generic-library/fdk-aac
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Merge remote-tracking branch 'aosp/master'

Browse Source
This commit is contained in:
Martin Storsjo 2016-08-25 22:12:15 +03:00
commit f9d0f65e65
75 changed files with 2590 additions and 817 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

BIN
documentation/aacDecoder.pdf
View File

Binary file not shown.

BIN
documentation/aacEncoder.pdf
View File

Binary file not shown.

3
libAACdec/include/aacdecoder_lib.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -382,6 +382,7 @@ typedef enum {
not exist. */
AAC_DEC_NEED_TO_RESTART = 0x200B, /*!< The decoder needs to be restarted, since the requiered configuration change cannot be
performed. */
AAC_DEC_OUTPUT_BUFFER_TOO_SMALL = 0x200C, /*!< The provided output buffer is too small. */
aac_dec_init_error_end = 0x2FFF,
/* Decode errors. Output buffer is valid but concealed. */

85
libAACdec/src/aac_rom.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -167,6 +167,36 @@ const SCHAR ExponentTable [4][14] =
} ;
/* 41 scfbands */
static const SHORT sfb_96_1024[42] =
{
0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44,
48, 52, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144,
156, 172, 188, 212, 240, 276, 320, 384, 448, 512, 576, 640,
704, 768, 832, 896, 960, 1024
};
/* 12 scfbands */
static const SHORT sfb_96_128[13] =
{
0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92,
128
};
/* 47 scfbands*/
static const SHORT sfb_64_1024[48] =
{
0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52,
56, 64, 72, 80, 88, 100, 112, 124, 140, 156, 172, 192, 216, 240,
268, 304, 344, 384, 424, 464, 504, 544, 584, 624, 664, 704, 744, 784,
824, 864, 904, 944, 984,1024
};
/* 12 scfbands */
static const SHORT sfb_64_128[13] =
{
0, 4, 8, 12, 16, 20, 24,
32, 40, 48, 64, 92, 128
};
/* 49 scfbands */
static const SHORT sfb_48_1024[50] = {
@ -239,6 +269,35 @@ static const SHORT sfb_8_128[16] =
};
static const SHORT sfb_96_960[42] =
{
0, 4, 8, 12, 16, 20, 24, 28, 32, 36,
40, 44, 48, 52, 56, 64, 72, 80, 88, 96,
108, 120, 132, 144, 156, 172, 188, 212, 240, 276,
320, 384, 448, 512, 576, 640, 704, 768, 832, 896,
960
}; /* 40 scfbands */
static const SHORT sfb_96_120[13] =
{
0, 4, 8, 12, 16, 20, 24, 32, 40, 48,
64, 92, 120
}; /* 12 scfbands */
static const SHORT sfb_64_960[47] =
{
0, 4, 8, 12, 16, 20, 24, 28, 32, 36,
40, 44, 48, 52, 56, 64, 72, 80, 88, 100,
112, 124, 140, 156, 172, 192, 216, 240, 268, 304,
344, 384, 424, 464, 504, 544, 584, 624, 664, 704,
744, 784, 824, 864, 904, 944, 960
}; /* 46 scfbands */
static const SHORT sfb_64_120[13] =
{
0, 4, 8, 12, 16, 20, 24, 32, 40, 48,
64, 92, 120
}; /* 12 scfbands */
static const SHORT sfb_48_960[50] =
{
@ -358,9 +417,9 @@ static const SHORT sfb_24_480[31] =
const SFB_INFO sfbOffsetTables[5][16] =
{
{
{ NULL, NULL, 0, 0 },
{ NULL, NULL, 0, 0 },
{ NULL, NULL, 0, 0 },
{ sfb_96_1024, sfb_96_128, 41, 12 },
{ sfb_96_1024, sfb_96_128, 41, 12 },
{ sfb_64_1024, sfb_64_128, 47, 12 },
{ sfb_48_1024, sfb_48_128, 49, 14 },
{ sfb_48_1024, sfb_48_128, 49, 14 },
{ sfb_32_1024, sfb_48_128, 51, 14 },
@ -372,9 +431,9 @@ const SFB_INFO sfbOffsetTables[5][16] =
{ sfb_8_1024, sfb_8_128, 40, 15 },
{ sfb_8_1024, sfb_8_128, 40, 15 },
}, {
{ NULL, NULL, 0, 0 },
{ NULL, NULL, 0, 0 },
{ NULL, NULL, 0, 0 },
{ sfb_96_960, sfb_96_120, 40, 12 },
{ sfb_96_960, sfb_96_120, 40, 12 },
{ sfb_64_960, sfb_64_120, 46, 12 },
{ sfb_48_960, sfb_48_120, 49, 14 },
{ sfb_48_960, sfb_48_120, 49, 14 },
{ sfb_32_960, sfb_48_120, 49, 14 },
@ -388,9 +447,9 @@ const SFB_INFO sfbOffsetTables[5][16] =
}, {
{ NULL, NULL, 0, 0 },
}, {
{ NULL, NULL, 0, 0 },
{ NULL, NULL, 0, 0 },
{ NULL, NULL, 0, 0 },
{ sfb_48_512, NULL, 36, 0 },
{ sfb_48_512, NULL, 36, 0 },
{ sfb_48_512, NULL, 36, 0 },
{ sfb_48_512, NULL, 36, 0 },
{ sfb_48_512, NULL, 36, 0},
{ sfb_32_512, NULL, 37, 0 },
@ -402,9 +461,9 @@ const SFB_INFO sfbOffsetTables[5][16] =
{ sfb_24_512, NULL, 31, 0 },
{ sfb_24_512, NULL, 31, 0 },
}, {
{ NULL, NULL, 0, 0 },
{ NULL, NULL, 0, 0 },
{ NULL, NULL, 0, 0 },
{ sfb_48_480, NULL, 35, 0 },
{ sfb_48_480, NULL, 35, 0 },
{ sfb_48_480, NULL, 35, 0 },
{ sfb_48_480, NULL, 35, 0 },
{ sfb_48_480, NULL, 35, 0 },
{ sfb_32_480, NULL, 37, 0 },

4
libAACdec/src/aacdec_drc.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved.
1. INTRODUCTION
@ -705,7 +705,7 @@ static int aacDecoder_drcExtractAndMap (
}
self->numPayloads = 0;
if (self->dvbAncDataAvailable)
if (self->dvbAncDataAvailable && self->numThreads < MAX_DRC_THREADS)
{ /* Append a DVB heavy compression payload thread if available. */
int bitsParsed;

54
libAACdec/src/aacdecoder.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -157,23 +157,9 @@ amm-info@iis.fraunhofer.de
#include "conceal.h"
#include "FDK_crc.h"
#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)
{
@ -552,8 +538,9 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self,
previous_element,
elIndex,
self->flags & AC_INDEP );
/* Enable SBR for implicit SBR signalling. */
if (sbrError == SBRDEC_OK) {
/* Enable SBR for implicit SBR signalling but only if no severe error happend. */
if ( (sbrError == SBRDEC_OK)
|| (sbrError == SBRDEC_PARSE_ERROR) ) {
self->sbrEnabled = 1;
}
} else {
@ -568,7 +555,7 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self,
FDKpushBiDirectional(hBs, *count);
*count = 0;
} else {
/* If this is not a fill element with a known length, we are screwed an no further parsing makes sense. */
/* If this is not a fill element with a known length, we are screwed and further parsing makes no sense. */
if (sbrError != SBRDEC_OK) {
self->frameOK = 0;
}
@ -847,12 +834,18 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
switch (asc->m_aot) {
case AOT_AAC_LC:
self->streamInfo.profile = 1;
break;
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;
}
case AOT_SBR:
case AOT_PS:
case AOT_ER_AAC_LD:
case AOT_ER_AAC_ELD:
case AOT_DRM_AAC:
break;
default:
@ -972,11 +965,20 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
if (asc->m_aot == AOT_ER_AAC_ELD) {
self->flags |= AC_ELD;
self->flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0; /* Need to set the SBR flag for backward-compatibility
reasons. Even if SBR is not supported. */
self->flags |= (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0;
self->flags |= (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) ? AC_LD_MPS : 0;
}
self->flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0;
self->flags |= (asc->m_epConfig >= 0) ? AC_ER : 0;
if ( asc->m_aot == AOT_DRM_AAC ) {
self->flags |= AC_DRM|AC_SBRCRC|AC_SCALABLE;
}
if ( (asc->m_aot == AOT_AAC_SCAL)
|| (asc->m_aot == AOT_ER_AAC_SCAL) ) {
self->flags |= AC_SCALABLE;
}
if (asc->m_sbrPresentFlag) {
@ -1162,6 +1164,9 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
/* Check sampling frequency */
switch ( self->streamInfo.aacSampleRate ) {
case 96000:
case 88200:
case 64000:
case 16000:
case 12000:
case 11025:
@ -1490,7 +1495,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
/* get the remaining bits of this frame */
bitCnt = transportDec_GetAuBitsRemaining(self->hInput, 0);
if ( (bitCnt > 0) && (self->flags & AC_SBR_PRESENT) && (self->flags & (AC_USAC|AC_RSVD50|AC_ELD)) )
if ( (bitCnt > 0) && (self->flags & AC_SBR_PRESENT) && (self->flags & (AC_USAC|AC_RSVD50|AC_ELD|AC_DRM)) )
{
SBR_ERROR err = SBRDEC_OK;
int elIdx, numChElements = el_cnt[ID_SCE] + el_cnt[ID_CPE];
@ -1528,6 +1533,13 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
}
if (self->flags & AC_DRM)
{
if ((bitCnt = (INT)FDKgetValidBits(bs)) != 0) {
FDKpushBiDirectional(bs, bitCnt);
}
}
if ( ! (self->flags & (AC_USAC|AC_RSVD50|AC_DRM)) )
{
while ( bitCnt > 7 ) {

4
libAACdec/src/aacdecoder.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -226,6 +226,8 @@ struct AAC_DECODER_INSTANCE {
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)*(2048)];
};

45
libAACdec/src/aacdecoder_lib.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -110,7 +110,7 @@ amm-info@iis.fraunhofer.de
/* Decoder library info */
#define AACDECODER_LIB_VL0 2
#define AACDECODER_LIB_VL1 5
#define AACDECODER_LIB_VL2 10
#define AACDECODER_LIB_VL2 17
#define AACDECODER_LIB_TITLE "AAC Decoder Lib"
#ifdef __ANDROID__
#define AACDECODER_LIB_BUILD_DATE ""
@ -181,8 +181,9 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_ConfigRaw (
break;
}
/* if baselayer is OK we continue decoding */
if(layer >= 1){
if(layer >= 1){
self->nrOfLayers = layer;
err = AAC_DEC_OK;
}
break;
}
@ -785,8 +786,8 @@ static INT aacDecoder_EstimateNumberOfLostFrames(HANDLE_AACDECODER self)
LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
HANDLE_AACDECODER self,
INT_PCM *pTimeData,
const INT timeDataSize,
INT_PCM *pTimeData_extern,
const INT timeDataSize_extern,
const UINT flags)
{
AAC_DECODER_ERROR ErrorStatus;
@ -796,12 +797,17 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
HANDLE_FDK_BITSTREAM hBs;
int fTpInterruption = 0; /* Transport originated interruption detection. */
int fTpConceal = 0; /* Transport originated concealment. */
INT_PCM *pTimeData = NULL;
INT timeDataSize = 0;
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;
}
@ -918,7 +924,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
if (self->sbrEnabled)
{
SBR_ERROR sbrError = SBRDEC_OK;
int chOutMapIdx = ((self->chMapIndex==0) && (self->streamInfo.numChannels<7)) ? self->streamInfo.numChannels : self->chMapIndex;
int chIdx, numCoreChannel = self->streamInfo.numChannels;
int chOutMapIdx = ((self->chMapIndex==0) && (numCoreChannel<7)) ? numCoreChannel : self->chMapIndex;
/* set params */
sbrDecoder_SetParam ( self->hSbrDecoder,
@ -978,10 +985,10 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
if (self->psPossible) {
self->flags |= AC_PS_PRESENT;
self->channelType[0] = ACT_FRONT;
self->channelType[1] = ACT_FRONT;
self->channelIndices[0] = 0;
self->channelIndices[1] = 1;
}
for (chIdx = numCoreChannel; chIdx < self->streamInfo.numChannels; chIdx+=1) {
self->channelType[chIdx] = ACT_FRONT;
self->channelIndices[chIdx] = chIdx;
}
}
}
@ -1006,7 +1013,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
self->channelOutputMapping,
(self->limiterEnableCurr) ? &pcmLimiterScale : NULL
);
if (dmxErr == PCMDMX_INVALID_MODE) {
if ( (ErrorStatus == AAC_DEC_OK)
&& (dmxErr == PCMDMX_INVALID_MODE) ) {
/* Announce the framework that the current combination of channel configuration and downmix
* settings are not know to produce a predictable behavior and thus maybe produce strange output. */
ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
@ -1051,6 +1059,19 @@ bail:
/* Update Statistics */
aacDecoder_UpdateBitStreamCounters(&self->streamInfo, hBs, nBits, ErrorStatus);
/* 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));
}
return ErrorStatus;
}
@ -1120,6 +1141,7 @@ LINKSPEC_CPP INT aacDecoder_GetLibInfo ( LIB_INFO *info )
/* Set flags */
info->flags = 0
| CAPF_AAC_LC
| CAPF_ER_AAC_SCAL
| CAPF_AAC_VCB11
| CAPF_AAC_HCR
| CAPF_AAC_RVLC
@ -1130,6 +1152,7 @@ LINKSPEC_CPP INT aacDecoder_GetLibInfo ( LIB_INFO *info )
| CAPF_AAC_MPEG4
| CAPF_AAC_DRM_BSFORMAT
| CAPF_AAC_1024
| CAPF_AAC_960

6
libAACdec/src/block.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -324,11 +324,11 @@ AAC_DECODER_ERROR CBlock_ReadSectionData(HANDLE_FDK_BITSTREAM bs,
if (flags & AC_ER_HCR) {
/* HCR input (long) -- collecting sideinfo (for HCR-_long_ only) */
pNumLinesInSec[numLinesInSecIdx] = BandOffsets[top] - BandOffsets[band];
numLinesInSecIdx++;
if (numLinesInSecIdx >= MAX_SFB_HCR) {
return AAC_DEC_PARSE_ERROR;
}
pNumLinesInSec[numLinesInSecIdx] = BandOffsets[top] - BandOffsets[band];
numLinesInSecIdx++;
if (sect_cb == BOOKSCL)
{
return AAC_DEC_INVALID_CODE_BOOK;

23
libAACenc/include/aacenc_lib.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -900,11 +900,7 @@ typedef enum
This configuration can be used only with stereo input audio data.
- 23: MPEG-4 AAC Low-Delay.
- 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.
- 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. */
combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter. */
AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE.
- CBR: Bitrate in bits/second.
@ -961,6 +957,16 @@ typedef enum
- 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not
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
types can be configured in encoder library:
- 0: raw access units
@ -1026,6 +1032,11 @@ typedef enum
- ADTS: Maximum number of sub frames restricted to 4.
- 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:
- 0: No protection. (default)
- 1: CRC active for ADTS bitstream format. */

32
libAACenc/src/aacEnc_rom.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -93,7 +93,7 @@ amm-info@iis.fraunhofer.de
/*
Huffman Tables
*/
const INT FDKaacEnc_huff_ltab1_2[3][3][3][3]=
const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3]=
{
{
{ {0x000b0009,0x00090007,0x000b0009}, {0x000a0008,0x00070006,0x000a0008}, {0x000b0009,0x00090008,0x000b0009} },
@ -113,7 +113,7 @@ const INT FDKaacEnc_huff_ltab1_2[3][3][3][3]=
};
const INT FDKaacEnc_huff_ltab3_4[3][3][3][3]=
const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3]=
{
{
{ {0x00010004,0x00040005,0x00080008}, {0x00040005,0x00050004,0x00080008}, {0x00090009,0x00090008,0x000a000b} },
@ -132,7 +132,7 @@ const INT FDKaacEnc_huff_ltab3_4[3][3][3][3]=
}
};
const INT FDKaacEnc_huff_ltab5_6[9][9]=
const ULONG FDKaacEnc_huff_ltab5_6[9][9]=
{
{0x000d000b, 0x000c000a, 0x000b0009, 0x000b0009, 0x000a0009, 0x000b0009, 0x000b0009, 0x000c000a, 0x000d000b},
{0x000c000a, 0x000b0009, 0x000a0008, 0x00090007, 0x00080007, 0x00090007, 0x000a0008, 0x000b0009, 0x000c000a},
@ -145,7 +145,7 @@ const INT FDKaacEnc_huff_ltab5_6[9][9]=
{0x000d000b, 0x000c000a, 0x000c0009, 0x000b0009, 0x000a0009, 0x000a0009, 0x000b0009, 0x000c000a, 0x000d000b}
};
const INT FDKaacEnc_huff_ltab7_8[8][8]=
const ULONG FDKaacEnc_huff_ltab7_8[8][8]=
{
{0x00010005, 0x00030004, 0x00060005, 0x00070006, 0x00080007, 0x00090008, 0x000a0009, 0x000b000a},
{0x00030004, 0x00040003, 0x00060004, 0x00070005, 0x00080006, 0x00080007, 0x00090007, 0x00090008},
@ -157,7 +157,7 @@ const INT FDKaacEnc_huff_ltab7_8[8][8]=
{0x000b000a, 0x000a0008, 0x000a0008, 0x000a0008, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c000a}
};
const INT FDKaacEnc_huff_ltab9_10[13][13]=
const ULONG FDKaacEnc_huff_ltab9_10[13][13]=
{
{0x00010006, 0x00030005, 0x00060006, 0x00080006, 0x00090007, 0x000a0008, 0x000a0009, 0x000b000a, 0x000b000a, 0x000c000a, 0x000c000b, 0x000d000b, 0x000d000c},
{0x00030005, 0x00040004, 0x00060004, 0x00070005, 0x00080006, 0x00080007, 0x00090007, 0x000a0008, 0x000a0008, 0x000a0009, 0x000b000a, 0x000c000a, 0x000c000b},
@ -392,7 +392,7 @@ const USHORT FDKaacEnc_huff_ctab11[21][17]=
{0x0046, 0x00ea, 0x0034, 0x00ea, 0x0011, 0x001b, 0x00a9, 0x0094, 0x00e2, 0x0031, 0x00d0, 0x00e5, 0x0007, 0x0070, 0x0069, 0x003e, 0x0021}
};
const INT FDKaacEnc_huff_ctabscf[121]=
const ULONG FDKaacEnc_huff_ctabscf[121]=
{
0x0003ffe8, 0x0003ffe6, 0x0003ffe7, 0x0003ffe5, 0x0007fff5, 0x0007fff1, 0x0007ffed, 0x0007fff6,
0x0007ffee, 0x0007ffef, 0x0007fff0, 0x0007fffc, 0x0007fffd, 0x0007ffff, 0x0007fffe, 0x0007fff7,
@ -657,11 +657,11 @@ const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_128 = {
*/
const FIXP_DBL FDKaacEnc_tnsEncCoeff3[8]=
{
0x81f1d201, 0x91261481, 0xadb92301, 0xd438af00, 0x00000000, 0x37898080, 0x64130dff, 0x7cca6fff
(FIXP_DBL)0x81f1d201, (FIXP_DBL)0x91261481, (FIXP_DBL)0xadb92301, (FIXP_DBL)0xd438af00, (FIXP_DBL)0x00000000, (FIXP_DBL)0x37898080, (FIXP_DBL)0x64130dff, (FIXP_DBL)0x7cca6fff
};
const FIXP_DBL FDKaacEnc_tnsCoeff3Borders[8]={
0x80000001 /*-4*/, 0x87b826df /*-3*/, 0x9df24154 /*-2*/, 0xbfffffe5 /*-1*/,
0xe9c5e578 /* 0*/, 0x1c7b90f0 /* 1*/, 0x4fce83a9 /* 2*/, 0x7352f2c3 /* 3*/
(FIXP_DBL)0x80000001 /*-4*/, (FIXP_DBL)0x87b826df /*-3*/, (FIXP_DBL)0x9df24154 /*-2*/, (FIXP_DBL)0xbfffffe5 /*-1*/,
(FIXP_DBL)0xe9c5e578 /* 0*/, (FIXP_DBL)0x1c7b90f0 /* 1*/, (FIXP_DBL)0x4fce83a9 /* 2*/, (FIXP_DBL)0x7352f2c3 /* 3*/
};
/*
@ -669,15 +669,15 @@ const FIXP_DBL FDKaacEnc_tnsCoeff3Borders[8]={
*/
const FIXP_DBL FDKaacEnc_tnsEncCoeff4[16]=
{
0x808bc881, 0x84e2e581, 0x8d6b4a01, 0x99da9201, 0xa9c45701, 0xbc9dde81, 0xd1c2d500, 0xe87ae540,
0x00000000, 0x1a9cd9c0, 0x340ff240, 0x4b3c8bff, 0x5f1f5e7f, 0x6ed9eb7f, 0x79bc387f, 0x7f4c7e7f
(FIXP_DBL)0x808bc881, (FIXP_DBL)0x84e2e581, (FIXP_DBL)0x8d6b4a01, (FIXP_DBL)0x99da9201, (FIXP_DBL)0xa9c45701, (FIXP_DBL)0xbc9dde81, (FIXP_DBL)0xd1c2d500, (FIXP_DBL)0xe87ae540,
(FIXP_DBL)0x00000000, (FIXP_DBL)0x1a9cd9c0, (FIXP_DBL)0x340ff240, (FIXP_DBL)0x4b3c8bff, (FIXP_DBL)0x5f1f5e7f, (FIXP_DBL)0x6ed9eb7f, (FIXP_DBL)0x79bc387f, (FIXP_DBL)0x7f4c7e7f
};
const FIXP_DBL FDKaacEnc_tnsCoeff4Borders[16]=
{
0x80000001 /*-8*/, 0x822deff0 /*-7*/, 0x88a4bfe6 /*-6*/, 0x932c159d /*-5*/,
0xa16827c2 /*-4*/, 0xb2dcde27 /*-3*/, 0xc6f20b91 /*-2*/, 0xdcf89c64 /*-1*/,
0xf4308ce1 /* 0*/, 0x0d613054 /* 1*/, 0x278dde80 /* 2*/, 0x4000001b /* 3*/,
0x55a6127b /* 4*/, 0x678dde8f /* 5*/, 0x74ef0ed7 /* 6*/, 0x7d33f0da /* 7*/
(FIXP_DBL)0x80000001 /*-8*/, (FIXP_DBL)0x822deff0 /*-7*/, (FIXP_DBL)0x88a4bfe6 /*-6*/, (FIXP_DBL)0x932c159d /*-5*/,
(FIXP_DBL)0xa16827c2 /*-4*/, (FIXP_DBL)0xb2dcde27 /*-3*/, (FIXP_DBL)0xc6f20b91 /*-2*/, (FIXP_DBL)0xdcf89c64 /*-1*/,
(FIXP_DBL)0xf4308ce1 /* 0*/, (FIXP_DBL)0x0d613054 /* 1*/, (FIXP_DBL)0x278dde80 /* 2*/, (FIXP_DBL)0x4000001b /* 3*/,
(FIXP_DBL)0x55a6127b /* 4*/, (FIXP_DBL)0x678dde8f /* 5*/, (FIXP_DBL)0x74ef0ed7 /* 6*/, (FIXP_DBL)0x7d33f0da /* 7*/
};
const FIXP_DBL FDKaacEnc_mTab_4_3Elc[512]={
FL2FXCONST_DBL(0.3968502629920499),FL2FXCONST_DBL(0.3978840634868335),FL2FXCONST_DBL(0.3989185359354711),FL2FXCONST_DBL(0.3999536794661432),

14
libAACenc/src/aacEnc_rom.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -105,11 +105,11 @@ amm-info@iis.fraunhofer.de
/*
Huffman Tables
*/
extern const INT FDKaacEnc_huff_ltab1_2[3][3][3][3];
extern const INT FDKaacEnc_huff_ltab3_4[3][3][3][3];
extern const INT FDKaacEnc_huff_ltab5_6[9][9];
extern const INT FDKaacEnc_huff_ltab7_8[8][8];
extern const INT FDKaacEnc_huff_ltab9_10[13][13];
extern const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3];
extern const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3];
extern const ULONG FDKaacEnc_huff_ltab5_6[9][9];
extern const ULONG FDKaacEnc_huff_ltab7_8[8][8];
extern const ULONG FDKaacEnc_huff_ltab9_10[13][13];
extern const UCHAR FDKaacEnc_huff_ltab11[17][17];
extern const UCHAR FDKaacEnc_huff_ltabscf[121];
extern const USHORT FDKaacEnc_huff_ctab1[3][3][3][3];
@ -123,7 +123,7 @@ extern const USHORT FDKaacEnc_huff_ctab8[8][8];
extern const USHORT FDKaacEnc_huff_ctab9[13][13];
extern const USHORT FDKaacEnc_huff_ctab10[13][13];
extern const USHORT FDKaacEnc_huff_ctab11[21][17];
extern const INT FDKaacEnc_huff_ctabscf[121];
extern const ULONG FDKaacEnc_huff_ctabscf[121];
/*
quantizer

89
libAACenc/src/aacenc.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -107,6 +107,39 @@ amm-info@iis.fraunhofer.de
#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,
INT framelength,
INT ancillaryRate,
@ -220,21 +253,19 @@ INT FDKaacEnc_GetVBRBitrate(INT bitrateMode, CHANNEL_MODE channelMode)
/**
* \brief Convert encoder bitreservoir value for transport library.
*
* \param bitrateMode Bitratemode used in current encoder instance. Se ::AACENC_BITRATE_MODE
* \param bitresTotal Encoder bitreservoir level in bits.
* \param hAacEnc Encoder handle
*
* \return Corrected bitreservoir level used in transport library.
*/
static INT FDKaacEnc_EncBitresToTpBitres(
const AACENC_BITRATE_MODE bitrateMode,
const INT bitresTotal
const HANDLE_AAC_ENC hAacEnc
)
{
INT transporBitreservoir = 0;
switch (bitrateMode) {
switch (hAacEnc->bitrateMode) {
case AACENC_BR_MODE_CBR:
transporBitreservoir = bitresTotal; /* encoder bitreservoir level */
transporBitreservoir = hAacEnc->qcKernel->bitResTot; /* encoder bitreservoir level */
break;
case AACENC_BR_MODE_VBR_1:
case AACENC_BR_MODE_VBR_2:
@ -253,6 +284,10 @@ static INT FDKaacEnc_EncBitresToTpBitres(
FDK_ASSERT(0);
}
if (hAacEnc->config->audioMuxVersion==2) {
transporBitreservoir = MIN_BUFSIZE_PER_EFF_CHAN * hAacEnc->channelMapping.nChannelsEff;
}
return transporBitreservoir;
}
@ -289,6 +324,7 @@ void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config)
config->minBitsPerFrame = -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->audioMuxVersion = -1; /* audio mux version not configured */
/* init tabs in fixpoint_math */
InitLdInt();
@ -435,7 +471,9 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
&averageBitsPerFrame,
config->bitrateMode,
config->nSubFrames
) != config->bitRate )
) != config->bitRate
&& !((config->bitrateMode>=1) && (config->bitrateMode<=5))
)
{
return AAC_ENC_UNSUPPORTED_BITRATE;
}
@ -562,7 +600,10 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
qcInit.averageBits = (averageBitsPerFrame+7)&~7;
qcInit.bitRes = 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
{
@ -573,9 +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 = (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 = (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;
@ -583,11 +626,9 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
qcInit.nSubFrames = config->nSubFrames;
qcInit.padding.paddingRest = config->sampleRate;
/* Calc meanPe */
bw_ratio = fDivNorm((FIXP_DBL)hAacEnc->bandwidth90dB, (FIXP_DBL)(config->sampleRate>>1), &qbw);
qbw = DFRACT_BITS-1-qbw;
/* 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 meanPe: qcInit.meanPe = 10.0f * FRAME_LEN_LONG * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */
bw_ratio = fDivNorm((FIXP_DBL)(10*config->framelength*hAacEnc->bandwidth90dB), (FIXP_DBL)(config->sampleRate), &qbw);
qcInit.meanPe = FDKmax((INT)scaleValue(bw_ratio, qbw+1-(DFRACT_BITS-1)), 1);
/* Calc maxBitFac */
mbfac = fDivNorm((MIN_BUFSIZE_PER_EFF_CHAN-744)*cm->nChannelsEff, qcInit.averageBits/qcInit.nSubFrames, &qmbfac);
@ -649,23 +690,7 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
if (ErrorStatus != AAC_ENC_OK)
goto bail;
/* Map virtual aot's to intern aot used in bitstream writer. */
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;
}
hAacEnc->aot = hAacEnc->config->audioObjectType;
/* common things */
@ -930,7 +955,7 @@ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc,
transportEnc_WriteAccessUnit(
hTpEnc,
totalBits,
FDKaacEnc_EncBitresToTpBitres(hAacEnc->bitrateMode, hAacEnc->qcKernel->bitResTot),
FDKaacEnc_EncBitresToTpBitres(hAacEnc),
cm->nChannelsEff);
/* write bitstream */

39
libAACenc/src/aacenc.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -98,6 +98,11 @@ amm-info@iis.fraunhofer.de
#include "sbr_encoder.h"
#define BITRES_MAX_LD 4000
#define BITRES_MIN_LD 500
#define BITRATE_MAX_LD 70000 /* Max assumed bitrate for bitres calculation */
#define BITRATE_MIN_LD 12000 /* Min assumed bitrate for bitres calculation */
#ifdef __cplusplus
extern "C" {
#endif
@ -205,6 +210,8 @@ struct AACENC_CONFIG {
INT maxBitsPerFrame; /* maximum number of bits in AU */
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 */
UCHAR useTns; /* flag: use temporal noise shaping */
@ -223,6 +230,36 @@ typedef struct {
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
* \param hTpEnc transport encoder handle

127
libAACenc/src/aacenc_lib.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -98,7 +98,7 @@ amm-info@iis.fraunhofer.de
/* Encoder library info */
#define AACENCODER_LIB_VL0 3
#define AACENCODER_LIB_VL1 4
#define AACENCODER_LIB_VL2 12
#define AACENCODER_LIB_VL2 22
#define AACENCODER_LIB_TITLE "AAC Encoder"
#ifdef __ANDROID__
#define AACENCODER_LIB_BUILD_DATE ""
@ -153,6 +153,7 @@ typedef struct {
UINT userAfterburner;
UINT userFramelength;
UINT userAncDataRate;
UINT userPeakBitrate;
UCHAR userTns; /*!< Use TNS coding. */
UCHAR userPns; /*!< Use PNS coding. */
@ -303,7 +304,7 @@ static AACENC_ERROR eldSbrConfigurator(
int i, cfgIdx = -1;
const ULONG channelBitrate = totalBitrate / FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff;
for (i=0; i<(sizeof(eldSbrAutoConfigTab)/sizeof(ELD_SBR_CONFIGURATOR)); i++) {
for (i=0; i<(int)(sizeof(eldSbrAutoConfigTab)/sizeof(ELD_SBR_CONFIGURATOR)); i++) {
if ( (samplingRate <= eldSbrAutoConfigTab[i].samplingRate)
&& (channelBitrate >= eldSbrAutoConfigTab[i].bitrateRange) )
{
@ -326,10 +327,7 @@ static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig)
{
INT sbrUsed = 0;
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) )
if ( (hAacConfig->audioObjectType==AOT_SBR) || (hAacConfig->audioObjectType==AOT_PS) )
{
sbrUsed = 1;
}
@ -345,10 +343,7 @@ static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType)
{
INT psUsed = 0;
if ( (audioObjectType==AOT_PS)
|| (audioObjectType==AOT_MP2_PS)
|| (audioObjectType==AOT_DABPLUS_PS)
|| (audioObjectType==AOT_DRM_MPEG_PS) )
if ( (audioObjectType==AOT_PS) )
{
psUsed = 1;
}
@ -373,8 +368,7 @@ static SBR_PS_SIGNALING getSbrSignalingMode(
sbrSignaling = SIG_IMPLICIT; /* default: implicit signaling */
}
if ((audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) ||
(audioObjectType==AOT_MP2_AAC_LC) || (audioObjectType==AOT_MP2_SBR) || (audioObjectType==AOT_MP2_PS) ) {
if ( (audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) ) {
switch (transportType) {
case TT_MP4_ADIF:
case TT_MP4_ADTS:
@ -430,22 +424,7 @@ static void FDKaacEnc_MapConfig(
cc->flags = 0;
/* Map virtual aot to transport aot. */
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;
}
transport_AOT = hAacConfig->audioObjectType;
if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) {
cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0;
@ -511,16 +490,7 @@ static void FDKaacEnc_MapConfig(
cc->samplingRate = hAacConfig->sampleRate;
/* Mpeg-4 signaling for transport library. */
switch ( hAacConfig->audioObjectType ) {
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;
}
cc->flags |= CC_MPEG_ID;
/* ER-tools signaling. */
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->userBitrate = hAacConfig->bitRate;
config->userBitrateMode = hAacConfig->bitrateMode;
config->userPeakBitrate = (UINT)-1;
config->userBandwidth = hAacConfig->bandWidth;
config->userTns = hAacConfig->useTns;
config->userPns = hAacConfig->usePns;
@ -792,12 +763,15 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
hAacConfig->syntaxFlags = 0;
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. */
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_SBR:
case AOT_PS:
@ -839,11 +813,16 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
switch ( hAacConfig->audioObjectType ) {
case AOT_ER_AAC_LD:
case AOT_ER_AAC_ELD:
if (config->userBitrateMode==8) {
hAacConfig->bitrateMode = 0;
}
if (config->userBitrateMode==0) {
hAacConfig->bitreservoir = 100*config->nChannels; /* default, reduced bitreservoir */
/* bitreservoir = (maxBitRes-minBitRes)/(maxBitRate-minBitrate)*(bitRate-minBitrate)+minBitRes; */
if ( isLowDelay(hAacConfig->audioObjectType) ) {
INT bitreservoir;
INT brPerChannel = hAacConfig->bitRate/hAacConfig->nChannels;
brPerChannel = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel));
FIXP_DBL slope = fDivNorm((brPerChannel-BITRATE_MIN_LD), BITRATE_MAX_LD-BITRATE_MIN_LD); /* calc slope for interpolation */
bitreservoir = fMultI(slope, (INT)(BITRES_MAX_LD-BITRES_MIN_LD)) + BITRES_MIN_LD; /* interpolate */
hAacConfig->bitreservoir = bitreservoir & ~7; /* align to bytes */
}
}
if (hAacConfig->bitrateMode!=0) {
return AACENC_INVALID_CONFIG;
@ -884,6 +863,18 @@ 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 */
if ( (hAacConfig->audioObjectType==AOT_ER_AAC_ELD)
&& (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio==0) )
@ -914,7 +905,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
}
else {
/* SBR ratio has been set by the user, so use it. */
hAacConfig->sbrRatio = config->userSbrRatio;
hAacConfig->sbrRatio = isSbrActive(hAacConfig) ? config->userSbrRatio : 0;
}
{
@ -1139,7 +1130,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder,
hAacConfig);
/* create flags for transport encoder */
if (config->userTpAmxv == 1) {
if (config->userTpAmxv != 0) {
flags |= TP_FLAG_LATM_AMV;
}
/* Clear output buffer */
@ -1569,7 +1560,7 @@ AACENC_ERROR aacEncEncode(
&& ((hAacEncoder->extParam.userChannelMode==MODE_1_2_2)||(hAacEncoder->extParam.userChannelMode==MODE_1_2_2_1)) )
{
/* 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) {
hAacEncoder->extParam.userPceAdditions = pceValue;
hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
@ -1785,19 +1776,16 @@ AACENC_ERROR aacEncoder_SetParam(
/* check if AOT matches the allocated modules */
switch ( value ) {
case AOT_PS:
case AOT_MP2_PS:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) {
err = AACENC_INVALID_CONFIG;
goto bail;
}
case AOT_SBR:
case AOT_MP2_SBR:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) {
err = AACENC_INVALID_CONFIG;
goto bail;
}
case AOT_AAC_LC:
case AOT_MP2_AAC_LC:
case AOT_ER_AAC_LD:
case AOT_ER_AAC_ELD:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) {
@ -1823,11 +1811,7 @@ AACENC_ERROR aacEncoder_SetParam(
if (settings->userBitrateMode != value) {
switch ( value ) {
case 0:
case 1:
case 2:
case 3:
case 4:
case 5:
case 1: case 2: case 3: case 4: case 5:
case 8:
settings->userBitrateMode = value;
hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
@ -1978,6 +1962,16 @@ AACENC_ERROR aacEncoder_SetParam(
hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
}
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:
if (settings->userTpNsubFrames != value) {
if (! ( (value>=1) && (value<=4) ) ) {
@ -2003,7 +1997,7 @@ AACENC_ERROR aacEncoder_SetParam(
break;
case AACENC_METADATA_MODE:
if ((UINT)settings->userMetaDataMode != value) {
if ( !((value>=0) && (value<=2)) ) {
if ( !(((INT)value>=0) && ((INT)value<=2)) ) {
err = AACENC_INVALID_CONFIG;
break;
}
@ -2011,6 +2005,12 @@ AACENC_ERROR aacEncoder_SetParam(
hAacEncoder->InitFlags |= AACENC_INIT_CONFIG;
}
break;
case AACENC_PEAK_BITRATE:
if (settings->userPeakBitrate != value) {
settings->userPeakBitrate = value;
hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
}
break;
default:
err = AACENC_UNSUPPORTED_PARAMETER;
break;
@ -2081,6 +2081,9 @@ UINT aacEncoder_GetParam(
case AACENC_HEADER_PERIOD:
value = (UINT)hAacEncoder->coderConfig.headerPeriod;
break;
case AACENC_AUDIOMUXVER:
value = (UINT)hAacEncoder->aacConfig.audioMuxVersion;
break;
case AACENC_TPSUBFRAMES:
value = (UINT)settings->userTpNsubFrames;
break;
@ -2093,6 +2096,12 @@ UINT aacEncoder_GetParam(
case AACENC_METADATA_MODE:
value = (hAacEncoder->metaDataAllowed==0) ? 0 : (UINT)settings->userMetaDataMode;
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:
//err = MPS_INVALID_PARAMETER;
break;

102
libAACenc/src/aacenc_tns.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -95,13 +95,7 @@ amm-info@iis.fraunhofer.de
#include "aacEnc_rom.h"
#include "aacenc_tns.h"
enum {
HIFILT = 0, /* index of higher filter */
LOFILT = 1 /* index of lower filter */
};
#define FILTER_DIRECTION 0
#define FILTER_DIRECTION 0 /* 0 = up, 1 = down */
static const FIXP_DBL acfWindowLong[12+3+1] = {
0x7fffffff,0x7fb80000,0x7ee00000,0x7d780000,0x7b800000,0x78f80000,0x75e00000,0x72380000,
@ -112,20 +106,6 @@ static const FIXP_DBL acfWindowShort[4+3+1] = {
0x7fffffff,0x7e000000,0x78000000,0x6e000000,0x60000000,0x4e000000,0x38000000,0x1e000000
};
typedef struct {
INT filterEnabled[MAX_NUM_OF_FILTERS];
INT threshOn[MAX_NUM_OF_FILTERS]; /* min. prediction gain for using tns TABUL*/
INT filterStartFreq[MAX_NUM_OF_FILTERS]; /* lowest freq for lpc TABUL*/
INT tnsLimitOrder[MAX_NUM_OF_FILTERS]; /* Limit for TNS order TABUL*/
INT tnsFilterDirection[MAX_NUM_OF_FILTERS]; /* Filtering direction, 0=up, 1=down TABUL */
INT acfSplit[MAX_NUM_OF_FILTERS];
FIXP_DBL tnsTimeResolution[MAX_NUM_OF_FILTERS]; /* TNS max. time resolution TABUL. Should be fract but MSVC won't compile then */
INT seperateFiltersAllowed;
} TNS_PARAMETER_TABULATED;
typedef struct{
INT bitRateFrom[2]; /* noneSbr=0, useSbr=1 */
INT bitRateTo[2]; /* noneSbr=0, useSbr=1 */
@ -373,6 +353,7 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate,
INT channels,
INT blockType,
INT granuleLength,
INT isLowDelay,
INT ldSbrPresent,
TNS_CONFIG *tC,
PSY_CONFIGURATION *pC,
@ -385,6 +366,8 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate,
if (channels <= 0)
return (AAC_ENCODER_ERROR)1;
tC->isLowDelay = isLowDelay;
/* initialize TNS filter flag, order, and coefficient resolution (in bits per coeff) */
tC->tnsActive = (active) ? TRUE : FALSE;
tC->maxOrder = (blockType == SHORT_WINDOW) ? 5 : 12; /* maximum: 7, 20 */
@ -450,27 +433,14 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate,
const TNS_PARAMETER_TABULATED* pCfg = FDKaacEnc_GetTnsParam(bitRate, channels, ldSbrPresent);
if ( pCfg != NULL ) {
FDKmemcpy(&(tC->confTab), pCfg, sizeof(tC->confTab));
tC->lpcStartBand[HIFILT] = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[HIFILT], sampleRate, pC->sfbCnt, pC->sfbOffset);
tC->lpcStartLine[HIFILT] = pC->sfbOffset[tC->lpcStartBand[HIFILT]];
tC->lpcStartBand[LOFILT] = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[LOFILT], sampleRate, pC->sfbCnt, pC->sfbOffset);
tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]];
tC->confTab.threshOn[HIFILT] = pCfg->threshOn[HIFILT];
tC->confTab.threshOn[LOFILT] = pCfg->threshOn[LOFILT];
tC->confTab.tnsLimitOrder[HIFILT] = pCfg->tnsLimitOrder[HIFILT];
tC->confTab.tnsLimitOrder[LOFILT] = pCfg->tnsLimitOrder[LOFILT];
tC->confTab.tnsFilterDirection[HIFILT] = pCfg->tnsFilterDirection[HIFILT];
tC->confTab.tnsFilterDirection[LOFILT] = pCfg->tnsFilterDirection[LOFILT];
tC->confTab.acfSplit[HIFILT] = pCfg->acfSplit[HIFILT];
tC->confTab.acfSplit[LOFILT] = pCfg->acfSplit[LOFILT];
tC->confTab.filterEnabled[HIFILT] = pCfg->filterEnabled[HIFILT];
tC->confTab.filterEnabled[LOFILT] = pCfg->filterEnabled[LOFILT];
tC->confTab.seperateFiltersAllowed = pCfg->seperateFiltersAllowed;
FDKaacEnc_CalcGaussWindow(tC->acfWindow[HIFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[HIFILT], TNS_TIMERES_SCALE);
FDKaacEnc_CalcGaussWindow(tC->acfWindow[LOFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[LOFILT], TNS_TIMERES_SCALE);
}
@ -614,6 +584,7 @@ static inline FIXP_DBL FDKaacEnc_AutoCorrNormFac(
static void FDKaacEnc_MergedAutoCorrelation(
const FIXP_DBL *spectrum,
const INT isLowDelay,
const FIXP_DBL acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER+3+1],
const INT lpcStartLine[MAX_NUM_OF_FILTERS],
const INT lpcStopLine,
@ -633,6 +604,8 @@ static void FDKaacEnc_MergedAutoCorrelation(
FDKmemclear(&_rxx1[0], sizeof(FIXP_DBL)*(maxOrder+1));
FDKmemclear(&_rxx2[0], sizeof(FIXP_DBL)*(maxOrder+1));
idx0 = idx1 = idx2 = idx3 = idx4 = 0;
/* MDCT line indices separating the 1st, 2nd, 3rd, and 4th analysis quarters */
if ( (acfSplit[LOFILT]==-1) || (acfSplit[HIFILT]==-1) ) {
/* autocorrelation function for 1st, 2nd, 3rd, and 4th quarter of the spectrum */
@ -676,17 +649,27 @@ static void FDKaacEnc_MergedAutoCorrelation(
/* compute energy normalization factors, i. e. 1/energy (saves some divisions) */
if (rxx1_0 != FL2FXCONST_DBL(0.f))
{
INT sc_fac1 = -1;
FIXP_DBL fac1 = FDKaacEnc_AutoCorrNormFac(rxx1_0, ((-2*sc1)+nsc1), &sc_fac1);
_rxx1[0] = scaleValue(fMult(rxx1_0,fac1),sc_fac1);
INT sc_fac1 = -1;
FIXP_DBL fac1 = FDKaacEnc_AutoCorrNormFac(rxx1_0, ((-2*sc1)+nsc1), &sc_fac1);
_rxx1[0] = scaleValue(fMult(rxx1_0,fac1),sc_fac1);
if (isLowDelay)
{
for (lag = 1; lag <= maxOrder; lag++) {
/* compute energy-normalized and windowed autocorrelation values at this lag */
FIXP_DBL x1 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, lag, nsc1);
_rxx1[lag] = fMult(scaleValue(fMult(x1,fac1),sc_fac1), acfWindow[LOFILT][lag]);
}
}
else
{
for (lag = 1; lag <= maxOrder; lag++) {
if ((3 * lag) <= maxOrder + 3) {
FIXP_DBL x1 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, lag, nsc1);
_rxx1[lag] = fMult(scaleValue(fMult(x1,fac1),sc_fac1), acfWindow[LOFILT][3*lag]);
}
}
}
}
/* auto corr over upper 3/4 of spectrum */
@ -762,8 +745,12 @@ INT FDKaacEnc_TnsDetect(
: &tnsData->dataRaw.Long.subBlockInfo;
tnsData->filtersMerged = FALSE;
tsbi->tnsActive = FALSE;
tsbi->predictionGain = 1000;
tsbi->tnsActive[HIFILT] = FALSE;
tsbi->predictionGain[HIFILT] = 1000;
tsbi->tnsActive[LOFILT] = FALSE;
tsbi->predictionGain[LOFILT] = 1000;
tnsInfo->numOfFilters[subBlockNumber] = 0;
tnsInfo->coefRes[subBlockNumber] = tC->coefRes;
for (i = 0; i < tC->maxOrder; i++) {
@ -779,6 +766,7 @@ INT FDKaacEnc_TnsDetect(
FDKaacEnc_MergedAutoCorrelation(
spectrum,
tC->isLowDelay,
tC->acfWindow,
tC->lpcStartLine,
tC->lpcStopLine,
@ -788,7 +776,7 @@ INT FDKaacEnc_TnsDetect(
rxx2);
/* compute higher TNS filter in lattice (ParCor) form with LeRoux-Gueguen algorithm */
tsbi->predictionGain = FDKaacEnc_AutoToParcor(rxx2, parcor_tmp, tC->confTab.tnsLimitOrder[HIFILT]);
tsbi->predictionGain[HIFILT] = FDKaacEnc_AutoToParcor(rxx2, parcor_tmp, tC->confTab.tnsLimitOrder[HIFILT]);
/* non-linear quantization of TNS lattice coefficients with given resolution */
FDKaacEnc_Parcor2Index(
@ -815,9 +803,9 @@ INT FDKaacEnc_TnsDetect(
tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[HIFILT];
/* disable TNS if predictionGain is less than 3dB or sumSqrCoef is too small */
if ((tsbi->predictionGain > tC->confTab.threshOn[HIFILT]) || (sumSqrCoef > (tC->confTab.tnsLimitOrder[HIFILT]/2 + 2)))
if ((tsbi->predictionGain[HIFILT] > tC->confTab.threshOn[HIFILT]) || (sumSqrCoef > (tC->confTab.tnsLimitOrder[HIFILT]/2 + 2)))
{
tsbi->tnsActive = TRUE;
tsbi->tnsActive[HIFILT] = TRUE;
tnsInfo->numOfFilters[subBlockNumber]++;
/* compute second filter for lower quarter; only allowed for long windows! */
@ -857,6 +845,7 @@ INT FDKaacEnc_TnsDetect(
|| ( (sumSqrCoef > 9) && (sumSqrCoef < 22 * tC->confTab.tnsLimitOrder[LOFILT]) ) )
{
/* compare lower to upper filter; if they are very similar, merge them */
tsbi->tnsActive[LOFILT] = TRUE;
sumSqrCoef = 0;
for (i = 0; i < tC->confTab.tnsLimitOrder[LOFILT]; i++) {
sumSqrCoef += FDKabs(tnsInfo->coef[subBlockNumber][HIFILT][i] - tnsInfo->coef[subBlockNumber][LOFILT][i]);
@ -884,6 +873,8 @@ INT FDKaacEnc_TnsDetect(
tnsInfo->numOfFilters[subBlockNumber]++;
}
} /* filter lower part */
tsbi->predictionGain[LOFILT]=predGain;
} /* second filter allowed */
} /* if predictionGain > 1437 ... */
} /* maxOrder > 0 && tnsActive */
@ -944,7 +935,7 @@ void FDKaacEnc_TnsSync(
INT doSync = 1, absDiffSum = 0;
/* if TNS is active in at least one channel, check if ParCor coefficients of higher filter are similar */
if (pSbInfoDestW->tnsActive || pSbInfoSrcW->tnsActive) {
if (pSbInfoDestW->tnsActive[HIFILT] || pSbInfoSrcW->tnsActive[HIFILT]) {
for (i = 0; i < tC->maxOrder; i++) {
absDiff = FDKabs(tnsInfoDest->coef[w][HIFILT][i] - tnsInfoSrc->coef[w][HIFILT][i]);
absDiffSum += absDiff;
@ -957,12 +948,12 @@ void FDKaacEnc_TnsSync(
if (doSync) {
/* if no significant difference was detected, synchronize coefficient sets */
if (pSbInfoSrcW->tnsActive) {
if (pSbInfoSrcW->tnsActive[HIFILT]) {
/* no dest filter, or more dest than source filters: use one dest filter */
if ((!pSbInfoDestW->tnsActive) ||
((pSbInfoDestW->tnsActive) && (tnsInfoDest->numOfFilters[w] > tnsInfoSrc->numOfFilters[w])))
if ((!pSbInfoDestW->tnsActive[HIFILT]) ||
((pSbInfoDestW->tnsActive[HIFILT]) && (tnsInfoDest->numOfFilters[w] > tnsInfoSrc->numOfFilters[w])))
{
pSbInfoDestW->tnsActive = tnsInfoDest->numOfFilters[w] = 1;
pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 1;
}
tnsDataDest->filtersMerged = tnsDataSrc->filtersMerged;
tnsInfoDest->order [w][HIFILT] = tnsInfoSrc->order [w][HIFILT];
@ -975,7 +966,7 @@ void FDKaacEnc_TnsSync(
}
}
else
pSbInfoDestW->tnsActive = tnsInfoDest->numOfFilters[w] = 0;
pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 0;
}
}
@ -1012,8 +1003,8 @@ INT FDKaacEnc_TnsEncode(
{
INT i, startLine, stopLine;
if ( ( (blockType == SHORT_WINDOW) && (!tnsData->dataRaw.Short.subBlockInfo[subBlockNumber].tnsActive) )
|| ( (blockType != SHORT_WINDOW) && (!tnsData->dataRaw.Long.subBlockInfo.tnsActive) ) )
if ( ( (blockType == SHORT_WINDOW) && (!tnsData->dataRaw.Short.subBlockInfo[subBlockNumber].tnsActive[HIFILT]) )
|| ( (blockType != SHORT_WINDOW) && (!tnsData->dataRaw.Long.subBlockInfo.tnsActive[HIFILT]) ) )
{
return 1;
}
@ -1129,8 +1120,9 @@ static INT FDKaacEnc_AutoToParcor(
FIXP_DBL *RESTRICT workBuffer = parcorWorkBuffer;
const FIXP_DBL autoCorr_0 = input[0];
FDKmemclear(reflCoeff,numOfCoeff*sizeof(FIXP_DBL));
if((FIXP_DBL)input[0] == FL2FXCONST_DBL(0.0)) {
FDKmemclear(reflCoeff,numOfCoeff*sizeof(FIXP_DBL));
return(predictionGain);
}

30
libAACenc/src/aacenc_tns.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -117,21 +117,25 @@ amm-info@iis.fraunhofer.de
#define MAX_NUM_OF_FILTERS 2
#define HIFILT 0 /* index of higher filter */
#define LOFILT 1 /* index of lower filter */
typedef struct{ /*stuff that is tabulated dependent on bitrate etc. */
INT filterEnabled[MAX_NUM_OF_FILTERS];
INT threshOn[MAX_NUM_OF_FILTERS]; /* min. prediction gain for using tns TABUL*/
INT tnsLimitOrder[MAX_NUM_OF_FILTERS]; /* Limit for TNS order TABUL*/
INT tnsFilterDirection[MAX_NUM_OF_FILTERS]; /* Filtering direction, 0=up, 1=down TABUL */
INT acfSplit[MAX_NUM_OF_FILTERS];
INT seperateFiltersAllowed;
}TNS_CONFIG_TABULATED;
typedef struct{ /* stuff that is tabulated dependent on bitrate etc. */
INT filterEnabled[MAX_NUM_OF_FILTERS];
INT threshOn[MAX_NUM_OF_FILTERS]; /* min. prediction gain for using tns TABUL*/
INT filterStartFreq[MAX_NUM_OF_FILTERS]; /* lowest freq for lpc TABUL*/
INT tnsLimitOrder[MAX_NUM_OF_FILTERS]; /* Limit for TNS order TABUL*/
INT tnsFilterDirection[MAX_NUM_OF_FILTERS]; /* Filtering direction, 0=up, 1=down TABUL */
INT acfSplit[MAX_NUM_OF_FILTERS];
FIXP_DBL tnsTimeResolution[MAX_NUM_OF_FILTERS]; /* TNS max. time resolution TABUL. Should be fract but MSVC won't compile then */
INT seperateFiltersAllowed;
} TNS_PARAMETER_TABULATED;
typedef struct { /*assigned at InitTime*/
TNS_CONFIG_TABULATED confTab;
TNS_PARAMETER_TABULATED confTab;
INT isLowDelay;
INT tnsActive;
INT maxOrder; /* max. order of tns filter */
INT coefRes;
@ -148,8 +152,8 @@ typedef struct { /*assigned at InitTime*/
typedef struct {
INT tnsActive;
INT predictionGain;
INT tnsActive[MAX_NUM_OF_FILTERS];
INT predictionGain[MAX_NUM_OF_FILTERS];
} TNS_SUBBLOCK_INFO;
typedef struct{ /*changed at runTime*/

97
libAACenc/src/adj_thr.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -130,14 +130,14 @@ the crash recovery strategy will be activated once.
typedef struct {
INT bitrate;
LONG bits2PeFactor_mono;
LONG bits2PeFactor_mono_slope;
LONG bits2PeFactor_stereo;
LONG bits2PeFactor_stereo_slope;
LONG bits2PeFactor_mono_scfOpt;
LONG bits2PeFactor_mono_scfOpt_slope;
LONG bits2PeFactor_stereo_scfOpt;
LONG bits2PeFactor_stereo_scfOpt_slope;
ULONG bits2PeFactor_mono;
ULONG bits2PeFactor_mono_slope;
ULONG bits2PeFactor_stereo;
ULONG bits2PeFactor_stereo_slope;
ULONG bits2PeFactor_mono_scfOpt;
ULONG bits2PeFactor_mono_scfOpt_slope;
ULONG bits2PeFactor_stereo_scfOpt;
ULONG bits2PeFactor_stereo_scfOpt_slope;
} BIT_PE_SFAC;
@ -153,10 +153,10 @@ static const BIT_PE_SFAC S_Bits2PeTab16000[] = {
{ 24000, 0x23D70A3D, 0x029F16B1, 0x2199999A, 0x07DD4413, 0x23D70A3D, 0x029F16B1, 0x2199999A, 0x07DD4413},
{ 32000, 0x247AE148, 0x11B1D92B, 0x23851EB8, 0x01F75105, 0x247AE148, 0x110A137F, 0x23851EB8, 0x01F75105},
{ 48000, 0x2D1EB852, 0x6833C600, 0x247AE148, 0x014F8B59, 0x2CCCCCCD, 0x68DB8BAC, 0x247AE148, 0x01F75105},
{ 64000, 0x60000000, 0x00000000, 0x251EB852, 0x154C985F, 0x60000000, 0x00000000, 0x2570A3D7, 0x154C985F},
{ 96000, 0x60000000, 0x00000000, 0x39EB851F, 0x088509C0, 0x60000000, 0x00000000, 0x3A3D70A4, 0x088509C0},
{128000, 0x60000000, 0x00000000, 0x423D70A4, 0x18A43BB4, 0x60000000, 0x00000000, 0x428F5C29, 0x181E03F7},
{148000, 0x60000000, 0x00000000, 0x5147AE14, 0x00000000, 0x60000000, 0x00000000, 0x5147AE14, 0x00000000}
{ 64000, 0x25c28f40, 0x00000000, 0x251EB852, 0x01480000, 0x25c28f40, 0x00000000, 0x2570A3D7, 0x01480000},
{ 96000, 0x25c28f40, 0x00000000, 0x26000000, 0x01000000, 0x25c28f40, 0x00000000, 0x26000000, 0x01000000},
{128000, 0x25c28f40, 0x00000000, 0x270a3d80, 0x01000000, 0x25c28f40, 0x00000000, 0x270a3d80, 0x01000000},
{148000, 0x25c28f40, 0x00000000, 0x28000000, 0x00000000, 0x25c28f40, 0x00000000, 0x28000000, 0x00000000}
};
static const BIT_PE_SFAC S_Bits2PeTab22050[] = {
@ -166,8 +166,8 @@ static const BIT_PE_SFAC S_Bits2PeTab22050[] = {
{ 48000, 0x23d70a3d, 0x014f8b59, 0x2199999a, 0x03eea20a, 0x23d70a3d, 0x14f8b59, 0x2199999a, 0x03eea20a},
{ 64000, 0x247ae148, 0x08d8ec96, 0x23851eb8, 0x00fba882, 0x247ae148, 0x88509c0, 0x23851eb8, 0x00fba882},
{ 96000, 0x2d1eb852, 0x3419e300, 0x247ae148, 0x00a7c5ac, 0x2ccccccd, 0x346dc5d6, 0x247ae148, 0x00fba882},
{128000, 0x60000000, 0x00000000, 0x251eb852, 0x029f16b1, 0x60000000, 0x00000000, 0x2570a3d7, 0x009f16b1},
{148000, 0x60000000, 0x00000000, 0x26b851ec, 0x00000000, 0x60000000, 0x00000000, 0x270a3d71, 0x00000000}
{128000, 0x25c28f40, 0x00000000, 0x251eb852, 0x029f16b1, 0x60000000, 0x25c28f40, 0x2570a3d7, 0x009f16b1},
{148000, 0x25c28f40, 0x00000000, 0x26b851ec, 0x00000000, 0x60000000, 0x25c28f40, 0x270a3d71, 0x00000000}
};
static const BIT_PE_SFAC S_Bits2PeTab24000[] = {
@ -178,21 +178,21 @@ static const BIT_PE_SFAC S_Bits2PeTab24000[] = {
{ 64000, 0x24cccccd, 0x05e5f30e, 0x22e147ae, 0x01a36e2f, 0x24cccccd, 0x05e5f30e, 0x23333333, 0x014f8b59},
{ 96000, 0x2a8f5c29, 0x24b33db0, 0x247ae148, 0x00fba882, 0x2a8f5c29, 0x26fe718b, 0x247ae148, 0x00fba882},
{128000, 0x4e666666, 0x1cd5f99c, 0x2570a3d7, 0x010c6f7a, 0x50a3d70a, 0x192a7371, 0x2570a3d7, 0x010c6f7a},
{148000, 0x60000000, 0x00000000, 0x26147ae1, 0x00000000, 0x60000000, 0x00000000, 0x26147ae1, 0x00000000}
{148000, 0x25c28f40, 0x00000000, 0x26147ae1, 0x00000000, 0x25c28f40, 0x00000000, 0x26147ae1, 0x00000000}
};
static const BIT_PE_SFAC S_Bits2PeTab32000[] = {
{ 16000, 0x1199999a, 0x20c49ba6, 0x00000000, 0x4577d955, 0x00000000, 0x60fe4799, 0x00000000, 0x00000000},
{ 24000, 0x1999999a, 0x0fba8827, 0x10f5c28f, 0x1b866e44, 0x17ae147b, 0x0fba8827, 0x00000000, 0x4d551d69},
{ 16000, 0x247ae140, 0xFFFFAC1E, 0x270a3d80, 0xFFFE9B7C, 0x14ccccc0, 0x000110A1, 0x15c28f60, 0xFFFEEF5F},
{ 24000, 0x23333340, 0x0fba8827, 0x21999980, 0x1b866e44, 0x18f5c280, 0x0fba8827, 0x119999a0, 0x4d551d69},
{ 32000, 0x1d70a3d7, 0x07357e67, 0x17ae147b, 0x09d49518, 0x1b851eb8, 0x0a7c5ac4, 0x12e147ae, 0x110a137f},
{ 48000, 0x20f5c28f, 0x049667b6, 0x1c7ae148, 0x053e2d62, 0x20a3d70a, 0x053e2d62, 0x1b333333, 0x05e5f30e},
{ 64000, 0x23333333, 0x029f16b1, 0x1f0a3d71, 0x02f2f987, 0x23333333, 0x029f16b1, 0x1e147ae1, 0x03eea20a},
{ 96000, 0x25c28f5c, 0x2c3c9eed, 0x21eb851f, 0x01f75105, 0x25c28f5c, 0x0a7c5ac4, 0x21eb851f, 0x01a36e2f},
{128000, 0x50f5c28f, 0x18a43bb4, 0x23d70a3d, 0x010c6f7a, 0x30000000, 0x168b5cc0, 0x23851eb8, 0x0192a737},
{148000, 0x60000000, 0x00000000, 0x247ae148, 0x00dfb23b, 0x3dc28f5c, 0x300f4aaf, 0x247ae148, 0x01bf6476},
{160000, 0x60000000, 0xb15b5740, 0x24cccccd, 0x053e2d62, 0x4f5c28f6, 0xbefd0072, 0x251eb852, 0x04fb1184},
{200000, 0x00000000, 0x00000000, 0x2b333333, 0x0836be91, 0x00000000, 0x00000000, 0x2b333333, 0x0890390f},
{320000, 0x00000000, 0x00000000, 0x4947ae14, 0x00000000, 0x00000000, 0x00000000, 0x4a8f5c29, 0x00000000}
{148000, 0x25c28f40, 0x00000000, 0x247ae148, 0x00dfb23b, 0x3dc28f5c, 0x300f4aaf, 0x247ae148, 0x01bf6476},
{160000, 0x25c28f40, 0xb15b5740, 0x24cccccd, 0x053e2d62, 0x4f5c28f6, 0xbefd0072, 0x251eb852, 0x04fb1184},
{200000, 0x25c28f40, 0x00000000, 0x2b333333, 0x0836be91, 0x25c28f40, 0x00000000, 0x2b333333, 0x0890390f},
{320000, 0x25c28f40, 0x00000000, 0x4947ae14, 0x00000000, 0x25c28f40, 0x00000000, 0x4a8f5c29, 0x00000000}
};
static const BIT_PE_SFAC S_Bits2PeTab44100[] = {
@ -205,8 +205,8 @@ static const BIT_PE_SFAC S_Bits2PeTab44100[] = {
{128000, 0x2ae147ae, 0x1b435265, 0x223d70a4, 0x0192a737, 0x2a3d70a4, 0x1040bfe4, 0x21eb851f, 0x0192a737},
{148000, 0x3b851eb8, 0x2832069c, 0x23333333, 0x00dfb23b, 0x3428f5c3, 0x2054c288, 0x22e147ae, 0x00dfb23b},
{160000, 0x4a3d70a4, 0xc32ebe5a, 0x23851eb8, 0x01d5c316, 0x40000000, 0xcb923a2b, 0x23333333, 0x01d5c316},
{200000, 0x00000000, 0x00000000, 0x25c28f5c, 0x0713f078, 0x00000000, 0x00000000, 0x2570a3d7, 0x072a4f17},
{320000, 0x00000000, 0x00000000, 0x3fae147b, 0x00000000, 0x00000000, 0x00000000, 0x3fae147b, 0x00000000}
{200000, 0x25c28f40, 0x00000000, 0x25c28f5c, 0x0713f078, 0x25c28f40, 0x00000000, 0x2570a3d7, 0x072a4f17},
{320000, 0x25c28f40, 0x00000000, 0x3fae147b, 0x00000000, 0x25c28f40, 0x00000000, 0x3fae147b, 0x00000000}
};
static const BIT_PE_SFAC S_Bits2PeTab48000[] = {
@ -219,8 +219,8 @@ static const BIT_PE_SFAC S_Bits2PeTab48000[] = {
{128000, 0x28f5c28f, 0x14727dcc, 0x2147ae14, 0x0218def4, 0x2851eb85, 0x0e27e0f0, 0x20f5c28f, 0x0218def4},
{148000, 0x3570a3d7, 0x1cd5f99c, 0x228f5c29, 0x01bf6476, 0x30f5c28f, 0x18777e75, 0x223d70a4, 0x01bf6476},
{160000, 0x40000000, 0xcb923a2b, 0x23333333, 0x0192a737, 0x39eb851f, 0xd08d4bae, 0x22e147ae, 0x0192a737},
{200000, 0x00000000, 0x00000000, 0x251eb852, 0x06775a1b, 0x00000000, 0x00000000, 0x24cccccd, 0x06a4175a},
{320000, 0x00000000, 0x00000000, 0x3ccccccd, 0x00000000, 0x00000000, 0x00000000, 0x3d1eb852, 0x00000000}
{200000, 0x25c28f40, 0x00000000, 0x251eb852, 0x06775a1b, 0x25c28f40, 0x00000000, 0x24cccccd, 0x06a4175a},
{320000, 0x25c28f40, 0x00000000, 0x3ccccccd, 0x00000000, 0x25c28f40, 0x00000000, 0x3d1eb852, 0x00000000}
};
static const BITS2PE_CFG_TAB bits2PeConfigTab[] = {
@ -258,6 +258,7 @@ static void FDKaacEnc_InitBits2PeFactor(
const INT nChannels,
const INT sampleRate,
const INT advancedBitsToPe,
const INT dZoneQuantEnable,
const INT invQuant
)
{
@ -329,7 +330,32 @@ static void FDKaacEnc_InitBits2PeFactor(
} /* advancedBitsToPe */
/* return bits2pe factor */
if (dZoneQuantEnable)
{
if(bit2PE_m >= (FL2FXCONST_DBL(0.6f))>>bit2PE_e)
{
/* Additional headroom for addition */
bit2PE_m >>= 1;
bit2PE_e += 1;
}
/* the quantTendencyCompensator compensates a lower bit consumption due to increasing the tendency to quantize low spectral values to the lower quantizer border for bitrates below a certain bitrate threshold --> see also function calcSfbDistLD in quantize.c */
if ((bitRate/nChannels > 32000) && (bitRate/nChannels <= 40000)) {
bit2PE_m += (FL2FXCONST_DBL(0.4f))>>bit2PE_e;
}
else if (bitRate/nChannels > 20000) {
bit2PE_m += (FL2FXCONST_DBL(0.3f))>>bit2PE_e;
}
else if (bitRate/nChannels >= 16000) {
bit2PE_m += (FL2FXCONST_DBL(0.3f))>>bit2PE_e;
}
else {
bit2PE_m += (FL2FXCONST_DBL(0.0f))>>bit2PE_e;
}
}
/***** 3.) Return bits2pe factor *****/
*bits2PeFactor_m = bit2PE_m;
*bits2PeFactor_e = bit2PE_e;
}
@ -1649,6 +1675,7 @@ static void FDKaacEnc_adaptThresholdsToPe(CHANNEL_MAPPING* cm,
QC_OUT_ELEMENT* qcElement[(8)],
PSY_OUT_ELEMENT* psyOutElement[(8)],
const INT desiredPe,
const INT maxIter2ndGuess,
const INT processElements,
const INT elementOffset)
{
@ -1733,7 +1760,7 @@ static void FDKaacEnc_adaptThresholdsToPe(CHANNEL_MAPPING* cm,
/* Part III: Iterate until bit constraints are met */
/* -------------------------------------------------- */
iter = 0;
while ((fixp_abs(redPeGlobal - desiredPe) > fMultI(FL2FXCONST_DBL(0.05f),desiredPe)) && (iter < 1)) {
while ((fixp_abs(redPeGlobal - desiredPe) > fMultI(FL2FXCONST_DBL(0.05f),desiredPe)) && (iter < maxIter2ndGuess)) {
INT desiredPeNoAHGlobal;
INT redPeNoAHGlobal = 0;
@ -2138,7 +2165,7 @@ static FIXP_DBL FDKaacEnc_bitresCalcBitFac(const INT bitresBits,
bresParam->clipSpendLow, bresParam->clipSpendHigh,
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);
bitresFac_fix = (UNITY>>1) - ((FIXP_DBL)bitSave>>1) + (tmp_fix>>1); qbres = (DFRACT_BITS-2);
@ -2225,7 +2252,8 @@ void FDKaacEnc_AdjThrInit(
INT nChannelsEff,
INT sampleRate,
INT advancedBitsToPe,
FIXP_DBL vbrQualFactor
FIXP_DBL vbrQualFactor,
const INT dZoneQuantEnable
)
{
INT i;
@ -2233,6 +2261,10 @@ void FDKaacEnc_AdjThrInit(
FIXP_DBL POINT8 = FL2FXCONST_DBL(0.8f);
FIXP_DBL POINT6 = FL2FXCONST_DBL(0.6f);
/* Max number of iterations in second guess is 3 for lowdelay aot and for configurations with
multiple audio elements in general, otherwise iteration value is always 1. */
hAdjThr->maxIter2ndGuess = (advancedBitsToPe!=0 || nElements>1) ? 3 : 1;
/* common for all elements: */
/* parameters for bitres control */
hAdjThr->bresParamLong.clipSaveLow = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */
@ -2313,10 +2345,11 @@ void FDKaacEnc_AdjThrInit(
FDKaacEnc_InitBits2PeFactor(
&atsElem->bits2PeFactor_m,
&atsElem->bits2PeFactor_e,
chBitrate, /* bitrate/channel*/
chBitrate*nChannelsEff, /* overall bitrate */
nChannelsEff, /* number of channels */
sampleRate,
advancedBitsToPe,
dZoneQuantEnable,
invQuant
);
@ -2545,6 +2578,7 @@ void FDKaacEnc_AdjustThresholds(ATS_ELEMENT* AdjThrStateElement[(8)],
QC_OUT* qcOut,
PSY_OUT_ELEMENT* psyOutElement[(8)],
INT CBRbitrateMode,
INT maxIter2ndGuess,
CHANNEL_MAPPING* cm)
{
int i;
@ -2570,6 +2604,7 @@ void FDKaacEnc_AdjustThresholds(ATS_ELEMENT* AdjThrStateElement[(8)],
qcElement,
psyOutElement,
qcElement[i]->grantedPeCorr,
maxIter2ndGuess,
1, /* Process only 1 element */
i); /* Process exactly THIS element */

6
libAACenc/src/adj_thr.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -118,7 +118,8 @@ void FDKaacEnc_AdjThrInit(ADJ_THR_STATE *hAdjThr,
INT nChannelsEff,
INT sampleRate,
INT advancedBitsToPe,
FIXP_DBL vbrQualFactor);
FIXP_DBL vbrQualFactor,
const INT dZoneQuantEnable);
void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
@ -140,6 +141,7 @@ void FDKaacEnc_AdjustThresholds(ATS_ELEMENT* AdjThrStateElement[(8)],
QC_OUT* qcOut,
PSY_OUT_ELEMENT* psyOutElement[(8)],
INT CBRbitrateMode,
INT maxIter2ndGuess,
CHANNEL_MAPPING* cm);
void FDKaacEnc_AdjThrClose(ADJ_THR_STATE** hAdjThr);

3
libAACenc/src/adj_thr_data.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -145,6 +145,7 @@ typedef struct {
typedef struct {
BRES_PARAM bresParamLong, bresParamShort;
ATS_ELEMENT* adjThrStateElem[(8)];
INT maxIter2ndGuess;
} ADJ_THR_STATE;
#endif

8
libAACenc/src/bandwidth.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -127,7 +127,7 @@ static const BANDWIDTH_TAB bandWidthTable_LD_24000[] = {
{ 8000, 2000, 2000},
{12000, 2000, 2300},
{16000, 2200, 2500},
{24000, 5650, 6400},
{24000, 5650, 7200},
{32000, 11600, 12000},
{40000, 12000, 16000},
{48000, 16000, 16000},
@ -138,10 +138,10 @@ static const BANDWIDTH_TAB bandWidthTable_LD_24000[] = {
static const BANDWIDTH_TAB bandWidthTable_LD_32000[] = {
{ 8000, 2000, 2000},
{12000, 2000, 2000},
{24000, 4250, 5200},
{24000, 4250, 7200},
{32000, 8400, 9000},
{40000, 9400, 11300},
{48000, 11900, 13700},
{48000, 11900, 14700},
{64000, 14800, 16000},
{76000, 16000, 16000},
{360001, 16000, 16000}

2
libAACenc/src/intensity.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2014 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved.
1. INTRODUCTION

47
libAACenc/src/pnsparam.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -107,6 +107,7 @@ typedef struct {
typedef struct {
ULONG brFrom;
ULONG brTo;
UCHAR S16000;
UCHAR S22050;
UCHAR S24000;
UCHAR S32000;
@ -115,25 +116,26 @@ typedef struct {
} AUTO_PNS_TAB;
static const AUTO_PNS_TAB levelTable_mono[]= {
{0, 11999, 1, 1, 1, 1, 1,},
{12000, 19999, 1, 1, 1, 1, 1,},
{20000, 28999, 2, 1, 1, 1, 1,},
{29000, 40999, 4, 4, 4, 2, 2,},
{41000, 55999, 9, 9, 7, 7, 7,},
{56000, 79999, 0, 0, 0, 9, 9,},
{80000, 99999, 0, 0, 0, 0, 0,},
{100000,999999, 0, 0, 0, 0, 0,},
{0, 11999, 0, 1, 1, 1, 1, 1,},
{12000, 19999, 0, 1, 1, 1, 1, 1,},
{20000, 28999, 0, 2, 1, 1, 1, 1,},
{29000, 40999, 0, 4, 4, 4, 2, 2,},
{41000, 55999, 0, 9, 9, 7, 7, 7,},
{56000, 61999, 0, 0, 0, 0, 9, 9,},
{62000, 75999, 0, 0, 0, 0, 0, 0,},
{76000, 92999, 0, 0, 0, 0, 0, 0,},
{93000, 999999, 0, 0, 0, 0, 0, 0,},
};
static const AUTO_PNS_TAB levelTable_stereo[]= {
{0, 11999, 1, 1, 1, 1, 1,},
{12000, 19999, 3, 1, 1, 1, 1,},
{20000, 28999, 3, 3, 3, 2, 2,},
{29000, 40999, 7, 6, 6, 5, 5,},
{41000, 55999, 9, 9, 7, 7, 7,},
{56000, 79999, 0, 0, 0, 0, 0,},
{80000, 99999, 0, 0, 0, 0, 0,},
{100000,999999, 0, 0, 0, 0, 0,},
{0, 11999, 0, 1, 1, 1, 1, 1,},
{12000, 19999, 0, 3, 1, 1, 1, 1,},
{20000, 28999, 0, 3, 3, 3, 2, 2,},
{29000, 40999, 0, 7, 6, 6, 5, 5,},
{41000, 55999, 0, 9, 9, 7, 7, 7,},
{56000, 79999, 0, 0, 0, 0, 0, 0,},
{80000, 99999, 0, 0, 0, 0, 0, 0,},
{100000,999999, 0, 0, 0, 0, 0, 0,},
};
@ -160,11 +162,11 @@ static const PNS_INFO_TAB pnsInfoTab[] = {
};
static const AUTO_PNS_TAB levelTable_lowComplexity[]= {
{0, 27999, 0, 0, 0, 0, 0,},
{28000, 31999, 2, 2, 2, 2, 2,},
{32000, 47999, 3, 3, 3, 3, 3,},
{48000, 48000, 4, 4, 4, 4, 4,},
{48001, 999999, 0, 0, 0, 0, 0,},
{0, 27999, 0, 0, 0, 0, 0, 0,},
{28000, 31999, 0, 2, 2, 2, 2, 2,},
{32000, 47999, 0, 3, 3, 3, 3, 3,},
{48000, 48000, 0, 4, 4, 4, 4, 4,},
{48001, 999999, 0, 0, 0, 0, 0, 0,},
};
/* conversion of old LC tuning tables to new (LD enc) structure (only entries which are actually used were converted) */
@ -211,6 +213,7 @@ int FDKaacEnc_lookUpPnsUse (int bitRate, int sampleRate, int numChan, const int
}
switch (sampleRate) {
case 16000: hUsePns = levelTable[i].S16000; break;
case 22050: hUsePns = levelTable[i].S22050; break;
case 24000: hUsePns = levelTable[i].S24000; break;
case 32000: hUsePns = levelTable[i].S32000; break;

5
libAACenc/src/psy_configuration.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -634,13 +634,14 @@ AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate,
if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine)
break;
}
psyConf->sfbActive = sfb;
psyConf->sfbActive = FDKmax(sfb, 1);
for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){
if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE)
break;
}
psyConf->sfbActiveLFE = sfb;
psyConf->sfbActive = FDKmax(psyConf->sfbActive, psyConf->sfbActiveLFE);
/* calculate minSnr */
FDKaacEnc_initMinSnr(bitrate,

29
libAACenc/src/psy_main.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -342,6 +342,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy,
tnsChannels,
LONG_WINDOW,
hPsy->granuleLength,
isLowDelay(audioObjectType),
(syntaxFlags&AC_SBR_PRESENT)?1:0,
&(hPsy->psyConf[0].tnsConf),
&hPsy->psyConf[0],
@ -362,6 +363,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy,
tnsChannels,
SHORT_WINDOW,
hPsy->granuleLength,
isLowDelay(audioObjectType),
(syntaxFlags&AC_SBR_PRESENT)?1:0,
&hPsy->psyConf[1].tnsConf,
&hPsy->psyConf[1],
@ -447,7 +449,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels,
INT totalChannels
)
{
INT commonWindow = 1;
const INT commonWindow = 1;
INT maxSfbPerGroup[(2)];
INT mdctSpectrum_e;
INT ch; /* counts through channels */
@ -621,7 +623,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels,
FDKmemclear(&psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset],
(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 */
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]);
@ -763,7 +765,8 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels,
/* Advance psychoacoustics: Tonality and TNS */
if (psyStatic[0]->isLFE) {
tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive = 0;
tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] = 0;
tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] = 0;
}
else
{
@ -815,15 +818,19 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels,
&hThisPsyConf[1]->tnsConf);
}
FDK_ASSERT(commonWindow=1); /* all checks for TNS do only work for common windows (which is always set)*/
FDK_ASSERT(1==commonWindow); /* all checks for TNS do only work for common windows (which is always set)*/
for(w = 0; w < nWindows[0]; w++)
{
if (isShortWindow[0])
tnsActive[w] = tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive ||
((channels == 2) ? tnsData[1]->dataRaw.Short.subBlockInfo[w].tnsActive : 0);
tnsActive[w] = tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[HIFILT] ||
tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[LOFILT] ||
tnsData[channels-1]->dataRaw.Short.subBlockInfo[w].tnsActive[HIFILT] ||
tnsData[channels-1]->dataRaw.Short.subBlockInfo[w].tnsActive[LOFILT];
else
tnsActive[w] = tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive ||
((channels == 2) ? tnsData[1]->dataRaw.Long.subBlockInfo.tnsActive : 0);
tnsActive[w] = tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] ||
tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] ||
tnsData[channels-1]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] ||
tnsData[channels-1]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT];
}
for(ch = 0; ch < channels; ch++) {
@ -1150,8 +1157,8 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels,
psyData[ch]->sfbMaxScaleSpec.Long,
sfbTonality[ch],
psyOutChannel[ch]->tnsInfo.order[0][0],
tnsData[ch]->dataRaw.Long.subBlockInfo.predictionGain,
tnsData[ch]->dataRaw.Long.subBlockInfo.tnsActive,
tnsData[ch]->dataRaw.Long.subBlockInfo.predictionGain[HIFILT],
tnsData[ch]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT],
psyOutChannel[ch]->sfbEnergyLdData,
psyOutChannel[ch]->noiseNrg );
} /* !isLFE */

4
libAACenc/src/qc_data.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -269,6 +269,8 @@ typedef struct
BITCNTR_STATE *hBitCounter;
ADJ_THR_STATE *hAdjThr;
INT dZoneQuantEnable; /* enable dead zone quantizer */
} QC_STATE;
#endif /* _QC_DATA_H */

55
libAACenc/src/qc_main.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -107,14 +107,11 @@ typedef struct {
} TAB_VBR_QUAL_FACTOR;
static const TAB_VBR_QUAL_FACTOR tableVbrQualFactor[] = {
{QCDATA_BR_MODE_CBR, FL2FXCONST_DBL(0.00f)},
{QCDATA_BR_MODE_VBR_1, FL2FXCONST_DBL(0.160f)}, /* 32 kbps mono AAC-LC + SBR + PS */
{QCDATA_BR_MODE_VBR_2, FL2FXCONST_DBL(0.148f)}, /* 64 kbps stereo AAC-LC + SBR */
{QCDATA_BR_MODE_VBR_3, FL2FXCONST_DBL(0.135f)}, /* 80 - 96 kbps stereo AAC-LC */
{QCDATA_BR_MODE_VBR_4, FL2FXCONST_DBL(0.111f)}, /* 128 kbps stereo AAC-LC */
{QCDATA_BR_MODE_VBR_5, FL2FXCONST_DBL(0.070f)}, /* 192 kbps stereo AAC-LC */
{QCDATA_BR_MODE_SFR, FL2FXCONST_DBL(0.00f)},
{QCDATA_BR_MODE_FF, FL2FXCONST_DBL(0.00f)}
{QCDATA_BR_MODE_VBR_5, FL2FXCONST_DBL(0.070f)} /* 192 kbps stereo AAC-LC */
};
static INT isConstantBitrateMode(
@ -369,6 +366,7 @@ QCNew_bail:
AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
struct QC_INIT *init)
{
int i;
hQC->maxBitsPerFrame = init->maxBits;
hQC->minBitsPerFrame = init->minBits;
hQC->nElements = init->channelMapping->nElements;
@ -382,7 +380,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
if ( isConstantBitrateMode(hQC->bitrateMode) ) {
INT bitresPerChannel = (hQC->bitResTotMax / init->channelMapping->nChannelsEff);
/* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */
hQC->bitDistributionMode = (bitresPerChannel>100) ? 0 : (bitresPerChannel>0) ? 1 : 2;
hQC->bitDistributionMode = (bitresPerChannel>BITRES_MIN_LD) ? 0 : (bitresPerChannel>0) ? 1 : 2;
}
else {
hQC->bitDistributionMode = 0; /* full bitreservoir */
@ -399,25 +397,22 @@ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
(init->averageBits/init->nSubFrames) - hQC->globHdrBits,
hQC->maxBitsPerFrame/init->channelMapping->nChannelsEff);
switch(hQC->bitrateMode){
case QCDATA_BR_MODE_CBR:
case QCDATA_BR_MODE_VBR_1:
case QCDATA_BR_MODE_VBR_2:
case QCDATA_BR_MODE_VBR_3:
case QCDATA_BR_MODE_VBR_4:
case QCDATA_BR_MODE_VBR_5:
case QCDATA_BR_MODE_SFR:
case QCDATA_BR_MODE_FF:
if((int)hQC->bitrateMode < (int)(sizeof(tableVbrQualFactor)/sizeof(TAB_VBR_QUAL_FACTOR))){
hQC->vbrQualFactor = (FIXP_DBL)tableVbrQualFactor[hQC->bitrateMode].vbrQualFactor;
} else {
hQC->vbrQualFactor = FL2FXCONST_DBL(0.f); /* default setting */
}
break;
case QCDATA_BR_MODE_INVALID:
default:
hQC->vbrQualFactor = FL2FXCONST_DBL(0.f);
hQC->vbrQualFactor = FL2FXCONST_DBL(0.f);
for (i=0; i<(int)(sizeof(tableVbrQualFactor)/sizeof(TAB_VBR_QUAL_FACTOR)); i++) {
if (hQC->bitrateMode==tableVbrQualFactor[i].bitrateMode) {
hQC->vbrQualFactor = (FIXP_DBL)tableVbrQualFactor[i].vbrQualFactor;
break;
}
}
if (init->channelMapping->nChannelsEff == 1 &&
(init->bitrate / init->channelMapping->nChannelsEff) < 32000 &&
init->advancedBitsToPe != 0
)
{
hQC->dZoneQuantEnable = 1;
} else {
hQC->dZoneQuantEnable = 0;
}
FDKaacEnc_AdjThrInit(
@ -429,7 +424,8 @@ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
init->channelMapping->nChannelsEff,
init->sampleRate, /* output sample rate */
init->advancedBitsToPe, /* if set, calc bits2PE factor depending on samplerate */
hQC->vbrQualFactor
hQC->vbrQualFactor,
hQC->dZoneQuantEnable
);
return AAC_ENC_OK;
@ -892,6 +888,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC,
qcOut[c],
psyOut[c]->psyOutElement,
isConstantBitrateMode(hQC->bitrateMode),
hQC->hAdjThr->maxIter2ndGuess,
cm);
} /* -end- sub frame counter */
@ -919,6 +916,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC,
FDKaacEnc_EstimateScaleFactors(psyOut[c]->psyOutElement[i]->psyOutChannel,
qcElement[c][i]->qcOutChannel,
hQC->invQuant,
hQC->dZoneQuantEnable,
cm->elInfo[i].nChannelsInEl);
@ -1013,7 +1011,8 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC,
qcOutCh->mdctSpectrum,
qcOutCh->globalGain,
qcOutCh->scf,
qcOutCh->quantSpec) ;
qcOutCh->quantSpec,
hQC->dZoneQuantEnable);
/*-------------------------------------------- */
if (FDKaacEnc_calcMaxValueInSfb(psyOutCh->sfbCnt,
@ -1263,6 +1262,8 @@ AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING* cm,
case QCDATA_BR_MODE_VBR_4:
case QCDATA_BR_MODE_VBR_5:
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;
case QCDATA_BR_MODE_CBR:
@ -1272,6 +1273,8 @@ AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING* cm,
/* processing fill-bits */
INT deltaBitRes = qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits ;
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;
} /* switch (qcKernel->bitrateMode) */

24
libAACenc/src/quantize.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -104,13 +104,19 @@ amm-info@iis.fraunhofer.de
static void FDKaacEnc_quantizeLines(INT gain,
INT noOfLines,
FIXP_DBL *mdctSpectrum,
SHORT *quaSpectrum)
SHORT *quaSpectrum,
INT dZoneQuantEnable)
{
int line;
FIXP_DBL k = FL2FXCONST_DBL(-0.0946f + 0.5f)>>16;
FIXP_DBL k = FL2FXCONST_DBL(0.0f);
FIXP_QTD quantizer = FDKaacEnc_quantTableQ[(-gain)&3];
INT quantizershift = ((-gain)>>2)+1;
const INT kShift=16;
if (dZoneQuantEnable)
k = FL2FXCONST_DBL(0.23f)>>kShift;
else
k = FL2FXCONST_DBL(-0.0946f + 0.5f)>>kShift;
for (line = 0; line < noOfLines; line++)
{
@ -263,7 +269,8 @@ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt,
FIXP_DBL *mdctSpectrum,
INT globalGain,
INT *scalefactors,
SHORT *quantizedSpectrum)
SHORT *quantizedSpectrum,
INT dZoneQuantEnable)
{
INT sfbOffs,sfb;
@ -280,7 +287,8 @@ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt,
FDKaacEnc_quantizeLines(globalGain - scalefactor, /* QSS */
sfbOffset[sfbOffs+sfb+1] - sfbOffset[sfbOffs+sfb],
mdctSpectrum + sfbOffset[sfbOffs+sfb],
quantizedSpectrum + sfbOffset[sfbOffs+sfb]);
quantizedSpectrum + sfbOffset[sfbOffs+sfb],
dZoneQuantEnable);
}
}
@ -296,7 +304,8 @@ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt,
FIXP_DBL FDKaacEnc_calcSfbDist(FIXP_DBL *mdctSpectrum,
SHORT *quantSpectrum,
INT noOfLines,
INT gain
INT gain,
INT dZoneQuantEnable
)
{
INT i,scale;
@ -311,7 +320,8 @@ FIXP_DBL FDKaacEnc_calcSfbDist(FIXP_DBL *mdctSpectrum,
FDKaacEnc_quantizeLines(gain,
1,
&mdctSpectrum[i],
&quantSpectrum[i]);
&quantSpectrum[i],
dZoneQuantEnable);
if (fAbs(quantSpectrum[i])>MAX_QUANT) {
return FL2FXCONST_DBL(0.0f);

8
libAACenc/src/quantize.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -102,12 +102,14 @@ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt,
INT sfbPerGroup,
INT *sfbOffset, FIXP_DBL *mdctSpectrum,
INT globalGain, INT *scalefactors,
SHORT *quantizedSpectrum);
SHORT *quantizedSpectrum,
INT dZoneQuantEnable);
FIXP_DBL FDKaacEnc_calcSfbDist(FIXP_DBL *mdctSpectrum,
SHORT *quantSpectrum,
INT noOfLines,
INT gain);
INT gain,
INT dZoneQuantEnable);
void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum,
SHORT *quantSpectrum,

73
libAACenc/src/sf_estim.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -363,7 +363,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
INT scf,
INT minScf,
FIXP_DBL *distLdData,
INT *minScfCalculated
INT *minScfCalculated,
INT dZoneQuantEnable
)
{
FIXP_DBL sfbDistLdData;
@ -375,7 +376,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
sfbDistLdData = FDKaacEnc_calcSfbDist(spec,
quantSpec,
sfbWidth,
scf);
scf,
dZoneQuantEnable);
*minScfCalculated = scf;
/* nmr > 1.25 -> try to improve nmr */
if (sfbDistLdData > (threshLdData-distFactorLdData)) {
@ -390,7 +392,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
sfbDistLdData = FDKaacEnc_calcSfbDist(spec,
quantSpecTmp,
sfbWidth,
scf);
scf,
dZoneQuantEnable);
if (sfbDistLdData < sfbDistBestLdData) {
scfBest = scf;
@ -408,7 +411,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
sfbDistLdData = FDKaacEnc_calcSfbDist(spec,
quantSpecTmp,
sfbWidth,
scf);
scf,
dZoneQuantEnable);
if (sfbDistLdData < sfbDistBestLdData) {
scfBest = scf;
@ -429,7 +433,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
sfbDistLdData = FDKaacEnc_calcSfbDist(spec,
quantSpecTmp,
sfbWidth,
scf);
scf,
dZoneQuantEnable);
if (sfbDistLdData < sfbDistAllowedLdData) {
*minScfCalculated = scfBest+1;
@ -454,6 +459,7 @@ static void FDKaacEnc_assimilateSingleScf(PSY_OUT_CHANNEL *psyOutChan,
QC_OUT_CHANNEL *qcOutChannel,
SHORT *quantSpec,
SHORT *quantSpecTmp,
INT dZoneQuantEnable,
INT *scf,
INT *minScf,
FIXP_DBL *sfbDist,
@ -570,7 +576,8 @@ static void FDKaacEnc_assimilateSingleScf(PSY_OUT_CHANNEL *psyOutChan,
sfbDistNew = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs,
quantSpecTmp+sfbOffs,
sfbWidth,
scfAct);
scfAct,
dZoneQuantEnable);
if (sfbDistNew < sfbDist[sfbAct]) {
/* success, replace scf by new one */
@ -629,6 +636,7 @@ static void FDKaacEnc_assimilateMultipleScf(PSY_OUT_CHANNEL *psyOutChan,
QC_OUT_CHANNEL *qcOutChannel,
SHORT *quantSpec,
SHORT *quantSpecTmp,
INT dZoneQuantEnable,
INT *scf,
INT *minScf,
FIXP_DBL *sfbDist,
@ -724,7 +732,8 @@ static void FDKaacEnc_assimilateMultipleScf(PSY_OUT_CHANNEL *psyOutChan,
sfbDistNew[sfb] = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs,
quantSpecTmp+sfbOffs,
sfbWidth,
scfAct);
scfAct,
dZoneQuantEnable);
if (sfbDistNew[sfb] >qcOutChannel->sfbThresholdLdData[sfb]) {
/* no improvement, skip further dist. calculations */
@ -768,6 +777,7 @@ static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(PSY_OUT_CHANNEL *psyOutCh
QC_OUT_CHANNEL *qcOutChannel,
SHORT *quantSpec,
SHORT *quantSpecTmp,
INT dZoneQuantEnable,
INT *scf,
INT *minScf,
FIXP_DBL *sfbDist,
@ -883,7 +893,8 @@ static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(PSY_OUT_CHANNEL *psyOutCh
sfbDistNew[sfb] = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs[sfb],
quantSpecTmp+sfbOffs[sfb],
sfbOffs[sfb+1]-sfbOffs[sfb],
scfNew);
scfNew,
dZoneQuantEnable);
if (sfbDistNew[sfb] > sfbDistMax[sfb]) {
/* no improvement, skip further dist. calculations */
@ -963,7 +974,8 @@ static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(PSY_OUT_CHANNEL *psyOutCh
sfbDistNew[sfb] = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs[sfb],
quantSpecTmp+sfbOffs[sfb],
sfbOffs[sfb+1]-sfbOffs[sfb],
scfNew);
scfNew,
dZoneQuantEnable);
if (sfbDistNew[sfb] > qcOutChannel->sfbThresholdLdData[sfb]) {
/* no improvement, skip further dist. calculations */
@ -1058,7 +1070,8 @@ FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(QC_OUT_CHANNEL *qcOutChannel,
INT *RESTRICT globalGain,
FIXP_DBL *RESTRICT sfbFormFactorLdData
,const INT invQuant,
SHORT *RESTRICT quantSpec
SHORT *RESTRICT quantSpec,
const INT dZoneQuantEnable
)
{
INT i, j, sfb, sfbOffs;
@ -1160,7 +1173,8 @@ FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(QC_OUT_CHANNEL *qcOutChannel,
quantSpecTmp+psyOutChannel->sfbOffsets[sfbOffs+sfb],
psyOutChannel->sfbOffsets[sfbOffs+sfb+1]-psyOutChannel->sfbOffsets[sfbOffs+sfb],
threshLdData, scfInt, minSfMaxQuant[sfbOffs+sfb],
&sfbDistLdData[sfbOffs+sfb], &minScfCalculated[sfbOffs+sfb]
&sfbDistLdData[sfbOffs+sfb], &minScfCalculated[sfbOffs+sfb],
dZoneQuantEnable
);
}
scf[sfbOffs+sfb] = scfInt;
@ -1187,20 +1201,32 @@ FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(QC_OUT_CHANNEL *qcOutChannel,
sfbNRelevantLines);
FDKaacEnc_assimilateSingleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, scf,
FDKaacEnc_assimilateSingleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp,
dZoneQuantEnable,
scf,
minSfMaxQuant, sfbDistLdData, sfbConstPePart,
sfbFormFactorLdData, sfbNRelevantLines, minScfCalculated, 1);
if(invQuant > 1) {
FDKaacEnc_assimilateMultipleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp,
dZoneQuantEnable,
scf,
minSfMaxQuant, sfbDistLdData, sfbConstPePart,
sfbFormFactorLdData, sfbNRelevantLines);
FDKaacEnc_assimilateMultipleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, scf,
minSfMaxQuant, sfbDistLdData, sfbConstPePart,
sfbFormFactorLdData, sfbNRelevantLines);
FDKaacEnc_assimilateMultipleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp,
dZoneQuantEnable,
scf,
minSfMaxQuant, sfbDistLdData, sfbConstPePart,
sfbFormFactorLdData, sfbNRelevantLines);
FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, scf,
minSfMaxQuant, sfbDistLdData, sfbConstPePart,
sfbFormFactorLdData, sfbNRelevantLines);
FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp,
dZoneQuantEnable,
scf,
minSfMaxQuant, sfbDistLdData, sfbConstPePart,
sfbFormFactorLdData, sfbNRelevantLines);
}
}
@ -1223,7 +1249,8 @@ FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(QC_OUT_CHANNEL *qcOutChannel,
FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+psyOutChannel->sfbOffsets[sfbOffs+sfb],
quantSpec+psyOutChannel->sfbOffsets[sfbOffs+sfb],
psyOutChannel->sfbOffsets[sfbOffs+sfb+1]-psyOutChannel->sfbOffsets[sfbOffs+sfb],
scf[sfbOffs+sfb]
scf[sfbOffs+sfb],
dZoneQuantEnable
);
}
}
@ -1281,6 +1308,7 @@ void
FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[],
QC_OUT_CHANNEL* qcOutChannel[],
const int invQuant,
const INT dZoneQuantEnable,
const int nChannels)
{
int ch;
@ -1293,7 +1321,8 @@ FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[],
&qcOutChannel[ch]->globalGain,
qcOutChannel[ch]->sfbFormFactorLdData
,invQuant,
qcOutChannel[ch]->quantSpec
qcOutChannel[ch]->quantSpec,
dZoneQuantEnable
);
}

3
libAACenc/src/sf_estim.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -110,6 +110,7 @@ void
FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[],
QC_OUT_CHANNEL* qcOutChannel[],
const int invQuant,
const INT dZoneQuantEnable,
const int nChannels);

3
libAACenc/src/tns_func.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -100,6 +100,7 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitrate,
INT channels,
INT blocktype,
INT granuleLength,
INT isLowDelay,
INT ldSbrPresent,
TNS_CONFIG *tnsConfig,
PSY_CONFIGURATION *psyConfig,

31
libFDK/include/fixpoint_math.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2014 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved.
1. INTRODUCTION
@ -94,6 +94,35 @@ amm-info@iis.fraunhofer.de
#include "common_fix.h"
#if !defined(FUNCTION_fIsLessThan)
/**
* \brief Compares two fixpoint values incl. scaling.
* \param a_m mantissa of the first input value.
* \param a_e exponent of the first input value.
* \param b_m mantissa of the second input value.
* \param b_e exponent of the second input value.
* \return non-zero if (a_m*2^a_e) < (b_m*2^b_e), 0 otherwise
*/
FDK_INLINE INT fIsLessThan(FIXP_DBL a_m, INT a_e, FIXP_DBL b_m, INT b_e)
{
if (a_e > b_e) {
return (b_m >> fMin(a_e-b_e, DFRACT_BITS-1) > a_m);
} else {
return (a_m >> fMin(b_e-a_e, DFRACT_BITS-1) < b_m);
}
}
FDK_INLINE INT fIsLessThan(FIXP_SGL a_m, INT a_e, FIXP_SGL b_m, INT b_e)
{
if (a_e > b_e) {
return (b_m >> fMin(a_e-b_e, FRACT_BITS-1) > a_m);
} else {
return (a_m >> fMin(b_e-a_e, FRACT_BITS-1) < b_m);
}
}
#endif
#define LD_DATA_SCALING (64.0f)
#define LD_DATA_SHIFT 6 /* pow(2, LD_DATA_SHIFT) = LD_DATA_SCALING */

4
libFDK/src/FDK_core.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -93,7 +93,7 @@ amm-info@iis.fraunhofer.de
/* FDK tools library info */
#define FDK_TOOLS_LIB_VL0 2
#define FDK_TOOLS_LIB_VL1 3
#define FDK_TOOLS_LIB_VL2 2
#define FDK_TOOLS_LIB_VL2 6
#define FDK_TOOLS_LIB_TITLE "FDK Tools"
#ifdef __ANDROID__
#define FDK_TOOLS_LIB_BUILD_DATE ""

212
libFDK/src/FDK_tools_rom.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -2039,19 +2039,6 @@ static const element_list_t node_aac_cpe = {
{ &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}
* epConfig = 0,1
@ -2249,7 +2236,7 @@ static const rbd_id_t el_aac_cpe1_epc1[] = {
ics_info,
ms,
ltp_data_present,
ltp_data,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
@ -2260,7 +2247,7 @@ static const rbd_id_t el_aac_cpe1_epc1[] = {
next_channel,
ltp_data_present,
ltp_data,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
@ -2303,7 +2290,178 @@ static const element_list_t node_aac_cpe_epc1 = {
{ &node_aac_cpe0_epc1, &node_aac_cpe1_epc1 }
};
/*
* AOT = 20
* epConfig = 0
*/
static const rbd_id_t el_scal_sce_epc0[] = {
ics_info, /* ESC 1 */
tns_data_present,
ltp_data_present,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
esc2_rvlc, /* ESC 2 */
tns_data, /* ESC 3 */
spectral_data, /* ESC 4 */
end_of_sequence
};
static const struct element_list node_scal_sce_epc0 = {
el_scal_sce_epc0,
{ NULL, NULL }
};
static const rbd_id_t el_scal_cpe_epc0[] = {
ics_info, /* ESC 0 */
ms,
tns_data_present, /* ESC 1 (ch 0) */
ltp_data_present,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
esc2_rvlc, /* ESC 2 (ch 0) */
tns_data, /* ESC 3 (ch 0) */
spectral_data, /* ESC 4 (ch 0) */
next_channel,
tns_data_present, /* ESC 1 (ch 1) */
ltp_data_present,
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
esc2_rvlc, /* ESC 2 (ch 1) */
tns_data, /* ESC 3 (ch 1) */
spectral_data, /* ESC 4 (ch 1) */
end_of_sequence
};
static const struct element_list node_scal_cpe_epc0 = {
el_scal_cpe_epc0,
{ NULL, NULL }
};
/*
* AOT = 20
* epConfig = 1
*/
static const rbd_id_t el_scal_sce_epc1[] = {
ics_info,
tns_data_present,
ltp_data_present,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
tns_data,
spectral_data,
end_of_sequence
};
static const struct element_list node_scal_sce_epc1 = {
el_scal_sce_epc1,
{ NULL, NULL }
};
static const rbd_id_t el_scal_cpe_epc1[] = {
ics_info,
ms,
tns_data_present,
ltp_data_present,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
next_channel,
tns_data_present,
ltp_data_present,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
next_channel,
tns_data,
next_channel,
tns_data,
next_channel,
spectral_data,
next_channel,
spectral_data,
end_of_sequence
};
static const struct element_list node_scal_cpe_epc1 = {
el_scal_cpe_epc1,
{ NULL, NULL }
};
/*
* Pseudo AOT for DRM/DRM+ (similar to AOT 20)
* Derived from epConfig = 1
*/
static const rbd_id_t el_drm_sce[] = {
drmcrc_start_reg,
ics_info,
tns_data_present,
ltp_data_present,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
tns_data,
drmcrc_end_reg,
spectral_data,
end_of_sequence
};
static const struct element_list node_drm_sce = {
el_drm_sce,
{ NULL, NULL }
};
static const rbd_id_t el_drm_cpe[] = {
drmcrc_start_reg,
ics_info,
ms,
tns_data_present,
ltp_data_present,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
next_channel,
tns_data_present,
ltp_data_present,
/* ltp_data, */
global_gain,
section_data,
scale_factor_data,
esc1_hcr,
next_channel,
tns_data,
next_channel,
tns_data,
drmcrc_end_reg,
next_channel,
spectral_data,
next_channel,
spectral_data,
end_of_sequence
};
static const struct element_list node_drm_cpe = {
el_drm_cpe,
{ NULL, NULL }
};
/*
* AOT = 39
@ -2418,6 +2576,19 @@ const element_list_t * getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epCo
return &node_aac_cpe_epc1;
}
break;
case AOT_ER_AAC_SCAL:
if (nChannels == 1) {
if (epConfig <= 0)
return &node_scal_sce_epc0;
else
return &node_scal_sce_epc1;
} else {
if (epConfig <= 0)
return &node_scal_cpe_epc0;
else
return &node_scal_cpe_epc1;
}
break;
case AOT_ER_AAC_ELD:
if (nChannels == 1) {
if (epConfig <= 0)
@ -2430,11 +2601,14 @@ const element_list_t * getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epCo
else
return &node_eld_cpe_epc1;
}
case AOT_MPEGS_RESIDUALS:
case AOT_DRM_AAC:
case AOT_DRM_SBR:
case AOT_DRM_MPEG_PS:
FDK_ASSERT(epConfig == 1);
if (nChannels == 1) {
return &node_mpegsres_sce;
return &node_drm_sce;
} else {
return &node_mpegsres_cpe;
return &node_drm_cpe;
}
break;
default:

78
libFDK/src/arm/qmf_arm.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -482,42 +482,42 @@ static void qmfSynPrototypeFirSlot1_filter(FIXP_QMF *RESTRICT realSlot,
B = p_flt[4]; /* Bottom=[8] Top=[9] */
A = p_fltm[3]; /* Bottom=[316] Top=[317] */
sta0 = sta[0]; /* save state[0] */
*sta++ = SMLAWT( sta[1], imag, B ); /* index=9...........319 */
*sta++ = SMLAWB( sta[1], real, A ); /* index=316...........6 */
*sta++ = SMLAWB( sta[1], imag, B ); /* index=8,18, ...318 */
sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=9...........319 */
sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=316...........6 */
sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=8,18, ...318 */
B = p_flt[3]; /* Bottom=[6] Top=[7] */
*sta++ = SMLAWT( sta[1], real, A ); /* index=317...........7 */
sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=317...........7 */
A = p_fltm[4]; /* Bottom=[318] Top=[319] */
*sta++ = SMLAWT( sta[1], imag, B ); /* index=7...........317 */
*sta++ = SMLAWB( sta[1], real, A ); /* index=318...........8 */
*sta++ = SMLAWB( sta[1], imag, B ); /* index=6...........316 */
sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=7...........317 */
sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=318...........8 */
sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=6...........316 */
B = p_flt[2]; /* Bottom=[X] Top=[5] */
*sta++ = SMLAWT( sta[1], real, A ); /* index=9...........319 */
sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=9...........319 */
A = p_fltm[2]; /* Bottom=[X] Top=[315] */
*sta++ = SMULWT( imag, B ); /* index=5,15, ... 315 */
sta[0] = SMULWT( imag, B ); sta++; /* index=5,15, ... 315 */
result = SMLAWT( sta0, real, A ); /* index=315...........5 */
*pMyTimeOut++ = result;
pMyTimeOut[0] = result; pMyTimeOut++;
real = *--realSlot;
imag = *--imagSlot;
A = p_fltm[0]; /* Bottom=[310] Top=[311] */
B = p_flt[7]; /* Bottom=[14] Top=[15] */
result = SMLAWB( sta[0], real, A ); /* index=310...........0 */
*sta++ = SMLAWB( sta[1], imag, B ); /* index=14..........324 */
*pMyTimeOut++ = result;
sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=14..........324 */
pMyTimeOut[0] = result; pMyTimeOut++;
B = p_flt[6]; /* Bottom=[12] Top=[13] */
*sta++ = SMLAWT( sta[1], real, A ); /* index=311...........1 */
sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=311...........1 */
A = p_fltm[1]; /* Bottom=[312] Top=[313] */
*sta++ = SMLAWT( sta[1], imag, B ); /* index=13..........323 */
*sta++ = SMLAWB( sta[1], real, A ); /* index=312...........2 */
*sta++ = SMLAWB( sta[1], imag, B ); /* index=12..........322 */
*sta++ = SMLAWT( sta[1], real, A ); /* index=313...........3 */
sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=13..........323 */
sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=312...........2 */
sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=12..........322 */
sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=313...........3 */
A = p_fltm[2]; /* Bottom=[314] Top=[315] */
B = p_flt[5]; /* Bottom=[10] Top=[11] */
*sta++ = SMLAWT( sta[1], imag, B ); /* index=11..........321 */
*sta++ = SMLAWB( sta[1], real, A ); /* index=314...........4 */
*sta++ = SMULWB( imag, B ); /* index=10..........320 */
sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=11..........321 */
sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=314...........4 */
sta[0] = SMULWB( imag, B ); sta++; /* index=10..........320 */
p_flt += 5;
@ -566,21 +566,21 @@ INT qmfSynPrototypeFirSlot2(
A = p_fltm[0]; /* Bottom=[310] Top=[311] */
B = p_flt[7]; /* Bottom=[14] Top=[15] */
result = SMLAWB( sta[0], real, A ); /* index=310...........0 */
*sta++ = SMLAWB( sta[1], imag, B ); /* index=14..........324 */
sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=14..........324 */
B = p_flt[6]; /* Bottom=[12] Top=[13] */
*sta++ = SMLAWT( sta[1], real, A ); /* index=311...........1 */
sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=311...........1 */
A = p_fltm[1]; /* Bottom=[312] Top=[313] */
*sta++ = SMLAWT( sta[1], imag, B ); /* index=13..........323 */
*sta++ = SMLAWB( sta[1], real, A ); /* index=312...........2 */
*sta++ = SMLAWB( sta[1], imag, B ); /* index=12..........322 */
*sta++ = SMLAWT( sta[1], real, A ); /* index=313...........3 */
sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=13..........323 */
sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=312...........2 */
sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=12..........322 */
sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=313...........3 */
A = p_fltm[2]; /* Bottom=[314] Top=[315] */
B = p_flt[5]; /* Bottom=[10] Top=[11] */
*sta++ = SMLAWT( sta[1], imag, B ); /* index=11..........321 */
*sta++ = SMLAWB( sta[1], real, A ); /* index=314...........4 */
*sta++ = SMULWB( imag, B ); /* index=10..........320 */
sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=11..........321 */
sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=314...........4 */
sta[0] = SMULWB( imag, B ); sta++; /* index=10..........320 */
*pMyTimeOut++ = result;
pMyTimeOut[0] = result; pMyTimeOut++;
p_fltm -= 5;
p_flt += 5;
@ -610,8 +610,8 @@ INT qmfSynPrototypeFirSlot2(
{
FIXP_DBL result1, result2;
result1 = *pMyTimeOut++;
result2 = *pMyTimeOut++;
result1 = pMyTimeOut[0]; pMyTimeOut++;
result2 = pMyTimeOut[0]; pMyTimeOut++;
result1 = fMult(result1,gain);
timeOut -= stride;
@ -635,8 +635,8 @@ INT qmfSynPrototypeFirSlot2(
timeOut[0] = result2 << scale;
#endif
result1 = *pMyTimeOut++;
result2 = *pMyTimeOut++;
result1 = pMyTimeOut[0]; pMyTimeOut++;
result2 = pMyTimeOut[0]; pMyTimeOut++;
result1 = fMult(result1,gain);
timeOut -= stride;
@ -666,8 +666,8 @@ INT qmfSynPrototypeFirSlot2(
for (no_channels>>=2; no_channels--;)
{
FIXP_DBL result1, result2;
result1 = *pMyTimeOut++;
result2 = *pMyTimeOut++;
result1 = pMyTimeOut[0]; pMyTimeOut++;
result2 = pMyTimeOut[0]; pMyTimeOut++;
timeOut -= stride;
if (result1 < 0) result1 += add_neg;
if (result1 < max_neg) result1 = max_neg;
@ -688,8 +688,8 @@ INT qmfSynPrototypeFirSlot2(
timeOut[0] = result2 << scale;
#endif
result1 = *pMyTimeOut++;
result2 = *pMyTimeOut++;
result1 = pMyTimeOut[0]; pMyTimeOut++;
result2 = pMyTimeOut[0]; pMyTimeOut++;
timeOut -= stride;
if (result1 < 0) result1 += add_neg;
if (result1 < max_neg) result1 = max_neg;

135
libMpegTPDec/src/tpdec_asc.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -1126,6 +1126,8 @@ TRANSPORTDEC_ERROR EldSpecificConfig_Parse(
if ( 0 != ld_sbr_header(asc, hBs, cb) ) {
return TRANSPORTDEC_PARSE_ERROR;
}
} else {
return TRANSPORTDEC_UNSUPPORTED_FORMAT;
}
}
esc->m_useLdQmfTimeAlign = 0;
@ -1146,7 +1148,7 @@ TRANSPORTDEC_ERROR EldSpecificConfig_Parse(
switch (eldExtType) {
default:
for(cnt=0; cnt<len; cnt++) {
for(cnt=0; cnt<eldExtLen; cnt++) {
FDKreadBits(hBs, 8 );
}
break;
@ -1372,4 +1374,133 @@ TRANSPORTDEC_ERROR AudioSpecificConfig_Parse(
return (ErrorStatus);
}
TRANSPORTDEC_ERROR DrmRawSdcAudioConfig_Parse(
CSAudioSpecificConfig *self,
HANDLE_FDK_BITSTREAM bs
)
{
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
AudioSpecificConfig_Init(self);
if ((INT)FDKgetValidBits(bs) < 20) {
ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
goto bail;
}
else {
/* DRM - Audio information data entity - type 9
- Short Id 2 bits
- Stream Id 2 bits
- audio coding 2 bits
- SBR flag 1 bit
- audio mode 2 bits
- audio sampling rate 3 bits
- text flag 1 bit
- enhancement flag 1 bit
- coder field 5 bits
- rfa 1 bit */
int audioCoding, audioMode, cSamplingFreq, coderField, sfIdx, sbrFlag;
/* Read the SDC field */
FDKreadBits(bs,4); /* Short and Stream Id */
audioCoding = FDKreadBits(bs, 2);
sbrFlag = FDKreadBits(bs, 1);
audioMode = FDKreadBits(bs, 2);
cSamplingFreq = FDKreadBits(bs, 3); /* audio sampling rate */
FDKreadBits(bs, 2); /* Text and enhancement flag */
coderField = FDKreadBits(bs, 5);
FDKreadBits(bs, 1); /* rfa */
/* Evaluate configuration and fill the ASC */
switch (cSamplingFreq) {
case 0: /* 8 kHz */
sfIdx = 11;
break;
case 1: /* 12 kHz */
sfIdx = 9;
break;
case 2: /* 16 kHz */
sfIdx = 8;
break;
case 3: /* 24 kHz */
sfIdx = 6;
break;
case 5: /* 48 kHz */
sfIdx = 3;
break;
case 4: /* reserved */
case 6: /* reserved */
case 7: /* reserved */
default:
ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
goto bail;
}
self->m_samplingFrequencyIndex = sfIdx;
self->m_samplingFrequency = SamplingRateTable[sfIdx];
if ( sbrFlag ) {
UINT i;
int tmp = -1;
self->m_sbrPresentFlag = 1;
self->m_extensionAudioObjectType = AOT_SBR;
self->m_extensionSamplingFrequency = self->m_samplingFrequency << 1;
for (i=0; i<(sizeof(SamplingRateTable)/sizeof(SamplingRateTable[0])); i++){
if (SamplingRateTable[i] == self->m_extensionSamplingFrequency){
tmp = i;
break;
}
}
self->m_extensionSamplingFrequencyIndex = tmp;
}
switch (audioCoding) {
case 0: /* AAC */
self->m_aot = AOT_DRM_AAC ; /* Set pseudo AOT for Drm AAC */
switch (audioMode) {
case 1: /* parametric stereo */
self->m_psPresentFlag = 1;
case 0: /* mono */
self->m_channelConfiguration = 1;
break;
case 2: /* stereo */
self->m_channelConfiguration = 2;
break;
default:
ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
goto bail;
}
self->m_vcb11Flag = 1;
self->m_hcrFlag = 1;
self->m_samplesPerFrame = 960;
self->m_epConfig = 1;
break;
case 1: /* CELP */
self->m_aot = AOT_ER_CELP;
self->m_channelConfiguration = 1;
break;
case 2: /* HVXC */
self->m_aot = AOT_ER_HVXC;
self->m_channelConfiguration = 1;
break;
case 3: /* reserved */
default:
ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
self->m_aot = AOT_NONE;
break;
}
if (self->m_psPresentFlag && !self->m_sbrPresentFlag) {
ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
goto bail;
}
}
bail:
return (ErrorStatus);
}

66
libAACenc/src/aacenc_hcr.h → libMpegTPDec/src/tpdec_drm.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -81,16 +81,66 @@ www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------------------------------------- */
/*************************** MPEG AAC Audio Encoder *************************
/***************************** MPEG-4 AAC Decoder **************************
Initial author: R. Boehm
contents/description: huffman codeword reordering
based on source from aacErrRobTrans
Author(s): Christian Griebel
Description: DRM transport stuff
******************************************************************************/
#ifndef _AACENC_HCR
#define _AACENC_HCR_H
#include "tpdec_drm.h"
#include "FDK_bitstream.h"
void drmRead_CrcInit(HANDLE_DRM pDrm) /*!< pointer to drm crc info stucture */
{
FDK_ASSERT(pDrm != NULL);
FDKcrcInit(&pDrm->crcInfo, 0x001d, 0xFFFF, 8);
}
int drmRead_CrcStartReg(
HANDLE_DRM pDrm, /*!< pointer to drm stucture */
HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */
int mBits /*!< number of bits in crc region */
)
{
FDK_ASSERT(pDrm != NULL);
FDKcrcReset(&pDrm->crcInfo);
pDrm->crcReadValue = FDKreadBits(hBs, 8);
return ( FDKcrcStartReg(&pDrm->crcInfo, hBs, mBits) );
}
void drmRead_CrcEndReg(
HANDLE_DRM pDrm, /*!< pointer to drm crc info stucture */
HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */
int reg /*!< crc region */
)
{
FDK_ASSERT(pDrm != NULL);
FDKcrcEndReg(&pDrm->crcInfo, hBs, reg);
}
TRANSPORTDEC_ERROR drmRead_CrcCheck( HANDLE_DRM pDrm )
{
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
USHORT crc;
crc = FDKcrcGetCRC(&pDrm->crcInfo) ^ 0xFF;
if (crc != pDrm->crcReadValue)
{
return (TRANSPORTDEC_CRC_ERROR);
}
return (ErrorStatus);
}
#endif /* ifndef _AACENC_HCR */

113
libAACenc/src/aacenc_hcr.cpp → libMpegTPDec/src/tpdec_drm.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -81,13 +81,114 @@ www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------------------------------------- */
/*************************** MPEG AAC Audio Encoder *************************
/***************************** MPEG-4 AAC Decoder **************************
Initial author: R. Boehm
contents/description: huffman codeword reordering
based on source from aacErrRobTrans
Author(s): Josef Hoepfl
Description: DRM interface
******************************************************************************/
#include "aacenc_hcr.h"
#ifndef TPDEC_DRM_H
#define TPDEC_DRM_H
#include "tpdec_lib.h"
#include "FDK_crc.h"
typedef struct {
FDK_CRCINFO crcInfo; /* CRC state info */
USHORT crcReadValue; /* CRC value read from bitstream data */
} STRUCT_DRM;
typedef STRUCT_DRM *HANDLE_DRM;
/*!
\brief Initialize DRM CRC
The function initialzes the crc buffer and the crc lookup table.
\return none
*/
void drmRead_CrcInit( HANDLE_DRM pDrm );
/**
* \brief Starts CRC region with a maximum number of bits
* If mBits is positive zero padding will be used for CRC calculation, if there
* are less than mBits bits available.
* If mBits is negative no zero padding is done.
* If mBits is zero the memory for the buffer is allocated dynamically, the
* number of bits is not limited.
*
* \param pDrm DRM data handle
* \param hBs bitstream handle, on which the CRC region referes to
* \param mBits max number of bits in crc region to be considered
*
* \return ID for the created region, -1 in case of an error
*/
int drmRead_CrcStartReg(
HANDLE_DRM pDrm,
HANDLE_FDK_BITSTREAM hBs,
int mBits
);
/**
* \brief Ends CRC region identified by reg
*
* \param pDrm DRM data handle
* \param hBs bitstream handle, on which the CRC region referes to
* \param reg CRC regions ID returned by drmRead_CrcStartReg()
*
* \return none
*/
void drmRead_CrcEndReg(
HANDLE_DRM pDrm,
HANDLE_FDK_BITSTREAM hBs,
int reg
);
/**
* \brief Check CRC
*
* Checks if the currently calculated CRC matches the CRC field read from the bitstream
* Deletes all CRC regions.
*
* \param pDrm DRM data handle
*
* \return Returns 0 if they are identical otherwise 1
*/
TRANSPORTDEC_ERROR drmRead_CrcCheck( HANDLE_DRM pDrm );
/**
* \brief Check if we have a valid DRM frame at the current bitbuffer position
*
* This function assumes enough bits in buffer for the current frame.
* It reads out the header bits to prepare the bitbuffer for the decode loop.
* In case the header bits show an invalid bitstream/frame, the whole frame is skipped.
*
* \param pDrm DRM data handle which is filled with parsed DRM header data
* \param bs handle of bitstream from whom the DRM header is read
*
* \return error status
*/
TRANSPORTDEC_ERROR drmRead_DecodeHeader(
HANDLE_DRM pDrm,
HANDLE_FDK_BITSTREAM bs
);
/**
* \brief Parse a Drm specific SDC audio config from a given bitstream handle.
*
* \param pAsc A pointer to an allocated CSAudioSpecificConfig struct.
* \param hBs Bitstream handle.
*
* \return Total element count including all SCE, CPE and LFE.
*/
TRANSPORTDEC_ERROR DrmRawSdcAudioConfig_Parse( CSAudioSpecificConfig *pAsc,
HANDLE_FDK_BITSTREAM hBs );
#endif /* TPDEC_DRM_H */

30
libMpegTPDec/src/tpdec_lib.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -102,6 +102,7 @@ amm-info@iis.fraunhofer.de
#include "tpdec_latm.h"
#include "tpdec_drm.h"
#define MODULE_NAME "transportDec"
@ -113,6 +114,7 @@ typedef union {
CLatmDemux latm;
STRUCT_DRM drm;
} transportdec_parser_t;
@ -182,6 +184,9 @@ HANDLE_TRANSPORTDEC transportDec_Open( const TRANSPORT_TYPE transportFmt, const
hInput->numberOfRawDataBlocks = 0;
break;
case TT_DRM:
drmRead_CrcInit(&hInput->parser.drm);
break;
case TT_MP4_LATM_MCP0:
case TT_MP4_LATM_MCP1:
@ -253,6 +258,18 @@ TRANSPORTDEC_ERROR transportDec_OutOfBandConfig(HANDLE_TRANSPORTDEC hTp, UCHAR *
}
}
break;
case TT_DRM:
fConfigFound = 1;
err = DrmRawSdcAudioConfig_Parse(&hTp->asc[layer], hBs);
if (err == TRANSPORTDEC_OK) {
int errC;
errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[layer]);
if (errC != 0) {
err = TRANSPORTDEC_PARSE_ERROR;
}
}
break;
}
if (err == TRANSPORTDEC_OK && fConfigFound) {
@ -1083,6 +1100,7 @@ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, c
break;
case TT_MP4_RAW:
case TT_DRM:
/* One Access Unit was filled into buffer.
So get the length out of the buffer. */
hTp->auLength[layer] = FDKgetValidBits(hBs);
@ -1100,7 +1118,6 @@ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, c
}
break;
case TT_RSVD50:
case TT_MP4_ADTS:
case TT_MP4_LOAS:
err = transportDec_readStream(hTp, layer);
@ -1284,6 +1301,7 @@ TRANSPORTDEC_ERROR transportDec_GetLibInfo( LIB_INFO *info )
| CAPF_LATM
| CAPF_LOAS
| CAPF_RAWPACKETS
| CAPF_DRM
;
return TRANSPORTDEC_OK; /* FDKERR_NOERROR; */
@ -1295,6 +1313,8 @@ int transportDec_CrcStartReg(HANDLE_TRANSPORTDEC pTp, INT mBits)
switch (pTp->transportFmt) {
case TT_MP4_ADTS:
return adtsRead_CrcStartReg(&pTp->parser.adts, &pTp->bitStream[0], mBits);
case TT_DRM:
return drmRead_CrcStartReg(&pTp->parser.drm, &pTp->bitStream[0], mBits);
default:
return 0;
}
@ -1306,6 +1326,9 @@ void transportDec_CrcEndReg(HANDLE_TRANSPORTDEC pTp, INT reg)
case TT_MP4_ADTS:
adtsRead_CrcEndReg(&pTp->parser.adts, &pTp->bitStream[0], reg);
break;
case TT_DRM:
drmRead_CrcEndReg(&pTp->parser.drm, &pTp->bitStream[0], reg);
break;
default:
break;
}
@ -1322,6 +1345,9 @@ TRANSPORTDEC_ERROR transportDec_CrcCheck(HANDLE_TRANSPORTDEC pTp)
transportDec_AdjustEndOfAccessUnit(pTp);
}
return adtsRead_CrcCheck(&pTp->parser.adts);
case TT_DRM:
return drmRead_CrcCheck(&pTp->parser.drm);
break;
default:
return TRANSPORTDEC_OK;
}

2
libMpegTPDec/src/version
View File

@ -2,7 +2,7 @@
/* library info */
#define TP_LIB_VL0 2
#define TP_LIB_VL1 3
#define TP_LIB_VL2 4
#define TP_LIB_VL2 7
#define TP_LIB_TITLE "MPEG Transport"
#ifdef __ANDROID__
#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
© 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.
1. INTRODUCTION
@ -296,6 +296,7 @@ CreateStreamMuxConfig(
USHORT coreFrameOffset=0;
hAss->taraBufferFullness = 0xFF;
hAss->audioMuxVersionA = 0; /* for future extensions */
hAss->streamMuxConfigBits = 0;
@ -339,13 +340,7 @@ CreateStreamMuxConfig(
hAss->streamMuxConfigBits+=1;
}
if( (useSameConfig == 0) || (transLayer==0) ) {
UINT bits;
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 );
const UINT alignAnchor = FDKgetValidBits(hBs);
transportEnc_writeASC(
hBs,
@ -353,19 +348,24 @@ CreateStreamMuxConfig(
cb
);
bits = FDKgetValidBits( hBs ) - bits;
if ( hAss->audioMuxVersion == 1 ) {
FDKpushBack(hBs, bits+2);
hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, bits );
UINT ascLen = transportEnc_LatmWriteValue(hBs, 0);
FDKbyteAlign(hBs, alignAnchor);
ascLen = FDKgetValidBits(hBs) - alignAnchor - ascLen;
FDKpushBack(hBs, FDKgetValidBits(hBs) - alignAnchor);
transportEnc_LatmWriteValue(hBs, ascLen);
transportEnc_writeASC(
hBs,
hAss->config[prog][layer],
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++;
@ -384,7 +384,6 @@ CreateStreamMuxConfig(
case AOT_ER_AAC_LD :
case AOT_ER_AAC_ELD :
case AOT_USAC:
case AOT_RSVD50:
p_linfo->frameLengthType = 0;
FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */

2
libMpegTPEnc/src/version
View File

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

3
libSBRdec/include/sbrdecoder.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -262,6 +262,7 @@ void sbrDecoder_drcDisable ( HANDLE_SBRDECODER self,
* into *count if a payload length is given (byPayLen > 0). If no SBR payload length is
* given (bsPayLen < 0) then the bit stream position on return will be random after this
* function call in case of errors, and any further decoding will be completely pointless.
* This function accepts either normal ordered SBR data or reverse ordered DRM SBR data.
*
* \param self SBR decoder handle.
* \param hBs Bit stream handle as data source.

204
libSBRdec/src/env_calc.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -151,13 +151,13 @@ typedef struct
}
ENV_CALC_NRGS;
/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
SCHAR *filtBuffer_e,
FIXP_DBL *NrgGain,
SCHAR *NrgGain_e,
int subbands);
/*static*/ void calcNrgPerSubband(FIXP_DBL **analysBufferReal,
static void calcNrgPerSubband(FIXP_DBL **analysBufferReal,
FIXP_DBL **analysBufferImag,
int lowSubband, int highSubband,
int start_pos, int next_pos,
@ -165,7 +165,7 @@ ENV_CALC_NRGS;
FIXP_DBL *nrgEst,
SCHAR *nrgEst_e );
/*static*/ void calcNrgPerSfb(FIXP_DBL **analysBufferReal,
static void calcNrgPerSfb(FIXP_DBL **analysBufferReal,
FIXP_DBL **analysBufferImag,
int nSfb,
UCHAR *freqBandTable,
@ -174,13 +174,13 @@ ENV_CALC_NRGS;
FIXP_DBL *nrg_est,
SCHAR *nrg_est_e );
/*static*/ void calcSubbandGain(FIXP_DBL nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
static void calcSubbandGain(FIXP_DBL nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
FIXP_DBL tmpNoise, SCHAR tmpNoise_e,
UCHAR sinePresentFlag,
UCHAR sineMapped,
int noNoiseFlag);
/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
static void calcAvgGain(ENV_CALC_NRGS* nrgs,
int lowSubband,
int highSubband,
FIXP_DBL *sumRef_m,
@ -188,7 +188,7 @@ ENV_CALC_NRGS;
FIXP_DBL *ptrAvgGain_m,
SCHAR *ptrAvgGain_e);
/*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal,
static void adjustTimeSlot_EldGrid(FIXP_DBL *ptrReal,
ENV_CALC_NRGS* nrgs,
UCHAR *ptrHarmIndex,
int lowSubbands,
@ -196,8 +196,17 @@ ENV_CALC_NRGS;
int scale_change,
int noNoiseFlag,
int *ptrPhaseIndex,
int fCldfb);
/*static*/ void adjustTimeSlotHQ(FIXP_DBL *ptrReal,
int scale_diff_low);
static void adjustTimeSlotLC(FIXP_DBL *ptrReal,
ENV_CALC_NRGS* nrgs,
UCHAR *ptrHarmIndex,
int lowSubbands,
int noSubbands,
int scale_change,
int noNoiseFlag,
int *ptrPhaseIndex);
static void adjustTimeSlotHQ(FIXP_DBL *ptrReal,
FIXP_DBL *ptrImag,
HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
ENV_CALC_NRGS* nrgs,
@ -224,7 +233,7 @@ ENV_CALC_NRGS;
Additionally, the flags in harmFlagsPrev are being updated by this function
for the next frame.
*/
/*static*/ void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */
static void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */
int nSfb, /*!< Number of bands in the table */
UCHAR *addHarmonics, /*!< vector with 1 flag per sfb */
int *harmFlagsPrev, /*!< Packed 'addHarmonics' */
@ -990,7 +999,6 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling
/* Prevent the smoothing filter from running on constant levels */
if (j-start_pos < smooth_length)
smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j-start_pos];
else
smooth_ratio = FL2FXCONST_SGL(0.0f);
@ -1007,7 +1015,8 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling
}
else
{
adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
if (flags & SBRDEC_ELD_GRID) {
adjustTimeSlot_EldGrid(&analysBufferReal[j][lowSubband],
pNrgs,
&h_sbr_cal_env->harmIndex,
lowSubband,
@ -1015,7 +1024,18 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling
scale_change,
noNoiseFlag,
&h_sbr_cal_env->phaseIndex,
(flags & SBRDEC_ELD_GRID));
EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale);
} else
{
adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
pNrgs,
&h_sbr_cal_env->harmIndex,
lowSubband,
noSubbands,
scale_change,
noNoiseFlag,
&h_sbr_cal_env->phaseIndex);
}
}
} // for
@ -1176,7 +1196,7 @@ resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to env
can be performed.
This function is called once for each envelope before adjusting.
*/
/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, /*!< bufferd gains */
static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, /*!< bufferd gains */
SCHAR *filtBuffer_e, /*!< exponents of bufferd gains */
FIXP_DBL *nrgGain, /*!< gains for current envelope */
SCHAR *nrgGain_e, /*!< exponents of gains for current envelope */
@ -1331,7 +1351,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output
This function is used when interpolFreq is true.
*/
/*static*/ void calcNrgPerSubband(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */
static void calcNrgPerSubband(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */
FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */
int lowSubband, /*!< Begin of the SBR frequency range */
int highSubband, /*!< High end of the SBR frequency range */
@ -1452,7 +1472,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output
This function is used when interpolFreq is false.
*/
/*static*/ void calcNrgPerSfb(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */
static void calcNrgPerSfb(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */
FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */
int nSfb, /*!< Number of scale factor bands */
UCHAR *freqBandTable, /*!< First Subband for each Sfb */
@ -1585,7 +1605,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output
The resulting energy gain is given by mantissa and exponent.
*/
/*static*/ void calcSubbandGain(FIXP_DBL nrgRef, /*!< Reference Energy according to envelope data */
static void calcSubbandGain(FIXP_DBL nrgRef, /*!< Reference Energy according to envelope data */
SCHAR nrgRef_e, /*!< Reference Energy according to envelope data (exponent) */
ENV_CALC_NRGS* nrgs,
int i,
@ -1689,7 +1709,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output
The result is used as a relative limit for all gains within the
current "limiter band" (a certain frequency range).
*/
/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
static void calcAvgGain(ENV_CALC_NRGS* nrgs,
int lowSubband, /*!< Begin of the limiter band */
int highSubband, /*!< High end of the limiter band */
FIXP_DBL *ptrSumRef,
@ -1728,21 +1748,101 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output
*ptrSumRef_e = sumRef_e;
}
static void adjustTimeSlot_EldGrid(
FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */
ENV_CALC_NRGS* nrgs,
UCHAR *ptrHarmIndex, /*!< Harmonic index */
int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */
int noSubbands, /*!< Number of QMF subbands */
int scale_change, /*!< Number of bits to shift adjusted samples */
int noNoiseFlag, /*!< Flag to suppress noise addition */
int *ptrPhaseIndex, /*!< Start index to random number array */
int scale_diff_low) /*!< */
{
int k;
FIXP_DBL signalReal, sbNoise;
int tone_count = 0;
FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */
FIXP_DBL *pNoiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */
FIXP_DBL *pSineLevel = nrgs->nrgSine; /*!< Sine levels */
int phaseIndex = *ptrPhaseIndex;
UCHAR harmIndex = *ptrHarmIndex;
static const INT harmonicPhase [2][4] = {
{ 1, 0, -1, 0},
{ 0, 1, 0, -1}
};
static const FIXP_DBL harmonicPhaseX [2][4] = {
{ FL2FXCONST_DBL(2.0*1.245183154539139e-001), FL2FXCONST_DBL(2.0*-1.123767859325028e-001), FL2FXCONST_DBL(2.0*-1.245183154539139e-001), FL2FXCONST_DBL(2.0* 1.123767859325028e-001) },
{ FL2FXCONST_DBL(2.0*1.245183154539139e-001), FL2FXCONST_DBL(2.0* 1.123767859325028e-001), FL2FXCONST_DBL(2.0*-1.245183154539139e-001), FL2FXCONST_DBL(2.0*-1.123767859325028e-001) }
};
for (k=0; k < noSubbands; k++) {
phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1);
if( (pSineLevel[0] != FL2FXCONST_DBL(0.0f)) || (noNoiseFlag == 1) ){
sbNoise = FL2FXCONST_DBL(0.0f);
} else {
sbNoise = pNoiseLevel[0];
}
signalReal = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change);
signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise)<<4);
signalReal += pSineLevel[0] * harmonicPhase[0][harmIndex];
*ptrReal = signalReal;
if (k == 0) {
*(ptrReal-1) += scaleValue(fMultDiv2(harmonicPhaseX[lowSubband&1][harmIndex], pSineLevel[0]), -scale_diff_low) ;
if (k < noSubbands - 1) {
*(ptrReal) += fMultDiv2(pSineLevel[1], harmonicPhaseX[(lowSubband+1)&1][harmIndex]);
}
}
if (k > 0 && k < noSubbands - 1 && tone_count < 16) {
*(ptrReal) += fMultDiv2(pSineLevel[- 1], harmonicPhaseX [(lowSubband+k)&1] [harmIndex]);
*(ptrReal) += fMultDiv2(pSineLevel[+ 1], harmonicPhaseX [(lowSubband+k+1)&1][harmIndex]);
}
if (k == noSubbands - 1 && tone_count < 16) {
if (k > 0) {
*(ptrReal) += fMultDiv2(pSineLevel[- 1], harmonicPhaseX [(lowSubband+k)&1][harmIndex]);
}
if (k + lowSubband + 1< 63) {
*(ptrReal+1) += fMultDiv2(pSineLevel[0], harmonicPhaseX[(lowSubband+k+1)&1][harmIndex]);
}
}
if(pSineLevel[0] != FL2FXCONST_DBL(0.0f)){
tone_count++;
}
ptrReal++;
pNoiseLevel++;
pGain++;
pSineLevel++;
}
*ptrHarmIndex = (harmIndex + 1) & 3;
*ptrPhaseIndex = phaseIndex & (SBR_NF_NO_RANDOM_VAL - 1);
}
/*!
\brief Amplify one timeslot of the signal with the calculated gains
and add the noisefloor.
*/
/*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */
static void adjustTimeSlotLC(FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */
ENV_CALC_NRGS* nrgs,
UCHAR *ptrHarmIndex, /*!< Harmonic index */
int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */
int noSubbands, /*!< Number of QMF subbands */
int scale_change, /*!< Number of bits to shift adjusted samples */
int noNoiseFlag, /*!< Flag to suppress noise addition */
int *ptrPhaseIndex, /*!< Start index to random number array */
int fCldfb) /*!< CLDFB 80 flag */
int *ptrPhaseIndex) /*!< Start index to random number array */
{
FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */
FIXP_DBL *pNoiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */
@ -1775,41 +1875,10 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output
sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f);
if (sineLevel!=FL2FXCONST_DBL(0.0f)) tone_count++;
else if (!noNoiseFlag)
/* Add noisefloor to the amplified signal */
signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
if (fCldfb) {
if (!(harmIndex&0x1)) {
/* harmIndex 0,2 */
signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
*ptrReal++ = signalReal;
}
else {
/* harmIndex 1,3 in combination with freqInvFlag */
int shift = (int) (scale_change+1);
shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);
FIXP_DBL tmp1 = scaleValue( fMultDiv2(C1_CLDFB, sineLevel), -shift );
FIXP_DBL tmp2 = fMultDiv2(C1_CLDFB, sineLevelNext);
/* save switch and compare operations and reduce to XOR statement */
if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
*(ptrReal-1) += tmp1;
signalReal -= tmp2;
} else {
*(ptrReal-1) -= tmp1;
signalReal += tmp2;
}
*ptrReal++ = signalReal;
freqInvFlag = !freqInvFlag;
}
} else
{
if (!(harmIndex&0x1)) {
/* harmIndex 0,2 */
@ -1933,8 +2002,9 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output
*ptrHarmIndex = (harmIndex + 1) & 3;
*ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1);
}
void adjustTimeSlotHQ(FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */
FIXP_DBL *RESTRICT ptrImag, /*!< Subband samples to be adjusted, imag part */
static void adjustTimeSlotHQ(
FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */
FIXP_DBL *RESTRICT ptrImag, /*!< Subband samples to be adjusted, imag part */
HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
ENV_CALC_NRGS* nrgs,
int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */
@ -2137,7 +2207,6 @@ ResetLimiterBands ( UCHAR *limiterBandTable, /*!< Resulting band borders in QM
UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1];
int patchBorders[MAX_NUM_PATCHES + 1];
int kx, k2;
FIXP_DBL temp;
int lowSubband = freqBandTable[0];
int highSubband = freqBandTable[noFreqBands];
@ -2169,13 +2238,32 @@ ResetLimiterBands ( UCHAR *limiterBandTable, /*!< Resulting band borders in QM
while (hiLimIndex <= tempNoLim) {
FIXP_DBL div_m, oct_m, temp;
INT div_e = 0, oct_e = 0, temp_e = 0;
k2 = workLimiterBandTable[hiLimIndex] + lowSubband;
kx = workLimiterBandTable[loLimIndex] + lowSubband;
temp = FX_SGL2FX_DBL(FDK_getNumOctavesDiv8(kx,k2)); /* Number of octaves */
temp = fMult(temp, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[limiterBands]);
div_m = fDivNorm(k2, kx, &div_e);
/* calculate number of octaves */
oct_m = fLog2(div_m, div_e, &oct_e);
/* multiply with limiterbands per octave */
/* values 1, 1.2, 2, 3 -> scale factor of 2 */
temp = fMultNorm(oct_m, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[limiterBands], &temp_e);
/* overall scale factor of temp ist addition of scalefactors from log2 calculation,
limiter bands scalefactor (2) and limiter bands multiplication */
temp_e += oct_e + 2;
/* div can be a maximum of 64 (k2 = 64 and kx = 1)
-> oct can be a maximum of 6
-> temp can be a maximum of 18 (as limiterBandsPerOctoave is a maximum factor of 3)
-> we need a scale factor of 5 for comparisson
*/
if (temp >> (5 - temp_e) < FL2FXCONST_DBL (0.49f) >> 5) {
if (temp < FL2FXCONST_DBL (0.49f)>>5) {
if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) {
workLimiterBandTable[hiLimIndex] = highSubband;
nBands--;

4
libSBRdec/src/env_dec.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -369,7 +369,7 @@ leanSbrConcealment(HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control d
FIXP_SGL step; /* speed of fade */
int i;
int currentStartPos = h_prev_data->stopPos - hHeaderData->numberTimeSlots;
int currentStartPos = FDKmax(0, h_prev_data->stopPos - hHeaderData->numberTimeSlots);
int currentStopPos = hHeaderData->numberTimeSlots;

4
libSBRdec/src/env_extr.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -327,7 +327,7 @@ sbrGetHeaderData (HANDLE_SBR_HEADER_DATA hHeaderData,
}
/* Look for new settings. IEC 14496-3, 4.6.18.3.1 */
if(hHeaderData->syncState != SBR_ACTIVE ||
if(hHeaderData->syncState < SBR_HEADER ||
lastHeader.startFreq != pBsData->startFreq ||
lastHeader.stopFreq != pBsData->stopFreq ||
lastHeader.freqScale != pBsData->freqScale ||

12
libSBRdec/src/env_extr.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -125,6 +125,7 @@ amm-info@iis.fraunhofer.de
typedef enum
{
HEADER_NOT_PRESENT,
HEADER_ERROR,
HEADER_OK,
HEADER_RESET
}
@ -132,10 +133,10 @@ SBR_HEADER_STATUS;
typedef enum
{
SBR_NOT_INITIALIZED,
UPSAMPLING,
SBR_HEADER,
SBR_ACTIVE
SBR_NOT_INITIALIZED = 0,
UPSAMPLING = 1,
SBR_HEADER = 2,
SBR_ACTIVE = 3
}
SBR_SYNC_STATE;
@ -179,6 +180,7 @@ typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA;
#define SBRDEC_LOW_POWER 16 /* Flag indicating that Low Power QMF mode shall be used. */
#define SBRDEC_PS_DECODED 32 /* Flag indicating that PS was decoded and rendered. */
#define SBRDEC_LD_MPS_QMF 512 /* Flag indicating that the LD-MPS QMF shall be used. */
#define SBRDEC_SYNTAX_DRM 2048 /* Flag indicating that DRM30/DRM+ reverse syntax is being used. */
#define SBRDEC_DOWNSAMPLE 8192 /* Flag indicating that the downsampling mode is used. */
#define SBRDEC_FLUSH 16384 /* Flag is used to flush all elements in use. */
#define SBRDEC_FORCE_RESET 32768 /* Flag is used to force a reset of all elements in use. */

62
libSBRdec/src/sbr_dec.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -225,7 +225,14 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec, /*!< hand
}
if (resetAnaQmf) {
int qmfErr = qmfInitAnalysisFilterBank (
QMF_FILTER_BANK prvAnaQmf;
int qmfErr;
/* Store current configuration */
FDKmemcpy(&prvAnaQmf, &hSbrDec->AnalysiscQMF, sizeof(QMF_FILTER_BANK));
/* Reset analysis QMF */
qmfErr = qmfInitAnalysisFilterBank (
&hSbrDec->AnalysiscQMF,
hSbrDec->anaQmfStates,
hSbrDec->AnalysiscQMF.no_col,
@ -234,13 +241,22 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec, /*!< hand
hSbrDec->AnalysiscQMF.no_channels,
anaQmfFlags | QMF_FLAG_KEEP_STATES
);
if (qmfErr != 0) {
FDK_ASSERT(0);
/* Restore old configuration of analysis QMF */
FDKmemcpy(&hSbrDec->AnalysiscQMF, &prvAnaQmf, sizeof(QMF_FILTER_BANK));
}
}
if (resetSynQmf) {
int qmfErr = qmfInitSynthesisFilterBank (
QMF_FILTER_BANK prvSynQmf;
int qmfErr;
/* Store current configuration */
FDKmemcpy(&prvSynQmf, &hSbrDec->SynthesisQMF, sizeof(QMF_FILTER_BANK));
/* Reset synthesis QMF */
qmfErr = qmfInitSynthesisFilterBank (
&hSbrDec->SynthesisQMF,
hSbrDec->pSynQmfStates,
hSbrDec->SynthesisQMF.no_col,
@ -251,7 +267,8 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec, /*!< hand
);
if (qmfErr != 0) {
FDK_ASSERT(0);
/* Restore old configuration of synthesis QMF */
FDKmemcpy(&hSbrDec->SynthesisQMF, &prvSynQmf, sizeof(QMF_FILTER_BANK));
}
}
}
@ -321,7 +338,8 @@ sbr_dec ( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */
const int applyProcessing, /*!< Flag for SBR operation */
HANDLE_PS_DEC h_ps_d,
const UINT flags
const UINT flags,
const int codecFrameSize
)
{
int i, slot, reserve;
@ -348,6 +366,33 @@ sbr_dec ( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
if (flags & SBRDEC_ELD_GRID) {
/* Choose the right low delay filter bank */
changeQmfType( hSbrDec, (flags & SBRDEC_LD_MPS_QMF) ? 1 : 0 );
/* If the LD-MPS QMF is not available delay the signal by (96-48*ldSbrSamplingRate)
* samples according to ISO/IEC 14496-3:2009/FDAM 2:2010(E) chapter 4.5.2.13. */
if ( (flags & SBRDEC_LD_MPS_QMF)
&& (hSbrDec->AnalysiscQMF.flags & QMF_FLAG_CLDFB) )
{
INT_PCM *pDlyBuf = hSbrDec->coreDelayBuf; /* DLYBUF */
int smpl, delay = 96 >> (!(flags & SBRDEC_DOWNSAMPLE) ? 1 : 0);
/* Create TMPBUF */
C_AALLOC_SCRATCH_START(pcmTemp, INT_PCM, (96));
/* Copy delay samples from INBUF to TMPBUF */
for (smpl = 0; smpl < delay; smpl += 1) {
pcmTemp[smpl] = timeIn[(codecFrameSize-delay+smpl)*strideIn];
}
/* Move input signal remainder to the very end of INBUF */
for (smpl = (codecFrameSize-delay-1)*strideIn; smpl >= 0; smpl -= strideIn) {
timeIn[smpl+delay] = timeIn[smpl];
}
/* Copy delayed samples from last frame from DLYBUF to the very beginning of INBUF */
for (smpl = 0; smpl < delay; smpl += 1) {
timeIn[smpl*strideIn] = pDlyBuf[smpl];
}
/* Copy TMPBUF to DLYBUF */
FDKmemcpy(pDlyBuf, pcmTemp, delay*sizeof(INT_PCM));
/* Destory TMPBUF */
C_AALLOC_SCRATCH_END(pcmTemp, INT_PCM, (96));
}
}
/*
@ -761,7 +806,7 @@ createSbrDec (SBR_CHANNEL * hSbrChannel,
{
int qmfErr;
/* Adapted QMF analysis post-twiddles for down-sampled HQ SBR */
const UINT downSampledFlag = (downsampleFac==2) ? QMF_FLAG_DOWNSAMPLED : 0;
const UINT downSampledFlag = (flags & SBRDEC_DOWNSAMPLE) ? QMF_FLAG_DOWNSAMPLED : 0;
qmfErr = qmfInitAnalysisFilterBank (
&hs->AnalysiscQMF,
@ -836,6 +881,9 @@ createSbrDec (SBR_CHANNEL * hSbrChannel,
}
}
/* Clear input delay line */
FDKmemclear(hs->coreDelayBuf, (96)*sizeof(INT_PCM));
/* assign qmf time slots */
assignTimeSlots( &hSbrChannel->SbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, qmfFlags & QMF_FLAG_LP);

8
libSBRdec/src/sbr_dec.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -118,6 +118,9 @@ typedef struct
FIXP_DBL * WorkBuffer1;
FIXP_DBL * WorkBuffer2;
/* Delayed time input signal needed to align CLDFD with LD-MPS QMF. */
INT_PCM coreDelayBuf[(96)];
/* QMF filter states */
FIXP_QAS anaQmfStates[(320)];
FIXP_QSS * pSynQmfStates;
@ -182,7 +185,8 @@ sbr_dec (HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */
const int applyProcessing, /*!< Flag for SBR operation */
HANDLE_PS_DEC h_ps_d,
const UINT flags
const UINT flags,
const int codecFrameSize
);

3
libSBRdec/src/sbr_ram.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -135,6 +135,7 @@ struct SBR_DECODER_INSTANCE
USHORT codecFrameSize;
UCHAR synDownsampleFac;
UCHAR numDelayFrames; /* The current number of additional delay frames used for processing. */
UCHAR numFlushedFrames; /* The variable counts the number of frames which are flushed consecutively. */
UINT flags;

11
libSBRdec/src/sbr_rom.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -189,6 +189,15 @@ const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4] =
FL2FXCONST_SGL(3.0f / 4.0f)
};
/*! Constants for calculating the number of limiter bands */
const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4] =
{
FL2FXCONST_DBL(1.0f / 4.0f),
FL2FXCONST_DBL(1.2f / 4.0f),
FL2FXCONST_DBL(2.0f / 4.0f),
FL2FXCONST_DBL(3.0f / 4.0f)
};
/*! Ratio of old gains and noise levels for the first 4 timeslots of an envelope */
const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4] = {
FL2FXCONST_SGL(0.66666666666666f),

3
libSBRdec/src/sbr_rom.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -124,6 +124,7 @@ extern const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRI
extern const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4];
extern const UCHAR FDK_sbrDecoder_sbr_limGains_e[4];
extern const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4];
extern const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4];
extern const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4];
extern const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2];
extern const FIXP_SGL harmonicPhaseX [2][4];

181
libSBRdec/src/sbrdecoder.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -128,6 +128,7 @@ amm-info@iis.fraunhofer.de
#include "lpp_tran.h"
#include "transcendent.h"
#include "FDK_crc.h"
#include "sbrdec_drc.h"
@ -137,7 +138,7 @@ amm-info@iis.fraunhofer.de
/* Decoder library info */
#define SBRDECODER_LIB_VL0 2
#define SBRDECODER_LIB_VL1 2
#define SBRDECODER_LIB_VL2 6
#define SBRDECODER_LIB_VL2 12
#define SBRDECODER_LIB_TITLE "SBR Decoder"
#ifdef __ANDROID__
#define SBRDECODER_LIB_BUILD_DATE ""
@ -194,6 +195,33 @@ static void copySbrHeader( HANDLE_SBR_HEADER_DATA hDst, const HANDLE_SBR_HEADER_
hDst->freqBandData.freqBandTable[1] = hDst->freqBandData.freqBandTableHi;
}
static int compareSbrHeader( const HANDLE_SBR_HEADER_DATA hHdr1, const HANDLE_SBR_HEADER_DATA hHdr2 )
{
int result = 0;
/* compare basic data */
result |= (hHdr1->syncState != hHdr2->syncState) ? 1 : 0;
result |= (hHdr1->status != hHdr2->status) ? 1 : 0;
result |= (hHdr1->frameErrorFlag != hHdr2->frameErrorFlag) ? 1 : 0;
result |= (hHdr1->numberTimeSlots != hHdr2->numberTimeSlots) ? 1 : 0;
result |= (hHdr1->numberOfAnalysisBands != hHdr2->numberOfAnalysisBands) ? 1 : 0;
result |= (hHdr1->timeStep != hHdr2->timeStep) ? 1 : 0;
result |= (hHdr1->sbrProcSmplRate != hHdr2->sbrProcSmplRate) ? 1 : 0;
/* compare bitstream data */
result |= FDKmemcmp( &hHdr1->bs_data, &hHdr2->bs_data, sizeof(SBR_HEADER_DATA_BS) );
result |= FDKmemcmp( &hHdr1->bs_info, &hHdr2->bs_info, sizeof(SBR_HEADER_DATA_BS_INFO) );
/* compare frequency band data */
result |= FDKmemcmp( &hHdr1->freqBandData, &hHdr2->freqBandData, (8+MAX_NUM_LIMITERS+1)*sizeof(UCHAR) );
result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableLo, hHdr2->freqBandData.freqBandTableLo, (MAX_FREQ_COEFFS/2+1)*sizeof(UCHAR) );
result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableHi, hHdr2->freqBandData.freqBandTableHi, (MAX_FREQ_COEFFS+1)*sizeof(UCHAR) );
result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableNoise, hHdr2->freqBandData.freqBandTableNoise, (MAX_NOISE_COEFFS+1)*sizeof(UCHAR) );
result |= FDKmemcmp( hHdr1->freqBandData.v_k_master, hHdr2->freqBandData.v_k_master, (MAX_FREQ_COEFFS+1)*sizeof(UCHAR) );
return result;
}
/*!
\brief Reset SBR decoder.
@ -318,7 +346,6 @@ SBR_ERROR sbrDecoder_ResetElement (
case AOT_PS:
case AOT_ER_AAC_SCAL:
case AOT_DRM_AAC:
case AOT_DRM_SURROUND:
if (CreatePsDec ( &self->hParametricStereoDec, samplesPerFrame )) {
sbrError = SBRDEC_CREATE_ERROR;
goto bail;
@ -392,6 +419,7 @@ int sbrDecoder_isCoreCodecValid(AUDIO_OBJECT_TYPE coreCodec)
case AOT_PS:
case AOT_ER_AAC_SCAL:
case AOT_ER_AAC_ELD:
case AOT_DRM_AAC:
return 1;
default:
return 0;
@ -464,6 +492,8 @@ SBR_ERROR sbrDecoder_InitElement (
self->flags = 0;
self->flags |= (coreCodec == AOT_ER_AAC_ELD) ? SBRDEC_ELD_GRID : 0;
self->flags |= (coreCodec == AOT_ER_AAC_SCAL) ? SBRDEC_SYNTAX_SCAL : 0;
self->flags |= (coreCodec == AOT_DRM_AAC) ? SBRDEC_SYNTAX_SCAL|SBRDEC_SYNTAX_DRM : 0;
/* Init SBR elements */
{
@ -503,7 +533,6 @@ SBR_ERROR sbrDecoder_InitElement (
case AOT_PS:
case AOT_ER_AAC_SCAL:
case AOT_DRM_AAC:
case AOT_DRM_SURROUND:
elChannels = 2;
break;
default:
@ -930,24 +959,73 @@ SBR_ERROR sbrDecoder_Parse(
)
{
SBR_DECODER_ELEMENT *hSbrElement;
HANDLE_SBR_HEADER_DATA hSbrHeader;
HANDLE_SBR_HEADER_DATA hSbrHeader = NULL;
HANDLE_SBR_CHANNEL *pSbrChannel;
SBR_FRAME_DATA *hFrameDataLeft;
SBR_FRAME_DATA *hFrameDataRight;
SBR_ERROR errorStatus = SBRDEC_OK;
SBR_SYNC_STATE initialSyncState;
SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT;
INT startPos;
INT CRCLen = 0;
HANDLE_FDK_BITSTREAM hBsOriginal = hBs;
FDK_CRCINFO crcInfo; /* shall be used for all other CRCs in the future (TBD) */
INT crcReg = 0;
USHORT drmSbrCrc = 0;
int stereo;
int fDoDecodeSbrData = 1;
int lastSlot, lastHdrSlot = 0, thisHdrSlot;
/* Reverse bits of DRM SBR payload */
if ( (self->flags & SBRDEC_SYNTAX_DRM) && *count > 0 )
{
UCHAR *bsBufferDrm = (UCHAR*)self->workBuffer1;
HANDLE_FDK_BITSTREAM hBsBwd = (HANDLE_FDK_BITSTREAM) (bsBufferDrm + (512));
int dataBytes, dataBits;
dataBits = *count;
if (dataBits > ((512)*8)) {
/* do not flip more data than needed */
dataBits = (512)*8;
}
dataBytes = (dataBits+7)>>3;
int j;
if ((j = (int)FDKgetValidBits(hBs)) != 8) {
FDKpushBiDirectional(hBs, (j-8));
}
j = 0;
for ( ; dataBytes > 0; dataBytes--)
{
int i;
UCHAR tmpByte;
UCHAR buffer = 0x00;
tmpByte = (UCHAR) FDKreadBits(hBs, 8);
for (i = 0; i < 4; i++) {
int shift = 2 * i + 1;
buffer |= (tmpByte & (0x08>>i)) << shift;
buffer |= (tmpByte & (0x10<<i)) >> shift;
}
bsBufferDrm[j++] = buffer;
FDKpushBack(hBs, 16);
}
FDKinitBitStream(hBsBwd, bsBufferDrm, (512), dataBits, BS_READER);
/* Use reversed data */
hBs = hBsBwd;
bsPayLen = *count;
}
/* Remember start position of SBR element */
startPos = FDKgetValidBits(hBs);
@ -972,7 +1050,6 @@ SBR_ERROR sbrDecoder_Parse(
hFrameDataLeft = &self->pSbrElement[elementIndex]->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
hFrameDataRight = &self->pSbrElement[elementIndex]->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
initialSyncState = hSbrHeader->syncState;
/* reset PS flag; will be set after PS was found */
self->flags &= ~SBRDEC_PS_DECODED;
@ -1008,12 +1085,19 @@ SBR_ERROR sbrDecoder_Parse(
*/
if (fDoDecodeSbrData)
{
if (crcFlag == 1) {
if (crcFlag) {
switch (self->coreCodec) {
case AOT_ER_AAC_ELD:
FDKpushFor (hBs, 10);
/* check sbrcrc later: we don't know the payload length now */
break;
case AOT_DRM_AAC:
drmSbrCrc = (USHORT)FDKreadBits(hBs, 8);
/* Setup CRC decoder */
FDKcrcInit(&crcInfo, 0x001d, 0xFFFF, 8);
/* Start CRC region */
crcReg = FDKcrcStartReg(&crcInfo, hBs, 0);
break;
default:
CRCLen = bsPayLen - 10; /* change: 0 => i */
if (CRCLen < 0) {
@ -1058,6 +1142,7 @@ SBR_ERROR sbrDecoder_Parse(
hSbrHeader->syncState = SBR_HEADER;
} else {
hSbrHeader->syncState = SBR_NOT_INITIALIZED;
headerStatus = HEADER_ERROR;
}
}
@ -1107,7 +1192,7 @@ SBR_ERROR sbrDecoder_Parse(
valBits = (INT)FDKgetValidBits(hBs);
}
if ( crcFlag == 1 ) {
if ( crcFlag ) {
switch (self->coreCodec) {
case AOT_ER_AAC_ELD:
{
@ -1119,6 +1204,14 @@ SBR_ERROR sbrDecoder_Parse(
FDKpushFor(hBs, crcLen);
}
break;
case AOT_DRM_AAC:
/* End CRC region */
FDKcrcEndReg(&crcInfo, hBs, crcReg);
/* Check CRC */
if ((FDKcrcGetCRC(&crcInfo)^0xFF) != drmSbrCrc) {
fDoDecodeSbrData = 0;
}
break;
default:
break;
}
@ -1169,8 +1262,25 @@ SBR_ERROR sbrDecoder_Parse(
}
bail:
if (errorStatus == SBRDEC_OK) {
if (headerStatus == HEADER_NOT_PRESENT) {
if ( self->flags & SBRDEC_SYNTAX_DRM )
{
hBs = hBsOriginal;
}
if ( (errorStatus == SBRDEC_OK)
|| ( (errorStatus == SBRDEC_PARSE_ERROR)
&& (headerStatus != HEADER_ERROR) ) )
{
int useOldHdr = ( (headerStatus == HEADER_NOT_PRESENT)
|| (headerStatus == HEADER_ERROR) ) ? 1 : 0;
if (!useOldHdr && (thisHdrSlot != lastHdrSlot)) {
useOldHdr |= ( compareSbrHeader( hSbrHeader,
&self->sbrHeader[elementIndex][lastHdrSlot] ) == 0 ) ? 1 : 0;
}
if (useOldHdr != 0) {
/* Use the old header for this frame */
hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = lastHdrSlot;
} else {
@ -1231,12 +1341,21 @@ sbrDecoder_DecodeElement (
int numElementChannels = hSbrElement->nChannels; /* Number of channels of the current SBR element */
if (self->flags & SBRDEC_FLUSH) {
/* Move frame pointer to the next slot which is up to be decoded/applied next */
hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1);
/* Update header and frame data pointer because they have already been set */
hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]];
hFrameDataLeft = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
if ( self->numFlushedFrames > self->numDelayFrames ) {
int hdrIdx;
/* No valid SBR payload available, hence switch to upsampling (in all headers) */
for (hdrIdx = 0; hdrIdx < ((1)+1); hdrIdx += 1) {
self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
}
}
else {
/* Move frame pointer to the next slot which is up to be decoded/applied next */
hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1);
/* Update header and frame data pointer because they have already been set */
hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]];
hFrameDataLeft = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
}
}
/* Update the header error flag */
@ -1356,7 +1475,8 @@ sbrDecoder_DecodeElement (
&pSbrChannel[0]->prevFrameData,
(hSbrHeader->syncState == SBR_ACTIVE),
h_ps_d,
self->flags
self->flags,
codecFrameSize
);
if (stereo) {
@ -1373,7 +1493,8 @@ sbrDecoder_DecodeElement (
&pSbrChannel[1]->prevFrameData,
(hSbrHeader->syncState == SBR_ACTIVE),
NULL,
self->flags
self->flags,
codecFrameSize
);
}
@ -1389,20 +1510,21 @@ sbrDecoder_DecodeElement (
if ( !(self->flags & SBRDEC_PS_DECODED) ) {
/* A decoder which is able to decode PS has to produce a stereo output even if no PS data is availble. */
/* So copy left channel to right channel. */
int copyFrameSize = codecFrameSize * 2 / self->synDownsampleFac;
if (interleaved) {
INT_PCM *ptr;
INT i;
FDK_ASSERT(strideOut == 2);
ptr = timeData;
for (i = codecFrameSize; i--; )
for (i = copyFrameSize>>1; i--; )
{
INT_PCM tmp; /* This temporal variable is required because some compilers can't do *ptr++ = *ptr++ correctly. */
tmp = *ptr++; *ptr++ = tmp;
tmp = *ptr++; *ptr++ = tmp;
}
} else {
FDKmemcpy( timeData+2*codecFrameSize, timeData, 2*codecFrameSize*sizeof(INT_PCM) );
FDKmemcpy( timeData+copyFrameSize, timeData, copyFrameSize*sizeof(INT_PCM) );
}
}
*numOutChannels = 2; /* Output minimum two channels when PS is enabled. */
@ -1466,14 +1588,23 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER self,
self->flags &= ~SBRDEC_PS_DECODED;
}
if ( self->flags & SBRDEC_FLUSH ) {
/* flushing is signalized, hence increment the flush frame counter */
self->numFlushedFrames++;
}
else {
/* no flushing is signalized, hence reset the flush frame counter */
self->numFlushedFrames = 0;
}
/* Loop over SBR elements */
for (sbrElementNum = 0; sbrElementNum<self->numSbrElements; sbrElementNum++)
{
int numElementChan;
if (psPossible && self->pSbrElement[sbrElementNum]->pSbrChannel[1] == NULL) {
errorStatus = SBRDEC_UNSUPPORTED_CONFIG;
goto bail;
/* Disable PS and try decoding SBR mono. */
psPossible = 0;
}
numElementChan = (self->pSbrElement[sbrElementNum]->elementID == ID_CPE) ? 2 : 1;
@ -1581,6 +1712,7 @@ INT sbrDecoder_GetLibInfo( LIB_INFO *info )
| CAPF_SBR_HQ
| CAPF_SBR_LP
| CAPF_SBR_PS_MPEG
| CAPF_SBR_DRM_BS
| CAPF_SBR_CONCEALMENT
| CAPF_SBR_DRC
;
@ -1609,6 +1741,9 @@ UINT sbrDecoder_GetDelay( const HANDLE_SBRDECODER self )
/* Low delay SBR: */
{
outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 32 : 64; /* QMF synthesis */
if (flags & SBRDEC_LD_MPS_QMF) {
outputDelay += 32;
}
}
}
else if (!IS_USAC(self->coreCodec)) {

19
libSBRenc/include/sbr_encoder.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -135,6 +135,12 @@ enum
SBR_SYNTAX_DRM_CRC = 0x0008
};
typedef enum
{
FREQ_RES_LOW = 0,
FREQ_RES_HIGH
} FREQ_RES;
typedef struct
{
CODEC_TYPE coreCoder; /*!< LC or ELD */
@ -168,8 +174,9 @@ typedef struct sbrConfiguration
INT dynBwSupported; /*!< Flag: support for dynamic bandwidth in this combination. */
INT parametricCoding; /*!< Flag: usage of parametric coding tool. */
INT downSampleFactor; /*!< Sampling rate relation between the SBR and the core encoder. */
int freq_res_fixfix[3]; /*!< Frequency resolution of envelopes in frame class FIXFIX
0=1 Env; 1=2 Env; 2=4 Env; */
FREQ_RES freq_res_fixfix[2];/*!< Frequency resolution of envelopes in frame class FIXFIX, for non-split case and split case */
UCHAR fResTransIsLow; /*!< Frequency resolution of envelopes in transient frames: low (0) or variable (1) */
/*
core coder dependent tuning parameters
*/
@ -221,6 +228,8 @@ typedef struct sbrConfiguration
INT sbr_interpol_freq; /*!< Flag: use interpolation in freq. direction. */
INT sbr_smoothing_length; /*!< Flag: choose length 4 or 0 (=on, off). */
UCHAR init_amp_res_FF;
FIXP_DBL threshold_AmpRes_FF_m;
SCHAR threshold_AmpRes_FF_e;
} sbrConfiguration, *sbrConfigurationPtr ;
typedef struct SBR_CONFIG_DATA
@ -237,7 +246,7 @@ typedef struct SBR_CONFIG_DATA
INT noQmfBands; /**< Number of QMF frequency bands. */
INT noQmfSlots; /**< Number of QMF slots. */
UCHAR *freqBandTable[2]; /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeefs actually needed for lowres. */
UCHAR *freqBandTable[2]; /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeffs actually needed for lowres. */
UCHAR *v_k_master; /**< Master BandTable where freqBandTable is derived from. */
@ -249,6 +258,8 @@ typedef struct SBR_CONFIG_DATA
INT xposCtrlSwitch; /**< Flag indicates whether to switch xpos ctrl on the fly. */
INT switchTransposers; /**< Flag indicates whether to switch xpos on the fly . */
UCHAR initAmpResFF;
FIXP_DBL thresholdAmpResFF_m;
SCHAR thresholdAmpResFF_e;
} SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA;
typedef struct {

6
libSBRenc/src/bit_sbr.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -283,9 +283,7 @@ void sbrEncoder_GetHeader(SBR_ENCODER *sbrEncoder,
INT element_index,
int fSendHeaders)
{
int bits;
bits = encodeSbrHeaderData (&sbrEncoder->sbrElement[element_index]->sbrHeaderData, hBs);
encodeSbrHeaderData (&sbrEncoder->sbrElement[element_index]->sbrHeaderData, hBs);
if (fSendHeaders == 0) {
/* Prevent header being embedded into the SBR payload. */

8
libSBRenc/src/bit_sbr.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -141,8 +141,8 @@ struct SBR_ENV_DATA
{
INT sbr_xpos_ctrl;
INT freq_res_fixfix;
FREQ_RES freq_res_fixfix[2];
UCHAR fResTransIsLow;
INVF_MODE sbr_invf_mode;
INVF_MODE sbr_invf_mode_vec[MAX_NUM_NOISE_VALUES];
@ -205,6 +205,8 @@ struct SBR_ENV_DATA
INT balance;
AMP_RES init_sbr_amp_res;
AMP_RES currentAmpResFF;
FIXP_DBL ton_HF[SBR_GLOBAL_TONALITY_VALUES]; /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */
FIXP_DBL global_tonality;
/* extended data */
INT extended_data;

175
libSBRenc/src/env_est.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -103,6 +103,114 @@ static const UCHAR panTable[2][10] = { { 0, 2, 4, 6, 8,12,16,20,24},
static const UCHAR maxIndex[2] = {9, 5};
/******************************************************************************
Functionname: FDKsbrEnc_GetTonality
******************************************************************************/
/***************************************************************************/
/*!
\brief Calculates complete energy per band from the energy values
of the QMF subsamples.
\brief quotaMatrix - calculated in FDKsbrEnc_CalculateTonalityQuotas()
\brief noEstPerFrame - number of estimations per frame
\brief startIndex - start index for the quota matrix
\brief Energies - energy matrix
\brief startBand - start band
\brief stopBand - number of QMF bands
\brief numberCols - number of QMF subsamples
\return mean tonality of the 5 bands with the highest energy
scaled by 2^(RELAXATION_SHIFT+2)*RELAXATION_FRACT
****************************************************************************/
static FIXP_DBL FDKsbrEnc_GetTonality(
const FIXP_DBL *const *quotaMatrix,
const INT noEstPerFrame,
const INT startIndex,
const FIXP_DBL *const *Energies,
const UCHAR startBand,
const INT stopBand,
const INT numberCols
)
{
UCHAR b, e, k;
INT no_enMaxBand[SBR_MAX_ENERGY_VALUES] = { -1, -1, -1, -1, -1 };
FIXP_DBL energyMax[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f) };
FIXP_DBL energyMaxMin = MAXVAL_DBL; /* min. energy in energyMax array */
UCHAR posEnergyMaxMin = 0; /* min. energy in energyMax array position */
FIXP_DBL tonalityBand[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f) };
FIXP_DBL globalTonality = FL2FXCONST_DBL(0.0f);
FIXP_DBL energyBand[QMF_CHANNELS];
INT maxNEnergyValues; /* max. number of max. energy values */
/*** Sum up energies for each band ***/
FDK_ASSERT(numberCols==15||numberCols==16);
/* numberCols is always 15 or 16 for ELD. In case of 16 bands, the
energyBands are initialized with the [15]th column.
The rest of the column energies are added in the next step. */
if (numberCols==15) {
for (b=startBand; b<stopBand; b++) {
energyBand[b]=FL2FXCONST_DBL(0.0f);
}
} else {
for (b=startBand; b<stopBand; b++) {
energyBand[b]=Energies[15][b]>>4;
}
}
for (k=0; k<15; k++) {
for (b=startBand; b<stopBand; b++) {
energyBand[b] += Energies[k][b]>>4;
}
}
/*** Determine 5 highest band-energies ***/
maxNEnergyValues = fMin(SBR_MAX_ENERGY_VALUES, stopBand-startBand);
/* Get min. value in energyMax array */
energyMaxMin = energyMax[0] = energyBand[startBand];
no_enMaxBand[0] = startBand;
posEnergyMaxMin = 0;
for (k=1; k<maxNEnergyValues; k++) {
energyMax[k] = energyBand[startBand+k];
no_enMaxBand[k] = startBand+k;
if (energyMaxMin > energyMax[k]) {
energyMaxMin = energyMax[k];
posEnergyMaxMin = k;
}
}
for (b=startBand+maxNEnergyValues; b<stopBand; b++) {
if (energyBand[b] > energyMaxMin) {
energyMax[posEnergyMaxMin] = energyBand[b];
no_enMaxBand[posEnergyMaxMin] = b;
/* Again, get min. value in energyMax array */
energyMaxMin = energyMax[0];
posEnergyMaxMin = 0;
for (k=1; k<maxNEnergyValues; k++) {
if (energyMaxMin > energyMax[k]) {
energyMaxMin = energyMax[k];
posEnergyMaxMin = k;
}
}
}
}
/*** End determine 5 highest band-energies ***/
/* Get tonality values for 5 highest energies */
for (e=0; e<maxNEnergyValues; e++) {
tonalityBand[e]=FL2FXCONST_DBL(0.0f);
for (k=0; k<noEstPerFrame; k++) {
tonalityBand[e] += quotaMatrix[startIndex + k][no_enMaxBand[e]] >> 1;
}
globalTonality += tonalityBand[e] >> 2; /* headroom of 2+1 (max. 5 additions) */
}
return globalTonality;
}
/***************************************************************************/
/*!
@ -919,10 +1027,42 @@ FDKsbrEnc_extractSbrEnvelope1 (
hEnvChan->qmfScale);
if(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
FIXP_DBL tonality = FDKsbrEnc_GetTonality (
hEnvChan->TonCorr.quotaMatrix,
hEnvChan->TonCorr.numberOfEstimatesPerFrame,
hEnvChan->TonCorr.startIndexMatrix,
sbrExtrEnv->YBuffer + sbrExtrEnv->YBufferWriteOffset,
h_con->freqBandTable[HI][0]+1,
h_con->noQmfBands,
sbrExtrEnv->no_cols
);
hEnvChan->encEnvData.ton_HF[1] = hEnvChan->encEnvData.ton_HF[0];
hEnvChan->encEnvData.ton_HF[0] = tonality;
/* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */
hEnvChan->encEnvData.global_tonality = (hEnvChan->encEnvData.ton_HF[0]>>1) + (hEnvChan->encEnvData.ton_HF[1]>>1);
}
/*
Transient detection COEFF Transform OK
*/
if(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
{
FDKsbrEnc_fastTransientDetect(
&hEnvChan->sbrFastTransientDetector,
sbrExtrEnv->YBuffer,
sbrExtrEnv->YBufferScale,
sbrExtrEnv->YBufferWriteOffset,
eData->transient_info
);
}
else
{
FDKsbrEnc_transientDetect(&hEnvChan->sbrTransientDetector,
sbrExtrEnv->YBuffer,
sbrExtrEnv->YBufferScale,
@ -931,6 +1071,7 @@ FDKsbrEnc_extractSbrEnvelope1 (
sbrExtrEnv->YBufferSzShift,
sbrExtrEnv->time_step,
hEnvChan->SbrEnvFrame.frameMiddleSlot);
}
@ -951,7 +1092,8 @@ FDKsbrEnc_extractSbrEnvelope1 (
sbrExtrEnv->YBufferSzShift,
h_con->nSfb[1],
sbrExtrEnv->time_step,
sbrExtrEnv->no_cols);
sbrExtrEnv->no_cols,
&hEnvChan->encEnvData.global_tonality);
}
@ -1128,12 +1270,26 @@ FDKsbrEnc_extractSbrEnvelope2 (
&& ( ed->nEnvelopes == 1 ) )
{
if (hEnvChan->encEnvData.ldGrid)
hEnvChan->encEnvData.currentAmpResFF = (AMP_RES)h_con->initAmpResFF;
else
if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
{
/* Note: global_tonaliy_float_value == ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0)));
threshold_float_value == ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0))); */
/* decision of SBR_AMP_RES */
if (fIsLessThan( /* global_tonality > threshold ? */
h_con->thresholdAmpResFF_m, h_con->thresholdAmpResFF_e,
hEnvChan->encEnvData.global_tonality, RELAXATION_SHIFT+2 )
)
{
hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5;
}
else {
hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0;
}
} else {
hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5;
}
if ( hEnvChan->encEnvData.currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) {
if ( hEnvChan->encEnvData.currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) {
FDKsbrEnc_InitSbrHuffmanTables(&hEnvChan->encEnvData,
&hEnvChan->sbrCodeEnvelope,
@ -1172,7 +1328,12 @@ FDKsbrEnc_extractSbrEnvelope2 (
}
/* Low energy in low band fix */
if ( hEnvChan->sbrTransientDetector.prevLowBandEnergy < hEnvChan->sbrTransientDetector.prevHighBandEnergy && hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03))
if ( hEnvChan->sbrTransientDetector.prevLowBandEnergy < hEnvChan->sbrTransientDetector.prevHighBandEnergy
&& hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03)
/* The fix needs the non-fast transient detector running.
It sets prevLowBandEnergy and prevHighBandEnergy. */
&& !(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
)
{
int i;

3
libSBRenc/src/env_est.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -127,6 +127,7 @@ typedef SBR_EXTRACT_ENVELOPE *HANDLE_SBR_EXTRACT_ENVELOPE;
struct ENV_CHANNEL
{
FAST_TRAN_DETECTOR sbrFastTransientDetector;
SBR_TRANSIENT_DETECTOR sbrTransientDetector;
SBR_CODE_ENVELOPE sbrCodeEnvelope;
SBR_CODE_ENVELOPE sbrCodeNoiseFloor;

82
libSBRenc/src/fram_gen.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -266,7 +266,7 @@ static void calcCtrlSignal (HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass,
static void ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid,
HANDLE_SBR_FRAME_INFO hFrameInfo,
INT freq_res_fixfix);
FREQ_RES *freq_res_fixfix);
/* table for 8 time slot index */
@ -341,8 +341,9 @@ static const FREQ_RES freqRes_table_16[16] = {
static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
HANDLE_SBR_GRID hSbrGrid,
int tranPosInternal,
int numberTimeSlots
);
int numberTimeSlots,
UCHAR fResTransIsLow
);
/*!
@ -402,11 +403,10 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
const int *v_tuningFreq = v_tuning + 3;
hSbrEnvFrame->v_tuningSegm = v_tuningSegm;
INT freq_res_fixfix = hSbrEnvFrame->freq_res_fixfix;
if (ldGrid) {
/* in case there was a transient at the very end of the previous frame, start with a transient envelope */
if(v_transient_info_pre[1] && (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance)){
if ( !tranFlag && v_transient_info_pre[1] && (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance) ){
tranFlag = 1;
tranPos = 0;
}
@ -529,7 +529,8 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
generateFixFixOnly ( &(hSbrEnvFrame->SbrFrameInfo),
&(hSbrEnvFrame->SbrGrid),
tranPosInternal,
numberTimeSlots
numberTimeSlots,
hSbrEnvFrame->fResTransIsLow
);
return &(hSbrEnvFrame->SbrFrameInfo);
@ -677,7 +678,7 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
---------------------------------------------------------------------------*/
ctrlSignal2FrameInfo (&hSbrEnvFrame->SbrGrid,
&hSbrEnvFrame->SbrFrameInfo,
freq_res_fixfix);
hSbrEnvFrame->freq_res_fixfix);
return &hSbrEnvFrame->SbrFrameInfo;
}
@ -692,7 +693,8 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
HANDLE_SBR_GRID hSbrGrid,
int tranPosInternal,
int numberTimeSlots
int numberTimeSlots,
UCHAR fResTransIsLow
)
{
int nEnv, i, k=0, tranIdx;
@ -727,8 +729,12 @@ static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
/* adjust segment-frequency-resolution according to the segment-length */
for (i=0; i<nEnv; i++){
k = hSbrFrameInfo->borders[i+1] - hSbrFrameInfo->borders[i];
hSbrFrameInfo->freqRes[i] = freqResTable[k];
hSbrGrid->v_f[i] = freqResTable[k];
if (!fResTransIsLow)
hSbrFrameInfo->freqRes[i] = freqResTable[k];
else
hSbrFrameInfo->freqRes[i] = FREQ_RES_LOW;
hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i];
}
hSbrFrameInfo->nEnvelopes = nEnv;
@ -765,15 +771,16 @@ static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
*******************************************************************************/
void
FDKsbrEnc_initFrameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
INT allowSpread,
INT numEnvStatic,
INT staticFraming,
INT timeSlots,
INT freq_res_fixfix
,int ldGrid
)
FDKsbrEnc_initFrameInfoGenerator (
HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
INT allowSpread,
INT numEnvStatic,
INT staticFraming,
INT timeSlots,
const FREQ_RES* freq_res_fixfix
,UCHAR fResTransIsLow,
INT ldGrid
)
{ /* FH 00-06-26 */
FDKmemclear(hSbrEnvFrame,sizeof(SBR_ENVELOPE_FRAME ));
@ -786,7 +793,9 @@ FDKsbrEnc_initFrameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
hSbrEnvFrame->allowSpread = allowSpread;
hSbrEnvFrame->numEnvStatic = numEnvStatic;
hSbrEnvFrame->staticFraming = staticFraming;
hSbrEnvFrame->freq_res_fixfix = freq_res_fixfix;
hSbrEnvFrame->freq_res_fixfix[0] = freq_res_fixfix[0];
hSbrEnvFrame->freq_res_fixfix[1] = freq_res_fixfix[1];
hSbrEnvFrame->fResTransIsLow = fResTransIsLow;
hSbrEnvFrame->length_v_bord = 0;
hSbrEnvFrame->length_v_bordFollow = 0;
@ -804,6 +813,7 @@ FDKsbrEnc_initFrameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
hSbrEnvFrame->dmin = 2;
hSbrEnvFrame->dmax = 16;
hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_512LD;
hSbrEnvFrame->SbrGrid.bufferFrameStart = 0;
} else
switch(timeSlots){
case NUMBER_TIME_SLOTS_1920:
@ -1862,19 +1872,28 @@ createDefFrameInfo(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, INT nEnv, INT nTimeSlots
Functionname: ctrlSignal2FrameInfo
*******************************************************************************
Description: Calculates frame_info struct from control signal.
Description: Convert "clear-text" sbr_grid() to "frame info" used by the
envelope and noise floor estimators.
This is basically (except for "low level" calculations) the
bitstream decoder defined in the MPEG-4 standard, sub clause
4.6.18.3.3, Time / Frequency Grid. See inline comments for
explanation of the shorten and noise border algorithms.
Arguments: hSbrGrid - source
hSbrFrameInfo - destination
freq_res_fixfix - frequency resolution for FIXFIX frames
Return: void; hSbrFrameInfo contains the updated FRAME_INFO struct
*******************************************************************************/
static void
ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid,
HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
INT freq_res_fixfix)
ctrlSignal2FrameInfo (
HANDLE_SBR_GRID hSbrGrid, /* input : the grid handle */
HANDLE_SBR_FRAME_INFO hSbrFrameInfo, /* output: the frame info handle */
FREQ_RES *freq_res_fixfix /* in/out: frequency resolution for FIXFIX frames */
)
{
INT frameSplit = 0;
INT nEnv = 0, border = 0, i, k, p /*?*/;
INT *v_r = hSbrGrid->bs_rel_bord;
INT *v_f = hSbrGrid->v_f;
@ -1887,17 +1906,10 @@ ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid,
case FIXFIX:
createDefFrameInfo(hSbrFrameInfo, hSbrGrid->bs_num_env, numberTimeSlots);
/* At this point all frequency resolutions are set to FREQ_RES_HIGH, so
* only if freq_res_fixfix is set to FREQ_RES_LOW, they all have to be
* changed.
* snd */
if (freq_res_fixfix == FREQ_RES_LOW) {
for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) {
hSbrFrameInfo->freqRes[i] = FREQ_RES_LOW;
}
frameSplit = (hSbrFrameInfo->nEnvelopes > 1);
for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) {
hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i] = freq_res_fixfix[frameSplit];
}
/* ELD: store current frequency resolution */
hSbrGrid->v_f[0] = hSbrFrameInfo->freqRes[0];
break;
case FIXVAR:

30
libSBRenc/src/fram_gen.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -89,6 +89,7 @@ amm-info@iis.fraunhofer.de
#define _FRAM_GEN_H
#include "sbr_def.h" /* for MAX_ENVELOPES and MAX_NOISE_ENVELOPES in struct FRAME_INFO and CODEC_TYPE */
#include "sbr_encoder.h" /* for FREQ_RES */
#define MAX_ENVELOPES_VARVAR MAX_ENVELOPES /*!< worst case number of envelopes in a VARVAR frame */
#define MAX_ENVELOPES_FIXVAR_VARFIX 4 /*!< worst case number of envelopes in VARFIX and FIXVAR frames */
@ -114,7 +115,7 @@ typedef enum {
#define NUMBER_TIME_SLOTS_1920 15
#define LD_PRETRAN_OFF 3
#define FRAME_MIDDLE_SLOT_512LD 0
#define FRAME_MIDDLE_SLOT_512LD 4
#define NUMBER_TIME_SLOTS_512LD 8
#define TRANSIENT_OFFSET_LD 0
@ -248,9 +249,10 @@ typedef struct
INT frameMiddleSlot; /*!< transient detector offset in SBR timeslots */
/* basic tuning parameters */
INT staticFraming; /*!< 1: run static framing in time, i.e. exclusive use of bs_frame_class = FIXFIX */
INT numEnvStatic; /*!< number of envelopes per frame for static framing */
INT freq_res_fixfix; /*!< envelope frequency resolution to use for bs_frame_class = FIXFIX */
INT staticFraming; /*!< 1: run static framing in time, i.e. exclusive use of bs_frame_class = FIXFIX */
INT numEnvStatic; /*!< number of envelopes per frame for static framing */
FREQ_RES freq_res_fixfix[2]; /*!< envelope frequency resolution to use for bs_frame_class = FIXFIX; single env and split */
UCHAR fResTransIsLow; /*!< frequency resolution for transient frames - always low (0) or according to table (1) */
/* expert tuning parameters */
const int *v_tuningSegm; /*!< segment lengths to use around transient */
@ -286,14 +288,16 @@ typedef SBR_ENVELOPE_FRAME *HANDLE_SBR_ENVELOPE_FRAME;
void
FDKsbrEnc_initFrameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
INT allowSpread,
INT numEnvStatic,
INT staticFraming,
INT timeSlots,
INT freq_res_fixfix
,int ldGrid
);
FDKsbrEnc_initFrameInfoGenerator (
HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
INT allowSpread,
INT numEnvStatic,
INT staticFraming,
INT timeSlots,
const FREQ_RES* freq_res_fixfix
,UCHAR fResTransIsLow,
INT ldGrid
);
HANDLE_SBR_FRAME_INFO
FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,

32
libSBRenc/src/mh_det.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -663,10 +663,27 @@ static void transientCleanUp(FIXP_DBL **quotaBuffer,
}
/**************************************************************************/
/*****************************************************************************/
/*!
\brief Do detection for one tonality estimate.
\brief Detection for one tonality estimate.
This is the actual missing harmonics detection, using information from the
previous detection.
If a missing harmonic was detected (in a previous frame) due to too high
tonality differences, but there was not enough tonality difference in the
current frame, the detection algorithm still continues to trace the strongest
tone in the scalefactor band (assuming that this is the tone that is going to
be replaced in the decoder). This is done to avoid abrupt endings of sines
fading out (e.g. in the glockenspiel).
The function also tries to estimate where one sine is going to be replaced
with multiple sines (due to the patching). This is done by comparing the
tonality flatness measure of the original and the SBR signal.
The function also tries to estimate (for the scalefactor bands only
containing one qmf subband) when a strong tone in the original will be
replaced by a strong tone in the adjacent QMF subband.
\return none.
@ -694,10 +711,10 @@ static void detection(FIXP_DBL *quotaBuffer,
for(i=0;i<nSfb;i++){
thresTemp = (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f))
? fixMax(fMult(mhThresh.decayGuideDiff,guideVectors.guideVectorDiff[i]), mhThresh.thresHoldDiffGuide)
? fMax(fMult(mhThresh.decayGuideDiff,guideVectors.guideVectorDiff[i]), mhThresh.thresHoldDiffGuide)
: mhThresh.thresHoldDiff;
thresTemp = fixMin(thresTemp, mhThresh.thresHoldDiff);
thresTemp = fMin(thresTemp, mhThresh.thresHoldDiff);
if(pDiffVecScfb[i] > thresTemp){
pHarmVec[i] = 1;
@ -813,8 +830,11 @@ static void detectionWithPrediction(FIXP_DBL **quotaBuffer,
if(newDetectionAllowed){
/* Since we don't want to use the transient region for detection (since the tonality values
tend to be a bit unreliable for this region) the guide-values are copied to the current
starting point. */
if(totNoEst > 1){
start = detectionStart;
start = detectionStart+1;
if (start != 0) {
FDKmemcpy(guideVectors[start].guideVectorDiff,guideVectors[0].guideVectorDiff,nSfb*sizeof(FIXP_DBL));

9
libSBRenc/src/nf_est.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
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
*****************************************************************/
if(!missingHarmonicFlag)
*noiseLevel = fMult(*noiseLevel, noiseFloorOffset)<<(NOISE_FLOOR_OFFSET_SCALING);
if(!missingHarmonicFlag) {
*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
@ -297,7 +298,7 @@ FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFlo
SCHAR *indexVector, /*!< Index vector to obtain the patched data. */
INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */
INT startIndex, /*!< Start index. */
int numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */
UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */
int transientFrame, /*!< A flag indicating if a transient is present. */
INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */
UINT sbrSyntaxFlags

4
libSBRenc/src/nf_est.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -119,7 +119,7 @@ FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFlo
SCHAR* indexVector, /*!< Index vector to obtain the patched data. */
INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */
INT startIndex, /*!< Start index. */
int numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */
UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */
INT transientFrame, /*!< A flag indicating if a transient is present. */
INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */
UINT sbrSyntaxFlags

36
libSBRenc/src/ps_bitenc.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -261,21 +261,23 @@ static const UINT opdDeltaTime_Code[] =
0x00000001, 0x00000002, 0x00000001, 0x00000007, 0x00000006, 0000000000, 0x00000002, 0x00000003
};
static const INT psBands[] =
static INT getNoBands(const INT mode)
{
PS_BANDS_COARSE,
PS_BANDS_MID
};
INT noBands = 0;
static INT getNoBands(UINT mode)
{
if(mode>=6)
return 0;
switch (mode) {
case 0: case 3: /* coarse */
noBands = PS_BANDS_COARSE;
break;
case 1: case 4: /* mid */
noBands = PS_BANDS_MID;
break;
case 2: case 5: /* fine not supported */
default: /* coarse as default */
noBands = PS_BANDS_COARSE;
}
if(mode>=3)
mode = mode-3;
return psBands[mode];
return noBands;
}
static INT getIIDRes(INT iidMode)
@ -524,7 +526,7 @@ static INT encodeIpdOpd(HANDLE_PS_OUT psOut,
bitCnt += FDKsbrEnc_EncodeIpd( hBitBuf,
psOut->ipd[env],
ipdLast,
getNoBands((UINT)psOut->iidMode),
getNoBands(psOut->iidMode),
psOut->deltaIPD[env],
&error);
@ -532,7 +534,7 @@ static INT encodeIpdOpd(HANDLE_PS_OUT psOut,
bitCnt += FDKsbrEnc_EncodeOpd( hBitBuf,
psOut->opd[env],
opdLast,
getNoBands((UINT)psOut->iidMode),
getNoBands(psOut->iidMode),
psOut->deltaOPD[env],
&error );
}
@ -661,7 +663,7 @@ INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut,
bitCnt += FDKsbrEnc_EncodeIid( hBitBuf,
psOut->iid[env],
iidLast,
getNoBands((UINT)psOut->iidMode),
getNoBands(psOut->iidMode),
(PS_IID_RESOLUTION)getIIDRes(psOut->iidMode),
psOut->deltaIID[env],
&error );
@ -677,7 +679,7 @@ INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut,
bitCnt += FDKsbrEnc_EncodeIcc( hBitBuf,
psOut->icc[env],
iccLast,
getNoBands((UINT)psOut->iccMode),
getNoBands(psOut->iccMode),
psOut->deltaICC[env],
&error);

16
libSBRenc/src/ps_encode.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
ｩ Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur F<EFBFBD>derung der angewandten Forschung e.V.
All rights reserved.
1. INTRODUCTION
@ -149,21 +149,21 @@ typedef enum {
static const FIXP_DBL iidQuant_fx[15] = {
0xce000000, 0xdc000000, 0xe4000000, 0xec000000, 0xf2000000, 0xf8000000, 0xfc000000, 0x00000000,
0x04000000, 0x08000000, 0x0e000000, 0x14000000, 0x1c000000, 0x24000000, 0x32000000
(FIXP_DBL)0xce000000, (FIXP_DBL)0xdc000000, (FIXP_DBL)0xe4000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf2000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000,
(FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, (FIXP_DBL)0x0e000000, (FIXP_DBL)0x14000000, (FIXP_DBL)0x1c000000, (FIXP_DBL)0x24000000, (FIXP_DBL)0x32000000
};
static const FIXP_DBL iidQuantFine_fx[31] = {
0x9c000001, 0xa6000001, 0xb0000001, 0xba000001, 0xc4000000, 0xce000000, 0xd4000000, 0xda000000,
0xe0000000, 0xe6000000, 0xec000000, 0xf0000000, 0xf4000000, 0xf8000000, 0xfc000000, 0x00000000,
0x04000000, 0x08000000, 0x0c000000, 0x10000000, 0x14000000, 0x1a000000, 0x20000000, 0x26000000,
0x2c000000, 0x32000000, 0x3c000000, 0x45ffffff, 0x4fffffff, 0x59ffffff, 0x63ffffff
(FIXP_DBL)0x9c000001, (FIXP_DBL)0xa6000001, (FIXP_DBL)0xb0000001, (FIXP_DBL)0xba000001, (FIXP_DBL)0xc4000000, (FIXP_DBL)0xce000000, (FIXP_DBL)0xd4000000, (FIXP_DBL)0xda000000,
(FIXP_DBL)0xe0000000, (FIXP_DBL)0xe6000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf0000000, (FIXP_DBL)0xf4000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000,
(FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, (FIXP_DBL)0x0c000000, (FIXP_DBL)0x10000000, (FIXP_DBL)0x14000000, (FIXP_DBL)0x1a000000, (FIXP_DBL)0x20000000, (FIXP_DBL)0x26000000,
(FIXP_DBL)0x2c000000, (FIXP_DBL)0x32000000, (FIXP_DBL)0x3c000000, (FIXP_DBL)0x45ffffff, (FIXP_DBL)0x4fffffff, (FIXP_DBL)0x59ffffff, (FIXP_DBL)0x63ffffff
};
static const FIXP_DBL iccQuant[8] = {
0x7fffffff, 0x77ef9d7f, 0x6babc97f, 0x4ceaf27f, 0x2f0ed3c0, 0x00000000, 0xb49ba601, 0x80000000
(FIXP_DBL)0x7fffffff, (FIXP_DBL)0x77ef9d7f, (FIXP_DBL)0x6babc97f, (FIXP_DBL)0x4ceaf27f, (FIXP_DBL)0x2f0ed3c0, (FIXP_DBL)0x00000000, (FIXP_DBL)0xb49ba601, (FIXP_DBL)0x80000000
};
static FDK_PSENC_ERROR InitPSData(

14
libSBRenc/src/sbr_def.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -122,6 +122,8 @@ amm-info@iis.fraunhofer.de
/************ Definitions ***************/
#define SBR_COMP_MODE_DELTA 0
#define SBR_COMP_MODE_CTS 1
#define SBR_MAX_ENERGY_VALUES 5
#define SBR_GLOBAL_TONALITY_VALUES 2
#define MAX_NUM_CHANNELS 2
@ -232,6 +234,8 @@ amm-info@iis.fraunhofer.de
#define FREQ 0
#define TIME 1
/* qmf data scaling */
#define QMF_SCALE_OFFSET 7
/* huffman tables */
#define CODE_BOOK_SCF_LAV00 60
@ -268,12 +272,4 @@ typedef enum
}
INVF_MODE;
typedef enum
{
FREQ_RES_LOW = 0,
FREQ_RES_HIGH
}
FREQ_RES;
#endif

126
libSBRenc/src/sbr_encoder.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -103,7 +103,7 @@ amm-info@iis.fraunhofer.de
#define SBRENCODER_LIB_VL0 3
#define SBRENCODER_LIB_VL1 3
#define SBRENCODER_LIB_VL2 4
#define SBRENCODER_LIB_VL2 12
@ -170,7 +170,6 @@ getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */
{
int i, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0;
UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE;
int isforThisCodec=0;
#define isForThisCore(i) \
( ( sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD ) || \
@ -413,6 +412,23 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
config->codecSettings.transFac = transFac;
config->codecSettings.standardBitrate = standardBitrate;
if (bitRate < 28000) {
config->threshold_AmpRes_FF_m = (FIXP_DBL)MAXVAL_DBL;
config->threshold_AmpRes_FF_e = 7;
}
else if (bitRate >= 28000 && bitRate <= 48000) {
/* The float threshold is 75
0.524288f is fractional part of RELAXATION, the quotaMatrix and therefore tonality are scaled by this
2/3 is because the original implementation divides the tonality values by 3, here it's divided by 2
128 compensates the necessary shiftfactor of 7 */
config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(75.0f*0.524288f/(2.0f/3.0f)/128.0f);
config->threshold_AmpRes_FF_e = 7;
}
else if (bitRate > 48000) {
config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(0);
config->threshold_AmpRes_FF_e = 0;
}
if (bitRate==0) {
/* map vbr quality to bitrate */
if (vbrMode < 30)
@ -468,6 +484,57 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
config->stereoMode = sbrTuningTable[idx].stereoMode ;
config->freqScale = sbrTuningTable[idx].freqScale ;
if (numChannels == 1) {
/* stereo case */
switch (core) {
case AOT_AAC_LC:
if (bitRate <= (useSpeechConfig?24000U:20000U)) {
config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */
config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */
}
break;
case AOT_ER_AAC_ELD:
if (bitRate < 36000)
config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */
if (bitRate < 26000) {
config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */
config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */
}
break;
default:
break;
}
}
else {
/* stereo case */
switch (core) {
case AOT_AAC_LC:
if (bitRate <= 28000) {
config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */
config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */
}
break;
case AOT_ER_AAC_ELD:
if (bitRate < 72000) {
config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */
}
if (bitRate < 52000) {
config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */
config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */
}
break;
default:
break;
}
if (bitRate <= 28000) {
/*
additionally restrict frequency resolution in FIXFIX frames
to further reduce SBR payload size */
config->freq_res_fixfix[0] = FREQ_RES_LOW;
config->freq_res_fixfix[1] = FREQ_RES_LOW;
}
}
/* adjust usage of parametric coding dependent on bitrate and speech config flag */
if (useSpeechConfig)
config->parametricCoding = 0;
@ -516,6 +583,7 @@ static UINT
FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
INT downSampleFactor,
UINT codecGranuleLen
,const INT isLowDelay
)
{
if ( (downSampleFactor < 1 || downSampleFactor > 2) ||
@ -526,7 +594,11 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
config->useWaveCoding = 0;
config->crcSbr = 0;
config->dynBwSupported = 1;
config->tran_thr = 13000;
if (isLowDelay)
config->tran_thr = 6000;
else
config->tran_thr = 13000;
config->parametricCoding = 1;
config->sbrFrameSize = codecGranuleLen * downSampleFactor;
@ -559,7 +631,9 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
config->noiseFloorOffset = 0;
config->startFreq = 5; /* 5.9 respectively 6.0 kHz at fs = 44.1/48 kHz */
config->stopFreq = 9; /* 16.2 respectively 16.8 kHz at fs = 44.1/48 kHz */
config->freq_res_fixfix[0] = FREQ_RES_HIGH; /* non-split case */
config->freq_res_fixfix[1] = FREQ_RES_HIGH; /* split case */
config->fResTransIsLow = 0; /* for transient frames, set variable frequency resolution according to freqResTable */
/* header_extra_1 */
config->freqScale = SBR_FREQ_SCALE_DEFAULT;
@ -854,7 +928,7 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder,
int clearOutput /*!< Do not consider any input signal */
)
{
HANDLE_SBR_ELEMENT hSbrElement = hEnvEncoder->sbrElement[iElement];
HANDLE_SBR_ELEMENT hSbrElement = NULL;
FDK_CRCINFO crcInfo;
INT crcReg;
INT ch;
@ -1207,7 +1281,10 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData,
FDK_ASSERT(params->e >= 0);
hEnv->encEnvData.freq_res_fixfix = 1;
hEnv->encEnvData.freq_res_fixfix[0] = params->freq_res_fixfix[0];
hEnv->encEnvData.freq_res_fixfix[1] = params->freq_res_fixfix[1];
hEnv->encEnvData.fResTransIsLow = params->fResTransIsLow;
hEnv->fLevelProtect = 0;
hEnv->encEnvData.ldGrid = (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? 1 : 0;
@ -1349,11 +1426,29 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData,
e,
params->stat,
timeSlots,
hEnv->encEnvData.freq_res_fixfix
,hEnv->encEnvData.ldGrid
hEnv->encEnvData.freq_res_fixfix,
hEnv->encEnvData.fResTransIsLow,
hEnv->encEnvData.ldGrid
);
if(sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
{
INT bandwidth_qmf_slot = (sbrConfigData->sampleFreq>>1) / (sbrConfigData->noQmfBands);
if(FDKsbrEnc_InitSbrFastTransientDetector(
&hEnv->sbrFastTransientDetector,
sbrConfigData->noQmfSlots,
bandwidth_qmf_slot,
sbrConfigData->noQmfBands,
sbrConfigData->freqBandTable[0][0]
))
return(1);
}
/* The transient detector has to be initialized also if the fast transient
detector was active, because the values from the transient detector
structure are used. */
if(FDKsbrEnc_InitSbrTransientDetector (&hEnv->sbrTransientDetector,
sbrConfigData->sbrSyntaxFlags,
sbrConfigData->frameSize,
sbrConfigData->sampleFreq,
params,
@ -1553,12 +1648,6 @@ INT FDKsbrEnc_EnvInit (
hSbrElement->sbrConfigData.sbrSyntaxFlags = 0;
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:
hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY;
break;
@ -1665,6 +1754,8 @@ INT FDKsbrEnc_EnvInit (
/* other switches */
hSbrElement->sbrConfigData.useWaveCoding = params->useWaveCoding;
hSbrElement->sbrConfigData.useParametricCoding = params->parametricCoding;
hSbrElement->sbrConfigData.thresholdAmpResFF_m = params->threshold_AmpRes_FF_m;
hSbrElement->sbrConfigData.thresholdAmpResFF_e = params->threshold_AmpRes_FF_e;
/* init freq band table */
if(updateFreqBandTable(&hSbrElement->sbrConfigData,
@ -1848,7 +1939,7 @@ 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;
}
if ( aot==AOT_ER_AAC_ELD ) {
@ -2006,7 +2097,8 @@ INT sbrEncoder_Init(
*/
if ( ! FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el],
*downSampleFactor,
coreFrameLength
coreFrameLength,
IS_LOWDELAY(aot)
) )
{
error = 1;

11
libSBRenc/src/sbr_rom.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -684,6 +684,9 @@ const sbrTuningTable_t sbrTuningTable[] =
/** AAC LOW DELAY SECTION **/
/* 24 kHz dual rate - 12kHz singlerate is not allowed (deactivated in FDKsbrEnc_IsSbrSettingAvail()) */
{ CODEC_AACLD, 8000, 32000, 12000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */
/*** mono ***/
/* 16/32 kHz dual rate not yet tuned ->alb copied from non LD tables*/
{ CODEC_AACLD, 16000, 18000, 16000, 1, 4, 5, 9, 7, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s wrr: tuned */
@ -702,10 +705,10 @@ const sbrTuningTable_t sbrTuningTable[] =
{ CODEC_AACLD, 52000, 64001, 22050, 1, 13,11,11,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 56 kbit/s */
/* 24/48 kHz dual rate */
{ CODEC_AACLD, 20000, 22000, 24000, 1, 4, 1, 8, 4, 2, 3, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
{ CODEC_AACLD, 20000, 22000, 24000, 1, 3, 4, 8, 8, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
{ CODEC_AACLD, 22000, 28000, 24000, 1, 3, 8, 8, 7, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
{ CODEC_AACLD, 28000, 36000, 24000, 1, 4, 8, 8, 7, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
{ CODEC_AACLD, 36000, 56000, 24000, 1, 8, 9, 9, 9, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
{ CODEC_AACLD, 36000, 56000, 24000, 1, 8, 9, 9, 8, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
{ CODEC_AACLD, 56000, 64001, 24000, 1, 13,11,11,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 kbit/s */
/* 32/64 kHz dual rate */ /* placebo settings */ /*jgr: new, copy from CODEC_AAC */
@ -722,7 +725,7 @@ const sbrTuningTable_t sbrTuningTable[] =
{ CODEC_AACLD, 100000,160001, 44100, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */
/* 48/96 kHz dual rate */ /* 32 and 40kbps line tuned for dual-rate SBR */
{ CODEC_AACLD, 36000, 60000, 48000, 1, 8, 7, 6, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
{ CODEC_AACLD, 36000, 60000, 48000, 1, 4, 7, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* nominal: 40 */
{ CODEC_AACLD, 60000, 72000, 48000, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
{ CODEC_AACLD, 72000,100000, 48000, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
{ CODEC_AACLD, 100000,160001, 48000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */

8
libSBRenc/src/ton_corr.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -682,7 +682,7 @@ FDKsbrEnc_InitTonCorrParamExtr (INT frameSize, /*!< Current
/*
Reset the patching and allocate memory for the quota matrix.
Assing parameters for the LPC analysis.
Assuming parameters for the LPC analysis.
*/
if (sbrCfg->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
switch (timeSlots) {
@ -690,7 +690,7 @@ FDKsbrEnc_InitTonCorrParamExtr (INT frameSize, /*!< Current
hTonCorr->lpcLength[0] = 8 - LPC_ORDER;
hTonCorr->lpcLength[1] = 7 - LPC_ORDER;
hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD;
hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 7;
hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 7 */
hTonCorr->frameStartIndexInvfEst = 0;
hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
break;
@ -698,7 +698,7 @@ FDKsbrEnc_InitTonCorrParamExtr (INT frameSize, /*!< Current
hTonCorr->lpcLength[0] = 8 - LPC_ORDER;
hTonCorr->lpcLength[1] = 8 - LPC_ORDER;
hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD;
hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 8;
hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 8 */
hTonCorr->frameStartIndexInvfEst = 0;
hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
break;

4
libSBRenc/src/ton_corr.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -118,7 +118,7 @@ typedef struct
INT bufferLength; /*!< Length of the r and i buffers. */
INT stepSize; /*!< Stride for the lpc estimate. */
INT numberOfEstimates; /*!< The total number of estiamtes, available in the quotaMatrix.*/
INT numberOfEstimatesPerFrame; /*!< The number of estimates per frame available in the quotaMatrix.*/
UINT numberOfEstimatesPerFrame; /*!< The number of estimates per frame available in the quotaMatrix.*/
INT lpcLength[2]; /*!< Segment length used for second order LPC analysis.*/
INT nextSample; /*!< Where to start the LPC analysis of the current frame.*/
INT move; /*!< How many estimates to move in the quotaMatrix, when buffering. */

496
libSBRenc/src/tran_det.cpp
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -89,7 +89,7 @@ amm-info@iis.fraunhofer.de
#include "genericStds.h"
#define NORM_QMF_ENERGY 5.684341886080801486968994140625e-14 /* 2^-44 */
#define NORM_QMF_ENERGY 9.31322574615479E-10 /* 2^-30 */
/* static FIXP_DBL ABS_THRES = fixMax( FL2FXCONST_DBL(1.28e5 * NORM_QMF_ENERGY), (FIXP_DBL)1) Minimum threshold for detecting changes */
#define ABS_THRES ((FIXP_DBL)16)
@ -106,22 +106,30 @@ amm-info@iis.fraunhofer.de
\return calculated value
*******************************************************************************/
#define NRG_SHIFT 3 /* for energy summation */
static FIXP_DBL spectralChange(FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS],
INT *scaleEnergies,
FIXP_DBL EnergyTotal,
INT nSfb,
INT start,
INT border,
INT stop)
INT YBufferWriteOffset,
INT stop,
INT *result_e)
{
INT i,j;
INT len1,len2;
FIXP_DBL delta,tmp0,tmp1,tmp2;
FIXP_DBL accu1,accu2,delta_sum,result;
SCHAR energies_e_diff[NUMBER_TIME_SLOTS_2304], energies_e, energyTotal_e=19, energies_e_add;
SCHAR prevEnergies_e_diff, newEnergies_e_diff;
FIXP_DBL tmp0,tmp1;
FIXP_DBL accu1,accu2,accu1_init,accu2_init;
FIXP_DBL delta, delta_sum;
INT accu_e, tmp_e;
FDK_ASSERT(scaleEnergies[0] >= 0);
delta_sum = FL2FXCONST_DBL(0.0f);
*result_e = 0;
/* equal for aac (would be not equal for mp3) */
len1 = border-start;
len2 = stop-border;
@ -130,43 +138,91 @@ static FIXP_DBL spectralChange(FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FRE
pos_weight = FL2FXCONST_DBL(0.5f) - (len1*GetInvInt(len1+len2));
pos_weight = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL - (fMult(pos_weight, pos_weight)<<2);
delta_sum = FL2FXCONST_DBL(0.0f);
/*** Calc scaling for energies ***/
FDK_ASSERT(scaleEnergies[0] >= 0);
FDK_ASSERT(scaleEnergies[1] >= 0);
energies_e = 19 - FDKmin(scaleEnergies[0], scaleEnergies[1]);
/* limit shift for energy accumulation, energies_e can be -10 min. */
if (energies_e < -10) {
energies_e_add = -10 - energies_e;
energies_e = -10;
} else if (energies_e > 17) {
energies_e_add = energies_e - 17;
energies_e = 17;
} else {
energies_e_add = 0;
}
/* compensate scaling differences between scaleEnergies[0] and scaleEnergies[1] */
prevEnergies_e_diff = scaleEnergies[0] - FDKmin(scaleEnergies[0], scaleEnergies[1]) + energies_e_add + NRG_SHIFT;
newEnergies_e_diff = scaleEnergies[1] - FDKmin(scaleEnergies[0], scaleEnergies[1]) + energies_e_add + NRG_SHIFT;
prevEnergies_e_diff = fMin(prevEnergies_e_diff, DFRACT_BITS-1);
newEnergies_e_diff = fMin(newEnergies_e_diff, DFRACT_BITS-1);
for (i=start; i<YBufferWriteOffset; i++) {
energies_e_diff[i] = prevEnergies_e_diff;
}
for (i=YBufferWriteOffset; i<stop; i++) {
energies_e_diff[i] = newEnergies_e_diff;
}
/* Sum up energies of all QMF-timeslots for both halfs */
FDK_ASSERT(len1<=8); /* otherwise an overflow is possible */
FDK_ASSERT(len2<=8); /* otherwise an overflow is possible */
/* init with some energy to prevent division by zero
and to prevent splitting for very low levels */
accu1_init = scaleValue((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY))),-energies_e);
accu2_init = scaleValue((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY))),-energies_e);
accu1_init = fMult(accu1_init, (FIXP_DBL)len1<<((DFRACT_BITS-1)-NRG_SHIFT-1))<<1;
accu2_init = fMult(accu2_init, (FIXP_DBL)len2<<((DFRACT_BITS-1)-NRG_SHIFT-1))<<1;
for (j=0; j<nSfb; j++) {
#define NRG_SCALE 3
/* init with some energy to prevent division by zero
and to prevent splitting for very low levels */
accu1 = ((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY*8.0/32))) << fixMin(scaleEnergies[0],25))>>NRG_SCALE; /* complex init for compare with original version */
accu2 = ((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY*8.0/32))) << fixMin(scaleEnergies[0],25))>>NRG_SCALE; /* can be simplified in dsp implementation */
accu1 = accu1_init;
accu2 = accu2_init;
accu_e = energies_e+3;
/* Sum up energies in first half */
for (i=start; i<border; i++) {
accu1 += (Energies[i][j]>>NRG_SCALE);
accu1 += scaleValue(Energies[i][j], -energies_e_diff[i]);
}
/* Sum up energies in second half */
for (i=border; i<stop; i++) {
accu2 += (Energies[i][j]>>NRG_SCALE);
accu2 += scaleValue(Energies[i][j], -energies_e_diff[i]);
}
/* Energy change in current band */
tmp0 = CalcLdData(accu2);
tmp1 = CalcLdData(accu1);
tmp2 = (tmp0 - tmp1 + CalcLdData(len1)-CalcLdData(len2));
delta = fixp_abs(fMult(tmp2, FL2FXCONST_DBL(0.6931471806f)));
#define LN2 FL2FXCONST_DBL(0.6931471806f) /* ln(2) */
tmp0 = fLog2(accu2, accu_e) - fLog2(accu1, accu_e);
tmp1 = fLog2((FIXP_DBL)len1, 31) - fLog2((FIXP_DBL)len2, 31);
delta = fMult(LN2, (tmp0 + tmp1));
delta = (FIXP_DBL)FDKabs( delta );
/* Weighting with amplitude ratio of this band */
result = (EnergyTotal == FL2FXCONST_DBL(0.0f))
? FL2FXCONST_DBL(0.f)
: FDKsbrEnc_LSI_divide_scale_fract( (accu1+accu2),
(EnergyTotal>>NRG_SCALE)+(FIXP_DBL)1,
(FIXP_DBL)MAXVAL_DBL >> fixMin(scaleEnergies[0],(DFRACT_BITS-1)) );
accu_e++;
accu1>>=1;
accu2>>=1;
if (accu_e & 1) {
accu_e++;
accu1>>=1;
accu2>>=1;
}
delta_sum += (FIXP_DBL)(fMult(sqrtFixp(result), delta));
delta_sum += fMult(sqrtFixp(accu1+accu2), delta);
*result_e = ((accu_e>>1) + LD_DATA_SHIFT);
}
energyTotal_e+=1; /* for a defined square result exponent, the exponent has to be even */
EnergyTotal<<=1;
delta_sum = fMult(delta_sum, invSqrtNorm2(EnergyTotal, &tmp_e));
*result_e = *result_e + (tmp_e-(energyTotal_e>>1));
return fMult(delta_sum, pos_weight);
}
@ -175,9 +231,12 @@ static FIXP_DBL spectralChange(FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FRE
*******************************************************************************
\brief Calculates total lowband energy
The return value nrgTotal is scaled by the factor (1/32.0)
The input values Energies[0] (low-band) are scaled by the factor
2^(14-*scaleEnergies[0])
The input values Energies[1] (high-band) are scaled by the factor
2^(14-*scaleEnergies[1])
\return total energy in the lowband
\return total energy in the lowband, scaled by the factor 2^19
*******************************************************************************/
static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies,
int *scaleEnergies,
@ -194,6 +253,7 @@ static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies,
int ts,k;
/* Sum up lowband energy from one frame at offset tran_off */
/* freqBandTable[LORES] has MAX_FREQ_COEFFS/2 +1 coeefs max. */
for (ts=tran_offdiv2; ts<YBufferWriteOffset; ts++) {
for (k = 0; k < freqBandTable[0]; k++) {
accu1 += Energies[ts][k] >> 6;
@ -201,12 +261,12 @@ static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies,
}
for (; ts<tran_offdiv2+(slots>>nrgSzShift); ts++) {
for (k = 0; k < freqBandTable[0]; k++) {
accu2 += Energies[ts][k] >> 6;
accu2 += Energies[ts][k] >> 9;
}
}
nrgTotal = ( (accu1 >> fixMin(scaleEnergies[0],(DFRACT_BITS-1)))
+ (accu2 >> fixMin(scaleEnergies[1],(DFRACT_BITS-1))) ) << (2);
nrgTotal = ( scaleValueSaturate(accu1, 1-scaleEnergies[0]) )
+ ( scaleValueSaturate(accu2, 4-scaleEnergies[1]) );
return(nrgTotal);
}
@ -222,21 +282,23 @@ static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies,
is 1 SBR-band. Therefore the data to be fed into the spectralChange
function is reduced.
The values EnergiesM are scaled by the factor (1/32.0) and scaleEnergies[0]
The return value nrgTotal is scaled by the factor (1/32.0)
The values EnergiesM are scaled by the factor (2^19-scaleEnergies[0]) for
slots<YBufferWriteOffset and by the factor (2^19-scaleEnergies[1]) for
slots>=YBufferWriteOffset.
\return total energy in the highband
\return total energy in the highband, scaled by factor 2^19
*******************************************************************************/
static FIXP_DBL addHighbandEnergies(FIXP_DBL **RESTRICT Energies, /*!< input */
INT *scaleEnergies,
INT YBufferWriteOffset,
FIXP_DBL EnergiesM[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS], /*!< Combined output */
UCHAR *RESTRICT freqBandTable,
INT nSfb,
INT sbrSlots,
INT timeStep)
{
INT i,j,k,slotIn,slotOut,scale;
INT i,j,k,slotIn,slotOut,scale[2];
INT li,ui;
FIXP_DBL nrgTotal;
FIXP_DBL accu = FL2FXCONST_DBL(0.0f);
@ -245,7 +307,7 @@ static FIXP_DBL addHighbandEnergies(FIXP_DBL **RESTRICT Energies, /*!< input */
combine QMF-bands to SBR-bands,
combine Left and Right channel */
for (slotOut=0; slotOut<sbrSlots; slotOut++) {
slotIn = 2*slotOut;
slotIn = timeStep*slotOut;
for (j=0; j<nSfb; j++) {
accu = FL2FXCONST_DBL(0.0f);
@ -262,19 +324,29 @@ static FIXP_DBL addHighbandEnergies(FIXP_DBL **RESTRICT Energies, /*!< input */
}
}
scale = fixMin(8,scaleEnergies[0]); /* scale energies down before add up */
/* scale energies down before add up */
scale[0] = fixMin(8,scaleEnergies[0]);
scale[1] = fixMin(8,scaleEnergies[1]);
if ((scaleEnergies[0]-1) > (DFRACT_BITS-1) )
if ((scaleEnergies[0]-scale[0]) > (DFRACT_BITS-1) || (scaleEnergies[1]-scale[0]) > (DFRACT_BITS-1))
nrgTotal = FL2FXCONST_DBL(0.0f);
else {
/* Now add all energies */
accu = FL2FXCONST_DBL(0.0f);
for (slotOut=0; slotOut<sbrSlots; slotOut++) {
for (slotOut=0; slotOut<YBufferWriteOffset; slotOut++) {
for (j=0; j<nSfb; j++) {
accu += (EnergiesM[slotOut][j] >> scale);
accu += (EnergiesM[slotOut][j] >> scale[0]);
}
}
nrgTotal = accu >> (scaleEnergies[0]-scale);
nrgTotal = accu >> (scaleEnergies[0]-scale[0]);
for (slotOut=YBufferWriteOffset; slotOut<sbrSlots; slotOut++) {
for (j=0; j<nSfb; j++) {
accu += (EnergiesM[slotOut][j] >> scale[0]);
}
}
nrgTotal = accu >> (scaleEnergies[1]-scale[1]);
}
return(nrgTotal);
@ -299,18 +371,23 @@ FDKsbrEnc_frameSplitter(FIXP_DBL **Energies,
int YBufferSzShift,
int nSfb,
int timeStep,
int no_cols)
int no_cols,
FIXP_DBL* tonality)
{
if (tran_vector[1]==0) /* no transient was detected */
{
FIXP_DBL delta;
FIXP_DBL EnergiesM[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS];
INT delta_e;
FIXP_DBL (*EnergiesM)[MAX_FREQ_COEFFS];
FIXP_DBL EnergyTotal,newLowbandEnergy,newHighbandEnergy;
INT border;
INT sbrSlots = fMultI(GetInvInt(timeStep),no_cols);
C_ALLOC_SCRATCH_START(_EnergiesM, FIXP_DBL, NUMBER_TIME_SLOTS_2304*MAX_FREQ_COEFFS)
FDK_ASSERT( sbrSlots * timeStep == no_cols );
EnergiesM = (FIXP_DBL(*)[MAX_FREQ_COEFFS])_EnergiesM;
/*
Get Lowband-energy over a range of 2 frames (Look half a frame back and ahead).
*/
@ -324,16 +401,13 @@ FDKsbrEnc_frameSplitter(FIXP_DBL **Energies,
newHighbandEnergy = addHighbandEnergies(Energies,
scaleEnergies,
YBufferWriteOffset,
EnergiesM,
freqBandTable,
nSfb,
sbrSlots,
timeStep);
if ( h_sbrTransientDetector->frameShift != 0 ) {
if (tran_vector[1]==0)
tran_vector[0] = 0;
} else
{
/* prevLowBandEnergy: Corresponds to 1 frame, starting with half a frame look-behind
newLowbandEnergy: Corresponds to 1 frame, starting in the middle of the current frame */
@ -343,23 +417,39 @@ FDKsbrEnc_frameSplitter(FIXP_DBL **Energies,
of a FIXFIX-frame with 2 envelopes. */
border = (sbrSlots+1) >> 1;
if ( (INT)EnergyTotal&0xffffffe0 && (scaleEnergies[0]<32 || scaleEnergies[1]<32) ) /* i.e. > 31 */ {
delta = spectralChange(EnergiesM,
scaleEnergies,
EnergyTotal,
nSfb,
0,
border,
sbrSlots);
YBufferWriteOffset,
sbrSlots,
&delta_e
);
} else {
delta = FL2FXCONST_DBL(0.0f);
delta_e = 0;
if (delta > (h_sbrTransientDetector->split_thr >> LD_DATA_SHIFT)) /* delta scaled by 1/64 */
/* set tonality to 0 when energy is very low, since the amplitude
resolution should then be low as well */
*tonality = FL2FXCONST_DBL(0.0f);
}
if ( fIsLessThan(h_sbrTransientDetector->split_thr_m, h_sbrTransientDetector->split_thr_e, delta, delta_e) ) {
tran_vector[0] = 1; /* Set flag for splitting */
else
} else {
tran_vector[0] = 0;
}
}
/* Update prevLowBandEnergy */
h_sbrTransientDetector->prevLowBandEnergy = newLowbandEnergy;
h_sbrTransientDetector->prevHighBandEnergy = newHighbandEnergy;
C_ALLOC_SCRATCH_END(_EnergiesM, FIXP_DBL, NUMBER_TIME_SLOTS_2304*MAX_FREQ_COEFFS)
}
}
@ -636,6 +726,7 @@ FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTran,
int
FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector,
UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */
INT frameSize,
INT sampleFreq,
sbrConfigurationPtr params,
@ -649,8 +740,8 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
{
INT totalBitrate = params->codecSettings.standardBitrate * params->codecSettings.nChannels;
INT codecBitrate = params->codecSettings.bitRate;
FIXP_DBL bitrateFactor_fix, framedur_fix;
INT scale_0, scale_1;
FIXP_DBL bitrateFactor_m, framedur_fix;
INT bitrateFactor_e, tmp_e;
FDKmemclear(h_sbrTransientDetector,sizeof(SBR_TRANSIENT_DETECTOR));
@ -658,11 +749,12 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
h_sbrTransientDetector->tran_off = tran_off;
if(codecBitrate) {
bitrateFactor_fix = fDivNorm((FIXP_DBL)totalBitrate, (FIXP_DBL)(codecBitrate<<2),&scale_0);
bitrateFactor_m = fDivNorm((FIXP_DBL)totalBitrate, (FIXP_DBL)(codecBitrate<<2),&bitrateFactor_e);
bitrateFactor_e += 2;
}
else {
bitrateFactor_fix = FL2FXCONST_DBL(1.0/4.0);
scale_0 = 0;
bitrateFactor_m = FL2FXCONST_DBL(1.0/4.0);
bitrateFactor_e = 2;
}
framedur_fix = fDivNorm(frameSize, sampleFreq);
@ -674,9 +766,13 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
FIXP_DBL tmp = framedur_fix - FL2FXCONST_DBL(0.010);
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), &tmp_e);
scale_1 = -(scale_1 + scale_0 + 2);
bitrateFactor_e = (tmp_e + bitrateFactor_e);
if(sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
bitrateFactor_e--; /* divide by 2 */
}
FDK_ASSERT(no_cols <= QMF_MAX_TIME_SLOTS);
FDK_ASSERT(no_rows <= QMF_CHANNELS);
@ -684,14 +780,8 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
h_sbrTransientDetector->no_cols = no_cols;
h_sbrTransientDetector->tran_thr = (FIXP_DBL)((params->tran_thr << (32-24-1)) / no_rows);
h_sbrTransientDetector->tran_fc = tran_fc;
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->split_thr_m = fMult(tmp, bitrateFactor_m);
h_sbrTransientDetector->split_thr_e = bitrateFactor_e;
h_sbrTransientDetector->no_rows = no_rows;
h_sbrTransientDetector->mode = params->tran_det_mode;
h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f);
@ -699,3 +789,281 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
return (0);
}
#define ENERGY_SCALING_SIZE 32
INT FDKsbrEnc_InitSbrFastTransientDetector(
HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector,
const INT time_slots_per_frame,
const INT bandwidth_qmf_slot,
const INT no_qmf_channels,
const INT sbr_qmf_1st_band
)
{
int i, e;
int buff_size;
FIXP_DBL myExp;
FIXP_DBL myExpSlot;
h_sbrFastTransientDetector->lookahead = TRAN_DET_LOOKAHEAD;
h_sbrFastTransientDetector->nTimeSlots = time_slots_per_frame;
buff_size = h_sbrFastTransientDetector->nTimeSlots + h_sbrFastTransientDetector->lookahead;
for(i=0; i< buff_size; i++) {
h_sbrFastTransientDetector->delta_energy[i] = FL2FXCONST_DBL(0.0f);
h_sbrFastTransientDetector->energy_timeSlots[i] = FL2FXCONST_DBL(0.0f);
h_sbrFastTransientDetector->lowpass_energy[i] = FL2FXCONST_DBL(0.0f);
h_sbrFastTransientDetector->transientCandidates[i] = 0;
}
FDK_ASSERT(bandwidth_qmf_slot > 0.f);
h_sbrFastTransientDetector->stopBand = fMin(TRAN_DET_STOP_FREQ/bandwidth_qmf_slot, no_qmf_channels);
h_sbrFastTransientDetector->startBand = fMin(sbr_qmf_1st_band, h_sbrFastTransientDetector->stopBand - TRAN_DET_MIN_QMFBANDS);
FDK_ASSERT(h_sbrFastTransientDetector->startBand < no_qmf_channels);
FDK_ASSERT(h_sbrFastTransientDetector->startBand < h_sbrFastTransientDetector->stopBand);
FDK_ASSERT(h_sbrFastTransientDetector->startBand > 1);
FDK_ASSERT(h_sbrFastTransientDetector->stopBand > 1);
/* the energy weighting and adding up has a headroom of 6 Bits,
so up to 64 bands can be added without potential overflow. */
FDK_ASSERT(h_sbrFastTransientDetector->stopBand - h_sbrFastTransientDetector->startBand <= 64);
/* QMF_HP_dB_SLOPE_FIX says that we want a 20 dB per 16 kHz HP filter.
The following lines map this to the QMF bandwidth. */
#define EXP_E 7 /* QMF_CHANNELS (=64) multiplications max, max. allowed sum is 0.5 */
myExp = fMultNorm(QMF_HP_dBd_SLOPE_FIX, (FIXP_DBL)bandwidth_qmf_slot, &e);
myExp = scaleValueSaturate(myExp, e+0+DFRACT_BITS-1-EXP_E);
myExpSlot = myExp;
for(i=0; i<QMF_CHANNELS; i++){
/* Calculate dBf over all qmf bands:
dBf = (10^(0.002266f/10*bw(slot)))^(band) =
= 2^(log2(10)*0.002266f/10*bw(slot)*band) =
= 2^(0.00075275f*bw(slot)*band) */
FIXP_DBL dBf_m; /* dBf mantissa */
INT dBf_e; /* dBf exponent */
INT tmp;
INT dBf_int; /* dBf integer part */
FIXP_DBL dBf_fract; /* dBf fractional part */
/* myExp*(i+1) = myExp_int - myExp_fract
myExp*(i+1) is split up here for better accuracy of CalcInvLdData(),
for its result can be split up into an integer and a fractional part */
/* Round up to next integer */
FIXP_DBL myExp_int = (myExpSlot & (FIXP_DBL)0xfe000000) + (FIXP_DBL)0x02000000;
/* This is the fractional part that needs to be substracted */
FIXP_DBL myExp_fract = myExp_int - myExpSlot;
/* Calc integer part */
dBf_int = CalcInvLdData(myExp_int);
/* The result needs to be re-scaled. The ld(myExp_int) had been scaled by EXP_E,
the CalcInvLdData expects the operand to be scaled by LD_DATA_SHIFT.
Therefore, the correctly scaled result is dBf_int^(2^(EXP_E-LD_DATA_SHIFT)),
which is dBf_int^2 */
dBf_int *= dBf_int;
/* Calc fractional part */
dBf_fract = CalcInvLdData(-myExp_fract);
/* The result needs to be re-scaled. The ld(myExp_fract) had been scaled by EXP_E,
the CalcInvLdData expects the operand to be scaled by LD_DATA_SHIFT.
Therefore, the correctly scaled result is dBf_fract^(2^(EXP_E-LD_DATA_SHIFT)),
which is dBf_fract^2 */
dBf_fract = fMultNorm(dBf_fract, dBf_fract, &tmp);
/* Get worst case scaling of multiplication result */
dBf_e = (DFRACT_BITS-1 - tmp) - CountLeadingBits(dBf_int);
/* Now multiply integer with fractional part of the result, thus resulting
in the overall accurate fractional result */
dBf_m = fMultNorm(dBf_int, dBf_fract, &e);
dBf_m = scaleValueSaturate(dBf_m, e+DFRACT_BITS-1+tmp-dBf_e);
myExpSlot += myExp;
/* Keep the results */
h_sbrFastTransientDetector->dBf_m[i] = dBf_m;
h_sbrFastTransientDetector->dBf_e[i] = dBf_e;
}
/* Make sure that dBf is greater than 1.0 (because it should be a highpass) */
/* ... */
return 0;
}
void FDKsbrEnc_fastTransientDetect(
const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector,
const FIXP_DBL *const *Energies,
const int *const scaleEnergies,
const INT YBufferWriteOffset,
UCHAR *const tran_vector
)
{
int timeSlot, band;
FIXP_DBL max_delta_energy; /* helper to store maximum energy ratio */
int max_delta_energy_scale; /* helper to store scale of maximum energy ratio */
int ind_max = 0; /* helper to store index of maximum energy ratio */
int isTransientInFrame = 0;
const int nTimeSlots = h_sbrFastTransientDetector->nTimeSlots;
const int lookahead = h_sbrFastTransientDetector->lookahead;
const int startBand = h_sbrFastTransientDetector->startBand;
const int stopBand = h_sbrFastTransientDetector->stopBand;
int * transientCandidates = h_sbrFastTransientDetector->transientCandidates;
FIXP_DBL * energy_timeSlots = h_sbrFastTransientDetector->energy_timeSlots;
int * energy_timeSlots_scale = h_sbrFastTransientDetector->energy_timeSlots_scale;
FIXP_DBL * delta_energy = h_sbrFastTransientDetector->delta_energy;
int * delta_energy_scale = h_sbrFastTransientDetector->delta_energy_scale;
const FIXP_DBL thr = TRAN_DET_THRSHLD;
const INT thr_scale = TRAN_DET_THRSHLD_SCALE;
/*reset transient info*/
tran_vector[2] = 0;
/* reset transient candidates */
FDKmemclear(transientCandidates+lookahead, nTimeSlots*sizeof(int));
for(timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) {
int i, norm;
FIXP_DBL tmpE = FL2FXCONST_DBL(0.0f);
int headroomEnSlot = DFRACT_BITS-1;
FIXP_DBL smallNRG = FL2FXCONST_DBL(1e-2f);
FIXP_DBL denominator;
INT denominator_scale;
/* determine minimum headroom of energy values for this timeslot */
for(band = startBand; band < stopBand; band++) {
int tmp_headroom = fNormz(Energies[timeSlot][band])-1;
if(tmp_headroom < headroomEnSlot){
headroomEnSlot = tmp_headroom;
}
}
for(i = 0, band = startBand; band < stopBand; band++, i++) {
/* energy is weighted by weightingfactor stored in dBf_m array */
/* dBf_m index runs from 0 to stopBand-startband */
/* energy shifted by calculated headroom for maximum precision */
FIXP_DBL weightedEnergy = fMult(Energies[timeSlot][band]<<headroomEnSlot, h_sbrFastTransientDetector->dBf_m[i]);
/* energy is added up */
/* shift by 6 to have a headroom for maximum 64 additions */
/* shift by dBf_e to handle weighting factor dependent scale factors */
tmpE += weightedEnergy >> (6 + (10 - h_sbrFastTransientDetector->dBf_e[i]));
}
/* store calculated energy for timeslot */
energy_timeSlots[timeSlot] = tmpE;
/* calculate overall scale factor for energy of this timeslot */
/* = original scale factor of energies (-scaleEnergies[0]+2*QMF_SCALE_OFFSET or -scaleEnergies[1]+2*QMF_SCALE_OFFSET */
/* depending on YBufferWriteOffset) */
/* + weighting factor scale (10) */
/* + adding up scale factor ( 6) */
/* - headroom of energy value (headroomEnSlot) */
if(timeSlot < YBufferWriteOffset){
energy_timeSlots_scale[timeSlot] = (-scaleEnergies[0]+2*QMF_SCALE_OFFSET) + (10+6) - headroomEnSlot;
} else {
energy_timeSlots_scale[timeSlot] = (-scaleEnergies[1]+2*QMF_SCALE_OFFSET) + (10+6) - headroomEnSlot;
}
/* Add a small energy to the denominator, thus making the transient
detection energy-dependent. Loud transients are being detected,
silent ones not. */
/* make sure that smallNRG does not overflow */
if ( -energy_timeSlots_scale[timeSlot-1] + 1 > 5 )
{
denominator = smallNRG;
denominator_scale = 0;
} else {
/* Leave an additional headroom of 1 bit for this addition. */
smallNRG = scaleValue(smallNRG, -(energy_timeSlots_scale[timeSlot-1] + 1));
denominator = (energy_timeSlots[timeSlot-1]>>1) + smallNRG;
denominator_scale = energy_timeSlots_scale[timeSlot-1]+1;
}
delta_energy[timeSlot] = fDivNorm(energy_timeSlots[timeSlot], denominator, &norm);
delta_energy_scale[timeSlot] = energy_timeSlots_scale[timeSlot] - denominator_scale + norm;
}
/*get transient candidates*/
/* For every timeslot, check if delta(E) exceeds the threshold. If it did,
it could potentially be marked as a transient candidate. However, the 2
slots before the current one must not be transients with an energy higher
than 1.4*E(current). If both aren't transients or if the energy of the
current timesolot is more than 1.4 times higher than the energy in the
last or the one before the last slot, it is marked as a transient.*/
FDK_ASSERT(lookahead >= 2);
for(timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) {
FIXP_DBL energy_cur_slot_weighted = fMult(energy_timeSlots[timeSlot],FL2FXCONST_DBL(1.0f/1.4f));
if( !fIsLessThan(delta_energy[timeSlot], delta_energy_scale[timeSlot], thr, thr_scale) &&
( ((transientCandidates[timeSlot-2]==0) && (transientCandidates[timeSlot-1]==0)) ||
!fIsLessThan(energy_cur_slot_weighted, energy_timeSlots_scale[timeSlot], energy_timeSlots[timeSlot-1], energy_timeSlots_scale[timeSlot-1] ) ||
!fIsLessThan(energy_cur_slot_weighted, energy_timeSlots_scale[timeSlot], energy_timeSlots[timeSlot-2], energy_timeSlots_scale[timeSlot-2] )
)
)
{
/* in case of strong transients, subsequent
* qmf slots might be recognized as transients. */
transientCandidates[timeSlot] = 1;
}
}
/*get transient with max energy*/
max_delta_energy = FL2FXCONST_DBL(0.0f);
max_delta_energy_scale = 0;
ind_max = 0;
isTransientInFrame = 0;
for(timeSlot = 0; timeSlot < nTimeSlots; timeSlot++) {
int scale = fMax(delta_energy_scale[timeSlot], max_delta_energy_scale);
if(transientCandidates[timeSlot] && ( (delta_energy[timeSlot] >> (scale - delta_energy_scale[timeSlot])) > (max_delta_energy >> (scale - max_delta_energy_scale)) ) ) {
max_delta_energy = delta_energy[timeSlot];
max_delta_energy_scale = scale;
ind_max = timeSlot;
isTransientInFrame = 1;
}
}
/*from all transient candidates take the one with the biggest energy*/
if(isTransientInFrame) {
tran_vector[0] = ind_max;
tran_vector[1] = 1;
} else {
/*reset transient info*/
tran_vector[0] = tran_vector[1] = 0;
}
/*check for transients in lookahead*/
for(timeSlot = nTimeSlots; timeSlot < nTimeSlots + lookahead; timeSlot++) {
if(transientCandidates[timeSlot]) {
tran_vector[2] = 1;
}
}
/*update buffers*/
for(timeSlot = 0; timeSlot < lookahead; timeSlot++) {
transientCandidates[timeSlot] = transientCandidates[nTimeSlots + timeSlot];
/* fixpoint stuff */
energy_timeSlots[timeSlot] = energy_timeSlots[nTimeSlots + timeSlot];
energy_timeSlots_scale[timeSlot] = energy_timeSlots_scale[nTimeSlots + timeSlot];
delta_energy[timeSlot] = delta_energy[nTimeSlots + timeSlot];
delta_energy_scale[timeSlot] = delta_energy_scale[nTimeSlots + timeSlot];
}
}

61
libSBRenc/src/tran_det.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -96,7 +96,8 @@ typedef struct
FIXP_DBL transients[QMF_MAX_TIME_SLOTS+(QMF_MAX_TIME_SLOTS/2)];
FIXP_DBL thresholds[QMF_CHANNELS];
FIXP_DBL tran_thr; /* Master threshold for transient signals */
FIXP_DBL split_thr; /* Threshold for splitting FIXFIX-frames into 2 env */
FIXP_DBL split_thr_m; /* Threshold for splitting FIXFIX-frames into 2 env */
INT split_thr_e; /* Scale for splitting threshold */
FIXP_DBL prevLowBandEnergy; /* Energy of low band */
FIXP_DBL prevHighBandEnergy; /* Energy of high band */
INT tran_fc; /* Number of lowband subbands to discard */
@ -112,6 +113,57 @@ SBR_TRANSIENT_DETECTOR;
typedef SBR_TRANSIENT_DETECTOR *HANDLE_SBR_TRANSIENT_DETECTOR;
#define TRAN_DET_LOOKAHEAD 2
#define TRAN_DET_START_FREQ 4500 /*start frequency for transient detection*/
#define TRAN_DET_STOP_FREQ 13500 /*stop frequency for transient detection*/
#define TRAN_DET_MIN_QMFBANDS 4 /* minimum qmf bands for transient detection */
#define QMF_HP_dBd_SLOPE_FIX FL2FXCONST_DBL(0.00075275f) /* 0.002266f/10 * log2(10) */
#define TRAN_DET_THRSHLD FL2FXCONST_DBL(3.2f/4.f)
#define TRAN_DET_THRSHLD_SCALE (2)
typedef struct
{
INT transientCandidates[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD];
INT nTimeSlots;
INT lookahead;
INT startBand;
INT stopBand;
FIXP_DBL dBf_m[QMF_CHANNELS];
INT dBf_e[QMF_CHANNELS];
FIXP_DBL energy_timeSlots[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD];
INT energy_timeSlots_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD];
FIXP_DBL delta_energy[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD];
INT delta_energy_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD];
FIXP_DBL lowpass_energy[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD];
INT lowpass_energy_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD];
#if defined (FTD_LOG)
FDKFILE *ftd_log;
#endif
}
FAST_TRAN_DETECTOR;
typedef FAST_TRAN_DETECTOR *HANDLE_FAST_TRAN_DET;
INT FDKsbrEnc_InitSbrFastTransientDetector(
HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector,
const INT time_slots_per_frame,
const INT bandwidth_qmf_slot,
const INT no_qmf_channels,
const INT sbr_qmf_1st_band
);
void FDKsbrEnc_fastTransientDetect(
const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector,
const FIXP_DBL *const *Energies,
const int *const scaleEnergies,
const INT YBufferWriteOffset,
UCHAR *const tran_vector
);
void
FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector,
FIXP_DBL **Energies,
@ -124,6 +176,7 @@ FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector,
int
FDKsbrEnc_InitSbrTransientDetector (HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector,
UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */
INT frameSize,
INT sampleFreq,
sbrConfigurationPtr params,
@ -145,6 +198,6 @@ FDKsbrEnc_frameSplitter(FIXP_DBL **Energies,
int YBufferSzShift,
int nSfb,
int timeStep,
int no_cols);
int no_cols,
FIXP_DBL* tonality);
#endif

44
libSYS/include/FDK_audio.h
View File

@ -2,7 +2,7 @@
/* -----------------------------------------------------------------------------------------------------------
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.
1. INTRODUCTION
@ -134,13 +134,7 @@ typedef enum
TT_MP4_LOAS = 10, /**< Audio Sync Stream. */
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 /**< */
TT_DRM = 12 /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
} TRANSPORT_TYPE;
@ -203,38 +197,22 @@ typedef enum
AOT_SAOC = 43, /**< SAOC */
AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround */
AOT_RSVD50 = 50, /**< Interim AOT for Rsvd50 */
/* 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_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_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 */
AOT_DRM_MPEG_PS = 145 /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */
} 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) \
((aot) == AOT_USAC \
|| (aot) == AOT_RSVD50)
((aot) == AOT_USAC)
#define IS_LOWDELAY(aot) \
((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
© 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.
1. INTRODUCTION
@ -99,7 +99,7 @@ amm-info@iis.fraunhofer.de
/* library info */
#define SYS_LIB_VL0 1
#define SYS_LIB_VL1 3
#define SYS_LIB_VL2 6
#define SYS_LIB_VL2 8
#define SYS_LIB_TITLE "System Integration Library"
#ifdef __ANDROID__
#define SYS_LIB_BUILD_DATE ""