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"
29 #include "mpegaudio.h"
30 #include <lame/lame.h>
32 #define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
33 typedef struct Mp3AudioContext {
34 lame_global_flags *gfp;
36 uint8_t buffer[BUFFER_SIZE];
44 static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
46 Mp3AudioContext *s = avctx->priv_data;
48 if (avctx->channels > 2)
51 s->stereo = avctx->channels > 1 ? 1 : 0;
53 if ((s->gfp = lame_init()) == NULL)
55 lame_set_in_samplerate(s->gfp, avctx->sample_rate);
56 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
57 lame_set_num_channels(s->gfp, avctx->channels);
58 if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
59 lame_set_quality(s->gfp, 5);
61 lame_set_quality(s->gfp, avctx->compression_level);
63 lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
64 lame_set_brate(s->gfp, avctx->bit_rate/1000);
65 if(avctx->flags & CODEC_FLAG_QSCALE) {
66 lame_set_brate(s->gfp, 0);
67 lame_set_VBR(s->gfp, vbr_default);
68 lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
70 lame_set_bWriteVbrTag(s->gfp,0);
71 lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
72 if (lame_init_params(s->gfp) < 0)
75 avctx->frame_size = lame_get_framesize(s->gfp);
77 if(!(avctx->coded_frame= avcodec_alloc_frame())) {
80 return AVERROR(ENOMEM);
82 avctx->coded_frame->key_frame= 1;
84 if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
85 int nelem = 2 * avctx->frame_size;
87 if(! (s->s32_data.left = av_malloc(nelem * sizeof(int)))) {
88 av_freep(&avctx->coded_frame);
91 return AVERROR(ENOMEM);
94 s->s32_data.right = s->s32_data.left + avctx->frame_size;
105 static const int sSampleRates[] = {
106 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
109 static const int sBitRates[2][3][15] = {
110 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
111 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
112 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
114 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
115 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
116 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
120 static const int sSamplesPerFrame[2][3] =
126 static const int sBitsPerSlot[3] = {
132 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
134 uint32_t header = AV_RB32(data);
135 int layerID = 3 - ((header >> 17) & 0x03);
136 int bitRateID = ((header >> 12) & 0x0f);
137 int sampleRateID = ((header >> 10) & 0x03);
138 int bitsPerSlot = sBitsPerSlot[layerID];
139 int isPadded = ((header >> 9) & 0x01);
140 static int const mode_tab[4]= {2,3,1,0};
141 int mode= mode_tab[(header >> 19) & 0x03];
143 int temp0, temp1, bitRate;
145 if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
149 if(!samplesPerFrame) samplesPerFrame= &temp0;
150 if(!sampleRate ) sampleRate = &temp1;
152 // *isMono = ((header >> 6) & 0x03) == 0x03;
154 *sampleRate = sSampleRates[sampleRateID]>>mode;
155 bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
156 *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
157 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
159 return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
162 static int MP3lame_encode_frame(AVCodecContext *avctx,
163 unsigned char *frame, int buf_size, void *data)
165 Mp3AudioContext *s = avctx->priv_data;
169 /* lame 3.91 dies on '1-channel interleaved' data */
172 lame_result= lame_encode_flush(
174 s->buffer + s->buffer_index,
175 BUFFER_SIZE - s->buffer_index
177 #if 2147483647 == INT_MAX
178 }else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){
181 int32_t *mp = rp + 2*avctx->frame_size;
182 int *wpl = s->s32_data.left;
183 int *wpr = s->s32_data.right;
190 lame_result = lame_encode_buffer_int(
195 s->buffer + s->buffer_index,
196 BUFFER_SIZE - s->buffer_index
199 lame_result = lame_encode_buffer_int(
204 s->buffer + s->buffer_index,
205 BUFFER_SIZE - s->buffer_index
211 lame_result = lame_encode_buffer_interleaved(
215 s->buffer + s->buffer_index,
216 BUFFER_SIZE - s->buffer_index
219 lame_result = lame_encode_buffer(
224 s->buffer + s->buffer_index,
225 BUFFER_SIZE - s->buffer_index
231 if(lame_result==-1) {
232 /* output buffer too small */
233 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);
238 s->buffer_index += lame_result;
240 if(s->buffer_index<4)
243 len= mp3len(s->buffer, NULL, NULL);
244 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
245 if(len <= s->buffer_index){
246 memcpy(frame, s->buffer, len);
247 s->buffer_index -= len;
249 memmove(s->buffer, s->buffer+len, s->buffer_index);
250 //FIXME fix the audio codec API, so we do not need the memcpy()
251 /*for(i=0; i<len; i++){
252 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
259 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
261 Mp3AudioContext *s = avctx->priv_data;
263 av_freep(&s->s32_data.left);
264 av_freep(&avctx->coded_frame);
271 AVCodec ff_libmp3lame_encoder = {
272 .name = "libmp3lame",
273 .type = AVMEDIA_TYPE_AUDIO,
275 .priv_data_size = sizeof(Mp3AudioContext),
276 .init = MP3lame_encode_init,
277 .encode = MP3lame_encode_frame,
278 .close = MP3lame_encode_close,
279 .capabilities= CODEC_CAP_DELAY,
280 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
281 #if 2147483647 == INT_MAX
285 .supported_samplerates= sSampleRates,
286 .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),