Encoder downsampled SBR

* AAC-Encoder

   - Introduce optional AACENC_SBR_RATIO encoder API parameter to configure
     dualrate or downsampled SBR explicitely. ELD makes use of downsampled SBR
     in default configuration.
     Modified file(s):
        documentation\aacEncoder.pdf
        libAACenc\include\aacenc_lib.h
        libAACenc\src\aacenc.h
        libAACenc\src\aacenc_lib.cpp

* SBR-Encoder

   - Implement downsampled SBR feature.
   - Revise sbr tuning parameter selection.
     Modified file(s):
        libSBRenc\include\sbr_encoder.h
        libSBRenc\src\bit_sbr.h
        libSBRenc\src\env_est.cpp
        libSBRenc\src\mh_det.cpp
        libSBRenc\src\nf_est.cpp
        libSBRenc\src\ps_main.cpp
        libSBRenc\src\sbr.h
        libSBRenc\src\sbr_def.h
        libSBRenc\src\sbr_encoder.cpp
        libSBRenc\src\sbr_rom.cpp
        libSBRenc\src\sbr_rom.h
        libSBRenc\src\sbrenc_freq_sca.cpp
        libSBRenc\src\sbrenc_freq_sca.h
        libSBRenc\src\ton_corr.cpp

Bug 9428126

Change-Id: I731720a10829272acaaf70b84525df00a09ff3d2

libAACenc/include/aacenc_lib.h

@@ -316,7 +316,8 @@ if the parameter was not set from extern. The bitrate depends on the number of e
 channels and sampling rate and is determined as follows.
 \code
 AAC-LC (AOT_AAC_LC): 1.5 bits per sample
-HE-AAC (AOT_SBR): 0.625 bits per sample
+HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr)
+HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr)
 HE-AAC v2 (AOT_PS): 0.5 bits per sample
 \endcode
 
@@ -416,8 +417,9 @@ determines the maximum allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a li
 look-up table is used.
 
 A good working point in terms of audio quality, sampling rate and bitrate, is at 1 to 1.5
-bits/audio sample for AAC-LC, 0.625 bits/audio sample for HE-AAC and 0.5 bits/audio sample
-for HE-AAC v2. For example for one channel with a sampling frequency of 48 kHz, the range from
+bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate HE-AAC, 1.125 bits/audio sample
+for downsampled HE-AAC and 0.5 bits/audio sample for HE-AAC v2.
+For example for one channel with a sampling frequency of 48 kHz, the range from
 48 kbit/s to 72 kbit/s achieves reasonable audio quality for AAC-LC.
 
 For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 16 kHz because then the
@@ -434,7 +436,7 @@ quality at that bitrate than HE-AAC or HE-AAC v2.
 The following table provides an overview of recommended encoder configuration parameters
 which we determined by virtue of numerous listening tests.
 
-\subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2.
+\subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode.
 \verbatim
 -----------------------------------------------------------------------------------
 Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
@@ -457,8 +459,8 @@ AAC LC + SBR       |  64000 - 128000  |  32.00, 44.10, 48.00  |     48.00  |
 -------------------+------------------+-----------------------+------------+-------
 AAC LC + SBR       |  64000 -  69999  |  32.00, 44.10, 48.00  |     32.00  | 5, 5.1
 AAC LC + SBR       |  70000 - 159999  |  32.00, 44.10, 48.00  |     44.10  | 5, 5.1
-AAC LC + SBR       | 160000 - 319999  |  32.00, 44.10, 48.00  |     48.00  | 5, 5.1
-AAC LC + SBR       | 320000 - 640000  |  64.00, 88.20, 96.00  |     96.00  | 5, 5.1
+AAC LC + SBR       | 160000 - 245999  |  32.00, 44.10, 48.00  |     48.00  |      5
+AAC LC + SBR       | 160000 - 265999  |  32.00, 44.10, 48.00  |     48.00  |    5.1
 -------------------+------------------+-----------------------+------------+-------
 AAC LC             |   8000 -  15999  | 11.025, 12.00, 16.00  |     12.00  |      1
 AAC LC             |  16000 -  23999  |                16.00  |     16.00  |      1
@@ -481,7 +483,7 @@ AAC LC             | 280000 - 800000  |  32.00, 44.10, 48.00  |     44.10  | 5,
 -----------------------------------------------------------------------------------
 \endverbatim \n
 
-\subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR.
+\subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR mode.
 \verbatim
 -----------------------------------------------------------------------------------
 Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
@@ -489,18 +491,20 @@ Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No.
                    |                  |                [kHz]  |      Rate  |
                    |                  |                       |     [kHz]  |
 -------------------+------------------+-----------------------+------------+-------
-ELD + SBR          |  16000 -  24999  |        32.00 - 44.10  |     32.00  |      1
+ELD + SBR          |  18000 -  24999  |        32.00 - 44.10  |     32.00  |      1
 ELD + SBR          |  25000 -  31999  |        32.00 - 48.00  |     32.00  |      1
 ELD + SBR          |  32000 -  64000  |        32.00 - 48.00  |     48.00  |      1
 -------------------+------------------+-----------------------+------------+-------
 ELD + SBR          |  32000 -  51999  |        32.00 - 48.00  |     44.10  |      2
 ELD + SBR          |  52000 - 128000  |        32.00 - 48.00  |     48.00  |      2
 -------------------+------------------+-----------------------+------------+-------
-ELD + SBR          |  72000 - 192000  |        44.10 - 48.00  |     48.00  |      3
+ELD + SBR          |  72000 - 160000  |        44.10 - 48.00  |     48.00  |      3
 -------------------+------------------+-----------------------+------------+-------
-ELD + SBR          |  96000 - 256000  |        44.10 - 48.00  |     48.00  |      4
+ELD + SBR          |  96000 - 212000  |        44.10 - 48.00  |     48.00  |      4
 -------------------+------------------+-----------------------+------------+-------
-ELD + SBR          | 120000 - 320000  |        44.10 - 48.00  |     48.00  |      5
+ELD + SBR          | 120000 - 246000  |        44.10 - 48.00  |     48.00  |      5
+-------------------+------------------+-----------------------+------------+-------
+ELD + SBR          | 120000 - 266000  |        44.10 - 48.00  |     48.00  |    5.1
 -------------------+------------------+-----------------------+------------+-------
 LD, ELD            |  16000 -  19999  |        16.00 - 24.00  |     16.00  |      1
 LD, ELD            |  20000 -  39999  |        16.00 - 32.00  |     24.00  |      1
@@ -531,13 +535,33 @@ LD, ELD            | 340000 - 960000  |        44.10 - 48.00  |     48.00  |
 -----------------------------------------------------------------------------------
 \endverbatim \n
 
+\subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode.
+\verbatim
+-----------------------------------------------------------------------------------
+Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
+                   |         [bit/s]  |       Sampling Rates  |    Sampl.  |  Chan.
+                   |                  |                [kHz]  |      Rate  |
+                   |                  |                       |     [kHz]  |
+-------------------+------------------+-----------------------+------------+-------
+ELD + SBR          |  18000 -  24999  |        16.00 - 22.05  |     22.05  |      1
+(downsampled SBR)  |  25000 -  35999  |        22.05 - 32.00  |     24.00  |      1
+                   |  36000 -  64000  |        32.00 - 48.00  |     32.00  |      1
+-----------------------------------------------------------------------------------
+\endverbatim \n
+
+
 \page ENCODERBEHAVIOUR Encoder Behaviour
 
 \section BEHAVIOUR_BANDWIDTH Bandwidth
 
 The FDK AAC encoder usually does not use the full frequency range of the input signal, but restricts the bandwidth
 according to certain library-internal settings. They can be changed in the table "bandWidthTable" in the
-file bandwidth.cpp (if available), or via command-line argument "-w" (see chapter \ref CommandLineUsage).
+file bandwidth.cpp (if available).
+
+The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the bandwidth explicitly.
+\code
+aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, value);
+\endcode
 
 However it is not recommended to change these settings, because they are based on numerious listening
 tests and careful tweaks to ensure the best overall encoding quality.
@@ -638,7 +662,6 @@ an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature.
 If SBR is activated, the encoder automatically deactivates PNS internally. If TNS is disabled but PNS is allowed,
 the encoder deactivates PNS calculation internally.
 
-
 */
 
 #ifndef _AAC_ENC_LIB_H_
@@ -875,6 +898,15 @@ typedef enum
                                                   - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). (default)
                                                   - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR). */
 
+  AACENC_SBR_RATIO                = 0x0108,  /*!<  Controls activation of downsampled SBR. With downsampled SBR, the delay will be
+                                                   shorter. On the other hand, for achieving the same quality level, downsampled SBR
+                                                   needs more bits than dual-rate SBR.
+                                                   With downsampled SBR, the AAC encoder will work at the same sampling rate as the
+                                                   SBR encoder (single rate).
+                                                   Downsampled SBR is supported for AAC-ELD and HE-AACv1.
+                                                   - 1: Downsampled SBR (default for ELD).
+                                                   - 2: Dual-rate SBR   (default for HE-AAC). */
+
   AACENC_AFTERBURNER              = 0x0200,  /*!< This parameter controls the use of the afterburner feature.
                                                   The afterburner is a type of analysis by synthesis algorithm which increases the
                                                   audio quality but also the required processing power. It is recommended to always

libAACenc/src/aacenc.h

@@ -204,6 +204,8 @@ struct AACENC_CONFIG {
   INT   maxBitsPerFrame;        /* maximum number of bits in AU */
   INT   bitreservoir;           /* size of bitreservoir */
 
+  UINT  sbrRatio;               /* sbr sampling rate ratio: dual- or single-rate */
+
   UCHAR useTns;                 /* flag: use temporal noise shaping */
   UCHAR usePns;                 /* flag: use perceptual noise substitution */
   UCHAR useIS;                  /* flag: use intensity coding */

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.
@@ -303,16 +322,18 @@ 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,
+INT sbrEncoder_Init(
+        HANDLE_SBR_ENCODER   hSbrEncoder,
         SBR_ELEMENT_INFO     elInfo[(6)],
         int                  noElements,
         INT_PCM             *inputBuffer,
-                     INT   *bandwidth,
-                     INT   *bufferOffset,
+        INT                 *coreBandwidth,
+        INT                 *inputBufferOffset,
         INT                 *numChannels,
         INT                 *sampleRate,
+        UINT                *downSampleFactor,
         INT                 *frameLength,
-                     AUDIO_OBJECT_TYPE *aot,
+        AUDIO_OBJECT_TYPE    aot,
         int                 *delay,
         int                  transformFactor,
         const int            headerPeriod,
@@ -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);

libSBRenc/src/bit_sbr.h

@@ -124,11 +124,6 @@ struct SBR_HEADER_DATA
   INT alterScale;
   INT freqScale;
 
-  /*
-    element of sbrdata
-  */
-  SR_MODE sampleRateMode;
-
   /*
     element of channelpairelement
   */

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++) {

libSBRenc/src/mh_det.cpp

@@ -1244,26 +1244,39 @@ FDKsbrEnc_InitSbrMissingHarmonicsDetector (
 
   FDK_ASSERT(totNoEst <= MAX_NO_OF_ESTIMATES);
 
+  if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+  {
     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:
+    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) {
     hs->mhParams = &paramsAacLd;

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)
 
 
 
@@ -484,10 +484,12 @@ 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 */
     exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp);
     tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp);
@@ -531,20 +533,26 @@ FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoise
         * 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 (h_sbrNoiseFloorEstimate->noNoiseBands > MAX_NUM_NOISE_COEFFS)
-          h_sbrNoiseFloorEstimate->noNoiseBands = MAX_NUM_NOISE_COEFFS;
 
-        if( h_sbrNoiseFloorEstimate->noNoiseBands==0)
-            h_sbrNoiseFloorEstimate->noNoiseBands=1;
+        if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) {
+          nNoiseBands = MAX_NUM_NOISE_COEFFS;
+        }
+
+        if( nNoiseBands == 0 ) {
+          nNoiseBands = 1;
+        }
+
+        h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands;
+
     }
 
 

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,

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 */

libSBRenc/src/sbr_def.h

@@ -268,13 +268,6 @@ typedef enum
 }
 INVF_MODE;
 
-typedef enum
-{
-  SINGLE_RATE,
-  DUAL_RATE
-}
-SR_MODE;
-
 typedef enum
 {
   FREQ_RES_LOW = 0,

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];
 

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,
+FDKsbrEnc_UpdateFreqScale(
+        UCHAR *v_k_master,
+        INT   *h_num_bands,
+        const  INT k0,
+        const  INT k2,
         const  INT freq_scale,
-                const INT alter_scale);
+        const  INT alter_scale
+        );
 
 INT
-FDKsbrEnc_UpdateHiRes(UCHAR *h_hires,
+FDKsbrEnc_UpdateHiRes(
+        UCHAR *h_hires,
         INT   *num_hires,
         UCHAR *v_k_master,
-            INT num_master ,
-            INT *xover_band,
-            SR_MODE drOrSr,
-            INT noQMFChannels);
+        INT    num_master,
+        INT   *xover_band
+        );
 
-void  FDKsbrEnc_UpdateLoRes(UCHAR * v_lores,
+void  FDKsbrEnc_UpdateLoRes(
+        UCHAR *v_lores,
         INT   *num_lores,
-                  UCHAR * v_hires,
-                  INT num_hires);
+        UCHAR *v_hires,
+        INT    num_hires
+        );
 
 INT
-FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
+FDKsbrEnc_FindStartAndStopBand(
+        const INT srSbr,
+        const INT srCore,
         const INT noChannels,
         const INT startFreq,
         const INT stop_freq,
-                     const SR_MODE sampleRateMode,
         INT *k0,
-                     INT *k2);
+        INT *k2
+        );
 
-INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs );
-INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT QMFbands, INT fs);
+INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore);
+INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT fsCore);
 #endif

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);