* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file mp3lameaudio.c
+ * @file
* Interface to libmp3lame for mp3 encoding.
*/
+#include "libavutil/intreadwrite.h"
+#include "libavutil/log.h"
+#include "libavutil/opt.h"
#include "avcodec.h"
#include "mpegaudio.h"
#include <lame/lame.h>
-#define BUFFER_SIZE (7200 + MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
+#define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
typedef struct Mp3AudioContext {
+ AVClass *class;
lame_global_flags *gfp;
int stereo;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
+ int reservoir;
} Mp3AudioContext;
static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
lame_set_in_samplerate(s->gfp, avctx->sample_rate);
lame_set_out_samplerate(s->gfp, avctx->sample_rate);
lame_set_num_channels(s->gfp, avctx->channels);
- /* lame 3.91 dies on quality != 5 */
- lame_set_quality(s->gfp, 5);
- /* lame 3.91 doesn't work in mono */
- lame_set_mode(s->gfp, JOINT_STEREO);
+ if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
+ lame_set_quality(s->gfp, 5);
+ } else {
+ lame_set_quality(s->gfp, avctx->compression_level);
+ }
+ lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
lame_set_brate(s->gfp, avctx->bit_rate/1000);
if(avctx->flags & CODEC_FLAG_QSCALE) {
lame_set_brate(s->gfp, 0);
lame_set_VBR(s->gfp, vbr_default);
- lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
+ lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
}
lame_set_bWriteVbrTag(s->gfp,0);
- lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
+#if FF_API_LAME_GLOBAL_OPTS
+ s->reservoir = avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR;
+#endif
+ lame_set_disable_reservoir(s->gfp, !s->reservoir);
if (lame_init_params(s->gfp) < 0)
goto err_close;
return -1;
}
-static const int sSampleRates[3] = {
- 44100, 48000, 32000
+static const int sSampleRates[] = {
+ 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
static const int sBitRates[2][3][15] = {
);
}
- if(lame_result==-1) {
- /* output buffer too small */
- 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);
- return 0;
+ if(lame_result < 0){
+ if(lame_result==-1) {
+ /* output buffer too small */
+ 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);
+ }
+ return -1;
}
s->buffer_index += lame_result;
if(s->buffer_index<4)
return 0;
- len= mp3len(s->buffer, NULL, NULL);
+ len= mp3len(s->buffer, NULL, NULL);
//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
- if(len <= s->buffer_index){
- memcpy(frame, s->buffer, len);
- s->buffer_index -= len;
+ if(len <= s->buffer_index){
+ memcpy(frame, s->buffer, len);
+ s->buffer_index -= len;
- memmove(s->buffer, s->buffer+len, s->buffer_index);
+ memmove(s->buffer, s->buffer+len, s->buffer_index);
//FIXME fix the audio codec API, so we do not need the memcpy()
/*for(i=0; i<len; i++){
av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
}*/
- return len;
- }else
- return 0;
+ return len;
+ }else
+ return 0;
}
static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
return 0;
}
+#define OFFSET(x) offsetof(Mp3AudioContext, x)
+#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
+static const AVOption options[] = {
+ { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
+ { NULL },
+};
+
+static const AVClass libmp3lame_class = {
+ .class_name = "libmp3lame encoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
-AVCodec libmp3lame_encoder = {
- "libmp3lame",
- CODEC_TYPE_AUDIO,
- CODEC_ID_MP3,
- sizeof(Mp3AudioContext),
- MP3lame_encode_init,
- MP3lame_encode_frame,
- MP3lame_encode_close,
+AVCodec ff_libmp3lame_encoder = {
+ .name = "libmp3lame",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_MP3,
+ .priv_data_size = sizeof(Mp3AudioContext),
+ .init = MP3lame_encode_init,
+ .encode = MP3lame_encode_frame,
+ .close = MP3lame_encode_close,
.capabilities= CODEC_CAP_DELAY,
- .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+ .supported_samplerates= sSampleRates,
.long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
+ .priv_class = &libmp3lame_class,
};
projects / ffmpeg / blobdiff