/*
* 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/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.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;
int buffer_index;
int buffer_size;
int reservoir;
int joint_stereo;
float *samples_flt[2];
AudioFrameQueue afq;
AVFloatDSPContext fdsp;
} LAMEContext;
static int realloc_buffer(LAMEContext *s)
{
if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
uint8_t *tmp;
int new_size = s->buffer_index + 2 * BUFFER_SIZE;
av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
new_size);
tmp = av_realloc(s->buffer, new_size);
if (!tmp) {
av_freep(&s->buffer);
s->buffer_size = s->buffer_index = 0;
return AVERROR(ENOMEM);
}
s->buffer = tmp;
s->buffer_size = new_size;
}
return 0;
}
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
av_freep(&s->samples_flt[0]);
av_freep(&s->samples_flt[1]);
av_freep(&s->buffer);
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);
lame_set_num_channels(s->gfp, avctx->channels);
lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : 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) { // VBR
lame_set_VBR(s->gfp, vbr_default);
lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
} else {
if (avctx->bit_rate) // CBR
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);
/* allocate float sample buffers */
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
int ch;
for (ch = 0; ch < avctx->channels; ch++) {
s->samples_flt[ch] = av_malloc(avctx->frame_size *
sizeof(*s->samples_flt[ch]));
if (!s->samples_flt[ch]) {
ret = AVERROR(ENOMEM);
goto error;
}
}
}
ret = realloc_buffer(s);
if (ret < 0)
goto error;
avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
return 0;
error:
mp3lame_encode_close(avctx);
return ret;
}
#define ENCODE_BUFFER(func, buf_type, buf_name) do { \
lame_result = func(s->gfp, \
(const buf_type *)buf_name[0], \
(const buf_type *)buf_name[1], frame->nb_samples, \
s->buffer + s->buffer_index, \
s->buffer_size - s->buffer_index); \
} while (0)
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, ch;
int lame_result;
if (frame) {
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_S16P:
ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
break;
case AV_SAMPLE_FMT_S32P:
ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
break;
case AV_SAMPLE_FMT_FLTP:
if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
return AVERROR(EINVAL);
}
for (ch = 0; ch < avctx->channels; ch++) {
s->fdsp.vector_fmul_scalar(s->samples_flt[ch],
(const float *)frame->data[ch],
32768.0f,
FFALIGN(frame->nb_samples, 8));
}
ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
break;
default:
return AVERROR_BUG;
}
} else {
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
s->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, s->buffer_size - s->buffer_index);
}
return -1;
}
s->buffer_index += lame_result;
ret = realloc_buffer(s);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
return ret;
}
/* 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)) < 0)
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, { .i64 = 1 }, 0, 1, AE },
{ "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 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",
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_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_S32P,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = libmp3lame_sample_rates,
.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
0 },
.priv_class = &libmp3lame_class,
.defaults = libmp3lame_defaults,
};