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+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
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) {
53 av_log(avctx, AV_LOG_ERROR,
54 "Invalid number of channels %d, must be <= 2\n", avctx->channels);
55 return AVERROR(EINVAL);
58 s->stereo = avctx->channels > 1 ? 1 : 0;
60 if ((s->gfp = lame_init()) == NULL)
62 lame_set_in_samplerate(s->gfp, avctx->sample_rate);
63 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
64 lame_set_num_channels(s->gfp, avctx->channels);
65 if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
66 lame_set_quality(s->gfp, 5);
68 lame_set_quality(s->gfp, avctx->compression_level);
70 lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
71 lame_set_brate(s->gfp, avctx->bit_rate / 1000);
72 if (avctx->flags & CODEC_FLAG_QSCALE) {
73 lame_set_brate(s->gfp, 0);
74 lame_set_VBR(s->gfp, vbr_default);
75 lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
77 lame_set_bWriteVbrTag(s->gfp,0);
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] = {
118 { 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
119 { 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
120 { 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 }
123 { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
124 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 },
125 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 }
129 static const int sSamplesPerFrame[2][3] = {
134 static const int sBitsPerSlot[3] = { 32, 8, 8 };
136 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
138 uint32_t header = AV_RB32(data);
139 int layerID = 3 - ((header >> 17) & 0x03);
140 int bitRateID = ((header >> 12) & 0x0f);
141 int sampleRateID = ((header >> 10) & 0x03);
142 int bitsPerSlot = sBitsPerSlot[layerID];
143 int isPadded = ((header >> 9) & 0x01);
144 static int const mode_tab[4] = { 2, 3, 1, 0 };
145 int mode = mode_tab[(header >> 19) & 0x03];
146 int mpeg_id = mode > 0;
147 int temp0, temp1, bitRate;
149 if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
154 if (!samplesPerFrame)
155 samplesPerFrame = &temp0;
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,
165 // "sr:%d br:%d spf:%d l:%d m:%d\n",
166 // *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
168 return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
171 static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
172 int buf_size, void *data)
174 Mp3AudioContext *s = avctx->priv_data;
178 /* lame 3.91 dies on '1-channel interleaved' data */
181 lame_result= lame_encode_flush(
183 s->buffer + s->buffer_index,
184 BUFFER_SIZE - s->buffer_index
186 #if 2147483647 == INT_MAX
187 }else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){
190 int32_t *mp = rp + 2*avctx->frame_size;
191 int *wpl = s->s32_data.left;
192 int *wpr = s->s32_data.right;
199 lame_result = lame_encode_buffer_int(
204 s->buffer + s->buffer_index,
205 BUFFER_SIZE - s->buffer_index
208 lame_result = lame_encode_buffer_int(
213 s->buffer + s->buffer_index,
214 BUFFER_SIZE - s->buffer_index
220 lame_result = lame_encode_buffer_interleaved(
224 s->buffer + s->buffer_index,
225 BUFFER_SIZE - s->buffer_index
228 lame_result = lame_encode_buffer(
233 s->buffer + s->buffer_index,
234 BUFFER_SIZE - s->buffer_index
239 if (lame_result < 0) {
240 if (lame_result == -1) {
241 /* output buffer too small */
242 av_log(avctx, AV_LOG_ERROR,
243 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
244 s->buffer_index, BUFFER_SIZE - s->buffer_index);
249 s->buffer_index += lame_result;
251 if (s->buffer_index < 4)
254 len = mp3len(s->buffer, NULL, NULL);
255 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n",
256 // avctx->frame_size, len, s->buffer_index);
257 if (len <= s->buffer_index) {
258 memcpy(frame, s->buffer, len);
259 s->buffer_index -= len;
261 memmove(s->buffer, s->buffer + len, s->buffer_index);
262 // FIXME fix the audio codec API, so we do not need the memcpy()
263 /*for(i=0; i<len; i++) {
264 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
271 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
273 Mp3AudioContext *s = avctx->priv_data;
275 av_freep(&s->s32_data.left);
276 av_freep(&avctx->coded_frame);
282 #define OFFSET(x) offsetof(Mp3AudioContext, x)
283 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
284 static const AVOption options[] = {
285 { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
289 static const AVClass libmp3lame_class = {
290 .class_name = "libmp3lame encoder",
291 .item_name = av_default_item_name,
293 .version = LIBAVUTIL_VERSION_INT,
296 AVCodec ff_libmp3lame_encoder = {
297 .name = "libmp3lame",
298 .type = AVMEDIA_TYPE_AUDIO,
300 .priv_data_size = sizeof(Mp3AudioContext),
301 .init = MP3lame_encode_init,
302 .encode = MP3lame_encode_frame,
303 .close = MP3lame_encode_close,
304 .capabilities = CODEC_CAP_DELAY,
305 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
306 #if 2147483647 == INT_MAX
309 AV_SAMPLE_FMT_NONE },
310 .supported_samplerates = sSampleRates,
311 .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
312 .priv_class = &libmp3lame_class,