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-rw-r--r--Makefile.am3
-rw-r--r--README.md9
-rwxr-xr-xbootstrap4
-rw-r--r--libAACenc/include/aacenc_lib.h6
-rw-r--r--libAACenc/src/aacenc.cpp44
-rw-r--r--libAACenc/src/aacenc_lib.cpp103
-rw-r--r--libAACenc/src/aacenc_tns.cpp21
-rw-r--r--libAACenc/src/bandwidth.cpp2
-rw-r--r--libAACenc/src/bitenc.cpp64
-rw-r--r--libAACenc/src/dyn_bits.cpp2
-rw-r--r--libAACenc/src/psy_configuration.cpp174
-rw-r--r--libAACenc/src/psy_const.h1
-rw-r--r--libAACenc/src/psy_main.cpp2
-rw-r--r--libAACenc/src/qc_main.cpp14
-rw-r--r--libFDK/src/FDK_crc.cpp94
-rw-r--r--libFDK/src/FDK_tools_rom.cpp3
-rw-r--r--libFDK/src/fft.cpp9
-rw-r--r--libFDK/src/mdct.cpp3
-rw-r--r--libMpegTPEnc/src/tpenc_dab.cpp467
-rw-r--r--libMpegTPEnc/src/tpenc_dab.h217
-rw-r--r--libMpegTPEnc/src/tpenc_lib.cpp52
-rw-r--r--libSBRenc/src/env_est.cpp1
-rw-r--r--libSBRenc/src/sbr_encoder.cpp2
-rw-r--r--libSYS/include/FDK_audio.h14
24 files changed, 1262 insertions, 49 deletions
diff --git a/Makefile.am b/Makefile.am
index 1ee2dfc..9ead8f1 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -127,7 +127,8 @@ MPEGTPENC_SRC = \
libMpegTPEnc/src/tpenc_adts.cpp \
libMpegTPEnc/src/tpenc_asc.cpp \
libMpegTPEnc/src/tpenc_latm.cpp \
- libMpegTPEnc/src/tpenc_lib.cpp
+ libMpegTPEnc/src/tpenc_lib.cpp \
+ libMpegTPEnc/src/tpenc_dab.cpp
PCMUTILS_SRC = \
libPCMutils/src/limiter.cpp \
diff --git a/README.md b/README.md
new file mode 100644
index 0000000..308f292
--- /dev/null
+++ b/README.md
@@ -0,0 +1,9 @@
+A patched version of fdk-aac with DAB+ support
+==============================================
+
+This is a modified version of fdk-aac that supports the AOTs
+required for DAB+ encoding.
+
+This library is a dependency for ODR-AudioEnc.
+
+See www.opendigitalradio.org for more
diff --git a/bootstrap b/bootstrap
new file mode 100755
index 0000000..a3394ab
--- /dev/null
+++ b/bootstrap
@@ -0,0 +1,4 @@
+#! /bin/sh
+
+autoreconf --install && \
+ echo "You can call ./configure now"
diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h
index 63c3697..de60a7a 100644
--- a/libAACenc/include/aacenc_lib.h
+++ b/libAACenc/include/aacenc_lib.h
@@ -90,7 +90,7 @@ amm-info@iis.fraunhofer.de
* \file aacenc_lib.h
* \brief FDK AAC Encoder library interface header file.
*
-\mainpage Introduction
+\page FDK AAC Encoder Library Introduction
\section Scope
@@ -923,6 +923,7 @@ typedef enum
AACENC_GRANULE_LENGTH = 0x0105, /*!< Core encoder (AAC) audio frame length in samples:
- 1024: Default configuration.
+ - 960: DRM/DAB+.
- 512: Default LD/ELD configuration.
- 480: Optional length in LD/ELD configuration. */
@@ -1028,8 +1029,9 @@ typedef enum
For AAC-ELD, the SBR information is transmitted in the ELDSpecific Config, which is part of the
AudioSpecificConfig. Therefore, the settings here will have no effect on AAC-ELD.*/
- AACENC_TPSUBFRAMES = 0x0303, /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1).
+ AACENC_TPSUBFRAMES = 0x0303, /*!< Number of sub frames in a transport frame for LOAS/LATM, DAB+ or ADTS (default 1).
- ADTS: Maximum number of sub frames restricted to 4.
+ - DAB+: Maximum number of sub frames restricted to 6.
- LOAS/LATM: Maximum number of sub frames restricted to 2.*/
AACENC_AUDIOMUXVER = 0x0304, /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, currently not implemented):
diff --git a/libAACenc/src/aacenc.cpp b/libAACenc/src/aacenc.cpp
index 5e8c08d..d7b06d0 100644
--- a/libAACenc/src/aacenc.cpp
+++ b/libAACenc/src/aacenc.cpp
@@ -87,7 +87,7 @@ amm-info@iis.fraunhofer.de
contents/description: fast aac coder functions
******************************************************************************/
-
+#include <stdio.h>
#include "aacenc.h"
#include "bitenc.h"
@@ -170,6 +170,7 @@ INT FDKaacEnc_LimitBitrate(
do {
prevBitRate = bitRate;
averageBitsPerFrame = (bitRate*(frameLength>>shift)) / (coreSamplingRate>>shift) / nSubFrames;
+ //fprintf(stderr, "FDKaacEnc_LimitBitrate(): averageBitsPerFrame=%d, prevBitRate=%d, nSubFrames=%d\n", averageBitsPerFrame, prevBitRate, bitRate);
if (pAverageBitsPerFrame != NULL) {
*pAverageBitsPerFrame = averageBitsPerFrame;
@@ -182,14 +183,18 @@ INT FDKaacEnc_LimitBitrate(
transportBits = 208;
}
+ //fprintf(stderr, "FDKaacEnc_LimitBitrate(): transportBits=%d, FDKmax(%d, %d)\n", transportBits, bitRate,
+ // ((((40 * nChannels) + transportBits) * (coreSamplingRate)) / frameLength));
bitRate = FDKmax(bitRate, ((((40 * nChannels) + transportBits) * (coreSamplingRate)) / frameLength) );
FDK_ASSERT(bitRate >= 0);
+ //fprintf(stderr, "FDKaacEnc_LimitBitrate(): FDKmin(%d, %d)\n", bitRate, ((nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN)*(coreSamplingRate>>shift)) / (frameLength>>shift));
bitRate = FDKmin(bitRate, ((nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN)*(coreSamplingRate>>shift)) / (frameLength>>shift)) ;
FDK_ASSERT(bitRate >= 0);
} while (prevBitRate != bitRate && iter++ < 3) ;
+ //fprintf(stderr, "FDKaacEnc_LimitBitrate(): bitRate=%d\n", bitRate);
return bitRate;
}
@@ -458,6 +463,36 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
return AAC_ENC_UNSUPPORTED_BITRATE;
}
+ INT superframe_size = 110*8*(config->bitRate/8000);
+ INT frames_per_superframe = 6;
+ INT staticBits = 0;
+ if((config->syntaxFlags & AC_DAB) && hTpEnc) {
+ staticBits = transportEnc_GetStaticBits(hTpEnc, 0);
+ switch(config->sampleRate) {
+ case 48000:
+ frames_per_superframe=6;
+ break;
+ case 32000:
+ frames_per_superframe=4;
+ break;
+ case 24000:
+ frames_per_superframe=3;
+ break;
+ case 16000:
+ frames_per_superframe=2;
+ break;
+ }
+
+ //config->nSubFrames = frames_per_superframe;
+ //fprintf(stderr, "DAB+ superframe size=%d\n", superframe_size);
+ config->bitRate = (superframe_size - 16*(frames_per_superframe-1) - staticBits) * 1000/120;
+ //fprintf(stderr, "DAB+ tuned bitrate=%d\n", config->bitRate);
+ config->maxBitsPerFrame = (superframe_size - 16*(frames_per_superframe-1) - staticBits) / frames_per_superframe;
+ config->maxBitsPerFrame += 7; /*padding*/
+ //config->bitreservoir=(superframe_size - 16*(frames_per_superframe-1) - staticBits - 2*8)/frames_per_superframe;
+ //fprintf(stderr, "DAB+ tuned maxBitsPerFrame=%d\n", (superframe_size - 16*(frames_per_superframe-1) - staticBits)/frames_per_superframe);
+ }
+
/* check bit rate */
if (FDKaacEnc_LimitBitrate(
@@ -489,6 +524,7 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
switch (config->framelength)
{
case 1024:
+ case 960: //TODO: DRM
if ( config->audioObjectType == AOT_ER_AAC_LD
|| config->audioObjectType == AOT_ER_AAC_ELD )
{
@@ -611,6 +647,7 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC hAacEnc,
qcInit.averageBits = (averageBitsPerFrame+7)&~7;
maxBitres = (MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff) - qcInit.averageBits;
qcInit.bitRes = (config->bitreservoir!=-1) ? FDKmin(config->bitreservoir, maxBitres) : maxBitres;
+ //fprintf(stderr, "qcInit.bitRes=%d\n", qcInit.bitRes);
qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes);
qcInit.maxBits = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits;
@@ -736,6 +773,7 @@ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc,
/* advance psychoacoustics */
for (el=0; el<cm->nElements; el++) {
ELEMENT_INFO elInfo = cm->elInfo[el];
+ //fprintf(stderr, "elInfo.elType=%d\n", elInfo.elType);
if ( (elInfo.elType == ID_SCE)
|| (elInfo.elType == ID_CPE)
@@ -896,10 +934,13 @@ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc,
/* adjust super frame bitrate */
avgTotalBits *= hAacEnc->config->nSubFrames;
+ //fprintf(stderr, "avgTotalBits=%d x %d\n", avgTotalBits, hAacEnc->config->nSubFrames);
}
/* Make first estimate of transport header overhead.
Take maximum possible frame size into account to prevent bitreservoir underrun. */
+
+ //fprintf(stderr, "avgTotalBits=%d, bitResTot=%d\n", avgTotalBits, hAacEnc->qcKernel->bitResTot);
hAacEnc->qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, avgTotalBits + hAacEnc->qcKernel->bitResTot);
@@ -951,6 +992,7 @@ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc,
/*-------------------------------------------- */
/* for ( all sub frames ) ... */
+ //fprintf(stderr, "totalBits=%d, qcOut->totalBits=%d, qcOut->totFillBits=%d\n", totalBits, qcOut->totalBits, qcOut->totFillBits);
/* write bitstream header */
transportEnc_WriteAccessUnit(
hTpEnc,
diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp
index fc58d6d..1ea9a5c 100644
--- a/libAACenc/src/aacenc_lib.cpp
+++ b/libAACenc/src/aacenc_lib.cpp
@@ -87,7 +87,7 @@ amm-info@iis.fraunhofer.de
contents/description: FDK HE-AAC Encoder interface library functions
****************************************************************************/
-
+#include <stdio.h>
#include "aacenc_lib.h"
#include "FDK_audio.h"
#include "aacenc.h"
@@ -327,7 +327,8 @@ static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig)
{
INT sbrUsed = 0;
- if ( (hAacConfig->audioObjectType==AOT_SBR) || (hAacConfig->audioObjectType==AOT_PS) )
+ if ( (hAacConfig->audioObjectType==AOT_SBR) || (hAacConfig->audioObjectType==AOT_PS)
+ || (hAacConfig->audioObjectType==AOT_DABPLUS_SBR) || (hAacConfig->audioObjectType==AOT_DABPLUS_PS) )
{
sbrUsed = 1;
}
@@ -343,7 +344,8 @@ static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType)
{
INT psUsed = 0;
- if ( (audioObjectType==AOT_PS) )
+ if ( (audioObjectType==AOT_PS)
+ || (audioObjectType==AOT_DABPLUS_PS) )
{
psUsed = 1;
}
@@ -368,7 +370,8 @@ static SBR_PS_SIGNALING getSbrSignalingMode(
sbrSignaling = SIG_IMPLICIT; /* default: implicit signaling */
}
- if ( (audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) ) {
+ if ((audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) ||
+ (audioObjectType==AOT_DABPLUS_SBR) || (audioObjectType==AOT_DABPLUS_PS) ) {
switch (transportType) {
case TT_MP4_ADIF:
case TT_MP4_ADTS:
@@ -424,7 +427,22 @@ static void FDKaacEnc_MapConfig(
cc->flags = 0;
- transport_AOT = hAacConfig->audioObjectType;
+ /* Map virtual aot to transport aot. */
+ switch (hAacConfig->audioObjectType) {
+ case AOT_DABPLUS_AAC_LC:
+ transport_AOT = AOT_AAC_LC;
+ break;
+ case AOT_DABPLUS_SBR:
+ transport_AOT = AOT_SBR;
+ cc->flags |= CC_SBR;
+ break;
+ case AOT_DABPLUS_PS:
+ transport_AOT = AOT_PS;
+ cc->flags |= CC_SBR;
+ break;
+ default:
+ transport_AOT = hAacConfig->audioObjectType;
+ }
if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) {
cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0;
@@ -465,9 +483,27 @@ static void FDKaacEnc_MapConfig(
cc->flags |= CC_IS_BASELAYER;
cc->channelMode = hAacConfig->channelMode;
- cc->nSubFrames = (hAacConfig->nSubFrames > 1 && extCfg->userTpNsubFrames == 1)
+ if(extCfg->userTpType == TT_DABPLUS && hAacConfig->nSubFrames==1) {
+ switch(hAacConfig->sampleRate) {
+ case 48000:
+ cc->nSubFrames=6;
+ break;
+ case 32000:
+ cc->nSubFrames=4;
+ break;
+ case 24000:
+ cc->nSubFrames=3;
+ break;
+ case 16000:
+ cc->nSubFrames=2;
+ break;
+ }
+ //fprintf(stderr, "hAacConfig->nSubFrames=%d hAacConfig->sampleRate=%d\n", hAacConfig->nSubFrames, hAacConfig->sampleRate);
+ } else {
+ cc->nSubFrames = (hAacConfig->nSubFrames > 1 && extCfg->userTpNsubFrames == 1)
? hAacConfig->nSubFrames
: extCfg->userTpNsubFrames;
+ }
cc->flags |= (extCfg->userTpProtection) ? CC_PROTECTION : 0;
@@ -724,6 +760,7 @@ INT aacEncoder_LimitBitrate(
FDK_ASSERT(bitRate > 0);
+ //fprintf(stderr, "aacEncoder_LimitBitrate(): bitRate=%d\n", bitRate);
return bitRate;
}
@@ -734,7 +771,7 @@ INT aacEncoder_LimitBitrate(
*
* \hAacEncoder Internal encoder config which is to be updated
* \param config User provided config (public struct)
- * \return ´returns always AAC_ENC_OK
+ * \return �returns always AAC_ENC_OK
*/
static
AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
@@ -777,10 +814,36 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
case AOT_PS:
config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_MP4_ADTS;
hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 1024;
- if (hAacConfig->framelength != 1024) {
+ if (hAacConfig->framelength != 1024 && hAacConfig->framelength != 960) {
return AACENC_INVALID_CONFIG;
}
break;
+ case AOT_DABPLUS_SBR:
+ case AOT_DABPLUS_PS:
+ hAacConfig->syntaxFlags |= ((config->userSbrEnabled) ? AC_SBR_PRESENT : 0);
+ case AOT_DABPLUS_AAC_LC:
+ config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_DABPLUS;
+ hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 960;
+ if (hAacConfig->framelength != 960) {
+ return AACENC_INVALID_CONFIG;
+ }
+ config->userTpSignaling=2;
+ if(config->userTpType == TT_DABPLUS)
+ hAacConfig->syntaxFlags |= AC_DAB;
+ break;
+#if 0
+ case AOT_ER_AAC_LC:
+ hAacConfig->epConfig = 0;
+ hAacConfig->syntaxFlags |= AC_ER;
+ hAacConfig->syntaxFlags |= ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0);
+ hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0);
+ config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS;
+ hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 1024;
+ if (hAacConfig->framelength != 1024 && hAacConfig->framelength != 960) {
+ return AACENC_INVALID_CONFIG;
+ }
+ break;
+#endif
case AOT_ER_AAC_LD:
hAacConfig->epConfig = 0;
hAacConfig->syntaxFlags |= AC_ER|AC_LD;
@@ -810,6 +873,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
break;
}
+
switch ( hAacConfig->audioObjectType ) {
case AOT_ER_AAC_LD:
case AOT_ER_AAC_ELD:
@@ -927,6 +991,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
+ //fprintf(stderr, "config->userBitrate=%d\n", config->userBitrate);
/* We need the frame length to call aacEncoder_LimitBitrate() */
hAacConfig->bitRate = aacEncoder_LimitBitrate(
NULL,
@@ -940,6 +1005,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
hAacConfig->sbrRatio,
hAacConfig->audioObjectType
);
+ //fprintf(stderr, "hAacConfig->bitRate=%d\n", hAacConfig->bitRate);
/* Configure PNS */
if ( ((hAacConfig->bitrateMode>=1) && (hAacConfig->bitrateMode<=5)) /* VBR without PNS. */
@@ -984,6 +1050,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
hAacEncoder->metaDataAllowed = 0;
}
+ //fprintf(stderr, "hAacEncoder->metaDataAllowed=%d\n", hAacEncoder->metaDataAllowed);
return err;
}
@@ -1294,8 +1361,10 @@ AACENC_ERROR aacEncOpen(
}
{ /* Get bitstream outputbuffer size */
+ UINT buffer_bytes_min = (hAacEncoder->nMaxSubFrames*hAacEncoder->nMaxAacChannels*6144)>>3;
+ buffer_bytes_min = fixMax(buffer_bytes_min, (UINT) (24 * 110)); // consider maximum DAB+ Superframe size
UINT ld_M;
- for (ld_M=1; (UINT)(1<<ld_M) < (hAacEncoder->nMaxSubFrames*hAacEncoder->nMaxAacChannels*6144)>>3; ld_M++) ;
+ for (ld_M=1; (UINT)(1<<ld_M) < buffer_bytes_min; ld_M++) ;
hAacEncoder->outBufferInBytes = (1<<ld_M); /* buffer has to be 2^n */
}
hAacEncoder->outBuffer = GetRam_bsOutbuffer();
@@ -1744,6 +1813,7 @@ AACENC_ERROR aacEncGetLibInfo(LIB_INFO *info)
/* Capability flags */
info[i].flags = 0
| CAPF_AAC_1024 | CAPF_AAC_LC
+ | CAPF_AAC_960
| CAPF_AAC_512
| CAPF_AAC_480
| CAPF_AAC_DRC
@@ -1776,16 +1846,20 @@ AACENC_ERROR aacEncoder_SetParam(
/* check if AOT matches the allocated modules */
switch ( value ) {
case AOT_PS:
+ case AOT_DRM_SBR: // Added mfeilen
+ case AOT_DABPLUS_PS:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) {
err = AACENC_INVALID_CONFIG;
goto bail;
}
case AOT_SBR:
+ case AOT_DABPLUS_SBR:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) {
err = AACENC_INVALID_CONFIG;
goto bail;
}
case AOT_AAC_LC:
+ case AOT_DABPLUS_AAC_LC:
case AOT_ER_AAC_LD:
case AOT_ER_AAC_ELD:
if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) {
@@ -1811,7 +1885,12 @@ 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 7:
case 8:
settings->userBitrateMode = value;
hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
@@ -1888,6 +1967,7 @@ AACENC_ERROR aacEncoder_SetParam(
if (settings->userFramelength != value) {
switch (value) {
case 1024:
+ case 960:
case 512:
case 480:
settings->userFramelength = value;
@@ -1927,6 +2007,7 @@ AACENC_ERROR aacEncoder_SetParam(
|| ((type==TT_MP4_LATM_MCP1) && ((flags&CAPF_LATM) && (flags&CAPF_RAWPACKETS)))
|| ((type==TT_MP4_LOAS) && (flags&CAPF_LOAS))
|| ((type==TT_MP4_RAW) && (flags&CAPF_RAWPACKETS))
+ || ((type==TT_DABPLUS) && ((flags&CAPF_DAB_AAC) && (flags&CAPF_RAWPACKETS)))
) )
{
err = AACENC_INVALID_CONFIG;
@@ -1974,7 +2055,7 @@ AACENC_ERROR aacEncoder_SetParam(
break;
case AACENC_TPSUBFRAMES:
if (settings->userTpNsubFrames != value) {
- if (! ( (value>=1) && (value<=4) ) ) {
+ if (! ( (value>=1) && (value<=6) ) ) {
err = AACENC_INVALID_CONFIG;
break;
}
diff --git a/libAACenc/src/aacenc_tns.cpp b/libAACenc/src/aacenc_tns.cpp
index 3026d69..f036518 100644
--- a/libAACenc/src/aacenc_tns.cpp
+++ b/libAACenc/src/aacenc_tns.cpp
@@ -166,6 +166,22 @@ static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab1024[] =
{ 8000, { 39, 14}}
};
+static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab960[] =
+{
+ { 96000, { 31, 9}},
+ { 88200, { 31, 9}},
+ { 64000, { 34, 10}},
+ { 48000, { 49, 14}},
+ { 44100, { 49, 14}},
+ { 32000, { 49, 14}},
+ { 24000, { 46, 15}},
+ { 22050, { 46, 14}},
+ { 16000, { 46, 15}},
+ { 12000, { 42, 15}},
+ { 11025, { 42, 15}},
+ { 8000, { 40, 15}}
+};
+
static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab480[] =
{
{ 48000, { 31, -1}},
@@ -261,6 +277,10 @@ static INT getTnsMaxBands(
int maxBandsTabSize = 0;
switch (granuleLength) {
+ case 960:
+ pMaxBandsTab = tnsMaxBandsTab960;
+ maxBandsTabSize = sizeof(tnsMaxBandsTab960)/sizeof(TNS_MAX_TAB_ENTRY);
+ break;
case 1024:
pMaxBandsTab = tnsMaxBandsTab1024;
maxBandsTabSize = sizeof(tnsMaxBandsTab1024)/sizeof(TNS_MAX_TAB_ENTRY);
@@ -386,6 +406,7 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate,
tC->lpcStopLine = pC->sfbOffset[tC->lpcStopBand];
switch (granuleLength) {
+ case 960:
case 1024:
/* TNS start line: skip lower MDCT lines to prevent artifacts due to filter mismatch */
tC->lpcStartBand[LOFILT] = (blockType == SHORT_WINDOW) ? 0 : ((sampleRate <= 8000) ? 2 : ((sampleRate < 18783) ? 4 : 8));
diff --git a/libAACenc/src/bandwidth.cpp b/libAACenc/src/bandwidth.cpp
index 6937362..af2a2cf 100644
--- a/libAACenc/src/bandwidth.cpp
+++ b/libAACenc/src/bandwidth.cpp
@@ -204,6 +204,7 @@ static INT GetBandwidthEntry(
INT bwTabSize = 0;
switch (frameLength) {
+ case 960:
case 1024:
pBwTab = bandWidthTable;
bwTabSize = sizeof(bandWidthTable)/sizeof(BANDWIDTH_TAB);
@@ -252,6 +253,7 @@ static INT GetBandwidthEntry(
chanBitRate < pBwTab[i+1].chanBitRate)
{
switch (frameLength) {
+ case 960:
case 1024:
bandwidth = (entryNo==0)
? pBwTab[i].bandWidthMono
diff --git a/libAACenc/src/bitenc.cpp b/libAACenc/src/bitenc.cpp
index 8e477aa..4552457 100644
--- a/libAACenc/src/bitenc.cpp
+++ b/libAACenc/src/bitenc.cpp
@@ -87,7 +87,7 @@ amm-info@iis.fraunhofer.de
contents/description: Bitstream encoder
******************************************************************************/
-
+#include <stdio.h>
#include "bitenc.h"
#include "bit_cnt.h"
#include "dyn_bits.h"
@@ -645,7 +645,7 @@ static INT FDKaacEnc_writeExtensionPayload( HANDLE_FDK_BITSTREAM hBitStream,
#define FILL_NIBBLE_BITS ( 4 )
INT extBitsUsed = 0;
-
+ //fprintf(stderr, "FDKaacEnc_writeExtensionPayload() extPayloadType=%d\n", extPayloadType);
if (extPayloadBits >= EXT_TYPE_BITS)
{
UCHAR fillByte = 0x00; /* for EXT_FIL and EXT_FILL_DATA */
@@ -749,7 +749,7 @@ static INT FDKaacEnc_writeDataStreamElement( HANDLE_TRANSPORTENC hTpEnc,
#define MAX_DSE_DATA_BYTES ( 510 )
INT dseBitsUsed = 0;
-
+ //fprintf(stderr, "FDKaacEnc_writeDataStreamElement() dataPayloadBytes=%d\n", dataPayloadBytes);
while (dataPayloadBytes > 0)
{
int esc_count = -1;
@@ -840,6 +840,7 @@ INT FDKaacEnc_writeExtensionData( HANDLE_TRANSPORTENC hTpEnc,
hBitStream = transportEnc_GetBitstream(hTpEnc);
}
+ //fprintf(stderr, "FDKaacEnc_writeExtensionData() pExtension->type=%d\n", pExtension->type);
if (syntaxFlags & (AC_SCALABLE|AC_ER))
{
if ( syntaxFlags & AC_DRM )
@@ -1196,6 +1197,7 @@ AAC_ENCODER_ERROR FDKaacEnc_ChannelElementWrite( HANDLE_TRANSPORTENC hTpEnc,
case esc1_hcr:
+ //TODO: DRM!
if (syntaxFlags & AC_ER_HCR)
{
error = AAC_ENC_UNKNOWN;
@@ -1301,6 +1303,23 @@ AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc,
frameBits = bitMarkUp = alignAnchor;
+ /* Write DSEs first in case of DAB */
+ for (n = 0; (n < qcOut->nExtensions) && (n < (2+2)); n++)
+ {
+ if ( (syntaxFlags & AC_DAB) &&
+ (qcOut->extension[n].type == EXT_DATA_ELEMENT) ) {
+ FDKaacEnc_writeExtensionData( hTpEnc,
+ &qcOut->extension[n],
+ 0,
+ alignAnchor,
+ syntaxFlags,
+ aot,
+ epConfig );
+ }
+
+ /* For EXT_FIL or EXT_FILL_DATA we could do an additional sanity check here */
+ }
+
/* Channel element loop */
for (i=0; i<channelMapping->nElements; i++) {
@@ -1392,7 +1411,7 @@ AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc,
if (channelElementExtensionWritten[i][n]==0)
{
- /* Write all ramaining extension payloads in element */
+ /* Write all remaining extension payloads in element */
FDKaacEnc_writeExtensionData( hTpEnc,
&qcOut->qcElement[i]->extension[n],
0,
@@ -1429,25 +1448,38 @@ AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc,
/* Add fill data / stuffing bits */
n = qcOut->nExtensions;
- qcOut->extension[n].type = EXT_FILL_DATA;
- qcOut->extension[n].nPayloadBits = qcOut->totFillBits;
- qcOut->nExtensions++;
+
+// if (!(syntaxFlags & AC_DAB)) {
+ qcOut->extension[n].type = EXT_FILL_DATA;
+ qcOut->extension[n].nPayloadBits = qcOut->totFillBits;
+ qcOut->nExtensions++;
+// } else {
+// doByteAlign = 0;
+// }
+ if (syntaxFlags & AC_DAB)
+ doByteAlign = 0;
/* Write global extension payload and fill data */
for (n = 0; (n < qcOut->nExtensions) && (n < (2+2)); n++)
{
- FDKaacEnc_writeExtensionData( hTpEnc,
- &qcOut->extension[n],
- 0,
- alignAnchor,
- syntaxFlags,
- aot,
- epConfig );
+ if ( !(syntaxFlags & AC_DAB) ||
+ ( (syntaxFlags & AC_DAB) &&
+ (qcOut->extension[n].type != EXT_DATA_ELEMENT)
+ )
+ ) {
+ FDKaacEnc_writeExtensionData( hTpEnc,
+ &qcOut->extension[n],
+ 0,
+ alignAnchor,
+ syntaxFlags,
+ aot,
+ epConfig );
+ }
/* For EXT_FIL or EXT_FILL_DATA we could do an additional sanity check here */
}
- if (!(syntaxFlags & (AC_SCALABLE|AC_ER))) {
+ if (!(syntaxFlags & (AC_SCALABLE|AC_ER|AC_DAB))) {
FDKwriteBits(hBs, ID_END, EL_ID_BITS);
}
@@ -1466,9 +1498,11 @@ AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc,
transportEnc_EndAccessUnit(hTpEnc, &frameBits);
if (frameBits != qcOut->totalBits + qcKernel->globHdrBits){
+ fprintf(stderr, "frameBits != qcOut->totalBits + qcKernel->globHdrBits: %d != %d + %d", frameBits, qcOut->totalBits, qcKernel->globHdrBits);
return AAC_ENC_WRITTEN_BITS_ERROR;
}
+ //fprintf(stderr, "ErrorStatus=%d", ErrorStatus);
return ErrorStatus;
}
diff --git a/libAACenc/src/dyn_bits.cpp b/libAACenc/src/dyn_bits.cpp
index 0c07109..3105fb4 100644
--- a/libAACenc/src/dyn_bits.cpp
+++ b/libAACenc/src/dyn_bits.cpp
@@ -416,6 +416,8 @@ static void FDKaacEnc_noiselessCounter(
break;
}
+ FDK_ASSERT(sideInfoTab != NULL);
+
sectionData->noOfSections = 0;
sectionData->huffmanBits = 0;
sectionData->sideInfoBits = 0;
diff --git a/libAACenc/src/psy_configuration.cpp b/libAACenc/src/psy_configuration.cpp
index 9a72c68..45b7d13 100644
--- a/libAACenc/src/psy_configuration.cpp
+++ b/libAACenc/src/psy_configuration.cpp
@@ -119,6 +119,161 @@ static const SFB_INFO_TAB sfbInfoTab[] = {
};
+
+
+const SFB_PARAM_LONG p_FDKaacEnc_8000_long_960 = {
+ 40,
+ { 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16,
+ 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28,
+ 28, 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 16 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_8000_short_120 = {
+ 15,
+ { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 12 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_11025_long_960 = {
+ 42,
+ { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12,
+ 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24,
+ 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_11025_short_120 = {
+ 15,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 12 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_12000_long_960 = {
+ 42,
+ { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12,
+ 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24,
+ 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_12000_short_120 = {
+ 15,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 12 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_16000_long_960 = {
+ 42,
+ { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12,
+ 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24,
+ 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_16000_short_120 = {
+ 15,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 12 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_22050_long_960 = {
+ 46,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8,
+ 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16,
+ 20, 20, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, 52, 52,
+ 64, 64, 64, 64 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_22050_short_120 = {
+ 15,
+ { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 12 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_24000_long_960 = {
+ 46,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8,
+ 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16,
+ 20, 20, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, 52, 52,
+ 64, 64, 64, 64 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_24000_short_120 = {
+ 15,
+ { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 12 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_32000_long_960 = {
+ 49,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12,
+ 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
+ 32, 32, 32, 32, 32, 32, 32, 32, 32 }
+};
+
+const SFB_PARAM_SHORT p_FDKaacEnc_32000_short_120 = {
+ 14,
+ { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 8 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_44100_long_960 = {
+ 49,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
+ 8, 8, 8, 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28,
+ 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
+ 32, 32, 32, 32, 32, 32, 32 }
+};
+
+const SFB_PARAM_SHORT p_FDKaacEnc_44100_short_120 = {
+ 14,
+ { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 8 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_48000_long_960 = {
+ 49,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
+ 8, 8, 8, 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28,
+ 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
+ 32, 32, 32, 32, 32, 32, 32 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_48000_short_120 = {
+ 14,
+ { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 8 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_64000_long_960 = {
+ 46,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 12,
+ 12, 16, 16, 16, 20, 24, 24, 28, 36, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40,
+ 40, 40, 40, 40, 40, 16 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_64000_short_120 = {
+ 12,
+ { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 28 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_88200_long_960 = {
+ 40,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12,
+ 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_88200_short_120 = {
+ 12,
+ { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 28 }
+};
+
+const SFB_PARAM_LONG p_FDKaacEnc_96000_long_960 = {
+ 40,
+ { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12,
+ 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64 }
+};
+const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_120 = {
+ 12,
+ { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 28 }
+};
+
+
+static const SFB_INFO_TAB sfbInfoTab960[] = {
+ { 8000, &p_FDKaacEnc_8000_long_960, &p_FDKaacEnc_8000_short_120},
+ {11025, &p_FDKaacEnc_11025_long_960, &p_FDKaacEnc_11025_short_120},
+ {12000, &p_FDKaacEnc_12000_long_960, &p_FDKaacEnc_12000_short_120},
+ {16000, &p_FDKaacEnc_16000_long_960, &p_FDKaacEnc_16000_short_120},
+ {22050, &p_FDKaacEnc_22050_long_960, &p_FDKaacEnc_22050_short_120},
+ {24000, &p_FDKaacEnc_24000_long_960, &p_FDKaacEnc_24000_short_120},
+ {32000, &p_FDKaacEnc_32000_long_960, &p_FDKaacEnc_32000_short_120},
+ {44100, &p_FDKaacEnc_44100_long_960, &p_FDKaacEnc_44100_short_120},
+ {48000, &p_FDKaacEnc_48000_long_960, &p_FDKaacEnc_48000_short_120},
+ {64000, &p_FDKaacEnc_64000_long_960, &p_FDKaacEnc_64000_short_120},
+ {88200, &p_FDKaacEnc_88200_long_960, &p_FDKaacEnc_88200_short_120},
+ {96000, &p_FDKaacEnc_96000_long_960, &p_FDKaacEnc_96000_short_120},
+};
+
+
/* 22050 and 24000 Hz */
static const SFB_PARAM_LONG p_22050_long_512 = {
31,
@@ -220,10 +375,13 @@ static AAC_ENCODER_ERROR FDKaacEnc_initSfbTable(LONG sampleRate, INT blockType,
*/
switch(granuleLength) {
case 1024:
- case 960:
sfbInfo = sfbInfoTab;
size = (INT)(sizeof(sfbInfoTab)/sizeof(SFB_INFO_TAB));
break;
+ case 960:
+ sfbInfo = sfbInfoTab960;
+ size = (INT)(sizeof(sfbInfoTab960)/sizeof(SFB_INFO_TAB));
+ break;
case 512:
sfbInfo = sfbInfoTabLD512;
size = sizeof(sfbInfoTabLD512);
@@ -307,9 +465,15 @@ static FIXP_DBL FDKaacEnc_BarcLineValue(INT noOfLines, INT fftLine, LONG samplin
case 1024:
center_freq = center_freq << 2; /* q13 */
break;
+ case 960:
+ center_freq = fMult(center_freq, INV480) << 3;
+ break;
case 128:
center_freq = center_freq << 5; /* q13 */
break;
+ case 120:
+ center_freq = fMult(center_freq, INV480) << 6;
+ break;
case 512:
center_freq = (fftLine * samplingFreq) << 3; // q13
break;
@@ -505,6 +669,14 @@ static void FDKaacEnc_initMinSnr(const LONG bitrate,
qperwin = qperwin - 9;
pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(480.f/512.f));
break;
+ case 960:
+ pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(960.f/1024.f));
+ qperwin = qperwin - 10;
+ break;
+ case 120:
+ pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(120.f/128.f));
+ qperwin = qperwin - 7;
+ break;
}
/* for short blocks it is assumed that more bits are available */
diff --git a/libAACenc/src/psy_const.h b/libAACenc/src/psy_const.h
index d9c9f43..15a69c9 100644
--- a/libAACenc/src/psy_const.h
+++ b/libAACenc/src/psy_const.h
@@ -96,6 +96,7 @@ amm-info@iis.fraunhofer.de
#define TRANS_FAC 8 /* encoder short long ratio */
+#define FRAME_LEN_LONG_960 (960)
#define FRAME_MAXLEN_SHORT ((1024)/TRANS_FAC)
#define FRAME_LEN_SHORT_128 ((1024)/TRANS_FAC)
diff --git a/libAACenc/src/psy_main.cpp b/libAACenc/src/psy_main.cpp
index 446c894..eeb0fa2 100644
--- a/libAACenc/src/psy_main.cpp
+++ b/libAACenc/src/psy_main.cpp
@@ -818,7 +818,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels,
&hThisPsyConf[1]->tnsConf);
}
- FDK_ASSERT(1==commonWindow); /* all checks for TNS do only work for common windows (which is always set)*/
+ FDK_ASSERT(commonWindow==1); /* all checks for TNS do only work for common windows (which is always set)*/
for(w = 0; w < nWindows[0]; w++)
{
if (isShortWindow[0])
diff --git a/libAACenc/src/qc_main.cpp b/libAACenc/src/qc_main.cpp
index 9cd73f6..1025894 100644
--- a/libAACenc/src/qc_main.cpp
+++ b/libAACenc/src/qc_main.cpp
@@ -87,7 +87,7 @@ amm-info@iis.fraunhofer.de
contents/description: Quantizing & coding
******************************************************************************/
-
+#include <stdio.h>
#include "qc_main.h"
#include "quantize.h"
#include "interface.h"
@@ -487,20 +487,24 @@ AAC_ENCODER_ERROR FDKaacEnc_AdjustBitrate(QC_STATE *RESTRICT hQC,
INT sampleRate, /* output sampling rate */
INT granuleLength) /* frame length */
{
- INT paddingOn;
+ INT paddingOn=0;
INT frameLen;
+ //fprintf(stderr, "hQC->padding.paddingRest=%d bytes! (before)\n", hQC->padding.paddingRest);
/* Do we need an extra padding byte? */
paddingOn = FDKaacEnc_framePadding(bitRate,
sampleRate,
granuleLength,
&hQC->padding.paddingRest);
+ //fprintf(stderr, "hQC->padding.paddingRest=%d bytes! (after)\n", hQC->padding.paddingRest);
frameLen = paddingOn + FDKaacEnc_calcFrameLen(bitRate,
sampleRate,
granuleLength,
FRAME_LEN_BYTES_INT);
+ //fprintf(stderr, "frameLen=%d bytes!\n", frameLen);
+
*avgTotalBits = frameLen<<3;
return AAC_ENC_OK;
@@ -1377,7 +1381,7 @@ AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption(CHANNEL_MAPPING *cm,
/* Get total consumed bits in AU */
qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + qcOut->totFillBits +
qcOut->elementExtBits + qcOut->globalExtBits;
-
+#if 1
if (qcKernel->bitrateMode==QCDATA_BR_MODE_CBR) {
/* Now we can get the exact transport bit amount, and hopefully it is equal to the estimated value */
@@ -1419,7 +1423,7 @@ AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption(CHANNEL_MAPPING *cm,
}
} /* MODE_CBR */
-
+#endif
/* Update exact number of consumed header bits. */
qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits);
@@ -1440,6 +1444,8 @@ AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption(CHANNEL_MAPPING *cm,
aot,
epConfig );
+ //fprintf(stderr, "FinalizeBitConsumption(): totFillBits=%d, qcOut->totFillBits=%d \n", totFillBits, qcOut->totFillBits);
+
/* now distribute extra fillbits and alignbits */
alignBits = 7 - (qcOut->staticBits + qcOut->usedDynBits + qcOut->elementExtBits
+ qcOut->totFillBits + qcOut->globalExtBits -1)%8;
diff --git a/libFDK/src/FDK_crc.cpp b/libFDK/src/FDK_crc.cpp
index 17d47ad..a4c6900 100644
--- a/libFDK/src/FDK_crc.cpp
+++ b/libFDK/src/FDK_crc.cpp
@@ -87,7 +87,7 @@ amm-info@iis.fraunhofer.de
contents/description: CRC calculation
******************************************************************************/
-
+#include <stdio.h>
#include "FDK_crc.h"
@@ -132,7 +132,7 @@ static const USHORT crcLookup_16_15_2_0[256] =
0x0220, 0x8225, 0x822f, 0x022a, 0x823b, 0x023e, 0x0234, 0x8231,
0x8213, 0x0216, 0x021c, 0x8219, 0x0208, 0x820d, 0x8207, 0x0202
};
-
+#if 1
/**
* \brief This table defines precalculated lookup tables for crc polynom x^16 + x^12 + x^5 + x^0.
*/
@@ -171,7 +171,83 @@ static const USHORT crcLookup_16_12_5_0[256] =
0xef1f, 0xff3e, 0xcf5d, 0xdf7c, 0xaf9b, 0xbfba, 0x8fd9, 0x9ff8,
0x6e17, 0x7e36, 0x4e55, 0x5e74, 0x2e93, 0x3eb2, 0x0ed1, 0x1ef0
};
+#else
+static const USHORT crcLookup_16_12_5_0[256] =
+{
+ 0x0000, 0x1021, 0x2042, 0x3063, 0x4084, 0x50a5, 0x60c6, 0x70e7,
+ 0x8108, 0x9129, 0xa14a, 0xb16b, 0xc18c, 0xd1ad, 0xe1ce, 0xf1ef,
+ 0x1231, 0x0210, 0x3273, 0x2252, 0x52b5, 0x4294, 0x72f7, 0x62d6,
+ 0x9339, 0x8318, 0xb37b, 0xa35a, 0xd3bd, 0xc39c, 0xf3ff, 0xe3de,
+ 0x2462, 0x3443, 0x0420, 0x1401, 0x64e6, 0x74c7, 0x44a4, 0x5485,
+ 0xa56a, 0xb54b, 0x8528, 0x9509, 0xe5ee, 0xf5cf, 0xc5ac, 0xd58d,
+ 0x3653, 0x2672, 0x1611, 0x0630, 0x76d7, 0x66f6, 0x5695, 0x46b4,
+ 0xb75b, 0xa77a, 0x9719, 0x8738, 0xf7df, 0xe7fe, 0xd79d, 0xc7bc,
+ 0x48c4, 0x58e5, 0x6886, 0x78a7, 0x0840, 0x1861, 0x2802, 0x3823,
+ 0xc9cc, 0xd9ed, 0xe98e, 0xf9af, 0x8948, 0x9969, 0xa90a, 0xb92b,
+ 0x5af5, 0x4ad4, 0x7ab7, 0x6a96, 0x1a71, 0x0a50, 0x3a33, 0x2a12,
+ 0xdbfd, 0xcbdc, 0xfbbf, 0xeb9e, 0x9b79, 0x8b58, 0xbb3b, 0xab1a,
+ 0x6ca6, 0x7c87, 0x4ce4, 0x5cc5, 0x2c22, 0x3c03, 0x0c60, 0x1c41,
+ 0xedae, 0xfd8f, 0xcdec, 0xddcd, 0xad2a, 0xbd0b, 0x8d68, 0x9d49,
+ 0x7e97, 0x6eb6, 0x5ed5, 0x4ef4, 0x3e13, 0x2e32, 0x1e51, 0x0e70,
+ 0xff9f, 0xefbe, 0xdfdd, 0xcffc, 0xbf1b, 0xaf3a, 0x9f59, 0x8f78,
+ 0x9188, 0x81a9, 0xb1ca, 0xa1eb, 0xd10c, 0xc12d, 0xf14e, 0xe16f,
+ 0x1080, 0x00a1, 0x30c2, 0x20e3, 0x5004, 0x4025, 0x7046, 0x6067,
+ 0x83b9, 0x9398, 0xa3fb, 0xb3da, 0xc33d, 0xd31c, 0xe37f, 0xf35e,
+ 0x02b1, 0x1290, 0x22f3, 0x32d2, 0x4235, 0x5214, 0x6277, 0x7256,
+ 0xb5ea, 0xa5cb, 0x95a8, 0x8589, 0xf56e, 0xe54f, 0xd52c, 0xc50d,
+ 0x34e2, 0x24c3, 0x14a0, 0x0481, 0x7466, 0x6447, 0x5424, 0x4405,
+ 0xa7db, 0xb7fa, 0x8799, 0x97b8, 0xe75f, 0xf77e, 0xc71d, 0xd73c,
+ 0x26d3, 0x36f2, 0x0691, 0x16b0, 0x6657, 0x7676, 0x4615, 0x5634,
+ 0xd94c, 0xc96d, 0xf90e, 0xe92f, 0x99c8, 0x89e9, 0xb98a, 0xa9ab,
+ 0x5844, 0x4865, 0x7806, 0x6827, 0x18c0, 0x08e1, 0x3882, 0x28a3,
+ 0xcb7d, 0xdb5c, 0xeb3f, 0xfb1e, 0x8bf9, 0x9bd8, 0xabbb, 0xbb9a,
+ 0x4a75, 0x5a54, 0x6a37, 0x7a16, 0x0af1, 0x1ad0, 0x2ab3, 0x3a92,
+ 0xfd2e, 0xed0f, 0xdd6c, 0xcd4d, 0xbdaa, 0xad8b, 0x9de8, 0x8dc9,
+ 0x7c26, 0x6c07, 0x5c64, 0x4c45, 0x3ca2, 0x2c83, 0x1ce0, 0x0cc1,
+ 0xef1f, 0xff3e, 0xcf5d, 0xdf7c, 0xaf9b, 0xbfba, 0x8fd9, 0x9ff8,
+ 0x6e17, 0x7e36, 0x4e55, 0x5e74, 0x2e93, 0x3eb2, 0x0ed1, 0x1ef0
+};
+#endif
+
+/**
+ * \brief This table defines precalculated lookup tables for crc polynom x^16 + x^14 + x^13 + x^12 + x^11 + x^5 + x^3 + x^2 + x^0.
+ */
+static const USHORT crcLookup_16_14_13_12_11_5_3_2_0[256] =
+{
+ 0x0000, 0x782f, 0xf05e, 0x8871, 0x9893, 0xe0bc, 0x68cd, 0x10e2,
+ 0x4909, 0x3126, 0xb957, 0xc178, 0xd19a, 0xa9b5, 0x21c4, 0x59eb,
+ 0x9212, 0xea3d, 0x624c, 0x1a63, 0x0a81, 0x72ae, 0xfadf, 0x82f0,
+ 0xdb1b, 0xa334, 0x2b45, 0x536a, 0x4388, 0x3ba7, 0xb3d6, 0xcbf9,
+ 0x5c0b, 0x2424, 0xac55, 0xd47a, 0xc498, 0xbcb7, 0x34c6, 0x4ce9,
+ 0x1502, 0x6d2d, 0xe55c, 0x9d73, 0x8d91, 0xf5be, 0x7dcf, 0x05e0,
+ 0xce19, 0xb636, 0x3e47, 0x4668, 0x568a, 0x2ea5, 0xa6d4, 0xdefb,
+ 0x8710, 0xff3f, 0x774e, 0x0f61, 0x1f83, 0x67ac, 0xefdd, 0x97f2,
+ 0xb816, 0xc039, 0x4848, 0x3067, 0x2085, 0x58aa, 0xd0db, 0xa8f4,
+ 0xf11f, 0x8930, 0x0141, 0x796e, 0x698c, 0x11a3, 0x99d2, 0xe1fd,
+ 0x2a04, 0x522b, 0xda5a, 0xa275, 0xb297, 0xcab8, 0x42c9, 0x3ae6,
+ 0x630d, 0x1b22, 0x9353, 0xeb7c, 0xfb9e, 0x83b1, 0x0bc0, 0x73ef,
+ 0xe41d, 0x9c32, 0x1443, 0x6c6c, 0x7c8e, 0x04a1, 0x8cd0, 0xf4ff,
+ 0xad14, 0xd53b, 0x5d4a, 0x2565, 0x3587, 0x4da8, 0xc5d9, 0xbdf6,
+ 0x760f, 0x0e20, 0x8651, 0xfe7e, 0xee9c, 0x96b3, 0x1ec2, 0x66ed,
+ 0x3f06, 0x4729, 0xcf58, 0xb777, 0xa795, 0xdfba, 0x57cb, 0x2fe4,
+ 0x0803, 0x702c, 0xf85d, 0x8072, 0x9090, 0xe8bf, 0x60ce, 0x18e1,
+ 0x410a, 0x3925, 0xb154, 0xc97b, 0xd999, 0xa1b6, 0x29c7, 0x51e8,
+ 0x9a11, 0xe23e, 0x6a4f, 0x1260, 0x0282, 0x7aad, 0xf2dc, 0x8af3,
+ 0xd318, 0xab37, 0x2346, 0x5b69, 0x4b8b, 0x33a4, 0xbbd5, 0xc3fa,
+ 0x5408, 0x2c27, 0xa456, 0xdc79, 0xcc9b, 0xb4b4, 0x3cc5, 0x44ea,
+ 0x1d01, 0x652e, 0xed5f, 0x9570, 0x8592, 0xfdbd, 0x75cc, 0x0de3,
+ 0xc61a, 0xbe35, 0x3644, 0x4e6b, 0x5e89, 0x26a6, 0xaed7, 0xd6f8,
+ 0x8f13, 0xf73c, 0x7f4d, 0x0762, 0x1780, 0x6faf, 0xe7de, 0x9ff1,
+ 0xb015, 0xc83a, 0x404b, 0x3864, 0x2886, 0x50a9, 0xd8d8, 0xa0f7,
+ 0xf91c, 0x8133, 0x0942, 0x716d, 0x618f, 0x19a0, 0x91d1, 0xe9fe,
+ 0x2207, 0x5a28, 0xd259, 0xaa76, 0xba94, 0xc2bb, 0x4aca, 0x32e5,
+ 0x6b0e, 0x1321, 0x9b50, 0xe37f, 0xf39d, 0x8bb2, 0x03c3, 0x7bec,
+ 0xec1e, 0x9431, 0x1c40, 0x646f, 0x748d, 0x0ca2, 0x84d3, 0xfcfc,
+ 0xa517, 0xdd38, 0x5549, 0x2d66, 0x3d84, 0x45ab, 0xcdda, 0xb5f5,
+ 0x7e0c, 0x0623, 0x8e52, 0xf67d, 0xe69f, 0x9eb0, 0x16c1, 0x6eee,
+ 0x3705, 0x4f2a, 0xc75b, 0xbf74, 0xaf96, 0xd7b9, 0x5fc8, 0x27e7,
+};
/*--------------- function declarations --------------------*/
@@ -207,6 +283,10 @@ void FDKcrcInit(
)
{
/* crc polynom example:
+ DAB+ FireCode:
+ x^16 + x^14 + x^13 + x^12 + x^11 + x^5 + x^3 + x^2 + x^0
+ (1) 0111 1000 0010 1101 -> 0x782d
+
x^16 + x^15 + x^2 + x^0 (1) 1000 0000 0000 0101 -> 0x8005
x^16 + x^12 + x^5 + x^0 (1) 0001 0000 0010 0001 -> 0x1021
x^8 + x^4 + x^3 + x^2 + x^0 (1) 0001 1101 -> 0x001d */
@@ -228,6 +308,9 @@ void FDKcrcInit(
case 0x1021:
hCrcInfo->pCrcLookup = crcLookup_16_12_5_0;
break;
+ case 0x782d:
+ hCrcInfo->pCrcLookup = crcLookup_16_14_13_12_11_5_3_2_0;
+ break;
case 0x001d:
default:
/* no lookup table */
@@ -372,10 +455,9 @@ static inline INT calcCrc_Bytes(
{
int i;
USHORT crc = *pCrc; /* get crc value */
-
if (hBs!=NULL) {
for (i=0; i<nBytes; i++) {
- crc = (crc<<8)^pCrcLookup[((crc>>8)^((UCHAR)FDKreadBits(hBs,8)))&0xFF];
+ crc = (crc<<8)^pCrcLookup[((crc>>8)^(FDKreadBits(hBs,8)))&0xFF];
}
}
else {
@@ -385,6 +467,7 @@ static inline INT calcCrc_Bytes(
}
*pCrc = crc; /* update crc value */
+ //fprintf(stderr, "\n\n crc[%d]=%04x\n", i, crc);
return (i);
}
@@ -420,7 +503,7 @@ static void crcCalc(
}
int bits, rBits;
- rBits = (rD->maxBits>=0) ? rD->maxBits : -rD->maxBits; /* ramaining bits */
+ rBits = (rD->maxBits>=0) ? rD->maxBits : -rD->maxBits; /* remaining bits */
if ((rD->maxBits>0) && (((INT)rD->bitBufCntBits>>3<<3)<rBits) ) {
bits = rD->bitBufCntBits;
}
@@ -454,6 +537,7 @@ static void crcCalc(
}
}
+ //fprintf(stderr, "\n\n crc=%04x\n", crc);
hCrcInfo->crcValue = crc;
}
diff --git a/libFDK/src/FDK_tools_rom.cpp b/libFDK/src/FDK_tools_rom.cpp
index f9121d1..a829485 100644
--- a/libFDK/src/FDK_tools_rom.cpp
+++ b/libFDK/src/FDK_tools_rom.cpp
@@ -2554,6 +2554,9 @@ const element_list_t * getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epCo
case AOT_AAC_LC:
case AOT_SBR:
case AOT_PS:
+ case AOT_DABPLUS_AAC_LC:
+ case AOT_DABPLUS_SBR:
+ case AOT_DABPLUS_PS:
FDK_ASSERT(epConfig == -1);
if (nChannels == 1) {
return &node_aac_sce;
diff --git a/libFDK/src/fft.cpp b/libFDK/src/fft.cpp
index 653a71a..ae3c98d 100644
--- a/libFDK/src/fft.cpp
+++ b/libFDK/src/fft.cpp
@@ -1078,7 +1078,8 @@ static inline void fft_apply_rot_vector(FIXP_DBL *RESTRICT pData, const int cl,
}
}
-#define FFT_TWO_STAGE_MACRO_ENABLE
+//FIXME:buggy for fft480
+//#define FFT_TWO_STAGE_MACRO_ENABLE
#ifdef FFT_TWO_STAGE_MACRO_ENABLE
@@ -1154,7 +1155,7 @@ static inline void fft_apply_rot_vector(FIXP_DBL *RESTRICT pData, const int cl,
#else /* FFT_TWO_STAGE_MACRO_ENABLE */
/* select either switch case of function pointer. */
-//#define FFT_TWO_STAGE_SWITCH_CASE
+#define FFT_TWO_STAGE_SWITCH_CASE
static inline void fftN2(
FIXP_DBL *pInput,
@@ -1206,7 +1207,7 @@ static inline void fftN2(
case 16: fft_16(pDst); break;
case 32: fft_32(pDst); break;
/*case 64: fft_64(pDst); break;*/
- case 128: fft_128(pDst); break;
+ /*case 128: fft_128(pDst); break;*/
}
#endif
pSrc += 2;
@@ -1245,7 +1246,7 @@ static inline void fftN2(
case 16: fft_16(pDst); break;
case 32: fft_32(pDst); break;
/*case 64: fft_64(pDst); break;*/
- case 128: fft_128(pDst); break;
+ /*case 128: fft_128(pDst); break;*/
}
#endif
diff --git a/libFDK/src/mdct.cpp b/libFDK/src/mdct.cpp
index 9a29aa1..9347a16 100644
--- a/libFDK/src/mdct.cpp
+++ b/libFDK/src/mdct.cpp
@@ -119,6 +119,9 @@ void imdct_gain(FIXP_DBL *pGain_m, int *pGain_e, int tl)
gain_e += -MDCT_OUTPUT_GAIN - log2_tl - MDCT_OUT_HEADROOM + 1;
+ FDK_ASSERT(log2_tl - 2 >= 0);
+ FDK_ASSERT(log2_tl - 2 < 8*sizeof(int));
+
/* Detect non-radix 2 transform length and add amplitude compensation factor
which cannot be included into the exponent above */
switch ( (tl) >> (log2_tl - 2) ) {
diff --git a/libMpegTPEnc/src/tpenc_dab.cpp b/libMpegTPEnc/src/tpenc_dab.cpp
new file mode 100644
index 0000000..202fecf
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_dab.cpp
@@ -0,0 +1,467 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+� Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
+ All rights reserved.
+
+ 1. INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2. COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3. NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4. DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5. CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** MPEG Audio Encoder **************************
+
+ Initial author: serge
+ contents/description: DAB Transport Headers support
+
+******************************************************************************/
+#include <stdio.h>
+#include "FDK_audio.h"
+#include "tpenc_dab.h"
+
+
+#include "tpenc_lib.h"
+#include "tpenc_asc.h"
+
+#include "common_fix.h"
+
+int dabWrite_CrcStartReg(
+ HANDLE_DAB pDab, /*!< pointer to dab stucture */
+ HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */
+ int mBits /*!< number of bits in crc region */
+ )
+{
+ //fprintf(stderr, "dabWrite_CrcStartReg(%p): bits in crc region=%d\n", hBs, mBits);
+ return ( FDKcrcStartReg(&pDab->crcInfo2, hBs, mBits) );
+}
+
+void dabWrite_CrcEndReg(
+ HANDLE_DAB pDab, /*!< pointer to dab crc info stucture */
+ HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */
+ int reg /*!< crc region */
+ )
+{
+ //fprintf(stderr, "dabWrite_CrcEndReg(%p): crc region=%d\n", hBs, reg);
+ FDKcrcEndReg(&pDab->crcInfo2, hBs, reg);
+}
+
+int dabWrite_GetHeaderBits( HANDLE_DAB hDab )
+{
+ int bits = 0;
+
+ if (hDab->currentBlock == 0) {
+ /* Static and variable header bits */
+ bits += 16; //header_firecode 16
+ bits += 8; //rfa=1, dac_rate=1, sbr_flag=1, aac_channel_mode=1, ps_flag=1, mpeg_surround_config=3
+ bits += 12 * hDab->num_raw_blocks; //au_start[1...num_aus] 12 bit AU start position markers
+
+ //4 byte alignment
+ if (hDab->dac_rate == 0 || hDab->sbr_flag == 0)
+ bits+=4;
+ //16sbr => 16 + 5 + 3 + 12*(2-1) => 36 => 40 bits 5
+ //24sbr => 16 + 5 + 3 + 12*(3-1) => 48 ok 6
+ //32sbr => 16 + 5 + 3 + 12*(4-1) => 60 => 64 bits 8
+ //48sbr => 16 + 5 + 3 + 12*(6-1) => 84 => 88 bits 11
+ }
+
+ /* Add raw data block CRC bits. Not really part of the header, put they cause bit overhead to be accounted. */
+ bits += 16;
+
+
+ return bits;
+}
+
+
+int dabWrite_CountTotalBitDemandHeader( HANDLE_DAB hDab, unsigned int streamDataLength )
+{
+ //fprintf(stderr, "streamDataLength=%d (%d bytes)\n", streamDataLength, streamDataLength >> 3);
+ return dabWrite_GetHeaderBits(hDab);
+}
+
+
+INT dabWrite_Init(HANDLE_DAB hDab, CODER_CONFIG *config)
+{
+ /* Sanity checks */
+ if((int)config->aot > 4
+ || (int)config->aot < 1 ) {
+ return -1;
+ }
+
+ /* Sanity checks DAB-specific */
+ if ( !(config->nSubFrames == 2 && config->samplingRate == 16000 && (config->flags & CC_SBR)) &&
+ !(config->nSubFrames == 3 && config->samplingRate == 24000 && (config->flags & CC_SBR)) &&
+ !(config->nSubFrames == 4 && config->samplingRate == 32000) &&
+ !(config->nSubFrames == 6 && config->samplingRate == 48000)) {
+ return -1;
+ }
+
+ hDab->dac_rate = 0;
+ hDab->aac_channel_mode=0;
+ hDab->sbr_flag = 0;
+ hDab->ps_flag = 0;
+ hDab->mpeg_surround_config=0;
+ hDab->subchannels_num=config->bitRate/8000;
+
+
+ if(config->samplingRate == 24000 || config->samplingRate == 48000)
+ hDab->dac_rate = 1;
+
+ if (config->extAOT==AOT_SBR || config->extAOT == AOT_PS)
+ hDab->sbr_flag = 1;
+
+ if(config->extAOT == AOT_PS)
+ hDab->ps_flag = 1;
+
+
+ if(config->channelMode == MODE_2)
+ hDab->aac_channel_mode = 1;
+
+ //fprintf(stderr, "hDab->dac_rate=%d\n", hDab->dac_rate);
+ //fprintf(stderr, "hDab->sbr_flag=%d\n", hDab->sbr_flag);
+ //fprintf(stderr, "hDab->ps_flag=%d\n", hDab->ps_flag);
+ //fprintf(stderr, "hDab->aac_channel_mode=%d\n", hDab->aac_channel_mode);
+ //fprintf(stderr, "hDab->subchannels_num=%d\n", hDab->subchannels_num);
+ //fprintf(stderr, "cc->nSubFrames=%d\n", config->nSubFrames);
+
+ hDab->num_raw_blocks=config->nSubFrames-1; /* 0 means 1 raw data block */
+
+ FDKcrcInit(&hDab->crcInfo, 0x1021, 0xFFFF, 16);
+ FDKcrcInit(&hDab->crcFire, 0x782d, 0, 16);
+ FDKcrcInit(&hDab->crcInfo2, 0x8005, 0xFFFF, 16);
+
+ hDab->currentBlock = 0;
+ hDab->headerBits = dabWrite_GetHeaderBits(hDab);
+
+ return 0;
+}
+
+int dabWrite_EncodeHeader(HANDLE_DAB hDab,
+ HANDLE_FDK_BITSTREAM hBitStream,
+ int buffer_fullness,
+ int frame_length)
+{
+ INT crcIndex = 0;
+
+
+ FDK_ASSERT(((frame_length+hDab->headerBits)/8)<0x2000); /*13 bit*/
+ FDK_ASSERT(buffer_fullness<0x800); /* 11 bit */
+
+ FDKcrcReset(&hDab->crcInfo);
+
+
+// fprintf(stderr, "dabWrite_EncodeHeader() hDab->currentBlock=%d, frame_length=%d, buffer_fullness=%d\n",
+// hDab->currentBlock, frame_length, buffer_fullness);
+
+// if (hDab->currentBlock == 0) {
+// //hDab->subFrameStartPrev=dabWrite_GetHeaderBits(hDab);
+// fprintf(stderr, "header bits[%d] [%d]\n", hDab->subFrameStartPrev, hDab->subFrameStartPrev >> 3);
+// FDKresetBitbuffer(hBitStream, BS_WRITER);
+// }
+
+ //hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
+// fprintf(stderr, "dabWrite_EncodeHeader() hDab->subFrameStartBit=%d [%d]\n", hDab->subFrameStartBit, hDab->subFrameStartBit >> 3);
+
+ //hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
+ /* Skip new header if this is raw data block 1..n */
+ if (hDab->currentBlock == 0)
+ {
+ FDKresetBitbuffer(hBitStream, BS_WRITER);
+// fprintf(stderr, "dabWrite_EncodeHeader() after FDKresetBitbuffer=%d [%d]\n", FDKgetValidBits(hBitStream), FDKgetValidBits(hBitStream) >> 3);
+
+ /* fixed header */
+ FDKwriteBits(hBitStream, 0, 16); //header_firecode
+ FDKwriteBits(hBitStream, 0, 1); //rfa
+ FDKwriteBits(hBitStream, hDab->dac_rate, 1);
+ FDKwriteBits(hBitStream, hDab->sbr_flag, 1);
+ FDKwriteBits(hBitStream, hDab->aac_channel_mode, 1);
+ FDKwriteBits(hBitStream, hDab->ps_flag, 1);
+ FDKwriteBits(hBitStream, hDab->mpeg_surround_config, 3);
+ /* variable header */
+ int i;
+ for(i=0; i<hDab->num_raw_blocks; i++)
+ FDKwriteBits(hBitStream, 0, 12);
+ /* padding */
+ if (hDab->dac_rate == 0 || hDab->sbr_flag == 0) {
+ FDKwriteBits(hBitStream, 0, 4);
+ }
+ } /* End of DAB header */
+
+ hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
+ FDK_ASSERT(FDKgetValidBits(hBitStream) % 8 == 0); //only aligned header
+
+// fprintf(stderr, "dabWrite_EncodeHeader() FDKgetValidBits(hBitStream)=%d [%d]\n", FDKgetValidBits(hBitStream), FDKgetValidBits(hBitStream) >> 3);
+ return 0;
+}
+
+int dabWrite_writeExtensionFillPayload(HANDLE_FDK_BITSTREAM hBitStream, int extPayloadBits)
+{
+#define EXT_TYPE_BITS ( 4 )
+#define DATA_EL_VERSION_BITS ( 4 )
+#define FILL_NIBBLE_BITS ( 4 )
+
+#define EXT_TYPE_BITS ( 4 )
+#define DATA_EL_VERSION_BITS ( 4 )
+#define FILL_NIBBLE_BITS ( 4 )
+
+ INT extBitsUsed = 0;
+ INT extPayloadType = EXT_FIL;
+ //fprintf(stderr, "FDKaacEnc_writeExtensionPayload() extPayloadType=%d\n", extPayloadType);
+ if (extPayloadBits >= EXT_TYPE_BITS)
+ {
+ UCHAR fillByte = 0x00; /* for EXT_FIL and EXT_FILL_DATA */
+
+ if (hBitStream != NULL) {
+ FDKwriteBits(hBitStream, extPayloadType, EXT_TYPE_BITS);
+ }
+ extBitsUsed += EXT_TYPE_BITS;
+
+ switch (extPayloadType) {
+ case EXT_FILL_DATA:
+ fillByte = 0xA5;
+ case EXT_FIL:
+ default:
+ if (hBitStream != NULL) {
+ int writeBits = extPayloadBits;
+ FDKwriteBits(hBitStream, 0x00, FILL_NIBBLE_BITS);
+ writeBits -= 8; /* acount for the extension type and the fill nibble */
+ while (writeBits >= 8) {
+ FDKwriteBits(hBitStream, fillByte, 8);
+ writeBits -= 8;
+ }
+ }
+ extBitsUsed += FILL_NIBBLE_BITS + (extPayloadBits & ~0x7) - 8;
+ break;
+ }
+ }
+
+ return (extBitsUsed);
+}
+
+void dabWrite_FillRawDataBlock(HANDLE_FDK_BITSTREAM hBitStream, int payloadBits)
+{
+ INT extBitsUsed = 0;
+#define EL_ID_BITS ( 3 )
+#define FILL_EL_COUNT_BITS ( 4 )
+#define FILL_EL_ESC_COUNT_BITS ( 8 )
+#define MAX_FILL_DATA_BYTES ( 269 )
+ while (payloadBits >= (EL_ID_BITS + FILL_EL_COUNT_BITS)) {
+ INT cnt, esc_count=-1, alignBits=7;
+
+ payloadBits -= EL_ID_BITS + FILL_EL_COUNT_BITS;
+ if (payloadBits >= 15*8) {
+ payloadBits -= FILL_EL_ESC_COUNT_BITS;
+ esc_count = 0; /* write esc_count even if cnt becomes smaller 15 */
+ }
+ alignBits = 0;
+
+ cnt = fixMin(MAX_FILL_DATA_BYTES, (payloadBits+alignBits)>>3);
+
+ if (cnt >= 15) {
+ esc_count = cnt - 15 + 1;
+ }
+
+ if (hBitStream != NULL) {
+ /* write bitstream */
+ FDKwriteBits(hBitStream, ID_FIL, EL_ID_BITS);
+ if (esc_count >= 0) {
+ FDKwriteBits(hBitStream, 15, FILL_EL_COUNT_BITS);
+ FDKwriteBits(hBitStream, esc_count, FILL_EL_ESC_COUNT_BITS);
+ } else {
+ FDKwriteBits(hBitStream, cnt, FILL_EL_COUNT_BITS);
+ }
+ }
+
+ extBitsUsed += EL_ID_BITS + FILL_EL_COUNT_BITS + ((esc_count>=0) ? FILL_EL_ESC_COUNT_BITS : 0);
+
+ cnt = fixMin(cnt*8, payloadBits); /* convert back to bits */
+#if 0
+ extBitsUsed += FDKaacEnc_writeExtensionPayload( hBitStream,
+ pExtension->type,
+ pExtension->pPayload,
+ cnt );
+#else
+ extBitsUsed += dabWrite_writeExtensionFillPayload(hBitStream, cnt);
+#endif
+ payloadBits -= cnt;
+ }
+}
+
+void dabWrite_EndRawDataBlock(HANDLE_DAB hDab,
+ HANDLE_FDK_BITSTREAM hBs,
+ int *pBits)
+{
+ FDK_BITSTREAM bsWriter;
+ INT crcIndex = 0;
+ USHORT crcData;
+ INT writeBits=0;
+ INT writeBitsNonLastBlock=0;
+ INT writeBitsLastBlock=0;
+#if 1
+ if (hDab->currentBlock == hDab->num_raw_blocks) {
+ //calculate byte-alignment before writing ID_FIL
+ if((FDKgetValidBits(hBs)+3) % 8){
+ writeBits = 8 - ((FDKgetValidBits(hBs)+3) % 8);
+ }
+
+ INT offset_end = hDab->subchannels_num*110*8 - 2*8 - 3;
+ writeBitsLastBlock = offset_end - FDKgetValidBits(hBs);
+ dabWrite_FillRawDataBlock(hBs, writeBitsLastBlock);
+ FDKsyncCache(hBs);
+ //fprintf(stderr, "FIL-element written=%d\n", writeBitsLastBlock);
+ writeBitsLastBlock=writeBits;
+ }
+#endif
+ FDKwriteBits(hBs, 7, 3); //finalize AU: ID_END
+ FDKsyncCache(hBs);
+ //byte-align (if ID_FIL doesn't align it).
+ if(FDKgetValidBits(hBs) % 8){
+ writeBits = 8 - (FDKgetValidBits(hBs) % 8);
+ FDKwriteBits(hBs, 0x00, writeBits);
+ FDKsyncCache(hBs);
+ }
+
+ //fake-written bits alignment for last AU
+ if (hDab->currentBlock == hDab->num_raw_blocks)
+ writeBits=writeBitsLastBlock;
+
+ INT frameLen = (FDKgetValidBits(hBs) - hDab->subFrameStartBit) >> 3;
+ //fprintf(stderr, "frame=%d, offset writeBits=%d\n", frameLen, writeBits);
+
+ FDK_ASSERT(FDKgetValidBits(hBs) % 8 == 0); //only aligned au's
+ FDK_ASSERT(hDab->subchannels_num*110*8 >= FDKgetValidBits(hBs)+2*8); //don't overlap superframe
+
+ FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
+ FDKpushFor(&bsWriter, hDab->subFrameStartBit);
+ FDKcrcReset(&hDab->crcInfo);
+ hDab->crcIndex = FDKcrcStartReg(&hDab->crcInfo, &bsWriter, 0);
+#if 0
+ if (hDab->currentBlock == hDab->num_raw_blocks) {
+ INT offset_size = hDab->subchannels_num*110*8 - 2*8 - FDKgetValidBits(hBs);
+ //fprintf(stderr, "offset_size=%d\n", offset_size >> 3);
+ FDKpushFor(hBs, offset_size);
+ }
+#endif
+
+ FDKpushFor(&bsWriter, FDKgetValidBits(hBs) - hDab->subFrameStartBit);
+ FDKcrcEndReg(&hDab->crcInfo, &bsWriter, hDab->crcIndex);
+ crcData = FDKcrcGetCRC(&hDab->crcInfo);
+ //fprintf(stderr, "crcData = %04x\n", crcData);
+ /* Write inverted CRC of current raw data block */
+ FDKwriteBits(hBs, crcData ^ 0xffff, 16);
+ FDKsyncCache(hBs);
+
+
+ /* Write distance to current data block */
+ if(hDab->currentBlock) {
+ FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
+ FDKpushFor(&bsWriter, 24 + (hDab->currentBlock-1)*12);
+ //fprintf(stderr, "FDKwriteBits() = %d\n", hDab->subFrameStartBit>>3);
+ FDKwriteBits(&bsWriter, (hDab->subFrameStartBit>>3), 12);
+ FDKsyncCache(&bsWriter);
+ }
+
+ /* Write FireCode */
+ if (hDab->currentBlock == hDab->num_raw_blocks) {
+ FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
+ FDKpushFor(&bsWriter, 16);
+
+ FDKcrcReset(&hDab->crcFire);
+ crcIndex = FDKcrcStartReg(&hDab->crcFire, &bsWriter, 72);
+ FDKpushFor(&bsWriter, 9*8); //9bytes
+ FDKcrcEndReg(&hDab->crcFire, &bsWriter, crcIndex);
+
+ crcData = FDKcrcGetCRC(&hDab->crcFire);
+ //fprintf(stderr, "Firecode: %04x\n", crcData);
+
+ FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
+ FDKwriteBits(&bsWriter, crcData, 16);
+ FDKsyncCache(&bsWriter);
+ }
+
+ if (hDab->currentBlock == 0)
+ *pBits += hDab->headerBits;
+ else
+ *pBits += 16;
+
+ *pBits += writeBits + 3; //size: ID_END + alignment
+
+ /* Correct *pBits to reflect the amount of bits of the current subframe */
+ *pBits -= hDab->subFrameStartBit;
+ /* Fixup CRC bits, since they come after each raw data block */
+
+ hDab->currentBlock++;
+ //fprintf(stderr, "dabWrite_EndRawDataBlock() *pBits=%d (%d)\n", *pBits, *pBits >> 3);
+}
+
diff --git a/libMpegTPEnc/src/tpenc_dab.h b/libMpegTPEnc/src/tpenc_dab.h
new file mode 100644
index 0000000..17b83c6
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_dab.h
@@ -0,0 +1,217 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+� Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
+ All rights reserved.
+
+ 1. INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2. COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3. NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4. DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5. CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** MPEG Audio Encoder **************************
+
+ Initial author: serge
+ contents/description: DAB Transport writer
+
+******************************************************************************/
+
+#ifndef TPENC_DAB_H
+#define TPENC_DAB_H
+
+
+
+#include "tp_data.h"
+
+#include "FDK_crc.h"
+
+typedef struct {
+ USHORT frame_length;
+ UCHAR dac_rate;
+ UCHAR aac_channel_mode;
+ UCHAR sbr_flag;
+ UCHAR ps_flag;
+ UCHAR mpeg_surround_config;
+ UCHAR num_raw_blocks;
+ UCHAR BufferFullnesStartFlag;
+ int subchannels_num;
+ int headerBits; /*!< Header bit demand for the current raw data block */
+ int currentBlock; /*!< Index of current raw data block */
+ int subFrameStartBit; /*!< Bit position where the current raw data block begins */
+ //int subFrameStartPrev; /*!< Bit position where the previous raw data block begins */
+ int crcIndex;
+ FDK_CRCINFO crcInfo;
+ FDK_CRCINFO crcFire;
+ FDK_CRCINFO crcInfo2;
+ USHORT tab[256];
+} STRUCT_DAB;
+
+typedef STRUCT_DAB *HANDLE_DAB;
+
+/**
+ * \brief Initialize DAB data structure
+ *
+ * \param hDab DAB data handle
+ * \param config a valid CODER_CONFIG struct from where the required
+ * information for the DAB header is extrated from
+ *
+ * \return 0 in case of success.
+ */
+INT dabWrite_Init(
+ HANDLE_DAB hDab,
+ CODER_CONFIG *config
+ );
+
+/**
+ * \brief Get the total bit overhead caused by DAB
+ *
+ * \hDab handle to DAB data
+ *
+ * \return Amount of additional bits required for the current raw data block
+ */
+int dabWrite_GetHeaderBits( HANDLE_DAB hDab );
+int dabWrite_CountTotalBitDemandHeader( HANDLE_DAB hDab, unsigned int streamDataLength );
+
+/**
+ * \brief Write an DAB header into the given bitstream. May not write a header
+ * in case of multiple raw data blocks.
+ *
+ * \param hDab DAB data handle
+ * \param hBitStream bitstream handle into which the DAB may be written into
+ * \param buffer_fullness the buffer fullness value for the DAB header
+ * \param the current raw data block length
+ *
+ * \return 0 in case of success.
+ */
+INT dabWrite_EncodeHeader(
+ HANDLE_DAB hDab,
+ HANDLE_FDK_BITSTREAM hBitStream,
+ int bufferFullness,
+ int frame_length
+ );
+/**
+ * \brief Finish a DAB raw data block
+ *
+ * \param hDab DAB data handle
+ * \param hBs bitstream handle into which the DAB may be written into
+ * \param pBits a pointer to a integer holding the current bitstream buffer bit count,
+ * which is corrected to the current raw data block boundary.
+ *
+ */
+void dabWrite_EndRawDataBlock(
+ HANDLE_DAB hDab,
+ HANDLE_FDK_BITSTREAM hBs,
+ int *bits
+ );
+
+
+/**
+ * \brief Start 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 pDab DAB data handle
+ * \param hBs bitstream handle of which the CRC region ends
+ * \param mBits limit of number of bits to be considered for the requested CRC region
+ *
+ * \return ID for the created region, -1 in case of an error
+ */
+int dabWrite_CrcStartReg(
+ HANDLE_DAB pDab,
+ HANDLE_FDK_BITSTREAM hBs,
+ int mBits
+ );
+
+/**
+ * \brief Ends CRC region identified by reg
+ *
+ * \param pDab DAB data handle
+ * \param hBs bitstream handle of which the CRC region ends
+ * \param reg a CRC region ID returned previously by dabWrite_CrcStartReg()
+ */
+void dabWrite_CrcEndReg(
+ HANDLE_DAB pDab,
+ HANDLE_FDK_BITSTREAM hBs,
+ int reg
+ );
+
+
+
+
+#endif /* TPENC_DAB_H */
+
diff --git a/libMpegTPEnc/src/tpenc_lib.cpp b/libMpegTPEnc/src/tpenc_lib.cpp
index 24fb32f..2c6f0c1 100644
--- a/libMpegTPEnc/src/tpenc_lib.cpp
+++ b/libMpegTPEnc/src/tpenc_lib.cpp
@@ -102,9 +102,9 @@ amm-info@iis.fraunhofer.de
#include "tpenc_adif.h"
-#include "tpenc_latm.h"
-
+#include "tpenc_dab.h"
+#include "tpenc_latm.h"
typedef struct {
int curSubFrame;
@@ -128,6 +128,8 @@ struct TRANSPORTENC
ADIF_INFO adif;
+ STRUCT_DAB dab;
+
LATM_STREAM latm;
RAWPACKETS_INFO raw;
@@ -203,6 +205,7 @@ static INT getPceRepetitionRate(
case TT_MP4_LOAS: /* PCE in ASC if chChonfig==0 */
case TT_MP4_LATM_MCP1: /* PCE in ASC if chChonfig==0 */
case TT_DRM: /* PCE not allowed in DRM */
+ case TT_DABPLUS:
default:
pceFrameCounter = -1; /* no PCE in raw_data_block */
}
@@ -221,6 +224,7 @@ static INT getPceRepetitionRate(
pceFrameCounter = headerPeriod;
break;
case TT_DRM: /* PCE not allowed in DRM */
+ case TT_DABPLUS:
default:
pceFrameCounter = -1; /* no PCE in raw_data_block */
} /* switch transportFmt */
@@ -287,6 +291,18 @@ TRANSPORTENC_ERROR transportEnc_Init(
}
break;
+ case TT_DABPLUS:
+ /* Sanity checks */
+ if ( ( hTpEnc->config.aot != AOT_AAC_LC)
+ ||(hTpEnc->config.samplesPerFrame != 960) )
+ {
+ return TRANSPORTENC_INVALID_PARAMETER;
+ }
+ if ( dabWrite_Init(&hTpEnc->writer.dab, &hTpEnc->config) != 0) {
+ return TRANSPORTENC_INVALID_PARAMETER;
+ }
+ break;
+
case TT_MP4_LOAS:
case TT_MP4_LATM_MCP0:
case TT_MP4_LATM_MCP1:
@@ -383,6 +399,17 @@ TRANSPORTENC_ERROR transportEnc_WriteAccessUnit(
frameUsedBits
);
break;
+ case TT_DABPLUS:
+ bufferFullness /= ncc; /* Number of Considered Channels */
+ bufferFullness /= 32;
+ bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */
+ dabWrite_EncodeHeader(
+ &hTp->writer.dab,
+ &hTp->bitStream,
+ bufferFullness,
+ frameUsedBits
+ );
+ break;
case TT_MP4_LOAS:
case TT_MP4_LATM_MCP0:
case TT_MP4_LATM_MCP1:
@@ -450,6 +477,9 @@ TRANSPORTENC_ERROR transportEnc_EndAccessUnit(HANDLE_TRANSPORTENC hTp, int *bits
case TT_MP4_ADTS:
adtsWrite_EndRawDataBlock(&hTp->writer.adts, &hTp->bitStream, bits);
break;
+ case TT_DABPLUS:
+ dabWrite_EndRawDataBlock(&hTp->writer.dab, &hTp->bitStream, bits);
+ break;
case TT_MP4_ADIF:
/* Substract ADIF header from AU bits, not to be considered. */
*bits -= adifWrite_GetHeaderBits(&hTp->writer.adif);
@@ -484,6 +514,14 @@ TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc, int *nbytes
*nbytes = 0;
}
break;
+ case TT_DABPLUS:
+ if (hTpEnc->writer.dab.currentBlock >= hTpEnc->writer.dab.num_raw_blocks+1) {
+ *nbytes = (FDKgetValidBits(hBs) + 7)>>3;
+ hTpEnc->writer.dab.currentBlock = 0;
+ } else {
+ *nbytes = 0;
+ }
+ break;
case TT_MP4_ADIF:
FDK_ASSERT((INT)FDKgetValidBits(hBs) >= 0);
*nbytes = (FDKgetValidBits(hBs) + 7)>>3;
@@ -518,6 +556,9 @@ INT transportEnc_GetStaticBits( HANDLE_TRANSPORTENC hTp, int auBits )
case TT_MP4_ADTS:
nbits = adtsWrite_GetHeaderBits(&hTp->writer.adts);
break;
+ case TT_DABPLUS:
+ nbits = dabWrite_CountTotalBitDemandHeader(&hTp->writer.dab, auBits);
+ break;
case TT_MP4_LOAS:
case TT_MP4_LATM_MCP0:
case TT_MP4_LATM_MCP1:
@@ -552,6 +593,9 @@ int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits)
case TT_MP4_ADTS:
crcReg = adtsWrite_CrcStartReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, mBits);
break;
+ case TT_DABPLUS:
+ crcReg = dabWrite_CrcStartReg(&hTpEnc->writer.dab, &hTpEnc->bitStream, mBits);
+ break;
default:
break;
}
@@ -565,6 +609,9 @@ void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg)
case TT_MP4_ADTS:
adtsWrite_CrcEndReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, reg);
break;
+ case TT_DABPLUS:
+ dabWrite_CrcEndReg(&hTpEnc->writer.dab, &hTpEnc->bitStream, reg);
+ break;
default:
break;
}
@@ -635,6 +682,7 @@ TRANSPORTENC_ERROR transportEnc_GetLibInfo( LIB_INFO *info )
| CAPF_LATM
| CAPF_LOAS
| CAPF_RAWPACKETS
+ | CAPF_DAB_AAC
;
return TRANSPORTENC_OK;
diff --git a/libSBRenc/src/env_est.cpp b/libSBRenc/src/env_est.cpp
index 4fcda51..06d7373 100644
--- a/libSBRenc/src/env_est.cpp
+++ b/libSBRenc/src/env_est.cpp
@@ -1147,6 +1147,7 @@ FDKsbrEnc_extractSbrEnvelope2 (
SBR_STEREO_MODE stereoMode = h_con->stereoMode;
int nChannels = h_con->nChannels;
+ FDK_ASSERT(nChannels <= MAX_NUM_CHANNELS);
const int *v_tuning;
static const int v_tuningHEAAC[6] = { 0, 2, 4, 0, 0, 0 };
diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp
index 71aab78..c7c5797 100644
--- a/libSBRenc/src/sbr_encoder.cpp
+++ b/libSBRenc/src/sbr_encoder.cpp
@@ -1939,7 +1939,7 @@ INT sbrEncoder_Init(
- if ( aot==AOT_PS ) {
+ if ( aot==AOT_PS || aot==AOT_DABPLUS_PS ) {
usePs = 1;
}
if ( aot==AOT_ER_AAC_ELD ) {
diff --git a/libSYS/include/FDK_audio.h b/libSYS/include/FDK_audio.h
index 98ded3b..9a787fb 100644
--- a/libSYS/include/FDK_audio.h
+++ b/libSYS/include/FDK_audio.h
@@ -134,7 +134,14 @@ typedef enum
TT_MP4_LOAS = 10, /**< Audio Sync Stream. */
- TT_DRM = 12 /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
+ TT_DRM = 12, /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
+ TT_DABPLUS = 13, /**< Digital Audio Broadcastong (DAB+) superframes bitstream format. */
+
+ TT_MP1_L1 = 16, /**< MPEG 1 Audio Layer 1 audio bitstream. */
+ TT_MP1_L2 = 17, /**< MPEG 1 Audio Layer 2 audio bitstream. */
+ TT_MP1_L3 = 18, /**< MPEG 1 Audio Layer 3 audio bitstream. */
+
+ TT_RSVD50 = 50 /**< */
} TRANSPORT_TYPE;
@@ -197,6 +204,11 @@ typedef enum
AOT_SAOC = 43, /**< SAOC */
AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround */
+ 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 */
+
+
/* Pseudo AOTs */
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) */
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-14 17:09:30 +0000