Mercurial > libavcodec.hg
annotate mp3lameaudio.c @ 4074:861a55e86ef0 libavcodec
factorize update state out
author | michael |
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date | Tue, 24 Oct 2006 09:34:13 +0000 |
parents | c8c591fe26f8 |
children | 85438e10d72d |
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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 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
429 | 8 * modify it under the terms of the GNU Lesser General Public |
9 * License as published by the Free Software Foundation; either | |
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10 * version 2.1 of the License, or (at your option) any later version. |
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11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
429 | 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
15 * Lesser General Public License for more details. | |
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16 * |
429 | 17 * You should have received a copy of the GNU Lesser General Public |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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20 */ |
2967 | 21 |
1106 | 22 /** |
23 * @file mp3lameaudio.c | |
24 * Interface to libmp3lame for mp3 encoding. | |
25 */ | |
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26 |
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27 #include "avcodec.h" |
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28 #include "mpegaudio.h" |
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29 #include <lame/lame.h> |
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30 |
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31 #define BUFFER_SIZE (2*MPA_FRAME_SIZE) |
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32 typedef struct Mp3AudioContext { |
2979 | 33 lame_global_flags *gfp; |
34 int stereo; | |
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35 uint8_t buffer[BUFFER_SIZE]; |
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36 int buffer_index; |
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37 } Mp3AudioContext; |
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38 |
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39 static int MP3lame_encode_init(AVCodecContext *avctx) |
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40 { |
2979 | 41 Mp3AudioContext *s = avctx->priv_data; |
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42 |
2979 | 43 if (avctx->channels > 2) |
44 return -1; | |
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45 |
2979 | 46 s->stereo = avctx->channels > 1 ? 1 : 0; |
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47 |
2979 | 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); | |
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58 if(avctx->flags & CODEC_FLAG_QSCALE) { |
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59 lame_set_brate(s->gfp, 0); |
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60 lame_set_VBR(s->gfp, vbr_default); |
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61 lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA); |
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62 } |
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63 lame_set_bWriteVbrTag(s->gfp,0); |
2979 | 64 if (lame_init_params(s->gfp) < 0) |
65 goto err_close; | |
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66 |
2979 | 67 avctx->frame_size = lame_get_framesize(s->gfp); |
2967 | 68 |
926 | 69 avctx->coded_frame= avcodec_alloc_frame(); |
70 avctx->coded_frame->key_frame= 1; | |
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71 |
2979 | 72 return 0; |
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73 |
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74 err_close: |
2979 | 75 lame_close(s->gfp); |
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76 err: |
2979 | 77 return -1; |
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78 } |
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79 |
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80 static const int sSampleRates[3] = { |
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81 44100, 48000, 32000 |
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82 }; |
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83 |
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84 static const int sBitRates[2][3][15] = { |
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85 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448}, |
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86 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384}, |
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87 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320} |
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88 }, |
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89 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256}, |
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90 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}, |
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91 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160} |
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92 }, |
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93 }; |
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94 |
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95 static const int sSamplesPerFrame[2][3] = |
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96 { |
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97 { 384, 1152, 1152 }, |
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98 { 384, 1152, 576 } |
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99 }; |
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100 |
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101 static const int sBitsPerSlot[3] = { |
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102 32, |
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103 8, |
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104 8 |
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105 }; |
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106 |
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107 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate) |
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108 { |
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109 uint8_t *dataTmp = (uint8_t *)data; |
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110 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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111 int layerID = 3 - ((header >> 17) & 0x03); |
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112 int bitRateID = ((header >> 12) & 0x0f); |
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113 int sampleRateID = ((header >> 10) & 0x03); |
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114 int bitsPerSlot = sBitsPerSlot[layerID]; |
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115 int isPadded = ((header >> 9) & 0x01); |
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116 static int const mode_tab[4]= {2,3,1,0}; |
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117 int mode= mode_tab[(header >> 19) & 0x03]; |
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118 int mpeg_id= mode>0; |
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119 int temp0, temp1, bitRate; |
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120 |
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121 if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) { |
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122 return -1; |
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123 } |
2967 | 124 |
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125 if(!samplesPerFrame) samplesPerFrame= &temp0; |
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126 if(!sampleRate ) sampleRate = &temp1; |
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127 |
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128 // *isMono = ((header >> 6) & 0x03) == 0x03; |
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129 |
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130 *sampleRate = sSampleRates[sampleRateID]>>mode; |
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131 bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000; |
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132 *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID]; |
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133 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode); |
2967 | 134 |
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135 return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded; |
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136 } |
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137 |
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138 int MP3lame_encode_frame(AVCodecContext *avctx, |
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139 unsigned char *frame, int buf_size, void *data) |
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140 { |
2979 | 141 Mp3AudioContext *s = avctx->priv_data; |
142 int len; | |
143 int lame_result; | |
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144 |
2979 | 145 /* lame 3.91 dies on '1-channel interleaved' data */ |
2091 | 146 |
147 if(data){ | |
148 if (s->stereo) { | |
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149 lame_result = lame_encode_buffer_interleaved( |
2967 | 150 s->gfp, |
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151 data, |
2967 | 152 avctx->frame_size, |
153 s->buffer + s->buffer_index, | |
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154 BUFFER_SIZE - s->buffer_index |
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155 ); |
2091 | 156 } else { |
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157 lame_result = lame_encode_buffer( |
2967 | 158 s->gfp, |
159 data, | |
160 data, | |
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161 avctx->frame_size, |
2967 | 162 s->buffer + s->buffer_index, |
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163 BUFFER_SIZE - s->buffer_index |
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164 ); |
2091 | 165 } |
166 }else{ | |
167 lame_result= lame_encode_flush( | |
2967 | 168 s->gfp, |
169 s->buffer + s->buffer_index, | |
2091 | 170 BUFFER_SIZE - s->buffer_index |
171 ); | |
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172 } |
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173 |
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174 if(lame_result==-1) { |
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175 /* output buffer too small */ |
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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); |
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177 return 0; |
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178 } |
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179 |
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180 s->buffer_index += lame_result; |
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181 |
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182 if(s->buffer_index<4) |
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183 return 0; |
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184 |
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185 len= mp3len(s->buffer, NULL, NULL); |
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186 //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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187 if(len <= s->buffer_index){ |
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188 memcpy(frame, s->buffer, len); |
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189 s->buffer_index -= len; |
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190 |
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191 memmove(s->buffer, s->buffer+len, s->buffer_index); |
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192 //FIXME fix the audio codec API, so we dont need the memcpy() |
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193 /*for(i=0; i<len; i++){ |
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194 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]); |
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195 }*/ |
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196 return len; |
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197 }else |
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198 return 0; |
259
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199 } |
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200 |
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201 int MP3lame_encode_close(AVCodecContext *avctx) |
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202 { |
2979 | 203 Mp3AudioContext *s = avctx->priv_data; |
2967 | 204 |
926 | 205 av_freep(&avctx->coded_frame); |
259
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206 |
2979 | 207 lame_close(s->gfp); |
208 return 0; | |
259
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209 } |
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210 |
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211 |
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212 AVCodec mp3lame_encoder = { |
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213 "mp3", |
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214 CODEC_TYPE_AUDIO, |
1448 | 215 CODEC_ID_MP3, |
259
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216 sizeof(Mp3AudioContext), |
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217 MP3lame_encode_init, |
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218 MP3lame_encode_frame, |
2091 | 219 MP3lame_encode_close, |
220 .capabilities= CODEC_CAP_DELAY, | |
259
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221 }; |