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 */

libAACenc/src/aacenc_lib.cpp

@@ -98,7 +98,7 @@ amm-info@iis.fraunhofer.de
 /* Encoder library info */
 #define AACENCODER_LIB_VL0 3
 #define AACENCODER_LIB_VL1 4
-#define AACENCODER_LIB_VL2 8
+#define AACENCODER_LIB_VL2 9
 #define AACENCODER_LIB_TITLE "AAC Encoder"
 #define AACENCODER_LIB_BUILD_DATE __DATE__
 #define AACENCODER_LIB_BUILD_TIME __TIME__
@@ -118,7 +118,7 @@ amm-info@iis.fraunhofer.de
 #define SBL(fl)            (fl/8)                 /*!< Short block length (hardcoded to 8 short blocks per long block) */
 #define BSLA(fl)           (4*SBL(fl)+SBL(fl)/2)  /*!< AAC block switching look-ahead */
 #define DELAY_AAC(fl)      (fl+BSLA(fl))          /*!< MDCT + blockswitching */
-#define DELAY_AACELD(fl)   ( (fl) + ((fl)/2)  )   /*!< ELD FB delay */
+#define DELAY_AACELD(fl)   ((fl)/2)               /*!< ELD FB delay (no framing delay included) */
 
 #define INPUTBUFFER_SIZE (1537+100+2048)
 
@@ -166,7 +166,8 @@ typedef struct {
 
     UCHAR             userMetaDataMode;      /*!< Meta data library configuration. */
 
-    UCHAR             userSbrEnabled;
+    UCHAR             userSbrEnabled;        /*!< Enable SBR for ELD. */
+    UINT              userSbrRatio;          /*!< SBR sampling rate ratio. Dual- or single-rate. */
 
 } USER_PARAM;
 
@@ -229,7 +230,7 @@ struct AACENCODER
 
    UINT                      encoder_modis;
 
-   /* Capabity flags */
+   /* Capability flags */
    UINT                      CAPF_tpEnc;
 
 } ;
@@ -255,6 +256,20 @@ static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig)
     return ( sbrUsed );
 }
 
+static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType)
+{
+    INT psUsed = 0;
+
+    if ( (audioObjectType==AOT_PS)
+      || (audioObjectType==AOT_MP2_PS)
+      || (audioObjectType==AOT_DABPLUS_PS)
+      || (audioObjectType==AOT_DRM_MPEG_PS) )
+    {
+        psUsed = 1;
+    }
+
+    return ( psUsed );
+}
 /****************************************************************************
                                Allocate Encoder
 ****************************************************************************/
@@ -449,6 +464,14 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig,
 
     config->userAncDataRate    = 0;
 
+    /* SBR rate is set to 0 here, which means it should be set automatically
+       in FDKaacEnc_AdjustEncSettings() if the user did not set a rate
+       expilicitely. */
+    config->userSbrRatio = 0;
+
+    /* SBR enable set to -1 means to inquire ELD audio configurator for reasonable configuration. */
+    config->userSbrEnabled     = -1;
+
     return AAC_ENC_OK;
 }
 
@@ -483,6 +506,7 @@ INT aacEncoder_LimitBitrate(
         INT bitRate,
         const INT nSubFrames,
         const INT sbrActive,
+        const INT sbrDownSampleRate,
         const AUDIO_OBJECT_TYPE aot
         )
 {
@@ -492,8 +516,7 @@ INT aacEncoder_LimitBitrate(
   FDKaacEnc_InitChannelMapping(channelMode, CH_ORDER_MPEG, &cm);
 
   if (sbrActive) {
-    /* Assume SBR rate ratio of 2:1 */
-    coreSamplingRate = samplingRate / 2;
+    coreSamplingRate = samplingRate >>  (sbrEncoder_IsSingleRatePossible(aot) ? (sbrDownSampleRate-1):1);
   } else {
     coreSamplingRate = samplingRate;
   }
@@ -669,19 +692,6 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
           break;
     }
 
-    /* We need the frame length to call aacEncoder_LimitBitrate() */
-    hAacConfig->bitRate = aacEncoder_LimitBitrate(
-              NULL,
-              hAacConfig->sampleRate,
-              hAacConfig->framelength,
-              hAacConfig->nChannels,
-              hAacConfig->channelMode,
-              config->userBitrate,
-              hAacConfig->nSubFrames,
-              isSbrActive(hAacConfig),
-              hAacConfig->audioObjectType
-              );
-
     switch ( hAacConfig->audioObjectType ) {
       case AOT_ER_AAC_LD:
       case AOT_ER_AAC_ELD:
@@ -699,24 +709,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
         break;
     }
 
-    if (hAacConfig->epConfig >= 0) {
-        hAacConfig->syntaxFlags |= AC_ER;
-         if (((INT)hAacConfig->channelMode < 1) || ((INT)hAacConfig->channelMode > 7)) {
-           return AACENC_INVALID_CONFIG;        /* Cannel config 0 not supported. */
-         }
-    }
-
-    if ( FDKaacEnc_DetermineEncoderMode(&hAacConfig->channelMode, hAacConfig->nChannels) != AAC_ENC_OK) {
-        return AACENC_INVALID_CONFIG;        /* nChannels doesn't match chMode, this is just a check-up */
-    }
-
-    if ( (hAacConfig->nChannels > hAacEncoder->nMaxAacChannels)
-      || ( (FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannelsEff > hAacEncoder->nMaxSbrChannels) &&
-            isSbrActive(hAacConfig) )
-         )
-    {
-        return AACENC_INVALID_CONFIG;      /* not enough channels allocated */
-    }
+    hAacConfig->bitRate = config->userBitrate;
 
     /* get bitrate in VBR configuration */
     if ( (hAacConfig->bitrateMode>=1) && (hAacConfig->bitrateMode<=5) ) {
@@ -727,25 +720,52 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
 
 
     /* Set default bitrate if no external bitrate declared. */
-    if (hAacConfig->bitRate==-1) {
+    if ( (hAacConfig->bitrateMode==0) && (config->userBitrate==(UINT)-1) ) {
         INT bitrate = FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannelsEff * hAacConfig->sampleRate;
-        switch (hAacConfig->audioObjectType)
+
+        if ( isPsActive(hAacConfig->audioObjectType) ) {
+          hAacConfig->bitRate = (bitrate>>1);                  /* 0.5 bit per sample */
+        }
+        else if ( isSbrActive(hAacConfig) )
+        {
+          if ( (config->userSbrRatio==2) || ((config->userSbrRatio==0)&&(hAacConfig->audioObjectType!=AOT_ER_AAC_ELD)) ) {
+            hAacConfig->bitRate = (bitrate + (bitrate>>2))>>1; /* 0.625 bits per sample */
+          }
+          if ( (config->userSbrRatio==1) || ((config->userSbrRatio==0)&&(hAacConfig->audioObjectType==AOT_ER_AAC_ELD)) ) {
+            hAacConfig->bitRate = (bitrate + (bitrate>>3));    /* 1.125 bits per sample */
+          }
+        } else
         {
-            case AOT_AAC_LC:
                 hAacConfig->bitRate = bitrate + (bitrate>>1);        /* 1.5 bits per sample */
-                break;
-            case AOT_SBR:
-                hAacConfig->bitRate = (bitrate + (bitrate>>2))>>1;   /* 0.625 bits per sample */
-                break;
-            case AOT_PS:
-                hAacConfig->bitRate = (bitrate>>1);                  /* 0.5 bit per sample */
-                break;
-            default:
-                hAacConfig->bitRate = bitrate;
-                break;
         }
     }
 
+    /* Initialize SBR parameters */
+    if ( (config->userSbrRatio==0) && (isSbrActive(hAacConfig)) ) {
+      /* Automatic SBR ratio configuration
+       * - downsampled SBR for ELD
+       * - otherwise always dualrate SBR
+       */
+        hAacConfig->sbrRatio = (hAacConfig->audioObjectType==AOT_ER_AAC_ELD) ? 1 : 2;
+    }
+    else {
+      /* SBR ratio has been set by the user, so use it. */
+      hAacConfig->sbrRatio = config->userSbrRatio;
+    }
+    /* We need the frame length to call aacEncoder_LimitBitrate() */
+    hAacConfig->bitRate = aacEncoder_LimitBitrate(
+              NULL,
+              hAacConfig->sampleRate,
+              hAacConfig->framelength,
+              hAacConfig->nChannels,
+              hAacConfig->channelMode,
+              hAacConfig->bitRate,
+              hAacConfig->nSubFrames,
+              isSbrActive(hAacConfig),
+              hAacConfig->sbrRatio,
+              hAacConfig->audioObjectType
+              );
+
     /* Configure PNS */
     if ( ((hAacConfig->bitrateMode>=1) && (hAacConfig->bitrateMode<=5)) /* VBR without PNS. */
         || (hAacConfig->useTns == 0) )                                  /* TNS required. */
@@ -753,6 +773,25 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
         hAacConfig->usePns = 0;
     }
 
+    if (hAacConfig->epConfig >= 0) {
+        hAacConfig->syntaxFlags |= AC_ER;
+         if (((INT)hAacConfig->channelMode < 1) || ((INT)hAacConfig->channelMode > 7)) {
+           return AACENC_INVALID_CONFIG;        /* Cannel config 0 not supported. */
+         }
+    }
+
+    if ( FDKaacEnc_DetermineEncoderMode(&hAacConfig->channelMode, hAacConfig->nChannels) != AAC_ENC_OK) {
+        return AACENC_INVALID_CONFIG;        /* nChannels doesn't match chMode, this is just a check-up */
+    }
+
+    if ( (hAacConfig->nChannels > hAacEncoder->nMaxAacChannels)
+      || ( (FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannelsEff > hAacEncoder->nMaxSbrChannels) &&
+            isSbrActive(hAacConfig) )
+         )
+    {
+        return AACENC_INVALID_CONFIG;      /* not enough channels allocated */
+    }
+
     /* Meta data restriction. */
     switch (hAacConfig->audioObjectType)
     {
@@ -846,8 +885,6 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
         SBR_ELEMENT_INFO sbrElInfo[(6)];
         CHANNEL_MAPPING channelMapping;
 
-        AUDIO_OBJECT_TYPE aot = hAacConfig->audioObjectType;
-
         if ( FDKaacEnc_InitChannelMapping(hAacConfig->channelMode,
                                           hAacConfig->channelOrder,
                                          &channelMapping) != AAC_ENC_OK )
@@ -875,8 +912,9 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
                                 &aacBufferOffset,
                                 &hAacConfig->nChannels,
                                 &hAacConfig->sampleRate,
+                                &hAacConfig->sbrRatio,
                                 &frameLength,
-                                &hAacConfig->audioObjectType,
+                                 hAacConfig->audioObjectType,
                                 &hAacEncoder->nDelay,
                                  (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) ? 1 : TRANS_FAC,
                                  (config->userTpHeaderPeriod!=0xFF) ? config->userTpHeaderPeriod : DEFAULT_HEADER_PERIOD_REPETITION_RATE,
@@ -884,7 +922,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
                                 );
 
         /* Suppress AOT reconfiguration and check error status. */
-        if ( sbrError || (hAacConfig->audioObjectType!=aot) ) {
+        if (sbrError) {
             return AACENC_INIT_SBR_ERROR;
         }
 
@@ -953,7 +991,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
         INT inputDataDelay = DELAY_AAC(hAacConfig->framelength);
 
         if ( isSbrActive(hAacConfig) && hSbrEncoder!=NULL) {
-            inputDataDelay = 2*inputDataDelay + sbrEncoder_GetInputDataDelay(*hSbrEncoder);
+          inputDataDelay = hAacConfig->sbrRatio*inputDataDelay + sbrEncoder_GetInputDataDelay(*hSbrEncoder);
         }
 
         if ( FDK_MetadataEnc_Init(hAacEncoder->hMetadataEnc,
@@ -1684,6 +1722,16 @@ AACENC_ERROR aacEncoder_SetParam(
           }
         }
         break;
+    case AACENC_SBR_RATIO:
+        if (settings->userSbrRatio != value) {
+            if (! ((value==0) || (value==1) || (value==2)) ) {
+              err = AACENC_INVALID_CONFIG;
+              break;
+            }
+            settings->userSbrRatio = value;
+            hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT;
+        }
+        break;
     case AACENC_SBR_MODE:
         if (settings->userSbrEnabled != value) {
             settings->userSbrEnabled = value;
@@ -1805,7 +1853,7 @@ UINT aacEncoder_GetParam(
         value = (UINT)hAacEncoder->aacConfig.bitrateMode;
         break;
     case AACENC_SAMPLERATE:
-        value = (UINT)settings->userSamplerate;
+        value = (UINT)hAacEncoder->coderConfig.extSamplingRate;
         break;
     case AACENC_CHANNELMODE:
         value = (UINT)hAacEncoder->aacConfig.channelMode;
@@ -1822,6 +1870,9 @@ UINT aacEncoder_GetParam(
     case AACENC_GRANULE_LENGTH:
         value = (UINT)hAacEncoder->aacConfig.framelength;
        break;
+    case AACENC_SBR_RATIO:
+        value = isSbrActive(&hAacEncoder->aacConfig) ? hAacEncoder->aacConfig.sbrRatio : 0;
+        break;
     case AACENC_SBR_MODE:
         value = (UINT) (hAacEncoder->aacConfig.syntaxFlags & AC_SBR_PRESENT) ? 1 : 0;
         break;

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);

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

@@ -1244,25 +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) {

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;
     }
 
 

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_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;
 }

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) },

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.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++;

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

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);