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-rw-r--r--libAACdec/include/aacdecoder_lib.h107
-rw-r--r--libAACdec/src/aacdec_drc.cpp6
-rw-r--r--libAACdec/src/aacdec_hcrs.cpp5
-rw-r--r--libAACdec/src/aacdecoder.cpp50
-rw-r--r--libAACdec/src/aacdecoder_lib.cpp60
-rw-r--r--libAACdec/src/block.cpp12
-rw-r--r--libAACdec/src/channel.cpp1
-rw-r--r--libAACdec/src/channelinfo.h2
-rw-r--r--libAACdec/src/conceal.cpp1
-rw-r--r--libAACdec/src/ldfiltbank.cpp8
-rw-r--r--libAACdec/src/usacdec_acelp.cpp12
-rw-r--r--libAACdec/src/usacdec_const.h1
-rw-r--r--libAACdec/src/usacdec_fac.cpp26
-rw-r--r--libAACdec/src/usacdec_fac.h2
-rw-r--r--libAACdec/src/usacdec_lpd.cpp4
15 files changed, 183 insertions, 114 deletions
diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h
index 3a9b910..6c2fda4 100644
--- a/libAACdec/include/aacdecoder_lib.h
+++ b/libAACdec/include/aacdecoder_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
@@ -741,66 +741,77 @@ typedef enum {
::CONCEAL_INTER. only some AOTs are
supported). \n */
AAC_DRC_BOOST_FACTOR =
- 0x0200, /*!< Dynamic Range Control: Scaling factor for boosting gain
- values. Defines how the boosting DRC factors (conveyed in the
- bitstream) will be applied to the decoded signal. The valid
- values range from 0 (don't apply boost factors) to 127 (fully
- apply boost factors). Default value is 0. */
- AAC_DRC_ATTENUATION_FACTOR =
- 0x0201, /*!< Dynamic Range Control: Scaling factor for attenuating gain
- values. Same as
- ::AAC_DRC_BOOST_FACTOR but for attenuating DRC factors. */
+ 0x0200, /*!< MPEG-4 / MPEG-D Dynamic Range Control (DRC): Scaling factor
+ for boosting gain values. Defines how the boosting DRC factors
+ (conveyed in the bitstream) will be applied to the decoded
+ signal. The valid values range from 0 (don't apply boost
+ factors) to 127 (fully apply boost factors). Default value is 0
+ for MPEG-4 DRC and 127 for MPEG-D DRC. */
+ AAC_DRC_ATTENUATION_FACTOR = 0x0201, /*!< MPEG-4 / MPEG-D DRC: Scaling factor
+ for attenuating gain values. Same as
+ ::AAC_DRC_BOOST_FACTOR but for
+ attenuating DRC factors. */
AAC_DRC_REFERENCE_LEVEL =
- 0x0202, /*!< Dynamic Range Control (DRC): Target reference level. Defines
- the level below full-scale (quantized in steps of 0.25dB) to
- which the output audio signal will be normalized to by the DRC
- module. The parameter controls loudness normalization for both
- MPEG-4 DRC and MPEG-D DRC. The valid values range from 40 (-10
- dBFS) to 127 (-31.75 dBFS). Any value smaller than 0 switches
- off loudness normalization and MPEG-4 DRC. By default, loudness
- normalization and MPEG-4 DRC is switched off. */
+ 0x0202, /*!< MPEG-4 / MPEG-D DRC: Target reference level / decoder target
+ loudness.\n Defines the level below full-scale (quantized in
+ steps of 0.25dB) to which the output audio signal will be
+ normalized to by the DRC module.\n The parameter controls
+ loudness normalization for both MPEG-4 DRC and MPEG-D DRC. The
+ valid values range from 40 (-10 dBFS) to 127 (-31.75 dBFS).\n
+ Example values:\n
+ 124 (-31 dBFS) for audio/video receivers (AVR) or other
+ devices allowing audio playback with high dynamic range,\n 96
+ (-24 dBFS) for TV sets or equivalent devices (default),\n 64
+ (-16 dBFS) for mobile devices where the dynamic range of audio
+ playback is restricted.\n Any value smaller than 0 switches off
+ loudness normalization and MPEG-4 DRC. */
AAC_DRC_HEAVY_COMPRESSION =
- 0x0203, /*!< Dynamic Range Control: En-/Disable DVB specific heavy
- compression (aka RF mode). If set to 1, the decoder will apply
- the compression values from the DVB specific ancillary data
- field. At the same time the MPEG-4 Dynamic Range Control tool
- will be disabled. By default, heavy compression is disabled. */
+ 0x0203, /*!< MPEG-4 DRC: En-/Disable DVB specific heavy compression (aka
+ RF mode). If set to 1, the decoder will apply the compression
+ values from the DVB specific ancillary data field. At the same
+ time the MPEG-4 Dynamic Range Control tool will be disabled. By
+ default, heavy compression is disabled. */
AAC_DRC_DEFAULT_PRESENTATION_MODE =
- 0x0204, /*!< Dynamic Range Control: Default presentation mode (DRC
- parameter handling). \n Defines the handling of the DRC
- parameters boost factor, attenuation factor and heavy
- compression, if no presentation mode is indicated in the
- bitstream.\n For options, see
- ::AAC_DRC_DEFAULT_PRESENTATION_MODE_OPTIONS.\n Default:
+ 0x0204, /*!< MPEG-4 DRC: Default presentation mode (DRC parameter
+ handling). \n Defines the handling of the DRC parameters boost
+ factor, attenuation factor and heavy compression, if no
+ presentation mode is indicated in the bitstream.\n For options,
+ see ::AAC_DRC_DEFAULT_PRESENTATION_MODE_OPTIONS.\n Default:
::AAC_DRC_PARAMETER_HANDLING_DISABLED */
AAC_DRC_ENC_TARGET_LEVEL =
- 0x0205, /*!< Dynamic Range Control: Encoder target level for light (i.e.
- not heavy) compression.\n If known, this declares the target
- reference level that was assumed at the encoder for calculation
- of limiting gains. The valid values range from 0 (full-scale)
- to 127 (31.75 dB below full-scale). This parameter is used only
+ 0x0205, /*!< MPEG-4 DRC: Encoder target level for light (i.e. not heavy)
+ compression.\n If known, this declares the target reference
+ level that was assumed at the encoder for calculation of
+ limiting gains. The valid values range from 0 (full-scale) to
+ 127 (31.75 dB below full-scale). This parameter is used only
with ::AAC_DRC_PARAMETER_HANDLING_ENABLED and ignored
otherwise.\n Default: 127 (worst-case assumption).\n */
+ AAC_UNIDRC_SET_EFFECT = 0x0206, /*!< MPEG-D DRC: Request a DRC effect type for
+ selection of a DRC set.\n Supported indices
+ are:\n -1: DRC off. Completely disables
+ MPEG-D DRC.\n 0: None (default). Disables
+ MPEG-D DRC, but automatically enables DRC
+ if necessary to prevent clipping.\n 1: Late
+ night\n 2: Noisy environment\n 3: Limited
+ playback range\n 4: Low playback level\n 5:
+ Dialog enhancement\n 6: General
+ compression. Used for generally enabling
+ MPEG-D DRC without particular request.\n */
+ AAC_UNIDRC_ALBUM_MODE =
+ 0x0207, /*!< MPEG-D DRC: Enable album mode. 0: Disabled (default), 1:
+ Enabled.\n Disabled album mode leads to application of gain
+ 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_TPDEC_CLEAR_BUFFER =
- 0x0603, /*!< Clear internal bit stream buffer of transport layers. The
- decoder will start decoding at new data passed after this event
- and any previous data is discarded. */
- AAC_UNIDRC_SET_EFFECT = 0x0903 /*!< MPEG-D DRC: Request a DRC effect type for
- selection of a DRC set.\n Supported indices
- are:\n -1: DRC off. Completely disables
- MPEG-D DRC.\n 0: None (default). Disables
- MPEG-D DRC, but automatically enables DRC if
- necessary to prevent clipping.\n 1: Late
- night\n 2: Noisy environment\n 3: Limited
- playback range\n 4: Low playback level\n 5:
- Dialog enhancement\n 6: General compression.
- Used for generally enabling MPEG-D DRC
- without particular request.\n */
+ 0x0603 /*!< Clear internal bit stream buffer of transport layers. The
+ decoder will start decoding at new data passed after this event
+ and any previous data is discarded. */
} AACDEC_PARAM;
diff --git a/libAACdec/src/aacdec_drc.cpp b/libAACdec/src/aacdec_drc.cpp
index 922a09e..4129d0f 100644
--- a/libAACdec/src/aacdec_drc.cpp
+++ b/libAACdec/src/aacdec_drc.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -174,9 +174,9 @@ void aacDecoder_drcInit(HANDLE_AAC_DRC self) {
pParams->usrCut = FL2FXCONST_DBL(0.0f);
pParams->boost = FL2FXCONST_DBL(0.0f);
pParams->usrBoost = FL2FXCONST_DBL(0.0f);
- pParams->targetRefLevel = -1;
+ pParams->targetRefLevel = 96;
pParams->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES;
- pParams->applyDigitalNorm = OFF;
+ pParams->applyDigitalNorm = ON;
pParams->applyHeavyCompression = OFF;
pParams->usrApplyHeavyCompression = OFF;
diff --git a/libAACdec/src/aacdec_hcrs.cpp b/libAACdec/src/aacdec_hcrs.cpp
index 1d5aa27..d2bc867 100644
--- a/libAACdec/src/aacdec_hcrs.cpp
+++ b/libAACdec/src/aacdec_hcrs.cpp
@@ -367,7 +367,10 @@ static UINT InitSegmentBitfield(UINT *pNumSegment,
UINT tempWord;
USHORT numValidSegment;
- *pNumWordForBitfield = ((*pNumSegment - 1) >> THIRTYTWO_LOG_DIV_TWO_LOG) + 1;
+ *pNumWordForBitfield =
+ (*pNumSegment == 0)
+ ? 0
+ : ((*pNumSegment - 1) >> THIRTYTWO_LOG_DIV_TWO_LOG) + 1;
/* loop over all words, which are completely used or only partial */
/* bit in pSegmentBitfield is zero if segment is empty; bit in
diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp
index 24907ee..7617937 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 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -234,7 +234,8 @@ void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self) {
MODE_HQ))) { /* MPS decoder does support the requested mode. */
break;
}
- } /* Fall-through: */
+ }
+ FDK_FALLTHROUGH;
default:
if (self->qmfModeUser == NOT_DEFINED) {
/* Revert in case mpegSurroundDecoder_SetParam() fails. */
@@ -538,13 +539,7 @@ static int CProgramConfigElement_Read(HANDLE_FDK_BITSTREAM bs,
sizeof(CProgramConfig)); /* Store the complete PCE */
pceStatus = 1; /* New PCE but no change of config */
break;
- case 2: /* The number of channels are identical but not the config */
- if (channelConfig == 0) {
- FDKmemcpy(pce, tmpPce,
- sizeof(CProgramConfig)); /* Store the complete PCE */
- pceStatus = 2; /* Decoder needs re-configuration */
- }
- break;
+ case 2: /* The number of channels are identical but not the config */
case -1: /* The channel configuration is completely different */
pceStatus = -1; /* Not supported! */
break;
@@ -775,7 +770,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PreRollExtensionPayloadParse(
/* For every AU get length and offset in the bitstream */
prerollAULength[i] = escapedValue(hBs, 16, 16, 0);
if (prerollAULength[i] > 0) {
- prerollAUOffset[i] = auStartAnchor - FDKgetValidBits(hBs);
+ prerollAUOffset[i] = auStartAnchor - (INT)FDKgetValidBits(hBs);
independencyFlag = FDKreadBit(hBs);
if (i == 0 && !independencyFlag) {
*numPrerollAU = 0;
@@ -938,6 +933,7 @@ static AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse(
case EXT_SBR_DATA_CRC:
crcFlag = 1;
+ FDK_FALLTHROUGH;
case EXT_SBR_DATA:
if (IS_CHANNEL_ELEMENT(previous_element)) {
SBR_ERROR sbrError;
@@ -1076,6 +1072,7 @@ static AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse(
* intentional. */
break;
}
+ FDK_FALLTHROUGH;
case EXT_FIL:
@@ -1108,12 +1105,13 @@ static AAC_DECODER_ERROR aacDecoder_ParseExplicitMpsAndSbr(
/* get the remaining bits of this frame */
bitCnt = transportDec_GetAuBitsRemaining(self->hInput, 0);
- if ((bitCnt > 0) && (self->flags[0] & AC_SBR_PRESENT) &&
+ if ((self->flags[0] & AC_SBR_PRESENT) &&
(self->flags[0] & (AC_USAC | AC_RSVD50 | AC_ELD | AC_DRM))) {
SBR_ERROR err = SBRDEC_OK;
int chElIdx, numChElements = el_cnt[ID_SCE] + el_cnt[ID_CPE] +
el_cnt[ID_LFE] + el_cnt[ID_USAC_SCE] +
el_cnt[ID_USAC_CPE] + el_cnt[ID_USAC_LFE];
+ INT bitCntTmp = bitCnt;
if (self->flags[0] & AC_USAC) {
chElIdx = numChElements - 1;
@@ -1123,6 +1121,7 @@ static AAC_DECODER_ERROR aacDecoder_ParseExplicitMpsAndSbr(
for (; chElIdx < numChElements; chElIdx += 1) {
MP4_ELEMENT_ID sbrType;
+ SBR_ERROR errTmp;
if (self->flags[0] & (AC_USAC)) {
FDK_ASSERT((self->elements[element_index] == ID_USAC_SCE) ||
(self->elements[element_index] == ID_USAC_CPE));
@@ -1132,19 +1131,21 @@ static AAC_DECODER_ERROR aacDecoder_ParseExplicitMpsAndSbr(
: ID_SCE;
} else
sbrType = self->elements[chElIdx];
- err = sbrDecoder_Parse(self->hSbrDecoder, bs, self->pDrmBsBuffer,
- self->drmBsBufferSize, &bitCnt, -1,
- self->flags[0] & AC_SBRCRC, sbrType, chElIdx,
- self->flags[0], self->elFlags);
- if (err != SBRDEC_OK) {
- break;
+ errTmp = sbrDecoder_Parse(self->hSbrDecoder, bs, self->pDrmBsBuffer,
+ self->drmBsBufferSize, &bitCnt, -1,
+ self->flags[0] & AC_SBRCRC, sbrType, chElIdx,
+ self->flags[0], self->elFlags);
+ if (errTmp != SBRDEC_OK) {
+ err = errTmp;
+ bitCntTmp = bitCnt;
+ bitCnt = 0;
}
}
switch (err) {
case SBRDEC_PARSE_ERROR:
/* Can not go on parsing because we do not
know the length of the SBR extension data. */
- FDKpushFor(bs, bitCnt);
+ FDKpushFor(bs, bitCntTmp);
bitCnt = 0;
break;
case SBRDEC_OK:
@@ -1495,11 +1496,13 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
switch (asc->m_aot) {
case AOT_AAC_LC:
self->streamInfo.profile = 1;
+ FDK_FALLTHROUGH;
case AOT_ER_AAC_SCAL:
if (asc->m_sc.m_gaSpecificConfig.m_layer > 0) {
/* aac_scalable_extension_element() currently not supported. */
return AAC_DEC_UNSUPPORTED_FORMAT;
}
+ FDK_FALLTHROUGH;
case AOT_SBR:
case AOT_PS:
case AOT_ER_AAC_LC:
@@ -1812,6 +1815,9 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
self->useLdQmfTimeAlign =
asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
}
+ if (self->sbrEnabled != asc->m_sbrPresentFlag) {
+ ascChanged = 1;
+ }
}
self->streamInfo.extAot = asc->m_extensionAudioObjectType;
@@ -3024,9 +3030,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
aacChannels = 0;
}
- if (TRANSPORTDEC_OK != transportDec_CrcCheck(self->hInput)) {
- ErrorStatus = AAC_DEC_CRC_ERROR;
- self->frameOK = 0;
+ if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) {
+ if (TRANSPORTDEC_OK != transportDec_CrcCheck(self->hInput)) {
+ ErrorStatus = AAC_DEC_CRC_ERROR;
+ self->frameOK = 0;
+ }
}
/* Ensure that in case of concealment a proper error status is set. */
diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index d98cf5a..86ec899 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_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
@@ -119,8 +119,8 @@ amm-info@iis.fraunhofer.de
/* Decoder library info */
#define AACDECODER_LIB_VL0 3
-#define AACDECODER_LIB_VL1 0
-#define AACDECODER_LIB_VL2 0
+#define AACDECODER_LIB_VL1 1
+#define AACDECODER_LIB_VL2 2
#define AACDECODER_LIB_TITLE "AAC Decoder Lib"
#ifdef __ANDROID__
#define AACDECODER_LIB_BUILD_DATE ""
@@ -368,9 +368,26 @@ static INT aacDecoder_CtrlCFGChangeCallback(
return errTp;
}
+static INT aacDecoder_SbrCallback(
+ void *handle, HANDLE_FDK_BITSTREAM hBs, const INT sampleRateIn,
+ const INT sampleRateOut, const INT samplesPerFrame,
+ const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID,
+ const INT elementIndex, const UCHAR harmonicSBR,
+ const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged,
+ const INT downscaleFactor) {
+ HANDLE_SBRDECODER self = (HANDLE_SBRDECODER)handle;
+
+ INT errTp = sbrDecoder_Header(self, hBs, sampleRateIn, sampleRateOut,
+ samplesPerFrame, coreCodec, elementID,
+ elementIndex, harmonicSBR, stereoConfigIndex,
+ configMode, configChanged, downscaleFactor);
+
+ return errTp;
+}
+
static INT aacDecoder_SscCallback(void *handle, HANDLE_FDK_BITSTREAM hBs,
const AUDIO_OBJECT_TYPE coreCodec,
- const INT samplingRate,
+ const INT samplingRate, const INT frameSize,
const INT stereoConfigIndex,
const INT coreSbrFrameLengthIndex,
const INT configBytes, const UCHAR configMode,
@@ -381,8 +398,8 @@ static INT aacDecoder_SscCallback(void *handle, HANDLE_FDK_BITSTREAM hBs,
err = mpegSurroundDecoder_Config(
(CMpegSurroundDecoder *)hAacDecoder->pMpegSurroundDecoder, hBs, coreCodec,
- samplingRate, stereoConfigIndex, coreSbrFrameLengthIndex, configBytes,
- configMode, configChanged);
+ samplingRate, frameSize, stereoConfigIndex, coreSbrFrameLengthIndex,
+ configBytes, configMode, configChanged);
switch (err) {
case MPS_UNSUPPORTED_CONFIG:
@@ -617,6 +634,7 @@ static AAC_DECODER_ERROR setConcealMethod(
switch (err) {
case PCMDMX_INVALID_HANDLE:
errorStatus = AAC_DEC_INVALID_HANDLE;
+ break;
case PCMDMX_OK:
break;
default:
@@ -805,11 +823,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_SetParam(
case AAC_DRC_ATTENUATION_FACTOR:
/* DRC compression factor (where 0 is no and 127 is max compression) */
errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_CUT_SCALE, value);
+ uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_COMPRESS,
+ value * (FL2FXCONST_DBL(0.5f / 127.0f)));
break;
case AAC_DRC_BOOST_FACTOR:
/* DRC boost factor (where 0 is no and 127 is max boost) */
errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_BOOST_SCALE, value);
+ uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_BOOST,
+ value * (FL2FXCONST_DBL(0.5f / 127.0f)));
break;
case AAC_DRC_REFERENCE_LEVEL:
@@ -853,6 +875,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_SetParam(
uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_EFFECT_TYPE,
(FIXP_DBL)value);
break;
+ case AAC_UNIDRC_ALBUM_MODE:
+ uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_ALBUM_MODE,
+ (FIXP_DBL)value);
+ break;
+
case AAC_TPDEC_CLEAR_BUFFER:
errTp = transportDec_SetParam(hTpDec, TPDEC_PARAM_RESET, 1);
self->streamInfo.numLostAccessUnits = 0;
@@ -959,7 +986,7 @@ LINKSPEC_CPP HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt,
goto bail;
}
aacDec->qmfModeUser = NOT_DEFINED;
- transportDec_RegisterSbrCallback(aacDec->hInput, (cbSbr_t)sbrDecoder_Header,
+ transportDec_RegisterSbrCallback(aacDec->hInput, aacDecoder_SbrCallback,
(void *)aacDec->hSbrDecoder);
if (mpegSurroundDecoder_Open(
@@ -1380,9 +1407,13 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
mpegSurroundDecoder_ConfigureQmfDomain(
(CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface,
(UINT)self->streamInfo.aacSampleRate, self->streamInfo.aot);
- self->qmfDomain.globalConf.nQmfTimeSlots_requested =
- self->streamInfo.aacSamplesPerFrame /
- self->qmfDomain.globalConf.nBandsAnalysis_requested;
+ if (self->qmfDomain.globalConf.nBandsAnalysis_requested > 0) {
+ self->qmfDomain.globalConf.nQmfTimeSlots_requested =
+ self->streamInfo.aacSamplesPerFrame /
+ self->qmfDomain.globalConf.nBandsAnalysis_requested;
+ } else {
+ self->qmfDomain.globalConf.nQmfTimeSlots_requested = 0;
+ }
}
self->qmfDomain.globalConf.TDinput = pTimeData;
@@ -1645,6 +1676,13 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
reverseOutChannelMap[ch] = ch;
}
+ /* Update sampleRate and frameSize. This may be necessary in case of
+ * implicit SBR signaling */
+ FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_SAMPLE_RATE,
+ self->streamInfo.sampleRate);
+ FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_FRAME_SIZE,
+ self->streamInfo.frameSize);
+
/* If SBR and/or MPS is active, the DRC gains are aligned to the QMF
domain signal before the QMF synthesis. Therefore the DRC gains
need to be delayed by the QMF synthesis delay. */
@@ -1865,7 +1903,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
} /* USAC DASH IPF flushing possible end */
if (accessUnit < numPrerollAU) {
- FDKpushBack(hBsAu, auStartAnchor - FDKgetValidBits(hBsAu));
+ FDKpushBack(hBsAu, auStartAnchor - (INT)FDKgetValidBits(hBsAu));
} else {
if ((self->buildUpStatus == AACDEC_RSV60_BUILD_UP_ON) ||
(self->buildUpStatus == AACDEC_RSV60_BUILD_UP_ON_IN_BAND) ||
diff --git a/libAACdec/src/block.cpp b/libAACdec/src/block.cpp
index 7d2a4b9..b3d09a6 100644
--- a/libAACdec/src/block.cpp
+++ b/libAACdec/src/block.cpp
@@ -127,9 +127,11 @@ amm-info@iis.fraunhofer.de
The function reads the escape sequence from the bitstream,
if the absolute value of the quantized coefficient has the
value 16.
- A limitation is implemented to maximal 31 bits to prevent endless loops.
- If it strikes, MAX_QUANTIZED_VALUE + 1 is returned, independent of the sign of
- parameter q.
+ A limitation is implemented to maximal 21 bits according to
+ ISO/IEC 14496-3:2009(E) 4.6.3.3.
+ This limits the escape prefix to a maximum of eight 1's.
+ If more than eight 1's are read, MAX_QUANTIZED_VALUE + 1 is
+ returned, independent of the sign of parameter q.
\return quantized coefficient
*/
@@ -139,11 +141,11 @@ LONG CBlock_GetEscape(HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */
if (fAbs(q) != 16) return (q);
LONG i, off;
- for (i = 4; i < 32; i++) {
+ for (i = 4; i < 13; i++) {
if (FDKreadBit(bs) == 0) break;
}
- if (i == 32) return (MAX_QUANTIZED_VALUE + 1);
+ if (i == 13) return (MAX_QUANTIZED_VALUE + 1);
off = FDKreadBits(bs, i);
i = off + (1 << i);
diff --git a/libAACdec/src/channel.cpp b/libAACdec/src/channel.cpp
index cfffd57..a020034 100644
--- a/libAACdec/src/channel.cpp
+++ b/libAACdec/src/channel.cpp
@@ -592,6 +592,7 @@ AAC_DECODER_ERROR CChannelElement_Read(
line: ~599 */
/* Note: The missing "break" is intentional here, since we need to call
* CBlock_ReadScaleFactorData(). */
+ FDK_FALLTHROUGH;
case scale_factor_data:
if (flags & AC_ER_RVLC) {
diff --git a/libAACdec/src/channelinfo.h b/libAACdec/src/channelinfo.h
index 45a288f..4523400 100644
--- a/libAACdec/src/channelinfo.h
+++ b/libAACdec/src/channelinfo.h
@@ -359,7 +359,7 @@ typedef struct {
shouldBeUnion {
struct {
FIXP_DBL fac_data0[LFAC];
- UCHAR fac_data_e[4];
+ SCHAR fac_data_e[4];
FIXP_DBL
*fac_data[4]; /* Pointers to unused parts of pSpectralCoefficient */
diff --git a/libAACdec/src/conceal.cpp b/libAACdec/src/conceal.cpp
index 569d672..5895cb8 100644
--- a/libAACdec/src/conceal.cpp
+++ b/libAACdec/src/conceal.cpp
@@ -1894,6 +1894,7 @@ INT CConcealment_TDFading(
case ConcealState_FadeIn:
idx = cntFadeFrames;
idx -= TDFadeInStopBeforeFullLevel;
+ FDK_FALLTHROUGH;
case ConcealState_Ok:
fadeFactor = pConcealParams->fadeInFactor;
idx = (concealState == ConcealState_Ok) ? -1 : idx;
diff --git a/libAACdec/src/ldfiltbank.cpp b/libAACdec/src/ldfiltbank.cpp
index 66a5914..c7d2928 100644
--- a/libAACdec/src/ldfiltbank.cpp
+++ b/libAACdec/src/ldfiltbank.cpp
@@ -216,6 +216,7 @@ int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctData, const int mdctData_e,
int scale = mdctData_e + MDCT_OUT_HEADROOM -
LDFB_HEADROOM; /* The LDFB_HEADROOM is compensated inside
multE2_DinvF_fdk() below */
+ int i;
/* Select LD window slope */
switch (N) {
@@ -261,10 +262,11 @@ int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctData, const int mdctData_e,
}
if (gain != (FIXP_DBL)0) {
- scaleValuesWithFactor(mdctData, gain, N, scale);
- } else {
- scaleValues(mdctData, N, scale);
+ for (i = 0; i < N; i++) {
+ mdctData[i] = fMult(mdctData[i], gain);
+ }
}
+ scaleValuesSaturate(mdctData, N, scale);
/* Since all exponent and factors have been applied, current exponent is zero.
*/
diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp
index af1f488..c836c6a 100644
--- a/libAACdec/src/usacdec_acelp.cpp
+++ b/libAACdec/src/usacdec_acelp.cpp
@@ -309,7 +309,7 @@ static FIXP_DBL calc_period_factor(FIXP_DBL exc[], FIXP_SGL gain_pit,
ener_exc = (FIXP_DBL)0;
for (int i = 0; i < L_SUBFR; i++) {
ener_exc += fPow2Div2(exc[i]) >> s;
- if (ener_exc > FL2FXCONST_DBL(0.5f)) {
+ if (ener_exc >= FL2FXCONST_DBL(0.5f)) {
ener_exc >>= 1;
s++;
}
@@ -579,11 +579,11 @@ void Syn_filt(const FIXP_LPC a[], /* (i) : a[m] prediction coefficients Q12 */
L_tmp = (FIXP_DBL)0;
for (j = 0; j < M_LP_FILTER_ORDER; j++) {
- L_tmp -= fMultDiv2(a[j], y[i - (j + 1)]);
+ L_tmp -= fMultDiv2(a[j], y[i - (j + 1)]) >> (LP_FILTER_SCALE - 1);
}
- L_tmp = scaleValue(L_tmp, a_exp + 1);
- y[i] = L_tmp + x[i];
+ L_tmp = scaleValue(L_tmp, a_exp + LP_FILTER_SCALE);
+ y[i] = fAddSaturate(L_tmp, x[i]);
}
return;
@@ -631,10 +631,10 @@ void E_UTIL_residu(const FIXP_LPC *a, const INT a_exp, FIXP_DBL *x, FIXP_DBL *y,
s = (FIXP_DBL)0;
for (j = 0; j < M_LP_FILTER_ORDER; j++) {
- s += fMultDiv2(a[j], x[i - j - 1]);
+ s += fMultDiv2(a[j], x[i - j - 1]) >> (LP_FILTER_SCALE - 1);
}
- s = scaleValue(s, a_exp + 1);
+ s = scaleValue(s, a_exp + LP_FILTER_SCALE);
y[i] = fAddSaturate(s, x[i]);
}
diff --git a/libAACdec/src/usacdec_const.h b/libAACdec/src/usacdec_const.h
index c7dbae7..f68e808 100644
--- a/libAACdec/src/usacdec_const.h
+++ b/libAACdec/src/usacdec_const.h
@@ -115,6 +115,7 @@ amm-info@iis.fraunhofer.de
/* definitions which are independent of coreCoderFrameLength */
#define M_LP_FILTER_ORDER 16 /* LP filter order */
+#define LP_FILTER_SCALE 4 /* LP filter scale */
#define PIT_MIN_12k8 34 /* Minimum pitch lag with resolution 1/4 */
#define PIT_MAX_12k8 231 /* Maximum pitch lag for fs=12.8kHz */
diff --git a/libAACdec/src/usacdec_fac.cpp b/libAACdec/src/usacdec_fac.cpp
index 71ce4a9..0d3d844 100644
--- a/libAACdec/src/usacdec_fac.cpp
+++ b/libAACdec/src/usacdec_fac.cpp
@@ -142,7 +142,7 @@ FIXP_DBL *CLpd_FAC_GetMemory(CAacDecoderChannelInfo *pAacDecoderChannelInfo,
return ptr;
}
-int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, UCHAR *pFacScale,
+int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, SCHAR *pFacScale,
int length, int use_gain, int frame) {
FIXP_DBL fac_gain;
int fac_gain_e = 0;
@@ -191,13 +191,11 @@ static void Syn_filt_zero(const FIXP_LPC a[], const INT a_exp, INT length,
L_tmp = (FIXP_DBL)0;
for (j = 0; j < fMin(i, M_LP_FILTER_ORDER); j++) {
- L_tmp -= fMultDiv2(a[j], x[i - (j + 1)]);
+ L_tmp -= fMultDiv2(a[j], x[i - (j + 1)]) >> (LP_FILTER_SCALE - 1);
}
- L_tmp = scaleValue(L_tmp, a_exp + 1);
-
- x[i] = scaleValueSaturate((x[i] >> 1) + (L_tmp >> 1),
- 1); /* Avoid overflow issues and saturate. */
+ L_tmp = scaleValue(L_tmp, a_exp + LP_FILTER_SCALE);
+ x[i] = fAddSaturate(x[i], L_tmp);
}
}
@@ -536,10 +534,12 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
/* Optional scaling of time domain - no yet windowed - of current spectrum */
if (total_gain != (FIXP_DBL)0) {
- scaleValuesWithFactor(pSpec, total_gain, tl, spec_scale[0] + scale);
- } else {
- scaleValues(pSpec, tl, spec_scale[0] + scale);
+ for (i = 0; i < tl; i++) {
+ pSpec[i] = fMult(pSpec[i], total_gain);
+ }
}
+ int loc_scale = fixmin_I(spec_scale[0] + scale, (INT)DFRACT_BITS - 1);
+ scaleValuesSaturate(pSpec, tl, loc_scale);
pOut1 += fl / 2 - 1;
pCurr = pSpec + tl - fl / 2;
@@ -625,10 +625,12 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
*/
/* and de-scale current spectrum signal (time domain, no yet windowed) */
if (total_gain != (FIXP_DBL)0) {
- scaleValuesWithFactor(pSpec, total_gain, tl, spec_scale[w] + scale);
- } else {
- scaleValues(pSpec, tl, spec_scale[w] + scale);
+ for (i = 0; i < tl; i++) {
+ pSpec[i] = fMult(pSpec[i], total_gain);
+ }
}
+ loc_scale = fixmin_I(spec_scale[w] + scale, (INT)DFRACT_BITS - 1);
+ scaleValuesSaturate(pSpec, tl, loc_scale);
if (noOutSamples <= nrSamples) {
/* Divert output first half to overlap buffer if we already got enough
diff --git a/libAACdec/src/usacdec_fac.h b/libAACdec/src/usacdec_fac.h
index bf13552..100a6fa 100644
--- a/libAACdec/src/usacdec_fac.h
+++ b/libAACdec/src/usacdec_fac.h
@@ -131,7 +131,7 @@ FIXP_DBL *CLpd_FAC_GetMemory(CAacDecoderChannelInfo *pAacDecoderChannelInfo,
* Always 0 for FD case.
* \return 0 on success, -1 on error.
*/
-int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, UCHAR *pFacScale,
+int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, SCHAR *pFacScale,
int length, int use_gain, int frame);
/**
diff --git a/libAACdec/src/usacdec_lpd.cpp b/libAACdec/src/usacdec_lpd.cpp
index 22069a6..e0a2631 100644
--- a/libAACdec/src/usacdec_lpd.cpp
+++ b/libAACdec/src/usacdec_lpd.cpp
@@ -418,6 +418,7 @@ void CLpd_AdaptLowFreqDeemph(FIXP_DBL x[], int lg, FIXP_DBL alfd_gains[],
FIXP_DBL tmp_pow2[32];
s = s * 2 + ALFDPOW2_SCALE;
+ s = fMin(31, s);
k = 8;
i_max = lg / 4; /* ALFD range = 1600Hz (lg = 6400Hz) */
@@ -1221,8 +1222,7 @@ AAC_DECODER_ERROR CLpdChannelStream_Read(
(INT)(samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM -
(INT)PIT_MIN_12k8;
- if (pSamplingRateInfo->samplingRate >
- FAC_FSCALE_MAX /* maximum allowed core sampling frequency */) {
+ if ((samplingRate < 6000) || (samplingRate > 24000)) {
error = AAC_DEC_PARSE_ERROR;
goto bail;
}
generated by cgit v1.2.3 (git 2.39.1) at 2025-01-11 03:43:21 +0000