mirror of
https://github.com/FFmpeg/FFmpeg.git
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215 lines
6.2 KiB
C
215 lines
6.2 KiB
C
/*
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* Interface to libmp3lame for mp3 encoding
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* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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/**
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* @file mp3lameaudio.c
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* Interface to libmp3lame for mp3 encoding.
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*/
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#include "avcodec.h"
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#include "mpegaudio.h"
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#include <lame/lame.h>
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#define BUFFER_SIZE (2*MPA_FRAME_SIZE)
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typedef struct Mp3AudioContext {
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lame_global_flags *gfp;
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int stereo;
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uint8_t buffer[BUFFER_SIZE];
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int buffer_index;
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} Mp3AudioContext;
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static int MP3lame_encode_init(AVCodecContext *avctx)
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{
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Mp3AudioContext *s = avctx->priv_data;
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if (avctx->channels > 2)
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return -1;
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s->stereo = avctx->channels > 1 ? 1 : 0;
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if ((s->gfp = lame_init()) == NULL)
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goto err;
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lame_set_in_samplerate(s->gfp, avctx->sample_rate);
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lame_set_out_samplerate(s->gfp, avctx->sample_rate);
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lame_set_num_channels(s->gfp, avctx->channels);
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/* lame 3.91 dies on quality != 5 */
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lame_set_quality(s->gfp, 5);
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/* lame 3.91 doesn't work in mono */
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lame_set_mode(s->gfp, JOINT_STEREO);
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lame_set_brate(s->gfp, avctx->bit_rate/1000);
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lame_set_bWriteVbrTag(s->gfp,0);
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if (lame_init_params(s->gfp) < 0)
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goto err_close;
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avctx->frame_size = lame_get_framesize(s->gfp);
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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return 0;
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err_close:
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lame_close(s->gfp);
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err:
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return -1;
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}
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static const int sSampleRates[3] = {
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44100, 48000, 32000
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};
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static const int sBitRates[2][3][15] = {
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{ { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
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{ 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
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{ 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
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},
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{ { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
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{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
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{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
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},
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};
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static const int sSamplesPerFrame[2][3] =
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{
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{ 384, 1152, 1152 },
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{ 384, 1152, 576 }
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};
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static const int sBitsPerSlot[3] = {
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32,
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8,
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8
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};
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static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
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{
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uint8_t *dataTmp = (uint8_t *)data;
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uint32_t header = ( (uint32_t)dataTmp[0] << 24 ) | ( (uint32_t)dataTmp[1] << 16 ) | ( (uint32_t)dataTmp[2] << 8 ) | (uint32_t)dataTmp[3];
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int layerID = 3 - ((header >> 17) & 0x03);
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int bitRateID = ((header >> 12) & 0x0f);
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int sampleRateID = ((header >> 10) & 0x03);
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int bitsPerSlot = sBitsPerSlot[layerID];
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int isPadded = ((header >> 9) & 0x01);
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static int const mode_tab[4]= {2,3,1,0};
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int mode= mode_tab[(header >> 19) & 0x03];
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int mpeg_id= mode>0;
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int temp0, temp1, bitRate;
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if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
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return -1;
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}
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if(!samplesPerFrame) samplesPerFrame= &temp0;
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if(!sampleRate ) sampleRate = &temp1;
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// *isMono = ((header >> 6) & 0x03) == 0x03;
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*sampleRate = sSampleRates[sampleRateID]>>mode;
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bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
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*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
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//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
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return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
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}
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int MP3lame_encode_frame(AVCodecContext *avctx,
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unsigned char *frame, int buf_size, void *data)
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{
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Mp3AudioContext *s = avctx->priv_data;
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int len;
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int lame_result;
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/* lame 3.91 dies on '1-channel interleaved' data */
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if(data){
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if (s->stereo) {
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lame_result = lame_encode_buffer_interleaved(
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s->gfp,
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data,
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avctx->frame_size,
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s->buffer + s->buffer_index,
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BUFFER_SIZE - s->buffer_index
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);
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} else {
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lame_result = lame_encode_buffer(
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s->gfp,
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data,
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data,
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avctx->frame_size,
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s->buffer + s->buffer_index,
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BUFFER_SIZE - s->buffer_index
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);
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}
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}else{
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lame_result= lame_encode_flush(
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s->gfp,
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s->buffer + s->buffer_index,
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BUFFER_SIZE - s->buffer_index
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);
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}
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if(lame_result==-1) {
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/* output buffer too small */
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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);
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return 0;
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}
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s->buffer_index += lame_result;
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if(s->buffer_index<4)
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return 0;
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len= mp3len(s->buffer, NULL, NULL);
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//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
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if(len <= s->buffer_index){
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memcpy(frame, s->buffer, len);
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s->buffer_index -= len;
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memmove(s->buffer, s->buffer+len, s->buffer_index);
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//FIXME fix the audio codec API, so we dont need the memcpy()
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/*for(i=0; i<len; i++){
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av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
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}*/
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return len;
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}else
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return 0;
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}
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int MP3lame_encode_close(AVCodecContext *avctx)
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{
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Mp3AudioContext *s = avctx->priv_data;
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av_freep(&avctx->coded_frame);
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lame_close(s->gfp);
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return 0;
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}
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AVCodec mp3lame_encoder = {
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"mp3",
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CODEC_TYPE_AUDIO,
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CODEC_ID_MP3,
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sizeof(Mp3AudioContext),
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MP3lame_encode_init,
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MP3lame_encode_frame,
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MP3lame_encode_close,
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.capabilities= CODEC_CAP_DELAY,
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};
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