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-rw-r--r--libSBRdec/src/arm/lpp_tran_arm.cpp159
-rw-r--r--libSBRdec/src/env_calc.cpp48
-rw-r--r--libSBRdec/src/hbe.cpp51
-rw-r--r--libSBRdec/src/lpp_tran.cpp112
-rw-r--r--libSBRdec/src/lpp_tran.h4
-rw-r--r--libSBRdec/src/sbr_dec.cpp24
-rw-r--r--libSBRdec/src/sbrdec_drc.cpp53
-rw-r--r--libSBRdec/src/sbrdec_freq_sca.cpp16
-rw-r--r--libSBRdec/src/sbrdecoder.cpp16
9 files changed, 163 insertions, 320 deletions
diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp
deleted file mode 100644
index db1948f..0000000
--- a/libSBRdec/src/arm/lpp_tran_arm.cpp
+++ /dev/null
@@ -1,159 +0,0 @@
-/* -----------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
-Forschung e.V. All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
-that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
-scheme for digital audio. This FDK AAC Codec software is intended to be used on
-a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
-general perceptual audio codecs. AAC-ELD is considered the best-performing
-full-bandwidth communications codec by independent studies and is widely
-deployed. AAC has been standardized by ISO and IEC as part of the MPEG
-specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including
-those of Fraunhofer) may be obtained through Via Licensing
-(www.vialicensing.com) or through the respective patent owners individually for
-the purpose of encoding or decoding bit streams in products that are compliant
-with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
-Android devices already license these patent claims through Via Licensing or
-directly from the patent owners, and therefore FDK AAC Codec software may
-already be covered under those patent licenses when it is used for those
-licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions
-with enhanced sound quality, are also available from Fraunhofer. Users are
-encouraged to check the Fraunhofer website for additional applications
-information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification,
-are permitted without payment of copyright license fees provided that you
-satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of
-the FDK AAC Codec or your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation
-and/or other materials provided with redistributions of the FDK AAC Codec or
-your modifications thereto in binary form. You must make available free of
-charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived
-from this library without prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute
-the FDK AAC Codec software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating
-that you changed the software and the date of any change. For modified versions
-of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
-must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
-AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
-limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
-Fraunhofer provides no warranty of patent non-infringement with respect to this
-software.
-
-You may use this FDK AAC Codec software or modifications thereto only for
-purposes that are authorized by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
-holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
-including but not limited to the implied warranties of merchantability and
-fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
-or consequential damages, including but not limited to procurement of substitute
-goods or services; loss of use, data, or profits, or business interruption,
-however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of
-this software, even if advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------ */
-
-/**************************** SBR decoder library ******************************
-
- Author(s): Arthur Tritthart
-
- Description: (ARM optimised) LPP transposer subroutines
-
-*******************************************************************************/
-
-#if defined(__arm__)
-
-#define FUNCTION_LPPTRANSPOSER_func1
-
-#ifdef FUNCTION_LPPTRANSPOSER_func1
-
-/* Note: This code requires only 43 cycles per iteration instead of 61 on
- * ARM926EJ-S */
-static void lppTransposer_func1(FIXP_DBL *lowBandReal, FIXP_DBL *lowBandImag,
- FIXP_DBL **qmfBufferReal,
- FIXP_DBL **qmfBufferImag, int loops, int hiBand,
- int dynamicScale, int descale, FIXP_SGL a0r,
- FIXP_SGL a0i, FIXP_SGL a1r, FIXP_SGL a1i,
- const int fPreWhitening,
- FIXP_DBL preWhiteningGain,
- int preWhiteningGains_sf) {
- FIXP_DBL real1, real2, imag1, imag2, accu1, accu2;
-
- real2 = lowBandReal[-2];
- real1 = lowBandReal[-1];
- imag2 = lowBandImag[-2];
- imag1 = lowBandImag[-1];
- for (int i = 0; i < loops; i++) {
- accu1 = fMultDiv2(a0r, real1);
- accu2 = fMultDiv2(a0i, imag1);
- accu1 = fMultAddDiv2(accu1, a1r, real2);
- accu2 = fMultAddDiv2(accu2, a1i, imag2);
- real2 = fMultDiv2(a1i, real2);
- accu1 = accu1 - accu2;
- accu1 = accu1 >> dynamicScale;
-
- accu2 = fMultAddDiv2(real2, a1r, imag2);
- real2 = real1;
- imag2 = imag1;
- accu2 = fMultAddDiv2(accu2, a0i, real1);
- real1 = lowBandReal[i];
- accu2 = fMultAddDiv2(accu2, a0r, imag1);
- imag1 = lowBandImag[i];
- accu2 = accu2 >> dynamicScale;
-
- accu1 <<= 1;
- accu2 <<= 1;
- accu1 += (real1 >> descale);
- accu2 += (imag1 >> descale);
- if (fPreWhitening) {
- accu1 = scaleValueSaturate(fMultDiv2(accu1, preWhiteningGain),
- preWhiteningGains_sf);
- accu2 = scaleValueSaturate(fMultDiv2(accu2, preWhiteningGain),
- preWhiteningGains_sf);
- }
- qmfBufferReal[i][hiBand] = accu1;
- qmfBufferImag[i][hiBand] = accu2;
- }
-}
-#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */
-
-#endif /* __arm__ */
diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
index 0b2f651..cefa612 100644
--- a/libSBRdec/src/env_calc.cpp
+++ b/libSBRdec/src/env_calc.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -664,7 +664,7 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag,
gain_sf[i] = mult_sf - total_power_low_sf + sf2;
gain[i] = sqrtFixp_lookup(gain[i], &gain_sf[i]);
if (gain_sf[i] < 0) {
- gain[i] >>= -gain_sf[i];
+ gain[i] >>= fMin(DFRACT_BITS - 1, -gain_sf[i]);
gain_sf[i] = 0;
}
} else {
@@ -683,11 +683,6 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag,
/* gain[i] = g_inter[i] */
for (i = 0; i < nbSubsample; ++i) {
- if (gain_sf[i] < 0) {
- gain[i] >>= -gain_sf[i];
- gain_sf[i] = 0;
- }
-
/* calculate: gain[i] = 1.0f + gamma * (gain[i] - 1.0f); */
FIXP_DBL one = (FIXP_DBL)MAXVAL_DBL >>
gain_sf[i]; /* to substract this from gain[i] */
@@ -755,23 +750,15 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag,
int gain_adj_sf = gain_adj_2_sf;
for (i = 0; i < nbSubsample; ++i) {
- gain[i] = fMult(gain[i], gain_adj);
- gain_sf[i] += gain_adj_sf;
-
- /* limit gain */
- if (gain_sf[i] > INTER_TES_SF_CHANGE) {
- gain[i] = (FIXP_DBL)MAXVAL_DBL;
- gain_sf[i] = INTER_TES_SF_CHANGE;
- }
- }
-
- for (i = 0; i < nbSubsample; ++i) {
- /* equalize gain[]'s scale factors */
- gain[i] >>= INTER_TES_SF_CHANGE - gain_sf[i];
+ int gain_e = fMax(
+ fMin(gain_sf[i] + gain_adj_sf - INTER_TES_SF_CHANGE, DFRACT_BITS - 1),
+ -(DFRACT_BITS - 1));
+ FIXP_DBL gain_final = fMult(gain[i], gain_adj);
+ gain_final = scaleValueSaturate(gain_final, gain_e);
for (j = lowSubband; j < highSubband; j++) {
- qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain[i]);
- qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain[i]);
+ qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain_final);
+ qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain_final);
}
}
} else { /* gamma_idx == 0 */
@@ -1398,6 +1385,17 @@ void calculateSbrEnvelope(
*/
noise_e = (start_pos < no_cols) ? adj_e : final_e;
+ if (start_pos >= no_cols) {
+ int diff = h_sbr_cal_env->filtBufferNoise_e - noise_e;
+ if (diff > 0) {
+ int s = getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands);
+ if (diff > s) {
+ final_e += diff - s;
+ noise_e = final_e;
+ }
+ }
+ }
+
/*
Convert energies to amplitude levels
*/
@@ -2741,6 +2739,9 @@ static void adjustTimeSlotHQ_GainAndNoise(
fMult(direct_ratio, noiseLevel[k]);
}
+ smoothedNoise = fMax(fMin(smoothedNoise, (FIXP_DBL)(MAXVAL_DBL / 2)),
+ (FIXP_DBL)(MINVAL_DBL / 2));
+
/*
The next 2 multiplications constitute the actual envelope adjustment
of the signal and should be carried out with full accuracy
@@ -2930,6 +2931,9 @@ static void adjustTimeSlotHQ(
fMult(direct_ratio, noiseLevel[k]);
}
+ smoothedNoise = fMax(fMin(smoothedNoise, (FIXP_DBL)(MAXVAL_DBL / 2)),
+ (FIXP_DBL)(MINVAL_DBL / 2));
+
/*
The next 2 multiplications constitute the actual envelope adjustment
of the signal and should be carried out with full accuracy
diff --git a/libSBRdec/src/hbe.cpp b/libSBRdec/src/hbe.cpp
index 77cb8af..9485823 100644
--- a/libSBRdec/src/hbe.cpp
+++ b/libSBRdec/src/hbe.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1400,42 +1400,27 @@ void QmfTransposerApply(HANDLE_HBE_TRANSPOSER hQmfTransposer,
if (shift_ov != 0) {
for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) {
- for (band = 0; band < QMF_SYNTH_CHANNELS; band++) {
- if (shift_ov >= 0) {
- hQmfTransposer->qmfHBEBufReal_F[i][band] <<= shift_ov;
- hQmfTransposer->qmfHBEBufImag_F[i][band] <<= shift_ov;
- } else {
- hQmfTransposer->qmfHBEBufReal_F[i][band] >>= (-shift_ov);
- hQmfTransposer->qmfHBEBufImag_F[i][band] >>= (-shift_ov);
- }
- }
+ scaleValuesSaturate(&hQmfTransposer->qmfHBEBufReal_F[i][0],
+ QMF_SYNTH_CHANNELS, shift_ov);
+ scaleValuesSaturate(&hQmfTransposer->qmfHBEBufImag_F[i][0],
+ QMF_SYNTH_CHANNELS, shift_ov);
}
- }
- if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) && shift_ov != 0) {
- for (i = timeStep * firstSlotOffsset; i < ov_len; i++) {
- for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand;
- band++) {
- if (shift_ov >= 0) {
- ppQmfBufferOutReal_F[i][band] <<= shift_ov;
- ppQmfBufferOutImag_F[i][band] <<= shift_ov;
- } else {
- ppQmfBufferOutReal_F[i][band] >>= (-shift_ov);
- ppQmfBufferOutImag_F[i][band] >>= (-shift_ov);
- }
+ if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) {
+ int nBands =
+ fMax(0, hQmfTransposer->stopBand - hQmfTransposer->startBand);
+
+ for (i = timeStep * firstSlotOffsset; i < ov_len; i++) {
+ scaleValuesSaturate(&ppQmfBufferOutReal_F[i][hQmfTransposer->startBand],
+ nBands, shift_ov);
+ scaleValuesSaturate(&ppQmfBufferOutImag_F[i][hQmfTransposer->startBand],
+ nBands, shift_ov);
}
- }
- /* shift lpc filterstates */
- for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) {
- for (band = 0; band < (64); band++) {
- if (shift_ov >= 0) {
- lpcFilterStatesReal[i][band] <<= shift_ov;
- lpcFilterStatesImag[i][band] <<= shift_ov;
- } else {
- lpcFilterStatesReal[i][band] >>= (-shift_ov);
- lpcFilterStatesImag[i][band] >>= (-shift_ov);
- }
+ /* shift lpc filterstates */
+ for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) {
+ scaleValuesSaturate(&lpcFilterStatesReal[i][0], (64), shift_ov);
+ scaleValuesSaturate(&lpcFilterStatesImag[i][0], (64), shift_ov);
}
}
}
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
index 93e1158..68a25bf 100644
--- a/libSBRdec/src/lpp_tran.cpp
+++ b/libSBRdec/src/lpp_tran.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -132,10 +132,6 @@ 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);
}
}
@@ -529,7 +527,7 @@ void lppTransposer(
if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+ alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphar[1] = -alphar[1];
}
@@ -557,7 +555,7 @@ void lppTransposer(
scale)) {
resetLPCCoeffs = 1;
} else {
- alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+ alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphai[1] = -alphai[1];
}
@@ -596,7 +594,7 @@ void lppTransposer(
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+ alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphar[0] = -alphar[0];
@@ -616,7 +614,7 @@ void lppTransposer(
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+ alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphai[0] = -alphai[0];
}
@@ -659,7 +657,7 @@ void lppTransposer(
INT scale;
FIXP_DBL result =
fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale);
- k1 = scaleValue(result, scale);
+ k1 = scaleValueSaturate(result, scale);
if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) {
k1 = -k1;
@@ -771,52 +769,50 @@ 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 */
@@ -1066,7 +1062,7 @@ void lppTransposerHBE(
if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+ alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphar[1] = -alphar[1];
}
@@ -1092,7 +1088,7 @@ void lppTransposerHBE(
(result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+ alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphai[1] = -alphai[1];
}
@@ -1121,7 +1117,7 @@ void lppTransposerHBE(
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+ alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphar[0] = -alphar[0];
@@ -1140,7 +1136,7 @@ void lppTransposerHBE(
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+ alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) {
alphai[0] = -alphai[0];
}
diff --git a/libSBRdec/src/lpp_tran.h b/libSBRdec/src/lpp_tran.h
index 51b4395..21c4101 100644
--- a/libSBRdec/src/lpp_tran.h
+++ b/libSBRdec/src/lpp_tran.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 - 2023 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -207,7 +207,7 @@ typedef struct {
inverse filtering levels */
PATCH_PARAM
- patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */
+ patchParam[MAX_NUM_PATCHES + 1]; /*!< new parameter set for patching */
WHITENING_FACTORS
whFactors; /*!< the pole moving factors for certain
whitening levels as indicated in the bitstream
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 */
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 @@ void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData,
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 @@ void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData,
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 @@ void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData,
if (hDrcData->winSequenceNext != 2) { /* long window */
int j = col - (numQmfSubSamples >> 1);
- if (hDrcData->drcInterpolationSchemeNext == 0) {
- INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
+ if (j < winBorderToColMap[15]) {
+ if (hDrcData->drcInterpolationSchemeNext == 0) {
+ INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
- alphaValue = (FIXP_DBL)(j * k);
- } else {
- if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) {
- alphaValue = (FIXP_DBL)MAXVAL_DBL;
+ alphaValue = (FIXP_DBL)(j * k);
+ } else {
+ if (j >=
+ (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) {
+ alphaValue = (FIXP_DBL)MAXVAL_DBL;
+ }
}
+ } else {
+ alphaValue = (FIXP_DBL)MAXVAL_DBL;
}
} else { /* short windows */
shortDrc = 1;
diff --git a/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp
index e187656..daa3554 100644
--- a/libSBRdec/src/sbrdec_freq_sca.cpp
+++ b/libSBRdec/src/sbrdec_freq_sca.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -765,9 +765,6 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) {
sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1],
nBandsHi);
- hFreq->nSfb[0] = nBandsLo;
- hFreq->nSfb[1] = nBandsHi;
-
/* Check index to freqBandTable[0] */
if (!(nBandsLo > 0) ||
(nBandsLo > (((hHeaderData->numberOfAnalysisBands == 16)
@@ -777,6 +774,9 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) {
return SBRDEC_UNSUPPORTED_CONFIG;
}
+ hFreq->nSfb[0] = nBandsLo;
+ hFreq->nSfb[1] = nBandsHi;
+
lsb = hFreq->freqBandTable[0][0];
usb = hFreq->freqBandTable[0][nBandsLo];
@@ -814,15 +814,15 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) {
if (intTemp == 0) intTemp = 1;
+ if (intTemp > MAX_NOISE_COEFFS) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
hFreq->nNfb = intTemp;
}
hFreq->nInvfBands = hFreq->nNfb;
- if (hFreq->nNfb > MAX_NOISE_COEFFS) {
- return SBRDEC_UNSUPPORTED_CONFIG;
- }
-
/* Get noise bands */
sbrdecDownSampleLoRes(hFreq->freqBandTableNoise, hFreq->nNfb,
hFreq->freqBandTable[0], nBandsLo);
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index b101a4a..7718695 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -961,8 +961,10 @@ SBR_ERROR sbrDecoder_SetParam(HANDLE_SBRDECODER self, const SBRDEC_PARAM param,
/* Set sync state UPSAMPLING for the corresponding slot.
This switches off bitstream parsing until a new header arrives. */
- hSbrHeader->syncState = UPSAMPLING;
- hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+ if (hSbrHeader->syncState != SBR_NOT_INITIALIZED) {
+ hSbrHeader->syncState = UPSAMPLING;
+ hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+ }
}
}
} break;
@@ -1371,7 +1373,9 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
}
if (headerStatus == HEADER_ERROR) {
/* Corrupt SBR info data, do not decode and switch to UPSAMPLING */
- hSbrHeader->syncState = UPSAMPLING;
+ hSbrHeader->syncState = hSbrHeader->syncState > UPSAMPLING
+ ? UPSAMPLING
+ : hSbrHeader->syncState;
fDoDecodeSbrData = 0;
sbrHeaderPresent = 0;
}
@@ -1610,7 +1614,9 @@ static SBR_ERROR sbrDecoder_DecodeElement(
/* No valid SBR payload available, hence switch to upsampling (in all
* headers) */
for (hdrIdx = 0; hdrIdx < ((1) + 1); hdrIdx += 1) {
- self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
+ if (self->sbrHeader[elementIndex][hdrIdx].syncState > UPSAMPLING) {
+ self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
+ }
}
} else {
/* Move frame pointer to the next slot which is up to be decoded/applied
generated by cgit v1.2.3 (git 2.46.0) at 2025-11-04 09:20:28 +0000