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/* $Id: encavcodeca.c,v 1.23 2005/10/13 23:47:06 titer Exp $
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 "hbffmpeg.h"
#include "downmix.h"
#include "libavcodec/audioconvert.h"
struct hb_work_private_s
{
hb_job_t * job;
AVCodecContext * context;
int out_discrete_channels;
int samples_per_frame;
int layout;
unsigned long input_samples;
unsigned long output_bytes;
hb_list_t * list;
uint8_t * buf;
};
static int encavcodecaInit( hb_work_object_t *, hb_job_t * );
static int encavcodecaWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
static void encavcodecaClose( hb_work_object_t * );
hb_work_object_t hb_encavcodeca =
{
WORK_ENCAVCODEC_AUDIO,
"AVCodec Audio encoder (libavcodec)",
encavcodecaInit,
encavcodecaWork,
encavcodecaClose
};
static int encavcodecaInit( hb_work_object_t * w, hb_job_t * job )
{
AVCodec * codec;
AVCodecContext * context;
hb_audio_t * audio = w->audio;
hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
w->private_data = pv;
pv->job = job;
pv->out_discrete_channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown);
codec = avcodec_find_encoder( w->codec_param );
if( !codec )
{
hb_log( "encavcodecaInit: avcodec_find_encoder "
"failed" );
return 1;
}
context = avcodec_alloc_context3(codec);
if ( w->codec_param == CODEC_ID_AAC )
{
int ret = hb_av_set_string( context, codec, "stereo_mode", "ms_off" );
/* Let avutil sanity check the options for us*/
if( ret == AVERROR_OPTION_NOT_FOUND )
hb_log( "avcodec options: Unknown option %s", "stereo_mode" );
if( ret == AVERROR(EINVAL) )
hb_log( "avcodec options: Bad argument %s=%s",
"stereo_mode", "ms_off" ? "ms_off" : "(null)" );
}
context->channel_layout = AV_CH_LAYOUT_STEREO;
switch( audio->config.out.mixdown )
{
case HB_AMIXDOWN_MONO:
context->channel_layout = AV_CH_LAYOUT_MONO;
pv->layout = HB_INPUT_CH_LAYOUT_MONO;
break;
case HB_AMIXDOWN_STEREO:
case HB_AMIXDOWN_DOLBY:
case HB_AMIXDOWN_DOLBYPLII:
context->channel_layout = AV_CH_LAYOUT_STEREO;
pv->layout = HB_INPUT_CH_LAYOUT_STEREO;
break;
case HB_AMIXDOWN_6CH:
context->channel_layout = AV_CH_LAYOUT_5POINT1;
pv->layout = HB_INPUT_CH_LAYOUT_3F2R | HB_INPUT_CH_LAYOUT_HAS_LFE;
break;
default:
hb_log("encavcodecaInit: bad mixdown" );
break;
}
if( audio->config.out.bitrate > 0 )
context->bit_rate = audio->config.out.bitrate * 1000;
else if( audio->config.out.quality >= 0 )
{
context->global_quality = audio->config.out.quality * FF_QP2LAMBDA;
context->flags |= CODEC_FLAG_QSCALE;
}
if( audio->config.out.compression_level >= 0 )
context->compression_level = audio->config.out.compression_level;
context->sample_rate = audio->config.out.samplerate;
context->channels = pv->out_discrete_channels;
// Try to set format to float. Fallback to whatever is supported.
hb_ff_set_sample_fmt( context, codec );
if( hb_avcodec_open( context, codec, 0 ) )
{
hb_log( "encavcodecaInit: avcodec_open failed" );
return 1;
}
pv->context = context;
audio->config.out.samples_per_frame = pv->samples_per_frame = context->frame_size;
pv->input_samples = pv->samples_per_frame * pv->out_discrete_channels;
// Set a reasonable maximum output size
pv->output_bytes = context->frame_size *
av_get_bytes_per_sample(context->sample_fmt) *
context->channels;
pv->buf = malloc( pv->input_samples * sizeof( float ) );
pv->list = hb_list_init();
if ( context->extradata )
{
memcpy( w->config->extradata.bytes, context->extradata, context->extradata_size );
w->config->extradata.length = context->extradata_size;
}
return 0;
}
/***********************************************************************
* Close
***********************************************************************
*
**********************************************************************/
// Some encoders (e.g. flac) require a final NULL encode in order to
// finalize things.
static void Finalize( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
hb_buffer_t * buf = hb_buffer_init( pv->output_bytes );
// Finalize with NULL input needed by FLAC to generate md5sum
// in context extradata
avcodec_encode_audio( pv->context, buf->data, buf->alloc, NULL );
hb_buffer_close( &buf );
}
static void encavcodecaClose( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
if ( pv )
{
if( pv->context )
{
Finalize( w );
hb_deep_log( 2, "encavcodeca: closing libavcodec" );
if ( pv->context->codec )
avcodec_flush_buffers( pv->context );
hb_avcodec_close( pv->context );
}
if ( pv->buf )
{
free( pv->buf );
pv->buf = NULL;
}
if ( pv->list )
hb_list_empty( &pv->list );
free( pv );
w->private_data = NULL;
}
}
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;
}
static hb_buffer_t * Flush( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
hb_buffer_t *first, *buf, *last;
first = last = buf = Encode( w );
while( buf )
{
last = buf;
buf->next = Encode( w );
buf = buf->next;
}
if( last )
{
last->next = hb_buffer_init( pv->output_bytes );
buf = last->next;
}
else
{
first = buf = hb_buffer_init( pv->output_bytes );
}
// Finalize with NULL input needed by FLAC to generate md5sum
// in context extradata
avcodec_encode_audio( pv->context, buf->data, buf->alloc, NULL );
buf->size = 0;
return first;
}
/***********************************************************************
* Work
***********************************************************************
*
**********************************************************************/
static int encavcodecaWork( 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 * in = *buf_in, * buf;
if ( in->size <= 0 )
{
/* EOF on input - send it downstream & say we're done */
*buf_out = Flush( w );
return HB_WORK_DONE;
}
if ( pv->context == NULL || pv->context->codec == NULL )
{
// No encoder context. Nothing we can do.
return HB_WORK_OK;
}
hb_list_add( pv->list, 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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