Esempio n. 1
0
static PyObject *simple_disconnect(simple *self) {
    if (self->s)
        pa_simple_free(self->s);
    self->s = NULL;

    return Py_BuildValue("i", 0);
}
Esempio n. 2
0
static void pulse_thread(void *context)
{
#ifdef __linux__
    prctl(PR_SET_NAME, "deadbeef-pulse", 0, 0, 0, 0);
#endif

    trace ("pulse thread started \n");
    while (!pulse_terminate)
    {
        if (state != OUTPUT_STATE_PLAYING || !deadbeef->streamer_ok_to_read (-1))
        {
            usleep(10000);
            continue;
        }

        int sample_size = plugin.fmt.channels * (plugin.fmt.bps / 8);
        char buf[buffer_size];

        in_callback = 1;
        int bytesread = deadbeef->streamer_read(buf, buffer_size);
        in_callback = 0;
        if (pulse_terminate) {
            break;
        }
        if (bytesread < 0) {
            bytesread = 0;
        }

        int error;

        int res = 0;
        if (bytesread > 0) {
            deadbeef->mutex_lock (mutex);
            res = pa_simple_write(s, buf, bytesread, &error);
            deadbeef->mutex_unlock(mutex);
        }

        if (pulse_terminate) {
            break;
        }

        if (res < 0)
        {
            usleep(10000);
        }
    }

    deadbeef->mutex_lock (mutex);
    state = OUTPUT_STATE_STOPPED;
    if (s)
    {
        pa_simple_drain (s, NULL);
        pa_simple_free(s);
        s = NULL;
    }
    pulse_terminate = 0;
    pulse_tid = 0;
    deadbeef->mutex_unlock (mutex);
    trace ("pulse_thread finished\n");
}
Esempio n. 3
0
int main(int argc, char*argv[]) {
    /* The sample type to use */
    static const pa_sample_spec ss = {
        .format = PA_SAMPLE_S16LE,
        .rate = 44100,
        .channels = 2
    };
    pa_simple *s = NULL;
    int ret = 1;
    int error;
    /* Create the recording stream */
    if (!(s = pa_simple_new(NULL, argv[0], PA_STREAM_RECORD, NULL, "record", &ss, NULL, NULL, &error))) {
        fprintf(stderr, __FILE__": pa_simple_new() failed: %s\n", pa_strerror(error));
        goto finish;
    }
    for (;;) {
        uint8_t buf[BUFSIZE];
        /* Record some data ... */
        if (pa_simple_read(s, buf, sizeof(buf), &error) < 0) {
            fprintf(stderr, __FILE__": pa_simple_read() failed: %s\n", pa_strerror(error));
            goto finish;
        }
        /* And write it to STDOUT */
        if (loop_write(STDOUT_FILENO, buf, sizeof(buf)) != sizeof(buf)) {
            fprintf(stderr, __FILE__": write() failed: %s\n", strerror(errno));
            goto finish;
        }
    }
    ret = 0;
finish:
    if (s)
        pa_simple_free(s);
    return ret;
}
Esempio n. 4
0
static void pulseaudio_deallocate_recorder(ALLEGRO_AUDIO_RECORDER *r)
{
   PULSEAUDIO_RECORDER *pa = (PULSEAUDIO_RECORDER *) r->extra;
   
   pa_simple_free(pa->s);
   al_free(r->extra);
}
Esempio n. 5
0
static void* pulse_thread(void* context)
{
	sem_t* init = (sem_t*)context;

	pa_sample_spec ss;
	ss.format = PA_SAMPLE_S16LE;
	ss.channels = 2;
	ss.rate = g_sample_rate;

	int err;
	g_pulse = pa_simple_new(NULL, g_appname, PA_STREAM_PLAYBACK, NULL, g_appname, &ss, NULL, NULL, &err);
	if (!g_pulse) {
		snprintf(g_lasterror, c_nlasterror, "failed to connect to pulse server: %d", err);
	}

	sem_post(init);
	if (!g_pulse)
		return 0;

	pa_simple* s = g_pulse;
	short samples[c_nsamples*2];

	g_running = true;
	while (g_running) {
		g_callback(samples, c_nsamples);
		if (0 > pa_simple_write(s, samples, sizeof(samples), NULL))
			break;
	}

	pa_simple_flush(s, NULL);
	pa_simple_free(s);
	g_pulse = 0;
	return 0;
}
Esempio n. 6
0
void PulseThread::Close()
{
	if(this->connection) {
		pa_simple_free(this->connection);
		this->connection = nullptr;
	}
}
Esempio n. 7
0
static void myshutdown()
{
	xmp_smix_off();

	if (s)
		pa_simple_free(s);
}
Esempio n. 8
0
void sound_close() {

	switch(sound_type) {
	case SOUND_NO:
	break;
	case SOUND_SDL:
	SDL_CloseAudio();
	break;
#ifdef D_SOUND_OSS
	case SOUND_OSS:
		close(audio_fd);
	break;
#endif
#ifdef D_SOUND_ALSA
	case SOUND_ALSA:
		snd_pcm_drain (_soundDevice);
		snd_pcm_close (_soundDevice);
	break;
#endif
#ifdef D_SOUND_PULSE
	case SOUND_PULSEAUDIO:
		pa_simple_free(pulse_s);
	break;
#endif
#ifdef GEKKO
	case SOUND_ASND:
		ASND_End();
	break;
#endif

	default:
	break;
	}
}
RTC::ReturnCode_t PulseAudioOutput::onFinalize()
{
  RTC_DEBUG(("onFinalize start"));
  is_active = false;
  m_mutex.lock();
  RTC_DEBUG(("onFinalize:mutex lock"));
  try {
    if ( !m_data.empty() ) {
      RTC_DEBUG(("onFinalize:queue buffer clear start."));
      m_data.clear();  //!< queue buffer clear
      RTC_DEBUG(("onFinalize:queue buffer clear finish."));
    }
    if ( m_simple ) {
      RTC_DEBUG(("onFinalize:simple connection object free start."));
      pa_simple_free( m_simple );
      RTC_DEBUG(("onFinalize:simple connection object free finish."));
      m_simple = NULL;
    }
  } catch (...) {
    RTC_WARN(("pa_simple_free() failed onDFinalize."));
  }
  m_mutex.unlock();
  RTC_DEBUG(("onFinalize:mutex unlock"));
  RTC_DEBUG(("onFinalize finish"));
  return RTC::RTC_OK;
}
 PulseAudioPCMOutputDriver::~PulseAudioPCMOutputDriver()
 {
     if(m_driver)
     {
         pa_simple_free(m_driver);
     }
 }
bool WaveOutPulseAudio::CloseWaveOutDevice()
{
    bool     tResult = false;
    int      tRes;

    LOG(LOG_VERBOSE, "Going to close..");

    if (mWaveOutOpened)
    {
        StopFilePlayback();
        Stop();

        if (mOutputStream != NULL)
        {
            LOG(LOG_VERBOSE, "..draining stream");
            if (pa_simple_drain(mOutputStream, &tRes) < 0)
            {
                LOG(LOG_ERROR, "Couldn't drain the output stream because %s(%d)", pa_strerror(tRes), tRes);
            }
            LOG(LOG_VERBOSE, "..closing stream");
            pa_simple_free(mOutputStream);
        }

        LOG(LOG_INFO, "...closed");

        mWaveOutOpened = false;
        tResult = true;
    }else
        LOG(LOG_INFO, "...wasn't open");

     ResetPacketStatistic();

    return tResult;
}
Esempio n. 12
0
void	pulseaudio_close_audio(void) {
	if (!pulseaudio_audiodev_is_open) return;
	if (pa_simple_drain(s, &error) < 0) {
		fprintf(stderr, __FILE__": pa_simple_drain() failed: %s\n", pa_strerror(error));
	}
	pa_simple_free(s);
	pulseaudio_audiodev_is_open = FALSE;
}
Esempio n. 13
0
void _closeRecord()
{
    if(!sr)
        return;

    pa_simple_free(sr);
    sr = NULL;
}
Esempio n. 14
0
void _closePlay()
{
    if(!sp)
        return;

    pa_simple_free(sp);
    sp = NULL;
}
bool WaveOutPulseAudio::OpenWaveOutDevice(int pSampleRate, int pOutputChannels)
{
    pa_sample_spec     tOutputFormat;
    int                tRes;
    pa_usec_t          tLatency;

    LOG(LOG_VERBOSE, "Trying to open the wave out device");

    if (mWaveOutOpened)
        return false;

    mSampleRate = pSampleRate;
    mAudioChannels = pOutputChannels;

    LOG(LOG_VERBOSE, "Desired device is %s", mDesiredDevice.c_str());

    if ((mDesiredDevice == "") || (mDesiredDevice == "auto") || (mDesiredDevice == "automatic"))
    {
        LOG(LOG_VERBOSE, "Using default audio device");
        mDesiredDevice = "";
    }

    tOutputFormat.format = PA_SAMPLE_S16LE;
    tOutputFormat.rate = mSampleRate;
    tOutputFormat.channels = mAudioChannels;

    // create a new playback stream
    if (!(mOutputStream = pa_simple_new(NULL, "Homer-Conferencing", PA_STREAM_PLAYBACK, (mDesiredDevice != "" ? mDesiredDevice.c_str() : NULL) /* dev Name */, GetStreamName().c_str(), &tOutputFormat, NULL, NULL, &tRes)))
    {
        LOG(LOG_ERROR, "Couldn't create PulseAudio stream because %s(%d)", pa_strerror(tRes), tRes);
        return false;
    }

    if ((tLatency = pa_simple_get_latency(mOutputStream, &tRes)) == (pa_usec_t) -1)
    {
        LOG(LOG_ERROR, "Couldn't determine the latency of the output stream because %s(%d)", pa_strerror(tRes), tRes);
        pa_simple_free(mOutputStream);
        mOutputStream = NULL;
        return false;
    }

    mCurrentDevice = mDesiredDevice;

    //######################################################
    //### give some verbose output
    //######################################################
    LOG(LOG_INFO, "PortAudio wave out opened...");
    LOG(LOG_INFO,"    ..sample rate: %d", mSampleRate);
    LOG(LOG_INFO,"    ..channels: %d", pOutputChannels);
    LOG(LOG_INFO,"    ..desired device: %s", mDesiredDevice.c_str());
    LOG(LOG_INFO,"    ..selected device: %s", mCurrentDevice.c_str());
    LOG(LOG_INFO,"    ..latency: %"PRIu64" seconds", (uint64_t)tLatency * 1000 * 1000);
    LOG(LOG_INFO,"    ..sample format: %d", PA_SAMPLE_S16LE);

    mWaveOutOpened = true;

    return true;
}
Esempio n. 16
0
void mpgplayer::run()
{
    if(mpg123_init() != MPG123_OK)
        qDebug("Error initilizing mpg123");
    const char **test = mpg123_supported_decoders();
    int error;
    mpg123_handle *mh = mpg123_new(test[0],&error);
    if(!mpg123_feature(MPG123_FEATURE_DECODE_LAYER3))
    {
        qDebug("You do not seem to have mp3 decoding support");
        return;
    }
    mpg123_format_none(mh);
    if(mpg123_format(mh,samplerate,MPG123_STEREO,MPG123_ENC_SIGNED_16)!=MPG123_OK)
        qDebug("Error in initilizing format decoder");
    qDebug(test[0]);
    mpg123_open(mh,"/home/eli/Projects/groove-evan/Animal.mp3");
    net = TData;
    pa_simple *s;
    pa_sample_spec ss;
    ss.format = PA_SAMPLE_S16NE;
    ss.rate = samplerate;
    ss.channels = 2;
    s =pa_simple_new(NULL,"Groove",PA_STREAM_PLAYBACK ,NULL,"Music",&ss,NULL,NULL,NULL);

    unsigned char bytes[1024];
    size_t bsize = 1024;
    size_t done = 0;
    bool stop = false;
    playing=true;
    while(!stop)
    {
        switch(net)
        {
        case TWait: usleep(100); break;
        case TData:
            if(mpg123_read(mh,bytes,bsize,&done)==MPG123_DONE)
            {
                net=TFinish;
            }
            pa_simple_write(s,bytes,done,&error);
            break;
        case TAbort:
            stop = true;
            break;
        case TFinish:
            pa_simple_drain(s,&error);
            stop = true;
            break;
        default: break;
        }
    }
    qDebug("Finsihed playback");
    pa_simple_free(s);

    mpg123_exit();
}
vis::PulseAudioSource::~PulseAudioSource()
{
#ifdef _ENABLE_PULSE
    if (m_pulseaudio_simple != nullptr)
    {
        pa_simple_free(m_pulseaudio_simple);
    }
#endif
}
Esempio n. 18
0
boolean
adin_close()
{
	if(r) {
		pa_simple_free(r);
		r = NULL;
	}
	return TRUE;
}
Esempio n. 19
0
void OutputPulseAudio::uninitialize()
{
    if (m_connection)
    {
        qDebug("OutputPulseAudio: closing connection");
        pa_simple_free(m_connection);
        m_connection = 0;
    }
}
Esempio n. 20
0
static void pulsedrv_close(void)
{
    int error = 0;
    if (pa_simple_flush(s, &error)) {
        log_error(LOG_DEFAULT, "pa_simple_flush(): %s", pa_strerror(error));
        /* don't stop */
    }
    pa_simple_free(s);
    s = NULL;
}
uint8_t  pulseSimpleAudioDevice::stop(void) 
{
int er;
    ADM_assert(instance);
    pa_simple_flush(INSTANCE,&er);
    pa_simple_free(INSTANCE);
    instance=NULL;
    printf("[PulseAudio] Stopped\n");
    return 1;
}
Esempio n. 22
0
void
ManglerPulse::close(bool drain) {/*{{{*/
   if (pulse_stream) {
        if (drain && pa_simple_drain(pulse_stream, &pulse_error) < 0) {
            fprintf(stderr, "pulse: pa_simple_drain() failed: %s\n", pa_strerror(pulse_error));
        }
        pa_simple_free(pulse_stream);
        pulse_stream = NULL;
    }
}/*}}}*/
Esempio n. 23
0
int main(int argc, char*argv[]) {
	/* The sample type to use */
	static const pa_sample_spec ss = {
		.format = PA_SAMPLE_S16LE,
		.rate = 44100,
		.channels = 2
	};
	pa_simple *inStream = NULL;
	pa_simple *outStream = NULL;

	int ret = 1;
	int error;
	/* Create the recording stream */
	if (!(inStream = pa_simple_new(NULL, argv[0], PA_STREAM_RECORD, NULL, "record", &ss, NULL, NULL, &error))) {
		fprintf(stderr, __FILE__": pa_simple_new() failed: %s\n", pa_strerror(error));
		goto finish;
	}
	/* Create the playback stream */
	if (!(outStream = pa_simple_new(NULL, argv[0], PA_STREAM_PLAYBACK, NULL, "playback", &ss, NULL, NULL, &error))) {
		fprintf(stderr, __FILE__": pa_simple_new() failed: %s\n", pa_strerror(error));
		goto finish;
	}
	for (;;) {
		uint8_t buf[BUFSIZE];
		/* Record some data ... */
		if (pa_simple_read(inStream, buf, sizeof(buf), &error) < 0) {
			fprintf(stderr, __FILE__": pa_simple_read() failed: %s\n", pa_strerror(error));
			goto finish;
		}
		if (pa_simple_write(outStream, buf, sizeof(buf), &error) < 0) {
			fprintf(stderr, __FILE__": pa_simple_write() failed: %s\n", pa_strerror(error));
			goto finish;
		}
	}
	ret = 0;
	finish:
	if (inStream)
		pa_simple_free(inStream);
	if (outStream)
		pa_simple_free(outStream);
	return ret;
}
Esempio n. 24
0
File: paaudio.c Progetto: CPFL/gxen
static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
{
    int error;
    static pa_sample_spec ss;
    struct audsettings obt_as = *as;
    PAVoiceIn *pa = (PAVoiceIn *) hw;

    ss.format = audfmt_to_pa (as->fmt, as->endianness);
    ss.channels = as->nchannels;
    ss.rate = as->freq;

    obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);

    pa->s = pa_simple_new (
        conf.server,
        "qemu",
        PA_STREAM_RECORD,
        conf.source,
        "pcm.capture",
        &ss,
        NULL,                   /* channel map */
        NULL,                   /* buffering attributes */
        &error
        );
    if (!pa->s) {
        qpa_logerr (error, "pa_simple_new for capture failed\n");
        goto fail1;
    }

    audio_pcm_init_info (&hw->info, &obt_as);
    hw->samples = conf.samples;
    pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
    if (!pa->pcm_buf) {
        dolog ("Could not allocate buffer (%d bytes)\n",
               hw->samples << hw->info.shift);
        goto fail2;
    }

    if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) {
        goto fail3;
    }

    return 0;

 fail3:
    free (pa->pcm_buf);
    pa->pcm_buf = NULL;
 fail2:
    pa_simple_free (pa->s);
    pa->s = NULL;
 fail1:
    return -1;
}
Esempio n. 25
0
UINT8 Pulse_Start(void* drvObj, UINT32 deviceID, AUDIO_OPTS* options, void* audDrvParam)
{
	DRV_PULSE* drv = (DRV_PULSE*)drvObj;
	UINT64 tempInt64;
	UINT8 retVal8;
	
	if (drv->devState != 0)
		return 0xD0;	// already running
	
	drv->audDrvPtr = audDrvParam;
	if (options == NULL)
		options = &defOptions;
	drv->pulseFmt.channels = options->numChannels;
	drv->pulseFmt.rate = options->sampleRate;
	
	tempInt64 = (UINT64)options->sampleRate * options->usecPerBuf;
	drv->bufSmpls = (UINT32)((tempInt64 + 500000) / 1000000);
	drv->bufSize = (options->numBitsPerSmpl * drv->pulseFmt.channels / 8) * drv->bufSmpls;
	drv->bufCount = options->numBuffers ? options->numBuffers : 10;
	
	if (options->numBitsPerSmpl == 8)
		drv->pulseFmt.format = PA_SAMPLE_U8;
	else if (options->numBitsPerSmpl == 16)
		drv->pulseFmt.format = PA_SAMPLE_S16NE;
	else if (options->numBitsPerSmpl == 24)
		drv->pulseFmt.format = PA_SAMPLE_S24NE;
	else if (options->numBitsPerSmpl == 32)
		drv->pulseFmt.format = PA_SAMPLE_S32NE;
	else
		return 0xCF;
	
	drv->canPause = 1;
	
	drv->hPulse = pa_simple_new(NULL, "libvgm", PA_STREAM_PLAYBACK, NULL, drv->streamDesc, &drv->pulseFmt, NULL, NULL, NULL);
	if(!drv->hPulse)
		return 0xC0;
	
	OSSignal_Reset(drv->hSignal);
	retVal8 = OSThread_Init(&drv->hThread, &PulseThread, drv);
	if (retVal8)
	{
		pa_simple_free(drv->hPulse);
		return 0xC8;	// CreateThread failed
	}
	
	drv->bufSpace = (UINT8*)malloc(drv->bufSize);
	
	drv->devState = 1;
	drv->pauseThread = 0x00;
	OSSignal_Signal(drv->hSignal);
	
	return AERR_OK;
}
Esempio n. 26
0
// Close an opened audio device, free any allocated buffers and
// reset any variables that reflect the current state.
void Audio_Pulse::close ()
{
    if (_audioHandle != NULL) {
        pa_simple_free(_audioHandle);
        _audioHandle = NULL;
    }

    if (_sampleBuffer != NULL) {
        delete [] _sampleBuffer;
        outOfOrder ();
    }
}
Esempio n. 27
0
static void
pulse_close (sw_handle * handle)
{
  struct pa_simple * pa ;

  pulse_drain(handle);

  if ((pa = (struct pa_simple *)handle->custom_data) == NULL)
    return;

  pa_simple_free(pa);
  handle->custom_data = NULL;
}
Esempio n. 28
0
void m1sdr_PlayStop(void)
{
	#if 0
	pa_operation *op;
	#endif

	#ifdef USE_SDL
	if (lnxdrv_apimode == 0) 
	{
		SDL_PauseAudio(1);
	}
	#endif

	if (lnxdrv_apimode == 1)
	{
//		snd_pcm_pause(pHandle, 1);
		snd_pcm_drop(pHandle);
	}

	#if PULSE_USE_SIMPLE
	if ((lnxdrv_apimode == 3) && (my_simple))
	{
		pa_simple_flush(my_simple, NULL);
		pa_simple_free(my_simple);
		my_simple = NULL;
	}
	#else
#if 0
	if (lnxdrv_apimode == 3)
	{
		op = pa_stream_drain(my_pa_stream, &pa_stream_drain_complete, NULL);
		if (op)
		{
			while (pa_operation_get_state(op) != PA_OPERATION_DONE)
			{
				if (pa_context_get_state(my_pa_context) != PA_CONTEXT_READY ||
				    pa_stream_get_state(my_pa_stream) != PA_STREAM_READY ||
				    pa_mainloop_iterate(my_pa_mainloop, 0, NULL) < 0)
				    {
				    	pa_operation_cancel(op);
					break;
				    }
			}
		}
	}
#endif
	#endif

	waveLogStop();
	oss_playing = 0;
}
Esempio n. 29
0
int audio_init(context_t *context) {
  int rc = 0;
  static const pa_sample_spec ss = {
    .format = PA_SAMPLE_S16LE,
    .rate = 44100,
    .channels = 1
  };
  int error;

  daemon_log(LOG_INFO,
             "Opening %s for input",
             context->input_device ?: "default source");

  context->pa_input = pa_simple_new(NULL,
                              "shusherd",
                              PA_STREAM_RECORD,
                              context->input_device,
                              "record",
                              &ss,
                              NULL,
                              NULL,
                              &error);
  if (!context->pa_input) {
    daemon_log(LOG_ERR, "pa_simple_new failed: %s", pa_strerror(error));
    assert(context->pa_input);
  }

  context->ebur128_state = ebur128_init(ss.channels, ss.rate, EBUR128_MODE_S);
  assert(context->ebur128_state);

  context->enable_processing = 1;

  rc = pthread_create(&context->audio_thread, NULL, audio_loop, (void *)context);
  if (rc) {
    daemon_log(LOG_ERR, "Unable to create audio thread: %d", rc);
    goto cleanup_ebur128_state;
  }

  return 0;

cleanup_ebur128_state:
  ebur128_destroy(&context->ebur128_state);
  return rc;
}

void audio_destroy(context_t *context) {
  context->enable_processing = 0;
  pthread_join(context->audio_thread, NULL);
  ebur128_destroy(&context->ebur128_state);
  pa_simple_free(context->pa_input);
}
Esempio n. 30
0
int main(int argc, char *argv[])
{
    FILE *fp;
    int n;
    int16_t buf[BB], out[BB];
    int error;
    static const pa_sample_spec ss = {
        .format = PA_SAMPLE_S16LE, .rate = 44100, .channels = 1};

    pa_simple *s = NULL;
    if (!(s = pa_simple_new(NULL, argv[0], PA_STREAM_PLAYBACK, NULL, "playback",
                            &ss, NULL, NULL, &error)))
    {
        fprintf(stderr, __FILE__ ": pa_simple_new() failed: %s\n",
                pa_strerror(error));
        goto finish;
    }

    fp = fopen("out.wav", "rb");
    if (!fp)
        exit(-1);

    fread(buf, 1, 44, fp);
    n = fread(buf, 2, BB, fp);
    while (!feof(fp))
    {
        highpass(buf, out, BB, 1.0f, 10);
        memcpy(buf, out, BB * 2);
        lowpass(buf, out, BB, 1.0f / 44100.0f, 0.00002);
        if (pa_simple_write(s, out, (size_t)n * 2, &error) < 0)
        {
            fprintf(stderr, __FILE__ ": pa_simple_write() failed: %s\n",
                    pa_strerror(error));
            goto finish;
        }

        n = fread(buf, 2, BB, fp);
    }

    if (pa_simple_drain(s, &error) < 0)
    {
        fprintf(stderr, __FILE__ ": pa_simple_drain() failed: %s\n",
                pa_strerror(error));
        goto finish;
    }

finish:
    if (s)
        pa_simple_free(s);
    return 0;
}