summaryrefslogtreecommitdiffstats
path: root/libSBRenc
diff options
context:
space:
mode:
Diffstat (limited to 'libSBRenc')
-rw-r--r--libSBRenc/include/sbr_encoder.h77
-rw-r--r--libSBRenc/src/bit_sbr.h5
-rw-r--r--libSBRenc/src/env_est.cpp3
-rw-r--r--libSBRenc/src/mh_det.cpp51
-rw-r--r--libSBRenc/src/nf_est.cpp34
-rw-r--r--libSBRenc/src/ps_main.cpp10
-rw-r--r--libSBRenc/src/sbr.h14
-rw-r--r--libSBRenc/src/sbr_def.h7
-rw-r--r--libSBRenc/src/sbr_encoder.cpp597
-rw-r--r--libSBRenc/src/sbr_rom.cpp371
-rw-r--r--libSBRenc/src/sbr_rom.h9
-rw-r--r--libSBRenc/src/sbrenc_freq_sca.cpp168
-rw-r--r--libSBRenc/src/sbrenc_freq_sca.h60
-rw-r--r--libSBRenc/src/ton_corr.cpp2
14 files changed, 808 insertions, 600 deletions
diff --git a/libSBRenc/include/sbr_encoder.h b/libSBRenc/include/sbr_encoder.h
index f8f2a5c..9d59ac7 100644
--- a/libSBRenc/include/sbr_encoder.h
+++ b/libSBRenc/include/sbr_encoder.h
@@ -101,6 +101,14 @@ amm-info@iis.fraunhofer.de
#define MAX_CODEC_FRAME_RATIO 2
#define MAX_PAYLOAD_SIZE 256
+typedef enum codecType
+{
+ CODEC_AAC=0,
+ CODEC_AACLD=1,
+ CODEC_UNSPECIFIED=99
+} CODEC_TYPE;
+
+
typedef struct
{
INT bitRate;
@@ -129,10 +137,11 @@ enum
typedef struct
{
+ CODEC_TYPE coreCoder; /*!< LC or ELD */
UINT bitrateFrom; /*!< inclusive */
UINT bitrateTo; /*!< exclusive */
- USHORT sampleRate; /*!< */
+ UINT sampleRate; /*!< */
UCHAR numChannels; /*!< */
UCHAR startFreq; /*!< bs_start_freq */
@@ -158,6 +167,7 @@ typedef struct sbrConfiguration
INT crcSbr; /*!< Flag: usage of SBR-CRC. */
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; */
/*
@@ -194,7 +204,6 @@ typedef struct sbrConfiguration
INT useSaPan; /*!< Flag: usage of SAPAN stereo. */
INT dynBwEnabled; /*!< Flag: usage of dynamic bandwidth. */
INT bParametricStereo; /*!< Flag: usage of parametric stereo coding tool. */
- INT bDownSampledSbr; /*!< Signal downsampled SBR is used. */
/*
header_extra1 configuration
@@ -214,7 +223,7 @@ typedef struct sbrConfiguration
UCHAR init_amp_res_FF;
} sbrConfiguration, *sbrConfigurationPtr ;
-typedef struct
+typedef struct SBR_CONFIG_DATA
{
UINT sbrSyntaxFlags; /**< SBR syntax flags derived from AOT. */
INT nChannels; /**< Number of channels. */
@@ -240,9 +249,7 @@ typedef struct
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;
-} SBR_CONFIG_DATA;
-
-typedef SBR_CONFIG_DATA *HANDLE_SBR_CONFIG_DATA;
+} SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA;
typedef struct {
MP4_ELEMENT_ID elType;
@@ -275,15 +282,26 @@ INT sbrEncoder_Open(
);
/**
- * \brief get closest working bit rate to specified desired bit rate for a single SBR element
- * \param bitRate the desired target bit rate
- * \param numChannels the amount of audio channels
- * \param coreSampleRate the sample rate of the core coder
- * \param the current Audio Object Type
- * \return closest working bit rate to bitRate value
+ * \brief Get closest working bitrate to specified desired
+ * bitrate for a single SBR element.
+ * \param bitRate The desired target bit rate
+ * \param numChannels The amount of audio channels
+ * \param coreSampleRate The sample rate of the core coder
+ * \param aot The current Audio Object Type
+ * \return Closest working bit rate to bitRate value
*/
UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot);
+
+/**
+ * \brief Check whether downsampled SBR single rate is possible
+ * with given audio object type.
+ * \param aot The Audio object type.
+ * \return 0 when downsampled SBR is not possible,
+ * 1 when downsampled SBR is possible.
+ */
+UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot);
+
/**
* \brief Initialize SBR Encoder instance.
* \param phSbrEncoder Pointer to a SBR Encoder instance.
@@ -294,6 +312,7 @@ UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate
* \param bufferOffset Returns the offset for the audio input data in order to do delay balancing.
* \param numChannels Input: Encoder input channels. output: core encoder channels.
* \param sampleRate Input: Encoder samplerate. output core encoder samplerate.
+ * \param downSampleFactor Input: Relation between SBR and core coder sampling rate;
* \param frameLength Input: Encoder frameLength. output core encoder frameLength.
* \param aot Input: Desired AOT. output AOT to be used after parameter checking.
* \param delay Input: core encoder delay. Output: total delay because of SBR.
@@ -303,21 +322,23 @@ UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate
* - (1-10) corresponds to header repetition rate in frames.
* \return 0 on success, and non-zero if failed.
*/
-INT sbrEncoder_Init( HANDLE_SBR_ENCODER hSbrEncoder,
- SBR_ELEMENT_INFO elInfo[(6)],
- int noElements,
- INT_PCM *inputBuffer,
- INT *bandwidth,
- INT *bufferOffset,
- INT *numChannels,
- INT *sampleRate,
- INT *frameLength,
- AUDIO_OBJECT_TYPE *aot,
- int *delay,
- int transformFactor,
- const int headerPeriod,
- ULONG statesInitFlag
- );
+INT sbrEncoder_Init(
+ HANDLE_SBR_ENCODER hSbrEncoder,
+ SBR_ELEMENT_INFO elInfo[(6)],
+ int noElements,
+ INT_PCM *inputBuffer,
+ INT *coreBandwidth,
+ INT *inputBufferOffset,
+ INT *numChannels,
+ INT *sampleRate,
+ UINT *downSampleFactor,
+ INT *frameLength,
+ AUDIO_OBJECT_TYPE aot,
+ int *delay,
+ int transformFactor,
+ const int headerPeriod,
+ ULONG statesInitFlag
+ );
/**
* \brief Do delay line buffers housekeeping. To be called after each encoded audio frame.
@@ -360,7 +381,7 @@ INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder,
* \param fSendHeaders Flag indicating that the SBR encoder should send more headers in the SBR payload or not.
* \return void
*/
-void sbrEncoder_GetHeader(SBR_ENCODER *sbrEncoder,
+void sbrEncoder_GetHeader(HANDLE_SBR_ENCODER sbrEncoder,
HANDLE_FDK_BITSTREAM hBs,
INT element_index,
int fSendHeaders);
diff --git a/libSBRenc/src/bit_sbr.h b/libSBRenc/src/bit_sbr.h
index 0629202..1ce2c1e 100644
--- a/libSBRenc/src/bit_sbr.h
+++ b/libSBRenc/src/bit_sbr.h
@@ -125,11 +125,6 @@ struct SBR_HEADER_DATA
INT freqScale;
/*
- element of sbrdata
- */
- SR_MODE sampleRateMode;
-
- /*
element of channelpairelement
*/
INT coupling;
diff --git a/libSBRenc/src/env_est.cpp b/libSBRenc/src/env_est.cpp
index 9924348..929f229 100644
--- a/libSBRenc/src/env_est.cpp
+++ b/libSBRenc/src/env_est.cpp
@@ -129,9 +129,6 @@ FDKsbrEnc_getEnergyFromCplxQmfData(FIXP_DBL **RESTRICT energyValues,/*!< the res
/* Get Scratch buffer */
C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, QMF_CHANNELS*QMF_MAX_TIME_SLOTS/2);
- FDK_ASSERT(numberBands <= QMF_CHANNELS);
- FDK_ASSERT(numberCols <= QMF_MAX_TIME_SLOTS);
-
/* Get max possible scaling of QMF data */
scale = DFRACT_BITS;
for (k=0; k<numberCols; k++) {
diff --git a/libSBRenc/src/mh_det.cpp b/libSBRenc/src/mh_det.cpp
index 739dd74..73d1b8b 100644
--- a/libSBRenc/src/mh_det.cpp
+++ b/libSBRenc/src/mh_det.cpp
@@ -1244,25 +1244,38 @@ FDKsbrEnc_InitSbrMissingHarmonicsDetector (
FDK_ASSERT(totNoEst <= MAX_NO_OF_ESTIMATES);
- switch(frameSize){
- case 2048:
- hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048;
- hs->timeSlots = NUMBER_TIME_SLOTS_2048;
- break;
- case 1920:
- hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920;
- hs->timeSlots = NUMBER_TIME_SLOTS_1920;
- break;
- case 1024:
- hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
- hs->timeSlots = 16;
- break;
- case 960:
- hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
- hs->timeSlots = 15;
- break;
- default:
- return -1;
+ if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+ {
+ switch(frameSize){
+ case 1024:
+ case 512:
+ hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+ hs->timeSlots = 16;
+ break;
+ case 960:
+ case 480:
+ hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+ hs->timeSlots = 15;
+ break;
+ default:
+ return -1;
+ }
+ } else
+ {
+ switch(frameSize){
+ case 2048:
+ case 1024:
+ hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048;
+ hs->timeSlots = NUMBER_TIME_SLOTS_2048;
+ break;
+ case 1920:
+ case 960:
+ hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920;
+ hs->timeSlots = NUMBER_TIME_SLOTS_1920;
+ break;
+ default:
+ return -1;
+ }
}
if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
diff --git a/libSBRenc/src/nf_est.cpp b/libSBRenc/src/nf_est.cpp
index 851584f..7a3c022 100644
--- a/libSBRenc/src/nf_est.cpp
+++ b/libSBRenc/src/nf_est.cpp
@@ -102,7 +102,7 @@ static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */
#define max(a,b) ( a > b ? a:b)
#endif
-#define NOISE_FLOOR_OFFSET_SCALING (3)
+#define NOISE_FLOOR_OFFSET_SCALING (4)
@@ -484,11 +484,13 @@ FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoise
tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING;
}
else {
- FDK_ASSERT(noiseFloorOffset<=8); /* because of NOISE_FLOOR_OFFSET_SCALING */
+ /* noiseFloorOffset has to be smaller than 12, because
+ the result of the calculation below must be smaller than 1:
+ (2^(noiseFloorOffset/3))*2^4<1 */
+ FDK_ASSERT(noiseFloorOffset<12);
- /* Assumes the noise floor offset in tuning table are in q31 */
- /* Currently the table contains only 0 for noise floor offset */
- /* Change the qformat here when non-zero values would be filled */
+ /* Assumes the noise floor offset in tuning table are in q31 */
+ /* Change the qformat here when non-zero values would be filled */
exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp);
tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp);
tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING);
@@ -527,24 +529,30 @@ FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoise
h_sbrNoiseFloorEstimate->noNoiseBands = 1;
}
else{
- /*
- * Calculate number of noise bands 1,2 or 3 bands/octave
+ /*
+ * Calculate number of noise bands 1,2 or 3 bands/octave
********************************************************/
FIXP_DBL tmp, ratio, lg2;
- INT ratio_e, qlg2;
+ INT ratio_e, qlg2, nNoiseBands;
ratio = fDivNorm(k2, kx, &ratio_e);
lg2 = fLog2(ratio, ratio_e, &qlg2);
tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2);
tmp = scaleValue(tmp, qlg2-23);
- h_sbrNoiseFloorEstimate->noNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
+ nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
+
+
+ if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) {
+ nNoiseBands = MAX_NUM_NOISE_COEFFS;
+ }
+
+ if( nNoiseBands == 0 ) {
+ nNoiseBands = 1;
+ }
- if (h_sbrNoiseFloorEstimate->noNoiseBands > MAX_NUM_NOISE_COEFFS)
- h_sbrNoiseFloorEstimate->noNoiseBands = MAX_NUM_NOISE_COEFFS;
+ h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands;
- if( h_sbrNoiseFloorEstimate->noNoiseBands==0)
- h_sbrNoiseFloorEstimate->noNoiseBands=1;
}
diff --git a/libSBRenc/src/ps_main.cpp b/libSBRenc/src/ps_main.cpp
index 76b759a..ab183e2 100644
--- a/libSBRenc/src/ps_main.cpp
+++ b/libSBRenc/src/ps_main.cpp
@@ -315,7 +315,7 @@ static FDK_PSENC_ERROR DownmixPSQmfData(
}
else {
int n, k;
- C_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, QMF_CHANNELS*2);
+ C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, 2*QMF_CHANNELS)
/* define scalings */
int dynQmfScale = fixMax(0, hParametricStereo->dmxScale-1); /* scale one bit more for addition of left and right */
@@ -400,8 +400,7 @@ static FDK_PSENC_ERROR DownmixPSQmfData(
*qmfScale = -downmixScale + 7;
- C_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, QMF_CHANNELS*2);
-
+ C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, 2*QMF_CHANNELS)
{
const INT noQmfSlots2 = hParametricStereo->noQmfSlots>>1;
@@ -475,10 +474,9 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing(
)
{
FDK_PSENC_ERROR error = PSENC_OK;
- INT noQmfBands = hParametricStereo->noQmfBands;
INT psQmfScale[MAX_PS_CHANNELS] = {0};
int psCh, i;
- C_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, QMF_CHANNELS*4);
+ C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, 4*QMF_CHANNELS)
for (psCh = 0; psCh<MAX_PS_CHANNELS; psCh ++) {
@@ -507,7 +505,7 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing(
} /* for psCh */
- C_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, QMF_CHANNELS*4);
+ C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, 4*QMF_CHANNELS)
/* find best scaling in new QMF and Hybrid data */
psFindBestScaling( hParametricStereo,
diff --git a/libSBRenc/src/sbr.h b/libSBRenc/src/sbr.h
index eed847f..cfc595e 100644
--- a/libSBRenc/src/sbr.h
+++ b/libSBRenc/src/sbr.h
@@ -105,7 +105,8 @@ amm-info@iis.fraunhofer.de
/* SBR bitstream delay */
#define DELAY_FRAMES 2
-typedef struct {
+
+typedef struct SBR_CHANNEL {
struct ENV_CHANNEL hEnvChannel;
//INT_PCM *pDSOutBuffer; /**< Pointer to downsampled audio output of SBR encoder */
DOWNSAMPLER downSampler;
@@ -113,7 +114,7 @@ typedef struct {
} SBR_CHANNEL;
typedef SBR_CHANNEL* HANDLE_SBR_CHANNEL;
-typedef struct {
+typedef struct SBR_ELEMENT {
HANDLE_SBR_CHANNEL sbrChannel[2];
QMF_FILTER_BANK *hQmfAnalysis[2];
SBR_CONFIG_DATA sbrConfigData;
@@ -126,10 +127,9 @@ typedef struct {
UCHAR payloadDelayLine[1+DELAY_FRAMES][MAX_PAYLOAD_SIZE];
UINT payloadDelayLineSize[1+DELAY_FRAMES]; /* Sizes in bits */
-} SBR_ELEMENT;
-typedef SBR_ELEMENT* HANDLE_SBR_ELEMENT;
+} SBR_ELEMENT, *HANDLE_SBR_ELEMENT;
-struct SBR_ENCODER
+typedef struct SBR_ENCODER
{
HANDLE_SBR_ELEMENT sbrElement[(6)];
HANDLE_SBR_CHANNEL pSbrChannel[(6)];
@@ -142,6 +142,7 @@ struct SBR_ENCODER
int bufferOffset; /* Offset for SBR parameter extraction in time domain input buffer. */
int downsampledOffset; /* Offset of downsampled/mixed output for core encoder. */
int downmixSize; /* Size in samples of downsampled/mixed output for core encoder. */
+ INT downSampleFactor; /* Sampling rate relation between the SBR and the core encoder. */
int fTimeDomainDownsampling; /* Flag signalling time domain downsampling instead of QMF downsampling. */
int nBitstrDelay; /* Amount of SBR frames to be delayed in bitstream domain. */
INT estimateBitrate; /* estimate bitrate of SBR encoder */
@@ -158,7 +159,8 @@ struct SBR_ENCODER
INT maxChannels;
INT supportPS;
-} ;
+
+} SBR_ENCODER;
#endif /* __SBR_H */
diff --git a/libSBRenc/src/sbr_def.h b/libSBRenc/src/sbr_def.h
index 97907c6..8b7cfc6 100644
--- a/libSBRenc/src/sbr_def.h
+++ b/libSBRenc/src/sbr_def.h
@@ -270,13 +270,6 @@ INVF_MODE;
typedef enum
{
- SINGLE_RATE,
- DUAL_RATE
-}
-SR_MODE;
-
-typedef enum
-{
FREQ_RES_LOW = 0,
FREQ_RES_HIGH
}
diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp
index d7ecae5..2d5bce2 100644
--- a/libSBRenc/src/sbr_encoder.cpp
+++ b/libSBRenc/src/sbr_encoder.cpp
@@ -83,7 +83,7 @@ amm-info@iis.fraunhofer.de
/*************************** Fraunhofer IIS FDK Tools ***********************
- Author(s): Andreas Ehret
+ Author(s): Andreas Ehret, Tobias Chalupka
Description: SBR encoder top level processing.
******************************************************************************/
@@ -103,7 +103,7 @@ amm-info@iis.fraunhofer.de
#define SBRENCODER_LIB_VL0 3
#define SBRENCODER_LIB_VL1 3
-#define SBRENCODER_LIB_VL2 2
+#define SBRENCODER_LIB_VL2 3
@@ -119,34 +119,30 @@ amm-info@iis.fraunhofer.de
(core2sbr delay ) ds (read, core and ds area)
*/
-#define DOWN_SMPL_FAC (2)
+#define SFB(dwnsmp) (32 << (dwnsmp-1)) /* SBR Frequency bands: 64 for dual-rate, 32 for single-rate */
+#define STS(fl) (((fl)==1024)?32:30) /* SBR Time Slots: 32 for core frame length 1024, 30 for core frame length 960 */
-#define SFL(fl) (fl*DOWN_SMPL_FAC) /* SBR frame length (hardcoded to downsample factor of 2) */
-#define STS(fl) (SFL(fl)/64) /* SBR Time Slots */
-
-#define DELAY_QMF_ANA (640 - 64) /* Full bandwidth */
-#define DELAY_QMF_ANAELD (32)
-#define DELAY_HYB_ANA (10*64) /* + 0.5 */
-#define DELAY_HYB_SYN (6*64 - 32)
-#define DELAY_QMF_SYNELD (32)
-#define DELAY_DEC_QMF (6*64) /* Decoder QMF overlap */
-#define DELAY_QMF_SYN (2) /* NO_POLY/2 */
-#define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */
+#define DELAY_QMF_ANA(dwnsmp) ((320<<((dwnsmp)-1)) - (32<<((dwnsmp)-1))) /* Full bandwidth */
+#define DELAY_HYB_ANA (10*64) /* + 0.5 */ /* */
+#define DELAY_HYB_SYN (6*64 - 32) /* */
+#define DELAY_QMF_POSTPROC(dwnsmp) (32*(dwnsmp)) /* QMF postprocessing delay */
+#define DELAY_DEC_QMF(dwnsmp) (6 * SFB(dwnsmp) ) /* Decoder QMF overlap */
+#define DELAY_QMF_SYN (2) /* NO_POLY/2=2.5, rounded down to 2 */
+#define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */
/* Delay in QMF paths */
-#define DELAY_SBR(fl) (DELAY_QMF_ANA + (64*STS(fl)-1) + DELAY_QMF_SYN)
-#define DELAY_PS(fl) (DELAY_QMF_ANA + DELAY_HYB_ANA + DELAY_DEC_QMF + (64*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN)
-#define DELAY_ELDSBR(fl) (DELAY_QMF_ANAELD + (((fl)+((fl)/2))*2 - 1) + DELAY_QMF_SYNELD)
+#define DELAY_SBR(fl,dwnsmp) (DELAY_QMF_ANA(dwnsmp) + (SFB(dwnsmp)*STS(fl) - 1) + DELAY_QMF_SYN)
+#define DELAY_PS(fl,dwnsmp) (DELAY_QMF_ANA(dwnsmp) + DELAY_HYB_ANA + DELAY_DEC_QMF(dwnsmp) + (SFB(dwnsmp)*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN)
+#define DELAY_ELDSBR(fl,dwnsmp) ( ( ((fl)/2)*(dwnsmp) ) - 1 + DELAY_QMF_POSTPROC(dwnsmp) )
/* Delay differences for SBR and SBR+PS */
- #define MAX_DS_FILTER_DELAY (34) /* the additional max downsampler filter delay (source fs) */
-#define DELAY_AAC2SBR(fl) ((/*RESAMPLER +*/ /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA + DELAY_DEC_QMF + DELAY_QMF_SYN) - DELAY_SBR(fl)) /* 1537 */
-#define DELAY_ELD2SBR(fl) ((/*RESAMPLER +*/ /*(DELAY_ELD(fl)*2) + */ DELAY_QMF_ANAELD + DELAY_QMF_SYNELD) - DELAY_ELDSBR(fl))
-
-#define DELAY_AAC2PS(fl) ((DELAY_QMF_ANA + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2)*/ + DELAY_QMF_ANA + DELAY_DEC_QMF + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl)) /* 2048 - 463*2 */
+#define MAX_DS_FILTER_DELAY (5) /* the additional max downsampler filter delay (source fs) */
+#define DELAY_AAC2SBR(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_QMF_SYN) - DELAY_SBR((fl),(dwnsmp)))
+#define DELAY_ELD2SBR(fl,dwnsmp) ((DELAY_QMF_POSTPROC(dwnsmp)) - DELAY_ELDSBR(fl,dwnsmp))
+#define DELAY_AAC2PS(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl,dwnsmp)) /* 2048 - 463*2 */
-/* Assumption: that the sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */
-#define MAX_SAMPLE_DELAY (DELAY_AAC2SBR(1024) + MAX_DS_FILTER_DELAY)
+/* Assumption: The sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */
+#define MAX_SAMPLE_DELAY (DELAY_AAC2SBR(1024,2) + MAX_DS_FILTER_DELAY) /* maximum delay: frame length of 1024 and dual-rate sbr */
/***************************************************************************/
@@ -172,41 +168,39 @@ getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */
UINT *pBitRateClosest
)
{
- int i, paramSetTop, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0;
+ int i, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0;
UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE;
+ int isforThisCodec=0;
- FDK_ASSERT(SBRENC_TUNING_SIZE == sizeof(sbrTuningTable)/sizeof(sbrTuningTable[0]));
-
- if (core == AOT_ER_AAC_ELD) {
- paramSetTop = SBRENC_TUNING_SIZE;
- i = SBRENC_AACLC_TUNING_SIZE;
- } else {
- paramSetTop = SBRENC_AACLC_TUNING_SIZE;
- i = 0;
- }
+ #define isForThisCore(i) \
+ ( ( sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD ) || \
+ ( sbrTuningTable[i].coreCoder == CODEC_AAC && core != AOT_ER_AAC_ELD ) )
- for (; i < paramSetTop ; i++) {
- if ( numChannels == sbrTuningTable [i].numChannels
- && sampleRate == sbrTuningTable [i].sampleRate )
+ for (i=0; i < sbrTuningTableSize ; i++) {
+ if ( isForThisCore(i) ) /* tuning table is for this core codec */
{
- found = 1;
- if ((bitrate >= sbrTuningTable [i].bitrateFrom) &&
- (bitrate < sbrTuningTable [i].bitrateTo)) {
- bitRateClosestLower = bitrate;
- bitRateClosestUpper = bitrate;
- //FDKprintf("entry %d\n", i);
- return i ;
- } else {
- if ( sbrTuningTable [i].bitrateFrom > bitrate ) {
- if (sbrTuningTable [i].bitrateFrom < bitRateClosestLower) {
- bitRateClosestLower = sbrTuningTable [i].bitrateFrom;
- bitRateClosestLowerIndex = i;
+ if ( numChannels == sbrTuningTable [i].numChannels
+ && sampleRate == sbrTuningTable [i].sampleRate )
+ {
+ found = 1;
+ if ((bitrate >= sbrTuningTable [i].bitrateFrom) &&
+ (bitrate < sbrTuningTable [i].bitrateTo)) {
+ bitRateClosestLower = bitrate;
+ bitRateClosestUpper = bitrate;
+ //FDKprintf("entry %d\n", i);
+ return i ;
+ } else {
+ if ( sbrTuningTable [i].bitrateFrom > bitrate ) {
+ if (sbrTuningTable [i].bitrateFrom < bitRateClosestLower) {
+ bitRateClosestLower = sbrTuningTable [i].bitrateFrom;
+ bitRateClosestLowerIndex = i;
+ }
}
- }
- if ( sbrTuningTable [i].bitrateTo <= bitrate ) {
- if (sbrTuningTable [i].bitrateTo > bitRateClosestUpper) {
- bitRateClosestUpper = sbrTuningTable [i].bitrateTo-1;
- bitRateClosestUpperIndex = i;
+ if ( sbrTuningTable [i].bitrateTo <= bitrate ) {
+ if (sbrTuningTable [i].bitrateTo > bitRateClosestUpper) {
+ bitRateClosestUpper = sbrTuningTable [i].bitrateTo-1;
+ bitRateClosestUpperIndex = i;
+ }
}
}
}
@@ -215,7 +209,7 @@ getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */
if (pBitRateClosest != NULL)
{
- /* Is there was at least one matching tuning entry found then pick the least distance bit rate */
+ /* If there was at least one matching tuning entry found then pick the least distance bit rate */
if (found)
{
int distanceUpper=DISTANCE_CEIL_VALUE, distanceLower=DISTANCE_CEIL_VALUE;
@@ -295,6 +289,52 @@ getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){
return INVALID_TABLE_IDX;
}
+/***************************************************************************/
+/*!
+
+ \brief In case of downsampled SBR we may need to lower the stop freq
+ of a tuning setting to fit into the lower half of the
+ spectrum ( which is sampleRate/4 )
+
+ \return the adapted stop frequency index (-1 -> error)
+
+ \ingroup SbrEncCfg
+
+****************************************************************************/
+static INT
+FDKsbrEnc_GetDownsampledStopFreq (
+ const INT sampleRateCore,
+ const INT startFreq,
+ INT stopFreq,
+ const INT downSampleFactor
+ )
+{
+ INT maxStopFreqRaw = sampleRateCore / 2;
+ INT startBand, stopBand;
+ HANDLE_ERROR_INFO err;
+
+ while (stopFreq > 0 && FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw) {
+ stopFreq--;
+ }
+
+ if (FDKsbrEnc_getSbrStopFreqRAW( stopFreq, sampleRateCore) > maxStopFreqRaw)
+ return -1;
+
+ err = FDKsbrEnc_FindStartAndStopBand (
+ sampleRateCore<<(downSampleFactor-1),
+ sampleRateCore,
+ 32<<(downSampleFactor-1),
+ startFreq,
+ stopFreq,
+ &startBand,
+ &stopBand
+ );
+ if (err)
+ return -1;
+
+ return stopFreq;
+}
+
/***************************************************************************/
/*!
@@ -307,15 +347,16 @@ getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){
****************************************************************************/
static UINT
-FDKsbrEnc_IsSbrSettingAvail (UINT bitrate, /*! the total bitrate in bits/sec */
- UINT vbrMode, /*! the vbr paramter, 0 means constant bitrate */
- UINT numOutputChannels,/*! the number of channels for the core coder */
- UINT sampleRateInput, /*! the input sample rate [in Hz] */
- AUDIO_OBJECT_TYPE core
- )
+FDKsbrEnc_IsSbrSettingAvail (
+ UINT bitrate, /*! the total bitrate in bits/sec */
+ UINT vbrMode, /*! the vbr paramter, 0 means constant bitrate */
+ UINT numOutputChannels, /*! the number of channels for the core coder */
+ UINT sampleRateInput, /*! the input sample rate [in Hz] */
+ UINT sampleRateCore, /*! the core's sampling rate */
+ AUDIO_OBJECT_TYPE core
+ )
{
INT idx = INVALID_TABLE_IDX;
- UINT sampleRateCore;
if (sampleRateInput < 16000)
return 0;
@@ -335,8 +376,6 @@ FDKsbrEnc_IsSbrSettingAvail (UINT bitrate, /*! the total bitrate in bit
bitrate *= numOutputChannels;
}
- /* try DOWN_SMPL_FAC of the input sampling rate */
- sampleRateCore = sampleRateInput/DOWN_SMPL_FAC;
idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core, NULL);
return (idx == INVALID_TABLE_IDX ? 0 : 1);
@@ -356,7 +395,8 @@ static UINT
FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modified */
UINT bitRate, /*! the total bitrate in bits/sec */
UINT numChannels, /*! the core coder number of channels */
- UINT fsCore, /*! the core coder sampling rate in Hz */
+ UINT sampleRateCore, /*! the core coder sampling rate in Hz */
+ UINT sampleRateSbr, /*! the sbr coder sampling rate in Hz */
UINT transFac, /*! the short block to long block ratio */
UINT standardBitrate, /*! the standard bitrate per channel in bits/sec */
UINT vbrMode, /*! the vbr paramter, 0 poor quality .. 100 high quality*/
@@ -366,15 +406,12 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
AUDIO_OBJECT_TYPE core) /* Core audio codec object type */
{
INT idx = INVALID_TABLE_IDX;
- UINT sampleRate;
-
- /* set the codec settings */
+ /* set the core codec settings */
config->codecSettings.bitRate = bitRate;
config->codecSettings.nChannels = numChannels;
- config->codecSettings.sampleFreq = fsCore;
+ config->codecSettings.sampleFreq = sampleRateCore;
config->codecSettings.transFac = transFac;
config->codecSettings.standardBitrate = standardBitrate;
- sampleRate = fsCore * DOWN_SMPL_FAC;
if (bitRate==0) {
/* map vbr quality to bitrate */
@@ -391,13 +428,13 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
bitRate *= numChannels;
/* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */
if (numChannels==1) {
- if (sampleRate==44100 || sampleRate==48000) {
+ if (sampleRateSbr==44100 || sampleRateSbr==48000) {
if (vbrMode<40) bitRate = 32000;
}
}
}
- idx = getSbrTuningTableIndex(bitRate,numChannels,fsCore, core, NULL);
+ idx = getSbrTuningTableIndex(bitRate,numChannels,sampleRateCore, core, NULL);
if (idx != INVALID_TABLE_IDX) {
config->startFreq = sbrTuningTable[idx].startFreq ;
@@ -407,6 +444,21 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
config->stopFreq = sbrTuningTable[idx].stopFreqSpeech;
}
+ /* Adapt stop frequency in case of downsampled SBR - only 32 bands then */
+ if (1 == config->downSampleFactor) {
+ INT dsStopFreq = FDKsbrEnc_GetDownsampledStopFreq(
+ sampleRateCore,
+ config->startFreq,
+ config->stopFreq,
+ config->downSampleFactor
+ );
+ if (dsStopFreq < 0) {
+ return 0;
+ }
+
+ config->stopFreq = dsStopFreq;
+ }
+
config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands ;
if (core == AOT_ER_AAC_ELD)
config->init_amp_res_FF = SBR_AMP_RES_1_5;
@@ -455,19 +507,20 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
description: initializes the SBR confifuration
returns: error status
input: - core codec type,
- - fac of SBR to core frame length,
+ - factor of SBR to core frame length,
- core frame length
output: initialized SBR configuration
*****************************************************************************/
static UINT
FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
- INT coreSbrFrameLenFac,
- UINT codecGranuleLen)
+ INT downSampleFactor,
+ UINT codecGranuleLen
+ )
{
- if ( (coreSbrFrameLenFac != 2) ||
- (codecGranuleLen*coreSbrFrameLenFac > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) )
- return(1);
+ if ( (downSampleFactor < 1 || downSampleFactor > 2) ||
+ (codecGranuleLen*downSampleFactor > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) )
+ return(0); /* error */
config->SendHeaderDataTime = 1000;
config->useWaveCoding = 0;
@@ -476,8 +529,8 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
config->tran_thr = 13000;
config->parametricCoding = 1;
- config->sbrFrameSize = codecGranuleLen * coreSbrFrameLenFac;
-
+ config->sbrFrameSize = codecGranuleLen * downSampleFactor;
+ config->downSampleFactor = downSampleFactor;
/* sbr default parameters */
config->sbr_data_extra = 0;
@@ -497,7 +550,6 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
config->sbr_xpos_level = 0;
config->useSaPan = 0;
config->dynBwEnabled = 0;
- config->bDownSampledSbr = 0;
/* the following parameters are overwritten by the FDKsbrEnc_AdjustSbrSettings() function since
@@ -645,46 +697,62 @@ void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder)
output: error info
*****************************************************************************/
-static INT updateFreqBandTable(HANDLE_SBR_CONFIG_DATA sbrConfigData,
- HANDLE_SBR_HEADER_DATA sbrHeaderData,
- INT noQmfChannels)
+static INT updateFreqBandTable(
+ HANDLE_SBR_CONFIG_DATA sbrConfigData,
+ HANDLE_SBR_HEADER_DATA sbrHeaderData,
+ const INT downSampleFactor
+ )
{
INT k0, k2;
- if(FDKsbrEnc_FindStartAndStopBand(sbrConfigData->sampleFreq,
- noQmfChannels,
- sbrHeaderData->sbr_start_frequency,
- sbrHeaderData->sbr_stop_frequency,
- sbrHeaderData->sampleRateMode,
- &k0, &k2))
+ if( FDKsbrEnc_FindStartAndStopBand (
+ sbrConfigData->sampleFreq,
+ sbrConfigData->sampleFreq >> (downSampleFactor-1),
+ sbrConfigData->noQmfBands,
+ sbrHeaderData->sbr_start_frequency,
+ sbrHeaderData->sbr_stop_frequency,
+ &k0,
+ &k2
+ )
+ )
return(1);
- if(FDKsbrEnc_UpdateFreqScale(sbrConfigData->v_k_master, &sbrConfigData->num_Master,
- k0, k2, sbrHeaderData->freqScale,
- sbrHeaderData->alterScale))
+ if( FDKsbrEnc_UpdateFreqScale(
+ sbrConfigData->v_k_master,
+ &sbrConfigData->num_Master,
+ k0,
+ k2,
+ sbrHeaderData->freqScale,
+ sbrHeaderData->alterScale
+ )
+ )
return(1);
sbrHeaderData->sbr_xover_band=0;
- if(FDKsbrEnc_UpdateHiRes(sbrConfigData->freqBandTable[HI],
- &sbrConfigData->nSfb[HI],
- sbrConfigData->v_k_master,
- sbrConfigData->num_Master ,
- &sbrHeaderData->sbr_xover_band,
- sbrHeaderData->sampleRateMode,
- noQmfChannels))
+ if( FDKsbrEnc_UpdateHiRes(
+ sbrConfigData->freqBandTable[HI],
+ &sbrConfigData->nSfb[HI],
+ sbrConfigData->v_k_master,
+ sbrConfigData->num_Master,
+ &sbrHeaderData->sbr_xover_band
+ )
+ )
return(1);
- FDKsbrEnc_UpdateLoRes(sbrConfigData->freqBandTable[LO],
- &sbrConfigData->nSfb[LO],
- sbrConfigData->freqBandTable[HI],
- sbrConfigData->nSfb[HI]);
+ FDKsbrEnc_UpdateLoRes(
+ sbrConfigData->freqBandTable[LO],
+ &sbrConfigData->nSfb[LO],
+ sbrConfigData->freqBandTable[HI],
+ sbrConfigData->nSfb[HI]
+ );
+
- sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / noQmfChannels+1)>>1;
+ sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / sbrConfigData->noQmfBands+1)>>1;
return (0);
}
@@ -866,7 +934,8 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder,
*/
if(updateFreqBandTable(&hSbrElement->sbrConfigData,
&hSbrElement->sbrHeaderData,
- hSbrElement->sbrConfigData.noQmfBands))
+ hEnvEncoder->downSampleFactor
+ ))
return(1);
@@ -891,8 +960,6 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder,
&crcInfo,
hSbrElement->sbrConfigData.sbrSyntaxFlags);
- INT error = noError;
-
/* Temporal Envelope Data */
SBR_FRAME_TEMP_DATA _fData;
SBR_FRAME_TEMP_DATA *fData = &_fData;
@@ -923,9 +990,9 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder,
if(hSbrElement->elInfo.fParametricStereo == 0)
{
- C_ALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2);
QMF_SCALE_FACTOR tmpScale;
FIXP_DBL **pQmfReal, **pQmfImag;
+ C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2)
/* Obtain pointers to QMF buffers. */
@@ -940,10 +1007,11 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder,
timeInStride,
qmfWorkBuffer );
- C_ALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2);
-
h_envChan->qmfScale = tmpScale.lb_scale + 7;
+
+ C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2)
+
} /* fParametricStereo == 0 */
@@ -952,6 +1020,8 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder,
*/
if (hSbrElement->elInfo.fParametricStereo)
{
+ INT error = noError;
+
/* Limit Parametric Stereo to one instance */
FDK_ASSERT(ch == 0);
@@ -1177,10 +1247,12 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData,
break;
case 2048:
case 1024:
+ case 512:
timeSlots = 16;
break;
case 1920:
case 960:
+ case 480:
timeSlots = 15;
break;
case 1152:
@@ -1221,9 +1293,9 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData,
tran_fc = params->tran_fc;
- if (tran_fc == 0)
- tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,64,sbrConfigData->sampleFreq));
-
+ if (tran_fc == 0) {
+ tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,params->codecSettings.sampleFreq));
+ }
tran_fc = (tran_fc*4*sbrConfigData->noQmfBands/sbrConfigData->sampleFreq + 1)>>1;
@@ -1233,11 +1305,11 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData,
} else
{
frameShift = 0;
- switch (params->sbrFrameSize) {
+ switch (timeSlots) {
/* The factor of 2 is by definition. */
- case 2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break;
- case 1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break;
- default: return 1; break;
+ case NUMBER_TIME_SLOTS_2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break;
+ case NUMBER_TIME_SLOTS_1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break;
+ default: return 1;
}
}
if ( FDKsbrEnc_InitExtractSbrEnvelope (&hEnv->sbrExtractEnvelope,
@@ -1330,7 +1402,6 @@ INT sbrEncoder_Open(
hSbrEncoder->pSBRdynamic_RAM = (UCHAR*)GetRam_SbrDynamic_RAM();
hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM;
-
for (i=0; i<nElements; i++) {
hSbrEncoder->sbrElement[i] = GetRam_SbrElement(i);
if (hSbrEncoder->sbrElement[i]==NULL) {
@@ -1463,7 +1534,8 @@ INT FDKsbrEnc_EnvInit (
int nBitstrDelay,
int nElement,
const int headerPeriod,
- ULONG statesInitFlag
+ ULONG statesInitFlag,
+ int fTimeDomainDownsampling
,UCHAR *dynamic_RAM
)
{
@@ -1497,8 +1569,16 @@ INT FDKsbrEnc_EnvInit (
hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC;
}
- hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS;
- hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize/hSbrElement->sbrConfigData.noQmfBands;
+ hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS>>(2-params->downSampleFactor);
+ switch (hSbrElement->sbrConfigData.noQmfBands)
+ {
+ case 64: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6;
+ break;
+ case 32: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>5;
+ break;
+ default: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6;
+ return(2);
+ }
FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, MAX_PAYLOAD_SIZE*sizeof(UCHAR), 0, BS_WRITER);
@@ -1514,10 +1594,7 @@ INT FDKsbrEnc_EnvInit (
hSbrElement->sbrConfigData.frameSize = params->sbrFrameSize;
- /* implicit rule for sampleRateMode */
- /* run in "multirate" mode where sbr fs is 2 * codec fs */
- hSbrElement->sbrHeaderData.sampleRateMode = DUAL_RATE;
- hSbrElement->sbrConfigData.sampleFreq = 2 * params->codecSettings.sampleFreq;
+ hSbrElement->sbrConfigData.sampleFreq = params->downSampleFactor * params->codecSettings.sampleFreq;
hSbrElement->sbrBitstreamData.CountSendHeaderData = 0;
if (params->SendHeaderDataTime > 0 ) {
@@ -1592,7 +1669,8 @@ INT FDKsbrEnc_EnvInit (
/* init freq band table */
if(updateFreqBandTable(&hSbrElement->sbrConfigData,
&hSbrElement->sbrHeaderData,
- hSbrElement->sbrConfigData.noQmfBands))
+ params->downSampleFactor
+ ))
{
return(1);
}
@@ -1632,6 +1710,9 @@ INT FDKsbrEnc_EnvInit (
hSbrElement->sbrConfigData.noQmfBands,
hSbrElement->sbrConfigData.noQmfBands,
qmfFlags );
+ if (0!=err) {
+ return err;
+ }
}
/* */
@@ -1653,7 +1734,7 @@ INT sbrEncoder_GetInBufferSize(int noChannels)
{
INT temp;
- temp = (1024*DOWN_SMPL_FAC);
+ temp = (2048);
temp += 1024 + MAX_SAMPLE_DELAY;
temp *= noChannels;
temp *= sizeof(INT_PCM);
@@ -1685,8 +1766,8 @@ INT FDKsbrEnc_DelayCompensation (
1
))
return -1;
- sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer);
}
+ sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer);
}
return 0;
}
@@ -1717,30 +1798,36 @@ UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate
return newBitRate;
}
+UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot)
+{
+ UINT isPossible=(AOT_PS==aot)?0:1;
+ return isPossible;
+}
INT sbrEncoder_Init(
- HANDLE_SBR_ENCODER hSbrEncoder,
- SBR_ELEMENT_INFO elInfo[(6)],
- int noElements,
- INT_PCM *inputBuffer,
- INT *coreBandwidth,
- INT *inputBufferOffset,
- INT *numChannels,
- INT *sampleRate,
- INT *frameLength,
- AUDIO_OBJECT_TYPE *aot,
- int *delay,
- int transformFactor,
- const int headerPeriod,
- ULONG statesInitFlag
- )
+ HANDLE_SBR_ENCODER hSbrEncoder,
+ SBR_ELEMENT_INFO elInfo[(6)],
+ int noElements,
+ INT_PCM *inputBuffer,
+ INT *coreBandwidth,
+ INT *inputBufferOffset,
+ INT *numChannels,
+ INT *coreSampleRate,
+ UINT *downSampleFactor,
+ INT *frameLength,
+ AUDIO_OBJECT_TYPE aot,
+ int *delay,
+ int transformFactor,
+ const int headerPeriod,
+ ULONG statesInitFlag
+ )
{
HANDLE_ERROR_INFO errorInfo = noError;
sbrConfiguration sbrConfig[(6)];
INT error = 0;
INT lowestBandwidth;
/* Save input parameters */
- INT inputSampleRate = *sampleRate;
+ INT inputSampleRate = *coreSampleRate;
int coreFrameLength = *frameLength;
int inputBandWidth = *coreBandwidth;
int inputChannels = *numChannels;
@@ -1748,20 +1835,26 @@ INT sbrEncoder_Init(
int downsampledOffset = 0;
int sbrOffset = 0;
int downsamplerDelay = 0;
- int downsample = 0;
+ int timeDomainDownsample = 0;
int nBitstrDelay = 0;
- int lowestSbrStartFreq, lowestSbrStopFreq;
+ int highestSbrStartFreq, highestSbrStopFreq;
int lowDelay = 0;
int usePs = 0;
/* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */
- if ( (*aot==AOT_PS) || (*aot==AOT_MP2_PS) || (*aot==AOT_DABPLUS_PS) || (*aot==AOT_DRM_MPEG_PS) ) {
+ if (!sbrEncoder_IsSingleRatePossible(aot)) {
+ *downSampleFactor = 2;
+ }
+
+
+
+ if ( (aot==AOT_PS) || (aot==AOT_MP2_PS) || (aot==AOT_DABPLUS_PS) || (aot==AOT_DRM_MPEG_PS) ) {
usePs = 1;
}
- if ( (*aot==AOT_ER_AAC_ELD) ) {
+ if ( (aot==AOT_ER_AAC_ELD) ) {
lowDelay = 1;
}
- else if ( (*aot==AOT_ER_AAC_LD) ) {
+ else if ( (aot==AOT_ER_AAC_LD) ) {
error = 1;
goto bail;
}
@@ -1776,25 +1869,25 @@ INT sbrEncoder_Init(
/* core encoder gets downmixed mono signal */
*numChannels = 1;
} else {
- switch (*aot) {
- case AOT_MP2_PS:
- *aot = AOT_MP2_SBR;
- break;
- case AOT_DABPLUS_PS:
- *aot = AOT_DABPLUS_SBR;
- break;
- case AOT_DRM_MPEG_PS:
- *aot = AOT_DRM_SBR;
- break;
- case AOT_PS:
- default:
- *aot = AOT_SBR;
- }
- usePs = 0;
+ error = 1;
+ goto bail;
}
} /* usePs */
- /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */
+ /* set the core's sample rate */
+ switch (*downSampleFactor) {
+ case 1:
+ *coreSampleRate = inputSampleRate;
+ break;
+ case 2:
+ *coreSampleRate = inputSampleRate>>1;
+ break;
+ default:
+ *coreSampleRate = inputSampleRate>>1;
+ return 0; /* return error */
+ }
+
+ /* check whether SBR setting is available for the current encoder configuration (bitrate, coreSampleRate) */
{
int delayDiff = 0;
int el, coreEl;
@@ -1807,54 +1900,37 @@ INT sbrEncoder_Init(
continue;
}
/* check if desired configuration is available */
- if ( !FDKsbrEnc_IsSbrSettingAvail (elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *aot) )
+ if ( !FDKsbrEnc_IsSbrSettingAvail (elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *coreSampleRate, aot) )
{
- /* otherwise - change to AAC-LC */
- switch (*aot) {
- case AOT_MP2_SBR:
- case AOT_MP2_PS:
- *aot = AOT_MP2_AAC_LC;
- break;
- case AOT_DABPLUS_SBR:
- case AOT_DABPLUS_PS:
- *aot = AOT_DABPLUS_AAC_LC;
- break;
- case AOT_DRM_SBR:
- case AOT_DRM_MPEG_PS:
- *aot = AOT_DRM_AAC;
- break;
- case AOT_ER_AAC_ELD:
- break;
- case AOT_SBR:
- case AOT_PS:
- default:
- *aot = AOT_AAC_LC;
- }
error = 1;
goto bail;
}
}
- *sampleRate /= DOWN_SMPL_FAC;
-
/* Determine Delay balancing and new encoder delay */
if (lowDelay) {
- downsample = 1; /* activate downsampler */
- delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_ELD2SBR(coreFrameLength);
- *delay = DELAY_ELDSBR(coreFrameLength);
+ {
+ delayDiff = (*delay * *downSampleFactor) + DELAY_ELD2SBR(coreFrameLength,*downSampleFactor);
+ *delay = DELAY_ELDSBR(coreFrameLength,*downSampleFactor);
+ }
}
else if (usePs) {
- delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_AAC2PS(coreFrameLength);
- *delay = DELAY_PS(coreFrameLength);
+ delayDiff = (*delay * *downSampleFactor) + DELAY_AAC2PS(coreFrameLength,*downSampleFactor);
+ *delay = DELAY_PS(coreFrameLength,*downSampleFactor);
}
else {
- downsample = 1; /* activate downsampler */
- delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_AAC2SBR(coreFrameLength);
- *delay = DELAY_SBR(coreFrameLength);
+ delayDiff = DELAY_AAC2SBR(coreFrameLength,*downSampleFactor);
+ delayDiff += (*delay * *downSampleFactor);
+ *delay = DELAY_SBR(coreFrameLength,*downSampleFactor);
+ }
+
+ if (!usePs) {
+ timeDomainDownsample = *downSampleFactor-1; /* activate time domain downsampler when downSampleFactor is != 1 */
}
+
/* Take care about downsampled data bound to the SBR path */
- if (!downsample && delayDiff > 0) {
+ if (!timeDomainDownsample && delayDiff > 0) {
/*
* We must tweak the balancing into a situation where the downsampled path
* is the one to be delayed, because delaying the QMF domain input, also delays
@@ -1863,12 +1939,15 @@ INT sbrEncoder_Init(
while ( delayDiff > 0 )
{
/* Encoder delay increases */
- *delay += coreFrameLength*DOWN_SMPL_FAC;
- /* Add one frame delay to SBR path */
- delayDiff -= coreFrameLength*DOWN_SMPL_FAC;
+ {
+ *delay += coreFrameLength * *downSampleFactor;
+ /* Add one frame delay to SBR path */
+ delayDiff -= coreFrameLength * *downSampleFactor;
+ }
nBitstrDelay += 1;
}
- } else {
+ } else
+ {
*delay += fixp_abs(delayDiff);
}
@@ -1876,32 +1955,33 @@ INT sbrEncoder_Init(
/* Delay AAC data */
delayDiff = -delayDiff;
/* Multiply downsampled offset by AAC core channels. Divide by 2 because of half samplerate of downsampled data. */
- downsampledOffset = (delayDiff*(*numChannels))/DOWN_SMPL_FAC;
+ FDK_ASSERT(*downSampleFactor>0 && *downSampleFactor<=2);
+ downsampledOffset = (delayDiff*(*numChannels))>>(*downSampleFactor-1);
sbrOffset = 0;
} else {
/* Delay SBR input */
- if ( delayDiff > (int)coreFrameLength*DOWN_SMPL_FAC )
+ if ( delayDiff > (int)coreFrameLength * (int)*downSampleFactor )
{
/* Do bitstream frame-wise delay balancing if we have more than SBR framelength samples delay difference */
- delayDiff -= coreFrameLength*DOWN_SMPL_FAC;
+ delayDiff -= coreFrameLength * *downSampleFactor;
nBitstrDelay = 1;
}
/* Multiply input offset by input channels */
sbrOffset = delayDiff*(*numChannels);
downsampledOffset = 0;
}
-
- hSbrEncoder->nBitstrDelay = nBitstrDelay;
- hSbrEncoder->nChannels = *numChannels;
- hSbrEncoder->frameSize = *frameLength*DOWN_SMPL_FAC;
- hSbrEncoder->fTimeDomainDownsampling = downsample;
- hSbrEncoder->estimateBitrate = 0;
- hSbrEncoder->inputDataDelay = 0;
+ hSbrEncoder->nBitstrDelay = nBitstrDelay;
+ hSbrEncoder->nChannels = *numChannels;
+ hSbrEncoder->frameSize = coreFrameLength * *downSampleFactor;
+ hSbrEncoder->fTimeDomainDownsampling = timeDomainDownsample;
+ hSbrEncoder->downSampleFactor = *downSampleFactor;
+ hSbrEncoder->estimateBitrate = 0;
+ hSbrEncoder->inputDataDelay = 0;
/* Open SBR elements */
el = -1;
- lowestSbrStartFreq = lowestSbrStopFreq = 9999;
+ highestSbrStartFreq = highestSbrStopFreq = 0;
lowestBandwidth = 99999;
/* Loop through each core encoder element and get a matching SBR element config */
@@ -1924,28 +2004,38 @@ INT sbrEncoder_Init(
/*
* Init sbrConfig structure
*/
- FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el],
- DOWN_SMPL_FAC,
- coreFrameLength);
+ if ( ! FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el],
+ *downSampleFactor,
+ coreFrameLength
+ ) )
+ {
+ error = 1;
+ goto bail;
+ }
+
/*
* Modify sbrConfig structure according to Element parameters
*/
- FDKsbrEnc_AdjustSbrSettings ( &sbrConfig[el],
- elInfo[coreEl].bitRate,
- elInfo[coreEl].nChannelsInEl,
- *sampleRate,
- transformFactor,
- 24000,
- 0,
- 0, /* useSpeechConfig */
- 0, /* lcsMode */
- usePs, /* bParametricStereo */
- *aot);
+ if ( ! FDKsbrEnc_AdjustSbrSettings (&sbrConfig[el],
+ elInfo[coreEl].bitRate,
+ elInfo[coreEl].nChannelsInEl,
+ *coreSampleRate,
+ inputSampleRate,
+ transformFactor,
+ 24000,
+ 0,
+ 0, /* useSpeechConfig */
+ 0, /* lcsMode */
+ usePs, /* bParametricStereo */
+ aot) )
+ {
+ error = 1;
+ goto bail;
+ }
/* Find common frequency border for all SBR elements */
- lowestSbrStartFreq = fixMin(lowestSbrStartFreq, sbrConfig[el].startFreq);
- lowestSbrStopFreq = fixMin(lowestSbrStopFreq, sbrConfig[el].stopFreq);
-
+ highestSbrStartFreq = fixMax(highestSbrStartFreq, sbrConfig[el].startFreq);
+ highestSbrStopFreq = fixMax(highestSbrStopFreq, sbrConfig[el].stopFreq);
} /* first element loop */
@@ -1961,22 +2051,24 @@ INT sbrEncoder_Init(
int bandwidth = *coreBandwidth;
/* Use lowest common bandwidth */
- sbrConfig[el].startFreq = lowestSbrStartFreq;
- sbrConfig[el].stopFreq = lowestSbrStopFreq;
+ sbrConfig[el].startFreq = highestSbrStartFreq;
+ sbrConfig[el].stopFreq = highestSbrStopFreq;
/* initialize SBR element, and get core bandwidth */
error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el],
&sbrConfig[el],
&bandwidth,
- *aot,
+ aot,
nBitstrDelay,
el,
headerPeriod,
- statesInitFlag
+ statesInitFlag,
+ hSbrEncoder->fTimeDomainDownsampling
,hSbrEncoder->dynamicRam
);
if (error != 0) {
+ error = 2;
goto bail;
}
@@ -1998,30 +2090,29 @@ INT sbrEncoder_Init(
for (ch=0; ch<hSbrEl->elInfo.nChannelsInEl; ch++)
{
- FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, DOWN_SMPL_FAC);
+ FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, *downSampleFactor);
+ FDK_ASSERT (hSbrEl->sbrChannel[ch]->downSampler.delay <=MAX_DS_FILTER_DELAY);
}
- FDK_ASSERT (hSbrEl->sbrChannel[0]->downSampler.delay <=MAX_DS_FILTER_DELAY && hSbrEl->sbrChannel[0]->downSampler.delay <=MAX_DS_FILTER_DELAY);
downsamplerDelay = hSbrEl->sbrChannel[0]->downSampler.delay;
} /* third element loop */
/* lfe */
- FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, DOWN_SMPL_FAC);
+ FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, *downSampleFactor);
/* Add the resampler additional delay to get the final delay and buffer offset values. */
- if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC)) {
+ if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))>>(*downSampleFactor-1))) {
sbrOffset += (downsamplerDelay - downsampledOffset) * (*numChannels) ;
*delay += downsamplerDelay - downsampledOffset;
downsampledOffset = 0;
} else {
- downsampledOffset -= (downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC;
+ downsampledOffset -= (downsamplerDelay * (*numChannels))>>(*downSampleFactor-1);
sbrOffset = 0;
}
hSbrEncoder->inputDataDelay = downsamplerDelay;
}
-
/* Assign core encoder Bandwidth */
*coreBandwidth = lowestBandwidth;
@@ -2035,7 +2126,7 @@ INT sbrEncoder_Init(
FDK_ASSERT(hSbrEncoder->noElements == 1);
INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL);
- psEncConfig.frameSize = *frameLength; //sbrConfig.sbrFrameSize;
+ psEncConfig.frameSize = coreFrameLength; //sbrConfig.sbrFrameSize;
psEncConfig.qmfFilterMode = 0;
psEncConfig.sbrPsDelay = 0;
@@ -2047,7 +2138,7 @@ INT sbrEncoder_Init(
/* calculation is not quite linear, increased number of envelopes causes more bits */
/* assume avg. 50 bits per frame for 10 stereo bands / 1 envelope configuration */
- hSbrEncoder->estimateBitrate += ( (((*sampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize));
+ hSbrEncoder->estimateBitrate += ( (((*coreSampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize));
} else {
error = ERROR(CDI, "Invalid ps tuning table index.");
@@ -2076,10 +2167,16 @@ INT sbrEncoder_Init(
errorInfo = handBack(errorInfo);
}
}
+
+ /* QMF analysis + Hybrid analysis + Hybrid synthesis + QMF synthesis + downsampled input buffer delay */
+ hSbrEncoder->inputDataDelay = (64*10/2) + (6*64) + (0) + (64*10/2-64+1) + ((*downSampleFactor)*downsampledOffset);
}
hSbrEncoder->downsampledOffset = downsampledOffset;
- hSbrEncoder->downmixSize = coreFrameLength*(*numChannels);
+ {
+ hSbrEncoder->downmixSize = coreFrameLength*(*numChannels);
+ }
+
hSbrEncoder->bufferOffset = sbrOffset;
/* Delay Compensation: fill bitstream delay buffer with zero input signal */
if ( hSbrEncoder->nBitstrDelay > 0 )
@@ -2090,7 +2187,7 @@ INT sbrEncoder_Init(
}
/* Set Output frame length */
- *frameLength = coreFrameLength*DOWN_SMPL_FAC;
+ *frameLength = coreFrameLength * *downSampleFactor;
/* Input buffer offset */
*inputBufferOffset = fixMax(sbrOffset, downsampledOffset);
@@ -2101,7 +2198,7 @@ INT sbrEncoder_Init(
bail:
/* Restore input settings */
- *sampleRate = inputSampleRate;
+ *coreSampleRate = inputSampleRate;
*frameLength = coreFrameLength;
*numChannels = inputChannels;
*coreBandwidth = inputBandWidth;
@@ -2139,8 +2236,8 @@ sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder,
}
}
- if ( (hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->fTimeDomainDownsampling) )
- {
+ if ( ( hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->downSampleFactor > 1) )
+ { /* lfe downsampler */
INT nOutSamples;
FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler,
@@ -2150,7 +2247,9 @@ sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder,
samples + hSbrEncoder->downsampledOffset + hSbrEncoder->lfeChIdx,
&nOutSamples,
hSbrEncoder->nChannels);
- } /* lfe downsampler */
+
+
+ }
return 0;
}
diff --git a/libSBRenc/src/sbr_rom.cpp b/libSBRenc/src/sbr_rom.cpp
index b1e3fb8..a2b6527 100644
--- a/libSBRenc/src/sbr_rom.cpp
+++ b/libSBRenc/src/sbr_rom.cpp
@@ -506,216 +506,277 @@ const UCHAR bookSbrNoiseBalanceL11T[25] =
/*
tuningTable
*/
-const sbrTuningTable_t sbrTuningTable[SBRENC_TUNING_SIZE] =
+const sbrTuningTable_t sbrTuningTable[] =
{
+ /* Some of the low bitrates are commented out here, this is because the
+ encoder could lose frames at those bitrates and throw an error because
+ it has insufficient bits to encode for some test items.
+ */
- /*** AAC ***/
+ /*** HE-AAC section ***/
/* sf,sfsp,sf,sfsp,nnb,nfo,saml,SM,FS*/
/*** mono ***/
/* 8/16 kHz dual rate */
- { 8000, 10000, 8000, 1, 7, 6, 11,10, 1, 0, 6, SBR_MONO, 3 },
- { 10000, 12000, 8000, 1, 11, 7, 13,12, 1, 0, 6, SBR_MONO, 3 },
- { 12000, 16001, 8000, 1, 14,10, 13,13, 1, 0, 6, SBR_MONO, 3 },
- { 16000, 24000, 8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
- { 24000, 32000, 8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
- { 32000, 48001, 8000, 1, 14,11, 15,15, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ /* bitrates higher than 48000 not supported by AAC core */
+ { CODEC_AAC, 8000, 10000, 8000, 1, 7, 6, 11,10, 1, 0, 6, SBR_MONO, 3 },
+ { CODEC_AAC, 10000, 12000, 8000, 1, 11, 7, 13,12, 1, 0, 6, SBR_MONO, 3 },
+ { CODEC_AAC, 12000, 16001, 8000, 1, 14,10, 13,13, 1, 0, 6, SBR_MONO, 3 },
+ { CODEC_AAC, 16000, 24000, 8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+ { CODEC_AAC, 24000, 32000, 8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+ { CODEC_AAC, 32000, 48001, 8000, 1, 14,11, 15,15, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ /* bitrates higher than 48000 not supported by AAC core */
/* 11/22 kHz dual rate */
- { 8000, 10000, 11025, 1, 5, 4, 6, 6, 1, 0, 6, SBR_MONO, 3 },
- { 10000, 12000, 11025, 1, 8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3 },
- { 12000, 16000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 },
- { 16000, 20000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* at such "high" bitrates it's better to upsample the input */
- { 20000, 24001, 11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* signal by a factor of 2 before sending it into the encoder */
- { 24000, 32000, 11025, 1, 14,10, 14, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
- { 32000, 48000, 11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
- { 48000, 64001, 11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
+ { CODEC_AAC, 8000, 10000, 11025, 1, 5, 4, 6, 6, 1, 0, 6, SBR_MONO, 3 },
+ { CODEC_AAC, 10000, 12000, 11025, 1, 8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3 },
+ { CODEC_AAC, 12000, 16000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 },
+ { CODEC_AAC, 16000, 20000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* at such "high" bitrates it's better to upsample the input */
+ { CODEC_AAC, 20000, 24001, 11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* signal by a factor of 2 before sending it into the encoder */
+ { CODEC_AAC, 24000, 32000, 11025, 1, 14,10, 14, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+ { CODEC_AAC, 32000, 48000, 11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+ { CODEC_AAC, 48000, 64001, 11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
/* 12/24 kHz dual rate */
- { 8000, 10000, 12000, 1, 4, 3, 6, 6, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */
- { 10000, 12000, 12000, 1, 7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
- { 12000, 16000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
- { 16000, 20000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ /* at such "high" bitrates it's better to upsample the input */
- { 20000, 24001, 12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ /* signal by a factor of 2 before sending it into the encoder */
- { 24000, 32000, 12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
- { 32000, 48000, 12000, 1, 14,10, 14,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
- { 48000, 64001, 12000, 1, 15,11, 15,11, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
+ { CODEC_AAC, 8000, 10000, 12000, 1, 4, 3, 6, 6, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */
+ { CODEC_AAC, 10000, 12000, 12000, 1, 7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
+ { CODEC_AAC, 12000, 16000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+ { CODEC_AAC, 16000, 20000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ /* at such "high" bitrates it's better to upsample the input */
+ { CODEC_AAC, 20000, 24001, 12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ /* signal by a factor of 2 before sending it into the encoder */
+ { CODEC_AAC, 24000, 32000, 12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+ { CODEC_AAC, 32000, 48000, 12000, 1, 14,10, 14,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+ { CODEC_AAC, 48000, 64001, 12000, 1, 14,11, 15,11, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
/* 16/32 kHz dual rate */
- { 8000, 10000, 16000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */
- { 10000, 12000, 16000, 1, 2, 1, 6, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
- { 12000, 16000, 16000, 1, 4, 2, 6, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
- { 16000, 18000, 16000, 1, 4, 2, 8, 3, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
- { 18000, 22000, 16000, 1, 6, 5,11, 7, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
- { 22000, 28000, 16000, 1, 10, 9,12, 8, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 16000, 1, 12,12,13,13, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 16000, 1, 14,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
- { 44000, 64001, 16000, 1, 15,15,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+ { CODEC_AAC, 8000, 10000, 16000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */
+ { CODEC_AAC, 10000, 12000, 16000, 1, 2, 1, 6, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
+ { CODEC_AAC, 12000, 16000, 16000, 1, 4, 2, 6, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+ { CODEC_AAC, 16000, 18000, 16000, 1, 4, 2, 8, 3, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+ { CODEC_AAC, 18000, 22000, 16000, 1, 6, 5,11, 7, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+ { CODEC_AAC, 22000, 28000, 16000, 1, 10, 9,12, 8, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 36000, 16000, 1, 12,12,13,13, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 16000, 1, 14,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 64001, 16000, 1, 14,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
/* 22.05/44.1 kHz dual rate */
- /* { 8000, 11369, 22050, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ /* nominal: 8 kbit/s */ /* encoder can not work stable at this extremely low bitrate */
- { 11369, 16000, 22050, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
- { 16000, 18000, 22050, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
- { 18000, 22000, 22050, 1, 4, 4, 8, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
- { 22000, 28000, 22050, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 22050, 1, 10,10, 9, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 22050, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
- { 44000, 64001, 22050, 1, 13,13,12,12, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+ /* { CODEC_AAC, 8000, 11369, 22050, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ /* nominal: 8 kbit/s */ /* encoder can not work stable at this extremely low bitrate */
+ { CODEC_AAC, 11369, 16000, 22050, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+ { CODEC_AAC, 16000, 18000, 22050, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+ { CODEC_AAC, 18000, 22000, 22050, 1, 4, 4, 8, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+ { CODEC_AAC, 22000, 28000, 22050, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 36000, 22050, 1, 10,10, 9, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 22050, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 64001, 22050, 1, 13,13,12,12, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
/* 24/48 kHz dual rate */
- /* { 8000, 12000, 24000, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ /* nominal: 8 kbit/s */ /* encoder can not work stable at this extremely low bitrate */
- { 12000, 16000, 24000, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
- { 16000, 18000, 24000, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
- { 18000, 22000, 24000, 1, 4, 3, 8, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
- { 22000, 28000, 24000, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 24000, 1, 10,10, 9, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 24000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
- { 44000, 64001, 24000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+ /* { CODEC_AAC, 8000, 12000, 24000, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ /* nominal: 8 kbit/s */ /* encoder can not work stable at this extremely low bitrate */
+ { CODEC_AAC, 12000, 16000, 24000, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+ { CODEC_AAC, 16000, 18000, 24000, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+ { CODEC_AAC, 18000, 22000, 24000, 1, 4, 3, 8, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+ { CODEC_AAC, 22000, 28000, 24000, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 36000, 24000, 1, 10,10, 9, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 24000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 64001, 24000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
/* 32/64 kHz dual rate */ /* placebo settings */
- { 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range */
- { 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range */
- { 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */
- { 72000,100000, 32000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
- { 100000,160001, 32000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
+ { CODEC_AAC, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range */
+ { CODEC_AAC, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range */
+ { CODEC_AAC, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */
+ { CODEC_AAC, 72000,100000, 32000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
+ { CODEC_AAC, 100000,160001, 32000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
/* 44.1/88.2 kHz dual rate */ /* placebo settings */
- { 24000, 36000, 44100, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
- { 36000, 60000, 44100, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range (multichannel rear) */
- { 60000, 72000, 44100, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */
- { 72000,100000, 44100, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
- { 100000,160001, 44100, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
+ { CODEC_AAC, 24000, 36000, 44100, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
+ { CODEC_AAC, 36000, 60000, 44100, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range (multichannel rear) */
+ { CODEC_AAC, 60000, 72000, 44100, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */
+ { CODEC_AAC, 72000,100000, 44100, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
+ { CODEC_AAC, 100000,160001, 44100, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
/* 48/96 kHz dual rate */ /* not yet finally tuned */
- { 32000, 36000, 48000, 1, 4, 4, 9, 9, 2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
- { 36000, 60000, 48000, 1, 7, 7,10,10, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
- { 60000, 72000, 48000, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
- { 72000,100000, 48000, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
- { 100000,160001, 48000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */
+ { CODEC_AAC, 32000, 36000, 48000, 1, 4, 4, 9, 9, 2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
+ { CODEC_AAC, 36000, 60000, 48000, 1, 7, 7,10,10, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
+ { CODEC_AAC, 60000, 72000, 48000, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
+ { CODEC_AAC, 72000,100000, 48000, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
+ { CODEC_AAC, 100000,160001, 48000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */
/*** stereo ***/
/* 08/16 kHz dual rate */
- { 16000, 24000, 8000, 2, 6, 6, 9, 7, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
- { 24000, 28000, 8000, 2, 9, 9, 11, 9, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 8000, 2, 11, 9, 11, 9, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 8000, 2, 13,11, 13,11, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 8000, 2, 14,12, 13,12, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
- { 52000, 60000, 8000, 2, 15,15, 13,13, 3, 0,-3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
- { 60000, 76000, 8000, 2, 15,15, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
- { 76000,128001, 8000, 2, 15,15, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AAC, 16000, 24000, 8000, 2, 6, 6, 9, 7, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+ { CODEC_AAC, 24000, 28000, 8000, 2, 9, 9, 11, 9, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 36000, 8000, 2, 11, 9, 11, 9, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 8000, 2, 13,11, 13,11, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 52000, 8000, 2, 14,12, 13,12, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+ { CODEC_AAC, 52000, 60000, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AAC, 60000, 76000, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AAC, 76000,128001, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
/* 11/22 kHz dual rate */
- { 16000, 24000, 11025, 2, 7, 5, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
- { 24000, 28000, 11025, 2, 10, 8,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 11025, 2, 12, 8,12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 11025, 2, 13, 9,13, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 11025, 2, 14,11,13,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
- { 52000, 60000, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
- { 60000, 76000, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
- { 76000,128001, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AAC, 16000, 24000, 11025, 2, 7, 5, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+ { CODEC_AAC, 24000, 28000, 11025, 2, 10, 8,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 36000, 11025, 2, 12, 8,12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 11025, 2, 13, 9,13, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 52000, 11025, 2, 14,11,13,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+ { CODEC_AAC, 52000, 60000, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AAC, 60000, 76000, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AAC, 76000,128001, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
/* 12/24 kHz dual rate */
- { 16000, 24000, 12000, 2, 6, 4, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
- { 24000, 28000, 12000, 2, 9, 7,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 12000, 2, 11, 7,12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 12000, 2, 12, 9,12, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 12000, 2, 13,12,13,12, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
- { 52000, 60000, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
- { 60000, 76000, 12000, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
- { 76000,128001, 12000, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AAC, 16000, 24000, 12000, 2, 6, 4, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+ { CODEC_AAC, 24000, 28000, 12000, 2, 9, 7,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 36000, 12000, 2, 11, 7,12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 12000, 2, 12, 9,12, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 52000, 12000, 2, 13,12,13,12, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+ { CODEC_AAC, 52000, 60000, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AAC, 60000, 76000, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AAC, 76000,128001, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
/* 16/32 kHz dual rate */
- { 16000, 24000, 16000, 2, 4, 2, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
- { 24000, 28000, 16000, 2, 8, 7,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 16000, 2, 10, 9,12,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 16000, 2, 13,13,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 16000, 2, 15,15,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
- { 52000, 60000, 16000, 2, 15,15,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
- { 60000, 76000, 16000, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
- { 76000,128001, 16000, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AAC, 16000, 24000, 16000, 2, 4, 2, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+ { CODEC_AAC, 24000, 28000, 16000, 2, 8, 7,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 36000, 16000, 2, 10, 9,12,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 16000, 2, 13,13,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 52000, 16000, 2, 14,14,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+ { CODEC_AAC, 52000, 60000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AAC, 60000, 76000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AAC, 76000,128001, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
/* 22.05/44.1 kHz dual rate */
- { 16000, 24000, 22050, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
- { 24000, 28000, 22050, 2, 5, 4, 6, 5, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
- { 28000, 32000, 22050, 2, 5, 4, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 28 kbit/s */
- { 32000, 36000, 22050, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 22050, 2, 10,10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 22050, 2, 12,12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
- { 52000, 60000, 22050, 2, 13,13,10,10, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
- { 60000, 76000, 22050, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
- { 76000,128001, 22050, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AAC, 16000, 24000, 22050, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+ { CODEC_AAC, 24000, 28000, 22050, 2, 5, 4, 6, 5, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 32000, 22050, 2, 5, 4, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 28 kbit/s */
+ { CODEC_AAC, 32000, 36000, 22050, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 22050, 2, 10,10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 52000, 22050, 2, 12,12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+ { CODEC_AAC, 52000, 60000, 22050, 2, 13,13,10,10, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AAC, 60000, 76000, 22050, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AAC, 76000,128001, 22050, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
/* 24/48 kHz dual rate */
- { 16000, 24000, 24000, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
- { 24000, 28000, 24000, 2, 5, 5, 6, 6, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 24000, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 24000, 2, 10,10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 24000, 2, 12,12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
- { 52000, 60000, 24000, 2, 13,13,10,10, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
- { 60000, 76000, 24000, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
- { 76000,128001, 24000, 2, 15,15,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AAC, 16000, 24000, 24000, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+ { CODEC_AAC, 24000, 28000, 24000, 2, 5, 5, 6, 6, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+ { CODEC_AAC, 28000, 36000, 24000, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AAC, 36000, 44000, 24000, 2, 10,10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AAC, 44000, 52000, 24000, 2, 12,12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+ { CODEC_AAC, 52000, 60000, 24000, 2, 13,13,10,10, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AAC, 60000, 76000, 24000, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AAC, 76000,128001, 24000, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
/* 32/64 kHz dual rate */ /* placebo settings */
- { 32000, 60000, 32000, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
- { 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
- { 80000,112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
- { 112000,144000, 32000, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
- { 144000,256001, 32000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+ { CODEC_AAC, 32000, 60000, 32000, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+ { CODEC_AAC, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+ { CODEC_AAC, 80000,112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
+ { CODEC_AAC, 112000,144000, 32000, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
+ { CODEC_AAC, 144000,256001, 32000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
/* 44.1/88.2 kHz dual rate */ /* placebo settings */
- { 32000, 60000, 44100, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
- { 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
- { 80000,112000, 44100, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
- { 112000,144000, 44100, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
- { 144000,256001, 44100, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+ { CODEC_AAC, 32000, 60000, 44100, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+ { CODEC_AAC, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+ { CODEC_AAC, 80000,112000, 44100, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
+ { CODEC_AAC, 112000,144000, 44100, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
+ { CODEC_AAC, 144000,256001, 44100, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
/* 48/96 kHz dual rate */ /* not yet finally tuned */
- { 36000, 60000, 48000, 2, 4, 4, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
- { 60000, 80000, 48000, 2, 7, 7, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */
- { 80000,112000, 48000, 2, 9, 9,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */
- { 112000,144000, 48000, 2, 11,11,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */
- { 144000,256001, 48000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 192 */
+ { CODEC_AAC, 36000, 60000, 48000, 2, 4, 4, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+ { CODEC_AAC, 60000, 80000, 48000, 2, 7, 7, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */
+ { CODEC_AAC, 80000,112000, 48000, 2, 9, 9,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */
+ { CODEC_AAC, 112000,144000, 48000, 2, 11,11,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */
+ { CODEC_AAC, 144000,256001, 48000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 192 */
+
/** AAC LOW DELAY SECTION **/
+ /*** 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 */
+ { CODEC_AACLD, 18000, 22000, 16000, 1, 7, 7,12,12, 1, 6, 9, SBR_MONO, 3 }, /* nominal: 20 kbit/s wrr: tuned */
+ { CODEC_AACLD, 22000, 28000, 16000, 1, 6, 6, 9, 9, 2, 3, 6, SBR_MONO, 3 }, /* nominal: 24 kbit/s wrr: tuned */
+ { CODEC_AACLD, 28000, 36000, 16000, 1, 8, 8,12, 7, 2, 9,12, SBR_MONO, 3 }, /* jgr: special */ /* wrr: tuned */
+ { CODEC_AACLD, 36000, 44000, 16000, 1, 10,14,12,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+ { CODEC_AACLD, 44000, 64001, 16000, 1, 11,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+
/* 22.05/44.1 kHz dual rate */
- { 18000, 22000, 22050, 1, 4, 4, 5, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
- { 22000, 28000, 22050, 1, 4, 4, 6, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 22050, 1, 7, 8, 8, 8, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 22050, 1, 9, 9, 9, 9, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 22050, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
- { 52000, 64001, 22050, 1, 12,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AACLD, 18000, 22000, 22050, 1, 4, 4, 5, 5, 2, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */
+ { CODEC_AACLD, 22000, 28000, 22050, 1, 5, 5, 6, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+ { CODEC_AACLD, 28000, 36000, 22050, 1, 7, 8, 8, 8, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AACLD, 36000, 44000, 22050, 1, 9, 9, 9, 9, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+ { CODEC_AACLD, 44000, 52000, 22050, 1, 12,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+ { 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 */
- { 20000, 22000, 24000, 1, 4, 4, 5, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
- { 22000, 28000, 24000, 1, 4, 4, 6, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
- { 28000, 36000, 24000, 1, 6, 8, 8, 8, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 24000, 1, 8, 9, 9, 9, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 24000, 1, 12,11,11,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
- { 52000, 64001, 24000, 1, 13,11,11,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+ { CODEC_AACLD, 20000, 22000, 24000, 1, 4, 1, 8, 4, 2, 3, 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, 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 */
+ { CODEC_AACLD, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range */
+ { CODEC_AACLD, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range */
+ { CODEC_AACLD, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */
+ { CODEC_AACLD, 72000,100000, 32000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
+ { CODEC_AACLD, 100000,160001, 32000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
+
+ /* 44/88 kHz dual rate */ /* not yet finally tuned */
+ { CODEC_AACLD, 36000, 60000, 44100, 1, 8, 7, 6, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
+ { CODEC_AACLD, 60000, 72000, 44100, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
+ { CODEC_AACLD, 72000,100000, 44100, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
+ { 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, 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 */
+
+ /*** stereo ***/
+ /* 16/32 kHz dual rate not yet tuned ->alb copied from non LD tables*/
+ { CODEC_AACLD, 32000, 36000, 16000, 2, 10, 9,12,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AACLD, 36000, 44000, 16000, 2, 13,13,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AACLD, 44000, 52000, 16000, 2, 10, 9,11, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* tune12 nominal: 48 kbit/s */
+ { CODEC_AACLD, 52000, 60000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AACLD, 60000, 76000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AACLD, 76000,128001, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
/* 22.05/44.1 kHz dual rate */
- { 32000, 36000, 22050, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 22050, 2, 5, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 22050, 2, 7,10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
- { 52000, 60000, 22050, 2, 9,11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
- { 60000, 76000, 22050, 2, 10,12,10,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
- { 76000, 82000, 22050, 2, 12,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
- { 82000,128001, 22050, 2, 13,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AACLD, 32000, 36000, 22050, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AACLD, 36000, 44000, 22050, 2, 5, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AACLD, 44000, 52000, 22050, 2, 7,10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+ { CODEC_AACLD, 52000, 60000, 22050, 2, 9,11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AACLD, 60000, 76000, 22050, 2, 10,12,10,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AACLD, 76000, 82000, 22050, 2, 12,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AACLD, 82000,128001, 22050, 2, 13,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
/* 24/48 kHz dual rate */
- { 32000, 36000, 24000, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
- { 36000, 44000, 24000, 2, 4, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
- { 44000, 52000, 24000, 2, 6,10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
- { 52000, 60000, 24000, 2, 9,11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
- { 60000, 76000, 24000, 2, 11,12,10,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
- { 76000, 88000, 24000, 2, 12,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
- { 88000,128001, 24000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 92 kbit/s */
-
+ { CODEC_AACLD, 32000, 36000, 24000, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+ { CODEC_AACLD, 36000, 44000, 24000, 2, 4, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+ { CODEC_AACLD, 44000, 52000, 24000, 2, 6,10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+ { CODEC_AACLD, 52000, 60000, 24000, 2, 9,11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+ { CODEC_AACLD, 60000, 76000, 24000, 2, 11,12,10,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+ { CODEC_AACLD, 76000, 88000, 24000, 2, 12,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+ { CODEC_AACLD, 88000,128001, 24000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 92 kbit/s */
+
+ /* 32/64 kHz dual rate */ /* placebo settings */ /*jgr: new, copy from CODEC_AAC */
+ { CODEC_AACLD, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+ { CODEC_AACLD, 80000,112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
+ { CODEC_AACLD, 112000,144000, 32000, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
+ { CODEC_AACLD, 144000,256001, 32000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+
+ /* 44.1/88.2 kHz dual rate */ /* placebo settings */ /*wrr: new, copy from CODEC_AAC */
+ { CODEC_AACLD, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+ { CODEC_AACLD, 80000,112000, 44100, 2, 10,10, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 11-08-29 */
+ { CODEC_AACLD, 112000,144000, 44100, 2, 12,12,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 11-08-29 */
+ { CODEC_AACLD, 144000,256001, 44100, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+
+ /* 48/96 kHz dual rate */ /* not yet finally tuned */ /*wrr: new, copy from CODEC_AAC */
+ { CODEC_AACLD, 60000, 80000, 48000, 2, 7, 7,10,10, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */
+ { CODEC_AACLD, 80000,112000, 48000, 2, 9, 9,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */
+ { CODEC_AACLD, 112000,144000, 48000, 2, 11,11,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */
+ { CODEC_AACLD, 144000,176000, 48000, 2, 12,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 09-10-19 */
+ { CODEC_AACLD, 176000,256001, 48000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 09-10-19 */
};
+const int sbrTuningTableSize = sizeof(sbrTuningTable)/sizeof(sbrTuningTable[0]);
+
const psTuningTable_t psTuningTable[4] =
{
{ 8000, 22000, PSENC_STEREO_BANDS_10, PSENC_NENV_1, FL2FXCONST_DBL(3.0f/4.0f) },
diff --git a/libSBRenc/src/sbr_rom.h b/libSBRenc/src/sbr_rom.h
index fbbdbba..afa924e 100644
--- a/libSBRenc/src/sbr_rom.h
+++ b/libSBRenc/src/sbr_rom.h
@@ -118,13 +118,8 @@ extern const UCHAR v_Huff_NoiseLevelL11T[63];
extern const INT bookSbrNoiseBalanceC11T[25];
extern const UCHAR bookSbrNoiseBalanceL11T[25];
-#define SBRENC_AACLC_TUNING_SIZE 124
-#define SBRENC_AACELD_TUNING_SIZE (26)
-#define SBRENC_AACELD2_TUNING_SIZE (26)
-
-#define SBRENC_TUNING_SIZE (SBRENC_AACLC_TUNING_SIZE + SBRENC_AACELD_TUNING_SIZE)
-
-extern const sbrTuningTable_t sbrTuningTable[SBRENC_TUNING_SIZE];
+extern const sbrTuningTable_t sbrTuningTable[];
+extern const int sbrTuningTableSize;
extern const psTuningTable_t psTuningTable[4];
diff --git a/libSBRenc/src/sbrenc_freq_sca.cpp b/libSBRenc/src/sbrenc_freq_sca.cpp
index b0b04fa..30bc5ca 100644
--- a/libSBRenc/src/sbrenc_freq_sca.cpp
+++ b/libSBRenc/src/sbrenc_freq_sca.cpp
@@ -84,6 +84,7 @@ amm-info@iis.fraunhofer.de
/*!
\file
\brief frequency scale
+ \author Tobias Chalupka
*/
#include "sbrenc_freq_sca.h"
@@ -92,10 +93,10 @@ amm-info@iis.fraunhofer.de
#include "genericStds.h"
/* StartFreq */
-static INT getStartFreq(INT fs, const INT start_freq);
+static INT getStartFreq(INT fsCore, const INT start_freq);
/* StopFreq */
-static INT getStopFreq(INT fs, const INT stop_freq, const INT noChannels);
+static INT getStopFreq(INT fsCore, const INT stop_freq);
static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor);
static void CalcBands(INT * diff, INT start , INT stop , INT num_bands);
@@ -115,7 +116,7 @@ static void cumSum(INT start_value, INT* diff, INT length, UCHAR *start_adress)
*******************************************************************************/
INT
-FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs)
+FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore)
{
INT result;
@@ -123,9 +124,9 @@ FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs)
return -1;
}
/* Update startFreq struct */
- result = getStartFreq(fs, startFreq);
+ result = getStartFreq(fsCore, startFreq);
- result = (result*fs/QMFbands+1)>>1;
+ result = (result*(fsCore>>5)+1)>>1; /* (result*fsSBR/QMFbands+1)>>1; */
return (result);
@@ -141,17 +142,16 @@ FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs)
Return:
*******************************************************************************/
-INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT QMFbands, INT fs)
+INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT fsCore)
{
INT result;
if ( stopFreq < 0 || stopFreq > 13)
return -1;
-
/* Uppdate stopFreq struct */
- result = getStopFreq( fs, stopFreq, QMFbands);
- result = (result*fs/QMFbands+1)>>1;
+ result = getStopFreq(fsCore, stopFreq);
+ result = (result*(fsCore>>5)+1)>>1; /* (result*fsSBR/QMFbands+1)>>1; */
return (result);
} /* End getSbrStopFreq */
@@ -162,69 +162,73 @@ INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT QMFbands, INT fs)
*******************************************************************************
Description:
- Arguments:
+ Arguments: fsCore - core sampling rate
+
Return:
*******************************************************************************/
static INT
-getStartFreq(INT fs, const INT start_freq)
+getStartFreq(INT fsCore, const INT start_freq)
{
INT k0_min;
- switch(fs){
- case 16000: k0_min = 24;
+ switch(fsCore){
+ case 8000: k0_min = 24; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
break;
- case 22050: k0_min = 17;
+ case 11025: k0_min = 17; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
break;
- case 24000: k0_min = 16;
+ case 12000: k0_min = 16; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
break;
- case 32000: k0_min = 16;
+ case 16000: k0_min = 16; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
break;
- case 44100: k0_min = 12;
+ case 22050: k0_min = 12; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
break;
- case 48000: k0_min = 11;
+ case 24000: k0_min = 11; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
break;
- case 64000: k0_min = 10;
+ case 32000: k0_min = 10; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
break;
- case 88200: k0_min = 7;
+ case 44100: k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
break;
- case 96000: k0_min = 7;
+ case 48000: k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
+ break;
+ case 96000: k0_min = 3; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
break;
default:
k0_min=11; /* illegal fs */
}
- switch (fs) {
+ switch (fsCore) {
- case 16000:
+ case 8000:
{
INT v_offset[]= {-8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7};
return (k0_min + v_offset[start_freq]);
}
- case 22050:
+ case 11025:
{
INT v_offset[]= {-5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13};
return (k0_min + v_offset[start_freq]);
}
- case 24000:
+ case 12000:
{
INT v_offset[]= {-5, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16};
return (k0_min + v_offset[start_freq]);
}
- case 32000:
+ case 16000:
{
INT v_offset[]= {-6, -4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16};
return (k0_min + v_offset[start_freq]);
}
- case 44100:
- case 48000:
- case 64000:
+ case 22050:
+ case 24000:
+ case 32000:
{
INT v_offset[]= {-4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20};
return (k0_min + v_offset[start_freq]);
}
- case 88200:
+ case 44100:
+ case 48000:
case 96000:
{
INT v_offset[]= {-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24};
@@ -249,13 +253,12 @@ getStartFreq(INT fs, const INT start_freq)
Return:
*******************************************************************************/
static INT
-getStopFreq(INT fs, const INT stop_freq, const INT noChannels)
+getStopFreq(INT fsCore, const INT stop_freq)
{
INT result,i;
INT k1_min;
INT v_dstop[13];
-
INT *v_stop_freq = NULL;
INT v_stop_freq_16[14] = {48,49,50,51,52,54,55,56,57,59,60,61,63,64};
INT v_stop_freq_22[14] = {35,37,38,40,42,44,46,48,51,53,56,58,61,64};
@@ -266,40 +269,45 @@ getStopFreq(INT fs, const INT stop_freq, const INT noChannels)
INT v_stop_freq_64[14] = {20,22,24,26,29,31,34,37,41,45,49,54,59,64};
INT v_stop_freq_88[14] = {15,17,19,21,23,26,29,33,37,41,46,51,57,64};
INT v_stop_freq_96[14] = {13,15,17,19,21,24,27,31,35,39,44,50,57,64};
+ INT v_stop_freq_192[14] = {7, 8,10,12,14,16,19,23,27,32,38,46,54,64};
- switch(fs){
- case 16000: k1_min = 48;
+ switch(fsCore){
+ case 8000: k1_min = 48;
v_stop_freq =v_stop_freq_16;
break;
- case 22050: k1_min = 35;
+ case 11025: k1_min = 35;
v_stop_freq =v_stop_freq_22;
break;
- case 24000: k1_min = 32;
+ case 12000: k1_min = 32;
v_stop_freq =v_stop_freq_24;
break;
- case 32000: k1_min = 32;
+ case 16000: k1_min = 32;
v_stop_freq =v_stop_freq_32;
break;
- case 44100: k1_min = 23;
+ case 22050: k1_min = 23;
v_stop_freq =v_stop_freq_44;
break;
- case 48000: k1_min = 21;
+ case 24000: k1_min = 21;
v_stop_freq =v_stop_freq_48;
break;
- case 64000: k1_min = 20;
+ case 32000: k1_min = 20;
v_stop_freq =v_stop_freq_64;
break;
- case 88200: k1_min = 15;
+ case 44100: k1_min = 15;
v_stop_freq =v_stop_freq_88;
break;
- case 96000: k1_min = 13;
+ case 48000: k1_min = 13;
v_stop_freq =v_stop_freq_96;
break;
+ case 96000: k1_min = 7;
+ v_stop_freq =v_stop_freq_192;
+ break;
default:
k1_min = 21; /* illegal fs */
}
-
+ /* if no valid core samplingrate is used this loop produces
+ a segfault, because v_stop_freq is not initialized */
/* Ensure increasing bandwidth */
for(i = 0; i <= 12; i++) {
v_dstop[i] = v_stop_freq[i+1] - v_stop_freq[i];
@@ -322,34 +330,41 @@ getStopFreq(INT fs, const INT stop_freq, const INT noChannels)
*******************************************************************************
Description:
- Arguments:
+ Arguments: srSbr SBR sampling freqency
+ srCore AAC core sampling freqency
+ noChannels Number of QMF channels
+ startFreq SBR start frequency in QMF bands
+ stopFreq SBR start frequency in QMF bands
- Return:
+ *k0 Output parameter
+ *k2 Output parameter
+
+ Return: Error code (0 is OK)
*******************************************************************************/
INT
-FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
- const INT noChannels,
- const INT startFreq,
- const INT stopFreq,
- const SR_MODE sampleRateMode,
- INT *k0,
- INT *k2)
+FDKsbrEnc_FindStartAndStopBand(
+ const INT srSbr,
+ const INT srCore,
+ const INT noChannels,
+ const INT startFreq,
+ const INT stopFreq,
+ INT *k0,
+ INT *k2
+ )
{
/* Update startFreq struct */
- *k0 = getStartFreq(samplingFreq, startFreq);
+ *k0 = getStartFreq(srCore, startFreq);
/* Test if start freq is outside corecoder range */
- if( ( sampleRateMode == 1 ) &&
- ( samplingFreq*noChannels <
- 2**k0 * samplingFreq) ) {
+ if( srSbr*noChannels < *k0 * srCore ) {
return (1); /* raise the cross-over frequency and/or lower the number
of target bands per octave (or lower the sampling frequency) */
}
/*Update stopFreq struct */
if ( stopFreq < 14 ) {
- *k2 = getStopFreq(samplingFreq, stopFreq, noChannels);
+ *k2 = getStopFreq(srCore, stopFreq);
} else if( stopFreq == 14 ) {
*k2 = 2 * *k0;
} else {
@@ -364,10 +379,10 @@ FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
/* Test for invalid k0 k2 combinations */
- if ( (samplingFreq == 44100) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS44100 ) )
+ if ( (srCore == 22050) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS44100 ) )
return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs=44.1kHz */
- if ( (samplingFreq >= 48000) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS48000 ) )
+ if ( (srCore >= 24000) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS48000 ) )
return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs>=48kHz */
if ((*k2 - *k0) > MAX_FREQ_COEFFS)
@@ -390,15 +405,19 @@ FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
Return:
*******************************************************************************/
INT
-FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands,
- const INT k0, const INT k2,
- const INT freqScale,
- const INT alterScale)
+FDKsbrEnc_UpdateFreqScale(
+ UCHAR *v_k_master,
+ INT *h_num_bands,
+ const INT k0,
+ const INT k2,
+ const INT freqScale,
+ const INT alterScale
+ )
{
INT b_p_o = 0; /* bands_per_octave */
- FIXP_DBL warp = FL2FXCONST_DBL(0.0f);
+ FIXP_DBL warp = FL2FXCONST_DBL(0.0f);
INT dk = 0;
/* Internal variables */
@@ -426,7 +445,7 @@ FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands,
warp = FL2FXCONST_DBL(1.0f/2.6f); /* 1.0/(1.3*2.0); */
- if(4*k2 >= 9*k0) /*two or more regions*/
+ if(4*k2 >= 9*k0) /*two or more regions (how many times the basis band is copied)*/
{
k1=2*k0;
@@ -592,30 +611,31 @@ modifyBands(INT max_band_previous, INT * diff, INT length)
*******************************************************************************
Description:
+
Arguments:
Return:
*******************************************************************************/
INT
-FDKsbrEnc_UpdateHiRes(UCHAR *h_hires, INT *num_hires,UCHAR * v_k_master,
- INT num_master , INT *xover_band, SR_MODE drOrSr,
- INT noQMFChannels)
+FDKsbrEnc_UpdateHiRes(
+ UCHAR *h_hires,
+ INT *num_hires,
+ UCHAR *v_k_master,
+ INT num_master,
+ INT *xover_band
+ )
{
INT i;
- INT divider;
INT max1,max2;
- /* Check if we use a Dual rate => diver=2 else 1 */
- divider = (drOrSr == DUAL_RATE) ? 2 : 1;
-
- if( (v_k_master[*xover_band] > (noQMFChannels/divider) ) ||
+ if( (v_k_master[*xover_band] > 32 ) || /* v_k_master[*xover_band] > noQMFChannels(dualRate)/divider */
( *xover_band > num_master ) ) {
/* xover_band error, too big for this startFreq. Will be clipped */
/* Calculate maximum value for xover_band */
max1=0;
max2=num_master;
- while( (v_k_master[max1+1] < (noQMFChannels/divider)) &&
+ while( (v_k_master[max1+1] < 32 ) && /* noQMFChannels(dualRate)/divider */
( (max1+1) < max2) )
{
max1++;
diff --git a/libSBRenc/src/sbrenc_freq_sca.h b/libSBRenc/src/sbrenc_freq_sca.h
index 51ed688..6f2bb84 100644
--- a/libSBRenc/src/sbrenc_freq_sca.h
+++ b/libSBRenc/src/sbrenc_freq_sca.h
@@ -96,34 +96,42 @@ amm-info@iis.fraunhofer.de
INT
-FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands,
- const INT k0, const INT k2,
- const INT freq_scale,
- const INT alter_scale);
+FDKsbrEnc_UpdateFreqScale(
+ UCHAR *v_k_master,
+ INT *h_num_bands,
+ const INT k0,
+ const INT k2,
+ const INT freq_scale,
+ const INT alter_scale
+ );
INT
-FDKsbrEnc_UpdateHiRes(UCHAR *h_hires,
- INT *num_hires,
- UCHAR *v_k_master,
- INT num_master ,
- INT *xover_band,
- SR_MODE drOrSr,
- INT noQMFChannels);
-
-void FDKsbrEnc_UpdateLoRes(UCHAR * v_lores,
- INT *num_lores,
- UCHAR * v_hires,
- INT num_hires);
+FDKsbrEnc_UpdateHiRes(
+ UCHAR *h_hires,
+ INT *num_hires,
+ UCHAR *v_k_master,
+ INT num_master,
+ INT *xover_band
+ );
+
+void FDKsbrEnc_UpdateLoRes(
+ UCHAR *v_lores,
+ INT *num_lores,
+ UCHAR *v_hires,
+ INT num_hires
+ );
INT
-FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
- const INT noChannels,
- const INT startFreq,
- const INT stop_freq,
- const SR_MODE sampleRateMode,
- INT *k0,
- INT *k2);
-
-INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs );
-INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT QMFbands, INT fs);
+FDKsbrEnc_FindStartAndStopBand(
+ const INT srSbr,
+ const INT srCore,
+ const INT noChannels,
+ const INT startFreq,
+ const INT stop_freq,
+ INT *k0,
+ INT *k2
+ );
+
+INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore);
+INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT fsCore);
#endif
diff --git a/libSBRenc/src/ton_corr.cpp b/libSBRenc/src/ton_corr.cpp
index 134a916..224da11 100644
--- a/libSBRenc/src/ton_corr.cpp
+++ b/libSBRenc/src/ton_corr.cpp
@@ -303,8 +303,6 @@ FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< H
}
}
- FDK_ASSERT(noEstPerFrame == 2);
-
C_ALLOC_SCRATCH_END(realBuf, FIXP_DBL, 2*BAND_V_SIZE*NUM_V_COMBINE);
C_ALLOC_SCRATCH_END(ac, ACORR_COEFS, 1);
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-23 11:14:51 +0000