void ADR_CALL
ALSAAudioDevice::update() {
int ret;
int sample_len;
int sample_left;
char* sample_buf;
sample_buf = m_buffer;
sample_len = m_buffer_size / 4;
sample_left = read(sample_len, sample_buf);
while (sample_left > 0) {
ret = snd_pcm_writei(m_pcm_handle, sample_buf, sample_left);
if (ret == -EAGAIN || (ret > 0 && ret < sample_left)) {
snd_pcm_wait(m_pcm_handle, 10);
} else if (ret == -ESTRPIPE) {
do {
snd_pcm_wait(m_pcm_handle, 10);
ret = snd_pcm_resume(m_pcm_handle);
} while (ret == -EAGAIN);
snd_pcm_prepare(m_pcm_handle);
} else if (ret == -EPIPE) {
snd_pcm_prepare(m_pcm_handle);
}
if (ret > 0) {
sample_buf += ret * 4;
sample_left -= ret;
}
}
}
static ssize_t
sa_alsa_read( simpleaudio *sa, void *buf, size_t nframes )
{
ssize_t frames_read = 0;
snd_pcm_t *pcm = (snd_pcm_t *)sa->backend_handle;
while ( frames_read < nframes ) {
ssize_t r;
void * data = buf+frames_read*sa->backend_framesize;
ssize_t count = nframes-frames_read;
r = snd_pcm_readi(pcm, data, count);
if ( r >= 0 ) {
frames_read += r;
if ( r != count )
fprintf(stderr, "#short+%zd#\n", r);
continue;
}
if (r == -EPIPE) { // Underrun
fprintf(stderr, "#");
snd_pcm_prepare(pcm);
} else {
fprintf(stderr, "snd_pcm_readi: %s\n", snd_strerror(r));
if (r == -EAGAIN || r== -ESTRPIPE)
snd_pcm_wait(pcm, 1000);
else
return r;
}
}
// fprintf(stderr,("[%zd]\n"), frames_read);
return frames_read;
}
static ssize_t pcm_read(u_char *data, size_t rcount)
{
ssize_t r;
size_t result = 0;
size_t count = rcount;
if (count != chunk_size) {
count = chunk_size;
}
while (count > 0) {
r = readi_func(handle, data, count);
if (r == -EAGAIN || (r >= 0 && (size_t)r < count)) {
snd_pcm_wait(handle, 1000);
} else if (r == -EPIPE) {
xrun();
} else if (r == -ESTRPIPE) {
suspend();
} else if (r < 0) {
error(_("read error: %s"), snd_strerror(r));
exit(EXIT_FAILURE);
}
if (r > 0) {
if (vumeter)
compute_max_peak(data, r * hwparams.channels);
result += r;
count -= r;
data += r * bits_per_frame / 8;
}
}
return rcount;
}
int audio_write_alsa(cst_audiodev *ad, void *samples, int num_frames)
{
size_t frame_size;
ssize_t res;
snd_pcm_t *pcm_handle;
char *buf = (char *) samples;
int frames_to_write;
frames_to_write = num_frames;
pcm_handle = (snd_pcm_t *) ad->platform_data;
frame_size = BELL_AUDIO_16BIT * ad->channels;
while (frames_to_write > 0)
{
res = snd_pcm_writei(pcm_handle, buf, frames_to_write); //Note pcm_handle open in blocked mode
if (res != frames_to_write)
{
if (res > 0 && res < frames_to_write)
{
snd_pcm_wait(pcm_handle, 100);
}
else if (recover_from_error(pcm_handle, res) < 0)
{
return -1;
}
}
if (res >0)
{
frames_to_write -= res;
buf += res * frame_size;
}
}
return num_frames;
}
static int write_to_pcm(int size, const struct snd_pcm_container *sndpcm,
int offset)
{
int err;
while (size > 0) {
err = snd_pcm_writei(sndpcm->handle, sndpcm->buffer + offset,
size);
if (err == -EAGAIN || (err >= 0 && err < size)) {
snd_pcm_wait(sndpcm->handle, 500);
} else if (err == -EPIPE) {
loge(E_WRITEPCM S_UNDERRUN, "%s(%d)",
snd_strerror(err), err);
snd_pcm_prepare(sndpcm->handle);
} else if (err < 0) {
loge(E_WRITEPCM, "%s(%d)", snd_strerror(err), err);
return -1;
}
if (err > 0) {
size -= err;
offset += err * sndpcm->frame_bits / 8;
}
}
return 0;
}
static int read_from_pcm(struct pcm_container *sndpcm,
int frames, struct bat *bat)
{
int err = 0;
int offset = 0;
int remain = frames;
while (remain > 0) {
err = snd_pcm_readi(sndpcm->handle,
sndpcm->buffer + offset, remain);
if (err == -EAGAIN || (err >= 0 && err < remain)) {
snd_pcm_wait(sndpcm->handle, 500);
} else if (err == -EPIPE) {
snd_pcm_prepare(sndpcm->handle);
fprintf(bat->err, _("Overrun: %s(%d)\n"),
snd_strerror(err), err);
} else if (err < 0) {
fprintf(bat->err, _("Read PCM device error: %s(%d)\n"),
snd_strerror(err), err);
return err;
}
if (err > 0) {
remain -= err;
offset += err * sndpcm->frame_bits / 8;
}
}
return 0;
}
static ssize_t
pcm_write (short *data, size_t count)
{
ssize_t r;
ssize_t result = 0;
while (count > 0)
{
r = snd_pcm_writei (AHandle, data, count);
if (r == -EAGAIN || (r >= 0 && (size_t) r < count))
{
snd_pcm_wait (AHandle, 1000);
}
else if (r == -EPIPE)
{
xrun ();
}
else if (r == -ESTRPIPE)
{
suspend ();
}
else if (r < 0)
{
fprintf (stderr, "write error: %s", snd_strerror (r));
exit (EXIT_FAILURE);
}
if (r > 0)
{
result += r;
count -= r;
data += r;
}
}
return result;
}
static size_t pcm_read(snd_pcm_uframes_t chunk_size, u_char *data, size_t rcount)
{
ssize_t r;
size_t result = 0;
size_t count = rcount;
if (count != chunk_size) {
count = chunk_size;
}
while (count > 0) {
r = snd_pcm_readi(handle, data, count);
if (r == -EAGAIN || (r >= 0 && (size_t)r < count)) {
snd_pcm_wait(handle, 1000);
}else if (r == -EPIPE){
xrun(SND_PCM_STREAM_CAPTURE);
}else if (r < 0) {
printf("read error: %s", snd_strerror(r));
exit(EXIT_FAILURE);
}
if (r > 0) {
result += r;
count -= r;
data += r * bits_per_frame / 8;
}
}
return result;
}
qint64 OutputALSA::writeAudio(unsigned char *data, qint64 maxSize)
{
if((maxSize = qMin(maxSize, m_prebuf_size - m_prebuf_fill)) > 0)
{
memmove(m_prebuf + m_prebuf_fill, data, maxSize);
m_prebuf_fill += maxSize;
}
snd_pcm_uframes_t l = snd_pcm_bytes_to_frames(pcm_handle, m_prebuf_fill);
while (l >= m_chunk_size)
{
snd_pcm_wait(pcm_handle, 10);
long m;
if ((m = alsa_write(m_prebuf, m_chunk_size)) >= 0)
{
l -= m;
m = snd_pcm_frames_to_bytes(pcm_handle, m); // convert frames to bytes
m_prebuf_fill -= m;
memmove(m_prebuf, m_prebuf + m, m_prebuf_fill); //move data to begin
}
else
return -1;
}
return maxSize;
}
static guint
gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstAlsaSink *alsa;
gint err;
gint cptr;
gint16 *ptr = data;
alsa = GST_ALSA_SINK (asink);
if (alsa->iec958 && alsa->need_swap) {
guint i;
GST_DEBUG_OBJECT (asink, "swapping bytes");
for (i = 0; i < length / 2; i++) {
ptr[i] = GUINT16_SWAP_LE_BE (ptr[i]);
}
}
GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);
cptr = length / alsa->bytes_per_sample;
GST_ALSA_SINK_LOCK (asink);
while (cptr > 0) {
/* start by doing a blocking wait for free space. Set the timeout
* to 4 times the period time */
err = snd_pcm_wait (alsa->handle, (4 * alsa->period_time / 1000));
if (err < 0) {
GST_DEBUG_OBJECT (asink, "wait error, %d", err);
} else {
GST_DELAY_SINK_LOCK (asink);
err = snd_pcm_writei (alsa->handle, ptr, cptr);
GST_DELAY_SINK_UNLOCK (asink);
}
GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);
if (err < 0) {
GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err));
if (err == -EAGAIN) {
continue;
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
goto write_error;
}
continue;
}
ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
cptr -= err;
}
GST_ALSA_SINK_UNLOCK (asink);
return length - (cptr * alsa->bytes_per_sample);
write_error:
{
GST_ALSA_SINK_UNLOCK (asink);
return length; /* skip one period */
}
}
static int
write_device(AudioDevice *ad, unsigned char *data, int size)
{
ALSA_data *alsa = (ALSA_data *)ad->private_data;
snd_pcm_sframes_t r;
ssize_t unit = snd_pcm_samples_to_bytes(alsa->fd, 1) * ad->channels;
snd_pcm_uframes_t count = size / unit;
while (count > 0) {
if ((r = snd_pcm_writei(alsa->fd, data, count)) == -EAGAIN) {
//debug_message_fnc(" EAGAIN\n");
snd_pcm_wait(alsa->fd, 1000);
} else if (r > 0) {
//debug_message_fnc(" wrote %d bytes\n", (int)(r * unit));
ad->bytes_written += r * unit;
count -= r;
data += r * unit;
} else if (r == -EPIPE) {
debug_message_fnc("EPIPE: ");
if (snd_pcm_state(alsa->fd) == SND_PCM_STATE_XRUN) {
if ((r = snd_pcm_prepare(alsa->fd)) < 0) {
debug_message("failed\n");
warning_fnc("snd_pcm_prepare() failed.");
} else {
debug_message("OK\n");
}
}
} else {
warning_fnc(" r = %d < 0...\n", (int)r);
}
}
return 1;
}
ssize_t SNDWAV_WritePcm(SNDPCMContainer_t *sndpcm, size_t wcount)
{
ssize_t r;
ssize_t result = 0;
uint8_t *data = sndpcm->data_buf;
if (wcount < sndpcm->chunk_size) {
snd_pcm_format_set_silence(sndpcm->format,
data + wcount * sndpcm->bits_per_frame / 8,
(sndpcm->chunk_size - wcount) * sndpcm->channels);
wcount = sndpcm->chunk_size;
}
while (wcount > 0) {
r = snd_pcm_writei(sndpcm->handle, data, wcount);
if (r == -EAGAIN || (r >= 0 && (size_t)r < wcount)) {
snd_pcm_wait(sndpcm->handle, 1000);
} else if (r == -EPIPE) {
snd_pcm_prepare(sndpcm->handle);
fprintf(stderr, "<<<<<<<<<<<<<<< Buffer Underrun >>>>>>>>>>>>>>>\n");
} else if (r == -ESTRPIPE) {
fprintf(stderr, "<<<<<<<<<<<<<<< Need suspend >>>>>>>>>>>>>>>\n");
} else if (r < 0) {
fprintf(stderr, "Error snd_pcm_writei: [%s]", snd_strerror(r));
exit(-1);
}
if (r > 0) {
result += r;
wcount -= r;
data += r * sndpcm->bits_per_frame / 8;
}
}
return result;
}
ssize_t SNDWAV_ReadPcm(SNDPCMContainer_t *sndpcm, size_t rcount)
{
ssize_t r;
size_t result = 0;
size_t count = rcount;
uint8_t *data = sndpcm->data_buf;
if (count != sndpcm->chunk_size) {
count = sndpcm->chunk_size;
}
while (count > 0) {
r = snd_pcm_readi(sndpcm->handle, data, count);
if (r == -EAGAIN || (r >= 0 && (size_t)r < count)) {
snd_pcm_wait(sndpcm->handle, 1000);
} else if (r == -EPIPE) {
snd_pcm_prepare(sndpcm->handle);
fprintf(stderr, "<<<<<<<<<<<<<<< Buffer Underrun >>>>>>>>>>>>>>>\n");
} else if (r == -ESTRPIPE) {
fprintf(stderr, "<<<<<<<<<<<<<<< Need suspend >>>>>>>>>>>>>>>\n");
} else if (r < 0) {
fprintf(stderr, "Error snd_pcm_writei: [%s]", snd_strerror(r));
exit(-1);
}
if (r > 0) {
result += r;
count -= r;
data += r * sndpcm->bits_per_frame / 8;
}
}
return rcount;
}
static ssize_t pcm_write(u_char *data, size_t count)
{
ssize_t r;
ssize_t result = 0;
if (sleep_min == 0 &&
count < chunk_size) {
snd_pcm_format_set_silence(hwparams.format, data + count * bits_per_frame / 8, (chunk_size - count) * hwparams.channels);
count = chunk_size;
}
while (count > 0) {
r = snd_pcm_writei(handle, data, count);
if (r == -EAGAIN || (r >= 0 && (size_t)r < count)) {
snd_pcm_wait(handle, 1000);
} else if (r == -EPIPE) {
xrun();
} else if (r == -ESTRPIPE) {
suspend();
} else if (r < 0) {
error(_("write error: %s"), snd_strerror(r));
exit(EXIT_FAILURE);
}
if (r > 0) {
result += r;
count -= r;
data += r * bits_per_frame / 8;
}
}
return result;
}
static snd_pcm_t *open_audiofd( char *device_name, int capture, int rate, int channels, int period, int nperiods ) {
int err;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
if ((err = snd_pcm_open(&(handle), device_name, capture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK )) < 0) {
printf("Capture open error: %s\n", snd_strerror(err));
return NULL;
}
if ((err = set_hwparams(handle, hwparams,SND_PCM_ACCESS_RW_INTERLEAVED, rate, channels, period, nperiods )) < 0) {
printf("Setting of hwparams failed: %s\n", snd_strerror(err));
return NULL;
}
if ((err = set_swparams(handle, swparams, period)) < 0) {
printf("Setting of swparams failed: %s\n", snd_strerror(err));
return NULL;
}
snd_pcm_start( handle );
snd_pcm_wait( handle, 200 );
return handle;
}
void PAPlayer::WaitForStream()
{
// should we wait for our other stream as well?
// currently we don't.
if (!m_pStream[m_currentStream])
{
snd_pcm_wait(m_pStream[m_currentStream], -1);
}
}
void write( const char *data, qint64 len )
{
if ( !handle )
return;
int count=0;
qLog(QAudioOutput)<<"frames to write out = "<<
snd_pcm_bytes_to_frames( handle, (int)len )<<" ("<<len<<") bytes";
while ( len > 0 ) {
int err=0;
int frames = snd_pcm_bytes_to_frames( handle, (int)len );
#ifdef ALSA_USE_AVAILABLE
if(frames < (int)period_size)
return;
int available = snd_pcm_avail_update(handle);
qLog(QAudioOutput) <<"available space = "<<available;
if(available == 0) {
while(available < frames) {
snd_pcm_wait(handle,period_size/1000);
usleep(period_size*10);
available = snd_pcm_avail_update(handle);
qLog(QAudioOutput) <<"->available space = "<<available;
count++;
if((count > 5)||(available < 0))
return;
}
}
#endif
err = snd_pcm_writei( handle, data, frames );
// Handle errors
if ( err >= 0 ) {
if(err == 0) count++;
int bytes = snd_pcm_frames_to_bytes( handle, err );
qLog(QAudioOutput) << QString("write out = %1").arg(bytes).toLatin1().constData();
data += bytes;
len -= bytes;
} else {
count++;
qLog(QAudioOutput) <<"err = "<<err;
err = xrun_recovery(err);
}
if(count > 5) {
qLog(QAudioOutput) <<"failing to write, close() and re-open() to try and recover!";
close();
open();
snd_pcm_prepare(handle);
break;
}
}
}
/* Play from alsa_buf as many chunks as possible. Move the remaining data
* to the beginning of the buffer. Return the number of bytes written
* or -1 on error. */
static int play_buf_chunks ()
{
int written = 0;
while (alsa_buf_fill >= chunk_size) {
int err;
err = snd_pcm_writei (handle, alsa_buf + written,
chunk_size / bytes_per_frame);
if (err == -EAGAIN) {
if (snd_pcm_wait(handle, 500) < 0)
logit ("snd_pcm_wait() failed");
}
else if (err == -EPIPE) {
logit ("underrun!");
if ((err = snd_pcm_prepare(handle)) < 0) {
error ("Can't recover after underrun: %s",
snd_strerror(err));
/* TODO: reopen the device */
return -1;
}
}
else if (err == -ESTRPIPE) {
logit ("Suspend, trying to resume");
while ((err = snd_pcm_resume(handle))
== -EAGAIN)
sleep (1);
if (err < 0) {
logit ("Failed, restarting");
if ((err = snd_pcm_prepare(handle))
< 0) {
error ("Failed to restart "
"device: %s.",
snd_strerror(err));
return -1;
}
}
}
else if (err < 0) {
error ("Can't play: %s", snd_strerror(err));
return -1;
}
else {
int written_bytes = err * bytes_per_frame;
written += written_bytes;
alsa_buf_fill -= written_bytes;
debug ("Played %d bytes", written_bytes);
}
}
debug ("%d bytes remain in alsa_buf", alsa_buf_fill);
memmove (alsa_buf, alsa_buf + written, alsa_buf_fill);
return written * bytes_per_frame;
}
void SoundOutput_alsa::wait()
{
if(handle == nullptr)
{
System::sleep(100);
return;
}
/* wait upto 1 second */
snd_pcm_wait(handle, 1000);
}
void CL_SoundOutput_alsa::wait()
{
if(handle == NULL)
{
CL_System::sleep(100);
return;
}
/* wait upto 1 second */
snd_pcm_wait(handle, 1000);
}
void AlsaLayer::audioCallback()
{
if (!playbackHandle_ or !captureHandle_)
return;
notifyIncomingCall();
snd_pcm_wait(playbackHandle_, 20);
int playbackAvailFrames = 0;
if (not safeUpdate(playbackHandle_, playbackAvailFrames))
return;
unsigned framesToGet = urgentRingBuffer_.availableForGet(MainBuffer::DEFAULT_ID);
if (framesToGet > 0) {
// Urgent data (dtmf, incoming call signal) come first.
framesToGet = std::min(framesToGet, (unsigned)playbackAvailFrames);
playbackBuff_.setFormat(audioFormat_);
playbackBuff_.resize(framesToGet);
urgentRingBuffer_.get(playbackBuff_, MainBuffer::DEFAULT_ID);
playbackBuff_.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
playbackBuff_.interleave(playbackIBuff_);
write(playbackIBuff_.data(), framesToGet, playbackHandle_);
// Consume the regular one as well (same amount of frames)
Manager::instance().getMainBuffer().discard(framesToGet, MainBuffer::DEFAULT_ID);
} else {
// regular audio data
playback(playbackAvailFrames);
}
if (ringtoneHandle_) {
AudioLoop *file_tone = Manager::instance().getTelephoneFile();
int ringtoneAvailFrames = 0;
if (not safeUpdate(ringtoneHandle_, ringtoneAvailFrames))
return;
playbackBuff_.setFormat(audioFormat_);
playbackBuff_.resize(ringtoneAvailFrames);
if (file_tone) {
DEBUG("playback gain %d", playbackGain_);
file_tone->getNext(playbackBuff_, playbackGain_);
}
playbackBuff_.interleave(playbackIBuff_);
write(playbackIBuff_.data(), ringtoneAvailFrames, ringtoneHandle_);
}
// Additionally handle the mic's audio stream
if (is_capture_running_)
capture();
}
static int
alsa_audio_deliver(audio_decoder_t *ad, int samples, int64_t pts, int epoch)
{
decoder_t *d = (decoder_t *)ad;
media_pipe_t *mp = ad->ad_mp;
int c;
retry:
c = snd_pcm_wait(d->h, 100);
if(c >= 0) {
c = snd_pcm_avail_update(d->h);
}
if(c == -EPIPE) {
snd_pcm_prepare(d->h);
usleep(100000);
TRACE(TRACE_DEBUG, "ALSA", "Audio underrun");
d->samples = 0;
goto retry;
}
c = MIN(d->max_frames_per_write, c);
uint8_t *planes[8] = {0};
planes[0] = d->tmp;
c = avresample_read(ad->ad_avr, planes, c);
snd_pcm_status_t *status;
int err;
snd_pcm_status_alloca(&status);
if ((err = snd_pcm_status(d->h, status)) >= 0) {
if(pts != AV_NOPTS_VALUE) {
snd_htimestamp_t hts;
snd_pcm_status_get_trigger_htstamp(status, &hts);
int64_t ts = hts.tv_sec * 1000000LL + hts.tv_nsec / 1000;
ts += d->samples * 1000000LL / ad->ad_out_sample_rate;
hts_mutex_lock(&mp->mp_clock_mutex);
mp->mp_audio_clock_avtime = ts;
mp->mp_audio_clock = pts;
mp->mp_audio_clock_epoch = epoch;
hts_mutex_unlock(&mp->mp_clock_mutex);
}
}
snd_pcm_sframes_t fr;
if(!snd_pcm_delay(d->h, &fr))
ad->ad_delay = 1000000L * fr / ad->ad_out_sample_rate;
c = snd_pcm_writei(d->h, d->tmp, c);
d->samples += c;
return 0;
}
unsigned int CAESinkALSA::AddPackets(uint8_t *data, unsigned int frames, bool hasAudio, bool blocking)
{
if (!m_pcm)
{
SoftResume();
if(!m_pcm)
return 0;
CLog::Log(LOGDEBUG, "CAESinkALSA - the grAEken is hunger, feed it (I am the downmost fallback - fix your code)");
}
int ret;
ret = snd_pcm_avail(m_pcm);
if (ret < 0)
{
HandleError("snd_pcm_avail", ret);
ret = 0;
}
if ((unsigned int)ret < frames)
{
if(blocking)
{
ret = snd_pcm_wait(m_pcm, m_timeout);
if (ret < 0)
HandleError("snd_pcm_wait", ret);
}
else
return 0;
}
ret = snd_pcm_writei(m_pcm, (void*)data, frames);
if (ret < 0)
{
HandleError("snd_pcm_writei(1)", ret);
ret = snd_pcm_writei(m_pcm, (void*)data, frames);
if (ret < 0)
{
HandleError("snd_pcm_writei(2)", ret);
ret = 0;
}
}
if ( ret > 0 && snd_pcm_state(m_pcm) == SND_PCM_STATE_PREPARED)
snd_pcm_start(m_pcm);
return ret;
}
int aplaypop()
{
if (pcm_handle == NULL && aplaypop_open() != 0)
return -1;
fprintf(stderr, "snd_pcm_state() = %s\n",
snd_pcm_state_name(snd_pcm_state(pcm_handle)));
int err = snd_pcm_prepare(pcm_handle);
if (err != 0) {
fprintf(stderr, "snd_pcm_start(): %s\n", snd_strerror(err));
return -1;
}
int16_t *ptr = (int16_t *)bin_data;
int len = sizeof(bin_data) / frame_bytes;
while (len > 0) {
// The unit of the buffersize depends on the function. Sometimes it is
// given in bytes, sometimes the number of frames has to be specified.
// One frame is the sample data vector for all channels.
// For 16 Bit stereo data, one frame has a length of four bytes.
snd_pcm_sframes_t frames = snd_pcm_writei(pcm_handle, ptr, len);
fprintf(stderr, " snd_pcm_writei(h, ptr, %d) = %ld\n", len, frames);
if (frames == -EAGAIN) {
frames = 0;
} else if (frames < 0) { // -EPIPE, -ESTRPIPE
frames = snd_pcm_recover(pcm_handle, frames, 1/*silent*/);
if (frames < 0) {
fprintf(stderr, "writei/recover failed: %s\n", snd_strerror(err));
break;
}
} else if (frames < len) {
fprintf(stderr, "Short write (expected %d, wrote %ld)\n", len, frames);
} else {
break; // All done
}
len -= frames * frame_bytes;
ptr += frames * frame_bytes;
snd_pcm_wait(pcm_handle, 10/*ms*/);
}
err = snd_pcm_drain(pcm_handle);
if (err != 0) {
fprintf(stderr, "snd_pcm_start(): %s\n", snd_strerror(err));
return -1;
}
return 0;
}
/*****************************************************************************
* Demux: Processes the audio frame
*****************************************************************************/
static int Demux( demux_t *p_demux )
{
demux_sys_t *p_sys = p_demux->p_sys;
block_t *p_block = NULL;
do
{
if( p_block )
{
es_out_Send( p_demux->out, p_sys->p_es, p_block );
p_block = NULL;
}
/* Wait for data */
int i_wait = snd_pcm_wait( p_sys->p_alsa_pcm, 10 ); /* See poll() comment in oss.c */
switch( i_wait )
{
case 1:
{
p_block = GrabAudio( p_demux );
if( p_block )
es_out_Control( p_demux->out, ES_OUT_SET_PCR, p_block->i_pts );
}
/* FIXME: this is a copy paste from below. Shouldn't be needed
* twice. */
case -EPIPE:
/* xrun */
snd_pcm_prepare( p_sys->p_alsa_pcm );
break;
case -ESTRPIPE:
{
/* suspend */
int i_resume = snd_pcm_resume( p_sys->p_alsa_pcm );
if( i_resume < 0 && i_resume != -EAGAIN ) snd_pcm_prepare( p_sys->p_alsa_pcm );
break;
}
/* </FIXME> */
}
} while( p_block && p_sys->i_next_demux_date > 0 &&
p_block->i_pts < p_sys->i_next_demux_date );
if( p_block )
es_out_Send( p_demux->out, p_sys->p_es, p_block );
return 1;
}
BOOL auTransferData(Audio* self, snd_pcm_sframes_t (*transfer)(snd_pcm_t*, void*, snd_pcm_uframes_t))
{
int numFramesTransferred = 0, error = 0;
int numFramesLeft = self->bufferNumFrames;
auSample_t* p = self->sampleBuffer;
while((numFramesLeft > 0) && self->threadShouldContinueRunning)
{
error = numFramesTransferred = transfer(self->device, p, numFramesLeft);
if(numFramesTransferred < 0)
{
//fprintf(stderr, "Audio.c: audio device error while transferring samples: %s, attempting to recover... ", snd_strerror(error));
switch(error)
{
case -EPIPE: //overflow / underflow
snd_pcm_wait(self->device, 100);
if((error = snd_pcm_avail(self->device)) < 0) //broken pipe
usleep(10000); //wait for more samples to come
else numFramesLeft = 0; //overrun, skip remaining samples;
error = snd_pcm_prepare(self->device);
break;
case -ESTRPIPE:
while(((error = snd_pcm_resume(self->device)) == -EAGAIN) && self->threadShouldContinueRunning)
sleep(1);
if(error == -ENOSYS) error = snd_pcm_prepare(self->device);
break;
}
if(error < 0)
{
//fprintf(stderr, "Aborting\n");
self->threadShouldContinueRunning = NO;
break;
}
else
{
//fprintf(stderr, "Okay\n");
numFramesTransferred = 0;
}
}
p += numFramesTransferred * self->numChannels;
numFramesLeft -= numFramesTransferred;
}
return (numFramesLeft == 0) ? YES : NO;
}
snd_pcm_sframes_t
XcapAlsaReadiQueue::readi(snd_pcm_t *handle, unsigned char *samples, int chunksize, int queue) {
unsigned char *qdata;
snd_pcm_sframes_t ret = 0;
// not initialized or error
if(n == 0 || this->chunksize != chunksize)
return -1;
if(samples == NULL)
return 0;
// no more queue
if(queue==0) {
// queue is empty
if(datapool.size() == 0)
return snd_pcm_readi(handle, samples, chunksize);
// queue is not empty, dequeue
qdata = datapool.front();
ret = *((snd_pcm_sframes_t*) qdata);
bcopy(qdata+sizeof(snd_pcm_sframes_t), samples, this->chunkbyte);
datapool.pop_front();
freepool.push_front(qdata);
}
// read and queue
if(freepool.size() == 0) {
ga_error("AlsaReadiQueue: no more queue, dropping eldest\n");
qdata = datapool.front();
datapool.pop_front();
} else {
qdata = freepool.front();
freepool.pop_front();
}
*((snd_pcm_sframes_t*) qdata) = snd_pcm_readi(handle,
qdata+sizeof(snd_pcm_sframes_t),
chunksize);
if(*((snd_pcm_sframes_t*) qdata) < 0) {
freepool.push_front(qdata);
if(ret > 0 && *((snd_pcm_sframes_t*) qdata) == -EAGAIN) {
ga_error("DEBUG: snd_pcm_wait in queue->readi (qsize=%d)\n", datapool.size());
snd_pcm_wait(handle, 1000);
return ret;
}
return *((snd_pcm_sframes_t*) qdata);
}
datapool.push_back(qdata);
//
return ret;
}
void *write_pcmdata(void *arg)
{
dbg("Enter write_pcmdata thread.....\n");
int ret;
msg_t pcm_data;
snd_pcm_sframes_t avail;
common_data_t *p_common_data = (common_data_t *)arg;
dbg("PCM handle name = '%s'\n",snd_pcm_name(p_common_data->handle));
dbg("period szie = %lu\n",p_common_data->period_size);
while (1) {
sem_wait(&p_common_data->queue.wait);
if((read_item_from_queue(&p_common_data->queue, &pcm_data)) == EMPTY){
dbg("queue is empty...\n");
//TODO:block?
continue;
}
snd_pcm_prepare(p_common_data->handle);
ret = snd_pcm_writei(p_common_data->handle, pcm_data.pcm_msg, p_common_data->period_size);
if (ret == -EAGAIN) { //means try again
/* wait 1000ms for pcm device to become ready */
dbg_alsa("EAGAIN error, Sleep for 1000ms\n");
snd_pcm_wait(p_common_data->handle, 1000);
}
else if (ret == -EPIPE) {
/* EPIPE means underrun */
snd_pcm_prepare(p_common_data->handle);
dbg_alsa("underrun occurred\n");
}
else if(ret == -ESTRPIPE) {
dbg_alsa("Need suspend\n");
}
else if (ret < 0) {
dbg_alsa("error from writei: %s\n", snd_strerror(ret));
}
ret = snd_pcm_state(p_common_data->handle);
dbg("state is %d\n", ret);
sem_post(&p_common_data->queue.full);
}
}
static void* alsa_audio_start(void *aux)
{
audio_fifo_t *af = aux;
snd_pcm_t *h = NULL;
int c;
int cur_channels = 0;
int cur_rate = 0;
audio_fifo_data_t *afd;
for (;;) {
afd = audio_get(af);
if (!h || cur_rate != afd->rate || cur_channels != afd->channels) {
if (h) snd_pcm_close(h);
cur_rate = afd->rate;
cur_channels = afd->channels;
h = alsa_open("default", cur_rate, cur_channels);
if (!h) {
fprintf(stderr, "Unable to open ALSA device (%d channels, %d Hz), dying\n",
cur_channels, cur_rate);
exit(1);
}
}
c = snd_pcm_wait(h, 1000);
if (c >= 0)
c = snd_pcm_avail_update(h);
if (c == -EPIPE)
snd_pcm_prepare(h);
snd_pcm_writei(h, afd->samples, afd->nsamples);
writeFile( &afd->samples );
zfree(afd);
}
}
void *
asource_threadproc(void *arg) {
int r;
unsigned char *fbuffer;
//
if(asource_init(NULL) < 0) {
exit(-1);
}
if((fbuffer = (unsigned char*) malloc(audioparam.chunk_bytes)) == NULL) {
ga_error("Audio source: malloc failed (%d bytes) - %s\n",
audioparam.chunk_bytes, strerror(errno));
exit(-1);
}
//
ga_error("Audio source thread started: tid=%ld\n", ga_gettid());
//
while(true) {
#ifdef WIN32
r = ga_wasapi_read(&audioparam, fbuffer, audioparam.chunk_size);
if(r < 0) {
ga_error("Audio source: WASAPI read failed.\n");
break;
}
#else
r = snd_pcm_readi(audioparam.handle, fbuffer, audioparam.chunk_size);
if(r == -EAGAIN) {
snd_pcm_wait(audioparam.handle, 1000);
continue;
} else if(r < 0) {
ga_error("Audio source: ALSA read failed - %s\n",
snd_strerror(r));
break;
}
#endif
audio_source_buffer_fill(fbuffer, r);
}
//
ga_error("audio capture thread terminated.\n");
//
return NULL;
}