/* Called whenever necessary by AudioConverterFillComplexBuffer */
static OSStatus inInputDataProc( AudioConverterRef converter, UInt32 *npackets,
AudioBufferList *buffers,
AudioStreamPacketDescription** ignored,
void *userdata )
{
hb_work_private_t *pv = userdata;
if( pv->ibytes == 0 )
{
*npackets = 0;
hb_log( "CoreAudio: no data to use in inInputDataProc" );
return 1;
}
if( pv->buf != NULL )
free( pv->buf );
uint64_t pts, pos;
buffers->mBuffers[0].mDataByteSize = MIN( *npackets * pv->isamplesiz, pv->ibytes );
buffers->mBuffers[0].mData = pv->buf = calloc( 1, buffers->mBuffers[0].mDataByteSize );
if( hb_list_bytes( pv->list ) >= buffers->mBuffers[0].mDataByteSize )
{
hb_list_getbytes( pv->list, buffers->mBuffers[0].mData,
buffers->mBuffers[0].mDataByteSize, &pts, &pos );
}
else
{
hb_log( "CoreAudio: Not enough data, exiting inInputDataProc" );
*npackets = 0;
return 1;
}
if( pv->ichanmap != &hb_qt_chan_map )
{
hb_layout_remap( pv->ichanmap, &hb_qt_chan_map, pv->layout,
(float*)buffers->mBuffers[0].mData,
buffers->mBuffers[0].mDataByteSize / pv->isamplesiz );
}
*npackets = buffers->mBuffers[0].mDataByteSize / pv->isamplesiz;
pv->ibytes -= buffers->mBuffers[0].mDataByteSize;
return noErr;
}
/* Called whenever necessary by AudioConverterFillComplexBuffer */
static OSStatus inInputDataProc(AudioConverterRef converter, UInt32 *npackets,
AudioBufferList *buffers,
AudioStreamPacketDescription **ignored,
void *userdata)
{
hb_work_private_t *pv = userdata;
if (!pv->ibytes)
{
*npackets = 0;
return 1;
}
if (pv->buf != NULL)
{
free(pv->buf);
}
buffers->mBuffers[0].mDataByteSize = MIN(pv->ibytes,
pv->isamplesiz * *npackets);
pv->buf = calloc(1, buffers->mBuffers[0].mDataByteSize);
buffers->mBuffers[0].mData = pv->buf;
if (hb_list_bytes(pv->list) >= buffers->mBuffers[0].mDataByteSize)
{
hb_list_getbytes(pv->list, buffers->mBuffers[0].mData,
buffers->mBuffers[0].mDataByteSize, NULL, NULL);
}
else
{
*npackets = 0;
return 1;
}
*npackets = buffers->mBuffers[0].mDataByteSize / pv->isamplesiz;
pv->ibytes -= buffers->mBuffers[0].mDataByteSize;
if (pv->ichanmap != &hb_qt_chan_map)
{
hb_layout_remap(pv->ichanmap, &hb_qt_chan_map, pv->layout,
(float*)buffers->mBuffers[0].mData, *npackets);
}
return noErr;
}
static hb_buffer_t * Encode( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
uint64_t pts, pos;
hb_audio_t * audio = w->audio;
hb_buffer_t * buf;
int ii;
if( hb_list_bytes( pv->list ) < pv->input_samples * sizeof( float ) )
{
return NULL;
}
hb_list_getbytes( pv->list, pv->buf, pv->input_samples * sizeof( float ),
&pts, &pos);
hb_chan_map_t *map = NULL;
if ( audio->config.in.codec == HB_ACODEC_AC3 )
{
map = &hb_ac3_chan_map;
}
else if ( audio->config.in.codec == HB_ACODEC_DCA )
{
map = &hb_qt_chan_map;
}
if ( map )
{
int layout;
switch (audio->config.out.mixdown)
{
case HB_AMIXDOWN_MONO:
layout = HB_INPUT_CH_LAYOUT_MONO;
break;
case HB_AMIXDOWN_STEREO:
case HB_AMIXDOWN_DOLBY:
case HB_AMIXDOWN_DOLBYPLII:
layout = HB_INPUT_CH_LAYOUT_STEREO;
break;
case HB_AMIXDOWN_6CH:
default:
layout = HB_INPUT_CH_LAYOUT_3F2R | HB_INPUT_CH_LAYOUT_HAS_LFE;
break;
}
hb_layout_remap( map, &hb_smpte_chan_map, layout,
(float*)pv->buf, AC3_SAMPLES_PER_FRAME);
}
for (ii = 0; ii < pv->input_samples; ii++)
{
// ffmpeg float samples are -1.0 to 1.0
pv->samples[ii] = ((float*)pv->buf)[ii] / 32768.0;
}
buf = hb_buffer_init( pv->output_bytes );
buf->size = avcodec_encode_audio( pv->context, buf->data, buf->alloc,
(short*)pv->samples );
buf->start = pts + 90000 * pos / pv->out_discrete_channels / sizeof( float ) / audio->config.out.samplerate;
buf->stop = buf->start + 90000 * AC3_SAMPLES_PER_FRAME / audio->config.out.samplerate;
buf->frametype = HB_FRAME_AUDIO;
if ( !buf->size )
{
hb_buffer_close( &buf );
return Encode( w );
}
else if (buf->size < 0)
{
hb_log( "encac3: avcodec_encode_audio failed" );
hb_buffer_close( &buf );
return NULL;
}
return buf;
}
static hb_buffer_t * Encode( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
uint64_t pts, pos;
hb_audio_t * audio = w->audio;
hb_buffer_t * buf;
if( hb_list_bytes( pv->list ) < pv->input_samples * sizeof( float ) )
{
return NULL;
}
hb_list_getbytes( pv->list, pv->buf, pv->input_samples * sizeof( float ),
&pts, &pos);
// XXX: ffaac fails to remap from the internal libav* channel map (SMPTE) to the native AAC channel map
// do it here - this hack should be removed if Libav fixes the bug
hb_chan_map_t * out_map = ( w->codec_param == CODEC_ID_AAC ) ? &hb_qt_chan_map : &hb_smpte_chan_map;
if ( audio->config.in.channel_map != out_map )
{
hb_layout_remap( audio->config.in.channel_map, out_map, pv->layout,
(float*)pv->buf, pv->samples_per_frame );
}
// Do we need to convert our internal float format?
if ( pv->context->sample_fmt != AV_SAMPLE_FMT_FLT )
{
int isamp, osamp;
AVAudioConvert *ctx;
isamp = av_get_bytes_per_sample( AV_SAMPLE_FMT_FLT );
osamp = av_get_bytes_per_sample( pv->context->sample_fmt );
ctx = av_audio_convert_alloc( pv->context->sample_fmt, 1,
AV_SAMPLE_FMT_FLT, 1,
NULL, 0 );
// get output buffer size then malloc a buffer
//nsamples = out_size / isamp;
//buffer = av_malloc( nsamples * sizeof(hb_sample_t) );
// we're doing straight sample format conversion which
// behaves as if there were only one channel.
const void * const ibuf[6] = { pv->buf };
void * const obuf[6] = { pv->buf };
const int istride[6] = { isamp };
const int ostride[6] = { osamp };
av_audio_convert( ctx, obuf, ostride, ibuf, istride, pv->input_samples );
av_audio_convert_free( ctx );
}
buf = hb_buffer_init( pv->output_bytes );
buf->size = avcodec_encode_audio( pv->context, buf->data, buf->alloc,
(short*)pv->buf );
buf->start = pts + 90000 * pos / pv->out_discrete_channels / sizeof( float ) / audio->config.out.samplerate;
buf->stop = buf->start + 90000 * pv->samples_per_frame / audio->config.out.samplerate;
buf->frametype = HB_FRAME_AUDIO;
if ( !buf->size )
{
hb_buffer_close( &buf );
return Encode( w );
}
else if (buf->size < 0)
{
hb_log( "encavcodeca: avcodec_encode_audio failed" );
hb_buffer_close( &buf );
return NULL;
}
return buf;
}
