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C++ pa_stream_new函数代码示例

原作者: [db:作者] 来自: [db:来源] 收藏 邀请

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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C++ pa_stream_set_state_callback函数代码示例发布时间：2022-05-30
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