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libavcodec/libmp3lame.c

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00001 /*
00002  * Interface to libmp3lame for mp3 encoding
00003  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
00004  *
00005  * This file is part of FFmpeg.
00006  *
00007  * FFmpeg is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * FFmpeg is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with FFmpeg; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00027 #include "avcodec.h"
00028 #include "mpegaudio.h"
00029 #include <lame/lame.h>
00030 
00031 #define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
00032 typedef struct Mp3AudioContext {
00033     lame_global_flags *gfp;
00034     int stereo;
00035     uint8_t buffer[BUFFER_SIZE];
00036     int buffer_index;
00037 } Mp3AudioContext;
00038 
00039 static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
00040 {
00041     Mp3AudioContext *s = avctx->priv_data;
00042 
00043     if (avctx->channels > 2)
00044         return -1;
00045 
00046     s->stereo = avctx->channels > 1 ? 1 : 0;
00047 
00048     if ((s->gfp = lame_init()) == NULL)
00049         goto err;
00050     lame_set_in_samplerate(s->gfp, avctx->sample_rate);
00051     lame_set_out_samplerate(s->gfp, avctx->sample_rate);
00052     lame_set_num_channels(s->gfp, avctx->channels);
00053     if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
00054         lame_set_quality(s->gfp, 5);
00055     } else {
00056         lame_set_quality(s->gfp, avctx->compression_level);
00057     }
00058     /* lame 3.91 doesn't work in mono */
00059     lame_set_mode(s->gfp, JOINT_STEREO);
00060     lame_set_brate(s->gfp, avctx->bit_rate/1000);
00061     if(avctx->flags & CODEC_FLAG_QSCALE) {
00062         lame_set_brate(s->gfp, 0);
00063         lame_set_VBR(s->gfp, vbr_default);
00064         lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
00065     }
00066     lame_set_bWriteVbrTag(s->gfp,0);
00067     lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
00068     if (lame_init_params(s->gfp) < 0)
00069         goto err_close;
00070 
00071     avctx->frame_size = lame_get_framesize(s->gfp);
00072 
00073     avctx->coded_frame= avcodec_alloc_frame();
00074     avctx->coded_frame->key_frame= 1;
00075 
00076     return 0;
00077 
00078 err_close:
00079     lame_close(s->gfp);
00080 err:
00081     return -1;
00082 }
00083 
00084 static const int sSampleRates[] = {
00085     44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
00086 };
00087 
00088 static const int sBitRates[2][3][15] = {
00089     {   {  0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
00090         {  0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
00091         {  0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
00092     },
00093     {   {  0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
00094         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
00095         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
00096     },
00097 };
00098 
00099 static const int sSamplesPerFrame[2][3] =
00100 {
00101     {  384,     1152,    1152 },
00102     {  384,     1152,     576 }
00103 };
00104 
00105 static const int sBitsPerSlot[3] = {
00106     32,
00107     8,
00108     8
00109 };
00110 
00111 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
00112 {
00113     uint32_t header = AV_RB32(data);
00114     int layerID = 3 - ((header >> 17) & 0x03);
00115     int bitRateID = ((header >> 12) & 0x0f);
00116     int sampleRateID = ((header >> 10) & 0x03);
00117     int bitsPerSlot = sBitsPerSlot[layerID];
00118     int isPadded = ((header >> 9) & 0x01);
00119     static int const mode_tab[4]= {2,3,1,0};
00120     int mode= mode_tab[(header >> 19) & 0x03];
00121     int mpeg_id= mode>0;
00122     int temp0, temp1, bitRate;
00123 
00124     if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
00125         return -1;
00126     }
00127 
00128     if(!samplesPerFrame) samplesPerFrame= &temp0;
00129     if(!sampleRate     ) sampleRate     = &temp1;
00130 
00131 //    *isMono = ((header >>  6) & 0x03) == 0x03;
00132 
00133     *sampleRate = sSampleRates[sampleRateID]>>mode;
00134     bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
00135     *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
00136 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
00137 
00138     return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
00139 }
00140 
00141 static int MP3lame_encode_frame(AVCodecContext *avctx,
00142                                 unsigned char *frame, int buf_size, void *data)
00143 {
00144     Mp3AudioContext *s = avctx->priv_data;
00145     int len;
00146     int lame_result;
00147 
00148     /* lame 3.91 dies on '1-channel interleaved' data */
00149 
00150     if(data){
00151         if (s->stereo) {
00152             lame_result = lame_encode_buffer_interleaved(
00153                 s->gfp,
00154                 data,
00155                 avctx->frame_size,
00156                 s->buffer + s->buffer_index,
00157                 BUFFER_SIZE - s->buffer_index
00158                 );
00159         } else {
00160             lame_result = lame_encode_buffer(
00161                 s->gfp,
00162                 data,
00163                 data,
00164                 avctx->frame_size,
00165                 s->buffer + s->buffer_index,
00166                 BUFFER_SIZE - s->buffer_index
00167                 );
00168         }
00169     }else{
00170         lame_result= lame_encode_flush(
00171                 s->gfp,
00172                 s->buffer + s->buffer_index,
00173                 BUFFER_SIZE - s->buffer_index
00174                 );
00175     }
00176 
00177     if(lame_result < 0){
00178         if(lame_result==-1) {
00179             /* output buffer too small */
00180             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);
00181         }
00182         return -1;
00183     }
00184 
00185     s->buffer_index += lame_result;
00186 
00187     if(s->buffer_index<4)
00188         return 0;
00189 
00190     len= mp3len(s->buffer, NULL, NULL);
00191 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
00192     if(len <= s->buffer_index){
00193         memcpy(frame, s->buffer, len);
00194         s->buffer_index -= len;
00195 
00196         memmove(s->buffer, s->buffer+len, s->buffer_index);
00197             //FIXME fix the audio codec API, so we do not need the memcpy()
00198 /*for(i=0; i<len; i++){
00199     av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
00200 }*/
00201         return len;
00202     }else
00203         return 0;
00204 }
00205 
00206 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
00207 {
00208     Mp3AudioContext *s = avctx->priv_data;
00209 
00210     av_freep(&avctx->coded_frame);
00211 
00212     lame_close(s->gfp);
00213     return 0;
00214 }
00215 
00216 
00217 AVCodec libmp3lame_encoder = {
00218     "libmp3lame",
00219     AVMEDIA_TYPE_AUDIO,
00220     CODEC_ID_MP3,
00221     sizeof(Mp3AudioContext),
00222     MP3lame_encode_init,
00223     MP3lame_encode_frame,
00224     MP3lame_encode_close,
00225     .capabilities= CODEC_CAP_DELAY,
00226     .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
00227     .supported_samplerates= sSampleRates,
00228     .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
00229 };

Generated on Fri Sep 16 2011 17:17:38 for FFmpeg by  doxygen 1.7.1