2 * Interface to libmp3lame for mp3 encoding
3 * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; 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;
38 uint8_t buffer[BUFFER_SIZE];
44 static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
46 Mp3AudioContext *s = avctx->priv_data;
48 av_freep(&avctx->coded_frame);
54 static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
56 Mp3AudioContext *s = avctx->priv_data;
59 if (avctx->channels > 2)
60 return AVERROR(EINVAL);
62 if ((s->gfp = lame_init()) == NULL)
63 return AVERROR(ENOMEM);
64 lame_set_in_samplerate(s->gfp, avctx->sample_rate);
65 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
66 lame_set_num_channels(s->gfp, avctx->channels);
67 if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
68 lame_set_quality(s->gfp, 5);
70 lame_set_quality(s->gfp, avctx->compression_level);
72 lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
73 lame_set_brate(s->gfp, avctx->bit_rate / 1000);
74 if (avctx->flags & CODEC_FLAG_QSCALE) {
75 lame_set_brate(s->gfp, 0);
76 lame_set_VBR(s->gfp, vbr_default);
77 lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
79 lame_set_bWriteVbrTag(s->gfp,0);
80 lame_set_disable_reservoir(s->gfp, !s->reservoir);
81 if (lame_init_params(s->gfp) < 0) {
86 avctx->frame_size = lame_get_framesize(s->gfp);
87 avctx->coded_frame = avcodec_alloc_frame();
88 if (!avctx->coded_frame) {
89 ret = AVERROR(ENOMEM);
95 MP3lame_encode_close(avctx);
99 static const int sSampleRates[] = {
100 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
103 static const int sBitRates[2][3][15] = {
105 { 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
106 { 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
107 { 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 }
110 { 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
111 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 },
112 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 }
116 static const int sSamplesPerFrame[2][3] = {
121 static const int sBitsPerSlot[3] = { 32, 8, 8 };
123 static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
125 uint32_t header = AV_RB32(data);
126 int layerID = 3 - ((header >> 17) & 0x03);
127 int bitRateID = ((header >> 12) & 0x0f);
128 int sampleRateID = ((header >> 10) & 0x03);
129 int bitsPerSlot = sBitsPerSlot[layerID];
130 int isPadded = ((header >> 9) & 0x01);
131 static int const mode_tab[4] = { 2, 3, 1, 0 };
132 int mode = mode_tab[(header >> 19) & 0x03];
133 int mpeg_id = mode > 0;
134 int temp0, temp1, bitRate;
136 if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
141 if (!samplesPerFrame)
142 samplesPerFrame = &temp0;
146 //*isMono = ((header >> 6) & 0x03) == 0x03;
148 *sampleRate = sSampleRates[sampleRateID] >> mode;
149 bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
150 *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
151 //av_log(NULL, AV_LOG_DEBUG,
152 // "sr:%d br:%d spf:%d l:%d m:%d\n",
153 // *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
155 return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
158 static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
159 int buf_size, void *data)
161 Mp3AudioContext *s = avctx->priv_data;
166 if (avctx->channels > 1) {
167 lame_result = lame_encode_buffer_interleaved(s->gfp, data,
169 s->buffer + s->buffer_index,
170 BUFFER_SIZE - s->buffer_index);
172 lame_result = lame_encode_buffer(s->gfp, data, data,
173 avctx->frame_size, s->buffer +
174 s->buffer_index, BUFFER_SIZE -
178 lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
179 BUFFER_SIZE - s->buffer_index);
182 if (lame_result < 0) {
183 if (lame_result == -1) {
184 /* output buffer too small */
185 av_log(avctx, AV_LOG_ERROR,
186 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
187 s->buffer_index, BUFFER_SIZE - s->buffer_index);
192 s->buffer_index += lame_result;
194 if (s->buffer_index < 4)
197 len = mp3len(s->buffer, NULL, NULL);
198 av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
200 if (len <= s->buffer_index) {
201 memcpy(frame, s->buffer, len);
202 s->buffer_index -= len;
204 memmove(s->buffer, s->buffer + len, s->buffer_index);
205 // FIXME fix the audio codec API, so we do not need the memcpy()
211 #define OFFSET(x) offsetof(Mp3AudioContext, x)
212 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
213 static const AVOption options[] = {
214 { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
218 static const AVClass libmp3lame_class = {
219 .class_name = "libmp3lame encoder",
220 .item_name = av_default_item_name,
222 .version = LIBAVUTIL_VERSION_INT,
225 AVCodec ff_libmp3lame_encoder = {
226 .name = "libmp3lame",
227 .type = AVMEDIA_TYPE_AUDIO,
229 .priv_data_size = sizeof(Mp3AudioContext),
230 .init = MP3lame_encode_init,
231 .encode = MP3lame_encode_frame,
232 .close = MP3lame_encode_close,
233 .capabilities = CODEC_CAP_DELAY,
234 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
235 AV_SAMPLE_FMT_NONE },
236 .supported_samplerates = sSampleRates,
237 .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
238 .priv_class = &libmp3lame_class,