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C++ pa_stream_connect_playback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_connect_playback函数的典型用法代码示例。如果您正苦于以下问题：C++ pa_stream_connect_playback函数的具体用法？C++ pa_stream_connect_playback怎么用？C++ pa_stream_connect_playback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_connect_playback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: 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2: 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3: pa_stream_connect_playback

JNIEXPORT jint
JNICALL Java_org_jitsi_impl_neomedia_pulseaudio_PA_stream_1connect_1playback
    (JNIEnv *env, jclass clazz, jlong s, jstring dev, jlong attr, jint flags,
        jlong volume, jlong syncStream)
{
    const char *devChars
        = dev ? (*env)->GetStringUTFChars(env, dev, NULL) : NULL;
    jint ret;

    if ((*env)->ExceptionCheck(env))
        ret = -1;
    else
    {
        ret
            = pa_stream_connect_playback(
                    (pa_stream *) (intptr_t) s,
                    devChars,
                    (const pa_buffer_attr *) (intptr_t) attr,
                    (pa_stream_flags_t) flags,
                    (const pa_cvolume *) (intptr_t) volume,
                    (pa_stream *) (intptr_t) syncStream);
        (*env)->ReleaseStringUTFChars(env, dev, devChars);
    }
    return ret;
}
开发者ID:DroidInteractiveSoftware，项目名称:libjitsi，代码行数:25，代码来源:org_jitsi_impl_neomedia_pulseaudio_PA.c


示例4: 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: 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9: 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10: 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11: 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];
                pa_format_info *formats[1];

                formats[0] = pa_format_info_new();
                formats[0]->encoding = PA_ENCODING_PCM;
                pa_format_info_set_sample_format(formats[0], PA_SAMPLE_FLOAT32);
                pa_format_info_set_rate(formats[0], SAMPLE_HZ);
                pa_format_info_set_channels(formats[0], 1);

                fprintf(stderr, "Creating stream %i\n", i);

                snprintf(name, sizeof(name), "stream #%i", i);

                streams[i] = pa_stream_new_extended(c, name, formats, 1, 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]);

                pa_format_info_free(formats[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:DryakhlyyZlodey，项目名称:pulseaudio，代码行数:50，代码来源:extended-test.c


示例12: 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13: 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14: 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15: 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16: 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17: pa_stream_new_with_proplist

static pa_stream *connect_playback_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, "Playback 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_playback(stream, device_name, attr, flags, NULL, NULL) < 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;
}
开发者ID:24BitGames，项目名称:LoomSDK，代码行数:39，代码来源:pulseaudio.c


示例18: 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


示例19: state_cb

// This callback gets called when our context changes state.  We really only
// care about when it's ready or if it has failed
void state_cb(pa_context *c, void *userdata) {
  pa_context_state_t state;
  int *pa_ready = userdata;

  printf("State changed\n");
  state = pa_context_get_state(c);
  switch  (state) {
    // There are just here for reference
  case PA_CONTEXT_UNCONNECTED:
  case PA_CONTEXT_CONNECTING:
  case PA_CONTEXT_AUTHORIZING:
  case PA_CONTEXT_SETTING_NAME:
  default:
    break;
  case PA_CONTEXT_FAILED:
  case PA_CONTEXT_TERMINATED:
    *pa_ready = 2;
    break;
  case PA_CONTEXT_READY: {
    pa_buffer_attr buffer_attr;

    if (verbose)
      printf("Connection established.%s\n", CLEAR_LINE);

    if (!(stream = pa_stream_new(c, "JanPlayback", &sample_spec, NULL))) {
      printf("pa_stream_new() failed: %s", pa_strerror(pa_context_errno(c)));
      exit(1); // 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;
    
    pa_cvolume cv;
      



    if (pa_stream_connect_playback(stream, NULL, &buffer_attr, flags,
				   NULL, 
				   NULL) < 0) {
      printf("pa_stream_connect_playback() failed: %s", pa_strerror(pa_context_errno(c)));
      exit(1); //goto fail;
    } else {
      printf("Set playback callback\n");
    }

    pa_stream_trigger(stream, stream_success, NULL);
  }

    break;
  }
}
开发者ID:wyongfei，项目名称:LinuxSound-HTMLSource，代码行数:73，代码来源:pacat2.c


示例20: tsmf_pulse_open_stream

static int
tsmf_pulse_open_stream(TSMFPulseAudioDevice * pulse)
{
	pa_stream_state_t state;
	pa_buffer_attr buffer_attr = { 0 };

	if (!pulse->context)
		return 1;

	LLOGLN(0, ("tsmf_pulse_open_stream:"));
	pa_threaded_mainloop_lock(pulse->mainloop);
	pulse->stream = pa_stream_new(pulse->context, "freerdp",
		&pulse->sample_spec, NULL);
	if (!pulse->stream)
	{
		pa_threaded_mainloop_unlock(pulse->mainloop);
		LLOGLN(0, ("tsmf_pulse_open_stream: pa_stream_new failed (%d)",
			pa_context_errno(pulse->context)));
		return 1;
	}
	pa_stream_set_state_callback(pulse->stream,
		tsmf_pulse_stream_state_callback, pulse);
	pa_stream_set_write_callback(pulse->stream,
		tsmf_pulse_stream_request_callback, pulse);
	buffer_attr.maxlength = pa_usec_to_bytes(500000, &pulse->sample_spec);
	buffer_attr.tlength = pa_usec_to_bytes(250000, &pulse->sample_spec);
	buffer_attr.prebuf = (uint32_t) -1;
	buffer_attr.minreq = (uint32_t) -1;
	buffer_attr.fragsize = (uint32_t) -1;
	if (pa_stream_connect_playback(pulse->stream,
		pulse->device[0] ? pulse->device : NULL, &buffer_attr,
		PA_STREAM_ADJUST_LATENCY | PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE,
		NULL, NULL) < 0)
	{
		pa_threaded_mainloop_unlock(pulse->mainloop);
		LLOGLN(0, ("tsmf_pulse_open_stream: pa_stream_connect_playback failed (%d)",
			pa_context_errno(pulse->context)));
		return 1;
	}

	for (;;)
	{
		state = pa_stream_get_state(pulse->stream);
		if (state == PA_STREAM_READY)
			break;
        if (!PA_STREAM_IS_GOOD(state))
		{
			LLOGLN(0, ("tsmf_pulse_open_stream: 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)
	{
		LLOGLN(0, ("tsmf_pulse_open_stream: connected"));
		return 0;
	}
	else
	{
		tsmf_pulse_close_stream(pulse);
		return 1;
	}
}
开发者ID:FreeRDP，项目名称:FreeRDP-old，代码行数:65，代码来源:tsmf_pulse.c



注：本文中的pa_stream_connect_playback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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