From 128abf6b1ace9c790ba7f42d976d3035cbe347b6 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 9 Aug 2019 17:06:37 +0200 Subject: Validate aacSampleRate after applying ELD downscale factor. Avoid division by zero. Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I3f7ece9451121d40fab2c97571c695c1ac62bd00 --- libAACdec/src/aacdecoder.cpp | 3 +++ 1 file changed, 3 insertions(+) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index 7617937..38ef4f8 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -1928,6 +1928,9 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, self->samplingRateInfo[0].samplingRate / self->downscaleFactor; self->streamInfo.aacSamplesPerFrame = asc->m_samplesPerFrame / self->downscaleFactor; + if (self->streamInfo.aacSampleRate <= 0) { + return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; + } } } -- cgit v1.2.3 From f26eb8af7c38047ad113072a6de38e069dc5cf3f Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 9 Aug 2019 17:06:59 +0200 Subject: Fix CpePersistentData memory leak. Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Iee5c93d61088bbda2328099ad20413adae3fa09b --- libAACdec/src/aacdecoder.cpp | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index 38ef4f8..b15fc80 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -1303,7 +1303,8 @@ static void CAacDecoder_DeInit(HANDLE_AACDECODER self, const int subStreamIndex) { int ch; int aacChannelOffset = 0, aacChannels = (8); - int numElements = (((8)) + (8)), elementOffset = 0; + int numElements = (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1), + elementOffset = 0; if (self == NULL) return; -- cgit v1.2.3 From 15965e3feb9025e1db65a7c9a7408b3970430cb8 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 9 Aug 2019 17:10:07 +0200 Subject: Limit too large shift exponent in CJointStereo_ApplyMS() Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Ia6ab654654e899368ba207a66eddf22b6b855635 --- libAACdec/src/stereo.cpp | 30 ++++++++++++++---------------- 1 file changed, 14 insertions(+), 16 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/stereo.cpp b/libAACdec/src/stereo.cpp index eed826b..a90ae35 100644 --- a/libAACdec/src/stereo.cpp +++ b/libAACdec/src/stereo.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -807,19 +807,17 @@ void CJointStereo_ApplyMS( for (int i = 0; i < windowLen; i++) { dmx_re_prev[i] = ((staticSpectralCoeffsL[index_offset + i] >> - srLeftChan) + + fMin(DFRACT_BITS - 1, srLeftChan + 1)) + (staticSpectralCoeffsR[index_offset + i] >> - srRightChan)) >> - 1; + fMin(DFRACT_BITS - 1, srRightChan + 1))); } } else { for (int i = 0; i < windowLen; i++) { dmx_re_prev[i] = ((staticSpectralCoeffsL[index_offset + i] >> - srLeftChan) - + fMin(DFRACT_BITS - 1, srLeftChan + 1)) - (staticSpectralCoeffsR[index_offset + i] >> - srRightChan)) >> - 1; + fMin(DFRACT_BITS - 1, srRightChan + 1))); } } } @@ -854,12 +852,13 @@ void CJointStereo_ApplyMS( if (window == 0) { if (dmx_re_prev_e < frameMaxScale) { if (mainband_flag == 0) { - scaleValues(dmx_re_prev, store_dmx_re_prev, windowLen, - -(frameMaxScale - dmx_re_prev_e)); + scaleValues( + dmx_re_prev, store_dmx_re_prev, windowLen, + -fMin(DFRACT_BITS - 1, (frameMaxScale - dmx_re_prev_e))); } else { - for (int i = 0; i < windowLen; i++) { - dmx_re_prev[i] >>= (frameMaxScale - dmx_re_prev_e); - } + scaleValues( + dmx_re_prev, windowLen, + -fMin(DFRACT_BITS - 1, (frameMaxScale - dmx_re_prev_e))); } } else { if (mainband_flag == 0) { @@ -873,10 +872,9 @@ void CJointStereo_ApplyMS( FDK_ASSERT(pAacDecoderChannelInfo[L]->icsInfo.WindowSequence == BLOCK_SHORT); if (specScaleL[window - 1] < frameMaxScale) { - for (int i = 0; i < windowLen; i++) { - dmx_re[windowLen * (window - 1) + i] >>= - (frameMaxScale - specScaleL[window - 1]); - } + scaleValues(&dmx_re[windowLen * (window - 1)], windowLen, + -fMin(DFRACT_BITS - 1, + (frameMaxScale - specScaleL[window - 1]))); } else { specScaleL[window] = specScaleL[window - 1]; specScaleR[window] = specScaleR[window - 1]; -- cgit v1.2.3 From da5b0dcb3c01197b3499e0b569babcb0f5a09bbe Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 9 Aug 2019 17:10:29 +0200 Subject: Restrict size of huffman escape sequence to prevent shift with too large exponent in DecodeEscapeSequence(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Ibcd182e313e9ef480e92619023bf424762b92e23 --- libAACdec/src/aacdec_hcr.cpp | 28 +++++++++++++++++++--------- 1 file changed, 19 insertions(+), 9 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdec_hcr.cpp b/libAACdec/src/aacdec_hcr.cpp index 6114756..a7e9cce 100644 --- a/libAACdec/src/aacdec_hcr.cpp +++ b/libAACdec/src/aacdec_hcr.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -137,7 +137,7 @@ static void DeriveNumberOfExtendedSortedSectionsInSets( static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, INT quantSpecCoef, INT *pLeftStartOfSegment, SCHAR *pRemainingBitsInSegment, - int *pNumDecodedBits); + int *pNumDecodedBits, UINT *errorWord); static int DecodePCW_Sign(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, UINT codebookDim, const SCHAR *pQuantVal, @@ -1179,8 +1179,8 @@ static void DecodePCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) { bs, pHcr->decInOut.bitstreamAnchor, pQuantizedSpectralCoefficients [quantizedSpectralCoefficientsIdx], - pLeftStartOfSegment, pRemainingBitsInSegment, - &numDecodedBits); + pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits, + &pHcr->decInOut.errorLog); } quantizedSpectralCoefficientsIdx++; if (quantizedSpectralCoefficientsIdx >= 1024) { @@ -1195,8 +1195,8 @@ static void DecodePCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) { bs, pHcr->decInOut.bitstreamAnchor, pQuantizedSpectralCoefficients [quantizedSpectralCoefficientsIdx], - pLeftStartOfSegment, pRemainingBitsInSegment, - &numDecodedBits); + pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits, + &pHcr->decInOut.errorLog); } quantizedSpectralCoefficientsIdx++; if (quantizedSpectralCoefficientsIdx >= 1024) { @@ -1386,7 +1386,7 @@ value == 16, a escapeSequence is decoded in two steps: static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, INT quantSpecCoef, INT *pLeftStartOfSegment, SCHAR *pRemainingBitsInSegment, - int *pNumDecodedBits) { + int *pNumDecodedBits, UINT *errorWord) { UINT i; INT sign; UINT escapeOnesCounter = 0; @@ -1400,6 +1400,9 @@ static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, FROM_LEFT_TO_RIGHT); *pRemainingBitsInSegment -= 1; *pNumDecodedBits += 1; + if (*pRemainingBitsInSegment < 0) { + return Q_VALUE_INVALID; + } if (carryBit != 0) { escapeOnesCounter += 1; @@ -1416,6 +1419,9 @@ static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, FROM_LEFT_TO_RIGHT); *pRemainingBitsInSegment -= 1; *pNumDecodedBits += 1; + if (*pRemainingBitsInSegment < 0) { + return Q_VALUE_INVALID; + } escape_word <<= 1; escape_word = escape_word | carryBit; @@ -1423,8 +1429,12 @@ static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, sign = (quantSpecCoef >= 0) ? 1 : -1; - quantSpecCoef = sign * (((INT)1 << escapeOnesCounter) + escape_word); - + if (escapeOnesCounter < 13) { + quantSpecCoef = sign * (((INT)1 << escapeOnesCounter) + escape_word); + } else { + *errorWord |= TOO_MANY_PCW_BODY_SIGN_ESC_BITS_DECODED; + quantSpecCoef = Q_VALUE_INVALID; + } return quantSpecCoef; } -- cgit v1.2.3 From 40d2a1d8b055ebe17b67195029029ecdc5bc3268 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 20 Sep 2019 13:58:23 +0200 Subject: Avoid integer overflows with pseudoLR in CAacDecoder_DecodeFrame(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I5c83d72b5c0f4cd1569b648f102c8c549a7a6ac2 --- libAACdec/src/aacdecoder.cpp | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index b15fc80..6b5a86c 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -3388,7 +3388,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( * LR) */ if ((aacChannels == 2) && bsPseudoLr) { int i, offset2; - const FIXP_SGL invSqrt2 = FL2FXCONST_SGL(0.707106781186547f); + const FIXP_SGL invSqrt2 = + FL2FXCONST_SGL(0.353553390593273f); /* scaled by -1 */ FIXP_PCM *pTD = pTimeData; offset2 = timeDataChannelOffset; @@ -3399,11 +3400,14 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( L = fMult(L, invSqrt2); R = fMult(R, invSqrt2); #if (SAMPLE_BITS == 16) - pTD[0] = FX_DBL2FX_PCM(fAddSaturate(L + R, (FIXP_DBL)0x8000)); - pTD[offset2] = FX_DBL2FX_PCM(fAddSaturate(L - R, (FIXP_DBL)0x8000)); + pTD[0] = (FIXP_SGL)SATURATE_RIGHT_SHIFT(L + R + (FIXP_DBL)(1 << 14), + 15, FRACT_BITS); + pTD[offset2] = (FIXP_SGL)SATURATE_RIGHT_SHIFT( + L - R + (FIXP_DBL)(1 << 14), 15, FRACT_BITS); #else - pTD[0] = FX_DBL2FX_PCM(L + R); - pTD[offset2] = FX_DBL2FX_PCM(L - R); + pTD[0] = SATURATE_LEFT_SHIFT(FX_DBL2FX_PCM(L + R), 1, DFRACT_BITS); + pTD[offset2] = + SATURATE_LEFT_SHIFT(FX_DBL2FX_PCM(L - R), 1, DFRACT_BITS); #endif pTD++; } -- cgit v1.2.3 From 1b4a3bedbb9d512c929c2fa191cc44c8dae8bce9 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 20 Sep 2019 14:00:27 +0200 Subject: Overcome potential integer overflows in M/S module for complex prediction. Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Ic0648203b6a61ebff8bcc93c4b1099e033bc6860 --- libAACdec/src/stereo.cpp | 40 +++++++++++++++------------------------- 1 file changed, 15 insertions(+), 25 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/stereo.cpp b/libAACdec/src/stereo.cpp index a90ae35..47f1a31 100644 --- a/libAACdec/src/stereo.cpp +++ b/libAACdec/src/stereo.cpp @@ -989,7 +989,7 @@ void CJointStereo_ApplyMS( } /* if ( pJointStereoData->complex_coef == 1 ) */ /* 4. upmix process */ - INT pred_dir = cplxPredictionData->pred_dir ? -1 : 1; + LONG pred_dir = cplxPredictionData->pred_dir ? -1 : 1; /* 0.1 in Q-3.34 */ const FIXP_DBL pointOne = 0x66666666; /* 0.8 */ /* Shift value for the downmix */ @@ -1039,34 +1039,24 @@ void CJointStereo_ApplyMS( the downmix. "dmx_re" and "specL" are two different pointers pointing to separate arrays, which may or may not contain the same data (with different scaling). - */ - - /* help1: alpha_re[i] * dmx_re[i] */ - FIXP_DBL help1 = fMultDiv2(alpha_re_tmp, *p2dmxRe++); - - /* tmp: dmx_im[i] */ - FIXP_DBL tmp = (*p2dmxIm++) << shift_dmx; - /* help2: alpha_im[i] * dmx_im[i] */ - FIXP_DBL help2 = fMultDiv2(alpha_im_tmp, tmp); - - /* help3: alpha_re[i] * dmx_re[i] + alpha_im[i] * dmx_im[i] */ - FIXP_DBL help3 = help1 + help2; + specL[i] = + (specL[i] + side); + specR[i] = -/+ (specL[i] - side); + */ + FIXP_DBL side, left, right; - /* side (= help4) = specR[i] - (dmx_re[i] * specL[i] + alpha_im[i] - * * dmx_im[i]) */ - FIXP_DBL help4 = *p2CoeffR - scaleValue(help3, help3_shift); + side = fMultAddDiv2(fMultDiv2(alpha_re_tmp, *p2dmxRe++), + alpha_im_tmp, (*p2dmxIm++) << shift_dmx); + side = ((*p2CoeffR) >> 2) - + (FIXP_DBL)SATURATE_SHIFT(side, -(help3_shift - 2), + DFRACT_BITS - 2); - /* We calculate the left and right output by using the helper - * function */ - /* specR[i] = -/+ (specL[i] - side); */ - *p2CoeffR = - (FIXP_DBL)((LONG)(*p2CoeffL - help4) * (LONG)pred_dir); - p2CoeffR++; + left = ((*p2CoeffL) >> 2) + side; + right = ((*p2CoeffL) >> 2) - side; + right = (FIXP_DBL)((LONG)right * pred_dir); - /* specL[i] = specL[i] + side; */ - *p2CoeffL = *p2CoeffL + help4; - p2CoeffL++; + *p2CoeffL++ = SATURATE_LEFT_SHIFT_ALT(left, 2, DFRACT_BITS); + *p2CoeffR++ = SATURATE_LEFT_SHIFT_ALT(right, 2, DFRACT_BITS); } } -- cgit v1.2.3 From 4c839c6a90d800d60f66f848f437ccdd800953a9 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 20 Sep 2019 14:01:01 +0200 Subject: Fix integer overflow in E_UTIL_preemph(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Iaaa0630e59d8e83e58b25168a3db04304485429b --- libAACdec/src/usacdec_acelp.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp index c836c6a..1ac8c9f 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -131,7 +131,7 @@ void E_UTIL_preemph(const FIXP_DBL *in, FIXP_DBL *out, INT L) { int i; for (i = 0; i < L; i++) { - out[i] = in[i] - fMult(PREEMPH_FAC, in[i - 1]); + out[i] = fAddSaturate(in[i], -fMult(PREEMPH_FAC, in[i - 1])); } return; -- cgit v1.2.3 From 2d292e9b2d116934f09e37c6553927582c9ee4fc Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 20 Sep 2019 14:01:32 +0200 Subject: Use SATURATE_LEFT_SHIFT in get_lsppol() to prevent integer overflow. Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Ie9ec52485a52b076a900b111b035289110c004f8 --- libAACdec/src/usacdec_lpc.cpp | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_lpc.cpp b/libAACdec/src/usacdec_lpc.cpp index 271463f..440fa87 100644 --- a/libAACdec/src/usacdec_lpc.cpp +++ b/libAACdec/src/usacdec_lpc.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1138,9 +1138,12 @@ static void get_lsppol(FIXP_LPC lsp[], FIXP_DBL f[], int n, int flag) { for (i = 2; i <= n; i++) { plsp += 2; b = -FX_LPC2FX_DBL(*plsp); - f[i] = ((fMultDiv2(b, f[i - 1]) << 1) + (f[i - 2])) << 1; + f[i] = SATURATE_LEFT_SHIFT((fMultDiv2(b, f[i - 1]) + (f[i - 2] >> 1)), 2, + DFRACT_BITS); for (j = i - 1; j > 1; j--) { - f[j] = f[j] + (fMultDiv2(b, f[j - 1]) << 2) + f[j - 2]; + f[j] = SATURATE_LEFT_SHIFT( + ((f[j] >> 2) + fMultDiv2(b, f[j - 1]) + (f[j - 2] >> 2)), 2, + DFRACT_BITS); } f[1] = f[1] + (b >> (SF_F - 1)); } -- cgit v1.2.3 From 4db9f39f6d1a4f89e3891d29345201a04eef43f0 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 20 Sep 2019 14:02:19 +0200 Subject: Avoid signed integer overflow in E_LPC_f_lsp_a_conversion(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: If2110ad246bb5b57ef76c9cd251874ecd4b05109 --- libAACdec/src/usacdec_lpc.cpp | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_lpc.cpp b/libAACdec/src/usacdec_lpc.cpp index 440fa87..278afbe 100644 --- a/libAACdec/src/usacdec_lpc.cpp +++ b/libAACdec/src/usacdec_lpc.cpp @@ -1170,6 +1170,9 @@ void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) { /*-----------------------------------------------------* * Multiply F1(z) by (1+z^-1) and F2(z) by (1-z^-1) * *-----------------------------------------------------*/ + scaleValues(f1, NC + 1, -2); + scaleValues(f2, NC + 1, -2); + for (i = NC; i > 0; i--) { f1[i] += f1[i - 1]; f2[i] -= f2[i - 1]; @@ -1178,13 +1181,8 @@ void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) { FIXP_DBL aDBL[M_LP_FILTER_ORDER]; for (i = 1, k = M_LP_FILTER_ORDER - 1; i <= NC; i++, k--) { - FIXP_DBL tmp1, tmp2; - - tmp1 = f1[i] >> 1; - tmp2 = f2[i] >> 1; - - aDBL[i - 1] = (tmp1 + tmp2); - aDBL[k] = (tmp1 - tmp2); + aDBL[i - 1] = f1[i] + f2[i]; + aDBL[k] = f1[i] - f2[i]; } int headroom_a = getScalefactor(aDBL, M_LP_FILTER_ORDER); @@ -1193,5 +1191,5 @@ void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) { a[i] = FX_DBL2FX_LPC(aDBL[i] << headroom_a); } - *a_exp = 8 - headroom_a; + *a_exp = SF_F + (2 - 1) - headroom_a; } -- cgit v1.2.3 From 3c377e33055267a007566a550820ae1cdb135331 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 20 Sep 2019 14:02:46 +0200 Subject: Prevent signed integer overflow in RE8_PPV(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Ic66ac4742d8e466431c5cf09d02b0d9c7f842df2 --- libAACdec/src/usacdec_lpc.cpp | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_lpc.cpp b/libAACdec/src/usacdec_lpc.cpp index 278afbe..d726798 100644 --- a/libAACdec/src/usacdec_lpc.cpp +++ b/libAACdec/src/usacdec_lpc.cpp @@ -231,7 +231,7 @@ void nearest_neighbor_2D8(FIXP_ZF x[8], int y[8]) { void RE8_PPV(FIXP_ZF x[], SHORT y[], int r) { int i, y0[8], y1[8]; FIXP_ZF x1[8], tmp; - FIXP_DBL e; + INT64 e; /* find the nearest neighbor y0 of x in 2D8 */ nearest_neighbor_2D8(x, y0); @@ -245,16 +245,16 @@ void RE8_PPV(FIXP_ZF x[], SHORT y[], int r) { } /* compute e0=||x-y0||^2 and e1=||x-y1||^2 */ - e = (FIXP_DBL)0; + e = 0; for (i = 0; i < 8; i++) { tmp = x[i] - INT2ZF(y0[i], 0); - e += fPow2Div2( + e += (INT64)fPow2Div2( tmp << r); /* shift left to ensure that no fract part bits get lost. */ tmp = x[i] - INT2ZF(y1[i], 0); - e -= fPow2Div2(tmp << r); + e -= (INT64)fPow2Div2(tmp << r); } /* select best candidate y0 or y1 to minimize distortion */ - if (e < (FIXP_DBL)0) { + if (e < 0) { for (i = 0; i < 8; i++) { y[i] = y0[i]; } -- cgit v1.2.3 From 2152ae1a5ef0511e0b38a981745eb910f2af9bca Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 20 Sep 2019 14:03:40 +0200 Subject: Suppress integer overflow in lsf_weight_2st() by using 64 bit multiplication with explicit 32 bit integer result. Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I82ce4858688e90abee0c44c4ada34a9a7b08342f --- libAACdec/src/usacdec_lpc.cpp | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_lpc.cpp b/libAACdec/src/usacdec_lpc.cpp index d726798..88601b7 100644 --- a/libAACdec/src/usacdec_lpc.cpp +++ b/libAACdec/src/usacdec_lpc.cpp @@ -565,7 +565,8 @@ static void lsf_weight_2st(FIXP_LPC *lsfq, FIXP_DBL *xq, int nk_mode) { /* add non-weighted residual LSF vector to LSF1st */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { w = (LONG)fMultDiv2(factor, sqrtFixp(fMult(d[i], d[i + 1]))); - lsfq[i] = fAddSaturate(lsfq[i], FX_DBL2FX_LPC((FIXP_DBL)(w * (LONG)xq[i]))); + lsfq[i] = fAddSaturate(lsfq[i], + FX_DBL2FX_LPC((FIXP_DBL)((INT64)w * (LONG)xq[i]))); } return; -- cgit v1.2.3 From ea9d3a049b7b3bb1a9523fe8cf79375140856359 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 18 Oct 2019 14:03:21 +0200 Subject: Prevent signed integer overflow in filtLP(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I8da32f4794274e2955936ccd42c009485fbe1972 --- libAACdec/src/usacdec_lpd.cpp | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_lpd.cpp b/libAACdec/src/usacdec_lpd.cpp index e0a2631..fa68ccb 100644 --- a/libAACdec/src/usacdec_lpd.cpp +++ b/libAACdec/src/usacdec_lpd.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -130,9 +130,10 @@ void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise, for (i = 0; i < stop; i++) { tmp = fMultDiv2(noise[i], filt[0]); // Filt in Q-1.16 for (j = 1; j <= len; j++) { - tmp += fMultDiv2((noise[i - j] + noise[i + j]), filt[j]); + tmp += fMult((noise[i - j] >> 1) + (noise[i + j] >> 1), filt[j]); } - syn_out[i] = (FIXP_PCM)(IMDCT_SCALE(syn[i] - tmp)); + syn_out[i] = (FIXP_PCM)(SATURATE_SHIFT( + (syn[i] >> 1) - (tmp >> 1), (MDCT_OUTPUT_SCALE - 1), PCM_OUT_BITS)); } } -- cgit v1.2.3 From f22eb9c1fa5bc594e6468a00566c11fc00654394 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 18 Oct 2019 14:03:39 +0200 Subject: Avoid signed integer overflow in bass_pf_1sf_delay(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I67bfab71987b31601a4666c8cf292f71bcb6799a --- libAACdec/src/usacdec_lpd.cpp | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_lpd.cpp b/libAACdec/src/usacdec_lpd.cpp index fa68ccb..5c3426d 100644 --- a/libAACdec/src/usacdec_lpd.cpp +++ b/libAACdec/src/usacdec_lpd.cpp @@ -336,17 +336,22 @@ void bass_pf_1sf_delay( { for (i = 0; i < lg; i++) { - /* scaled with SF_SYNTH + gain_sf + 1 */ + /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2: + * one additional shift of syn values + fMult => fMultDiv2 */ noise_in[i] = - (fMult(gainSGL, syn[i + i_subfr] - (syn[i + i_subfr - T] >> 1) - - (syn[i + i_subfr + T] >> 1))) >> - s1; + scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) - + (syn[i + i_subfr - T] >> 2) - + (syn[i + i_subfr + T] >> 2)), + 2 - s1); } for (i = lg; i < L_SUBFR; i++) { - /* scaled with SF_SYNTH + gain_sf + 1 */ + /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2: + * one additional shift of syn values + fMult => fMultDiv2 */ noise_in[i] = - (fMult(gainSGL, syn[i + i_subfr] - syn[i + i_subfr - T])) >> s1; + scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) - + (syn[i + i_subfr - T] >> 1)), + 2 - s1); } } } else { -- cgit v1.2.3 From 3634955ca5efcfe7a8565ef1ba676531b3123ca1 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 18 Oct 2019 14:03:59 +0200 Subject: Fix integer overflow in complex multiplication called from CLpd_FAC_Acelp2Mdct(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I7cd234698821cf13b163d323cfd90fdccaaec3c1 --- libAACdec/src/usacdec_fac.cpp | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_fac.cpp b/libAACdec/src/usacdec_fac.cpp index 0d3d844..0e28fbb 100644 --- a/libAACdec/src/usacdec_fac.cpp +++ b/libAACdec/src/usacdec_fac.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -668,9 +668,9 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; - cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); - *pOut0 = IMDCT_SCALE_DBL(x0); - *pOut1 = IMDCT_SCALE_DBL(-x1); + cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); + *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); + *pOut1 = IMDCT_SCALE_DBL_LSH1(-x1); pOut0++; pOut1--; } @@ -680,9 +680,9 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; - cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); - *pOut0 = IMDCT_SCALE_DBL(x0); - *pOut1 = IMDCT_SCALE_DBL(x1); + cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); + *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); + *pOut1 = IMDCT_SCALE_DBL_LSH1(x1); pOut0++; pOut1--; } @@ -691,9 +691,9 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; - cplxMult(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]); - *pOut0 = IMDCT_SCALE_DBL(x0); - *pOut1 = IMDCT_SCALE_DBL(x1); + cplxMultDiv2(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]); + *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); + *pOut1 = IMDCT_SCALE_DBL_LSH1(x1); pOut0++; pOut1--; } -- cgit v1.2.3 From 920ecc7487645192e9b9dee7c5261b5ff8b81be9 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 18 Oct 2019 14:04:15 +0200 Subject: Prevent signed integer overflows in CLpd_FAC_Acelp2Mdct(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I216f97b0c75a9076f3963036b098af37b390c5bb --- libAACdec/src/usacdec_fac.cpp | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_fac.cpp b/libAACdec/src/usacdec_fac.cpp index 0e28fbb..3c0b1ce 100644 --- a/libAACdec/src/usacdec_fac.cpp +++ b/libAACdec/src/usacdec_fac.cpp @@ -493,9 +493,7 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, /* Div2 is compensated by table scaling */ x = fMultDiv2(pTmp2[i], FacWindowZir[w]); x += fMultDiv2(pTmp1[-i - 1], FacWindowSynth[w]); - x += pFAC_and_FAC_ZIR[i]; - pOut1[i] = x; - + pOut1[i] = fAddSaturate(x, pFAC_and_FAC_ZIR[i]); w++; } } @@ -552,7 +550,7 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, FDK_ASSERT((pOut1 >= hMdct->overlap.time && pOut1 < hMdct->overlap.time + hMdct->ov_size) || (pOut1 >= output && pOut1 < output + 1024)); - *pOut1 += IMDCT_SCALE_DBL(-x1); + *pOut1 = fAddSaturate(*pOut1, IMDCT_SCALE_DBL(-x1)); pOut1--; } @@ -578,7 +576,7 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, FIXP_DBL x = -(*pCurr--); /* 5) (item 4) Synthesis filter Zir component, FAC ZIR (another one). */ if (i < f_len) { - x += *pF++; + x = fAddSaturate(x, *pF++); } FDK_ASSERT((pOut1 >= hMdct->overlap.time && @@ -705,7 +703,7 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, FIXP_DBL *pOut = pOut0 - fl / 2; FDK_ASSERT(fl / 2 <= 128); for (i = 0; i < fl / 2; i++) { - pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]); + pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i])); } hMdct->pFacZir = NULL; } -- cgit v1.2.3 From ade11a1d6f3954ded789afcdac8ef4c80f9bb2f9 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Fri, 18 Oct 2019 14:04:50 +0200 Subject: Prevent signed integer overflows in CLpd_FAC_Mdct2Acelp(). Bug: 131430997 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I348fe4be577ac39f961352902d138a8e07982096 --- libAACdec/src/usacdec_fac.cpp | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_fac.cpp b/libAACdec/src/usacdec_fac.cpp index 3c0b1ce..b246171 100644 --- a/libAACdec/src/usacdec_fac.cpp +++ b/libAACdec/src/usacdec_fac.cpp @@ -344,7 +344,7 @@ INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac, /* Overlap Add */ x0 = -fMult(*pOvl--, pWindow[i].v.re); - *pOut0 += IMDCT_SCALE_DBL(x0); + *pOut0 = fAddSaturate(*pOut0, IMDCT_SCALE_DBL(x0)); pOut0++; } } else { @@ -354,7 +354,7 @@ INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac, /* Overlap Add */ x0 = fMult(*pOvl--, pWindow[i].v.re); - *pOut0 += IMDCT_SCALE_DBL(x0); + *pOut0 = fAddSaturate(*pOut0, IMDCT_SCALE_DBL(x0)); pOut0++; } } @@ -362,7 +362,7 @@ INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac, 0) { /* this should only happen for ACELP -> TCX20 -> ACELP transition */ FIXP_DBL *pOut = pOut0 - fl / 2; /* fl/2 == fac_length */ for (i = 0; i < fl / 2; i++) { - pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]); + pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i])); } hMdct->pFacZir = NULL; } -- cgit v1.2.3 From fc8faa38d90380adfb61b2b280a36c73369c9aff Mon Sep 17 00:00:00 2001 From: Jean-Michel Trivi Date: Mon, 21 Oct 2019 14:52:48 -0700 Subject: FDK: USAC CLpd_Reset uses rate constants Use constants defined in libAACdec/src/usacdec_const.h Test: atest DecoderTestXheAac Change-Id: Idf1aa9fefb92b1abf64f50b044b2490b0d31b426 --- libAACdec/src/usacdec_lpd.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_lpd.cpp b/libAACdec/src/usacdec_lpd.cpp index 5c3426d..de0c2de 100644 --- a/libAACdec/src/usacdec_lpd.cpp +++ b/libAACdec/src/usacdec_lpd.cpp @@ -1228,7 +1228,7 @@ AAC_DECODER_ERROR CLpdChannelStream_Read( (INT)(samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM - (INT)PIT_MIN_12k8; - if ((samplingRate < 6000) || (samplingRate > 24000)) { + if ((samplingRate < FAC_FSCALE_MIN) || (samplingRate > FAC_FSCALE_MAX)) { error = AAC_DEC_PARSE_ERROR; goto bail; } -- cgit v1.2.3 From 07b5fd99418373e7c622030735bd47a0b153a9ec Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Tue, 29 Oct 2019 13:05:02 +0100 Subject: Use one additional bit headroom to prevent signed integer overflow in BuildAdaptiveExcitation(). Bug: 145666984 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I5c881238562c3d9f7cd8d77a8c52f7231126587f --- libAACdec/src/usacdec_acelp.cpp | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp index 1ac8c9f..a8dadc0 100644 --- a/libAACdec/src/usacdec_acelp.cpp +++ b/libAACdec/src/usacdec_acelp.cpp @@ -465,7 +465,9 @@ void BuildAdaptiveExcitation( /* Note: code[L_SUBFR] and exc2[L_SUBFR] share the same memory! If exc2[i] is written, code[i] will be destroyed! */ -#define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC) +#define SF_HEADROOM (1) +#define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC - SF_HEADROOM) +#define SF_GAIN_P2 (SF_GAIN_P - SF_HEADROOM) int i; FIXP_DBL tmp, cpe, code_smooth_prev, code_smooth; @@ -477,8 +479,8 @@ void BuildAdaptiveExcitation( cpe = (period_fac >> (2 - SF_PFAC)) + FL2FXCONST_DBL(0.25f); /* u'(n) */ - tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); /* v(0)*g_p */ - *exc++ = tmp + (fMultDiv2(code[0], gain_code) << SF); + tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1); /* v(0)*g_p */ + *exc++ = (tmp + (fMultDiv2(code[0], gain_code) << SF)) << SF_HEADROOM; /* u(n) */ code_smooth_prev = fMultDiv2(*code++, gain_code_smoothed) @@ -487,15 +489,15 @@ void BuildAdaptiveExcitation( code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; /* c(1) * g_sc */ tmp += code_smooth_prev; /* tmp = v(0)*g_p + c(0)*g_sc */ cpe_code_smooth = fMultDiv2(cpe, code_smooth); - *exc2++ = tmp - cpe_code_smooth; + *exc2++ = (tmp - cpe_code_smooth) << SF_HEADROOM; cpe_code_smooth_prev = fMultDiv2(cpe, code_smooth_prev); i = L_SUBFR - 2; do /* ARM926: 22 cycles per iteration */ { /* u'(n) */ - tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); - *exc++ = tmp + (fMultDiv2(code_i, gain_code) << SF); + tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1); + *exc++ = (tmp + (fMultDiv2(code_i, gain_code) << SF)) << SF_HEADROOM; /* u(n) */ tmp += code_smooth; /* += g_sc * c(i) */ tmp -= cpe_code_smooth_prev; @@ -503,16 +505,17 @@ void BuildAdaptiveExcitation( code_i = *code++; code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; cpe_code_smooth = fMultDiv2(cpe, code_smooth); - *exc2++ = tmp - cpe_code_smooth; /* tmp - c_pe * g_sc * c(i+1) */ + *exc2++ = (tmp - cpe_code_smooth) + << SF_HEADROOM; /* tmp - c_pe * g_sc * c(i+1) */ } while (--i != 0); /* u'(n) */ - tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); - *exc = tmp + (fMultDiv2(code_i, gain_code) << SF); + tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1); + *exc = (tmp + (fMultDiv2(code_i, gain_code) << SF)) << SF_HEADROOM; /* u(n) */ tmp += code_smooth; tmp -= cpe_code_smooth_prev; - *exc2++ = tmp; + *exc2++ = tmp << SF_HEADROOM; return; } -- cgit v1.2.3 From 089d368ddb5ce41c397fdd05034a1e67d519b836 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Tue, 29 Oct 2019 13:06:16 +0100 Subject: Prevent signed integer overflows in ELD filterbank, multE2_DinvF_fdk(). Bug: 145669388 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Iaba69b58666833c90e05b296c502ba6509a087a9 --- libAACdec/src/ldfiltbank.cpp | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/ldfiltbank.cpp b/libAACdec/src/ldfiltbank.cpp index c7d2928..1898557 100644 --- a/libAACdec/src/ldfiltbank.cpp +++ b/libAACdec/src/ldfiltbank.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -131,10 +131,12 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb, FIXP_DBL z0, z2, tmp; z2 = x[N / 2 + i]; - z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1)); + z0 = fAddSaturate(z2, + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1))); - z[N / 2 + i] = x[N / 2 - 1 - i] + - (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1)); + z[N / 2 + i] = fAddSaturate( + x[N / 2 - 1 - i], + (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1))); tmp = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) + fMultDiv2(z[i], fb[N + N / 2 + i])); @@ -159,10 +161,12 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb, FIXP_DBL z0, z2, tmp0, tmp1; z2 = x[N / 2 + i]; - z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1)); + z0 = fAddSaturate(z2, + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1))); - z[N / 2 + i] = x[N / 2 - 1 - i] + - (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1)); + z[N / 2 + i] = fAddSaturate( + x[N / 2 - 1 - i], + (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1))); tmp0 = (fMultDiv2(z[N / 2 + i], fb[N / 2 - 1 - i]) + fMultDiv2(z[i], fb[N / 2 + i])); -- cgit v1.2.3 From 2334454bc85e205b0d9f23ba6fbd1a966bf2c364 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Tue, 29 Oct 2019 13:09:38 +0100 Subject: Add sanity check for huffman escape sequences in HCR tool. Bug: 145669389 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Iaa624f3040269b5c84883bc3ee38cad7cd88e54d --- libAACdec/src/aacdec_hcrs.cpp | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdec_hcrs.cpp b/libAACdec/src/aacdec_hcrs.cpp index d2bc867..44b32a5 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1324,6 +1324,10 @@ UINT Hcr_State_BODY_SIGN_ESC__ESC_PREFIX(HANDLE_FDK_BITSTREAM bs, void *ptr) { /* count ones and store sum in escapePrefixUp */ if (carryBit == 1) { escapePrefixUp += 1; /* update conter for ones */ + if (escapePrefixUp > 8) { + pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__ESC_PREFIX; + return BODY_SIGN_ESC__ESC_PREFIX; + } /* store updated counter in sideinfo of current codeword */ pEscapeSequenceInfo[codewordOffset] &= -- cgit v1.2.3 From 9ab63ce1512780752eb5f54bfd91d132a8611b2a Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Wed, 13 Nov 2019 16:06:56 +0100 Subject: Prevent negation of INT_MIN in CConcealment_ApplyRandomSign(). Bug: 146937601 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I9747110eca96b37604df996ef5e86ea58e2d8932 --- libAACdec/src/conceal.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/conceal.cpp b/libAACdec/src/conceal.cpp index 5895cb8..ae98874 100644 --- a/libAACdec/src/conceal.cpp +++ b/libAACdec/src/conceal.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1618,7 +1618,7 @@ static void CConcealment_ApplyRandomSign(int randomPhase, FIXP_DBL *spec, } if (packedSign & 0x1) { - spec[i] = -spec[i]; + spec[i] = -fMax(spec[i], (FIXP_DBL)(MINVAL_DBL + 1)); } packedSign >>= 1; -- cgit v1.2.3 From cee316ab3ebde92047b5e76d15c00768b92cb890 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Wed, 13 Nov 2019 16:09:34 +0100 Subject: Fix too large shift exponent in CConcealment_InterpolateBuffer(). Bug: 146938361 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Idb0a4e2c87962e453a991f0a573155ace6e9bf40 --- libAACdec/src/conceal.cpp | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/conceal.cpp b/libAACdec/src/conceal.cpp index ae98874..379e63a 100644 --- a/libAACdec/src/conceal.cpp +++ b/libAACdec/src/conceal.cpp @@ -1228,7 +1228,6 @@ static void CConcealment_InterpolateBuffer(FIXP_DBL *spectrum, int sfb, line = 0; int fac_shift; int fac_mod; - FIXP_DBL accu; for (sfb = 0; sfb < sfbCnt; sfb++) { fac_shift = @@ -1236,15 +1235,11 @@ static void CConcealment_InterpolateBuffer(FIXP_DBL *spectrum, fac_mod = fac_shift & 3; fac_shift = (fac_shift >> 2) + 1; fac_shift += *pSpecScalePrv - fixMax(*pSpecScalePrv, *pSpecScaleAct); + fac_shift = fMax(fMin(fac_shift, DFRACT_BITS - 1), -(DFRACT_BITS - 1)); for (; line < pSfbOffset[sfb + 1]; line++) { - accu = fMult(*(spectrum + line), facMod4Table[fac_mod]); - if (fac_shift < 0) { - accu >>= -fac_shift; - } else { - accu <<= fac_shift; - } - *(spectrum + line) = accu; + FIXP_DBL accu = fMult(*(spectrum + line), facMod4Table[fac_mod]); + *(spectrum + line) = scaleValue(accu, fac_shift); } } *pSpecScaleOut = fixMax(*pSpecScalePrv, *pSpecScaleAct); -- cgit v1.2.3 From 7388c0732d9cfabd5757f4eda9ea15608a5db126 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Wed, 13 Nov 2019 16:10:21 +0100 Subject: Reject HE-AAC SBR with a ratio greater than 2:1 and reject USAC streams containing legacy DRC info. Bug: 146937553 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: I1a23d986160c1de07a7159ac026f57d821d3ff1d --- libAACdec/src/aacdecoder.cpp | 39 ++++++++++++++++++++++++++++++++------- 1 file changed, 32 insertions(+), 7 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index 6b5a86c..bd12d96 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1848,6 +1848,12 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, self->streamInfo.extSamplingRate / self->downscaleFactor; } } + if ((asc->m_aot == AOT_AAC_LC) && (asc->m_sbrPresentFlag == 1) && + (asc->m_extensionSamplingFrequency > (2 * asc->m_samplingFrequency))) { + return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* Core decoder supports at most a + 1:2 upsampling for HE-AAC and + HE-AACv2 */ + } /* --------- vcb11 ------------ */ self->flags[streamIndex] |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0; @@ -2366,6 +2372,13 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, goto bail; } + if (*configChanged) { + if (asc->m_aot == AOT_USAC) { + self->hDrcInfo->enable = 0; + self->hDrcInfo->progRefLevelPresent = 0; + } + } + if (asc->m_aot == AOT_USAC) { pcmLimiter_SetAttack(self->hLimiter, (5)); pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f)); @@ -3172,9 +3185,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( self->hDrcInfo, bs, self->pAacDecoderStaticChannelInfo, pce->ElementInstanceTag, drcChMap, aacChannels); if (mapped > 0) { - /* If at least one DRC thread has been mapped to a channel threre was DRC - * data in the bitstream. */ - self->flags[streamIndex] |= AC_DRC_PRESENT; + if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) { + /* If at least one DRC thread has been mapped to a channel there was DRC + * data in the bitstream. */ + self->flags[streamIndex] |= AC_DRC_PRESENT; + } else { + self->hDrcInfo->enable = 0; + self->hDrcInfo->progRefLevelPresent = 0; + ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT; + } } /* Create a reverse mapping table */ @@ -3419,9 +3438,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( self->hDrcInfo, bs, self->pAacDecoderStaticChannelInfo, pce->ElementInstanceTag, drcChMap, aacChannels); if (mapped > 0) { - /* If at least one DRC thread has been mapped to a channel threre was DRC - * data in the bitstream. */ - self->flags[streamIndex] |= AC_DRC_PRESENT; + if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) { + /* If at least one DRC thread has been mapped to a channel there was DRC + * data in the bitstream. */ + self->flags[streamIndex] |= AC_DRC_PRESENT; + } else { + self->hDrcInfo->enable = 0; + self->hDrcInfo->progRefLevelPresent = 0; + ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT; + } } } -- cgit v1.2.3 From 00285819a2099a5bd0b0ebcdaebf8ee88f02546c Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Thu, 19 Dec 2019 17:20:20 +0100 Subject: Fix AACDEC_INTR handling for USAC configuration with multiple pre-roll AUs Bug: 148384920 Test: atest DecoderTestXheAac DecoderTestAacDrc Change-Id: I88cd6da0b18c73f7b521ea58ba8b8f364278b64f --- libAACdec/src/aacdecoder_lib.cpp | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp index 86ec899..e90dbef 100644 --- a/libAACdec/src/aacdecoder_lib.cpp +++ b/libAACdec/src/aacdecoder_lib.cpp @@ -1269,9 +1269,9 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, } /* Signal bit stream interruption to other modules if required. */ - if (fTpInterruption || (flags & AACDEC_INTR)) { + if (fTpInterruption || ((flags & AACDEC_INTR) && (accessUnit == 0))) { aacDecoder_SignalInterruption(self); - if (!(flags & AACDEC_INTR)) { + if (!((flags & AACDEC_INTR) && (accessUnit == 0))) { ErrorStatus = AAC_DEC_TRANSPORT_SYNC_ERROR; goto bail; } @@ -1768,7 +1768,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, if (self->streamInfo.extAot != AOT_AAC_SLS) { INT pcmLimiterScale = 0; PCMDMX_ERROR dmxErr = PCMDMX_OK; - if (flags & (AACDEC_INTR)) { + if ((flags & AACDEC_INTR) && (accessUnit == 0)) { /* delete data from the past (e.g. mixdown coeficients) */ pcmDmx_Reset(self->hPcmUtils, PCMDMX_RESET_BS_DATA); } -- cgit v1.2.3 From 31f66f6d3ffcfa429c170b2c35250982d11f5082 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Thu, 19 Dec 2019 17:21:07 +0100 Subject: Extend decoder API with audio output loudness info (FDKdec v3.1.3). Bug: 148385721 Test: atest DecoderTestXheAac DecoderTestAacDrc Change-Id: I68b09883def21baef259c9ab914922567ab8cee3 --- documentation/aacDecoder.pdf | Bin 488412 -> 494126 bytes libAACdec/include/aacdecoder_lib.h | 28 ++++++++++++++++-------- libAACdec/src/aacdecoder.cpp | 2 ++ libAACdec/src/aacdecoder_lib.cpp | 34 +++++++++++++++++++++++++++++- libDRCdec/include/FDK_drcDecLib.h | 11 +++++++--- libDRCdec/src/FDK_drcDecLib.cpp | 2 ++ libDRCdec/src/drcDec_selectionProcess.cpp | 11 +++++----- libDRCdec/src/drcDec_selectionProcess.h | 2 ++ 8 files changed, 71 insertions(+), 19 deletions(-) (limited to 'libAACdec') diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf index eb2a75e..f2131ed 100644 Binary files a/documentation/aacDecoder.pdf and b/documentation/aacDecoder.pdf differ diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h index 6c2fda4..6084bd3 100644 --- a/libAACdec/include/aacdecoder_lib.h +++ b/libAACdec/include/aacdecoder_lib.h @@ -892,15 +892,25 @@ typedef struct { 1770. If no level has been found in the bitstream the value is -1. */ SCHAR - drcPresMode; /*!< DRC presentation mode. According to ETSI TS 101 154, - this field indicates whether light (MPEG-4 Dynamic Range - Control tool) or heavy compression (DVB heavy - compression) dynamic range control shall take priority - on the outputs. For details, see ETSI TS 101 154, table - C.33. Possible values are: \n -1: No corresponding - metadata found in the bitstream \n 0: DRC presentation - mode not indicated \n 1: DRC presentation mode 1 \n 2: - DRC presentation mode 2 \n 3: Reserved */ + drcPresMode; /*!< DRC presentation mode. According to ETSI TS 101 154, + this field indicates whether light (MPEG-4 Dynamic Range + Control tool) or heavy compression (DVB heavy + compression) dynamic range control shall take priority + on the outputs. For details, see ETSI TS 101 154, table + C.33. Possible values are: \n -1: No corresponding + metadata found in the bitstream \n 0: DRC presentation + mode not indicated \n 1: DRC presentation mode 1 \n 2: + DRC presentation mode 2 \n 3: Reserved */ + INT outputLoudness; /*!< Audio output loudness in steps of -0.25 dB. Range: 0 + (0 dBFS) to 231 (-57.75 dBFS).\n A value of -1 + indicates that no loudness metadata is present.\n If + loudness normalization is active, the value corresponds + to the target loudness value set with + ::AAC_DRC_REFERENCE_LEVEL.\n If loudness normalization + is not active, the output loudness value corresponds to + the loudness metadata given in the bitstream.\n + Loudness metadata can originate from MPEG-4 DRC or + MPEG-D DRC. */ } CStreamInfo; diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index bd12d96..f747b2d 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -1225,6 +1225,8 @@ static void CStreamInfoInit(CStreamInfo *pStreamInfo) { pStreamInfo->drcProgRefLev = -1; /* set program reference level to not indicated */ pStreamInfo->drcPresMode = -1; /* default: presentation mode not indicated */ + + pStreamInfo->outputLoudness = -1; /* default: no loudness metadata present */ } /*! diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp index e90dbef..2ba0e86 100644 --- a/libAACdec/src/aacdecoder_lib.cpp +++ b/libAACdec/src/aacdecoder_lib.cpp @@ -120,7 +120,7 @@ amm-info@iis.fraunhofer.de /* Decoder library info */ #define AACDECODER_LIB_VL0 3 #define AACDECODER_LIB_VL1 1 -#define AACDECODER_LIB_VL2 2 +#define AACDECODER_LIB_VL2 3 #define AACDECODER_LIB_TITLE "AAC Decoder Lib" #ifdef __ANDROID__ #define AACDECODER_LIB_BUILD_DATE "" @@ -1764,6 +1764,38 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, } } } + if (FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_IS_ACTIVE)) { + /* return output loudness information for MPEG-D DRC */ + LONG outputLoudness = + FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_OUTPUT_LOUDNESS); + if (outputLoudness == DRC_DEC_LOUDNESS_NOT_PRESENT) { + /* no valid MPEG-D DRC loudness value contained */ + self->streamInfo.outputLoudness = -1; + } else { + if (outputLoudness > 0) { + /* positive output loudness values (very unusual) are limited to 0 + * dB */ + self->streamInfo.outputLoudness = 0; + } else { + self->streamInfo.outputLoudness = + -(INT)outputLoudness >> + 22; /* negate and scale from e = 7 to e = (31-2) */ + } + } + } else { + /* return output loudness information for MPEG-4 DRC */ + if (self->streamInfo.drcProgRefLev < + 0) { /* no MPEG-4 DRC loudness metadata contained */ + self->streamInfo.outputLoudness = -1; + } else { + if (self->defaultTargetLoudness < + 0) { /* loudness normalization is off */ + self->streamInfo.outputLoudness = self->streamInfo.drcProgRefLev; + } else { + self->streamInfo.outputLoudness = self->defaultTargetLoudness; + } + } + } if (self->streamInfo.extAot != AOT_AAC_SLS) { INT pcmLimiterScale = 0; diff --git a/libDRCdec/include/FDK_drcDecLib.h b/libDRCdec/include/FDK_drcDecLib.h index 2d28d23..79f8566 100644 --- a/libDRCdec/include/FDK_drcDecLib.h +++ b/libDRCdec/include/FDK_drcDecLib.h @@ -114,6 +114,8 @@ amm-info@iis.fraunhofer.de extern "C" { #endif +#define DRC_DEC_LOUDNESS_NOT_PRESENT (LONG)0x7FFFFFFE + typedef struct s_drc_decoder* HANDLE_DRC_DECODER; typedef struct s_uni_drc_interface* HANDLE_UNI_DRC_INTERFACE; typedef struct s_selection_process_output* HANDLE_SEL_PROC_OUTPUT; @@ -150,9 +152,12 @@ typedef enum { DRC_DEC_IS_ACTIVE, /**< MPEG-D DRC payload is present and at least one of Dynamic Range Control (DRC) or Loudness Normalization (LN) is activated */ - DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED /**< number of output channels if - appropriate downmixInstruction exists - */ + DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED, /**< number of output channels if + appropriate downmixInstruction + exists */ + DRC_DEC_OUTPUT_LOUDNESS /**< output loudness in dB, with exponent e = 7, or + DRC_DEC_LOUDNESS_NOT_PRESENT if no loudness is + contained in the bitstream */ } DRC_DEC_USERPARAM; typedef enum { diff --git a/libDRCdec/src/FDK_drcDecLib.cpp b/libDRCdec/src/FDK_drcDecLib.cpp index e43279f..83b5773 100644 --- a/libDRCdec/src/FDK_drcDecLib.cpp +++ b/libDRCdec/src/FDK_drcDecLib.cpp @@ -519,6 +519,8 @@ LONG FDK_drcDec_GetParam(HANDLE_DRC_DECODER hDrcDec, } case DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED: return (LONG)hDrcDec->selProcOutput.targetChannelCount; + case DRC_DEC_OUTPUT_LOUDNESS: + return (LONG)hDrcDec->selProcOutput.outputLoudness; default: return 0; } diff --git a/libDRCdec/src/drcDec_selectionProcess.cpp b/libDRCdec/src/drcDec_selectionProcess.cpp index fe6034c..46ed740 100644 --- a/libDRCdec/src/drcDec_selectionProcess.cpp +++ b/libDRCdec/src/drcDec_selectionProcess.cpp @@ -103,8 +103,6 @@ amm-info@iis.fraunhofer.de #include "drcDec_selectionProcess.h" #include "drcDec_tools.h" -#define UNDEFINED_LOUDNESS_VALUE (FIXP_DBL) MAXVAL_DBL - typedef enum { DETR_NONE = 0, DETR_NIGHT = 1, @@ -1136,9 +1134,8 @@ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement8( FIXP_DBL loudnessDeviationMax = ((FIXP_DBL)hSelProcInput->loudnessDeviationMax) << (DFRACT_BITS - 1 - 7); - ; - if (hSelProcInput->loudnessNormalizationOn) { + { retVal = _getLoudness(hLoudnessInfoSet, hSelProcInput->albumMode, hSelProcInput->loudnessMeasurementMethod, hSelProcInput->loudnessMeasurementSystem, @@ -1147,9 +1144,10 @@ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement8( hSelProcInput->downmixIdRequested[downmixIdIndex], &loudnessNormalizationGainDb, &loudness); if (retVal) return (retVal); - } else { + } + + if (!hSelProcInput->loudnessNormalizationOn) { loudnessNormalizationGainDb = (FIXP_DBL)0; - loudness = UNDEFINED_LOUDNESS_VALUE; } retVal = _getSignalPeakLevel( @@ -2070,6 +2068,7 @@ static DRCDEC_SELECTION_PROCESS_RETURN _generateOutputInfo( pSelectionData->loudnessNormalizationGainDbAdjusted + hSelProcInput->loudnessNormalizationGainModificationDb; hSelProcOutput->outputPeakLevelDb = pSelectionData->outputPeakLevel; + hSelProcOutput->outputLoudness = pSelectionData->outputLoudness; hSelProcOutput->boost = boost; hSelProcOutput->compress = compress; diff --git a/libDRCdec/src/drcDec_selectionProcess.h b/libDRCdec/src/drcDec_selectionProcess.h index 420bae6..0f878cf 100644 --- a/libDRCdec/src/drcDec_selectionProcess.h +++ b/libDRCdec/src/drcDec_selectionProcess.h @@ -111,6 +111,8 @@ amm-info@iis.fraunhofer.de typedef struct s_drcdec_selection_process* HANDLE_DRC_SELECTION_PROCESS; +#define UNDEFINED_LOUDNESS_VALUE (FIXP_DBL)(MAXVAL_DBL - 1) + typedef enum { DRCDEC_SELECTION_PROCESS_NO_ERROR = 0, -- cgit v1.2.3 From 1d451971ca16a2ba3655dec3d23115b0a240a8bb Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Thu, 19 Dec 2019 17:21:57 +0100 Subject: Revise and straighten out useless references from decoder documentation. Bug: 149481606 Test: doc update in header, so just make Change-Id: I656e76762ee336931d4f1938e5c39c8ad7f65954 --- documentation/aacDecoder.pdf | Bin 494126 -> 490978 bytes libAACdec/include/aacdecoder_lib.h | 116 ++++++++++++++----------------------- 2 files changed, 44 insertions(+), 72 deletions(-) (limited to 'libAACdec') diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf index f2131ed..d04a7d6 100644 Binary files a/documentation/aacDecoder.pdf and b/documentation/aacDecoder.pdf differ diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h index 6084bd3..56f4ec1 100644 --- a/libAACdec/include/aacdecoder_lib.h +++ b/libAACdec/include/aacdecoder_lib.h @@ -164,9 +164,6 @@ The contents of each file is described in detail in this document. All header files are provided for usage in specific C/C++ programs. The main AAC decoder library API functions are located in aacdecoder_lib.h header file. -In binary releases the decoder core resides in statically linkable libraries, -for example libAACdec.a. - \section Calling_Sequence Calling Sequence @@ -174,19 +171,7 @@ The following sequence is necessary for proper decoding of ISO/MPEG-2/4 AAC, HE-AAC v2, or MPEG-D USAC bitstreams. In the following description, input stream read and output write function details are left out, since they may be implemented in a variety of configurations depending on the user's specific -requirements. The example implementation uses file-based input/output, and in -such case one may call mpegFileRead_Open() to open an input file and to allocate -memory for the required structures, and the corresponding mpegFileRead_Close() -to close opened files and to de-allocate associated structures. -mpegFileRead_Open() will attempt to detect the bitstream format and in case of -MPEG-4 file format or Raw Packets file format (a proprietary Fraunhofer IIS file -format suitable only for testing) it will read the Audio Specific Config data -(ASC). An unsuccessful attempt to recognize the bitstream format requires the -user to provide this information manually. For any other bitstream formats that -are usually applicable in streaming applications, the decoder itself will try to -synchronize and parse the given bitstream fragment using the FDK transport -library. Hence, for streaming applications (without file access) this step is -not necessary. +requirements. -# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder @@ -205,19 +190,17 @@ do { working memory (a client-supplied input buffer "inBuffer" in framework). This buffer will be used to load AAC bitstream data to the decoder. Only when all data in this buffer has been processed will the decoder signal an empty buffer. -For file-based input, you may invoke mpegFileRead_Read() to acquire new -bitstream data. -# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer with the client-supplied bitstream input buffer. Note, if the data loaded in to the internal buffer is not sufficient to decode a frame, aacDecoder_DecodeFrame() will return ::AAC_DEC_NOT_ENOUGH_BITS until a sufficient amount of data is loaded in to the internal buffer. For streaming formats (ADTS, LOAS), it is acceptable to load more than one frame to the -decoder. However, for RAW file format (Fraunhofer IIS proprietary format), only -one frame may be loaded to the decoder per aacDecoder_DecodeFrame() call. For -least amount of communication delay, fill and decode should be performed on a -frame by frame basis. \code ErrorStatus = aacDecoder_Fill(aacDecoderInfo, -inBuffer, bytesRead, bytesValid); \endcode +decoder. However, for packed based formats, only one frame may be loaded to the +decoder per aacDecoder_DecodeFrame() call. For least amount of communication +delay, fill and decode should be performed on a frame by frame basis. \code + ErrorStatus = aacDecoder_Fill(aacDecoderInfo, inBuffer, bytesRead, +bytesValid); \endcode -# Call aacDecoder_DecodeFrame(). This function decodes one frame and writes decoded PCM audio data to a client-supplied buffer. It is the client's responsibility to allocate a buffer which is large enough to hold the decoded @@ -225,12 +208,9 @@ output data. \code ErrorStatus = aacDecoder_DecodeFrame(aacDecoderInfo, TimeData, OUT_BUF_SIZE, flags); \endcode If the bitstream configuration (number of channels, sample rate, frame size) is not known a priori, you may call aacDecoder_GetStreamInfo() to retrieve a structure that contains this -information. You may use this data to initialize an audio output device. In the -example program, if the number of channels or the sample rate has changed since -program start or the previously decoded frame, the audio output device is then -re-initialized. If WAVE file output is chosen, a new WAVE file for each new -stream configuration is be created. \code p_si = -aacDecoder_GetStreamInfo(aacDecoderInfo); \endcode +information. You may use this data to initialize an audio output device. \code + p_si = aacDecoder_GetStreamInfo(aacDecoderInfo); +\endcode -# Repeat steps 5 to 7 until no data is available to decode any more, or in case of error. \code } while (bytesRead[0] > 0 || doFlush || doBsFlush || forceContinue); \endcode @@ -239,7 +219,7 @@ structures. \code aacDecoder_Close(aacDecoderInfo); \endcode \image latex decode.png "Decode calling sequence" width=11cm -\image latex change_source.png "Change data source sequence" width 5cm +\image latex change_source.png "Change data source sequence" width=5cm \image latex conceal.png "Error concealment sequence" width=14cm @@ -296,16 +276,14 @@ input buffer, and one to hold the decoded output PCM sample data. In resource limited applications, the output buffer may be reused as an external input buffer prior to the subsequence aacDecoder_Fill() function call. -The external input buffer is set in the example program and its size is defined -by ::IN_BUF_SIZE. You may freely choose different buffer sizes. To feed the data -to the decoder-internal input buffer, use the function aacDecoder_Fill(). This -function returns important information regarding the number of bytes in the -external input buffer that have not yet been copied into the internal input -buffer (variable bytesValid). Once the external buffer has been fully copied, it -can be completely re-filled again. In case you wish to refill the buffer while -there are unprocessed bytes (bytesValid is unequal 0), you should preserve the -unconsumed data. However, we recommend to refill the buffer only when bytesValid -returns 0. +To feed the data to the decoder-internal input buffer, use the +function aacDecoder_Fill(). This function returns important information +regarding the number of bytes in the external input buffer that have not yet +been copied into the internal input buffer (variable bytesValid). Once the +external buffer has been fully copied, it can be completely re-filled again. In +case you wish to refill the buffer while there are unprocessed bytes (bytesValid +is unequal 0), you should preserve the unconsumed data. However, we recommend to +refill the buffer only when bytesValid returns 0. The bytesValid parameter is an input and output parameter to the FDK decoder. As an input, it signals how many valid bytes are available in the external buffer. @@ -340,10 +318,7 @@ explanation, please refer to ISO/IEC 13818-7:2005(E), chapter 8.5.3.2. In case a Program Config is included in the audio configuration, the channel mapping described within it will be adopted. -In case of MPEG-D Surround the channel mapping will follow the same criteria -described in ISO/IEC 13818-7:2005(E), but adding corresponding top channels (if -available) to the channel types in order to avoid ambiguity. The examples below -explain these aspects in detail. +The examples below explain these aspects in detail. \section OutputFormatChange Changing the audio output format @@ -689,9 +664,7 @@ typedef enum { 2. If the parameter value is greater than that of ::AAC_PCM_MAX_OUTPUT_CHANNELS both will be set to the same value. \n - 3. This parameter does not affect MPEG Surround processing. - \n - 4. This parameter will be ignored if the number of encoded + 3. This parameter will be ignored if the number of encoded audio channels is greater than 8. */ AAC_PCM_MAX_OUTPUT_CHANNELS = 0x0012, /*!< Maximum number of PCM output channels. If lower than the @@ -718,11 +691,7 @@ typedef enum { 2. If the parameter value is greater than zero but smaller than ::AAC_PCM_MIN_OUTPUT_CHANNELS both will be set to same value. \n - 3. The operating mode of the MPEG Surround module will be - set accordingly. \n - 4. Setting this parameter with any value will disable the - binaural processing of the MPEG Surround module - 5. This parameter will be ignored if the number of encoded + 3. This parameter will be ignored if the number of encoded audio channels is greater than 8. */ AAC_METADATA_PROFILE = 0x0020, /*!< See ::AAC_MD_PROFILE for all available values. */ @@ -803,11 +772,11 @@ typedef enum { sequences for fading in and out, if provided in the bitstream.\n Enabled album mode makes use of dedicated album loudness information, if provided in the bitstream.\n */ - AAC_QMF_LOWPOWER = 0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing - mode. \n -1: Use internal default. Implies MPEG - Surround partially complex accordingly. \n 0: - Use complex QMF data mode. \n 1: Use real (low - power) QMF data mode. \n */ + AAC_QMF_LOWPOWER = + 0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing mode. \n + -1: Use internal default. \n + 0: Use complex QMF data mode. \n + 1: Use real (low power) QMF data mode. \n */ AAC_TPDEC_CLEAR_BUFFER = 0x0603 /*!< Clear internal bit stream buffer of transport layers. The decoder will start decoding at new data passed after this event @@ -1038,21 +1007,24 @@ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_Fill(HANDLE_AACDECODER self, const UINT bufferSize[], UINT *bytesValid); -#define AACDEC_CONCEAL \ - 1 /*!< Flag for aacDecoder_DecodeFrame(): Trigger the built-in error \ - concealment module to generate a substitute signal for one lost frame. \ - New input data will not be considered. */ -#define AACDEC_FLUSH \ - 2 /*!< Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all \ - delayed audio without having new input data. Thus new input data will \ - not be considered.*/ -#define AACDEC_INTR \ - 4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data \ - discontinuity. Resync any internals as necessary. */ -#define AACDEC_CLRHIST \ - 8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and \ - history buffers. CAUTION: This can cause discontinuities in the output \ - signal. */ +/** Flag for aacDecoder_DecodeFrame(): Trigger the built-in error concealment + * module to generate a substitute signal for one lost frame. New input data + * will not be considered. + */ +#define AACDEC_CONCEAL 1 +/** Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all delayed + * audio without having new input data. Thus new input data will not be + * considered. + */ +#define AACDEC_FLUSH 2 +/** Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data + * discontinuity. Resync any internals as necessary. + */ +#define AACDEC_INTR 4 +/** Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history + * buffers. CAUTION: This can cause discontinuities in the output signal. + */ +#define AACDEC_CLRHIST 8 /** * \brief Decode one audio frame -- cgit v1.2.3 From 90c29e0808d2221ae747bad7385036c20b5662b0 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Thu, 19 Dec 2019 17:27:40 +0100 Subject: Revise memory overlay usage and remove deprecated buffers and overlay tags. Bug: 149514474 Test: atest DecoderTestXheAac DecoderTestAacDrc Change-Id: I7aea2898a3c49e06209fae89d734939c100e1184 --- libAACdec/src/aac_ram.cpp | 10 +--------- libAACdec/src/aac_ram.h | 7 +------ libFDK/include/FDK_qmf_domain.h | 3 +-- libFDK/src/FDK_qmf_domain.cpp | 14 +++++--------- libSBRdec/src/sbr_ram.cpp | 5 +---- 5 files changed, 9 insertions(+), 30 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aac_ram.cpp b/libAACdec/src/aac_ram.cpp index e13167d..aa8f6a6 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -105,12 +105,7 @@ amm-info@iis.fraunhofer.de #define WORKBUFFER1_TAG 0 #define WORKBUFFER2_TAG 1 - -#define WORKBUFFER3_TAG 4 -#define WORKBUFFER4_TAG 5 - #define WORKBUFFER5_TAG 6 - #define WORKBUFFER6_TAG 7 /*! The structure AAC_DECODER_INSTANCE is the top level structure holding all @@ -169,9 +164,6 @@ C_ALLOC_MEM2(TimeDataFlush, INT_PCM, TIME_DATA_FLUSH_SIZE, (8)) C_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL, ((8) * 1024), SECT_DATA_L2, WORKBUFFER2_TAG) -C_ALLOC_MEM_OVERLAY(WorkBufferCore3, FIXP_DBL, WB_SECTION_SIZE, SECT_DATA_L2, - WORKBUFFER3_TAG) -C_AALLOC_MEM(WorkBufferCore4, FIXP_DBL, WB_SECTION_SIZE) C_ALLOC_MEM_OVERLAY(WorkBufferCore6, SCHAR, fMax((INT)(sizeof(FIXP_DBL) * WB_SECTION_SIZE), (INT)sizeof(CAacDecoderCommonData)), diff --git a/libAACdec/src/aac_ram.h b/libAACdec/src/aac_ram.h index a861e25..b9b95b7 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -136,12 +136,7 @@ H_ALLOC_MEM(TimeDataFlush, INT_PCM) H_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1) H_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL) - -H_ALLOC_MEM_OVERLAY(WorkBufferCore3, FIXP_DBL) -H_ALLOC_MEM(WorkBufferCore4, FIXP_DBL) - H_ALLOC_MEM_OVERLAY(WorkBufferCore5, PCM_DEC) - H_ALLOC_MEM_OVERLAY(WorkBufferCore6, SCHAR) #endif /* #ifndef AAC_RAM_H */ diff --git a/libFDK/include/FDK_qmf_domain.h b/libFDK/include/FDK_qmf_domain.h index d8bc8ce..aa247e9 100644 --- a/libFDK/include/FDK_qmf_domain.h +++ b/libFDK/include/FDK_qmf_domain.h @@ -123,11 +123,10 @@ typedef enum { #define QMF_WB_SECTION_SIZE (1024 * 2) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore1, FIXP_DBL) -H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore2, FIXP_DBL) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore3, FIXP_DBL) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore4, FIXP_DBL) -H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore5, FIXP_DBL) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore6, FIXP_DBL) +H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore7, FIXP_DBL) #define QMF_DOMAIN_MAX_ANALYSIS_QMF_BANDS (64) #define QMF_DOMAIN_MAX_SYNTHESIS_QMF_BANDS (QMF_MAX_SYNTHESIS_BANDS) diff --git a/libFDK/src/FDK_qmf_domain.cpp b/libFDK/src/FDK_qmf_domain.cpp index 5e1d35f..71e2ef5 100644 --- a/libFDK/src/FDK_qmf_domain.cpp +++ b/libFDK/src/FDK_qmf_domain.cpp @@ -106,25 +106,21 @@ amm-info@iis.fraunhofer.de #include "common_fix.h" #define WORKBUFFER1_TAG 0 -#define WORKBUFFER2_TAG 1 - #define WORKBUFFER3_TAG 4 #define WORKBUFFER4_TAG 5 -#define WORKBUFFER5_TAG 6 #define WORKBUFFER6_TAG 7 +#define WORKBUFFER7_TAG 8 C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore1, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L1, WORKBUFFER1_TAG) -C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore2, FIXP_DBL, QMF_WB_SECTION_SIZE, - SECT_DATA_L2, WORKBUFFER2_TAG) C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore3, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L2, WORKBUFFER3_TAG) C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore4, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L2, WORKBUFFER4_TAG) -C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore5, FIXP_DBL, QMF_WB_SECTION_SIZE, - SECT_DATA_L2, WORKBUFFER5_TAG) C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore6, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L2, WORKBUFFER6_TAG) +C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore7, FIXP_DBL, QMF_WB_SECTION_SIZE, + SECT_DATA_L2, WORKBUFFER7_TAG) /*! Analysis states buffer.
Dimension: #((8) + (1)) */ @@ -949,7 +945,7 @@ QMF_DOMAIN_ERROR FDK_QmfDomain_Configure(HANDLE_FDK_QMF_DOMAIN hqd) { if ((size > 4 * QMF_WB_SECTION_SIZE) && (pWorkBuffer[4] == NULL)) { /* get work buffer of size QMF_WB_SECTION_SIZE */ - pWorkBuffer[4] = GetQmfWorkBufferCore5(); + pWorkBuffer[4] = GetQmfWorkBufferCore7(); } /* 8. distribute workbuffer over processing channels */ @@ -995,7 +991,7 @@ static void FDK_QmfDomain_FreeWorkBuffer(HANDLE_FDK_QMF_DOMAIN hqd) { if (pWorkBuffer[1]) FreeQmfWorkBufferCore1(&pWorkBuffer[1]); if (pWorkBuffer[2]) FreeQmfWorkBufferCore3(&pWorkBuffer[2]); if (pWorkBuffer[3]) FreeQmfWorkBufferCore4(&pWorkBuffer[3]); - if (pWorkBuffer[4]) FreeQmfWorkBufferCore5(&pWorkBuffer[4]); + if (pWorkBuffer[4]) FreeQmfWorkBufferCore7(&pWorkBuffer[4]); } void FDK_QmfDomain_FreeMem(HANDLE_FDK_QMF_DOMAIN hqd) { diff --git a/libSBRdec/src/sbr_ram.cpp b/libSBRdec/src/sbr_ram.cpp index 8b35fd2..a759d71 100644 --- a/libSBRdec/src/sbr_ram.cpp +++ b/libSBRdec/src/sbr_ram.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -109,9 +109,6 @@ amm-info@iis.fraunhofer.de #include "sbr_ram.h" -#define WORKBUFFER1_TAG 2 -#define WORKBUFFER2_TAG 3 - /*! \name StaticSbrData -- cgit v1.2.3 From 57c9355de0269afb462ad4a8aa8814f6a6486ff1 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Thu, 19 Dec 2019 17:27:53 +0100 Subject: Create new applyDrcLevelNormalization() function for subsequent usage. Bug: 149514474 Test: atest DecoderTestXheAac DecoderTestAacDrc Change-Id: I23cc3081ff71830bc96766a31131594499dbdd04 --- libAACdec/src/aacdec_drc.cpp | 165 +++++++++++++++++++++++++++++++++++++++ libAACdec/src/aacdec_drc.h | 50 +++++++++++- libAACdec/src/aacdec_drc_types.h | 9 ++- 3 files changed, 222 insertions(+), 2 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdec_drc.cpp b/libAACdec/src/aacdec_drc.cpp index 4129d0f..6066a64 100644 --- a/libAACdec/src/aacdec_drc.cpp +++ b/libAACdec/src/aacdec_drc.cpp @@ -149,6 +149,20 @@ static INT convert_drcParam(FIXP_DBL param_dbl) { return (INT)param_long; } +/*! +\brief Reset DRC information + +\self Handle of DRC info + +\return none +*/ +void aacDecoder_drcReset(HANDLE_AAC_DRC self) { + self->applyExtGain = 0; + self->additionalGainPrev = AACDEC_DRC_GAIN_INIT_VALUE; + self->additionalGainFilterState = AACDEC_DRC_GAIN_INIT_VALUE; + self->additionalGainFilterState1 = AACDEC_DRC_GAIN_INIT_VALUE; +} + /*! \brief Initialize DRC information @@ -192,6 +206,8 @@ void aacDecoder_drcInit(HANDLE_AAC_DRC self) { self->progRefLevelPresent = 0; self->presMode = -1; self->uniDrcPrecedence = 0; + + aacDecoder_drcReset(self); } /*! @@ -1353,3 +1369,152 @@ void aacDecoder_drcGetInfo(HANDLE_AAC_DRC self, SCHAR *pPresMode, } } } + +/** + * \brief Apply DRC Level Normalization. + * + * This function prepares/applies the gain values for the DRC Level + * Normalization and returns the exponent of the time data. The following two + * cases are handled: + * + * - Limiter enabled: + * The input data must be interleaved. + * One gain per sample is written to the buffer pGainPerSample. + * If necessary the time data is rescaled. + * + * - Limiter disabled: + * The input data can be interleaved or deinterleaved. + * The gain values are applied to the time data. + * If necessary the time data is rescaled. + * + * \param hDrcInfo [i/o] handle to drc data structure. + * \param samplesIn [i/o] pointer to time data. + * \param pGain [i ] pointer to gain to be applied to + * the time data. + * \param pGainPerSample [o ] pointer to the gain per sample to + * be applied to the time data in the limiter. + * \param gain_scale [i ] exponent to be applied to the time + * data. + * \param gain_delay [i ] delay[samples] with which the gains + * in pGain shall be applied (gain_delay <= nSamples). + * \param nSamples [i ] number of samples per frame. + * \param channels [i ] number of channels. + * \param stride [i ] channel stride of time data. + * \param limiterEnabled [i ] 1 if limiter is enabled, otherwise + * 0. + * + * \return exponent of time data + */ +INT applyDrcLevelNormalization(HANDLE_AAC_DRC hDrcInfo, PCM_DEC *samplesIn, + FIXP_DBL *pGain, FIXP_DBL *pGainPerSample, + const INT gain_scale, const UINT gain_delay, + const UINT nSamples, const UINT channels, + const UINT stride, const UINT limiterEnabled) { + UINT i; + INT additionalGain_scaling; + FIXP_DBL additionalGain; + + FDK_ASSERT(gain_delay <= nSamples); + + FIXP_DBL additionalGainSmoothState = hDrcInfo->additionalGainFilterState; + FIXP_DBL additionalGainSmoothState1 = hDrcInfo->additionalGainFilterState1; + + if (!gain_delay) { + additionalGain = pGain[0]; + + /* Apply the additional scaling gain_scale[0] that has no delay and no + * smoothing */ + additionalGain_scaling = + fMin(gain_scale, CntLeadingZeros(additionalGain) - 1); + additionalGain = scaleValue(additionalGain, additionalGain_scaling); + + /* if it's not possible to fully apply gain_scale to additionalGain, apply + * it to the input signal */ + additionalGain_scaling -= gain_scale; + + if (additionalGain_scaling) { + scaleValuesSaturate(samplesIn, channels * nSamples, + -additionalGain_scaling); + } + + if (limiterEnabled) { + FDK_ASSERT(pGainPerSample != NULL); + + for (i = 0; i < nSamples; i++) { + pGainPerSample[i] = additionalGain; + } + } else { + for (i = 0; i < channels * nSamples; i++) { + samplesIn[i] = FIXP_DBL2PCM_DEC(fMult(samplesIn[i], additionalGain)); + } + } + } else { + UINT inc; + FIXP_DBL additionalGainUnfiltered; + + inc = (stride == 1) ? channels : 1; + + for (i = 0; i < nSamples; i++) { + if (i < gain_delay) { + additionalGainUnfiltered = hDrcInfo->additionalGainPrev; + } else { + additionalGainUnfiltered = pGain[0]; + } + + /* Smooth additionalGain */ + + /* [b,a] = butter(1, 0.01) */ + static const FIXP_SGL b[] = {FL2FXCONST_SGL(0.015466 * 2.0), + FL2FXCONST_SGL(0.015466 * 2.0)}; + static const FIXP_SGL a[] = {(FIXP_SGL)MAXVAL_SGL, + FL2FXCONST_SGL(-0.96907)}; + + additionalGain = -fMult(additionalGainSmoothState, a[1]) + + fMultDiv2(additionalGainUnfiltered, b[0]) + + fMultDiv2(additionalGainSmoothState1, b[1]); + additionalGainSmoothState1 = additionalGainUnfiltered; + additionalGainSmoothState = additionalGain; + + /* Apply the additional scaling gain_scale[0] that has no delay and no + * smoothing */ + additionalGain_scaling = + fMin(gain_scale, CntLeadingZeros(additionalGain) - 1); + additionalGain = scaleValue(additionalGain, additionalGain_scaling); + + /* if it's not possible to fully apply gain_scale[0] to additionalGain, + * apply it to the input signal */ + additionalGain_scaling -= gain_scale; + + if (limiterEnabled) { + FDK_ASSERT(stride == 1); + FDK_ASSERT(pGainPerSample != NULL); + + if (additionalGain_scaling) { + scaleValuesSaturate(samplesIn, channels, -additionalGain_scaling); + } + + pGainPerSample[i] = additionalGain; + } else { + if (additionalGain_scaling) { + for (UINT k = 0; k < channels; k++) { + scaleValuesSaturate(&samplesIn[k * stride], 1, + -additionalGain_scaling); + } + } + + for (UINT k = 0; k < channels; k++) { + samplesIn[k * stride] = + FIXP_DBL2PCM_DEC(fMult(samplesIn[k * stride], additionalGain)); + } + } + + samplesIn += inc; + } + } + + hDrcInfo->additionalGainPrev = pGain[0]; + hDrcInfo->additionalGainFilterState = additionalGainSmoothState; + hDrcInfo->additionalGainFilterState1 = additionalGainSmoothState1; + + return (AACDEC_DRC_GAIN_SCALING); +} diff --git a/libAACdec/src/aacdec_drc.h b/libAACdec/src/aacdec_drc.h index 924ec6f..1873c5b 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -109,6 +109,11 @@ amm-info@iis.fraunhofer.de #include "channel.h" #include "FDK_bitstream.h" +#define AACDEC_DRC_GAIN_SCALING (11) /* Scaling of DRC gains */ +#define AACDEC_DRC_GAIN_INIT_VALUE \ + (FL2FXCONST_DBL( \ + 1.0f / (1 << AACDEC_DRC_GAIN_SCALING))) /* Init value for DRC gains */ + #define AACDEC_DRC_DFLT_EXPIRY_FRAMES \ (0) /* Default DRC data expiry time in AAC frames */ @@ -136,6 +141,8 @@ typedef enum { /** * \brief DRC module interface functions */ +void aacDecoder_drcReset(HANDLE_AAC_DRC self); + void aacDecoder_drcInit(HANDLE_AAC_DRC self); void aacDecoder_drcInitChannelData(CDrcChannelData *pDrcChannel); @@ -189,4 +196,45 @@ int aacDecoder_drcEpilog( void aacDecoder_drcGetInfo(HANDLE_AAC_DRC self, SCHAR *pPresMode, SCHAR *pProgRefLevel); +/** + * \brief Apply DRC Level Normalization. + * + * This function prepares/applies the gain values for the DRC Level + * Normalization and returns the exponent of the time data. The following two + * cases are handled: + * + * - Limiter enabled: + * The input data must be interleaved. + * One gain per sample is written to the buffer pGainPerSample. + * If necessary the time data is rescaled. + * + * - Limiter disabled: + * The input data can be interleaved or deinterleaved. + * The gain values are applied to the time data. + * If necessary the time data is rescaled. + * + * \param hDrcInfo [i/o] handle to drc data structure. + * \param samplesIn [i/o] pointer to time data. + * \param pGain [i ] pointer to gain to be applied to + * the time data. + * \param pGainPerSample [o ] pointer to the gain per sample to + * be applied to the time data in the limiter. + * \param gain_scale [i ] exponent to be applied to the time + * data. + * \param gain_delay [i ] delay[samples] with which the gains + * in pGain shall be applied (gain_delay <= nSamples). + * \param nSamples [i ] number of samples per frame. + * \param channels [i ] number of channels. + * \param stride [i ] channel stride of time data. + * \param limiterEnabled [i ] 1 if limiter is enabled, otherwise + * 0. + * + * \return exponent of time data + */ +INT applyDrcLevelNormalization(HANDLE_AAC_DRC hDrcInfo, PCM_DEC *samplesIn, + FIXP_DBL *pGain, FIXP_DBL *pGainPerSample, + const INT gain_scale, const UINT gain_delay, + const UINT nSamples, const UINT channels, + const UINT stride, const UINT limiterEnabled); + #endif /* AACDEC_DRC_H */ diff --git a/libAACdec/src/aacdec_drc_types.h b/libAACdec/src/aacdec_drc_types.h index 76c35d0..873bd60 100644 --- a/libAACdec/src/aacdec_drc_types.h +++ b/libAACdec/src/aacdec_drc_types.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -213,6 +213,13 @@ typedef struct { uniDrcPrecedence; /* Flag for signalling that uniDrc is active and takes precedence over legacy DRC */ + UCHAR applyExtGain; /* Flag is 1 if extGain has to be applied, otherwise 0. */ + + FIXP_DBL additionalGainPrev; /* Gain of previous frame to be applied to the + time data */ + FIXP_DBL additionalGainFilterState; /* Filter state for the gain smoothing */ + FIXP_DBL additionalGainFilterState1; /* Filter state for the gain smoothing */ + } CDrcInfo; typedef CDrcInfo *HANDLE_AAC_DRC; -- cgit v1.2.3 From e016635f0d3a5c7532b00711ce461f97a13f7bc2 Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Thu, 19 Dec 2019 17:28:15 +0100 Subject: Avoid decoder internal clipping by converting the whole audio sample data path from 16 to 32 bit data width (FDKdec v3.2.0). Bug: 149514474 Test: atest DecoderTestXheAac DecoderTestAacDrc Change-Id: I8a504ab709e42e27a61fe29840212953742283a5 --- documentation/aacDecoder.pdf | Bin 490978 -> 490978 bytes libAACdec/src/FDK_delay.cpp | 26 ++--- libAACdec/src/FDK_delay.h | 6 +- libAACdec/src/aac_rom.h | 3 +- libAACdec/src/aacdecoder.cpp | 49 ++++---- libAACdec/src/aacdecoder.h | 11 +- libAACdec/src/aacdecoder_lib.cpp | 232 +++++++++++++++++++++---------------- libAACdec/src/block.cpp | 12 +- libAACdec/src/block.h | 8 +- libAACdec/src/conceal.cpp | 42 ++++--- libAACdec/src/conceal.h | 4 +- libAACdec/src/conceal_types.h | 4 +- libAACdec/src/ldfiltbank.cpp | 35 +++--- libAACdec/src/ldfiltbank.h | 5 +- libAACdec/src/usacdec_lpd.cpp | 34 +++--- libAACdec/src/usacdec_lpd.h | 19 +-- libFDK/include/FDK_qmf_domain.h | 5 - libFDK/include/mdct.h | 14 +-- libFDK/src/FDK_core.cpp | 4 +- libFDK/src/FDK_qmf_domain.cpp | 12 +- libPCMutils/include/limiter.h | 36 ++---- libPCMutils/src/limiter.cpp | 116 ++++++++----------- libPCMutils/src/version.h | 4 +- libSACdec/include/sac_dec_lib.h | 13 ++- libSACdec/src/sac_dec.cpp | 23 ++-- libSACdec/src/sac_dec.h | 5 +- libSACdec/src/sac_dec_interface.h | 8 +- libSACdec/src/sac_dec_lib.cpp | 24 ++-- libSACdec/src/sac_process.cpp | 30 +++-- libSACdec/src/sac_qmf.cpp | 4 +- libSACdec/src/sac_qmf.h | 4 +- libSACdec/src/sac_reshapeBBEnv.cpp | 18 +-- libSACdec/src/sac_rom.h | 15 --- libSBRdec/include/sbrdecoder.h | 15 ++- libSBRdec/src/HFgen_preFlat.cpp | 4 +- libSBRdec/src/hbe.cpp | 2 +- libSBRdec/src/hbe.h | 10 +- libSBRdec/src/sbr_dec.cpp | 26 +++-- libSBRdec/src/sbr_dec.h | 11 +- libSBRdec/src/sbr_ram.h | 5 +- libSBRdec/src/sbrdecoder.cpp | 32 ++--- 41 files changed, 488 insertions(+), 442 deletions(-) (limited to 'libAACdec') diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf index d04a7d6..cc7cf41 100644 Binary files a/documentation/aacDecoder.pdf and b/documentation/aacDecoder.pdf differ diff --git a/libAACdec/src/FDK_delay.cpp b/libAACdec/src/FDK_delay.cpp index 0ab1a66..0cc869c 100644 --- a/libAACdec/src/FDK_delay.cpp +++ b/libAACdec/src/FDK_delay.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -113,7 +113,7 @@ INT FDK_Delay_Create(FDK_SignalDelay* data, const USHORT delay, if (delay > 0) { data->delay_line = - (INT_PCM*)FDKcalloc(num_channels * delay, sizeof(INT_PCM)); + (PCM_DEC*)FDKcalloc(num_channels * delay, sizeof(PCM_DEC)); if (data->delay_line == NULL) { return -1; } @@ -126,36 +126,36 @@ INT FDK_Delay_Create(FDK_SignalDelay* data, const USHORT delay, return 0; } -void FDK_Delay_Apply(FDK_SignalDelay* data, FIXP_PCM* time_buffer, +void FDK_Delay_Apply(FDK_SignalDelay* data, PCM_DEC* time_buffer, const UINT frame_length, const UCHAR channel) { FDK_ASSERT(data != NULL); if (data->delay > 0) { - C_ALLOC_SCRATCH_START(tmp, FIXP_PCM, MAX_FRAME_LENGTH) + C_ALLOC_SCRATCH_START(tmp, PCM_DEC, MAX_FRAME_LENGTH) FDK_ASSERT(frame_length <= MAX_FRAME_LENGTH); FDK_ASSERT(channel < data->num_channels); FDK_ASSERT(time_buffer != NULL); if (frame_length >= data->delay) { FDKmemcpy(tmp, &time_buffer[frame_length - data->delay], - data->delay * sizeof(FIXP_PCM)); + data->delay * sizeof(PCM_DEC)); FDKmemmove(&time_buffer[data->delay], &time_buffer[0], - (frame_length - data->delay) * sizeof(FIXP_PCM)); + (frame_length - data->delay) * sizeof(PCM_DEC)); FDKmemcpy(&time_buffer[0], &data->delay_line[channel * data->delay], - data->delay * sizeof(FIXP_PCM)); + data->delay * sizeof(PCM_DEC)); FDKmemcpy(&data->delay_line[channel * data->delay], tmp, - data->delay * sizeof(FIXP_PCM)); + data->delay * sizeof(PCM_DEC)); } else { - FDKmemcpy(tmp, &time_buffer[0], frame_length * sizeof(FIXP_PCM)); + FDKmemcpy(tmp, &time_buffer[0], frame_length * sizeof(PCM_DEC)); FDKmemcpy(&time_buffer[0], &data->delay_line[channel * data->delay], - frame_length * sizeof(FIXP_PCM)); + frame_length * sizeof(PCM_DEC)); FDKmemcpy(&data->delay_line[channel * data->delay], &data->delay_line[channel * data->delay + frame_length], - (data->delay - frame_length) * sizeof(FIXP_PCM)); + (data->delay - frame_length) * sizeof(PCM_DEC)); FDKmemcpy(&data->delay_line[channel * data->delay + (data->delay - frame_length)], - tmp, frame_length * sizeof(FIXP_PCM)); + tmp, frame_length * sizeof(PCM_DEC)); } - C_ALLOC_SCRATCH_END(tmp, FIXP_PCM, MAX_FRAME_LENGTH) + C_ALLOC_SCRATCH_END(tmp, PCM_DEC, MAX_FRAME_LENGTH) } return; diff --git a/libAACdec/src/FDK_delay.h b/libAACdec/src/FDK_delay.h index f89c3a2..6317d9d 100644 --- a/libAACdec/src/FDK_delay.h +++ b/libAACdec/src/FDK_delay.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -109,7 +109,7 @@ amm-info@iis.fraunhofer.de * Structure representing one delay element for multiple channels. */ typedef struct { - INT_PCM* delay_line; /*!< Pointer which stores allocated delay line. */ + PCM_DEC* delay_line; /*!< Pointer which stores allocated delay line. */ USHORT delay; /*!< Delay required in samples (per channel). */ UCHAR num_channels; /*!< Number of channels to delay. */ } FDK_SignalDelay; @@ -137,7 +137,7 @@ INT FDK_Delay_Create(FDK_SignalDelay* data, const USHORT delay, * * \return void */ -void FDK_Delay_Apply(FDK_SignalDelay* data, FIXP_PCM* time_buffer, +void FDK_Delay_Apply(FDK_SignalDelay* data, PCM_DEC* time_buffer, const UINT frame_length, const UCHAR channel); /** diff --git a/libAACdec/src/aac_rom.h b/libAACdec/src/aac_rom.h index ffaf951..7a1597c 100644 --- a/libAACdec/src/aac_rom.h +++ b/libAACdec/src/aac_rom.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -108,6 +108,7 @@ amm-info@iis.fraunhofer.de #include "aacdec_hcr_types.h" #include "aacdec_hcrs.h" +#define PCM_AAC LONG #define PCM_DEC FIXP_DBL #define MAXVAL_PCM_DEC MAXVAL_DBL #define MINVAL_PCM_DEC MINVAL_DBL diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index f747b2d..6a0254d 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -1281,6 +1281,7 @@ LINKSPEC_CPP HANDLE_AACDECODER CAacDecoder_Open( /* Set default frame delay */ aacDecoder_drcSetParam(self->hDrcInfo, DRC_BS_DELAY, CConcealment_GetDelay(&self->concealCommonData)); + self->workBufferCore1 = (FIXP_DBL *)GetWorkBufferCore1(); self->workBufferCore2 = GetWorkBufferCore2(); if (self->workBufferCore2 == NULL) goto bail; @@ -1456,6 +1457,10 @@ LINKSPEC_CPP void CAacDecoder_Close(HANDLE_AACDECODER self) { FreeDrcInfo(&self->hDrcInfo); } + if (self->workBufferCore1 != NULL) { + FreeWorkBufferCore1((CWorkBufferCore1 **)&self->workBufferCore1); + } + /* Free WorkBufferCore2 */ if (self->workBufferCore2 != NULL) { FreeWorkBufferCore2(&self->workBufferCore2); @@ -1493,6 +1498,8 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, UCHAR downscaleFactor = self->downscaleFactor; UCHAR downscaleFactorInBS = self->downscaleFactorInBS; + self->aacOutDataHeadroom = (3); + // set profile and check for supported aot // leave profile on default (=-1) for all other supported MPEG-4 aot's except // aot=2 (=AAC-LC) @@ -2394,7 +2401,7 @@ bail: } LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( - HANDLE_AACDECODER self, const UINT flags, FIXP_PCM *pTimeData, + HANDLE_AACDECODER self, const UINT flags, PCM_DEC *pTimeData, const INT timeDataSize, const int timeDataChannelOffset) { AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; @@ -3170,10 +3177,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( FDKmemcpy(drcChMap, self->chMapping, (8) * sizeof(UCHAR)); } - /* Turn on/off DRC modules level normalization in digital domain depending - * on the limiter status. */ - aacDecoder_drcSetParam(self->hDrcInfo, APPLY_NORMALIZATION, - (self->limiterEnableCurr) ? 0 : 1); + /* Turn off DRC modules level normalization in digital domain. */ + aacDecoder_drcSetParam(self->hDrcInfo, APPLY_NORMALIZATION, 0); /* deactivate legacy DRC in case uniDrc is active, i.e. uniDrc payload is * present and one of DRC or Loudness Normalization is switched on */ @@ -3325,9 +3330,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( &pAacDecoderStaticChannelInfo->drcData); } } + /* The DRC module demands to be called with the gain field holding the * gain scale. */ - self->extGain[0] = (FIXP_DBL)TDL_GAIN_SCALING; + self->extGain[0] = (FIXP_DBL)AACDEC_DRC_GAIN_SCALING; + /* DRC processing */ aacDecoder_drcApply( self->hDrcInfo, self->hSbrDecoder, pAacDecoderChannelInfo, @@ -3343,7 +3350,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( if (self->flushStatus && (self->flushCnt > 0) && !(flags & AACDEC_CONCEAL)) { FDKmemclear(pTimeData + offset, - sizeof(FIXP_PCM) * self->streamInfo.aacSamplesPerFrame); + sizeof(PCM_DEC) * self->streamInfo.aacSamplesPerFrame); } else switch (pAacDecoderChannelInfo->renderMode) { case AACDEC_RENDER_IMDCT: @@ -3355,7 +3362,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( !frameOk_butConceal), pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1 ->mdctOutTemp, - self->elFlags[el], elCh); + self->aacOutDataHeadroom, self->elFlags[el], elCh); self->extGainDelay = self->streamInfo.aacSamplesPerFrame; break; @@ -3376,7 +3383,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( &self->samplingRateInfo[streamIndex], (self->frameOK && !(flags & AACDEC_CONCEAL) && !frameOk_butConceal), - flags, self->flags[streamIndex]); + self->aacOutDataHeadroom, flags, self->flags[streamIndex]); self->extGainDelay = self->streamInfo.aacSamplesPerFrame; break; @@ -3388,7 +3395,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( if (!CConceal_TDFading_Applied[c]) { CConceal_TDFading_Applied[c] = CConcealment_TDFading( self->streamInfo.aacSamplesPerFrame, - &self->pAacDecoderStaticChannelInfo[c], pTimeData + offset, 0); + &self->pAacDecoderStaticChannelInfo[c], self->aacOutDataHeadroom, + pTimeData + offset, 0); if (c + 1 < (8) && c < aacChannels - 1) { /* update next TDNoise Seed to avoid muting in case of Parametric * Stereo */ @@ -3409,27 +3417,18 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( * LR) */ if ((aacChannels == 2) && bsPseudoLr) { int i, offset2; - const FIXP_SGL invSqrt2 = - FL2FXCONST_SGL(0.353553390593273f); /* scaled by -1 */ - FIXP_PCM *pTD = pTimeData; + const FIXP_SGL invSqrt2 = FL2FXCONST_SGL(0.707106781186547f); + PCM_DEC *pTD = pTimeData; offset2 = timeDataChannelOffset; for (i = 0; i < self->streamInfo.aacSamplesPerFrame; i++) { - FIXP_DBL L = FX_PCM2FX_DBL(pTD[0]); - FIXP_DBL R = FX_PCM2FX_DBL(pTD[offset2]); + FIXP_DBL L = PCM_DEC2FIXP_DBL(pTD[0]); + FIXP_DBL R = PCM_DEC2FIXP_DBL(pTD[offset2]); L = fMult(L, invSqrt2); R = fMult(R, invSqrt2); -#if (SAMPLE_BITS == 16) - pTD[0] = (FIXP_SGL)SATURATE_RIGHT_SHIFT(L + R + (FIXP_DBL)(1 << 14), - 15, FRACT_BITS); - pTD[offset2] = (FIXP_SGL)SATURATE_RIGHT_SHIFT( - L - R + (FIXP_DBL)(1 << 14), 15, FRACT_BITS); -#else - pTD[0] = SATURATE_LEFT_SHIFT(FX_DBL2FX_PCM(L + R), 1, DFRACT_BITS); - pTD[offset2] = - SATURATE_LEFT_SHIFT(FX_DBL2FX_PCM(L - R), 1, DFRACT_BITS); -#endif + pTD[0] = L + R; + pTD[offset2] = L - R; pTD++; } } diff --git a/libAACdec/src/aacdecoder.h b/libAACdec/src/aacdecoder.h index 20f4c45..bd1f38f 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -191,6 +191,9 @@ struct AAC_DECODER_INSTANCE { INT outputInterleaved; /*!< PCM output format (interleaved/none interleaved). */ + INT aacOutDataHeadroom; /*!< Headroom of the output time signal to prevent + clipping */ + HANDLE_TRANSPORTDEC hInput; /*!< Transport layer handle. */ SamplingRateInfo @@ -235,6 +238,7 @@ struct AAC_DECODER_INSTANCE { CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[(8)]; /*!< Persistent channel memory */ + FIXP_DBL *workBufferCore1; FIXP_DBL *workBufferCore2; PCM_DEC *pTimeData2; INT timeData2Size; @@ -311,11 +315,10 @@ This structure is allocated once for each CPE. */ UCHAR limiterEnableUser; /*!< The limiter configuration requested by the library user */ UCHAR limiterEnableCurr; /*!< The current limiter configuration. */ + FIXP_DBL extGain[1]; /*!< Gain that must be applied to the output signal. */ UINT extGainDelay; /*!< Delay that must be accounted for extGain. */ - INT_PCM pcmOutputBuffer[(8) * (1024 * 2)]; - HANDLE_DRC_DECODER hUniDrcDecoder; UCHAR multibandDrcPresent; UCHAR numTimeSlots; @@ -427,7 +430,7 @@ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, \return error status */ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_DecodeFrame( - HANDLE_AACDECODER self, const UINT flags, FIXP_PCM *pTimeData, + HANDLE_AACDECODER self, const UINT flags, PCM_DEC *pTimeData, const INT timeDataSize, const int timeDataChannelOffset); /* Free config dependent AAC memory */ diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp index 2ba0e86..f5ce7e0 100644 --- a/libAACdec/src/aacdecoder_lib.cpp +++ b/libAACdec/src/aacdecoder_lib.cpp @@ -119,8 +119,8 @@ amm-info@iis.fraunhofer.de /* Decoder library info */ #define AACDECODER_LIB_VL0 3 -#define AACDECODER_LIB_VL1 1 -#define AACDECODER_LIB_VL2 3 +#define AACDECODER_LIB_VL1 2 +#define AACDECODER_LIB_VL2 0 #define AACDECODER_LIB_TITLE "AAC Decoder Lib" #ifdef __ANDROID__ #define AACDECODER_LIB_BUILD_DATE "" @@ -1131,35 +1131,31 @@ static INT aacDecoder_EstimateNumberOfLostFrames(HANDLE_AACDECODER self) { return n; } -LINKSPEC_CPP AAC_DECODER_ERROR -aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, - const INT timeDataSize_extern, const UINT flags) { +LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self, + INT_PCM *pTimeData, + const INT timeDataSize, + const UINT flags) { AAC_DECODER_ERROR ErrorStatus; INT layer; INT nBits; + INT timeData2Size; + INT timeData3Size; + INT timeDataHeadroom; HANDLE_FDK_BITSTREAM hBs; int fTpInterruption = 0; /* Transport originated interruption detection. */ int fTpConceal = 0; /* Transport originated concealment. */ - INT_PCM *pTimeData = NULL; - INT timeDataSize = 0; UINT accessUnit = 0; UINT numAccessUnits = 1; UINT numPrerollAU = 0; - int fEndAuNotAdjusted = 0; /* The end of the access unit was not adjusted */ - int applyCrossfade = 1; /* flag indicates if flushing was possible */ - FIXP_PCM *pTimeDataFixpPcm; /* Signal buffer for decoding process before PCM - processing */ - INT timeDataFixpPcmSize; - PCM_DEC *pTimeDataPcmPost; /* Signal buffer for PCM post-processing */ - INT timeDataPcmPostSize; + int fEndAuNotAdjusted = 0; /* The end of the access unit was not adjusted */ + int applyCrossfade = 1; /* flag indicates if flushing was possible */ + PCM_DEC *pTimeData2; + PCM_AAC *pTimeData3; if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } - pTimeData = self->pcmOutputBuffer; - timeDataSize = sizeof(self->pcmOutputBuffer) / sizeof(*self->pcmOutputBuffer); - if (flags & AACDEC_INTR) { self->streamInfo.numLostAccessUnits = 0; } @@ -1315,19 +1311,23 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, /* Use limiter configuration as requested. */ self->limiterEnableCurr = self->limiterEnableUser; } - /* reset limiter gain on a per frame basis */ - self->extGain[0] = FL2FXCONST_DBL(1.0f / (float)(1 << TDL_GAIN_SCALING)); - pTimeDataFixpPcm = pTimeData; - timeDataFixpPcmSize = timeDataSize; + /* reset DRC level normalization gain on a per frame basis */ + self->extGain[0] = AACDEC_DRC_GAIN_INIT_VALUE; + + pTimeData2 = self->pTimeData2; + timeData2Size = self->timeData2Size / sizeof(PCM_DEC); + pTimeData3 = (PCM_AAC *)self->pTimeData2; + timeData3Size = self->timeData2Size / sizeof(PCM_AAC); ErrorStatus = CAacDecoder_DecodeFrame( self, flags | (fTpConceal ? AACDEC_CONCEAL : 0) | ((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH : 0), - pTimeDataFixpPcm + 0, timeDataFixpPcmSize, - self->streamInfo.aacSamplesPerFrame + 0); + pTimeData2 + 0, timeData2Size, self->streamInfo.aacSamplesPerFrame + 0); + + timeDataHeadroom = self->aacOutDataHeadroom; /* if flushing for USAC DASH IPF was not possible go on with decoding * preroll */ @@ -1352,7 +1352,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, } } - /* If the current pTimeDataFixpPcm does not contain a valid signal, there + /* If the current pTimeData2 does not contain a valid signal, there * nothing else we can do, so bail. */ if (!IS_OUTPUT_VALID(ErrorStatus)) { goto bail; @@ -1366,10 +1366,10 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, self->streamInfo.numChannels = self->streamInfo.aacNumChannels; { - FDK_Delay_Apply(&self->usacResidualDelay, - pTimeDataFixpPcm + - 1 * (self->streamInfo.aacSamplesPerFrame + 0) + 0, - self->streamInfo.frameSize, 0); + FDK_Delay_Apply( + &self->usacResidualDelay, + pTimeData2 + 1 * (self->streamInfo.aacSamplesPerFrame + 0) + 0, + self->streamInfo.frameSize, 0); } /* Setting of internal MPS state; may be reset in CAacDecoder_SyncQmfMode @@ -1416,8 +1416,6 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, } } - self->qmfDomain.globalConf.TDinput = pTimeData; - switch (FDK_QmfDomain_Configure(&self->qmfDomain)) { default: case QMF_DOMAIN_INIT_ERROR: @@ -1474,18 +1472,18 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, (self->mpsEnableCurr) ? 2 : 0); - INT_PCM *input; - input = (INT_PCM *)self->workBufferCore2; - FDKmemcpy(input, pTimeData, - sizeof(INT_PCM) * (self->streamInfo.numChannels) * + PCM_AAC *input; + input = (PCM_AAC *)self->workBufferCore2; + FDKmemcpy(input, pTimeData3, + sizeof(PCM_AAC) * (self->streamInfo.numChannels) * (self->streamInfo.frameSize)); /* apply SBR processing */ - sbrError = sbrDecoder_Apply(self->hSbrDecoder, input, pTimeData, - timeDataSize, &self->streamInfo.numChannels, - &self->streamInfo.sampleRate, - &self->mapDescr, self->chMapIndex, - self->frameOK, &self->psPossible); + sbrError = sbrDecoder_Apply( + self->hSbrDecoder, input, pTimeData3, timeData3Size, + &self->streamInfo.numChannels, &self->streamInfo.sampleRate, + &self->mapDescr, self->chMapIndex, self->frameOK, &self->psPossible, + self->aacOutDataHeadroom, &timeDataHeadroom); if (sbrError == SBRDEC_OK) { /* Update data in streaminfo structure. Assume that the SBR upsampling @@ -1564,10 +1562,11 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, if (err == 0) { err = mpegSurroundDecoder_Apply( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, - (INT_PCM *)self->workBufferCore2, pTimeData, timeDataSize, + (PCM_AAC *)self->workBufferCore2, pTimeData3, timeData3Size, self->streamInfo.aacSamplesPerFrame, &nChannels, &frameSize, self->streamInfo.sampleRate, self->streamInfo.aot, - self->channelType, self->channelIndices, &self->mapDescr); + self->channelType, self->channelIndices, &self->mapDescr, + self->aacOutDataHeadroom, &timeDataHeadroom); } if (err == MPS_OUTPUT_BUFFER_TOO_SMALL) { @@ -1590,8 +1589,8 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, self->streamInfo.frameSize = self->mpsFrameSizeLast; /* ... and clear output buffer so that potentially corrupted data does * not reach the framework. */ - FDKmemclear(pTimeData, self->mpsOutChannelsLast * - self->mpsFrameSizeLast * sizeof(INT_PCM)); + FDKmemclear(pTimeData3, self->mpsOutChannelsLast * + self->mpsFrameSizeLast * sizeof(PCM_AAC)); /* Additionally proclaim that this frame had errors during decoding. */ ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; @@ -1612,11 +1611,11 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, 1); /* apply SBR processing */ - sbrError = sbrDecoder_Apply(self->hSbrDecoder, pTimeData, pTimeData, - timeDataSize, &self->streamInfo.numChannels, - &self->streamInfo.sampleRate, - &self->mapDescr, self->chMapIndex, - self->frameOK, &self->psPossible); + sbrError = sbrDecoder_Apply( + self->hSbrDecoder, pTimeData3, pTimeData3, timeData3Size, + &self->streamInfo.numChannels, &self->streamInfo.sampleRate, + &self->mapDescr, self->chMapIndex, self->frameOK, &self->psPossible, + self->aacOutDataHeadroom, &timeDataHeadroom); if (sbrError == SBRDEC_OK) { /* Update data in streaminfo structure. Assume that the SBR upsampling @@ -1644,17 +1643,15 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, } } - /* Use dedicated memory for PCM postprocessing */ - pTimeDataPcmPost = self->pTimeData2; - timeDataPcmPostSize = self->timeData2Size; - { - const int size = - self->streamInfo.frameSize * self->streamInfo.numChannels; - FDK_ASSERT(timeDataPcmPostSize >= size); - for (int i = 0; i < size; i++) { - pTimeDataPcmPost[i] = - (PCM_DEC)FX_PCM2PCM_DEC(pTimeData[i]) >> PCM_OUT_HEADROOM; + if ((INT)PCM_OUT_HEADROOM != timeDataHeadroom) { + for (int i = ((self->streamInfo.frameSize * + self->streamInfo.numChannels) - + 1); + i >= 0; i--) { + pTimeData2[i] = + (PCM_DEC)pTimeData3[i] >> (PCM_OUT_HEADROOM - timeDataHeadroom); + } } } @@ -1709,22 +1706,21 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, if ((self->streamInfo.numChannels > 1) && (0 || (self->sbrEnabled) || (self->mpsEnableCurr))) { /* interleaving/deinterleaving is performed on upper part of - * pTimeDataPcmPost. Check if this buffer is large enough. */ - if (timeDataPcmPostSize < - (INT)(2 * self->streamInfo.numChannels * - self->streamInfo.frameSize * sizeof(PCM_DEC))) { + * pTimeData2. Check if this buffer is large enough. */ + if (timeData2Size < (INT)(2 * self->streamInfo.numChannels * + self->streamInfo.frameSize)) { ErrorStatus = AAC_DEC_UNKNOWN; goto bail; } needsDeinterleaving = 1; drcWorkBuffer = - (FIXP_DBL *)pTimeDataPcmPost + + (FIXP_DBL *)pTimeData2 + self->streamInfo.numChannels * self->streamInfo.frameSize; FDK_deinterleave( - pTimeDataPcmPost, drcWorkBuffer, self->streamInfo.numChannels, + pTimeData2, drcWorkBuffer, self->streamInfo.numChannels, self->streamInfo.frameSize, self->streamInfo.frameSize); } else { - drcWorkBuffer = (FIXP_DBL *)pTimeDataPcmPost; + drcWorkBuffer = pTimeData2; } /* prepare Loudness Normalisation gain */ @@ -1759,7 +1755,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, if (needsDeinterleaving) { FDK_interleave( - drcWorkBuffer, pTimeDataPcmPost, self->streamInfo.numChannels, + drcWorkBuffer, pTimeData2, self->streamInfo.numChannels, self->streamInfo.frameSize, self->streamInfo.frameSize); } } @@ -1799,6 +1795,9 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, if (self->streamInfo.extAot != AOT_AAC_SLS) { INT pcmLimiterScale = 0; + INT interleaved = 0; + interleaved |= (self->sbrEnabled) ? 1 : 0; + interleaved |= (self->mpsEnableCurr) ? 1 : 0; PCMDMX_ERROR dmxErr = PCMDMX_OK; if ((flags & AACDEC_INTR) && (accessUnit == 0)) { /* delete data from the past (e.g. mixdown coeficients) */ @@ -1811,17 +1810,12 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, } } - INT interleaved = 0; - interleaved |= (self->sbrEnabled) ? 1 : 0; - interleaved |= (self->mpsEnableCurr) ? 1 : 0; - /* do PCM post processing */ - dmxErr = pcmDmx_ApplyFrame( - self->hPcmUtils, pTimeDataPcmPost, timeDataFixpPcmSize, - self->streamInfo.frameSize, &self->streamInfo.numChannels, - interleaved, self->channelType, self->channelIndices, - &self->mapDescr, - (self->limiterEnableCurr) ? &pcmLimiterScale : NULL); + dmxErr = pcmDmx_ApplyFrame(self->hPcmUtils, pTimeData2, timeData2Size, + self->streamInfo.frameSize, + &self->streamInfo.numChannels, interleaved, + self->channelType, self->channelIndices, + &self->mapDescr, &pcmLimiterScale); if (dmxErr == PCMDMX_OUTPUT_BUFFER_TOO_SMALL) { ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL; goto bail; @@ -1833,13 +1827,35 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, 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; @@ -1848,44 +1864,72 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern, /* Set actual signal parameters */ pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels); pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate); - pcmLimiterScale += PCM_OUT_HEADROOM; if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) || (self->mpsEnableCurr)) { - pInterleaveBuffer = (PCM_LIM *)pTimeDataPcmPost; + pInterleaveBuffer = (PCM_LIM *)pTimeData2; } else { - pInterleaveBuffer = (PCM_LIM *)pTimeData; + pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2; + /* applyLimiter requests for interleaved data */ /* Interleave ouput buffer */ - FDK_interleave(pTimeDataPcmPost, pInterleaveBuffer, + 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, - self->extGain, &pcmLimiterScale, 1, - self->extGainDelay, self->streamInfo.frameSize); + 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, pTimeDataPcmPost, + pTimeData, pTimeData2, self->streamInfo.frameSize * self->streamInfo.numChannels, - PCM_OUT_HEADROOM); + pcmLimiterScale); } else { scaleValuesSaturate( - (INT_PCM *)self->workBufferCore2, pTimeDataPcmPost, + (INT_PCM *)self->workBufferCore2, pTimeData2, self->streamInfo.frameSize * self->streamInfo.numChannels, - PCM_OUT_HEADROOM); + pcmLimiterScale); /* Interleave ouput buffer */ FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData, self->streamInfo.numChannels, @@ -1981,20 +2025,8 @@ bail: ErrorStatus = AAC_DEC_UNKNOWN; } - /* Check whether external output buffer is large enough. */ - if (timeDataSize_extern < - self->streamInfo.numChannels * self->streamInfo.frameSize) { - ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL; - } - - /* Update external output buffer. */ - if (IS_OUTPUT_VALID(ErrorStatus)) { - FDKmemcpy(pTimeData_extern, pTimeData, - self->streamInfo.numChannels * self->streamInfo.frameSize * - sizeof(*pTimeData)); - } else { - FDKmemclear(pTimeData_extern, - timeDataSize_extern * sizeof(*pTimeData_extern)); + if (!IS_OUTPUT_VALID(ErrorStatus)) { + FDKmemclear(pTimeData, timeDataSize * sizeof(*pTimeData)); } return ErrorStatus; diff --git a/libAACdec/src/block.cpp b/libAACdec/src/block.cpp index b3d09a6..0bca577 100644 --- a/libAACdec/src/block.cpp +++ b/libAACdec/src/block.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1015,9 +1015,9 @@ FIXP_DBL get_gain(const FIXP_DBL *x, const FIXP_DBL *y, int n) { void CBlock_FrequencyToTime( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[], + CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[], const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1, - UINT elFlags, INT elCh) { + const INT aacOutDataHeadroom, UINT elFlags, INT elCh) { int fr, fl, tl, nSpec; #if defined(FDK_ASSERT_ENABLE) @@ -1213,6 +1213,7 @@ void CBlock_FrequencyToTime( bass_pf_1sf_delay(p2_synth, pitch, pit_gain, frameLen, (LpdSfd + 2) * L_SUBFR + BPF_SFD * L_SUBFR, frameLen - (LpdSfd + 4) * L_SUBFR, outSamples, + aacOutDataHeadroom, pAacDecoderStaticChannelInfo->mem_bpf); } @@ -1236,7 +1237,8 @@ void CBlock_FrequencyToTime( ? MLT_FLAG_CURR_ALIAS_SYMMETRY : 0); - scaleValuesSaturate(outSamples, tmp, frameLen, MDCT_OUT_HEADROOM); + scaleValuesSaturate(outSamples, tmp, frameLen, + MDCT_OUT_HEADROOM - aacOutDataHeadroom); } } @@ -1251,7 +1253,7 @@ void CBlock_FrequencyToTime( #include "ldfiltbank.h" void CBlock_FrequencyToTimeLowDelay( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[], + CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[], const short frameLen) { InvMdctTransformLowDelay_fdk( SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), diff --git a/libAACdec/src/block.h b/libAACdec/src/block.h index f0f56cd..f5118a2 100644 --- a/libAACdec/src/block.h +++ b/libAACdec/src/block.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -218,16 +218,16 @@ void ApplyTools(CAacDecoderChannelInfo *pAacDecoderChannelInfo[], */ void CBlock_FrequencyToTime( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[], + CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[], const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1, - UINT elFlags, INT elCh); + const INT aacOutDataHeadroom, UINT elFlags, INT elCh); /** * \brief Transform double lapped MDCT (AAC-ELD) spectral data into time domain. */ void CBlock_FrequencyToTimeLowDelay( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[], + CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[], const short frameLen); AAC_DECODER_ERROR CBlock_InverseQuantizeSpectralData( diff --git a/libAACdec/src/conceal.cpp b/libAACdec/src/conceal.cpp index 379e63a..0939bb5 100644 --- a/libAACdec/src/conceal.cpp +++ b/libAACdec/src/conceal.cpp @@ -226,7 +226,7 @@ static void CConcealment_ApplyRandomSign(int iRandomPhase, FIXP_DBL *spec, /* TimeDomainFading */ static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart, - FIXP_DBL fadeStop, FIXP_PCM *pcmdata); + FIXP_DBL fadeStop, PCM_DEC *pcmdata); static void CConcealment_TDFadeFillFadingStations(FIXP_DBL *fadingStations, int *fadingSteps, FIXP_DBL fadeStop, @@ -242,7 +242,9 @@ static int CConcealment_ApplyFadeOut( static int CConcealment_TDNoise_Random(ULONG *seed); static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo, - const int len, FIXP_PCM *const pcmdata); + const int len, + const INT aacOutDataHeadroom, + PCM_DEC *const pcmdata); static BLOCK_TYPE CConcealment_GetWinSeq(int prevWinSeq) { BLOCK_TYPE newWinSeq = BLOCK_LONG; @@ -1844,7 +1846,7 @@ Target fading level is determined by fading index cntFadeFrames. INT CConcealment_TDFading( int len, CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo, - FIXP_PCM *pcmdata, FIXP_PCM *pcmdata_1) { + const INT aacOutDataHeadroom, PCM_DEC *pcmdata, PCM_DEC *pcmdata_1) { /* Do the fading in Time domain based on concealment states and core mode */ @@ -1957,7 +1959,8 @@ INT CConcealment_TDFading( start += len; } } - CConcealment_TDNoise_Apply(pConcealmentInfo, len, pcmdata); + CConcealment_TDNoise_Apply(pConcealmentInfo, len, aacOutDataHeadroom, + pcmdata); /* Save end-of-frame attenuation and fading type */ pConcealmentInfo->lastFadingType = fadingType; @@ -1969,12 +1972,11 @@ INT CConcealment_TDFading( /* attenuate pcmdata in Time Domain Fading process */ static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart, - FIXP_DBL fadeStop, FIXP_PCM *pcmdata) { + FIXP_DBL fadeStop, PCM_DEC *pcmdata) { int i; FIXP_DBL dStep; FIXP_DBL dGain; FIXP_DBL dGain_apply; - int bitshift = (DFRACT_BITS - SAMPLE_BITS); /* set start energy */ dGain = fadeStart; @@ -1987,7 +1989,7 @@ static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart, */ dGain_apply = fMax((FIXP_DBL)0, dGain); /* finally, attenuate samples */ - pcmdata[i] = (FIXP_PCM)((fMult(pcmdata[i], (dGain_apply))) >> bitshift); + pcmdata[i] = FIXP_DBL2PCM_DEC(fMult(pcmdata[i], dGain_apply)); } } @@ -2050,9 +2052,11 @@ static int CConcealment_TDNoise_Random(ULONG *seed) { } static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo, - const int len, FIXP_PCM *const pcmdata) { - FIXP_PCM *states = pConcealmentInfo->TDNoiseStates; - FIXP_PCM noiseVal; + const int len, + const INT aacOutDataHeadroom, + PCM_DEC *const pcmdata) { + PCM_DEC *states = pConcealmentInfo->TDNoiseStates; + PCM_DEC noiseVal; FIXP_DBL noiseValLong; FIXP_SGL *coef = pConcealmentInfo->TDNoiseCoef; FIXP_DBL TDNoiseAtt; @@ -2070,18 +2074,20 @@ static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo, /* create filtered noise */ states[2] = states[1]; states[1] = states[0]; - states[0] = ((FIXP_PCM)CConcealment_TDNoise_Random(&seed)); + states[0] = + FIXP_DBL2PCM_DEC((FIXP_DBL)CConcealment_TDNoise_Random(&seed)); noiseValLong = fMult(states[0], coef[0]) + fMult(states[1], coef[1]) + fMult(states[2], coef[2]); - noiseVal = FX_DBL2FX_PCM(fMult(noiseValLong, TDNoiseAtt)); + noiseVal = FIXP_DBL2PCM_DEC(fMult(noiseValLong, TDNoiseAtt) >> + aacOutDataHeadroom); /* add filtered noise - check for clipping, before */ - if (noiseVal > (FIXP_PCM)0 && - pcmdata[ii] > (FIXP_PCM)MAXVAL_FIXP_PCM - noiseVal) { - noiseVal = noiseVal * (FIXP_PCM)-1; - } else if (noiseVal < (FIXP_PCM)0 && - pcmdata[ii] < (FIXP_PCM)MINVAL_FIXP_PCM - noiseVal) { - noiseVal = noiseVal * (FIXP_PCM)-1; + if (noiseVal > (PCM_DEC)0 && + pcmdata[ii] > (PCM_DEC)MAXVAL_PCM_DEC - noiseVal) { + noiseVal = noiseVal * (PCM_DEC)-1; + } else if (noiseVal < (PCM_DEC)0 && + pcmdata[ii] < (PCM_DEC)MINVAL_PCM_DEC - noiseVal) { + noiseVal = noiseVal * (PCM_DEC)-1; } pcmdata[ii] += noiseVal; diff --git a/libAACdec/src/conceal.h b/libAACdec/src/conceal.h index e01a796..0c002a5 100644 --- a/libAACdec/src/conceal.h +++ b/libAACdec/src/conceal.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -147,6 +147,6 @@ int CConcealment_GetLastFrameOk(CConcealmentInfo *hConcealmentInfo, INT CConcealment_TDFading( int len, CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo, - FIXP_PCM *pcmdata, FIXP_PCM *pcmdata_1); + const INT aacOutDataHeadroom, PCM_DEC *pcmdata, PCM_DEC *pcmdata_1); #endif /* #ifndef CONCEAL_H */ diff --git a/libAACdec/src/conceal_types.h b/libAACdec/src/conceal_types.h index d90374e..36e7dec 100644 --- a/libAACdec/src/conceal_types.h +++ b/libAACdec/src/conceal_types.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -194,7 +194,7 @@ typedef struct { FIXP_DBL last_tcx_gain; INT last_tcx_gain_e; ULONG TDNoiseSeed; - FIXP_PCM TDNoiseStates[3]; + PCM_DEC TDNoiseStates[3]; FIXP_SGL TDNoiseCoef[3]; FIXP_SGL TDNoiseAtt; diff --git a/libAACdec/src/ldfiltbank.cpp b/libAACdec/src/ldfiltbank.cpp index 1898557..13e61a5 100644 --- a/libAACdec/src/ldfiltbank.cpp +++ b/libAACdec/src/ldfiltbank.cpp @@ -112,17 +112,20 @@ amm-info@iis.fraunhofer.de #if defined(__arm__) #endif -static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb, +static void multE2_DinvF_fdk(PCM_DEC *output, FIXP_DBL *x, const FIXP_WTB *fb, FIXP_DBL *z, const int N) { int i; - /* scale for FIXP_DBL -> INT_PCM conversion. */ - const int scale = (DFRACT_BITS - SAMPLE_BITS) - LDFB_HEADROOM; -#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0) + /* scale for FIXP_DBL -> PCM_DEC conversion: */ + const int scale = (DFRACT_BITS - PCM_OUT_BITS) - LDFB_HEADROOM + (3); + +#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0) FIXP_DBL rnd_val_wts0 = (FIXP_DBL)0; FIXP_DBL rnd_val_wts1 = (FIXP_DBL)0; +#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - WTS0 - 1) > 0) if (-WTS0 - 1 + scale) rnd_val_wts0 = (FIXP_DBL)(1 << (-WTS0 - 1 + scale - 1)); +#endif if (-WTS1 - 1 + scale) rnd_val_wts1 = (FIXP_DBL)(1 << (-WTS1 - 1 + scale - 1)); #endif @@ -141,16 +144,16 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb, tmp = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) + fMultDiv2(z[i], fb[N + N / 2 + i])); -#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0) +#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0) FDK_ASSERT((-WTS1 - 1 + scale) >= 0); FDK_ASSERT(tmp <= ((FIXP_DBL)0x7FFFFFFF - rnd_val_wts1)); /* rounding must not cause overflow */ - output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT( + output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT( tmp + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS); #else FDK_ASSERT((WTS1 + 1 - scale) >= 0); output[(N * 3 / 4 - 1 - i)] = - (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS); + (PCM_DEC)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS); #endif z[i] = z0; @@ -173,22 +176,22 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb, tmp1 = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) + fMultDiv2(z[i], fb[N + N / 2 + i])); -#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0) +#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0) FDK_ASSERT((-WTS0 - 1 + scale) >= 0); FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF - rnd_val_wts0)); /* rounding must not cause overflow */ FDK_ASSERT(tmp1 <= ((FIXP_DBL)0x7FFFFFFF - rnd_val_wts1)); /* rounding must not cause overflow */ - output[(i - N / 4)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT( + output[(i - N / 4)] = (PCM_DEC)SATURATE_RIGHT_SHIFT( tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS); - output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT( + output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT( tmp1 + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS); #else FDK_ASSERT((WTS0 + 1 - scale) >= 0); output[(i - N / 4)] = - (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS); + (PCM_DEC)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS); output[(N * 3 / 4 - 1 - i)] = - (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS); + (PCM_DEC)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS); #endif z[i] = z0; z[N + i] = z2; @@ -198,22 +201,22 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb, for (i = 0; i < N / 4; i++) { FIXP_DBL tmp0 = fMultDiv2(z[i], fb[N / 2 + i]); -#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0) +#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0) FDK_ASSERT((-WTS0 - 1 + scale) >= 0); FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF - rnd_val_wts0)); /* rounding must not cause overflow */ - output[(N * 3 / 4 + i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT( + output[(N * 3 / 4 + i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT( tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS); #else FDK_ASSERT((WTS0 + 1 - scale) >= 0); output[(N * 3 / 4 + i)] = - (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS); + (PCM_DEC)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS); #endif } } int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctData, const int mdctData_e, - FIXP_PCM *output, FIXP_DBL *fs_buffer, + PCM_DEC *output, FIXP_DBL *fs_buffer, const int N) { const FIXP_WTB *coef; FIXP_DBL gain = (FIXP_DBL)0; diff --git a/libAACdec/src/ldfiltbank.h b/libAACdec/src/ldfiltbank.h index b63da6b..02971d0 100644 --- a/libAACdec/src/ldfiltbank.h +++ b/libAACdec/src/ldfiltbank.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -104,9 +104,10 @@ amm-info@iis.fraunhofer.de #define LDFILTBANK_H #include "common_fix.h" +#include "aac_rom.h" int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctdata_m, const int mdctdata_e, - FIXP_PCM *mdctOut, FIXP_DBL *fs_buffer, + PCM_DEC *mdctOut, FIXP_DBL *fs_buffer, const int frameLength); #endif diff --git a/libAACdec/src/usacdec_lpd.cpp b/libAACdec/src/usacdec_lpd.cpp index de0c2de..fbf6fab 100644 --- a/libAACdec/src/usacdec_lpd.cpp +++ b/libAACdec/src/usacdec_lpd.cpp @@ -122,18 +122,21 @@ amm-info@iis.fraunhofer.de #include "ac_arith_coder.h" -void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise, - const FIXP_SGL *filt, INT stop, int len) { +void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise, + const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop, + int len) { INT i, j; FIXP_DBL tmp; + FDK_ASSERT((aacOutDataHeadroom - 1) >= -(MDCT_OUTPUT_SCALE)); + for (i = 0; i < stop; i++) { tmp = fMultDiv2(noise[i], filt[0]); // Filt in Q-1.16 for (j = 1; j <= len; j++) { tmp += fMult((noise[i - j] >> 1) + (noise[i + j] >> 1), filt[j]); } - syn_out[i] = (FIXP_PCM)(SATURATE_SHIFT( - (syn[i] >> 1) - (tmp >> 1), (MDCT_OUTPUT_SCALE - 1), PCM_OUT_BITS)); + syn_out[i] = (PCM_DEC)( + IMDCT_SCALE((syn[i] >> 1) - (tmp >> 1), aacOutDataHeadroom - 1)); } } @@ -143,8 +146,10 @@ void bass_pf_1sf_delay( FIXP_DBL *pit_gain, const int frame_length, /* (i) : frame length (should be 768|1024) */ const INT l_frame, - const INT l_next, /* (i) : look ahead for symmetric filtering */ - FIXP_PCM *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */ + const INT l_next, /* (i) : look ahead for symmetric filtering */ + PCM_DEC *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */ + const INT aacOutDataHeadroom, /* (i) : headroom of the output time signal to + prevent clipping */ FIXP_DBL mem_bpf[]) /* i/o : memory state [L_FILT+L_SUBFR] */ { INT i, sf, i_subfr, T, T2, lg; @@ -370,7 +375,7 @@ void bass_pf_1sf_delay( { filtLP(&syn[i_subfr - L_SUBFR], &synth_out[i_subfr], noise, - fdk_dec_filt_lp, L_SUBFR, L_FILT); + fdk_dec_filt_lp, aacOutDataHeadroom, L_SUBFR, L_FILT); } } @@ -383,9 +388,9 @@ void bass_pf_1sf_delay( /* Output scaling of the BPF memory */ scaleValues(mem_bpf, (L_FILT + L_SUBFR), -1); /* Copy the rest of the signal (after the fac) */ - scaleValuesSaturate((FIXP_PCM *)&synth_out[l_frame], - (FIXP_DBL *)&syn[l_frame - L_SUBFR], - (frame_length - l_frame), MDCT_OUT_HEADROOM); + scaleValuesSaturate( + (PCM_DEC *)&synth_out[l_frame], (FIXP_DBL *)&syn[l_frame - L_SUBFR], + (frame_length - l_frame), MDCT_OUT_HEADROOM - aacOutDataHeadroom); } return; @@ -1552,9 +1557,9 @@ void CLpdChannelStream_Decode( AAC_DECODER_ERROR CLpd_RenderTimeSignal( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData, - INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, UINT flags, - UINT strmFlags) { + CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData, + INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, + const INT aacOutDataHeadroom, UINT flags, UINT strmFlags) { UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; AAC_DECODER_ERROR error = AAC_DEC_OK; int k, i_offset; @@ -2017,7 +2022,8 @@ AAC_DECODER_ERROR CLpd_RenderTimeSignal( { bass_pf_1sf_delay(p2_synth, pitch, pit_gain, lFrame, lFrame / facFB, mod[nbDiv - 1] ? (SynDelay - (lDiv / 2)) : SynDelay, - pTimeData, pAacDecoderStaticChannelInfo->mem_bpf); + pTimeData, aacOutDataHeadroom, + pAacDecoderStaticChannelInfo->mem_bpf); } } diff --git a/libAACdec/src/usacdec_lpd.h b/libAACdec/src/usacdec_lpd.h index 3e7938d..448dc55 100644 --- a/libAACdec/src/usacdec_lpd.h +++ b/libAACdec/src/usacdec_lpd.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -140,13 +140,14 @@ void CLpdChannelStream_Decode( * \param pTimeData pointer to output buffer * \param samplesPerFrame amount of output samples * \param pSamplingRateInfo holds the sampling rate information - * \param pWorkBuffer1 pointer to work buffer for temporal data + * \param aacOutDataHeadroom headroom of the output time signal to prevent + * clipping */ AAC_DECODER_ERROR CLpd_RenderTimeSignal( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData, + CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData, INT samplesPerFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, - UINT flags, UINT strmFlags); + const INT aacOutDataHeadroom, UINT flags, UINT strmFlags); static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) { if (fNotShortBlock) { @@ -156,8 +157,9 @@ static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) { } } -void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise, - const FIXP_SGL *filt, INT stop, int len); +void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise, + const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop, + int len); /** * \brief perform a low-frequency pitch enhancement on time domain signal @@ -171,13 +173,14 @@ void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise, * \param[in] l_frame length of filtering, must be multiple of L_SUBFR * \param[in] l_next length of allowed look ahead on syn[i], i < l_frame+l_next * \param[out] synth_out pointer to time domain output signal + * \param[in] headroom of the output time signal to prevent clipping * \param[in,out] mem_bpf pointer to filter memory (L_FILT+L_SUBFR) */ void bass_pf_1sf_delay(FIXP_DBL syn[], const INT T_sf[], FIXP_DBL *pit_gain, const int frame_length, const INT l_frame, - const INT l_next, FIXP_PCM *synth_out, - FIXP_DBL mem_bpf[]); + const INT l_next, PCM_DEC *synth_out, + const INT aacOutDataHeadroom, FIXP_DBL mem_bpf[]); /** * \brief random sign generator for FD and TCX noise filling diff --git a/libFDK/include/FDK_qmf_domain.h b/libFDK/include/FDK_qmf_domain.h index aa247e9..0e83da3 100644 --- a/libFDK/include/FDK_qmf_domain.h +++ b/libFDK/include/FDK_qmf_domain.h @@ -160,8 +160,6 @@ H_ALLOC_MEM(QmfSlotsImag32, FIXP_DBL *) H_ALLOC_MEM(QmfOverlapBuffer16, FIXP_DBL) H_ALLOC_MEM(QmfOverlapBuffer32, FIXP_DBL) -#define QDOM_PCM INT_PCM - /** * Structure to hold the configuration data which is global whithin a QMF domain * instance. @@ -181,9 +179,6 @@ typedef struct { park a channel if only one processing channel is available. */ UCHAR parkChannel_requested; - QDOM_PCM - *TDinput; /*!< Pointer to time domain data used as input for the QMF - analysis. */ FIXP_DBL * pWorkBuffer[QMF_MAX_WB_SECTIONS]; /*!< Pointerarray to volatile memory. */ UINT flags; /*!< Flags to be set on all QMF analysis/synthesis filter diff --git a/libFDK/include/mdct.h b/libFDK/include/mdct.h index 1382374..e7cf3ad 100644 --- a/libFDK/include/mdct.h +++ b/libFDK/include/mdct.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -106,18 +106,16 @@ amm-info@iis.fraunhofer.de #include "common_fix.h" #define MDCT_OUT_HEADROOM 2 /* Output additional headroom */ -#define PCM_OUT_BITS SAMPLE_BITS + +#define PCM_OUT_BITS DFRACT_BITS #define PCM_OUT_HEADROOM 8 /* Must have the same values as DMXH_HEADROOM */ -#define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - SAMPLE_BITS)) +#define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - PCM_OUT_BITS)) /* Refer to "Output word length" in ISO/IEC 14496-3:2008(E) 23.2.3.6 */ #define MDCT_OUTPUT_GAIN 16 -#if (MDCT_OUTPUT_SCALE >= 0) -#define IMDCT_SCALE(x) SATURATE_RIGHT_SHIFT(x, MDCT_OUTPUT_SCALE, PCM_OUT_BITS) -#else -#define IMDCT_SCALE(x) SATURATE_LEFT_SHIFT(x, -MDCT_OUTPUT_SCALE, PCM_OUT_BITS) -#endif +#define IMDCT_SCALE(x, s) \ + SATURATE_RIGHT_SHIFT((x), ((s) + MDCT_OUTPUT_SCALE), PCM_OUT_BITS) #define IMDCT_SCALE_DBL(x) (FIXP_DBL)(x) #define IMDCT_SCALE_DBL_LSH1(x) SATURATE_LEFT_SHIFT_ALT((x), 1, DFRACT_BITS) diff --git a/libFDK/src/FDK_core.cpp b/libFDK/src/FDK_core.cpp index 75ea8a2..2f77179 100644 --- a/libFDK/src/FDK_core.cpp +++ b/libFDK/src/FDK_core.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -104,7 +104,7 @@ amm-info@iis.fraunhofer.de /* FDK tools library info */ #define FDK_TOOLS_LIB_VL0 3 -#define FDK_TOOLS_LIB_VL1 0 +#define FDK_TOOLS_LIB_VL1 1 #define FDK_TOOLS_LIB_VL2 0 #define FDK_TOOLS_LIB_TITLE "FDK Tools" #ifdef __ANDROID__ diff --git a/libFDK/src/FDK_qmf_domain.cpp b/libFDK/src/FDK_qmf_domain.cpp index 71e2ef5..77c5ca2 100644 --- a/libFDK/src/FDK_qmf_domain.cpp +++ b/libFDK/src/FDK_qmf_domain.cpp @@ -637,10 +637,10 @@ void FDK_QmfDomain_GetSlot(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch, const int ts, if (pQmfOutImag == NULL) { for (; b < fMin(lsb, stop_band); b++) { - pQmfOutReal[b] = scaleValue(real[b], lb_sf); + pQmfOutReal[b] = scaleValueSaturate(real[b], lb_sf); } for (; b < fMin(usb, stop_band); b++) { - pQmfOutReal[b] = scaleValue(real[b], hb_sf); + pQmfOutReal[b] = scaleValueSaturate(real[b], hb_sf); } for (; b < stop_band; b++) { pQmfOutReal[b] = (FIXP_DBL)0; @@ -648,12 +648,12 @@ void FDK_QmfDomain_GetSlot(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch, const int ts, } else { FDK_ASSERT(imag != NULL); for (; b < fMin(lsb, stop_band); b++) { - pQmfOutReal[b] = scaleValue(real[b], lb_sf); - pQmfOutImag[b] = scaleValue(imag[b], lb_sf); + pQmfOutReal[b] = scaleValueSaturate(real[b], lb_sf); + pQmfOutImag[b] = scaleValueSaturate(imag[b], lb_sf); } for (; b < fMin(usb, stop_band); b++) { - pQmfOutReal[b] = scaleValue(real[b], hb_sf); - pQmfOutImag[b] = scaleValue(imag[b], hb_sf); + pQmfOutReal[b] = scaleValueSaturate(real[b], hb_sf); + pQmfOutImag[b] = scaleValueSaturate(imag[b], hb_sf); } for (; b < stop_band; b++) { pQmfOutReal[b] = (FIXP_DBL)0; diff --git a/libPCMutils/include/limiter.h b/libPCMutils/include/limiter.h index fab7226..419e891 100644 --- a/libPCMutils/include/limiter.h +++ b/libPCMutils/include/limiter.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -109,8 +109,6 @@ amm-info@iis.fraunhofer.de #define TDL_ATTACK_DEFAULT_MS (15) /* default attack time in ms */ #define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */ -#define TDL_GAIN_SCALING (15) /* scaling of gain value. */ - #ifdef __cplusplus extern "C" { #endif @@ -128,10 +126,7 @@ struct TDLimiter { unsigned int maxBufIdx, delayBufIdx; FIXP_DBL smoothState0; FIXP_DBL minGain; - - FIXP_DBL additionalGainPrev; - FIXP_DBL additionalGainFilterState; - FIXP_DBL additionalGainFilterState1; + INT scaling; }; typedef enum { @@ -255,27 +250,16 @@ TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter, /****************************************************************************** * pcmLimiter_Apply * - * limiter: limiter handle * - * pGain : pointer to gains to be applied to the signal before limiting, * - * which are downscaled by TDL_GAIN_SCALING bit. * - * These gains are delayed by gain_delay, and smoothed. * - * Smoothing is done by a butterworth lowpass filter with a cutoff * - * frequency which is fixed with respect to the sampling rate. * - * It is a substitute for the smoothing due to windowing and * - * overlap/add, if a gain is applied in frequency domain. * - * gain_scale: pointer to scaling exponents to be applied to the signal before * - * limiting, without delay and without smoothing * - * gain_size: number of elements in pGain, currently restricted to 1 * - * gain_delay: delay [samples] with which the gains in pGain shall be applied * - * gain_delay <= nSamples * - * samples: input/output buffer containing interleaved samples * - * precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits * - * nSamples: number of samples per channel * + * limiter: limiter handle * + * samplesIn: pointer to input buffer containing interleaved samples * + * samplesOut: pointer to output buffer containing interleaved samples * + * pGainPerSample: pointer to gains for each sample * + * scaling: scaling of output samples * + * nSamples: number of samples per channel * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, - INT_PCM* samplesOut, FIXP_DBL* pGain, - const INT* gain_scale, const UINT gain_size, - const UINT gain_delay, const UINT nSamples); + INT_PCM* samplesOut, FIXP_DBL* pGainPerSample, + const INT scaling, const UINT nSamples); #endif /* #ifndef LIMITER_H */ diff --git a/libPCMutils/src/limiter.cpp b/libPCMutils/src/limiter.cpp index 60e3e3b..598dc0c 100644 --- a/libPCMutils/src/limiter.cpp +++ b/libPCMutils/src/limiter.cpp @@ -152,7 +152,7 @@ TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs, limiter->attack = attack; limiter->attackConst = attackConst; limiter->releaseConst = releaseConst; - limiter->threshold = threshold >> TDL_GAIN_SCALING; + limiter->threshold = threshold; limiter->channels = maxChannels; limiter->maxChannels = maxChannels; limiter->sampleRate = maxSampleRate; @@ -165,18 +165,13 @@ TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs, /* apply limiter */ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, - INT_PCM* samplesOut, FIXP_DBL* RESTRICT pGain, - const INT* RESTRICT gain_scale, - const UINT gain_size, const UINT gain_delay, - const UINT nSamples) { + INT_PCM* samplesOut, FIXP_DBL* pGainPerSample, + const INT scaling, const UINT nSamples) { unsigned int i, j; - FIXP_DBL tmp1; FIXP_DBL tmp2; - FIXP_DBL tmp, old, gain, additionalGain = 0, additionalGainUnfiltered; + FIXP_DBL tmp, old, gain, additionalGain = 0; FIXP_DBL minGain = FL2FXCONST_DBL(1.0f / (1 << 1)); - - FDK_ASSERT(gain_size == 1); - FDK_ASSERT(gain_delay <= nSamples); + UINT additionalGainAvailable = 1; if (limiter == NULL) return TDLIMIT_INVALID_HANDLE; @@ -185,7 +180,7 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, unsigned int attack = limiter->attack; FIXP_DBL attackConst = limiter->attackConst; FIXP_DBL releaseConst = limiter->releaseConst; - FIXP_DBL threshold = limiter->threshold; + FIXP_DBL threshold = limiter->threshold >> scaling; FIXP_DBL max = limiter->max; FIXP_DBL* maxBuf = limiter->maxBuf; @@ -195,55 +190,34 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, unsigned int delayBufIdx = limiter->delayBufIdx; FIXP_DBL smoothState0 = limiter->smoothState0; - FIXP_DBL additionalGainSmoothState = limiter->additionalGainFilterState; - FIXP_DBL additionalGainSmoothState1 = limiter->additionalGainFilterState1; - if (!gain_delay) { - additionalGain = pGain[0]; - if (gain_scale[0] > 0) { - additionalGain <<= gain_scale[0]; - } else { - additionalGain >>= -gain_scale[0]; - } + if (limiter->scaling != scaling) { + scaleValuesSaturate(delayBuf, attack * channels, + limiter->scaling - scaling); + scaleValuesSaturate(maxBuf, attack + 1, limiter->scaling - scaling); + max = scaleValueSaturate(max, limiter->scaling - scaling); + limiter->scaling = scaling; } - for (i = 0; i < nSamples; i++) { - if (gain_delay) { - if (i < gain_delay) { - additionalGainUnfiltered = limiter->additionalGainPrev; - } else { - additionalGainUnfiltered = pGain[0]; - } + if (pGainPerSample == NULL) { + additionalGainAvailable = 0; + } - /* Smooth additionalGain */ - /* [b,a] = butter(1, 0.01) */ - static const FIXP_SGL b[] = {FL2FXCONST_SGL(0.015466 * 2.0), - FL2FXCONST_SGL(0.015466 * 2.0)}; - static const FIXP_SGL a[] = {(FIXP_SGL)MAXVAL_SGL, - FL2FXCONST_SGL(-0.96907)}; - additionalGain = -fMult(additionalGainSmoothState, a[1]) + - fMultDiv2(additionalGainUnfiltered, b[0]) + - fMultDiv2(additionalGainSmoothState1, b[1]); - additionalGainSmoothState1 = additionalGainUnfiltered; - additionalGainSmoothState = additionalGain; - - /* Apply the additional scaling that has no delay and no smoothing */ - if (gain_scale[0] > 0) { - additionalGain <<= gain_scale[0]; - } else { - additionalGain >>= -gain_scale[0]; - } - } + for (i = 0; i < nSamples; i++) { /* get maximum absolute sample value of all channels, including the * additional gain. */ - tmp1 = (FIXP_DBL)0; + tmp = (FIXP_DBL)0; for (j = 0; j < channels; j++) { tmp2 = PCM_LIM2FIXP_DBL(samplesIn[j]); tmp2 = (tmp2 == (FIXP_DBL)MINVAL_DBL) ? (FIXP_DBL)MAXVAL_DBL : fAbs(tmp2); - tmp1 = fMax(tmp1, tmp2); + tmp = fMax(tmp, tmp2); + } + + if (additionalGainAvailable) { + additionalGain = pGainPerSample[i]; + tmp = fMult(tmp, additionalGain); } - tmp = fMult(tmp1, additionalGain); /* set threshold as lower border to save calculations in running maximum * algorithm */ @@ -314,22 +288,42 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, /* lookahead delay, apply gain */ for (j = 0; j < channels; j++) { tmp = p_delayBuf[j]; - p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain); + + if (additionalGainAvailable) { + p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain); + } else { + p_delayBuf[j] = PCM_LIM2FIXP_DBL(samplesIn[j]); + } /* Apply gain to delayed signal */ tmp = fMultDiv2(tmp, gain); - +#if (SAMPLE_BITS == DFRACT_BITS) + samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM( + (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling + 1, DFRACT_BITS)); +#else samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT( - tmp, TDL_GAIN_SCALING + 1, DFRACT_BITS)); + tmp + ((FIXP_DBL)0x8000 >> (scaling + 1)), scaling + 1, + DFRACT_BITS)); +#endif } gain >>= 1; } else { /* lookahead delay, apply gain=1.0f */ for (j = 0; j < channels; j++) { tmp = p_delayBuf[j]; - p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain); + if (additionalGainAvailable) { + p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain); + } else { + p_delayBuf[j] = PCM_LIM2FIXP_DBL(samplesIn[j]); + } + +#if (SAMPLE_BITS == DFRACT_BITS) + samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM( + (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling, DFRACT_BITS)); +#else samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT( - tmp, TDL_GAIN_SCALING, DFRACT_BITS)); + tmp + ((FIXP_DBL)0x8000 >> scaling), scaling, DFRACT_BITS)); +#endif } } @@ -354,13 +348,9 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, limiter->delayBufIdx = delayBufIdx; limiter->smoothState0 = smoothState0; - limiter->additionalGainFilterState = additionalGainSmoothState; - limiter->additionalGainFilterState1 = additionalGainSmoothState1; limiter->minGain = minGain; - limiter->additionalGainPrev = pGain[0]; - return TDLIMIT_OK; } } @@ -370,7 +360,7 @@ TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter, FIXP_DBL threshold) { if (limiter == NULL) return TDLIMIT_INVALID_HANDLE; - limiter->threshold = threshold >> TDL_GAIN_SCALING; + limiter->threshold = threshold; return TDLIMIT_OK; } @@ -384,13 +374,7 @@ TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter) { limiter->cor = FL2FXCONST_DBL(1.0f / (1 << 1)); limiter->smoothState0 = FL2FXCONST_DBL(1.0f / (1 << 1)); limiter->minGain = FL2FXCONST_DBL(1.0f / (1 << 1)); - - limiter->additionalGainPrev = - FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING)); - limiter->additionalGainFilterState = - FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING)); - limiter->additionalGainFilterState1 = - FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING)); + limiter->scaling = 0; FDKmemset(limiter->maxBuf, 0, (limiter->attack + 1) * sizeof(FIXP_DBL)); FDKmemset(limiter->delayBuf, 0, diff --git a/libPCMutils/src/version.h b/libPCMutils/src/version.h index fa31af1..05371f8 100644 --- a/libPCMutils/src/version.h +++ b/libPCMutils/src/version.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -105,7 +105,7 @@ amm-info@iis.fraunhofer.de /* library info */ #define PCMUTIL_LIB_VL0 3 -#define PCMUTIL_LIB_VL1 0 +#define PCMUTIL_LIB_VL1 1 #define PCMUTIL_LIB_VL2 0 #define PCMUTIL_LIB_TITLE "PCM Utility Lib" #ifdef __ANDROID__ diff --git a/libSACdec/include/sac_dec_lib.h b/libSACdec/include/sac_dec_lib.h index 9913279..1827504 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -252,7 +252,7 @@ typedef enum { scenario. Default parameter value is 3 frames. */ } SACDEC_PARAM; -#define PCM_MPS INT_PCM +#define PCM_MPS LONG /** * \brief MPEG Surround decoder handle. @@ -401,17 +401,22 @@ int mpegSurroundDecoder_Parse(CMpegSurroundDecoder *pMpegSurroundDecoder, * for each output audio channel is stored into. * \param mapDescr Channep map descriptor for output channel mapping * to be used (From MPEG PCE ordering to whatever is required). + * \param inDataHeadroom Headroom of SAC input time signal to prevent + * clipping. + * \param outDataHeadroom Pointer to headroom of SAC output time signal to + * prevent clipping. * * \return Error code. */ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder, - INT_PCM *input, PCM_MPS *pTimeData, + PCM_MPS *input, PCM_MPS *pTimeData, const int timeDataSize, int timeDataFrameSize, int *nChannels, int *frameSize, int sampleRate, AUDIO_OBJECT_TYPE coreCodec, AUDIO_CHANNEL_TYPE channelType[], UCHAR channelIndices[], - const FDK_channelMapDescr *const mapDescr); + const FDK_channelMapDescr *const mapDescr, + const INT inDataHeadroom, INT *outDataHeadroom); /** * \brief Deallocate a MPEG Surround decoder instance. diff --git a/libSACdec/src/sac_dec.cpp b/libSACdec/src/sac_dec.cpp index c1832f1..a7b50df 100644 --- a/libSACdec/src/sac_dec.cpp +++ b/libSACdec/src/sac_dec.cpp @@ -766,7 +766,7 @@ SACDEC_ERROR FDK_SpatialDecInit(spatialDec *self, SPATIAL_BS_FRAME *frame, /* output scaling */ for (nCh = 0; nCh < self->numOutputChannelsAT; nCh++) { - int outputScale = 0, outputGain_e = 0, scale = 0; + int outputScale = 0, outputGain_e = 0, scale = -(8) + (1); FIXP_DBL outputGain_m = getChGain(self, nCh, &outputGain_e); if (!isTwoChMode(self->upmixType) && !bypassMode) { @@ -775,7 +775,7 @@ SACDEC_ERROR FDK_SpatialDecInit(spatialDec *self, SPATIAL_BS_FRAME *frame, synthesis qmf */ } - scale = outputScale; + scale += outputScale; qmfChangeOutScalefactor(&self->pQmfDomain->QmfDomainOut[nCh].fb, scale); qmfChangeOutGain(&self->pQmfDomain->QmfDomainOut[nCh].fb, outputGain_m, @@ -1223,18 +1223,24 @@ static SACDEC_ERROR SpatialDecApplyParameterSets( !(self->stereoConfigIndex == 3)) { for (i = 0; i < self->qmfBands; i++) { self_qmfResidualReal__FDK_0_0[i] = - fMult(self_qmfResidualReal__FDK_0_0[i] << 1, + fMult(scaleValueSaturate(self_qmfResidualReal__FDK_0_0[i], + 1 + self->sacInDataHeadroom - (1)), self->clipProtectGain__FDK); self_qmfResidualImag__FDK_0_0[i] = - fMult(self_qmfResidualImag__FDK_0_0[i] << 1, + fMult(scaleValueSaturate(self_qmfResidualImag__FDK_0_0[i], + 1 + self->sacInDataHeadroom - (1)), self->clipProtectGain__FDK); } } else { for (i = 0; i < self->qmfBands; i++) { - self_qmfResidualReal__FDK_0_0[i] = fMult( - self_qmfResidualReal__FDK_0_0[i], self->clipProtectGain__FDK); - self_qmfResidualImag__FDK_0_0[i] = fMult( - self_qmfResidualImag__FDK_0_0[i], self->clipProtectGain__FDK); + self_qmfResidualReal__FDK_0_0[i] = + fMult(scaleValueSaturate(self_qmfResidualReal__FDK_0_0[i], + self->sacInDataHeadroom - (1)), + self->clipProtectGain__FDK); + self_qmfResidualImag__FDK_0_0[i] = + fMult(scaleValueSaturate(self_qmfResidualImag__FDK_0_0[i], + self->sacInDataHeadroom - (1)), + self->clipProtectGain__FDK); } } } @@ -1416,6 +1422,7 @@ SACDEC_ERROR SpatialDecApplyFrame( FDK_ASSERT(self != NULL); FDK_ASSERT(pControlFlags != NULL); FDK_ASSERT(pcmOutBuf != NULL); + FDK_ASSERT(self->sacInDataHeadroom >= (1)); self->errInt = err; /* Init internal error */ diff --git a/libSACdec/src/sac_dec.h b/libSACdec/src/sac_dec.h index 992acad..1c3df71 100644 --- a/libSACdec/src/sac_dec.h +++ b/libSACdec/src/sac_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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -523,6 +523,9 @@ struct spatialDec_struct { new frame after SSC change (aka decodeAfterConfigHasChangedFlag). */ SpatialDecConcealmentInfo concealInfo; + + INT sacInDataHeadroom; /* Headroom of the SAC input time signal to prevent + clipping */ }; #define SACDEC_SYNTAX_MPS 1 diff --git a/libSACdec/src/sac_dec_interface.h b/libSACdec/src/sac_dec_interface.h index a2eea92..05a9a75 100644 --- a/libSACdec/src/sac_dec_interface.h +++ b/libSACdec/src/sac_dec_interface.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -249,10 +249,10 @@ typedef struct { } MEM_REQUIREMENTS; -#define PCM_MPS INT_PCM -#define PCM_MPSF FIXP_PCM +#define PCM_MPS LONG +#define PCM_MPSF FIXP_DBL -#define FIXP_DBL2PCM_MPS(x) ((INT_PCM)FX_DBL2FX_PCM(x)) +#define FIXP_DBL2PCM_MPS(x) ((LONG)(x)) /* exposed functions (library interface) */ diff --git a/libSACdec/src/sac_dec_lib.cpp b/libSACdec/src/sac_dec_lib.cpp index bf6dedf..856a923 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -1507,15 +1507,17 @@ bail: } int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder, - INT_PCM *input, PCM_MPS *pTimeData, + PCM_MPS *input, PCM_MPS *pTimeData, const int timeDataSize, int timeDataFrameSize, int *nChannels, int *frameSize, int sampleRate, AUDIO_OBJECT_TYPE coreCodec, AUDIO_CHANNEL_TYPE channelType[], UCHAR channelIndices[], - const FDK_channelMapDescr *const mapDescr) { + const FDK_channelMapDescr *const mapDescr, + const INT inDataHeadroom, INT *outDataHeadroom) { SACDEC_ERROR err = MPS_OK; PCM_MPS *pTimeOut = pTimeData; + PCM_MPS *TDinput = NULL; UINT initControlFlags = 0, controlFlags = 0; int timeDataRequiredSize = 0; int newData; @@ -1534,6 +1536,9 @@ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder, return MPS_NOTOK; } + pMpegSurroundDecoder->pSpatialDec->sacInDataHeadroom = inDataHeadroom; + *outDataHeadroom = (INT)(8); + pMpegSurroundDecoder->pSpatialDec->pConfigCurrent = &pMpegSurroundDecoder ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameDecode]; @@ -1682,8 +1687,7 @@ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder, (timeDataFrameSize * pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsSynthesis) / pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsAnalysis; - pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput = - pTimeData + timeDataFrameSizeOut - timeDataFrameSize; + TDinput = pTimeData + timeDataFrameSizeOut - timeDataFrameSize; for (int i = *nChannels - 1; i >= 0; i--) { FDKmemmove(pTimeData + (i + 1) * timeDataFrameSizeOut - timeDataFrameSize, pTimeData + timeDataFrameSize * i, @@ -1694,8 +1698,8 @@ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder, } else { if (pMpegSurroundDecoder->mpegSurroundUseTimeInterface) { FDKmemcpy(input, pTimeData, - sizeof(INT_PCM) * (*nChannels) * (*frameSize)); - pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput = input; + sizeof(PCM_MPS) * (*nChannels) * (*frameSize)); + TDinput = input; } } @@ -1707,8 +1711,8 @@ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder, &pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameDecode], pMpegSurroundDecoder->mpegSurroundUseTimeInterface ? INPUTMODE_TIME : INPUTMODE_QMF_SBR, - pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput, NULL, NULL, - pTimeOut, *frameSize, &controlFlags, *nChannels, mapDescr); + TDinput, NULL, NULL, pTimeOut, *frameSize, &controlFlags, *nChannels, + mapDescr); *nChannels = pMpegSurroundDecoder->pSpatialDec->numOutputChannelsAT; if (err != @@ -1781,7 +1785,7 @@ void mpegSurroundDecoder_Close(CMpegSurroundDecoder *pMpegSurroundDecoder) { } #define SACDEC_VL0 2 -#define SACDEC_VL1 0 +#define SACDEC_VL1 1 #define SACDEC_VL2 0 int mpegSurroundDecoder_GetLibInfo(LIB_INFO *info) { diff --git a/libSACdec/src/sac_process.cpp b/libSACdec/src/sac_process.cpp index 95128f3..22091a9 100644 --- a/libSACdec/src/sac_process.cpp +++ b/libSACdec/src/sac_process.cpp @@ -187,8 +187,12 @@ SACDEC_ERROR SpatialDecQMFAnalysis(spatialDec *self, const PCM_MPS *inData, if (!isTwoChMode(self->upmixType) && !bypassMode) { int i; for (i = 0; i < self->qmfBands; i++) { - qmfReal[ch][i] = fMult(qmfReal[ch][i], self->clipProtectGain__FDK); - qmfImag[ch][i] = fMult(qmfImag[ch][i], self->clipProtectGain__FDK); + qmfReal[ch][i] = fMult( + scaleValueSaturate(qmfReal[ch][i], self->sacInDataHeadroom - (1)), + self->clipProtectGain__FDK); + qmfImag[ch][i] = fMult( + scaleValueSaturate(qmfImag[ch][i], self->sacInDataHeadroom - (1)), + self->clipProtectGain__FDK); } } } @@ -216,16 +220,17 @@ SACDEC_ERROR SpatialDecFeedQMF(spatialDec *self, FIXP_DBL **qmfInDataReal, /* Write Input data to pQmfRealAnalysis. */ if (self->bShareDelayWithSBR) { - FDK_QmfDomain_GetSlot( - &self->pQmfDomain->QmfDomainIn[ch], ts + HYBRID_FILTER_DELAY, 0, - MAX_QMF_BANDS_TO_HYBRID, pQmfRealAnalysis, pQmfImagAnalysis, 15); + FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], + ts + HYBRID_FILTER_DELAY, 0, + MAX_QMF_BANDS_TO_HYBRID, pQmfRealAnalysis, + pQmfImagAnalysis, 15 + (1)); FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts, MAX_QMF_BANDS_TO_HYBRID, self->qmfBands, - pQmfRealAnalysis, pQmfImagAnalysis, 15); + pQmfRealAnalysis, pQmfImagAnalysis, 15 + (1)); } else { FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts, 0, self->qmfBands, pQmfRealAnalysis, - pQmfImagAnalysis, 15); + pQmfImagAnalysis, 15 + (1)); } if (ts == self->pQmfDomain->globalConf.nQmfTimeSlots - 1) { /* Is currently also needed in case we dont have any overlap. We need to @@ -501,8 +506,8 @@ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding( for (pb = 0, qs = 3; pb < 2; pb++) { INT s; FIXP_DBL maxVal; - FIXP_SGL mReal1; - FIXP_SGL mReal0, mImag0; + FIXP_DBL mReal1; + FIXP_DBL mReal0, mImag0; FIXP_DBL iReal0, iImag0, iReal1; iReal0 = interpolateParameter(alpha, MReal0[pb], MRealPrev0[pb]); @@ -515,9 +520,9 @@ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding( s = fMax(CntLeadingZeros(maxVal) - 1, 0); s = fMin(s, scale_param_m2); - mReal0 = FX_DBL2FX_SGL(iReal0 << s); - mImag0 = FX_DBL2FX_SGL(iImag0 << s); - mReal1 = FX_DBL2FX_SGL(iReal1 << s); + mReal0 = iReal0 << s; + mImag0 = iImag0 << s; + mReal1 = iReal1 << s; s = scale_param_m2 - s; @@ -934,6 +939,7 @@ SACDEC_ERROR SpatialDecSynthesis(spatialDec *self, const INT ts, self->pQmfDomain->QmfDomainIn[outCh].scaling.lb_scale -= self->clipProtectGainSF__FDK; + self->pQmfDomain->QmfDomainIn[outCh].scaling.lb_scale -= (1); } else { /* Call the QMF synthesis for dry. */ err = CalculateSpaceSynthesisQmf(&self->pQmfDomain->QmfDomainOut[outCh], diff --git a/libSACdec/src/sac_qmf.cpp b/libSACdec/src/sac_qmf.cpp index a075490..fd7599d 100644 --- a/libSACdec/src/sac_qmf.cpp +++ b/libSACdec/src/sac_qmf.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -110,7 +110,7 @@ amm-info@iis.fraunhofer.de SACDEC_ERROR CalculateSpaceSynthesisQmf( const HANDLE_FDK_QMF_DOMAIN_OUT hQmfDomainOutCh, const FIXP_DBL *Sr, - const FIXP_DBL *Si, const INT stride, INT_PCM *timeSig) { + const FIXP_DBL *Si, const INT stride, PCM_MPS *timeSig) { SACDEC_ERROR err = MPS_OK; if (hQmfDomainOutCh == NULL) { diff --git a/libSACdec/src/sac_qmf.h b/libSACdec/src/sac_qmf.h index d1dc837..5cd573e 100644 --- a/libSACdec/src/sac_qmf.h +++ b/libSACdec/src/sac_qmf.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -124,7 +124,7 @@ amm-info@iis.fraunhofer.de */ SACDEC_ERROR CalculateSpaceSynthesisQmf( const HANDLE_FDK_QMF_DOMAIN_OUT hQmfDomainOutCh, const FIXP_DBL *Sr, - const FIXP_DBL *Si, const INT stride, INT_PCM *timeSig); + const FIXP_DBL *Si, const INT stride, PCM_MPS *timeSig); /** * \brief Convert audio input data to qmf representation. diff --git a/libSACdec/src/sac_reshapeBBEnv.cpp b/libSACdec/src/sac_reshapeBBEnv.cpp index b44ce4e..272d009 100644 --- a/libSACdec/src/sac_reshapeBBEnv.cpp +++ b/libSACdec/src/sac_reshapeBBEnv.cpp @@ -253,17 +253,17 @@ static inline void slotAmp(FIXP_DBL *RESTRICT slotAmp_dry, dry = wet = FL2FXCONST_DBL(0.0f); for (qs = 0; qs < cplxBands; qs++) { - dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs]) + - fPow2Div2(pHybOutputImagDry[qs])); - wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs]) + - fPow2Div2(pHybOutputImagWet[qs])); + dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)) + + fPow2Div2(pHybOutputImagDry[qs] << (1))); + wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)) + + fPow2Div2(pHybOutputImagWet[qs] << (1))); } for (; qs < hybBands; qs++) { - dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs])); - wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs])); + dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1))); + wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1))); } - *slotAmp_dry = dry; - *slotAmp_wet = wet; + *slotAmp_dry = dry >> (2 * (1)); + *slotAmp_wet = wet >> (2 * (1)); } #if defined(__aarch64__) @@ -327,7 +327,7 @@ static void extractBBEnv(spatialDec *self, INT inp, INT start, INT channels, INT shapeActiv = 1; INT hybBands = fixMin(42, self->hybridBands); - INT staticScale = self->staticDecScale; + INT staticScale = self->staticDecScale + (1); INT cplxBands; cplxBands = fixMin(42, self->hybridBands); diff --git a/libSACdec/src/sac_rom.h b/libSACdec/src/sac_rom.h index 38f17f2..d317856 100644 --- a/libSACdec/src/sac_rom.h +++ b/libSACdec/src/sac_rom.h @@ -111,21 +111,12 @@ amm-info@iis.fraunhofer.de #include "machine_type.h" /* Global ROM table data type: */ -#ifndef ARCH_PREFER_MULT_32x32 -#define FIXP_CFG FIXP_SGL -#define FX_CFG2FX_DBL FX_SGL2FX_DBL -#define FX_CFG2FX_SGL -#define CFG(a) (FX_DBL2FXCONST_SGL(a)) -#define FL2FXCONST_CFG FL2FXCONST_SGL -#define FX_DBL2FX_CFG(x) FX_DBL2FX_SGL((FIXP_DBL)(x)) -#else #define FIXP_CFG FIXP_DBL #define FX_CFG2FX_DBL #define FX_CFG2FX_SGL FX_DBL2FX_SGL #define CFG(a) FIXP_DBL(a) #define FL2FXCONST_CFG FL2FXCONST_DBL #define FX_DBL2FX_CFG(x) ((FIXP_DBL)(x)) -#endif /* others */ #define SCALE_INV_ICC (2) @@ -133,15 +124,9 @@ amm-info@iis.fraunhofer.de #define QCC_SCALE 1 #define M1M2_DATA FIXP_DBL -#ifndef ARCH_PREFER_MULT_32x32 -#define M1M2_CDATA FIXP_SGL -#define M1M2_CDATA2FX_DBL(a) FX_SGL2FX_DBL(a) -#define FX_DBL2M1M2_CDATA(a) FX_DBL2FX_SGL(a) -#else #define M1M2_CDATA FIXP_DBL #define M1M2_CDATA2FX_DBL(a) (a) #define FX_DBL2M1M2_CDATA(a) (a) -#endif #define CLIP_PROTECT_GAIN_0(x) FL2FXCONST_CFG(((x) / (float)(1 << 0))) #define CLIP_PROTECT_GAIN_1(x) FL2FXCONST_CFG(((x) / (float)(1 << 1))) diff --git a/libSBRdec/include/sbrdecoder.h b/libSBRdec/include/sbrdecoder.h index cc55572..c09c985 100644 --- a/libSBRdec/include/sbrdecoder.h +++ b/libSBRdec/include/sbrdecoder.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -361,15 +361,20 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs, * error (0: core decoder found errors, 1: no errors). * \param psDecoded Pointer to a buffer holding a flag. Input: PS is * possible, Output: PS has been rendered. + * \param inDataHeadroom Headroom of the SBR input time signal to prevent + * clipping. + * \param outDataHeadroom Pointer to headroom of the SBR output time signal to + * prevent clipping. * * \return Error code. */ -SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input, - INT_PCM *timeData, const int timeDataSize, - int *numChannels, int *sampleRate, +SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData, + const int timeDataSize, int *numChannels, + int *sampleRate, const FDK_channelMapDescr *const mapDescr, const int mapIdx, const int coreDecodedOk, - UCHAR *psDecoded); + UCHAR *psDecoded, const INT inDataHeadroom, + INT *outDataHeadroom); /** * \brief Close SBR decoder instance and free memory. diff --git a/libSBRdec/src/HFgen_preFlat.cpp b/libSBRdec/src/HFgen_preFlat.cpp index 268011e..ad4caba 100644 --- a/libSBRdec/src/HFgen_preFlat.cpp +++ b/libSBRdec/src/HFgen_preFlat.cpp @@ -897,10 +897,10 @@ void sbrDecoder_calculateGainVec(FIXP_DBL **sourceBufferReal, for (i = startSample; i < stopSample; i++) { maxVal |= (FIXP_DBL)((LONG)(sourceBufferReal[i][loBand]) ^ - ((LONG)sourceBufferReal[i][loBand] >> (SAMPLE_BITS - 1))); + ((LONG)sourceBufferReal[i][loBand] >> (DFRACT_BITS - 1))); maxVal |= (FIXP_DBL)((LONG)(sourceBufferImag[i][loBand]) ^ - ((LONG)sourceBufferImag[i][loBand] >> (SAMPLE_BITS - 1))); + ((LONG)sourceBufferImag[i][loBand] >> (DFRACT_BITS - 1))); } if (maxVal != FL2FX_DBL(0.0f)) { diff --git a/libSBRdec/src/hbe.cpp b/libSBRdec/src/hbe.cpp index 2284c42..d210bb6 100644 --- a/libSBRdec/src/hbe.cpp +++ b/libSBRdec/src/hbe.cpp @@ -957,7 +957,7 @@ QmfTransposerCreate(HANDLE_HBE_TRANSPOSER* hQmfTransposer, const int frameSize, hQmfTran->qmfOutBufSize = 2 * (hQmfTran->noCols / 2 + QMF_WIN_LEN - 1); hQmfTran->inBuf_F = - (INT_PCM*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(INT_PCM)); + (LONG*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(LONG)); /* buffered time signal needs to be delayed by synthesis_size; max * synthesis_size = 20; */ if (hQmfTran->inBuf_F == NULL) { diff --git a/libSBRdec/src/hbe.h b/libSBRdec/src/hbe.h index fdffe1e..3556783 100644 --- a/libSBRdec/src/hbe.h +++ b/libSBRdec/src/hbe.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -132,6 +132,9 @@ typedef enum { } KEEP_STATES_SYNCED_MODE; struct hbeTransposer { + FIXP_DBL anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE]; + FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE]; + int xOverQmf[MAX_NUM_PATCHES_HBE]; int maxStretch; @@ -144,7 +147,7 @@ struct hbeTransposer { int stopBand; int bSbr41; - INT_PCM *inBuf_F; + LONG *inBuf_F; FIXP_DBL **qmfInBufReal_F; FIXP_DBL **qmfInBufImag_F; @@ -156,9 +159,6 @@ struct hbeTransposer { FIXP_DBL const *synthesisQmfPreModCos_F; FIXP_DBL const *synthesisQmfPreModSin_F; - FIXP_QAS anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE]; - FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE]; - FIXP_DBL **qmfHBEBufReal_F; FIXP_DBL **qmfHBEBufImag_F; diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp index 30611e7..b1fb0da 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -259,17 +259,18 @@ static void copyHarmonicSpectrum(int *xOverQmf, FIXP_DBL **qmfReal, void sbr_dec( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - INT_PCM *timeIn, /*!< pointer to input time signal */ - INT_PCM *timeOut, /*!< pointer to output time signal */ + LONG *timeIn, /*!< pointer to input time signal */ + LONG *timeOut, /*!< pointer to output time signal */ HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */ - INT_PCM *timeOutRight, /*!< pointer to output time signal */ + LONG *timeOutRight, /*!< pointer to output time signal */ const int strideOut, /*!< Time data traversal strideOut */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */ const int applyProcessing, /*!< Flag for SBR operation */ - HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize) { + HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize, + const INT sbrInDataHeadroom) { int i, slot, reserve; int saveLbScale; int lastSlotOffs; @@ -278,7 +279,7 @@ void sbr_dec( /* temporary pointer / variable for QMF; required as we want to use temporary buffer creating one frame delay for HBE in LP mode */ - INT_PCM *pTimeInQmf = timeIn; + LONG *pTimeInQmf = timeIn; /* Number of QMF timeslots in the overlap buffer: */ int ov_len = hSbrDec->LppTrans.pSettings->overlap; @@ -341,8 +342,8 @@ void sbr_dec( } else { C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2 * (64)); qmfAnalysisFiltering(&hSbrDec->qmfDomainInCh->fb, pReal, pImag, - &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf, 0, 1, - qmfTemp); + &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf, + 0 + sbrInDataHeadroom, 1, qmfTemp); C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2 * (64)); } @@ -658,7 +659,7 @@ void sbr_dec( if (!(flags & SBRDEC_PS_DECODED)) { if (!(flags & SBRDEC_SKIP_QMF_SYN)) { - int outScalefactor = 0; + int outScalefactor = -(8); if (h_ps_d != NULL) { h_ps_d->procFrameBased = 1; /* we here do frame based processing */ @@ -743,6 +744,7 @@ void sbr_dec( */ FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <= QMF_MAX_SYNTHESIS_BANDS); + qmfChangeOutScalefactor(synQmfRight, -(8)); FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates, 9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis * sizeof(FIXP_QSS)); @@ -814,7 +816,8 @@ void sbr_dec( : scaleFactorLowBand_no_ov, scaleFactorHighBand, synQmf->lsb, synQmf->usb); - outScalefactorL = outScalefactorR = 1; /* psDiffScale! (MPEG-PS) */ + outScalefactorL = outScalefactorR = + 1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */ } sbrDecoder_drcApplySlot(/* right channel */ @@ -831,6 +834,9 @@ void sbr_dec( outScalefactorL += maxShift; if (!(flags & SBRDEC_SKIP_QMF_SYN)) { + qmfChangeOutScalefactor(synQmf, -(8)); + qmfChangeOutScalefactor(synQmfRight, -(8)); + qmfSynthesisFilteringSlot( synQmfRight, rQmfReal, /* QMF real buffer */ rQmfImag, /* QMF imag buffer */ diff --git a/libSBRdec/src/sbr_dec.h b/libSBRdec/src/sbr_dec.h index 156da03..eb9c394 100644 --- a/libSBRdec/src/sbr_dec.h +++ b/libSBRdec/src/sbr_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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -176,17 +176,18 @@ typedef SBR_CHANNEL *HANDLE_SBR_CHANNEL; void sbr_dec( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ - INT_PCM *timeIn, /*!< pointer to input time signal */ - INT_PCM *timeOut, /*!< pointer to output time signal */ + LONG *timeIn, /*!< pointer to input time signal */ + LONG *timeOut, /*!< pointer to output time signal */ HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */ - INT_PCM *timeOutRight, /*!< pointer to output time signal */ + LONG *timeOutRight, /*!< pointer to output time signal */ INT strideOut, /*!< Time data traversal strideOut */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */ const int applyProcessing, /*!< Flag for SBR operation */ - HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize); + HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize, + const INT sbrInDataHeadroom); SBR_ERROR createSbrDec(SBR_CHANNEL *hSbrChannel, HANDLE_SBR_HEADER_DATA hHeaderData, diff --git a/libSBRdec/src/sbr_ram.h b/libSBRdec/src/sbr_ram.h index e00f8b5..452f835 100644 --- a/libSBRdec/src/sbr_ram.h +++ b/libSBRdec/src/sbr_ram.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 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -170,6 +170,9 @@ struct SBR_DECODER_INSTANCE { flushed consecutively. */ UINT flags; + + INT sbrInDataHeadroom; /* Headroom of the SBR input time signal to prevent + clipping */ }; H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT) diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp index 52403f6..6e4aad4 100644 --- a/libSBRdec/src/sbrdecoder.cpp +++ b/libSBRdec/src/sbrdecoder.cpp @@ -155,7 +155,7 @@ amm-info@iis.fraunhofer.de /* Decoder library info */ #define SBRDECODER_LIB_VL0 3 -#define SBRDECODER_LIB_VL1 0 +#define SBRDECODER_LIB_VL1 1 #define SBRDECODER_LIB_VL2 0 #define SBRDECODER_LIB_TITLE "SBR Decoder" #ifdef __ANDROID__ @@ -1570,10 +1570,10 @@ bail: * \return SBRDEC_OK if successfull, else error code */ static SBR_ERROR sbrDecoder_DecodeElement( - HANDLE_SBRDECODER self, QDOM_PCM *input, INT_PCM *timeData, - const int timeDataSize, const FDK_channelMapDescr *const mapDescr, - const int mapIdx, int channelIndex, const int elementIndex, - const int numInChannels, int *numOutChannels, const int psPossible) { + HANDLE_SBRDECODER self, LONG *input, LONG *timeData, const int timeDataSize, + const FDK_channelMapDescr *const mapDescr, const int mapIdx, + int channelIndex, const int elementIndex, const int numInChannels, + int *numOutChannels, const int psPossible) { SBR_DECODER_ELEMENT *hSbrElement = self->pSbrElement[elementIndex]; HANDLE_SBR_CHANNEL *pSbrChannel = self->pSbrElement[elementIndex]->pSbrChannel; @@ -1743,7 +1743,7 @@ static SBR_ERROR sbrDecoder_DecodeElement( timeData + offset1, strideOut, hSbrHeader, hFrameDataLeft, &pSbrChannel[0]->prevFrameData, (hSbrHeader->syncState == SBR_ACTIVE), h_ps_d, self->flags, - codecFrameSize); + codecFrameSize, self->sbrInDataHeadroom); if (stereo) { /* Process right channel */ @@ -1751,7 +1751,7 @@ static SBR_ERROR sbrDecoder_DecodeElement( timeData + offset1, NULL, NULL, strideOut, hSbrHeader, hFrameDataRight, &pSbrChannel[1]->prevFrameData, (hSbrHeader->syncState == SBR_ACTIVE), NULL, self->flags, - codecFrameSize); + codecFrameSize, self->sbrInDataHeadroom); } C_ALLOC_SCRATCH_END(pPsScratch, struct PS_DEC_COEFFICIENTS, 1) @@ -1771,14 +1771,14 @@ static SBR_ERROR sbrDecoder_DecodeElement( int copyFrameSize = codecFrameSize * self->pQmfDomain->QmfDomainOut->fb.no_channels; copyFrameSize /= self->pQmfDomain->QmfDomainIn->fb.no_channels; - INT_PCM *ptr; + LONG *ptr; INT i; FDK_ASSERT(strideOut == 2); ptr = timeData; for (i = copyFrameSize >> 1; i--;) { - INT_PCM tmp; /* This temporal variable is required because some - compilers can't do *ptr++ = *ptr++ correctly. */ + LONG tmp; /* This temporal variable is required because some compilers + can't do *ptr++ = *ptr++ correctly. */ tmp = *ptr++; *ptr++ = tmp; tmp = *ptr++; @@ -1791,12 +1791,13 @@ static SBR_ERROR sbrDecoder_DecodeElement( return errorStatus; } -SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input, - INT_PCM *timeData, const int timeDataSize, - int *numChannels, int *sampleRate, +SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData, + const int timeDataSize, int *numChannels, + int *sampleRate, const FDK_channelMapDescr *const mapDescr, const int mapIdx, const int coreDecodedOk, - UCHAR *psDecoded) { + UCHAR *psDecoded, const INT inDataHeadroom, + INT *outDataHeadroom) { SBR_ERROR errorStatus = SBRDEC_OK; int psPossible; @@ -1833,6 +1834,9 @@ SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input, psPossible = 0; } + self->sbrInDataHeadroom = inDataHeadroom; + *outDataHeadroom = (INT)(8); + /* Make sure that even if no SBR data was found/parsed *psDecoded is returned * 1 if psPossible was 0. */ if (psPossible == 0) { -- cgit v1.2.3 From 261fadc3a2221829196158824a512f11268756ef Mon Sep 17 00:00:00 2001 From: Fraunhofer IIS FDK Date: Thu, 19 Dec 2019 17:28:33 +0100 Subject: Remove outdated apply_normalization parameter from legacy DRC module. Bug: 149514474 Test: atest DecoderTestXheAac DecoderTestAacDrc Change-Id: I9b894007fc9d80da6aaee09d84c2b356f2c06ff7 --- libAACdec/src/aacdec_drc.cpp | 22 +++------------------- libAACdec/src/aacdec_drc.h | 1 - libAACdec/src/aacdec_drc_types.h | 1 - libAACdec/src/aacdecoder.cpp | 3 --- 4 files changed, 3 insertions(+), 24 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/src/aacdec_drc.cpp b/libAACdec/src/aacdec_drc.cpp index 6066a64..b6f5b49 100644 --- a/libAACdec/src/aacdec_drc.cpp +++ b/libAACdec/src/aacdec_drc.cpp @@ -190,7 +190,6 @@ void aacDecoder_drcInit(HANDLE_AAC_DRC self) { pParams->usrBoost = FL2FXCONST_DBL(0.0f); pParams->targetRefLevel = 96; pParams->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES; - pParams->applyDigitalNorm = ON; pParams->applyHeavyCompression = OFF; pParams->usrApplyHeavyCompression = OFF; @@ -274,11 +273,8 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam(HANDLE_AAC_DRC self, return AAC_DEC_INVALID_HANDLE; } if (value < 0) { - self->params.applyDigitalNorm = OFF; self->params.targetRefLevel = -1; } else { - /* ref_level must be between 0 and MAX_REFERENCE_LEVEL, inclusive */ - self->params.applyDigitalNorm = ON; if (self->params.targetRefLevel != (SCHAR)value) { self->params.targetRefLevel = (SCHAR)value; self->progRefLevel = (SCHAR)value; /* Always set the program reference @@ -289,16 +285,6 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam(HANDLE_AAC_DRC self, self->update = 1; } break; - case APPLY_NORMALIZATION: - if ((value != OFF) && (value != ON)) { - return AAC_DEC_SET_PARAM_FAIL; - } - if (self == NULL) { - return AAC_DEC_INVALID_HANDLE; - } - /* Store new parameter value */ - self->params.applyDigitalNorm = (UCHAR)value; - break; case APPLY_HEAVY_COMPRESSION: if ((value != OFF) && (value != ON)) { return AAC_DEC_SET_PARAM_FAIL; @@ -926,11 +912,9 @@ void aacDecoder_drcApply(HANDLE_AAC_DRC self, void *pSbrDec, FDK_ASSERT(0); } } - if (self->params.applyDigitalNorm == OFF) { - /* Reset normalization gain since this module must not apply it */ - norm_mantissa = FL2FXCONST_DBL(0.5f); - norm_exponent = 1; - } + /* Reset normalization gain since this module must not apply it */ + norm_mantissa = FL2FXCONST_DBL(0.5f); + norm_exponent = 1; /* calc scale factors */ for (band = 0; band < numBands; band++) { diff --git a/libAACdec/src/aacdec_drc.h b/libAACdec/src/aacdec_drc.h index 1873c5b..76a44d6 100644 --- a/libAACdec/src/aacdec_drc.h +++ b/libAACdec/src/aacdec_drc.h @@ -130,7 +130,6 @@ typedef enum { TARGET_REF_LEVEL, DRC_BS_DELAY, DRC_DATA_EXPIRY_FRAME, - APPLY_NORMALIZATION, APPLY_HEAVY_COMPRESSION, DEFAULT_PRESENTATION_MODE, ENCODER_TARGET_LEVEL, diff --git a/libAACdec/src/aacdec_drc_types.h b/libAACdec/src/aacdec_drc_types.h index 873bd60..d4393f7 100644 --- a/libAACdec/src/aacdec_drc_types.h +++ b/libAACdec/src/aacdec_drc_types.h @@ -168,7 +168,6 @@ typedef struct { UINT expiryFrame; UCHAR bsDelayEnable; - UCHAR applyDigitalNorm; AACDEC_DRC_PARAMETER_HANDLING defaultPresentationMode; UCHAR encoderTargetLevel; diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index 6a0254d..965631b 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -3177,9 +3177,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( FDKmemcpy(drcChMap, self->chMapping, (8) * sizeof(UCHAR)); } - /* Turn off DRC modules level normalization in digital domain. */ - aacDecoder_drcSetParam(self->hDrcInfo, APPLY_NORMALIZATION, 0); - /* deactivate legacy DRC in case uniDrc is active, i.e. uniDrc payload is * present and one of DRC or Loudness Normalization is switched on */ aacDecoder_drcSetParam( -- cgit v1.2.3 From 7f7bbce892d9e8ed45e4ee2ee7dcefc1cd87a763 Mon Sep 17 00:00:00 2001 From: Anuj Joshi Date: Fri, 27 Mar 2020 10:43:08 +0530 Subject: Stop using __DATE__/__TIME__ on all builds Test: mmma external/libaac Bug: 151595970 Change-Id: Icd937cad3e4e2f70a5486cca424544eb410be26f --- Android.bp | 1 + libAACdec/src/aacdecoder_lib.cpp | 2 +- libAACenc/src/aacenc_lib.cpp | 2 +- libDRCdec/src/FDK_drcDecLib.cpp | 2 +- libFDK/src/FDK_core.cpp | 2 +- libMpegTPDec/src/tpdec_lib.cpp | 2 +- libMpegTPEnc/src/tpenc_lib.cpp | 2 +- libPCMutils/src/version.h | 2 +- libSACdec/src/sac_dec_lib.cpp | 2 +- libSACenc/src/sacenc_lib.cpp | 2 +- libSBRdec/src/sbrdecoder.cpp | 2 +- libSBRenc/src/sbr_encoder.cpp | 2 +- 12 files changed, 12 insertions(+), 11 deletions(-) (limited to 'libAACdec') diff --git a/Android.bp b/Android.bp index 80a0347..4d52b71 100644 --- a/Android.bp +++ b/Android.bp @@ -23,6 +23,7 @@ cc_library_static { "-Wuninitialized", "-Wno-self-assign", "-Wno-implicit-fallthrough", + "-DSUPPRESS_BUILD_DATE_INFO", ], sanitize: { misc_undefined:[ diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp index 86ec899..3a1fc48 100644 --- a/libAACdec/src/aacdecoder_lib.cpp +++ b/libAACdec/src/aacdecoder_lib.cpp @@ -122,7 +122,7 @@ amm-info@iis.fraunhofer.de #define AACDECODER_LIB_VL1 1 #define AACDECODER_LIB_VL2 2 #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 diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp index 0f0094f..a152c0d 100644 --- a/libAACenc/src/aacenc_lib.cpp +++ b/libAACenc/src/aacenc_lib.cpp @@ -112,7 +112,7 @@ amm-info@iis.fraunhofer.de #define AACENCODER_LIB_VL1 0 #define AACENCODER_LIB_VL2 0 #define AACENCODER_LIB_TITLE "AAC Encoder" -#ifdef __ANDROID__ +#ifdef SUPPRESS_BUILD_DATE_INFO #define AACENCODER_LIB_BUILD_DATE "" #define AACENCODER_LIB_BUILD_TIME "" #else diff --git a/libDRCdec/src/FDK_drcDecLib.cpp b/libDRCdec/src/FDK_drcDecLib.cpp index 4f8ebc7..98a73ca 100644 --- a/libDRCdec/src/FDK_drcDecLib.cpp +++ b/libDRCdec/src/FDK_drcDecLib.cpp @@ -112,7 +112,7 @@ amm-info@iis.fraunhofer.de #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/libFDK/src/FDK_core.cpp b/libFDK/src/FDK_core.cpp index 75ea8a2..52f8915 100644 --- a/libFDK/src/FDK_core.cpp +++ b/libFDK/src/FDK_core.cpp @@ -107,7 +107,7 @@ amm-info@iis.fraunhofer.de #define FDK_TOOLS_LIB_VL1 0 #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/libMpegTPDec/src/tpdec_lib.cpp b/libMpegTPDec/src/tpdec_lib.cpp index 506aed3..0b8ff24 100644 --- a/libMpegTPDec/src/tpdec_lib.cpp +++ b/libMpegTPDec/src/tpdec_lib.cpp @@ -1750,7 +1750,7 @@ TRANSPORTDEC_ERROR transportDec_GetLibInfo(LIB_INFO *info) { 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/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 @@ TRANSPORTENC_ERROR transportEnc_GetLibInfo(LIB_INFO *info) { 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 fa31af1..8e537ee 100644 --- a/libPCMutils/src/version.h +++ b/libPCMutils/src/version.h @@ -108,7 +108,7 @@ amm-info@iis.fraunhofer.de #define PCMUTIL_LIB_VL1 0 #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/src/sac_dec_lib.cpp b/libSACdec/src/sac_dec_lib.cpp index bf6dedf..a07e1c9 100644 --- a/libSACdec/src/sac_dec_lib.cpp +++ b/libSACdec/src/sac_dec_lib.cpp @@ -1800,7 +1800,7 @@ int mpegSurroundDecoder_GetLibInfo(LIB_INFO *info) { info += i; info->module_id = FDK_MPSDEC; -#ifdef __ANDROID__ +#ifdef SUPPRESS_BUILD_DATE_INFO info->build_date = ""; info->build_time = ""; #else 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 @@ Description of file contents #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/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp index c827ced..89a2338 100644 --- a/libSBRdec/src/sbrdecoder.cpp +++ b/libSBRdec/src/sbrdecoder.cpp @@ -160,7 +160,7 @@ amm-info@iis.fraunhofer.de #define SBRDECODER_LIB_VL1 0 #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/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp index df9e996..c1e083f 100644 --- a/libSBRenc/src/sbr_encoder.cpp +++ b/libSBRenc/src/sbr_encoder.cpp @@ -2560,7 +2560,7 @@ INT sbrEncoder_GetLibInfo(LIB_INFO *info) { 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 -- cgit v1.2.3 From 559379e08bb0cf26092a5d3cbbbfe0396792cbad Mon Sep 17 00:00:00 2001 From: Martin Storsjo Date: Tue, 17 Nov 2020 12:41:48 +0200 Subject: Update the version numbers in the public installed headers --- libAACdec/include/aacdecoder_lib.h | 4 ++-- libAACenc/include/aacenc_lib.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'libAACdec') diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h index 6572541..06272df 100644 --- a/libAACdec/include/aacdecoder_lib.h +++ b/libAACdec/include/aacdecoder_lib.h @@ -434,8 +434,8 @@ Where N equals to CStreamInfo::frameSize . #include "genericStds.h" #define AACDECODER_LIB_VL0 3 -#define AACDECODER_LIB_VL1 1 -#define AACDECODER_LIB_VL2 2 +#define AACDECODER_LIB_VL1 2 +#define AACDECODER_LIB_VL2 0 /** * \brief AAC decoder error codes. diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h index f1965aa..4d6a327 100644 --- a/libAACenc/include/aacenc_lib.h +++ b/libAACenc/include/aacenc_lib.h @@ -1007,7 +1007,7 @@ internally. #define AACENCODER_LIB_VL0 4 #define AACENCODER_LIB_VL1 0 -#define AACENCODER_LIB_VL2 0 +#define AACENCODER_LIB_VL2 1 /** * AAC encoder error codes. -- cgit v1.2.3