本文整理汇总了C++中pa_stream_new函数的典型用法代码示例。如果您正苦于以下问题：C++ pa_stream_new函数的具体用法？C++ pa_stream_new怎么用？C++ pa_stream_new使用的例子？那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了pa_stream_new函数的20个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于我们的系统推荐出更棒的C++代码示例。
示例1: pulse_output_setup_stream
/**
* Create, set up and connect a context.
*
* Caller must lock the main loop.
*
* @return true on success, false on error
*/
static bool
pulse_output_setup_stream(struct pulse_output *po, const pa_sample_spec *ss,
GError **error_r)
{
assert(po != NULL);
assert(po->context != NULL);
po->stream = pa_stream_new(po->context, po->name, ss, NULL);
if (po->stream == NULL) {
g_set_error(error_r, pulse_output_quark(), 0,
"pa_stream_new() has failed: %s",
pa_strerror(pa_context_errno(po->context)));
return false;
}
#if PA_CHECK_VERSION(0,9,8)
pa_stream_set_suspended_callback(po->stream,
pulse_output_stream_suspended_cb, po);
#endif
pa_stream_set_state_callback(po->stream,
pulse_output_stream_state_cb, po);
pa_stream_set_write_callback(po->stream,
pulse_output_stream_write_cb, po);
return true;
}
开发者ID:Acidburn0zzz，项目名称:mpd，代码行数:34，代码来源:pulse_output_plugin.c
示例2: pulse_write_preprocess
static void pulse_write_preprocess(MSFilter *f){
PulseWriteState *s=(PulseWriteState*)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;
s->fragsize=latency_req*(float)s->channels*(float)s->rate*2;
attr.maxlength=-1;
attr.tlength=s->fragsize;
attr.prebuf=-1;
attr.minreq=-1;
attr.fragsize=-1;
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_playback(s->stream,NULL,&attr, PA_STREAM_ADJUST_LATENCY,NULL,NULL);
pa_threaded_mainloop_unlock(pa_loop);
if (err!=0){
ms_error("pa_stream_connect_playback() failed");
}
}
开发者ID:LaughingAngus，项目名称:linphone-vs2008，代码行数:32，代码来源:pulseaudio.c
示例3: pa_context_get_state
void PulseAudioDriver::ctx_state_callback(pa_context* ctx, void* udata)
{
PulseAudioDriver* self = (PulseAudioDriver*)udata;
pa_context_state s = pa_context_get_state(ctx);
if (s == PA_CONTEXT_READY)
{
pa_sample_spec spec;
spec.format = PA_SAMPLE_S16LE;
spec.rate = self->m_sample_rate;
spec.channels = 2;
self->m_stream = pa_stream_new(ctx, "Hydrogen", &spec, 0);
pa_stream_set_state_callback(self->m_stream, stream_state_callback, self);
pa_stream_set_write_callback(self->m_stream, stream_write_callback, self);
pa_buffer_attr bufattr;
bufattr.fragsize = (uint32_t)-1;
bufattr.maxlength = self->m_buffer_size * 4;
bufattr.minreq = 0;
bufattr.prebuf = (uint32_t)-1;
bufattr.tlength = self->m_buffer_size * 4;
pa_stream_connect_playback(self->m_stream, 0, &bufattr, pa_stream_flags_t(0), 0, 0);
}
else if (s == PA_CONTEXT_FAILED)
pa_mainloop_quit(self->m_main_loop, 1);
}
开发者ID:AdamFf，项目名称:hydrogen，代码行数:25，代码来源:pulse_audio_driver.cpp
示例4: context_state_callback
/**
* 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);
}
开发者ID:muggenhor，项目名称:GNUnet，代码行数:62，代码来源:gnunet-helper-audio-record.c
示例5: pa_threaded_mainloop_lock
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;
}
开发者ID:juanquintela，项目名称:qemu，代码行数:59，代码来源:paaudio.c
示例6: pa_stream_new
void WavegenClient::onConnectionEstablishedFirstTime() {
// create stream:
pa_sample_spec ss;
ss.channels = 1;
ss.format = PA_SAMPLE_S16LE;
ss.rate = SAMPLE_RATE;
paStream_ = pa_stream_new(paContext_, name_.c_str(), &ss, nullptr);
pa_stream_connect_playback(paStream_, nullptr, nullptr, pa_stream_flags::PA_STREAM_NOFLAGS, nullptr, nullptr);
}
开发者ID:bog2k3，项目名称:soundspace，代码行数:9，代码来源:WavegenClient.cpp
示例7: eventd_sound_pulseaudio_play_data
void
eventd_sound_pulseaudio_play_data(EventdSoundPulseaudioContext *context, gpointer data, gsize length, gint format, guint32 rate, guint8 channels)
{
pa_sample_spec sample_spec;
pa_stream *stream;
EventdSoundPulseaudioEventData *event_data;
if ( data == NULL )
return;
if ( ( context == NULL ) || ( pa_context_get_state(context->context) != PA_CONTEXT_READY ) )
{
g_free(data);
return;
}
switch ( format )
{
case SF_FORMAT_PCM_16:
case SF_FORMAT_PCM_U8:
case SF_FORMAT_PCM_S8:
sample_spec.format = PA_SAMPLE_S16NE;
break;
case SF_FORMAT_PCM_24:
sample_spec.format = PA_SAMPLE_S24NE;
break;
case SF_FORMAT_PCM_32:
sample_spec.format = PA_SAMPLE_S32NE;
break;
case SF_FORMAT_FLOAT:
case SF_FORMAT_DOUBLE:
sample_spec.format = PA_SAMPLE_FLOAT32NE;
break;
default:
g_warning("Unsupported format");
return;
}
sample_spec.rate = rate;
sample_spec.channels = channels;
if ( ! pa_sample_spec_valid(&sample_spec) )
{
g_warning("Invalid spec");
return;
}
stream = pa_stream_new(context->context, "sndfile plugin playback", &sample_spec, NULL);
event_data = g_new0(EventdSoundPulseaudioEventData, 1);
event_data->data = data;
event_data->length = length;
pa_stream_set_state_callback(stream, _eventd_sound_pulseaudio_stream_state_callback, event_data);
pa_stream_connect_playback(stream, NULL, NULL, 0, NULL, NULL);
}
开发者ID:worr，项目名称:eventd，代码行数:56，代码来源:pulseaudio.c
示例8: pa_mainloop_new
bool PulseAudio::init(bool)
{
pa_ml = pa_mainloop_new();
pa_mainloop_api* pa_mlapi = pa_mainloop_get_api(pa_ml);
pa_context* pa_ctx = pa_context_new(pa_mlapi, "MuseScore");
if (pa_context_connect(pa_ctx, NULL, pa_context_flags_t(0), NULL) != 0)
qDebug("PulseAudio Context Connect Failed with Error: %s", pa_strerror(pa_context_errno(pa_ctx)));
int pa_ready = 0;
pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);
while (pa_ready == 0)
pa_mainloop_iterate(pa_ml, 1, NULL);
if (pa_ready == 2)
return false;
ss.rate = _sampleRate;
ss.channels = 2;
ss.format = PA_SAMPLE_FLOAT32LE;
pa_stream* playstream = pa_stream_new(pa_ctx, "Playback", &ss, NULL);
if (!playstream) {
qDebug("pa_stream_new failed");
return false;
}
pa_stream_set_write_callback(playstream, paCallback, this);
bufattr.fragsize = (uint32_t)-1;
bufattr.maxlength = FRAMES * 2 * sizeof(float);
bufattr.minreq = FRAMES * 1 * sizeof(float); // pa_usec_to_bytes(0, &ss);
bufattr.prebuf = (uint32_t)-1;
bufattr.tlength = bufattr.maxlength;
int r = pa_stream_connect_playback(playstream, NULL, &bufattr,
pa_stream_flags_t(PA_STREAM_INTERPOLATE_TIMING
| PA_STREAM_ADJUST_LATENCY
| PA_STREAM_AUTO_TIMING_UPDATE),
NULL, NULL);
if (r < 0) {
// Old pulse audio servers don't like the ADJUST_LATENCY flag, so retry without that
r = pa_stream_connect_playback(playstream, NULL, &bufattr,
pa_stream_flags_t(PA_STREAM_INTERPOLATE_TIMING
| PA_STREAM_AUTO_TIMING_UPDATE),
NULL, NULL);
}
if (r < 0) {
qDebug("pa_stream_connect_playback failed");
pa_context_disconnect(pa_ctx);
pa_context_unref(pa_ctx);
pa_mainloop_free(pa_ml);
pa_ml = 0;
return false;
}
return true;
}
开发者ID:AdrianShe，项目名称:MuseScore，代码行数:55，代码来源:pulseaudio.cpp
示例9: qDebug
void QPulseAudioThread::reconnect(SourceContainer::const_iterator pos = s_sourceList.end())
{
if (s_sourceList.empty())
return;
if (pos != s_sourceList.end()) {
s_sourcePosition = pos;
qDebug() << "reconnecting with" << *pos;
} else
s_sourcePosition = scanForPlaybackMonitor();
if (s_sourcePosition == s_sourceList.end()) {
s_sourcePosition = s_sourceList.begin();
}
if (stream && (pa_stream_get_state(stream) == PA_STREAM_READY)) {
//qDebug() << "disconnect";
pa_stream_disconnect ( stream );
// pa_stream_unref(stream);
//qDebug() << "* return *";
}
if ( ! ( stream = pa_stream_new ( context, stream_name, &sample_spec, channel_map_set ? &channel_map : NULL ) ) ) {
fprintf ( stderr, "pa_stream_new() failed: %s\n", pa_strerror ( pa_context_errno ( context ) ) );
return;
}
pa_stream_set_state_callback
( stream, stream_state_callback, &s_sourceList );
pa_stream_set_read_callback ( stream, stream_read_callback, &s_sourceList );
pa_stream_set_moved_callback(stream, stream_moved_callback, &s_sourceList );
switch (pa_stream_get_state(stream)) {
case PA_STREAM_UNCONNECTED:// The stream is not yet connected to any sink or source.
qDebug() << "unconnected: connecting...";
connectHelper(s_sourcePosition);
break;
case PA_STREAM_CREATING ://The stream is being created.
break;
case PA_STREAM_READY :// The stream is established, you may pass audio data to it now.
qDebug() << "stream is still ready, waiting for callback...";
break;
case PA_STREAM_FAILED :// An error occured that made the stream invalid.
qDebug() << "stream is now invalid. great.";
break;
case PA_STREAM_TERMINATED:// The stream has been terminated cleanly.
qDebug() << "terminated...";
break;
}
}
开发者ID:flair2005，项目名称:scribble，代码行数:53，代码来源:QPulseAudioThread.cpp
示例10: context_state_callback
/*
* Context state callbacks
*
* A 'context' represents the connection handle between a PulseAudio
* client and its server. It multiplexes everything in that connection
* including data streams , bi-directional commands, and events.
*/
static void context_state_callback(pa_context *context, void *userdata) {
struct context *ctx = userdata;
struct audio_file *file;
pa_stream *stream;
int ret;
assert(ctx);
assert((file = ctx->file));
switch (pa_context_get_state(context)) {
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_SETTING_NAME:
break;
case PA_CONTEXT_READY:
out("Connection established with PulseAudio sound server");
for (int i = 0; i < 256; i++) {
stream = pa_stream_new(context, "playback stream", &file->spec, NULL);
if (!stream)
goto fail;
pa_stream_set_state_callback(stream, stream_state_callback, userdata);
pa_stream_set_write_callback(stream, stream_write_callback, userdata);
/* Connect this stream with a sink chosen by PulseAudio */
ret = pa_stream_connect_playback(stream, NULL, NULL, 0, NULL, NULL);
if (ret < 0) {
error("pa_stream_connect_playback() failed: %s",
pa_strerror(pa_context_errno(context)));
goto fail;
}
}
break;
case PA_CONTEXT_TERMINATED:
exit(EXIT_SUCCESS);
break;
case PA_CONTEXT_FAILED:
default:
error("PulseAudio context connection failure: %s",
pa_strerror(pa_context_errno(context)));
goto fail;
}
return;
fail:
quit(ctx, EXIT_FAILURE);
}
开发者ID:a-darwish，项目名称:malicious-pulseaudio-clients，代码行数:60，代码来源:kill_server_quickly_open_write_streams.c
示例11: pa_mainloop_new
bool PulseAudio::PulseInit()
{
m_pa_error = 0;
m_pa_connected = 0;
// create pulseaudio main loop and context
// also register the async state callback which is called when the connection to the pa server has changed
m_pa_ml = pa_mainloop_new();
m_pa_mlapi = pa_mainloop_get_api(m_pa_ml);
m_pa_ctx = pa_context_new(m_pa_mlapi, "dolphin-emu");
m_pa_error = pa_context_connect(m_pa_ctx, nullptr, PA_CONTEXT_NOFLAGS, nullptr);
pa_context_set_state_callback(m_pa_ctx, StateCallback, this);
// wait until we're connected to the pulseaudio server
while (m_pa_connected == 0 && m_pa_error >= 0)
m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
if (m_pa_connected == 2 || m_pa_error < 0)
{
ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
return false;
}
// create a new audio stream with our sample format
// also connect the callbacks for this stream
pa_sample_spec ss;
ss.format = PA_SAMPLE_S16LE;
ss.channels = 2;
ss.rate = m_mixer->GetSampleRate();
m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, nullptr);
pa_stream_set_write_callback(m_pa_s, WriteCallback, this);
pa_stream_set_underflow_callback(m_pa_s, UnderflowCallback, this);
// connect this audio stream to the default audio playback
// limit buffersize to reduce latency
m_pa_ba.fragsize = -1;
m_pa_ba.maxlength = -1; // max buffer, so also max latency
m_pa_ba.minreq = -1; // don't read every byte, try to group them _a bit_
m_pa_ba.prebuf = -1; // start as early as possible
m_pa_ba.tlength = BUFFER_SIZE; // designed latency, only change this flag for low latency output
pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
m_pa_error = pa_stream_connect_playback(m_pa_s, nullptr, &m_pa_ba, flags, nullptr, nullptr);
if (m_pa_error < 0)
{
ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
return false;
}
INFO_LOG(AUDIO, "Pulse successfully initialized");
return true;
}
开发者ID:comex，项目名称:Dolphin-work，代码行数:51，代码来源:PulseAudioStream.cpp
示例12: instream_open_pa
static int instream_open_pa(struct SoundIoPrivate *si, struct SoundIoInStreamPrivate *is) {
struct SoundIoInStreamPulseAudio *ispa = &is->backend_data.pulseaudio;
struct SoundIoInStream *instream = &is->pub;
if ((unsigned)instream->layout.channel_count > PA_CHANNELS_MAX)
return SoundIoErrorIncompatibleBackend;
if (!instream->name)
instream->name = "SoundIoInStream";
struct SoundIoPulseAudio *sipa = &si->backend_data.pulseaudio;
SOUNDIO_ATOMIC_STORE(ispa->stream_ready, false);
pa_threaded_mainloop_lock(sipa->main_loop);
pa_sample_spec sample_spec;
sample_spec.format = to_pulseaudio_format(instream->format);
sample_spec.rate = instream->sample_rate;
sample_spec.channels = instream->layout.channel_count;
pa_channel_map channel_map = to_pulseaudio_channel_map(&instream->layout);
ispa->stream = pa_stream_new(sipa->pulse_context, instream->name, &sample_spec, &channel_map);
if (!ispa->stream) {
pa_threaded_mainloop_unlock(sipa->main_loop);
instream_destroy_pa(si, is);
return SoundIoErrorNoMem;
}
pa_stream *stream = ispa->stream;
pa_stream_set_state_callback(stream, recording_stream_state_callback, is);
pa_stream_set_read_callback(stream, recording_stream_read_callback, is);
ispa->buffer_attr.maxlength = UINT32_MAX;
ispa->buffer_attr.tlength = UINT32_MAX;
ispa->buffer_attr.prebuf = 0;
ispa->buffer_attr.minreq = UINT32_MAX;
ispa->buffer_attr.fragsize = UINT32_MAX;
if (instream->software_latency > 0.0) {
int bytes_per_second = instream->bytes_per_frame * instream->sample_rate;
int buffer_length = instream->bytes_per_frame *
ceil_dbl_to_int(instream->software_latency * bytes_per_second / (double)instream->bytes_per_frame);
ispa->buffer_attr.fragsize = buffer_length;
}
pa_threaded_mainloop_unlock(sipa->main_loop);
return 0;
}
开发者ID:IceDragon200，项目名称:libsoundio，代码行数:50，代码来源:pulseaudio.c
示例13: pa_stream_new
static pa_stream *connect_playback_stream(ALCdevice *device,
pa_stream_flags_t flags, pa_buffer_attr *attr, pa_sample_spec *spec,
pa_channel_map *chanmap)
{
pulse_data *data = device->ExtraData;
pa_stream_state_t state;
pa_stream *stream;
stream = pa_stream_new(data->context, "Playback Stream", spec, chanmap);
if(!stream)
{
ERR("pa_stream_new() failed: %s\n",
pa_strerror(pa_context_errno(data->context)));
return NULL;
}
pa_stream_set_state_callback(stream, stream_state_callback, data->loop);
AL_PRINT("Attempting flags 0x%x\n", flags);
if (attr)
{
AL_PRINT("maxlength: %d tlegth: %d prebuf: %d minreq: %d fragsize: %d\n",
attr->maxlength, attr->tlength, attr->prebuf, attr->minreq, attr->fragsize);
}
if(pa_stream_connect_playback(stream, data->device_name, attr, flags, NULL, NULL) < 0)
{
ERR("Stream did not connect: %s\n",
pa_strerror(pa_context_errno(data->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(data->context)));
pa_stream_unref(stream);
return NULL;
}
pa_threaded_mainloop_wait(data->loop);
}
pa_stream_set_state_callback(stream, NULL, NULL);
return stream;
}
开发者ID:siana，项目名称:2p-openal，代码行数:49，代码来源:pulseaudio.c
示例14: pa_mainloop_new
bool PulseAudio::init(bool)
{
pa_ml = pa_mainloop_new();
pa_mainloop_api* pa_mlapi = pa_mainloop_get_api(pa_ml);
pa_context* pa_ctx = pa_context_new(pa_mlapi, "MuseScore");
if (pa_context_connect(pa_ctx, NULL, pa_context_flags_t(0), NULL) != 0) {
qDebug("PulseAudio Context Connect Failed with Error: %s", pa_strerror(pa_context_errno(pa_ctx)));
return false;
}
int pa_ready = 0;
pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);
while (pa_ready == 0)
pa_mainloop_iterate(pa_ml, 1, NULL);
if (pa_ready == 2)
return false;
ss.rate = _sampleRate;
ss.channels = 2;
ss.format = PA_SAMPLE_FLOAT32LE;
pa_stream* playstream = pa_stream_new(pa_ctx, "Playback", &ss, NULL);
if (!playstream) {
qDebug("pa_stream_new failed: %s", pa_strerror(pa_context_errno(pa_ctx)));
return false;
}
pa_stream_set_write_callback(playstream, paCallback, this);
bufattr.fragsize = (uint32_t)-1;
bufattr.maxlength = FRAMES * 2 * sizeof(float);
bufattr.minreq = FRAMES * 1 * sizeof(float); // pa_usec_to_bytes(0, &ss);
bufattr.prebuf = (uint32_t)-1;
bufattr.tlength = bufattr.maxlength;
int r = pa_stream_connect_playback(playstream, nullptr, &bufattr,
PA_STREAM_NOFLAGS, nullptr, nullptr);
if (r < 0) {
qDebug("pa_stream_connect_playback failed");
pa_context_disconnect(pa_ctx);
pa_context_unref(pa_ctx);
pa_mainloop_free(pa_ml);
pa_ml = 0;
return false;
}
return true;
}
开发者ID:CammyVee，项目名称:MuseScore，代码行数:48，代码来源:pulseaudio.cpp
示例15: preload_sample
static bool preload_sample(struct audio_service *service, struct play_feedback_data *pfd)
{
bool result = false;
struct stat st;
pa_sample_spec spec;
char *sample_path;
if (!pfd || !pfd->name)
return false;
if (g_slist_find(sample_list, pfd->name)) {
play_feedback_sample(pfd);
play_feedback_data_free(pfd);
return true;
}
sample_path = g_strdup_printf("%s/%s.pcm", SAMPLE_PATH, pfd->name);
if (stat(sample_path, &st) != 0)
goto cleanup;
pfd->sample_length = st.st_size;
spec.format = PA_SAMPLE_S16LE;
spec.rate = 44100;
spec.channels = 1;
pfd->fd = open(sample_path, O_RDONLY);
if (pfd->fd < 0)
goto cleanup;
pfd->sample_stream = pa_stream_new(service->context, pfd->name, &spec, NULL);
if (!pfd->sample_stream)
goto cleanup;
pa_stream_set_state_callback(pfd->sample_stream, preload_stream_state_cb, pfd);
pa_stream_set_write_callback(pfd->sample_stream, preload_stream_write_cb, pfd);
pa_stream_connect_upload(pfd->sample_stream, pfd->sample_length);
result = true;
cleanup:
g_free(sample_path);
return result;
}
开发者ID:nizovn，项目名称:audio-service，代码行数:46，代码来源:audio_service.c
示例16: audiostream_
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);
}
开发者ID:dyfet，项目名称:sflphone，代码行数:44，代码来源:audiostream.cpp
示例17: pa_stream_new
static pa_stream *connect_playback_stream(ALCdevice *device,
pa_stream_flags_t flags, pa_buffer_attr *attr, pa_sample_spec *spec,
pa_channel_map *chanmap)
{
pulse_data *data = device->ExtraData;
pa_stream_state_t state;
pa_stream *stream;
stream = pa_stream_new(data->context, "Playback Stream", spec, chanmap);
if(!stream)
{
ERR("pa_stream_new() failed: %s\n",
pa_strerror(pa_context_errno(data->context)));
return NULL;
}
pa_stream_set_state_callback(stream, stream_state_callback, data->loop);
if(pa_stream_connect_playback(stream, data->device_name, attr, flags, NULL, NULL) < 0)
{
ERR("Stream did not connect: %s\n",
pa_strerror(pa_context_errno(data->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(data->context)));
pa_stream_unref(stream);
return NULL;
}
pa_threaded_mainloop_wait(data->loop);
}
pa_stream_set_state_callback(stream, NULL, NULL);
return stream;
}
开发者ID:chairbender，项目名称:OpenAL3dAudioForAndroid，代码行数:42，代码来源:pulseaudio.c
示例18: sa_stream_open
int
sa_stream_open(sa_stream_t *s) {
if (s == NULL) {
return SA_ERROR_NO_INIT;
}
if (s->stream != NULL) {
return SA_ERROR_INVALID;
}
pa_threaded_mainloop_lock(s->m);
if (!(s->stream = pa_stream_new(s->context, s->client_name, &s->sample_spec, NULL))) {
fprintf(stderr, "pa_stream_new() failed: %s\n", pa_strerror(pa_context_errno(s->context)));
goto unlock_and_fail;
}
pa_stream_set_state_callback(s->stream, stream_state_callback, s);
pa_stream_set_write_callback(s->stream, stream_write_callback, s);
pa_stream_set_latency_update_callback(s->stream, stream_latency_update_callback, s);
if (pa_stream_connect_playback(s->stream, NULL, NULL, 0, NULL, NULL) < 0) {
fprintf(stderr, "pa_stream_connect_playback() failed: %s\n", pa_strerror(pa_context_errno(s->context)));
goto unlock_and_fail;
}
/* Wait until the stream is ready */
pa_threaded_mainloop_wait(s->m);
if (pa_stream_get_state(s->stream) != PA_STREAM_READY) {
fprintf(stderr, "Failed to connect stream: %s", pa_strerror(pa_context_errno(s->context)));
goto unlock_and_fail;
}
pa_threaded_mainloop_unlock(s->m);
if (!s->stream)
return SA_ERROR_NO_DEVICE;
return SA_SUCCESS;
unlock_and_fail:
pa_threaded_mainloop_unlock(s->m);
return SA_ERROR_NO_DEVICE;
}
开发者ID:MozillaOnline，项目名称:gecko-dev，代码行数:41，代码来源:sydney_audio_pulseaudio.c
示例19: context_state_callback
/* This is called whenever the context status changes */
static void context_state_callback(pa_context *c, void *userdata) {
fail_unless(c != NULL);
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 i;
fprintf(stderr, "Connection established.\n");
for (i = 0; i < NSTREAMS; i++) {
char name[64];
fprintf(stderr, "Creating stream %i\n", i);
snprintf(name, sizeof(name), "stream #%i", i);
streams[i] = pa_stream_new(c, name, &sample_spec, NULL);
fail_unless(streams[i] != NULL);
pa_stream_set_state_callback(streams[i], stream_state_callback, (void*) (long) i);
pa_stream_connect_playback(streams[i], NULL, &buffer_attr, PA_STREAM_START_CORKED, NULL, i == 0 ? NULL : streams[0]);
}
break;
}
case PA_CONTEXT_TERMINATED:
mainloop_api->quit(mainloop_api, 0);
break;
case PA_CONTEXT_FAILED:
default:
fprintf(stderr, "Context error: %s\n", pa_strerror(pa_context_errno(c)));
fail();
}
}
开发者ID:Distrotech，项目名称:pulseaudio，代码行数:41，代码来源:sync-playback.c
示例20: audio_init
void audio_init() {
pa_threaded_mainloop *pa_ml=pa_threaded_mainloop_new();
pa_mainloop_api *pa_mlapi=pa_threaded_mainloop_get_api(pa_ml);
pa_context *pa_ctx=pa_context_new(pa_mlapi, "te");
pa_context_connect(pa_ctx, NULL, 0, NULL);
int pa_ready = 0;
pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);
pa_threaded_mainloop_start(pa_ml);
while(pa_ready==0) { ; }
printf("audio ready\n");
if (pa_ready == 2) {
pa_context_disconnect(pa_ctx);
pa_context_unref(pa_ctx);
pa_threaded_mainloop_free(pa_ml);
}
pa_sample_spec ss;
ss.rate=96000;
ss.channels=1;
ss.format=PA_SAMPLE_S24_32LE;
ps=pa_stream_new(pa_ctx,"Playback",&ss,NULL);
pa_stream_set_write_callback(ps,audio_request_cb,NULL);
pa_stream_set_underflow_callback(ps,audio_underflow_cb,NULL);
pa_buffer_attr bufattr;
bufattr.fragsize = (uint32_t)-1;
bufattr.maxlength = pa_usec_to_bytes(20000,&ss);
bufattr.minreq = pa_usec_to_bytes(0,&ss);
bufattr.prebuf = 0;
bufattr.tlength = pa_usec_to_bytes(20000,&ss);
pa_stream_connect_playback(ps,NULL,&bufattr,
PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_ADJUST_LATENCY|PA_STREAM_AUTO_TIMING_UPDATE|PA_STREAM_START_CORKED,NULL,NULL);
}
开发者ID:ITikhonov，项目名称:tem，代码行数:37，代码来源:te.c
注：本文中的pa_stream_new函数示例由纯净天空整理自Github/MSDocs等源码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。 |
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