libavcodec/libmp3lame.c - vendor/opensource/ffmpeg - Git at Google

 /*
  * Interface to libmp3lame for mp3 encoding
  * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
  *
  * 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 = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
 };
	/*
	* Interface to libmp3lame for mp3 encoding
	* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
	*
	* 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 = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
	.long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
	};