コード例 #1
0
SDL_Aout *SDL_AoutAndroid_CreateForOpenSLES()
{
    SDLTRACE("%s\n", __func__);
    SDL_Aout *aout = SDL_Aout_CreateInternal(sizeof(SDL_Aout_Opaque));
    if (!aout)
        return NULL;

    SDL_Aout_Opaque *opaque = aout->opaque;
    opaque->wakeup_cond = SDL_CreateCond();
    opaque->wakeup_mutex = SDL_CreateMutex();

    int ret = 0;

    SLObjectItf slObject = NULL;
    ret = slCreateEngine(&slObject, 0, NULL, 0, NULL, NULL);
    CHECK_OPENSL_ERROR(ret, "%s: slCreateEngine() failed", __func__);
    opaque->slObject = slObject;

    ret = (*slObject)->Realize(slObject, SL_BOOLEAN_FALSE);
    CHECK_OPENSL_ERROR(ret, "%s: slObject->Realize() failed", __func__);

    SLEngineItf slEngine = NULL;
    ret = (*slObject)->GetInterface(slObject, SL_IID_ENGINE, &slEngine);
    CHECK_OPENSL_ERROR(ret, "%s: slObject->GetInterface() failed", __func__);
    opaque->slEngine = slEngine;

    SLObjectItf slOutputMixObject = NULL;
    const SLInterfaceID ids1[] = {SL_IID_VOLUME};
    const SLboolean req1[] = {SL_BOOLEAN_FALSE};
    ret = (*slEngine)->CreateOutputMix(slEngine, &slOutputMixObject, 1, ids1, req1);
    CHECK_OPENSL_ERROR(ret, "%s: slEngine->CreateOutputMix() failed", __func__);
    opaque->slOutputMixObject = slOutputMixObject;

    ret = (*slOutputMixObject)->Realize(slOutputMixObject, SL_BOOLEAN_FALSE);
    CHECK_OPENSL_ERROR(ret, "%s: slOutputMixObject->Realize() failed", __func__);

    aout->free_l       = aout_free_l;
    aout->opaque_class = &g_opensles_class;
    aout->open_audio   = aout_open_audio;
    aout->pause_audio  = aout_pause_audio;
    aout->flush_audio  = aout_flush_audio;
    aout->close_audio  = aout_close_audio;
    aout->set_volume   = aout_set_volume;
    aout->func_get_latency_seconds = aout_get_latency_seconds;

    return aout;
fail:
    aout_free_l(aout);
    return NULL;
}
コード例 #2
0
ファイル: ao_opensl.c プロジェクト: reillylen/dttv-android
static int Open(dtaudio_output_t *aout) {
    aout_sys_t *sys;
    SLresult result;

    dtaudio_para_t *para = &aout->para;
    sys = (aout_sys_t *) malloc(sizeof(*sys));
    if (unlikely(sys == NULL))
        return -1;

    sys->p_so_handle = dlopen("libOpenSLES.so", RTLD_NOW);
    if (sys->p_so_handle == NULL) {
        goto error;
    }

    sys->slCreateEnginePtr = dlsym(sys->p_so_handle, "slCreateEngine");
    if (unlikely(sys->slCreateEnginePtr == NULL)) {
        goto error;
    }

#define OPENSL_DLSYM(dest, name)                       \
    do {                                                       \
        const SLInterfaceID *sym = dlsym(sys->p_so_handle, "SL_IID_"name);        \
        if (unlikely(sym == NULL))                             \
        {                                                      \
            goto error;                                        \
        }                                                      \
        sys->dest = *sym;                                           \
    } while(0)

    OPENSL_DLSYM(SL_IID_ANDROIDSIMPLEBUFFERQUEUE, "ANDROIDSIMPLEBUFFERQUEUE");
    OPENSL_DLSYM(SL_IID_ENGINE, "ENGINE");
    OPENSL_DLSYM(SL_IID_PLAY, "PLAY");
    OPENSL_DLSYM(SL_IID_VOLUME, "VOLUME");
#undef OPENSL_DLSYM

    // create engine
    result = sys->slCreateEnginePtr(&sys->engineObject, 0, NULL, 0, NULL, NULL);
    CHECK_OPENSL_ERROR("Failed to create engine");

    // realize the engine in synchronous mode
    result = Realize(sys->engineObject, SL_BOOLEAN_FALSE);
    CHECK_OPENSL_ERROR("Failed to realize engine");

    // get the engine interface, needed to create other objects
    result = GetInterface(sys->engineObject, sys->SL_IID_ENGINE, &sys->engineEngine);
    CHECK_OPENSL_ERROR("Failed to get the engine interface");

    // create output mix, with environmental reverb specified as a non-required interface
    const SLInterfaceID ids1[] = {sys->SL_IID_VOLUME};
    const SLboolean req1[] = {SL_BOOLEAN_FALSE};
    result = CreateOutputMix(sys->engineEngine, &sys->outputMixObject, 1, ids1, req1);
    CHECK_OPENSL_ERROR("Failed to create output mix");

    // realize the output mix in synchronous mode
    result = Realize(sys->outputMixObject, SL_BOOLEAN_FALSE);
    CHECK_OPENSL_ERROR("Failed to realize output mix");

    dt_lock_init(&sys->lock, NULL);

    if (buf_init(&sys->dbt, para->dst_samplerate * 4 / 10) < 0) // 100ms
        return -1;
    aout->ao_priv = (void *) sys;
    return 0;

    error:
    if (sys->outputMixObject)
        Destroy(sys->outputMixObject);
    if (sys->engineObject)
        Destroy(sys->engineObject);
    if (sys->p_so_handle)
        dlclose(sys->p_so_handle);
    free(sys);
    return -1;
}
コード例 #3
0
ファイル: ao_opensl.c プロジェクト: reillylen/dttv-android
static int Start(dtaudio_output_t *aout) {
    SLresult result;

    aout_sys_t *sys = (aout_sys_t *) aout->ao_priv;
    dtaudio_para_t *para = &aout->para;

    // configure audio source - this defines the number of samples you can enqueue.
    SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
            SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
            OPENSLES_BUFFERS
    };

    int mask;

    if (para->dst_channels > 1)
        mask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
    else
        mask = SL_SPEAKER_FRONT_CENTER;


    SLDataFormat_PCM format_pcm;
    format_pcm.formatType = SL_DATAFORMAT_PCM;
    format_pcm.numChannels = para->dst_channels;
    //format_pcm.samplesPerSec    = ((SLuint32) para->dst_samplerate * 1000) ;
    format_pcm.samplesPerSec = ((SLuint32) convertSampleRate(para->dst_samplerate));
    format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
    format_pcm.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
    format_pcm.channelMask = mask;
    format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;

    SLDataSource audioSrc = {&loc_bufq, &format_pcm};

    // configure audio sink
    SLDataLocator_OutputMix loc_outmix = {
            SL_DATALOCATOR_OUTPUTMIX,
            sys->outputMixObject
    };
    SLDataSink audioSnk = {&loc_outmix, NULL};

    //create audio player
    const SLInterfaceID ids2[] = {sys->SL_IID_ANDROIDSIMPLEBUFFERQUEUE, sys->SL_IID_VOLUME};
    static const SLboolean req2[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
    result = CreateAudioPlayer(sys->engineEngine, &sys->playerObject, &audioSrc,
                               &audioSnk, sizeof(ids2) / sizeof(*ids2),
                               ids2, req2);
    if (unlikely(result != SL_RESULT_SUCCESS)) { // error
        return -1;
        /* Try again with a more sensible samplerate */
#if 0
        fmt->i_rate = 44100;
        format_pcm.samplesPerSec = ((SLuint32) 44100 * 1000) ;
        result = CreateAudioPlayer(sys->engineEngine, &sys->playerObject, &audioSrc,
                &audioSnk, sizeof(ids2) / sizeof(*ids2),
                ids2, req2);
#endif
    }
    CHECK_OPENSL_ERROR("Failed to create audio player");

    result = Realize(sys->playerObject, SL_BOOLEAN_FALSE);
    CHECK_OPENSL_ERROR("Failed to realize player object.");

    result = GetInterface(sys->playerObject, sys->SL_IID_PLAY, &sys->playerPlay);
    CHECK_OPENSL_ERROR("Failed to get player interface.");

    result = GetInterface(sys->playerObject, sys->SL_IID_VOLUME, &sys->volumeItf);
    CHECK_OPENSL_ERROR("failed to get volume interface.");

    result = GetInterface(sys->playerObject, sys->SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
                          &sys->playerBufferQueue);
    CHECK_OPENSL_ERROR("Failed to get buff queue interface");

    result = RegisterCallback(sys->playerBufferQueue, PlayedCallback,
                              (void *) aout);
    CHECK_OPENSL_ERROR("Failed to register buff queue callback.");

    // set the player's state to playing
    result = SetPlayState(sys->playerPlay, SL_PLAYSTATE_PLAYING);
    CHECK_OPENSL_ERROR("Failed to switch to playing state");

    /* XXX: rounding shouldn't affect us at normal sampling rate */
    sys->rate = para->dst_samplerate;
    sys->samples_per_buf = OPENSLES_BUFLEN * para->dst_samplerate / 1000;
    sys->buf = malloc(OPENSLES_BUFFERS * sys->samples_per_buf * bytesPerSample(aout));
    if (!sys->buf)
        goto error;

    sys->started = 0;
    sys->next_buf = 0;

    sys->samples = 0;
    SetPositionUpdatePeriod(sys->playerPlay, AOUT_MIN_PREPARE_TIME * 1000 / CLOCK_FREQ);
    return 0;

    error:
    if (sys->playerObject) {
        Destroy(sys->playerObject);
        sys->playerObject = NULL;
    }

    return -1;
}
コード例 #4
0
static int aout_open_audio(SDL_Aout *aout, const SDL_AudioSpec *desired, SDL_AudioSpec *obtained)
{
    SDLTRACE("%s\n", __func__);
    assert(desired);
    SDLTRACE("aout_open_audio()\n");
    SDL_Aout_Opaque  *opaque     = aout->opaque;
    SLEngineItf       slEngine   = opaque->slEngine;
    SLDataFormat_PCM *format_pcm = &opaque->format_pcm;
    int               ret = 0;

    opaque->spec = *desired;

    // config audio src
    SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
        SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
        OPENSLES_BUFFERS
    };

    int native_sample_rate = audiotrack_get_native_output_sample_rate(NULL);
    ALOGI("OpenSL-ES: native sample rate %d Hz\n", native_sample_rate);

    CHECK_COND_ERROR((desired->format == AUDIO_S16SYS), "%s: not AUDIO_S16SYS", __func__);
    CHECK_COND_ERROR((desired->channels == 2 || desired->channels == 1), "%s: not 1,2 channel", __func__);
    CHECK_COND_ERROR((desired->freq >= 8000 && desired->freq <= 48000), "%s: unsupport freq %d Hz", __func__, desired->freq);
    if (SDL_Android_GetApiLevel() < IJK_API_21_LOLLIPOP &&
        native_sample_rate > 0 &&
        desired->freq < native_sample_rate) {
        // Don't try to play back a sample rate higher than the native one,
        // since OpenSL ES will try to use the fast path, which AudioFlinger
        // will reject (fast path can't do resampling), and will end up with
        // too small buffers for the resampling. See http://b.android.com/59453
        // for details. This bug is still present in 4.4. If it is fixed later
        // this workaround could be made conditional.
        //
        // by VLC/android_opensles.c
        ALOGW("OpenSL-ES: force resample %lu to native sample rate %d\n",
              (unsigned long) format_pcm->samplesPerSec / 1000,
              (int) native_sample_rate);
        format_pcm->samplesPerSec = native_sample_rate * 1000;
    }

    format_pcm->formatType       = SL_DATAFORMAT_PCM;
    format_pcm->numChannels      = desired->channels;
    format_pcm->samplesPerSec    = desired->freq * 1000; // milli Hz
    // format_pcm->numChannels      = 2;
    // format_pcm->samplesPerSec    = SL_SAMPLINGRATE_44_1;

    format_pcm->bitsPerSample    = SL_PCMSAMPLEFORMAT_FIXED_16;
    format_pcm->containerSize    = SL_PCMSAMPLEFORMAT_FIXED_16;
    switch (desired->channels) {
    case 2:
        format_pcm->channelMask  = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
        break;
    case 1:
        format_pcm->channelMask  = SL_SPEAKER_FRONT_CENTER;
        break;
    default:
        ALOGE("%s, invalid channel %d", __func__, desired->channels);
        goto fail;
    }
    format_pcm->endianness       = SL_BYTEORDER_LITTLEENDIAN;

    SLDataSource audio_source = {&loc_bufq, format_pcm};

    // config audio sink
    SLDataLocator_OutputMix loc_outmix = {
        SL_DATALOCATOR_OUTPUTMIX,
        opaque->slOutputMixObject
    };
    SLDataSink audio_sink = {&loc_outmix, NULL};

    SLObjectItf slPlayerObject = NULL;
    const SLInterfaceID ids2[] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_VOLUME, SL_IID_PLAY };
    static const SLboolean req2[] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE };
    ret = (*slEngine)->CreateAudioPlayer(slEngine, &slPlayerObject, &audio_source,
                                &audio_sink, sizeof(ids2) / sizeof(*ids2),
                                ids2, req2);
    CHECK_OPENSL_ERROR(ret, "%s: slEngine->CreateAudioPlayer() failed", __func__);
    opaque->slPlayerObject = slPlayerObject;

    ret = (*slPlayerObject)->Realize(slPlayerObject, SL_BOOLEAN_FALSE);
    CHECK_OPENSL_ERROR(ret, "%s: slPlayerObject->Realize() failed", __func__);

    ret = (*slPlayerObject)->GetInterface(slPlayerObject, SL_IID_PLAY, &opaque->slPlayItf);
    CHECK_OPENSL_ERROR(ret, "%s: slPlayerObject->GetInterface(SL_IID_PLAY) failed", __func__);

    ret = (*slPlayerObject)->GetInterface(slPlayerObject, SL_IID_VOLUME, &opaque->slVolumeItf);
    CHECK_OPENSL_ERROR(ret, "%s: slPlayerObject->GetInterface(SL_IID_VOLUME) failed", __func__);

    ret = (*slPlayerObject)->GetInterface(slPlayerObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &opaque->slBufferQueueItf);
    CHECK_OPENSL_ERROR(ret, "%s: slPlayerObject->GetInterface(SL_IID_ANDROIDSIMPLEBUFFERQUEUE) failed", __func__);

    ret = (*opaque->slBufferQueueItf)->RegisterCallback(opaque->slBufferQueueItf, aout_opensles_callback, (void*)aout);
    CHECK_OPENSL_ERROR(ret, "%s: slBufferQueueItf->RegisterCallback() failed", __func__);

    // set the player's state to playing
    // ret = (*opaque->slPlayItf)->SetPlayState(opaque->slPlayItf, SL_PLAYSTATE_PLAYING);
    // CHECK_OPENSL_ERROR(ret, "%s: slBufferQueueItf->slPlayItf() failed", __func__);

    opaque->bytes_per_frame   = format_pcm->numChannels * format_pcm->bitsPerSample / 8;
    opaque->milli_per_buffer  = OPENSLES_BUFLEN;
    opaque->frames_per_buffer = opaque->milli_per_buffer * format_pcm->samplesPerSec / 1000000; // samplesPerSec is in milli
    opaque->bytes_per_buffer  = opaque->bytes_per_frame * opaque->frames_per_buffer;
    opaque->buffer_capacity   = OPENSLES_BUFFERS * opaque->bytes_per_buffer;
    ALOGI("OpenSL-ES: bytes_per_frame  = %d bytes\n",  (int)opaque->bytes_per_frame);
    ALOGI("OpenSL-ES: milli_per_buffer = %d ms\n",     (int)opaque->milli_per_buffer);
    ALOGI("OpenSL-ES: frame_per_buffer = %d frames\n", (int)opaque->frames_per_buffer);
    ALOGI("OpenSL-ES: bytes_per_buffer = %d bytes\n",  (int)opaque->bytes_per_buffer);
    ALOGI("OpenSL-ES: buffer_capacity  = %d bytes\n",  (int)opaque->buffer_capacity);
    opaque->buffer          = malloc(opaque->buffer_capacity);
    CHECK_COND_ERROR(opaque->buffer, "%s: failed to alloc buffer %d\n", __func__, (int)opaque->buffer_capacity);

    // (*opaque->slPlayItf)->SetPositionUpdatePeriod(opaque->slPlayItf, 1000);

    // enqueue empty buffer to start play
    memset(opaque->buffer, 0, opaque->buffer_capacity);
    for(int i = 0; i < OPENSLES_BUFFERS; ++i) {
        ret = (*opaque->slBufferQueueItf)->Enqueue(opaque->slBufferQueueItf, opaque->buffer + i * opaque->bytes_per_buffer, opaque->bytes_per_buffer);
        CHECK_OPENSL_ERROR(ret, "%s: slBufferQueueItf->Enqueue(000...) failed", __func__);
    }

    opaque->pause_on = 1;
    opaque->abort_request = 0;
    opaque->audio_tid = SDL_CreateThreadEx(&opaque->_audio_tid, aout_thread, aout, "ff_aout_opensles");
    CHECK_COND_ERROR(opaque->audio_tid, "%s: failed to SDL_CreateThreadEx", __func__);

    if (obtained) {
        *obtained      = *desired;
        obtained->size = opaque->buffer_capacity;
        obtained->freq = format_pcm->samplesPerSec / 1000;
    }

    return opaque->buffer_capacity;
fail:
    aout_close_audio(aout);
    return -1;
}