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-rw-r--r--libAACenc/include/aacenc_lib.h95
-rw-r--r--libAACenc/src/aacenc.cpp11
-rw-r--r--libAACenc/src/aacenc.h1
-rw-r--r--libAACenc/src/aacenc_lib.cpp2
-rw-r--r--libAACenc/src/aacenc_tns.cpp23
-rw-r--r--libAACenc/src/band_nrg.cpp14
-rw-r--r--libAACenc/src/grp_data.cpp12
-rw-r--r--libAACenc/src/line_pe.cpp2
-rw-r--r--libAACenc/src/qc_main.cpp54
-rw-r--r--libAACenc/src/quantize.cpp32
-rw-r--r--libAACenc/src/tns_param.cpp93
-rw-r--r--libAACenc/src/tns_param.h96
12 files changed, 88 insertions, 347 deletions
diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h
index 4635119..862dcb5 100644
--- a/libAACenc/include/aacenc_lib.h
+++ b/libAACenc/include/aacenc_lib.h
@@ -424,102 +424,11 @@ For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 1
AAC-LC core encoder operates in dual rate mode at its lowest possible sampling frequency, which is 8 kHz.
HE-AAC v2 requires stereo input audio data.
-The following table lists the supported bitrates for AAC-LC, HE-AAC and HE-AAC v2 encoding depending
-on input sampling frequency ("Hz") and number of input channels ("chan"). The minimum and maximum
-allowed bitrate ("BR Min", "BR Max") is given in bits per second.
-In case the desired combination of bitrate and sampling frequency is not available ("NA") for HE-AAC or
-HE-AAC v2 then the encoder will automatically switch to AAC-LC and give a command line warning.
Please note that in HE-AAC or HE-AAC v2 mode the encoder supports much higher bitrates than are
appropriate for HE-AAC or HE-AAC v2. For example, at a bitrate of more than 64 kbit/s for a stereo
audio signal at 44.1 kHz it usually makes sense to use AAC-LC, which will produce better audio
quality at that bitrate than HE-AAC or HE-AAC v2.
-
-\verbatim
- Config AAC-LC HE-AAC (SBR) HE-AACv2 (SBR+PS)
-
- Hz chan BR Min BR Max BR Min BR Max BR Min BR Max
-
-8000 1 758 48000 NA NA NA NA
-11025 1 1045 66150 NA NA NA NA
-12000 1 1137 72000 NA NA NA NA
-16000 1 1516 96000 8000 48000 NA NA
-22050 1 2089 132300 8000 64000 NA NA
-24000 1 2274 144000 8000 64000 NA NA
-32000 1 3032 192000 8000 64000 NA NA
-44100 1 4178 264576 8000 64000 NA NA
-48000 1 4547 288000 12000 64000 NA NA
-64000 1 6063 384000 24000 160000 NA NA
-88200 1 8355 529200 24000 160000 NA NA
-96000 1 9094 576000 24000 160000 NA NA
------------------------------------------------------------------------------------
-8000 2 1071 96000 NA NA NA NA
-11025 2 1476 132300 NA NA NA NA
-12000 2 1606 144000 NA NA NA NA
-16000 2 2141 192000 16000 96000 8000 48000
-22050 2 2951 264600 16000 128000 8000 64000
-24000 2 3211 288000 16000 128000 8000 64000
-32000 2 4282 384000 16000 128000 8000 64000
-44100 2 5900 529152 16000 128000 8000 64000
-48000 2 6422 576000 16000 128000 12000 64000
-64000 2 8563 768000 32000 256000 24000 160000
-88200 2 11801 1058400 32000 256000 24000 160000
-96000 2 12844 1152000 32000 256000 24000 160000
------------------------------------------------------------------------------------
-8000 3 1383 144000 NA NA NA NA
-11025 3 1906 198450 NA NA NA NA
-12000 3 2075 216000 NA NA NA NA
-16000 3 2766 288000 26667 120000 NA NA
-22050 3 3812 396900 26667 160000 NA NA
-24000 3 4149 432000 26667 160000 NA NA
-32000 3 5532 576000 26667 160000 NA NA
-44100 3 7623 793728 26667 160000 NA NA
-48000 3 8297 864000 29996 160000 NA NA
-64000 3 11063 1152000 59996 400000 NA NA
-88200 3 15246 1587600 59996 400000 NA NA
-96000 3 16594 1728000 59996 400000 NA NA
------------------------------------------------------------------------------------
-8000 4 1696 192000 NA NA NA NA
-11025 4 2337 264600 NA NA NA NA
-12000 4 2543 288000 NA NA NA NA
-16000 4 3391 384000 40000 160000 NA NA
-22050 4 4673 529200 40000 213330 NA NA
-24000 4 5086 576000 40000 213330 NA NA
-32000 4 6782 768000 40000 213330 NA NA
-44100 4 9345 1058304 40000 213330 NA NA
-48000 4 10172 1152000 40000 213330 NA NA
-64000 4 13563 1536000 80000 533330 NA NA
-88200 4 18691 2116800 80000 533330 NA NA
-96000 4 20344 2304000 80000 533330 NA NA
------------------------------------------------------------------------------------
-8000 5 2008 240000 NA NA NA NA
-11025 5 2768 330750 NA NA NA NA
-12000 5 3012 360000 NA NA NA NA
-16000 5 4016 480000 43244 184612 NA NA
-22050 5 5535 661500 43244 246152 NA NA
-24000 5 6024 720000 43244 246152 NA NA
-32000 5 8032 960000 43244 246152 NA NA
-44100 5 11068 1322880 43244 246152 NA NA
-48000 5 12047 1440000 46140 246152 NA NA
-64000 5 16063 1920000 92296 615384 NA NA
-88200 5 22137 2646000 92296 615384 NA NA
-96000 5 24094 2880000 92296 615384 NA NA
------------------------------------------------------------------------------------
-8000 5.1 2321 240000 NA NA NA NA
-11025 5.1 3198 330750 NA NA NA NA
-12000 5.1 3481 360000 NA NA NA NA
-16000 5.1 4641 480000 45715 199990 NA NA
-22050 5.1 6396 661500 45715 266658 NA NA
-24000 5.1 6961 720000 45715 266658 NA NA
-32000 5.1 9282 960000 45715 266658 NA NA
-44100 5.1 12790 1322880 45715 266658 NA NA
-48000 5.1 13922 1440000 49982 266658 NA NA
-64000 5.1 18563 1920000 99982 666658 NA NA
-88200 5.1 25582 2646000 99982 666658 NA NA
-96000 5.1 27844 2880000 99982 666658 NA NA
-
-\endverbatim \n
-
\section reommendedConfig Recommended Sampling Rate and Bitrate Combinations
The following table provides an overview of recommended encoder configuration parameters
@@ -956,8 +865,8 @@ typedef enum
AACENC_GRANULE_LENGTH = 0x0105, /*!< Core encoder (AAC) audio frame length in samples:
- 1024: Default configuration.
- - 512: Optional length in LD/ELD configuration.
- - 480: Default LD/ELD configuration. */
+ - 512: Default LD/ELD configuration.
+ - 480: Optional length in LD/ELD configuration. */
AACENC_CHANNELMODE = 0x0106, /*!< Set explicit channel mode. Channel mode must match with number of input channels.
- 1-6: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */
diff --git a/libAACenc/src/aacenc.cpp b/libAACenc/src/aacenc.cpp
index 85083cd..d3f36aa 100644
--- a/libAACenc/src/aacenc.cpp
+++ b/libAACenc/src/aacenc.cpp
@@ -149,7 +149,7 @@ INT FDKaacEnc_LimitBitrate(
transportBits = 208;
}
- bitRate = FDKmax(bitRate, ((((40 * nChannels) + transportBits + frameLength) * (coreSamplingRate)) / frameLength) );
+ bitRate = FDKmax(bitRate, ((((40 * nChannels) + transportBits) * (coreSamplingRate)) / frameLength) );
FDK_ASSERT(bitRate >= 0);
bitRate = FDKmin(bitRate, ((nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN)*(coreSamplingRate>>shift)) / (frameLength>>shift)) ;
@@ -280,7 +280,7 @@ void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config)
config->useTns = TNS_ENABLE_MASK; /* tns enabled completly */
config->usePns = 1; /* depending on channelBitrate this might be set to 0 later */
config->useIS = 1; /* Intensity Stereo Configuration */
- config->framelength = DEFAULT_FRAMELENGTH; /* used frame size */
+ config->framelength = -1; /* Framesize not configured */
config->syntaxFlags = 0; /* default syntax with no specialities */
config->epConfig = -1; /* no ER syntax -> no additional error protection */
config->nSubFrames = 1; /* default, no sub frames */
@@ -451,11 +451,8 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
switch (config->framelength)
{
case 1024:
- if ( config->audioObjectType != AOT_AAC_LC
- && config->audioObjectType != AOT_SBR
- && config->audioObjectType != AOT_PS
- && config->audioObjectType != AOT_ER_AAC_LC
- && config->audioObjectType != AOT_AAC_SCAL )
+ if ( config->audioObjectType == AOT_ER_AAC_LD
+ || config->audioObjectType == AOT_ER_AAC_ELD )
{
return AAC_ENC_INVALID_FRAME_LENGTH;
}
diff --git a/libAACenc/src/aacenc.h b/libAACenc/src/aacenc.h
index bb49019..517b0dc 100644
--- a/libAACenc/src/aacenc.h
+++ b/libAACenc/src/aacenc.h
@@ -153,7 +153,6 @@ typedef enum {
/*-------------------------- defines --------------------------------------*/
#define ANC_DATA_BUFFERSIZE 1024 /* ancBuffer size */
-#define DEFAULT_FRAMELENGTH 1024 /* size of AAC core frame in (new) PCM samples */
#define MAX_TOTAL_EXT_PAYLOADS (((6) * (1)) + (2+2))
diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp
index 9647094..a4291d5 100644
--- a/libAACenc/src/aacenc_lib.cpp
+++ b/libAACenc/src/aacenc_lib.cpp
@@ -98,7 +98,7 @@ amm-info@iis.fraunhofer.de
/* Encoder library info */
#define AACENCODER_LIB_VL0 3
#define AACENCODER_LIB_VL1 3
-#define AACENCODER_LIB_VL2 1
+#define AACENCODER_LIB_VL2 3
#define AACENCODER_LIB_TITLE "AAC Encoder"
#define AACENCODER_LIB_BUILD_DATE __DATE__
#define AACENCODER_LIB_BUILD_TIME __TIME__
diff --git a/libAACenc/src/aacenc_tns.cpp b/libAACenc/src/aacenc_tns.cpp
index fbff424..d6339fc 100644
--- a/libAACenc/src/aacenc_tns.cpp
+++ b/libAACenc/src/aacenc_tns.cpp
@@ -1067,11 +1067,11 @@ static void FDKaacEnc_CalcGaussWindow(
const INT timeResolution_e
)
{
- #define PI_SCALE (2)
- #define PI_FIX FL2FXCONST_DBL(3.1416f/(float)(1<<PI_SCALE))
+ #define PI_E (2)
+ #define PI_M FL2FXCONST_DBL(3.1416f/(float)(1<<PI_E))
- #define EULER_SCALE (2)
- #define EULER_FIX FL2FXCONST_DBL(2.7183/(float)(1<<EULER_SCALE))
+ #define EULER_E (2)
+ #define EULER_M FL2FXCONST_DBL(2.7183/(float)(1<<EULER_E))
#define COEFF_LOOP_SCALE (4)
@@ -1083,9 +1083,9 @@ static void FDKaacEnc_CalcGaussWindow(
* gaussExp = PI * samplingRate * 0.001f * timeResolution / transformResolution;
* gaussExp = -0.5f * gaussExp * gaussExp;
*/
- gaussExp_m = fMultNorm(timeResolution, fMult(PI_FIX, fDivNorm( (FIXP_DBL)(samplingRate), (FIXP_DBL)(LONG)(transformResolution*1000.f), &e1)), &e2);
+ gaussExp_m = fMultNorm(timeResolution, fMult(PI_M, fDivNorm( (FIXP_DBL)(samplingRate), (FIXP_DBL)(LONG)(transformResolution*1000.f), &e1)), &e2);
gaussExp_m = -fPow2Div2(gaussExp_m);
- gaussExp_e = 2*(e1+e2+timeResolution_e+PI_SCALE);
+ gaussExp_e = 2*(e1+e2+timeResolution_e+PI_E);
FDK_ASSERT( winSize < (1<<COEFF_LOOP_SCALE) );
@@ -1095,13 +1095,13 @@ static void FDKaacEnc_CalcGaussWindow(
for( i=0; i<winSize; i++) {
win[i] = fPow(
- EULER_FIX,
- EULER_SCALE,
+ EULER_M,
+ EULER_E,
fMult(gaussExp_m, fPow2((i*FL2FXCONST_DBL(1.f/(float)(1<<COEFF_LOOP_SCALE)) + FL2FXCONST_DBL(.5f/(float)(1<<COEFF_LOOP_SCALE))))),
gaussExp_e + 2*COEFF_LOOP_SCALE,
&e1);
- win[i] = scaleValue(win[i], e1);
+ win[i] = scaleValueSaturate(win[i], e1);
}
}
@@ -1157,7 +1157,10 @@ static INT FDKaacEnc_AutoToParcor(
workBuffer++;
}
- tmp = fMult((FIXP_DBL)((LONG)TNS_PREDGAIN_SCALE<<21), fDivNorm(autoCorr_0, input[0], &scale));
+ tmp = fMult((FIXP_DBL)((LONG)TNS_PREDGAIN_SCALE<<21), fDivNorm(fAbs(autoCorr_0), fAbs(input[0]), &scale));
+ if ( fMultDiv2(autoCorr_0, input[0])<FL2FXCONST_DBL(0.0f) ) {
+ tmp = -tmp;
+ }
predictionGain = (LONG)scaleValue(tmp,scale-21);
return (predictionGain);
diff --git a/libAACenc/src/band_nrg.cpp b/libAACenc/src/band_nrg.cpp
index c672c6c..0e46b45 100644
--- a/libAACenc/src/band_nrg.cpp
+++ b/libAACenc/src/band_nrg.cpp
@@ -260,21 +260,21 @@ FDKaacEnc_CalcBandEnergyOptimShort(const FIXP_DBL *RESTRICT mdctSpectrum,
for(i=0; i<numBands; i++)
{
- int leadingBits = fixMax(0,sfbMaxScaleSpec[i]-4); /* max sfbWidth = 96 ; 2^7=128 => 7/2 = 4 (spc*spc) */
+ int leadingBits = sfbMaxScaleSpec[i]-3; /* max sfbWidth = 36 ; 2^6=64 => 6/2 = 3 (spc*spc) */
FIXP_DBL tmp = FL2FXCONST_DBL(0.0);
for (j=bandOffset[i];j<bandOffset[i+1];j++)
{
- FIXP_DBL spec = mdctSpectrum[j]<<leadingBits;
+ FIXP_DBL spec = scaleValue(mdctSpectrum[j],leadingBits);
tmp = fPow2AddDiv2(tmp, spec);
}
- bandEnergy[i] = scaleValueSaturate(tmp, 1);
+ bandEnergy[i] = tmp;
}
for(i=0; i<numBands; i++)
{
- INT scale = 2*fixMax(0,sfbMaxScaleSpec[i]-4); /* max sfbWidth = 96 ; 2^7=128 => 7/2 = 4 (spc*spc) */
- scale = fixMin(scale,(DFRACT_BITS-1));
- bandEnergy[i] >>= scale;
+ INT scale = (2*(sfbMaxScaleSpec[i]-3))-1; /* max sfbWidth = 36 ; 2^6=64 => 6/2 = 3 (spc*spc) */
+ scale = fixMax(fixMin(scale,(DFRACT_BITS-1)),-(DFRACT_BITS-1));
+ bandEnergy[i] = scaleValueSaturate(bandEnergy[i], -scale);
}
}
@@ -343,7 +343,7 @@ void FDKaacEnc_CalcBandNrgMSOpt(const FIXP_DBL *RESTRICT mdctSpectrumLeft,
{
/* using the minimal scaling of left and right channel can cause very small energies;
check ldNrg before subtract scaling multiplication: fract*INT we don't need fMult */
-
+
int minus = scale*FL2FXCONST_DBL(1.0/64);
if (bandEnergyMidLdData[i] != FL2FXCONST_DBL(-1.0f))
diff --git a/libAACenc/src/grp_data.cpp b/libAACenc/src/grp_data.cpp
index 88a01ba..4355295 100644
--- a/libAACenc/src/grp_data.cpp
+++ b/libAACenc/src/grp_data.cpp
@@ -94,6 +94,10 @@ amm-info@iis.fraunhofer.de
* this routine does not work in-place
*/
+static inline FIXP_DBL nrgAddSaturate(const FIXP_DBL a, const FIXP_DBL b) {
+ return ( (a>=(FIXP_DBL)MAXVAL_DBL-b) ? (FIXP_DBL)MAXVAL_DBL : (a + b) );
+}
+
void
FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out */
SFB_THRESHOLD *sfbThreshold, /* in-out */
@@ -177,7 +181,7 @@ FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out
FIXP_DBL thresh = sfbThreshold->Short[wnd][sfb];
for (j=1; j<groupLen[grp]; j++)
{
- thresh = fAddSaturate(thresh, sfbThreshold->Short[wnd+j][sfb]);
+ thresh = nrgAddSaturate(thresh, sfbThreshold->Short[wnd+j][sfb]);
}
sfbThreshold->Long[i++] = thresh;
}
@@ -195,7 +199,7 @@ FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out
FIXP_DBL energy = sfbEnergy->Short[wnd][sfb];
for (j=1; j<groupLen[grp]; j++)
{
- energy = fAddSaturate(energy, sfbEnergy->Short[wnd+j][sfb]);
+ energy = nrgAddSaturate(energy, sfbEnergy->Short[wnd+j][sfb]);
}
sfbEnergy->Long[i++] = energy;
}
@@ -213,7 +217,7 @@ FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out
FIXP_DBL energy = sfbEnergyMS->Short[wnd][sfb];
for (j=1; j<groupLen[grp]; j++)
{
- energy = fAddSaturate(energy, sfbEnergyMS->Short[wnd+j][sfb]);
+ energy = nrgAddSaturate(energy, sfbEnergyMS->Short[wnd+j][sfb]);
}
sfbEnergyMS->Long[i++] = energy;
}
@@ -231,7 +235,7 @@ FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out
FIXP_DBL energy = sfbSpreadEnergy->Short[wnd][sfb];
for (j=1; j<groupLen[grp]; j++)
{
- energy = fAddSaturate(energy, sfbSpreadEnergy->Short[wnd+j][sfb]);
+ energy = nrgAddSaturate(energy, sfbSpreadEnergy->Short[wnd+j][sfb]);
}
sfbSpreadEnergy->Long[i++] = energy;
}
diff --git a/libAACenc/src/line_pe.cpp b/libAACenc/src/line_pe.cpp
index ed5ee7f..7014bcb 100644
--- a/libAACenc/src/line_pe.cpp
+++ b/libAACenc/src/line_pe.cpp
@@ -122,6 +122,8 @@ void FDKaacEnc_prepareSfbPe(PE_CHANNEL_DATA *peChanData,
avgFormFactorLdData = ((-sfbEnergyLdData[sfbGrp+sfb]>>1) + (CalcLdInt(sfbWidth)>>1))>>1;
peChanData->sfbNLines[sfbGrp+sfb] =
(INT)CalcInvLdData( (sfbFormFactorLdData[sfbGrp+sfb] + formFacScaling) + avgFormFactorLdData);
+ /* Make sure sfbNLines is never greater than sfbWidth due to unaccuracies (e.g. sfbEnergyLdData[sfbGrp+sfb] = 0x80000000) */
+ peChanData->sfbNLines[sfbGrp+sfb] = fMin(sfbWidth, peChanData->sfbNLines[sfbGrp+sfb]);
}
else {
peChanData->sfbNLines[sfbGrp+sfb] = 0;
diff --git a/libAACenc/src/qc_main.cpp b/libAACenc/src/qc_main.cpp
index 749398a..d7e76c7 100644
--- a/libAACenc/src/qc_main.cpp
+++ b/libAACenc/src/qc_main.cpp
@@ -797,7 +797,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC,
{
int i, c;
AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK;
- INT avgTotalDynBits = 0; /* maximal allowd dynamic bits for all frames */
+ INT avgTotalDynBits = 0; /* maximal allowed dynamic bits for all frames */
INT totalAvailableBits = 0;
INT nSubFrames = 1;
@@ -1092,7 +1092,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC,
int sumBitsConsumedTotal = FDKaacEnc_getTotalConsumedBits(qcOut, qcElement, cm, hQC->globHdrBits, nSubFrames);
/* in all frames are valid dynamic bits */
- if (sumBitsConsumedTotal < totalAvailableBits && (decreaseBitConsumption==1) && checkMinFrameBitsDemand(qcOut,hQC->minBitsPerFrame,nSubFrames)
+ if ( ((sumBitsConsumedTotal < totalAvailableBits) || qcOut[c]->usedDynBits==0) && (decreaseBitConsumption==1) && checkMinFrameBitsDemand(qcOut,hQC->minBitsPerFrame,nSubFrames)
/*()*/ )
{
quantizationDone = 1; /* exit bit adjustment */
@@ -1365,42 +1365,54 @@ AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption(CHANNEL_MAPPING *cm,
QC_OUT_EXTENSION fillExtPayload;
INT totFillBits, alignBits;
- {
- int exactTpBits;
- int max_iter = 3;
+ /* Get total consumed bits in AU */
+ qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + qcOut->totFillBits +
+ qcOut->elementExtBits + qcOut->globalExtBits;
- /* Get total consumed bits in AU */
- qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + qcOut->totFillBits +
- qcOut->elementExtBits + qcOut->globalExtBits;
+ if (qcKernel->bitrateMode==QCDATA_BR_MODE_CBR) {
/* Now we can get the exact transport bit amount, and hopefully it is equal to the estimated value */
- exactTpBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits);
+ INT exactTpBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits);
if (exactTpBits != qcKernel->globHdrBits) {
INT diffFillBits = 0;
+ /* How many bits can be taken by bitreservoir */
+ const INT bitresSpace = qcKernel->bitResTotMax - (qcKernel->bitResTot + (qcOut->grantedDynBits - (qcOut->usedDynBits + qcOut->totFillBits) ) );
+
/* Number of bits which can be moved to bitreservoir. */
- INT bitsToBitres = qcKernel->globHdrBits - exactTpBits;
+ const INT bitsToBitres = qcKernel->globHdrBits - exactTpBits;
+ FDK_ASSERT(bitsToBitres>=0); /* is always positive */
- if (bitsToBitres>0) {
- /* if bitreservoir can not take all bits, move ramaining bits to fillbits */
- diffFillBits = FDKmax(0, bitsToBitres - (qcKernel->bitResTotMax-qcKernel->bitResTot));
- }
- else if (bitsToBitres<0) {
- /* if bits mus be taken from bitreservoir, reduce fillbits first. */
- diffFillBits = (FDKmax(FDKmax(bitsToBitres, -qcKernel->bitResTot), -qcOut->totFillBits));
- }
+ /* If bitreservoir can not take all bits, move ramaining bits to fillbits */
+ diffFillBits = FDKmax(0, bitsToBitres - bitresSpace);
+
+ /* Assure previous alignment */
+ diffFillBits = (diffFillBits+7)&~7;
- diffFillBits = (diffFillBits+7)&~7; /* assure previous alignment */
+ /* Move as many bits as possible to bitreservoir */
+ qcKernel->bitResTot += (bitsToBitres-diffFillBits);
+ /* Write remaing bits as fill bits */
qcOut->totFillBits += diffFillBits;
qcOut->totalBits += diffFillBits;
qcOut->grantedDynBits += diffFillBits;
- /* new header bits */
+ /* Get new header bits */
qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits);
+
+ if (qcKernel->globHdrBits != exactTpBits) {
+ /* In previous step, fill bits and corresponding total bits were changed when bitreservoir was completely filled.
+ Now we can take the too much taken bits caused by header overhead from bitreservoir.
+ */
+ qcKernel->bitResTot -= (qcKernel->globHdrBits - exactTpBits);
+ }
}
- }
+
+ } /* MODE_CBR */
+
+ /* Update exact number of consumed header bits. */
+ qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits);
/* Save total fill bits and distribut to alignment and fill bits */
totFillBits = qcOut->totFillBits;
diff --git a/libAACenc/src/quantize.cpp b/libAACenc/src/quantize.cpp
index dc85a6d..a1698a8 100644
--- a/libAACenc/src/quantize.cpp
+++ b/libAACenc/src/quantize.cpp
@@ -127,10 +127,7 @@ static void FDKaacEnc_quantizeLines(INT gain,
accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex],FDKaacEnc_quantTableE[totalShift&3]);
totalShift = (16-4)-(3*(totalShift>>2));
FDK_ASSERT(totalShift >=0); /* MAX_QUANT_VIOLATION */
- if (totalShift < 32)
- accu>>=totalShift;
- else
- accu = 0;
+ accu>>=totalShift;
quaSpectrum[line] = (SHORT)(-((LONG)(k + accu) >> (DFRACT_BITS-1-16)));
}
else if(accu > FL2FXCONST_DBL(0.0f))
@@ -143,10 +140,7 @@ static void FDKaacEnc_quantizeLines(INT gain,
accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex],FDKaacEnc_quantTableE[totalShift&3]);
totalShift = (16-4)-(3*(totalShift>>2));
FDK_ASSERT(totalShift >=0); /* MAX_QUANT_VIOLATION */
- if (totalShift < 32)
- accu>>=totalShift;
- else
- accu = 0;
+ accu>>=totalShift;
quaSpectrum[line] = (SHORT)((LONG)(k + accu) >> (DFRACT_BITS-1-16));
}
else
@@ -319,6 +313,9 @@ FIXP_DBL FDKaacEnc_calcSfbDist(FIXP_DBL *mdctSpectrum,
&mdctSpectrum[i],
&quantSpectrum[i]);
+ if (fAbs(quantSpectrum[i])>MAX_QUANT) {
+ return FL2FXCONST_DBL(0.0f);
+ }
/* inverse quantization */
FDKaacEnc_invQuantizeLines(gain,1,&quantSpectrum[i],&invQuantSpec);
@@ -361,15 +358,22 @@ void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum,
FIXP_DBL invQuantSpec;
FIXP_DBL diff;
- *en = FL2FXCONST_DBL(0.0f);
- *dist = FL2FXCONST_DBL(0.0f);
+ FIXP_DBL energy = FL2FXCONST_DBL(0.0f);
+ FIXP_DBL distortion = FL2FXCONST_DBL(0.0f);
for (i=0; i<noOfLines; i++) {
+
+ if (fAbs(quantSpectrum[i])>MAX_QUANT) {
+ *en = FL2FXCONST_DBL(0.0f);
+ *dist = FL2FXCONST_DBL(0.0f);
+ return;
+ }
+
/* inverse quantization */
FDKaacEnc_invQuantizeLines(gain,1,&quantSpectrum[i],&invQuantSpec);
/* energy */
- *en += fPow2(invQuantSpec);
+ energy += fPow2(invQuantSpec);
/* dist */
diff = fixp_abs(fixp_abs(invQuantSpec) - fixp_abs(mdctSpectrum[i]>>1));
@@ -382,10 +386,10 @@ void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum,
diff = scaleValue(diff, -scale);
- *dist += diff;
+ distortion += diff;
}
- *en = CalcLdData(*en)+FL2FXCONST_DBL(0.03125f);
- *dist = CalcLdData(*dist);
+ *en = CalcLdData(energy)+FL2FXCONST_DBL(0.03125f);
+ *dist = CalcLdData(distortion);
}
diff --git a/libAACenc/src/tns_param.cpp b/libAACenc/src/tns_param.cpp
deleted file mode 100644
index 3c04c51..0000000
--- a/libAACenc/src/tns_param.cpp
+++ /dev/null
@@ -1,93 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
- Initial author: M.Werner
- contents/description: TNS parameters
-
-******************************************************************************/
-
-#include "tns_param.h"
-
-
diff --git a/libAACenc/src/tns_param.h b/libAACenc/src/tns_param.h
deleted file mode 100644
index b191b5c..0000000
--- a/libAACenc/src/tns_param.h
+++ /dev/null
@@ -1,96 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
- Initial author: Alex Goeschel
- contents/description: Temporal noise shaping
-
-******************************************************************************/
-
-#ifndef _TNS_PARAM_H
-#define _TNS_PARAM_H
-
-
-
-#endif /* _TNS_PARAM_H */
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-07 19:06:44 +0000