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-rw-r--r--libAACdec/src/usacdec_acelp.cpp4
-rw-r--r--libFDK/include/nlc_dec.h5
-rw-r--r--libFDK/src/autocorr2nd.cpp43
-rw-r--r--libFDK/src/nlc_dec.cpp28
-rw-r--r--libMpegTPDec/src/tpdec_asc.cpp10
-rw-r--r--libPCMutils/src/limiter.cpp5
-rw-r--r--libPCMutils/src/pcmdmx_lib.cpp56
-rw-r--r--libSACdec/src/sac_bitdec.cpp17
-rw-r--r--libSACdec/src/sac_stp.cpp15
-rw-r--r--libSBRdec/src/arm/lpp_tran_arm.cpp159
-rw-r--r--libSBRdec/src/lpp_tran.cpp94
-rw-r--r--libSBRdec/src/sbr_dec.cpp24
12 files changed, 162 insertions, 298 deletions
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 @@ static void ConcealPitchLag(CAcelpStaticMem *acelp_mem, const int PIT_MAX,
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/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 @@ typedef enum {
#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/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 @@ amm-info@iis.fraunhofer.de
#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 @@ INT autoCorr2nd_real(
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 @@ INT autoCorr2nd_cplx(
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 @@ INT autoCorr2nd_cplx(
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/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 @@ bail:
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 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
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 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
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 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
case HUFF_2D:
- switch (*cdg_scheme & PAIR_MASK) {
+ switch (*pairing_scheme) {
case FREQ_PAIR:
if (out_data_1 != NULL) {
@@ -843,7 +841,7 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
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 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
}
/* 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 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
}
}
- 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/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp
index e46cb32..8f77017 100644
--- a/libMpegTPDec/src/tpdec_asc.cpp
+++ b/libMpegTPDec/src/tpdec_asc.cpp
@@ -1694,8 +1694,7 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
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) {
@@ -1712,7 +1711,6 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
}
}
- extElement->usacExtElementType = usacExtElementType;
int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16);
extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength;
INT bsAnchor;
@@ -1746,8 +1744,10 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
}
} 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. */
@@ -1776,7 +1776,7 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc,
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. */
@@ -1787,7 +1787,7 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc,
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 */
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 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
(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 @@ static PCM_DMX_CHANNEL_MODE getChMode4Plain(
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 @@ static PCMDMX_ERROR getChannelMode(
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 @@ static PCMDMX_ERROR getChannelMode(
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 @@ static PCMDMX_ERROR getChannelMode(
(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/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 @@ SACDEC_ERROR SpatialDecParseSpecificConfig(
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_stp.cpp b/libSACdec/src/sac_stp.cpp
index b328c82..be332c7 100644
--- a/libSACdec/src/sac_stp.cpp
+++ b/libSACdec/src/sac_stp.cpp
@@ -252,12 +252,15 @@ inline void combineSignalCplxScale2(FIXP_DBL *hybOutputRealDry,
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 >> 1) +
+ (fMultDiv2(*hybOutputRealWet, scaleX) << SF_SCALE),
+ 1, DFRACT_BITS);
+ *hybOutputImagDry = SATURATE_LEFT_SHIFT(
+ (*hybOutputImagDry >> 1) +
+ (fMultDiv2(*hybOutputImagWet, scaleX) << SF_SCALE),
+ 1, DFRACT_BITS);
+ ;
hybOutputRealDry++, hybOutputRealWet++;
hybOutputImagDry++, hybOutputImagWet++;
}
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/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
index 93e1158..113b1de 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -132,10 +132,6 @@ amm-info@iis.fraunhofer.de
#include "HFgen_preFlat.h"
-#if defined(__arm__)
-#include "arm/lpp_tran_arm.cpp"
-#endif
-
#define LPC_SCALE_FACTOR 2
/*!
@@ -220,19 +216,21 @@ static inline void calc_qmfBufferReal(FIXP_DBL **qmfBufferReal,
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 (int i = 0; i < stopSample - startSample; i++) {
+ FIXP_DBL accu;
- for (i = 0; i < stopSample - startSample; i++) {
- accu1 = fMultDiv2(a1r, lowBandReal[i]);
- accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1);
- accu1 = accu1 >> dynamicScale;
+ accu = fMultDiv2(a1r, lowBandReal[i]) + fMultDiv2(a0r, lowBandReal[i + 1]);
+ accu = (lowBandReal[i + 2] >> descale) + (accu >> dynscale);
- accu1 <<= 1;
- accu2 = (lowBandReal[i + 2] >> descale);
- qmfBufferReal[i + startSample][hiBand] = accu1 + accu2;
+ qmfBufferReal[i + startSample][hiBand] =
+ SATURATE_LEFT_SHIFT(accu, rescale, DFRACT_BITS);
}
}
@@ -771,52 +769,50 @@ void lppTransposer(
} 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 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);
- accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);
+ 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 >> 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 */
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 @@ void sbr_dec(
} 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 @@ void sbr_dec(
*/
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 @@ void sbr_dec(
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 @@ void sbr_dec(
? 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 */
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-28 01:59:50 +0000