frame.c

《Visual C++小波变换技术与工程实践》作者：靳济芳。书上的代码。第3章：语音的去噪处理

/*
 * FAAC - Freeware Advanced Audio Coder
 * Copyright (C) 2001 Menno Bakker
 *
 * 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 2.1 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
 *
 * $Id: frame.c,v 1.4 2002/02/25 22:26:43 dmackie Exp $
 */

/*
 * CHANGES:
 *  2001/01/17: menno: Added frequency cut off filter.
 *  2001/02/28: menno: Added Temporal Noise Shaping.
 *  2001/03/05: menno: Added Long Term Prediction.
 *  2001/05/01: menno: Added backward prediction.
 *
 */

#include <stdio.h>
#include <stdlib.h>

#include "frame.h"
#include "coder.h"
#include "joint.h"
#include "channels.h"
#include "bitstream.h"
#include "filtbank.h"
#include "aacquant.h"
#include "util.h"
#include "huffman.h"
#include "psych.h"
#include "tns.h"
#include "ltp.h"
#include "backpred.h"


faacEncConfigurationPtr FAACAPI faacEncGetCurrentConfiguration(faacEncHandle hEncoder)
{
	faacEncConfigurationPtr config = &(hEncoder->config);

	return config;
}

int FAACAPI faacEncSetConfiguration(faacEncHandle hEncoder,
									faacEncConfigurationPtr config)
{
	hEncoder->config.allowMidside = config->allowMidside;
	hEncoder->config.useLfe = config->useLfe;
	hEncoder->config.useTns = config->useTns;
	hEncoder->config.aacObjectType = config->aacObjectType;
	hEncoder->config.mpegVersion = config->mpegVersion;

	/* No SSR supported for now */
	if (hEncoder->config.aacObjectType == SSR)
		return 0;

	/* LTP only with MPEG4 */
	if ((hEncoder->config.aacObjectType == LTP) && (hEncoder->config.mpegVersion != MPEG4))
		return 0;

	/* Re-init TNS for new profile */
	TnsInit(hEncoder);

	/* Check for correct bitrate */
	if (config->bitRate > MaxBitrate(hEncoder->sampleRate))
		return 0;
	if (config->bitRate < MinBitrate())
		return 0;

	/* Bitrate check passed */
	hEncoder->config.bitRate = config->bitRate;

	/* OK */
	return 1;
}

faacEncHandle FAACAPI faacEncOpen(unsigned long sampleRate,
							unsigned int numChannels,
							unsigned long *inputSamples,
							unsigned long *maxOutputBytes)
{
	unsigned int channel;
	faacEncHandle hEncoder;

	*inputSamples = 1024*numChannels;
	*maxOutputBytes = (6144/8)*numChannels;

	hEncoder = (faacEncStruct*)AllocMemory(sizeof(faacEncStruct));
	SetMemory(hEncoder, 0, sizeof(faacEncStruct));

	hEncoder->numChannels = numChannels;
	hEncoder->sampleRate = sampleRate;
	hEncoder->sampleRateIdx = GetSRIndex(sampleRate);

	/* 以缺省值初始化各变量*/
	hEncoder->frameNum = 0;
	hEncoder->flushFrame = 0;

	/* 缺省设置*/
	hEncoder->config.mpegVersion = MPEG4;
	hEncoder->config.aacObjectType = LTP;
	hEncoder->config.allowMidside = 1;
	hEncoder->config.useLfe = 0;
	hEncoder->config.useTns = 0;
	hEncoder->config.bitRate = 64000; 	/*缺省的每信道比特率 */
	hEncoder->config.bandWidth = 18000; 	/*缺省的带宽*/
#ifdef MPEG-4IP
	hEncoder->config.useAdts = 1;
#endif

	/* 根据已有参数寻找合适的采样频率*/
	hEncoder->srInfo = &srInfo[hEncoder->sampleRateIdx];

	for (channel = 0; channel < numChannels; channel++) {
		hEncoder->coderInfo[channel].prev_window_shape = SINE_WINDOW;
		hEncoder->coderInfo[channel].window_shape = SINE_WINDOW;
		hEncoder->coderInfo[channel].block_type = ONLY_LONG_WINDOW;
		hEncoder->coderInfo[channel].num_window_groups = 1;
		hEncoder->coderInfo[channel].window_group_length[0] = 1;

		hEncoder->coderInfo[channel].max_pred_sfb = GetMaxPredSfb(hEncoder->sampleRateIdx);

		hEncoder->sampleBuff[channel] = NULL;
		hEncoder->nextSampleBuff[channel] = NULL;
		hEncoder->next2SampleBuff[channel] = NULL;
		hEncoder->ltpTimeBuff[channel] = (double*)AllocMemory(2*BLOCK_LEN_LONG*sizeof(double));
		SetMemory(hEncoder->ltpTimeBuff[channel], 0, 2*BLOCK_LEN_LONG*sizeof(double));
	}

	/*初始化编码器各函数*/
	PsyInit(&hEncoder->gpsyInfo, hEncoder->psyInfo, hEncoder->numChannels,
		hEncoder->sampleRate, hEncoder->sampleRateIdx);

	FilterBankInit(hEncoder);

    TnsInit(hEncoder);

	LtpInit(hEncoder);

	PredInit(hEncoder);

	AACQuantizeInit(hEncoder->coderInfo, hEncoder->numChannels);

	HuffmanInit(hEncoder->coderInfo, hEncoder->numChannels);

	/*返回编码器句柄值*/
	return hEncoder;
}


int FAACAPI faacEncClose(faacEncHandle hEncoder)
{
	unsigned int channel;

	/* 编码器关闭函数*/
	PsyEnd(&hEncoder->gpsyInfo, hEncoder->psyInfo, hEncoder->numChannels);
//关闭滤波器空间
	FilterBankEnd(hEncoder);
//关闭Ltp编码器
	LtpEnd(hEncoder);
//关闭AAC量化器
	AACQuantizeEnd(hEncoder->coderInfo, hEncoder->numChannels);
//关闭霍夫曼编码器
	HuffmanEnd(hEncoder->coderInfo, hEncoder->numChannels);

	/* 释放保留的存储空间*/
	for (channel = 0; channel < hEncoder->numChannels; channel++) {
		if (hEncoder->ltpTimeBuff[channel]) FreeMemory(hEncoder->ltpTimeBuff[channel]);
		if (hEncoder->sampleBuff[channel]) FreeMemory(hEncoder->sampleBuff[channel]);
		if (hEncoder->nextSampleBuff[channel]) FreeMemory(hEncoder->nextSampleBuff[channel]);
	}

	/* 释放编码器空间*/
	if (hEncoder) FreeMemory(hEncoder);

	return 0;
}


int FAACAPI faacEncEncode(faacEncHandle hEncoder,
					 short *inputBuffer,
					 unsigned int samplesInput,
					 unsigned char *outputBuffer,
					 unsigned int bufferSize)
{
	unsigned int channel, i;
	int sb, frameBytes;
	unsigned int bitsToUse, offset;
	BitStream *bitStream;		 /*用于写音频帧的比特流数据*/
	TnsInfo *tnsInfo_for_LTP;
	TnsInfo *tnsDecInfo;

	/* 函数体内部参数复制*/
	ChannelInfo *channelInfo = hEncoder->channelInfo;
	CoderInfo *coderInfo = hEncoder->coderInfo;
	unsigned int numChannels = hEncoder->numChannels;
	unsigned int sampleRate = hEncoder->sampleRate;
	unsigned int aacObjectType = hEncoder->config.aacObjectType;
	unsigned int mpegVersion = hEncoder->config.mpegVersion;
	unsigned int useLfe = hEncoder->config.useLfe;
	unsigned int useTns = hEncoder->config.useTns;
	unsigned int allowMidside = hEncoder->config.allowMidside;
	unsigned int bitRate = hEncoder->config.bitRate;
	unsigned int bandWidth = hEncoder->config.bandWidth;

	/* 转到下一个音频帧*/
	hEncoder->frameNum++;

	if (samplesInput == 0)
		hEncoder->flushFrame++;

	/* 2次帧刷新后，所有的采样都被编码，返回0字节被写入*/
	if (hEncoder->flushFrame == 2)
		return 0;

	/* 决定信道的设置*/
	GetChannelInfo(channelInfo, numChannels, useLfe);

	/* 对当前的采样存储空间进行一次Update */
	for (channel = 0; channel < numChannels; channel++) {
		if (hEncoder->sampleBuff[channel]) {
			for(i = 0; i < FRAME_LEN; i++) {
				hEncoder->ltpTimeBuff[channel][i] =	hEncoder->sampleBuff[channel][i];
			}
		}
		if (hEncoder->nextSampleBuff[channel]) {
			for(i = 0; i < FRAME_LEN; i++) {
				hEncoder->ltpTimeBuff[channel][FRAME_LEN + i] =
					hEncoder->nextSampleBuff[channel][i];
			}
		}

		if (hEncoder->sampleBuff[channel])
			FreeMemory(hEncoder->sampleBuff[channel]);
		hEncoder->sampleBuff[channel] = hEncoder->nextSampleBuff[channel];
		hEncoder->nextSampleBuff[channel] = (double*)AllocMemory(FRAME_LEN*sizeof(double));

		if (samplesInput == 0) { 	/* 开始刷新存储区*/
			for (i = 0; i < FRAME_LEN; i++)
				hEncoder->nextSampleBuff[channel][i] = 0.0;
		} else {
			for (i = 0; i < (int)(samplesInput/numChannels); i++)
				hEncoder->nextSampleBuff[channel][i] = 
					(double)inputBuffer[(i*numChannels)+channel];
			for (i = (int)(samplesInput/numChannels); i < FRAME_LEN; i++)
				hEncoder->nextSampleBuff[channel][i] = 0.0;
		}

		/* 心理声学部分*/
		/* 更新存储区并对新的采样值做快速Fourier变换*/
		PsyBufferUpdate(&hEncoder->gpsyInfo, &hEncoder->psyInfo[channel],
			hEncoder->nextSampleBuff[channel]);
	}

	if (hEncoder->frameNum <= 1) /* 仍然填写输出存储区*/
		return 0;

	/* 心理声学部分*/
	PsyCalculate(channelInfo, &hEncoder->gpsyInfo, hEncoder->psyInfo,
		hEncoder->srInfo->cb_width_long, hEncoder->srInfo->num_cb_long,
		hEncoder->srInfo->cb_width_short,
		hEncoder->srInfo->num_cb_short, numChannels);

	BlockSwitch(coderInfo, hEncoder->psyInfo, numChannels);

	/* AAC滤波器簇以及MDCT */
	for (channel = 0; channel < numChannels; channel++) {
		int k;

		FilterBank(hEncoder,
			&coderInfo[channel],
			hEncoder->sampleBuff[channel],
			hEncoder->freqBuff[channel],