Ejemplo n.º 1
0
/***********************************************************************
 * hb_work_encCoreAudio_init
 ***********************************************************************
 *
 **********************************************************************/
int encCoreAudioInit(hb_work_object_t *w, hb_job_t *job, enum AAC_MODE mode)
{
    hb_work_private_t *pv = calloc(1, sizeof(hb_work_private_t));
    hb_audio_t *audio = w->audio;
    AudioStreamBasicDescription input, output;
    UInt32 tmp, tmpsiz = sizeof(tmp);
    OSStatus err;

    w->private_data = pv;
    pv->job = job;

    // pass the number of channels used into the private work data
    pv->nchannels =
        hb_mixdown_get_discrete_channel_count(audio->config.out.mixdown);

    bzero(&input, sizeof(AudioStreamBasicDescription));
    input.mSampleRate = (Float64)audio->config.out.samplerate;
    input.mFormatID = kAudioFormatLinearPCM;
    input.mFormatFlags = (kLinearPCMFormatFlagIsFloat|kAudioFormatFlagsNativeEndian);
    input.mBytesPerPacket = 4 * pv->nchannels;
    input.mFramesPerPacket = 1;
    input.mBytesPerFrame = input.mBytesPerPacket * input.mFramesPerPacket;
    input.mChannelsPerFrame = pv->nchannels;
    input.mBitsPerChannel = 32;

    bzero(&output, sizeof(AudioStreamBasicDescription));
    switch (mode)
    {
        case AAC_MODE_HE:
            output.mFormatID = kAudioFormatMPEG4AAC_HE;
            break;
        case AAC_MODE_LC:
        default:
            output.mFormatID = kAudioFormatMPEG4AAC;
            break;
    }
    output.mSampleRate = (Float64)audio->config.out.samplerate;
    output.mChannelsPerFrame = pv->nchannels;
    // let CoreAudio decide the rest

    // initialise encoder
    err = AudioConverterNew(&input, &output, &pv->converter);
    if (err != noErr)
    {
        // Retry without the samplerate
        bzero(&output, sizeof(AudioStreamBasicDescription));
        switch (mode)
        {
            case AAC_MODE_HE:
                output.mFormatID = kAudioFormatMPEG4AAC_HE;
                break;
            case AAC_MODE_LC:
            default:
                output.mFormatID = kAudioFormatMPEG4AAC;
                break;
        }
        output.mChannelsPerFrame = pv->nchannels;

        err = AudioConverterNew(&input, &output, &pv->converter);

        if (err != noErr)
        {
            hb_log("Error creating an AudioConverter err=%"PRId64" output.mBytesPerFrame=%"PRIu64"",
                   (int64_t)err, (uint64_t)output.mBytesPerFrame);
            *job->done_error = HB_ERROR_UNKNOWN;
            *job->die = 1;
            return -1;
        }
    }

    // set encoder quality to maximum
    tmp = kAudioConverterQuality_Max;
    AudioConverterSetProperty(pv->converter, kAudioConverterCodecQuality,
                              sizeof(tmp), &tmp);

    if (audio->config.out.bitrate > 0)
    {
        // set encoder bitrate control mode to constrained variable
        tmp = kAudioCodecBitRateControlMode_VariableConstrained;
        AudioConverterSetProperty(pv->converter,
                                  kAudioCodecPropertyBitRateControlMode,
                                  sizeof(tmp), &tmp);

        // get available bitrates
        AudioValueRange *bitrates;
        ssize_t bitrateCounts;
        err = AudioConverterGetPropertyInfo(pv->converter,
                                            kAudioConverterApplicableEncodeBitRates,
                                            &tmpsiz, NULL);
        bitrates = malloc(tmpsiz);
        err = AudioConverterGetProperty(pv->converter,
                                        kAudioConverterApplicableEncodeBitRates,
                                        &tmpsiz, bitrates);
        bitrateCounts = tmpsiz / sizeof(AudioValueRange);

        // set bitrate
        tmp = audio->config.out.bitrate * 1000;
        if (tmp < bitrates[0].mMinimum)
            tmp = bitrates[0].mMinimum;
        if (tmp > bitrates[bitrateCounts-1].mMinimum)
            tmp = bitrates[bitrateCounts-1].mMinimum;
        free(bitrates);
        if (tmp != audio->config.out.bitrate * 1000)
        {
            hb_log("encCoreAudioInit: sanitizing track %d audio bitrate %d to %"PRIu32"",
                   audio->config.out.track, audio->config.out.bitrate, tmp / 1000);
        }
        AudioConverterSetProperty(pv->converter,
                                  kAudioConverterEncodeBitRate,
                                  sizeof(tmp), &tmp);
    }
    else if (audio->config.out.quality >= 0)
    {
        if (mode != AAC_MODE_LC)
        {
            hb_error("encCoreAudioInit: internal error, VBR set but not applicable");
            return 1;
        }
        // set encoder bitrate control mode to variable
        tmp = kAudioCodecBitRateControlMode_Variable;
        AudioConverterSetProperty(pv->converter,
                                  kAudioCodecPropertyBitRateControlMode,
                                  sizeof(tmp), &tmp);

        // set quality
        tmp = audio->config.out.quality;
        AudioConverterSetProperty(pv->converter,
                                  kAudioCodecPropertySoundQualityForVBR,
                                  sizeof(tmp), &tmp);
    }
    else
    {
        hb_error("encCoreAudioInit: internal error, bitrate/quality not set");
        return 1;
    }

    // get real input
    tmpsiz = sizeof(input);
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterCurrentInputStreamDescription,
                              &tmpsiz, &input);
    // get real output
    tmpsiz = sizeof(output);
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterCurrentOutputStreamDescription,
                              &tmpsiz, &output);

    // set sizes
    pv->isamplesiz  = input.mBytesPerPacket;
    pv->isamples    = output.mFramesPerPacket;
    pv->osamplerate = output.mSampleRate;
    audio->config.out.samples_per_frame = pv->isamples;

    // channel remapping
    pv->remap = hb_audio_remap_init(AV_SAMPLE_FMT_FLT, &hb_aac_chan_map,
                                    audio->config.in.channel_map);
    if (pv->remap == NULL)
    {
        hb_error("encCoreAudioInit: hb_audio_remap_init() failed");
    }
    uint64_t layout = hb_ff_mixdown_xlat(audio->config.out.mixdown, NULL);
    hb_audio_remap_set_channel_layout(pv->remap, layout);

    // get maximum output size
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterPropertyMaximumOutputPacketSize,
                              &tmpsiz, &tmp);
    pv->omaxpacket = tmp;

    // get magic cookie (elementary stream descriptor)
    tmp = HB_CONFIG_MAX_SIZE;
    AudioConverterGetProperty(pv->converter,
                              kAudioConverterCompressionMagicCookie,
                              &tmp, w->config->extradata.bytes);
    // CoreAudio returns a complete ESDS, but we only need
    // the DecoderSpecific info.
    UInt8* buffer = NULL;
    ReadESDSDescExt(w->config->extradata.bytes, &buffer, &tmpsiz, 0);
    w->config->extradata.length = tmpsiz;
    memmove(w->config->extradata.bytes, buffer, w->config->extradata.length);
    free(buffer);

    pv->list = hb_list_init();
    pv->buf = NULL;

    return 0;
}
Ejemplo n.º 2
0
/***********************************************************************
 * hb_work_encCoreAudio_init
 ***********************************************************************
 *
 **********************************************************************/
int encCoreAudioInit( hb_work_object_t * w, hb_job_t * job, enum AAC_MODE mode )
{
    hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
    hb_audio_t * audio = w->audio;
    AudioStreamBasicDescription input, output;
    UInt32 tmp, tmpsiz = sizeof( tmp );
    OSStatus err;

    w->private_data = pv;
    pv->job = job;

    // pass the number of channels used into the private work data
    pv->nchannels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT( audio->config.out.mixdown );

    bzero( &input, sizeof( AudioStreamBasicDescription ) );
    input.mSampleRate = ( Float64 ) audio->config.out.samplerate;
    input.mFormatID = kAudioFormatLinearPCM;
    input.mFormatFlags = kLinearPCMFormatFlagIsFloat | kAudioFormatFlagsNativeEndian;
    input.mBytesPerPacket = 4 * pv->nchannels;
    input.mFramesPerPacket = 1;
    input.mBytesPerFrame = input.mBytesPerPacket * input.mFramesPerPacket;
    input.mChannelsPerFrame = pv->nchannels;
    input.mBitsPerChannel = 32;

    bzero( &output, sizeof( AudioStreamBasicDescription ) );
    switch ( mode ) 
    {
        case AAC_MODE_HE:
            output.mFormatID = kAudioFormatMPEG4AAC_HE;
            break;
        case AAC_MODE_LC:
        default:
            output.mFormatID = kAudioFormatMPEG4AAC;
            break;
    }
    output.mSampleRate = ( Float64 ) audio->config.out.samplerate;
    output.mChannelsPerFrame = pv->nchannels;
    // let CoreAudio decide the rest...

    // initialise encoder
    err = AudioConverterNew( &input, &output, &pv->converter );
    if( err != noErr)
    {
        // Retry without the samplerate
        bzero( &output, sizeof( AudioStreamBasicDescription ) );
        switch ( mode )
        {
            case AAC_MODE_HE:
                output.mFormatID = kAudioFormatMPEG4AAC_HE;
                break;
            case AAC_MODE_LC:
            default:
                output.mFormatID = kAudioFormatMPEG4AAC;
                break;
        }
        output.mChannelsPerFrame = pv->nchannels;

        err = AudioConverterNew( &input, &output, &pv->converter );

        if( err != noErr)
        {
            hb_log( "Error creating an AudioConverter err=%"PRId64" %"PRIu64, (int64_t)err, (uint64_t)output.mBytesPerFrame );
            *job->die = 1;
            return 0;
        }
    }

    if( ( audio->config.out.mixdown == HB_AMIXDOWN_6CH ) && ( audio->config.in.codec == HB_ACODEC_AC3) )
    {
        SInt32 channelMap[6] = { 2, 1, 3, 4, 5, 0 };
        AudioConverterSetProperty( pv->converter, kAudioConverterChannelMap,
                                   sizeof( channelMap ), channelMap );
    }

    // set encoder quality to maximum
    tmp = kAudioConverterQuality_Max;
    AudioConverterSetProperty( pv->converter, kAudioConverterCodecQuality,
                               sizeof( tmp ), &tmp );

    // set encoder bitrate control mode to constrained variable
    tmp = kAudioCodecBitRateControlMode_VariableConstrained;
    AudioConverterSetProperty( pv->converter, kAudioCodecPropertyBitRateControlMode,
                               sizeof( tmp ), &tmp );

    // get available bitrates
    AudioValueRange *bitrates;
    ssize_t bitrateCounts;
    err = AudioConverterGetPropertyInfo( pv->converter, kAudioConverterApplicableEncodeBitRates,
                                         &tmpsiz, NULL);
    bitrates = malloc( tmpsiz );
    err = AudioConverterGetProperty( pv->converter, kAudioConverterApplicableEncodeBitRates,
                                     &tmpsiz, bitrates);
    bitrateCounts = tmpsiz / sizeof( AudioValueRange );

    // set bitrate
    tmp = audio->config.out.bitrate * 1000;
    if( tmp < bitrates[0].mMinimum )
        tmp = bitrates[0].mMinimum;
    if( tmp > bitrates[bitrateCounts-1].mMinimum )
        tmp = bitrates[bitrateCounts-1].mMinimum;
    free( bitrates );
    if( tmp != audio->config.out.bitrate * 1000 )
        hb_log( "encca_aac: sanitizing track %d audio bitrate %d to %"PRIu32"", 
                audio->config.out.track, audio->config.out.bitrate, tmp/1000 );
    AudioConverterSetProperty( pv->converter, kAudioConverterEncodeBitRate,
                               sizeof( tmp ), &tmp );

    // get real input
    tmpsiz = sizeof( input );
    AudioConverterGetProperty( pv->converter,
                               kAudioConverterCurrentInputStreamDescription,
                               &tmpsiz, &input );
    // get real output
    tmpsiz = sizeof( output );
    AudioConverterGetProperty( pv->converter,
                               kAudioConverterCurrentOutputStreamDescription,
                               &tmpsiz, &output );

    // set sizes
    pv->isamplesiz  = input.mBytesPerPacket;
    pv->isamples    = output.mFramesPerPacket;
    pv->osamplerate = output.mSampleRate;

    // get maximum output size
    AudioConverterGetProperty( pv->converter,
                               kAudioConverterPropertyMaximumOutputPacketSize,
                               &tmpsiz, &tmp );
    pv->omaxpacket = tmp;

    // get magic cookie (elementary stream descriptor)
    tmp = HB_CONFIG_MAX_SIZE;
    AudioConverterGetProperty( pv->converter,
                               kAudioConverterCompressionMagicCookie,
                               &tmp, w->config->aac.bytes );
    // CoreAudio returns a complete ESDS, but we only need
    // the DecoderSpecific info.
    UInt8* buffer = NULL;
    ReadESDSDescExt(w->config->aac.bytes, &buffer, &tmpsiz, 0);
    w->config->aac.length = tmpsiz;
    memmove( w->config->aac.bytes, buffer,
             w->config->aac.length );

    pv->list = hb_list_init();
    pv->buf = NULL;

    return 0;
}