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-rw-r--r--libAACdec/src/FDK_delay.cpp26
-rw-r--r--libAACdec/src/FDK_delay.h6
-rw-r--r--libAACdec/src/aac_rom.h3
-rw-r--r--libAACdec/src/aacdecoder.cpp49
-rw-r--r--libAACdec/src/aacdecoder.h11
-rw-r--r--libAACdec/src/aacdecoder_lib.cpp232
-rw-r--r--libAACdec/src/block.cpp12
-rw-r--r--libAACdec/src/block.h8
-rw-r--r--libAACdec/src/conceal.cpp42
-rw-r--r--libAACdec/src/conceal.h4
-rw-r--r--libAACdec/src/conceal_types.h4
-rw-r--r--libAACdec/src/ldfiltbank.cpp35
-rw-r--r--libAACdec/src/ldfiltbank.h5
-rw-r--r--libAACdec/src/usacdec_lpd.cpp34
-rw-r--r--libAACdec/src/usacdec_lpd.h19
15 files changed, 273 insertions, 217 deletions
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
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-28 15:46:16 +0000