Mercurial > libavcodec.hg
comparison libmp3lame.c @ 5101:c3f2379b80db libavcodec
Give all wrappers for external libraries names starting with lib.
author | diego |
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date | Wed, 06 Jun 2007 00:14:18 +0000 |
parents | mp3lameaudio.c@bff60ecc02f9 |
children | 1deb3e53da27 |
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5100:daff9ebd2e0b | 5101:c3f2379b80db |
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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 (2*MPA_FRAME_SIZE) | |
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 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 if (lame_init_params(s->gfp) < 0) | |
65 goto err_close; | |
66 | |
67 avctx->frame_size = lame_get_framesize(s->gfp); | |
68 | |
69 avctx->coded_frame= avcodec_alloc_frame(); | |
70 avctx->coded_frame->key_frame= 1; | |
71 | |
72 return 0; | |
73 | |
74 err_close: | |
75 lame_close(s->gfp); | |
76 err: | |
77 return -1; | |
78 } | |
79 | |
80 static const int sSampleRates[3] = { | |
81 44100, 48000, 32000 | |
82 }; | |
83 | |
84 static const int sBitRates[2][3][15] = { | |
85 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448}, | |
86 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384}, | |
87 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320} | |
88 }, | |
89 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256}, | |
90 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}, | |
91 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160} | |
92 }, | |
93 }; | |
94 | |
95 static const int sSamplesPerFrame[2][3] = | |
96 { | |
97 { 384, 1152, 1152 }, | |
98 { 384, 1152, 576 } | |
99 }; | |
100 | |
101 static const int sBitsPerSlot[3] = { | |
102 32, | |
103 8, | |
104 8 | |
105 }; | |
106 | |
107 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate) | |
108 { | |
109 uint32_t header = AV_RB32(data); | |
110 int layerID = 3 - ((header >> 17) & 0x03); | |
111 int bitRateID = ((header >> 12) & 0x0f); | |
112 int sampleRateID = ((header >> 10) & 0x03); | |
113 int bitsPerSlot = sBitsPerSlot[layerID]; | |
114 int isPadded = ((header >> 9) & 0x01); | |
115 static int const mode_tab[4]= {2,3,1,0}; | |
116 int mode= mode_tab[(header >> 19) & 0x03]; | |
117 int mpeg_id= mode>0; | |
118 int temp0, temp1, bitRate; | |
119 | |
120 if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) { | |
121 return -1; | |
122 } | |
123 | |
124 if(!samplesPerFrame) samplesPerFrame= &temp0; | |
125 if(!sampleRate ) sampleRate = &temp1; | |
126 | |
127 // *isMono = ((header >> 6) & 0x03) == 0x03; | |
128 | |
129 *sampleRate = sSampleRates[sampleRateID]>>mode; | |
130 bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000; | |
131 *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID]; | |
132 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode); | |
133 | |
134 return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded; | |
135 } | |
136 | |
137 static int MP3lame_encode_frame(AVCodecContext *avctx, | |
138 unsigned char *frame, int buf_size, void *data) | |
139 { | |
140 Mp3AudioContext *s = avctx->priv_data; | |
141 int len; | |
142 int lame_result; | |
143 | |
144 /* lame 3.91 dies on '1-channel interleaved' data */ | |
145 | |
146 if(data){ | |
147 if (s->stereo) { | |
148 lame_result = lame_encode_buffer_interleaved( | |
149 s->gfp, | |
150 data, | |
151 avctx->frame_size, | |
152 s->buffer + s->buffer_index, | |
153 BUFFER_SIZE - s->buffer_index | |
154 ); | |
155 } else { | |
156 lame_result = lame_encode_buffer( | |
157 s->gfp, | |
158 data, | |
159 data, | |
160 avctx->frame_size, | |
161 s->buffer + s->buffer_index, | |
162 BUFFER_SIZE - s->buffer_index | |
163 ); | |
164 } | |
165 }else{ | |
166 lame_result= lame_encode_flush( | |
167 s->gfp, | |
168 s->buffer + s->buffer_index, | |
169 BUFFER_SIZE - s->buffer_index | |
170 ); | |
171 } | |
172 | |
173 if(lame_result==-1) { | |
174 /* output buffer too small */ | |
175 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); | |
176 return 0; | |
177 } | |
178 | |
179 s->buffer_index += lame_result; | |
180 | |
181 if(s->buffer_index<4) | |
182 return 0; | |
183 | |
184 len= mp3len(s->buffer, NULL, NULL); | |
185 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index); | |
186 if(len <= s->buffer_index){ | |
187 memcpy(frame, s->buffer, len); | |
188 s->buffer_index -= len; | |
189 | |
190 memmove(s->buffer, s->buffer+len, s->buffer_index); | |
191 //FIXME fix the audio codec API, so we dont need the memcpy() | |
192 /*for(i=0; i<len; i++){ | |
193 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]); | |
194 }*/ | |
195 return len; | |
196 }else | |
197 return 0; | |
198 } | |
199 | |
200 static int MP3lame_encode_close(AVCodecContext *avctx) | |
201 { | |
202 Mp3AudioContext *s = avctx->priv_data; | |
203 | |
204 av_freep(&avctx->coded_frame); | |
205 | |
206 lame_close(s->gfp); | |
207 return 0; | |
208 } | |
209 | |
210 | |
211 AVCodec mp3lame_encoder = { | |
212 "mp3", | |
213 CODEC_TYPE_AUDIO, | |
214 CODEC_ID_MP3, | |
215 sizeof(Mp3AudioContext), | |
216 MP3lame_encode_init, | |
217 MP3lame_encode_frame, | |
218 MP3lame_encode_close, | |
219 .capabilities= CODEC_CAP_DELAY, | |
220 }; |