ref: 7ab1c9ece6e293660fe3c81effa45058a795fc17
parent: 735bb0c27c9089ad8178656123fee8a25ff6fa30
author: menno <menno>
date: Fri Aug 8 06:02:09 EDT 2003
Small fix
--- a/libfaac/frame.c
+++ b/libfaac/frame.c
@@ -16,7 +16,7 @@
* 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 1.40 2003/08/07 08:17:00 knik Exp $
+ * $Id: frame.c,v 1.41 2003/08/08 10:02:09 menno Exp $
*/
/*
@@ -51,8 +51,8 @@
static char *libfaacName = FAAC_VERSION " (" __DATE__ ")";static char *libCopyright =
"FAAC - Freeware Advanced Audio Coder (http://www.audiocoding.com/)\n"
- " Portions Copyright (C) 2001 Menno Bakker\n"
- " Portions Copyright (C) 2002,2003 Krzysztof Nikiel\n"
+ " Portions Copyright (C) 2001 Menno Bakker\n"
+ " Portions 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[] = {@@ -71,7 +71,7 @@
if((hEncoder == NULL) || (ppBuffer == NULL) || (pSizeOfDecoderSpecificInfo == NULL)) {return -1;
}
-
+
if(hEncoder->config.mpegVersion == MPEG2){return -2; /* not supported */
}
@@ -119,22 +119,23 @@
hEncoder->config.useTns = config->useTns;
hEncoder->config.aacObjectType = config->aacObjectType;
hEncoder->config.mpegVersion = config->mpegVersion;
- hEncoder->config.outputFormat = config->outputFormat;
+ hEncoder->config.outputFormat = config->outputFormat;
+ hEncoder->config.inputFormat = config->inputFormat;
- assert((hEncoder->config.outputFormat == 0) || (hEncoder->config.outputFormat == 1));
+ 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;
+ 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;
- }
+ default:
+ return 0;
+ break;
+ }
/* No SSR supported for now */
if (hEncoder->config.aacObjectType == SSR)
@@ -159,16 +160,16 @@
{int i;
static struct {- int rate; // per channel at 44100 sampling frequency
+ int rate; // per channel at 44100 sampling frequency
int cutoff;
} rates[] =
{- {26500, 5000},- {32000, 7000},- {41000, 10000},- {64000, 16000},- {84500, 20000},- {0, 0}+ {26500, 5000},+ {32000, 7000},+ {41000, 10000},+ {64000, 16000},+ {84500, 20000},+ {0, 0}};
int f0, f1;
int r0, r1;
@@ -181,28 +182,28 @@
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;
+ 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;
+ config->bitRate = r0;
if (f1 > f0)
- config->bandWidth =
- pow((double)config->bitRate / r1,
- log((double)f1 / f0) / log ((double)r1 / r0)) * (double)f1;
+ config->bandWidth =
+ pow((double)config->bitRate / r1,
+ log((double)f1 / f0) / log ((double)r1 / r0)) * (double)f1;
else
- config->bandWidth = f1;
+ config->bandWidth = f1;
config->bandWidth =
- (double)config->bandWidth * hEncoder->sampleRate / 44100;
+ (double)config->bandWidth * hEncoder->sampleRate / 44100;
config->bitRate = (double)config->bitRate * hEncoder->sampleRate / 44100;
}
@@ -212,7 +213,7 @@
{config->bandWidth = bwfac * hEncoder->sampleRate;
if (config->bandWidth > bwmax)
- config->bandWidth = bwmax;
+ config->bandWidth = bwmax;
}
hEncoder->config.bandWidth = config->bandWidth;
@@ -236,9 +237,9 @@
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);
+ 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;
@@ -285,17 +286,17 @@
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;
+ /*
+ 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;
+ /*
+ 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];
@@ -321,7 +322,7 @@
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);
+ hEncoder->srInfo->num_cb_short);
FilterBankInit(hEncoder);
@@ -357,15 +358,15 @@
/* Free remaining buffer memory */
for (channel = 0; channel < hEncoder->numChannels; channel++) {if (hEncoder->ltpTimeBuff[channel])
- FreeMemory(hEncoder->ltpTimeBuff[channel]);
+ FreeMemory(hEncoder->ltpTimeBuff[channel]);
if (hEncoder->sampleBuff[channel])
- FreeMemory(hEncoder->sampleBuff[channel]);
+ FreeMemory(hEncoder->sampleBuff[channel]);
if (hEncoder->nextSampleBuff[channel])
- FreeMemory(hEncoder->nextSampleBuff[channel]);
+ FreeMemory(hEncoder->nextSampleBuff[channel]);
if (hEncoder->next2SampleBuff[channel])
- FreeMemory (hEncoder->next2SampleBuff[channel]);
+ FreeMemory (hEncoder->next2SampleBuff[channel]);
if (hEncoder->next3SampleBuff[channel])
- FreeMemory (hEncoder->next3SampleBuff[channel]);
+ FreeMemory (hEncoder->next3SampleBuff[channel]);
}
/* Free handle */
@@ -418,7 +419,7 @@
/* Update current sample buffers */
for (channel = 0; channel < numChannels; channel++) {- double *tmp;
+ double *tmp;
if (hEncoder->sampleBuff[channel]) { for(i = 0; i < FRAME_LEN; i++) {@@ -432,53 +433,53 @@
}
}
- if (!hEncoder->sampleBuff[channel])
- hEncoder->sampleBuff[channel] = (double*)AllocMemory(FRAME_LEN*sizeof(double));
- tmp = hEncoder->sampleBuff[channel];
+ 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;
+ hEncoder->next3SampleBuff[channel] = tmp;
if (samplesInput == 0)
- { - /* start flushing*/
+ {+ /* start flushing*/
for (i = 0; i < FRAME_LEN; i++)
hEncoder->next3SampleBuff[channel][i] = 0.0;
- }
- else
- { - /* handle the various input formats */
- switch( hEncoder->config.inputFormat )
- {- case FAAC_INPUT_16BIT:
- for (i = 0; i < (int)(samplesInput/numChannels); i++)
- {- hEncoder->next3SampleBuff[channel][i] =
- (double)((short*)inputBuffer)[(i*numChannels)+channel];
- }
- break;
+ }
+ else
+ {+ /* handle the various input formats */
+ switch( hEncoder->config.inputFormat )
+ {+ case FAAC_INPUT_16BIT:
+ for (i = 0; i < (int)(samplesInput/numChannels); i++)
+ {+ hEncoder->next3SampleBuff[channel][i] =
+ (double)((short*)inputBuffer)[(i*numChannels)+channel];
+ }
+ break;
- case FAAC_INPUT_32BIT:
- for (i = 0; i < (int)(samplesInput/numChannels); i++)
- {- hEncoder->next3SampleBuff[channel][i] =
- (1.0/256) * (double)inputBuffer[(i*numChannels)+channel];
- }
- break;
+ case FAAC_INPUT_32BIT:
+ for (i = 0; i < (int)(samplesInput/numChannels); i++)
+ {+ hEncoder->next3SampleBuff[channel][i] =
+ (1.0/256) * (double)inputBuffer[(i*numChannels)+channel];
+ }
+ break;
- case FAAC_INPUT_FLOAT:
- for (i = 0; i < (int)(samplesInput/numChannels); i++)
- {- hEncoder->next3SampleBuff[channel][i] =
- ((float*)inputBuffer)[(i*numChannels)+channel];
- }
- break;
+ case FAAC_INPUT_FLOAT:
+ for (i = 0; i < (int)(samplesInput/numChannels); i++)
+ {+ hEncoder->next3SampleBuff[channel][i] =
+ ((float*)inputBuffer)[(i*numChannels)+channel];
+ }
+ break;
- default:
- return -1; /* invalid input format */
- break;
- }
+ default:
+ return -1; /* invalid input format */
+ break;
+ }
for (i = (int)(samplesInput/numChannels); i < FRAME_LEN; i++)
hEncoder->next3SampleBuff[channel][i] = 0.0;
@@ -485,14 +486,14 @@
}
/* 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->gpsyInfo, &hEncoder->psyInfo[channel],
- hEncoder->next3SampleBuff[channel], bandWidth);
- }
+ /* 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->gpsyInfo, &hEncoder->psyInfo[channel],
+ hEncoder->next3SampleBuff[channel], bandWidth);
}
+ }
if (hEncoder->frameNum <= 3) /* Still filling up the buffers */
return 0;
@@ -623,11 +624,11 @@
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]);
+ SortForGrouping(&coderInfo[channel],
+ &hEncoder->psyInfo[channel],
+ &channelInfo[channel],
+ hEncoder->srInfo->cb_width_short,
+ hEncoder->freqBuff[channel]);
}
CalcAvgEnrg(&coderInfo[channel], hEncoder->freqBuff[channel]);
@@ -634,7 +635,7 @@
// reduce LFE bandwidth
if (!channelInfo[channel].cpe && channelInfo[channel].lfe)
{- coderInfo[channel].nr_of_sfb = coderInfo[channel].max_sfb = 3;
+ coderInfo[channel].nr_of_sfb = coderInfo[channel].max_sfb = 3;
}
}
@@ -660,19 +661,19 @@
// 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;
+ 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 = &coderInfo[channel];
+ cir = &coderInfo[channelInfo[channel].paired_ch];
- cil->max_sfb = cir->max_sfb = max(cil->max_sfb, cir->max_sfb);
+ 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);
@@ -838,6 +839,9 @@
/*
$Log: frame.c,v $
+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)