ref: a45b7e2efb17f0ace411d91fa90657a167248c6e
dir: /libfaac/frame.c/
/*
* FAAC - Freeware Advanced Audio Coder
* Copyright (C) 2001 Menno Bakker
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* $Id: frame.c,v 1.9 2001/02/12 14:39:14 menno Exp $
*/
/*
* CHANGES:
* 2001/01/17: menno: Added frequency cut off filter.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <memory.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"
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;
/* 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 int channel;
faacEncHandle hEncoder;
hEncoder = (faacEncStruct*)malloc(sizeof(faacEncStruct));
memset(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;
hEncoder->aacProfile = 0; /* MAIN = 0, LOW = 1, SSR = 2 */
hEncoder->config.allowMidside = 1;
hEncoder->config.useLfe = 0;
hEncoder->config.bitRate = 64000; /* default bitrate / channel */
hEncoder->config.bandWidth = 18000; /* default bandwidth */
/* 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;
hEncoder->sampleBuff[channel] = NULL;
hEncoder->nextSampleBuff[channel] = NULL;
hEncoder->next2SampleBuff[channel] = NULL;
}
/* Initialize coder functions */
PsyInit(&hEncoder->gpsyInfo, hEncoder->psyInfo, hEncoder->numChannels,
hEncoder->sampleRate, hEncoder->sampleRateIdx);
FilterBankInit(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 */
PsyEnd(&hEncoder->gpsyInfo, hEncoder->psyInfo, hEncoder->numChannels);
FilterBankEnd(hEncoder);
AACQuantizeEnd();
HuffmanEnd(hEncoder->coderInfo, hEncoder->numChannels);
/* Free remaining buffer memory */
for (channel = 0; channel < hEncoder->numChannels; channel++) {
if (hEncoder->sampleBuff[channel]) free(hEncoder->sampleBuff[channel]);
if (hEncoder->nextSampleBuff[channel]) free(hEncoder->nextSampleBuff[channel]);
}
/* Free handle */
if (hEncoder) free(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 */
/* 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 useLfe = hEncoder->config.useLfe;
unsigned int allowMidside = hEncoder->config.allowMidside;
unsigned int bitRate = hEncoder->config.bitRate;
unsigned int bandWidth = hEncoder->config.bandWidth;
/* 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 == 2)
return 0;
/* Determine the channel configuration */
GetChannelInfo(channelInfo, numChannels, useLfe);
/* Update current sample buffers */
for (channel = 0; channel < numChannels; channel++) {
if (hEncoder->sampleBuff[channel])
free(hEncoder->sampleBuff[channel]);
hEncoder->sampleBuff[channel] = hEncoder->nextSampleBuff[channel];
hEncoder->nextSampleBuff[channel] = (double*)malloc(FRAME_LEN*sizeof(double));
if (samplesInput == 0) { /* start flushing*/
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;
}
/* Psychoacoustics */
/* Update buffers and run FFT on new samples */
PsyBufferUpdate(&hEncoder->gpsyInfo, &hEncoder->psyInfo[channel],
hEncoder->nextSampleBuff[channel]);
}
if (hEncoder->frameNum <= 1) /* Still filling up the buffers */
return 0;
/* Psychoacoustics */
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 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]);
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;
}
}
MSEncode(coderInfo, channelInfo, hEncoder->freqBuff, numChannels, allowMidside);
/* Quantize and code the signal */
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);
}
}
/* 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-1, 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, 32, 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 }
};