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/* encfaac.c
Copyright (c) 2003-2012 HandBrake Team
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 v2.
For full terms see the file COPYING file or visit http://www.gnu.org/licenses/gpl-2.0.html
*/
#include "hb.h"
#include "faac.h"
struct hb_work_private_s
{
hb_job_t * job;
faacEncHandle * faac;
unsigned long input_samples;
unsigned long output_bytes;
uint8_t * buf;
uint8_t * obuf;
hb_list_t * list;
double pts;
double framedur;
int out_discrete_channels;
};
int encfaacInit( hb_work_object_t *, hb_job_t * );
int encfaacWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
void encfaacClose( hb_work_object_t * );
hb_work_object_t hb_encfaac =
{
WORK_ENCFAAC,
"AAC encoder (libfaac)",
encfaacInit,
encfaacWork,
encfaacClose
};
static const int valid_rates[] =
{
22050, 24000, 32000, 44100, 48000, 0
};
static int find_samplerate( int rate )
{
int i;
for ( i = 0; valid_rates[i] && rate > valid_rates[i]; ++i )
{
}
return i;
}
/***********************************************************************
* hb_work_encfaac_init
***********************************************************************
*
**********************************************************************/
int encfaacInit( hb_work_object_t * w, hb_job_t * job )
{
hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
hb_audio_t * audio = w->audio;
faacEncConfigurationPtr cfg;
uint8_t * bytes;
unsigned long length;
w->private_data = pv;
pv->job = job;
/* pass the number of channels used into the private work data */
pv->out_discrete_channels = hb_mixdown_get_discrete_channel_count( audio->config.out.mixdown );
/* if the sample rate is 'auto' and that has given us an invalid output */
/* rate, map it to the next highest output rate or 48K if above the highest. */
int rate_index = find_samplerate(audio->config.out.samplerate);
if ( audio->config.out.samplerate != valid_rates[rate_index] )
{
int rate = valid_rates[valid_rates[rate_index]? rate_index : rate_index - 1];
hb_log( "encfaac changing output samplerate from %d to %d",
audio->config.out.samplerate, rate );
audio->config.out.samplerate = rate;
/* if the new rate is over the max bandwidth per channel limit */
/* lower the bandwidth. */
double bw = audio->config.out.bitrate * 1000 / pv->out_discrete_channels;
if ( bw > (double)rate * (6144./1024.) )
{
int newbr = (double)rate * (6.144/1024.) * pv->out_discrete_channels;
hb_log( "encfaac changing output bitrate from %d to %d",
audio->config.out.bitrate, newbr );
audio->config.out.bitrate = newbr;
}
}
pv->faac = faacEncOpen( audio->config.out.samplerate, pv->out_discrete_channels,
&pv->input_samples, &pv->output_bytes );
pv->buf = malloc( pv->input_samples * sizeof( float ) );
pv->obuf = malloc( pv->output_bytes );
pv->framedur = 90000.0 * pv->input_samples /
( audio->config.out.samplerate * pv->out_discrete_channels );
audio->config.out.samples_per_frame = pv->input_samples / pv->out_discrete_channels;
cfg = faacEncGetCurrentConfiguration( pv->faac );
cfg->mpegVersion = MPEG4;
cfg->aacObjectType = LOW;
cfg->allowMidside = 1;
if (pv->out_discrete_channels == 6) {
/* we are preserving 5.1 audio into 6-channel AAC,
so indicate that we have an lfe channel */
cfg->useLfe = 1;
} else {
cfg->useLfe = 0;
}
cfg->useTns = 0;
cfg->bitRate = audio->config.out.bitrate * 1000 / pv->out_discrete_channels; /* Per channel */
cfg->bandWidth = 0;
cfg->outputFormat = 0;
cfg->inputFormat = FAAC_INPUT_FLOAT;
if( ( audio->config.out.mixdown == HB_AMIXDOWN_6CH ) && ( audio->config.in.channel_map != &hb_qt_chan_map ) )
{
if( audio->config.in.channel_map == &hb_ac3_chan_map )
{
cfg->channel_map[0] = 2;
cfg->channel_map[1] = 1;
cfg->channel_map[2] = 3;
cfg->channel_map[3] = 4;
cfg->channel_map[4] = 5;
cfg->channel_map[5] = 0;
}
else if( audio->config.in.channel_map == &hb_smpte_chan_map )
{
cfg->channel_map[0] = 2;
cfg->channel_map[1] = 0;
cfg->channel_map[2] = 1;
cfg->channel_map[3] = 4;
cfg->channel_map[4] = 5;
cfg->channel_map[5] = 3;
}
}
if( !faacEncSetConfiguration( pv->faac, cfg ) )
{
hb_log( "faacEncSetConfiguration failed" );
*job->die = 1;
return 0;
}
if( faacEncGetDecoderSpecificInfo( pv->faac, &bytes, &length ) < 0 )
{
hb_log( "faacEncGetDecoderSpecificInfo failed" );
*job->die = 1;
return 0;
}
memcpy( w->config->extradata.bytes, bytes, length );
w->config->extradata.length = length;
free( bytes );
pv->list = hb_list_init();
return 0;
}
/***********************************************************************
* Close
***********************************************************************
*
**********************************************************************/
void encfaacClose( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
if ( pv )
{
if ( pv->faac )
{
faacEncClose( pv->faac );
pv->faac = NULL;
}
if ( pv->buf )
{
free( pv->buf );
pv->buf = NULL;
}
if ( pv->obuf )
{
free( pv->obuf );
pv->obuf = NULL;
}
if ( pv->list )
hb_list_empty( &pv->list );
free( pv );
w->private_data = NULL;
}
}
/***********************************************************************
* Encode
***********************************************************************
*
**********************************************************************/
static hb_buffer_t * Encode( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
if( hb_list_bytes( pv->list ) < pv->input_samples * sizeof( float ) )
{
/* Need more data */
return NULL;
}
uint64_t pts, pos;
hb_list_getbytes( pv->list, pv->buf, pv->input_samples * sizeof( float ),
&pts, &pos );
int i;
float *fltBuf = (float*)pv->buf;
for ( i = 0; i < pv->input_samples; i++ )
fltBuf[i] *= 32768.0;
int size = faacEncEncode( pv->faac, (int32_t *)pv->buf, pv->input_samples,
pv->obuf, pv->output_bytes );
// AAC needs four frames before it can start encoding so we'll get nothing
// on the first three calls to the encoder.
if ( size > 0 )
{
hb_buffer_t * buf = hb_buffer_init( size );
memcpy( buf->data, pv->obuf, size );
buf->size = size;
buf->s.start = pv->pts;
pv->pts += pv->framedur;
buf->s.stop = pv->pts;
buf->s.type = AUDIO_BUF;
buf->s.frametype = HB_FRAME_AUDIO;
return buf;
}
return NULL;
}
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->input_samples * sizeof(float) * 4 - nbytes;
if ( pad > 0 )
{
hb_buffer_t *tmp = hb_buffer_init( pad );
memset( tmp->data, 0, pad );
hb_list_add( pv->list, tmp );
}
// There are up to three frames buffered in the encoder plus one
// in our list buffer so four calls to Encode should get them all.
hb_buffer_t *bufout = NULL, *buf = NULL;
while ( hb_list_bytes( pv->list ) >= pv->input_samples * sizeof(float) )
{
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 encfaacWork( 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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