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SBR/AAC encoder updates, code clean up

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* SBR-Encoder

   - Prevent noise level overflow in noise floor detection.
   - Saturate threshold calculation in transient detection.
     Modified file(s):
        libSBRenc/src/nf_est.cpp
        libSBRenc/src/sbr_encoder.cpp
        libSBRenc/src/tran_det.cpp

* AAC-Encoder

   - Expand input data range of GetInvInt() function. There was an encoder
     assert observed in non-default bitrate configuration.
     Modified file(s):
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/intensity.cpp
        libFDK/include/fixpoint_math.h
        libFDK/src/FDK_core.cpp
        libFDK/src/FDK_tools_rom.cpp

   - Make sure that the encoder is stable with regard to very low audio bandwidth
     confguration parameter value.
   - Fix lowdelay blending for low audio bandwidth.
     Modified file(s):
        libAACenc/src/aacenc.cpp
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/adj_thr.cpp
        libAACenc/src/psy_configuration.cpp
        libAACenc/src/psy_main.cpp

   - Disable pseudo surround flag in case metadata matrix mixdown index is
     present in program config element.
     Modified file(s):
        libAACenc/src/aacenc_lib.cpp

   - Enable variable bitrate mode in encoder api.
   - Add AACENC_PEAK_BITRATE parameter to encoder api.
   - Add AACENC_AUDIOMUXVER parameter to encoder api.
     Modified file(s):
        libAACenc/include/aacenc_lib.h
        libAACenc/src/aacenc.cpp
        libAACenc/src/aacenc.h
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/qc_main.cpp
        libMpegTPEnc/src/tpenc_latm.cpp
        libMpegTPEnc/src/version

* FDK-Sources

   - Code clean up. Remove unneeded pseudo audio object types and transport types.
     Modified file(s):
        libAACdec/src/aacdecoder.cpp
        libAACdec/src/aacdecoder_lib.cpp
        libAACenc/include/aacenc_lib.h
        libAACenc/src/aacenc.cpp
        libAACenc/src/aacenc_lib.cpp
        libFDK/src/FDK_tools_rom.cpp
        libMpegTPDec/src/tpdec_lib.cpp
        libMpegTPDec/src/version
        libMpegTPEnc/src/tpenc_latm.cpp
        libMpegTPEnc/src/version
        libSBRdec/src/sbrdecoder.cpp
        libSBRenc/src/sbr_encoder.cpp
        libSYS/include/FDK_audio.h
        libSYS/src/genericStds.cpp

Change-Id: I807a53cb7f48c9ee7563cb8da1d0c52221576ca6
This commit is contained in:
Jean-Michel Trivi 2016-04-04 16:06:48 -07:00
parent ef30836651
commit e1c78ed73f
25 changed files with 227 additions and 215 deletions
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documentation/aacEncoder.pdf
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17
libAACdec/src/aacdecoder.cpp
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@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -159,21 +159,6 @@ amm-info@iis.fraunhofer.de
#define CAN_DO_PS(aot) \
((aot) == AOT_AAC_LC \
|| (aot) == AOT_SBR \
|| (aot) == AOT_PS \
|| (aot) == AOT_ER_BSAC \
|| (aot) == AOT_DRM_AAC)
#define IS_USAC(aot) \
((aot) == AOT_USAC \
|| (aot) == AOT_RSVD50)
#define IS_LOWDELAY(aot) \
((aot) == AOT_ER_AAC_LD \
|| (aot) == AOT_ER_AAC_ELD)
void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self) void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self)
{ {

4
libAACdec/src/aacdecoder_lib.cpp
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@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -110,7 +110,7 @@ amm-info@iis.fraunhofer.de
/* Decoder library info */ /* Decoder library info */
#define AACDECODER_LIB_VL0 2 #define AACDECODER_LIB_VL0 2
#define AACDECODER_LIB_VL1 5 #define AACDECODER_LIB_VL1 5
#define AACDECODER_LIB_VL2 10 #define AACDECODER_LIB_VL2 11
#define AACDECODER_LIB_TITLE "AAC Decoder Lib" #define AACDECODER_LIB_TITLE "AAC Decoder Lib"
#ifdef __ANDROID__ #ifdef __ANDROID__
#define AACDECODER_LIB_BUILD_DATE "" #define AACDECODER_LIB_BUILD_DATE ""

23
libAACenc/include/aacenc_lib.h
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@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -897,11 +897,7 @@ typedef enum
This configuration can be used only with stereo input audio data. This configuration can be used only with stereo input audio data.
- 23: MPEG-4 AAC Low-Delay. - 23: MPEG-4 AAC Low-Delay.
- 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in
combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter. combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter. */
- 129: MPEG-2 AAC Low Complexity.
- 132: MPEG-2 AAC Low Complexity with Spectral Band Replication (HE-AAC).
- 156: MPEG-2 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2).
This configuration can be used only with stereo input audio data. */
AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE. AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE.
- CBR: Bitrate in bits/second. - CBR: Bitrate in bits/second.
@ -958,6 +954,16 @@ typedef enum
- 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not - 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not
touch this value to avoid degraded audio quality) */ touch this value to avoid degraded audio quality) */
AACENC_PEAK_BITRATE = 0x0207, /*!< Peak bitrate configuration parameter to adjust maximum bits per audio frame. Bitrate is in bits/second.
The peak bitrate will internally be limited to the chosen bitrate ::AACENC_BITRATE as lower limit
and the number_of_effective_channels*6144 bit as upper limit.
Setting the peak bitrate equal to ::AACENC_BITRATE does not necessarily mean that the audio frames
will be of constant size. Since the peak bitate is in bits/second, the frame sizes can vary by
one byte in one or the other direction over various frames. However, it is not recommended to reduce
the peak pitrate to ::AACENC_BITRATE - it would disable the bitreservoir, which would affect the
audio quality by a large amount. */
AACENC_TRANSMUX = 0x0300, /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following AACENC_TRANSMUX = 0x0300, /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following
types can be configured in encoder library: types can be configured in encoder library:
- 0: raw access units - 0: raw access units
@ -1023,6 +1029,11 @@ typedef enum
- ADTS: Maximum number of sub frames restricted to 4. - ADTS: Maximum number of sub frames restricted to 4.
- LOAS/LATM: Maximum number of sub frames restricted to 2.*/ - LOAS/LATM: Maximum number of sub frames restricted to 2.*/
AACENC_AUDIOMUXVER = 0x0304, /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, currently not implemented):
- 0: Default, no transmission of tara Buffer fullness, no ASC length and including actual latm Buffer fullnes.
- 1: Transmission of tara Buffer fullness, ASC length and actual latm Buffer fullness.
- 2: Transmission of tara Buffer fullness, ASC length and maximum level of latm Buffer fullness. */
AACENC_PROTECTION = 0x0306, /*!< Configure protection in tranpsort layer: AACENC_PROTECTION = 0x0306, /*!< Configure protection in tranpsort layer:
- 0: No protection. (default) - 0: No protection. (default)
- 1: CRC active for ADTS bitstream format. */ - 1: CRC active for ADTS bitstream format. */

83
libAACenc/src/aacenc.cpp
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@ -107,6 +107,39 @@ amm-info@iis.fraunhofer.de
#define MIN_BUFSIZE_PER_EFF_CHAN 6144 #define MIN_BUFSIZE_PER_EFF_CHAN 6144
INT FDKaacEnc_CalcBitsPerFrame(
const INT bitRate,
const INT frameLength,
const INT samplingRate
)
{
int shift = 0;
while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength
&& (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate)
{
shift++;
}
return (bitRate*(frameLength>>shift)) / (samplingRate>>shift);
}
INT FDKaacEnc_CalcBitrate(
const INT bitsPerFrame,
const INT frameLength,
const INT samplingRate
)
{
int shift = 0;
while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength
&& (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate)
{
shift++;
}
return (bitsPerFrame * (samplingRate>>shift)) / ( frameLength>>shift) ;
}
static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary(INT bitRate, static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary(INT bitRate,
INT framelength, INT framelength,
INT ancillaryRate, INT ancillaryRate,
@ -220,21 +253,19 @@ INT FDKaacEnc_GetVBRBitrate(INT bitrateMode, CHANNEL_MODE channelMode)
/** /**
* \brief Convert encoder bitreservoir value for transport library. * \brief Convert encoder bitreservoir value for transport library.
* *
* \param bitrateMode Bitratemode used in current encoder instance. Se ::AACENC_BITRATE_MODE * \param hAacEnc Encoder handle
* \param bitresTotal Encoder bitreservoir level in bits.
* *
* \return Corrected bitreservoir level used in transport library. * \return Corrected bitreservoir level used in transport library.
*/ */
static INT FDKaacEnc_EncBitresToTpBitres( static INT FDKaacEnc_EncBitresToTpBitres(
const AACENC_BITRATE_MODE bitrateMode, const HANDLE_AAC_ENC hAacEnc
const INT bitresTotal
) )
{ {
INT transporBitreservoir = 0; INT transporBitreservoir = 0;
switch (bitrateMode) { switch (hAacEnc->bitrateMode) {
case AACENC_BR_MODE_CBR: case AACENC_BR_MODE_CBR:
transporBitreservoir = bitresTotal; /* encoder bitreservoir level */ transporBitreservoir = hAacEnc->qcKernel->bitResTot; /* encoder bitreservoir level */
break; break;
case AACENC_BR_MODE_VBR_1: case AACENC_BR_MODE_VBR_1:
case AACENC_BR_MODE_VBR_2: case AACENC_BR_MODE_VBR_2:
@ -253,6 +284,10 @@ static INT FDKaacEnc_EncBitresToTpBitres(
FDK_ASSERT(0); FDK_ASSERT(0);
} }
if (hAacEnc->config->audioMuxVersion==2) {
transporBitreservoir = MIN_BUFSIZE_PER_EFF_CHAN * hAacEnc->channelMapping.nChannelsEff;
}
return transporBitreservoir; return transporBitreservoir;
} }
@ -289,6 +324,7 @@ void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config)
config->minBitsPerFrame = -1; /* minum number of bits in each AU */ config->minBitsPerFrame = -1; /* minum number of bits in each AU */
config->maxBitsPerFrame = -1; /* minum number of bits in each AU */ config->maxBitsPerFrame = -1; /* minum number of bits in each AU */
config->bitreservoir = -1; /* default, uninitialized value */ config->bitreservoir = -1; /* default, uninitialized value */
config->audioMuxVersion = -1; /* audio mux version not configured */
/* init tabs in fixpoint_math */ /* init tabs in fixpoint_math */
InitLdInt(); InitLdInt();
@ -564,7 +600,10 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
qcInit.averageBits = (averageBitsPerFrame+7)&~7; qcInit.averageBits = (averageBitsPerFrame+7)&~7;
qcInit.bitRes = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff; qcInit.bitRes = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff;
qcInit.maxBits = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff; qcInit.maxBits = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff;
qcInit.minBits = 0; qcInit.maxBits = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits;
qcInit.maxBits = fixMax(qcInit.maxBits, (averageBitsPerFrame+7)&~7);
qcInit.minBits = (config->minBitsPerFrame!=-1) ? config->minBitsPerFrame : 0;
qcInit.minBits = fixMin(qcInit.minBits, averageBitsPerFrame&~7);
} }
else else
{ {
@ -575,9 +614,11 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes); qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes);
qcInit.maxBits = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits; qcInit.maxBits = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits;
qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, fixMax(qcInit.maxBits, (averageBitsPerFrame+7+8)&~7));
qcInit.minBits = fixMax(0, ((averageBitsPerFrame-1)&~7)-qcInit.bitRes-transportEnc_GetStaticBits(hTpEnc, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes)); qcInit.minBits = fixMax(0, ((averageBitsPerFrame-1)&~7)-qcInit.bitRes-transportEnc_GetStaticBits(hTpEnc, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes));
qcInit.minBits = (config->minBitsPerFrame!=-1) ? fixMax(qcInit.minBits, config->minBitsPerFrame) : qcInit.minBits; qcInit.minBits = (config->minBitsPerFrame!=-1) ? fixMax(qcInit.minBits, config->minBitsPerFrame) : qcInit.minBits;
qcInit.minBits = fixMin(qcInit.minBits, (averageBitsPerFrame - transportEnc_GetStaticBits(hTpEnc, qcInit.maxBits))&~7);
} }
qcInit.sampleRate = config->sampleRate; qcInit.sampleRate = config->sampleRate;
@ -585,11 +626,9 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
qcInit.nSubFrames = config->nSubFrames; qcInit.nSubFrames = config->nSubFrames;
qcInit.padding.paddingRest = config->sampleRate; qcInit.padding.paddingRest = config->sampleRate;
/* Calc meanPe */ /* Calc meanPe: qcInit.meanPe = 10.0f * FRAME_LEN_LONG * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */
bw_ratio = fDivNorm((FIXP_DBL)hAacEnc->bandwidth90dB, (FIXP_DBL)(config->sampleRate>>1), &qbw); bw_ratio = fDivNorm((FIXP_DBL)(10*config->framelength*hAacEnc->bandwidth90dB), (FIXP_DBL)(config->sampleRate), &qbw);
qbw = DFRACT_BITS-1-qbw; qcInit.meanPe = FDKmax((INT)scaleValue(bw_ratio, qbw+1-(DFRACT_BITS-1)), 1);
/* qcInit.meanPe = 10.0f * FRAME_LEN_LONG * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */
qcInit.meanPe = fMult(bw_ratio, (FIXP_DBL)((10*config->framelength)<<16)) >> (qbw-15);
/* Calc maxBitFac */ /* Calc maxBitFac */
mbfac = fDivNorm((MIN_BUFSIZE_PER_EFF_CHAN-744)*cm->nChannelsEff, qcInit.averageBits/qcInit.nSubFrames, &qmbfac); mbfac = fDivNorm((MIN_BUFSIZE_PER_EFF_CHAN-744)*cm->nChannelsEff, qcInit.averageBits/qcInit.nSubFrames, &qmbfac);
@ -651,23 +690,7 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
if (ErrorStatus != AAC_ENC_OK) if (ErrorStatus != AAC_ENC_OK)
goto bail; goto bail;
/* Map virtual aot's to intern aot used in bitstream writer. */ hAacEnc->aot = hAacEnc->config->audioObjectType;
switch (hAacEnc->config->audioObjectType) {
case AOT_MP2_AAC_LC:
case AOT_DABPLUS_AAC_LC:
hAacEnc->aot = AOT_AAC_LC;
break;
case AOT_MP2_SBR:
case AOT_DABPLUS_SBR:
hAacEnc->aot = AOT_SBR;
break;
case AOT_MP2_PS:
case AOT_DABPLUS_PS:
hAacEnc->aot = AOT_PS;
break;
default:
hAacEnc->aot = hAacEnc->config->audioObjectType;
}
/* common things */ /* common things */
@ -932,7 +955,7 @@ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc,
transportEnc_WriteAccessUnit( transportEnc_WriteAccessUnit(
hTpEnc, hTpEnc,
totalBits, totalBits,
FDKaacEnc_EncBitresToTpBitres(hAacEnc->bitrateMode, hAacEnc->qcKernel->bitResTot), FDKaacEnc_EncBitresToTpBitres(hAacEnc),
cm->nChannelsEff); cm->nChannelsEff);
/* write bitstream */ /* write bitstream */

34
libAACenc/src/aacenc.h
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@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -204,6 +204,8 @@ struct AACENC_CONFIG {
INT maxBitsPerFrame; /* maximum number of bits in AU */ INT maxBitsPerFrame; /* maximum number of bits in AU */
INT bitreservoir; /* size of bitreservoir */ INT bitreservoir; /* size of bitreservoir */
INT audioMuxVersion; /* audio mux version in loas/latm transport format */
UINT sbrRatio; /* sbr sampling rate ratio: dual- or single-rate */ UINT sbrRatio; /* sbr sampling rate ratio: dual- or single-rate */
UCHAR useTns; /* flag: use temporal noise shaping */ UCHAR useTns; /* flag: use temporal noise shaping */
@ -222,6 +224,36 @@ typedef struct {
typedef struct AAC_ENC *HANDLE_AAC_ENC; typedef struct AAC_ENC *HANDLE_AAC_ENC;
/**
* \brief Calculate framesize in bits for given bit rate, frame length and sampling rate.
*
* \param bitRate Ttarget bitrate in bits per second.
* \param frameLength Number of audio samples in one frame.
* \param samplingRate Sampling rate in Hz.
*
* \return Framesize in bits per frame.
*/
INT FDKaacEnc_CalcBitsPerFrame(
const INT bitRate,
const INT frameLength,
const INT samplingRate
);
/**
* \brief Calculate bitrate in bits per second for given framesize, frame length and sampling rate.
*
* \param bitsPerFrame Framesize in bits per frame.
* \param frameLength Number of audio samples in one frame.
* \param samplingRate Sampling rate in Hz.
*
* \return Bitrate in bits per second.
*/
INT FDKaacEnc_CalcBitrate(
const INT bitsPerFrame,
const INT frameLength,
const INT samplingRate
);
/** /**
* \brief Limit given bit rate to a valid value * \brief Limit given bit rate to a valid value
* \param hTpEnc transport encoder handle * \param hTpEnc transport encoder handle

101
libAACenc/src/aacenc_lib.cpp
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@ -98,7 +98,7 @@ amm-info@iis.fraunhofer.de
/* Encoder library info */ /* Encoder library info */
#define AACENCODER_LIB_VL0 3 #define AACENCODER_LIB_VL0 3
#define AACENCODER_LIB_VL1 4 #define AACENCODER_LIB_VL1 4
#define AACENCODER_LIB_VL2 14 #define AACENCODER_LIB_VL2 19
#define AACENCODER_LIB_TITLE "AAC Encoder" #define AACENCODER_LIB_TITLE "AAC Encoder"
#ifdef __ANDROID__ #ifdef __ANDROID__
#define AACENCODER_LIB_BUILD_DATE "" #define AACENCODER_LIB_BUILD_DATE ""
@ -153,6 +153,7 @@ typedef struct {
UINT userAfterburner; UINT userAfterburner;
UINT userFramelength; UINT userFramelength;
UINT userAncDataRate; UINT userAncDataRate;
UINT userPeakBitrate;
UCHAR userTns; /*!< Use TNS coding. */ UCHAR userTns; /*!< Use TNS coding. */
UCHAR userPns; /*!< Use PNS coding. */ UCHAR userPns; /*!< Use PNS coding. */
@ -326,10 +327,7 @@ static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig)
{ {
INT sbrUsed = 0; INT sbrUsed = 0;
if ( (hAacConfig->audioObjectType==AOT_SBR) || (hAacConfig->audioObjectType==AOT_PS) if ( (hAacConfig->audioObjectType==AOT_SBR) || (hAacConfig->audioObjectType==AOT_PS) )
|| (hAacConfig->audioObjectType==AOT_MP2_SBR) || (hAacConfig->audioObjectType==AOT_MP2_PS)
|| (hAacConfig->audioObjectType==AOT_DABPLUS_SBR) || (hAacConfig->audioObjectType==AOT_DABPLUS_PS)
|| (hAacConfig->audioObjectType==AOT_DRM_SBR) || (hAacConfig->audioObjectType==AOT_DRM_MPEG_PS) )
{ {
sbrUsed = 1; sbrUsed = 1;
} }
@ -345,10 +343,7 @@ static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType)
{ {
INT psUsed = 0; INT psUsed = 0;
if ( (audioObjectType==AOT_PS) if ( (audioObjectType==AOT_PS) )
|| (audioObjectType==AOT_MP2_PS)
|| (audioObjectType==AOT_DABPLUS_PS)
|| (audioObjectType==AOT_DRM_MPEG_PS) )
{ {
psUsed = 1; psUsed = 1;
} }
@ -373,8 +368,7 @@ static SBR_PS_SIGNALING getSbrSignalingMode(
sbrSignaling = SIG_IMPLICIT; /* default: implicit signaling */ sbrSignaling = SIG_IMPLICIT; /* default: implicit signaling */
} }
if ((audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) || if ( (audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) ) {
(audioObjectType==AOT_MP2_AAC_LC) || (audioObjectType==AOT_MP2_SBR) || (audioObjectType==AOT_MP2_PS) ) {
switch (transportType) { switch (transportType) {
case TT_MP4_ADIF: case TT_MP4_ADIF:
case TT_MP4_ADTS: case TT_MP4_ADTS:
@ -430,22 +424,7 @@ static void FDKaacEnc_MapConfig(
cc->flags = 0; cc->flags = 0;
/* Map virtual aot to transport aot. */ transport_AOT = hAacConfig->audioObjectType;
switch (hAacConfig->audioObjectType) {
case AOT_MP2_AAC_LC:
transport_AOT = AOT_AAC_LC;
break;
case AOT_MP2_SBR:
transport_AOT = AOT_SBR;
cc->flags |= CC_SBR;
break;
case AOT_MP2_PS:
transport_AOT = AOT_PS;
cc->flags |= CC_SBR;
break;
default:
transport_AOT = hAacConfig->audioObjectType;
}
if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) { if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) {
cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0; cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0;
@ -511,16 +490,7 @@ static void FDKaacEnc_MapConfig(
cc->samplingRate = hAacConfig->sampleRate; cc->samplingRate = hAacConfig->sampleRate;
/* Mpeg-4 signaling for transport library. */ /* Mpeg-4 signaling for transport library. */
switch ( hAacConfig->audioObjectType ) { cc->flags |= CC_MPEG_ID;
case AOT_MP2_AAC_LC:
case AOT_MP2_SBR:
case AOT_MP2_PS:
cc->flags &= ~CC_MPEG_ID; /* Required for ADTS. */
cc->extAOT = AOT_NULL_OBJECT;
break;
default:
cc->flags |= CC_MPEG_ID;
}
/* ER-tools signaling. */ /* ER-tools signaling. */
cc->flags |= (hAacConfig->syntaxFlags & AC_ER_VCB11) ? CC_VCB11 : 0; cc->flags |= (hAacConfig->syntaxFlags & AC_ER_VCB11) ? CC_VCB11 : 0;
@ -585,6 +555,7 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig,
config->userChannelMode = hAacConfig->channelMode; config->userChannelMode = hAacConfig->channelMode;
config->userBitrate = hAacConfig->bitRate; config->userBitrate = hAacConfig->bitRate;
config->userBitrateMode = hAacConfig->bitrateMode; config->userBitrateMode = hAacConfig->bitrateMode;
config->userPeakBitrate = (UINT)-1;
config->userBandwidth = hAacConfig->bandWidth; config->userBandwidth = hAacConfig->bandWidth;
config->userTns = hAacConfig->useTns; config->userTns = hAacConfig->useTns;
config->userPns = hAacConfig->usePns; config->userPns = hAacConfig->usePns;
@ -792,12 +763,15 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
hAacConfig->syntaxFlags = 0; hAacConfig->syntaxFlags = 0;
hAacConfig->epConfig = -1; hAacConfig->epConfig = -1;
if (config->userTpType==TT_MP4_LATM_MCP1 || config->userTpType==TT_MP4_LATM_MCP0 || config->userTpType==TT_MP4_LOAS) {
hAacConfig->audioMuxVersion = config->userTpAmxv;
}
else {
hAacConfig->audioMuxVersion = -1;
}
/* Adapt internal AOT when necessary. */ /* Adapt internal AOT when necessary. */
switch ( hAacConfig->audioObjectType ) { switch ( hAacConfig->audioObjectType ) {
case AOT_MP2_AAC_LC:
case AOT_MP2_SBR:
case AOT_MP2_PS:
hAacConfig->usePns = 0;
case AOT_AAC_LC: case AOT_AAC_LC:
case AOT_SBR: case AOT_SBR:
case AOT_PS: case AOT_PS:
@ -884,6 +858,18 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
} }
} }
if ((hAacConfig->bitrateMode >= 0) && (hAacConfig->bitrateMode <= 5)) {
if ((INT)config->userPeakBitrate != -1) {
hAacConfig->maxBitsPerFrame = (FDKaacEnc_CalcBitsPerFrame(fMax(hAacConfig->bitRate, (INT)config->userPeakBitrate), hAacConfig->framelength, hAacConfig->sampleRate) + 7)&~7;
}
else {
hAacConfig->maxBitsPerFrame = -1;
}
if (hAacConfig->audioMuxVersion==2) {
hAacConfig->minBitsPerFrame = fMin(32*8, FDKaacEnc_CalcBitsPerFrame(hAacConfig->bitRate, hAacConfig->framelength, hAacConfig->sampleRate))&~7;
}
}
/* Initialize SBR parameters */ /* Initialize SBR parameters */
if ( (hAacConfig->audioObjectType==AOT_ER_AAC_ELD) if ( (hAacConfig->audioObjectType==AOT_ER_AAC_ELD)
&& (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio==0) ) && (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio==0) )
@ -1139,7 +1125,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder,
hAacConfig); hAacConfig);
/* create flags for transport encoder */ /* create flags for transport encoder */
if (config->userTpAmxv == 1) { if (config->userTpAmxv != 0) {
flags |= TP_FLAG_LATM_AMV; flags |= TP_FLAG_LATM_AMV;
} }
/* Clear output buffer */ /* Clear output buffer */
@ -1569,7 +1555,7 @@ AACENC_ERROR aacEncEncode(
&& ((hAacEncoder->extParam.userChannelMode==MODE_1_2_2)||(hAacEncoder->extParam.userChannelMode==MODE_1_2_2_1)) ) && ((hAacEncoder->extParam.userChannelMode==MODE_1_2_2)||(hAacEncoder->extParam.userChannelMode==MODE_1_2_2_1)) )
{ {
/* Set matrix mixdown coefficient. */ /* Set matrix mixdown coefficient. */
UINT pceValue = (UINT)( (1<<3) | ((matrix_mixdown_idx&0x3)<<1) | 1 ); UINT pceValue = (UINT)( (0<<3) | ((matrix_mixdown_idx&0x3)<<1) | 1 );
if (hAacEncoder->extParam.userPceAdditions != pceValue) { if (hAacEncoder->extParam.userPceAdditions != pceValue) {
hAacEncoder->extParam.userPceAdditions = pceValue; hAacEncoder->extParam.userPceAdditions = pceValue;
hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
@ -1785,19 +1771,16 @@ AACENC_ERROR aacEncoder_SetParam(
/* check if AOT matches the allocated modules */ /* check if AOT matches the allocated modules */
switch ( value ) { switch ( value ) {
case AOT_PS: case AOT_PS:
case AOT_MP2_PS:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) { if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) {
err = AACENC_INVALID_CONFIG; err = AACENC_INVALID_CONFIG;
goto bail; goto bail;
} }
case AOT_SBR: case AOT_SBR:
case AOT_MP2_SBR:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) { if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) {
err = AACENC_INVALID_CONFIG; err = AACENC_INVALID_CONFIG;
goto bail; goto bail;
} }
case AOT_AAC_LC: case AOT_AAC_LC:
case AOT_MP2_AAC_LC:
case AOT_ER_AAC_LD: case AOT_ER_AAC_LD:
case AOT_ER_AAC_ELD: case AOT_ER_AAC_ELD:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) { if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) {
@ -1823,6 +1806,7 @@ AACENC_ERROR aacEncoder_SetParam(
if (settings->userBitrateMode != value) { if (settings->userBitrateMode != value) {
switch ( value ) { switch ( value ) {
case 0: case 0:
case 1: case 2: case 3: case 4: case 5:
case 8: case 8:
settings->userBitrateMode = value; settings->userBitrateMode = value;
hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
@ -1973,6 +1957,16 @@ AACENC_ERROR aacEncoder_SetParam(
hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
} }
break; break;
case AACENC_AUDIOMUXVER:
if (settings->userTpAmxv != value) {
if ( !((value==0) || (value==1) || (value==2)) ) {
err = AACENC_INVALID_CONFIG;
break;
}
settings->userTpAmxv = value;
hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
}
break;
case AACENC_TPSUBFRAMES: case AACENC_TPSUBFRAMES:
if (settings->userTpNsubFrames != value) { if (settings->userTpNsubFrames != value) {
if (! ( (value>=1) && (value<=4) ) ) { if (! ( (value>=1) && (value<=4) ) ) {
@ -2006,6 +2000,12 @@ AACENC_ERROR aacEncoder_SetParam(
hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG;
} }
break; break;
case AACENC_PEAK_BITRATE:
if (settings->userPeakBitrate != value) {
settings->userPeakBitrate = value;
hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
}
break;
default: default:
err = AACENC_UNSUPPORTED_PARAMETER; err = AACENC_UNSUPPORTED_PARAMETER;
break; break;
@ -2076,6 +2076,9 @@ UINT aacEncoder_GetParam(
case AACENC_HEADER_PERIOD: case AACENC_HEADER_PERIOD:
value = (UINT)hAacEncoder->coderConfig.headerPeriod; value = (UINT)hAacEncoder->coderConfig.headerPeriod;
break; break;
case AACENC_AUDIOMUXVER:
value = (UINT)hAacEncoder->aacConfig.audioMuxVersion;
break;
case AACENC_TPSUBFRAMES: case AACENC_TPSUBFRAMES:
value = (UINT)settings->userTpNsubFrames; value = (UINT)settings->userTpNsubFrames;
break; break;
@ -2088,6 +2091,12 @@ UINT aacEncoder_GetParam(
case AACENC_METADATA_MODE: case AACENC_METADATA_MODE:
value = (hAacEncoder->metaDataAllowed==0) ? 0 : (UINT)settings->userMetaDataMode; value = (hAacEncoder->metaDataAllowed==0) ? 0 : (UINT)settings->userMetaDataMode;
break; break;
case AACENC_PEAK_BITRATE:
value = (UINT)-1; /* peak bitrate parameter is meaningless */
if ( ((INT)hAacEncoder->extParam.userPeakBitrate!=-1) ) {
value = (UINT)(fMax((INT)hAacEncoder->extParam.userPeakBitrate, hAacEncoder->aacConfig.bitRate)); /* peak bitrate parameter is in use */
}
break;
default: default:
//err = MPS_INVALID_PARAMETER; //err = MPS_INVALID_PARAMETER;
break; break;

2
libAACenc/src/adj_thr.cpp
View File

@ -2138,7 +2138,7 @@ static FIXP_DBL FDKaacEnc_bitresCalcBitFac(const INT bitresBits,
bresParam->clipSpendLow, bresParam->clipSpendHigh, bresParam->clipSpendLow, bresParam->clipSpendHigh,
bresParam->minBitSpend, bresParam->maxBitSpend, bitspend_slope); bresParam->minBitSpend, bresParam->maxBitSpend, bitspend_slope);
pe_pers = fDivNorm(pex - adjThrChan->peMin, adjThrChan->peMax - adjThrChan->peMin); pe_pers = (pex > adjThrChan->peMin) ? fDivNorm(pex - adjThrChan->peMin, adjThrChan->peMax - adjThrChan->peMin) : 0;
tmp_fix = fMult(((FIXP_DBL)bitSpend + (FIXP_DBL)bitSave), pe_pers); tmp_fix = fMult(((FIXP_DBL)bitSpend + (FIXP_DBL)bitSave), pe_pers);
bitresFac_fix = (UNITY>>1) - ((FIXP_DBL)bitSave>>1) + (tmp_fix>>1); qbres = (DFRACT_BITS-2); bitresFac_fix = (UNITY>>1) - ((FIXP_DBL)bitSave>>1) + (tmp_fix>>1); qbres = (DFRACT_BITS-2);

3
libAACenc/src/intensity.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -325,7 +325,6 @@ FDKaacEnc_prepareIntensityDecision(const FIXP_DBL *sfbEnergyLeft,
channelCorr[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); channelCorr[sfb + sfboffs] = FL2FXCONST_DBL(0.0f);
FDK_ASSERT(50 >= 49);
/* max width of scalefactorband is 96; width's are always even */ /* max width of scalefactorband is 96; width's are always even */
/* inv_n is scaled with factor 2 to compensate fMultDiv2() in subsequent loops */ /* inv_n is scaled with factor 2 to compensate fMultDiv2() in subsequent loops */
inv_n = GetInvInt((sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs])>>1); inv_n = GetInvInt((sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs])>>1);

5
libAACenc/src/psy_configuration.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -634,13 +634,14 @@ AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate,
if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine) if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine)
break; break;
} }
psyConf->sfbActive = sfb; psyConf->sfbActive = FDKmax(sfb, 1);
for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){ for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){
if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE) if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE)
break; break;
} }
psyConf->sfbActiveLFE = sfb; psyConf->sfbActiveLFE = sfb;
psyConf->sfbActive = FDKmax(psyConf->sfbActive, psyConf->sfbActiveLFE);
/* calculate minSnr */ /* calculate minSnr */
FDKaacEnc_initMinSnr(bitrate, FDKaacEnc_initMinSnr(bitrate,

2
libAACenc/src/psy_main.cpp
View File

@ -621,7 +621,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels,
FDKmemclear(&psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset], FDKmemclear(&psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset],
(windowLength[ch]-psyData[ch]->lowpassLine)*sizeof(FIXP_DBL)); (windowLength[ch]-psyData[ch]->lowpassLine)*sizeof(FIXP_DBL));
if (hPsyConfLong->filterbank != FB_LC) { if ( (hPsyConfLong->filterbank != FB_LC) && (psyData[ch]->lowpassLine >= FADE_OUT_LEN) ) {
/* Do blending to reduce gibbs artifacts */ /* Do blending to reduce gibbs artifacts */
for (int i=0; i<FADE_OUT_LEN; i++) { for (int i=0; i<FADE_OUT_LEN; i++) {
psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i] = fMult(psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i], fadeOutFactor[i]); psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i] = fMult(psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i], fadeOutFactor[i]);

4
libAACenc/src/qc_main.cpp
View File

@ -1248,6 +1248,8 @@ AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING* cm,
case QCDATA_BR_MODE_VBR_4: case QCDATA_BR_MODE_VBR_4:
case QCDATA_BR_MODE_VBR_5: case QCDATA_BR_MODE_VBR_5:
qcOut[0]->totFillBits = (qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits)&7; /* precalculate alignment bits */ qcOut[0]->totFillBits = (qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits)&7; /* precalculate alignment bits */
qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits;
qcOut[0]->totFillBits += ( fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7;
break; break;
case QCDATA_BR_MODE_CBR: case QCDATA_BR_MODE_CBR:
@ -1257,6 +1259,8 @@ AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING* cm,
/* processing fill-bits */ /* processing fill-bits */
INT deltaBitRes = qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits ; INT deltaBitRes = qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits ;
qcOut[0]->totFillBits = fixMax((deltaBitRes&7), (deltaBitRes - (fixMax(0,bitResSpace-7)&~7))); qcOut[0]->totFillBits = fixMax((deltaBitRes&7), (deltaBitRes - (fixMax(0,bitResSpace-7)&~7)));
qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits;
qcOut[0]->totFillBits += ( fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7;
break; break;
} /* switch (qcKernel->bitrateMode) */ } /* switch (qcKernel->bitrateMode) */

12
libFDK/include/fixpoint_math.h
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -438,11 +438,11 @@ inline FIXP_DBL fAddSaturate(const FIXP_DBL a, const FIXP_DBL b)
/***************************************************************************** /*****************************************************************************
array for 1/n, n=1..50 array for 1/n, n=1..80
****************************************************************************/ ****************************************************************************/
extern const FIXP_DBL invCount[50]; extern const FIXP_DBL invCount[80];
LNK_SECTION_INITCODE LNK_SECTION_INITCODE
inline void InitInvInt(void) {} inline void InitInvInt(void) {}
@ -450,14 +450,14 @@ inline FIXP_DBL fAddSaturate(const FIXP_DBL a, const FIXP_DBL b)
/** /**
* \brief Calculate the value of 1/i where i is a integer value. It supports * \brief Calculate the value of 1/i where i is a integer value. It supports
* input values from 1 upto 50. * input values from 1 upto 80.
* \param intValue Integer input value. * \param intValue Integer input value.
* \param FIXP_DBL representation of 1/intValue * \param FIXP_DBL representation of 1/intValue
*/ */
inline FIXP_DBL GetInvInt(int intValue) inline FIXP_DBL GetInvInt(int intValue)
{ {
FDK_ASSERT((intValue > 0) && (intValue < 50)); FDK_ASSERT((intValue > 0) && (intValue < 80));
FDK_ASSERT(intValue<50); FDK_ASSERT(intValue<80);
return invCount[intValue]; return invCount[intValue];
} }

2
libFDK/src/FDK_core.cpp
View File

@ -93,7 +93,7 @@ amm-info@iis.fraunhofer.de
/* FDK tools library info */ /* FDK tools library info */
#define FDK_TOOLS_LIB_VL0 2 #define FDK_TOOLS_LIB_VL0 2
#define FDK_TOOLS_LIB_VL1 3 #define FDK_TOOLS_LIB_VL1 3
#define FDK_TOOLS_LIB_VL2 3 #define FDK_TOOLS_LIB_VL2 4
#define FDK_TOOLS_LIB_TITLE "FDK Tools" #define FDK_TOOLS_LIB_TITLE "FDK Tools"
#ifdef __ANDROID__ #ifdef __ANDROID__
#define FDK_TOOLS_LIB_BUILD_DATE "" #define FDK_TOOLS_LIB_BUILD_DATE ""

32
libFDK/src/FDK_tools_rom.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -1902,7 +1902,7 @@ const USHORT sqrt_tab[49]={
0xb504}; 0xb504};
LNK_SECTION_CONSTDATA_L1 LNK_SECTION_CONSTDATA_L1
const FIXP_DBL invCount[50]= /* This could be 16-bit wide */ const FIXP_DBL invCount[80]= /* This could be 16-bit wide */
{ {
0x00000000, 0x7fffffff, 0x40000000, 0x2aaaaaab, 0x20000000, 0x00000000, 0x7fffffff, 0x40000000, 0x2aaaaaab, 0x20000000,
0x1999999a, 0x15555555, 0x12492492, 0x10000000, 0x0e38e38e, 0x1999999a, 0x15555555, 0x12492492, 0x10000000, 0x0e38e38e,
@ -1913,7 +1913,13 @@ const FIXP_DBL invCount[50]= /* This could be 16-bit wide */
0x04444444, 0x04210842, 0x04000000, 0x03e0f83e, 0x03c3c3c4, 0x04444444, 0x04210842, 0x04000000, 0x03e0f83e, 0x03c3c3c4,
0x03a83a84, 0x038e38e4, 0x03759f23, 0x035e50d8, 0x03483483, 0x03a83a84, 0x038e38e4, 0x03759f23, 0x035e50d8, 0x03483483,
0x03333333, 0x031f3832, 0x030c30c3, 0x02fa0be8, 0x02e8ba2f, 0x03333333, 0x031f3832, 0x030c30c3, 0x02fa0be8, 0x02e8ba2f,
0x02d82d83, 0x02c8590b, 0x02b93105, 0x02aaaaab, 0x029cbc15 0x02d82d83, 0x02c8590b, 0x02b93105, 0x02aaaaab, 0x029cbc15,
0x028f5c29, 0x02828283, 0x02762762, 0x026a439f, 0x025ed098,
0x0253c825, 0x02492492, 0x023ee090, 0x0234f72c, 0x022b63cc,
0x02222222, 0x02192e2a, 0x02108421, 0x02082082, 0x02000000,
0x01f81f82, 0x01f07c1f, 0x01e9131b, 0x01e1e1e2, 0x01dae607,
0x01d41d42, 0x01cd8569, 0x01c71c72, 0x01c0e070, 0x01bacf91,
0x01b4e81b, 0x01af286c, 0x01a98ef6, 0x01a41a42, 0x019ec8e9
}; };
@ -2033,19 +2039,6 @@ static const element_list_t node_aac_cpe = {
{ &node_aac_cpe0, &node_aac_cpe1 } { &node_aac_cpe0, &node_aac_cpe1 }
}; };
#define el_mpegsres_sce &el_aac_sce[2]
static const element_list_t node_mpegsres_sce = {
el_mpegsres_sce,
{ NULL, NULL }
};
static const element_list_t node_mpegsres_cpe = {
el_aac_cpe1,
{ NULL, NULL }
};
/* /*
* AOT C- {17,23} * AOT C- {17,23}
* epConfig = 0,1 * epConfig = 0,1
@ -2424,13 +2417,6 @@ const element_list_t * getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epCo
else else
return &node_eld_cpe_epc1; return &node_eld_cpe_epc1;
} }
case AOT_MPEGS_RESIDUALS:
if (nChannels == 1) {
return &node_mpegsres_sce;
} else {
return &node_mpegsres_cpe;
}
break;
default: default:
break; break;
} }

3
libMpegTPDec/src/tpdec_lib.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -1100,7 +1100,6 @@ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, c
} }
break; break;
case TT_RSVD50:
case TT_MP4_ADTS: case TT_MP4_ADTS:
case TT_MP4_LOAS: case TT_MP4_LOAS:
err = transportDec_readStream(hTp, layer); err = transportDec_readStream(hTp, layer);

2
libMpegTPDec/src/version
View File

@ -2,7 +2,7 @@
/* library info */ /* library info */
#define TP_LIB_VL0 2 #define TP_LIB_VL0 2
#define TP_LIB_VL1 3 #define TP_LIB_VL1 3
#define TP_LIB_VL2 4 #define TP_LIB_VL2 5
#define TP_LIB_TITLE "MPEG Transport" #define TP_LIB_TITLE "MPEG Transport"
#ifdef __ANDROID__ #ifdef __ANDROID__
#define TP_LIB_BUILD_DATE "" #define TP_LIB_BUILD_DATE ""

27
libMpegTPEnc/src/tpenc_latm.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -296,6 +296,7 @@ CreateStreamMuxConfig(
USHORT coreFrameOffset=0; USHORT coreFrameOffset=0;
hAss->taraBufferFullness = 0xFF;
hAss->audioMuxVersionA = 0; /* for future extensions */ hAss->audioMuxVersionA = 0; /* for future extensions */
hAss->streamMuxConfigBits = 0; hAss->streamMuxConfigBits = 0;
@ -339,13 +340,7 @@ CreateStreamMuxConfig(
hAss->streamMuxConfigBits+=1; hAss->streamMuxConfigBits+=1;
} }
if( (useSameConfig == 0) || (transLayer==0) ) { if( (useSameConfig == 0) || (transLayer==0) ) {
UINT bits; const UINT alignAnchor = FDKgetValidBits(hBs);
if ( hAss->audioMuxVersion == 1 ) {
FDKpushFor(hBs, 2); /* align to ASC, even if we do not know the length of the "ascLen" field yet */
}
bits = FDKgetValidBits( hBs );
transportEnc_writeASC( transportEnc_writeASC(
hBs, hBs,
@ -353,19 +348,24 @@ CreateStreamMuxConfig(
cb cb
); );
bits = FDKgetValidBits( hBs ) - bits;
if ( hAss->audioMuxVersion == 1 ) { if ( hAss->audioMuxVersion == 1 ) {
FDKpushBack(hBs, bits+2); UINT ascLen = transportEnc_LatmWriteValue(hBs, 0);
hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, bits ); FDKbyteAlign(hBs, alignAnchor);
ascLen = FDKgetValidBits(hBs) - alignAnchor - ascLen;
FDKpushBack(hBs, FDKgetValidBits(hBs) - alignAnchor);
transportEnc_LatmWriteValue(hBs, ascLen);
transportEnc_writeASC( transportEnc_writeASC(
hBs, hBs,
hAss->config[prog][layer], hAss->config[prog][layer],
cb cb
); );
FDKbyteAlign(hBs, alignAnchor); /* asc length fillbits */
} }
hAss->streamMuxConfigBits += bits; /* add asc length to smc summary */ hAss->streamMuxConfigBits += FDKgetValidBits(hBs) - alignAnchor; /* add asc length to smc summary */
} }
transLayer++; transLayer++;
@ -384,7 +384,6 @@ CreateStreamMuxConfig(
case AOT_ER_AAC_LD : case AOT_ER_AAC_LD :
case AOT_ER_AAC_ELD : case AOT_ER_AAC_ELD :
case AOT_USAC: case AOT_USAC:
case AOT_RSVD50:
p_linfo->frameLengthType = 0; p_linfo->frameLengthType = 0;
FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */ FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */

2
libMpegTPEnc/src/version
View File

@ -2,7 +2,7 @@
/* library info */ /* library info */
#define TP_LIB_VL0 2 #define TP_LIB_VL0 2
#define TP_LIB_VL1 3 #define TP_LIB_VL1 3
#define TP_LIB_VL2 4 #define TP_LIB_VL2 6
#define TP_LIB_TITLE "MPEG Transport" #define TP_LIB_TITLE "MPEG Transport"
#ifdef __ANDROID__ #ifdef __ANDROID__
#define TP_LIB_BUILD_DATE "" #define TP_LIB_BUILD_DATE ""

6
libSBRdec/src/sbrdecoder.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -137,7 +137,7 @@ amm-info@iis.fraunhofer.de
/* Decoder library info */ /* Decoder library info */
#define SBRDECODER_LIB_VL0 2 #define SBRDECODER_LIB_VL0 2
#define SBRDECODER_LIB_VL1 2 #define SBRDECODER_LIB_VL1 2
#define SBRDECODER_LIB_VL2 6 #define SBRDECODER_LIB_VL2 7
#define SBRDECODER_LIB_TITLE "SBR Decoder" #define SBRDECODER_LIB_TITLE "SBR Decoder"
#ifdef __ANDROID__ #ifdef __ANDROID__
#define SBRDECODER_LIB_BUILD_DATE "" #define SBRDECODER_LIB_BUILD_DATE ""
@ -318,7 +318,6 @@ SBR_ERROR sbrDecoder_ResetElement (
case AOT_PS: case AOT_PS:
case AOT_ER_AAC_SCAL: case AOT_ER_AAC_SCAL:
case AOT_DRM_AAC: case AOT_DRM_AAC:
case AOT_DRM_SURROUND:
if (CreatePsDec ( &self->hParametricStereoDec, samplesPerFrame )) { if (CreatePsDec ( &self->hParametricStereoDec, samplesPerFrame )) {
sbrError = SBRDEC_CREATE_ERROR; sbrError = SBRDEC_CREATE_ERROR;
goto bail; goto bail;
@ -503,7 +502,6 @@ SBR_ERROR sbrDecoder_InitElement (
case AOT_PS: case AOT_PS:
case AOT_ER_AAC_SCAL: case AOT_ER_AAC_SCAL:
case AOT_DRM_AAC: case AOT_DRM_AAC:
case AOT_DRM_SURROUND:
elChannels = 2; elChannels = 2;
break; break;
default: default:

7
libSBRenc/src/nf_est.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -268,8 +268,9 @@ qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel, /*!< Pointer to v
/* /*
* Add a noise floor offset to compensate for bias in the detector * Add a noise floor offset to compensate for bias in the detector
*****************************************************************/ *****************************************************************/
if(!missingHarmonicFlag) if(!missingHarmonicFlag) {
*noiseLevel = fMult(*noiseLevel, noiseFloorOffset)<<(NOISE_FLOOR_OFFSET_SCALING); *noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset), (FIXP_DBL)MAXVAL_DBL>>NOISE_FLOOR_OFFSET_SCALING) << NOISE_FLOOR_OFFSET_SCALING;
}
/* /*
* check to see that we don't exceed the maximum allowed level * check to see that we don't exceed the maximum allowed level

10
libSBRenc/src/sbr_encoder.cpp
View File

@ -103,7 +103,7 @@ amm-info@iis.fraunhofer.de
#define SBRENCODER_LIB_VL0 3 #define SBRENCODER_LIB_VL0 3
#define SBRENCODER_LIB_VL1 3 #define SBRENCODER_LIB_VL1 3
#define SBRENCODER_LIB_VL2 6 #define SBRENCODER_LIB_VL2 8
@ -1552,12 +1552,6 @@ INT FDKsbrEnc_EnvInit (
hSbrElement->sbrConfigData.sbrSyntaxFlags = 0; hSbrElement->sbrConfigData.sbrSyntaxFlags = 0;
switch (aot) { switch (aot) {
case AOT_DRM_MPEG_PS:
case AOT_DRM_SBR:
hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_SCALABLE;
hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_DRM_CRC;
hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC;
break;
case AOT_ER_AAC_ELD: case AOT_ER_AAC_ELD:
hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY; hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY;
break; break;
@ -1847,7 +1841,7 @@ INT sbrEncoder_Init(
if ( (aot==AOT_PS) || (aot==AOT_MP2_PS) || (aot==AOT_DABPLUS_PS) || (aot==AOT_DRM_MPEG_PS) ) { if ( (aot==AOT_PS) ) {
usePs = 1; usePs = 1;
} }
if ( (aot==AOT_ER_AAC_ELD) ) { if ( (aot==AOT_ER_AAC_ELD) ) {

13
libSBRenc/src/tran_det.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -676,7 +676,7 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
tmp = fixMax(tmp, FL2FXCONST_DBL(0.0001)); tmp = fixMax(tmp, FL2FXCONST_DBL(0.0001));
tmp = fDivNorm(FL2FXCONST_DBL(0.000075), fPow2(tmp), &scale_1); tmp = fDivNorm(FL2FXCONST_DBL(0.000075), fPow2(tmp), &scale_1);
scale_1 = -(scale_1 + scale_0 + 2); scale_1 = (scale_1 + scale_0 + 2);
FDK_ASSERT(no_cols <= QMF_MAX_TIME_SLOTS); FDK_ASSERT(no_cols <= QMF_MAX_TIME_SLOTS);
FDK_ASSERT(no_rows <= QMF_CHANNELS); FDK_ASSERT(no_rows <= QMF_CHANNELS);
@ -684,14 +684,7 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
h_sbrTransientDetector->no_cols = no_cols; h_sbrTransientDetector->no_cols = no_cols;
h_sbrTransientDetector->tran_thr = (FIXP_DBL)((params->tran_thr << (32-24-1)) / no_rows); h_sbrTransientDetector->tran_thr = (FIXP_DBL)((params->tran_thr << (32-24-1)) / no_rows);
h_sbrTransientDetector->tran_fc = tran_fc; h_sbrTransientDetector->tran_fc = tran_fc;
h_sbrTransientDetector->split_thr = scaleValueSaturate(fMult(tmp, bitrateFactor_fix), scale_1);
if (scale_1>=0) {
h_sbrTransientDetector->split_thr = fMult(tmp, bitrateFactor_fix) >> scale_1;
}
else {
h_sbrTransientDetector->split_thr = fMult(tmp, bitrateFactor_fix) << (-scale_1);
}
h_sbrTransientDetector->no_rows = no_rows; h_sbrTransientDetector->no_rows = no_rows;
h_sbrTransientDetector->mode = params->tran_det_mode; h_sbrTransientDetector->mode = params->tran_det_mode;
h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f); h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f);

44
libSYS/include/FDK_audio.h
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -134,13 +134,7 @@ typedef enum
TT_MP4_LOAS = 10, /**< Audio Sync Stream. */ TT_MP4_LOAS = 10, /**< Audio Sync Stream. */
TT_DRM = 12, /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */ TT_DRM = 12 /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
TT_MP1_L1 = 16, /**< MPEG 1 Audio Layer 1 audio bitstream. */
TT_MP1_L2 = 17, /**< MPEG 1 Audio Layer 2 audio bitstream. */
TT_MP1_L3 = 18, /**< MPEG 1 Audio Layer 3 audio bitstream. */
TT_RSVD50 = 50 /**< */
} TRANSPORT_TYPE; } TRANSPORT_TYPE;
@ -203,38 +197,22 @@ typedef enum
AOT_SAOC = 43, /**< SAOC */ AOT_SAOC = 43, /**< SAOC */
AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround */ AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround */
AOT_RSVD50 = 50, /**< Interim AOT for Rsvd50 */
/* Pseudo AOTs */ /* Pseudo AOTs */
AOT_MP2_AAC_MAIN = 128, /**< Virtual AOT MP2 Main profile */
AOT_MP2_AAC_LC = 129, /**< Virtual AOT MP2 Low Complexity profile */
AOT_MP2_AAC_SSR = 130, /**< Virtual AOT MP2 Scalable Sampling Rate profile */
AOT_MP2_SBR = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR */
AOT_DAB = 134, /**< Virtual AOT for DAB (Layer2 with scalefactor CRC) */
AOT_DABPLUS_AAC_LC = 135, /**< Virtual AOT for DAB plus AAC-LC */
AOT_DABPLUS_SBR = 136, /**< Virtual AOT for DAB plus HE-AAC */
AOT_DABPLUS_PS = 137, /**< Virtual AOT for DAB plus HE-AAC v2 */
AOT_PLAIN_MP1 = 140, /**< Virtual AOT for plain mp1 */
AOT_PLAIN_MP2 = 141, /**< Virtual AOT for plain mp2 */
AOT_PLAIN_MP3 = 142, /**< Virtual AOT for plain mp3 */
AOT_DRM_AAC = 143, /**< Virtual AOT for DRM (ER-AAC-SCAL without SBR) */ AOT_DRM_AAC = 143, /**< Virtual AOT for DRM (ER-AAC-SCAL without SBR) */
AOT_DRM_SBR = 144, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR) */ AOT_DRM_SBR = 144, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR) */
AOT_DRM_MPEG_PS = 145, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */ AOT_DRM_MPEG_PS = 145 /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */
AOT_DRM_SURROUND = 146, /**< Virtual AOT for DRM Surround (ER-AAC-SCAL (+SBR) +MPS) */
AOT_MP2_PS = 156, /**< Virtual AOT MP2 Low Complexity Profile with SBR and PS */
AOT_MPEGS_RESIDUALS = 256 /**< Virtual AOT for MPEG Surround residuals */
} AUDIO_OBJECT_TYPE; } AUDIO_OBJECT_TYPE;
#define CAN_DO_PS(aot) \
((aot) == AOT_AAC_LC \
|| (aot) == AOT_SBR \
|| (aot) == AOT_PS \
|| (aot) == AOT_ER_BSAC \
|| (aot) == AOT_DRM_AAC)
#define IS_USAC(aot) \ #define IS_USAC(aot) \
((aot) == AOT_USAC \ ((aot) == AOT_USAC)
|| (aot) == AOT_RSVD50)
#define IS_LOWDELAY(aot) \ #define IS_LOWDELAY(aot) \
((aot) == AOT_ER_AAC_LD \ ((aot) == AOT_ER_AAC_LD \

4
libSYS/src/genericStds.cpp
View File

@ -2,7 +2,7 @@
/* ----------------------------------------------------------------------------------------------------------- /* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. © Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved. All rights reserved.
1. INTRODUCTION 1. INTRODUCTION
@ -99,7 +99,7 @@ amm-info@iis.fraunhofer.de
/* library info */ /* library info */
#define SYS_LIB_VL0 1 #define SYS_LIB_VL0 1
#define SYS_LIB_VL1 3 #define SYS_LIB_VL1 3
#define SYS_LIB_VL2 6 #define SYS_LIB_VL2 7
#define SYS_LIB_TITLE "System Integration Library" #define SYS_LIB_TITLE "System Integration Library"
#ifdef __ANDROID__ #ifdef __ANDROID__
#define SYS_LIB_BUILD_DATE "" #define SYS_LIB_BUILD_DATE ""