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ffmpeg / libavcodec / libmp3lame.c @ 2912e87a

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/*
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 * Interface to libmp3lame for mp3 encoding
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 * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
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 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with Libav; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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/**
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 * @file
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 * Interface to libmp3lame for mp3 encoding.
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 */
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#include "avcodec.h"
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#include "mpegaudio.h"
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#include <lame/lame.h>
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#define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
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typedef struct Mp3AudioContext {
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    lame_global_flags *gfp;
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    int stereo;
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    uint8_t buffer[BUFFER_SIZE];
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    int buffer_index;
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} Mp3AudioContext;
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static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
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{
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    Mp3AudioContext *s = avctx->priv_data;
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    if (avctx->channels > 2)
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        return -1;
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    s->stereo = avctx->channels > 1 ? 1 : 0;
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    if ((s->gfp = lame_init()) == NULL)
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        goto err;
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    lame_set_in_samplerate(s->gfp, avctx->sample_rate);
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    lame_set_out_samplerate(s->gfp, avctx->sample_rate);
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    lame_set_num_channels(s->gfp, avctx->channels);
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    if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
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        lame_set_quality(s->gfp, 5);
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    } else {
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        lame_set_quality(s->gfp, avctx->compression_level);
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    }
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    lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
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    lame_set_brate(s->gfp, avctx->bit_rate/1000);
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    if(avctx->flags & CODEC_FLAG_QSCALE) {
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        lame_set_brate(s->gfp, 0);
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        lame_set_VBR(s->gfp, vbr_default);
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        lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
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    }
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    lame_set_bWriteVbrTag(s->gfp,0);
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    lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
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    if (lame_init_params(s->gfp) < 0)
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        goto err_close;
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    avctx->frame_size = lame_get_framesize(s->gfp);
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    avctx->coded_frame= avcodec_alloc_frame();
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    avctx->coded_frame->key_frame= 1;
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    return 0;
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err_close:
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    lame_close(s->gfp);
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err:
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    return -1;
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}
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static const int sSampleRates[] = {
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    44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
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};
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static const int sBitRates[2][3][15] = {
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    {   {  0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
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        {  0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
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        {  0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
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    },
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    {   {  0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
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        {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
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        {  0,  8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
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    },
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};
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static const int sSamplesPerFrame[2][3] =
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{
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    {  384,     1152,    1152 },
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    {  384,     1152,     576 }
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};
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static const int sBitsPerSlot[3] = {
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    32,
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    8,
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    8
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};
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static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
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{
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    uint32_t header = AV_RB32(data);
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    int layerID = 3 - ((header >> 17) & 0x03);
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    int bitRateID = ((header >> 12) & 0x0f);
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    int sampleRateID = ((header >> 10) & 0x03);
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    int bitsPerSlot = sBitsPerSlot[layerID];
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    int isPadded = ((header >> 9) & 0x01);
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    static int const mode_tab[4]= {2,3,1,0};
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    int mode= mode_tab[(header >> 19) & 0x03];
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    int mpeg_id= mode>0;
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    int temp0, temp1, bitRate;
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    if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
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        return -1;
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    }
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    if(!samplesPerFrame) samplesPerFrame= &temp0;
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    if(!sampleRate     ) sampleRate     = &temp1;
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//    *isMono = ((header >>  6) & 0x03) == 0x03;
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    *sampleRate = sSampleRates[sampleRateID]>>mode;
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    bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
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    *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
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//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
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    return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
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}
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static int MP3lame_encode_frame(AVCodecContext *avctx,
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                                unsigned char *frame, int buf_size, void *data)
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{
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    Mp3AudioContext *s = avctx->priv_data;
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    int len;
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    int lame_result;
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    /* lame 3.91 dies on '1-channel interleaved' data */
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    if(data){
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        if (s->stereo) {
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            lame_result = lame_encode_buffer_interleaved(
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                s->gfp,
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                data,
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                avctx->frame_size,
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                s->buffer + s->buffer_index,
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                BUFFER_SIZE - s->buffer_index
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                );
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        } else {
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            lame_result = lame_encode_buffer(
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                s->gfp,
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                data,
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                data,
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                avctx->frame_size,
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                s->buffer + s->buffer_index,
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                BUFFER_SIZE - s->buffer_index
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                );
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        }
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    }else{
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        lame_result= lame_encode_flush(
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                s->gfp,
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                s->buffer + s->buffer_index,
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                BUFFER_SIZE - s->buffer_index
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                );
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    }
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    if(lame_result < 0){
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        if(lame_result==-1) {
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            /* output buffer too small */
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            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);
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        }
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        return -1;
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    }
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    s->buffer_index += lame_result;
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    if(s->buffer_index<4)
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        return 0;
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    len= mp3len(s->buffer, NULL, NULL);
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//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
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    if(len <= s->buffer_index){
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        memcpy(frame, s->buffer, len);
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        s->buffer_index -= len;
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        memmove(s->buffer, s->buffer+len, s->buffer_index);
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            //FIXME fix the audio codec API, so we do not need the memcpy()
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/*for(i=0; i<len; i++){
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    av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
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}*/
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        return len;
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    }else
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        return 0;
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}
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static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
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{
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    Mp3AudioContext *s = avctx->priv_data;
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    av_freep(&avctx->coded_frame);
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    lame_close(s->gfp);
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    return 0;
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}
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AVCodec ff_libmp3lame_encoder = {
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    "libmp3lame",
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    AVMEDIA_TYPE_AUDIO,
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    CODEC_ID_MP3,
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    sizeof(Mp3AudioContext),
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    MP3lame_encode_init,
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    MP3lame_encode_frame,
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    MP3lame_encode_close,
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    .capabilities= CODEC_CAP_DELAY,
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    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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    .supported_samplerates= sSampleRates,
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    .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
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};