2 * Interface to libmp3lame for mp3 encoding
3 * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
5 * This file is part of FFmpeg.
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.
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.
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
24 * Interface to libmp3lame for mp3 encoding.
27 #include "libavutil/intreadwrite.h"
28 #include "libavutil/log.h"
29 #include "libavutil/opt.h"
31 #include "mpegaudio.h"
32 #include <lame/lame.h>
34 #define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
35 typedef struct Mp3AudioContext {
37 lame_global_flags *gfp;
39 uint8_t buffer[BUFFER_SIZE];
48 static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
50 Mp3AudioContext *s = avctx->priv_data;
52 if (avctx->channels > 2)
55 s->stereo = avctx->channels > 1 ? 1 : 0;
57 if ((s->gfp = lame_init()) == NULL)
59 lame_set_in_samplerate(s->gfp, avctx->sample_rate);
60 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
61 lame_set_num_channels(s->gfp, avctx->channels);
62 if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
63 lame_set_quality(s->gfp, 5);
65 lame_set_quality(s->gfp, avctx->compression_level);
67 lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
68 lame_set_brate(s->gfp, avctx->bit_rate/1000);
69 if(avctx->flags & CODEC_FLAG_QSCALE) {
70 lame_set_brate(s->gfp, 0);
71 lame_set_VBR(s->gfp, vbr_default);
72 lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
74 lame_set_bWriteVbrTag(s->gfp,0);
75 #if FF_API_LAME_GLOBAL_OPTS
76 s->reservoir = avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR;
78 lame_set_disable_reservoir(s->gfp, !s->reservoir);
79 if (lame_init_params(s->gfp) < 0)
82 avctx->frame_size = lame_get_framesize(s->gfp);
84 if(!(avctx->coded_frame= avcodec_alloc_frame())) {
87 return AVERROR(ENOMEM);
89 avctx->coded_frame->key_frame= 1;
91 if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
92 int nelem = 2 * avctx->frame_size;
94 if(! (s->s32_data.left = av_malloc(nelem * sizeof(int)))) {
95 av_freep(&avctx->coded_frame);
98 return AVERROR(ENOMEM);
101 s->s32_data.right = s->s32_data.left + avctx->frame_size;
112 static const int sSampleRates[] = {
113 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
116 static const int sBitRates[2][3][15] = {
117 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
118 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
119 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
121 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
122 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
123 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
127 static const int sSamplesPerFrame[2][3] =
133 static const int sBitsPerSlot[3] = {
139 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
141 uint32_t header = AV_RB32(data);
142 int layerID = 3 - ((header >> 17) & 0x03);
143 int bitRateID = ((header >> 12) & 0x0f);
144 int sampleRateID = ((header >> 10) & 0x03);
145 int bitsPerSlot = sBitsPerSlot[layerID];
146 int isPadded = ((header >> 9) & 0x01);
147 static int const mode_tab[4]= {2,3,1,0};
148 int mode= mode_tab[(header >> 19) & 0x03];
150 int temp0, temp1, bitRate;
152 if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
156 if(!samplesPerFrame) samplesPerFrame= &temp0;
157 if(!sampleRate ) sampleRate = &temp1;
159 // *isMono = ((header >> 6) & 0x03) == 0x03;
161 *sampleRate = sSampleRates[sampleRateID]>>mode;
162 bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
163 *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
164 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
166 return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
169 static int MP3lame_encode_frame(AVCodecContext *avctx,
170 unsigned char *frame, int buf_size, void *data)
172 Mp3AudioContext *s = avctx->priv_data;
176 /* lame 3.91 dies on '1-channel interleaved' data */
179 lame_result= lame_encode_flush(
181 s->buffer + s->buffer_index,
182 BUFFER_SIZE - s->buffer_index
184 #if 2147483647 == INT_MAX
185 }else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){
188 int32_t *mp = rp + 2*avctx->frame_size;
189 int *wpl = s->s32_data.left;
190 int *wpr = s->s32_data.right;
197 lame_result = lame_encode_buffer_int(
202 s->buffer + s->buffer_index,
203 BUFFER_SIZE - s->buffer_index
206 lame_result = lame_encode_buffer_int(
211 s->buffer + s->buffer_index,
212 BUFFER_SIZE - s->buffer_index
218 lame_result = lame_encode_buffer_interleaved(
222 s->buffer + s->buffer_index,
223 BUFFER_SIZE - s->buffer_index
226 lame_result = lame_encode_buffer(
231 s->buffer + s->buffer_index,
232 BUFFER_SIZE - s->buffer_index
238 if(lame_result==-1) {
239 /* output buffer too small */
240 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);
245 s->buffer_index += lame_result;
247 if(s->buffer_index<4)
250 len= mp3len(s->buffer, NULL, NULL);
251 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
252 if(len <= s->buffer_index){
253 memcpy(frame, s->buffer, len);
254 s->buffer_index -= len;
256 memmove(s->buffer, s->buffer+len, s->buffer_index);
257 //FIXME fix the audio codec API, so we do not need the memcpy()
258 /*for(i=0; i<len; i++){
259 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
266 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
268 Mp3AudioContext *s = avctx->priv_data;
270 av_freep(&s->s32_data.left);
271 av_freep(&avctx->coded_frame);
277 #define OFFSET(x) offsetof(Mp3AudioContext, x)
278 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
279 static const AVOption options[] = {
280 { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
284 static const AVClass libmp3lame_class = {
285 .class_name = "libmp3lame encoder",
286 .item_name = av_default_item_name,
288 .version = LIBAVUTIL_VERSION_INT,
291 AVCodec ff_libmp3lame_encoder = {
292 .name = "libmp3lame",
293 .type = AVMEDIA_TYPE_AUDIO,
295 .priv_data_size = sizeof(Mp3AudioContext),
296 .init = MP3lame_encode_init,
297 .encode = MP3lame_encode_frame,
298 .close = MP3lame_encode_close,
299 .capabilities= CODEC_CAP_DELAY,
300 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
301 #if 2147483647 == INT_MAX
305 .supported_samplerates= sSampleRates,
306 .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
307 .priv_class = &libmp3lame_class,