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/channel_layout.h"
30 #include "libavutil/common.h"
31 #include "libavutil/float_dsp.h"
32 #include "libavutil/intreadwrite.h"
33 #include "libavutil/log.h"
34 #include "libavutil/opt.h"
36 #include "audio_frame_queue.h"
38 #include "mpegaudio.h"
39 #include "mpegaudiodecheader.h"
41 #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
43 typedef struct LAMEContext {
45 AVCodecContext *avctx;
46 lame_global_flags *gfp;
51 float *samples_flt[2];
53 AVFloatDSPContext fdsp;
57 static int realloc_buffer(LAMEContext *s)
59 if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
61 int new_size = s->buffer_index + 2 * BUFFER_SIZE;
63 av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
65 tmp = av_realloc(s->buffer, new_size);
68 s->buffer_size = s->buffer_index = 0;
69 return AVERROR(ENOMEM);
72 s->buffer_size = new_size;
77 static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
79 LAMEContext *s = avctx->priv_data;
81 #if FF_API_OLD_ENCODE_AUDIO
82 av_freep(&avctx->coded_frame);
84 av_freep(&s->samples_flt[0]);
85 av_freep(&s->samples_flt[1]);
88 ff_af_queue_close(&s->afq);
94 static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
96 LAMEContext *s = avctx->priv_data;
101 /* initialize LAME and get defaults */
102 if ((s->gfp = lame_init()) == NULL)
103 return AVERROR(ENOMEM);
105 lame_set_num_channels(s->gfp, avctx->channels);
106 lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
109 lame_set_in_samplerate (s->gfp, avctx->sample_rate);
110 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
112 /* algorithmic quality */
113 if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
114 lame_set_quality(s->gfp, 5);
116 lame_set_quality(s->gfp, avctx->compression_level);
119 if (avctx->flags & CODEC_FLAG_QSCALE) {
120 lame_set_VBR(s->gfp, vbr_default);
121 lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
124 lame_set_brate(s->gfp, avctx->bit_rate / 1000);
127 /* do not get a Xing VBR header frame from LAME */
128 lame_set_bWriteVbrTag(s->gfp,0);
130 /* bit reservoir usage */
131 lame_set_disable_reservoir(s->gfp, !s->reservoir);
133 /* set specified parameters */
134 if (lame_init_params(s->gfp) < 0) {
139 /* get encoder delay */
140 avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
141 ff_af_queue_init(avctx, &s->afq);
143 avctx->frame_size = lame_get_framesize(s->gfp);
145 #if FF_API_OLD_ENCODE_AUDIO
146 avctx->coded_frame = avcodec_alloc_frame();
147 if (!avctx->coded_frame) {
148 ret = AVERROR(ENOMEM);
153 /* allocate float sample buffers */
154 if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
156 for (ch = 0; ch < avctx->channels; ch++) {
157 s->samples_flt[ch] = av_malloc(avctx->frame_size *
158 sizeof(*s->samples_flt[ch]));
159 if (!s->samples_flt[ch]) {
160 ret = AVERROR(ENOMEM);
166 ret = realloc_buffer(s);
170 avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
174 mp3lame_encode_close(avctx);
178 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
179 lame_result = func(s->gfp, \
180 (const buf_type *)buf_name[0], \
181 (const buf_type *)buf_name[1], frame->nb_samples, \
182 s->buffer + s->buffer_index, \
183 s->buffer_size - s->buffer_index); \
186 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
187 const AVFrame *frame, int *got_packet_ptr)
189 LAMEContext *s = avctx->priv_data;
195 switch (avctx->sample_fmt) {
196 case AV_SAMPLE_FMT_S16P:
197 ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
199 case AV_SAMPLE_FMT_S32P:
200 ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
202 case AV_SAMPLE_FMT_FLTP:
203 if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
204 av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
205 return AVERROR(EINVAL);
207 for (ch = 0; ch < avctx->channels; ch++) {
208 s->fdsp.vector_fmul_scalar(s->samples_flt[ch],
209 (const float *)frame->data[ch],
211 FFALIGN(frame->nb_samples, 8));
213 ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
219 lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
220 BUFFER_SIZE - s->buffer_index);
222 if (lame_result < 0) {
223 if (lame_result == -1) {
224 av_log(avctx, AV_LOG_ERROR,
225 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
226 s->buffer_index, s->buffer_size - s->buffer_index);
230 s->buffer_index += lame_result;
231 ret = realloc_buffer(s);
233 av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
237 /* add current frame to the queue */
239 if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
243 /* Move 1 frame from the LAME buffer to the output packet, if available.
244 We have to parse the first frame header in the output buffer to
245 determine the frame size. */
246 if (s->buffer_index < 4)
248 if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
249 av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
252 len = hdr.frame_size;
253 av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
255 if (len <= s->buffer_index) {
256 if ((ret = ff_alloc_packet(avpkt, len))) {
257 av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
260 memcpy(avpkt->data, s->buffer, len);
261 s->buffer_index -= len;
262 memmove(s->buffer, s->buffer + len, s->buffer_index);
264 /* Get the next frame pts/duration */
265 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
274 #define OFFSET(x) offsetof(LAMEContext, x)
275 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
276 static const AVOption options[] = {
277 { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
281 static const AVClass libmp3lame_class = {
282 .class_name = "libmp3lame encoder",
283 .item_name = av_default_item_name,
285 .version = LIBAVUTIL_VERSION_INT,
288 static const AVCodecDefault libmp3lame_defaults[] = {
293 static const int libmp3lame_sample_rates[] = {
294 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
297 AVCodec ff_libmp3lame_encoder = {
298 .name = "libmp3lame",
299 .type = AVMEDIA_TYPE_AUDIO,
300 .id = AV_CODEC_ID_MP3,
301 .priv_data_size = sizeof(LAMEContext),
302 .init = mp3lame_encode_init,
303 .encode2 = mp3lame_encode_frame,
304 .close = mp3lame_encode_close,
305 .capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
306 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
309 AV_SAMPLE_FMT_NONE },
310 .supported_samplerates = libmp3lame_sample_rates,
311 .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
314 .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
315 .priv_class = &libmp3lame_class,
316 .defaults = libmp3lame_defaults,