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 <lame/lame.h>
29 #include "libavutil/intreadwrite.h"
30 #include "libavutil/log.h"
31 #include "libavutil/opt.h"
34 #include "mpegaudio.h"
35 #include "mpegaudiodecheader.h"
37 #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
39 typedef struct LAMEContext {
41 AVCodecContext *avctx;
42 lame_global_flags *gfp;
43 uint8_t buffer[BUFFER_SIZE];
46 void *planar_samples[2];
50 static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
52 LAMEContext *s = avctx->priv_data;
54 av_freep(&avctx->coded_frame);
55 av_freep(&s->planar_samples[0]);
56 av_freep(&s->planar_samples[1]);
62 static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
64 LAMEContext *s = avctx->priv_data;
69 /* initialize LAME and get defaults */
70 if ((s->gfp = lame_init()) == NULL)
71 return AVERROR(ENOMEM);
74 if (avctx->channels > 2) {
75 ret = AVERROR(EINVAL);
78 lame_set_num_channels(s->gfp, avctx->channels);
79 lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
82 lame_set_in_samplerate (s->gfp, avctx->sample_rate);
83 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
85 /* algorithmic quality */
86 if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
87 lame_set_quality(s->gfp, 5);
89 lame_set_quality(s->gfp, avctx->compression_level);
92 if (avctx->flags & CODEC_FLAG_QSCALE) {
93 lame_set_VBR(s->gfp, vbr_default);
94 lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
97 lame_set_brate(s->gfp, avctx->bit_rate / 1000);
100 /* do not get a Xing VBR header frame from LAME */
101 lame_set_bWriteVbrTag(s->gfp,0);
103 /* bit reservoir usage */
104 lame_set_disable_reservoir(s->gfp, !s->reservoir);
106 /* set specified parameters */
107 if (lame_init_params(s->gfp) < 0) {
112 avctx->frame_size = lame_get_framesize(s->gfp);
113 avctx->coded_frame = avcodec_alloc_frame();
114 if (!avctx->coded_frame) {
115 ret = AVERROR(ENOMEM);
120 if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
121 avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
123 for (ch = 0; ch < avctx->channels; ch++) {
124 s->planar_samples[ch] = av_malloc(avctx->frame_size *
125 av_get_bytes_per_sample(avctx->sample_fmt));
126 if (!s->planar_samples[ch]) {
127 ret = AVERROR(ENOMEM);
135 mp3lame_encode_close(avctx);
139 #define DEINTERLEAVE(type, scale) do { \
141 for (ch = 0; ch < s->avctx->channels; ch++) { \
142 const type *input = samples; \
143 type *output = s->planar_samples[ch]; \
145 for (i = 0; i < s->avctx->frame_size; i++) { \
146 output[i] = *input * scale; \
147 input += s->avctx->channels; \
152 static int encode_frame_int16(LAMEContext *s, void *samples)
154 if (s->avctx->channels > 1) {
155 return lame_encode_buffer_interleaved(s->gfp, samples,
156 s->avctx->frame_size,
157 s->buffer + s->buffer_index,
158 BUFFER_SIZE - s->buffer_index);
160 return lame_encode_buffer(s->gfp, samples, NULL, s->avctx->frame_size,
161 s->buffer + s->buffer_index,
162 BUFFER_SIZE - s->buffer_index);
166 static int encode_frame_int32(LAMEContext *s, void *samples)
168 DEINTERLEAVE(int32_t, 1);
170 return lame_encode_buffer_int(s->gfp,
171 s->planar_samples[0], s->planar_samples[1],
172 s->avctx->frame_size,
173 s->buffer + s->buffer_index,
174 BUFFER_SIZE - s->buffer_index);
177 static int encode_frame_float(LAMEContext *s, void *samples)
179 DEINTERLEAVE(float, 32768.0f);
181 return lame_encode_buffer_float(s->gfp,
182 s->planar_samples[0], s->planar_samples[1],
183 s->avctx->frame_size,
184 s->buffer + s->buffer_index,
185 BUFFER_SIZE - s->buffer_index);
188 static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
189 int buf_size, void *data)
191 LAMEContext *s = avctx->priv_data;
197 switch (avctx->sample_fmt) {
198 case AV_SAMPLE_FMT_S16:
199 lame_result = encode_frame_int16(s, data);
201 case AV_SAMPLE_FMT_S32:
202 lame_result = encode_frame_int32(s, data);
204 case AV_SAMPLE_FMT_FLT:
205 lame_result = encode_frame_float(s, data);
211 lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
212 BUFFER_SIZE - s->buffer_index);
214 if (lame_result < 0) {
215 if (lame_result == -1) {
216 av_log(avctx, AV_LOG_ERROR,
217 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
218 s->buffer_index, BUFFER_SIZE - s->buffer_index);
222 s->buffer_index += lame_result;
224 /* Move 1 frame from the LAME buffer to the output packet, if available.
225 We have to parse the first frame header in the output buffer to
226 determine the frame size. */
227 if (s->buffer_index < 4)
229 if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
230 av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
233 len = hdr.frame_size;
234 av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
236 if (len <= s->buffer_index) {
237 memcpy(frame, s->buffer, len);
238 s->buffer_index -= len;
239 memmove(s->buffer, s->buffer + len, s->buffer_index);
245 #define OFFSET(x) offsetof(LAMEContext, x)
246 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
247 static const AVOption options[] = {
248 { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
252 static const AVClass libmp3lame_class = {
253 .class_name = "libmp3lame encoder",
254 .item_name = av_default_item_name,
256 .version = LIBAVUTIL_VERSION_INT,
259 static const AVCodecDefault libmp3lame_defaults[] = {
264 static const int libmp3lame_sample_rates[] = {
265 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
268 AVCodec ff_libmp3lame_encoder = {
269 .name = "libmp3lame",
270 .type = AVMEDIA_TYPE_AUDIO,
272 .priv_data_size = sizeof(LAMEContext),
273 .init = mp3lame_encode_init,
274 .encode = mp3lame_encode_frame,
275 .close = mp3lame_encode_close,
276 .capabilities = CODEC_CAP_DELAY,
277 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
280 AV_SAMPLE_FMT_NONE },
281 .supported_samplerates = libmp3lame_sample_rates,
282 .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
283 .priv_class = &libmp3lame_class,
284 .defaults = libmp3lame_defaults,