libavcodec/libmp3lame.c

   1 /*
   2  * Interface to libmp3lame for mp3 encoding
   3  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
   4  *
   5  * This file is part of FFmpeg.
   6  *
   7  * FFmpeg is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * FFmpeg is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with FFmpeg; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 /**
  23  * @file
  24  * Interface to libmp3lame for mp3 encoding.
  25  */
  26
  27 #include "avcodec.h"
  28 #include "mpegaudio.h"
  29 #include <lame/lame.h>
  30
  31 #define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
  32 typedef struct Mp3AudioContext {
  33     lame_global_flags *gfp;
  34     int stereo;
  35     uint8_t buffer[BUFFER_SIZE];
  36     int buffer_index;
  37 } Mp3AudioContext;
  38
  39 static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
  40 {
  41     Mp3AudioContext *s = avctx->priv_data;
  42
  43     if (avctx->channels > 2)
  44         return -1;
  45
  46     s->stereo = avctx->channels > 1 ? 1 : 0;
  47
  48     if ((s->gfp = lame_init()) == NULL)
  49         goto err;
  50     lame_set_in_samplerate(s->gfp, avctx->sample_rate);
  51     lame_set_out_samplerate(s->gfp, avctx->sample_rate);
  52     lame_set_num_channels(s->gfp, avctx->channels);
  53     if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
  54         lame_set_quality(s->gfp, 5);
  55     } else {
  56         lame_set_quality(s->gfp, avctx->compression_level);
  57     }
  58     lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
  59     lame_set_brate(s->gfp, avctx->bit_rate/1000);
  60     if(avctx->flags & CODEC_FLAG_QSCALE) {
  61         lame_set_brate(s->gfp, 0);
  62         lame_set_VBR(s->gfp, vbr_default);
  63         lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
  64     }
  65     lame_set_bWriteVbrTag(s->gfp,0);
  66     lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
  67     if (lame_init_params(s->gfp) < 0)
  68         goto err_close;
  69
  70     avctx->frame_size = lame_get_framesize(s->gfp);
  71
  72     avctx->coded_frame= avcodec_alloc_frame();
  73     avctx->coded_frame->key_frame= 1;
  74
  75     return 0;
  76
  77 err_close:
  78     lame_close(s->gfp);
  79 err:
  80     return -1;
  81 }
  82
  83 static const int sSampleRates[] = {
  84     44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
  85 };
  86
  87 static const int sBitRates[2][3][15] = {
  88     {   {  0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
  89         {  0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
  90         {  0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
  91     },
  92     {   {  0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
  93         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
  94         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
  95     },
  96 };
  97
  98 static const int sSamplesPerFrame[2][3] =
  99 {
 100     {  384,     1152,    1152 },
 101     {  384,     1152,     576 }
 102 };
 103
 104 static const int sBitsPerSlot[3] = {
 105     32,
 106     8,
 107     8
 108 };
 109
 110 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
 111 {
 112     uint32_t header = AV_RB32(data);
 113     int layerID = 3 - ((header >> 17) & 0x03);
 114     int bitRateID = ((header >> 12) & 0x0f);
 115     int sampleRateID = ((header >> 10) & 0x03);
 116     int bitsPerSlot = sBitsPerSlot[layerID];
 117     int isPadded = ((header >> 9) & 0x01);
 118     static int const mode_tab[4]= {2,3,1,0};
 119     int mode= mode_tab[(header >> 19) & 0x03];
 120     int mpeg_id= mode>0;
 121     int temp0, temp1, bitRate;
 122
 123     if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
 124         return -1;
 125     }
 126
 127     if(!samplesPerFrame) samplesPerFrame= &temp0;
 128     if(!sampleRate     ) sampleRate     = &temp1;
 129
 130 //    *isMono = ((header >>  6) & 0x03) == 0x03;
 131
 132     *sampleRate = sSampleRates[sampleRateID]>>mode;
 133     bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
 134     *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
 135 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
 136
 137     return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
 138 }
 139
 140 static int MP3lame_encode_frame(AVCodecContext *avctx,
 141                                 unsigned char *frame, int buf_size, void *data)
 142 {
 143     Mp3AudioContext *s = avctx->priv_data;
 144     int len;
 145     int lame_result;
 146
 147     /* lame 3.91 dies on '1-channel interleaved' data */
 148
 149     if(data){
 150         if (s->stereo) {
 151             lame_result = lame_encode_buffer_interleaved(
 152                 s->gfp,
 153                 data,
 154                 avctx->frame_size,
 155                 s->buffer + s->buffer_index,
 156                 BUFFER_SIZE - s->buffer_index
 157                 );
 158         } else {
 159             lame_result = lame_encode_buffer(
 160                 s->gfp,
 161                 data,
 162                 data,
 163                 avctx->frame_size,
 164                 s->buffer + s->buffer_index,
 165                 BUFFER_SIZE - s->buffer_index
 166                 );
 167         }
 168     }else{
 169         lame_result= lame_encode_flush(
 170                 s->gfp,
 171                 s->buffer + s->buffer_index,
 172                 BUFFER_SIZE - s->buffer_index
 173                 );
 174     }
 175
 176     if(lame_result < 0){
 177         if(lame_result==-1) {
 178             /* output buffer too small */
 179             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);
 180         }
 181         return -1;
 182     }
 183
 184     s->buffer_index += lame_result;
 185
 186     if(s->buffer_index<4)
 187         return 0;
 188
 189     len= mp3len(s->buffer, NULL, NULL);
 190 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
 191     if(len <= s->buffer_index){
 192         memcpy(frame, s->buffer, len);
 193         s->buffer_index -= len;
 194
 195         memmove(s->buffer, s->buffer+len, s->buffer_index);
 196             //FIXME fix the audio codec API, so we do not need the memcpy()
 197 /*for(i=0; i<len; i++){
 198     av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
 199 }*/
 200         return len;
 201     }else
 202         return 0;
 203 }
 204
 205 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
 206 {
 207     Mp3AudioContext *s = avctx->priv_data;
 208
 209     av_freep(&avctx->coded_frame);
 210
 211     lame_close(s->gfp);
 212     return 0;
 213 }
 214
 215
 216 AVCodec libmp3lame_encoder = {
 217     "libmp3lame",
 218     AVMEDIA_TYPE_AUDIO,
 219     CODEC_ID_MP3,
 220     sizeof(Mp3AudioContext),
 221     MP3lame_encode_init,
 222     MP3lame_encode_frame,
 223     MP3lame_encode_close,
 224     .capabilities= CODEC_CAP_DELAY,
 225     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
 226     .supported_samplerates= sSampleRates,
 227     .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
 228 };