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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 "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 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
typedef struct Mp3AudioContext {
AVClass *class;
lame_global_flags *gfp;
int stereo;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
struct {
int *left;
int *right;
} s32_data;
int reservoir;
} Mp3AudioContext;
static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR,
"Invalid number of channels %d, must be <= 2\n", avctx->channels);
return AVERROR(EINVAL);
}
s->stereo = avctx->channels > 1 ? 1 : 0;
if ((s->gfp = lame_init()) == NULL)
goto err;
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, 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_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)
goto err_close;
avctx->frame_size = lame_get_framesize(s->gfp);
if(!(avctx->coded_frame= avcodec_alloc_frame())) {
lame_close(s->gfp);
return AVERROR(ENOMEM);
}
if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
int nelem = 2 * avctx->frame_size;
if(! (s->s32_data.left = av_malloc(nelem * sizeof(int)))) {
av_freep(&avctx->coded_frame);
lame_close(s->gfp);
return AVERROR(ENOMEM);
}
s->s32_data.right = s->s32_data.left + avctx->frame_size;
}
return 0;
err_close:
lame_close(s->gfp);
err:
return -1;
}
static const int sSampleRates[] = {
44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
static const int sBitRates[2][3][15] = {
{
{ 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
{ 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
{ 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 }
},
{
{ 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 },
{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 }
},
};
static const int sSamplesPerFrame[2][3] = {
{ 384, 1152, 1152 },
{ 384, 1152, 576 }
};
static const int sBitsPerSlot[3] = { 32, 8, 8 };
static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
{
uint32_t header = AV_RB32(data);
int layerID = 3 - ((header >> 17) & 0x03);
int bitRateID = ((header >> 12) & 0x0f);
int sampleRateID = ((header >> 10) & 0x03);
int bitsPerSlot = sBitsPerSlot[layerID];
int isPadded = ((header >> 9) & 0x01);
static int const mode_tab[4] = { 2, 3, 1, 0 };
int mode = mode_tab[(header >> 19) & 0x03];
int mpeg_id = mode > 0;
int temp0, temp1, bitRate;
if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
sampleRateID == 3) {
return -1;
}
if (!samplesPerFrame)
samplesPerFrame = &temp0;
if (!sampleRate)
sampleRate = &temp1;
//*isMono = ((header >> 6) & 0x03) == 0x03;
*sampleRate = sSampleRates[sampleRateID] >> mode;
bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
//av_log(NULL, AV_LOG_DEBUG,
// "sr:%d br:%d spf:%d l:%d m:%d\n",
// *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
}
static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
{
Mp3AudioContext *s = avctx->priv_data;
int len;
int lame_result;
/* lame 3.91 dies on '1-channel interleaved' data */
if (!data){
lame_result= lame_encode_flush(
s->gfp,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
#if 2147483647 == INT_MAX
}else if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt){
if (s->stereo) {
int32_t *rp = data;
int32_t *mp = rp + 2*avctx->frame_size;
int *wpl = s->s32_data.left;
int *wpr = s->s32_data.right;
while (rp < mp) {
*wpl++ = *rp++;
*wpr++ = *rp++;
}
lame_result = lame_encode_buffer_int(
s->gfp,
s->s32_data.left,
s->s32_data.right,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
} else {
lame_result = lame_encode_buffer_int(
s->gfp,
data,
data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
}
#endif
}else{
if (s->stereo) {
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
);
}
}
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);
//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;
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;
}
static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
av_freep(&s->s32_data.left);
av_freep(&avctx->coded_frame);
lame_close(s->gfp);
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,
#if 2147483647 == INT_MAX
AV_SAMPLE_FMT_S32,
#endif
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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