ifdef AUDIOHW_MUTE_ON_PAUSE
if (pause) audiohw_mute(true);
#endif
@ -566,14 +650,15 @@ void pcm_play_dma_pause(bool pause)
void pcm_play_dma_stop(void)
{
snd_pcm_nonblock(handle, 0);
snd_pcm_drain(handle);
snd_pcm_nonblock(handle, 1);
// last_sample_rate = 0;
logf("PCM DMA stop (%d)", snd_pcm_state(handle));
int err = snd_pcm_drain(handle);
if (err < 0)
if (err < 0)
logf("Drain failed: %s", snd_strerror(err));
#ifdef AUDIOHW_MUTE_ON_PAUSE
audiohw_mute(true);
#endif
logf("PCM DMA stopped");
}
void pcm_play_dma_start(const void *addr, size_t size)
@ -591,40 +676,67 @@ void pcm_play_dma_start(const void *addr, size_t size)
while (1)
{
snd_pcm_state_t state = snd_pcm_state(handle);
logf("PCM State %d", state);
logf("PCM State %d", state);
switch (state)
{
case SND_PCM_STATE_RUNNING:
#if defined(AUDIOHW_MUTE_ON_PAUSE)
audiohw_mute(false);
#endif
return;
case SND_PCM_STATE_XRUN:
{
logf("Trying to recover from error\n");
logf("Trying to recover from error");
int err = snd_pcm_recover(handle, -EPIPE, 0);
if (err < 0)
logf("Recovery failed: %s\n", snd_strerror(err));
logf("Recovery failed: %s", snd_strerror(err));
continue;
}
case SND_PCM_STATE_SETUP:
{
int err = snd_pcm_prepare(handle);
if (err < 0)
logf("Prepare error: %s\n", snd_strerror(err));
/* fall through */
logf("Prepare error: %s", snd_strerror(err));
}
/* fall through */
case SND_PCM_STATE_PREPARED:
{ /* prepared state, we need to fill the buffer with silence before
* starting */
int err = async_rw(handle);
if (err < 0) {
logf("Start error: %s\n", snd_strerror(err));
{
int err;
#if 0
/* fill buffer with silence to initiate playback without noisy click */
snd_pcm_sframes_t sample_size = buffer_size;
sample_t *samples = calloc(1, sample_size * channels * sizeof(sample_t));
snd_pcm_format_set_silence(format, samples, sample_size);
err = snd_pcm_writei(handle, samples, sample_size);
free(samples);
if (err != (ssize_t)sample_size)
{
logf("Initial write error: written %i expected %li", err, sample_size);
return;
}
#if defined(AUDIOHW_MUTE_ON_PAUSE)
audiohw_mute(false);
#else
/* Fill buffer with proper sample data */
while (snd_pcm_avail_update(handle) >= period_size)
{
if (copy_frames(true))
{
err = snd_pcm_writei(handle, frames, period_size);
if (err < 0 && err != period_size && err != -EAGAIN)
{
logf("Write error: written %i expected %li", err, period_size);
break;
}
}
}
#endif
if (err == 0)
return;
err = snd_pcm_start(handle);
if (err < 0) {
logf("start error: %s", snd_strerror(err));
}
break;
}
case SND_PCM_STATE_PAUSED:
@ -636,7 +748,7 @@ void pcm_play_dma_start(const void *addr, size_t size)
/* run until drained */
continue;
default:
logf("Unhandled state: %s\n", snd_pcm_state_name(state));
logf("Unhandled state: %s", snd_pcm_state_name(state));
return;
}
}
@ -676,40 +788,100 @@ int pcm_alsa_get_rate(void)
#ifdef HAVE_RECORDING
void pcm_rec_lock(void)
{
pcm_play_lock();
}
void pcm_rec_unlock(void)
{
pcm_play_unlock();
}
void pcm_rec_dma_init(void)
{
logf("PCM REC DMA Init");
open_hwdev(capture_dev, SND_PCM_STREAM_CAPTURE);
}
void pcm_rec_dma_close(void)
{
logf("Rec DMA Close");
close_hwdev();
}
void pcm_rec_dma_start(void *start, size_t size)
{
(void)start;
(void)size;
logf("PCM REC DMA start (%p %d)", start, size);
pcm_dma_apply_settings_nolock();
pcm_data_rec = start;
pcm_size = size;
if (!handle) return;
while (1)
{
snd_pcm_state_t state = snd_pcm_state(handle);
switch (state)
{
case SND_PCM_STATE_RUNNING:
return;
case SND_PCM_STATE_XRUN:
{
logf("Trying to recover from error");
int err = snd_pcm_recover(handle, -EPIPE, 0);
if (err < 0)
panicf("Recovery failed: %s", snd_strerror(err));
continue;
}
case SND_PCM_STATE_SETUP:
{
int err = snd_pcm_prepare(handle);
if (err < 0)
panicf("Prepare error: %s", snd_strerror(err));
}
/* fall through */
case SND_PCM_STATE_PREPARED:
{
int err = snd_pcm_start(handle);
if (err < 0)
panicf("Start error: %s", snd_strerror(err));
return;
}
case SND_PCM_STATE_PAUSED:
{ /* paused, simply resume */
pcm_play_dma_pause(0);
return;
}
case SND_PCM_STATE_DRAINING:
/* run until drained */
continue;
default:
logf("Unhandled state: %s", snd_pcm_state_name(state));
return;
}
}
}
void pcm_rec_dma_stop(void)
{
logf("Rec DMA Stop");
close_hwdev();
}
const void * pcm_rec_dma_get_peak_buffer(void)
{
return NULL;
uintptr_t addr = (uintptr_t)pcm_data_rec;
return (void*)((addr + 3) & ~3);
}
#ifdef SIMULATOR
void audiohw_set_recvol(int left, int right, int type)
{
(void)left;
(void)right;
(void)type;
}
#endif
#endif /* HAVE_RECORDING */

6
firmware/target/hosted/pcm-alsa.h
View File

@ -28,6 +28,12 @@
void pcm_alsa_set_digital_volume(int vol_db_l, int vol_db_r);
#endif
/* These two should be invoked in your audiohw_preinit() call! */
void pcm_alsa_set_playback_device(const char *device);
#if defined(HAVE_RECORDING)
void pcm_alsa_set_capture_device(const char *device);
#endif
int pcm_alsa_get_rate(void);
#endif /* __PCM_ALSA_RB_H__ */