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/* This file is part of the HandBrake source code.
Homepage: <http://handbrake.fr/>.
It may be used under the terms of the GNU General Public License. */
#include "hb.h"
#include "downmix.h"
#include <AudioToolbox/AudioToolbox.h>
#include <CoreAudio/CoreAudio.h>
enum AAC_MODE { AAC_MODE_LC, AAC_MODE_HE };
int encCoreAudioInitLC( hb_work_object_t *, hb_job_t * );
int encCoreAudioInitHE( hb_work_object_t *, hb_job_t * );
int encCoreAudioInit( hb_work_object_t *, hb_job_t *, enum AAC_MODE mode );
int encCoreAudioWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
void encCoreAudioClose( hb_work_object_t * );
hb_work_object_t hb_encca_aac =
{
WORK_ENC_CA_AAC,
"AAC encoder (Apple)",
encCoreAudioInitLC,
encCoreAudioWork,
encCoreAudioClose
};
hb_work_object_t hb_encca_haac =
{
WORK_ENC_CA_HAAC,
"HE-AAC encoder (Apple)",
encCoreAudioInitHE,
encCoreAudioWork,
encCoreAudioClose
};
struct hb_work_private_s
{
hb_job_t *job;
AudioConverterRef converter;
uint8_t *obuf;
uint8_t *buf;
hb_list_t *list;
unsigned long isamples, isamplesiz, omaxpacket, nchannels;
uint64_t pts, ibytes;
Float64 osamplerate;
int layout;
hb_chan_map_t *ichanmap;
};
#define MP4ESDescrTag 0x03
#define MP4DecConfigDescrTag 0x04
#define MP4DecSpecificDescrTag 0x05
// based off of mov_mp4_read_descr_len from mov.c in ffmpeg's libavformat
static int readDescrLen(UInt8 **buffer)
{
int len = 0;
int count = 4;
while (count--) {
int c = *(*buffer)++;
len = (len << 7) | (c & 0x7f);
if (!(c & 0x80))
break;
}
return len;
}
// based off of mov_mp4_read_descr from mov.c in ffmpeg's libavformat
static int readDescr(UInt8 **buffer, int *tag)
{
*tag = *(*buffer)++;
return readDescrLen(buffer);
}
// based off of mov_read_esds from mov.c in ffmpeg's libavformat
static long ReadESDSDescExt(void* descExt, UInt8 **buffer, UInt32 *size, int versionFlags)
{
UInt8 *esds = (UInt8 *) descExt;
int tag, len;
*size = 0;
if (versionFlags)
esds += 4; // version + flags
readDescr(&esds, &tag);
esds += 2; // ID
if (tag == MP4ESDescrTag)
esds++; // priority
readDescr(&esds, &tag);
if (tag == MP4DecConfigDescrTag) {
esds++; // object type id
esds++; // stream type
esds += 3; // buffer size db
esds += 4; // max bitrate
esds += 4; // average bitrate
len = readDescr(&esds, &tag);
if (tag == MP4DecSpecificDescrTag) {
*buffer = calloc(1, len + 8);
if (*buffer) {
memcpy(*buffer, esds, len);
*size = len;
}
}
}
return noErr;
}
/***********************************************************************
* hb_work_encCoreAudio_init switches
***********************************************************************
*
**********************************************************************/
int encCoreAudioInitLC( hb_work_object_t * w, hb_job_t * job )
{
return encCoreAudioInit( w, job, AAC_MODE_LC );
}
int encCoreAudioInitHE( hb_work_object_t * w, hb_job_t * job )
{
return encCoreAudioInit( w, job, AAC_MODE_HE );
}
/***********************************************************************
* 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;
}
}
// 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 = audio->config.out.samples_per_frame = output.mFramesPerPacket;
pv->osamplerate = output.mSampleRate;
// set channel map and layout (for remapping)
pv->ichanmap = audio->config.in.channel_map;
switch( audio->config.out.mixdown )
{
case HB_AMIXDOWN_MONO:
pv->layout = HB_INPUT_CH_LAYOUT_MONO;
break;
case HB_AMIXDOWN_STEREO:
case HB_AMIXDOWN_DOLBY:
case HB_AMIXDOWN_DOLBYPLII:
pv->layout = HB_INPUT_CH_LAYOUT_STEREO;
break;
case HB_AMIXDOWN_6CH:
default:
pv->layout = HB_INPUT_CH_LAYOUT_3F2R | HB_INPUT_CH_LAYOUT_HAS_LFE;
break;
}
// 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;
}
/***********************************************************************
* Close
***********************************************************************
*
**********************************************************************/
void encCoreAudioClose( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
if( pv->converter )
{
AudioConverterDispose( pv->converter );
hb_list_empty( &pv->list );
free( pv->obuf );
free( pv->buf );
free( pv );
w->private_data = NULL;
}
}
/* 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;
}
/***********************************************************************
* Encode
***********************************************************************
*
**********************************************************************/
static hb_buffer_t * Encode( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
UInt32 npackets = 1;
/* check if we need more data */
if( ( pv->ibytes = hb_list_bytes( pv->list ) ) < pv->isamples * pv->isamplesiz )
return NULL;
hb_buffer_t * obuf;
AudioStreamPacketDescription odesc = { 0 };
AudioBufferList obuflist = { .mNumberBuffers = 1,
.mBuffers = { { .mNumberChannels = pv->nchannels } },
};
obuf = hb_buffer_init( pv->omaxpacket );
obuflist.mBuffers[0].mDataByteSize = obuf->size;
obuflist.mBuffers[0].mData = obuf->data;
OSStatus err = AudioConverterFillComplexBuffer( pv->converter, inInputDataProc, pv,
&npackets, &obuflist, &odesc );
if( err ) {
hb_log( "CoreAudio: Not enough data" );
return NULL;
}
if( odesc.mDataByteSize == 0 || npackets == 0 ) {
hb_log( "CoreAudio: 0 packets returned " );
return NULL;
}
obuf->start = pv->pts;
pv->pts += 90000LL * pv->isamples / pv->osamplerate;
obuf->stop = pv->pts;
obuf->size = odesc.mDataByteSize;
obuf->frametype = HB_FRAME_AUDIO;
return obuf;
}
static hb_buffer_t *Flush( hb_work_object_t *w, hb_buffer_t *bufin )
{
hb_work_private_t *pv = w->private_data;
// pad whatever data we have out to four input frames.
int nbytes = hb_list_bytes( pv->list );
int pad = pv->isamples * pv->isamplesiz - nbytes;
if ( pad > 0 )
{
hb_buffer_t *tmp = hb_buffer_init( pad );
memset( tmp->data, 0, pad );
hb_list_add( pv->list, tmp );
}
hb_buffer_t *bufout = NULL, *buf = NULL;
while ( hb_list_bytes( pv->list ) >= pv->isamples * pv->isamplesiz )
{
hb_buffer_t *b = Encode( w );
if ( b )
{
if ( bufout == NULL )
{
bufout = b;
}
else
{
buf->next = b;
}
buf = b;
}
}
// add the eof marker to the end of our buf chain
if ( buf )
buf->next = bufin;
else
bufout = bufin;
return bufout;
}
/***********************************************************************
* Work
***********************************************************************
*
**********************************************************************/
int encCoreAudioWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
hb_buffer_t ** buf_out )
{
hb_work_private_t * pv = w->private_data;
hb_buffer_t * buf;
if( (*buf_in)->size <= 0 )
{
// EOF on input. Finish encoding what we have buffered then send
// it & the eof downstream.
*buf_out = Flush( w, *buf_in );
*buf_in = NULL;
return HB_WORK_DONE;
}
hb_list_add( pv->list, *buf_in );
*buf_in = NULL;
*buf_out = buf = Encode( w );
while( buf )
{
buf->next = Encode( w );
buf = buf->next;
}
return HB_WORK_OK;
}
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