void
AudioStream::stream_state_callback(pa_stream* s, void* /*user_data*/)
{
char str[PA_SAMPLE_SPEC_SNPRINT_MAX];
switch (pa_stream_get_state(s)) {
case PA_STREAM_CREATING:
DEBUG("Stream is creating...");
break;
case PA_STREAM_TERMINATED:
DEBUG("Stream is terminating...");
break;
case PA_STREAM_READY:
DEBUG("Stream successfully created, connected to %s", pa_stream_get_device_name(s));
DEBUG("maxlength %u", pa_stream_get_buffer_attr(s)->maxlength);
DEBUG("tlength %u", pa_stream_get_buffer_attr(s)->tlength);
DEBUG("prebuf %u", pa_stream_get_buffer_attr(s)->prebuf);
DEBUG("minreq %u", pa_stream_get_buffer_attr(s)->minreq);
DEBUG("fragsize %u", pa_stream_get_buffer_attr(s)->fragsize);
DEBUG("samplespec %s", pa_sample_spec_snprint(str, sizeof(str), pa_stream_get_sample_spec(s)));
break;
case PA_STREAM_UNCONNECTED:
DEBUG("Stream unconnected");
break;
case PA_STREAM_FAILED:
default:
ERROR("Sink/Source doesn't exists: %s" , pa_strerror(pa_context_errno(pa_stream_get_context(s))));
break;
}
}
/**
* Pulseaudio stream state callback
*/
static void
stream_state_callback (pa_stream * s, void *userdata)
{
GNUNET_assert (NULL != s);
switch (pa_stream_get_state (s))
{
case PA_STREAM_CREATING:
case PA_STREAM_TERMINATED:
break;
case PA_STREAM_READY:
{
const pa_buffer_attr *a;
char cmt[PA_CHANNEL_MAP_SNPRINT_MAX];
char sst[PA_SAMPLE_SPEC_SNPRINT_MAX];
GNUNET_log (GNUNET_ERROR_TYPE_INFO,
_("Stream successfully created.\n"));
if (!(a = pa_stream_get_buffer_attr (s)))
{
GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
_("pa_stream_get_buffer_attr() failed: %s\n"),
pa_strerror (pa_context_errno
(pa_stream_get_context (s))));
}
else
{
GNUNET_log (GNUNET_ERROR_TYPE_INFO,
_("Buffer metrics: maxlength=%u, fragsize=%u\n"),
a->maxlength, a->fragsize);
}
GNUNET_log (GNUNET_ERROR_TYPE_INFO,
_("Using sample spec '%s', channel map '%s'.\n"),
pa_sample_spec_snprint (sst, sizeof (sst),
pa_stream_get_sample_spec (s)),
pa_channel_map_snprint (cmt, sizeof (cmt),
pa_stream_get_channel_map (s)));
GNUNET_log (GNUNET_ERROR_TYPE_INFO,
_("Connected to device %s (%u, %ssuspended).\n"),
pa_stream_get_device_name (s),
pa_stream_get_device_index (s),
pa_stream_is_suspended (s) ? "" : "not ");
}
break;
case PA_STREAM_FAILED:
default:
GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
_("Stream error: %s\n"),
pa_strerror (pa_context_errno (pa_stream_get_context (s))));
quit (1);
}
}
// Return the current latency in seconds
static float get_delay(struct ao *ao)
{
/* This code basically does what pa_stream_get_latency() _should_
* do, but doesn't due to multiple known bugs in PulseAudio (at
* PulseAudio version 2.1). In particular, the timing interpolation
* mode (PA_STREAM_INTERPOLATE_TIMING) can return completely bogus
* values, and the non-interpolating code has a bug causing too
* large results at end of stream (so a stream never seems to finish).
* This code can still return wrong values in some cases due to known
* PulseAudio bugs that can not be worked around on the client side.
*
* We always query the server for latest timing info. This may take
* too long to work well with remote audio servers, but at least
* this should be enough to fix the normal local playback case.
*/
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
if (!waitop(priv, pa_stream_update_timing_info(priv->stream, NULL, NULL))) {
GENERIC_ERR_MSG(priv->context, "pa_stream_update_timing_info() failed");
return 0;
}
pa_threaded_mainloop_lock(priv->mainloop);
const pa_timing_info *ti = pa_stream_get_timing_info(priv->stream);
if (!ti) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG(priv->context, "pa_stream_get_timing_info() failed");
return 0;
}
const struct pa_sample_spec *ss = pa_stream_get_sample_spec(priv->stream);
if (!ss) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG(priv->context, "pa_stream_get_sample_spec() failed");
return 0;
}
// data left in PulseAudio's main buffers (not written to sink yet)
int64_t latency = pa_bytes_to_usec(ti->write_index - ti->read_index, ss);
// since this info may be from a while ago, playback has progressed since
latency -= ti->transport_usec;
// data already moved from buffers to sink, but not played yet
int64_t sink_latency = ti->sink_usec;
if (!ti->playing)
/* At the end of a stream, part of the data "left" in the sink may
* be padding silence after the end; that should be subtracted to
* get the amount of real audio from our stream. This adjustment
* is missing from Pulseaudio's own get_latency calculations
* (as of PulseAudio 2.1). */
sink_latency -= pa_bytes_to_usec(ti->since_underrun, ss);
if (sink_latency > 0)
latency += sink_latency;
if (latency < 0)
latency = 0;
pa_threaded_mainloop_unlock(priv->mainloop);
return latency / 1e6;
}
/* This routine is called whenever the stream state changes */
static void stream_state_callback(pa_stream *s, void *userdata) {
pa_assert(s);
switch (pa_stream_get_state(s)) {
case PA_STREAM_CREATING:
case PA_STREAM_TERMINATED:
break;
case PA_STREAM_READY:
if (verbose) {
const pa_buffer_attr *a;
char cmt[PA_CHANNEL_MAP_SNPRINT_MAX], sst[PA_SAMPLE_SPEC_SNPRINT_MAX];
pa_log(_("Stream successfully created."));
if (!(a = pa_stream_get_buffer_attr(s)))
pa_log(_("pa_stream_get_buffer_attr() failed: %s"), pa_strerror(pa_context_errno(pa_stream_get_context(s))));
else {
if (mode == PLAYBACK)
pa_log(_("Buffer metrics: maxlength=%u, tlength=%u, prebuf=%u, minreq=%u"), a->maxlength, a->tlength, a->prebuf, a->minreq);
else {
pa_assert(mode == RECORD);
pa_log(_("Buffer metrics: maxlength=%u, fragsize=%u"), a->maxlength, a->fragsize);
}
}
pa_log(_("Using sample spec '%s', channel map '%s'."),
pa_sample_spec_snprint(sst, sizeof(sst), pa_stream_get_sample_spec(s)),
pa_channel_map_snprint(cmt, sizeof(cmt), pa_stream_get_channel_map(s)));
pa_log(_("Connected to device %s (%u, %ssuspended)."),
pa_stream_get_device_name(s),
pa_stream_get_device_index(s),
pa_stream_is_suspended(s) ? "" : "not ");
}
break;
case PA_STREAM_FAILED:
default:
pa_log(_("Stream error: %s"), pa_strerror(pa_context_errno(pa_stream_get_context(s))));
quit(1);
}
}
void PulseAudioSystem::write_callback(pa_stream *s, size_t bytes, void *userdata) {
PulseAudioSystem *pas = reinterpret_cast<PulseAudioSystem *>(userdata);
Q_ASSERT(s == pas->pasOutput);
AudioOutputPtr ao = g.ao;
PulseAudioOutput *pao = dynamic_cast<PulseAudioOutput *>(ao.get());
unsigned char buffer[bytes];
if (! pao) {
// Transitioning, but most likely transitions back, so just zero.
memset(buffer, 0, bytes);
pa_stream_write(s, buffer, bytes, NULL, 0, PA_SEEK_RELATIVE);
pas->wakeup();
return;
}
const pa_sample_spec *pss = pa_stream_get_sample_spec(s);
const pa_channel_map *pcm = pa_stream_get_channel_map(pas->pasOutput);
if (!pa_sample_spec_equal(pss, &pao->pss) || !pa_channel_map_equal(pcm, &pao->pcm)) {
pao->pss = *pss;
pao->pcm = *pcm;
if (pss->format == PA_SAMPLE_FLOAT32NE)
pao->eSampleFormat = PulseAudioOutput::SampleFloat;
else
pao->eSampleFormat = PulseAudioOutput::SampleShort;
pao->iMixerFreq = pss->rate;
pao->iChannels = pss->channels;
unsigned int chanmasks[pss->channels];
for (int i=0;i<pss->channels;++i) {
unsigned int cm = 0;
switch (pcm->map[i]) {
case PA_CHANNEL_POSITION_LEFT:
cm = SPEAKER_FRONT_LEFT;
break;
case PA_CHANNEL_POSITION_RIGHT:
cm = SPEAKER_FRONT_RIGHT;
break;
case PA_CHANNEL_POSITION_CENTER:
cm = SPEAKER_FRONT_CENTER;
break;
case PA_CHANNEL_POSITION_REAR_LEFT:
cm = SPEAKER_BACK_LEFT;
break;
case PA_CHANNEL_POSITION_REAR_RIGHT:
cm = SPEAKER_BACK_RIGHT;
break;
case PA_CHANNEL_POSITION_REAR_CENTER:
cm = SPEAKER_BACK_CENTER;
break;
case PA_CHANNEL_POSITION_LFE:
cm = SPEAKER_LOW_FREQUENCY;
break;
case PA_CHANNEL_POSITION_SIDE_LEFT:
cm = SPEAKER_SIDE_LEFT;
break;
case PA_CHANNEL_POSITION_SIDE_RIGHT:
cm = SPEAKER_SIDE_RIGHT;
break;
case PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER:
cm = SPEAKER_FRONT_LEFT_OF_CENTER;
break;
case PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER:
cm = SPEAKER_FRONT_RIGHT_OF_CENTER;
break;
default:
cm = 0;
break;
}
chanmasks[i] = cm;
}
pao->initializeMixer(chanmasks);
}
const unsigned int iSampleSize = pao->iSampleSize;
const unsigned int samples = static_cast<unsigned int>(bytes) / iSampleSize;
bool oldAttenuation = pas->bAttenuating;
// do we have some mixed output?
if (pao->mix(buffer, samples)) {
// attenuate if instructed to or it's in settings
pas->bAttenuating = (g.bAttenuateOthers || g.s.bAttenuateOthers);
} else {
memset(buffer, 0, bytes);
// attenuate if intructed to (self-activated)
pas->bAttenuating = g.bAttenuateOthers;
}
// if the attenuation state has changed
if (oldAttenuation != pas->bAttenuating) {
pas->setVolumes();
}
pa_stream_write(s, buffer, iSampleSize * samples, NULL, 0, PA_SEEK_RELATIVE);
}
void PulseAudioSystem::read_callback(pa_stream *s, size_t bytes, void *userdata) {
PulseAudioSystem *pas = reinterpret_cast<PulseAudioSystem *>(userdata);
size_t length = bytes;
const void *data = NULL;
pa_stream_peek(s, &data, &length);
if (data == NULL && length > 0) {
qWarning("PulseAudio: pa_stream_peek reports no data at current read index.");
} else if (data == NULL && length == 0) {
qWarning("PulseAudio: pa_stream_peek reports empty memblockq.");
} else if (data == NULL || length == 0) {
qWarning("PulseAudio: invalid pa_stream_peek state encountered.");
return;
}
AudioInputPtr ai = g.ai;
PulseAudioInput *pai = dynamic_cast<PulseAudioInput *>(ai.get());
if (! pai) {
if (length > 0) {
pa_stream_drop(s);
}
pas->wakeup();
return;
}
const pa_sample_spec *pss = pa_stream_get_sample_spec(s);
if (s == pas->pasInput) {
if (!pa_sample_spec_equal(pss, &pai->pssMic)) {
pai->pssMic = *pss;
pai->iMicFreq = pss->rate;
pai->iMicChannels = pss->channels;
if (pss->format == PA_SAMPLE_FLOAT32NE)
pai->eMicFormat = PulseAudioInput::SampleFloat;
else
pai->eMicFormat = PulseAudioInput::SampleShort;
pai->initializeMixer();
}
if (data != NULL) {
pai->addMic(data, static_cast<unsigned int>(length) / pai->iMicSampleSize);
}
} else if (s == pas->pasSpeaker) {
if (!pa_sample_spec_equal(pss, &pai->pssEcho)) {
pai->pssEcho = *pss;
pai->iEchoFreq = pss->rate;
pai->iEchoChannels = pss->channels;
if (pss->format == PA_SAMPLE_FLOAT32NE)
pai->eEchoFormat = PulseAudioInput::SampleFloat;
else
pai->eEchoFormat = PulseAudioInput::SampleShort;
pai->initializeMixer();
}
if (data != NULL) {
pai->addEcho(data, static_cast<unsigned int>(length) / pai->iEchoSampleSize);
}
}
if (length > 0) {
pa_stream_drop(s);
}
}
void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
api->defer_enable(pade, false);
if (! bSourceDone || ! bSinkDone || ! bServerDone)
return;
AudioInputPtr ai = g.ai;
AudioOutputPtr ao = g.ao;
AudioInput *raw_ai = ai.get();
AudioOutput *raw_ao = ao.get();
PulseAudioInput *pai = dynamic_cast<PulseAudioInput *>(raw_ai);
PulseAudioOutput *pao = dynamic_cast<PulseAudioOutput *>(raw_ao);
if (raw_ao) {
QString odev = outputDevice();
pa_stream_state ost = pasOutput ? pa_stream_get_state(pasOutput) : PA_STREAM_TERMINATED;
bool do_stop = false;
bool do_start = false;
if (! pao && (ost == PA_STREAM_READY)) {
do_stop = true;
} else if (pao) {
switch (ost) {
case PA_STREAM_TERMINATED: {
if (pasOutput)
pa_stream_unref(pasOutput);
pa_sample_spec pss = qhSpecMap.value(odev);
pa_channel_map pcm = qhChanMap.value(odev);
if ((pss.format != PA_SAMPLE_FLOAT32NE) && (pss.format != PA_SAMPLE_S16NE))
pss.format = PA_SAMPLE_FLOAT32NE;
if (pss.rate == 0)
pss.rate = SAMPLE_RATE;
if ((pss.channels == 0) || (! g.s.doPositionalAudio()))
pss.channels = 1;
pasOutput = pa_stream_new(pacContext, mumble_sink_input, &pss, (pss.channels == 1) ? NULL : &pcm);
pa_stream_set_state_callback(pasOutput, stream_callback, this);
pa_stream_set_write_callback(pasOutput, write_callback, this);
}
case PA_STREAM_UNCONNECTED:
do_start = true;
break;
case PA_STREAM_READY: {
if (g.s.iOutputDelay != iDelayCache) {
do_stop = true;
} else if (g.s.doPositionalAudio() != bPositionalCache) {
do_stop = true;
} else if (odev != qsOutputCache) {
do_stop = true;
}
break;
}
default:
break;
}
}
if (do_stop) {
qWarning("PulseAudio: Stopping output");
pa_stream_disconnect(pasOutput);
iSinkId = -1;
} else if (do_start) {
qWarning("PulseAudio: Starting output: %s", qPrintable(odev));
pa_buffer_attr buff;
const pa_sample_spec *pss = pa_stream_get_sample_spec(pasOutput);
const size_t sampleSize = (pss->format == PA_SAMPLE_FLOAT32NE) ? sizeof(float) : sizeof(short);
const unsigned int iBlockLen = ((pao->iFrameSize * pss->rate) / SAMPLE_RATE) * pss->channels * static_cast<unsigned int>(sampleSize);
buff.tlength = iBlockLen * (g.s.iOutputDelay+1);
buff.minreq = iBlockLen;
buff.maxlength = -1;
buff.prebuf = -1;
buff.fragsize = iBlockLen;
iDelayCache = g.s.iOutputDelay;
bPositionalCache = g.s.doPositionalAudio();
qsOutputCache = odev;
pa_stream_connect_playback(pasOutput, qPrintable(odev), &buff, PA_STREAM_ADJUST_LATENCY, NULL, NULL);
pa_context_get_sink_info_by_name(pacContext, qPrintable(odev), sink_info_callback, this);
}
}
if (raw_ai) {
QString idev = inputDevice();
pa_stream_state ist = pasInput ? pa_stream_get_state(pasInput) : PA_STREAM_TERMINATED;
bool do_stop = false;
bool do_start = false;
if (! pai && (ist == PA_STREAM_READY)) {
do_stop = true;
} else if (pai) {
switch (ist) {
case PA_STREAM_TERMINATED: {
if (pasInput)
pa_stream_unref(pasInput);
pa_sample_spec pss = qhSpecMap.value(idev);
if ((pss.format != PA_SAMPLE_FLOAT32NE) && (pss.format != PA_SAMPLE_S16NE))
pss.format = PA_SAMPLE_FLOAT32NE;
if (pss.rate == 0)
pss.rate = SAMPLE_RATE;
pss.channels = 1;
pasInput = pa_stream_new(pacContext, "Microphone", &pss, NULL);
pa_stream_set_state_callback(pasInput, stream_callback, this);
pa_stream_set_read_callback(pasInput, read_callback, this);
}
case PA_STREAM_UNCONNECTED:
do_start = true;
break;
case PA_STREAM_READY: {
if (idev != qsInputCache) {
do_stop = true;
}
break;
}
default:
break;
}
}
if (do_stop) {
qWarning("PulseAudio: Stopping input");
pa_stream_disconnect(pasInput);
} else if (do_start) {
qWarning("PulseAudio: Starting input %s",qPrintable(idev));
pa_buffer_attr buff;
const pa_sample_spec *pss = pa_stream_get_sample_spec(pasInput);
const size_t sampleSize = (pss->format == PA_SAMPLE_FLOAT32NE) ? sizeof(float) : sizeof(short);
const unsigned int iBlockLen = ((pai->iFrameSize * pss->rate) / SAMPLE_RATE) * pss->channels * static_cast<unsigned int>(sampleSize);
buff.tlength = iBlockLen;
buff.minreq = iBlockLen;
buff.maxlength = -1;
buff.prebuf = -1;
buff.fragsize = iBlockLen;
qsInputCache = idev;
pa_stream_connect_record(pasInput, qPrintable(idev), &buff, PA_STREAM_ADJUST_LATENCY);
}
}
if (raw_ai) {
QString odev = outputDevice();
QString edev = qhEchoMap.value(odev);
pa_stream_state est = pasSpeaker ? pa_stream_get_state(pasSpeaker) : PA_STREAM_TERMINATED;
bool do_stop = false;
bool do_start = false;
if ((! pai || ! g.s.doEcho()) && (est == PA_STREAM_READY)) {
do_stop = true;
} else if (pai && g.s.doEcho()) {
switch (est) {
case PA_STREAM_TERMINATED: {
if (pasSpeaker)
pa_stream_unref(pasSpeaker);
pa_sample_spec pss = qhSpecMap.value(edev);
pa_channel_map pcm = qhChanMap.value(edev);
if ((pss.format != PA_SAMPLE_FLOAT32NE) && (pss.format != PA_SAMPLE_S16NE))
pss.format = PA_SAMPLE_FLOAT32NE;
if (pss.rate == 0)
pss.rate = SAMPLE_RATE;
if ((pss.channels == 0) || (! g.s.bEchoMulti))
pss.channels = 1;
pasSpeaker = pa_stream_new(pacContext, mumble_echo, &pss, (pss.channels == 1) ? NULL : &pcm);
pa_stream_set_state_callback(pasSpeaker, stream_callback, this);
pa_stream_set_read_callback(pasSpeaker, read_callback, this);
}
case PA_STREAM_UNCONNECTED:
do_start = true;
break;
case PA_STREAM_READY: {
if (g.s.bEchoMulti != bEchoMultiCache) {
do_stop = true;
} else if (edev != qsEchoCache) {
do_stop = true;
}
break;
}
default:
break;
}
}
if (do_stop) {
qWarning("PulseAudio: Stopping echo");
pa_stream_disconnect(pasSpeaker);
} else if (do_start) {
qWarning("PulseAudio: Starting echo: %s", qPrintable(edev));
pa_buffer_attr buff;
const pa_sample_spec *pss = pa_stream_get_sample_spec(pasSpeaker);
const size_t sampleSize = (pss->format == PA_SAMPLE_FLOAT32NE) ? sizeof(float) : sizeof(short);
const unsigned int iBlockLen = ((pai->iFrameSize * pss->rate) / SAMPLE_RATE) * pss->channels * static_cast<unsigned int>(sampleSize);
buff.tlength = iBlockLen;
buff.minreq = iBlockLen;
buff.maxlength = -1;
buff.prebuf = -1;
buff.fragsize = iBlockLen;
bEchoMultiCache = g.s.bEchoMulti;
qsEchoCache = edev;
pa_stream_connect_record(pasSpeaker, qPrintable(edev), &buff, PA_STREAM_ADJUST_LATENCY);
}
}
}
FXbool PulseOutput::configure(const AudioFormat & fmt){
const pa_sample_spec * config=nullptr;
pa_operation *operation=nullptr;
if (!open())
return false;
if (stream && fmt==af)
return true;
if (stream) {
pa_stream_disconnect(stream);
pa_stream_unref(stream);
stream=nullptr;
}
pa_sample_spec spec;
pa_channel_map cmap;
if (!to_pulse_format(fmt,spec.format))
goto failed;
spec.rate = fmt.rate;
spec.channels = fmt.channels;
// setup channel map
pa_channel_map_init(&cmap);
cmap.channels = fmt.channels;
for (FXint i=0;i<fmt.channels;i++) {
switch(fmt.channeltype(i)) {
case Channel::None : cmap.map[i] = PA_CHANNEL_POSITION_INVALID; break;
case Channel::Mono : cmap.map[i] = PA_CHANNEL_POSITION_MONO; break;
case Channel::FrontLeft : cmap.map[i] = PA_CHANNEL_POSITION_FRONT_LEFT; break;
case Channel::FrontRight : cmap.map[i] = PA_CHANNEL_POSITION_FRONT_RIGHT; break;
case Channel::FrontCenter : cmap.map[i] = PA_CHANNEL_POSITION_FRONT_CENTER; break;
case Channel::BackLeft : cmap.map[i] = PA_CHANNEL_POSITION_REAR_LEFT; break;
case Channel::BackRight : cmap.map[i] = PA_CHANNEL_POSITION_REAR_RIGHT; break;
case Channel::BackCenter : cmap.map[i] = PA_CHANNEL_POSITION_REAR_CENTER; break;
case Channel::SideLeft : cmap.map[i] = PA_CHANNEL_POSITION_SIDE_LEFT; break;
case Channel::SideRight : cmap.map[i] = PA_CHANNEL_POSITION_SIDE_RIGHT; break;
case Channel::LFE : cmap.map[i] = PA_CHANNEL_POSITION_LFE; break;
default: goto failed;
}
}
stream = pa_stream_new(pulse_context,"Goggles Music Manager",&spec,&cmap);
if (stream == nullptr)
goto failed;
#ifdef DEBUG
pa_stream_set_state_callback(stream,stream_state_callback,this);
#endif
//pa_stream_set_write_callback(stream,stream_write_callback,this);
if (pa_stream_connect_playback(stream,nullptr,nullptr,PA_STREAM_NOFLAGS,nullptr,nullptr)<0)
goto failed;
/// Wait until stream is ready
pa_stream_state_t state;
while((state=pa_stream_get_state(stream))!=PA_STREAM_READY) {
if (state==PA_STREAM_FAILED || state==PA_STREAM_TERMINATED){
goto failed;
}
context->wait_plugin_events();
}
/// Get Actual Format
config = pa_stream_get_sample_spec(stream);
if (!to_gap_format(config->format,af))
goto failed;
af.channels=config->channels;
af.rate=config->rate;
af.channelmap=fmt.channelmap;
/// Get Current Volume
operation = pa_context_get_sink_input_info(pulse_context,pa_stream_get_index(stream),sink_info_callback,this);
if (operation) pa_operation_unref(operation);
return true;
failed:
GM_DEBUG_PRINT("[pulse] Unsupported pulse configuration:\n");
fmt.debug();
return false;
}
static ALCboolean pulse_reset_playback(ALCdevice *device) //{{{
{
pulse_data *data = device->ExtraData;
pa_stream_flags_t flags = 0;
pa_channel_map chanmap;
pa_threaded_mainloop_lock(data->loop);
if(!(device->Flags&DEVICE_CHANNELS_REQUEST))
{
pa_operation *o;
o = pa_context_get_sink_info_by_name(data->context, data->device_name, sink_info_callback, device);
while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
pa_threaded_mainloop_wait(data->loop);
pa_operation_unref(o);
}
if(!(device->Flags&DEVICE_FREQUENCY_REQUEST))
flags |= PA_STREAM_FIX_RATE;
data->frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->attr.prebuf = -1;
data->attr.fragsize = -1;
data->attr.minreq = device->UpdateSize * data->frame_size;
data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
data->attr.maxlength = -1;
flags |= PA_STREAM_EARLY_REQUESTS;
flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE;
switch(device->FmtType)
{
case DevFmtByte:
device->FmtType = DevFmtUByte;
/* fall-through */
case DevFmtUByte:
data->spec.format = PA_SAMPLE_U8;
break;
case DevFmtUShort:
device->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
data->spec.format = PA_SAMPLE_S16NE;
break;
case DevFmtFloat:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
}
data->spec.rate = device->Frequency;
data->spec.channels = ChannelsFromDevFmt(device->FmtChans);
if(pa_sample_spec_valid(&data->spec) == 0)
{
ERR("Invalid sample format\n");
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
{
ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
SetDefaultWFXChannelOrder(device);
data->stream = connect_playback_stream(device, flags, &data->attr, &data->spec, &chanmap);
if(!data->stream)
{
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
pa_stream_set_state_callback(data->stream, stream_state_callback2, device);
data->spec = *(pa_stream_get_sample_spec(data->stream));
if(device->Frequency != data->spec.rate)
{
pa_operation *o;
if((device->Flags&DEVICE_FREQUENCY_REQUEST))
ERR("Failed to set frequency %dhz, got %dhz instead\n", device->Frequency, data->spec.rate);
device->Flags &= ~DEVICE_FREQUENCY_REQUEST;
/* Server updated our playback rate, so modify the buffer attribs
* accordingly. */
data->attr.minreq = (ALuint64)(data->attr.minreq/data->frame_size) *
data->spec.rate / device->Frequency * data->frame_size;
data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
o = pa_stream_set_buffer_attr(data->stream, &data->attr,
stream_success_callback, device);
while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
pa_threaded_mainloop_wait(data->loop);
pa_operation_unref(o);
device->Frequency = data->spec.rate;
}
#if PA_CHECK_VERSION(0,9,15)
if(pa_stream_set_buffer_attr_callback)
pa_stream_set_buffer_attr_callback(data->stream, stream_buffer_attr_callback, device);
#endif
pa_stream_set_moved_callback(data->stream, stream_device_callback, device);
pa_stream_set_write_callback(data->stream, stream_write_callback, device);
pa_stream_set_underflow_callback(data->stream, stream_signal_callback, device);
data->attr = *(pa_stream_get_buffer_attr(data->stream));
ERR("PulseAudio returned minreq=%d, tlength=%d\n", data->attr.minreq, data->attr.tlength);
device->UpdateSize = data->attr.minreq / data->frame_size;
device->NumUpdates = (data->attr.tlength/data->frame_size) / device->UpdateSize;
while(device->NumUpdates <= 2)
{
pa_operation *o;
ERR("minreq too high - expect lag or break up\n");
/* Server gave a comparatively large minreq, so modify the tlength. */
device->NumUpdates = 2;
data->attr.tlength = data->attr.minreq * device->NumUpdates;
o = pa_stream_set_buffer_attr(data->stream, &data->attr,
stream_success_callback, device);
while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
pa_threaded_mainloop_wait(data->loop);
pa_operation_unref(o);
data->attr = *(pa_stream_get_buffer_attr(data->stream));
ERR("PulseAudio returned minreq=%d, tlength=%d", data->attr.minreq, data->attr.tlength);
device->UpdateSize = data->attr.minreq / data->frame_size;
device->NumUpdates = (data->attr.tlength/data->frame_size) / device->UpdateSize;
}
data->thread = StartThread(PulseProc, device);
if(!data->thread)
{
#if PA_CHECK_VERSION(0,9,15)
if(pa_stream_set_buffer_attr_callback)
pa_stream_set_buffer_attr_callback(data->stream, NULL, NULL);
#endif
pa_stream_set_moved_callback(data->stream, NULL, NULL);
pa_stream_set_write_callback(data->stream, NULL, NULL);
pa_stream_set_underflow_callback(data->stream, NULL, NULL);
pa_stream_disconnect(data->stream);
pa_stream_unref(data->stream);
data->stream = NULL;
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
pa_threaded_mainloop_unlock(data->loop);
return ALC_TRUE;
} //}}}
/*****************************************************************************
* Open: open the audio device
*****************************************************************************/
static int Open ( vlc_object_t *p_this )
{
aout_instance_t *p_aout = (aout_instance_t *)p_this;
struct aout_sys_t * p_sys;
struct pa_sample_spec ss;
const struct pa_buffer_attr *buffer_attr;
struct pa_buffer_attr a;
struct pa_channel_map map;
/* Allocate structures */
p_aout->output.p_sys = p_sys = calloc( 1, sizeof( aout_sys_t ) );
if( p_sys == NULL )
return VLC_ENOMEM;
PULSE_DEBUG( "Pulse start initialization");
ss.channels = aout_FormatNbChannels( &p_aout->output.output ); /* Get the input stream channel count */
/* Setup the pulse audio stream based on the input stream count */
switch(ss.channels)
{
case 8:
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
| AOUT_CHAN_MIDDLELEFT | AOUT_CHAN_MIDDLERIGHT
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT
| AOUT_CHAN_LFE;
break;
case 6:
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT
| AOUT_CHAN_LFE;
break;
case 4:
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT;
break;
case 2:
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
break;
case 1:
p_aout->output.output.i_physical_channels = AOUT_CHAN_CENTER;
break;
default:
msg_Err(p_aout,"Invalid number of channels");
goto fail;
}
/* Add a quick command line info message */
msg_Info(p_aout, "No. of Audio Channels: %d", ss.channels);
ss.rate = p_aout->output.output.i_rate;
ss.format = PA_SAMPLE_FLOAT32NE;
p_aout->output.output.i_format = VLC_CODEC_FL32;
if (!pa_sample_spec_valid(&ss)) {
msg_Err(p_aout,"Invalid sample spec");
goto fail;
}
/* Reduce overall latency to 200mS to reduce audible clicks
* Also pulse minreq and internal buffers are now 20mS which reduces resampling
*/
a.tlength = pa_bytes_per_second(&ss)/5;
a.maxlength = a.tlength * 2;
a.prebuf = a.tlength / 2;
a.minreq = a.tlength / 10;
/* Buffer size is 20mS */
p_sys->buffer_size = a.minreq;
/* Initialise the speaker map setup above */
pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_ALSA);
if (!(p_sys->mainloop = pa_threaded_mainloop_new())) {
msg_Err(p_aout, "Failed to allocate main loop");
goto fail;
}
if (!(p_sys->context = pa_context_new(pa_threaded_mainloop_get_api(p_sys->mainloop), _( PULSE_CLIENT_NAME )))) {
msg_Err(p_aout, "Failed to allocate context");
goto fail;
}
pa_context_set_state_callback(p_sys->context, context_state_cb, p_aout);
PULSE_DEBUG( "Pulse before context connect");
if (pa_context_connect(p_sys->context, NULL, 0, NULL) < 0) {
msg_Err(p_aout, "Failed to connect to server: %s", pa_strerror(pa_context_errno(p_sys->context)));
goto fail;
}
PULSE_DEBUG( "Pulse after context connect");
pa_threaded_mainloop_lock(p_sys->mainloop);
if (pa_threaded_mainloop_start(p_sys->mainloop) < 0) {
msg_Err(p_aout, "Failed to start main loop");
goto unlock_and_fail;
}
msg_Dbg(p_aout, "Pulse mainloop started");
/* Wait until the context is ready */
pa_threaded_mainloop_wait(p_sys->mainloop);
if (pa_context_get_state(p_sys->context) != PA_CONTEXT_READY) {
msg_Err(p_aout, "Failed to connect to server: %s", pa_strerror(pa_context_errno(p_sys->context)));
goto unlock_and_fail;
}
if (!(p_sys->stream = pa_stream_new(p_sys->context, "audio stream", &ss, &map))) {
msg_Err(p_aout, "Failed to create stream: %s", pa_strerror(pa_context_errno(p_sys->context)));
goto unlock_and_fail;
}
PULSE_DEBUG( "Pulse after new stream");
pa_stream_set_state_callback(p_sys->stream, stream_state_cb, p_aout);
pa_stream_set_write_callback(p_sys->stream, stream_request_cb, p_aout);
pa_stream_set_latency_update_callback(p_sys->stream, stream_latency_update_cb, p_aout);
if (pa_stream_connect_playback(p_sys->stream, NULL, &a, PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE|PA_STREAM_ADJUST_LATENCY, NULL, NULL) < 0) {
msg_Err(p_aout, "Failed to connect stream: %s", pa_strerror(pa_context_errno(p_sys->context)));
goto unlock_and_fail;
}
PULSE_DEBUG("Pulse stream connect");
/* Wait until the stream is ready */
pa_threaded_mainloop_wait(p_sys->mainloop);
msg_Dbg(p_aout,"Pulse stream connected");
if (pa_stream_get_state(p_sys->stream) != PA_STREAM_READY) {
msg_Err(p_aout, "Failed to connect to server: %s", pa_strerror(pa_context_errno(p_sys->context)));
goto unlock_and_fail;
}
PULSE_DEBUG("Pulse after stream get status");
pa_threaded_mainloop_unlock(p_sys->mainloop);
buffer_attr = pa_stream_get_buffer_attr(p_sys->stream);
p_aout->output.i_nb_samples = buffer_attr->minreq / pa_frame_size(&ss);
p_aout->output.pf_play = Play;
aout_VolumeSoftInit(p_aout);
msg_Dbg(p_aout, "Pulse initialized successfully");
{
char cmt[PA_CHANNEL_MAP_SNPRINT_MAX], sst[PA_SAMPLE_SPEC_SNPRINT_MAX];
msg_Dbg(p_aout, "Buffer metrics: maxlength=%u, tlength=%u, prebuf=%u, minreq=%u", buffer_attr->maxlength, buffer_attr->tlength, buffer_attr->prebuf, buffer_attr->minreq);
msg_Dbg(p_aout, "Using sample spec '%s', channel map '%s'.",
pa_sample_spec_snprint(sst, sizeof(sst), pa_stream_get_sample_spec(p_sys->stream)),
pa_channel_map_snprint(cmt, sizeof(cmt), pa_stream_get_channel_map(p_sys->stream)));
msg_Dbg(p_aout, "Connected to device %s (%u, %ssuspended).",
pa_stream_get_device_name(p_sys->stream),
pa_stream_get_device_index(p_sys->stream),
pa_stream_is_suspended(p_sys->stream) ? "" : "not ");
}
return VLC_SUCCESS;
unlock_and_fail:
msg_Dbg(p_aout, "Pulse initialization unlock and fail");
if (p_sys->mainloop)
pa_threaded_mainloop_unlock(p_sys->mainloop);
fail:
msg_Err(p_aout, "Pulse initialization failed");
uninit(p_aout);
return VLC_EGENERIC;
}
static ALCboolean pulse_reset_playback(ALCdevice *device)
{
pulse_data *data = device->ExtraData;
pa_stream_flags_t flags = 0;
pa_channel_map chanmap;
pa_threaded_mainloop_lock(data->loop);
if(data->stream)
{
#if PA_CHECK_VERSION(0,9,15)
if(pa_stream_set_buffer_attr_callback)
pa_stream_set_buffer_attr_callback(data->stream, NULL, NULL);
#endif
pa_stream_disconnect(data->stream);
pa_stream_unref(data->stream);
data->stream = NULL;
}
if(!(device->Flags&DEVICE_CHANNELS_REQUEST))
{
pa_operation *o;
o = pa_context_get_sink_info_by_name(data->context, data->device_name, sink_info_callback, device);
WAIT_FOR_OPERATION(o, data->loop);
}
if(!(device->Flags&DEVICE_FREQUENCY_REQUEST))
flags |= PA_STREAM_FIX_RATE;
flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE;
flags |= PA_STREAM_ADJUST_LATENCY;
flags |= PA_STREAM_START_CORKED;
flags |= PA_STREAM_DONT_MOVE;
switch(device->FmtType)
{
case DevFmtByte:
device->FmtType = DevFmtUByte;
/* fall-through */
case DevFmtUByte:
data->spec.format = PA_SAMPLE_U8;
break;
case DevFmtUShort:
device->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
data->spec.format = PA_SAMPLE_S16NE;
break;
case DevFmtUInt:
device->FmtType = DevFmtInt;
/* fall-through */
case DevFmtInt:
data->spec.format = PA_SAMPLE_S32NE;
break;
case DevFmtFloat:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
}
data->spec.rate = device->Frequency;
data->spec.channels = ChannelsFromDevFmt(device->FmtChans);
if(pa_sample_spec_valid(&data->spec) == 0)
{
ERR("Invalid sample format\n");
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
{
ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
SetDefaultWFXChannelOrder(device);
data->attr.fragsize = -1;
data->attr.prebuf = 0;
data->attr.minreq = device->UpdateSize * pa_frame_size(&data->spec);
data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
data->attr.maxlength = -1;
data->stream = connect_playback_stream(data->device_name, data->loop,
data->context, flags, &data->attr,
&data->spec, &chanmap);
if(!data->stream)
{
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
pa_stream_set_state_callback(data->stream, stream_state_callback2, device);
data->spec = *(pa_stream_get_sample_spec(data->stream));
if(device->Frequency != data->spec.rate)
{
pa_operation *o;
/* Server updated our playback rate, so modify the buffer attribs
* accordingly. */
data->attr.minreq = (ALuint64)device->UpdateSize * data->spec.rate /
device->Frequency * pa_frame_size(&data->spec);
data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
data->attr.prebuf = 0;
o = pa_stream_set_buffer_attr(data->stream, &data->attr,
stream_success_callback, device);
WAIT_FOR_OPERATION(o, data->loop);
device->Frequency = data->spec.rate;
}
#if PA_CHECK_VERSION(0,9,15)
if(pa_stream_set_buffer_attr_callback)
pa_stream_set_buffer_attr_callback(data->stream, stream_buffer_attr_callback, device);
#endif
stream_buffer_attr_callback(data->stream, device);
device->NumUpdates = device->UpdateSize*device->NumUpdates /
(data->attr.minreq/pa_frame_size(&data->spec));
device->NumUpdates = maxu(device->NumUpdates, 2);
device->UpdateSize = data->attr.minreq / pa_frame_size(&data->spec);
pa_threaded_mainloop_unlock(data->loop);
return ALC_TRUE;
}
static int
pulseaudio_check_passthru(audio_decoder_t *ad, int codec_id)
{
decoder_t *d = (decoder_t *)ad;
int e;
switch(codec_id) {
case AV_CODEC_ID_DTS:
e = PA_ENCODING_DTS_IEC61937;
break;
case AV_CODEC_ID_AC3:
e = PA_ENCODING_AC3_IEC61937;
break;
case AV_CODEC_ID_EAC3:
e = PA_ENCODING_EAC3_IEC61937;
break;
#if 0
case AV_CODEC_ID_MP1:
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
e = PA_ENCODING_MPEG_IEC61937;
break;
#endif
default:
return 0;
}
pa_threaded_mainloop_lock(mainloop);
if(pulseaudio_make_context_ready()) {
pa_threaded_mainloop_unlock(mainloop);
return 0;
}
if(d->s) {
pa_stream_disconnect(d->s);
pa_stream_unref(d->s);
}
pa_format_info *vec[1];
vec[0] = pa_format_info_new();
vec[0]->encoding = e;
pa_format_info_set_rate(vec[0], 48000);
pa_format_info_set_channels(vec[0], 2);
d->s = pa_stream_new_extended(ctx, "Showtime playback", vec, 1, NULL);
int flags = 0;
pa_stream_set_state_callback(d->s, stream_state_callback, d);
pa_stream_set_write_callback(d->s, stream_write_callback, d);
flags |= PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_INTERPOLATE_TIMING |
PA_STREAM_NOT_MONOTONIC;
pa_stream_connect_playback(d->s, NULL, NULL, flags, NULL, NULL);
while(1) {
switch(pa_stream_get_state(d->s)) {
case PA_STREAM_UNCONNECTED:
case PA_STREAM_CREATING:
pa_threaded_mainloop_wait(mainloop);
continue;
case PA_STREAM_READY:
pa_threaded_mainloop_unlock(mainloop);
ad->ad_tile_size = 512;
d->ss = *pa_stream_get_sample_spec(d->s);
return 1;
case PA_STREAM_TERMINATED:
case PA_STREAM_FAILED:
pa_threaded_mainloop_unlock(mainloop);
return 0;
}
}
}
static int Open(vlc_object_t *obj)
{
demux_t *demux = (demux_t *)obj;
demux_sys_t *sys = malloc(sizeof (*sys));
if (unlikely(sys == NULL))
return VLC_ENOMEM;
sys->context = vlc_pa_connect(obj, &sys->mainloop);
if (sys->context == NULL) {
free(sys);
return VLC_EGENERIC;
}
sys->stream = NULL;
sys->es = NULL;
sys->discontinuity = false;
sys->caching = INT64_C(1000) * var_InheritInteger(obj, "live-caching");
demux->p_sys = sys;
/* Stream parameters */
struct pa_sample_spec ss;
ss.format = PA_SAMPLE_S16NE;
ss.rate = 48000;
ss.channels = 2;
assert(pa_sample_spec_valid(&ss));
struct pa_channel_map map;
map.channels = 2;
map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
assert(pa_channel_map_valid(&map));
const pa_stream_flags_t flags = PA_STREAM_INTERPOLATE_TIMING
| PA_STREAM_AUTO_TIMING_UPDATE
| PA_STREAM_ADJUST_LATENCY
| PA_STREAM_FIX_FORMAT
| PA_STREAM_FIX_RATE
/*| PA_STREAM_FIX_CHANNELS*/;
const char *dev = NULL;
if (demux->psz_location != NULL && demux->psz_location[0] != '\0')
dev = demux->psz_location;
struct pa_buffer_attr attr = {
.maxlength = -1,
.fragsize = pa_usec_to_bytes(sys->caching, &ss) / 2,
};
es_format_t fmt;
/* Create record stream */
pa_stream *s;
pa_operation *op;
pa_threaded_mainloop_lock(sys->mainloop);
s = pa_stream_new(sys->context, "audio stream", &ss, &map);
if (s == NULL)
goto error;
sys->stream = s;
pa_stream_set_state_callback(s, stream_state_cb, sys->mainloop);
pa_stream_set_read_callback(s, stream_read_cb, demux);
pa_stream_set_buffer_attr_callback(s, stream_buffer_attr_cb, demux);
pa_stream_set_moved_callback(s, stream_moved_cb, demux);
pa_stream_set_overflow_callback(s, stream_overflow_cb, demux);
pa_stream_set_started_callback(s, stream_started_cb, demux);
pa_stream_set_suspended_callback(s, stream_suspended_cb, demux);
pa_stream_set_underflow_callback(s, stream_underflow_cb, demux);
if (pa_stream_connect_record(s, dev, &attr, flags) < 0
|| stream_wait(s, sys->mainloop)) {
vlc_pa_error(obj, "cannot connect record stream", sys->context);
goto error;
}
/* The ES should be initialized before stream_read_cb(), but how? */
const struct pa_sample_spec *pss = pa_stream_get_sample_spec(s);
if ((unsigned)pss->format >= sizeof (fourccs) / sizeof (fourccs[0])) {
msg_Err(obj, "unknown PulseAudio sample format %u",
(unsigned)pss->format);
goto error;
}
vlc_fourcc_t format = fourccs[pss->format];
if (format == 0) { /* FIXME: should renegotiate something else */
msg_Err(obj, "unsupported PulseAudio sample format %u",
(unsigned)pss->format);
goto error;
}
es_format_Init(&fmt, AUDIO_ES, format);
fmt.audio.i_physical_channels = fmt.audio.i_original_channels =
AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
fmt.audio.i_channels = ss.channels;
fmt.audio.i_rate = pss->rate;
fmt.audio.i_bitspersample = aout_BitsPerSample(format);
fmt.audio.i_blockalign = fmt.audio.i_bitspersample * ss.channels / 8;
fmt.i_bitrate = fmt.audio.i_bitspersample * ss.channels * pss->rate;
sys->framesize = fmt.audio.i_blockalign;
sys->es = es_out_Add (demux->out, &fmt);
/* Update the buffer attributes according to actual format */
attr.fragsize = pa_usec_to_bytes(sys->caching, pss) / 2;
op = pa_stream_set_buffer_attr(s, &attr, stream_success_cb, sys->mainloop);
if (likely(op != NULL)) {
while (pa_operation_get_state(op) == PA_OPERATION_RUNNING)
pa_threaded_mainloop_wait(sys->mainloop);
pa_operation_unref(op);
}
stream_buffer_attr_cb(s, demux);
pa_threaded_mainloop_unlock(sys->mainloop);
demux->pf_demux = NULL;
demux->pf_control = Control;
return VLC_SUCCESS;
error:
pa_threaded_mainloop_unlock(sys->mainloop);
Close(obj);
return VLC_EGENERIC;
}
static void Close (vlc_object_t *obj)
{
demux_t *demux = (demux_t *)obj;
demux_sys_t *sys = demux->p_sys;
pa_stream *s = sys->stream;
if (likely(s != NULL)) {
pa_threaded_mainloop_lock(sys->mainloop);
pa_stream_disconnect(s);
pa_stream_set_state_callback(s, NULL, NULL);
pa_stream_set_read_callback(s, NULL, NULL);
pa_stream_set_buffer_attr_callback(s, NULL, NULL);
pa_stream_set_moved_callback(s, NULL, NULL);
pa_stream_set_overflow_callback(s, NULL, NULL);
pa_stream_set_started_callback(s, NULL, NULL);
pa_stream_set_suspended_callback(s, NULL, NULL);
pa_stream_set_underflow_callback(s, NULL, NULL);
pa_stream_unref(s);
pa_threaded_mainloop_unlock(sys->mainloop);
}
vlc_pa_disconnect(obj, sys->context, sys->mainloop);
free(sys);
}