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 mp3lameaudio.c
  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 + 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     /* lame 3.91 dies on quality != 5 */
  54     lame_set_quality(s->gfp, 5);
  55     /* lame 3.91 doesn't work in mono */
  56     lame_set_mode(s->gfp, JOINT_STEREO);
  57     lame_set_brate(s->gfp, avctx->bit_rate/1000);
  58     if(avctx->flags & CODEC_FLAG_QSCALE) {
  59         lame_set_brate(s->gfp, 0);
  60         lame_set_VBR(s->gfp, vbr_default);
  61         lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
  62     }
  63     lame_set_bWriteVbrTag(s->gfp,0);
  64     lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
  65     if (lame_init_params(s->gfp) < 0)
  66         goto err_close;
  67
  68     avctx->frame_size = lame_get_framesize(s->gfp);
  69
  70     avctx->coded_frame= avcodec_alloc_frame();
  71     avctx->coded_frame->key_frame= 1;
  72
  73     return 0;
  74
  75 err_close:
  76     lame_close(s->gfp);
  77 err:
  78     return -1;
  79 }
  80
  81 static const int sSampleRates[3] = {
  82     44100, 48000,  32000
  83 };
  84
  85 static const int sBitRates[2][3][15] = {
  86     {   {  0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
  87         {  0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
  88         {  0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
  89     },
  90     {   {  0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
  91         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
  92         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
  93     },
  94 };
  95
  96 static const int sSamplesPerFrame[2][3] =
  97 {
  98     {  384,     1152,    1152 },
  99     {  384,     1152,     576 }
 100 };
 101
 102 static const int sBitsPerSlot[3] = {
 103     32,
 104     8,
 105     8
 106 };
 107
 108 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
 109 {
 110     uint32_t header = AV_RB32(data);
 111     int layerID = 3 - ((header >> 17) & 0x03);
 112     int bitRateID = ((header >> 12) & 0x0f);
 113     int sampleRateID = ((header >> 10) & 0x03);
 114     int bitsPerSlot = sBitsPerSlot[layerID];
 115     int isPadded = ((header >> 9) & 0x01);
 116     static int const mode_tab[4]= {2,3,1,0};
 117     int mode= mode_tab[(header >> 19) & 0x03];
 118     int mpeg_id= mode>0;
 119     int temp0, temp1, bitRate;
 120
 121     if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
 122         return -1;
 123     }
 124
 125     if(!samplesPerFrame) samplesPerFrame= &temp0;
 126     if(!sampleRate     ) sampleRate     = &temp1;
 127
 128 //    *isMono = ((header >>  6) & 0x03) == 0x03;
 129
 130     *sampleRate = sSampleRates[sampleRateID]>>mode;
 131     bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
 132     *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
 133 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
 134
 135     return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
 136 }
 137
 138 static int MP3lame_encode_frame(AVCodecContext *avctx,
 139                                 unsigned char *frame, int buf_size, void *data)
 140 {
 141     Mp3AudioContext *s = avctx->priv_data;
 142     int len;
 143     int lame_result;
 144
 145     /* lame 3.91 dies on '1-channel interleaved' data */
 146
 147     if(data){
 148         if (s->stereo) {
 149             lame_result = lame_encode_buffer_interleaved(
 150                 s->gfp,
 151                 data,
 152                 avctx->frame_size,
 153                 s->buffer + s->buffer_index,
 154                 BUFFER_SIZE - s->buffer_index
 155                 );
 156         } else {
 157             lame_result = lame_encode_buffer(
 158                 s->gfp,
 159                 data,
 160                 data,
 161                 avctx->frame_size,
 162                 s->buffer + s->buffer_index,
 163                 BUFFER_SIZE - s->buffer_index
 164                 );
 165         }
 166     }else{
 167         lame_result= lame_encode_flush(
 168                 s->gfp,
 169                 s->buffer + s->buffer_index,
 170                 BUFFER_SIZE - s->buffer_index
 171                 );
 172     }
 173
 174     if(lame_result==-1) {
 175         /* output buffer too small */
 176         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);
 177         return 0;
 178     }
 179
 180     s->buffer_index += lame_result;
 181
 182     if(s->buffer_index<4)
 183         return 0;
 184
 185         len= mp3len(s->buffer, NULL, NULL);
 186 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
 187         if(len <= s->buffer_index){
 188             memcpy(frame, s->buffer, len);
 189             s->buffer_index -= len;
 190
 191             memmove(s->buffer, s->buffer+len, s->buffer_index);
 192             //FIXME fix the audio codec API, so we do not need the memcpy()
 193 /*for(i=0; i<len; i++){
 194     av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
 195 }*/
 196             return len;
 197         }else
 198             return 0;
 199 }
 200
 201 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
 202 {
 203     Mp3AudioContext *s = avctx->priv_data;
 204
 205     av_freep(&avctx->coded_frame);
 206
 207     lame_close(s->gfp);
 208     return 0;
 209 }
 210
 211
 212 AVCodec libmp3lame_encoder = {
 213     "libmp3lame",
 214     CODEC_TYPE_AUDIO,
 215     CODEC_ID_MP3,
 216     sizeof(Mp3AudioContext),
 217     MP3lame_encode_init,
 218     MP3lame_encode_frame,
 219     MP3lame_encode_close,
 220     .capabilities= CODEC_CAP_DELAY,
 221     .long_name= "libmp3lame MP3 (MPEG audio layer 3)",
 222 };