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split stream into valid mp3 frames, at least flv & nut absolutely need this, but probably most other formats too

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Originally committed as revision 2942 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer 2004-04-01 17:07:06 +00:00
parent cac0a56c55
commit 2f996b8397
1 changed files with 93 additions and 14 deletions
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107
libavcodec/mp3lameaudio.c
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@ -26,12 +26,14 @@
#include "mpegaudio.h"
#include <lame/lame.h>
#define BUFFER_SIZE (2*MPA_FRAME_SIZE)
typedef struct Mp3AudioContext {
lame_global_flags *gfp;
int stereo;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
} Mp3AudioContext;
static int MP3lame_encode_init(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
@ -68,30 +70,107 @@ err:
return -1;
}
static const int sSampleRates[3] = {
44100, 48000, 32000
};
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, 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 sBitsPerSlot[3] = {
32,
8,
8
};
static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
{
uint8_t *dataTmp = (uint8_t *)data;
uint32_t header = ( (uint32_t)dataTmp[0] << 24 ) | ( (uint32_t)dataTmp[1] << 16 ) | ( (uint32_t)dataTmp[2] << 8 ) | (uint32_t)dataTmp[3];
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 temp0, temp1, bitRate;
if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
return -1;
}
if(!samplesPerFrame) samplesPerFrame= &temp0;
if(!sampleRate ) sampleRate = &temp1;
// *isMono = ((header >> 6) & 0x03) == 0x03;
*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);
return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
}
int MP3lame_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
Mp3AudioContext *s = avctx->priv_data;
int num, i;
//av_log(avctx, AV_LOG_DEBUG, "%X %d %X\n", (int)frame, buf_size, (int)data);
// if(data==NULL)
// return lame_encode_flush(s->gfp, frame, buf_size);
int len, i;
/* lame 3.91 dies on '1-channel interleaved' data */
if (s->stereo) {
num = lame_encode_buffer_interleaved(s->gfp, data,
MPA_FRAME_SIZE, frame, buf_size);
s->buffer_index += lame_encode_buffer_interleaved(
s->gfp,
data,
MPA_FRAME_SIZE,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
} else {
num = lame_encode_buffer(s->gfp, data, data, MPA_FRAME_SIZE,
frame, buf_size);
s->buffer_index += lame_encode_buffer(
s->gfp,
data,
data,
MPA_FRAME_SIZE,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
}
if(s->buffer_index<4)
return 0;
/*av_log(avctx, AV_LOG_DEBUG, "in:%d out:%d\n", MPA_FRAME_SIZE, num);
for(i=0; i<num; i++){
len= mp3len(s->buffer, NULL, NULL);
//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", MPA_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 dont need the memcpy()
//FIXME fix the audio codec API, so we can output multiple packets if we have them
/*for(i=0; i<len; i++){
av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
}*/
}
return num;
return len;
}else
return 0;
}
int MP3lame_encode_close(AVCodecContext *avctx)