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/*
* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
* This file is part of Libav.
*
* 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.
*
* 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @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 "mpegaudiodecheader.h"
#include <lame/lame.h>
#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
typedef struct Mp3AudioContext {
AVClass *class;
lame_global_flags *gfp;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
int reservoir;
} Mp3AudioContext;
static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame);
lame_close(s->gfp);
return 0;
}
static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
int ret;
if (avctx->channels > 2)
return AVERROR(EINVAL);
if ((s->gfp = lame_init()) == NULL)
return AVERROR(ENOMEM);
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);
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, avctx->channels > 1 ? 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_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
}
lame_set_bWriteVbrTag(s->gfp,0);
lame_set_disable_reservoir(s->gfp, !s->reservoir);
if (lame_init_params(s->gfp) < 0) {
ret = -1;
goto error;
}
avctx->frame_size = lame_get_framesize(s->gfp);
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
return 0;
error:
MP3lame_encode_close(avctx);
return ret;
}
static const int sSampleRates[] = {
44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
{
Mp3AudioContext *s = avctx->priv_data;
MPADecodeHeader hdr;
int len;
int lame_result;
if (data) {
if (avctx->channels > 1) {
lame_result = lame_encode_buffer_interleaved(s->gfp, data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
} else {
lame_result = lame_encode_buffer(s->gfp, data, data,
avctx->frame_size, s->buffer +
s->buffer_index, BUFFER_SIZE -
s->buffer_index);
}
} else {
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
if (lame_result < 0) {
if (lame_result == -1) {
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;
if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
return -1;
}
len = hdr.frame_size;
av_dlog(avctx, "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;
memmove(s->buffer, s->buffer + len, s->buffer_index);
return len;
} else
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 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 = (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,
};
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