@ -31,7 +31,7 @@
# include "mpegaudio.h"
# include <lame/lame.h>
# define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
# define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
typedef struct Mp3AudioContext {
AVClass * class ;
lame_global_flags * gfp ;
@ -62,17 +62,17 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
lame_set_in_samplerate ( s - > gfp , avctx - > sample_rate ) ;
lame_set_out_samplerate ( s - > gfp , avctx - > sample_rate ) ;
lame_set_num_channels ( s - > gfp , avctx - > channels ) ;
if ( avctx - > compression_level = = FF_COMPRESSION_DEFAULT ) {
if ( avctx - > compression_level = = FF_COMPRESSION_DEFAULT ) {
lame_set_quality ( s - > gfp , 5 ) ;
} else {
lame_set_quality ( s - > gfp , avctx - > compression_level ) ;
}
lame_set_mode ( s - > gfp , s - > stereo ? JOINT_STEREO : MONO ) ;
lame_set_brate ( s - > gfp , avctx - > bit_rate / 1000 ) ;
if ( avctx - > flags & CODEC_FLAG_QSCALE ) {
lame_set_brate ( s - > gfp , avctx - > bit_rate / 1000 ) ;
if ( avctx - > flags & CODEC_FLAG_QSCALE ) {
lame_set_brate ( s - > gfp , 0 ) ;
lame_set_VBR ( s - > gfp , vbr_default ) ;
lame_set_VBR_quality ( s - > gfp , avctx - > global_quality / ( float ) FF_QP2LAMBDA ) ;
lame_set_VBR_quality ( s - > gfp , avctx - > global_quality / ( float ) FF_QP2LAMBDA ) ;
}
lame_set_bWriteVbrTag ( s - > gfp , 0 ) ;
# if FF_API_LAME_GLOBAL_OPTS
@ -82,14 +82,14 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
if ( lame_init_params ( s - > gfp ) < 0 )
goto err_close ;
avctx - > frame_size = lame_get_framesize ( s - > gfp ) ;
avctx - > frame_size = lame_get_framesize ( s - > gfp ) ;
if ( ! ( avctx - > coded_frame = avcodec_alloc_frame ( ) ) ) {
lame_close ( s - > gfp ) ;
return AVERROR ( ENOMEM ) ;
}
avctx - > coded_frame - > key_frame = 1 ;
avctx - > coded_frame - > key_frame = 1 ;
if ( AV_SAMPLE_FMT_S32 = = avctx - > sample_fmt & & s - > stereo ) {
int nelem = 2 * avctx - > frame_size ;
@ -117,60 +117,62 @@ static const int sSampleRates[] = {
} ;
static const int sBitRates [ 2 ] [ 3 ] [ 15 ] = {
{ { 0 , 32 , 64 , 96 , 128 , 160 , 192 , 224 , 256 , 288 , 320 , 352 , 384 , 416 , 448 } ,
{ 0 , 32 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 160 , 192 , 224 , 256 , 320 , 384 } ,
{ 0 , 32 , 40 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 160 , 192 , 224 , 256 , 320 }
{
{ 0 , 32 , 64 , 96 , 128 , 160 , 192 , 224 , 256 , 288 , 320 , 352 , 384 , 416 , 448 } ,
{ 0 , 32 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 160 , 192 , 224 , 256 , 320 , 384 } ,
{ 0 , 32 , 40 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 160 , 192 , 224 , 256 , 320 }
} ,
{ { 0 , 32 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 144 , 160 , 176 , 192 , 224 , 256 } ,
{ 0 , 8 , 16 , 24 , 32 , 40 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 144 , 160 } ,
{ 0 , 8 , 16 , 24 , 32 , 40 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 144 , 160 }
{
{ 0 , 32 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 144 , 160 , 176 , 192 , 224 , 256 } ,
{ 0 , 8 , 16 , 24 , 32 , 40 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 144 , 160 } ,
{ 0 , 8 , 16 , 24 , 32 , 40 , 48 , 56 , 64 , 80 , 96 , 112 , 128 , 144 , 160 }
} ,
} ;
static const int sSamplesPerFrame [ 2 ] [ 3 ] =
{
{ 384 , 1152 , 1152 } ,
{ 384 , 1152 , 576 }
static const int sSamplesPerFrame [ 2 ] [ 3 ] = {
{ 384 , 1152 , 1152 } ,
{ 384 , 1152 , 576 }
} ;
static const int sBitsPerSlot [ 3 ] = {
32 ,
8 ,
8
} ;
static const int sBitsPerSlot [ 3 ] = { 32 , 8 , 8 } ;
static int mp3len ( void * data , int * samplesPerFrame , int * sampleRate )
{
uint32_t header = AV_RB32 ( data ) ;
int layerID = 3 - ( ( header > > 17 ) & 0x03 ) ;
int bitRateID = ( ( header > > 12 ) & 0x0f ) ;
uint32_t header = AV_RB32 ( data ) ;
int layerID = 3 - ( ( header > > 17 ) & 0x03 ) ;
int bitRateID = ( ( header > > 12 ) & 0x0f ) ;
int sampleRateID = ( ( header > > 10 ) & 0x03 ) ;
int bitsPerSlot = sBitsPerSlot [ layerID ] ;
int isPadded = ( ( header > > 9 ) & 0x01 ) ;
static int const mode_tab [ 4 ] = { 2 , 3 , 1 , 0 } ;
int mode = mode_tab [ ( header > > 19 ) & 0x03 ] ;
int mpeg_id = mode > 0 ;
int bitsPerSlot = sBitsPerSlot [ layerID ] ;
int isPadded = ( ( header > > 9 ) & 0x01 ) ;
static int const mode_tab [ 4 ] = { 2 , 3 , 1 , 0 } ;
int mode = mode_tab [ ( header > > 19 ) & 0x03 ] ;
int mpeg_id = mode > 0 ;
int temp0 , temp1 , bitRate ;
if ( ( ( header > > 21 ) & 0x7ff ) ! = 0x7ff | | mode = = 3 | | layerID = = 3 | | sampleRateID = = 3 ) {
if ( ( ( header > > 21 ) & 0x7ff ) ! = 0x7ff | | mode = = 3 | | layerID = = 3 | |
sampleRateID = = 3 ) {
return - 1 ;
}
if ( ! samplesPerFrame ) samplesPerFrame = & temp0 ;
if ( ! sampleRate ) sampleRate = & temp1 ;
if ( ! samplesPerFrame )
samplesPerFrame = & temp0 ;
if ( ! sampleRate )
sampleRate = & temp1 ;
// *isMono = ((header >> 6) & 0x03) == 0x03;
//*isMono = ((header >> 6) & 0x03) == 0x03;
* sampleRate = sSampleRates [ sampleRateID ] > > mode ;
bitRate = sBitRates [ mpeg_id ] [ layerID ] [ bitRateID ] * 1000 ;
* sampleRate = sSampleRates [ sampleRateID ] > > mode ;
bitRate = sBitRates [ mpeg_id ] [ layerID ] [ bitRateID ] * 1000 ;
* samplesPerFrame = sSamplesPerFrame [ mpeg_id ] [ layerID ] ;
//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
//av_log(NULL, AV_LOG_DEBUG,
// "sr:%d br:%d spf:%d l:%d m:%d\n",
// *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
return * samplesPerFrame * bitRate / ( bitsPerSlot * * sampleRate ) + isPadded ;
}
static int MP3lame_encode_frame ( AVCodecContext * avctx ,
unsigned char * frame , int buf_size , void * data )
static int MP3lame_encode_frame ( AVCodecContext * avctx , unsigned char * frame ,
int buf_size , void * data )
{
Mp3AudioContext * s = avctx - > priv_data ;
int len ;
@ -178,7 +180,7 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
/* lame 3.91 dies on '1-channel interleaved' data */
if ( ! data ) {
if ( ! data ) {
lame_result = lame_encode_flush (
s - > gfp ,
s - > buffer + s - > buffer_index ,
@ -237,32 +239,35 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
}
}
if ( lame_result < 0 ) {
if ( lame_result = = - 1 ) {
if ( lame_result < 0 ) {
if ( lame_result = = - 1 ) {
/* output buffer too small */
av_log ( avctx , AV_LOG_ERROR , " lame: output buffer too small (buffer index: %d, free bytes: %d) \n " , s - > buffer_index , BUFFER_SIZE - s - > buffer_index ) ;
av_log ( avctx , AV_LOG_ERROR ,
" lame: output buffer too small (buffer index: %d, free bytes: %d) \n " ,
s - > buffer_index , BUFFER_SIZE - s - > buffer_index ) ;
}
return - 1 ;
}
s - > buffer_index + = lame_result ;
if ( s - > buffer_index < 4 )
if ( s - > buffer_index < 4 )
return 0 ;
len = mp3len ( s - > buffer , NULL , NULL ) ;
//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
if ( len < = s - > buffer_index ) {
len = mp3len ( s - > buffer , NULL , NULL ) ;
//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n",
// avctx->frame_size, len, s->buffer_index);
if ( len < = s - > buffer_index ) {
memcpy ( frame , s - > buffer , len ) ;
s - > buffer_index - = len ;
memmove ( s - > buffer , s - > buffer + len , s - > buffer_index ) ;
//FIXME fix the audio codec API, so we do not need the memcpy()
/*for(i=0; i<len; i++){
av_log ( avctx , AV_LOG_DEBUG , " %2X " , frame [ i ] ) ;
} */
memmove ( s - > buffer , s - > buffer + len , s - > buffer_index ) ;
// FIXME fix the audio codec API, so we do not need the memcpy()
/*for(i=0; i<len; i++) {
av_log ( avctx , AV_LOG_DEBUG , " %2X " , frame [ i ] ) ;
} */
return len ;
} else
} else
return 0 ;
}
@ -280,7 +285,7 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
# define OFFSET(x) offsetof(Mp3AudioContext, x)
# define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options [ ] = {
{ " reservoir " , " Use bit reservoir. " , OFFSET ( reservoir ) , AV_OPT_TYPE_INT , { 1 } , 0 , 1 , AE } ,
{ " reservoir " , " Use bit reservoir. " , OFFSET ( reservoir ) , AV_OPT_TYPE_INT , { 1 } , 0 , 1 , AE } ,
{ NULL } ,
} ;
@ -292,20 +297,20 @@ static const AVClass libmp3lame_class = {
} ;
AVCodec ff_libmp3lame_encoder = {
. name = " libmp3lame " ,
. type = AVMEDIA_TYPE_AUDIO ,
. id = CODEC_ID_MP3 ,
. priv_data_size = sizeof ( Mp3AudioContext ) ,
. init = MP3lame_encode_init ,
. encode = MP3lame_encode_frame ,
. close = MP3lame_encode_close ,
. capabilities = CODEC_CAP_DELAY ,
. sample_fmts = ( const enum AVSampleFormat [ ] ) { AV_SAMPLE_FMT_S16 ,
. name = " libmp3lame " ,
. type = AVMEDIA_TYPE_AUDIO ,
. id = CODEC_ID_MP3 ,
. priv_data_size = sizeof ( Mp3AudioContext ) ,
. init = MP3lame_encode_init ,
. encode = MP3lame_encode_frame ,
. close = MP3lame_encode_close ,
. capabilities = CODEC_CAP_DELAY ,
. sample_fmts = ( const enum AVSampleFormat [ ] ) { AV_SAMPLE_FMT_S16 ,
# if 2147483647 == INT_MAX
AV_SAMPLE_FMT_S32 ,
# endif
AV_SAMPLE_FMT_NONE } ,
. supported_samplerates = sSampleRates ,
. long_name = NULL_IF_CONFIG_SMALL ( " libmp3lame MP3 (MPEG audio layer 3) " ) ,
. priv_class = & libmp3lame_class ,
AV_SAMPLE_FMT_NONE } ,
. supported_samplerates = sSampleRates ,
. long_name = NULL_IF_CONFIG_SMALL ( " libmp3lame MP3 (MPEG audio layer 3) " ) ,
. priv_class = & libmp3lame_class ,
} ;
Michael Niedermayer
14 years ago