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 * );
+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 =
{
struct hb_work_private_s
{
+ uint8_t *buf;
hb_job_t *job;
+ hb_list_t *list;
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;
};
{
int len = 0;
int count = 4;
- while (count--) {
+ while (count--)
+ {
int c = *(*buffer)++;
len = (len << 7) | (c & 0x7f);
if (!(c & 0x80))
// 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;
+ UInt8 *esds = (UInt8*)descExt;
int tag, len;
*size = 0;
esds++; // priority
readDescr(&esds, &tag);
- if (tag == MP4DecConfigDescrTag) {
+ if (tag == MP4DecConfigDescrTag)
+ {
esds++; // object type id
esds++; // stream type
esds += 3; // buffer size db
esds += 4; // average bitrate
len = readDescr(&esds, &tag);
- if (tag == MP4DecSpecificDescrTag) {
+ if (tag == MP4DecSpecificDescrTag)
+ {
*buffer = calloc(1, len + 8);
- if (*buffer) {
+ if (*buffer)
+ {
memcpy(*buffer, esds, len);
*size = len;
}
***********************************************************************
*
**********************************************************************/
-int encCoreAudioInitLC( hb_work_object_t * w, hb_job_t * job )
+int encCoreAudioInitLC(hb_work_object_t *w, hb_job_t *job)
{
- return encCoreAudioInit( w, job, AAC_MODE_LC );
+ return encCoreAudioInit(w, job, AAC_MODE_LC);
}
-int encCoreAudioInitHE( hb_work_object_t * w, hb_job_t * job )
+int encCoreAudioInitHE(hb_work_object_t *w, hb_job_t *job)
{
- return encCoreAudioInit( w, job, AAC_MODE_HE );
+ return encCoreAudioInit(w, job, AAC_MODE_HE);
}
/***********************************************************************
***********************************************************************
*
**********************************************************************/
-int encCoreAudioInit( hb_work_object_t * w, hb_job_t * job, enum AAC_MODE mode )
+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;
+ 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 );
+ 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 );
+ pv->nchannels =
+ HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown);
- bzero( &input, sizeof( AudioStreamBasicDescription ) );
- input.mSampleRate = ( Float64 ) audio->config.out.samplerate;
+ bzero(&input, sizeof(AudioStreamBasicDescription));
+ input.mSampleRate = (Float64)audio->config.out.samplerate;
input.mFormatID = kAudioFormatLinearPCM;
- input.mFormatFlags = kLinearPCMFormatFlagIsFloat | kAudioFormatFlagsNativeEndian;
+ 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 )
+ bzero(&output, sizeof(AudioStreamBasicDescription));
+ switch (mode)
{
case AAC_MODE_HE:
output.mFormatID = kAudioFormatMPEG4AAC_HE;
output.mFormatID = kAudioFormatMPEG4AAC;
break;
}
- output.mSampleRate = ( Float64 ) audio->config.out.samplerate;
+ output.mSampleRate = (Float64)audio->config.out.samplerate;
output.mChannelsPerFrame = pv->nchannels;
- // let CoreAudio decide the rest...
+ // let CoreAudio decide the rest
// initialise encoder
- err = AudioConverterNew( &input, &output, &pv->converter );
- if( err != noErr)
+ err = AudioConverterNew(&input, &output, &pv->converter);
+ if (err != noErr)
{
// Retry without the samplerate
- bzero( &output, sizeof( AudioStreamBasicDescription ) );
- switch ( mode )
+ bzero(&output, sizeof(AudioStreamBasicDescription));
+ switch (mode)
{
case AAC_MODE_HE:
output.mFormatID = kAudioFormatMPEG4AAC_HE;
}
output.mChannelsPerFrame = pv->nchannels;
- err = AudioConverterNew( &input, &output, &pv->converter );
+ err = AudioConverterNew(&input, &output, &pv->converter);
- if( err != noErr)
+ if (err != noErr)
{
- hb_log( "Error creating an AudioConverter err=%"PRId64" %"PRIu64, (int64_t)err, (uint64_t)output.mBytesPerFrame );
+ hb_log("Error creating an AudioConverter err=%"PRId64" output.mBytesPerFrame=%"PRIu64"",
+ (int64_t)err, (uint64_t)output.mBytesPerFrame);
*job->die = 1;
- return 0;
+ return -1;
}
}
// set encoder quality to maximum
tmp = kAudioConverterQuality_Max;
- AudioConverterSetProperty( pv->converter, kAudioConverterCodecQuality,
- sizeof( tmp ), &tmp );
+ AudioConverterSetProperty(pv->converter, kAudioConverterCodecQuality,
+ sizeof(tmp), &tmp);
- if( audio->config.out.bitrate > 0 )
+ if (audio->config.out.bitrate > 0)
{
// set encoder bitrate control mode to constrained variable
tmp = kAudioCodecBitRateControlMode_VariableConstrained;
- AudioConverterSetProperty( pv->converter, kAudioCodecPropertyBitRateControlMode,
- sizeof( tmp ), &tmp );
+ 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 );
+ 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 )
+ if (tmp < bitrates[0].mMinimum)
tmp = bitrates[0].mMinimum;
- if( tmp > bitrates[bitrateCounts-1].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 );
+ 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);
}
- else if( audio->config.out.quality >= 0 )
+ else if (audio->config.out.quality >= 0)
{
- if( mode != AAC_MODE_LC )
+ if (mode != AAC_MODE_LC)
{
- hb_log( "encCoreAudioInit: internal error, VBR set but not applicable" );
+ hb_log("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 );
+ AudioConverterSetProperty(pv->converter, kAudioCodecPropertyBitRateControlMode,
+ sizeof(tmp), &tmp);
// set quality
tmp = audio->config.out.quality;
- AudioConverterSetProperty( pv->converter, kAudioCodecPropertySoundQualityForVBR,
- sizeof( tmp ), &tmp );
+ AudioConverterSetProperty(pv->converter, kAudioCodecPropertySoundQualityForVBR,
+ sizeof(tmp), &tmp);
}
else
{
- hb_log( "encCoreAudioInit: internal error, bitrate/quality not set" );
+ hb_log("encCoreAudioInit: internal error, bitrate/quality not set");
return 1;
}
// get real input
- tmpsiz = sizeof( input );
- AudioConverterGetProperty( pv->converter,
- kAudioConverterCurrentInputStreamDescription,
- &tmpsiz, &input );
+ tmpsiz = sizeof(input);
+ AudioConverterGetProperty(pv->converter,
+ kAudioConverterCurrentInputStreamDescription,
+ &tmpsiz, &input);
// get real output
- tmpsiz = sizeof( output );
- AudioConverterGetProperty( pv->converter,
- kAudioConverterCurrentOutputStreamDescription,
- &tmpsiz, &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->isamples = output.mFramesPerPacket;
pv->osamplerate = output.mSampleRate;
+ audio->config.out.samples_per_frame = pv->isamples;
// set channel map and layout (for remapping)
pv->ichanmap = audio->config.in.channel_map;
- switch( audio->config.out.mixdown )
+ switch (audio->config.out.mixdown)
{
case HB_AMIXDOWN_MONO:
pv->layout = HB_INPUT_CH_LAYOUT_MONO;
}
// get maximum output size
- AudioConverterGetProperty( pv->converter,
- kAudioConverterPropertyMaximumOutputPacketSize,
- &tmpsiz, &tmp );
+ 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 );
+ 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 );
+ memmove(w->config->extradata.bytes, buffer, w->config->extradata.length);
pv->list = hb_list_init();
pv->buf = NULL;
***********************************************************************
*
**********************************************************************/
-void encCoreAudioClose( hb_work_object_t * w )
+void encCoreAudioClose(hb_work_object_t *w)
{
- hb_work_private_t * pv = w->private_data;
+ hb_work_private_t *pv = w->private_data;
- if( pv->converter )
+ if (pv != NULL)
{
- AudioConverterDispose( pv->converter );
- hb_list_empty( &pv->list );
- free( pv->obuf );
- free( pv->buf );
- free( pv );
+ if (pv->converter)
+ {
+ AudioConverterDispose(pv->converter);
+ }
+ if (pv->buf != NULL)
+ {
+ free(pv->buf);
+ }
+ hb_list_empty(&pv->list);
+ free(pv);
w->private_data = NULL;
}
}
/* Called whenever necessary by AudioConverterFillComplexBuffer */
-static OSStatus inInputDataProc( AudioConverterRef converter, UInt32 *npackets,
- AudioBufferList *buffers,
- AudioStreamPacketDescription** ignored,
- void *userdata )
+static OSStatus inInputDataProc(AudioConverterRef converter, UInt32 *npackets,
+ AudioBufferList *buffers,
+ AudioStreamPacketDescription **ignored,
+ void *userdata)
{
hb_work_private_t *pv = userdata;
- if( pv->ibytes == 0 )
+ if (!pv->ibytes)
{
*npackets = 0;
- hb_log( "CoreAudio: no data to use in inInputDataProc" );
return 1;
}
- if( pv->buf != NULL )
- free( pv->buf );
+ 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 );
+ 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 )
+ 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 );
+ hb_list_getbytes(pv->list, buffers->mBuffers[0].mData,
+ buffers->mBuffers[0].mDataByteSize, NULL, NULL);
}
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;
+ 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 )
+static hb_buffer_t* Encode(hb_work_object_t *w)
{
- hb_work_private_t * pv = w->private_data;
+ 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 )
+ if ((pv->ibytes = hb_list_bytes(pv->list)) < pv->isamples * pv->isamplesiz)
+ {
return NULL;
+ }
- hb_buffer_t * obuf;
+ hb_buffer_t *obuf;
AudioStreamPacketDescription odesc = { 0 };
- AudioBufferList obuflist = { .mNumberBuffers = 1,
- .mBuffers = { { .mNumberChannels = pv->nchannels } },
- };
+ AudioBufferList obuflist =
+ {
+ .mNumberBuffers = 1,
+ .mBuffers = { { .mNumberChannels = pv->nchannels } },
+ };
- obuf = hb_buffer_init( pv->omaxpacket );
+ 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;
+ OSStatus err = AudioConverterFillComplexBuffer(pv->converter,
+ inInputDataProc, pv,
+ &npackets, &obuflist, &odesc);
+
+ if (err != noErr && err != 1)
+ {
+ hb_log("encCoreAudio: unexpected error in AudioConverterFillComplexBuffer()");
}
- if( odesc.mDataByteSize == 0 || npackets == 0 ) {
- hb_log( "CoreAudio: 0 packets returned " );
+ // only drop the output buffer if it's actually empty
+ if (!odesc.mDataByteSize || !npackets)
+ {
+ hb_log("encCoreAudio: 0 packets returned");
return NULL;
}
- obuf->start = pv->pts;
+ obuf->size = odesc.mDataByteSize;
+ obuf->start = pv->pts;
pv->pts += 90000LL * pv->isamples / pv->osamplerate;
- obuf->stop = pv->pts;
- obuf->size = odesc.mDataByteSize;
+ obuf->stop = pv->pts;
obuf->frametype = HB_FRAME_AUDIO;
return obuf;
}
-static hb_buffer_t *Flush( hb_work_object_t *w, hb_buffer_t *bufin )
+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 nbytes = hb_list_bytes(pv->list);
int pad = pv->isamples * pv->isamplesiz - nbytes;
- if ( pad > 0 )
+ 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 *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 )
+ while (hb_list_bytes(pv->list) >= pv->isamples * pv->isamplesiz)
{
- hb_buffer_t *b = Encode( w );
- if ( b )
+ hb_buffer_t *b = Encode(w);
+ if (b != NULL)
{
- if ( bufout == NULL )
+ if (bufout == NULL)
{
bufout = b;
}
}
}
// add the eof marker to the end of our buf chain
- if ( buf )
+ if (buf != NULL)
+ {
buf->next = bufin;
+ }
else
+ {
bufout = bufin;
+ }
return bufout;
}
***********************************************************************
*
**********************************************************************/
-int encCoreAudioWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
- hb_buffer_t ** buf_out )
+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;
+ hb_work_private_t *pv = w->private_data;
+ hb_buffer_t *buf;
- if( (*buf_in)->size <= 0 )
+ 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_out = Flush(w, *buf_in);
*buf_in = NULL;
return HB_WORK_DONE;
}
- hb_list_add( pv->list, *buf_in );
+ hb_list_add(pv->list, *buf_in);
*buf_in = NULL;
- *buf_out = buf = Encode( w );
+ *buf_out = buf = Encode(w);
- while( buf )
+ while (buf != NULL)
{
- buf->next = Encode( w );
+ buf->next = Encode(w);
buf = buf->next;
}
projects / handbrake / commitdiff