view mp3lameaudio.c @ 2497:69adfbbdcdeb libavcodec

- samples from mplayer ftp in the "adv" profile seem to have profile=2, which isn't the advanced one; and indeed, using adv. profile parser fails. Using normal parser works, and that's what is done - attempt at taking care of stride for NORM2 bitplane decoding - duplication of much code from msmpeg4.c; this code isn't yet used, but goes down as far as the block layer (mainly Transform Type stuff, the remains are wild editing without checking). Unusable yet, and lacks the AC decoding (but a step further in bitstream parsing) patch by anonymous

/*
 * Interface to libmp3lame for mp3 encoding
 * Copyright (c) 2002 Lennert Buytenhek <[e-mail]>
 *
 * This library 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 of the License, or (at your option) any later version.
 *
 * This library 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 this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  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);
        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)
{
    uint8_t *dataTmp = (uint8_t *)data;
    uint32_t header = ( (uint32_t)dataTmp[0] << 24 ) | ( (uint32_t)dataTmp[1] << 16 ) | ( (uint32_t)dataTmp[2] << 8 ) | (uint32_t)dataTmp[3];
    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;
}

int MP3lame_encode_frame(AVCodecContext *avctx,
                     unsigned char *frame, int buf_size, void *data)
{
	Mp3AudioContext *s = avctx->priv_data;
	int len, i;
	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 dont need the memcpy()
/*for(i=0; i<len; i++){
    av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
}*/
            return len;
        }else
            return 0;
}

int MP3lame_encode_close(AVCodecContext *avctx)
{
	Mp3AudioContext *s = avctx->priv_data;
    
        av_freep(&avctx->coded_frame);

	lame_close(s->gfp);
	return 0;
}


AVCodec mp3lame_encoder = {
    "mp3",
    CODEC_TYPE_AUDIO,
    CODEC_ID_MP3,
    sizeof(Mp3AudioContext),
    MP3lame_encode_init,
    MP3lame_encode_frame,
    MP3lame_encode_close,
    .capabilities= CODEC_CAP_DELAY,
};