view libmp3lame.c @ 5306:abc5c130b448 libavcodec

AC-3 decoder, soc revision 32, Jul 17 09:37:32 2006 UTC by cloud9 Latest commit. There is no error in parsing and or recovering transform coefficients. Double checked with ac3dec. Getting consistent results with the bit allocation routine and transform coefficients. The code is able to parse valid ac3 bitstreams without error from start to end. I have also implemented the imdct when block switching is not enabled. However, can anybody provide an insight into how to convert float samples to int16_t ? lrint is of no help cuz it produces output -1, 0 or 1 whereas the output should be between -32768 to 32767.

/*
 * 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 mp3lameaudio.c
 * Interface to libmp3lame for mp3 encoding.
 */

#include "avcodec.h"
#include "mpegaudio.h"
#include <lame/lame.h>

#define BUFFER_SIZE (2*MPA_FRAME_SIZE)
typedef struct Mp3AudioContext {
    lame_global_flags *gfp;
    int stereo;
    uint8_t buffer[BUFFER_SIZE];
    int buffer_index;
} Mp3AudioContext;

static 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);
    /* lame 3.91 dies on quality != 5 */
    lame_set_quality(s->gfp, 5);
    /* 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);
    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==-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 0;
    }

    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 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,
};