Mercurial > libavcodec.hg
annotate mp3lameaudio.c @ 2718:158dc437c783 libavcodec
read Huffman tables from Theora header (patch courtesy of
Matthieu Castet <castet.matthieu at free.fr>)
author | melanson |
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date | Thu, 19 May 2005 23:59:10 +0000 |
parents | e25782262d7d |
children | bceeca1bb30f |
rev | line source |
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1 /* |
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2 * Interface to libmp3lame for mp3 encoding |
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3 * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org> |
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4 * |
429 | 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. | |
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9 * |
429 | 10 * This library is distributed in the hope that it will be useful, |
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11 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
429 | 12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
13 * Lesser General Public License for more details. | |
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14 * |
429 | 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 | |
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18 */ |
1106 | 19 |
20 /** | |
21 * @file mp3lameaudio.c | |
22 * Interface to libmp3lame for mp3 encoding. | |
23 */ | |
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24 |
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25 #include "avcodec.h" |
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26 #include "mpegaudio.h" |
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27 #include <lame/lame.h> |
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28 |
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29 #define BUFFER_SIZE (2*MPA_FRAME_SIZE) |
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30 typedef struct Mp3AudioContext { |
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31 lame_global_flags *gfp; |
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32 int stereo; |
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33 uint8_t buffer[BUFFER_SIZE]; |
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34 int buffer_index; |
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35 } Mp3AudioContext; |
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36 |
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37 static int MP3lame_encode_init(AVCodecContext *avctx) |
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38 { |
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39 Mp3AudioContext *s = avctx->priv_data; |
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40 |
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41 if (avctx->channels > 2) |
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42 return -1; |
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43 |
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44 s->stereo = avctx->channels > 1 ? 1 : 0; |
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45 |
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46 if ((s->gfp = lame_init()) == NULL) |
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47 goto err; |
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48 lame_set_in_samplerate(s->gfp, avctx->sample_rate); |
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49 lame_set_out_samplerate(s->gfp, avctx->sample_rate); |
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50 lame_set_num_channels(s->gfp, avctx->channels); |
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51 /* lame 3.91 dies on quality != 5 */ |
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52 lame_set_quality(s->gfp, 5); |
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53 /* lame 3.91 doesn't work in mono */ |
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54 lame_set_mode(s->gfp, JOINT_STEREO); |
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55 lame_set_brate(s->gfp, avctx->bit_rate/1000); |
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56 lame_set_bWriteVbrTag(s->gfp,0); |
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57 if (lame_init_params(s->gfp) < 0) |
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58 goto err_close; |
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59 |
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60 avctx->frame_size = lame_get_framesize(s->gfp); |
926 | 61 |
62 avctx->coded_frame= avcodec_alloc_frame(); | |
63 avctx->coded_frame->key_frame= 1; | |
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64 |
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65 return 0; |
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66 |
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67 err_close: |
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68 lame_close(s->gfp); |
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69 err: |
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70 return -1; |
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71 } |
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72 |
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73 static const int sSampleRates[3] = { |
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74 44100, 48000, 32000 |
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75 }; |
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76 |
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77 static const int sBitRates[2][3][15] = { |
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78 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448}, |
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79 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384}, |
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80 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320} |
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81 }, |
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82 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256}, |
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83 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}, |
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84 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160} |
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85 }, |
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86 }; |
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87 |
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88 static const int sSamplesPerFrame[2][3] = |
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89 { |
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90 { 384, 1152, 1152 }, |
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91 { 384, 1152, 576 } |
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92 }; |
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93 |
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94 static const int sBitsPerSlot[3] = { |
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95 32, |
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96 8, |
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97 8 |
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98 }; |
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99 |
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100 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate) |
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101 { |
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102 uint8_t *dataTmp = (uint8_t *)data; |
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103 uint32_t header = ( (uint32_t)dataTmp[0] << 24 ) | ( (uint32_t)dataTmp[1] << 16 ) | ( (uint32_t)dataTmp[2] << 8 ) | (uint32_t)dataTmp[3]; |
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104 int layerID = 3 - ((header >> 17) & 0x03); |
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105 int bitRateID = ((header >> 12) & 0x0f); |
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106 int sampleRateID = ((header >> 10) & 0x03); |
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107 int bitsPerSlot = sBitsPerSlot[layerID]; |
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108 int isPadded = ((header >> 9) & 0x01); |
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109 static int const mode_tab[4]= {2,3,1,0}; |
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110 int mode= mode_tab[(header >> 19) & 0x03]; |
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111 int mpeg_id= mode>0; |
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112 int temp0, temp1, bitRate; |
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113 |
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114 if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) { |
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115 return -1; |
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116 } |
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117 |
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118 if(!samplesPerFrame) samplesPerFrame= &temp0; |
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119 if(!sampleRate ) sampleRate = &temp1; |
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120 |
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121 // *isMono = ((header >> 6) & 0x03) == 0x03; |
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122 |
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123 *sampleRate = sSampleRates[sampleRateID]>>mode; |
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124 bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000; |
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125 *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID]; |
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126 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode); |
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127 |
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128 return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded; |
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129 } |
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130 |
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131 int MP3lame_encode_frame(AVCodecContext *avctx, |
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132 unsigned char *frame, int buf_size, void *data) |
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133 { |
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134 Mp3AudioContext *s = avctx->priv_data; |
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135 int len; |
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136 int lame_result; |
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137 |
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138 /* lame 3.91 dies on '1-channel interleaved' data */ |
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140 if(data){ | |
141 if (s->stereo) { | |
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142 lame_result = lame_encode_buffer_interleaved( |
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143 s->gfp, |
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144 data, |
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145 avctx->frame_size, |
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146 s->buffer + s->buffer_index, |
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147 BUFFER_SIZE - s->buffer_index |
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148 ); |
2091 | 149 } else { |
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150 lame_result = lame_encode_buffer( |
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151 s->gfp, |
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152 data, |
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153 data, |
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154 avctx->frame_size, |
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155 s->buffer + s->buffer_index, |
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156 BUFFER_SIZE - s->buffer_index |
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157 ); |
2091 | 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 ); | |
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165 } |
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166 |
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167 if(lame_result==-1) { |
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168 /* output buffer too small */ |
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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); |
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170 return 0; |
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171 } |
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172 |
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173 s->buffer_index += lame_result; |
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174 |
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175 if(s->buffer_index<4) |
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176 return 0; |
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177 |
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178 len= mp3len(s->buffer, NULL, NULL); |
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179 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index); |
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180 if(len <= s->buffer_index){ |
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181 memcpy(frame, s->buffer, len); |
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182 s->buffer_index -= len; |
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183 |
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184 memmove(s->buffer, s->buffer+len, s->buffer_index); |
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185 //FIXME fix the audio codec API, so we dont need the memcpy() |
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186 /*for(i=0; i<len; i++){ |
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187 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]); |
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188 }*/ |
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189 return len; |
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190 }else |
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191 return 0; |
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192 } |
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193 |
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194 int MP3lame_encode_close(AVCodecContext *avctx) |
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195 { |
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196 Mp3AudioContext *s = avctx->priv_data; |
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198 av_freep(&avctx->coded_frame); | |
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199 |
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200 lame_close(s->gfp); |
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201 return 0; |
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202 } |
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203 |
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204 |
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205 AVCodec mp3lame_encoder = { |
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206 "mp3", |
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207 CODEC_TYPE_AUDIO, |
1448 | 208 CODEC_ID_MP3, |
259
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209 sizeof(Mp3AudioContext), |
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210 MP3lame_encode_init, |
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211 MP3lame_encode_frame, |
2091 | 212 MP3lame_encode_close, |
213 .capabilities= CODEC_CAP_DELAY, | |
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214 }; |