Mercurial > libavcodec.hg
annotate mp3lameaudio.c @ 3526:7dc8e4a12105 libavcodec
New functions in DSPContext for VC-1 decoding
author | kostya |
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date | Sun, 30 Jul 2006 03:45:02 +0000 |
parents | 0b546eab515d |
children | c8c591fe26f8 |
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 | |
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17 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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18 */ |
2967 | 19 |
1106 | 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 { |
2979 | 31 lame_global_flags *gfp; |
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 { |
2979 | 39 Mp3AudioContext *s = avctx->priv_data; |
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40 |
2979 | 41 if (avctx->channels > 2) |
42 return -1; | |
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43 |
2979 | 44 s->stereo = avctx->channels > 1 ? 1 : 0; |
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45 |
2979 | 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); | |
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56 if(avctx->flags & CODEC_FLAG_QSCALE) { |
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57 lame_set_brate(s->gfp, 0); |
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58 lame_set_VBR(s->gfp, vbr_default); |
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59 lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA); |
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60 } |
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61 lame_set_bWriteVbrTag(s->gfp,0); |
2979 | 62 if (lame_init_params(s->gfp) < 0) |
63 goto err_close; | |
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64 |
2979 | 65 avctx->frame_size = lame_get_framesize(s->gfp); |
2967 | 66 |
926 | 67 avctx->coded_frame= avcodec_alloc_frame(); |
68 avctx->coded_frame->key_frame= 1; | |
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69 |
2979 | 70 return 0; |
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71 |
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72 err_close: |
2979 | 73 lame_close(s->gfp); |
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74 err: |
2979 | 75 return -1; |
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76 } |
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77 |
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78 static const int sSampleRates[3] = { |
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79 44100, 48000, 32000 |
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80 }; |
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81 |
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82 static const int sBitRates[2][3][15] = { |
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83 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448}, |
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84 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384}, |
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85 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320} |
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86 }, |
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87 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256}, |
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88 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}, |
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89 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160} |
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90 }, |
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91 }; |
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92 |
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93 static const int sSamplesPerFrame[2][3] = |
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94 { |
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95 { 384, 1152, 1152 }, |
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96 { 384, 1152, 576 } |
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97 }; |
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98 |
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99 static const int sBitsPerSlot[3] = { |
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100 32, |
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101 8, |
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102 8 |
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103 }; |
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104 |
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105 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate) |
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106 { |
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107 uint8_t *dataTmp = (uint8_t *)data; |
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108 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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109 int layerID = 3 - ((header >> 17) & 0x03); |
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110 int bitRateID = ((header >> 12) & 0x0f); |
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111 int sampleRateID = ((header >> 10) & 0x03); |
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112 int bitsPerSlot = sBitsPerSlot[layerID]; |
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113 int isPadded = ((header >> 9) & 0x01); |
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114 static int const mode_tab[4]= {2,3,1,0}; |
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115 int mode= mode_tab[(header >> 19) & 0x03]; |
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116 int mpeg_id= mode>0; |
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117 int temp0, temp1, bitRate; |
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118 |
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119 if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) { |
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120 return -1; |
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121 } |
2967 | 122 |
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123 if(!samplesPerFrame) samplesPerFrame= &temp0; |
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124 if(!sampleRate ) sampleRate = &temp1; |
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125 |
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126 // *isMono = ((header >> 6) & 0x03) == 0x03; |
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127 |
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128 *sampleRate = sSampleRates[sampleRateID]>>mode; |
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129 bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000; |
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130 *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID]; |
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131 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode); |
2967 | 132 |
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133 return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded; |
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134 } |
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135 |
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136 int MP3lame_encode_frame(AVCodecContext *avctx, |
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137 unsigned char *frame, int buf_size, void *data) |
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138 { |
2979 | 139 Mp3AudioContext *s = avctx->priv_data; |
140 int len; | |
141 int lame_result; | |
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142 |
2979 | 143 /* lame 3.91 dies on '1-channel interleaved' data */ |
2091 | 144 |
145 if(data){ | |
146 if (s->stereo) { | |
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147 lame_result = lame_encode_buffer_interleaved( |
2967 | 148 s->gfp, |
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149 data, |
2967 | 150 avctx->frame_size, |
151 s->buffer + s->buffer_index, | |
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152 BUFFER_SIZE - s->buffer_index |
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153 ); |
2091 | 154 } else { |
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155 lame_result = lame_encode_buffer( |
2967 | 156 s->gfp, |
157 data, | |
158 data, | |
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159 avctx->frame_size, |
2967 | 160 s->buffer + s->buffer_index, |
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161 BUFFER_SIZE - s->buffer_index |
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162 ); |
2091 | 163 } |
164 }else{ | |
165 lame_result= lame_encode_flush( | |
2967 | 166 s->gfp, |
167 s->buffer + s->buffer_index, | |
2091 | 168 BUFFER_SIZE - s->buffer_index |
169 ); | |
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170 } |
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171 |
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172 if(lame_result==-1) { |
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173 /* output buffer too small */ |
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174 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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175 return 0; |
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176 } |
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177 |
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178 s->buffer_index += lame_result; |
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179 |
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180 if(s->buffer_index<4) |
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181 return 0; |
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182 |
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183 len= mp3len(s->buffer, NULL, NULL); |
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184 //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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185 if(len <= s->buffer_index){ |
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186 memcpy(frame, s->buffer, len); |
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187 s->buffer_index -= len; |
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188 |
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189 memmove(s->buffer, s->buffer+len, s->buffer_index); |
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190 //FIXME fix the audio codec API, so we dont need the memcpy() |
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191 /*for(i=0; i<len; i++){ |
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192 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]); |
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193 }*/ |
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194 return len; |
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195 }else |
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196 return 0; |
259
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197 } |
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198 |
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199 int MP3lame_encode_close(AVCodecContext *avctx) |
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200 { |
2979 | 201 Mp3AudioContext *s = avctx->priv_data; |
2967 | 202 |
926 | 203 av_freep(&avctx->coded_frame); |
259
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204 |
2979 | 205 lame_close(s->gfp); |
206 return 0; | |
259
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207 } |
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208 |
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209 |
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210 AVCodec mp3lame_encoder = { |
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211 "mp3", |
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212 CODEC_TYPE_AUDIO, |
1448 | 213 CODEC_ID_MP3, |
259
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214 sizeof(Mp3AudioContext), |
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215 MP3lame_encode_init, |
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216 MP3lame_encode_frame, |
2091 | 217 MP3lame_encode_close, |
218 .capabilities= CODEC_CAP_DELAY, | |
259
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219 }; |