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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 "libavutil/intreadwrite.h"
00028 #include "avcodec.h"
00029 #include "mpegaudio.h"
00030 #include <lame/lame.h>
00031 
00032 #define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
00033 typedef struct Mp3AudioContext {
00034     lame_global_flags *gfp;
00035     int stereo;
00036     uint8_t buffer[BUFFER_SIZE];
00037     int buffer_index;
00038     struct {
00039         int *left;
00040         int *right;
00041     } s32_data;
00042 } Mp3AudioContext;
00043 
00044 static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
00045 {
00046     Mp3AudioContext *s = avctx->priv_data;
00047 
00048     if (avctx->channels > 2)
00049         return -1;
00050 
00051     s->stereo = avctx->channels > 1 ? 1 : 0;
00052 
00053     if ((s->gfp = lame_init()) == NULL)
00054         goto err;
00055     lame_set_in_samplerate(s->gfp, avctx->sample_rate);
00056     lame_set_out_samplerate(s->gfp, avctx->sample_rate);
00057     lame_set_num_channels(s->gfp, avctx->channels);
00058     if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
00059         lame_set_quality(s->gfp, 5);
00060     } else {
00061         lame_set_quality(s->gfp, avctx->compression_level);
00062     }
00063     lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
00064     lame_set_brate(s->gfp, avctx->bit_rate/1000);
00065     if(avctx->flags & CODEC_FLAG_QSCALE) {
00066         lame_set_brate(s->gfp, 0);
00067         lame_set_VBR(s->gfp, vbr_default);
00068         lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
00069     }
00070     lame_set_bWriteVbrTag(s->gfp,0);
00071     lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
00072     if (lame_init_params(s->gfp) < 0)
00073         goto err_close;
00074 
00075     avctx->frame_size = lame_get_framesize(s->gfp);
00076 
00077     if(!(avctx->coded_frame= avcodec_alloc_frame())) {
00078         lame_close(s->gfp);
00079 
00080         return AVERROR(ENOMEM);
00081     }
00082     avctx->coded_frame->key_frame= 1;
00083 
00084     if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
00085         int nelem = 2 * avctx->frame_size;
00086 
00087         if(! (s->s32_data.left = av_malloc(nelem * sizeof(int)))) {
00088             av_freep(&avctx->coded_frame);
00089             lame_close(s->gfp);
00090 
00091             return AVERROR(ENOMEM);
00092         }
00093 
00094         s->s32_data.right = s->s32_data.left + avctx->frame_size;
00095     }
00096 
00097     return 0;
00098 
00099 err_close:
00100     lame_close(s->gfp);
00101 err:
00102     return -1;
00103 }
00104 
00105 static const int sSampleRates[] = {
00106     44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
00107 };
00108 
00109 static const int sBitRates[2][3][15] = {
00110     {   {  0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
00111         {  0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
00112         {  0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
00113     },
00114     {   {  0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
00115         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
00116         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
00117     },
00118 };
00119 
00120 static const int sSamplesPerFrame[2][3] =
00121 {
00122     {  384,     1152,    1152 },
00123     {  384,     1152,     576 }
00124 };
00125 
00126 static const int sBitsPerSlot[3] = {
00127     32,
00128     8,
00129     8
00130 };
00131 
00132 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
00133 {
00134     uint32_t header = AV_RB32(data);
00135     int layerID = 3 - ((header >> 17) & 0x03);
00136     int bitRateID = ((header >> 12) & 0x0f);
00137     int sampleRateID = ((header >> 10) & 0x03);
00138     int bitsPerSlot = sBitsPerSlot[layerID];
00139     int isPadded = ((header >> 9) & 0x01);
00140     static int const mode_tab[4]= {2,3,1,0};
00141     int mode= mode_tab[(header >> 19) & 0x03];
00142     int mpeg_id= mode>0;
00143     int temp0, temp1, bitRate;
00144 
00145     if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
00146         return -1;
00147     }
00148 
00149     if(!samplesPerFrame) samplesPerFrame= &temp0;
00150     if(!sampleRate     ) sampleRate     = &temp1;
00151 
00152 //    *isMono = ((header >>  6) & 0x03) == 0x03;
00153 
00154     *sampleRate = sSampleRates[sampleRateID]>>mode;
00155     bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
00156     *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
00157 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
00158 
00159     return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
00160 }
00161 
00162 static int MP3lame_encode_frame(AVCodecContext *avctx,
00163                                 unsigned char *frame, int buf_size, void *data)
00164 {
00165     Mp3AudioContext *s = avctx->priv_data;
00166     int len;
00167     int lame_result;
00168 
00169     /* lame 3.91 dies on '1-channel interleaved' data */
00170 
00171     if(!data){
00172         lame_result= lame_encode_flush(
00173                 s->gfp,
00174                 s->buffer + s->buffer_index,
00175                 BUFFER_SIZE - s->buffer_index
00176                 );
00177 #if 2147483647 == INT_MAX
00178     }else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){
00179         if (s->stereo) {
00180             int32_t *rp = data;
00181             int32_t *mp = rp + 2*avctx->frame_size;
00182             int *wpl = s->s32_data.left;
00183             int *wpr = s->s32_data.right;
00184 
00185             while (rp < mp) {
00186                 *wpl++ = *rp++;
00187                 *wpr++ = *rp++;
00188             }
00189 
00190             lame_result = lame_encode_buffer_int(
00191                 s->gfp,
00192                 s->s32_data.left,
00193                 s->s32_data.right,
00194                 avctx->frame_size,
00195                 s->buffer + s->buffer_index,
00196                 BUFFER_SIZE - s->buffer_index
00197                 );
00198         } else {
00199             lame_result = lame_encode_buffer_int(
00200                 s->gfp,
00201                 data,
00202                 data,
00203                 avctx->frame_size,
00204                 s->buffer + s->buffer_index,
00205                 BUFFER_SIZE - s->buffer_index
00206                 );
00207         }
00208 #endif
00209     }else{
00210         if (s->stereo) {
00211             lame_result = lame_encode_buffer_interleaved(
00212                 s->gfp,
00213                 data,
00214                 avctx->frame_size,
00215                 s->buffer + s->buffer_index,
00216                 BUFFER_SIZE - s->buffer_index
00217                 );
00218         } else {
00219             lame_result = lame_encode_buffer(
00220                 s->gfp,
00221                 data,
00222                 data,
00223                 avctx->frame_size,
00224                 s->buffer + s->buffer_index,
00225                 BUFFER_SIZE - s->buffer_index
00226                 );
00227         }
00228     }
00229 
00230     if(lame_result < 0){
00231         if(lame_result==-1) {
00232             /* output buffer too small */
00233             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);
00234         }
00235         return -1;
00236     }
00237 
00238     s->buffer_index += lame_result;
00239 
00240     if(s->buffer_index<4)
00241         return 0;
00242 
00243     len= mp3len(s->buffer, NULL, NULL);
00244 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
00245     if(len <= s->buffer_index){
00246         memcpy(frame, s->buffer, len);
00247         s->buffer_index -= len;
00248 
00249         memmove(s->buffer, s->buffer+len, s->buffer_index);
00250             //FIXME fix the audio codec API, so we do not need the memcpy()
00251 /*for(i=0; i<len; i++){
00252     av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
00253 }*/
00254         return len;
00255     }else
00256         return 0;
00257 }
00258 
00259 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
00260 {
00261     Mp3AudioContext *s = avctx->priv_data;
00262 
00263     av_freep(&s->s32_data.left);
00264     av_freep(&avctx->coded_frame);
00265 
00266     lame_close(s->gfp);
00267     return 0;
00268 }
00269 
00270 
00271 AVCodec ff_libmp3lame_encoder = {
00272     "libmp3lame",
00273     AVMEDIA_TYPE_AUDIO,
00274     CODEC_ID_MP3,
00275     sizeof(Mp3AudioContext),
00276     MP3lame_encode_init,
00277     MP3lame_encode_frame,
00278     MP3lame_encode_close,
00279     .capabilities= CODEC_CAP_DELAY,
00280     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
00281 #if 2147483647 == INT_MAX
00282     AV_SAMPLE_FMT_S32,
00283 #endif
00284     AV_SAMPLE_FMT_NONE},
00285     .supported_samplerates= sSampleRates,
00286     .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
00287 };

Generated on Wed Apr 11 2012 07:31:33 for FFmpeg by  doxygen 1.7.1