static void pulse_read_preprocess(MSFilter *f){
PulseReadState *s=(PulseReadState *)f->data;
int err;
pa_sample_spec pss;
pa_buffer_attr attr;
if (context==NULL) return;
pss.format=PA_SAMPLE_S16LE;
pss.channels=s->channels;
pss.rate=s->rate;
attr.maxlength=-1;
attr.tlength=-1;
attr.prebuf=-1;
attr.minreq=-1;
attr.fragsize=s->fragsize=latency_req*(float)s->channels*(float)s->rate*2;
s->stream=pa_stream_new(context,"phone",&pss,NULL);
if (s->stream==NULL){
ms_error("pa_stream_new() failed: %s",pa_strerror(pa_context_errno(context)));
return;
}
pa_threaded_mainloop_lock(pa_loop);
err=pa_stream_connect_record(s->stream,NULL,&attr, PA_STREAM_ADJUST_LATENCY);
pa_threaded_mainloop_unlock(pa_loop);
if (err!=0){
ms_error("pa_stream_connect_record() failed");
}
}
JNIEXPORT void JNICALL
Java_com_harrcharr_pulse_Stream_connectRecord(
JNIEnv *jenv, jobject jstream, jstring jdev) {
pa_buffer_attr attr;
pa_stream *stream = get_stream_ptr(jenv, jstream);
memset(&attr, 0, sizeof(attr));
attr.fragsize = sizeof(float);
attr.maxlength = (uint32_t) -1;
const char *dev;
dev = (*jenv)->GetStringUTFChars(jenv, jdev, NULL);
if (dev == NULL) {
return; /* OutOfMemoryError already thrown */
}
if (pa_stream_connect_record(stream, dev, &attr,
(pa_stream_flags_t) (PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND|
PA_STREAM_DONT_MOVE|PA_STREAM_PEAK_DETECT|
PA_STREAM_ADJUST_LATENCY)) < 0) {
LOGE("Failed to connect to stream");
// Throw an exception to java
}
(*jenv)->ReleaseStringUTFChars(jenv, jdev, dev);
}
/**
* Start recording
*
* We request the default format used by pulse here because the data will be
* converted and possibly re-sampled by obs anyway.
*
* For now we request a buffer length of 25ms although pulse seems to ignore
* this setting for monitor streams. For "real" input streams this should work
* fine though.
*/
static int_fast32_t pulse_start_recording(struct pulse_data *data)
{
if (pulse_get_server_info(pulse_server_info, (void *) data) < 0) {
blog(LOG_ERROR, "Unable to get server info !");
return -1;
}
if (pulse_get_source_info(pulse_source_info, data->device,
(void *) data) < 0) {
blog(LOG_ERROR, "Unable to get source info !");
return -1;
}
pa_sample_spec spec;
spec.format = data->format;
spec.rate = data->samples_per_sec;
spec.channels = data->channels;
if (!pa_sample_spec_valid(&spec)) {
blog(LOG_ERROR, "Sample spec is not valid");
return -1;
}
data->speakers = pulse_channels_to_obs_speakers(spec.channels);
data->bytes_per_frame = pa_frame_size(&spec);
data->stream = pulse_stream_new(obs_source_get_name(data->source),
&spec, NULL);
if (!data->stream) {
blog(LOG_ERROR, "Unable to create stream");
return -1;
}
pulse_lock();
pa_stream_set_read_callback(data->stream, pulse_stream_read,
(void *) data);
pulse_unlock();
pa_buffer_attr attr;
attr.fragsize = pa_usec_to_bytes(25000, &spec);
attr.maxlength = (uint32_t) -1;
attr.minreq = (uint32_t) -1;
attr.prebuf = (uint32_t) -1;
attr.tlength = (uint32_t) -1;
pa_stream_flags_t flags = PA_STREAM_ADJUST_LATENCY;
pulse_lock();
int_fast32_t ret = pa_stream_connect_record(data->stream, data->device,
&attr, flags);
pulse_unlock();
if (ret < 0) {
pulse_stop_recording(data);
blog(LOG_ERROR, "Unable to connect to stream");
return -1;
}
blog(LOG_INFO, "Started recording from '%s'", data->device);
return 0;
}
/*
* Initialize pulseAudio
*/
static int init_pa(recorder_context_t *rctx)
{
pa_mainloop_api *pa_mlapi;
pa_proplist *ctx_properties = pa_proplist_new();
pa_proplist *stream_properties = pa_proplist_new();
int retval = 0;
rctx->pa_ml = pa_mainloop_new();
pa_mlapi = pa_mainloop_get_api(rctx->pa_ml);
rctx->pa_ctx = pa_context_new_with_proplist(pa_mlapi,
"NoApp recorder", ctx_properties);
pa_context_connect(rctx->pa_ctx, NULL, 0, NULL);
rctx->pa_ready = 0;
pa_context_set_state_callback(rctx->pa_ctx, pa_state_cb, rctx);
while (rctx->pa_ready == 0){
pa_mainloop_iterate(rctx->pa_ml, 1, NULL);
}
if (rctx->pa_ready == 2){
retval = -1;
goto exit;
}
rctx->recording_stream = pa_stream_new_with_proplist(rctx->pa_ctx,
"NoApp recorder", &rctx->pa_ss,
NULL, stream_properties);
retval = pa_stream_connect_record(rctx->recording_stream, NULL, NULL, 0);
if (retval < 0){
Log(LOG_ERR, "pa_stream_connect_playback failed\n");
goto exit;
}
exit:
return retval;
}
static int instream_start_pa(SoundIoPrivate *si, SoundIoInStreamPrivate *is) {
SoundIoInStream *instream = &is->pub;
SoundIoInStreamPulseAudio *ispa = &is->backend_data.pulseaudio;
SoundIoPulseAudio *sipa = &si->backend_data.pulseaudio;
pa_threaded_mainloop_lock(sipa->main_loop);
pa_stream_flags_t flags = PA_STREAM_AUTO_TIMING_UPDATE;
if (instream->software_latency > 0.0)
flags = (pa_stream_flags_t) (flags|PA_STREAM_ADJUST_LATENCY);
int err = pa_stream_connect_record(ispa->stream,
instream->device->id,
&ispa->buffer_attr, flags);
if (err) {
pa_threaded_mainloop_unlock(sipa->main_loop);
return SoundIoErrorOpeningDevice;
}
while (!ispa->stream_ready)
pa_threaded_mainloop_wait(sipa->main_loop);
pa_operation *update_timing_info_op = pa_stream_update_timing_info(ispa->stream, timing_update_callback, si);
if ((err = perform_operation(si, update_timing_info_op))) {
pa_threaded_mainloop_unlock(sipa->main_loop);
return err;
}
pa_threaded_mainloop_unlock(sipa->main_loop);
return 0;
}
static int instream_start_pa(struct SoundIoPrivate *si, struct SoundIoInStreamPrivate *is) {
struct SoundIoInStream *instream = &is->pub;
struct SoundIoInStreamPulseAudio *ispa = &is->backend_data.pulseaudio;
struct SoundIoPulseAudio *sipa = &si->backend_data.pulseaudio;
pa_threaded_mainloop_lock(sipa->main_loop);
pa_stream_flags_t flags = (pa_stream_flags_t)(PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_INTERPOLATE_TIMING);
int err = pa_stream_connect_record(ispa->stream,
instream->device->id,
&ispa->buffer_attr, flags);
if (err) {
pa_threaded_mainloop_unlock(sipa->main_loop);
return SoundIoErrorOpeningDevice;
}
while (!SOUNDIO_ATOMIC_LOAD(ispa->stream_ready))
pa_threaded_mainloop_wait(sipa->main_loop);
pa_operation *update_timing_info_op = pa_stream_update_timing_info(ispa->stream, timing_update_callback, si);
if ((err = perform_operation(si, update_timing_info_op))) {
pa_threaded_mainloop_unlock(sipa->main_loop);
return err;
}
pa_threaded_mainloop_unlock(sipa->main_loop);
return 0;
}
/**
* Pulseaudio context state callback
*/
static void
context_state_callback (pa_context * c,
void *userdata)
{
GNUNET_assert (c);
switch (pa_context_get_state (c))
{
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
case PA_CONTEXT_READY:
{
int r;
pa_buffer_attr na;
GNUNET_assert (!stream_in);
GNUNET_log (GNUNET_ERROR_TYPE_INFO,
_("Connection established.\n"));
if (! (stream_in =
pa_stream_new (c, "GNUNET_VoIP recorder", &sample_spec, NULL)))
{
GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
_("pa_stream_new() failed: %s\n"),
pa_strerror (pa_context_errno (c)));
goto fail;
}
pa_stream_set_state_callback (stream_in, &stream_state_callback, NULL);
pa_stream_set_read_callback (stream_in, &stream_read_callback, NULL);
memset (&na, 0, sizeof (na));
na.maxlength = UINT32_MAX;
na.fragsize = pcm_length;
if ((r = pa_stream_connect_record (stream_in, NULL, &na,
PA_STREAM_ADJUST_LATENCY)) < 0)
{
GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
_("pa_stream_connect_record() failed: %s\n"),
pa_strerror (pa_context_errno (c)));
goto fail;
}
break;
}
case PA_CONTEXT_TERMINATED:
quit (0);
break;
case PA_CONTEXT_FAILED:
default:
GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
_("Connection failure: %s\n"),
pa_strerror (pa_context_errno (c)));
goto fail;
}
return;
fail:
quit (1);
}
static pa_stream *qpa_simple_new (
paaudio *g,
const char *name,
pa_stream_direction_t dir,
const char *dev,
const pa_sample_spec *ss,
const pa_channel_map *map,
const pa_buffer_attr *attr,
int *rerror)
{
int r;
pa_stream *stream;
pa_threaded_mainloop_lock (g->mainloop);
stream = pa_stream_new (g->context, name, ss, map);
if (!stream) {
goto fail;
}
pa_stream_set_state_callback (stream, stream_state_cb, g);
pa_stream_set_read_callback (stream, stream_request_cb, g);
pa_stream_set_write_callback (stream, stream_request_cb, g);
if (dir == PA_STREAM_PLAYBACK) {
r = pa_stream_connect_playback (stream, dev, attr,
PA_STREAM_INTERPOLATE_TIMING
#ifdef PA_STREAM_ADJUST_LATENCY
|PA_STREAM_ADJUST_LATENCY
#endif
|PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
} else {
r = pa_stream_connect_record (stream, dev, attr,
PA_STREAM_INTERPOLATE_TIMING
#ifdef PA_STREAM_ADJUST_LATENCY
|PA_STREAM_ADJUST_LATENCY
#endif
|PA_STREAM_AUTO_TIMING_UPDATE);
}
if (r < 0) {
goto fail;
}
pa_threaded_mainloop_unlock (g->mainloop);
return stream;
fail:
pa_threaded_mainloop_unlock (g->mainloop);
if (stream) {
pa_stream_unref (stream);
}
*rerror = pa_context_errno (g->context);
return NULL;
}
/*
* start recording
*/
static int_fast32_t pulse_start_recording(struct pulse_data *data)
{
if (pulse_get_server_info(pulse_server_info, (void *) data) < 0) {
blog(LOG_ERROR, "pulse-input: Unable to get server info !");
return -1;
}
pa_sample_spec spec;
spec.format = data->format;
spec.rate = data->samples_per_sec;
spec.channels = data->channels;
if (!pa_sample_spec_valid(&spec)) {
blog(LOG_ERROR, "pulse-input: Sample spec is not valid");
return -1;
}
data->bytes_per_frame = pa_frame_size(&spec);
blog(LOG_DEBUG, "pulse-input: %u bytes per frame",
(unsigned int) data->bytes_per_frame);
data->stream = pulse_stream_new(obs_source_getname(data->source),
&spec, NULL);
if (!data->stream) {
blog(LOG_ERROR, "pulse-input: Unable to create stream");
return -1;
}
pulse_lock();
pa_stream_set_read_callback(data->stream, pulse_stream_read,
(void *) data);
pulse_unlock();
pa_buffer_attr attr;
attr.fragsize = get_buffer_size(data, 250);
attr.maxlength = (uint32_t) -1;
attr.minreq = (uint32_t) -1;
attr.prebuf = (uint32_t) -1;
attr.tlength = (uint32_t) -1;
pa_stream_flags_t flags =
PA_STREAM_INTERPOLATE_TIMING
| PA_STREAM_AUTO_TIMING_UPDATE
| PA_STREAM_ADJUST_LATENCY;
pulse_lock();
int_fast32_t ret = pa_stream_connect_record(data->stream, data->device,
&attr, flags);
pulse_unlock();
if (ret < 0) {
blog(LOG_ERROR, "pulse-input: Unable to connect to stream");
return -1;
}
blog(LOG_DEBUG, "pulse-input: Recording started");
return 0;
}
static int m_pa_stream_connect(pa_context *pa_ctx)
{
if (pa_context_get_server_protocol_version (pa_ctx) < 13) {
return -1;
}
printf("server version: %d\n", pa_context_get_server_protocol_version(pa_ctx));
if (s) {
pa_stream_disconnect(s);
pa_stream_unref(s);
}
pa_proplist *proplist;
pa_buffer_attr attr;
pa_sample_spec ss;
int res;
//char dev_name[40];
// pa_sample_spec
ss.channels = 1;
ss.format = PA_SAMPLE_FLOAT32;
ss.rate = 25;
// pa_buffer_attr
memset(&attr, 0, sizeof(attr));
attr.fragsize = sizeof(float);
attr.maxlength = (uint32_t) -1;
// pa_proplist
proplist = pa_proplist_new ();
pa_proplist_sets (proplist, PA_PROP_APPLICATION_ID, "Deepin Sound Settings");
// create new stream
if (!(s = pa_stream_new_with_proplist(pa_ctx, "Deepin Sound Settings", &ss, NULL, proplist))) {
fprintf(stderr, "pa_stream_new error\n");
return -2;
}
pa_proplist_free(proplist);
pa_stream_set_read_callback(s, on_monitor_read_callback, NULL);
pa_stream_set_suspended_callback(s, on_monitor_suspended_callback, NULL);
res = pa_stream_connect_record(s, NULL, &attr,
(pa_stream_flags_t) (PA_STREAM_DONT_MOVE
|PA_STREAM_PEAK_DETECT
|PA_STREAM_ADJUST_LATENCY));
if (res < 0) {
fprintf(stderr, "Failed to connect monitoring stream\n");
return -3;
}
return 0;
}
/*
* Create a new pulse audio stream and connect to it
*
* Return a negative value on error
*/
static int pulse_connect_stream(struct pulse_data *data)
{
pa_sample_spec spec;
spec.format = data->format;
spec.rate = data->samples_per_sec;
spec.channels = get_audio_channels(data->speakers);
if (!pa_sample_spec_valid(&spec)) {
blog(LOG_ERROR, "pulse-input: Sample spec is not valid");
return -1;
}
data->bytes_per_frame = pa_frame_size(&spec);
blog(LOG_DEBUG, "pulse-input: %u bytes per frame",
(unsigned int) data->bytes_per_frame);
pa_buffer_attr attr;
attr.fragsize = get_buffer_size(data, 250);
attr.maxlength = (uint32_t) -1;
attr.minreq = (uint32_t) -1;
attr.prebuf = (uint32_t) -1;
attr.tlength = (uint32_t) -1;
data->stream = pa_stream_new_with_proplist(data->context,
obs_source_getname(data->source), &spec, NULL, data->props);
if (!data->stream) {
blog(LOG_ERROR, "pulse-input: Unable to create stream");
return -1;
}
pa_stream_flags_t flags =
PA_STREAM_INTERPOLATE_TIMING
| PA_STREAM_AUTO_TIMING_UPDATE
| PA_STREAM_ADJUST_LATENCY;
if (pa_stream_connect_record(data->stream, NULL, &attr, flags) < 0) {
blog(LOG_ERROR, "pulse-input: Unable to connect to stream");
return -1;
}
for (;;) {
pulse_iterate(data);
pa_stream_state_t state = pa_stream_get_state(data->stream);
if (state == PA_STREAM_READY) {
blog(LOG_DEBUG, "pulse-input: Stream ready");
break;
}
if (!PA_STREAM_IS_GOOD(state)) {
blog(LOG_ERROR, "pulse-input: Stream connect failed");
return -1;
}
}
return 0;
}
void QPulseAudioThread::connectHelper (SourceContainer::const_iterator pos)
{
Q_ASSERT(stream);
pa_stream_flags_t flags = ( pa_stream_flags_t ) 0;
// qDebug() << "start2 ";
assert (pos != s_sourceList.end());
qDebug() << "connectHelper: " << *pos;
int r;
if ( ( ( r = pa_stream_connect_record (stream, (*pos).toStdString().c_str(), NULL, flags ) ) ) < 0 ) {
fprintf ( stderr, "pa_stream_connect_record() failed: %s\n", pa_strerror ( pa_context_errno ( context ) ) );
}
}
AudioStream::AudioStream(pa_context *c, pa_threaded_mainloop *m, const char *desc, int type, int smplrate, std::string& deviceName)
: audiostream_(0), mainloop_(m)
{
static const pa_channel_map channel_map = {
1,
{ PA_CHANNEL_POSITION_MONO },
};
pa_sample_spec sample_spec = {
PA_SAMPLE_S16LE, // PA_SAMPLE_FLOAT32LE,
smplrate,
1
};
assert(pa_sample_spec_valid(&sample_spec));
assert(pa_channel_map_valid(&channel_map));
audiostream_ = pa_stream_new(c, desc, &sample_spec, &channel_map);
if (!audiostream_) {
ERROR("%s: pa_stream_new() failed : %s" , desc, pa_strerror(pa_context_errno(c)));
throw std::runtime_error("Could not create stream\n");
}
pa_buffer_attr attributes;
attributes.maxlength = pa_usec_to_bytes(160 * PA_USEC_PER_MSEC, &sample_spec);
attributes.tlength = pa_usec_to_bytes(80 * PA_USEC_PER_MSEC, &sample_spec);
attributes.prebuf = 0;
attributes.fragsize = pa_usec_to_bytes(80 * PA_USEC_PER_MSEC, &sample_spec);
attributes.minreq = (uint32_t) -1;
pa_threaded_mainloop_lock(mainloop_);
if (type == PLAYBACK_STREAM || type == RINGTONE_STREAM)
pa_stream_connect_playback(audiostream_, deviceName == "" ? NULL : deviceName.c_str(), &attributes,
(pa_stream_flags_t)(PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE), NULL, NULL);
else if (type == CAPTURE_STREAM)
pa_stream_connect_record(audiostream_, deviceName == "" ? NULL : deviceName.c_str(), &attributes,
(pa_stream_flags_t)(PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE));
pa_threaded_mainloop_unlock(mainloop_);
pa_stream_set_state_callback(audiostream_, stream_state_callback, NULL);
}
static pa_stream *connect_record_stream(const char *device_name,
pa_threaded_mainloop *loop, pa_context *context,
pa_stream_flags_t flags, pa_buffer_attr *attr, pa_sample_spec *spec,
pa_channel_map *chanmap)
{
pa_stream_state_t state;
pa_stream *stream;
stream = pa_stream_new_with_proplist(context, "Capture Stream", spec, chanmap, prop_filter);
if(!stream)
{
ERR("pa_stream_new_with_proplist() failed: %s\n", pa_strerror(pa_context_errno(context)));
return NULL;
}
pa_stream_set_state_callback(stream, stream_state_callback, loop);
if(pa_stream_connect_record(stream, device_name, attr, flags) < 0)
{
ERR("Stream did not connect: %s\n", pa_strerror(pa_context_errno(context)));
pa_stream_unref(stream);
return NULL;
}
while((state=pa_stream_get_state(stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
ERR("Stream did not get ready: %s\n", pa_strerror(pa_context_errno(context)));
pa_stream_unref(stream);
return NULL;
}
pa_threaded_mainloop_wait(loop);
}
pa_stream_set_state_callback(stream, NULL, NULL);
return stream;
}
JNIEXPORT jint JNICALL
Java_org_jitsi_impl_neomedia_pulseaudio_PA_stream_1connect_1record
(JNIEnv *env, jclass clazz, jlong s, jstring dev, jlong attr, jint flags)
{
const char *devChars
= dev ? (*env)->GetStringUTFChars(env, dev, NULL) : NULL;
jint ret;
if ((*env)->ExceptionCheck(env))
ret = -1;
else
{
ret
= pa_stream_connect_record(
(pa_stream *) (intptr_t) s,
devChars,
(const pa_buffer_attr *) (intptr_t) attr,
(pa_stream_flags_t) flags);
(*env)->ReleaseStringUTFChars(env, dev, devChars);
}
return ret;
}
static void __context_get_sink_info_callback(pa_context* context,
const pa_sink_info* info, int is_last, void* data) {
guac_client* client = (guac_client*) data;
pa_stream* stream;
pa_sample_spec spec;
pa_buffer_attr attr;
/* Stop if end of list reached */
if (is_last)
return;
guac_client_log_info(client, "Starting streaming from \"%s\"",
info->description);
/* Set format */
spec.format = PA_SAMPLE_S16LE;
spec.rate = GUAC_VNC_AUDIO_RATE;
spec.channels = GUAC_VNC_AUDIO_CHANNELS;
attr.maxlength = -1;
attr.fragsize = GUAC_VNC_AUDIO_FRAGMENT_SIZE;
/* Create stream */
stream = pa_stream_new(context, "Guacamole Audio", &spec, NULL);
/* Set stream callbacks */
pa_stream_set_state_callback(stream, __stream_state_callback, client);
pa_stream_set_read_callback(stream, __stream_read_callback, client);
/* Start stream */
pa_stream_connect_record(stream, info->monitor_source_name, &attr,
PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND
| PA_STREAM_ADJUST_LATENCY);
}
static void audin_pulse_open(IAudinDevice* device, AudinReceive receive, void* user_data)
{
pa_stream_state_t state;
pa_buffer_attr buffer_attr = { 0 };
AudinPulseDevice* pulse = (AudinPulseDevice*) device;
if (!pulse->context)
return;
if (!pulse->sample_spec.rate || pulse->stream)
return;
DEBUG_DVC("");
pulse->receive = receive;
pulse->user_data = user_data;
pa_threaded_mainloop_lock(pulse->mainloop);
pulse->stream = pa_stream_new(pulse->context, "freerdp_audin",
&pulse->sample_spec, NULL);
if (!pulse->stream)
{
pa_threaded_mainloop_unlock(pulse->mainloop);
DEBUG_DVC("pa_stream_new failed (%d)",
pa_context_errno(pulse->context));
return;
}
pulse->bytes_per_frame = pa_frame_size(&pulse->sample_spec);
pa_stream_set_state_callback(pulse->stream,
audin_pulse_stream_state_callback, pulse);
pa_stream_set_read_callback(pulse->stream,
audin_pulse_stream_request_callback, pulse);
buffer_attr.maxlength = (uint32_t) -1;
buffer_attr.tlength = (uint32_t) -1;
buffer_attr.prebuf = (uint32_t) -1;
buffer_attr.minreq = (uint32_t) -1;
/* 500ms latency */
buffer_attr.fragsize = pa_usec_to_bytes(500000, &pulse->sample_spec);
if (pa_stream_connect_record(pulse->stream,
pulse->device_name[0] ? pulse->device_name : NULL,
&buffer_attr, PA_STREAM_ADJUST_LATENCY) < 0)
{
pa_threaded_mainloop_unlock(pulse->mainloop);
DEBUG_WARN("pa_stream_connect_playback failed (%d)",
pa_context_errno(pulse->context));
return;
}
for (;;)
{
state = pa_stream_get_state(pulse->stream);
if (state == PA_STREAM_READY)
break;
if (!PA_STREAM_IS_GOOD(state))
{
DEBUG_WARN("bad stream state (%d)",
pa_context_errno(pulse->context));
break;
}
pa_threaded_mainloop_wait(pulse->mainloop);
}
pa_threaded_mainloop_unlock(pulse->mainloop);
if (state == PA_STREAM_READY)
{
memset(&pulse->adpcm, 0, sizeof(ADPCM));
pulse->buffer = xzalloc(pulse->bytes_per_frame * pulse->frames_per_packet);
pulse->buffer_frames = 0;
DEBUG_DVC("connected");
}
else
{
audin_pulse_close(device);
}
}
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);
}
}
}
/* This is called whenever the context status changes */
static void context_state_callback(pa_context *c, void *userdata) {
pa_assert(c);
switch (pa_context_get_state(c)) {
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
case PA_CONTEXT_READY: {
pa_buffer_attr buffer_attr;
pa_assert(c);
pa_assert(!stream);
if (verbose)
pa_log(_("Connection established.%s"), CLEAR_LINE);
if (!(stream = pa_stream_new_with_proplist(c, NULL, &sample_spec, &channel_map, proplist))) {
pa_log(_("pa_stream_new() failed: %s"), pa_strerror(pa_context_errno(c)));
goto fail;
}
pa_stream_set_state_callback(stream, stream_state_callback, NULL);
pa_stream_set_write_callback(stream, stream_write_callback, NULL);
pa_stream_set_read_callback(stream, stream_read_callback, NULL);
pa_stream_set_suspended_callback(stream, stream_suspended_callback, NULL);
pa_stream_set_moved_callback(stream, stream_moved_callback, NULL);
pa_stream_set_underflow_callback(stream, stream_underflow_callback, NULL);
pa_stream_set_overflow_callback(stream, stream_overflow_callback, NULL);
pa_stream_set_started_callback(stream, stream_started_callback, NULL);
pa_stream_set_event_callback(stream, stream_event_callback, NULL);
pa_stream_set_buffer_attr_callback(stream, stream_buffer_attr_callback, NULL);
pa_zero(buffer_attr);
buffer_attr.maxlength = (uint32_t) -1;
buffer_attr.prebuf = (uint32_t) -1;
if (latency_msec > 0) {
buffer_attr.fragsize = buffer_attr.tlength = pa_usec_to_bytes(latency_msec * PA_USEC_PER_MSEC, &sample_spec);
flags |= PA_STREAM_ADJUST_LATENCY;
} else if (latency > 0) {
buffer_attr.fragsize = buffer_attr.tlength = (uint32_t) latency;
flags |= PA_STREAM_ADJUST_LATENCY;
} else
buffer_attr.fragsize = buffer_attr.tlength = (uint32_t) -1;
if (process_time_msec > 0) {
buffer_attr.minreq = pa_usec_to_bytes(process_time_msec * PA_USEC_PER_MSEC, &sample_spec);
} else if (process_time > 0)
buffer_attr.minreq = (uint32_t) process_time;
else
buffer_attr.minreq = (uint32_t) -1;
if (mode == PLAYBACK) {
pa_cvolume cv;
if (pa_stream_connect_playback(stream, device, &buffer_attr, flags, volume_is_set ? pa_cvolume_set(&cv, sample_spec.channels, volume) : NULL, NULL) < 0) {
pa_log(_("pa_stream_connect_playback() failed: %s"), pa_strerror(pa_context_errno(c)));
goto fail;
}
} else {
if (pa_stream_connect_record(stream, device, latency > 0 ? &buffer_attr : NULL, flags) < 0) {
pa_log(_("pa_stream_connect_record() failed: %s"), pa_strerror(pa_context_errno(c)));
goto fail;
}
}
break;
}
case PA_CONTEXT_TERMINATED:
quit(0);
break;
case PA_CONTEXT_FAILED:
default:
pa_log(_("Connection failure: %s"), pa_strerror(pa_context_errno(c)));
goto fail;
}
return;
fail:
quit(1);
}
/*
* Iterate the main loop while recording is on.
* This function runs under it's own thread called by audio_pulse_start
* args:
* data - pointer to user data (audio context)
*
* asserts:
* data is not null
*
* returns: pointer to error code
*/
static void *pulse_read_audio(void *data)
{
audio_context_t *audio_ctx = (audio_context_t *) data;
/*assertions*/
assert(audio_ctx != NULL);
if(verbosity > 0)
printf("AUDIO: (pulseaudio) read thread started\n");
pa_mainloop *pa_ml;
pa_mainloop_api *pa_mlapi;
pa_buffer_attr bufattr;
pa_sample_spec ss;
pa_stream_flags_t flags = 0;
int r;
int pa_ready = 0;
/* Create a mainloop API and connection to the default server */
pa_ml = pa_mainloop_new();
pa_mlapi = pa_mainloop_get_api(pa_ml);
pa_ctx = pa_context_new(pa_mlapi, "guvcview Pulse API");
if(pa_context_connect(pa_ctx, NULL, 0, NULL) < 0)
{
fprintf(stderr,"AUDIO: PULSE - unable to connect to server: pa_context_connect failed\n");
finish(pa_ctx, pa_ml);
return ((void *) -1);
}
/*
* This function defines a callback so the server will tell us it's state.
* Our callback will wait for the state to be ready. The callback will
* modify the variable to 1 so we know when we have a connection and it's
* ready.
* If there's an error, the callback will set pa_ready to 2
*/
pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);
/*
* This function defines a time event callback (called every TIME_EVENT_USEC)
*/
//pa_context_rttime_new(pa_ctx, pa_rtclock_now() + TIME_EVENT_USEC, time_event_callback, NULL);
/*
* We can't do anything until PA is ready, so just iterate the mainloop
* and continue
*/
while (pa_ready == 0)
{
pa_mainloop_iterate(pa_ml, 1, NULL);
}
if (pa_ready == 2)
{
finish(pa_ctx, pa_ml);
return ((void *) -1);
}
/* set the sample spec (frame rate, channels and format) */
ss.rate = audio_ctx->samprate;
ss.channels = audio_ctx->channels;
ss.format = PA_SAMPLE_FLOAT32LE; /*for PCM -> PA_SAMPLE_S16LE*/
recordstream = pa_stream_new(pa_ctx, "Record", &ss, NULL);
if (!recordstream)
fprintf(stderr, "AUDIO: (pulseaudio) pa_stream_new failed (chan:%d rate:%d)\n",
ss.channels, ss.rate);
/* define the callbacks */
pa_stream_set_read_callback(recordstream, stream_request_cb, (void *) audio_ctx);
// Set properties of the record buffer
pa_zero(bufattr);
/* optimal value for all is (uint32_t)-1 ~= 2 sec */
bufattr.maxlength = (uint32_t) -1;
bufattr.prebuf = (uint32_t) -1;
bufattr.minreq = (uint32_t) -1;
if (audio_ctx->latency > 0)
{
bufattr.fragsize = bufattr.tlength = pa_usec_to_bytes((audio_ctx->latency * 1000) * PA_USEC_PER_MSEC, &ss);
flags |= PA_STREAM_ADJUST_LATENCY;
}
else
bufattr.fragsize = bufattr.tlength = (uint32_t) -1;
flags |= PA_STREAM_INTERPOLATE_TIMING;
flags |= PA_STREAM_AUTO_TIMING_UPDATE;
char * dev = audio_ctx->list_devices[audio_ctx->device].name;
if(verbosity > 0)
printf("AUDIO: (pulseaudio) connecting to device %s\n\t (channels %d rate %d)\n",
dev, ss.channels, ss.rate);
r = pa_stream_connect_record(recordstream, dev, &bufattr, flags);
if (r < 0)
{
fprintf(stderr, "AUDIO: (pulseaudio) skip latency adjustment\n");
/*
* Old pulse audio servers don't like the ADJUST_LATENCY flag,
* so retry without that
*/
r = pa_stream_connect_record(recordstream, dev, &bufattr,
PA_STREAM_INTERPOLATE_TIMING|
PA_STREAM_AUTO_TIMING_UPDATE);
}
if (r < 0)
{
fprintf(stderr, "AUDIO: (pulseaudio) pa_stream_connect_record failed\n");
finish(pa_ctx, pa_ml);
return ((void *) -1);
}
get_latency(recordstream);
/*
* Iterate the main loop while streaming. The second argument is whether
* or not the iteration should block until something is ready to be
* done. Set it to zero for non-blocking.
*/
while (audio_ctx->stream_flag == AUDIO_STRM_ON)
{
pa_mainloop_iterate(pa_ml, 1, NULL);
}
if(verbosity > 0)
printf("AUDIO: (pulseaudio) stream terminated(%i)\n", audio_ctx->stream_flag);
pa_stream_disconnect (recordstream);
pa_stream_unref (recordstream);
finish(pa_ctx, pa_ml);
return ((void *) 0);
}
static void conn_state(pa_context *context, void *arg)
{
int err;
struct pulse_conn_t *conn = arg;
pa_context_state_t state = pa_context_get_state(context);
switch(state) {
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING: break;
case PA_CONTEXT_AUTHORIZING: break;
case PA_CONTEXT_SETTING_NAME: break;
case PA_CONTEXT_FAILED:
case PA_CONTEXT_TERMINATED:
break;
case PA_CONTEXT_READY:
{
pa_sample_spec spec;
pa_buffer_attr attr;
spec.rate = conn->conf.rate;
spec.format = PA_SAMPLE_FLOAT32NE;
if(conn->conf.in > 0) {
spec.channels = conn->conf.in;
conn->record = pa_stream_new(conn->context, "Record", &spec, NULL);
pa_stream_set_read_callback(conn->record, conn_record, conn);
pa_stream_set_overflow_callback(conn->record, conn_overflow, conn);
}
else
conn->record = NULL;
if(conn->conf.out > 0) {
spec.channels = conn->conf.out;
conn->playback = pa_stream_new(conn->context, "Playback", &spec, NULL);
pa_stream_set_write_callback(conn->playback, conn_playback, conn);
pa_stream_set_underflow_callback(conn->playback, conn_underflow, conn);
}
else
conn->playback = NULL;
attr.fragsize = sizeof(float) * conn->lat;
attr.maxlength = (uint32_t)-1;
attr.maxlength = sizeof(float) * conn->lat;
attr.minreq = 0;
attr.prebuf = 0;
attr.tlength = sizeof(float) * conn->lat;
if(conn->record != NULL) {
err = pa_stream_connect_record(conn->record, NULL, &attr, PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
if(err < 0)
fprintf(stderr, "Failed to connect to recorder."), exit(1);
}
if(conn->playback != NULL) {
err = pa_stream_connect_playback(conn->playback, NULL, &attr, PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
if(err < 0)
fprintf(stderr, "Failed to connect to playback."), exit(1);
}
}
break;
}
}
krad_pulse_t *kradpulse_create(krad_audio_t *kradaudio) {
krad_pulse_t *kradpulse;
if ((kradpulse = calloc (1, sizeof (krad_pulse_t))) == NULL) {
fprintf(stderr, "mem alloc fail\n");
exit (1);
}
kradpulse->kradaudio = kradaudio;
kradpulse->samples[0] = malloc(24 * 8192);
kradpulse->samples[1] = malloc(24 * 8192);
kradpulse->interleaved_samples = malloc(48 * 8192);
kradpulse->capture_samples[0] = malloc(24 * 8192);
kradpulse->capture_samples[1] = malloc(24 * 8192);
kradpulse->capture_interleaved_samples = malloc(48 * 8192);
kradpulse->latency = 20000; // start latency in micro seconds
kradpulse->underflows = 0;
kradpulse->pa_ready = 0;
kradpulse->retval = 0;
// Create a mainloop API and connection to the default server
kradpulse->pa_ml = pa_mainloop_new();
kradpulse->pa_mlapi = pa_mainloop_get_api(kradpulse->pa_ml);
kradpulse->pa_ctx = pa_context_new(kradpulse->pa_mlapi, kradpulse->kradaudio->name);
pa_context_connect(kradpulse->pa_ctx, NULL, 0, NULL);
// This function defines a callback so the server will tell us it's state.
// Our callback will wait for the state to be ready. The callback will
// modify the variable to 1 so we know when we have a connection and it's
// ready.
// If there's an error, the callback will set pa_ready to 2
pa_context_set_state_callback(kradpulse->pa_ctx, kradpulse_state_cb, kradpulse);
// We can't do anything until PA is ready, so just iterate the mainloop
// and continue
while (kradpulse->pa_ready == 0) {
pa_mainloop_iterate(kradpulse->pa_ml, 1, NULL);
}
if (kradpulse->pa_ready == 2) {
kradpulse->retval = -1;
printf("pulseaudio fail\n");
exit(1);
}
kradpulse->ss.rate = 44100;
kradpulse->ss.channels = 2;
kradpulse->ss.format = PA_SAMPLE_FLOAT32LE;
kradpulse->bufattr.fragsize = (uint32_t)-1;
kradpulse->bufattr.maxlength = pa_usec_to_bytes(kradpulse->latency, &kradpulse->ss);
kradpulse->bufattr.minreq = pa_usec_to_bytes(0, &kradpulse->ss);
kradpulse->bufattr.prebuf = (uint32_t)-1;
kradpulse->bufattr.tlength = pa_usec_to_bytes(kradpulse->latency, &kradpulse->ss);
if ((kradaudio->direction == KOUTPUT) || (kradaudio->direction == KDUPLEX)) {
kradpulse->playstream = pa_stream_new(kradpulse->pa_ctx, "Playback", &kradpulse->ss, NULL);
if (!kradpulse->playstream) {
printf("playback pa_stream_new failed\n");
exit(1);
}
pa_stream_set_write_callback(kradpulse->playstream, kradpulse_playback_cb, kradpulse);
pa_stream_set_underflow_callback(kradpulse->playstream, kradpulse_stream_underflow_cb, kradpulse);
kradpulse->r = pa_stream_connect_playback(kradpulse->playstream, NULL, &kradpulse->bufattr,
PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
if (kradpulse->r < 0) {
printf("pa_stream_connect_playback failed\n");
kradpulse->retval = -1;
printf("pulseaudio fail\n");
exit(1);
}
}
if ((kradaudio->direction == KINPUT) || (kradaudio->direction == KDUPLEX)) {
kradpulse->capturestream = pa_stream_new(kradpulse->pa_ctx, "Capture", &kradpulse->ss, NULL);
if (!kradpulse->capturestream) {
printf("capture pa_stream_new failed\n");
exit(1);
}
pa_stream_set_read_callback(kradpulse->capturestream, kradpulse_capture_cb, kradpulse);
pa_stream_set_underflow_callback(kradpulse->capturestream, kradpulse_stream_underflow_cb, kradpulse);
kradpulse->r = pa_stream_connect_record(kradpulse->capturestream, NULL, &kradpulse->bufattr, PA_STREAM_NOFLAGS );
if (kradpulse->r < 0) {
printf("pa_stream_connect_capture failed\n");
kradpulse->retval = -1;
printf("pulseaudio fail\n");
exit(1);
}
}
kradaudio->sample_rate = kradpulse->ss.rate;
pthread_create( &kradpulse->loop_thread, NULL, kradpulse_loop_thread, kradpulse);
return kradpulse;
}
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_stream_flags_t flags;
pa_operation *o;
GstAudioClock *clock;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
{
GstAudioRingBufferSpec s = *spec;
const pa_channel_map *m;
m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &s);
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(pulsesrc)->ringbuffer, s.info.position);
}
/* enable event notifications */
GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
if (!(o = pa_context_subscribe (pulsesrc->context,
PA_SUBSCRIPTION_MASK_SOURCE_OUTPUT, NULL, NULL))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_subscribe() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pa_operation_unref (o);
wanted.maxlength = -1;
wanted.tlength = -1;
wanted.prebuf = 0;
wanted.minreq = -1;
wanted.fragsize = spec->segsize;
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d", wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", wanted.prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize);
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
PA_STREAM_START_CORKED;
if (pulsesrc->mute_set && pulsesrc->mute)
flags |= PA_STREAM_START_MUTED;
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
flags) < 0) {
goto connect_failed;
}
/* our clock will now start from 0 again */
clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
gst_audio_clock_reset (clock, 0);
pulsesrc->corked = TRUE;
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pulsesrc->stream);
if (!PA_STREAM_IS_GOOD (state))
goto stream_is_bad;
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
pulsesrc->stream_connected = TRUE;
/* store the source output index so it can be accessed via a property */
pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
if (pulsesrc->volume_set) {
gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
pulsesrc->volume_set = FALSE;
}
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pulsesrc->stream);
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d (wanted: %d)",
actual->tlength, wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", actual->prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d (wanted %d)", actual->minreq,
wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d (wanted %d)",
actual->fragsize, wanted.fragsize);
if (actual->fragsize >= wanted.fragsize) {
spec->segsize = actual->fragsize;
} else {
spec->segsize = actual->fragsize * (wanted.fragsize / actual->fragsize);
}
spec->segtotal = actual->maxlength / spec->segsize;
if (!pulsesrc->paused) {
GST_DEBUG_OBJECT (pulsesrc, "uncorking because we are playing");
gst_pulsesrc_set_corked (pulsesrc, FALSE, FALSE);
}
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
/* ERRORS */
connect_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
stream_is_bad:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
unlock_and_fail:
{
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
}
static ALCenum pulse_open_capture(ALCdevice *device, const ALCchar *device_name) //{{{
{
char *pulse_name = NULL;
pulse_data *data;
pa_stream_flags_t flags = 0;
pa_stream_state_t state;
pa_channel_map chanmap;
if(!allCaptureDevNameMap)
probe_devices(AL_TRUE);
if(!device_name)
device_name = pulse_device;
else if(strcmp(device_name, pulse_device) != 0)
{
ALuint i;
for(i = 0;i < numCaptureDevNames;i++)
{
if(strcmp(device_name, allCaptureDevNameMap[i].name) == 0)
{
pulse_name = allCaptureDevNameMap[i].device_name;
break;
}
}
if(i == numCaptureDevNames)
return ALC_INVALID_VALUE;
}
if(pulse_open(device, device_name) == ALC_FALSE)
return ALC_INVALID_VALUE;
data = device->ExtraData;
pa_threaded_mainloop_lock(data->loop);
data->samples = device->UpdateSize * device->NumUpdates;
data->frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->samples = maxu(data->samples, 100 * device->Frequency / 1000);
if(!(data->ring = CreateRingBuffer(data->frame_size, data->samples)))
{
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
data->attr.minreq = -1;
data->attr.prebuf = -1;
data->attr.maxlength = data->samples * data->frame_size;
data->attr.tlength = -1;
data->attr.fragsize = minu(data->samples, 50*device->Frequency/1000) *
data->frame_size;
data->spec.rate = device->Frequency;
data->spec.channels = ChannelsFromDevFmt(device->FmtChans);
switch(device->FmtType)
{
case DevFmtUByte:
data->spec.format = PA_SAMPLE_U8;
break;
case DevFmtShort:
data->spec.format = PA_SAMPLE_S16NE;
break;
case DevFmtFloat:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
case DevFmtByte:
case DevFmtUShort:
ERR("Capture format type %#x capture not supported on PulseAudio\n", device->FmtType);
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
if(pa_sample_spec_valid(&data->spec) == 0)
{
ERR("Invalid sample format\n");
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
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);
goto fail;
}
data->stream = pa_stream_new(data->context, "Capture Stream", &data->spec, &chanmap);
if(!data->stream)
{
ERR("pa_stream_new() failed: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
pa_stream_set_state_callback(data->stream, stream_state_callback, data->loop);
flags |= PA_STREAM_START_CORKED|PA_STREAM_ADJUST_LATENCY;
if(pa_stream_connect_record(data->stream, pulse_name, &data->attr, flags) < 0)
{
ERR("Stream did not connect: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_stream_unref(data->stream);
data->stream = NULL;
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
while((state=pa_stream_get_state(data->stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
ERR("Stream did not get ready: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_stream_unref(data->stream);
data->stream = NULL;
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
pa_threaded_mainloop_wait(data->loop);
}
pa_stream_set_state_callback(data->stream, stream_state_callback2, device);
pa_threaded_mainloop_unlock(data->loop);
return ALC_NO_ERROR;
fail:
pulse_close(device);
return ALC_INVALID_VALUE;
} //}}}
int main(int argc, const char *argv[])
{
pa_mainloop *pa_ml = NULL;
pa_mainloop_api *pa_mlapi = NULL;
pa_operation *pa_op = NULL;
pa_context *pa_ctx = NULL;
int pa_ready = 0;
int state = 0;
pa_ml = pa_mainloop_new();
pa_mlapi = pa_mainloop_get_api(pa_ml);
pa_ctx = pa_context_new(pa_mlapi, "deepin");
pa_context_connect(pa_ctx, NULL, 0, NULL);
pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);
for (;;) {
if (0 == pa_ready) {
pa_mainloop_iterate(pa_ml, 1, NULL);
continue;
}
if (2 == pa_ready) {
pa_context_disconnect(pa_ctx);
pa_context_unref(pa_ctx);
pa_mainloop_free(pa_ml);
return -1;
}
switch (state) {
case 0:
if (pa_context_get_server_protocol_version (pa_ctx) < 13) {
return -1;
}
printf("server version: %d\n", pa_context_get_server_protocol_version(pa_ctx));
pa_stream *s = NULL;
pa_proplist *proplist;
pa_buffer_attr attr;
pa_sample_spec ss;
int res;
char dev_name[40];
// pa_sample_spec
ss.channels = 1;
ss.format = PA_SAMPLE_FLOAT32;
ss.rate = 25;
// pa_buffer_attr
memset(&attr, 0, sizeof(attr));
attr.fragsize = sizeof(float);
attr.maxlength = (uint32_t) -1;
// pa_proplist
proplist = pa_proplist_new ();
pa_proplist_sets (proplist, PA_PROP_APPLICATION_ID, "deepin.sound");
// create new stream
if (!(s = pa_stream_new_with_proplist(pa_ctx, "Peak detect", &ss, NULL, proplist))) {
fprintf(stderr, "pa_stream_new error\n");
return -2;
}
pa_proplist_free(proplist);
/*pa_stream_set_monitor_stream(s, 26);*/
pa_stream_set_read_callback(s, on_monitor_read_callback, NULL);
pa_stream_set_suspended_callback(s, on_monitor_suspended_callback, NULL);
res = pa_stream_connect_record(s, NULL, &attr,
(pa_stream_flags_t) (PA_STREAM_DONT_MOVE
|PA_STREAM_PEAK_DETECT
|PA_STREAM_ADJUST_LATENCY));
if (res < 0) {
fprintf(stderr, "Failed to connect monitoring stream\n");
return -3;
}
state++;
break;
case 1:
usleep(100);
break;
case 2:
return 0;
break;
default:
return -1;
}
pa_mainloop_iterate(pa_ml, 1, NULL);
}
return 0;
}
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
wanted.maxlength = -1;
wanted.tlength = -1;
wanted.prebuf = 0;
wanted.minreq = -1;
wanted.fragsize = spec->segsize;
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d", wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", wanted.prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize);
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
PA_STREAM_INTERPOLATE_TIMING |
PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONOUS |
#ifdef HAVE_PULSE_0_9_11
PA_STREAM_ADJUST_LATENCY |
#endif
PA_STREAM_START_CORKED) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pulsesrc->corked = TRUE;
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pulsesrc->stream);
if (!PA_STREAM_IS_GOOD (state)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pulsesrc->stream);
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d (wanted: %d)",
actual->tlength, wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", actual->prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d (wanted %d)", actual->minreq,
wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d (wanted %d)",
actual->fragsize, wanted.fragsize);
if (actual->fragsize >= wanted.fragsize) {
spec->segsize = actual->fragsize;
} else {
spec->segsize = actual->fragsize * (wanted.fragsize / actual->fragsize);
}
spec->segtotal = actual->maxlength / spec->segsize;
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
unlock_and_fail:
{
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
}
pa_simple* pa_simple_new(
const char *server,
const char *name,
pa_stream_direction_t dir,
const char *dev,
const char *stream_name,
const pa_sample_spec *ss,
const pa_channel_map *map,
const pa_buffer_attr *attr,
int *rerror) {
pa_simple *p;
int error = PA_ERR_INTERNAL, r;
CHECK_VALIDITY_RETURN_ANY(rerror, !server || *server, PA_ERR_INVALID, NULL);
CHECK_VALIDITY_RETURN_ANY(rerror, dir == PA_STREAM_PLAYBACK || dir == PA_STREAM_RECORD, PA_ERR_INVALID, NULL);
CHECK_VALIDITY_RETURN_ANY(rerror, !dev || *dev, PA_ERR_INVALID, NULL);
CHECK_VALIDITY_RETURN_ANY(rerror, ss && pa_sample_spec_valid(ss), PA_ERR_INVALID, NULL);
CHECK_VALIDITY_RETURN_ANY(rerror, !map || (pa_channel_map_valid(map) && map->channels == ss->channels), PA_ERR_INVALID, NULL)
p = pa_xnew0(pa_simple, 1);
p->direction = dir;
if (!(p->mainloop = pa_threaded_mainloop_new()))
goto fail;
if (!(p->context = pa_context_new(pa_threaded_mainloop_get_api(p->mainloop), name)))
goto fail;
pa_context_set_state_callback(p->context, context_state_cb, p);
if (pa_context_connect(p->context, server, 0, NULL) < 0) {
error = pa_context_errno(p->context);
goto fail;
}
pa_threaded_mainloop_lock(p->mainloop);
if (pa_threaded_mainloop_start(p->mainloop) < 0)
goto unlock_and_fail;
for (;;) {
pa_context_state_t state;
state = pa_context_get_state(p->context);
if (state == PA_CONTEXT_READY)
break;
if (!PA_CONTEXT_IS_GOOD(state)) {
error = pa_context_errno(p->context);
goto unlock_and_fail;
}
/* Wait until the context is ready */
pa_threaded_mainloop_wait(p->mainloop);
}
if (!(p->stream = pa_stream_new(p->context, stream_name, ss, map))) {
error = pa_context_errno(p->context);
goto unlock_and_fail;
}
pa_stream_set_state_callback(p->stream, stream_state_cb, p);
pa_stream_set_read_callback(p->stream, stream_request_cb, p);
pa_stream_set_write_callback(p->stream, stream_request_cb, p);
pa_stream_set_latency_update_callback(p->stream, stream_latency_update_cb, p);
if (dir == PA_STREAM_PLAYBACK)
r = pa_stream_connect_playback(p->stream, dev, attr,
PA_STREAM_INTERPOLATE_TIMING
|PA_STREAM_ADJUST_LATENCY
|PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
else
r = pa_stream_connect_record(p->stream, dev, attr,
PA_STREAM_INTERPOLATE_TIMING
|PA_STREAM_ADJUST_LATENCY
|PA_STREAM_AUTO_TIMING_UPDATE);
if (r < 0) {
error = pa_context_errno(p->context);
goto unlock_and_fail;
}
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state(p->stream);
if (state == PA_STREAM_READY)
break;
if (!PA_STREAM_IS_GOOD(state)) {
error = pa_context_errno(p->context);
goto unlock_and_fail;
}
/* Wait until the stream is ready */
pa_threaded_mainloop_wait(p->mainloop);
}
pa_threaded_mainloop_unlock(p->mainloop);
return p;
unlock_and_fail:
pa_threaded_mainloop_unlock(p->mainloop);
fail:
if (rerror)
*rerror = error;
pa_simple_free(p);
return NULL;
}
bool QPulseAudioInput::open()
{
if (m_opened)
return true;
QPulseAudioEngine *pulseEngine = QPulseAudioEngine::instance();
if (!pulseEngine->context() || pa_context_get_state(pulseEngine->context()) != PA_CONTEXT_READY) {
setError(QAudio::FatalError);
setState(QAudio::StoppedState);
return false;
}
pa_sample_spec spec = QPulseAudioInternal::audioFormatToSampleSpec(m_format);
if (!pa_sample_spec_valid(&spec)) {
setError(QAudio::OpenError);
setState(QAudio::StoppedState);
return false;
}
m_spec = spec;
#ifdef DEBUG_PULSE
// QTime now(QTime::currentTime());
// qDebug()<<now.second()<<"s "<<now.msec()<<"ms :open()";
#endif
if (m_streamName.isNull())
m_streamName = QString(QLatin1String("QtmPulseStream-%1-%2")).arg(::getpid()).arg(quintptr(this)).toUtf8();
#ifdef DEBUG_PULSE
qDebug() << "Format: " << QPulseAudioInternal::sampleFormatToQString(spec.format);
qDebug() << "Rate: " << spec.rate;
qDebug() << "Channels: " << spec.channels;
qDebug() << "Frame size: " << pa_frame_size(&spec);
#endif
pulseEngine->lock();
pa_channel_map channel_map;
pa_channel_map_init_extend(&channel_map, spec.channels, PA_CHANNEL_MAP_DEFAULT);
if (!pa_channel_map_compatible(&channel_map, &spec))
qWarning() << "Channel map doesn't match sample specification!";
m_stream = pa_stream_new(pulseEngine->context(), m_streamName.constData(), &spec, &channel_map);
pa_stream_set_state_callback(m_stream, inputStreamStateCallback, this);
pa_stream_set_read_callback(m_stream, inputStreamReadCallback, this);
pa_stream_set_underflow_callback(m_stream, inputStreamUnderflowCallback, this);
pa_stream_set_overflow_callback(m_stream, inputStreamOverflowCallback, this);
m_periodSize = pa_usec_to_bytes(PeriodTimeMs*1000, &spec);
int flags = 0;
pa_buffer_attr buffer_attr;
buffer_attr.maxlength = (uint32_t) -1;
buffer_attr.prebuf = (uint32_t) -1;
buffer_attr.tlength = (uint32_t) -1;
buffer_attr.minreq = (uint32_t) -1;
flags |= PA_STREAM_ADJUST_LATENCY;
if (m_bufferSize > 0)
buffer_attr.fragsize = (uint32_t) m_bufferSize;
else
buffer_attr.fragsize = (uint32_t) m_periodSize;
if (pa_stream_connect_record(m_stream, m_device.data(), &buffer_attr, (pa_stream_flags_t)flags) < 0) {
qWarning() << "pa_stream_connect_record() failed!";
pa_stream_unref(m_stream);
m_stream = 0;
pulseEngine->unlock();
setError(QAudio::OpenError);
setState(QAudio::StoppedState);
return false;
}
while (pa_stream_get_state(m_stream) != PA_STREAM_READY)
pa_threaded_mainloop_wait(pulseEngine->mainloop());
const pa_buffer_attr *actualBufferAttr = pa_stream_get_buffer_attr(m_stream);
m_periodSize = actualBufferAttr->fragsize;
m_periodTime = pa_bytes_to_usec(m_periodSize, &spec) / 1000;
if (actualBufferAttr->tlength != (uint32_t)-1)
m_bufferSize = actualBufferAttr->tlength;
pulseEngine->unlock();
connect(pulseEngine, &QPulseAudioEngine::contextFailed, this, &QPulseAudioInput::onPulseContextFailed);
m_opened = true;
m_timer->start(m_periodTime);
m_clockStamp.restart();
m_timeStamp.restart();
m_elapsedTimeOffset = 0;
m_totalTimeValue = 0;
return true;
}
AudioStream::AudioStream(pa_context *c,
pa_threaded_mainloop *m,
const char *desc,
int type,
unsigned samplrate,
const PaDeviceInfos* infos,
bool ec)
: audiostream_(0), mainloop_(m)
{
const pa_channel_map channel_map = infos->channel_map;
pa_sample_spec sample_spec = {
PA_SAMPLE_S16LE, // PA_SAMPLE_FLOAT32LE,
samplrate,
channel_map.channels
};
RING_DBG("%s: trying to create stream with device %s (%dHz, %d channels)", desc, infos->name.c_str(), samplrate, channel_map.channels);
assert(pa_sample_spec_valid(&sample_spec));
assert(pa_channel_map_valid(&channel_map));
std::unique_ptr<pa_proplist, decltype(pa_proplist_free)&> pl (pa_proplist_new(), pa_proplist_free);
pa_proplist_sets(pl.get(), PA_PROP_FILTER_WANT, "echo-cancel");
audiostream_ = pa_stream_new_with_proplist(c, desc, &sample_spec, &channel_map, ec ? pl.get() : nullptr);
if (!audiostream_) {
RING_ERR("%s: pa_stream_new() failed : %s" , desc, pa_strerror(pa_context_errno(c)));
throw std::runtime_error("Could not create stream\n");
}
pa_buffer_attr attributes;
attributes.maxlength = pa_usec_to_bytes(160 * PA_USEC_PER_MSEC, &sample_spec);
attributes.tlength = pa_usec_to_bytes(80 * PA_USEC_PER_MSEC, &sample_spec);
attributes.prebuf = 0;
attributes.fragsize = pa_usec_to_bytes(80 * PA_USEC_PER_MSEC, &sample_spec);
attributes.minreq = (uint32_t) -1;
{
PulseMainLoopLock lock(mainloop_);
const pa_stream_flags_t flags = static_cast<pa_stream_flags_t>(PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
if (type == PLAYBACK_STREAM || type == RINGTONE_STREAM) {
pa_stream_connect_playback(audiostream_,
infos->name.empty() ? NULL : infos->name.c_str(),
&attributes,
flags,
NULL, NULL);
} else if (type == CAPTURE_STREAM) {
pa_stream_connect_record(audiostream_,
infos->name.empty() ? NULL : infos->name.c_str(),
&attributes,
flags);
}
}
pa_stream_set_state_callback(audiostream_, [](pa_stream* s, void* user_data){
static_cast<AudioStream*>(user_data)->stateChanged(s);
}, this);
pa_stream_set_moved_callback(audiostream_, [](pa_stream* s, void* user_data){
static_cast<AudioStream*>(user_data)->moved(s);
}, this);
}
static int drvHostPulseAudioOpen(bool fIn, const char *pszName,
pa_sample_spec *pSampleSpec, pa_buffer_attr *pBufAttr,
pa_stream **ppStream)
{
AssertPtrReturn(pszName, VERR_INVALID_POINTER);
AssertPtrReturn(pSampleSpec, VERR_INVALID_POINTER);
AssertPtrReturn(pBufAttr, VERR_INVALID_POINTER);
AssertPtrReturn(ppStream, VERR_INVALID_POINTER);
if (!pa_sample_spec_valid(pSampleSpec))
{
LogRel(("PulseAudio: Unsupported sample specification for stream \"%s\"\n",
pszName));
return VERR_NOT_SUPPORTED;
}
int rc = VINF_SUCCESS;
pa_stream *pStream = NULL;
uint32_t flags = PA_STREAM_NOFLAGS;
LogFunc(("Opening \"%s\", rate=%dHz, channels=%d, format=%s\n",
pszName, pSampleSpec->rate, pSampleSpec->channels,
pa_sample_format_to_string(pSampleSpec->format)));
pa_threaded_mainloop_lock(g_pMainLoop);
do
{
if (!(pStream = pa_stream_new(g_pContext, pszName, pSampleSpec,
NULL /* pa_channel_map */)))
{
LogRel(("PulseAudio: Could not create stream \"%s\"\n", pszName));
rc = VERR_NO_MEMORY;
break;
}
pa_stream_set_state_callback(pStream, drvHostPulseAudioCbStreamState, NULL);
#if PA_API_VERSION >= 12
/* XXX */
flags |= PA_STREAM_ADJUST_LATENCY;
#endif
#if 0
/* Not applicable as we don't use pa_stream_get_latency() and pa_stream_get_time(). */
flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE;
#endif
/* No input/output right away after the stream was started. */
flags |= PA_STREAM_START_CORKED;
if (fIn)
{
LogFunc(("Input stream attributes: maxlength=%d fragsize=%d\n",
pBufAttr->maxlength, pBufAttr->fragsize));
if (pa_stream_connect_record(pStream, /*dev=*/NULL, pBufAttr, (pa_stream_flags_t)flags) < 0)
{
LogRel(("PulseAudio: Could not connect input stream \"%s\": %s\n",
pszName, pa_strerror(pa_context_errno(g_pContext))));
rc = VERR_AUDIO_BACKEND_INIT_FAILED;
break;
}
}
else
{
LogFunc(("Output buffer attributes: maxlength=%d tlength=%d prebuf=%d minreq=%d\n",
pBufAttr->maxlength, pBufAttr->tlength, pBufAttr->prebuf, pBufAttr->minreq));
if (pa_stream_connect_playback(pStream, /*dev=*/NULL, pBufAttr, (pa_stream_flags_t)flags,
/*cvolume=*/NULL, /*sync_stream=*/NULL) < 0)
{
LogRel(("PulseAudio: Could not connect playback stream \"%s\": %s\n",
pszName, pa_strerror(pa_context_errno(g_pContext))));
rc = VERR_AUDIO_BACKEND_INIT_FAILED;
break;
}
}
/* Wait until the stream is ready. */
for (;;)
{
if (!g_fAbortMainLoop)
pa_threaded_mainloop_wait(g_pMainLoop);
g_fAbortMainLoop = false;
pa_stream_state_t sstate = pa_stream_get_state(pStream);
if (sstate == PA_STREAM_READY)
break;
else if ( sstate == PA_STREAM_FAILED
|| sstate == PA_STREAM_TERMINATED)
{
LogRel(("PulseAudio: Failed to initialize stream \"%s\" (state %ld)\n",
pszName, sstate));
rc = VERR_AUDIO_BACKEND_INIT_FAILED;
break;
}
}
if (RT_FAILURE(rc))
break;
const pa_buffer_attr *pBufAttrObtained = pa_stream_get_buffer_attr(pStream);
AssertPtr(pBufAttrObtained);
memcpy(pBufAttr, pBufAttrObtained, sizeof(pa_buffer_attr));
if (fIn)
LogFunc(("Obtained record buffer attributes: maxlength=%RU32, fragsize=%RU32\n",
pBufAttr->maxlength, pBufAttr->fragsize));
else
LogFunc(("Obtained playback buffer attributes: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d\n",
pBufAttr->maxlength, pBufAttr->tlength, pBufAttr->prebuf, pBufAttr->minreq));
}
while (0);
if ( RT_FAILURE(rc)
&& pStream)
pa_stream_disconnect(pStream);
pa_threaded_mainloop_unlock(g_pMainLoop);
if (RT_FAILURE(rc))
{
if (pStream)
pa_stream_unref(pStream);
}
else
*ppStream = pStream;
LogFlowFuncLeaveRC(rc);
return rc;
}
