mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-28 20:53:54 +02:00
246 lines
8.1 KiB
C
246 lines
8.1 KiB
C
/*
|
|
* 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 <lame/lame.h>
|
|
|
|
#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)
|
|
{
|
|
Mp3AudioContext *s = avctx->priv_data;
|
|
|
|
if (avctx->channels > 2)
|
|
return -1;
|
|
|
|
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);
|
|
#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;
|
|
|
|
avctx->frame_size = lame_get_framesize(s->gfp);
|
|
avctx->coded_frame = avcodec_alloc_frame();
|
|
avctx->coded_frame->key_frame = 1;
|
|
|
|
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) {
|
|
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);
|
|
}
|
|
} 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) {
|
|
/* 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(&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,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.supported_samplerates = sSampleRates,
|
|
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
|
|
.priv_class = &libmp3lame_class,
|
|
};
|