Snap for 7277594 from edbf93589b1786557383451f2d6d34fd85d8e73d to mainline-documentsui-release
Change-Id: I0b92789091acd39674317db5b61c2fc442c1a83f
diff --git a/Android.bp b/Android.bp
index 1cf54fd..3f42c19 100644
--- a/Android.bp
+++ b/Android.bp
@@ -23,6 +23,7 @@
"-Wuninitialized",
"-Wno-self-assign",
"-Wno-implicit-fallthrough",
+ "-DSUPPRESS_BUILD_DATE_INFO",
],
sanitize: {
misc_undefined:[
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..c18e5e9 100644
--- a/libAACdec/src/aacdecoder.cpp
+++ b/libAACdec/src/aacdecoder.cpp
@@ -568,7 +568,7 @@
\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 +584,7 @@
}
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 +606,7 @@
\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 +622,15 @@
}
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 +753,12 @@
/* 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 +1402,31 @@
}
/*!
+ * \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(*elFlags) * (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1));
+ }
+
+ self->flags[streamIndex] = flags;
+}
+
+/*!
* \brief CAacDecoder_CtrlCFGChange Set config change parameters.
*
* \param self [i] handle to AACDECODER structure
@@ -1493,6 +1523,9 @@
const int streamIndex = 0;
INT flushChannels = 0;
+ UINT flags;
+ UINT elFlags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)];
+
if (!self) return AAC_DEC_INVALID_HANDLE;
UCHAR downscaleFactor = self->downscaleFactor;
@@ -1649,8 +1682,8 @@
}
/* 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)) {
@@ -1700,31 +1733,30 @@
}
}
- 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 +1823,17 @@
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 {
@@ -1838,8 +1878,8 @@
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;
@@ -1865,51 +1905,47 @@
}
/* --------- 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;
+ 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;
}
}
- 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)) {
@@ -1960,6 +1996,10 @@
if (ascChanged != 0) {
*configChanged = 1;
}
+
+ CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex,
+ elementOffset);
+
return err;
}
@@ -1988,7 +2028,7 @@
}
if (usacStereoConfigIndex == 3) {
- self->flags[streamIndex] |= AC_USAC_SCFGI3;
+ flags |= AC_USAC_SCFGI3;
}
}
break;
@@ -2069,14 +2109,14 @@
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 +2216,14 @@
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 +2234,7 @@
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 +2246,7 @@
}
}
- if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA)) {
+ if (flags & (AC_USAC | AC_RSV603DA)) {
int complexStereoPredPossible = 0;
ch = aacChannelsOffset;
chIdx = aacChannelsOffsetIdx;
@@ -2223,7 +2262,7 @@
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 +2399,6 @@
}
}
- /* Update externally visible copy of flags */
- self->streamInfo.flags = self->flags[0];
-
if (*configChanged) {
int drcDecSampleRate, drcDecFrameSize;
@@ -2383,8 +2419,7 @@
if (*configChanged) {
if (asc->m_aot == AOT_USAC) {
- self->hDrcInfo->enable = 0;
- self->hDrcInfo->progRefLevelPresent = 0;
+ aacDecoder_drcDisable(self->hDrcInfo);
}
}
@@ -2393,6 +2428,12 @@
pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f));
}
+ CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex,
+ elementOffset);
+
+ /* Update externally visible copy of flags */
+ self->streamInfo.flags = self->flags[0];
+
return err;
bail:
@@ -3194,11 +3235,12 @@
* 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 +3483,12 @@
* 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 f5ce7e0..9d36d10 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -122,7 +122,7 @@
#define AACDECODER_LIB_VL1 2
#define AACDECODER_LIB_VL2 0
#define AACDECODER_LIB_TITLE "AAC Decoder Lib"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
#define AACDECODER_LIB_BUILD_DATE ""
#define AACDECODER_LIB_BUILD_TIME ""
#else
@@ -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);
+ }
}
}
@@ -1794,8 +1813,7 @@
}
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 +1844,38 @@
* 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 +1917,116 @@
((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/libAACenc/src/aacenc.cpp b/libAACenc/src/aacenc.cpp
index b6f733d..1af8a2e 100644
--- a/libAACenc/src/aacenc.cpp
+++ b/libAACenc/src/aacenc.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
@@ -245,6 +245,46 @@
return bitrate;
}
+/*-----------------------------------------------------------------------------
+
+ functionname: FDKaacEnc_AdjustVBRBitrateMode
+ description: Adjust bitrate mode to given bitrate parameter
+ input params: int vbrQuality (VBR0, VBR1, VBR2)
+ bitrate
+ channelMode
+ returns: vbr bitrate mode
+
+ ------------------------------------------------------------------------------*/
+AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode(
+ AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode) {
+ AACENC_BITRATE_MODE newBitrateMode = bitrateMode;
+
+ if (bitrate != -1) {
+ const INT monoStereoMode =
+ (FDKaacEnc_GetMonoStereoMode(channelMode) == EL_MODE_STEREO) ? 1 : 0;
+ const INT nChannelsEff =
+ FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff;
+ newBitrateMode = AACENC_BR_MODE_INVALID;
+
+ for (int idx = (int)(sizeof(configTabVBR) / sizeof(*configTabVBR)) - 1;
+ idx >= 0; idx--) {
+ if (bitrate >=
+ configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff) {
+ if (configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff <
+ FDKaacEnc_GetVBRBitrate(bitrateMode, channelMode)) {
+ newBitrateMode = configTabVBR[idx].bitrateMode;
+ } else {
+ newBitrateMode = bitrateMode;
+ }
+ break;
+ }
+ }
+ }
+
+ return AACENC_BR_MODE_IS_VBR(newBitrateMode) ? newBitrateMode
+ : AACENC_BR_MODE_INVALID;
+}
+
/**
* \brief Convert encoder bitreservoir value for transport library.
*
@@ -397,7 +437,6 @@
FIXP_DBL mbfac, bw_ratio;
QC_INIT qcInit;
INT averageBitsPerFrame = 0;
- int bitresMin = 0; /* the bitreservoir is always big for AAC-LC */
const CHANNEL_MODE prevChannelMode = hAacEnc->encoderMode;
if (config == NULL) return AAC_ENC_INVALID_HANDLE;
@@ -553,7 +592,6 @@
qcInit.minBits = fixMin(qcInit.minBits, averageBitsPerFrame & ~7);
} else {
INT bitreservoir = -1; /* default bitreservoir size*/
- bitresMin = BITRES_MIN;
if (isLowDelay(config->audioObjectType)) {
INT brPerChannel = config->bitRate / config->nChannels;
brPerChannel = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel));
@@ -567,7 +605,6 @@
bitreservoir = fMultI(slope, (INT)(BITRES_MAX_LD - BITRES_MIN_LD)) +
BITRES_MIN_LD; /* interpolate */
bitreservoir = bitreservoir & ~7; /* align to bytes */
- bitresMin = BITRES_MIN_LD;
}
int maxBitres;
@@ -604,7 +641,8 @@
qcInit.nSubFrames = config->nSubFrames;
qcInit.padding.paddingRest = config->sampleRate;
- if (qcInit.maxBits - qcInit.averageBits >= bitresMin * config->nChannels) {
+ if (qcInit.maxBits - qcInit.averageBits >=
+ ((qcInit.isLowDelay) ? BITRES_MIN_LD : BITRES_MIN) * config->nChannels) {
qcInit.bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */
} else if (qcInit.maxBits > qcInit.averageBits) {
qcInit.bitResMode = AACENC_BR_MODE_REDUCED; /* reduced bitreservoir */
diff --git a/libAACenc/src/aacenc.h b/libAACenc/src/aacenc.h
index 0e0d8c1..b7e0ef2 100644
--- a/libAACenc/src/aacenc.h
+++ b/libAACenc/src/aacenc.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
@@ -335,6 +335,19 @@
/*-----------------------------------------------------------------------------
+ functionname: FDKaacEnc_AdjustVBRBitrateMode
+ description: Adjust bitrate mode to given bitrate parameter
+ input params: int vbrQuality (VBR0, VBR1, VBR2)
+ bitrate
+ channelMode
+ returns: vbr bitrate mode
+
+ ------------------------------------------------------------------------------*/
+AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode(
+ AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode);
+
+/*-----------------------------------------------------------------------------
+
functionname: FDKaacEnc_AacInitDefaultConfig
description: gives reasonable default configuration
returns: ---
diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp
index 2c2010f..caa62c5 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -112,7 +112,7 @@
#define AACENCODER_LIB_VL1 0
#define AACENCODER_LIB_VL2 1
#define AACENCODER_LIB_TITLE "AAC Encoder"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
#define AACENCODER_LIB_BUILD_DATE ""
#define AACENCODER_LIB_BUILD_TIME ""
#else
@@ -1028,6 +1028,13 @@
case AACENC_BR_MODE_VBR_3:
case AACENC_BR_MODE_VBR_4:
case AACENC_BR_MODE_VBR_5:
+ /* Adjust bitrate mode in case given peak bitrate is lower than expected
+ * VBR bitrate. */
+ if ((INT)config->userPeakBitrate != -1) {
+ hAacConfig->bitrateMode = FDKaacEnc_AdjustVBRBitrateMode(
+ hAacConfig->bitrateMode, config->userPeakBitrate,
+ hAacConfig->channelMode);
+ }
/* Get bitrate in VBR configuration */
/* In VBR mode; SBR-modul depends on bitrate, core encoder on bitrateMode.
*/
@@ -1235,7 +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;
@@ -1777,8 +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
@@ -1820,12 +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;
diff --git a/libAACenc/src/qc_main.cpp b/libAACenc/src/qc_main.cpp
index bcfaa23..9a42550 100644
--- a/libAACenc/src/qc_main.cpp
+++ b/libAACenc/src/qc_main.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
@@ -373,13 +373,8 @@
hQC->invQuant = init->invQuant;
hQC->maxIterations = init->maxIterations;
- if (isConstantBitrateMode(hQC->bitrateMode)) {
- /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir
- */
- hQC->bitResMode = init->bitResMode;
- } else {
- hQC->bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */
- }
+ /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */
+ hQC->bitResMode = init->bitResMode;
hQC->padding.paddingRest = init->padding.paddingRest;
@@ -800,10 +795,15 @@
INT avgTotalDynBits = 0; /* maximal allowed dynamic bits for all frames */
INT totalAvailableBits = 0;
INT nSubFrames = 1;
+ const INT isCBRAdjustment = (isConstantBitrateMode(hQC->bitrateMode) ||
+ (hQC->bitResMode != AACENC_BR_MODE_FULL))
+ ? 1
+ : 0;
/*-------------------------------------------- */
/* redistribute total bitreservoir to elements */
- ErrorStatus = FDKaacEnc_BitResRedistribution(hQC, cm, avgTotalBits);
+ ErrorStatus = FDKaacEnc_BitResRedistribution(
+ hQC, cm, (isCBRAdjustment == 0) ? hQC->maxBitsPerFrame : avgTotalBits);
if (ErrorStatus != AAC_ENC_OK) {
return ErrorStatus;
}
@@ -831,33 +831,22 @@
/*-------------------------------------------- */
/*-------------------------------------------- */
- if (isConstantBitrateMode(hQC->bitrateMode)) {
- /* calc granted dynamic bits for sub frame and
- distribute it to each element */
- ErrorStatus = FDKaacEnc_prepareBitDistribution(
- hQC, psyOut, qcOut, cm, qcElement, avgTotalBits, &totalAvailableBits,
- &avgTotalDynBits);
+ /* calc granted dynamic bits for sub frame and
+ distribute it to each element */
+ ErrorStatus = FDKaacEnc_prepareBitDistribution(
+ hQC, psyOut, qcOut, cm, qcElement,
+ (isCBRAdjustment == 0) ? hQC->maxBitsPerFrame : avgTotalBits,
+ &totalAvailableBits, &avgTotalDynBits);
- if (ErrorStatus != AAC_ENC_OK) {
- return ErrorStatus;
- }
- } else {
- qcOut[0]->grantedDynBits =
- ((hQC->maxBitsPerFrame - (hQC->globHdrBits)) & ~7) -
- (qcOut[0]->globalExtBits + qcOut[0]->staticBits +
- qcOut[0]->elementExtBits);
- qcOut[0]->maxDynBits = qcOut[0]->grantedDynBits;
-
- totalAvailableBits = hQC->maxBitsPerFrame;
- avgTotalDynBits = 0;
+ if (ErrorStatus != AAC_ENC_OK) {
+ return ErrorStatus;
}
/* for ( all sub frames ) ... */
for (c = 0; c < nSubFrames; c++) {
/* for CBR and VBR mode */
FDKaacEnc_AdjustThresholds(hQC->hAdjThr, qcElement[c], qcOut[c],
- psyOut[c]->psyOutElement,
- isConstantBitrateMode(hQC->bitrateMode), cm);
+ psyOut[c]->psyOutElement, isCBRAdjustment, cm);
} /* -end- sub frame counter */
diff --git a/libDRCdec/src/FDK_drcDecLib.cpp b/libDRCdec/src/FDK_drcDecLib.cpp
index 83b5773..26e5b78 100644
--- a/libDRCdec/src/FDK_drcDecLib.cpp
+++ b/libDRCdec/src/FDK_drcDecLib.cpp
@@ -112,7 +112,7 @@
#define DRCDEC_LIB_VL1 1
#define DRCDEC_LIB_VL2 0
#define DRCDEC_LIB_TITLE "MPEG-D DRC Decoder Lib"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
#define DRCDEC_LIB_BUILD_DATE ""
#define DRCDEC_LIB_BUILD_TIME ""
#else
diff --git a/libDRCdec/src/drcDec_reader.cpp b/libDRCdec/src/drcDec_reader.cpp
index 367a352..b3ec187 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) {
diff --git a/libFDK/src/FDK_core.cpp b/libFDK/src/FDK_core.cpp
index 2f77179..48db17e 100644
--- a/libFDK/src/FDK_core.cpp
+++ b/libFDK/src/FDK_core.cpp
@@ -107,7 +107,7 @@
#define FDK_TOOLS_LIB_VL1 1
#define FDK_TOOLS_LIB_VL2 0
#define FDK_TOOLS_LIB_TITLE "FDK Tools"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
#define FDK_TOOLS_LIB_BUILD_DATE ""
#define FDK_TOOLS_LIB_BUILD_TIME ""
#else
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/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/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..e46cb32 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) {
@@ -1633,6 +1778,10 @@
int numConfigExtensions;
CONFIG_EXT_ID 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++) {
@@ -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_lib.cpp b/libMpegTPDec/src/tpdec_lib.cpp
index ca35184..091d011 100644
--- a/libMpegTPDec/src/tpdec_lib.cpp
+++ b/libMpegTPDec/src/tpdec_lib.cpp
@@ -1769,7 +1769,7 @@
info += i;
info->module_id = FDK_TPDEC;
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
info->build_date = "";
info->build_time = "";
#else
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/libMpegTPEnc/src/tpenc_lib.cpp b/libMpegTPEnc/src/tpenc_lib.cpp
index 14ea5fe..77c19b5 100644
--- a/libMpegTPEnc/src/tpenc_lib.cpp
+++ b/libMpegTPEnc/src/tpenc_lib.cpp
@@ -647,7 +647,7 @@
info->module_id = FDK_TPENC;
info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2);
LIB_VERSION_STRING(info);
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
info->build_date = "";
info->build_time = "";
#else
diff --git a/libPCMutils/src/version.h b/libPCMutils/src/version.h
index 05371f8..871aa90 100644
--- a/libPCMutils/src/version.h
+++ b/libPCMutils/src/version.h
@@ -108,7 +108,7 @@
#define PCMUTIL_LIB_VL1 1
#define PCMUTIL_LIB_VL2 0
#define PCMUTIL_LIB_TITLE "PCM Utility Lib"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
#define PCMUTIL_LIB_BUILD_DATE ""
#define PCMUTIL_LIB_BUILD_TIME ""
#else
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_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 856a923..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:
@@ -1804,7 +1819,7 @@
info += i;
info->module_id = FDK_MPSDEC;
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
info->build_date = "";
info->build_time = "";
#else
diff --git a/libSACdec/src/sac_stp.cpp b/libSACdec/src/sac_stp.cpp
index bb66277..b328c82 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -369,15 +369,15 @@
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/libSACenc/src/sacenc_lib.cpp b/libSACenc/src/sacenc_lib.cpp
index d6a1658..fcfe39b 100644
--- a/libSACenc/src/sacenc_lib.cpp
+++ b/libSACenc/src/sacenc_lib.cpp
@@ -130,7 +130,7 @@
#define SACENC_LIB_VL1 0
#define SACENC_LIB_VL2 0
#define SACENC_LIB_TITLE "MPEG Surround Encoder"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
#define SACENC_LIB_BUILD_DATE ""
#define SACENC_LIB_BUILD_TIME ""
#else
diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
index 0b2f651..ad5edfe 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1398,6 +1398,17 @@
*/
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 +2752,9 @@
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 +2944,9 @@
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/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/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index 55f929f..b101a4a 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -158,7 +158,7 @@
#define SBRDECODER_LIB_VL1 1
#define SBRDECODER_LIB_VL2 0
#define SBRDECODER_LIB_TITLE "SBR Decoder"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
#define SBRDECODER_LIB_BUILD_DATE ""
#define SBRDECODER_LIB_BUILD_TIME ""
#else
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 df9e996..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;
@@ -2560,7 +2559,7 @@
info->version =
LIB_VERSION(SBRENCODER_LIB_VL0, SBRENCODER_LIB_VL1, SBRENCODER_LIB_VL2);
LIB_VERSION_STRING(info);
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
info->build_date = "";
info->build_time = "";
#else
