Merge "Merge Android 12"
diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf
index cc7cf41..3d4699e 100644
--- a/documentation/aacDecoder.pdf
+++ b/documentation/aacDecoder.pdf
Binary files differ
diff --git a/documentation/aacEncoder.pdf b/documentation/aacEncoder.pdf
index 77b8f4c..a47708a 100644
--- a/documentation/aacEncoder.pdf
+++ b/documentation/aacEncoder.pdf
Binary files differ
diff --git a/fuzzer/aac_dec_fuzzer.cpp b/fuzzer/aac_dec_fuzzer.cpp
index b5545fc..c970197 100644
--- a/fuzzer/aac_dec_fuzzer.cpp
+++ b/fuzzer/aac_dec_fuzzer.cpp
@@ -118,7 +118,8 @@
INT_PCM outputBuf[kMaxOutBufferSize];
do {
mErrorCode =
- aacDecoder_DecodeFrame(mAacDecoderHandle, outputBuf, sizeof(outputBuf), 0);
+ aacDecoder_DecodeFrame(mAacDecoderHandle, outputBuf,
+ kMaxOutBufferSize /*size in number of INT_PCM, not bytes*/, 0);
} while (mErrorCode == AAC_DEC_OK);
UINT offset = inputSize - valid;
data += offset;
diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h
index 56f4ec1..d7928c0 100644
--- a/libAACdec/include/aacdecoder_lib.h
+++ b/libAACdec/include/aacdecoder_lib.h
@@ -1032,7 +1032,7 @@
* \param self AAC decoder handle.
* \param pTimeData Pointer to external output buffer where the decoded PCM
* samples will be stored into.
- * \param timeDataSize Size of external output buffer.
+ * \param timeDataSize Size of external output buffer in PCM samples.
* \param flags Bit field with flags for the decoder: \n
* (flags & AACDEC_CONCEAL) == 1: Do concealment. \n
* (flags & AACDEC_FLUSH) == 2: Discard input data. Flush
diff --git a/libAACdec/src/aac_ram.cpp b/libAACdec/src/aac_ram.cpp
index aa8f6a6..fac1540 100644
--- a/libAACdec/src/aac_ram.cpp
+++ b/libAACdec/src/aac_ram.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -148,7 +148,7 @@
/*! The buffer holds time samples for the crossfade in case of an USAC DASH IPF
config change Dimension: (8)
*/
-C_ALLOC_MEM2(TimeDataFlush, INT_PCM, TIME_DATA_FLUSH_SIZE, (8))
+C_ALLOC_MEM2(TimeDataFlush, PCM_DEC, TIME_DATA_FLUSH_SIZE, (8))
/* @} */
diff --git a/libAACdec/src/aac_ram.h b/libAACdec/src/aac_ram.h
index b9b95b7..395b2b2 100644
--- a/libAACdec/src/aac_ram.h
+++ b/libAACdec/src/aac_ram.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -132,7 +132,7 @@
H_ALLOC_MEM(SpectralCoeffs, FIXP_DBL)
H_ALLOC_MEM(SpecScale, SHORT)
-H_ALLOC_MEM(TimeDataFlush, INT_PCM)
+H_ALLOC_MEM(TimeDataFlush, PCM_DEC)
H_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1)
H_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL)
diff --git a/libAACdec/src/aacdec_drc.cpp b/libAACdec/src/aacdec_drc.cpp
index b6f5b49..760a9ba 100644
--- a/libAACdec/src/aacdec_drc.cpp
+++ b/libAACdec/src/aacdec_drc.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -150,6 +150,19 @@
}
/*!
+\brief Disable DRC
+
+\self Handle of DRC info
+
+\return none
+*/
+void aacDecoder_drcDisable(HANDLE_AAC_DRC self) {
+ self->enable = 0;
+ self->applyExtGain = 0;
+ self->progRefLevelPresent = 0;
+}
+
+/*!
\brief Reset DRC information
\self Handle of DRC info
diff --git a/libAACdec/src/aacdec_drc.h b/libAACdec/src/aacdec_drc.h
index 76a44d6..2bb945d 100644
--- a/libAACdec/src/aacdec_drc.h
+++ b/libAACdec/src/aacdec_drc.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -140,6 +140,8 @@
/**
* \brief DRC module interface functions
*/
+void aacDecoder_drcDisable(HANDLE_AAC_DRC self);
+
void aacDecoder_drcReset(HANDLE_AAC_DRC self);
void aacDecoder_drcInit(HANDLE_AAC_DRC self);
diff --git a/libAACdec/src/aacdec_hcrs.cpp b/libAACdec/src/aacdec_hcrs.cpp
index 44b32a5..5e3f9ac 100644
--- a/libAACdec/src/aacdec_hcrs.cpp
+++ b/libAACdec/src/aacdec_hcrs.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -173,7 +173,9 @@
pHcr->segmentInfo.readDirection = FROM_RIGHT_TO_LEFT;
/* Process sets subsequently */
+ numSet = fMin(numSet, (UCHAR)MAX_HCR_SETS);
for (currentSet = 1; currentSet < numSet; currentSet++) {
+
/* step 1 */
numCodeword -=
*pNumSegment; /* number of remaining non PCWs [for all sets] */
diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp
index 965631b..d5f0cea 100644
--- a/libAACdec/src/aacdecoder.cpp
+++ b/libAACdec/src/aacdecoder.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -494,6 +494,75 @@
return error;
}
+static INT findElementInstanceTag(
+ INT elementTag, MP4_ELEMENT_ID elementId,
+ CAacDecoderChannelInfo **pAacDecoderChannelInfo, INT nChannels,
+ MP4_ELEMENT_ID *pElementIdTab, INT nElements) {
+ int el, chCnt = 0;
+
+ for (el = 0; el < nElements; el++) {
+ switch (pElementIdTab[el]) {
+ case ID_CPE:
+ case ID_SCE:
+ case ID_LFE:
+ if ((elementTag == pAacDecoderChannelInfo[chCnt]->ElementInstanceTag) &&
+ (elementId == pElementIdTab[el])) {
+ return 1; /* element instance tag found */
+ }
+ chCnt += (pElementIdTab[el] == ID_CPE) ? 2 : 1;
+ break;
+ default:
+ break;
+ }
+ if (chCnt >= nChannels) break;
+ if (pElementIdTab[el] == ID_END) break;
+ }
+
+ return 0; /* element instance tag not found */
+}
+
+static INT validateElementInstanceTags(
+ CProgramConfig *pce, CAacDecoderChannelInfo **pAacDecoderChannelInfo,
+ INT nChannels, MP4_ELEMENT_ID *pElementIdTab, INT nElements) {
+ if (nChannels >= pce->NumChannels) {
+ for (int el = 0; el < pce->NumFrontChannelElements; el++) {
+ if (!findElementInstanceTag(pce->FrontElementTagSelect[el],
+ pce->FrontElementIsCpe[el] ? ID_CPE : ID_SCE,
+ pAacDecoderChannelInfo, nChannels,
+ pElementIdTab, nElements)) {
+ return 0; /* element instance tag not in raw_data_block() */
+ }
+ }
+ for (int el = 0; el < pce->NumSideChannelElements; el++) {
+ if (!findElementInstanceTag(pce->SideElementTagSelect[el],
+ pce->SideElementIsCpe[el] ? ID_CPE : ID_SCE,
+ pAacDecoderChannelInfo, nChannels,
+ pElementIdTab, nElements)) {
+ return 0; /* element instance tag not in raw_data_block() */
+ }
+ }
+ for (int el = 0; el < pce->NumBackChannelElements; el++) {
+ if (!findElementInstanceTag(pce->BackElementTagSelect[el],
+ pce->BackElementIsCpe[el] ? ID_CPE : ID_SCE,
+ pAacDecoderChannelInfo, nChannels,
+ pElementIdTab, nElements)) {
+ return 0; /* element instance tag not in raw_data_block() */
+ }
+ }
+ for (int el = 0; el < pce->NumLfeChannelElements; el++) {
+ if (!findElementInstanceTag(pce->LfeElementTagSelect[el], ID_LFE,
+ pAacDecoderChannelInfo, nChannels,
+ pElementIdTab, nElements)) {
+ return 0; /* element instance tag not in raw_data_block() */
+ }
+ }
+ } else {
+ return 0; /* too less decoded audio channels */
+ }
+
+ return 1; /* all element instance tags found in raw_data_block() */
+}
+
/*!
\brief Read Program Config Element
@@ -568,7 +637,7 @@
\return Error code
*/
LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
- const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+ const PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
const INT frameSize, const INT interleaved) {
int i, ch, s1, s2;
AAC_DECODER_ERROR ErrorStatus;
@@ -584,7 +653,7 @@
}
for (ch = 0; ch < numChannels; ch++) {
- const INT_PCM *pIn = &pTimeData[ch * s1];
+ const PCM_DEC *pIn = &pTimeData[ch * s1];
for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
pTimeDataFlush[ch][i] = *pIn;
pIn += s2;
@@ -606,7 +675,7 @@
\return Error code
*/
LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
- INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+ PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
const INT frameSize, const INT interleaved) {
int i, ch, s1, s2;
AAC_DECODER_ERROR ErrorStatus;
@@ -622,15 +691,15 @@
}
for (ch = 0; ch < numChannels; ch++) {
- INT_PCM *pIn = &pTimeData[ch * s1];
+ PCM_DEC *pIn = &pTimeData[ch * s1];
for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
FIXP_SGL alpha = (FIXP_SGL)i
<< (FRACT_BITS - 1 - TIME_DATA_FLUSH_SIZE_SF);
- FIXP_DBL time = FX_PCM2FX_DBL(*pIn);
- FIXP_DBL timeFlush = FX_PCM2FX_DBL(pTimeDataFlush[ch][i]);
+ FIXP_DBL time = PCM_DEC2FIXP_DBL(*pIn);
+ FIXP_DBL timeFlush = PCM_DEC2FIXP_DBL(pTimeDataFlush[ch][i]);
- *pIn = (INT_PCM)(FIXP_PCM)FX_DBL2FX_PCM(
- timeFlush - fMult(timeFlush, alpha) + fMult(time, alpha));
+ *pIn = FIXP_DBL2PCM_DEC(timeFlush - fMult(timeFlush, alpha) +
+ fMult(time, alpha));
pIn += s2;
}
}
@@ -753,7 +822,12 @@
/* We are interested in preroll AUs if an explicit or an implicit config
* change is signalized in other words if the build up status is set. */
if (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON) {
- self->applyCrossfade |= FDKreadBit(hBs);
+ UCHAR applyCrossfade = FDKreadBit(hBs);
+ if (applyCrossfade) {
+ self->applyCrossfade |= AACDEC_CROSSFADE_BITMASK_PREROLL;
+ } else {
+ self->applyCrossfade &= ~AACDEC_CROSSFADE_BITMASK_PREROLL;
+ }
FDKreadBit(hBs); /* reserved */
/* Read num preroll AU's */
*numPrerollAU = escapedValue(hBs, 2, 4, 0);
@@ -1397,6 +1471,27 @@
}
/*!
+ * \brief CAacDecoder_AcceptFlags Accept flags and element flags
+ *
+ * \param self [o] handle to AACDECODER structure
+ * \param asc [i] handle to ASC structure
+ * \param flags [i] flags
+ * \param elFlags [i] pointer to element flags
+ * \param streamIndex [i] stream index
+ * \param elementOffset [i] element offset
+ *
+ * \return void
+ */
+static void CAacDecoder_AcceptFlags(HANDLE_AACDECODER self,
+ const CSAudioSpecificConfig *asc,
+ UINT flags, UINT *elFlags, int streamIndex,
+ int elementOffset) {
+ FDKmemcpy(self->elFlags, elFlags, sizeof(self->elFlags));
+
+ self->flags[streamIndex] = flags;
+}
+
+/*!
* \brief CAacDecoder_CtrlCFGChange Set config change parameters.
*
* \param self [i] handle to AACDECODER structure
@@ -1493,6 +1588,15 @@
const int streamIndex = 0;
INT flushChannels = 0;
+ UINT flags;
+ /* elFlags[(3*MAX_CHANNELS + (MAX_CHANNELS)/2 + 4 * (MAX_TRACKS) + 1]
+ where MAX_CHANNELS is (8*2) and MAX_TRACKS is 1 */
+ UINT elFlags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)];
+
+ UCHAR sbrEnabled = self->sbrEnabled;
+ UCHAR sbrEnabledPrev = self->sbrEnabledPrev;
+ UCHAR mpsEnableCurr = self->mpsEnableCurr;
+
if (!self) return AAC_DEC_INVALID_HANDLE;
UCHAR downscaleFactor = self->downscaleFactor;
@@ -1649,8 +1753,8 @@
}
/* Set syntax flags */
- self->flags[streamIndex] = 0;
- { FDKmemclear(self->elFlags, sizeof(self->elFlags)); }
+ flags = 0;
+ { FDKmemclear(elFlags, sizeof(elFlags)); }
if ((asc->m_channelConfiguration > 0) || IS_USAC(asc->m_aot)) {
if (IS_USAC(asc->m_aot)) {
@@ -1676,7 +1780,7 @@
asc->m_sc.m_usacConfig.m_usacNumElements;
}
- self->mpsEnableCurr = 0;
+ mpsEnableCurr = 0;
for (int _el = 0;
_el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements;
_el++) {
@@ -1696,35 +1800,34 @@
self->usacStereoConfigIndex[el] =
asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex;
if (self->elements[el] == ID_USAC_CPE) {
- self->mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0;
+ mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0;
}
}
- self->elFlags[el] |=
- (asc->m_sc.m_usacConfig.element[_el].m_noiseFilling)
- ? AC_EL_USAC_NOISE
- : 0;
- self->elFlags[el] |=
+ elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_noiseFilling)
+ ? AC_EL_USAC_NOISE
+ : 0;
+ elFlags[el] |=
(asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex > 0)
? AC_EL_USAC_MPS212
: 0;
- self->elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_interTes)
- ? AC_EL_USAC_ITES
- : 0;
- self->elFlags[el] |=
+ elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_interTes)
+ ? AC_EL_USAC_ITES
+ : 0;
+ elFlags[el] |=
(asc->m_sc.m_usacConfig.element[_el].m_pvc) ? AC_EL_USAC_PVC : 0;
- self->elFlags[el] |=
+ elFlags[el] |=
(asc->m_sc.m_usacConfig.element[_el].usacElementType == ID_USAC_LFE)
? AC_EL_USAC_LFE
: 0;
- self->elFlags[el] |=
+ elFlags[el] |=
(asc->m_sc.m_usacConfig.element[_el].usacElementType == ID_USAC_LFE)
? AC_EL_LFE
: 0;
if ((asc->m_sc.m_usacConfig.element[_el].usacElementType ==
ID_USAC_CPE) &&
((self->usacStereoConfigIndex[el] == 0))) {
- self->elFlags[el] |= AC_EL_USAC_CP_POSSIBLE;
+ elFlags[el] |= AC_EL_USAC_CP_POSSIBLE;
}
}
@@ -1791,9 +1894,17 @@
downscaleFactorInBS =
asc->m_samplingFrequency /
asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency;
- if (downscaleFactorInBS == 1 || downscaleFactorInBS == 2 ||
- downscaleFactorInBS == 3 || downscaleFactorInBS == 4) {
+ if ((downscaleFactorInBS == 1 || downscaleFactorInBS == 2 ||
+ (downscaleFactorInBS == 3 &&
+ asc->m_sc.m_eldSpecificConfig.m_frameLengthFlag) ||
+ downscaleFactorInBS == 4) &&
+ ((asc->m_samplingFrequency %
+ asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency) ==
+ 0)) {
downscaleFactor = downscaleFactorInBS;
+ } else {
+ downscaleFactorInBS = 1;
+ downscaleFactor = 1;
}
}
} else {
@@ -1825,7 +1936,7 @@
self->useLdQmfTimeAlign =
asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
}
- if (self->sbrEnabled != asc->m_sbrPresentFlag) {
+ if (sbrEnabled != asc->m_sbrPresentFlag) {
ascChanged = 1;
}
}
@@ -1838,16 +1949,16 @@
if (configMode & AC_CM_ALLOC_MEM) {
self->streamInfo.extSamplingRate = asc->m_extensionSamplingFrequency;
}
- self->flags[streamIndex] |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0;
- self->flags[streamIndex] |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0;
+ flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0;
+ flags |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0;
if (asc->m_sbrPresentFlag) {
- self->sbrEnabled = 1;
- self->sbrEnabledPrev = 1;
+ sbrEnabled = 1;
+ sbrEnabledPrev = 1;
} else {
- self->sbrEnabled = 0;
- self->sbrEnabledPrev = 0;
+ sbrEnabled = 0;
+ sbrEnabledPrev = 0;
}
- if (self->sbrEnabled && asc->m_extensionSamplingFrequency) {
+ if (sbrEnabled && asc->m_extensionSamplingFrequency) {
if (downscaleFactor != 1 && (downscaleFactor)&1) {
return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* SBR needs an even downscale
factor */
@@ -1865,51 +1976,47 @@
}
/* --------- vcb11 ------------ */
- self->flags[streamIndex] |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0;
+ flags |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0;
/* ---------- rvlc ------------ */
- self->flags[streamIndex] |= (asc->m_rvlcFlag) ? AC_ER_RVLC : 0;
+ flags |= (asc->m_rvlcFlag) ? AC_ER_RVLC : 0;
/* ----------- hcr ------------ */
- self->flags[streamIndex] |= (asc->m_hcrFlag) ? AC_ER_HCR : 0;
+ flags |= (asc->m_hcrFlag) ? AC_ER_HCR : 0;
if (asc->m_aot == AOT_ER_AAC_ELD) {
- self->mpsEnableCurr = 0;
- self->flags[streamIndex] |= AC_ELD;
- self->flags[streamIndex] |=
- (asc->m_sbrPresentFlag)
- ? AC_SBR_PRESENT
- : 0; /* Need to set the SBR flag for backward-compatibility
- reasons. Even if SBR is not supported. */
- self->flags[streamIndex] |=
- (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0;
- self->flags[streamIndex] |=
- (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) ? AC_MPS_PRESENT
- : 0;
+ mpsEnableCurr = 0;
+ flags |= AC_ELD;
+ flags |= (asc->m_sbrPresentFlag)
+ ? AC_SBR_PRESENT
+ : 0; /* Need to set the SBR flag for backward-compatibility
+ reasons. Even if SBR is not supported. */
+ flags |= (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0;
+ flags |= (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign)
+ ? AC_MPS_PRESENT
+ : 0;
if (self->mpsApplicable) {
- self->mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
+ mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
}
}
- self->flags[streamIndex] |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0;
- self->flags[streamIndex] |= (asc->m_epConfig >= 0) ? AC_ER : 0;
+ flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0;
+ flags |= (asc->m_epConfig >= 0) ? AC_ER : 0;
if (asc->m_aot == AOT_USAC) {
- self->flags[streamIndex] |= AC_USAC;
- self->flags[streamIndex] |=
- (asc->m_sc.m_usacConfig.element[0].m_stereoConfigIndex > 0)
- ? AC_MPS_PRESENT
- : 0;
+ flags |= AC_USAC;
+ flags |= (asc->m_sc.m_usacConfig.element[0].m_stereoConfigIndex > 0)
+ ? AC_MPS_PRESENT
+ : 0;
}
if (asc->m_aot == AOT_DRM_AAC) {
- self->flags[streamIndex] |= AC_DRM | AC_SBRCRC | AC_SCALABLE;
+ flags |= AC_DRM | AC_SBRCRC | AC_SCALABLE;
}
if (asc->m_aot == AOT_DRM_SURROUND) {
- self->flags[streamIndex] |=
- AC_DRM | AC_SBRCRC | AC_SCALABLE | AC_MPS_PRESENT;
+ flags |= AC_DRM | AC_SBRCRC | AC_SCALABLE | AC_MPS_PRESENT;
FDK_ASSERT(!asc->m_psPresentFlag);
}
if ((asc->m_aot == AOT_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_SCAL)) {
- self->flags[streamIndex] |= AC_SCALABLE;
+ flags |= AC_SCALABLE;
}
if ((asc->m_epConfig >= 0) && (asc->m_channelConfiguration <= 0)) {
@@ -1960,13 +2067,17 @@
if (ascChanged != 0) {
*configChanged = 1;
}
+
+ CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex,
+ elementOffset);
+
return err;
}
/* set AC_USAC_SCFGI3 globally if any usac element uses */
switch (asc->m_aot) {
case AOT_USAC:
- if (self->sbrEnabled) {
+ if (sbrEnabled) {
for (int _el = 0;
_el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements;
_el++) {
@@ -1988,7 +2099,7 @@
}
if (usacStereoConfigIndex == 3) {
- self->flags[streamIndex] |= AC_USAC_SCFGI3;
+ flags |= AC_USAC_SCFGI3;
}
}
break;
@@ -2003,7 +2114,7 @@
*/
switch (asc->m_aot) {
case AOT_USAC:
- if (self->sbrEnabled) {
+ if (sbrEnabled) {
const UCHAR map_sbrRatio_2_nAnaBands[] = {16, 24, 32};
FDK_ASSERT(asc->m_sc.m_usacConfig.m_sbrRatioIndex > 0);
@@ -2031,11 +2142,11 @@
}
break;
case AOT_ER_AAC_ELD:
- if (self->mpsEnableCurr &&
+ if (mpsEnableCurr &&
asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) {
- SAC_INPUT_CONFIG sac_interface =
- (self->sbrEnabled && self->hSbrDecoder) ? SAC_INTERFACE_QMF
- : SAC_INTERFACE_TIME;
+ SAC_INPUT_CONFIG sac_interface = (sbrEnabled && self->hSbrDecoder)
+ ? SAC_INTERFACE_QMF
+ : SAC_INTERFACE_TIME;
mpegSurroundDecoder_ConfigureQmfDomain(
(CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface,
(UINT)self->streamInfo.aacSampleRate, asc->m_aot);
@@ -2069,14 +2180,14 @@
ch = aacChannelsOffset;
int _numElements;
_numElements = (((8)) + (8));
- if (self->flags[streamIndex] & (AC_RSV603DA | AC_USAC)) {
+ if (flags & (AC_RSV603DA | AC_USAC)) {
_numElements = (int)asc->m_sc.m_usacConfig.m_usacNumElements;
}
for (int _el = 0; _el < _numElements; _el++) {
int el_channels = 0;
int el = elementOffset + _el;
- if (self->flags[streamIndex] &
+ if (flags &
(AC_ER | AC_LD | AC_ELD | AC_RSV603DA | AC_USAC | AC_RSVD50)) {
if (ch >= ascChannels) {
break;
@@ -2176,15 +2287,14 @@
if (self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer == NULL) {
goto bail;
}
- if (self->flags[streamIndex] &
- (AC_USAC | AC_RSVD50 | AC_RSV603DA /*|AC_BSAC*/)) {
+ if (flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA /*|AC_BSAC*/)) {
self->pAacDecoderStaticChannelInfo[ch]->hArCo = CArco_Create();
if (self->pAacDecoderStaticChannelInfo[ch]->hArCo == NULL) {
goto bail;
}
}
- if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) {
+ if (!(flags & (AC_USAC | AC_RSV603DA))) {
CPns_UpdateNoiseState(
&self->pAacDecoderChannelInfo[ch]->data.aac.PnsData,
&self->pAacDecoderStaticChannelInfo[ch]->pnsCurrentSeed,
@@ -2195,7 +2305,7 @@
chIdx++;
}
- if (self->flags[streamIndex] & AC_USAC) {
+ if (flags & AC_USAC) {
for (int _ch = 0; _ch < flushChannels; _ch++) {
ch = aacChannelsOffset + _ch;
if (self->pTimeDataFlush[ch] == NULL) {
@@ -2207,7 +2317,7 @@
}
}
- if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA)) {
+ if (flags & (AC_USAC | AC_RSV603DA)) {
int complexStereoPredPossible = 0;
ch = aacChannelsOffset;
chIdx = aacChannelsOffsetIdx;
@@ -2223,7 +2333,7 @@
elCh = 1;
}
- if (self->elFlags[el2] & AC_EL_USAC_CP_POSSIBLE) {
+ if (elFlags[el2] & AC_EL_USAC_CP_POSSIBLE) {
complexStereoPredPossible = 1;
if (self->cpeStaticData[el2] == NULL) {
self->cpeStaticData[el2] = GetCpePersistentData();
@@ -2360,9 +2470,6 @@
}
}
- /* Update externally visible copy of flags */
- self->streamInfo.flags = self->flags[0];
-
if (*configChanged) {
int drcDecSampleRate, drcDecFrameSize;
@@ -2383,8 +2490,7 @@
if (*configChanged) {
if (asc->m_aot == AOT_USAC) {
- self->hDrcInfo->enable = 0;
- self->hDrcInfo->progRefLevelPresent = 0;
+ aacDecoder_drcDisable(self->hDrcInfo);
}
}
@@ -2393,6 +2499,15 @@
pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f));
}
+ CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex,
+ elementOffset);
+ self->sbrEnabled = sbrEnabled;
+ self->sbrEnabledPrev = sbrEnabledPrev;
+ self->mpsEnableCurr = mpsEnableCurr;
+
+ /* Update externally visible copy of flags */
+ self->streamInfo.flags = self->flags[0];
+
return err;
bail:
@@ -2927,6 +3042,24 @@
} /* while ( (type != ID_END) ... ) */
+ if (!(self->flags[streamIndex] &
+ (AC_USAC | AC_RSVD50 | AC_RSV603DA | AC_BSAC | AC_LD | AC_ELD | AC_ER |
+ AC_SCALABLE)) &&
+ (self->streamInfo.channelConfig == 0) && pce->isValid &&
+ (ErrorStatus == AAC_DEC_OK) && self->frameOK &&
+ !(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) {
+ /* Check whether all PCE listed element instance tags are present in
+ * raw_data_block() */
+ if (!validateElementInstanceTags(
+ &self->pce, self->pAacDecoderChannelInfo, aacChannels,
+ channel_elements,
+ fMin(channel_element_count, (int)(sizeof(channel_elements) /
+ sizeof(*channel_elements))))) {
+ ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
+ self->frameOK = 0;
+ }
+ }
+
if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) {
/* float decoder checks if bitsLeft is in range 0-7; only prerollAUs are
* byteAligned with respect to the first bit */
@@ -3194,11 +3327,12 @@
* data in the bitstream. */
self->flags[streamIndex] |= AC_DRC_PRESENT;
} else {
- self->hDrcInfo->enable = 0;
- self->hDrcInfo->progRefLevelPresent = 0;
ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT;
}
}
+ if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA)) {
+ aacDecoder_drcDisable(self->hDrcInfo);
+ }
/* Create a reverse mapping table */
UCHAR Reverse_chMapping[((8) * 2)];
@@ -3441,11 +3575,12 @@
* data in the bitstream. */
self->flags[streamIndex] |= AC_DRC_PRESENT;
} else {
- self->hDrcInfo->enable = 0;
- self->hDrcInfo->progRefLevelPresent = 0;
ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT;
}
}
+ if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA)) {
+ aacDecoder_drcDisable(self->hDrcInfo);
+ }
}
/* Add additional concealment delay */
diff --git a/libAACdec/src/aacdecoder.h b/libAACdec/src/aacdecoder.h
index bd1f38f..002807f 100644
--- a/libAACdec/src/aacdecoder.h
+++ b/libAACdec/src/aacdecoder.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -172,6 +172,12 @@
AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
};
+#define AACDEC_CROSSFADE_BITMASK_OFF \
+ ((UCHAR)0) /*!< No cross-fade between frames shall be applied at next \
+ config change. */
+#define AACDEC_CROSSFADE_BITMASK_PREROLL \
+ ((UCHAR)1 << 1) /*!< applyCrossfade is signaled in AudioPreRoll */
+
typedef struct {
/* Usac Extension Elements */
USAC_EXT_ELEMENT_TYPE usacExtElementType[(3)];
@@ -325,7 +331,7 @@
UINT loudnessInfoSetPosition[3];
SCHAR defaultTargetLoudness;
- INT_PCM
+ PCM_DEC
*pTimeDataFlush[((8) * 2)]; /*!< Pointer to the flushed time data which
will be used for the crossfade in case of
an USAC DASH IPF config change */
@@ -341,8 +347,8 @@
start position in the
bitstream */
INT accessUnit; /*!< Number of the actual processed preroll accessUnit */
- UCHAR applyCrossfade; /*!< if set crossfade for seamless stream switching is
- applied */
+ UCHAR applyCrossfade; /*!< If any bit is set, cross-fade for seamless stream
+ switching is applied */
FDK_SignalDelay usacResidualDelay; /*!< Delay residual signal to compensate
for eSBR delay of DMX signal in case of
@@ -439,12 +445,12 @@
/* Prepare crossfade for USAC DASH IPF config change */
LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
- const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+ const PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
const INT frameSize, const INT interleaved);
/* Apply crossfade for USAC DASH IPF config change */
LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
- INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+ PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
const INT frameSize, const INT interleaved);
/* Set flush and build up mode */
diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index 0f281eb..0c83191 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -385,21 +385,19 @@
return errTp;
}
-static INT aacDecoder_SscCallback(void *handle, HANDLE_FDK_BITSTREAM hBs,
- const AUDIO_OBJECT_TYPE coreCodec,
- const INT samplingRate, const INT frameSize,
- const INT stereoConfigIndex,
- const INT coreSbrFrameLengthIndex,
- const INT configBytes, const UCHAR configMode,
- UCHAR *configChanged) {
+static INT aacDecoder_SscCallback(
+ void *handle, HANDLE_FDK_BITSTREAM hBs, const AUDIO_OBJECT_TYPE coreCodec,
+ const INT samplingRate, const INT frameSize, const INT numChannels,
+ const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex,
+ const INT configBytes, const UCHAR configMode, UCHAR *configChanged) {
SACDEC_ERROR err;
TRANSPORTDEC_ERROR errTp;
HANDLE_AACDECODER hAacDecoder = (HANDLE_AACDECODER)handle;
err = mpegSurroundDecoder_Config(
(CMpegSurroundDecoder *)hAacDecoder->pMpegSurroundDecoder, hBs, coreCodec,
- samplingRate, frameSize, stereoConfigIndex, coreSbrFrameLengthIndex,
- configBytes, configMode, configChanged);
+ samplingRate, frameSize, numChannels, stereoConfigIndex,
+ coreSbrFrameLengthIndex, configBytes, configMode, configChanged);
switch (err) {
case MPS_UNSUPPORTED_CONFIG:
@@ -443,12 +441,23 @@
TRANSPORTDEC_ERROR errTp;
HANDLE_AACDECODER hAacDecoder = (HANDLE_AACDECODER)handle;
DRC_DEC_CODEC_MODE drcDecCodecMode = DRC_DEC_CODEC_MODE_UNDEFINED;
+ UCHAR dummyBuffer[4] = {0};
+ FDK_BITSTREAM dummyBs;
+ HANDLE_FDK_BITSTREAM hReadBs;
if (subStreamIndex != 0) {
return TRANSPORTDEC_OK;
}
- else if (aot == AOT_USAC) {
+ if (hBs == NULL) {
+ /* use dummy zero payload to clear memory */
+ hReadBs = &dummyBs;
+ FDKinitBitStream(hReadBs, dummyBuffer, 4, 24);
+ } else {
+ hReadBs = hBs;
+ }
+
+ if (aot == AOT_USAC) {
drcDecCodecMode = DRC_DEC_MPEG_D_USAC;
}
@@ -457,10 +466,10 @@
if (payloadType == 0) /* uniDrcConfig */
{
- err = FDK_drcDec_ReadUniDrcConfig(hAacDecoder->hUniDrcDecoder, hBs);
+ err = FDK_drcDec_ReadUniDrcConfig(hAacDecoder->hUniDrcDecoder, hReadBs);
} else /* loudnessInfoSet */
{
- err = FDK_drcDec_ReadLoudnessInfoSet(hAacDecoder->hUniDrcDecoder, hBs);
+ err = FDK_drcDec_ReadLoudnessInfoSet(hAacDecoder->hUniDrcDecoder, hReadBs);
hAacDecoder->loudnessInfoSetPosition[1] = payloadStart;
hAacDecoder->loudnessInfoSetPosition[2] = fullPayloadLength;
}
@@ -822,6 +831,9 @@
case AAC_DRC_ATTENUATION_FACTOR:
/* DRC compression factor (where 0 is no and 127 is max compression) */
+ if ((value < 0) || (value > 127)) {
+ return AAC_DEC_SET_PARAM_FAIL;
+ }
errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_CUT_SCALE, value);
uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_COMPRESS,
value * (FL2FXCONST_DBL(0.5f / 127.0f)));
@@ -829,6 +841,9 @@
case AAC_DRC_BOOST_FACTOR:
/* DRC boost factor (where 0 is no and 127 is max boost) */
+ if ((value < 0) || (value > 127)) {
+ return AAC_DEC_SET_PARAM_FAIL;
+ }
errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_BOOST_SCALE, value);
uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_BOOST,
value * (FL2FXCONST_DBL(0.5f / 127.0f)));
@@ -1151,6 +1166,8 @@
int applyCrossfade = 1; /* flag indicates if flushing was possible */
PCM_DEC *pTimeData2;
PCM_AAC *pTimeData3;
+ INT pcmLimiterScale = 0;
+ INT interleaved = 0;
if (self == NULL) {
return AAC_DEC_INVALID_HANDLE;
@@ -1173,8 +1190,10 @@
aacDecoder_FreeMemCallback(self, &asc);
self->streamInfo.numChannels = 0;
/* 3) restore AudioSpecificConfig */
- transportDec_OutOfBandConfig(self->hInput, asc.config,
- (asc.configBits + 7) >> 3, 0);
+ if (asc.configBits <= (TP_USAC_MAX_CONFIG_LEN << 3)) {
+ transportDec_OutOfBandConfig(self->hInput, asc.config,
+ (asc.configBits + 7) >> 3, 0);
+ }
}
}
@@ -1607,6 +1626,11 @@
/* set params */
sbrDecoder_SetParam(self->hSbrDecoder, SBR_SYSTEM_BITSTREAM_DELAY,
self->sbrParams.bsDelay);
+ sbrDecoder_SetParam(
+ self->hSbrDecoder, SBR_FLUSH_DATA,
+ (flags & AACDEC_FLUSH) |
+ ((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH
+ : 0));
sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, 1);
@@ -1794,8 +1818,7 @@
}
if (self->streamInfo.extAot != AOT_AAC_SLS) {
- INT pcmLimiterScale = 0;
- INT interleaved = 0;
+ interleaved = 0;
interleaved |= (self->sbrEnabled) ? 1 : 0;
interleaved |= (self->mpsEnableCurr) ? 1 : 0;
PCMDMX_ERROR dmxErr = PCMDMX_OK;
@@ -1826,145 +1849,38 @@
* predictable behavior and thus maybe produce strange output. */
ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
}
-
- pcmLimiterScale += PCM_OUT_HEADROOM;
-
- if (flags & AACDEC_CLRHIST) {
- if (!(self->flags[0] & AC_USAC)) {
- /* Reset DRC data */
- aacDecoder_drcReset(self->hDrcInfo);
- /* Delete the delayed signal. */
- pcmLimiter_Reset(self->hLimiter);
- }
- }
-
- /* Set applyExtGain if DRC processing is enabled and if
- progRefLevelPresent is present for the first time. Consequences: The
- headroom of the output signal can be set to AACDEC_DRC_GAIN_SCALING
- only for audio formats which support legacy DRC Level Normalization.
- For all other audio formats the headroom of the output
- signal is set to PCM_OUT_HEADROOM. */
- if (self->hDrcInfo->enable &&
- (self->hDrcInfo->progRefLevelPresent == 1)) {
- self->hDrcInfo->applyExtGain |= 1;
- }
-
- /* Check whether time data buffer is large enough. */
- if (timeDataSize <
- (self->streamInfo.numChannels * self->streamInfo.frameSize)) {
- ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
- goto bail;
- }
-
- if (self->limiterEnableCurr) {
- /* use workBufferCore2 buffer for interleaving */
- PCM_LIM *pInterleaveBuffer;
- int blockLength = self->streamInfo.frameSize;
-
- /* Set actual signal parameters */
- pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels);
- pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate);
-
- if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
- (self->mpsEnableCurr)) {
- pInterleaveBuffer = (PCM_LIM *)pTimeData2;
- } else {
- pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2;
-
- /* applyLimiter requests for interleaved data */
- /* Interleave ouput buffer */
- FDK_interleave(pTimeData2, pInterleaveBuffer,
- self->streamInfo.numChannels, blockLength,
- self->streamInfo.frameSize);
- }
-
- FIXP_DBL *pGainPerSample = NULL;
-
- if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
- pGainPerSample = self->workBufferCore1;
-
- if ((INT)GetRequiredMemWorkBufferCore1() <
- (INT)(self->streamInfo.frameSize * sizeof(FIXP_DBL))) {
- ErrorStatus = AAC_DEC_UNKNOWN;
- goto bail;
- }
-
- pcmLimiterScale = applyDrcLevelNormalization(
- self->hDrcInfo, (PCM_DEC *)pInterleaveBuffer, self->extGain,
- pGainPerSample, pcmLimiterScale, self->extGainDelay,
- self->streamInfo.frameSize, self->streamInfo.numChannels, 1, 1);
- }
-
- pcmLimiter_Apply(self->hLimiter, pInterleaveBuffer, pTimeData,
- pGainPerSample, pcmLimiterScale,
- self->streamInfo.frameSize);
-
- {
- /* Announce the additional limiter output delay */
- self->streamInfo.outputDelay += pcmLimiter_GetDelay(self->hLimiter);
- }
- } else {
- if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
- pcmLimiterScale = applyDrcLevelNormalization(
- self->hDrcInfo, pTimeData2, self->extGain, NULL,
- pcmLimiterScale, self->extGainDelay, self->streamInfo.frameSize,
- self->streamInfo.numChannels,
- (interleaved || (self->streamInfo.numChannels == 1))
- ? 1
- : self->streamInfo.frameSize,
- 0);
- }
-
- /* If numChannels = 1 we do not need interleaving. The same applies if
- SBR or MPS are used, since their output is interleaved already
- (resampled or not) */
- if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
- (self->mpsEnableCurr)) {
- scaleValuesSaturate(
- pTimeData, pTimeData2,
- self->streamInfo.frameSize * self->streamInfo.numChannels,
- pcmLimiterScale);
-
- } else {
- scaleValuesSaturate(
- (INT_PCM *)self->workBufferCore2, pTimeData2,
- self->streamInfo.frameSize * self->streamInfo.numChannels,
- pcmLimiterScale);
- /* Interleave ouput buffer */
- FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData,
- self->streamInfo.numChannels,
- self->streamInfo.frameSize,
- self->streamInfo.frameSize);
- }
- }
- } /* if (self->streamInfo.extAot != AOT_AAC_SLS)*/
+ }
if (self->flags[0] & AC_USAC) {
if (self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON &&
!(flags & AACDEC_CONCEAL)) {
- CAacDecoder_PrepareCrossFade(pTimeData, self->pTimeDataFlush,
+ CAacDecoder_PrepareCrossFade(pTimeData2, self->pTimeDataFlush,
self->streamInfo.numChannels,
- self->streamInfo.frameSize, 1);
+ self->streamInfo.frameSize, interleaved);
}
/* prepare crossfade buffer for fade in */
- if (!applyCrossfade && self->applyCrossfade &&
+ if (!applyCrossfade &&
+ (self->applyCrossfade != AACDEC_CROSSFADE_BITMASK_OFF) &&
!(flags & AACDEC_CONCEAL)) {
for (int ch = 0; ch < self->streamInfo.numChannels; ch++) {
for (int i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
- self->pTimeDataFlush[ch][i] = 0;
+ self->pTimeDataFlush[ch][i] = (PCM_DEC)0;
}
}
applyCrossfade = 1;
}
- if (applyCrossfade && self->applyCrossfade &&
+ if (applyCrossfade &&
+ (self->applyCrossfade != AACDEC_CROSSFADE_BITMASK_OFF) &&
!(accessUnit < numPrerollAU) &&
(self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON)) {
- CAacDecoder_ApplyCrossFade(pTimeData, self->pTimeDataFlush,
+ CAacDecoder_ApplyCrossFade(pTimeData2, self->pTimeDataFlush,
self->streamInfo.numChannels,
- self->streamInfo.frameSize, 1);
- self->applyCrossfade = 0;
+ self->streamInfo.frameSize, interleaved);
+ self->applyCrossfade =
+ AACDEC_CROSSFADE_BITMASK_OFF; /* disable cross-fade between frames
+ at nect config change */
}
}
@@ -2006,6 +1922,116 @@
((self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON) &&
!(flags & AACDEC_CONCEAL)));
+ if (self->streamInfo.extAot != AOT_AAC_SLS) {
+ pcmLimiterScale += PCM_OUT_HEADROOM;
+
+ if (flags & AACDEC_CLRHIST) {
+ if (!(self->flags[0] & AC_USAC)) {
+ /* Reset DRC data */
+ aacDecoder_drcReset(self->hDrcInfo);
+ /* Delete the delayed signal. */
+ pcmLimiter_Reset(self->hLimiter);
+ }
+ }
+
+ /* Set applyExtGain if DRC processing is enabled and if progRefLevelPresent
+ is present for the first time. Consequences: The headroom of the output
+ signal can be set to AACDEC_DRC_GAIN_SCALING only for audio formats which
+ support legacy DRC Level Normalization. For all other audio formats the
+ headroom of the output signal is set to PCM_OUT_HEADROOM. */
+ if (self->hDrcInfo->enable && (self->hDrcInfo->progRefLevelPresent == 1)) {
+ self->hDrcInfo->applyExtGain |= 1;
+ }
+
+ /* Check whether time data buffer is large enough. */
+ if (timeDataSize <
+ (self->streamInfo.numChannels * self->streamInfo.frameSize)) {
+ ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
+ goto bail;
+ }
+
+ if (self->limiterEnableCurr) {
+ /* use workBufferCore2 buffer for interleaving */
+ PCM_LIM *pInterleaveBuffer;
+ int blockLength = self->streamInfo.frameSize;
+
+ /* Set actual signal parameters */
+ pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels);
+ pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate);
+
+ if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
+ (self->mpsEnableCurr)) {
+ pInterleaveBuffer = (PCM_LIM *)pTimeData2;
+ } else {
+ pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2;
+
+ /* applyLimiter requests for interleaved data */
+ /* Interleave ouput buffer */
+ FDK_interleave(pTimeData2, pInterleaveBuffer,
+ self->streamInfo.numChannels, blockLength,
+ self->streamInfo.frameSize);
+ }
+
+ FIXP_DBL *pGainPerSample = NULL;
+
+ if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
+ pGainPerSample = self->workBufferCore1;
+
+ if ((INT)GetRequiredMemWorkBufferCore1() <
+ (INT)(self->streamInfo.frameSize * sizeof(FIXP_DBL))) {
+ ErrorStatus = AAC_DEC_UNKNOWN;
+ goto bail;
+ }
+
+ pcmLimiterScale = applyDrcLevelNormalization(
+ self->hDrcInfo, (PCM_DEC *)pInterleaveBuffer, self->extGain,
+ pGainPerSample, pcmLimiterScale, self->extGainDelay,
+ self->streamInfo.frameSize, self->streamInfo.numChannels, 1, 1);
+ }
+
+ pcmLimiter_Apply(self->hLimiter, pInterleaveBuffer, pTimeData,
+ pGainPerSample, pcmLimiterScale,
+ self->streamInfo.frameSize);
+
+ {
+ /* Announce the additional limiter output delay */
+ self->streamInfo.outputDelay += pcmLimiter_GetDelay(self->hLimiter);
+ }
+ } else {
+ if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
+ pcmLimiterScale = applyDrcLevelNormalization(
+ self->hDrcInfo, pTimeData2, self->extGain, NULL, pcmLimiterScale,
+ self->extGainDelay, self->streamInfo.frameSize,
+ self->streamInfo.numChannels,
+ (interleaved || (self->streamInfo.numChannels == 1))
+ ? 1
+ : self->streamInfo.frameSize,
+ 0);
+ }
+
+ /* If numChannels = 1 we do not need interleaving. The same applies if SBR
+ or MPS are used, since their output is interleaved already (resampled or
+ not) */
+ if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
+ (self->mpsEnableCurr)) {
+ scaleValuesSaturate(
+ pTimeData, pTimeData2,
+ self->streamInfo.frameSize * self->streamInfo.numChannels,
+ pcmLimiterScale);
+
+ } else {
+ scaleValuesSaturate(
+ (INT_PCM *)self->workBufferCore2, pTimeData2,
+ self->streamInfo.frameSize * self->streamInfo.numChannels,
+ pcmLimiterScale);
+ /* Interleave ouput buffer */
+ FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData,
+ self->streamInfo.numChannels, self->streamInfo.frameSize,
+ self->streamInfo.frameSize);
+ }
+ }
+ } /* if (self->streamInfo.extAot != AOT_AAC_SLS)*/
+
bail:
/* error in renderer part occurred, ErrorStatus was set to invalid output */
diff --git a/libAACdec/src/channel.cpp b/libAACdec/src/channel.cpp
index a020034..7e62bfb 100644
--- a/libAACdec/src/channel.cpp
+++ b/libAACdec/src/channel.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -265,7 +265,9 @@
stereo prediction since scaling has already been carried out. */
int max_sfb_ste = (INT)(pAacDecoderChannelInfo[L]->icsInfo.max_sfb_ste);
- if ((!CP_active) || (CP_active && (max_sfb_ste < noSfbs)) ||
+ if (!(CP_active && (max_sfb_ste == noSfbs)) ||
+ !(CP_active &&
+ !(pAacDecoderChannelInfo[ch]->pDynData->TnsData.Active)) ||
((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) &&
(pAacDecoderChannelInfo[L]->pDynData->specificTo.usac.tns_on_lr ==
0))) {
diff --git a/libAACdec/src/rvlc.cpp b/libAACdec/src/rvlc.cpp
index b7a9be1..0b80364 100644
--- a/libAACdec/src/rvlc.cpp
+++ b/libAACdec/src/rvlc.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -628,7 +628,7 @@
SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd;
SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc;
- UCHAR *pEscEscCnt = &(pRvlc->numDecodedEscapeWordsEsc);
+ UCHAR escEscCnt = pRvlc->numDecodedEscapeWordsEsc;
UCHAR *pEscBwdCnt = &(pRvlc->numDecodedEscapeWordsBwd);
pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_bwd);
@@ -636,7 +636,7 @@
*pEscBwdCnt = 0;
pRvlc->direction = BWD;
- pScfEsc += *pEscEscCnt - 1; /* set pScfEsc to last entry */
+ pScfEsc += escEscCnt - 1; /* set pScfEsc to last entry */
pRvlc->firstScf = 0;
pRvlc->firstNrg = 0;
pRvlc->firstIs = 0;
@@ -651,7 +651,7 @@
}
dpcm -= TABLE_OFFSET;
if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
- if (pRvlc->length_of_rvlc_escapes) {
+ if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
pRvlc->conceal_min = bnds;
return;
} else {
@@ -694,7 +694,7 @@
}
dpcm -= TABLE_OFFSET;
if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
- if (pRvlc->length_of_rvlc_escapes) {
+ if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
pScfBwd[bnds] = position;
pRvlc->conceal_min = fMax(0, bnds - offset);
return;
@@ -731,7 +731,8 @@
}
dpcm -= TABLE_OFFSET;
if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
- if (pRvlc->length_of_rvlc_escapes) {
+ if ((pRvlc->length_of_rvlc_escapes) ||
+ (*pEscBwdCnt >= escEscCnt)) {
pScfBwd[bnds] = noisenrg;
pRvlc->conceal_min = fMax(0, bnds - offset);
return;
@@ -762,7 +763,7 @@
}
dpcm -= TABLE_OFFSET;
if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
- if (pRvlc->length_of_rvlc_escapes) {
+ if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
pScfBwd[bnds] = factor;
pRvlc->conceal_min = fMax(0, bnds - offset);
return;
diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp
index a8dadc0..ca1a6a2 100644
--- a/libAACdec/src/usacdec_acelp.cpp
+++ b/libAACdec/src/usacdec_acelp.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -719,7 +719,7 @@
UCHAR *pold_T0_frac = &acelp_mem->old_T0_frac;
if ((int)*pold_T0 >= PIT_MAX) {
- *pold_T0 = (UCHAR)(PIT_MAX - 5);
+ *pold_T0 = (USHORT)(PIT_MAX - 5);
}
*pT0 = (int)*pold_T0;
*pT0_frac = (int)*pold_T0_frac;
diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h
index 71f7556..f0f23b4 100644
--- a/libAACenc/include/aacenc_lib.h
+++ b/libAACenc/include/aacenc_lib.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1643,7 +1643,7 @@
*
* \return
* - AACENC_OK, on succes.
- * - AACENC_INIT_ERROR, on failure.
+ * - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure.
*/
AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder,
AACENC_InfoStruct *pInfo);
diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp
index 0ae329b..c11db27 100644
--- a/libAACenc/src/aacenc_lib.cpp
+++ b/libAACenc/src/aacenc_lib.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1242,7 +1242,7 @@
INT aacenc_SscCallback(void *self, HANDLE_FDK_BITSTREAM hBs,
const AUDIO_OBJECT_TYPE coreCodec,
const INT samplingRate, const INT frameSize,
- const INT stereoConfigIndex,
+ const INT numChannels, const INT stereoConfigIndex,
const INT coreSbrFrameLengthIndex, const INT configBytes,
const UCHAR configMode, UCHAR *configChanged) {
HANDLE_AACENCODER hAacEncoder = (HANDLE_AACENCODER)self;
@@ -1784,8 +1784,8 @@
hAacEncoder->nSamplesRead));
INT_PCM *pIn =
hAacEncoder->inputBuffer +
- (hAacEncoder->inputBufferOffset + hAacEncoder->nSamplesRead) /
- hAacEncoder->aacConfig.nChannels;
+ hAacEncoder->inputBufferOffset / hAacEncoder->aacConfig.nChannels +
+ hAacEncoder->nSamplesRead / hAacEncoder->extParam.nChannels;
newSamples -=
(newSamples %
hAacEncoder->extParam
@@ -1827,12 +1827,13 @@
/* clear out until end-of-buffer */
if (nZeros) {
+ INT_PCM *pIn =
+ hAacEncoder->inputBuffer +
+ hAacEncoder->inputBufferOffset /
+ hAacEncoder->aacConfig.nChannels +
+ hAacEncoder->nSamplesRead / hAacEncoder->extParam.nChannels;
for (i = 0; i < (int)hAacEncoder->extParam.nChannels; i++) {
- FDKmemclear(hAacEncoder->inputBuffer +
- i * hAacEncoder->inputBufferSizePerChannel +
- (hAacEncoder->inputBufferOffset +
- hAacEncoder->nSamplesRead) /
- hAacEncoder->extParam.nChannels,
+ FDKmemclear(pIn + i * hAacEncoder->inputBufferSizePerChannel,
sizeof(INT_PCM) * nZeros);
}
hAacEncoder->nZerosAppended += nZeros;
@@ -2520,6 +2521,11 @@
AACENC_InfoStruct *pInfo) {
AACENC_ERROR err = AACENC_OK;
+ if ((hAacEncoder == NULL) || (pInfo == NULL)) {
+ err = AACENC_INVALID_HANDLE;
+ goto bail;
+ }
+
FDKmemclear(pInfo, sizeof(AACENC_InfoStruct));
pInfo->confSize = 64; /* pre-initialize */
diff --git a/libDRCdec/src/drcDec_reader.cpp b/libDRCdec/src/drcDec_reader.cpp
index 367a352..b080f50 100644
--- a/libDRCdec/src/drcDec_reader.cpp
+++ b/libDRCdec/src/drcDec_reader.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -512,10 +512,13 @@
fMin(tmpNNodes, (UCHAR)16) * sizeof(GAIN_NODE));
}
- hUniDrcGain->uniDrcGainExtPresent = FDKreadBits(hBs, 1);
- if (hUniDrcGain->uniDrcGainExtPresent == 1) {
- err = _readUniDrcGainExtension(hBs, &(hUniDrcGain->uniDrcGainExtension));
- if (err) return err;
+ if (pCoef && (gainSequenceCount ==
+ pCoef->gainSequenceCount)) { /* all sequences have been read */
+ hUniDrcGain->uniDrcGainExtPresent = FDKreadBits(hBs, 1);
+ if (hUniDrcGain->uniDrcGainExtPresent == 1) {
+ err = _readUniDrcGainExtension(hBs, &(hUniDrcGain->uniDrcGainExtension));
+ if (err) return err;
+ }
}
if (err == DE_OK && gainSequenceCount > 0) {
@@ -914,7 +917,7 @@
firFilterOrder;
int uniqueEqSubbandGainsCount, eqSubbandGainRepresentation,
eqSubbandGainCount;
- EQ_SUBBAND_GAIN_FORMAT eqSubbandGainFormat;
+ int eqSubbandGainFormat;
eqDelayMaxPresent = FDKreadBits(hBs, 1);
if (eqDelayMaxPresent) {
@@ -955,7 +958,7 @@
uniqueEqSubbandGainsCount = FDKreadBits(hBs, 6);
if (uniqueEqSubbandGainsCount > 0) {
eqSubbandGainRepresentation = FDKreadBits(hBs, 1);
- eqSubbandGainFormat = (EQ_SUBBAND_GAIN_FORMAT)FDKreadBits(hBs, 4);
+ eqSubbandGainFormat = FDKreadBits(hBs, 4);
switch (eqSubbandGainFormat) {
case GF_QMF32:
eqSubbandGainCount = 32;
diff --git a/libFDK/include/nlc_dec.h b/libFDK/include/nlc_dec.h
index cca97f1..aded569 100644
--- a/libFDK/include/nlc_dec.h
+++ b/libFDK/include/nlc_dec.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -159,9 +159,6 @@
#ifndef HUFFDEC_PARAMS
#define HUFFDEC_PARMS
-#define PAIR_SHIFT 4
-#define PAIR_MASK 0xf
-
#define MAX_ENTRIES 168
#define HANDLE_HUFF_NODE const SHORT(*)[MAX_ENTRIES][2]
diff --git a/libFDK/src/FDK_hybrid.cpp b/libFDK/src/FDK_hybrid.cpp
index 08d32a8..d208abd 100644
--- a/libFDK/src/FDK_hybrid.cpp
+++ b/libFDK/src/FDK_hybrid.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -539,11 +539,11 @@
i6 = pQmfImag[pReadIdx[6]] >> 2;
FDK_ASSERT((invert == 0) || (invert == 1));
- mHybridReal[0 + invert] = (r6 + r1) << 1;
- mHybridImag[0 + invert] = (i6 + i1) << 1;
+ mHybridReal[0 + invert] = SATURATE_LEFT_SHIFT((r6 + r1), 1, DFRACT_BITS);
+ mHybridImag[0 + invert] = SATURATE_LEFT_SHIFT((i6 + i1), 1, DFRACT_BITS);
- mHybridReal[1 - invert] = (r6 - r1) << 1;
- mHybridImag[1 - invert] = (i6 - i1) << 1;
+ mHybridReal[1 - invert] = SATURATE_LEFT_SHIFT((r6 - r1), 1, DFRACT_BITS);
+ mHybridImag[1 - invert] = SATURATE_LEFT_SHIFT((i6 - i1), 1, DFRACT_BITS);
}
static void fourChannelFiltering(const FIXP_DBL *const pQmfReal,
@@ -766,15 +766,15 @@
mHybridReal[3] = pfft[FFT_IDX_R(1)] << sc;
mHybridImag[3] = pfft[FFT_IDX_I(1)] << sc;
- mHybridReal[4] = pfft[FFT_IDX_R(2)] << sc;
- mHybridReal[4] += pfft[FFT_IDX_R(5)] << sc;
- mHybridImag[4] = pfft[FFT_IDX_I(2)] << sc;
- mHybridImag[4] += pfft[FFT_IDX_I(5)] << sc;
+ mHybridReal[4] = SATURATE_LEFT_SHIFT(
+ (pfft[FFT_IDX_R(2)] + pfft[FFT_IDX_R(5)]), sc, DFRACT_BITS);
+ mHybridImag[4] = SATURATE_LEFT_SHIFT(
+ (pfft[FFT_IDX_I(2)] + pfft[FFT_IDX_I(5)]), sc, DFRACT_BITS);
- mHybridReal[5] = pfft[FFT_IDX_R(3)] << sc;
- mHybridReal[5] += pfft[FFT_IDX_R(4)] << sc;
- mHybridImag[5] = pfft[FFT_IDX_I(3)] << sc;
- mHybridImag[5] += pfft[FFT_IDX_I(4)] << sc;
+ mHybridReal[5] = SATURATE_LEFT_SHIFT(
+ (pfft[FFT_IDX_R(3)] + pfft[FFT_IDX_R(4)]), sc, DFRACT_BITS);
+ mHybridImag[5] = SATURATE_LEFT_SHIFT(
+ (pfft[FFT_IDX_I(3)] + pfft[FFT_IDX_I(4)]), sc, DFRACT_BITS);
} else {
for (k = 0; k < 8; k++) {
mHybridReal[k] = pfft[FFT_IDX_R(k)] << sc;
diff --git a/libFDK/src/autocorr2nd.cpp b/libFDK/src/autocorr2nd.cpp
index 718a555..8c5673c 100644
--- a/libFDK/src/autocorr2nd.cpp
+++ b/libFDK/src/autocorr2nd.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -102,11 +102,6 @@
#include "autocorr2nd.h"
-/* If the accumulator does not provide enough overflow bits,
- products have to be shifted down in the autocorrelation below. */
-#define SHIFT_FACTOR (5)
-#define SHIFT >> (SHIFT_FACTOR)
-
/*!
*
* \brief Calculate second order autocorrelation using 2 accumulators
@@ -126,45 +121,49 @@
const FIXP_DBL *realBuf = reBuffer;
+ const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)(len / 2)), 1);
/*
r11r,r22r
r01r,r12r
r02r
*/
pReBuf = realBuf - 2;
- accu5 = ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3]))
- SHIFT);
+ accu5 =
+ ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >>
+ len_scale);
pReBuf++;
/* len must be even */
- accu1 = fPow2Div2(pReBuf[0]) SHIFT;
- accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) SHIFT;
+ accu1 = fPow2Div2(pReBuf[0]) >> len_scale;
+ accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) >> len_scale;
pReBuf++;
for (j = (len - 2) >> 1; j != 0; j--, pReBuf += 2) {
- accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) SHIFT);
+ accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) >> len_scale);
- accu3 += ((fMultDiv2(pReBuf[0], pReBuf[1]) +
- fMultDiv2(pReBuf[1], pReBuf[2])) SHIFT);
+ accu3 +=
+ ((fMultDiv2(pReBuf[0], pReBuf[1]) + fMultDiv2(pReBuf[1], pReBuf[2])) >>
+ len_scale);
- accu5 += ((fMultDiv2(pReBuf[0], pReBuf[2]) +
- fMultDiv2(pReBuf[1], pReBuf[3])) SHIFT);
+ accu5 +=
+ ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >>
+ len_scale);
}
- accu2 = (fPow2Div2(realBuf[-2]) SHIFT);
+ accu2 = (fPow2Div2(realBuf[-2]) >> len_scale);
accu2 += accu1;
- accu1 += (fPow2Div2(realBuf[len - 2]) SHIFT);
+ accu1 += (fPow2Div2(realBuf[len - 2]) >> len_scale);
- accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) SHIFT);
+ accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) >> len_scale);
accu4 += accu3;
- accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) SHIFT);
+ accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) >> len_scale);
mScale = CntLeadingZeros(
(accu1 | accu2 | fAbs(accu3) | fAbs(accu4) | fAbs(accu5))) -
1;
- autoCorrScaling = mScale - 1 - SHIFT_FACTOR; /* -1 because of fMultDiv2*/
+ autoCorrScaling = mScale - 1 - len_scale; /* -1 because of fMultDiv2*/
/* Scale to common scale factor */
ac->r11r = accu1 << mScale;
@@ -190,7 +189,7 @@
const FIXP_DBL *imBuffer, /*!< Pointer to imag part of input samples */
const int len /*!< Number of input samples (should be smaller than 128) */
) {
- int j, autoCorrScaling, mScale, len_scale;
+ int j, autoCorrScaling, mScale;
FIXP_DBL accu0, accu1, accu2, accu3, accu4, accu5, accu6, accu7, accu8;
@@ -199,7 +198,7 @@
const FIXP_DBL *realBuf = reBuffer;
const FIXP_DBL *imagBuf = imBuffer;
- (len > 64) ? (len_scale = 6) : (len_scale = 5);
+ const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)len), 1);
/*
r00r,
r11r,r22r
diff --git a/libFDK/src/dct.cpp b/libFDK/src/dct.cpp
index bd26736..35507b5 100644
--- a/libFDK/src/dct.cpp
+++ b/libFDK/src/dct.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -305,9 +305,8 @@
{
for (i = 0; i < M; i++) {
- tmp[i] = pDat[2 * i] >> 1; /* dit_fft expects 1 bit scaled input values */
- tmp[L - 1 - i] =
- pDat[2 * i + 1] >> 1; /* dit_fft expects 1 bit scaled input values */
+ tmp[i] = pDat[2 * i] >> 2;
+ tmp[L - 1 - i] = pDat[2 * i + 1] >> 2;
}
}
@@ -337,15 +336,14 @@
a1 = ((pTmp_0[0] >> 1) + (pTmp_1[0] >> 1));
a2 = ((pTmp_0[1] >> 1) - (pTmp_1[1] >> 1));
- cplxMultDiv2(&accu3, &accu4, (a1 + accu2), -(accu1 + a2),
- sin_twiddle[i * inc]);
- pDat[L - i] = accu4;
- pDat[i] = accu3;
+ cplxMult(&accu3, &accu4, (accu1 + a2), (a1 + accu2), sin_twiddle[i * inc]);
+ pDat[L - i] = -accu3;
+ pDat[i] = accu4;
- cplxMultDiv2(&accu3, &accu4, (a1 - accu2), -(accu1 - a2),
- sin_twiddle[(M - i) * inc]);
- pDat[M + i] = accu4;
- pDat[M - i] = accu3;
+ cplxMult(&accu3, &accu4, (accu1 - a2), (a1 - accu2),
+ sin_twiddle[(M - i) * inc]);
+ pDat[M + i] = -accu3;
+ pDat[M - i] = accu4;
/* Create index helper variables for (4*i)*inc indexed equivalent values of
* short tables. */
@@ -356,12 +354,12 @@
}
}
- cplxMultDiv2(&accu1, &accu2, tmp[M], tmp[M + 1], sin_twiddle[(M / 2) * inc]);
+ cplxMult(&accu1, &accu2, tmp[M], tmp[M + 1], sin_twiddle[(M / 2) * inc]);
pDat[L - (M / 2)] = accu2;
pDat[M / 2] = accu1;
- pDat[0] = (tmp[0] >> 1) + (tmp[1] >> 1);
- pDat[M] = fMult(((tmp[0] >> 1) - (tmp[1] >> 1)),
+ pDat[0] = tmp[0] + tmp[1];
+ pDat[M] = fMult(tmp[0] - tmp[1],
sin_twiddle[M * inc].v.re); /* cos((PI/(2*L))*M); */
*pDat_e += 2;
diff --git a/libFDK/src/nlc_dec.cpp b/libFDK/src/nlc_dec.cpp
index 6e98ce0..3733d98 100644
--- a/libFDK/src/nlc_dec.cpp
+++ b/libFDK/src/nlc_dec.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -568,12 +568,12 @@
static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
SCHAR* out_data_2, DATA_TYPE data_type,
DIFF_TYPE diff_type_1, DIFF_TYPE diff_type_2,
- int num_val, CODING_SCHEME* cdg_scheme, int ldMode) {
+ int num_val, PAIRING* pairing_scheme, int ldMode) {
ERROR_t err = HUFFDEC_OK;
+ CODING_SCHEME coding_scheme = HUFF_1D;
DIFF_TYPE diff_type;
int i = 0;
- ULONG data = 0;
SCHAR pair_vec[28][2];
@@ -596,15 +596,13 @@
int hufYY;
/* Coding scheme */
- data = FDKreadBits(strm, 1);
- *cdg_scheme = (CODING_SCHEME)(data << PAIR_SHIFT);
+ coding_scheme = (CODING_SCHEME)FDKreadBits(strm, 1);
- if (*cdg_scheme >> PAIR_SHIFT == HUFF_2D) {
+ if (coding_scheme == HUFF_2D) {
if ((out_data_1 != NULL) && (out_data_2 != NULL) && (ldMode == 0)) {
- data = FDKreadBits(strm, 1);
- *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | data);
+ *pairing_scheme = (PAIRING)FDKreadBits(strm, 1);
} else {
- *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | FREQ_PAIR);
+ *pairing_scheme = FREQ_PAIR;
}
}
@@ -613,7 +611,7 @@
hufYY2 = diff_type_2;
}
- switch (*cdg_scheme >> PAIR_SHIFT) {
+ switch (coding_scheme) {
case HUFF_1D:
p0_flag[0] = (diff_type_1 == DIFF_FREQ);
p0_flag[1] = (diff_type_2 == DIFF_FREQ);
@@ -634,7 +632,7 @@
case HUFF_2D:
- switch (*cdg_scheme & PAIR_MASK) {
+ switch (*pairing_scheme) {
case FREQ_PAIR:
if (out_data_1 != NULL) {
@@ -843,7 +841,7 @@
SCHAR* pDataVec[2] = {NULL, NULL};
DIFF_TYPE diff_type[2] = {DIFF_FREQ, DIFF_FREQ};
- CODING_SCHEME cdg_scheme = HUFF_1D;
+ PAIRING pairing = FREQ_PAIR;
DIRECTION direction = BACKWARDS;
switch (data_type) {
@@ -959,7 +957,7 @@
}
/* Huffman decoding */
err = huff_decode(strm, pDataVec[0], pDataVec[1], data_type, diff_type[0],
- diff_type[1], dataBands, &cdg_scheme,
+ diff_type[1], dataBands, &pairing,
(DECODER == SAOC_DECODER));
if (err != HUFFDEC_OK) {
return HUFFDEC_NOTOK;
@@ -986,8 +984,8 @@
}
}
- mixed_time_pair = (diff_type[0] != diff_type[1]) &&
- ((cdg_scheme & PAIR_MASK) == TIME_PAIR);
+ mixed_time_pair =
+ (diff_type[0] != diff_type[1]) && (pairing == TIME_PAIR);
if (direction == BACKWARDS) {
if (diff_type[0] == DIFF_FREQ) {
diff --git a/libMpegTPDec/include/tp_data.h b/libMpegTPDec/include/tp_data.h
index b015332..b63087a 100644
--- a/libMpegTPDec/include/tp_data.h
+++ b/libMpegTPDec/include/tp_data.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -368,7 +368,7 @@
typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM,
const AUDIO_OBJECT_TYPE coreCodec,
const INT samplingRate, const INT frameSize,
- const INT stereoConfigIndex,
+ const INT numChannels, const INT stereoConfigIndex,
const INT coreSbrFrameLengthIndex, const INT configBytes,
const UCHAR configMode, UCHAR *configChanged);
diff --git a/libMpegTPDec/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp
index 82f840e..8f77017 100644
--- a/libMpegTPDec/src/tpdec_asc.cpp
+++ b/libMpegTPDec/src/tpdec_asc.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -266,11 +266,118 @@
return (err);
}
+/**
+ * \brief Sanity checks for program config element.
+ * Check order of elements according to ISO/IEC 13818-7:2003(E),
+ * chapter 8.5.1
+ *
+ * \param pPce pointer to program config element.
+ *
+ * \return 0 if successful, otherwise 1.
+ */
+static int CProgramConfig_Check(CProgramConfig *pPce) {
+ INT i;
+ INT err = 0;
+ INT numBackChannels[3] = {0};
+ INT numSideChannels[3] = {0};
+ INT numFrontChannels[3] = {0};
+ UCHAR *pCpeFront = pPce->FrontElementIsCpe;
+ UCHAR *pCpeSide = pPce->SideElementIsCpe;
+ UCHAR *pCpeBack = pPce->BackElementIsCpe;
+ UCHAR *pHeight;
+
+ pHeight = pPce->BackElementHeightInfo;
+ for (i = 0; i < pPce->NumBackChannelElements; i++) {
+ numBackChannels[*pHeight] += pPce->BackElementIsCpe[i] ? 2 : 1;
+ pHeight++;
+ }
+ pHeight = pPce->SideElementHeightInfo;
+ for (i = 0; i < pPce->NumSideChannelElements; i++) {
+ numSideChannels[*pHeight] += pPce->SideElementIsCpe[i] ? 2 : 1;
+ pHeight++;
+ }
+ pHeight = pPce->FrontElementHeightInfo;
+ for (i = 0; i < pPce->NumFrontChannelElements; i++) {
+ numFrontChannels[*pHeight] += pPce->FrontElementIsCpe[i] ? 2 : 1;
+ pHeight++;
+ }
+
+ /* 0 = normal height channels, 1 = top height channels, 2 = bottom height
+ * channels */
+ for (i = 0; i < 3; i++) {
+ /* if number of channels is odd => first element must be a SCE (front center
+ * channel) */
+ if (numFrontChannels[i] & 1) {
+ if (*pCpeFront++ == ID_CPE) {
+ err = 1;
+ goto bail;
+ }
+ numFrontChannels[i]--;
+ }
+ while (numFrontChannels[i] > 0) {
+ /* must be CPE or paired SCE */
+ if (*pCpeFront++ == ID_SCE) {
+ if (*pCpeFront++ == ID_CPE) {
+ err = 1;
+ goto bail;
+ }
+ }
+ numFrontChannels[i] -= 2;
+ };
+
+ /* in case that a top center surround channel (Ts) is transmitted the number
+ * of channels can be odd */
+ if (i != 1) {
+ /* number of channels must be even */
+ if (numSideChannels[i] & 1) {
+ err = 1;
+ goto bail;
+ }
+ while (numSideChannels[i] > 0) {
+ /* must be CPE or paired SCE */
+ if (*pCpeSide++ == ID_SCE) {
+ if (*pCpeSide++ == ID_CPE) {
+ err = 1;
+ goto bail;
+ }
+ }
+ numSideChannels[i] -= 2;
+ };
+ }
+
+ while (numBackChannels[i] > 1) {
+ /* must be CPE or paired SCE */
+ if (*pCpeBack++ == ID_SCE) {
+ if (*pCpeBack++ == ID_CPE) {
+ err = 1;
+ goto bail;
+ }
+ }
+ numBackChannels[i] -= 2;
+ };
+ /* if number of channels is odd => last element must be a SCE (back center
+ * channel) */
+ if (numBackChannels[i]) {
+ if (*pCpeBack++ == ID_CPE) {
+ err = 1;
+ goto bail;
+ }
+ }
+ }
+
+bail:
+
+ return err;
+}
+
void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs,
UINT alignmentAnchor) {
- int i, err = 0;
+ int i;
int commentBytes;
+ UCHAR tag, isCpe;
+ UCHAR checkElementTagSelect[3][PC_FSB_CHANNELS_MAX] = {{0}};
+ pPce->isValid = 1;
pPce->NumEffectiveChannels = 0;
pPce->NumChannels = 0;
pPce->ElementInstanceTag = (UCHAR)FDKreadBits(bs, 4);
@@ -297,28 +404,60 @@
}
for (i = 0; i < pPce->NumFrontChannelElements; i++) {
- pPce->FrontElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1);
- pPce->FrontElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4);
+ pPce->FrontElementIsCpe[i] = isCpe = (UCHAR)FDKreadBits(bs, 1);
+ pPce->FrontElementTagSelect[i] = tag = (UCHAR)FDKreadBits(bs, 4);
pPce->NumChannels += pPce->FrontElementIsCpe[i] ? 2 : 1;
+
+ /* Check element instance tag according to ISO/IEC 13818-7:2003(E),
+ * chapter 8.2.1.1 */
+ if (checkElementTagSelect[isCpe][tag] == 0) {
+ checkElementTagSelect[isCpe][tag] = 1;
+ } else {
+ pPce->isValid = 0;
+ }
}
for (i = 0; i < pPce->NumSideChannelElements; i++) {
- pPce->SideElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1);
- pPce->SideElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4);
+ pPce->SideElementIsCpe[i] = isCpe = (UCHAR)FDKreadBits(bs, 1);
+ pPce->SideElementTagSelect[i] = tag = (UCHAR)FDKreadBits(bs, 4);
pPce->NumChannels += pPce->SideElementIsCpe[i] ? 2 : 1;
+
+ /* Check element instance tag according to ISO/IEC 13818-7:2003(E),
+ * chapter 8.2.1.1 */
+ if (checkElementTagSelect[isCpe][tag] == 0) {
+ checkElementTagSelect[isCpe][tag] = 1;
+ } else {
+ pPce->isValid = 0;
+ }
}
for (i = 0; i < pPce->NumBackChannelElements; i++) {
- pPce->BackElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1);
- pPce->BackElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4);
+ pPce->BackElementIsCpe[i] = isCpe = (UCHAR)FDKreadBits(bs, 1);
+ pPce->BackElementTagSelect[i] = tag = (UCHAR)FDKreadBits(bs, 4);
pPce->NumChannels += pPce->BackElementIsCpe[i] ? 2 : 1;
+
+ /* Check element instance tag according to ISO/IEC 13818-7:2003(E),
+ * chapter 8.2.1.1 */
+ if (checkElementTagSelect[isCpe][tag] == 0) {
+ checkElementTagSelect[isCpe][tag] = 1;
+ } else {
+ pPce->isValid = 0;
+ }
}
pPce->NumEffectiveChannels = pPce->NumChannels;
for (i = 0; i < pPce->NumLfeChannelElements; i++) {
- pPce->LfeElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4);
+ pPce->LfeElementTagSelect[i] = tag = (UCHAR)FDKreadBits(bs, 4);
pPce->NumChannels += 1;
+
+ /* Check element instance tag according to ISO/IEC 13818-7:2003(E),
+ * chapter 8.2.1.1 */
+ if (checkElementTagSelect[2][tag] == 0) {
+ checkElementTagSelect[2][tag] = 1;
+ } else {
+ pPce->isValid = 0;
+ }
}
for (i = 0; i < pPce->NumAssocDataElements; i++) {
@@ -336,7 +475,15 @@
commentBytes = pPce->CommentFieldBytes;
/* Search for height info extension and read it if available */
- err = CProgramConfig_ReadHeightExt(pPce, bs, &commentBytes, alignmentAnchor);
+ if (CProgramConfig_ReadHeightExt(pPce, bs, &commentBytes, alignmentAnchor)) {
+ pPce->isValid = 0;
+ }
+
+ /* Check order of elements according to ISO / IEC 13818 - 7:2003(E),
+ * chapter 8.5.1 */
+ if (CProgramConfig_Check(pPce)) {
+ pPce->isValid = 0;
+ }
for (i = 0; i < commentBytes; i++) {
UCHAR text;
@@ -347,8 +494,6 @@
pPce->Comment[i] = text;
}
}
-
- pPce->isValid = (err) ? 0 : 1;
}
/*
@@ -1415,7 +1560,7 @@
cb->cbSscData, hBs, asc->m_aot,
asc->m_samplingFrequency << esc->m_sbrSamplingRate,
asc->m_samplesPerFrame << esc->m_sbrSamplingRate,
- 1, /* stereoConfigIndex */
+ asc->m_channelConfiguration, 1, /* stereoConfigIndex */
-1, /* nTimeSlots: read from bitstream */
eldExtLen, asc->configMode, &asc->SacConfigChanged);
if (ErrorStatus != TRANSPORTDEC_OK) {
@@ -1549,8 +1694,7 @@
const AUDIO_OBJECT_TYPE aot) {
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
- USAC_EXT_ELEMENT_TYPE usacExtElementType =
- (USAC_EXT_ELEMENT_TYPE)escapedValue(hBs, 4, 8, 16);
+ UINT usacExtElementType = escapedValue(hBs, 4, 8, 16);
/* recurve extension elements which are invalid for USAC */
if (aot == AOT_USAC) {
@@ -1567,7 +1711,6 @@
}
}
- extElement->usacExtElementType = usacExtElementType;
int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16);
extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength;
INT bsAnchor;
@@ -1601,8 +1744,10 @@
}
} break;
default:
+ usacExtElementType = ID_EXT_ELE_UNKNOWN;
break;
}
+ extElement->usacExtElementType = (USAC_EXT_ELEMENT_TYPE)usacExtElementType;
/* Adjust bit stream position. This is required because of byte alignment and
* unhandled extensions. */
@@ -1631,14 +1776,18 @@
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
int numConfigExtensions;
- CONFIG_EXT_ID usacConfigExtType;
+ UINT usacConfigExtType;
int usacConfigExtLength;
+ int loudnessInfoSetIndex =
+ -1; /* index of loudnessInfoSet config extension. -1 if not contained. */
+ int tmp_subStreamIndex = 0;
+ AUDIO_OBJECT_TYPE tmp_aot = AOT_USAC;
numConfigExtensions = (int)escapedValue(hBs, 2, 4, 8) + 1;
for (int confExtIdx = 0; confExtIdx < numConfigExtensions; confExtIdx++) {
INT nbits;
int loudnessInfoSetConfigExtensionPosition = FDKgetValidBits(hBs);
- usacConfigExtType = (CONFIG_EXT_ID)escapedValue(hBs, 4, 8, 16);
+ usacConfigExtType = escapedValue(hBs, 4, 8, 16);
usacConfigExtLength = (int)escapedValue(hBs, 4, 8, 16);
/* Start bit position of config extension */
@@ -1662,10 +1811,12 @@
ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc(
cb->cbUniDrcData, hBs, usacConfigExtLength,
1, /* loudnessInfoSet */
- 0, loudnessInfoSetConfigExtensionPosition, AOT_USAC);
+ tmp_subStreamIndex, loudnessInfoSetConfigExtensionPosition,
+ tmp_aot);
if (ErrorStatus != TRANSPORTDEC_OK) {
return ErrorStatus;
}
+ loudnessInfoSetIndex = confExtIdx;
}
} break;
default:
@@ -1681,6 +1832,17 @@
FDKpushFor(hBs, usacConfigExtLength);
}
+ if (loudnessInfoSetIndex == -1 && cb->cbUniDrc != NULL) {
+ /* no loudnessInfoSet contained. Clear the loudnessInfoSet struct by feeding
+ * an empty config extension */
+ ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc(
+ cb->cbUniDrcData, NULL, 0, 1 /* loudnessInfoSet */, tmp_subStreamIndex,
+ 0, tmp_aot);
+ if (ErrorStatus != TRANSPORTDEC_OK) {
+ return ErrorStatus;
+ }
+ }
+
return ErrorStatus;
}
@@ -1697,6 +1859,8 @@
int channelElementIdx =
0; /* index for elements which contain audio channels (sce, cpe, lfe) */
SC_CHANNEL_CONFIG sc_chan_config = {0, 0, 0, 0};
+ int uniDrcElement =
+ -1; /* index of uniDrc extension element. -1 if not contained. */
numberOfElements = (int)escapedValue(hBs, 4, 8, 16) + 1;
usc->m_usacNumElements = numberOfElements;
@@ -1827,6 +1991,8 @@
/* Mps212Config() ISO/IEC FDIS 23003-3 */
if (cb->cbSsc(cb->cbSscData, hBs, asc->m_aot,
asc->m_extensionSamplingFrequency, samplesPerFrame,
+ 1, /* only downmix channels (residual channels are
+ not counted) */
usc->element[i].m_stereoConfigIndex,
usc->m_coreSbrFrameLengthIndex,
0, /* don't know the length */
@@ -1870,6 +2036,10 @@
case ID_USAC_EXT:
ErrorStatus = extElementConfig(&usc->element[i].extElement, hBs, cb, 0,
asc->m_samplesPerFrame, 0, asc->m_aot);
+ if (usc->element[i].extElement.usacExtElementType ==
+ ID_EXT_ELE_UNI_DRC) {
+ uniDrcElement = i;
+ }
if (ErrorStatus) {
return ErrorStatus;
@@ -1898,6 +2068,18 @@
}
}
+ if (uniDrcElement == -1 && cb->cbUniDrc != NULL) {
+ /* no uniDrcConfig contained. Clear the uniDrcConfig struct by feeding an
+ * empty extension element */
+ int subStreamIndex = 0;
+ ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc(
+ cb->cbUniDrcData, NULL, 0, 0 /* uniDrcConfig */, subStreamIndex, 0,
+ asc->m_aot);
+ if (ErrorStatus != TRANSPORTDEC_OK) {
+ return ErrorStatus;
+ }
+ }
+
return ErrorStatus;
}
@@ -1984,6 +2166,14 @@
if (err != TRANSPORTDEC_OK) {
return err;
}
+ } else if (cb->cbUniDrc != NULL) {
+ /* no loudnessInfoSet contained. Clear the loudnessInfoSet struct by feeding
+ * an empty config extension */
+ err = (TRANSPORTDEC_ERROR)cb->cbUniDrc(
+ cb->cbUniDrcData, NULL, 0, 1 /* loudnessInfoSet */, 0, 0, asc->m_aot);
+ if (err != TRANSPORTDEC_OK) {
+ return err;
+ }
}
/* sanity check whether number of channels signaled in UsacDecoderConfig()
@@ -1996,9 +2186,11 @@
/* Copy UsacConfig() to asc->m_sc.m_usacConfig.UsacConfig[] buffer. */
INT configSize_bits = (INT)FDKgetValidBits(hBs) - nbits;
- StoreConfigAsBitstream(hBs, configSize_bits,
- asc->m_sc.m_usacConfig.UsacConfig,
- TP_USAC_MAX_CONFIG_LEN);
+ if (StoreConfigAsBitstream(hBs, configSize_bits,
+ asc->m_sc.m_usacConfig.UsacConfig,
+ TP_USAC_MAX_CONFIG_LEN)) {
+ return TRANSPORTDEC_PARSE_ERROR;
+ }
asc->m_sc.m_usacConfig.UsacConfigBits = fAbs(configSize_bits);
return err;
@@ -2219,7 +2411,7 @@
case AOT_MPEGS:
if (cb->cbSsc != NULL) {
if (cb->cbSsc(cb->cbSscData, bs, self->m_aot, self->m_samplingFrequency,
- self->m_samplesPerFrame, 1,
+ self->m_samplesPerFrame, self->m_channelConfiguration, 1,
-1, /* nTimeSlots: read from bitstream */
0, /* don't know the length */
self->configMode, &self->SacConfigChanged)) {
@@ -2300,8 +2492,10 @@
/* Copy config() to asc->config[] buffer. */
if ((ErrorStatus == TRANSPORTDEC_OK) && (self->m_aot == AOT_USAC)) {
INT configSize_bits = (INT)FDKgetValidBits(bs) - (INT)ascStartAnchor;
- StoreConfigAsBitstream(bs, configSize_bits, self->config,
- TP_USAC_MAX_CONFIG_LEN);
+ if (StoreConfigAsBitstream(bs, configSize_bits, self->config,
+ TP_USAC_MAX_CONFIG_LEN)) {
+ return TRANSPORTDEC_PARSE_ERROR;
+ }
self->configBits = fAbs(configSize_bits);
}
@@ -2415,6 +2609,8 @@
cb->cbSscData, hBs,
AOT_DRM_USAC, /* syntax differs from MPEG Mps212Config() */
asc->m_extensionSamplingFrequency, samplesPerFrame,
+ 1, /* only downmix channels (residual channels are not
+ counted) */
usc->element[elemIdx].m_stereoConfigIndex,
usc->m_coreSbrFrameLengthIndex, 0, /* don't know the length */
asc->configMode, &asc->SacConfigChanged);
diff --git a/libMpegTPDec/src/tpdec_latm.cpp b/libMpegTPDec/src/tpdec_latm.cpp
index 3b71db8..c32be54 100644
--- a/libMpegTPDec/src/tpdec_latm.cpp
+++ b/libMpegTPDec/src/tpdec_latm.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -591,6 +591,18 @@
return (ErrorStatus);
}
+static int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) {
+ int len = 0, tmp = 255;
+ int validBytes = (int)FDKgetValidBits(bs) >> 3;
+
+ while (tmp == 255 && validBytes-- > 0) {
+ tmp = (int)FDKreadBits(bs, 8);
+ len += tmp;
+ }
+
+ return ((tmp == 255) ? -1 : (len << 3));
+}
+
TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs,
CLatmDemux *pLatmDemux) {
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
@@ -602,11 +614,17 @@
FDK_ASSERT(pLatmDemux->m_numLayer[prog] <= LATM_MAX_LAYER);
for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) {
LATM_LAYER_INFO *p_linfo = &pLatmDemux->m_linfo[prog][lay];
+ int auChunkLengthInfo = 0;
switch (p_linfo->m_frameLengthType) {
case 0:
- p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs);
- totalPayloadBits += p_linfo->m_frameLengthInBits;
+ auChunkLengthInfo = CLatmDemux_ReadAuChunkLengthInfo(bs);
+ if (auChunkLengthInfo >= 0) {
+ p_linfo->m_frameLengthInBits = (UINT)auChunkLengthInfo;
+ totalPayloadBits += p_linfo->m_frameLengthInBits;
+ } else {
+ return TRANSPORTDEC_PARSE_ERROR;
+ }
break;
case 3:
case 5:
@@ -627,23 +645,6 @@
return (ErrorStatus);
}
-int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) {
- UCHAR endFlag;
- int len = 0;
-
- do {
- UCHAR tmp = (UCHAR)FDKreadBits(bs, 8);
- endFlag = (tmp < 255);
-
- len += tmp;
-
- } while (endFlag == 0);
-
- len <<= 3; /* convert from bytes to bits */
-
- return len;
-}
-
UINT CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux, const UINT prog,
const UINT layer) {
UINT nFrameLenBits = 0;
diff --git a/libMpegTPDec/src/tpdec_latm.h b/libMpegTPDec/src/tpdec_latm.h
index 6af553d..8b8c971 100644
--- a/libMpegTPDec/src/tpdec_latm.h
+++ b/libMpegTPDec/src/tpdec_latm.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -151,8 +151,6 @@
AudioPreRoll */
} CLatmDemux;
-int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs);
-
TRANSPORTDEC_ERROR CLatmDemux_Read(HANDLE_FDK_BITSTREAM bs,
CLatmDemux *pLatmDemux, TRANSPORT_TYPE tt,
CSTpCallBacks *pTpDecCallbacks,
diff --git a/libMpegTPEnc/include/tp_data.h b/libMpegTPEnc/include/tp_data.h
index 00de356..464c485 100644
--- a/libMpegTPEnc/include/tp_data.h
+++ b/libMpegTPEnc/include/tp_data.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -368,7 +368,7 @@
typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM,
const AUDIO_OBJECT_TYPE coreCodec,
const INT samplingRate, const INT frameSize,
- const INT stereoConfigIndex,
+ const INT numChannels, const INT stereoConfigIndex,
const INT coreSbrFrameLengthIndex, const INT configBytes,
const UCHAR configMode, UCHAR *configChanged);
diff --git a/libMpegTPEnc/src/tpenc_asc.cpp b/libMpegTPEnc/src/tpenc_asc.cpp
index 0b484a0..9591ba8 100644
--- a/libMpegTPEnc/src/tpenc_asc.cpp
+++ b/libMpegTPEnc/src/tpenc_asc.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -795,7 +795,7 @@
const INT eldExtLen =
(cb->cbSsc(cb->cbSscData, NULL, config->aot, config->extSamplingRate, 0,
- 0, 0, 0, 0, NULL) +
+ 0, 0, 0, 0, 0, NULL) +
7) >>
3;
INT cnt = eldExtLen;
@@ -818,7 +818,7 @@
}
cb->cbSsc(cb->cbSscData, hBs, config->aot, config->extSamplingRate, 0, 0, 0,
- 0, 0, NULL);
+ 0, 0, 0, NULL);
}
if (config->downscaleSamplingRate != 0 &&
diff --git a/libPCMutils/src/limiter.cpp b/libPCMutils/src/limiter.cpp
index 598dc0c..c6b8687 100644
--- a/libPCMutils/src/limiter.cpp
+++ b/libPCMutils/src/limiter.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -322,7 +322,8 @@
(FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling, DFRACT_BITS));
#else
samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(
- tmp + ((FIXP_DBL)0x8000 >> scaling), scaling, DFRACT_BITS));
+ (tmp >> 1) + ((FIXP_DBL)0x8000 >> (scaling + 1)), scaling + 1,
+ DFRACT_BITS));
#endif
}
}
diff --git a/libPCMutils/src/pcmdmx_lib.cpp b/libPCMutils/src/pcmdmx_lib.cpp
index 2070dbc..fca12ce 100644
--- a/libPCMutils/src/pcmdmx_lib.cpp
+++ b/libPCMutils/src/pcmdmx_lib.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -494,13 +494,40 @@
return plainChMode;
}
-static inline UINT getIdxSum(UCHAR numCh) {
- UINT result = 0;
- int i;
- for (i = 1; i < numCh; i += 1) {
- result += i;
+/** Validates the channel indices of all channels present in the bitstream.
+ * The channel indices have to be consecutive and unique for each audio channel
+ *type.
+ * @param [in] The total number of channels of the given configuration.
+ * @param [in] The total number of channels of the current audio channel type of
+ *the given configuration.
+ * @param [in] Audio channel type to be examined.
+ * @param [in] Array holding the corresponding channel types for each channel.
+ * @param [in] Array holding the corresponding channel type indices for each
+ *channel.
+ * @returns Returns 1 on success, returns 0 on error.
+ **/
+static UINT validateIndices(UINT numChannels, UINT numChannelsPlaneAndGrp,
+ AUDIO_CHANNEL_TYPE aChType,
+ const AUDIO_CHANNEL_TYPE channelType[],
+ const UCHAR channelIndices[]) {
+ for (UINT reqValue = 0; reqValue < numChannelsPlaneAndGrp; reqValue++) {
+ int found = FALSE;
+ for (UINT i = 0; i < numChannels; i++) {
+ if (channelType[i] == aChType) {
+ if (channelIndices[i] == reqValue) {
+ if (found == TRUE) {
+ return 0; /* Found channel index a second time */
+ } else {
+ found = TRUE; /* Found channel index */
+ }
+ }
+ }
+ }
+ if (found == FALSE) {
+ return 0; /* Did not find channel index */
+ }
}
- return result;
+ return 1; /* Successfully validated channel indices */
}
/** Evaluate a given channel configuration and extract a packed channel mode. In
@@ -523,7 +550,6 @@
UCHAR offsetTable[(8)], /* out */
PCM_DMX_CHANNEL_MODE *chMode /* out */
) {
- UINT idxSum[(3)][(4)];
UCHAR numCh[(3)][(4)];
UCHAR mapped[(8)];
PCM_DMX_SPEAKER_POSITION spkrPos[(8)];
@@ -538,7 +564,6 @@
FDK_ASSERT(chMode != NULL);
/* For details see ISO/IEC 13818-7:2005(E), 8.5.3 Channel configuration */
- FDKmemclear(idxSum, (3) * (4) * sizeof(UINT));
FDKmemclear(numCh, (3) * (4) * sizeof(UCHAR));
FDKmemclear(mapped, (8) * sizeof(UCHAR));
FDKmemclear(spkrPos, (8) * sizeof(PCM_DMX_SPEAKER_POSITION));
@@ -552,19 +577,22 @@
(channelType[ch] & 0x0F) - 1,
0); /* Assign all undefined channels (ACT_NONE) to front channels. */
numCh[channelType[ch] >> 4][chGrp] += 1;
- idxSum[channelType[ch] >> 4][chGrp] += channelIndices[ch];
}
- if (numChannels > TWO_CHANNEL) {
+
+ {
int chGrp;
/* Sanity check on the indices */
for (chGrp = 0; chGrp < (4); chGrp += 1) {
int plane;
for (plane = 0; plane < (3); plane += 1) {
- if (idxSum[plane][chGrp] != getIdxSum(numCh[plane][chGrp])) {
+ if (numCh[plane][chGrp] == 0) continue;
+ AUDIO_CHANNEL_TYPE aChType =
+ (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF));
+ if (!validateIndices(numChannels, numCh[plane][chGrp], aChType,
+ channelType, channelIndices)) {
unsigned idxCnt = 0;
for (ch = 0; ch < numChannels; ch += 1) {
- if (channelType[ch] ==
- (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF))) {
+ if (channelType[ch] == aChType) {
channelIndices[ch] = idxCnt++;
}
}
diff --git a/libSACdec/include/sac_dec_lib.h b/libSACdec/include/sac_dec_lib.h
index 1827504..5aad4e0 100644
--- a/libSACdec/include/sac_dec_lib.h
+++ b/libSACdec/include/sac_dec_lib.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -316,8 +316,8 @@
SACDEC_ERROR mpegSurroundDecoder_Config(
CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs,
AUDIO_OBJECT_TYPE coreCodec, INT samplingRate, INT frameSize,
- INT stereoConfigIndex, INT coreSbrFrameLengthIndex, INT configBytes,
- const UCHAR configMode, UCHAR *configChanged);
+ INT numChannels, INT stereoConfigIndex, INT coreSbrFrameLengthIndex,
+ INT configBytes, const UCHAR configMode, UCHAR *configChanged);
SACDEC_ERROR
mpegSurroundDecoder_ConfigureQmfDomain(
diff --git a/libSACdec/src/sac_bitdec.cpp b/libSACdec/src/sac_bitdec.cpp
index 4485ccf..25b3d9e 100644
--- a/libSACdec/src/sac_bitdec.cpp
+++ b/libSACdec/src/sac_bitdec.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -488,12 +488,17 @@
pSpatialSpecificConfig->freqRes =
(SPATIALDEC_FREQ_RES)freqResTable_LD[bsFreqRes];
- pSpatialSpecificConfig->treeConfig =
- (SPATIALDEC_TREE_CONFIG)FDKreadBits(bitstream, 4);
+ {
+ UINT treeConfig = FDKreadBits(bitstream, 4);
- if (pSpatialSpecificConfig->treeConfig != SPATIALDEC_MODE_RSVD7) {
- err = MPS_UNSUPPORTED_CONFIG;
- goto bail;
+ switch (treeConfig) {
+ case SPATIALDEC_MODE_RSVD7:
+ pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG)treeConfig;
+ break;
+ default:
+ err = MPS_UNSUPPORTED_CONFIG;
+ goto bail;
+ }
}
{
diff --git a/libSACdec/src/sac_dec.cpp b/libSACdec/src/sac_dec.cpp
index a7b50df..a26e251 100644
--- a/libSACdec/src/sac_dec.cpp
+++ b/libSACdec/src/sac_dec.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1098,6 +1098,28 @@
}
}
+/**
+ * \brief Set internal error and reset error status
+ *
+ * \param self spatialDec handle.
+ * \param bypassMode pointer to bypassMode.
+ * \param err error status.
+ *
+ * \return error status.
+ */
+static SACDEC_ERROR SpatialDecSetInternalError(spatialDec *self,
+ int *bypassMode,
+ SACDEC_ERROR err) {
+ *bypassMode = 1;
+
+ if (self->errInt == MPS_OK) {
+ /* store internal error before it gets overwritten */
+ self->errInt = err;
+ }
+
+ return MPS_OK;
+}
+
/*******************************************************************************
Functionname: SpatialDecApplyParameterSets
*******************************************************************************
@@ -1118,7 +1140,7 @@
const FDK_channelMapDescr *const mapDescr) {
SACDEC_ERROR err = MPS_OK;
- FIXP_SGL alpha;
+ FIXP_SGL alpha = FL2FXCONST_SGL(0.0);
int ts;
int ch;
@@ -1141,20 +1163,22 @@
ts++, ts_io++) {
int currSlot = frame->paramSlot[ps];
+ err = (currSlot < ts) ? MPS_WRONG_PARAMETERSETS : MPS_OK;
+ if (err != MPS_OK) {
+ err = SpatialDecSetInternalError(self, &bypassMode, err);
+ }
+
/*
* Get new parameter set
*/
if (ts == prevSlot + 1) {
- err = SpatialDecCalculateM1andM2(self, ps,
- frame); /* input: ottCLD, ottICC, ... */
- /* output: M1param(Real/Imag), M2(Real/Imag) */
- if (err != MPS_OK) {
- bypassMode = 1;
- if (self->errInt == MPS_OK) {
- /* store internal error befor it gets overwritten */
- self->errInt = err;
+ if (bypassMode == 0) {
+ err = SpatialDecCalculateM1andM2(
+ self, ps, frame); /* input: ottCLD, ottICC, ... */
+ /* output: M1param(Real/Imag), M2(Real/Imag) */
+ if (err != MPS_OK) {
+ err = SpatialDecSetInternalError(self, &bypassMode, err);
}
- err = MPS_OK;
}
if ((ps == 0) && (self->bOverwriteM1M2prev != 0)) {
@@ -1168,13 +1192,16 @@
self->bOverwriteM1M2prev = 0;
}
- SpatialDecSmoothM1andM2(
- self, frame,
- ps); /* input: M1param(Real/Imag)(Prev), M2(Real/Imag)(Prev) */
- /* output: M1param(Real/Imag), M2(Real/Imag) */
+ if (bypassMode == 0) {
+ SpatialDecSmoothM1andM2(
+ self, frame,
+ ps); /* input: M1param(Real/Imag)(Prev), M2(Real/Imag)(Prev) */
+ } /* output: M1param(Real/Imag), M2(Real/Imag) */
}
- alpha = FX_DBL2FX_SGL(fDivNorm(ts - prevSlot, currSlot - prevSlot));
+ if (bypassMode == 0) {
+ alpha = FX_DBL2FX_SGL(fDivNorm(ts - prevSlot, currSlot - prevSlot));
+ }
switch (mode) {
case INPUTMODE_QMF_SBR:
@@ -1182,15 +1209,17 @@
self->bShareDelayWithSBR = 0; /* We got no hybrid delay */
else
self->bShareDelayWithSBR = 1;
- SpatialDecFeedQMF(self, qmfInDataReal, qmfInDataImag, ts_io, bypassMode,
- self->qmfInputReal__FDK, self->qmfInputImag__FDK,
- self->numInputChannels);
+ SpatialDecFeedQMF(
+ self, qmfInDataReal, qmfInDataImag, ts_io, bypassMode,
+ self->qmfInputReal__FDK, self->qmfInputImag__FDK,
+ (bypassMode) ? numInputChannels : self->numInputChannels);
break;
case INPUTMODE_TIME:
self->bShareDelayWithSBR = 0;
- SpatialDecQMFAnalysis(self, inData, ts_io, bypassMode,
- self->qmfInputReal__FDK, self->qmfInputImag__FDK,
- self->numInputChannels);
+ SpatialDecQMFAnalysis(
+ self, inData, ts_io, bypassMode, self->qmfInputReal__FDK,
+ self->qmfInputImag__FDK,
+ (bypassMode) ? numInputChannels : self->numInputChannels);
break;
default:
break;
@@ -1360,7 +1389,7 @@
} /* !self->tempShapeConfig == 1 */
} /* !bypassMode */
- if (self->phaseCoding == 1) {
+ if ((self->phaseCoding == 1) && (bypassMode == 0)) {
/* only if bsPhaseCoding == 1 and bsResidualCoding == 0 */
SpatialDecApplyPhase(
diff --git a/libSACdec/src/sac_dec_lib.cpp b/libSACdec/src/sac_dec_lib.cpp
index 57446f8..d30131f 100644
--- a/libSACdec/src/sac_dec_lib.cpp
+++ b/libSACdec/src/sac_dec_lib.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -700,9 +700,10 @@
SACDEC_ERROR mpegSurroundDecoder_Config(
CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs,
AUDIO_OBJECT_TYPE coreCodec, INT samplingRate, INT frameSize,
- INT stereoConfigIndex, INT coreSbrFrameLengthIndex, INT configBytes,
- const UCHAR configMode, UCHAR *configChanged) {
+ INT numChannels, INT stereoConfigIndex, INT coreSbrFrameLengthIndex,
+ INT configBytes, const UCHAR configMode, UCHAR *configChanged) {
SACDEC_ERROR err = MPS_OK;
+ INT nInputChannels;
SPATIAL_SPECIFIC_CONFIG spatialSpecificConfig;
SPATIAL_SPECIFIC_CONFIG *pSsc =
&pMpegSurroundDecoder->spatialSpecificConfigBackup;
@@ -716,12 +717,18 @@
err = SpatialDecParseMps212Config(
hBs, &spatialSpecificConfig, samplingRate, coreCodec,
stereoConfigIndex, coreSbrFrameLengthIndex);
+ nInputChannels = spatialSpecificConfig.nInputChannels;
pSsc = &spatialSpecificConfig;
} else {
err = SpatialDecParseMps212Config(
hBs, &pMpegSurroundDecoder->spatialSpecificConfigBackup,
samplingRate, coreCodec, stereoConfigIndex,
coreSbrFrameLengthIndex);
+ nInputChannels =
+ pMpegSurroundDecoder->spatialSpecificConfigBackup.nInputChannels;
+ }
+ if ((err == MPS_OK) && (numChannels != nInputChannels)) {
+ err = MPS_PARSE_ERROR;
}
break;
case AOT_ER_AAC_ELD:
@@ -731,11 +738,19 @@
* into temporarily allocated structure */
err = SpatialDecParseSpecificConfig(hBs, &spatialSpecificConfig,
configBytes, coreCodec);
+ nInputChannels = spatialSpecificConfig.nInputChannels;
pSsc = &spatialSpecificConfig;
} else {
err = SpatialDecParseSpecificConfig(
hBs, &pMpegSurroundDecoder->spatialSpecificConfigBackup,
configBytes, coreCodec);
+ nInputChannels =
+ pMpegSurroundDecoder->spatialSpecificConfigBackup.nInputChannels;
+ }
+ /* check number of channels for channel_configuration > 0 */
+ if ((err == MPS_OK) && (numChannels > 0) &&
+ (numChannels != nInputChannels)) {
+ err = MPS_PARSE_ERROR;
}
break;
default:
diff --git a/libSACdec/src/sac_process.cpp b/libSACdec/src/sac_process.cpp
index 22091a9..33a1647 100644
--- a/libSACdec/src/sac_process.cpp
+++ b/libSACdec/src/sac_process.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -517,12 +517,11 @@
maxVal = fAbs(iReal0) | fAbs(iImag0);
maxVal |= fAbs(iReal1);
- s = fMax(CntLeadingZeros(maxVal) - 1, 0);
- s = fMin(s, scale_param_m2);
+ s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2);
- mReal0 = iReal0 << s;
- mImag0 = iImag0 << s;
- mReal1 = iReal1 << s;
+ mReal0 = scaleValue(iReal0, s);
+ mImag0 = scaleValue(iImag0, s);
+ mReal1 = scaleValue(iReal1, s);
s = scale_param_m2 - s;
@@ -562,12 +561,11 @@
maxVal = fAbs(iReal0) | fAbs(iImag0);
maxVal |= fAbs(iReal1);
- s = fMax(CntLeadingZeros(maxVal) - 1, 0);
- s = fMin(s, scale_param_m2);
+ s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2);
- mReal0 = FX_DBL2FX_SGL(iReal0 << s);
- mImag0 = FX_DBL2FX_SGL(iImag0 << s);
- mReal1 = FX_DBL2FX_SGL(iReal1 << s);
+ mReal0 = FX_DBL2FX_SGL(scaleValue(iReal0, s));
+ mImag0 = FX_DBL2FX_SGL(scaleValue(iImag0, s));
+ mReal1 = FX_DBL2FX_SGL(scaleValue(iReal1, s));
s = scale_param_m2 - s;
diff --git a/libSACdec/src/sac_reshapeBBEnv.cpp b/libSACdec/src/sac_reshapeBBEnv.cpp
index 272d009..72f4e58 100644
--- a/libSACdec/src/sac_reshapeBBEnv.cpp
+++ b/libSACdec/src/sac_reshapeBBEnv.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -241,29 +241,56 @@
}
}
-static inline void slotAmp(FIXP_DBL *RESTRICT slotAmp_dry,
- FIXP_DBL *RESTRICT slotAmp_wet,
- FIXP_DBL *RESTRICT pHybOutputRealDry,
- FIXP_DBL *RESTRICT pHybOutputImagDry,
- FIXP_DBL *RESTRICT pHybOutputRealWet,
- FIXP_DBL *RESTRICT pHybOutputImagWet, INT cplxBands,
- INT hybBands) {
- INT qs;
+static inline void slotAmp(
+ FIXP_DBL *RESTRICT slotAmp_dry, INT *RESTRICT slotAmp_dry_e,
+ FIXP_DBL *RESTRICT slotAmp_wet, INT *RESTRICT slotAmp_wet_e,
+ FIXP_DBL *RESTRICT pHybOutputRealDry, FIXP_DBL *RESTRICT pHybOutputImagDry,
+ FIXP_DBL *RESTRICT pHybOutputRealWet, FIXP_DBL *RESTRICT pHybOutputImagWet,
+ INT cplxBands, INT hybBands) {
+ INT qs, s1, s2, headroom_dry, headroom_wet;
FIXP_DBL dry, wet;
+ /* headroom can be reduced by 1 bit due to use of fPow2Div2 */
+ s1 = DFRACT_BITS - 1 - CntLeadingZeros(hybBands + cplxBands);
+ headroom_dry = fMin(getScalefactor(pHybOutputRealDry, hybBands),
+ getScalefactor(pHybOutputImagDry, cplxBands));
+ headroom_wet = fMin(getScalefactor(pHybOutputRealWet, hybBands),
+ getScalefactor(pHybOutputImagWet, cplxBands));
+
dry = wet = FL2FXCONST_DBL(0.0f);
for (qs = 0; qs < cplxBands; qs++) {
- dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)) +
- fPow2Div2(pHybOutputImagDry[qs] << (1)));
- wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)) +
- fPow2Div2(pHybOutputImagWet[qs] << (1)));
+ /* sum up dry part */
+ dry += (fPow2Div2(pHybOutputRealDry[qs] << headroom_dry) >> s1);
+ dry += (fPow2Div2(pHybOutputImagDry[qs] << headroom_dry) >> s1);
+ /* sum up wet part */
+ wet += (fPow2Div2(pHybOutputRealWet[qs] << headroom_wet) >> s1);
+ wet += (fPow2Div2(pHybOutputImagWet[qs] << headroom_wet) >> s1);
}
for (; qs < hybBands; qs++) {
- dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)));
- wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)));
+ dry += (fPow2Div2(pHybOutputRealDry[qs] << headroom_dry) >> s1);
+ wet += (fPow2Div2(pHybOutputRealWet[qs] << headroom_wet) >> s1);
}
- *slotAmp_dry = dry >> (2 * (1));
- *slotAmp_wet = wet >> (2 * (1));
+
+ /* consider fPow2Div2() */
+ s1 += 1;
+
+ /* normalize dry part, ensure that exponent is even */
+ s2 = fixMax(0, CntLeadingZeros(dry) - 1);
+ *slotAmp_dry = dry << s2;
+ *slotAmp_dry_e = s1 - s2 - 2 * headroom_dry;
+ if (*slotAmp_dry_e & 1) {
+ *slotAmp_dry = *slotAmp_dry >> 1;
+ *slotAmp_dry_e += 1;
+ }
+
+ /* normalize wet part, ensure that exponent is even */
+ s2 = fixMax(0, CntLeadingZeros(wet) - 1);
+ *slotAmp_wet = wet << s2;
+ *slotAmp_wet_e = s1 - s2 - 2 * headroom_wet;
+ if (*slotAmp_wet_e & 1) {
+ *slotAmp_wet = *slotAmp_wet >> 1;
+ *slotAmp_wet_e += 1;
+ }
}
#if defined(__aarch64__)
@@ -533,6 +560,7 @@
INT ts) {
INT ch, scale;
INT dryFacSF, slotAmpSF;
+ INT slotAmp_dry_e, slotAmp_wet_e;
FIXP_DBL tmp, dryFac, envShape;
FIXP_DBL slotAmp_dry, slotAmp_wet, slotAmp_ratio;
FIXP_DBL envDry[MAX_OUTPUT_CHANNELS], envDmx[2];
@@ -594,22 +622,25 @@
dryFacSF = SF_SHAPE + 2 * dryFacSF;
}
+ slotAmp_dry_e = slotAmp_wet_e = 0;
+
/* calculate slotAmp_dry and slotAmp_wet */
- slotAmp(&slotAmp_dry, &slotAmp_wet, &self->hybOutputRealDry__FDK[ch][6],
+ slotAmp(&slotAmp_dry, &slotAmp_dry_e, &slotAmp_wet, &slotAmp_wet_e,
+ &self->hybOutputRealDry__FDK[ch][6],
&self->hybOutputImagDry__FDK[ch][6],
&self->hybOutputRealWet__FDK[ch][6],
&self->hybOutputImagWet__FDK[ch][6], cplxBands, hybBands);
+ /* exponents must be even due to subsequent square root calculation */
+ FDK_ASSERT(((slotAmp_dry_e & 1) == 0) && ((slotAmp_wet_e & 1) == 0));
+
/* slotAmp_ratio will be scaled by slotAmpSF bits */
if (slotAmp_dry != FL2FXCONST_DBL(0.0f)) {
- sc = fixMax(0, CntLeadingZeros(slotAmp_wet) - 1);
- sc = sc - (sc & 1);
-
- slotAmp_wet = sqrtFixp(slotAmp_wet << sc);
+ slotAmp_wet = sqrtFixp(slotAmp_wet);
slotAmp_dry = invSqrtNorm2(slotAmp_dry, &slotAmpSF);
slotAmp_ratio = fMult(slotAmp_wet, slotAmp_dry);
- slotAmpSF = slotAmpSF - (sc >> 1);
+ slotAmpSF = slotAmpSF + (slotAmp_wet_e >> 1) - (slotAmp_dry_e >> 1);
}
/* calculate common scale factor */
diff --git a/libSACdec/src/sac_stp.cpp b/libSACdec/src/sac_stp.cpp
index bb66277..0e6affa 100644
--- a/libSACdec/src/sac_stp.cpp
+++ b/libSACdec/src/sac_stp.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -229,15 +229,13 @@
int n;
FIXP_DBL scaleY;
for (n = bands - 1; n >= 0; n--) {
- scaleY = fMultDiv2(scaleX, *pBP);
+ scaleY = fMult(scaleX, *pBP);
*hybOutputRealDry = SATURATE_LEFT_SHIFT(
- (*hybOutputRealDry >> 1) +
- (fMultDiv2(*hybOutputRealWet, scaleY) << (SF_SCALE + 1)),
- 1, DFRACT_BITS);
+ (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleY),
+ SF_SCALE, DFRACT_BITS);
*hybOutputImagDry = SATURATE_LEFT_SHIFT(
- (*hybOutputImagDry >> 1) +
- (fMultDiv2(*hybOutputImagWet, scaleY) << (SF_SCALE + 1)),
- 1, DFRACT_BITS);
+ (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleY),
+ SF_SCALE, DFRACT_BITS);
hybOutputRealDry++, hybOutputRealWet++;
hybOutputImagDry++, hybOutputImagWet++;
pBP++;
@@ -252,12 +250,12 @@
int n;
for (n = bands - 1; n >= 0; n--) {
- *hybOutputRealDry =
- *hybOutputRealDry +
- (fMultDiv2(*hybOutputRealWet, scaleX) << (SF_SCALE + 1));
- *hybOutputImagDry =
- *hybOutputImagDry +
- (fMultDiv2(*hybOutputImagWet, scaleX) << (SF_SCALE + 1));
+ *hybOutputRealDry = SATURATE_LEFT_SHIFT(
+ (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleX),
+ SF_SCALE, DFRACT_BITS);
+ *hybOutputImagDry = SATURATE_LEFT_SHIFT(
+ (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleX),
+ SF_SCALE, DFRACT_BITS);
hybOutputRealDry++, hybOutputRealWet++;
hybOutputImagDry++, hybOutputImagWet++;
}
@@ -369,15 +367,15 @@
hStpDec->update_old_ener = 1;
for (ch = 0; ch < self->numInputChannels; ch++) {
hStpDec->oldDryEnerLD64[ch] =
- CalcLdData(hStpDec->runDryEner[ch] + ABS_THR__FDK);
+ CalcLdData(fAddSaturate(hStpDec->runDryEner[ch], ABS_THR__FDK));
}
for (ch = 0; ch < self->numOutputChannels; ch++) {
if (self->treeConfig == TREE_212)
hStpDec->oldWetEnerLD64[ch] =
- CalcLdData(hStpDec->runWetEner[ch] + ABS_THR__FDK);
+ CalcLdData(fAddSaturate(hStpDec->runWetEner[ch], ABS_THR__FDK));
else
hStpDec->oldWetEnerLD64[ch] =
- CalcLdData(hStpDec->runWetEner[ch] + ABS_THR2__FDK);
+ CalcLdData(fAddSaturate(hStpDec->runWetEner[ch], ABS_THR2__FDK));
}
} else {
hStpDec->update_old_ener++;
diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp
deleted file mode 100644
index db1948f..0000000
--- a/libSBRdec/src/arm/lpp_tran_arm.cpp
+++ /dev/null
@@ -1,159 +0,0 @@
-/* -----------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2018 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
------------------------------------------------------------------------------ */
-
-/**************************** SBR decoder library ******************************
-
- Author(s): Arthur Tritthart
-
- Description: (ARM optimised) LPP transposer subroutines
-
-*******************************************************************************/
-
-#if defined(__arm__)
-
-#define FUNCTION_LPPTRANSPOSER_func1
-
-#ifdef FUNCTION_LPPTRANSPOSER_func1
-
-/* Note: This code requires only 43 cycles per iteration instead of 61 on
- * ARM926EJ-S */
-static void lppTransposer_func1(FIXP_DBL *lowBandReal, FIXP_DBL *lowBandImag,
- FIXP_DBL **qmfBufferReal,
- FIXP_DBL **qmfBufferImag, int loops, int hiBand,
- int dynamicScale, int descale, FIXP_SGL a0r,
- FIXP_SGL a0i, FIXP_SGL a1r, FIXP_SGL a1i,
- const int fPreWhitening,
- FIXP_DBL preWhiteningGain,
- int preWhiteningGains_sf) {
- FIXP_DBL real1, real2, imag1, imag2, accu1, accu2;
-
- real2 = lowBandReal[-2];
- real1 = lowBandReal[-1];
- imag2 = lowBandImag[-2];
- imag1 = lowBandImag[-1];
- for (int i = 0; i < loops; i++) {
- accu1 = fMultDiv2(a0r, real1);
- accu2 = fMultDiv2(a0i, imag1);
- accu1 = fMultAddDiv2(accu1, a1r, real2);
- accu2 = fMultAddDiv2(accu2, a1i, imag2);
- real2 = fMultDiv2(a1i, real2);
- accu1 = accu1 - accu2;
- accu1 = accu1 >> dynamicScale;
-
- accu2 = fMultAddDiv2(real2, a1r, imag2);
- real2 = real1;
- imag2 = imag1;
- accu2 = fMultAddDiv2(accu2, a0i, real1);
- real1 = lowBandReal[i];
- accu2 = fMultAddDiv2(accu2, a0r, imag1);
- imag1 = lowBandImag[i];
- accu2 = accu2 >> dynamicScale;
-
- accu1 <<= 1;
- accu2 <<= 1;
- accu1 += (real1 >> descale);
- accu2 += (imag1 >> descale);
- if (fPreWhitening) {
- accu1 = scaleValueSaturate(fMultDiv2(accu1, preWhiteningGain),
- preWhiteningGains_sf);
- accu2 = scaleValueSaturate(fMultDiv2(accu2, preWhiteningGain),
- preWhiteningGains_sf);
- }
- qmfBufferReal[i][hiBand] = accu1;
- qmfBufferImag[i][hiBand] = accu2;
- }
-}
-#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */
-
-#endif /* __arm__ */
diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
index 0b2f651..cefa612 100644
--- a/libSBRdec/src/env_calc.cpp
+++ b/libSBRdec/src/env_calc.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -664,7 +664,7 @@
gain_sf[i] = mult_sf - total_power_low_sf + sf2;
gain[i] = sqrtFixp_lookup(gain[i], &gain_sf[i]);
if (gain_sf[i] < 0) {
- gain[i] >>= -gain_sf[i];
+ gain[i] >>= fMin(DFRACT_BITS - 1, -gain_sf[i]);
gain_sf[i] = 0;
}
} else {
@@ -683,11 +683,6 @@
/* gain[i] = g_inter[i] */
for (i = 0; i < nbSubsample; ++i) {
- if (gain_sf[i] < 0) {
- gain[i] >>= -gain_sf[i];
- gain_sf[i] = 0;
- }
-
/* calculate: gain[i] = 1.0f + gamma * (gain[i] - 1.0f); */
FIXP_DBL one = (FIXP_DBL)MAXVAL_DBL >>
gain_sf[i]; /* to substract this from gain[i] */
@@ -755,23 +750,15 @@
int gain_adj_sf = gain_adj_2_sf;
for (i = 0; i < nbSubsample; ++i) {
- gain[i] = fMult(gain[i], gain_adj);
- gain_sf[i] += gain_adj_sf;
-
- /* limit gain */
- if (gain_sf[i] > INTER_TES_SF_CHANGE) {
- gain[i] = (FIXP_DBL)MAXVAL_DBL;
- gain_sf[i] = INTER_TES_SF_CHANGE;
- }
- }
-
- for (i = 0; i < nbSubsample; ++i) {
- /* equalize gain[]'s scale factors */
- gain[i] >>= INTER_TES_SF_CHANGE - gain_sf[i];
+ int gain_e = fMax(
+ fMin(gain_sf[i] + gain_adj_sf - INTER_TES_SF_CHANGE, DFRACT_BITS - 1),
+ -(DFRACT_BITS - 1));
+ FIXP_DBL gain_final = fMult(gain[i], gain_adj);
+ gain_final = scaleValueSaturate(gain_final, gain_e);
for (j = lowSubband; j < highSubband; j++) {
- qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain[i]);
- qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain[i]);
+ qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain_final);
+ qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain_final);
}
}
} else { /* gamma_idx == 0 */
@@ -1398,6 +1385,17 @@
*/
noise_e = (start_pos < no_cols) ? adj_e : final_e;
+ if (start_pos >= no_cols) {
+ int diff = h_sbr_cal_env->filtBufferNoise_e - noise_e;
+ if (diff > 0) {
+ int s = getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands);
+ if (diff > s) {
+ final_e += diff - s;
+ noise_e = final_e;
+ }
+ }
+ }
+
/*
Convert energies to amplitude levels
*/
@@ -2741,6 +2739,9 @@
fMult(direct_ratio, noiseLevel[k]);
}
+ smoothedNoise = fMax(fMin(smoothedNoise, (FIXP_DBL)(MAXVAL_DBL / 2)),
+ (FIXP_DBL)(MINVAL_DBL / 2));
+
/*
The next 2 multiplications constitute the actual envelope adjustment
of the signal and should be carried out with full accuracy
@@ -2930,6 +2931,9 @@
fMult(direct_ratio, noiseLevel[k]);
}
+ smoothedNoise = fMax(fMin(smoothedNoise, (FIXP_DBL)(MAXVAL_DBL / 2)),
+ (FIXP_DBL)(MINVAL_DBL / 2));
+
/*
The next 2 multiplications constitute the actual envelope adjustment
of the signal and should be carried out with full accuracy
diff --git a/libSBRdec/src/hbe.cpp b/libSBRdec/src/hbe.cpp
index d210bb6..f2452ea 100644
--- a/libSBRdec/src/hbe.cpp
+++ b/libSBRdec/src/hbe.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1400,42 +1400,27 @@
if (shift_ov != 0) {
for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) {
- for (band = 0; band < QMF_SYNTH_CHANNELS; band++) {
- if (shift_ov >= 0) {
- hQmfTransposer->qmfHBEBufReal_F[i][band] <<= shift_ov;
- hQmfTransposer->qmfHBEBufImag_F[i][band] <<= shift_ov;
- } else {
- hQmfTransposer->qmfHBEBufReal_F[i][band] >>= (-shift_ov);
- hQmfTransposer->qmfHBEBufImag_F[i][band] >>= (-shift_ov);
- }
- }
- }
- }
-
- if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) && shift_ov != 0) {
- for (i = timeStep * firstSlotOffsset; i < ov_len; i++) {
- for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand;
- band++) {
- if (shift_ov >= 0) {
- ppQmfBufferOutReal_F[i][band] <<= shift_ov;
- ppQmfBufferOutImag_F[i][band] <<= shift_ov;
- } else {
- ppQmfBufferOutReal_F[i][band] >>= (-shift_ov);
- ppQmfBufferOutImag_F[i][band] >>= (-shift_ov);
- }
- }
+ scaleValuesSaturate(&hQmfTransposer->qmfHBEBufReal_F[i][0],
+ QMF_SYNTH_CHANNELS, shift_ov);
+ scaleValuesSaturate(&hQmfTransposer->qmfHBEBufImag_F[i][0],
+ QMF_SYNTH_CHANNELS, shift_ov);
}
- /* shift lpc filterstates */
- for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) {
- for (band = 0; band < (64); band++) {
- if (shift_ov >= 0) {
- lpcFilterStatesReal[i][band] <<= shift_ov;
- lpcFilterStatesImag[i][band] <<= shift_ov;
- } else {
- lpcFilterStatesReal[i][band] >>= (-shift_ov);
- lpcFilterStatesImag[i][band] >>= (-shift_ov);
- }
+ if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) {
+ int nBands =
+ fMax(0, hQmfTransposer->stopBand - hQmfTransposer->startBand);
+
+ for (i = timeStep * firstSlotOffsset; i < ov_len; i++) {
+ scaleValuesSaturate(&ppQmfBufferOutReal_F[i][hQmfTransposer->startBand],
+ nBands, shift_ov);
+ scaleValuesSaturate(&ppQmfBufferOutImag_F[i][hQmfTransposer->startBand],
+ nBands, shift_ov);
+ }
+
+ /* shift lpc filterstates */
+ for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) {
+ scaleValuesSaturate(&lpcFilterStatesReal[i][0], (64), shift_ov);
+ scaleValuesSaturate(&lpcFilterStatesImag[i][0], (64), shift_ov);
}
}
}
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
index 93e1158..68a25bf 100644
--- a/libSBRdec/src/lpp_tran.cpp
+++ b/libSBRdec/src/lpp_tran.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -132,10 +132,6 @@
#include "HFgen_preFlat.h"
-#if defined(__arm__)
-#include "arm/lpp_tran_arm.cpp"
-#endif
-
#define LPC_SCALE_FACTOR 2
/*!
@@ -220,19 +216,21 @@
const FIXP_DBL *const lowBandReal,
const int startSample,
const int stopSample, const UCHAR hiBand,
- const int dynamicScale, const int descale,
+ const int dynamicScale,
const FIXP_SGL a0r, const FIXP_SGL a1r) {
- FIXP_DBL accu1, accu2;
- int i;
+ const int dynscale = fixMax(0, dynamicScale - 1) + 1;
+ const int rescale = -fixMin(0, dynamicScale - 1) + 1;
+ const int descale =
+ fixMin(DFRACT_BITS - 1, LPC_SCALE_FACTOR + dynamicScale + rescale);
- for (i = 0; i < stopSample - startSample; i++) {
- accu1 = fMultDiv2(a1r, lowBandReal[i]);
- accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1);
- accu1 = accu1 >> dynamicScale;
+ for (int i = 0; i < stopSample - startSample; i++) {
+ FIXP_DBL accu;
- accu1 <<= 1;
- accu2 = (lowBandReal[i + 2] >> descale);
- qmfBufferReal[i + startSample][hiBand] = accu1 + accu2;
+ accu = fMultDiv2(a1r, lowBandReal[i]) + fMultDiv2(a0r, lowBandReal[i + 1]);
+ accu = (lowBandReal[i + 2] >> descale) + (accu >> dynscale);
+
+ qmfBufferReal[i + startSample][hiBand] =
+ SATURATE_LEFT_SHIFT(accu, rescale, DFRACT_BITS);
}
}
@@ -529,7 +527,7 @@
if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+ alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphar[1] = -alphar[1];
}
@@ -557,7 +555,7 @@
scale)) {
resetLPCCoeffs = 1;
} else {
- alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+ alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphai[1] = -alphai[1];
}
@@ -596,7 +594,7 @@
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+ alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphar[0] = -alphar[0];
@@ -616,7 +614,7 @@
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+ alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphai[0] = -alphai[0];
}
@@ -659,7 +657,7 @@
INT scale;
FIXP_DBL result =
fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale);
- k1 = scaleValue(result, scale);
+ k1 = scaleValueSaturate(result, scale);
if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) {
k1 = -k1;
@@ -771,52 +769,50 @@
} else { /* bw <= 0 */
if (!useLP) {
- int descale =
- fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale));
-#ifdef FUNCTION_LPPTRANSPOSER_func1
- lppTransposer_func1(
- lowBandReal + LPC_ORDER + startSample,
- lowBandImag + LPC_ORDER + startSample,
- qmfBufferReal + startSample, qmfBufferImag + startSample,
- stopSample - startSample, (int)hiBand, dynamicScale, descale, a0r,
- a0i, a1r, a1i, fPreWhitening, preWhiteningGains[loBand],
- preWhiteningGains_exp[loBand] + 1);
-#else
+ const int dynscale = fixMax(0, dynamicScale - 2) + 1;
+ const int rescale = -fixMin(0, dynamicScale - 2) + 1;
+ const int descale = fixMin(DFRACT_BITS - 1,
+ LPC_SCALE_FACTOR + dynamicScale + rescale);
+
for (i = startSample; i < stopSample; i++) {
FIXP_DBL accu1, accu2;
- accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) -
- fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) +
- fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) -
- fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >>
- dynamicScale;
- accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) +
- fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) +
- fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) +
- fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >>
- dynamicScale;
+ accu1 = ((fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) -
+ fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1])) >>
+ 1) +
+ ((fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) -
+ fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >>
+ 1);
+ accu2 = ((fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) +
+ fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1])) >>
+ 1) +
+ ((fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) +
+ fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >>
+ 1);
- accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);
- accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);
+ accu1 =
+ (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 >> dynscale);
+ accu2 =
+ (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 >> dynscale);
if (fPreWhitening) {
- accu1 = scaleValueSaturate(
+ qmfBufferReal[i][hiBand] = scaleValueSaturate(
fMultDiv2(accu1, preWhiteningGains[loBand]),
- preWhiteningGains_exp[loBand] + 1);
- accu2 = scaleValueSaturate(
+ preWhiteningGains_exp[loBand] + 1 + rescale);
+ qmfBufferImag[i][hiBand] = scaleValueSaturate(
fMultDiv2(accu2, preWhiteningGains[loBand]),
- preWhiteningGains_exp[loBand] + 1);
+ preWhiteningGains_exp[loBand] + 1 + rescale);
+ } else {
+ qmfBufferReal[i][hiBand] =
+ SATURATE_LEFT_SHIFT(accu1, rescale, DFRACT_BITS);
+ qmfBufferImag[i][hiBand] =
+ SATURATE_LEFT_SHIFT(accu2, rescale, DFRACT_BITS);
}
- qmfBufferReal[i][hiBand] = accu1;
- qmfBufferImag[i][hiBand] = accu2;
}
-#endif
} else {
FDK_ASSERT(dynamicScale >= 0);
calc_qmfBufferReal(
qmfBufferReal, &(lowBandReal[LPC_ORDER + startSample - 2]),
- startSample, stopSample, hiBand, dynamicScale,
- fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r,
- a1r);
+ startSample, stopSample, hiBand, dynamicScale, a0r, a1r);
}
} /* bw <= 0 */
@@ -1066,7 +1062,7 @@
if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+ alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphar[1] = -alphar[1];
}
@@ -1092,7 +1088,7 @@
(result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+ alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphai[1] = -alphai[1];
}
@@ -1121,7 +1117,7 @@
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+ alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphar[0] = -alphar[0];
@@ -1140,7 +1136,7 @@
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+ alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) {
alphai[0] = -alphai[0];
}
diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp
index b1fb0da..919e9bb 100644
--- a/libSBRdec/src/sbr_dec.cpp
+++ b/libSBRdec/src/sbr_dec.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -713,7 +713,8 @@
} else { /* (flags & SBRDEC_PS_DECODED) */
INT sdiff;
- INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
+ INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov,
+ outScalefactor, outScalefactorR, outScalefactorL;
HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->qmfDomainOutCh->fb;
HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->qmfDomainOutCh->fb;
@@ -744,7 +745,7 @@
*/
FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <=
QMF_MAX_SYNTHESIS_BANDS);
- qmfChangeOutScalefactor(synQmfRight, -(8));
+ synQmfRight->outScalefactor = synQmf->outScalefactor;
FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates,
9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis *
sizeof(FIXP_QSS));
@@ -788,9 +789,11 @@
FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel,
sizeof(SBRDEC_DRC_CHANNEL));
- for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
+ outScalefactor = maxShift - (8);
+ outScalefactorL = outScalefactorR =
+ sbrInDataHeadroom + 1; /* +1: psDiffScale! (MPEG-PS) */
- INT outScalefactorR, outScalefactorL;
+ for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
/* qmf timeslot of right channel */
FIXP_DBL *rQmfReal = pWorkBuffer;
@@ -815,27 +818,20 @@
? scaleFactorLowBand_ov
: scaleFactorLowBand_no_ov,
scaleFactorHighBand, synQmf->lsb, synQmf->usb);
-
- outScalefactorL = outScalefactorR =
- 1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */
}
sbrDecoder_drcApplySlot(/* right channel */
&hSbrDecRight->sbrDrcChannel, rQmfReal,
rQmfImag, i, synQmfRight->no_col, maxShift);
- outScalefactorR += maxShift;
-
sbrDecoder_drcApplySlot(/* left channel */
&hSbrDec->sbrDrcChannel, *(pLowBandReal + i),
*(pLowBandImag + i), i, synQmf->no_col,
maxShift);
- outScalefactorL += maxShift;
-
if (!(flags & SBRDEC_SKIP_QMF_SYN)) {
- qmfChangeOutScalefactor(synQmf, -(8));
- qmfChangeOutScalefactor(synQmfRight, -(8));
+ qmfChangeOutScalefactor(synQmf, outScalefactor);
+ qmfChangeOutScalefactor(synQmfRight, outScalefactor);
qmfSynthesisFilteringSlot(
synQmfRight, rQmfReal, /* QMF real buffer */
diff --git a/libSBRdec/src/sbrdec_drc.cpp b/libSBRdec/src/sbrdec_drc.cpp
index 2d73f32..089d046 100644
--- a/libSBRdec/src/sbrdec_drc.cpp
+++ b/libSBRdec/src/sbrdec_drc.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -233,14 +233,19 @@
if (hDrcData->winSequenceCurr != 2) { /* long window */
int j = col + (numQmfSubSamples >> 1);
- if (hDrcData->drcInterpolationSchemeCurr == 0) {
- INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
+ if (j < winBorderToColMap[15]) {
+ if (hDrcData->drcInterpolationSchemeCurr == 0) {
+ INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
- alphaValue = (FIXP_DBL)(j * k);
- } else {
- if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeCurr]) {
- alphaValue = (FIXP_DBL)MAXVAL_DBL;
+ alphaValue = (FIXP_DBL)(j * k);
+ } else {
+ if (j >=
+ (int)winBorderToColMap[hDrcData->drcInterpolationSchemeCurr]) {
+ alphaValue = (FIXP_DBL)MAXVAL_DBL;
+ }
}
+ } else {
+ alphaValue = (FIXP_DBL)MAXVAL_DBL;
}
} else { /* short windows */
shortDrc = 1;
@@ -254,14 +259,19 @@
if (hDrcData->winSequenceNext != 2) { /* next: long window */
int j = col - (numQmfSubSamples >> 1);
- if (hDrcData->drcInterpolationSchemeNext == 0) {
- INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
+ if (j < winBorderToColMap[15]) {
+ if (hDrcData->drcInterpolationSchemeNext == 0) {
+ INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
- alphaValue = (FIXP_DBL)(j * k);
- } else {
- if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) {
- alphaValue = (FIXP_DBL)MAXVAL_DBL;
+ alphaValue = (FIXP_DBL)(j * k);
+ } else {
+ if (j >=
+ (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) {
+ alphaValue = (FIXP_DBL)MAXVAL_DBL;
+ }
}
+ } else {
+ alphaValue = (FIXP_DBL)MAXVAL_DBL;
}
fact_mag = hDrcData->nextFact_mag;
@@ -289,14 +299,19 @@
if (hDrcData->winSequenceNext != 2) { /* long window */
int j = col - (numQmfSubSamples >> 1);
- if (hDrcData->drcInterpolationSchemeNext == 0) {
- INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
+ if (j < winBorderToColMap[15]) {
+ if (hDrcData->drcInterpolationSchemeNext == 0) {
+ INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
- alphaValue = (FIXP_DBL)(j * k);
- } else {
- if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) {
- alphaValue = (FIXP_DBL)MAXVAL_DBL;
+ alphaValue = (FIXP_DBL)(j * k);
+ } else {
+ if (j >=
+ (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) {
+ alphaValue = (FIXP_DBL)MAXVAL_DBL;
+ }
}
+ } else {
+ alphaValue = (FIXP_DBL)MAXVAL_DBL;
}
} else { /* short windows */
shortDrc = 1;
diff --git a/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp
index e187656..daa3554 100644
--- a/libSBRdec/src/sbrdec_freq_sca.cpp
+++ b/libSBRdec/src/sbrdec_freq_sca.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -765,9 +765,6 @@
sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1],
nBandsHi);
- hFreq->nSfb[0] = nBandsLo;
- hFreq->nSfb[1] = nBandsHi;
-
/* Check index to freqBandTable[0] */
if (!(nBandsLo > 0) ||
(nBandsLo > (((hHeaderData->numberOfAnalysisBands == 16)
@@ -777,6 +774,9 @@
return SBRDEC_UNSUPPORTED_CONFIG;
}
+ hFreq->nSfb[0] = nBandsLo;
+ hFreq->nSfb[1] = nBandsHi;
+
lsb = hFreq->freqBandTable[0][0];
usb = hFreq->freqBandTable[0][nBandsLo];
@@ -814,15 +814,15 @@
if (intTemp == 0) intTemp = 1;
+ if (intTemp > MAX_NOISE_COEFFS) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
hFreq->nNfb = intTemp;
}
hFreq->nInvfBands = hFreq->nNfb;
- if (hFreq->nNfb > MAX_NOISE_COEFFS) {
- return SBRDEC_UNSUPPORTED_CONFIG;
- }
-
/* Get noise bands */
sbrdecDownSampleLoRes(hFreq->freqBandTableNoise, hFreq->nNfb,
hFreq->freqBandTable[0], nBandsLo);
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index b101a4a..7718695 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -961,8 +961,10 @@
/* Set sync state UPSAMPLING for the corresponding slot.
This switches off bitstream parsing until a new header arrives. */
- hSbrHeader->syncState = UPSAMPLING;
- hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+ if (hSbrHeader->syncState != SBR_NOT_INITIALIZED) {
+ hSbrHeader->syncState = UPSAMPLING;
+ hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+ }
}
}
} break;
@@ -1371,7 +1373,9 @@
}
if (headerStatus == HEADER_ERROR) {
/* Corrupt SBR info data, do not decode and switch to UPSAMPLING */
- hSbrHeader->syncState = UPSAMPLING;
+ hSbrHeader->syncState = hSbrHeader->syncState > UPSAMPLING
+ ? UPSAMPLING
+ : hSbrHeader->syncState;
fDoDecodeSbrData = 0;
sbrHeaderPresent = 0;
}
@@ -1610,7 +1614,9 @@
/* No valid SBR payload available, hence switch to upsampling (in all
* headers) */
for (hdrIdx = 0; hdrIdx < ((1) + 1); hdrIdx += 1) {
- self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
+ if (self->sbrHeader[elementIndex][hdrIdx].syncState > UPSAMPLING) {
+ self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
+ }
}
} else {
/* Move frame pointer to the next slot which is up to be decoded/applied
diff --git a/libSBRenc/src/env_est.cpp b/libSBRenc/src/env_est.cpp
index 0eb8425..cc8780a 100644
--- a/libSBRenc/src/env_est.cpp
+++ b/libSBRenc/src/env_est.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1267,6 +1267,7 @@
sbrExtrEnv->pre_transient_info[1] = ed->transient_info[1]; /* tran_flag */
hEnvChan->encEnvData.noOfEnvelopes = ed->nEnvelopes =
ed->frame_info->nEnvelopes; /* number of envelopes of current frame */
+ hEnvChan->encEnvData.currentAmpResFF = (AMP_RES)h_con->initAmpResFF;
/*
Check if the current frame is divided into one envelope only. If so, set
@@ -1274,8 +1275,9 @@
*/
if ((hEnvChan->encEnvData.hSbrBSGrid->frameClass == FIXFIX) &&
(ed->nEnvelopes == 1)) {
+ AMP_RES currentAmpResFF = SBR_AMP_RES_1_5;
if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
- /* Note: global_tonaliy_float_value ==
+ /* Note: global_tonality_float_value ==
((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0)));
threshold_float_value ==
((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0)));
@@ -1289,14 +1291,13 @@
} else {
hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0;
}
- } else
- hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5;
+ currentAmpResFF = hEnvChan->encEnvData.currentAmpResFF;
+ }
- if (hEnvChan->encEnvData.currentAmpResFF !=
- hEnvChan->encEnvData.init_sbr_amp_res) {
+ if (currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) {
FDKsbrEnc_InitSbrHuffmanTables(
&hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope,
- &hEnvChan->sbrCodeNoiseFloor, hEnvChan->encEnvData.currentAmpResFF);
+ &hEnvChan->sbrCodeNoiseFloor, currentAmpResFF);
}
} else {
if (sbrHeaderData->sbr_amp_res != hEnvChan->encEnvData.init_sbr_amp_res) {
@@ -1355,6 +1356,13 @@
}
}
+ if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY &&
+ stereoMode == SBR_SWITCH_LRC &&
+ h_envChan[0]->encEnvData.currentAmpResFF !=
+ h_envChan[1]->encEnvData.currentAmpResFF) {
+ stereoMode = SBR_LEFT_RIGHT;
+ }
+
/*
Extract envelope of current frame.
*/
diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp
index c1e083f..c3da072 100644
--- a/libSBRenc/src/sbr_encoder.cpp
+++ b/libSBRenc/src/sbr_encoder.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1450,8 +1450,6 @@
params->deltaTAcrossFrames, 0, 0))
return (1);
- sbrConfigData->initAmpResFF = params->init_amp_res_FF;
-
if (FDKsbrEnc_InitSbrHuffmanTables(&hEnv->encEnvData, &hEnv->sbrCodeEnvelope,
&hEnv->sbrCodeNoiseFloor,
sbrHeaderData->sbr_amp_res))
@@ -1749,6 +1747,7 @@
hSbrElement->sbrHeaderData.sbr_data_extra = 1;
hSbrElement->sbrHeaderData.sbr_amp_res = (AMP_RES)params->amp_res;
+ hSbrElement->sbrConfigData.initAmpResFF = params->init_amp_res_FF;
/* header_extra_1 */
hSbrElement->sbrHeaderData.freqScale = params->freqScale;
