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/*
* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/**
* @file mp3lameaudio.c
* Interface to libmp3lame for mp3 encoding.
*/
#include "avcodec.h"
#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;
if (avctx->channels > 2)
return -1;
s->stereo = avctx->channels > 1 ? 1 : 0;
if ((s->gfp = lame_init()) == NULL)
goto err;
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);
/* lame 3.91 dies on quality != 5 */
lame_set_quality(s->gfp, 5);
/* lame 3.91 doesn't work in mono */
lame_set_mode(s->gfp, JOINT_STEREO);
lame_set_brate(s->gfp, avctx->bit_rate/1000);
lame_set_bWriteVbrTag(s->gfp,0);
if (lame_init_params(s->gfp) < 0)
goto err_close;
avctx->frame_size = lame_get_framesize(s->gfp);
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
err_close:
lame_close(s->gfp);
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 len, i;
int lame_result;
/* lame 3.91 dies on '1-channel interleaved' data */
if(data){
if (s->stereo) {
lame_result = lame_encode_buffer_interleaved(
s->gfp,
data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
} else {
lame_result = lame_encode_buffer(
s->gfp,
data,
data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
}
}else{
lame_result= lame_encode_flush(
s->gfp,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
}
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);
return 0;
}
s->buffer_index += lame_result;
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){
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()
/*for(i=0; i<len; i++){
av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
}*/
return len;
}else
return 0;
}
int MP3lame_encode_close(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame);
lame_close(s->gfp);
return 0;
}
AVCodec mp3lame_encoder = {
"mp3",
CODEC_TYPE_AUDIO,
CODEC_ID_MP3,
sizeof(Mp3AudioContext),
MP3lame_encode_init,
MP3lame_encode_frame,
MP3lame_encode_close,
.capabilities= CODEC_CAP_DELAY,
};
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