@ -257,6 +257,8 @@ typedef struct AVOutputStream {
/* audio only */
int audio_resample ;
ReSampleContext * resample ; /* for audio resampling */
int reformat_pair ;
AVAudioConvert * reformat_ctx ;
AVFifoBuffer fifo ; /* for compression: one audio fifo per codec */
FILE * logfile ;
} AVOutputStream ;
@ -513,6 +515,7 @@ static void do_audio_out(AVFormatContext *s,
uint8_t * buftmp ;
static uint8_t * audio_buf = NULL ;
static uint8_t * audio_out = NULL ;
static uint8_t * audio_out2 = NULL ;
const int audio_out_size = 4 * MAX_AUDIO_PACKET_SIZE ;
int size_out , frame_bytes , ret ;
@ -541,6 +544,26 @@ static void do_audio_out(AVFormatContext *s,
}
}
# define MAKE_SFMT_PAIR(a,b) ((a)+SAMPLE_FMT_NB*(b))
if ( dec - > sample_fmt ! = enc - > sample_fmt & &
MAKE_SFMT_PAIR ( enc - > sample_fmt , dec - > sample_fmt ) ! = ost - > reformat_pair ) {
if ( ! audio_out2 )
audio_out2 = av_malloc ( audio_out_size ) ;
if ( ! audio_out2 )
av_exit ( 1 ) ;
if ( ost - > reformat_ctx )
av_audio_convert_free ( ost - > reformat_ctx ) ;
ost - > reformat_ctx = av_audio_convert_alloc ( enc - > sample_fmt , 1 ,
dec - > sample_fmt , 1 , NULL , 0 ) ;
if ( ! ost - > reformat_ctx ) {
fprintf ( stderr , " Cannot convert %s sample format to %s sample format \n " ,
avcodec_get_sample_fmt_name ( dec - > sample_fmt ) ,
avcodec_get_sample_fmt_name ( enc - > sample_fmt ) ) ;
av_exit ( 1 ) ;
}
ost - > reformat_pair = MAKE_SFMT_PAIR ( enc - > sample_fmt , dec - > sample_fmt ) ;
}
if ( audio_sync_method ) {
double delta = get_sync_ipts ( ost ) * enc - > sample_rate - ost - > sync_opts
- av_fifo_size ( & ost - > fifo ) / ( ost - > st - > codec - > channels * 2 ) ;
@ -599,6 +622,22 @@ static void do_audio_out(AVFormatContext *s,
size_out = size ;
}
if ( dec - > sample_fmt ! = enc - > sample_fmt ) {
const void * ibuf [ 6 ] = { buftmp } ;
void * obuf [ 6 ] = { audio_out2 } ;
int istride [ 6 ] = { av_get_bits_per_sample_format ( dec - > sample_fmt ) / 8 } ;
int ostride [ 6 ] = { av_get_bits_per_sample_format ( enc - > sample_fmt ) / 8 } ;
int len = size_out / istride [ 0 ] ;
if ( av_audio_convert ( ost - > reformat_ctx , obuf , ostride , ibuf , istride , len ) < 0 ) {
printf ( " av_audio_convert() failed \n " ) ;
return ;
}
buftmp = audio_out2 ;
/* FIXME: existing code assume that size_out equals framesize*channels*2
remove this legacy cruft */
size_out = len * 2 ;
}
/* now encode as many frames as possible */
if ( enc - > frame_size > 1 ) {
/* output resampled raw samples */
@ -1726,6 +1765,7 @@ static int av_encode(AVFormatContext **output_files,
case CODEC_TYPE_AUDIO :
if ( av_fifo_init ( & ost - > fifo , 1024 ) )
goto fail ;
ost - > reformat_pair = MAKE_SFMT_PAIR ( SAMPLE_FMT_NONE , SAMPLE_FMT_NONE ) ;
ost - > audio_resample = codec - > sample_rate ! = icodec - > sample_rate | | audio_sync_method > 1 ;
icodec - > request_channels = codec - > channels ;
ist - > decoding_needed = 1 ;
@ -2162,6 +2202,8 @@ static int av_encode(AVFormatContext **output_files,
sws_freeContext ( ost - > img_resample_ctx ) ;
if ( ost - > resample )
audio_resample_close ( ost - > resample ) ;
if ( ost - > reformat_ctx )
av_audio_convert_free ( ost - > reformat_ctx ) ;
av_free ( ost ) ;
}
}
@ -2763,6 +2805,7 @@ static void opt_input_file(const char *filename)
ap - > width = frame_width + frame_padleft + frame_padright ;
ap - > height = frame_height + frame_padtop + frame_padbottom ;
ap - > pix_fmt = frame_pix_fmt ;
// ap->sample_fmt = audio_sample_fmt; //FIXME:not implemented in libavformat
ap - > channel = video_channel ;
ap - > standard = video_standard ;
ap - > video_codec_id = find_codec_or_die ( video_codec_name , CODEC_TYPE_VIDEO , 0 ) ;
@ -2827,6 +2870,7 @@ static void opt_input_file(const char *filename)
//fprintf(stderr, "\nInput Audio channels: %d", enc->channels);
audio_channels = enc - > channels ;
audio_sample_rate = enc - > sample_rate ;
audio_sample_fmt = enc - > sample_fmt ;
if ( audio_disable )
ic - > streams [ i ] - > discard = AVDISCARD_ALL ;
break ;
@ -3109,6 +3153,7 @@ static void new_audio_stream(AVFormatContext *oc)
st - > stream_copy = 1 ;
audio_enc - > channels = audio_channels ;
} else {
AVCodec * codec ;
codec_id = av_guess_codec ( oc - > oformat , NULL , oc - > filename , NULL , CODEC_TYPE_AUDIO ) ;
set_context_opts ( audio_enc , avctx_opts [ CODEC_TYPE_AUDIO ] , AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM ) ;
@ -3116,6 +3161,7 @@ static void new_audio_stream(AVFormatContext *oc)
if ( audio_codec_name )
codec_id = find_codec_or_die ( audio_codec_name , CODEC_TYPE_AUDIO , 1 ) ;
audio_enc - > codec_id = codec_id ;
codec = avcodec_find_encoder ( codec_id ) ;
if ( audio_qscale > QSCALE_NONE ) {
audio_enc - > flags | = CODEC_FLAG_QSCALE ;
@ -3123,6 +3169,17 @@ static void new_audio_stream(AVFormatContext *oc)
}
audio_enc - > thread_count = thread_count ;
audio_enc - > channels = audio_channels ;
audio_enc - > sample_fmt = audio_sample_fmt ;
if ( codec & & codec - > sample_fmts ) {
const enum SampleFormat * p = codec - > sample_fmts ;
for ( ; * p ! = - 1 ; p + + ) {
if ( * p = = audio_enc - > sample_fmt )
break ;
}
if ( * p = = - 1 )
audio_enc - > sample_fmt = codec - > sample_fmts [ 0 ] ;
}
}
audio_enc - > sample_rate = audio_sample_rate ;
audio_enc - > time_base = ( AVRational ) { 1 , audio_sample_rate } ;
@ -3793,6 +3850,7 @@ static const OptionDef options[] = {
{ " vol " , OPT_INT | HAS_ARG | OPT_AUDIO , { ( void * ) & audio_volume } , " change audio volume (256=normal) " , " volume " } , //
{ " newaudio " , OPT_AUDIO , { ( void * ) opt_new_audio_stream } , " add a new audio stream to the current output stream " } ,
{ " alang " , HAS_ARG | OPT_STRING | OPT_AUDIO , { ( void * ) & audio_language } , " set the ISO 639 language code (3 letters) of the current audio stream " , " code " } ,
{ " sample_fmt " , HAS_ARG | OPT_EXPERT | OPT_AUDIO , { ( void * ) opt_audio_sample_fmt } , " set sample format, 'list' as argument shows all the sample formats supported " , " format " } ,
/* subtitle options */
{ " sn " , OPT_BOOL | OPT_SUBTITLE , { ( void * ) & subtitle_disable } , " disable subtitle " } ,
Peter Ross
17 years ago