/*
* 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 1.60 2004/04/13 13:47:33 danchr 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 <assert.h>
#include <math.h>
#include "frame.h"
#include "coder.h"
#include "midside.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"
#include "version.h"
#if FAAC_RELEASE
static char *libfaacName = FAAC_VERSION;
#else
static char *libfaacName = FAAC_VERSION " (" __DATE__ ") UNSTABLE";
#endif
static char *libCopyright =
"FAAC - Freeware Advanced Audio Coder (http://www.audiocoding.com/)\n"
" Copyright (C) 1999,2000,2001 Menno Bakker\n"
" Copyright (C) 2002,2003 Krzysztof Nikiel\n"
"This software is based on the ISO MPEG-4 reference source code.\n";
static const psymodellist_t psymodellist[] = {
{&psymodel2, "knipsycho psychoacoustic"},
{NULL}
};
static SR_INFO srInfo[12+1];
// base bandwidth for q=100
static const int bwbase = 16000;
// bandwidth multiplier (for quantiser)
static const int bwmult = 120;
// max bandwidth/samplerate ratio
static const double bwfac = 0.45;
int FAACAPI faacEncGetVersion( char **faac_id_string,
char **faac_copyright_string)
{
if (faac_id_string)
*faac_id_string = libfaacName;
if (faac_copyright_string)
*faac_copyright_string = libCopyright;
return FAAC_CFG_VERSION;
}
int FAACAPI faacEncGetDecoderSpecificInfo(faacEncHandle hEncoder,unsigned char** ppBuffer,unsigned long* pSizeOfDecoderSpecificInfo)
{
BitStream* pBitStream = NULL;
if((hEncoder == NULL) || (ppBuffer == NULL) || (pSizeOfDecoderSpecificInfo == NULL)) {
return -1;
}
if(hEncoder->config.mpegVersion == MPEG2){
return -2; /* not supported */
}
*pSizeOfDecoderSpecificInfo = 2;
*ppBuffer = malloc(2);
if(*ppBuffer != NULL){
memset(*ppBuffer,0,*pSizeOfDecoderSpecificInfo);
pBitStream = OpenBitStream(*pSizeOfDecoderSpecificInfo, *ppBuffer);
PutBit(pBitStream, hEncoder->config.aacObjectType + 1, 5);
/*
temporary fix,
when object type defines will be changed to values defined by ISO 14496-3
"+ 1" shall be removed
/AV
*/
PutBit(pBitStream, hEncoder->sampleRateIdx, 4);
PutBit(pBitStream, hEncoder->numChannels, 4);
CloseBitStream(pBitStream);
return 0;
} else {
return -3;
}
}
faacEncConfigurationPtr FAACAPI faacEncGetCurrentConfiguration(faacEncHandle hEncoder)
{
faacEncConfigurationPtr config = &(hEncoder->config);
return config;
}
int FAACAPI faacEncSetConfiguration(faacEncHandle hEncoder,
faacEncConfigurationPtr config)
{
int i;
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;
hEncoder->config.outputFormat = config->outputFormat;
hEncoder->config.inputFormat = config->inputFormat;
hEncoder->config.shortctl = config->shortctl;
assert((hEncoder->config.outputFormat == 0) || (hEncoder->config.outputFormat == 1));
switch( hEncoder->config.inputFormat )
{
case FAAC_INPUT_16BIT:
//case FAAC_INPUT_24BIT:
case FAAC_INPUT_32BIT:
case FAAC_INPUT_FLOAT:
break;
default:
return 0;
break;
}
/* 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 0
if (config->bitRate < MinBitrate())
return 0;
#endif
if (config->bitRate && !config->bandWidth)
{
static struct {
int rate; // per channel at 44100 sampling frequency
int cutoff;
} rates[] = {
{29500, 5000},
{37500, 7000},
{47000, 10000},
{64000, 16000},
{76000, 20000},
{0, 0}
};
int f0, f1;
int r0, r1;
config->quantqual = 100;
config->bitRate = (double)config->bitRate * 44100 / hEncoder->sampleRate;
f0 = f1 = rates[0].cutoff;
r0 = r1 = rates[0].rate;
for (i = 0; rates[i].rate; i++)
{
f0 = f1;
f1 = rates[i].cutoff;
r0 = r1;
r1 = rates[i].rate;
if (rates[i].rate >= config->bitRate)
break;
}
if (config->bitRate > r1)
config->bitRate = r1;
if (config->bitRate < r0)
config->bitRate = r0;
if (f1 > f0)
config->bandWidth =
pow((double)config->bitRate / r1,
log((double)f1 / f0) / log ((double)r1 / r0)) * (double)f1;
else
config->bandWidth = f1;
config->bandWidth =
(double)config->bandWidth * hEncoder->sampleRate / 44100;
config->bitRate = (double)config->bitRate * hEncoder->sampleRate / 44100;
if (config->bandWidth > bwbase)
config->bandWidth = bwbase;
}
hEncoder->config.bitRate = config->bitRate;
if (!config->bandWidth)
{
config->bandWidth = (config->quantqual - 100) * bwmult + bwbase;
}
hEncoder->config.bandWidth = config->bandWidth;
// check bandwidth
if (hEncoder->config.bandWidth < 100)
hEncoder->config.bandWidth = 100;
if (hEncoder->config.bandWidth > (hEncoder->sampleRate / 2))
hEncoder->config.bandWidth = hEncoder->sampleRate / 2;
if (config->quantqual > 500)
config->quantqual = 500;
if (config->quantqual < 10)
config->quantqual = 10;
hEncoder->config.quantqual = config->quantqual;
/* set quantization quality */
hEncoder->aacquantCfg.quality = config->quantqual;
// reset psymodel
hEncoder->psymodel->PsyEnd(&hEncoder->gpsyInfo, hEncoder->psyInfo, hEncoder->numChannels);
if (config->psymodelidx >= (sizeof(psymodellist) / sizeof(psymodellist[0]) - 1))
config->psymodelidx = (sizeof(psymodellist) / sizeof(psymodellist[0])) - 2;
hEncoder->config.psymodelidx = config->psymodelidx;
hEncoder->psymodel = psymodellist[hEncoder->config.psymodelidx].model;
hEncoder->psymodel->PsyInit(&hEncoder->gpsyInfo, hEncoder->psyInfo, hEncoder->numChannels,
hEncoder->sampleRate, hEncoder->srInfo->cb_width_long,
hEncoder->srInfo->num_cb_long, hEncoder->srInfo->cb_width_short,
hEncoder->srInfo->num_cb_short);
/* load channel_map */
for( i = 0; i < 64; i++ )
hEncoder->config.channel_map[i] = config->channel_map[i];
/* 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);
/* Initialize variables to default values */
hEncoder->frameNum = 0;
hEncoder->flushFrame = 0;
/* Default configuration */
hEncoder->config.version = FAAC_CFG_VERSION;
hEncoder->config.name = libfaacName;
hEncoder->config.copyright = libCopyright;
hEncoder->config.mpegVersion = MPEG4;
hEncoder->config.aacObjectType = LTP;
hEncoder->config.allowMidside = 1;
hEncoder->config.useLfe = 1;
hEncoder->config.useTns = 0;
hEncoder->config.bitRate = 0; /* default bitrate / channel */
hEncoder->config.bandWidth = bwfac * hEncoder->sampleRate;
if (hEncoder->config.bandWidth > bwbase)
hEncoder->config.bandWidth = bwbase;
hEncoder->config.quantqual = 100;
hEncoder->config.psymodellist = (psymodellist_t *)psymodellist;
hEncoder->config.psymodelidx = 0;
hEncoder->psymodel =
hEncoder->config.psymodellist[hEncoder->config.psymodelidx].model;
hEncoder->config.shortctl = SHORTCTL_NORMAL;
/* default channel map is straight-through */
for( channel = 0; channel < 64; channel++ )
hEncoder->config.channel_map[channel] = channel;
/*
by default we have to be compatible with all previous software
which assumes that we will generate ADTS
/AV
*/
hEncoder->config.outputFormat = 1;
/*
be compatible with software which assumes 24bit in 32bit PCM
*/
hEncoder->config.inputFormat = FAAC_INPUT_32BIT;
/* find correct sampling rate depending parameters */
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;
/* FIXME: Use sr_idx here */
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));
}
/* Initialize coder functions */
fft_initialize( &hEncoder->fft_tables );
hEncoder->psymodel->PsyInit(&hEncoder->gpsyInfo, hEncoder->psyInfo, hEncoder->numChannels,
hEncoder->sampleRate, hEncoder->srInfo->cb_width_long,
hEncoder->srInfo->num_cb_long, hEncoder->srInfo->cb_width_short,
hEncoder->srInfo->num_cb_short);
FilterBankInit(hEncoder);
TnsInit(hEncoder);
LtpInit(hEncoder);
PredInit(hEncoder);
AACQuantizeInit(hEncoder->coderInfo, hEncoder->numChannels,
&(hEncoder->aacquantCfg));
HuffmanInit(hEncoder->coderInfo, hEncoder->numChannels);
/* Return handle */
return hEncoder;
}
int FAACAPI faacEncClose(faacEncHandle hEncoder)
{
unsigned int channel;
/* Deinitialize coder functions */
hEncoder->psymodel->PsyEnd(&hEncoder->gpsyInfo, hEncoder->psyInfo, hEncoder->numChannels);
FilterBankEnd(hEncoder);
LtpEnd(hEncoder);
AACQuantizeEnd(hEncoder->coderInfo, hEncoder->numChannels,
&(hEncoder->aacquantCfg));
HuffmanEnd(hEncoder->coderInfo, hEncoder->numChannels);
fft_terminate( &hEncoder->fft_tables );
/* Free remaining buffer memory */
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]);
if (hEncoder->next2SampleBuff[channel])
FreeMemory (hEncoder->next2SampleBuff[channel]);
if (hEncoder->next3SampleBuff[channel])
FreeMemory (hEncoder->next3SampleBuff[channel]);
}
/* Free handle */
if (hEncoder)
FreeMemory(hEncoder);
return 0;
}
int FAACAPI faacEncEncode(faacEncHandle hEncoder,
int32_t *inputBuffer,
unsigned int samplesInput,
unsigned char *outputBuffer,
unsigned int bufferSize
)
{
unsigned int channel, i;
int sb, frameBytes;
unsigned int offset;
BitStream *bitStream; /* bitstream used for writing the frame to */
TnsInfo *tnsInfo_for_LTP;
TnsInfo *tnsDecInfo;
/* local copy's of parameters */
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 bandWidth = hEncoder->config.bandWidth;
unsigned int shortctl = hEncoder->config.shortctl;
/* Increase frame number */
hEncoder->frameNum++;
if (samplesInput == 0)
hEncoder->flushFrame++;
/* After 4 flush frames all samples have been encoded,
return 0 bytes written */
if (hEncoder->flushFrame > 4)
return 0;
/* Determine the channel configuration */
GetChannelInfo(channelInfo, numChannels, useLfe);
/* Update current sample buffers */
for (channel = 0; channel < numChannels; channel++)
{
double *tmp;
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])
hEncoder->sampleBuff[channel] = (double*)AllocMemory(FRAME_LEN*sizeof(double));
tmp = hEncoder->sampleBuff[channel];
hEncoder->sampleBuff[channel] = hEncoder->nextSampleBuff[channel];
hEncoder->nextSampleBuff[channel] = hEncoder->next2SampleBuff[channel];
hEncoder->next2SampleBuff[channel] = hEncoder->next3SampleBuff[channel];
hEncoder->next3SampleBuff[channel] = tmp;
if (samplesInput == 0)
{
/* start flushing*/
for (i = 0; i < FRAME_LEN; i++)
hEncoder->next3SampleBuff[channel][i] = 0.0;
}
else
{
int samples_per_channel = samplesInput/numChannels;
/* handle the various input formats and channel remapping */
switch( hEncoder->config.inputFormat )
{
case FAAC_INPUT_16BIT:
{
short *input_channel = (short*)inputBuffer + hEncoder->config.channel_map[channel];
for (i = 0; i < samples_per_channel; i++)
{
hEncoder->next3SampleBuff[channel][i] = (double)*input_channel;
input_channel += numChannels;
}
}
break;
case FAAC_INPUT_32BIT:
{
int32_t *input_channel = (int32_t*)inputBuffer + hEncoder->config.channel_map[channel];
for (i = 0; i < samples_per_channel; i++)
{
hEncoder->next3SampleBuff[channel][i] = (1.0/256) * (double)*input_channel;
input_channel += numChannels;
}
}
break;
case FAAC_INPUT_FLOAT:
{
float *input_channel = (float*)inputBuffer + hEncoder->config.channel_map[channel];
for (i = 0; i < samples_per_channel; i++)
{
hEncoder->next3SampleBuff[channel][i] = (double)*input_channel;
input_channel += numChannels;
}
}
break;
default:
return -1; /* invalid input format */
break;
}
for (i = (int)(samplesInput/numChannels); i < FRAME_LEN; i++)
hEncoder->next3SampleBuff[channel][i] = 0.0;
}
/* Psychoacoustics */
/* Update buffers and run FFT on new samples */
/* LFE psychoacoustic can run without it */
if (!channelInfo[channel].lfe || channelInfo[channel].cpe)
{
hEncoder->psymodel->PsyBufferUpdate(
&hEncoder->fft_tables,
&hEncoder->gpsyInfo,
&hEncoder->psyInfo[channel],
hEncoder->next3SampleBuff[channel],
bandWidth,
hEncoder->srInfo->cb_width_short,
hEncoder->srInfo->num_cb_short);
}
}
if (hEncoder->frameNum <= 3) /* Still filling up the buffers */
return 0;
/* Psychoacoustics */
hEncoder->psymodel->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);
hEncoder->psymodel->BlockSwitch(coderInfo, hEncoder->psyInfo, numChannels);
/* force block type */
if (shortctl == SHORTCTL_NOSHORT)
{
for (channel = 0; channel < numChannels; channel++)
{
coderInfo[channel].block_type = ONLY_LONG_WINDOW;
}
}
if (shortctl == SHORTCTL_NOLONG)
{
for (channel = 0; channel < numChannels; channel++)
{
coderInfo[channel].block_type = ONLY_SHORT_WINDOW;
}
}
/* AAC Filterbank, MDCT with overlap and add */
for (channel = 0; channel < numChannels; channel++) {
int k;
FilterBank(hEncoder,
&coderInfo[channel],
hEncoder->sampleBuff[channel],
hEncoder->freqBuff[channel],
hEncoder->overlapBuff[channel],
MOVERLAPPED);
if (coderInfo[channel].block_type == ONLY_SHORT_WINDOW) {
for (k = 0; k < 8; k++) {
specFilter(hEncoder->freqBuff[channel]+k*BLOCK_LEN_SHORT,
sampleRate, bandWidth, BLOCK_LEN_SHORT);
}
} else {
specFilter(hEncoder->freqBuff[channel], sampleRate,
bandWidth, BLOCK_LEN_LONG);
}
}
/* TMP: Build sfb offset table and other stuff */
for (channel = 0; channel < numChannels; channel++) {
channelInfo[channel].msInfo.is_present = 0;
if (coderInfo[channel].block_type == ONLY_SHORT_WINDOW) {
coderInfo[channel].max_sfb = hEncoder->srInfo->num_cb_short;
coderInfo[channel].nr_of_sfb = hEncoder->srInfo->num_cb_short;
coderInfo[channel].num_window_groups = 1;
coderInfo[channel].window_group_length[0] = 8;
coderInfo[channel].window_group_length[1] = 0;
coderInfo[channel].window_group_length[2] = 0;
coderInfo[channel].window_group_length[3] = 0;
coderInfo[channel].window_group_length[4] = 0;
coderInfo[channel].window_group_length[5] = 0;
coderInfo[channel].window_group_length[6] = 0;
coderInfo[channel].window_group_length[7] = 0;
offset = 0;
for (sb = 0; sb < coderInfo[channel].nr_of_sfb; sb++) {
coderInfo[channel].sfb_offset[sb] = offset;
offset += hEncoder->srInfo->cb_width_short[sb];
}
coderInfo[channel].sfb_offset[coderInfo[channel].nr_of_sfb] = offset;
} else {
coderInfo[channel].max_sfb = hEncoder->srInfo->num_cb_long;
coderInfo[channel].nr_of_sfb = hEncoder->srInfo->num_cb_long;
coderInfo[channel].num_window_groups = 1;
coderInfo[channel].window_group_length[0] = 1;
offset = 0;
for (sb = 0; sb < coderInfo[channel].nr_of_sfb; sb++) {
coderInfo[channel].sfb_offset[sb] = offset;
offset += hEncoder->srInfo->cb_width_long[sb];
}
coderInfo[channel].sfb_offset[coderInfo[channel].nr_of_sfb] = offset;
}
}
/* Perform TNS analysis and filtering */
for (channel = 0; channel < numChannels; channel++) {
if ((!channelInfo[channel].lfe) && (useTns)) {
TnsEncode(&(coderInfo[channel].tnsInfo),
coderInfo[channel].max_sfb,
coderInfo[channel].max_sfb,
coderInfo[channel].block_type,
coderInfo[channel].sfb_offset,
hEncoder->freqBuff[channel]);
} else {
coderInfo[channel].tnsInfo.tnsDataPresent = 0; /* TNS not used for LFE */
}
}
for(channel = 0; channel < numChannels; channel++)
{
if((coderInfo[channel].tnsInfo.tnsDataPresent != 0) && (useTns))
tnsInfo_for_LTP = &(coderInfo[channel].tnsInfo);
else
tnsInfo_for_LTP = NULL;
if(channelInfo[channel].present && (!channelInfo[channel].lfe) &&
(coderInfo[channel].block_type != ONLY_SHORT_WINDOW) &&
(mpegVersion == MPEG4) && (aacObjectType == LTP))
{
LtpEncode(hEncoder,
&coderInfo[channel],
&(coderInfo[channel].ltpInfo),
tnsInfo_for_LTP,
hEncoder->freqBuff[channel],
hEncoder->ltpTimeBuff[channel]);
} else {
coderInfo[channel].ltpInfo.global_pred_flag = 0;
}
}
for(channel = 0; channel < numChannels; channel++)
{
if ((aacObjectType == MAIN) && (!channelInfo[channel].lfe)) {
int numPredBands = min(coderInfo[channel].max_pred_sfb, coderInfo[channel].nr_of_sfb);
PredCalcPrediction(hEncoder->freqBuff[channel],
coderInfo[channel].requantFreq,
coderInfo[channel].block_type,
numPredBands,
(coderInfo[channel].block_type==ONLY_SHORT_WINDOW)?
hEncoder->srInfo->cb_width_short:hEncoder->srInfo->cb_width_long,
coderInfo,
channelInfo,
channel);
} else {
coderInfo[channel].pred_global_flag = 0;
}
}
for (channel = 0; channel < numChannels; channel++) {
if (coderInfo[channel].block_type == ONLY_SHORT_WINDOW) {
SortForGrouping(&coderInfo[channel],
&hEncoder->psyInfo[channel],
&channelInfo[channel],
hEncoder->srInfo->cb_width_short,
hEncoder->freqBuff[channel]);
}
CalcAvgEnrg(&coderInfo[channel], hEncoder->freqBuff[channel]);
// reduce LFE bandwidth
if (!channelInfo[channel].cpe && channelInfo[channel].lfe)
{
coderInfo[channel].nr_of_sfb = coderInfo[channel].max_sfb = 3;
}
}
MSEncode(coderInfo, channelInfo, hEncoder->freqBuff, numChannels, allowMidside);
/* Quantize and code the signal */
for (channel = 0; channel < numChannels; channel++) {
if (coderInfo[channel].block_type == ONLY_SHORT_WINDOW) {
AACQuantize(&coderInfo[channel], &hEncoder->psyInfo[channel],
&channelInfo[channel], hEncoder->srInfo->cb_width_short,
hEncoder->srInfo->num_cb_short, hEncoder->freqBuff[channel],
&(hEncoder->aacquantCfg));
} else {
AACQuantize(&coderInfo[channel], &hEncoder->psyInfo[channel],
&channelInfo[channel], hEncoder->srInfo->cb_width_long,
hEncoder->srInfo->num_cb_long, hEncoder->freqBuff[channel],
&(hEncoder->aacquantCfg));
}
}
// fix max_sfb in CPE mode
for (channel = 0; channel < numChannels; channel++)
{
if (channelInfo[channel].present
&& (channelInfo[channel].cpe)
&& (channelInfo[channel].ch_is_left))
{
CoderInfo *cil, *cir;
cil = &coderInfo[channel];
cir = &coderInfo[channelInfo[channel].paired_ch];
cil->max_sfb = cir->max_sfb = max(cil->max_sfb, cir->max_sfb);
cil->nr_of_sfb = cir->nr_of_sfb = cil->max_sfb;
}
}
MSReconstruct(coderInfo, channelInfo, numChannels);
for (channel = 0; channel < numChannels; channel++)
{
/* If short window, reconstruction not needed for prediction */
if ((coderInfo[channel].block_type == ONLY_SHORT_WINDOW)) {
int sind;
for (sind = 0; sind < 1024; sind++) {
coderInfo[channel].requantFreq[sind] = 0.0;
}
} else {
if((coderInfo[channel].tnsInfo.tnsDataPresent != 0) && (useTns))
tnsDecInfo = &(coderInfo[channel].tnsInfo);
else
tnsDecInfo = NULL;
if ((!channelInfo[channel].lfe) && (aacObjectType == LTP)) { /* no reconstruction needed for LFE channel*/
LtpReconstruct(&coderInfo[channel], &(coderInfo[channel].ltpInfo),
coderInfo[channel].requantFreq);
if(tnsDecInfo != NULL)
TnsDecodeFilterOnly(&(coderInfo[channel].tnsInfo), coderInfo[channel].nr_of_sfb,
coderInfo[channel].max_sfb, coderInfo[channel].block_type,
coderInfo[channel].sfb_offset, coderInfo[channel].requantFreq);
IFilterBank(hEncoder, &coderInfo[channel],
coderInfo[channel].requantFreq,
coderInfo[channel].ltpInfo.time_buffer,
coderInfo[channel].ltpInfo.ltp_overlap_buffer,
MOVERLAPPED);
LtpUpdate(&(coderInfo[channel].ltpInfo),
coderInfo[channel].ltpInfo.time_buffer,
coderInfo[channel].ltpInfo.ltp_overlap_buffer,
BLOCK_LEN_LONG);
}
}
}
/* Write the AAC bitstream */
bitStream = OpenBitStream(bufferSize, outputBuffer);
WriteBitstream(hEncoder, coderInfo, channelInfo, bitStream, numChannels);
/* Close the bitstream and return the number of bytes written */
frameBytes = CloseBitStream(bitStream);
/* Adjust quality to get correct average bitrate */
if (hEncoder->config.bitRate)
{
double fix;
int desbits = numChannels * (hEncoder->config.bitRate * 1024)
/ hEncoder->sampleRate;
int diff = (frameBytes * 8) - desbits;
hEncoder->bitDiff += diff;
fix = (double)hEncoder->bitDiff / desbits;
fix *= 0.01;
fix = max(fix, -0.2);
fix = min(fix, 0.2);
if (((diff > 0) && (fix > 0.0)) || ((diff < 0) && (fix < 0.0)))
{
hEncoder->aacquantCfg.quality *= (1.0 - fix);
if (hEncoder->aacquantCfg.quality > 300)
hEncoder->aacquantCfg.quality = 300;
if (hEncoder->aacquantCfg.quality < 50)
hEncoder->aacquantCfg.quality = 50;
}
}
return frameBytes;
}
/* Scalefactorband data table */
static SR_INFO srInfo[12+1] =
{
{ 96000, 41, 12,
{
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28,
36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
},{
4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
}
}, { 88200, 41, 12,
{
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28,
36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
},{
4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
}
}, { 64000, 47, 12,
{
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
8, 8, 8, 8, 12, 12, 12, 16, 16, 16, 20, 24, 24, 28,
36, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40,
40, 40, 40, 40, 40
},{
4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 32
}
}, { 48000, 49, 14,
{
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32,
32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 96
}, {
4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
}
}, { 44100, 49, 14,
{
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32,
32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 96
}, {
4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
}
}, { 32000, 51, 14,
{
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
8, 8, 8, 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28,
28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
32, 32, 32, 32, 32, 32, 32, 32, 32
},{
4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
}
}, { 24000, 47, 15,
{
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
8, 8, 8, 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, 32,
36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
}, {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
}
}, { 22050, 47, 15,
{
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
8, 8, 8, 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, 32,
36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
}, {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
}
}, { 16000, 43, 15,
{
8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12,
12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24,
24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
}, {
4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
}
}, { 12000, 43, 15,
{
8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12,
12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24,
24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
}, {
4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
}
}, { 11025, 43, 15,
{
8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12,
12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24,
24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
}, {
4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
}
}, { 8000, 40, 15,
{
12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16,
16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28,
28, 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
}, {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
}
},
{ -1 }
};
/*
$Log: frame.c,v $
Revision 1.60 2004/04/13 13:47:33 danchr
clarify release <> unstable status
Revision 1.59 2004/04/02 14:56:17 danchr
fix name clash w/ libavcodec: fft_init -> fft_initialize
bump version number to 1.24 beta
Revision 1.58 2004/03/17 13:34:20 danchr
Automatic, untuned setting of lowpass for VBR.
Revision 1.57 2004/03/15 20:16:42 knik
fixed copyright notice
Revision 1.56 2004/01/23 10:22:26 stux
*** empty log message ***
Revision 1.55 2003/12/17 20:59:55 knik
changed default cutoff to 16k
Revision 1.54 2003/11/24 18:09:12 knik
A safe version of faacEncGetVersion() without string length problem.
Removed Stux from copyright notice. I don't think he contributed something very
substantial to faac and this is not the right place to list all contributors.
Revision 1.53 2003/11/16 05:02:52 stux
moved global tables from fft.c into hEncoder FFT_Tables. Add fft_init and fft_terminate, flowed through all necessary changes. This should remove at least one instance of a memory leak, and fix some thread-safety problems. Version update to 1.23.3
Revision 1.52 2003/11/15 08:13:42 stux
added FaacEncGetVersion(), version 1.23.2, added myself to faacCopyright :-P, does vanity know no bound ;)
Revision 1.51 2003/11/10 17:48:00 knik
Allowed independent bitRate and bandWidth setting.
Small fixes.
Revision 1.50 2003/10/29 10:31:25 stux
Added channel_map to FaacEncHandle, facilitates free generalised channel remapping in the faac core. Default is straight-through, should be *zero* performance hit... and even probably an immeasurable performance gain, updated FAAC_CFG_VERSION to 104 and FAAC_VERSION to 1.22.0
Revision 1.49 2003/10/12 16:43:39 knik
average bitrate control made more stable
Revision 1.48 2003/10/12 14:29:53 knik
more accurate average bitrate control
Revision 1.47 2003/09/24 16:26:54 knik
faacEncStruct: quantizer specific data enclosed in AACQuantCfg structure.
Added config option to enforce block type.
Revision 1.46 2003/09/07 16:48:31 knik
Updated psymodel call. Updated bitrate/cutoff mapping table.
Revision 1.45 2003/08/23 15:02:13 knik
last frame moved back to the library
Revision 1.44 2003/08/15 11:42:08 knik
removed single silent flush frame
Revision 1.43 2003/08/11 09:43:47 menno
thread safety, some tables added to the encoder context
Revision 1.42 2003/08/09 11:39:30 knik
LFE support enabled by default
Revision 1.41 2003/08/08 10:02:09 menno
Small fix
Revision 1.40 2003/08/07 08:17:00 knik
Better LFE support (reduced bandwidth)
Revision 1.39 2003/08/02 11:32:10 stux
added config.inputFormat, and associated defines and code, faac now handles native endian 16bit, 24bit and float input. Added faacEncGetDecoderSpecificInfo to the dll exports, needed for MP4. Updated DLL .dsp to compile without error. Updated CFG_VERSION to 102. Version number might need to be updated as the API has technically changed. Did not update libfaac.pdf
Revision 1.38 2003/07/10 19:17:01 knik
24-bit input
Revision 1.37 2003/06/26 19:20:09 knik
Mid/Side support.
Copyright info moved from frontend.
Fixed memory leak.
Revision 1.36 2003/05/12 17:53:16 knik
updated ABR table
Revision 1.35 2003/05/10 09:39:55 knik
added approximate ABR setting
modified default cutoff
Revision 1.34 2003/05/01 09:31:39 knik
removed ISO psyodel
disabled m/s coding
fixed default bandwidth
reduced max_sfb check
Revision 1.33 2003/04/13 08:37:23 knik
version number moved to version.h
Revision 1.32 2003/03/27 17:08:23 knik
added quantizer quality and bandwidth setting
Revision 1.31 2002/10/11 18:00:15 menno
small bugfix
Revision 1.30 2002/10/08 18:53:01 menno
Fixed some memory leakage
Revision 1.29 2002/08/19 16:34:43 knik
added one additional flush frame
fixed sample buffer memory allocation
*/
|
