view libmp3lame.c @ 10602:987373501407 libavcodec

Fix an issue uncovered by commit 20623: The init functions of mpc7 and mpc8 check whether the vlc has been initialized already and return early if this is the case (eg by calling init a second time). But avctx->sample_fmt and channel_layout is set after the vlc initialization, causing it not to be set on the second call of init. Move all manipulations of avctx before the initialization of the vlc, so that it is always set.

/*
 * Interface to libmp3lame for mp3 encoding
 * Copyright (c) 2002 Lennert Buytenhek <[e-mail]>
 *
 * 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 libavcodec/libmp3lame.c
 * Interface to libmp3lame for mp3 encoding.
 */

#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 {
    lame_global_flags *gfp;
    int stereo;
    uint8_t buffer[BUFFER_SIZE];
    int buffer_index;
} 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 3.91 doesn't work in mono */
    lame_set_mode(s->gfp, JOINT_STEREO);
    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_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
    }
    lame_set_bWriteVbrTag(s->gfp,0);
    lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
    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[3] = {
    44100, 48000,  32000
};

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;
}


AVCodec libmp3lame_encoder = {
    "libmp3lame",
    CODEC_TYPE_AUDIO,
    CODEC_ID_MP3,
    sizeof(Mp3AudioContext),
    MP3lame_encode_init,
    MP3lame_encode_frame,
    MP3lame_encode_close,
    .capabilities= CODEC_CAP_DELAY,
    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
    .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
};