Update fdk to v2.0.2

9 files changed, 344 insertions, 85 deletions

diff --git a/fdk-aac/libFDK/include/FDK_qmf_domain.h b/fdk-aac/libFDK/include/FDK_qmf_domain.h
index 5c12682..0e83da3 100644
--- a/fdk-aac/libFDK/include/FDK_qmf_domain.h
+++ b/fdk-aac/libFDK/include/FDK_qmf_domain.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -123,11 +123,10 @@ typedef enum {
 #define QMF_WB_SECTION_SIZE (1024 * 2)
 
 H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore1, FIXP_DBL)
-H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore2, FIXP_DBL)
 H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore3, FIXP_DBL)
 H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore4, FIXP_DBL)
-H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore5, FIXP_DBL)
 H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore6, FIXP_DBL)
+H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore7, FIXP_DBL)
 
 #define QMF_DOMAIN_MAX_ANALYSIS_QMF_BANDS (64)
 #define QMF_DOMAIN_MAX_SYNTHESIS_QMF_BANDS (QMF_MAX_SYNTHESIS_BANDS)
@@ -145,15 +144,15 @@ H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore6, FIXP_DBL)
 #define QMF_DOMAIN_OV_TIMESLOTS_16 (3)
 #define QMF_DOMAIN_OV_TIMESLOTS_32 (6)
 
-H_ALLOC_MEM(AnaQmfStates, FIXP_QAS)
+H_ALLOC_MEM(AnaQmfStates, FIXP_DBL)
 H_ALLOC_MEM(SynQmfStates, FIXP_QSS)
 H_ALLOC_MEM(QmfSlotsReal, FIXP_DBL *)
 H_ALLOC_MEM(QmfSlotsImag, FIXP_DBL *)
 H_ALLOC_MEM(QmfOverlapBuffer, FIXP_DBL)
 
-H_ALLOC_MEM(AnaQmfStates16, FIXP_QAS)
-H_ALLOC_MEM(AnaQmfStates24, FIXP_QAS)
-H_ALLOC_MEM(AnaQmfStates32, FIXP_QAS)
+H_ALLOC_MEM(AnaQmfStates16, FIXP_DBL)
+H_ALLOC_MEM(AnaQmfStates24, FIXP_DBL)
+H_ALLOC_MEM(AnaQmfStates32, FIXP_DBL)
 H_ALLOC_MEM(QmfSlotsReal16, FIXP_DBL *)
 H_ALLOC_MEM(QmfSlotsReal32, FIXP_DBL *)
 H_ALLOC_MEM(QmfSlotsImag16, FIXP_DBL *)
@@ -161,8 +160,6 @@ H_ALLOC_MEM(QmfSlotsImag32, FIXP_DBL *)
 H_ALLOC_MEM(QmfOverlapBuffer16, FIXP_DBL)
 H_ALLOC_MEM(QmfOverlapBuffer32, FIXP_DBL)
 
-#define QDOM_PCM INT_PCM
-
 /**
  * Structure to hold the configuration data which is global whithin a QMF domain
  * instance.
@@ -182,9 +179,6 @@ typedef struct {
                   park a channel if only one processing channel is
                   available. */
   UCHAR parkChannel_requested;
-  QDOM_PCM
-  *TDinput; /*!< Pointer to time domain data used as input for the QMF
-               analysis. */
   FIXP_DBL *
       pWorkBuffer[QMF_MAX_WB_SECTIONS]; /*!< Pointerarray to volatile memory. */
   UINT flags; /*!< Flags to be set on all QMF analysis/synthesis filter
@@ -244,7 +238,7 @@ typedef struct {
                                 (workBuf_nTimeSlots * workBuf_nBands * CMPLX_MOD). */
   USHORT workBufferOffset;   /*!< Offset within work buffer. */
   USHORT workBufferSectSize; /*!< Size of work buffer section. */
-  FIXP_QAS *
+  FIXP_DBL *
       pAnaQmfStates; /*!< Pointer to QMF analysis states (persistent memory). */
   FIXP_DBL
   *pOverlapBuffer;        /*!< Pointer to QMF overlap/delay memory (persistent
diff --git a/fdk-aac/libFDK/include/fixpoint_math.h b/fdk-aac/libFDK/include/fixpoint_math.h
index 3805892..51df4d7 100644
--- a/fdk-aac/libFDK/include/fixpoint_math.h
+++ b/fdk-aac/libFDK/include/fixpoint_math.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -171,6 +171,19 @@ extern const FIXP_DBL invSqrtTab[SQRT_VALUES];
  * \return non-zero if (a_m*2^a_e) < (b_m*2^b_e), 0 otherwise
  */
 FDK_INLINE INT fIsLessThan(FIXP_DBL a_m, INT a_e, FIXP_DBL b_m, INT b_e) {
+  INT n;
+
+  n = fixnorm_D(a_m);
+  a_m <<= n;
+  a_e -= n;
+
+  n = fixnorm_D(b_m);
+  b_m <<= n;
+  b_e -= n;
+
+  if (a_m == (FIXP_DBL)0) a_e = b_e;
+  if (b_m == (FIXP_DBL)0) b_e = a_e;
+
   if (a_e > b_e) {
     return ((b_m >> fMin(a_e - b_e, DFRACT_BITS - 1)) > a_m);
   } else {
@@ -179,6 +192,19 @@ FDK_INLINE INT fIsLessThan(FIXP_DBL a_m, INT a_e, FIXP_DBL b_m, INT b_e) {
 }
 
 FDK_INLINE INT fIsLessThan(FIXP_SGL a_m, INT a_e, FIXP_SGL b_m, INT b_e) {
+  INT n;
+
+  n = fixnorm_S(a_m);
+  a_m <<= n;
+  a_e -= n;
+
+  n = fixnorm_S(b_m);
+  b_m <<= n;
+  b_e -= n;
+
+  if (a_m == (FIXP_SGL)0) a_e = b_e;
+  if (b_m == (FIXP_SGL)0) b_e = a_e;
+
   if (a_e > b_e) {
     return ((b_m >> fMin(a_e - b_e, FRACT_BITS - 1)) > a_m);
   } else {
@@ -545,15 +571,20 @@ inline INT fMultIceil(FIXP_DBL a, INT b) {
   m = fMultNorm(a, (FIXP_DBL)b, &m_e);
 
   if (m_e < (INT)0) {
-    if (m_e > (INT)-DFRACT_BITS) {
+    if (m_e > (INT) - (DFRACT_BITS - 1)) {
       mi = (m >> (-m_e));
       if ((LONG)m & ((1 << (-m_e)) - 1)) {
         mi = mi + (FIXP_DBL)1;
       }
     } else {
-      mi = (FIXP_DBL)1;
-      if (m < (FIXP_DBL)0) {
-        mi = (FIXP_DBL)0;
+      if (m > (FIXP_DBL)0) {
+        mi = (FIXP_DBL)1;
+      } else {
+        if ((m_e == -(DFRACT_BITS - 1)) && (m == (FIXP_DBL)MINVAL_DBL)) {
+          mi = (FIXP_DBL)-1;
+        } else {
+          mi = (FIXP_DBL)0;
+        }
       }
     }
   } else {
@@ -744,7 +775,7 @@ FIXP_DBL fPow(FIXP_DBL base_m, INT base_e, FIXP_DBL exp_m, INT exp_e,
 
 /**
  * \brief return (base_m * 2^base_e) ^ N
- * \param base_m mantissa of the base
+ * \param base_m mantissa of the base. Must not be negative.
  * \param base_e exponent of the base
  * \param N power to be calculated of the base
  * \param result_e pointer to a INT where the exponent of the result will be
diff --git a/fdk-aac/libFDK/include/mdct.h b/fdk-aac/libFDK/include/mdct.h
index 1382374..e7cf3ad 100644
--- a/fdk-aac/libFDK/include/mdct.h
+++ b/fdk-aac/libFDK/include/mdct.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -106,18 +106,16 @@ amm-info@iis.fraunhofer.de
 #include "common_fix.h"
 
 #define MDCT_OUT_HEADROOM 2 /* Output additional headroom */
-#define PCM_OUT_BITS SAMPLE_BITS
+
+#define PCM_OUT_BITS DFRACT_BITS
 #define PCM_OUT_HEADROOM 8 /* Must have the same values as DMXH_HEADROOM */
 
-#define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - SAMPLE_BITS))
+#define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - PCM_OUT_BITS))
 /* Refer to "Output word length" in ISO/IEC 14496-3:2008(E) 23.2.3.6 */
 #define MDCT_OUTPUT_GAIN 16
 
-#if (MDCT_OUTPUT_SCALE >= 0)
-#define IMDCT_SCALE(x) SATURATE_RIGHT_SHIFT(x, MDCT_OUTPUT_SCALE, PCM_OUT_BITS)
-#else
-#define IMDCT_SCALE(x) SATURATE_LEFT_SHIFT(x, -MDCT_OUTPUT_SCALE, PCM_OUT_BITS)
-#endif
+#define IMDCT_SCALE(x, s) \
+  SATURATE_RIGHT_SHIFT((x), ((s) + MDCT_OUTPUT_SCALE), PCM_OUT_BITS)
 #define IMDCT_SCALE_DBL(x) (FIXP_DBL)(x)
 #define IMDCT_SCALE_DBL_LSH1(x) SATURATE_LEFT_SHIFT_ALT((x), 1, DFRACT_BITS)
 
diff --git a/fdk-aac/libFDK/include/mips/abs_mips.h b/fdk-aac/libFDK/include/mips/abs_mips.h
index dbb2063..5644bc0 100644
--- a/fdk-aac/libFDK/include/mips/abs_mips.h
+++ b/fdk-aac/libFDK/include/mips/abs_mips.h
@@ -107,7 +107,7 @@ amm-info@iis.fraunhofer.de
 
 #if defined(__GNUC__) && defined(__mips__)
 
-#if defined(__mips_dsp)
+#if defined(__mips_dsp) && !defined(__mips16)
 #define FUNCTION_fixabs_D
 #define FUNCTION_fixabs_I
 #define FUNCTION_fixabs_S
diff --git a/fdk-aac/libFDK/include/mips/scale_mips.h b/fdk-aac/libFDK/include/mips/scale_mips.h
index 3c141fc..882ef35 100644
--- a/fdk-aac/libFDK/include/mips/scale_mips.h
+++ b/fdk-aac/libFDK/include/mips/scale_mips.h
@@ -103,7 +103,7 @@ amm-info@iis.fraunhofer.de
 #ifndef SCALE_MIPS_H
 #define SCALE_MIPS_H
 
-#if defined(__mips_dsp)
+#if defined(__mips_dsp) && !defined(__mips16)
 
 /*!
  *
diff --git a/fdk-aac/libFDK/include/qmf.h b/fdk-aac/libFDK/include/qmf.h
index 609c6f1..78770b3 100644
--- a/fdk-aac/libFDK/include/qmf.h
+++ b/fdk-aac/libFDK/include/qmf.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -116,6 +116,7 @@ amm-info@iis.fraunhofer.de
 
 #define FIXP_QAS FIXP_PCM
 #define QAS_BITS SAMPLE_BITS
+#define INT_PCM_QMFIN INT_PCM
 
 #define FIXP_QSS FIXP_DBL
 #define QSS_BITS DFRACT_BITS
@@ -201,66 +202,70 @@ struct QMF_FILTER_BANK {
 
 typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK;
 
+int qmfInitAnalysisFilterBank(
+    HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
+    FIXP_QAS *pFilterStates,      /*!< Pointer to filter state buffer */
+    int noCols,                   /*!< Number of time slots  */
+    int lsb,                      /*!< Number of lower bands */
+    int usb,                      /*!< Number of upper bands */
+    int no_channels,              /*!< Number of critically sampled bands */
+    int flags);                   /*!< Flags */
+#if SAMPLE_BITS == 16
+
+int qmfInitAnalysisFilterBank(
+    HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
+    FIXP_DBL *pFilterStates,      /*!< Pointer to filter state buffer */
+    int noCols,                   /*!< Number of time slots  */
+    int lsb,                      /*!< Number of lower bands */
+    int usb,                      /*!< Number of upper bands */
+    int no_channels,              /*!< Number of critically sampled bands */
+    int flags);                   /*!< Flags */
+#endif
+
 void qmfAnalysisFiltering(
     HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank   */
     FIXP_DBL **qmfReal,            /*!< Pointer to real subband slots */
     FIXP_DBL **qmfImag,            /*!< Pointer to imag subband slots */
     QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data     */
-    const LONG *timeIn,            /*!< Time signal */
+    const INT_PCM *timeIn,         /*!< Time signal */
     const int timeIn_e,            /*!< Exponent of audio data        */
     const int stride,              /*!< Stride factor of audio data   */
-    FIXP_DBL *pWorkBuffer          /*!< pointer to temporary working buffer */
+    FIXP_DBL *pWorkBuffer          /*!< pointer to temporal working buffer */
 );
+#if SAMPLE_BITS == 16
 
 void qmfAnalysisFiltering(
     HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank   */
     FIXP_DBL **qmfReal,            /*!< Pointer to real subband slots */
     FIXP_DBL **qmfImag,            /*!< Pointer to imag subband slots */
     QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data     */
-    const INT_PCM *timeIn,         /*!< Time signal */
+    const LONG *timeIn,            /*!< Time signal */
     const int timeIn_e,            /*!< Exponent of audio data        */
     const int stride,              /*!< Stride factor of audio data   */
-    FIXP_DBL *pWorkBuffer          /*!< pointer to temporal working buffer */
-);
-
-void qmfSynthesisFiltering(
-    HANDLE_QMF_FILTER_BANK synQmf,       /*!< Handle of Qmf Synthesis Bank  */
-    FIXP_DBL **QmfBufferReal,            /*!< Pointer to real subband slots */
-    FIXP_DBL **QmfBufferImag,            /*!< Pointer to imag subband slots */
-    const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data     */
-    const int ov_len,                    /*!< Length of band overlap        */
-    INT_PCM *timeOut,                    /*!< Time signal */
-    const INT stride,                    /*!< Stride factor of audio data   */
-    FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be
-                             aligned */
+    FIXP_DBL *pWorkBuffer          /*!< pointer to temporary working buffer */
 );
-
-int qmfInitAnalysisFilterBank(
-    HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
-    FIXP_QAS *pFilterStates,      /*!< Pointer to filter state buffer */
-    int noCols,                   /*!< Number of time slots  */
-    int lsb,                      /*!< Number of lower bands */
-    int usb,                      /*!< Number of upper bands */
-    int no_channels,              /*!< Number of critically sampled bands */
-    int flags);                   /*!< Flags */
+#endif
 
 void qmfAnalysisFilteringSlot(
     HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank  */
     FIXP_DBL *qmfReal,             /*!< Low and High band, real */
     FIXP_DBL *qmfImag,             /*!< Low and High band, imag */
-    const LONG *timeIn,            /*!< Pointer to input */
+    const INT_PCM *timeIn,         /*!< Pointer to input */
     const int stride,              /*!< stride factor of input */
-    FIXP_DBL *pWorkBuffer          /*!< pointer to temporary working buffer */
+    FIXP_DBL *pWorkBuffer          /*!< pointer to temporal working buffer */
 );
+#if SAMPLE_BITS == 16
 
 void qmfAnalysisFilteringSlot(
     HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank  */
     FIXP_DBL *qmfReal,             /*!< Low and High band, real */
     FIXP_DBL *qmfImag,             /*!< Low and High band, imag */
-    const INT_PCM *timeIn,         /*!< Pointer to input */
+    const LONG *timeIn,            /*!< Pointer to input */
     const int stride,              /*!< stride factor of input */
-    FIXP_DBL *pWorkBuffer          /*!< pointer to temporal working buffer */
+    FIXP_DBL *pWorkBuffer          /*!< pointer to temporary working buffer */
 );
+#endif
+
 int qmfInitSynthesisFilterBank(
     HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
     FIXP_QSS *pFilterStates,      /*!< Pointer to filter state buffer */
@@ -270,12 +275,46 @@ int qmfInitSynthesisFilterBank(
     int no_channels,              /*!< Number of critically sampled bands */
     int flags);                   /*!< Flags */
 
+void qmfSynthesisFiltering(
+    HANDLE_QMF_FILTER_BANK synQmf,       /*!< Handle of Qmf Synthesis Bank  */
+    FIXP_DBL **QmfBufferReal,            /*!< Pointer to real subband slots */
+    FIXP_DBL **QmfBufferImag,            /*!< Pointer to imag subband slots */
+    const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data     */
+    const int ov_len,                    /*!< Length of band overlap        */
+    INT_PCM *timeOut,                    /*!< Time signal */
+    const INT stride,                    /*!< Stride factor of audio data   */
+    FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be
+                             aligned */
+);
+#if SAMPLE_BITS == 16
+
+void qmfSynthesisFiltering(
+    HANDLE_QMF_FILTER_BANK synQmf,       /*!< Handle of Qmf Synthesis Bank  */
+    FIXP_DBL **QmfBufferReal,            /*!< Pointer to real subband slots */
+    FIXP_DBL **QmfBufferImag,            /*!< Pointer to imag subband slots */
+    const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data     */
+    const int ov_len,                    /*!< Length of band overlap        */
+    LONG *timeOut,                       /*!< Time signal */
+    const int timeOut_e,                 /*!< Target exponent for timeOut  */
+    FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */
+);
+#endif
+
 void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf,
                                const FIXP_DBL *realSlot,
                                const FIXP_DBL *imagSlot,
                                const int scaleFactorLowBand,
                                const int scaleFactorHighBand, INT_PCM *timeOut,
                                const int timeOut_e, FIXP_DBL *pWorkBuffer);
+#if SAMPLE_BITS == 16
+
+void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf,
+                               const FIXP_DBL *realSlot,
+                               const FIXP_DBL *imagSlot,
+                               const int scaleFactorLowBand,
+                               const int scaleFactorHighBand, LONG *timeOut,
+                               const int timeOut_e, FIXP_DBL *pWorkBuffer);
+#endif
 
 void qmfChangeOutScalefactor(
     HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
@@ -291,11 +330,5 @@ void qmfChangeOutGain(
     FIXP_DBL outputGain,           /*!< New gain for output data (mantissa) */
     int outputGainScale            /*!< New gain for output data (exponent) */
 );
-void qmfSynPrototypeFirSlot(
-    HANDLE_QMF_FILTER_BANK qmf,
-    FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */
-    FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */
-    INT_PCM *RESTRICT timeOut,   /*!< Time domain data */
-    const int timeOut_e);
 
 #endif /*ifndef QMF_H       */
diff --git a/fdk-aac/libFDK/include/qmf_pcm.h b/fdk-aac/libFDK/include/qmf_pcm.h
index f24e0cd..5da53db 100644
--- a/fdk-aac/libFDK/include/qmf_pcm.h
+++ b/fdk-aac/libFDK/include/qmf_pcm.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -402,4 +402,220 @@ void qmfSynthesisFiltering(
                               timeOut + (i * L * stride), stride, pWorkBuffer);
   } /* no_col loop  i  */
 }
+
+/*!
+ *
+ * \brief Create QMF filter bank instance
+ *
+ *
+ * \return 0 if successful
+ *
+ */
+int qmfInitAnalysisFilterBank(
+    HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Returns handle */
+    FIXP_QAS *pFilterStates,      /*!< Handle to filter states */
+    int noCols,                   /*!< Number of timeslots per frame */
+    int lsb,                      /*!< lower end of QMF */
+    int usb,                      /*!< upper end of QMF */
+    int no_channels,              /*!< Number of channels (bands) */
+    int flags)                    /*!< Low Power flag */
+{
+  int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb,
+                              no_channels, flags, 0);
+  if (!(flags & QMF_FLAG_KEEP_STATES) && (h_Qmf->FilterStates != NULL)) {
+    FDKmemclear(h_Qmf->FilterStates,
+                (2 * QMF_NO_POLY - 1) * h_Qmf->no_channels * sizeof(FIXP_QAS));
+  }
+
+  FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->lsb);
+
+  return err;
+}
+
+#ifndef FUNCTION_qmfAnaPrototypeFirSlot
+/*!
+  \brief Perform Analysis Prototype Filtering on a single slot of input data.
+*/
+static void qmfAnaPrototypeFirSlot(
+    FIXP_DBL *analysisBuffer,
+    INT no_channels, /*!< Number channels of analysis filter */
+    const FIXP_PFT *p_filter, INT p_stride, /*!< Stride of analysis filter    */
+    FIXP_QAS *RESTRICT pFilterStates) {
+  INT k;
+
+  FIXP_DBL accu;
+  const FIXP_PFT *RESTRICT p_flt = p_filter;
+  FIXP_DBL *RESTRICT pData_0 = analysisBuffer + 2 * no_channels - 1;
+  FIXP_DBL *RESTRICT pData_1 = analysisBuffer;
+
+  FIXP_QAS *RESTRICT sta_0 = (FIXP_QAS *)pFilterStates;
+  FIXP_QAS *RESTRICT sta_1 =
+      (FIXP_QAS *)pFilterStates + (2 * QMF_NO_POLY * no_channels) - 1;
+  INT pfltStep = QMF_NO_POLY * (p_stride);
+  INT staStep1 = no_channels << 1;
+  INT staStep2 = (no_channels << 3) - 1; /* Rewind one less */
+
+  /* FIR filters 127..64 0..63 */
+  for (k = 0; k < no_channels; k++) {
+    accu = fMultDiv2(p_flt[0], *sta_1);
+    sta_1 -= staStep1;
+    accu += fMultDiv2(p_flt[1], *sta_1);
+    sta_1 -= staStep1;
+    accu += fMultDiv2(p_flt[2], *sta_1);
+    sta_1 -= staStep1;
+    accu += fMultDiv2(p_flt[3], *sta_1);
+    sta_1 -= staStep1;
+    accu += fMultDiv2(p_flt[4], *sta_1);
+    *pData_1++ = (accu << 1);
+    sta_1 += staStep2;
+
+    p_flt += pfltStep;
+    accu = fMultDiv2(p_flt[0], *sta_0);
+    sta_0 += staStep1;
+    accu += fMultDiv2(p_flt[1], *sta_0);
+    sta_0 += staStep1;
+    accu += fMultDiv2(p_flt[2], *sta_0);
+    sta_0 += staStep1;
+    accu += fMultDiv2(p_flt[3], *sta_0);
+    sta_0 += staStep1;
+    accu += fMultDiv2(p_flt[4], *sta_0);
+    *pData_0-- = (accu << 1);
+    sta_0 -= staStep2;
+  }
+}
+#endif /* !defined(FUNCTION_qmfAnaPrototypeFirSlot) */
+
+#ifndef FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric
+/*!
+  \brief Perform Analysis Prototype Filtering on a single slot of input data.
+*/
+static void qmfAnaPrototypeFirSlot_NonSymmetric(
+    FIXP_DBL *analysisBuffer,
+    int no_channels, /*!< Number channels of analysis filter */
+    const FIXP_PFT *p_filter, int p_stride, /*!< Stride of analysis filter    */
+    FIXP_QAS *RESTRICT pFilterStates) {
+  const FIXP_PFT *RESTRICT p_flt = p_filter;
+  int p, k;
+
+  for (k = 0; k < 2 * no_channels; k++) {
+    FIXP_DBL accu = (FIXP_DBL)0;
+
+    p_flt += QMF_NO_POLY * (p_stride - 1);
+
+    /*
+      Perform FIR-Filter
+    */
+    for (p = 0; p < QMF_NO_POLY; p++) {
+      accu += fMultDiv2(*p_flt++, pFilterStates[2 * no_channels * p]);
+    }
+    analysisBuffer[2 * no_channels - 1 - k] = (accu << 1);
+    pFilterStates++;
+  }
+}
+#endif /* FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric */
+
+/*
+ * \brief Perform one QMF slot analysis of the time domain data of timeIn
+ *        with specified stride and stores the real part of the subband
+ *        samples in rSubband, and the imaginary part in iSubband
+ *
+ *        Note: anaQmf->lsb can be greater than anaQmf->no_channels in case
+ *        of implicit resampling (USAC with reduced 3/4 core frame length).
+ */
+void qmfAnalysisFilteringSlot(
+    HANDLE_QMF_FILTER_BANK anaQmf,        /*!< Handle of Qmf Synthesis Bank  */
+    FIXP_DBL *qmfReal,                    /*!< Low and High band, real */
+    FIXP_DBL *qmfImag,                    /*!< Low and High band, imag */
+    const INT_PCM_QMFIN *RESTRICT timeIn, /*!< Pointer to input */
+    const int stride,                     /*!< stride factor of input */
+    FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */
+) {
+  int offset = anaQmf->no_channels * (QMF_NO_POLY * 2 - 1);
+  /*
+    Feed time signal into oldest anaQmf->no_channels states
+  */
+  {
+    FIXP_QAS *FilterStatesAnaTmp = ((FIXP_QAS *)anaQmf->FilterStates) + offset;
+
+    /* Feed and scale actual time in slot */
+    for (int i = anaQmf->no_channels >> 1; i != 0; i--) {
+      /* Place INT_PCM value left aligned in scaledTimeIn */
+      *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;
+      timeIn += stride;
+      *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;
+      timeIn += stride;
+    }
+  }
+
+  if (anaQmf->flags & QMF_FLAG_NONSYMMETRIC) {
+    qmfAnaPrototypeFirSlot_NonSymmetric(pWorkBuffer, anaQmf->no_channels,
+                                        anaQmf->p_filter, anaQmf->p_stride,
+                                        (FIXP_QAS *)anaQmf->FilterStates);
+  } else {
+    qmfAnaPrototypeFirSlot(pWorkBuffer, anaQmf->no_channels, anaQmf->p_filter,
+                           anaQmf->p_stride, (FIXP_QAS *)anaQmf->FilterStates);
+  }
+
+  if (anaQmf->flags & QMF_FLAG_LP) {
+    if (anaQmf->flags & QMF_FLAG_CLDFB)
+      qmfForwardModulationLP_odd(anaQmf, pWorkBuffer, qmfReal);
+    else
+      qmfForwardModulationLP_even(anaQmf, pWorkBuffer, qmfReal);
+
+  } else {
+    qmfForwardModulationHQ(anaQmf, pWorkBuffer, qmfReal, qmfImag);
+  }
+  /*
+    Shift filter states
+
+    Should be realized with modulo addressing on a DSP instead of a true buffer
+    shift
+  */
+  FDKmemmove(anaQmf->FilterStates,
+             (FIXP_QAS *)anaQmf->FilterStates + anaQmf->no_channels,
+             offset * sizeof(FIXP_QAS));
+}
+
+/*!
+ *
+ * \brief Perform complex-valued subband filtering of the time domain
+ *        data of timeIn and stores the real part of the subband
+ *        samples in rAnalysis, and the imaginary part in iAnalysis
+ * The qmf coefficient table is symmetric. The symmetry is expoited by
+ * shrinking the coefficient table to half the size. The addressing mode
+ * takes care of the symmetries.
+ *
+ *
+ * \sa PolyphaseFiltering
+ */
+void qmfAnalysisFiltering(
+    HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */
+    FIXP_DBL **qmfReal,            /*!< Pointer to real subband slots */
+    FIXP_DBL **qmfImag,            /*!< Pointer to imag subband slots */
+    QMF_SCALE_FACTOR *scaleFactor,
+    const INT_PCM_QMFIN *timeIn, /*!< Time signal */
+    const int timeIn_e, const int stride,
+    FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */
+) {
+  int i;
+  int no_channels = anaQmf->no_channels;
+
+  scaleFactor->lb_scale =
+      -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - timeIn_e;
+  scaleFactor->lb_scale -= anaQmf->filterScale;
+
+  for (i = 0; i < anaQmf->no_col; i++) {
+    FIXP_DBL *qmfImagSlot = NULL;
+
+    if (!(anaQmf->flags & QMF_FLAG_LP)) {
+      qmfImagSlot = qmfImag[i];
+    }
+
+    qmfAnalysisFilteringSlot(anaQmf, qmfReal[i], qmfImagSlot, timeIn, stride,
+                             pWorkBuffer);
+
+    timeIn += no_channels * stride;
+
+  } /* no_col loop  i  */
+}
 #endif /* QMF_PCM_H */
diff --git a/fdk-aac/libFDK/include/scale.h b/fdk-aac/libFDK/include/scale.h
index fcfc116..a58614e 100644
--- a/fdk-aac/libFDK/include/scale.h
+++ b/fdk-aac/libFDK/include/scale.h
@@ -129,15 +129,13 @@ void scaleCplxValues(FIXP_DBL *r_dst, FIXP_DBL *i_dst, const FIXP_DBL *r_src,
 void scaleValuesWithFactor(FIXP_DBL *vector, FIXP_DBL factor, INT len,
                            INT scalefactor);
 void scaleValuesSaturate(FIXP_DBL *vector, INT len, INT scalefactor);
-void scaleValuesSaturate(FIXP_DBL *dst, FIXP_DBL *src, INT len,
+void scaleValuesSaturate(FIXP_DBL *dst, const FIXP_DBL *src, INT len,
                          INT scalefactor);
-void scaleValuesSaturate(FIXP_SGL *dst, FIXP_DBL *src, INT len,
+void scaleValuesSaturate(FIXP_SGL *dst, const FIXP_DBL *src, INT len,
                          INT scalefactor);
-void scaleValuesSaturate(INT_PCM *dst, FIXP_DBL *src, INT len, INT scalefactor);
 void scaleValuesSaturate(FIXP_SGL *vector, INT len, INT scalefactor);
-void scaleValuesSaturate(FIXP_SGL *dst, FIXP_SGL *src, INT len,
+void scaleValuesSaturate(FIXP_SGL *dst, const FIXP_SGL *src, INT len,
                          INT scalefactor);
-void scaleValuesSaturate(INT_PCM *dst, INT_PCM *src, INT len, INT scalefactor);
 INT getScalefactorShort(const SHORT *vector, INT len);
 INT getScalefactorPCM(const INT_PCM *vector, INT len, INT stride);
 INT getScalefactor(const FIXP_DBL *vector, INT len);
@@ -241,31 +239,20 @@ inline void scaleValueInPlace(FIXP_DBL *value, /*!< Value */
 
 #ifndef SATURATE_RIGHT_SHIFT
 #define SATURATE_RIGHT_SHIFT(src, scale, dBits)                            \
-  (((scale) >= 8*sizeof(LONG)) ? (LONG)0 :                                 \
   ((((LONG)(src) >> (scale)) > (LONG)(((1U) << ((dBits)-1)) - 1))          \
        ? (LONG)(((1U) << ((dBits)-1)) - 1)                                 \
        : (((LONG)(src) >> (scale)) < ~((LONG)(((1U) << ((dBits)-1)) - 1))) \
              ? ~((LONG)(((1U) << ((dBits)-1)) - 1))                        \
-             : ((LONG)(src) >> (scale))))
-#endif
-
-#ifndef SATURATE_LEFT_MAX
-#define SATURATE_LEFT_MAX(src, dBits)                 \
-  (((LONG)(src) > 0)                                  \
-       ? (LONG)(((1U) << ((dBits)-1)) - 1)            \
-       : ((LONG)(src) < 0)                            \
-             ? ~((LONG)(((1U) << ((dBits)-1)) - 1))   \
-             : (LONG)0)
+             : ((LONG)(src) >> (scale)))
 #endif
 
 #ifndef SATURATE_LEFT_SHIFT
 #define SATURATE_LEFT_SHIFT(src, scale, dBits)                           \
-  (((scale) >= 8*sizeof(LONG)) ? SATURATE_LEFT_MAX(src, dBits) :         \
   (((LONG)(src) > ((LONG)(((1U) << ((dBits)-1)) - 1) >> (scale)))        \
        ? (LONG)(((1U) << ((dBits)-1)) - 1)                               \
        : ((LONG)(src) < ~((LONG)(((1U) << ((dBits)-1)) - 1) >> (scale))) \
              ? ~((LONG)(((1U) << ((dBits)-1)) - 1))                      \
-             : ((LONG)(src) << (scale))))
+             : ((LONG)(src) << (scale)))
 #endif
 
 #ifndef SATURATE_SHIFT
diff --git a/fdk-aac/libFDK/include/scramble.h b/fdk-aac/libFDK/include/scramble.h
index f07ebed..e7cead9 100644
--- a/fdk-aac/libFDK/include/scramble.h
+++ b/fdk-aac/libFDK/include/scramble.h
@@ -108,7 +108,7 @@ amm-info@iis.fraunhofer.de
 #if defined(__arm__)
 #include "arm/scramble_arm.h"
 
-#elif defined(__mips__) && defined(__mips_dsp)
+#elif defined(__mips__) && defined(__mips_dsp) && !defined(__mips16)
 #include "mips/scramble_mips.h"
 
 #endif