int sndout_alsa_write_nb(const void *samples, int len)
{
snd_pcm_sframes_t left;
int ret;
len /= 2;
if (channels == 2)
len /= 2;
left = snd_pcm_avail(handle);
if (left >= 0 && left < len)
return 0;
ret = snd_pcm_writei(handle, samples, len);
if (ret < 0) {
ret = snd_pcm_recover(handle, ret, 1);
if (ret != 0 && failure_counter++ < 5)
fprintf(stderr, PFX "snd_pcm_recover: %d\n", ret);
if (silent_period)
snd_pcm_writei(handle, silent_period, period_size);
snd_pcm_writei(handle, samples, len);
}
return len;
}
void FFTWidget::processFFT()
{
int avail = snd_pcm_avail(capture_handle);
if(avail<0)
{
fprintf (stderr, "Cannot read audio samples (%s)\n",
snd_strerror (avail));
exit(1);
}
if(avail > (2*1024))
{
short input[2*1024];
int err;
if ((err = snd_pcm_readi(capture_handle, input, 1024)) != 1024)
{
fprintf(stderr, "read from audio interface failed (%s)\n",
snd_strerror (err));
exit(1);
}
for(int i=0; i<1024; i++)
{
buffer[i]= window_function[i] * std::complex<float>(input[2*i+1], input[2*i]);
}
FFT::dft(buffer);
update();
}
}
size_t write_avail()
{
snd_pcm_sframes_t rc = snd_pcm_avail(pcm);
if (rc < 0)
{
runnable = false;
return 0;
}
return snd_pcm_frames_to_bytes(pcm, rc) / sizeof(T);
}
void audio_buffer_stats(struct audio *audio, int *samples, int *stutter)
{
/* Accuracy is like... irrelevant */
*stutter = audio->underruns;
audio->underruns = 0;
*samples = snd_pcm_avail(audio->pcm);
if (*samples < 0) {
*samples = 0;
}
}
qint64 ALSAWriter::write(const QByteArray &arr)
{
if (!readyWrite())
return 0;
const int samples = arr.size() / sizeof(float);
const int to_write = samples / channels;
const int bytes = samples * sample_size;
if (int_samples.size() < bytes)
int_samples.resize(bytes);
switch (sample_size)
{
case 4:
convert_samples((const float *)arr.constData(), samples, (qint32 *)int_samples.constData(), mustSwapChn ? channels : 0);
break;
case 2:
convert_samples((const float *)arr.constData(), samples, (qint16 *)int_samples.constData(), mustSwapChn ? channels : 0);
break;
case 1:
convert_samples((const float *)arr.constData(), samples, (qint8 *)int_samples.constData(), mustSwapChn ? channels : 0);
break;
}
switch (snd_pcm_state(snd))
{
case SND_PCM_STATE_XRUN:
if (!snd_pcm_prepare(snd))
{
const int silence = snd_pcm_avail(snd) - to_write;
if (silence > 0)
{
QByteArray silenceArr(silence * channels * sample_size, 0);
snd_pcm_writei(snd, silenceArr.constData(), silence);
}
}
break;
case SND_PCM_STATE_PAUSED:
snd_pcm_pause(snd, false);
break;
default:
break;
}
int ret = snd_pcm_writei(snd, int_samples.constData(), to_write);
if (ret < 0 && ret != -EPIPE && snd_pcm_recover(snd, ret, false))
{
QMPlay2Core.logError("ALSA :: " + tr("Playback error"));
err = true;
return 0;
}
return arr.size();
}
void sndout_alsa_wait(void)
{
snd_pcm_sframes_t left;
while (1)
{
left = snd_pcm_avail(handle);
if (left < 0 || left >= buffer_size / 2)
break;
usleep(4000);
}
}
static size_t alsa_write_avail(void *data)
{
alsa_t *alsa = (alsa_t*)data;
snd_pcm_sframes_t avail = snd_pcm_avail(alsa->pcm);
if (avail < 0)
{
//RARCH_WARN("[ALSA]: snd_pcm_avail() failed: %s\n", snd_strerror(avail));
return alsa->buffer_size;
}
return snd_pcm_frames_to_bytes(alsa->pcm, avail);
}
static int alsa_bufferspace(void)
{
#ifdef HAVE_SND_PCM_AVAIL
snd_pcm_sframes_t space = snd_pcm_avail(handle);
#else
snd_pcm_sframes_t space = snd_pcm_avail_update(handle);
#endif
/* keep alsa values real. Values < 0 mean errors, call to alsa_write
* will resume. */
if (space < 0 || space > alsa_bufsize)
space = alsa_bufsize;
return space;
}
inline void audio_play(char* outbuf, int samples, void* priv_data)
{
#ifdef DEBUGALSA
snd_pcm_sframes_t avail=snd_pcm_avail (alsa_handle);
syslog(LOG_DEBUG,"AUDIO_PLAY: Available buffer in ALSA:%d, Sample size %d\n",(int)avail,samples);
#endif
int err = snd_pcm_writei(alsa_handle, outbuf, samples);
if (err < 0)
{
err = snd_pcm_recover(alsa_handle, err, 0);
if (err < 0)
syslog(LOG_DEBUG, "AUDIO_PLAY: snd_pcm_writei failed: %s\n", snd_strerror(err));
}
}
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;
}
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;
}
void AudioProvider::update(AudioData& audioData)
{
audioData.channels = channels;
audioData.sampleRate = sampleRate;
unsigned available = std::min((unsigned) snd_pcm_avail(handle), maxFrames);
audioData.samples.resize(available * channels);
int status = snd_pcm_readi(handle, audioData.samples.data(), available);
if(status < 0)
{
OUTPUT_WARNING("Lost audio stream (" << status << "), recovering...");
snd_pcm_recover(handle, status, 1);
ASSERT(channels <= 4);
short buf[4];
VERIFY(snd_pcm_readi(handle, buf, 1) >= 0);
audioData.samples.clear();
}
}
INT64 DAUDIO_GetBytePosition(void* id, int isSource, INT64 javaBytePos) {
AlsaPcmInfo* info = (AlsaPcmInfo*) id;
int ret;
INT64 result = javaBytePos;
snd_pcm_state_t state;
state = snd_pcm_state(info->handle);
if (state != SND_PCM_STATE_XRUN) {
#ifdef GET_POSITION_METHOD2
snd_timestamp_t* ts;
snd_pcm_uframes_t framesAvail;
// note: slight race condition if this is called simultaneously from 2 threads
ret = snd_pcm_status(info->handle, info->positionStatus);
if (ret != 0) {
ERROR1("ERROR in snd_pcm_status: %s\n", snd_strerror(ret));
result = javaBytePos;
} else {
// calculate from time value, or from available bytes
framesAvail = snd_pcm_status_get_avail(info->positionStatus);
result = estimatePositionFromAvail(info, isSource, javaBytePos, framesAvail * info->frameSize);
}
#endif
#ifdef GET_POSITION_METHOD3
snd_pcm_uframes_t framesAvail;
ret = snd_pcm_avail(info->handle, &framesAvail);
if (ret != 0) {
ERROR1("ERROR in snd_pcm_avail: %s\n", snd_strerror(ret));
result = javaBytePos;
} else {
result = estimatePositionFromAvail(info, isSource, javaBytePos, framesAvail * info->frameSize);
}
#endif
#ifdef GET_POSITION_METHOD1
result = estimatePositionFromAvail(info, isSource, javaBytePos, DAUDIO_GetAvailable(id, isSource));
#endif
}
//printf("getbyteposition: javaBytePos=%d , return=%d\n", (int) javaBytePos, (int) result);
return result;
}
unsigned int CAESinkALSA::AddPackets(uint8_t *data, unsigned int frames, bool hasAudio)
{
if (!m_pcm)
return 0;
if (snd_pcm_state(m_pcm) == SND_PCM_STATE_PREPARED)
snd_pcm_start(m_pcm);
int ret;
ret = snd_pcm_avail(m_pcm);
if (ret < 0)
{
HandleError("snd_pcm_avail", ret);
ret = 0;
}
if ((unsigned int)ret < frames);
{
ret = snd_pcm_wait(m_pcm, m_timeout);
if (ret < 0)
HandleError("snd_pcm_wait", ret);
}
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;
}
}
return ret;
}
static
void *
audio_thread(void *param)
{
struct pollfd *fds = NULL;
nfds_t nfds = 0;
static char buf[16 * 1024];
ppb_message_loop_mark_thread_unsuitable();
nfds = do_rebuild_fds(&fds);
pthread_barrier_wait(&stream_list_update_barrier);
if (nfds == 0)
goto quit;
while (1) {
if (g_atomic_int_get(&terminate_thread))
goto quit;
int res = poll(fds, nfds, 10 * 1000);
if (res == -1) {
if (errno == EINTR)
continue;
trace_error("%s, poll, errno=%d\n", __func__, errno);
continue;
}
if (res == 0 || fds == NULL)
continue;
if (fds[0].revents)
drain_wakeup_pipe(fds[0].fd);
if (g_atomic_int_get(&rebuild_fds)) {
nfds = do_rebuild_fds(&fds);
pthread_barrier_wait(&stream_list_update_barrier);
if (nfds == 0)
goto quit;
}
for (uintptr_t k = 1; k < nfds; k ++) {
unsigned short revents = 0;
audio_stream *as = g_hash_table_lookup(stream_by_fd_ht, GINT_TO_POINTER(fds[k].fd));
// check if stream was deleted
if (!as)
continue;
snd_pcm_poll_descriptors_revents(as->pcm, &fds[k], 1, &revents);
if (revents & (~(POLLIN | POLLOUT))) {
trace_warning("%s, revents have unexpected flags set (%u)\n", __func__,
(unsigned int)revents);
recover_pcm(as->pcm);
}
if (revents & (POLLIN | POLLOUT)) {
int paused = g_atomic_int_get(&as->paused);
snd_pcm_sframes_t frame_count = snd_pcm_avail(as->pcm);
if (revents & POLLIN) {
// POLLIN
const size_t frame_size = 1 * sizeof(int16_t); // mono 16-bit
const size_t max_segment_length = MIN(as->sample_frame_count * frame_size,
sizeof(buf));
size_t to_process = frame_count * frame_size;
while (to_process > 0) {
snd_pcm_sframes_t frames_read;
const size_t segment_length = MIN(to_process, max_segment_length);
frames_read = snd_pcm_readi(as->pcm, buf, segment_length / frame_size);
if (frames_read < 0) {
trace_warning("%s, snd_pcm_readi error %d\n", __func__,
(int)frames_read);
recover_pcm(as->pcm);
continue;
}
if (!paused && as->capture_cb)
as->capture_cb(buf, frames_read * frame_size, 0, as->cb_user_data);
to_process -= frames_read * frame_size;
}
} else {
// POLLOUT
const size_t frame_size = 2 * sizeof(int16_t); // stereo 16-bit
const size_t max_segment_length = MIN(as->sample_frame_count * frame_size,
sizeof(buf));
size_t to_process = frame_count * frame_size;
while (to_process > 0) {
snd_pcm_sframes_t frames_written;
const size_t segment_length = MIN(to_process, max_segment_length);
if (paused || !as->playback_cb)
memset(buf, 0, segment_length);
else
as->playback_cb(buf, segment_length, 0, as->cb_user_data);
frames_written = snd_pcm_writei(as->pcm, buf, segment_length / frame_size);
if (frames_written < 0) {
trace_warning("%s, snd_pcm_writei error %d\n", __func__,
(int)frames_written);
recover_pcm(as->pcm);
continue;
}
to_process -= frames_written * frame_size;
}
}
}
}
}
quit:
free(fds);
return NULL;
}
int main(int argc, char *argv[]) {
const char *dev;
int r, cap, count = 0;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_status_t *status;
snd_pcm_t *pcm;
unsigned rate = 44100;
unsigned periods = 2;
snd_pcm_uframes_t boundary, buffer_size = 44100/10; /* 100s */
int dir = 1;
struct timespec start, last_timestamp = { 0, 0 };
uint64_t start_us, last_us = 0;
snd_pcm_sframes_t last_avail = 0, last_delay = 0;
struct pollfd *pollfds;
int n_pollfd;
int64_t sample_count = 0;
struct sched_param sp;
r = -1;
#ifdef _POSIX_PRIORITY_SCHEDULING
sp.sched_priority = 5;
r = pthread_setschedparam(pthread_self(), SCHED_RR, &sp);
#endif
if (r)
printf("Could not get RT prio. :(\n");
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_status_alloca(&status);
r = clock_gettime(CLOCK_MONOTONIC, &start);
assert(r == 0);
start_us = timespec_us(&start);
dev = argc > 1 ? argv[1] : "front:AudioPCI";
cap = argc > 2 ? atoi(argv[2]) : 0;
if (cap == 0)
r = snd_pcm_open(&pcm, dev, SND_PCM_STREAM_PLAYBACK, 0);
else
r = snd_pcm_open(&pcm, dev, SND_PCM_STREAM_CAPTURE, 0);
assert(r == 0);
r = snd_pcm_hw_params_any(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 0);
assert(r == 0);
r = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
assert(r == 0);
r = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE);
assert(r == 0);
r = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, NULL);
assert(r == 0);
r = snd_pcm_hw_params_set_channels(pcm, hwparams, 2);
assert(r == 0);
r = snd_pcm_hw_params_set_periods_integer(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_set_periods_near(pcm, hwparams, &periods, &dir);
assert(r == 0);
r = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, &buffer_size);
assert(r == 0);
r = snd_pcm_hw_params(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_current(pcm, hwparams);
assert(r == 0);
r = snd_pcm_sw_params_current(pcm, swparams);
assert(r == 0);
if (cap == 0)
r = snd_pcm_sw_params_set_avail_min(pcm, swparams, 1);
else
r = snd_pcm_sw_params_set_avail_min(pcm, swparams, 0);
assert(r == 0);
r = snd_pcm_sw_params_set_period_event(pcm, swparams, 0);
assert(r == 0);
r = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
assert(r == 0);
r = snd_pcm_sw_params_set_start_threshold(pcm, swparams, buffer_size);
assert(r == 0);
r = snd_pcm_sw_params_get_boundary(swparams, &boundary);
assert(r == 0);
r = snd_pcm_sw_params_set_stop_threshold(pcm, swparams, boundary);
assert(r == 0);
r = snd_pcm_sw_params_set_tstamp_mode(pcm, swparams, SND_PCM_TSTAMP_ENABLE);
assert(r == 0);
r = snd_pcm_sw_params(pcm, swparams);
assert(r == 0);
r = snd_pcm_prepare(pcm);
assert(r == 0);
r = snd_pcm_sw_params_current(pcm, swparams);
assert(r == 0);
/* assert(snd_pcm_hw_params_is_monotonic(hwparams) > 0); */
n_pollfd = snd_pcm_poll_descriptors_count(pcm);
assert(n_pollfd > 0);
pollfds = malloc(sizeof(struct pollfd) * n_pollfd);
assert(pollfds);
r = snd_pcm_poll_descriptors(pcm, pollfds, n_pollfd);
assert(r == n_pollfd);
printf("Starting. Buffer size is %u frames\n", (unsigned int) buffer_size);
if (cap) {
r = snd_pcm_start(pcm);
assert(r == 0);
}
for (;;) {
snd_pcm_sframes_t avail, delay;
struct timespec now, timestamp;
unsigned short revents;
int handled = 0;
uint64_t now_us, timestamp_us;
snd_pcm_state_t state;
unsigned long long pos;
r = poll(pollfds, n_pollfd, 0);
assert(r >= 0);
r = snd_pcm_poll_descriptors_revents(pcm, pollfds, n_pollfd, &revents);
assert(r == 0);
if (cap == 0)
assert((revents & ~POLLOUT) == 0);
else
assert((revents & ~POLLIN) == 0);
avail = snd_pcm_avail(pcm);
assert(avail >= 0);
r = snd_pcm_status(pcm, status);
assert(r == 0);
/* This assertion fails from time to time. ALSA seems to be broken */
/* assert(avail == (snd_pcm_sframes_t) snd_pcm_status_get_avail(status)); */
/* printf("%lu %lu\n", (unsigned long) avail, (unsigned long) snd_pcm_status_get_avail(status)); */
snd_pcm_status_get_htstamp(status, ×tamp);
delay = snd_pcm_status_get_delay(status);
state = snd_pcm_status_get_state(status);
r = clock_gettime(CLOCK_MONOTONIC, &now);
assert(r == 0);
assert(!revents || avail > 0);
if ((!cap && avail) || (cap && (unsigned)avail >= buffer_size)) {
snd_pcm_sframes_t sframes;
static const uint16_t psamples[2] = { 0, 0 };
uint16_t csamples[2];
if (cap == 0)
sframes = snd_pcm_writei(pcm, psamples, 1);
else
sframes = snd_pcm_readi(pcm, csamples, 1);
assert(sframes == 1);
handled = 1;
sample_count++;
}
if (!handled &&
memcmp(×tamp, &last_timestamp, sizeof(timestamp)) == 0 &&
avail == last_avail &&
delay == last_delay) {
/* This is boring */
continue;
}
now_us = timespec_us(&now);
timestamp_us = timespec_us(×tamp);
if (cap == 0)
pos = (unsigned long long) ((sample_count - handled - delay) * 1000000LU / 44100);
else
pos = (unsigned long long) ((sample_count - handled + delay) * 1000000LU / 44100);
if (count++ % 50 == 0)
printf("Elapsed\tCPU\tALSA\tPos\tSamples\tavail\tdelay\trevents\thandled\tstate\n");
printf("%llu\t%llu\t%llu\t%llu\t%llu\t%li\t%li\t%i\t%i\t%i\n",
(unsigned long long) (now_us - last_us),
(unsigned long long) (now_us - start_us),
(unsigned long long) (timestamp_us ? timestamp_us - start_us : 0),
pos,
(unsigned long long) sample_count,
(signed long) avail,
(signed long) delay,
revents,
handled,
state);
if (cap == 0)
/** When this assert is hit, most likely something bad
* happened, i.e. the avail jumped suddenly. */
assert((unsigned) avail <= buffer_size);
last_avail = avail;
last_delay = delay;
last_timestamp = timestamp;
last_us = now_us;
}
return 0;
}
/**
* The process callback for this JACK application.
* It is called by JACK at the appropriate times.
*/
int process (jack_nframes_t nframes, void *arg) {
int rlen;
int err;
snd_pcm_sframes_t delay = target_delay;
int i;
delay = (num_periods*period_size)-snd_pcm_avail( alsa_handle ) ;
delay -= jack_frames_since_cycle_start( client );
// Do it the hard way.
// this is for compensating xruns etc...
if( delay > (target_delay+max_diff) ) {
snd_pcm_rewind( alsa_handle, delay - target_delay );
output_new_delay = (int) delay;
delay = target_delay;
// Set the resample_rate... we need to adjust the offset integral, to do this.
// first look at the PI controller, this code is just a special case, which should never execute once
// everything is swung in.
offset_integral = - (resample_mean - static_resample_factor) * catch_factor * catch_factor2;
// Also clear the array. we are beginning a new control cycle.
for( i=0; i<smooth_size; i++ )
offset_array[i] = 0.0;
}
if( delay < (target_delay-max_diff) ) {
output_new_delay = (int) delay;
while ((target_delay-delay) > 0) {
snd_pcm_uframes_t to_write = ((target_delay-delay) > 512) ? 512 : (target_delay-delay);
snd_pcm_writei( alsa_handle, tmpbuf, to_write );
delay += to_write;
}
delay = target_delay;
// Set the resample_rate... we need to adjust the offset integral, to do this.
offset_integral = - (resample_mean - static_resample_factor) * catch_factor * catch_factor2;
// Also clear the array. we are beginning a new control cycle.
for( i=0; i<smooth_size; i++ )
offset_array[i] = 0.0;
}
/* ok... now we should have target_delay +- max_diff on the alsa side.
*
* calculate the number of frames, we want to get.
*/
double offset = delay - target_delay;
// Save offset.
offset_array[(offset_differential_index++)% smooth_size ] = offset;
// Build the mean of the windowed offset array
// basically fir lowpassing.
double smooth_offset = 0.0;
for( i=0; i<smooth_size; i++ )
smooth_offset +=
offset_array[ (i + offset_differential_index-1) % smooth_size] * window_array[i];
smooth_offset /= (double) smooth_size;
// this is the integral of the smoothed_offset
offset_integral += smooth_offset;
// Clamp offset.
// the smooth offset still contains unwanted noise
// which would go straigth onto the resample coeff.
// it only used in the P component and the I component is used for the fine tuning anyways.
if( fabs( smooth_offset ) < pclamp )
smooth_offset = 0.0;
// ok. now this is the PI controller.
// u(t) = K * ( e(t) + 1/T \int e(t') dt' )
// K = 1/catch_factor and T = catch_factor2
double current_resample_factor = static_resample_factor - smooth_offset / (double) catch_factor - offset_integral / (double) catch_factor / (double)catch_factor2;
// now quantize this value around resample_mean, so that the noise which is in the integral component doesnt hurt.
current_resample_factor = floor( (current_resample_factor - resample_mean) * controlquant + 0.5 ) / controlquant + resample_mean;
// Output "instrumentatio" gonna change that to real instrumentation in a few.
output_resampling_factor = (float) current_resample_factor;
output_diff = (float) smooth_offset;
output_integral = (float) offset_integral;
output_offset = (float) offset;
// Clamp a bit.
if( current_resample_factor < resample_lower_limit ) current_resample_factor = resample_lower_limit;
if( current_resample_factor > resample_upper_limit ) current_resample_factor = resample_upper_limit;
// Now Calculate how many samples we need.
rlen = ceil( ((double)nframes) * current_resample_factor )+2;
assert( rlen > 2 );
// Calculate resample_mean so we can init ourselves to saner values.
resample_mean = 0.9999 * resample_mean + 0.0001 * current_resample_factor;
/*
* now this should do it...
*/
outbuf = alloca( rlen * formats[format].sample_size * num_channels );
resampbuf = alloca( rlen * sizeof( float ) );
/*
* render jack ports to the outbuf...
*/
int chn = 0;
JSList *node = playback_ports;
JSList *src_node = playback_srcs;
SRC_DATA src;
while ( node != NULL)
{
jack_port_t *port = (jack_port_t *) node->data;
float *buf = jack_port_get_buffer (port, nframes);
SRC_STATE *src_state = src_node->data;
src.data_in = buf;
src.input_frames = nframes;
src.data_out = resampbuf;
src.output_frames = rlen;
src.end_of_input = 0;
src.src_ratio = current_resample_factor;
src_process( src_state, &src );
formats[format].jack_to_soundcard( outbuf + format[formats].sample_size * chn, resampbuf, src.output_frames_gen, num_channels*format[formats].sample_size, NULL);
src_node = jack_slist_next (src_node);
node = jack_slist_next (node);
chn++;
}
// now write the output...
again:
err = snd_pcm_writei(alsa_handle, outbuf, src.output_frames_gen);
//err = snd_pcm_writei(alsa_handle, outbuf, src.output_frames_gen);
if( err < 0 ) {
printf( "err = %d\n", err );
if (xrun_recovery(alsa_handle, err) < 0) {
printf("Write error: %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
goto again;
}
return 0;
}
void S9xAlsaSoundDriver::samples_available()
{
snd_pcm_sframes_t frames_written, frames;
int bytes;
frames = snd_pcm_avail(pcm);
if (frames < 0)
{
frames = snd_pcm_recover(pcm, frames, 1);
return;
}
if (Settings.DynamicRateControl)
{
S9xUpdateDynamicRate(snd_pcm_frames_to_bytes(pcm, frames),
output_buffer_size);
}
int snes_frames_available = S9xGetSampleCount() >> 1;
if (Settings.DynamicRateControl && !Settings.SoundSync)
{
// Using rate control, we should always keep the emulator's sound buffers empty to
// maintain an accurate measurement.
if (frames < snes_frames_available)
{
S9xClearSamples();
return;
}
}
if (Settings.SoundSync && !Settings.TurboMode && !Settings.Mute)
{
snd_pcm_nonblock(pcm, 0);
frames = snes_frames_available;
}
else
{
snd_pcm_nonblock(pcm, 1);
frames = MIN(frames, snes_frames_available);
}
bytes = snd_pcm_frames_to_bytes(pcm, frames);
if (bytes <= 0)
return;
if (sound_buffer_size < bytes || sound_buffer == NULL)
{
sound_buffer = (uint8 *)realloc(sound_buffer, bytes);
sound_buffer_size = bytes;
}
S9xMixSamples(sound_buffer, frames * 2);
frames_written = 0;
while (frames_written < frames)
{
int result;
result = snd_pcm_writei(pcm,
sound_buffer +
snd_pcm_frames_to_bytes(pcm, frames_written),
frames - frames_written);
if (result < 0)
{
result = snd_pcm_recover(pcm, result, 1);
if (result < 0)
{
break;
}
}
else
{
frames_written += result;
}
}
}
static
void
data_available_for_stream(pa_mainloop_api *a, pa_io_event *ioe, int fd, pa_io_event_flags_t events,
void *userdata)
{
pa_stream *s = userdata;
snd_pcm_sframes_t frame_count;
size_t frame_size = pa_frame_size(&s->ss);
char buf[16 * 1024];
int paused = g_atomic_int_get(&s->paused);
if (events & (PA_IO_EVENT_INPUT | PA_IO_EVENT_OUTPUT)) {
#if HAVE_SND_PCM_AVAIL
frame_count = snd_pcm_avail(s->ph);
#else
snd_pcm_hwsync(s->ph);
frame_count = snd_pcm_avail_update(s->ph);
#endif
if (frame_count < 0) {
if (frame_count == -EBADFD) {
// stream was closed
return;
}
int cnt = 0, ret;
do {
cnt ++;
ret = snd_pcm_recover(s->ph, frame_count, 1);
} while (ret == -1 && errno == EINTR && cnt < 5);
#if HAVE_SND_PCM_AVAIL
frame_count = snd_pcm_avail(s->ph);
#else
snd_pcm_hwsync(s->ph);
frame_count = snd_pcm_avail_update(s->ph);
#endif
if (frame_count < 0) {
trace_error("%s, can't recover after failed snd_pcm_avail (%d)\n", __func__,
(int)frame_count);
return;
}
}
} else {
return;
}
if (events & PA_IO_EVENT_OUTPUT) {
if (paused) {
// client stream is corked. Pass silence to ALSA
size_t bytecnt = MIN(sizeof(buf), frame_count * frame_size);
memset(buf, 0, bytecnt);
snd_pcm_writei(s->ph, buf, bytecnt / frame_size);
} else {
size_t writable_size = pa_stream_writable_size(s);
if (s->write_cb && writable_size > 0)
s->write_cb(s, writable_size, s->write_cb_userdata);
size_t bytecnt = MIN(sizeof(buf), frame_count * frame_size);
bytecnt = ringbuffer_read(s->rb, buf, bytecnt);
if (bytecnt == 0) {
// application is not ready yet, play silence
bytecnt = MIN(sizeof(buf), frame_count * frame_size);
memset(buf, 0, bytecnt);
}
snd_pcm_writei(s->ph, buf, bytecnt / frame_size);
}
}
if (events & PA_IO_EVENT_INPUT) {
if (paused) {
// client stream is corked. Read data from ALSA and discard them
size_t bytecnt = MIN(sizeof(buf), frame_count * frame_size);
snd_pcm_readi(s->ph, buf, bytecnt / frame_size);
} else {
size_t bytecnt = ringbuffer_writable_size(s->rb);
if (bytecnt == 0) {
// ringbuffer is full because app doesn't read data fast enough.
// Make some room
ringbuffer_drop(s->rb, frame_count * frame_size);
bytecnt = ringbuffer_writable_size(s->rb);
}
bytecnt = MIN(bytecnt, frame_count * frame_size);
bytecnt = MIN(bytecnt, sizeof(buf));
if (bytecnt > 0) {
snd_pcm_readi(s->ph, buf, bytecnt / frame_size);
ringbuffer_write(s->rb, buf, bytecnt);
}
size_t readable_size = pa_stream_readable_size(s);
if (s->read_cb && readable_size > 0)
s->read_cb(s, readable_size, s->read_cb_userdata);
}
}
}
bool
ALSAPCMPlayer::Start(PCMDataSource &_source)
{
const unsigned new_sample_rate = _source.GetSampleRate();
if ((nullptr != source) &&
alsa_handle &&
(source->GetSampleRate() == new_sample_rate)) {
/* just change the source / resume playback */
bool success = false;
DispatchWait(io_service, [this, &_source, &success]() {
bool recovered_from_underrun = false;
switch (snd_pcm_state(alsa_handle.get())) {
case SND_PCM_STATE_XRUN:
if (0 != snd_pcm_prepare(alsa_handle.get()))
return;
else {
recovered_from_underrun = true;
success = true;
}
break;
case SND_PCM_STATE_RUNNING:
success = true;
break;
default:
return;
}
if (success) {
source = &_source;
if (recovered_from_underrun) {
const size_t n = buffer_size / channels;
const size_t n_read = FillPCMBuffer(buffer.get(), n);
if (!WriteFrames(n_read)) {
success = false;
return;
}
}
if (success)
StartEventHandling();
}
});
if (success)
return true;
}
Stop();
assert(!alsa_handle);
AlsaHandleUniquePtr new_alsa_handle = MakeAlsaHandleUniquePtr();
{
const char *alsa_device = ALSAEnv::GetALSADeviceName();
snd_pcm_t *raw_alsa_handle;
int alsa_error = snd_pcm_open(&raw_alsa_handle, alsa_device,
SND_PCM_STREAM_PLAYBACK, 0);
if (alsa_error < 0) {
LogFormat("snd_pcm_open(0x%p, \"%s\", SND_PCM_STREAM_PLAYBACK, 0) "
"failed: %s",
&alsa_handle, alsa_device, snd_strerror(alsa_error));
return false;
}
new_alsa_handle = MakeAlsaHandleUniquePtr(raw_alsa_handle);
assert(new_alsa_handle);
}
unsigned latency = ALSAEnv::GetALSALatency();
channels = 1;
bool big_endian_source = _source.IsBigEndian();
if (!SetParameters(*new_alsa_handle, new_sample_rate, big_endian_source,
latency, channels))
return false;
snd_pcm_sframes_t n_available = snd_pcm_avail(new_alsa_handle.get());
if (n_available <= 0) {
LogFormat("snd_pcm_avail(0x%p) failed: %ld - %s",
new_alsa_handle.get(),
static_cast<long>(n_available),
snd_strerror(static_cast<int>(n_available)));
return false;
}
buffer_size = static_cast<snd_pcm_uframes_t>(n_available * channels);
buffer = std::unique_ptr<int16_t[]>(new int16_t[buffer_size]);
/* Why does Boost.Asio make it so hard to register a set of of standard
poll() descriptors (struct pollfd)? */
int poll_fds_count = snd_pcm_poll_descriptors_count(new_alsa_handle.get());
if (poll_fds_count < 1) {
LogFormat("snd_pcm_poll_descriptors_count(0x%p) returned %d",
new_alsa_handle.get(),
poll_fds_count);
return false;
}
std::unique_ptr<struct pollfd[]> poll_fds(
new struct pollfd[poll_fds_count]);
BOOST_VERIFY(
poll_fds_count ==
snd_pcm_poll_descriptors(new_alsa_handle.get(),
poll_fds.get(),
static_cast<unsigned>(poll_fds_count)));
for (int i = 0; i < poll_fds_count; ++i) {
if ((poll_fds[i].events & POLLIN) || (poll_fds[i].events & POLLPRI)) {
read_poll_descs.emplace_back(
boost::asio::posix::stream_descriptor(io_service, poll_fds[i].fd));
}
if (poll_fds[i].events & POLLOUT) {
write_poll_descs.emplace_back(
boost::asio::posix::stream_descriptor(io_service, poll_fds[i].fd));
}
}
source = &_source;
size_t n_read = FillPCMBuffer(buffer.get(), static_cast<size_t>(n_available));
if (0 == n_read) {
LogFormat("ALSA PCMPlayer started with data source which "
"does not deliver any data");
return false;
}
if (!WriteFrames(*new_alsa_handle, buffer.get(),
static_cast<size_t>(n_available), false))
return false;
alsa_handle = std::move(new_alsa_handle);
StartEventHandling();
return true;
}
