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ffmpeg/libavcodec/libmp3lame.c

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/*
* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
* This file is part of FFmpeg.
*
* FFmpeg 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,
* 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
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Interface to libmp3lame for mp3 encoding.
*/
#include <lame/lame.h>
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
typedef struct LAMEContext {
AVClass *class;
AVCodecContext *avctx;
lame_global_flags *gfp;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
int reservoir;
void *planar_samples[2];
AudioFrameQueue afq;
} LAMEContext;
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&s->planar_samples[0]);
av_freep(&s->planar_samples[1]);
ff_af_queue_close(&s->afq);
lame_close(s->gfp);
return 0;
}
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
int ret;
s->avctx = avctx;
/* initialize LAME and get defaults */
if ((s->gfp = lame_init()) == NULL)
return AVERROR(ENOMEM);
/* channels */
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR,
"Invalid number of channels %d, must be <= 2\n", avctx->channels);
ret = AVERROR(EINVAL);
goto error;
}
lame_set_num_channels(s->gfp, avctx->channels);
lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
/* sample rate */
lame_set_in_samplerate (s->gfp, avctx->sample_rate);
lame_set_out_samplerate(s->gfp, avctx->sample_rate);
/* algorithmic quality */
if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
lame_set_quality(s->gfp, 5);
else
lame_set_quality(s->gfp, avctx->compression_level);
/* rate control */
if (avctx->flags & CODEC_FLAG_QSCALE) {
lame_set_VBR(s->gfp, vbr_default);
lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
} else {
if (avctx->bit_rate)
lame_set_brate(s->gfp, avctx->bit_rate / 1000);
}
/* do not get a Xing VBR header frame from LAME */
lame_set_bWriteVbrTag(s->gfp,0);
/* bit reservoir usage */
lame_set_disable_reservoir(s->gfp, !s->reservoir);
/* set specified parameters */
if (lame_init_params(s->gfp) < 0) {
ret = -1;
goto error;
}
/* get encoder delay */
avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
ff_af_queue_init(avctx, &s->afq);
avctx->frame_size = lame_get_framesize(s->gfp);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
/* sample format */
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
int ch;
for (ch = 0; ch < avctx->channels; ch++) {
s->planar_samples[ch] = av_malloc(avctx->frame_size *
av_get_bytes_per_sample(avctx->sample_fmt));
if (!s->planar_samples[ch]) {
ret = AVERROR(ENOMEM);
goto error;
}
}
}
return 0;
error:
mp3lame_encode_close(avctx);
return ret;
}
#define DEINTERLEAVE(type, scale) do { \
int ch, i; \
for (ch = 0; ch < s->avctx->channels; ch++) { \
const type *input = samples; \
type *output = s->planar_samples[ch]; \
input += ch; \
for (i = 0; i < nb_samples; i++) { \
output[i] = *input * scale; \
input += s->avctx->channels; \
} \
} \
} while (0)
static int encode_frame_int16(LAMEContext *s, void *samples, int nb_samples)
{
if (s->avctx->channels > 1) {
return lame_encode_buffer_interleaved(s->gfp, samples,
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
} else {
return lame_encode_buffer(s->gfp, samples, NULL, nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
}
static int encode_frame_int32(LAMEContext *s, void *samples, int nb_samples)
{
DEINTERLEAVE(int32_t, 1);
return lame_encode_buffer_int(s->gfp,
s->planar_samples[0], s->planar_samples[1],
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
static int encode_frame_float(LAMEContext *s, void *samples, int nb_samples)
{
DEINTERLEAVE(float, 32768.0f);
return lame_encode_buffer_float(s->gfp,
s->planar_samples[0], s->planar_samples[1],
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr;
int len, ret;
int lame_result;
if (frame) {
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_S16:
lame_result = encode_frame_int16(s, frame->data[0], frame->nb_samples);
break;
case AV_SAMPLE_FMT_S32:
lame_result = encode_frame_int32(s, frame->data[0], frame->nb_samples);
break;
case AV_SAMPLE_FMT_FLT:
lame_result = encode_frame_float(s, frame->data[0], frame->nb_samples);
break;
default:
return AVERROR_BUG;
}
} 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;
/* add current frame to the queue */
if (frame) {
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
}
/* Move 1 frame from the LAME buffer to the output packet, if available.
We have to parse the first frame header in the output buffer to
determine the frame size. */
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) {
if ((ret = ff_alloc_packet2(avctx, avpkt, len)))
return ret;
memcpy(avpkt->data, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index);
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = len;
*got_packet_ptr = 1;
}
return 0;
}
#define OFFSET(x) offsetof(LAMEContext, 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,
};
static const AVCodecDefault libmp3lame_defaults[] = {
{ "b", "0" },
{ NULL },
};
static const int libmp3lame_sample_rates[] = {
44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
AVCodec ff_libmp3lame_encoder = {
.name = "libmp3lame",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_MP3,
.priv_data_size = sizeof(LAMEContext),
.init = mp3lame_encode_init,
.encode2 = mp3lame_encode_frame,
.close = mp3lame_encode_close,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = libmp3lame_sample_rates,
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class,
.defaults = libmp3lame_defaults,
};
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