quicktime/ffmpeg/libavcodec/mp3lameaudio.c

   1 /*
   2  * Interface to libmp3lame for mp3 encoding
   3  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
   4  *
   5  * This library is free software; you can redistribute it and/or
   6  * modify it under the terms of the GNU Lesser General Public
   7  * License as published by the Free Software Foundation; either
   8  * version 2 of the License, or (at your option) any later version.
   9  *
  10  * This library is distributed in the hope that it will be useful,
  11  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  13  * Lesser General Public License for more details.
  14  *
  15  * You should have received a copy of the GNU Lesser General Public
  16  * License along with this library; if not, write to the Free Software
  17  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  18  */
  19
  20 /**
  21  * @file mp3lameaudio.c
  22  * Interface to libmp3lame for mp3 encoding.
  23  */
  24
  25 #include "avcodec.h"
  26 #include "mpegaudio.h"
  27 #include <lame/lame.h>
  28
  29 #define BUFFER_SIZE (2*MPA_FRAME_SIZE)
  30 typedef struct Mp3AudioContext {
  31         lame_global_flags *gfp;
  32         int stereo;
  33         uint8_t buffer[BUFFER_SIZE];
  34         int buffer_index;
  35 } Mp3AudioContext;
  36
  37 static int MP3lame_encode_init(AVCodecContext *avctx)
  38 {
  39         Mp3AudioContext *s = avctx->priv_data;
  40
  41         if (avctx->channels > 2)
  42                 return -1;
  43
  44         s->stereo = avctx->channels > 1 ? 1 : 0;
  45
  46         if ((s->gfp = lame_init()) == NULL)
  47                 goto err;
  48         lame_set_in_samplerate(s->gfp, avctx->sample_rate);
  49         lame_set_out_samplerate(s->gfp, avctx->sample_rate);
  50         lame_set_num_channels(s->gfp, avctx->channels);
  51         /* lame 3.91 dies on quality != 5 */
  52         lame_set_quality(s->gfp, 5);
  53         /* lame 3.91 doesn't work in mono */
  54         lame_set_mode(s->gfp, JOINT_STEREO);
  55         lame_set_brate(s->gfp, avctx->bit_rate/1000);
  56         lame_set_bWriteVbrTag(s->gfp,0);
  57         if (lame_init_params(s->gfp) < 0)
  58                 goto err_close;
  59
  60         avctx->frame_size = lame_get_framesize(s->gfp);
  61
  62         avctx->coded_frame= avcodec_alloc_frame();
  63         avctx->coded_frame->key_frame= 1;
  64
  65         return 0;
  66
  67 err_close:
  68         lame_close(s->gfp);
  69 err:
  70         return -1;
  71 }
  72
  73 static const int sSampleRates[3] = {
  74     44100, 48000,  32000
  75 };
  76
  77 static const int sBitRates[2][3][15] = {
  78     {   {  0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
  79         {  0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
  80         {  0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
  81     },
  82     {   {  0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
  83         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
  84         {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
  85     },
  86 };
  87
  88 static const int sSamplesPerFrame[2][3] =
  89 {
  90     {  384,     1152,    1152 },
  91     {  384,     1152,     576 }
  92 };
  93
  94 static const int sBitsPerSlot[3] = {
  95     32,
  96     8,
  97     8
  98 };
  99
 100 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
 101 {
 102     uint8_t *dataTmp = (uint8_t *)data;
 103     uint32_t header = ( (uint32_t)dataTmp[0] << 24 ) | ( (uint32_t)dataTmp[1] << 16 ) | ( (uint32_t)dataTmp[2] << 8 ) | (uint32_t)dataTmp[3];
 104     int layerID = 3 - ((header >> 17) & 0x03);
 105     int bitRateID = ((header >> 12) & 0x0f);
 106     int sampleRateID = ((header >> 10) & 0x03);
 107     int bitsPerSlot = sBitsPerSlot[layerID];
 108     int isPadded = ((header >> 9) & 0x01);
 109     static int const mode_tab[4]= {2,3,1,0};
 110     int mode= mode_tab[(header >> 19) & 0x03];
 111     int mpeg_id= mode>0;
 112     int temp0, temp1, bitRate;
 113
 114     if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
 115         return -1;
 116     }
 117
 118     if(!samplesPerFrame) samplesPerFrame= &temp0;
 119     if(!sampleRate     ) sampleRate     = &temp1;
 120
 121 //    *isMono = ((header >>  6) & 0x03) == 0x03;
 122
 123     *sampleRate = sSampleRates[sampleRateID]>>mode;
 124     bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
 125     *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
 126 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
 127
 128     return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
 129 }
 130
 131 int MP3lame_encode_frame(AVCodecContext *avctx,
 132                      unsigned char *frame, int buf_size, void *data)
 133 {
 134         Mp3AudioContext *s = avctx->priv_data;
 135         int len;
 136         int lame_result;
 137
 138         /* lame 3.91 dies on '1-channel interleaved' data */
 139
 140     if(data){
 141         if (s->stereo) {
 142             lame_result = lame_encode_buffer_interleaved(
 143                 s->gfp,
 144                 data,
 145                 avctx->frame_size,
 146                 s->buffer + s->buffer_index,
 147                 BUFFER_SIZE - s->buffer_index
 148                 );
 149         } else {
 150             lame_result = lame_encode_buffer(
 151                 s->gfp,
 152                 data,
 153                 data,
 154                 avctx->frame_size,
 155                 s->buffer + s->buffer_index,
 156                 BUFFER_SIZE - s->buffer_index
 157                 );
 158         }
 159     }else{
 160         lame_result= lame_encode_flush(
 161                 s->gfp,
 162                 s->buffer + s->buffer_index,
 163                 BUFFER_SIZE - s->buffer_index
 164                 );
 165     }
 166
 167     if(lame_result==-1) {
 168         /* output buffer too small */
 169         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);
 170         return 0;
 171     }
 172
 173     s->buffer_index += lame_result;
 174
 175     if(s->buffer_index<4)
 176         return 0;
 177
 178         len= mp3len(s->buffer, NULL, NULL);
 179 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
 180         if(len <= s->buffer_index){
 181             memcpy(frame, s->buffer, len);
 182             s->buffer_index -= len;
 183
 184             memmove(s->buffer, s->buffer+len, s->buffer_index);
 185             //FIXME fix the audio codec API, so we dont need the memcpy()
 186 /*for(i=0; i<len; i++){
 187     av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
 188 }*/
 189             return len;
 190         }else
 191             return 0;
 192 }
 193
 194 int MP3lame_encode_close(AVCodecContext *avctx)
 195 {
 196         Mp3AudioContext *s = avctx->priv_data;
 197
 198         av_freep(&avctx->coded_frame);
 199
 200         lame_close(s->gfp);
 201         return 0;
 202 }
 203
 204
 205 AVCodec mp3lame_encoder = {
 206     "mp3",
 207     CODEC_TYPE_AUDIO,
 208     CODEC_ID_MP3,
 209     sizeof(Mp3AudioContext),
 210     MP3lame_encode_init,
 211     MP3lame_encode_frame,
 212     MP3lame_encode_close,
 213     .capabilities= CODEC_CAP_DELAY,
 214 };