ref: 0feb3cab9fd2395f7de759c7dc36124a27269b9b
dir: /libfaac/frame.c/
/*
* 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 1.35 2003/05/10 09:39:55 knik 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 "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"
#include "version.h"
static char *libfaacName = FAAC_VERSION " (" __DATE__ ")";
static const psymodellist_t psymodellist[] = {
{&psymodel2, "knipsycho psychoacoustic"},
{NULL}
};
static SR_INFO srInfo[12+1];
static const int bwmax = 16000;
static const double bwfac = 0.45;
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)
{
hEncoder->config.allowMidside = 0;//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;
assert((hEncoder->config.outputFormat == 0) || (hEncoder->config.outputFormat == 1));
/* 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)
{
int i;
static struct {
int rate; // per channel at 44100 sampling frequency
int cutoff;
} rates[] =
{
{33500, 5000},
{41000, 7000},
{49000, 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;
}
hEncoder->config.bitRate = config->bitRate;
if (!config->bandWidth)
{
config->bandWidth = bwfac * hEncoder->sampleRate;
if (config->bandWidth > bwmax)
config->bandWidth = bwmax;
}
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;
// 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);
/* 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.mpegVersion = MPEG4;
hEncoder->config.aacObjectType = LTP;
hEncoder->config.allowMidside = 1;
hEncoder->config.useLfe = 0;
hEncoder->config.useTns = 0;
hEncoder->config.bitRate = 0; /* default bitrate / channel */
hEncoder->config.bandWidth = bwfac * hEncoder->sampleRate;
if (hEncoder->config.bandWidth > bwmax)
hEncoder->config.bandWidth = bwmax;
hEncoder->config.quantqual = 100;
hEncoder->config.psymodellist = psymodellist;
hEncoder->config.psymodelidx = 0;
hEncoder->psymodel =
hEncoder->config.psymodellist[hEncoder->config.psymodelidx].model;
/*
by default we have to be compatible with all previous software
which assumes that we will generate ADTS
/AV
*/
hEncoder->config.outputFormat = 1;
/* 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 */
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);
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);
HuffmanEnd(hEncoder->coderInfo, hEncoder->numChannels);
/* 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]);
}
/* Free handle */
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; /* 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 bitRate = hEncoder->config.bitRate;
unsigned int bandWidth = hEncoder->config.bandWidth;
unsigned int quantqual = hEncoder->config.quantqual;
/* 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 {
for (i = 0; i < (int)(samplesInput/numChannels); i++)
hEncoder->next3SampleBuff[channel][i] =
(double)inputBuffer[(i*numChannels)+channel];
for (i = (int)(samplesInput/numChannels); i < FRAME_LEN; i++)
hEncoder->next3SampleBuff[channel][i] = 0.0;
}
/* Psychoacoustics */
/* Update buffers and run FFT on new samples */
hEncoder->psymodel->PsyBufferUpdate(&hEncoder->gpsyInfo, &hEncoder->psyInfo[channel],
hEncoder->next3SampleBuff[channel], bandWidth);
}
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);
/* 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;
}
}
MSEncode(coderInfo, channelInfo, hEncoder->freqBuff, numChannels, allowMidside);
/* Quantize and code the signal */
if (quantqual)
bitsToUse = quantqual;
else
bitsToUse = (int)(bitRate*FRAME_LEN/sampleRate+0.5);
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], bitsToUse);
} else {
AACQuantize(&coderInfo[channel], &hEncoder->psyInfo[channel],
&channelInfo[channel], hEncoder->srInfo->cb_width_long,
hEncoder->srInfo->num_cb_long, hEncoder->freqBuff[channel], bitsToUse);
}
}
// 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);
#ifdef _DEBUG
printf("%4d %4d\n", hEncoder->frameNum-3, frameBytes);
#endif
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 1.35 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
*/