Snap for 7474514 from 26eed59258d1a172c7bcfcdf9c562c7d50b5d726 to mainline-media-swcodec-releaseandroid-mainline-12.0.0_r91android-mainline-12.0.0_r75android-mainline-12.0.0_r47android-mainline-12.0.0_r30android-mainline-12.0.0_r13android-mainline-12.0.0_r105android12-mainline-media-swcodec-release

Change-Id: I1606212aae9d967e2a10b82a1e88900255d9a017

34 files changed, 1341 insertions, 480 deletions

diff --git a/Android.bp b/Android.bp
index 3f42c19..0c67186 100644
--- a/Android.bp
+++ b/Android.bp
@@ -1,6 +1,40 @@
+// *** THIS PACKAGE HAS SPECIAL LICENSING CONDITIONS.  PLEASE
+//     CONSULT THE OWNERS AND opensource-licensing@google.com BEFORE
+//     DEPENDING ON IT IN YOUR PROJECT. ***
+package {
+    default_applicable_licenses: ["external_aac_license"],
+}
+
+// Added automatically by a large-scale-change that took the approach of
+// 'apply every license found to every target'. While this makes sure we respect
+// every license restriction, it may not be entirely correct.
+//
+// e.g. GPL in an MIT project might only apply to the contrib/ directory.
+//
+// Please consider splitting the single license below into multiple licenses,
+// taking care not to lose any license_kind information, and overriding the
+// default license using the 'licenses: [...]' property on targets as needed.
+//
+// For unused files, consider creating a 'fileGroup' with "//visibility:private"
+// to attach the license to, and including a comment whether the files may be
+// used in the current project.
+// See: http://go/android-license-faq
+license {
+    name: "external_aac_license",
+    visibility: [":__subpackages__"],
+    license_kinds: [
+        "SPDX-license-identifier-Apache-2.0",
+        "legacy_by_exception_only", // by exception only
+    ],
+    license_text: [
+        "NOTICE",
+    ],
+}
+
 cc_library_static {
     name: "libFraunhoferAAC",
     vendor_available: true,
+    host_supported: true,
     srcs: [
         "libAACdec/src/*.cpp",
         "libAACenc/src/*.cpp",
@@ -26,13 +60,12 @@ cc_library_static {
         "-DSUPPRESS_BUILD_DATE_INFO",
     ],
     sanitize: {
-        misc_undefined:[
-           "unsigned-integer-overflow",
-           "signed-integer-overflow",
-           "bounds",
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+            "bounds",
         ],
-        // Enable CFI if this becomes a shared library.
-        // cfi: true,
+        cfi: true,
     },
     shared_libs: [
         "liblog",
@@ -53,6 +86,12 @@ cc_library_static {
         "libSACenc/include",
     ],
 
+    target: {
+        darwin: {
+            enabled: false,
+        },
+    },
+
     apex_available: [
         "//apex_available:platform",
         "com.android.bluetooth.updatable",
diff --git a/METADATA b/METADATA
new file mode 100644
index 0000000..5c12860
--- /dev/null
+++ b/METADATA
@@ -0,0 +1,3 @@
+third_party {
+  license_type: BY_EXCEPTION_ONLY
+}
diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf
index cc7cf41..3d4699e 100644
--- a/documentation/aacDecoder.pdf
+++ b/documentation/aacDecoder.pdf
diff --git a/documentation/aacEncoder.pdf b/documentation/aacEncoder.pdf
index 77b8f4c..a47708a 100644
--- a/documentation/aacEncoder.pdf
+++ b/documentation/aacEncoder.pdf
diff --git a/fuzzer/Android.bp b/fuzzer/Android.bp
new file mode 100644
index 0000000..6739798
--- /dev/null
+++ b/fuzzer/Android.bp
@@ -0,0 +1,82 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *****************************************************************************
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore
+ */
+
+package {
+    // See: http://go/android-license-faq
+    // A large-scale-change added 'default_applicable_licenses' to import
+    // all of the 'license_kinds' from "external_aac_license"
+    // to get the below license kinds:
+    //   SPDX-license-identifier-Apache-2.0
+    default_applicable_licenses: ["external_aac_license"],
+}
+
+cc_defaults {
+    name: "aac_fuzz_defaults",
+    host_supported: true,
+
+    static_libs: [
+        "libFraunhoferAAC",
+    ],
+
+    target: {
+        darwin: {
+            enabled: false,
+        },
+    },
+
+    fuzz_config: {
+        cc: [
+            "android-media-fuzzing-reports@google.com",
+        ],
+        componentid: 155276,
+    },
+}
+
+cc_fuzz {
+    name: "aac_dec_fuzzer",
+
+    srcs: [
+        "aac_dec_fuzzer.cpp",
+    ],
+
+    static_libs: [
+        "liblog",
+    ],
+
+    defaults: [
+        "aac_fuzz_defaults"
+    ],
+}
+
+cc_fuzz {
+    name: "aac_enc_fuzzer",
+
+    srcs: [
+        "aac_enc_fuzzer.cpp",
+    ],
+
+    defaults: [
+        "aac_fuzz_defaults"
+    ],
+
+    include_dirs: [
+        "external/aac/libAACenc/"
+    ],
+}
diff --git a/fuzzer/README.md b/fuzzer/README.md
new file mode 100644
index 0000000..b8cc260
--- /dev/null
+++ b/fuzzer/README.md
@@ -0,0 +1,150 @@
+# Fuzzer for libFraunhoferAAC decoder
+
+## Plugin Design Considerations
+The fuzzer plugin for aac decoder is designed based on the understanding of the
+codec and tries to achieve the following:
+
+##### Maximize code coverage
+
+This fuzzer makes use of the following config parameters:
+1. Transport type (parameter name: `TRANSPORT_TYPE`)
+
+| Parameter| Valid Values| Configured Value|
+|------------- |-------------| ----- |
+| `TRANSPORT_TYPE` | 0.`TT_UNKNOWN  ` 1.`TT_MP4_RAW ` 2.`TT_MP4_ADIF ` 3.`TT_MP4_ADTS ` 4.`TT_MP4_LATM_MCP1 ` 5.`TT_MP4_LATM_MCP0  ` 6.`TT_MP4_LOAS ` 7.`TT_DRM ` | `TT_MP4_ADIF ` |
+
+Note: Value of `TRANSPORT_TYPE` could be set to any of these values.
+It is set to `TT_MP4_ADIF` in the fuzzer plugin.
+
+##### Maximize utilization of input data
+The plugin feeds the entire input data to the codec using a loop.
+ * If the decode operation was successful, the input is advanced by an
+   offset calculated using valid bytes.
+ * If the decode operation was un-successful, the input is advanced by 1 byte
+   till it reaches a valid frame or end of stream.
+
+This ensures that the plugin tolerates any kind of input (empty, huge,
+malformed, etc) and doesnt `exit()` on any input and thereby increasing the
+chance of identifying vulnerabilities.
+
+## Build
+
+This describes steps to build aac_dec_fuzzer binary.
+
+## Android
+
+### Steps to build
+Build the fuzzer
+```
+  $ mm -j$(nproc) aac_dec_fuzzer
+```
+
+### Steps to run
+Create a directory CORPUS_DIR and copy some aac files to that folder.
+Push this directory to device.
+
+To run on device
+```
+  $ adb sync data
+  $ adb shell /data/fuzz/arm64/aac_dec_fuzzer/aac_dec_fuzzer CORPUS_DIR
+```
+To run on host
+```
+  $ $ANDROID_HOST_OUT/fuzz/x86_64/aac_dec_fuzzer/aac_dec_fuzzer CORPUS_DIR
+```
+
+# Fuzzer for libFraunhoferAAC encoder
+
+## Plugin Design Considerations
+The fuzzer plugin for aac encoder is designed based on the understanding of the
+codec and tries to achieve the following:
+
+##### Maximize code coverage
+
+The configuration parameters are not hardcoded, but instead selected based on
+incoming data. This ensures more code paths are reached by the fuzzer.
+
+Following arguments are passed to aacEncoder_SetParam to set the respective AACENC_PARAM parameter:
+
+| AACENC_PARAM param| Valid Values| Configured Value|
+|-------------| ----- |----- |
+|`AACENC_SBR_MODE` | `-1 ` `0 ` `1 ` `2 ` | Calculated using first byte of data |
+|`AACENC_AOT` |`AOT_NONE ` `AOT_NULL_OBJECT ` `AOT_AAC_MAIN ` `AOT_AAC_LC ` `AOT_AAC_SSR ` `AOT_AAC_LTP ` `AOT_SBR ` `AOT_AAC_SCAL ` `AOT_TWIN_VQ ` `AOT_CELP ` `AOT_HVXC ` `AOT_RSVD_10 ` `AOT_RSVD_11 ` `AOT_TTSI ` `AOT_MAIN_SYNTH ` `AOT_WAV_TAB_SYNTH ` `AOT_GEN_MIDI ` `AOT_ALG_SYNTH_AUD_FX ` `AOT_ER_AAC_LC ` `AOT_RSVD_18 ` `AOT_ER_AAC_LTP ` `AOT_ER_AAC_SCAL ` `AOT_ER_TWIN_VQ ` `AOT_ER_BSAC ` `AOT_ER_AAC_LD ` `AOT_ER_CELP ` `AOT_ER_HVXC ` `AOT_ER_HILN ` `AOT_ER_PARA ` `AOT_RSVD_28 ` `AOT_PS ` `AOT_MPEGS ` `AOT_ESCAPE ` `AOT_MP3ONMP4_L1 ` `AOT_MP3ONMP4_L2 ` `AOT_MP3ONMP4_L3 ` `AOT_RSVD_35 ` `AOT_RSVD_36 ` `AOT_AAC_SLS ` `AOT_SLS ` `AOT_ER_AAC_ELD ` `AOT_USAC ` `AOT_SAOC ` `AOT_LD_MPEGS ` `AOT_MP2_AAC_LC ` `AOT_MP2_SBR ` `AOT_DRM_AAC ` `AOT_DRM_SBR ` `AOT_DRM_MPEG_PS ` `AOT_DRM_SURROUND ` `AOT_DRM_USAC `  | Calculated using second byte of data  |
+|`AACENC_SAMPLERATE` |  `8000 ` `11025 ` `12000 ` `16000 ` `22050 ` `24000 ` `32000 ` `44100 ` `48000 ` `64000 ` `88200 ` `96000 `| Calculated using third byte of data  |
+|`AACENC_BITRATE` | In range `8000 ` to `960000 ` | Calculated using fourth, fifth and sixth byte of data  |
+|`AACENC_CHANNELMODE` | `MODE_1 ` `MODE_2 ` `MODE_1_2 ` `MODE_1_2_1 ` `MODE_1_2_2 ` `MODE_1_2_2_1 ` `MODE_1_2_2_2_1 ` `MODE_6_1 `  `MODE_7_1_BACK ` `MODE_7_1_TOP_FRONT ` `MODE_7_1_REAR_SURROUND ` `MODE_7_1_FRONT_CENTER ` `MODE_212 ` | Calculated using seventh byte of data |
+|`AACENC_TRANSMUX` | `TT_MP4_RAW ` `TT_MP4_ADIF ` `TT_MP4_ADTS ` `TT_MP4_LATM_MCP1 ` `TT_MP4_LATM_MCP0 ` `TT_MP4_LOAS ` `TT_DRM `  | Calculated using eight byte of data  |`AACENC_SBR_RATIO` |`-1 ` `0 ` `1 ` `2 ` | Calculated using ninth byte of data |
+|`AACENC_BITRATEMODE` |`AACENC_BR_MODE_INVALID ` `AACENC_BR_MODE_CBR ` `AACENC_BR_MODE_VBR_1 ` `AACENC_BR_MODE_VBR_2 ` `AACENC_BR_MODE_VBR_3 ` `AACENC_BR_MODE_VBR_4 ` `AACENC_BR_MODE_VBR_5 ` `AACENC_BR_MODE_FF ` `AACENC_BR_MODE_SFR `  | Calculated using thirty-fourth byte of data |
+|`AACENC_GRANULE_LENGTH` |`120 ` `128 ` `240 ` `256 ` `480 ` `512 ` `1024 ` | Calculated using thirty-fifth byte of data |
+|`AACENC_CHANNELORDER` |`CH_ORDER_MPEG ` `CH_ORDER_WAV ` | Calculated using thirty-sixth byte of data |
+|`AACENC_AFTERBURNER` |`0 ` `1 ` | Calculated using thirty-seventh byte of data |
+|`AACENC_BANDWIDTH` |`0 ` `1`  | Calculated using thirty-eigth byte of data |
+|` AACENC_IDX_PEAK_BITRATE` | In range `8000 ` to `960000 ` | Calculated using thirty-ninth byte of data |
+|` AACENC_HEADER_PERIOD` |In range `0 ` to `255 ` | Calculated using fortieth byte of data |
+|` AACENC_SIGNALING_MODE` |`-1 ` `0 ` `1 ` `2 ` `3 `  | Calculated using forty-first byte of data |
+|` AACENC_TPSUBFRAMES` |In range `0 ` to `255 ` | Calculated using forty-second byte of data |
+|` AACENC_AUDIOMUXVER` |`-1 ` `0 ` `1 ` `2 ` | Calculated using forty-third byte of data |
+|` AACENC_PROTECTION` |`0 ` `1 ` | Calculated using forty-fourth of data |
+|`AACENC_ANCILLARY_BITRATE` |In range `0 ` to `960000 `| Calculated using forty-fifth byte of data |
+|`AACENC_METADATA_MODE ` |`0 ` `1 ` `2 ` `3 ` | Calculated using forty-sixth byte of data |
+
+Following values are configured to set up the meta data represented by the class variable `mMetaData ` :
+
+| Variable name| Possible Values| Configured Value|
+|------------- | ----- |----- |
+| `drc_profile`   | `AACENC_METADATA_DRC_NONE ` `AACENC_METADATA_DRC_FILMSTANDARD ` `AACENC_METADATA_DRC_FILMLIGHT ` `AACENC_METADATA_DRC_MUSICSTANDARD ` `AACENC_METADATA_DRC_MUSICLIGHT ` `AACENC_METADATA_DRC_SPEECH ` `AACENC_METADATA_DRC_NOT_PRESENT `  | Calculated using tenth byte of data |
+| `comp_profile`   | `AACENC_METADATA_DRC_NONE ` `AACENC_METADATA_DRC_FILMSTANDARD ` `AACENC_METADATA_DRC_FILMLIGHT ` `AACENC_METADATA_DRC_MUSICSTANDARD ` `AACENC_METADATA_DRC_MUSICLIGHT ` `AACENC_METADATA_DRC_SPEECH ` `AACENC_METADATA_DRC_NOT_PRESENT `  | Calculated using eleventh byte of data |
+| `drc_TargetRefLevel`   | In range `0 ` to `255 `  | Calculated using twelfth byte of data |
+| `comp_TargetRefLevel`   | In range `0 ` to `255 `  | Calculated using thirteenth byte of data |
+| `prog_ref_level_present`   | `0 ` `1 `  | Calculated using fourteenth byte of data |
+| `prog_ref_level`   | In range `0 ` to `255 `  | Calculated using fifteenth byte of data |
+| `PCE_mixdown_idx_present`   | `0 ` `1 `   | Calculated using sixteenth byte of data |
+| `ETSI_DmxLvl_present`   | `0 ` `1 `   | Calculated using seventeenth byte of data |
+| `centerMixLevel`   | In range `0 ` to `7 `  | Calculated using eighteenth byte of data |
+| `surroundMixLevel`   | In range `0 ` to `7 `  | Calculated using nineteenth byte of data |
+| `dolbySurroundMode`   | In range `0 ` to `2 `   | Calculated using twentieth byte of data |
+| `drcPresentationMode`   | In range `0 ` to `2 `   | Calculated using twenty-first byte of data |
+| `extAncDataEnable`   | `0 ` `1 `  | Calculated using twenty-second byte of data |
+| `extDownmixLevelEnable`   | `0 ` `1 `  | Calculated using twenty-third byte of data |
+| `extDownmixLevel_A`   | In range `0 ` to `7 `  | Calculated using twenty-fourth byte of data |
+| `extDownmixLevel_B`   | In range `0 ` to `7 `  | Calculated using twenty-fifth byte of data |
+| `dmxGainEnable`   |  `0 ` `1 `   | Calculated using twenty-sixth byte of data |
+| `dmxGain5`   | In range `0 ` to `255 `  | Calculated using twenty-seventh byte of data |
+| `dmxGain2`   | In range `0 ` to `255 `  | Calculated using twenty-eighth byte of data |
+| `lfeDmxEnable`   | `0 ` `1 `  | Calculated using twenty-ninth byte of data |
+| `lfeDmxLevel`   | In range `0 ` to `15 `  | Calculated using thirtieth byte of data |
+
+Indexes `mInBufferIdx_1`, `mInBufferIdx_2`  and `mInBufferIdx_3`(in range `0 ` to `2`) are calculated using the thirty-first, thirty-second and thirty-third byte respectively.
+
+##### Maximize utilization of input data
+The plugin feeds the entire input data to the codec and continues with the encoding even on a failure. This ensures that the plugin tolerates any kind of input (empty, huge, malformed, etc) and doesnt `exit()` on any input and thereby increasing the chance of identifying vulnerabilities.
+
+## Build
+
+This describes steps to build aac_enc_fuzzer binary.
+
+## Android
+
+### Steps to build
+Build the fuzzer
+```
+  $ mm -j$(nproc) aac_enc_fuzzer
+```
+
+### Steps to run
+Create a directory CORPUS_DIR and copy some raw files to that folder.
+Push this directory to device.
+
+To run on device
+```
+  $ adb sync data
+  $ adb shell /data/fuzz/arm64/aac_enc_fuzzer/aac_enc_fuzzer CORPUS_DIR
+```
+To run on host
+```
+  $ $ANDROID_HOST_OUT/fuzz/x86_64/aac_enc_fuzzer/aac_enc_fuzzer CORPUS_DIR
+```
+
+## References:
+ * http://llvm.org/docs/LibFuzzer.html
+ * https://github.com/google/oss-fuzz
diff --git a/fuzzer/aac_dec_fuzzer.cpp b/fuzzer/aac_dec_fuzzer.cpp
new file mode 100644
index 0000000..c970197
--- /dev/null
+++ b/fuzzer/aac_dec_fuzzer.cpp
@@ -0,0 +1,141 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *****************************************************************************
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore
+ */
+
+#include <stdint.h>
+#include <string.h>
+#include <algorithm>
+#include "aacdecoder_lib.h"
+
+constexpr uint8_t kNumberOfLayers = 1;
+constexpr uint8_t kMaxChannelCount = 8;
+constexpr uint32_t kMaxConfigurationSize = 1024;
+constexpr uint32_t kMaxOutBufferSize = 2048 * kMaxChannelCount;
+
+// Value indicating the start of AAC Header Segment
+constexpr const char *kAacSegStartSeq = "AAC_STRT";
+constexpr uint8_t kAacSegStartSeqLen = sizeof(kAacSegStartSeq);
+// Value indicating the end of AAC Header Segment
+constexpr const char *kAacSegEndSeq = "AAC_ENDS";
+constexpr uint8_t kAacSegEndSeqLen = sizeof(kAacSegEndSeq);
+
+// Number of bytes used to signal the length of the header
+constexpr uint8_t kHeaderLengthBytes = 2;
+// Minimum size of an AAC header is 2
+// Minimum data required is
+// strlen(AAC_STRT) + strlen(AAC_ENDS) + kHeaderLengthBytes + 2;
+constexpr UINT kMinDataSize = kAacSegStartSeqLen + kAacSegEndSeqLen + kHeaderLengthBytes + 2;
+
+UINT getHeaderSize(UCHAR *data, UINT size) {
+  if (size < kMinDataSize) {
+    return 0;
+  }
+
+  int32_t result = memcmp(data, kAacSegStartSeq, kAacSegStartSeqLen);
+  if (result) {
+    return 0;
+  }
+  data += kAacSegStartSeqLen;
+  size -= kAacSegStartSeqLen;
+
+  uint32_t headerLengthInBytes = (data[0] << 8 | data[1]) & 0xFFFF;
+  data += kHeaderLengthBytes;
+  size -= kHeaderLengthBytes;
+
+  if (headerLengthInBytes + kAacSegEndSeqLen > size) {
+    return 0;
+  }
+
+  data += headerLengthInBytes;
+  size -= headerLengthInBytes;
+  result = memcmp(data, kAacSegEndSeq, kAacSegEndSeqLen);
+  if (result) {
+    return 0;
+  }
+
+  return std::min(headerLengthInBytes, kMaxConfigurationSize);
+}
+
+class Codec {
+ public:
+  Codec() = default;
+  ~Codec() { deInitDecoder(); }
+  bool initDecoder();
+  void decodeFrames(UCHAR *data, UINT size);
+  void deInitDecoder();
+
+ private:
+  HANDLE_AACDECODER mAacDecoderHandle = nullptr;
+  AAC_DECODER_ERROR mErrorCode = AAC_DEC_OK;
+};
+
+bool Codec::initDecoder() {
+  mAacDecoderHandle = aacDecoder_Open(TT_MP4_ADIF, kNumberOfLayers);
+  if (!mAacDecoderHandle) {
+    return false;
+  }
+  return true;
+}
+
+void Codec::deInitDecoder() {
+  aacDecoder_Close(mAacDecoderHandle);
+  mAacDecoderHandle = nullptr;
+}
+
+void Codec::decodeFrames(UCHAR *data, UINT size) {
+  UINT headerSize = getHeaderSize(data, size);
+  if (headerSize != 0) {
+    data += kAacSegStartSeqLen + kHeaderLengthBytes;
+    size -= kAacSegStartSeqLen + kHeaderLengthBytes;
+    aacDecoder_ConfigRaw(mAacDecoderHandle, &data, &headerSize);
+    data += headerSize + kAacSegEndSeqLen;
+    size -= headerSize + kAacSegEndSeqLen;
+  }
+  while (size > 0) {
+    UINT inputSize = size;
+    UINT valid = size;
+    mErrorCode = aacDecoder_Fill(mAacDecoderHandle, &data, &inputSize, &valid);
+    if (mErrorCode != AAC_DEC_OK) {
+      ++data;
+      --size;
+    } else {
+      INT_PCM outputBuf[kMaxOutBufferSize];
+      do {
+        mErrorCode =
+            aacDecoder_DecodeFrame(mAacDecoderHandle, outputBuf,
+                    kMaxOutBufferSize /*size in number of INT_PCM, not bytes*/, 0);
+      } while (mErrorCode == AAC_DEC_OK);
+      UINT offset = inputSize - valid;
+      data += offset;
+      size = valid;
+    }
+  }
+}
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
+  Codec *codec = new Codec();
+  if (!codec) {
+    return 0;
+  }
+  if (codec->initDecoder()) {
+    codec->decodeFrames((UCHAR *)(data), static_cast<UINT>(size));
+  }
+  delete codec;
+  return 0;
+}
diff --git a/fuzzer/aac_enc_fuzzer.cpp b/fuzzer/aac_enc_fuzzer.cpp
new file mode 100644
index 0000000..5a35d70
--- /dev/null
+++ b/fuzzer/aac_enc_fuzzer.cpp
@@ -0,0 +1,479 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *****************************************************************************
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore
+ */
+
+#include <string>
+#include "aacenc_lib.h"
+#include "src/aacenc.h"
+
+using namespace std;
+
+// IN_AUDIO_DATA, IN_ANCILLRY_DATA and IN_METADATA_SETUP
+constexpr size_t kMaxBuffers = 3;
+
+constexpr size_t kMaxOutputBufferSize = 8192;
+
+constexpr uint32_t kMinBitRate = 8000;
+constexpr uint32_t kMaxBitRate = 960000;
+
+constexpr int32_t kSampleRates[] = {8000,  11025, 12000, 16000, 22050, 24000,
+                                    32000, 44100, 48000, 64000, 88200, 96000};
+constexpr size_t kSampleRatesSize = size(kSampleRates);
+
+constexpr CHANNEL_MODE kChannelModes[] = {MODE_1,
+                                          MODE_2,
+                                          MODE_1_2,
+                                          MODE_1_2_1,
+                                          MODE_1_2_2,
+                                          MODE_1_2_2_1,
+                                          MODE_1_2_2_2_1,
+                                          MODE_6_1,
+                                          MODE_7_1_BACK,
+                                          MODE_7_1_TOP_FRONT,
+                                          MODE_7_1_REAR_SURROUND,
+                                          MODE_7_1_FRONT_CENTER,
+                                          MODE_212};
+constexpr size_t kChannelModesSize = size(kChannelModes);
+
+constexpr TRANSPORT_TYPE kIdentifiers[] = {
+    TT_MP4_RAW, TT_MP4_ADIF, TT_MP4_ADTS, TT_MP4_LATM_MCP1, TT_MP4_LATM_MCP0, TT_MP4_LOAS, TT_DRM};
+constexpr size_t kIdentifiersSize = size(kIdentifiers);
+
+constexpr AUDIO_OBJECT_TYPE kAudioObjectTypes[] = {AOT_NONE,        AOT_NULL_OBJECT,
+                                                   AOT_AAC_MAIN,    AOT_AAC_LC,
+                                                   AOT_AAC_SSR,     AOT_AAC_LTP,
+                                                   AOT_SBR,         AOT_AAC_SCAL,
+                                                   AOT_TWIN_VQ,     AOT_CELP,
+                                                   AOT_HVXC,        AOT_RSVD_10,
+                                                   AOT_RSVD_11,     AOT_TTSI,
+                                                   AOT_MAIN_SYNTH,  AOT_WAV_TAB_SYNTH,
+                                                   AOT_GEN_MIDI,    AOT_ALG_SYNTH_AUD_FX,
+                                                   AOT_ER_AAC_LC,   AOT_RSVD_18,
+                                                   AOT_ER_AAC_LTP,  AOT_ER_AAC_SCAL,
+                                                   AOT_ER_TWIN_VQ,  AOT_ER_BSAC,
+                                                   AOT_ER_AAC_LD,   AOT_ER_CELP,
+                                                   AOT_ER_HVXC,     AOT_ER_HILN,
+                                                   AOT_ER_PARA,     AOT_RSVD_28,
+                                                   AOT_PS,          AOT_MPEGS,
+                                                   AOT_ESCAPE,      AOT_MP3ONMP4_L1,
+                                                   AOT_MP3ONMP4_L2, AOT_MP3ONMP4_L3,
+                                                   AOT_RSVD_35,     AOT_RSVD_36,
+                                                   AOT_AAC_SLS,     AOT_SLS,
+                                                   AOT_ER_AAC_ELD,  AOT_USAC,
+                                                   AOT_SAOC,        AOT_LD_MPEGS,
+                                                   AOT_MP2_AAC_LC,  AOT_MP2_SBR,
+                                                   AOT_DRM_AAC,     AOT_DRM_SBR,
+                                                   AOT_DRM_MPEG_PS, AOT_DRM_SURROUND,
+                                                   AOT_DRM_USAC};
+
+constexpr size_t kAudioObjectTypesSize = size(kAudioObjectTypes);
+
+constexpr int32_t kSbrRatios[] = {-1, 0, 1, 2};
+constexpr size_t kSbrRatiosSize = size(kSbrRatios);
+
+constexpr int32_t kBitRateModes[] = {
+    AACENC_BR_MODE_INVALID, AACENC_BR_MODE_CBR,   AACENC_BR_MODE_VBR_1,
+    AACENC_BR_MODE_VBR_2,   AACENC_BR_MODE_VBR_3, AACENC_BR_MODE_VBR_4,
+    AACENC_BR_MODE_VBR_5,   AACENC_BR_MODE_FF,    AACENC_BR_MODE_SFR};
+constexpr size_t kBitRateModesSize = size(kBitRateModes);
+
+constexpr int32_t kGranuleLengths[] = {120, 128, 240, 256, 480, 512, 1024};
+constexpr size_t kGranuleLengthsSize = size(kGranuleLengths);
+
+constexpr int32_t kChannelOrder[] = {CH_ORDER_MPEG, CH_ORDER_WAV};
+constexpr size_t kChannelOrderSize = size(kChannelOrder);
+
+constexpr int32_t kSignalingModes[] = {-1, 0, 1, 2, 3};
+constexpr size_t kSignalingModesSize = size(kSignalingModes);
+
+constexpr int32_t kAudioMuxVer[] = {-1, 0, 1, 2};
+constexpr size_t kAudioMuxVerSize = size(kAudioMuxVer);
+
+constexpr int32_t kSbrModes[] = {-1, 0, 1, 2};
+constexpr size_t kSbrModesSize = size(kSbrModes);
+
+constexpr AACENC_METADATA_DRC_PROFILE kMetaDataDrcProfiles[] = {
+    AACENC_METADATA_DRC_NONE,       AACENC_METADATA_DRC_FILMSTANDARD,
+    AACENC_METADATA_DRC_FILMLIGHT,  AACENC_METADATA_DRC_MUSICSTANDARD,
+    AACENC_METADATA_DRC_MUSICLIGHT, AACENC_METADATA_DRC_SPEECH,
+    AACENC_METADATA_DRC_NOT_PRESENT};
+constexpr size_t kMetaDataDrcProfilesSize = size(kMetaDataDrcProfiles);
+
+enum {
+    IDX_SBR_MODE = 0,
+    IDX_AAC_AOT,
+    IDX_SAMPLE_RATE,
+    IDX_BIT_RATE_1,
+    IDX_BIT_RATE_2,
+    IDX_BIT_RATE_3,
+    IDX_CHANNEL,
+    IDX_IDENTIFIER,
+    IDX_SBR_RATIO,
+    IDX_METADATA_DRC_PROFILE,
+    IDX_METADATA_COMP_PROFILE,
+    IDX_METADATA_DRC_TARGET_REF_LEVEL,
+    IDX_METADATA_COMP_TARGET_REF_LEVEL,
+    IDX_METADATA_PROG_LEVEL_PRESENT,
+    IDX_METADATA_PROG_LEVEL,
+    IDX_METADATA_PCE_MIXDOWN_IDX_PRESENT,
+    IDX_METADATA_ETSI_DMXLVL_PRESENT,
+    IDX_METADATA_CENTER_MIX_LEVEL,
+    IDX_METADATA_SURROUND_MIX_LEVEL,
+    IDX_METADATA_DOLBY_SURROUND_MODE,
+    IDX_METADATA_DRC_PRESENTATION_MODE,
+    IDX_METADATA_EXT_ANC_DATA_ENABLE,
+    IDX_METADATA_EXT_DOWNMIX_LEVEL_ENABLE,
+    IDX_METADATA_EXT_DOWNMIX_LEVEL_A,
+    IDX_METADATA_EXT_DOWNMIX_LEVEL_B,
+    IDX_METADATA_DMX_GAIN_ENABLE,
+    IDX_METADATA_DMX_GAIN_5,
+    IDX_METADATA_DMX_GAIN_2,
+    IDX_METADATA_LFE_DMX_ENABLE,
+    IDX_METADATA_LFE_DMX_LEVEL,
+    IDX_IN_BUFFER_INDEX_1,
+    IDX_IN_BUFFER_INDEX_2,
+    IDX_IN_BUFFER_INDEX_3,
+    IDX_BIT_RATE_MODE,
+    IDX_GRANULE_LENGTH,
+    IDX_CHANNELORDER,
+    IDX_AFTERBURNER,
+    IDX_BANDWIDTH,
+    IDX_PEAK_BITRATE,
+    IDX_HEADER_PERIOD,
+    IDX_SIGNALING_MODE,
+    IDX_TPSUBFRAMES,
+    IDX_AUDIOMUXVER,
+    IDX_PROTECTION,
+    IDX_ANCILLARY_BITRATE,
+    IDX_METADATA_MODE,
+    IDX_LAST
+};
+
+template <typename type1, typename type2, typename type3>
+auto generateNumberInRangeFromData(type1 data, type2 min, type3 max) -> decltype(max) {
+    return (data % (1 + max - min)) + min;
+}
+
+class Codec {
+   public:
+    ~Codec() { deInitEncoder(); }
+    bool initEncoder(uint8_t **dataPtr, size_t *sizePtr);
+    void encodeFrames(const uint8_t *data, size_t size);
+    void deInitEncoder();
+
+   private:
+    template <typename type1, typename type2, typename type3>
+    void setAACParam(type1 data, const AACENC_PARAM aacParam, type2 min, type2 max,
+                     const type3 *array = nullptr);
+    void setupMetaData(uint8_t *data);
+
+    HANDLE_AACENCODER mEncoder = nullptr;
+    AACENC_MetaData mMetaData = {};
+    uint32_t mInBufferIdx_1 = 0;
+    uint32_t mInBufferIdx_2 = 0;
+    uint32_t mInBufferIdx_3 = 0;
+};
+
+void Codec::setupMetaData(uint8_t *data) {
+    uint32_t drcProfileIndex = generateNumberInRangeFromData(data[IDX_METADATA_DRC_PROFILE], 0,
+                                                             kMetaDataDrcProfilesSize - 1);
+    AACENC_METADATA_DRC_PROFILE drcProfile = kMetaDataDrcProfiles[drcProfileIndex];
+    mMetaData.drc_profile = drcProfile;
+
+    uint32_t compProfileIndex = generateNumberInRangeFromData(data[IDX_METADATA_COMP_PROFILE], 0,
+                                                              kMetaDataDrcProfilesSize - 1);
+    AACENC_METADATA_DRC_PROFILE compProfile = kMetaDataDrcProfiles[compProfileIndex];
+    mMetaData.comp_profile = compProfile;
+
+    INT drcTargetRefLevel =
+        generateNumberInRangeFromData(data[IDX_METADATA_DRC_TARGET_REF_LEVEL], 0, UINT8_MAX);
+    mMetaData.drc_TargetRefLevel = drcTargetRefLevel;
+
+    INT compTargetRefLevel =
+        generateNumberInRangeFromData(data[IDX_METADATA_COMP_TARGET_REF_LEVEL], 0, UINT8_MAX);
+    mMetaData.comp_TargetRefLevel = compTargetRefLevel;
+
+    INT isProgRefLevelPresent =
+        generateNumberInRangeFromData(data[IDX_METADATA_PROG_LEVEL_PRESENT], 0, 1);
+    mMetaData.prog_ref_level_present = isProgRefLevelPresent;
+
+    INT progRefLevel = generateNumberInRangeFromData(data[IDX_METADATA_PROG_LEVEL], 0, UINT8_MAX);
+    mMetaData.prog_ref_level = progRefLevel;
+
+    UCHAR isPCEMixdownIdxPresent =
+        generateNumberInRangeFromData(data[IDX_METADATA_PCE_MIXDOWN_IDX_PRESENT], 0, 1);
+    mMetaData.PCE_mixdown_idx_present = isPCEMixdownIdxPresent;
+
+    UCHAR isETSIDmxLvlPresent =
+        generateNumberInRangeFromData(data[IDX_METADATA_ETSI_DMXLVL_PRESENT], 0, 1);
+    mMetaData.ETSI_DmxLvl_present = isETSIDmxLvlPresent;
+
+    SCHAR centerMixLevel = generateNumberInRangeFromData(data[IDX_METADATA_CENTER_MIX_LEVEL], 0, 7);
+    mMetaData.centerMixLevel = centerMixLevel;
+
+    SCHAR surroundMixLevel =
+        generateNumberInRangeFromData(data[IDX_METADATA_SURROUND_MIX_LEVEL], 0, 7);
+    mMetaData.surroundMixLevel = surroundMixLevel;
+
+    UCHAR dolbySurroundMode =
+        generateNumberInRangeFromData(data[IDX_METADATA_DOLBY_SURROUND_MODE], 0, 2);
+    mMetaData.dolbySurroundMode = dolbySurroundMode;
+
+    UCHAR drcPresentationMode =
+        generateNumberInRangeFromData(data[IDX_METADATA_DRC_PRESENTATION_MODE], 0, 2);
+    mMetaData.drcPresentationMode = drcPresentationMode;
+
+    UCHAR extAncDataEnable =
+        generateNumberInRangeFromData(data[IDX_METADATA_EXT_ANC_DATA_ENABLE], 0, 1);
+    mMetaData.ExtMetaData.extAncDataEnable = extAncDataEnable;
+
+    UCHAR extDownmixLevelEnable =
+        generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_ENABLE], 0, 1);
+    mMetaData.ExtMetaData.extDownmixLevelEnable = extDownmixLevelEnable;
+
+    UCHAR extDownmixLevel_A =
+        generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_A], 0, 7);
+    mMetaData.ExtMetaData.extDownmixLevel_A = extDownmixLevel_A;
+
+    UCHAR extDownmixLevel_B =
+        generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_B], 0, 7);
+    mMetaData.ExtMetaData.extDownmixLevel_B = extDownmixLevel_B;
+
+    UCHAR dmxGainEnable = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_ENABLE], 0, 1);
+    mMetaData.ExtMetaData.dmxGainEnable = dmxGainEnable;
+
+    INT dmxGain5 = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_5], 0, UINT8_MAX);
+    mMetaData.ExtMetaData.dmxGain5 = dmxGain5;
+
+    INT dmxGain2 = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_2], 0, UINT8_MAX);
+    mMetaData.ExtMetaData.dmxGain2 = dmxGain2;
+
+    UCHAR lfeDmxEnable = generateNumberInRangeFromData(data[IDX_METADATA_LFE_DMX_ENABLE], 0, 1);
+    mMetaData.ExtMetaData.lfeDmxEnable = lfeDmxEnable;
+
+    UCHAR lfeDmxLevel = generateNumberInRangeFromData(data[IDX_METADATA_LFE_DMX_LEVEL], 0, 15);
+    mMetaData.ExtMetaData.lfeDmxLevel = lfeDmxLevel;
+}
+
+template <typename type1, typename type2, typename type3>
+void Codec::setAACParam(type1 data, const AACENC_PARAM aacParam, type2 min, type2 max,
+                        const type3 *array) {
+    auto value = 0;
+    if (array) {
+        uint32_t index = generateNumberInRangeFromData(data, min, max);
+        value = array[index];
+    } else {
+        value = generateNumberInRangeFromData(data, min, max);
+    }
+    aacEncoder_SetParam(mEncoder, aacParam, value);
+    (void)aacEncoder_GetParam(mEncoder, aacParam);
+}
+
+bool Codec::initEncoder(uint8_t **dataPtr, size_t *sizePtr) {
+    uint8_t *data = *dataPtr;
+
+    if (AACENC_OK != aacEncOpen(&mEncoder, 0, 0)) {
+        return false;
+    }
+
+    setAACParam<uint8_t, size_t, int32_t>(data[IDX_SBR_MODE], AACENC_SBR_MODE, 0, kSbrModesSize - 1,
+                                          kSbrModes);
+
+    setAACParam<uint8_t, size_t, int32_t>(data[IDX_SBR_RATIO], AACENC_SBR_RATIO, 0,
+                                          kSbrRatiosSize - 1, kSbrRatios);
+
+    setAACParam<uint8_t, size_t, AUDIO_OBJECT_TYPE>(data[IDX_AAC_AOT], AACENC_AOT, 0,
+                                                    kAudioObjectTypesSize - 1, kAudioObjectTypes);
+
+    setAACParam<uint8_t, size_t, int32_t>(data[IDX_SAMPLE_RATE], AACENC_SAMPLERATE, 0,
+                                          kSampleRatesSize - 1, kSampleRates);
+
+    uint32_t tempValue =
+        (data[IDX_BIT_RATE_1] << 16) | (data[IDX_BIT_RATE_2] << 8) | data[IDX_BIT_RATE_3];
+    setAACParam<uint8_t, uint32_t, uint32_t>(tempValue, AACENC_BITRATE, kMinBitRate, kMaxBitRate);
+
+    setAACParam<uint8_t, size_t, CHANNEL_MODE>(data[IDX_CHANNEL], AACENC_CHANNELMODE, 0,
+                                               kChannelModesSize - 1, kChannelModes);
+
+    setAACParam<uint8_t, size_t, TRANSPORT_TYPE>(data[IDX_IDENTIFIER], AACENC_TRANSMUX, 0,
+                                                 kIdentifiersSize - 1, kIdentifiers);
+
+    setAACParam<uint8_t, size_t, int32_t>(data[IDX_BIT_RATE_MODE], AACENC_BITRATEMODE, 0,
+                                          kBitRateModesSize - 1, kBitRateModes);
+
+    setAACParam<uint8_t, size_t, int32_t>(data[IDX_GRANULE_LENGTH], AACENC_GRANULE_LENGTH, 0,
+                                          kGranuleLengthsSize - 1, kGranuleLengths);
+
+    setAACParam<uint8_t, size_t, int32_t>(data[IDX_CHANNELORDER], AACENC_CHANNELORDER, 0,
+                                          kChannelOrderSize - 1, kChannelOrder);
+
+    setAACParam<uint8_t, int32_t, int32_t>(data[IDX_AFTERBURNER], AACENC_AFTERBURNER, 0, 1);
+
+    setAACParam<uint8_t, int32_t, int32_t>(data[IDX_BANDWIDTH], AACENC_BANDWIDTH, 0, 1);
+
+    setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_PEAK_BITRATE], AACENC_PEAK_BITRATE,
+                                             kMinBitRate, kMinBitRate);
+
+    setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_HEADER_PERIOD], AACENC_HEADER_PERIOD, 0,
+                                             UINT8_MAX);
+
+    setAACParam<uint8_t, size_t, int32_t>(data[IDX_SIGNALING_MODE], AACENC_SIGNALING_MODE, 0,
+                                          kSignalingModesSize - 1, kSignalingModes);
+
+    setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_TPSUBFRAMES], AACENC_TPSUBFRAMES, 0,
+                                             UINT8_MAX);
+
+    setAACParam<uint8_t, size_t, int32_t>(data[IDX_AUDIOMUXVER], AACENC_AUDIOMUXVER, 0,
+                                          kAudioMuxVerSize - 1, kAudioMuxVer);
+
+    setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_PROTECTION], AACENC_PROTECTION, 0, 1);
+
+    setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_ANCILLARY_BITRATE], AACENC_ANCILLARY_BITRATE,
+                                             0, kMaxBitRate);
+
+    setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_METADATA_MODE], AACENC_METADATA_MODE, 0, 3);
+
+    AACENC_InfoStruct encInfo;
+    aacEncInfo(mEncoder, &encInfo);
+
+    mInBufferIdx_1 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_1], 0, kMaxBuffers - 1);
+    mInBufferIdx_2 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_2], 0, kMaxBuffers - 1);
+    mInBufferIdx_3 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_3], 0, kMaxBuffers - 1);
+
+    setupMetaData(data);
+
+    // Not re-using the data which was used for configuration for encoding
+    *dataPtr += IDX_LAST;
+    *sizePtr -= IDX_LAST;
+
+    return true;
+}
+
+static void deleteBuffers(uint8_t **buffers, size_t size) {
+    for (size_t n = 0; n < size; ++n) {
+        delete[] buffers[n];
+    }
+    delete[] buffers;
+}
+
+void Codec::encodeFrames(const uint8_t *data, size_t size) {
+    uint8_t *audioData = (uint8_t *)data;
+    uint8_t *ancData = (uint8_t *)data;
+    size_t audioSize = size;
+    size_t ancSize = size;
+
+    while ((audioSize > 0) && (ancSize > 0)) {
+        AACENC_InArgs inargs;
+        memset(&inargs, 0, sizeof(inargs));
+        inargs.numInSamples = audioSize / sizeof(int16_t);
+        inargs.numAncBytes = ancSize;
+
+        void *buffers[] = {(void *)audioData, (void *)ancData, &mMetaData};
+        INT bufferIds[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA, IN_METADATA_SETUP};
+        INT bufferSizes[] = {static_cast<INT>(audioSize), static_cast<INT>(ancSize),
+                             static_cast<INT>(sizeof(mMetaData))};
+        INT bufferElSizes[] = {sizeof(int16_t), sizeof(UCHAR), sizeof(AACENC_MetaData)};
+
+        void *inBuffer[kMaxBuffers] = {};
+        INT inBufferIds[kMaxBuffers] = {};
+        INT inBufferSize[kMaxBuffers] = {};
+        INT inBufferElSize[kMaxBuffers] = {};
+        for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) {
+            uint32_t Idxs[] = {mInBufferIdx_1, mInBufferIdx_2, mInBufferIdx_3};
+            inBuffer[buffer] = buffers[Idxs[buffer]];
+            inBufferIds[buffer] = bufferIds[Idxs[buffer]];
+            inBufferSize[buffer] = bufferSizes[Idxs[buffer]];
+            inBufferElSize[buffer] = bufferElSizes[Idxs[buffer]];
+        }
+
+        AACENC_BufDesc inBufDesc;
+        inBufDesc.numBufs = kMaxBuffers;
+        inBufDesc.bufs = (void **)&inBuffer;
+        inBufDesc.bufferIdentifiers = inBufferIds;
+        inBufDesc.bufSizes = inBufferSize;
+        inBufDesc.bufElSizes = inBufferElSize;
+
+        uint8_t **outPtrRef = new uint8_t *[kMaxBuffers];
+        for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) {
+            outPtrRef[buffer] = new uint8_t[kMaxOutputBufferSize];
+        }
+
+        void *outBuffer[kMaxBuffers];
+        INT outBufferIds[kMaxBuffers];
+        INT outBufferSize[kMaxBuffers];
+        INT outBufferElSize[kMaxBuffers];
+
+        for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) {
+            outBuffer[buffer] = outPtrRef[buffer];
+            outBufferIds[buffer] = OUT_BITSTREAM_DATA;
+            outBufferSize[buffer] = (INT)kMaxOutputBufferSize;
+            outBufferElSize[buffer] = sizeof(UCHAR);
+        }
+
+        AACENC_BufDesc outBufDesc;
+        outBufDesc.numBufs = kMaxBuffers;
+        outBufDesc.bufs = (void **)&outBuffer;
+        outBufDesc.bufferIdentifiers = outBufferIds;
+        outBufDesc.bufSizes = outBufferSize;
+        outBufDesc.bufElSizes = outBufferElSize;
+
+        AACENC_OutArgs outargs = {};
+        aacEncEncode(mEncoder, &inBufDesc, &outBufDesc, &inargs, &outargs);
+
+        if (outargs.numOutBytes == 0) {
+            if (audioSize > 0) {
+                ++audioData;
+                --audioSize;
+            }
+            if (ancSize > 0) {
+                ++ancData;
+                --ancSize;
+            }
+        } else {
+            size_t audioConsumed = outargs.numInSamples * sizeof(int16_t);
+            audioData += audioConsumed;
+            audioSize -= audioConsumed;
+
+            size_t ancConsumed = outargs.numAncBytes;
+            ancData += ancConsumed;
+            ancSize -= ancConsumed;
+        }
+        deleteBuffers(outPtrRef, kMaxBuffers);
+
+        // break out of loop if only metadata was sent in all the input buffers
+        // as sending it multiple times in a loop is redundant.
+        if ((mInBufferIdx_1 == kMaxBuffers - 1) && (mInBufferIdx_2 == kMaxBuffers - 1) &&
+            (mInBufferIdx_3 == kMaxBuffers - 1)) {
+            break;
+        }
+    }
+}
+
+void Codec::deInitEncoder() { aacEncClose(&mEncoder); }
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
+    if (size < IDX_LAST) {
+        return 0;
+    }
+    Codec encoder;
+    if (encoder.initEncoder(const_cast<uint8_t **>(&data), &size)) {
+        encoder.encodeFrames(data, size);
+    }
+    return 0;
+}
diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h
index 56f4ec1..d7928c0 100644
--- a/libAACdec/include/aacdecoder_lib.h
+++ b/libAACdec/include/aacdecoder_lib.h
@@ -1032,7 +1032,7 @@ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_Fill(HANDLE_AACDECODER self,
  * \param self          AAC decoder handle.
  * \param pTimeData     Pointer to external output buffer where the decoded PCM
  * samples will be stored into.
- * \param timeDataSize  Size of external output buffer.
+ * \param timeDataSize  Size of external output buffer in PCM samples.
  * \param flags         Bit field with flags for the decoder: \n
  *                      (flags & AACDEC_CONCEAL) == 1: Do concealment. \n
  *                      (flags & AACDEC_FLUSH) == 2: Discard input data. Flush
diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp
index c18e5e9..d5f0cea 100644
--- a/libAACdec/src/aacdecoder.cpp
+++ b/libAACdec/src/aacdecoder.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -494,6 +494,75 @@ static AAC_DECODER_ERROR CDataStreamElement_Read(HANDLE_AACDECODER self,
   return error;
 }
 
+static INT findElementInstanceTag(
+    INT elementTag, MP4_ELEMENT_ID elementId,
+    CAacDecoderChannelInfo **pAacDecoderChannelInfo, INT nChannels,
+    MP4_ELEMENT_ID *pElementIdTab, INT nElements) {
+  int el, chCnt = 0;
+
+  for (el = 0; el < nElements; el++) {
+    switch (pElementIdTab[el]) {
+      case ID_CPE:
+      case ID_SCE:
+      case ID_LFE:
+        if ((elementTag == pAacDecoderChannelInfo[chCnt]->ElementInstanceTag) &&
+            (elementId == pElementIdTab[el])) {
+          return 1; /* element instance tag found */
+        }
+        chCnt += (pElementIdTab[el] == ID_CPE) ? 2 : 1;
+        break;
+      default:
+        break;
+    }
+    if (chCnt >= nChannels) break;
+    if (pElementIdTab[el] == ID_END) break;
+  }
+
+  return 0; /* element instance tag not found */
+}
+
+static INT validateElementInstanceTags(
+    CProgramConfig *pce, CAacDecoderChannelInfo **pAacDecoderChannelInfo,
+    INT nChannels, MP4_ELEMENT_ID *pElementIdTab, INT nElements) {
+  if (nChannels >= pce->NumChannels) {
+    for (int el = 0; el < pce->NumFrontChannelElements; el++) {
+      if (!findElementInstanceTag(pce->FrontElementTagSelect[el],
+                                  pce->FrontElementIsCpe[el] ? ID_CPE : ID_SCE,
+                                  pAacDecoderChannelInfo, nChannels,
+                                  pElementIdTab, nElements)) {
+        return 0; /* element instance tag not in raw_data_block() */
+      }
+    }
+    for (int el = 0; el < pce->NumSideChannelElements; el++) {
+      if (!findElementInstanceTag(pce->SideElementTagSelect[el],
+                                  pce->SideElementIsCpe[el] ? ID_CPE : ID_SCE,
+                                  pAacDecoderChannelInfo, nChannels,
+                                  pElementIdTab, nElements)) {
+        return 0; /* element instance tag not in raw_data_block() */
+      }
+    }
+    for (int el = 0; el < pce->NumBackChannelElements; el++) {
+      if (!findElementInstanceTag(pce->BackElementTagSelect[el],
+                                  pce->BackElementIsCpe[el] ? ID_CPE : ID_SCE,
+                                  pAacDecoderChannelInfo, nChannels,
+                                  pElementIdTab, nElements)) {
+        return 0; /* element instance tag not in raw_data_block() */
+      }
+    }
+    for (int el = 0; el < pce->NumLfeChannelElements; el++) {
+      if (!findElementInstanceTag(pce->LfeElementTagSelect[el], ID_LFE,
+                                  pAacDecoderChannelInfo, nChannels,
+                                  pElementIdTab, nElements)) {
+        return 0; /* element instance tag not in raw_data_block() */
+      }
+    }
+  } else {
+    return 0; /* too less decoded audio channels */
+  }
+
+  return 1; /* all element instance tags found in raw_data_block() */
+}
+
 /*!
   \brief Read Program Config Element
 
@@ -1417,11 +1486,7 @@ static void CAacDecoder_AcceptFlags(HANDLE_AACDECODER self,
                                     const CSAudioSpecificConfig *asc,
                                     UINT flags, UINT *elFlags, int streamIndex,
                                     int elementOffset) {
-  {
-    FDKmemcpy(
-        self->elFlags, elFlags,
-        sizeof(*elFlags) * (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1));
-  }
+  FDKmemcpy(self->elFlags, elFlags, sizeof(self->elFlags));
 
   self->flags[streamIndex] = flags;
 }
@@ -1524,8 +1589,14 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
   INT flushChannels = 0;
 
   UINT flags;
+  /* elFlags[(3*MAX_CHANNELS + (MAX_CHANNELS)/2 + 4 * (MAX_TRACKS) + 1]
+     where MAX_CHANNELS is (8*2) and MAX_TRACKS is 1 */
   UINT elFlags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)];
 
+  UCHAR sbrEnabled = self->sbrEnabled;
+  UCHAR sbrEnabledPrev = self->sbrEnabledPrev;
+  UCHAR mpsEnableCurr = self->mpsEnableCurr;
+
   if (!self) return AAC_DEC_INVALID_HANDLE;
 
   UCHAR downscaleFactor = self->downscaleFactor;
@@ -1709,7 +1780,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
             asc->m_sc.m_usacConfig.m_usacNumElements;
       }
 
-      self->mpsEnableCurr = 0;
+      mpsEnableCurr = 0;
       for (int _el = 0;
            _el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements;
            _el++) {
@@ -1729,7 +1800,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
           self->usacStereoConfigIndex[el] =
               asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex;
           if (self->elements[el] == ID_USAC_CPE) {
-            self->mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0;
+            mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0;
           }
         }
 
@@ -1865,7 +1936,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
       self->useLdQmfTimeAlign =
           asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
     }
-    if (self->sbrEnabled != asc->m_sbrPresentFlag) {
+    if (sbrEnabled != asc->m_sbrPresentFlag) {
       ascChanged = 1;
     }
   }
@@ -1881,13 +1952,13 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
   flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0;
   flags |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0;
   if (asc->m_sbrPresentFlag) {
-    self->sbrEnabled = 1;
-    self->sbrEnabledPrev = 1;
+    sbrEnabled = 1;
+    sbrEnabledPrev = 1;
   } else {
-    self->sbrEnabled = 0;
-    self->sbrEnabledPrev = 0;
+    sbrEnabled = 0;
+    sbrEnabledPrev = 0;
   }
-  if (self->sbrEnabled && asc->m_extensionSamplingFrequency) {
+  if (sbrEnabled && asc->m_extensionSamplingFrequency) {
     if (downscaleFactor != 1 && (downscaleFactor)&1) {
       return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* SBR needs an even downscale
                                                   factor */
@@ -1914,7 +1985,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
   flags |= (asc->m_hcrFlag) ? AC_ER_HCR : 0;
 
   if (asc->m_aot == AOT_ER_AAC_ELD) {
-    self->mpsEnableCurr = 0;
+    mpsEnableCurr = 0;
     flags |= AC_ELD;
     flags |= (asc->m_sbrPresentFlag)
                  ? AC_SBR_PRESENT
@@ -1925,7 +1996,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
                  ? AC_MPS_PRESENT
                  : 0;
     if (self->mpsApplicable) {
-      self->mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
+      mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
     }
   }
   flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0;
@@ -2006,7 +2077,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
   /* set AC_USAC_SCFGI3 globally if any usac element uses */
   switch (asc->m_aot) {
     case AOT_USAC:
-      if (self->sbrEnabled) {
+      if (sbrEnabled) {
         for (int _el = 0;
              _el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements;
              _el++) {
@@ -2043,7 +2114,7 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
     */
     switch (asc->m_aot) {
       case AOT_USAC:
-        if (self->sbrEnabled) {
+        if (sbrEnabled) {
           const UCHAR map_sbrRatio_2_nAnaBands[] = {16, 24, 32};
 
           FDK_ASSERT(asc->m_sc.m_usacConfig.m_sbrRatioIndex > 0);
@@ -2071,11 +2142,11 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
         }
         break;
       case AOT_ER_AAC_ELD:
-        if (self->mpsEnableCurr &&
+        if (mpsEnableCurr &&
             asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) {
-          SAC_INPUT_CONFIG sac_interface =
-              (self->sbrEnabled && self->hSbrDecoder) ? SAC_INTERFACE_QMF
-                                                      : SAC_INTERFACE_TIME;
+          SAC_INPUT_CONFIG sac_interface = (sbrEnabled && self->hSbrDecoder)
+                                               ? SAC_INTERFACE_QMF
+                                               : SAC_INTERFACE_TIME;
           mpegSurroundDecoder_ConfigureQmfDomain(
               (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface,
               (UINT)self->streamInfo.aacSampleRate, asc->m_aot);
@@ -2430,6 +2501,9 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
 
   CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex,
                           elementOffset);
+  self->sbrEnabled = sbrEnabled;
+  self->sbrEnabledPrev = sbrEnabledPrev;
+  self->mpsEnableCurr = mpsEnableCurr;
 
   /* Update externally visible copy of flags */
   self->streamInfo.flags = self->flags[0];
@@ -2968,6 +3042,24 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
 
   } /* while ( (type != ID_END) ... ) */
 
+  if (!(self->flags[streamIndex] &
+        (AC_USAC | AC_RSVD50 | AC_RSV603DA | AC_BSAC | AC_LD | AC_ELD | AC_ER |
+         AC_SCALABLE)) &&
+      (self->streamInfo.channelConfig == 0) && pce->isValid &&
+      (ErrorStatus == AAC_DEC_OK) && self->frameOK &&
+      !(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) {
+    /* Check whether all PCE listed element instance tags are present in
+     * raw_data_block() */
+    if (!validateElementInstanceTags(
+            &self->pce, self->pAacDecoderChannelInfo, aacChannels,
+            channel_elements,
+            fMin(channel_element_count, (int)(sizeof(channel_elements) /
+                                              sizeof(*channel_elements))))) {
+      ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
+      self->frameOK = 0;
+    }
+  }
+
   if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) {
     /* float decoder checks if bitsLeft is in range 0-7; only prerollAUs are
      * byteAligned with respect to the first bit */
diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index 9d36d10..0c83191 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1626,6 +1626,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,
         /* set params */
         sbrDecoder_SetParam(self->hSbrDecoder, SBR_SYSTEM_BITSTREAM_DELAY,
                             self->sbrParams.bsDelay);
+        sbrDecoder_SetParam(
+            self->hSbrDecoder, SBR_FLUSH_DATA,
+            (flags & AACDEC_FLUSH) |
+                ((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH
+                                                                  : 0));
 
         sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, 1);
 
diff --git a/libAACdec/src/rvlc.cpp b/libAACdec/src/rvlc.cpp
index b7a9be1..0b80364 100644
--- a/libAACdec/src/rvlc.cpp
+++ b/libAACdec/src/rvlc.cpp
@@ -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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -628,7 +628,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc,
 
   SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd;
   SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc;
-  UCHAR *pEscEscCnt = &(pRvlc->numDecodedEscapeWordsEsc);
+  UCHAR escEscCnt = pRvlc->numDecodedEscapeWordsEsc;
   UCHAR *pEscBwdCnt = &(pRvlc->numDecodedEscapeWordsBwd);
 
   pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_bwd);
@@ -636,7 +636,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc,
 
   *pEscBwdCnt = 0;
   pRvlc->direction = BWD;
-  pScfEsc += *pEscEscCnt - 1; /* set pScfEsc to last entry */
+  pScfEsc += escEscCnt - 1; /* set pScfEsc to last entry */
   pRvlc->firstScf = 0;
   pRvlc->firstNrg = 0;
   pRvlc->firstIs = 0;
@@ -651,7 +651,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc,
     }
     dpcm -= TABLE_OFFSET;
     if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
-      if (pRvlc->length_of_rvlc_escapes) {
+      if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
         pRvlc->conceal_min = bnds;
         return;
       } else {
@@ -694,7 +694,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc,
           }
           dpcm -= TABLE_OFFSET;
           if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
-            if (pRvlc->length_of_rvlc_escapes) {
+            if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
               pScfBwd[bnds] = position;
               pRvlc->conceal_min = fMax(0, bnds - offset);
               return;
@@ -731,7 +731,8 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc,
             }
             dpcm -= TABLE_OFFSET;
             if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
-              if (pRvlc->length_of_rvlc_escapes) {
+              if ((pRvlc->length_of_rvlc_escapes) ||
+                  (*pEscBwdCnt >= escEscCnt)) {
                 pScfBwd[bnds] = noisenrg;
                 pRvlc->conceal_min = fMax(0, bnds - offset);
                 return;
@@ -762,7 +763,7 @@ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc,
           }
           dpcm -= TABLE_OFFSET;
           if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
-            if (pRvlc->length_of_rvlc_escapes) {
+            if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
               pScfBwd[bnds] = factor;
               pRvlc->conceal_min = fMax(0, bnds - offset);
               return;
diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp
index a8dadc0..ca1a6a2 100644
--- a/libAACdec/src/usacdec_acelp.cpp
+++ b/libAACdec/src/usacdec_acelp.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -719,7 +719,7 @@ static void ConcealPitchLag(CAcelpStaticMem *acelp_mem, const int PIT_MAX,
   UCHAR *pold_T0_frac = &acelp_mem->old_T0_frac;
 
   if ((int)*pold_T0 >= PIT_MAX) {
-    *pold_T0 = (UCHAR)(PIT_MAX - 5);
+    *pold_T0 = (USHORT)(PIT_MAX - 5);
   }
   *pT0 = (int)*pold_T0;
   *pT0_frac = (int)*pold_T0_frac;
diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h
index 71f7556..f0f23b4 100644
--- a/libAACenc/include/aacenc_lib.h
+++ b/libAACenc/include/aacenc_lib.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1643,7 +1643,7 @@ AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder,
  *
  * \return
  *          - AACENC_OK, on succes.
- *          - AACENC_INIT_ERROR, on failure.
+ *          - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure.
  */
 AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder,
                         AACENC_InfoStruct *pInfo);
diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp
index caa62c5..c11db27 100644
--- a/libAACenc/src/aacenc_lib.cpp
+++ b/libAACenc/src/aacenc_lib.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -2521,6 +2521,11 @@ AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder,
                         AACENC_InfoStruct *pInfo) {
   AACENC_ERROR err = AACENC_OK;
 
+  if ((hAacEncoder == NULL) || (pInfo == NULL)) {
+    err = AACENC_INVALID_HANDLE;
+    goto bail;
+  }
+
   FDKmemclear(pInfo, sizeof(AACENC_InfoStruct));
   pInfo->confSize = 64; /* pre-initialize */
 
diff --git a/libDRCdec/src/drcDec_reader.cpp b/libDRCdec/src/drcDec_reader.cpp
index b3ec187..b080f50 100644
--- a/libDRCdec/src/drcDec_reader.cpp
+++ b/libDRCdec/src/drcDec_reader.cpp
@@ -917,7 +917,7 @@ static void _skipEqCoefficients(HANDLE_FDK_BITSTREAM hBs) {
       firFilterOrder;
   int uniqueEqSubbandGainsCount, eqSubbandGainRepresentation,
       eqSubbandGainCount;
-  EQ_SUBBAND_GAIN_FORMAT eqSubbandGainFormat;
+  int eqSubbandGainFormat;
 
   eqDelayMaxPresent = FDKreadBits(hBs, 1);
   if (eqDelayMaxPresent) {
@@ -958,7 +958,7 @@ static void _skipEqCoefficients(HANDLE_FDK_BITSTREAM hBs) {
   uniqueEqSubbandGainsCount = FDKreadBits(hBs, 6);
   if (uniqueEqSubbandGainsCount > 0) {
     eqSubbandGainRepresentation = FDKreadBits(hBs, 1);
-    eqSubbandGainFormat = (EQ_SUBBAND_GAIN_FORMAT)FDKreadBits(hBs, 4);
+    eqSubbandGainFormat = FDKreadBits(hBs, 4);
     switch (eqSubbandGainFormat) {
       case GF_QMF32:
         eqSubbandGainCount = 32;
diff --git a/libFDK/include/nlc_dec.h b/libFDK/include/nlc_dec.h
index cca97f1..aded569 100644
--- a/libFDK/include/nlc_dec.h
+++ b/libFDK/include/nlc_dec.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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -159,9 +159,6 @@ typedef enum {
 #ifndef HUFFDEC_PARAMS
 #define HUFFDEC_PARMS
 
-#define PAIR_SHIFT 4
-#define PAIR_MASK 0xf
-
 #define MAX_ENTRIES 168
 #define HANDLE_HUFF_NODE const SHORT(*)[MAX_ENTRIES][2]
 
diff --git a/libFDK/src/autocorr2nd.cpp b/libFDK/src/autocorr2nd.cpp
index 718a555..8c5673c 100644
--- a/libFDK/src/autocorr2nd.cpp
+++ b/libFDK/src/autocorr2nd.cpp
@@ -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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -102,11 +102,6 @@ amm-info@iis.fraunhofer.de
 
 #include "autocorr2nd.h"
 
-/*  If the accumulator does not provide enough overflow bits,
-    products have to be shifted down in the autocorrelation below. */
-#define SHIFT_FACTOR (5)
-#define SHIFT >> (SHIFT_FACTOR)
-
 /*!
  *
  * \brief Calculate second order autocorrelation using 2 accumulators
@@ -126,45 +121,49 @@ INT autoCorr2nd_real(
 
   const FIXP_DBL *realBuf = reBuffer;
 
+  const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)(len / 2)), 1);
   /*
     r11r,r22r
     r01r,r12r
     r02r
   */
   pReBuf = realBuf - 2;
-  accu5 = ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3]))
-               SHIFT);
+  accu5 =
+      ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >>
+       len_scale);
   pReBuf++;
 
   /* len must be even */
-  accu1 = fPow2Div2(pReBuf[0]) SHIFT;
-  accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) SHIFT;
+  accu1 = fPow2Div2(pReBuf[0]) >> len_scale;
+  accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) >> len_scale;
   pReBuf++;
 
   for (j = (len - 2) >> 1; j != 0; j--, pReBuf += 2) {
-    accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) SHIFT);
+    accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) >> len_scale);
 
-    accu3 += ((fMultDiv2(pReBuf[0], pReBuf[1]) +
-               fMultDiv2(pReBuf[1], pReBuf[2])) SHIFT);
+    accu3 +=
+        ((fMultDiv2(pReBuf[0], pReBuf[1]) + fMultDiv2(pReBuf[1], pReBuf[2])) >>
+         len_scale);
 
-    accu5 += ((fMultDiv2(pReBuf[0], pReBuf[2]) +
-               fMultDiv2(pReBuf[1], pReBuf[3])) SHIFT);
+    accu5 +=
+        ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >>
+         len_scale);
   }
 
-  accu2 = (fPow2Div2(realBuf[-2]) SHIFT);
+  accu2 = (fPow2Div2(realBuf[-2]) >> len_scale);
   accu2 += accu1;
 
-  accu1 += (fPow2Div2(realBuf[len - 2]) SHIFT);
+  accu1 += (fPow2Div2(realBuf[len - 2]) >> len_scale);
 
-  accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) SHIFT);
+  accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) >> len_scale);
   accu4 += accu3;
 
-  accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) SHIFT);
+  accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) >> len_scale);
 
   mScale = CntLeadingZeros(
                (accu1 | accu2 | fAbs(accu3) | fAbs(accu4) | fAbs(accu5))) -
            1;
-  autoCorrScaling = mScale - 1 - SHIFT_FACTOR; /* -1 because of fMultDiv2*/
+  autoCorrScaling = mScale - 1 - len_scale; /* -1 because of fMultDiv2*/
 
   /* Scale to common scale factor */
   ac->r11r = accu1 << mScale;
@@ -190,7 +189,7 @@ INT autoCorr2nd_cplx(
     const FIXP_DBL *imBuffer, /*!< Pointer to imag part of input samples */
     const int len /*!< Number of input samples (should be smaller than 128) */
 ) {
-  int j, autoCorrScaling, mScale, len_scale;
+  int j, autoCorrScaling, mScale;
 
   FIXP_DBL accu0, accu1, accu2, accu3, accu4, accu5, accu6, accu7, accu8;
 
@@ -199,7 +198,7 @@ INT autoCorr2nd_cplx(
   const FIXP_DBL *realBuf = reBuffer;
   const FIXP_DBL *imagBuf = imBuffer;
 
-  (len > 64) ? (len_scale = 6) : (len_scale = 5);
+  const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)len), 1);
   /*
     r00r,
     r11r,r22r
diff --git a/libFDK/src/nlc_dec.cpp b/libFDK/src/nlc_dec.cpp
index 6e98ce0..3733d98 100644
--- a/libFDK/src/nlc_dec.cpp
+++ b/libFDK/src/nlc_dec.cpp
@@ -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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -568,12 +568,12 @@ bail:
 static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
                            SCHAR* out_data_2, DATA_TYPE data_type,
                            DIFF_TYPE diff_type_1, DIFF_TYPE diff_type_2,
-                           int num_val, CODING_SCHEME* cdg_scheme, int ldMode) {
+                           int num_val, PAIRING* pairing_scheme, int ldMode) {
   ERROR_t err = HUFFDEC_OK;
+  CODING_SCHEME coding_scheme = HUFF_1D;
   DIFF_TYPE diff_type;
 
   int i = 0;
-  ULONG data = 0;
 
   SCHAR pair_vec[28][2];
 
@@ -596,15 +596,13 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
   int hufYY;
 
   /* Coding scheme */
-  data = FDKreadBits(strm, 1);
-  *cdg_scheme = (CODING_SCHEME)(data << PAIR_SHIFT);
+  coding_scheme = (CODING_SCHEME)FDKreadBits(strm, 1);
 
-  if (*cdg_scheme >> PAIR_SHIFT == HUFF_2D) {
+  if (coding_scheme == HUFF_2D) {
     if ((out_data_1 != NULL) && (out_data_2 != NULL) && (ldMode == 0)) {
-      data = FDKreadBits(strm, 1);
-      *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | data);
+      *pairing_scheme = (PAIRING)FDKreadBits(strm, 1);
     } else {
-      *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | FREQ_PAIR);
+      *pairing_scheme = FREQ_PAIR;
     }
   }
 
@@ -613,7 +611,7 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
     hufYY2 = diff_type_2;
   }
 
-  switch (*cdg_scheme >> PAIR_SHIFT) {
+  switch (coding_scheme) {
     case HUFF_1D:
       p0_flag[0] = (diff_type_1 == DIFF_FREQ);
       p0_flag[1] = (diff_type_2 == DIFF_FREQ);
@@ -634,7 +632,7 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
 
     case HUFF_2D:
 
-      switch (*cdg_scheme & PAIR_MASK) {
+      switch (*pairing_scheme) {
         case FREQ_PAIR:
 
           if (out_data_1 != NULL) {
@@ -843,7 +841,7 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
   SCHAR* pDataVec[2] = {NULL, NULL};
 
   DIFF_TYPE diff_type[2] = {DIFF_FREQ, DIFF_FREQ};
-  CODING_SCHEME cdg_scheme = HUFF_1D;
+  PAIRING pairing = FREQ_PAIR;
   DIRECTION direction = BACKWARDS;
 
   switch (data_type) {
@@ -959,7 +957,7 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
     }
     /* Huffman decoding */
     err = huff_decode(strm, pDataVec[0], pDataVec[1], data_type, diff_type[0],
-                      diff_type[1], dataBands, &cdg_scheme,
+                      diff_type[1], dataBands, &pairing,
                       (DECODER == SAOC_DECODER));
     if (err != HUFFDEC_OK) {
       return HUFFDEC_NOTOK;
@@ -986,8 +984,8 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
         }
       }
 
-      mixed_time_pair = (diff_type[0] != diff_type[1]) &&
-                        ((cdg_scheme & PAIR_MASK) == TIME_PAIR);
+      mixed_time_pair =
+          (diff_type[0] != diff_type[1]) && (pairing == TIME_PAIR);
 
       if (direction == BACKWARDS) {
         if (diff_type[0] == DIFF_FREQ) {
diff --git a/libMpegTPDec/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp
index e46cb32..8f77017 100644
--- a/libMpegTPDec/src/tpdec_asc.cpp
+++ b/libMpegTPDec/src/tpdec_asc.cpp
@@ -1694,8 +1694,7 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
                                            const AUDIO_OBJECT_TYPE aot) {
   TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
 
-  USAC_EXT_ELEMENT_TYPE usacExtElementType =
-      (USAC_EXT_ELEMENT_TYPE)escapedValue(hBs, 4, 8, 16);
+  UINT usacExtElementType = escapedValue(hBs, 4, 8, 16);
 
   /* recurve extension elements which are invalid for USAC */
   if (aot == AOT_USAC) {
@@ -1712,7 +1711,6 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
     }
   }
 
-  extElement->usacExtElementType = usacExtElementType;
   int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16);
   extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength;
   INT bsAnchor;
@@ -1746,8 +1744,10 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
       }
     } break;
     default:
+      usacExtElementType = ID_EXT_ELE_UNKNOWN;
       break;
   }
+  extElement->usacExtElementType = (USAC_EXT_ELEMENT_TYPE)usacExtElementType;
 
   /* Adjust bit stream position. This is required because of byte alignment and
    * unhandled extensions. */
@@ -1776,7 +1776,7 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc,
   TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
 
   int numConfigExtensions;
-  CONFIG_EXT_ID usacConfigExtType;
+  UINT usacConfigExtType;
   int usacConfigExtLength;
   int loudnessInfoSetIndex =
       -1; /* index of loudnessInfoSet config extension. -1 if not contained. */
@@ -1787,7 +1787,7 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc,
   for (int confExtIdx = 0; confExtIdx < numConfigExtensions; confExtIdx++) {
     INT nbits;
     int loudnessInfoSetConfigExtensionPosition = FDKgetValidBits(hBs);
-    usacConfigExtType = (CONFIG_EXT_ID)escapedValue(hBs, 4, 8, 16);
+    usacConfigExtType = escapedValue(hBs, 4, 8, 16);
     usacConfigExtLength = (int)escapedValue(hBs, 4, 8, 16);
 
     /* Start bit position of config extension */
diff --git a/libMpegTPDec/src/tpdec_latm.cpp b/libMpegTPDec/src/tpdec_latm.cpp
index 3b71db8..c32be54 100644
--- a/libMpegTPDec/src/tpdec_latm.cpp
+++ b/libMpegTPDec/src/tpdec_latm.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -591,6 +591,18 @@ bail:
   return (ErrorStatus);
 }
 
+static int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) {
+  int len = 0, tmp = 255;
+  int validBytes = (int)FDKgetValidBits(bs) >> 3;
+
+  while (tmp == 255 && validBytes-- > 0) {
+    tmp = (int)FDKreadBits(bs, 8);
+    len += tmp;
+  }
+
+  return ((tmp == 255) ? -1 : (len << 3));
+}
+
 TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs,
                                                     CLatmDemux *pLatmDemux) {
   TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
@@ -602,11 +614,17 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs,
       FDK_ASSERT(pLatmDemux->m_numLayer[prog] <= LATM_MAX_LAYER);
       for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) {
         LATM_LAYER_INFO *p_linfo = &pLatmDemux->m_linfo[prog][lay];
+        int auChunkLengthInfo = 0;
 
         switch (p_linfo->m_frameLengthType) {
           case 0:
-            p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs);
-            totalPayloadBits += p_linfo->m_frameLengthInBits;
+            auChunkLengthInfo = CLatmDemux_ReadAuChunkLengthInfo(bs);
+            if (auChunkLengthInfo >= 0) {
+              p_linfo->m_frameLengthInBits = (UINT)auChunkLengthInfo;
+              totalPayloadBits += p_linfo->m_frameLengthInBits;
+            } else {
+              return TRANSPORTDEC_PARSE_ERROR;
+            }
             break;
           case 3:
           case 5:
@@ -627,23 +645,6 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs,
   return (ErrorStatus);
 }
 
-int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) {
-  UCHAR endFlag;
-  int len = 0;
-
-  do {
-    UCHAR tmp = (UCHAR)FDKreadBits(bs, 8);
-    endFlag = (tmp < 255);
-
-    len += tmp;
-
-  } while (endFlag == 0);
-
-  len <<= 3; /* convert from bytes to bits */
-
-  return len;
-}
-
 UINT CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux, const UINT prog,
                                      const UINT layer) {
   UINT nFrameLenBits = 0;
diff --git a/libMpegTPDec/src/tpdec_latm.h b/libMpegTPDec/src/tpdec_latm.h
index 6af553d..8b8c971 100644
--- a/libMpegTPDec/src/tpdec_latm.h
+++ b/libMpegTPDec/src/tpdec_latm.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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -151,8 +151,6 @@ typedef struct {
                                   AudioPreRoll */
 } CLatmDemux;
 
-int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs);
-
 TRANSPORTDEC_ERROR CLatmDemux_Read(HANDLE_FDK_BITSTREAM bs,
                                    CLatmDemux *pLatmDemux, TRANSPORT_TYPE tt,
                                    CSTpCallBacks *pTpDecCallbacks,
diff --git a/libPCMutils/src/pcmdmx_lib.cpp b/libPCMutils/src/pcmdmx_lib.cpp
index 2070dbc..fca12ce 100644
--- a/libPCMutils/src/pcmdmx_lib.cpp
+++ b/libPCMutils/src/pcmdmx_lib.cpp
@@ -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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -494,13 +494,40 @@ static PCM_DMX_CHANNEL_MODE getChMode4Plain(
   return plainChMode;
 }
 
-static inline UINT getIdxSum(UCHAR numCh) {
-  UINT result = 0;
-  int i;
-  for (i = 1; i < numCh; i += 1) {
-    result += i;
+/** Validates the channel indices of all channels present in the bitstream.
+ * The channel indices have to be consecutive and unique for each audio channel
+ *type.
+ * @param [in] The total number of channels of the given configuration.
+ * @param [in] The total number of channels of the current audio channel type of
+ *the given configuration.
+ * @param [in] Audio channel type to be examined.
+ * @param [in] Array holding the corresponding channel types for each channel.
+ * @param [in] Array holding the corresponding channel type indices for each
+ *channel.
+ * @returns Returns 1 on success, returns 0 on error.
+ **/
+static UINT validateIndices(UINT numChannels, UINT numChannelsPlaneAndGrp,
+                            AUDIO_CHANNEL_TYPE aChType,
+                            const AUDIO_CHANNEL_TYPE channelType[],
+                            const UCHAR channelIndices[]) {
+  for (UINT reqValue = 0; reqValue < numChannelsPlaneAndGrp; reqValue++) {
+    int found = FALSE;
+    for (UINT i = 0; i < numChannels; i++) {
+      if (channelType[i] == aChType) {
+        if (channelIndices[i] == reqValue) {
+          if (found == TRUE) {
+            return 0; /* Found channel index a second time */
+          } else {
+            found = TRUE; /* Found channel index */
+          }
+        }
+      }
+    }
+    if (found == FALSE) {
+      return 0; /* Did not find channel index */
+    }
   }
-  return result;
+  return 1; /* Successfully validated channel indices */
 }
 
 /** Evaluate a given channel configuration and extract a packed channel mode. In
@@ -523,7 +550,6 @@ static PCMDMX_ERROR getChannelMode(
     UCHAR offsetTable[(8)],                 /* out */
     PCM_DMX_CHANNEL_MODE *chMode            /* out */
 ) {
-  UINT idxSum[(3)][(4)];
   UCHAR numCh[(3)][(4)];
   UCHAR mapped[(8)];
   PCM_DMX_SPEAKER_POSITION spkrPos[(8)];
@@ -538,7 +564,6 @@ static PCMDMX_ERROR getChannelMode(
   FDK_ASSERT(chMode != NULL);
 
   /* For details see ISO/IEC 13818-7:2005(E), 8.5.3 Channel configuration */
-  FDKmemclear(idxSum, (3) * (4) * sizeof(UINT));
   FDKmemclear(numCh, (3) * (4) * sizeof(UCHAR));
   FDKmemclear(mapped, (8) * sizeof(UCHAR));
   FDKmemclear(spkrPos, (8) * sizeof(PCM_DMX_SPEAKER_POSITION));
@@ -552,19 +577,22 @@ static PCMDMX_ERROR getChannelMode(
         (channelType[ch] & 0x0F) - 1,
         0); /* Assign all undefined channels (ACT_NONE) to front channels. */
     numCh[channelType[ch] >> 4][chGrp] += 1;
-    idxSum[channelType[ch] >> 4][chGrp] += channelIndices[ch];
   }
-  if (numChannels > TWO_CHANNEL) {
+
+  {
     int chGrp;
     /* Sanity check on the indices */
     for (chGrp = 0; chGrp < (4); chGrp += 1) {
       int plane;
       for (plane = 0; plane < (3); plane += 1) {
-        if (idxSum[plane][chGrp] != getIdxSum(numCh[plane][chGrp])) {
+        if (numCh[plane][chGrp] == 0) continue;
+        AUDIO_CHANNEL_TYPE aChType =
+            (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF));
+        if (!validateIndices(numChannels, numCh[plane][chGrp], aChType,
+                             channelType, channelIndices)) {
           unsigned idxCnt = 0;
           for (ch = 0; ch < numChannels; ch += 1) {
-            if (channelType[ch] ==
-                (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF))) {
+            if (channelType[ch] == aChType) {
               channelIndices[ch] = idxCnt++;
             }
           }
diff --git a/libSACdec/src/sac_bitdec.cpp b/libSACdec/src/sac_bitdec.cpp
index 4485ccf..25b3d9e 100644
--- a/libSACdec/src/sac_bitdec.cpp
+++ b/libSACdec/src/sac_bitdec.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -488,12 +488,17 @@ SACDEC_ERROR SpatialDecParseSpecificConfig(
   pSpatialSpecificConfig->freqRes =
       (SPATIALDEC_FREQ_RES)freqResTable_LD[bsFreqRes];
 
-  pSpatialSpecificConfig->treeConfig =
-      (SPATIALDEC_TREE_CONFIG)FDKreadBits(bitstream, 4);
+  {
+    UINT treeConfig = FDKreadBits(bitstream, 4);
 
-  if (pSpatialSpecificConfig->treeConfig != SPATIALDEC_MODE_RSVD7) {
-    err = MPS_UNSUPPORTED_CONFIG;
-    goto bail;
+    switch (treeConfig) {
+      case SPATIALDEC_MODE_RSVD7:
+        pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG)treeConfig;
+        break;
+      default:
+        err = MPS_UNSUPPORTED_CONFIG;
+        goto bail;
+    }
   }
 
   {
diff --git a/libSACdec/src/sac_process.cpp b/libSACdec/src/sac_process.cpp
index 22091a9..33a1647 100644
--- a/libSACdec/src/sac_process.cpp
+++ b/libSACdec/src/sac_process.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -517,12 +517,11 @@ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding(
       maxVal = fAbs(iReal0) | fAbs(iImag0);
       maxVal |= fAbs(iReal1);
 
-      s = fMax(CntLeadingZeros(maxVal) - 1, 0);
-      s = fMin(s, scale_param_m2);
+      s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2);
 
-      mReal0 = iReal0 << s;
-      mImag0 = iImag0 << s;
-      mReal1 = iReal1 << s;
+      mReal0 = scaleValue(iReal0, s);
+      mImag0 = scaleValue(iImag0, s);
+      mReal1 = scaleValue(iReal1, s);
 
       s = scale_param_m2 - s;
 
@@ -562,12 +561,11 @@ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding(
       maxVal = fAbs(iReal0) | fAbs(iImag0);
       maxVal |= fAbs(iReal1);
 
-      s = fMax(CntLeadingZeros(maxVal) - 1, 0);
-      s = fMin(s, scale_param_m2);
+      s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2);
 
-      mReal0 = FX_DBL2FX_SGL(iReal0 << s);
-      mImag0 = FX_DBL2FX_SGL(iImag0 << s);
-      mReal1 = FX_DBL2FX_SGL(iReal1 << s);
+      mReal0 = FX_DBL2FX_SGL(scaleValue(iReal0, s));
+      mImag0 = FX_DBL2FX_SGL(scaleValue(iImag0, s));
+      mReal1 = FX_DBL2FX_SGL(scaleValue(iReal1, s));
 
       s = scale_param_m2 - s;
 
diff --git a/libSACdec/src/sac_reshapeBBEnv.cpp b/libSACdec/src/sac_reshapeBBEnv.cpp
index 272d009..72f4e58 100644
--- a/libSACdec/src/sac_reshapeBBEnv.cpp
+++ b/libSACdec/src/sac_reshapeBBEnv.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -241,29 +241,56 @@ static inline void combineDryWet(FIXP_DBL *RESTRICT pReal,
   }
 }
 
-static inline void slotAmp(FIXP_DBL *RESTRICT slotAmp_dry,
-                           FIXP_DBL *RESTRICT slotAmp_wet,
-                           FIXP_DBL *RESTRICT pHybOutputRealDry,
-                           FIXP_DBL *RESTRICT pHybOutputImagDry,
-                           FIXP_DBL *RESTRICT pHybOutputRealWet,
-                           FIXP_DBL *RESTRICT pHybOutputImagWet, INT cplxBands,
-                           INT hybBands) {
-  INT qs;
+static inline void slotAmp(
+    FIXP_DBL *RESTRICT slotAmp_dry, INT *RESTRICT slotAmp_dry_e,
+    FIXP_DBL *RESTRICT slotAmp_wet, INT *RESTRICT slotAmp_wet_e,
+    FIXP_DBL *RESTRICT pHybOutputRealDry, FIXP_DBL *RESTRICT pHybOutputImagDry,
+    FIXP_DBL *RESTRICT pHybOutputRealWet, FIXP_DBL *RESTRICT pHybOutputImagWet,
+    INT cplxBands, INT hybBands) {
+  INT qs, s1, s2, headroom_dry, headroom_wet;
   FIXP_DBL dry, wet;
 
+  /* headroom can be reduced by 1 bit due to use of fPow2Div2 */
+  s1 = DFRACT_BITS - 1 - CntLeadingZeros(hybBands + cplxBands);
+  headroom_dry = fMin(getScalefactor(pHybOutputRealDry, hybBands),
+                      getScalefactor(pHybOutputImagDry, cplxBands));
+  headroom_wet = fMin(getScalefactor(pHybOutputRealWet, hybBands),
+                      getScalefactor(pHybOutputImagWet, cplxBands));
+
   dry = wet = FL2FXCONST_DBL(0.0f);
   for (qs = 0; qs < cplxBands; qs++) {
-    dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)) +
-                                fPow2Div2(pHybOutputImagDry[qs] << (1)));
-    wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)) +
-                                fPow2Div2(pHybOutputImagWet[qs] << (1)));
+    /* sum up dry part */
+    dry += (fPow2Div2(pHybOutputRealDry[qs] << headroom_dry) >> s1);
+    dry += (fPow2Div2(pHybOutputImagDry[qs] << headroom_dry) >> s1);
+    /* sum up wet part */
+    wet += (fPow2Div2(pHybOutputRealWet[qs] << headroom_wet) >> s1);
+    wet += (fPow2Div2(pHybOutputImagWet[qs] << headroom_wet) >> s1);
   }
   for (; qs < hybBands; qs++) {
-    dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)));
-    wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)));
+    dry += (fPow2Div2(pHybOutputRealDry[qs] << headroom_dry) >> s1);
+    wet += (fPow2Div2(pHybOutputRealWet[qs] << headroom_wet) >> s1);
+  }
+
+  /* consider fPow2Div2() */
+  s1 += 1;
+
+  /* normalize dry part, ensure that exponent is even */
+  s2 = fixMax(0, CntLeadingZeros(dry) - 1);
+  *slotAmp_dry = dry << s2;
+  *slotAmp_dry_e = s1 - s2 - 2 * headroom_dry;
+  if (*slotAmp_dry_e & 1) {
+    *slotAmp_dry = *slotAmp_dry >> 1;
+    *slotAmp_dry_e += 1;
+  }
+
+  /* normalize wet part, ensure that exponent is even */
+  s2 = fixMax(0, CntLeadingZeros(wet) - 1);
+  *slotAmp_wet = wet << s2;
+  *slotAmp_wet_e = s1 - s2 - 2 * headroom_wet;
+  if (*slotAmp_wet_e & 1) {
+    *slotAmp_wet = *slotAmp_wet >> 1;
+    *slotAmp_wet_e += 1;
   }
-  *slotAmp_dry = dry >> (2 * (1));
-  *slotAmp_wet = wet >> (2 * (1));
 }
 
 #if defined(__aarch64__)
@@ -533,6 +560,7 @@ void SpatialDecReshapeBBEnv(spatialDec *self, const SPATIAL_BS_FRAME *frame,
                             INT ts) {
   INT ch, scale;
   INT dryFacSF, slotAmpSF;
+  INT slotAmp_dry_e, slotAmp_wet_e;
   FIXP_DBL tmp, dryFac, envShape;
   FIXP_DBL slotAmp_dry, slotAmp_wet, slotAmp_ratio;
   FIXP_DBL envDry[MAX_OUTPUT_CHANNELS], envDmx[2];
@@ -594,22 +622,25 @@ void SpatialDecReshapeBBEnv(spatialDec *self, const SPATIAL_BS_FRAME *frame,
         dryFacSF = SF_SHAPE + 2 * dryFacSF;
       }
 
+      slotAmp_dry_e = slotAmp_wet_e = 0;
+
       /* calculate slotAmp_dry and slotAmp_wet */
-      slotAmp(&slotAmp_dry, &slotAmp_wet, &self->hybOutputRealDry__FDK[ch][6],
+      slotAmp(&slotAmp_dry, &slotAmp_dry_e, &slotAmp_wet, &slotAmp_wet_e,
+              &self->hybOutputRealDry__FDK[ch][6],
               &self->hybOutputImagDry__FDK[ch][6],
               &self->hybOutputRealWet__FDK[ch][6],
               &self->hybOutputImagWet__FDK[ch][6], cplxBands, hybBands);
 
+      /* exponents must be even due to subsequent square root calculation */
+      FDK_ASSERT(((slotAmp_dry_e & 1) == 0) && ((slotAmp_wet_e & 1) == 0));
+
       /* slotAmp_ratio will be scaled by slotAmpSF bits */
       if (slotAmp_dry != FL2FXCONST_DBL(0.0f)) {
-        sc = fixMax(0, CntLeadingZeros(slotAmp_wet) - 1);
-        sc = sc - (sc & 1);
-
-        slotAmp_wet = sqrtFixp(slotAmp_wet << sc);
+        slotAmp_wet = sqrtFixp(slotAmp_wet);
         slotAmp_dry = invSqrtNorm2(slotAmp_dry, &slotAmpSF);
 
         slotAmp_ratio = fMult(slotAmp_wet, slotAmp_dry);
-        slotAmpSF = slotAmpSF - (sc >> 1);
+        slotAmpSF = slotAmpSF + (slotAmp_wet_e >> 1) - (slotAmp_dry_e >> 1);
       }
 
       /* calculate common scale factor */
diff --git a/libSACdec/src/sac_stp.cpp b/libSACdec/src/sac_stp.cpp
index b328c82..0e6affa 100644
--- a/libSACdec/src/sac_stp.cpp
+++ b/libSACdec/src/sac_stp.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -229,15 +229,13 @@ inline void combineSignalCplxScale1(FIXP_DBL *hybOutputRealDry,
   int n;
   FIXP_DBL scaleY;
   for (n = bands - 1; n >= 0; n--) {
-    scaleY = fMultDiv2(scaleX, *pBP);
+    scaleY = fMult(scaleX, *pBP);
     *hybOutputRealDry = SATURATE_LEFT_SHIFT(
-        (*hybOutputRealDry >> 1) +
-            (fMultDiv2(*hybOutputRealWet, scaleY) << (SF_SCALE + 1)),
-        1, DFRACT_BITS);
+        (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleY),
+        SF_SCALE, DFRACT_BITS);
     *hybOutputImagDry = SATURATE_LEFT_SHIFT(
-        (*hybOutputImagDry >> 1) +
-            (fMultDiv2(*hybOutputImagWet, scaleY) << (SF_SCALE + 1)),
-        1, DFRACT_BITS);
+        (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleY),
+        SF_SCALE, DFRACT_BITS);
     hybOutputRealDry++, hybOutputRealWet++;
     hybOutputImagDry++, hybOutputImagWet++;
     pBP++;
@@ -252,12 +250,12 @@ inline void combineSignalCplxScale2(FIXP_DBL *hybOutputRealDry,
   int n;
 
   for (n = bands - 1; n >= 0; n--) {
-    *hybOutputRealDry =
-        *hybOutputRealDry +
-        (fMultDiv2(*hybOutputRealWet, scaleX) << (SF_SCALE + 1));
-    *hybOutputImagDry =
-        *hybOutputImagDry +
-        (fMultDiv2(*hybOutputImagWet, scaleX) << (SF_SCALE + 1));
+    *hybOutputRealDry = SATURATE_LEFT_SHIFT(
+        (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleX),
+        SF_SCALE, DFRACT_BITS);
+    *hybOutputImagDry = SATURATE_LEFT_SHIFT(
+        (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleX),
+        SF_SCALE, DFRACT_BITS);
     hybOutputRealDry++, hybOutputRealWet++;
     hybOutputImagDry++, hybOutputImagWet++;
   }
diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp
deleted file mode 100644
index db1948f..0000000
--- a/libSBRdec/src/arm/lpp_tran_arm.cpp
+++ /dev/null
@@ -1,159 +0,0 @@
-/* -----------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
-Forschung e.V. All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
-that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
-scheme for digital audio. This FDK AAC Codec software is intended to be used on
-a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
-general perceptual audio codecs. AAC-ELD is considered the best-performing
-full-bandwidth communications codec by independent studies and is widely
-deployed. AAC has been standardized by ISO and IEC as part of the MPEG
-specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including
-those of Fraunhofer) may be obtained through Via Licensing
-(www.vialicensing.com) or through the respective patent owners individually for
-the purpose of encoding or decoding bit streams in products that are compliant
-with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
-Android devices already license these patent claims through Via Licensing or
-directly from the patent owners, and therefore FDK AAC Codec software may
-already be covered under those patent licenses when it is used for those
-licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions
-with enhanced sound quality, are also available from Fraunhofer. Users are
-encouraged to check the Fraunhofer website for additional applications
-information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification,
-are permitted without payment of copyright license fees provided that you
-satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of
-the FDK AAC Codec or your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation
-and/or other materials provided with redistributions of the FDK AAC Codec or
-your modifications thereto in binary form. You must make available free of
-charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived
-from this library without prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute
-the FDK AAC Codec software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating
-that you changed the software and the date of any change. For modified versions
-of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
-must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
-AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
-limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
-Fraunhofer provides no warranty of patent non-infringement with respect to this
-software.
-
-You may use this FDK AAC Codec software or modifications thereto only for
-purposes that are authorized by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
-holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
-including but not limited to the implied warranties of merchantability and
-fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
-or consequential damages, including but not limited to procurement of substitute
-goods or services; loss of use, data, or profits, or business interruption,
-however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of
-this software, even if advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------ */
-
-/**************************** SBR decoder library ******************************
-
-   Author(s):   Arthur Tritthart
-
-   Description: (ARM optimised) LPP transposer subroutines
-
-*******************************************************************************/
-
-#if defined(__arm__)
-
-#define FUNCTION_LPPTRANSPOSER_func1
-
-#ifdef FUNCTION_LPPTRANSPOSER_func1
-
-/* Note: This code requires only 43 cycles per iteration instead of 61 on
- * ARM926EJ-S */
-static void lppTransposer_func1(FIXP_DBL *lowBandReal, FIXP_DBL *lowBandImag,
-                                FIXP_DBL **qmfBufferReal,
-                                FIXP_DBL **qmfBufferImag, int loops, int hiBand,
-                                int dynamicScale, int descale, FIXP_SGL a0r,
-                                FIXP_SGL a0i, FIXP_SGL a1r, FIXP_SGL a1i,
-                                const int fPreWhitening,
-                                FIXP_DBL preWhiteningGain,
-                                int preWhiteningGains_sf) {
-  FIXP_DBL real1, real2, imag1, imag2, accu1, accu2;
-
-  real2 = lowBandReal[-2];
-  real1 = lowBandReal[-1];
-  imag2 = lowBandImag[-2];
-  imag1 = lowBandImag[-1];
-  for (int i = 0; i < loops; i++) {
-    accu1 = fMultDiv2(a0r, real1);
-    accu2 = fMultDiv2(a0i, imag1);
-    accu1 = fMultAddDiv2(accu1, a1r, real2);
-    accu2 = fMultAddDiv2(accu2, a1i, imag2);
-    real2 = fMultDiv2(a1i, real2);
-    accu1 = accu1 - accu2;
-    accu1 = accu1 >> dynamicScale;
-
-    accu2 = fMultAddDiv2(real2, a1r, imag2);
-    real2 = real1;
-    imag2 = imag1;
-    accu2 = fMultAddDiv2(accu2, a0i, real1);
-    real1 = lowBandReal[i];
-    accu2 = fMultAddDiv2(accu2, a0r, imag1);
-    imag1 = lowBandImag[i];
-    accu2 = accu2 >> dynamicScale;
-
-    accu1 <<= 1;
-    accu2 <<= 1;
-    accu1 += (real1 >> descale);
-    accu2 += (imag1 >> descale);
-    if (fPreWhitening) {
-      accu1 = scaleValueSaturate(fMultDiv2(accu1, preWhiteningGain),
-                                 preWhiteningGains_sf);
-      accu2 = scaleValueSaturate(fMultDiv2(accu2, preWhiteningGain),
-                                 preWhiteningGains_sf);
-    }
-    qmfBufferReal[i][hiBand] = accu1;
-    qmfBufferImag[i][hiBand] = accu2;
-  }
-}
-#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */
-
-#endif /* __arm__ */
diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
index ad5edfe..cefa612 100644
--- a/libSBRdec/src/env_calc.cpp
+++ b/libSBRdec/src/env_calc.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -664,7 +664,7 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag,
         gain_sf[i] = mult_sf - total_power_low_sf + sf2;
         gain[i] = sqrtFixp_lookup(gain[i], &gain_sf[i]);
         if (gain_sf[i] < 0) {
-          gain[i] >>= -gain_sf[i];
+          gain[i] >>= fMin(DFRACT_BITS - 1, -gain_sf[i]);
           gain_sf[i] = 0;
         }
       } else {
@@ -683,11 +683,6 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag,
 
     /* gain[i] = g_inter[i] */
     for (i = 0; i < nbSubsample; ++i) {
-      if (gain_sf[i] < 0) {
-        gain[i] >>= -gain_sf[i];
-        gain_sf[i] = 0;
-      }
-
       /* calculate: gain[i] = 1.0f + gamma * (gain[i] - 1.0f); */
       FIXP_DBL one = (FIXP_DBL)MAXVAL_DBL >>
                      gain_sf[i]; /* to substract this from gain[i] */
@@ -755,23 +750,15 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag,
     int gain_adj_sf = gain_adj_2_sf;
 
     for (i = 0; i < nbSubsample; ++i) {
-      gain[i] = fMult(gain[i], gain_adj);
-      gain_sf[i] += gain_adj_sf;
-
-      /* limit gain */
-      if (gain_sf[i] > INTER_TES_SF_CHANGE) {
-        gain[i] = (FIXP_DBL)MAXVAL_DBL;
-        gain_sf[i] = INTER_TES_SF_CHANGE;
-      }
-    }
-
-    for (i = 0; i < nbSubsample; ++i) {
-      /* equalize gain[]'s scale factors */
-      gain[i] >>= INTER_TES_SF_CHANGE - gain_sf[i];
+      int gain_e = fMax(
+          fMin(gain_sf[i] + gain_adj_sf - INTER_TES_SF_CHANGE, DFRACT_BITS - 1),
+          -(DFRACT_BITS - 1));
+      FIXP_DBL gain_final = fMult(gain[i], gain_adj);
+      gain_final = scaleValueSaturate(gain_final, gain_e);
 
       for (j = lowSubband; j < highSubband; j++) {
-        qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain[i]);
-        qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain[i]);
+        qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain_final);
+        qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain_final);
       }
     }
   } else { /* gamma_idx == 0 */
diff --git a/libSBRdec/src/hbe.cpp b/libSBRdec/src/hbe.cpp
index d210bb6..f2452ea 100644
--- a/libSBRdec/src/hbe.cpp
+++ b/libSBRdec/src/hbe.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1400,42 +1400,27 @@ void QmfTransposerApply(HANDLE_HBE_TRANSPOSER hQmfTransposer,
 
   if (shift_ov != 0) {
     for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) {
-      for (band = 0; band < QMF_SYNTH_CHANNELS; band++) {
-        if (shift_ov >= 0) {
-          hQmfTransposer->qmfHBEBufReal_F[i][band] <<= shift_ov;
-          hQmfTransposer->qmfHBEBufImag_F[i][band] <<= shift_ov;
-        } else {
-          hQmfTransposer->qmfHBEBufReal_F[i][band] >>= (-shift_ov);
-          hQmfTransposer->qmfHBEBufImag_F[i][band] >>= (-shift_ov);
-        }
-      }
+      scaleValuesSaturate(&hQmfTransposer->qmfHBEBufReal_F[i][0],
+                          QMF_SYNTH_CHANNELS, shift_ov);
+      scaleValuesSaturate(&hQmfTransposer->qmfHBEBufImag_F[i][0],
+                          QMF_SYNTH_CHANNELS, shift_ov);
     }
-  }
 
-  if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) && shift_ov != 0) {
-    for (i = timeStep * firstSlotOffsset; i < ov_len; i++) {
-      for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand;
-           band++) {
-        if (shift_ov >= 0) {
-          ppQmfBufferOutReal_F[i][band] <<= shift_ov;
-          ppQmfBufferOutImag_F[i][band] <<= shift_ov;
-        } else {
-          ppQmfBufferOutReal_F[i][band] >>= (-shift_ov);
-          ppQmfBufferOutImag_F[i][band] >>= (-shift_ov);
-        }
+    if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) {
+      int nBands =
+          fMax(0, hQmfTransposer->stopBand - hQmfTransposer->startBand);
+
+      for (i = timeStep * firstSlotOffsset; i < ov_len; i++) {
+        scaleValuesSaturate(&ppQmfBufferOutReal_F[i][hQmfTransposer->startBand],
+                            nBands, shift_ov);
+        scaleValuesSaturate(&ppQmfBufferOutImag_F[i][hQmfTransposer->startBand],
+                            nBands, shift_ov);
       }
-    }
 
-    /* shift lpc filterstates */
-    for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) {
-      for (band = 0; band < (64); band++) {
-        if (shift_ov >= 0) {
-          lpcFilterStatesReal[i][band] <<= shift_ov;
-          lpcFilterStatesImag[i][band] <<= shift_ov;
-        } else {
-          lpcFilterStatesReal[i][band] >>= (-shift_ov);
-          lpcFilterStatesImag[i][band] >>= (-shift_ov);
-        }
+      /* shift lpc filterstates */
+      for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) {
+        scaleValuesSaturate(&lpcFilterStatesReal[i][0], (64), shift_ov);
+        scaleValuesSaturate(&lpcFilterStatesImag[i][0], (64), shift_ov);
       }
     }
   }
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
index 93e1158..68a25bf 100644
--- a/libSBRdec/src/lpp_tran.cpp
+++ b/libSBRdec/src/lpp_tran.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -132,10 +132,6 @@ amm-info@iis.fraunhofer.de
 
 #include "HFgen_preFlat.h"
 
-#if defined(__arm__)
-#include "arm/lpp_tran_arm.cpp"
-#endif
-
 #define LPC_SCALE_FACTOR 2
 
 /*!
@@ -220,19 +216,21 @@ static inline void calc_qmfBufferReal(FIXP_DBL **qmfBufferReal,
                                       const FIXP_DBL *const lowBandReal,
                                       const int startSample,
                                       const int stopSample, const UCHAR hiBand,
-                                      const int dynamicScale, const int descale,
+                                      const int dynamicScale,
                                       const FIXP_SGL a0r, const FIXP_SGL a1r) {
-  FIXP_DBL accu1, accu2;
-  int i;
+  const int dynscale = fixMax(0, dynamicScale - 1) + 1;
+  const int rescale = -fixMin(0, dynamicScale - 1) + 1;
+  const int descale =
+      fixMin(DFRACT_BITS - 1, LPC_SCALE_FACTOR + dynamicScale + rescale);
+
+  for (int i = 0; i < stopSample - startSample; i++) {
+    FIXP_DBL accu;
 
-  for (i = 0; i < stopSample - startSample; i++) {
-    accu1 = fMultDiv2(a1r, lowBandReal[i]);
-    accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1);
-    accu1 = accu1 >> dynamicScale;
+    accu = fMultDiv2(a1r, lowBandReal[i]) + fMultDiv2(a0r, lowBandReal[i + 1]);
+    accu = (lowBandReal[i + 2] >> descale) + (accu >> dynscale);
 
-    accu1 <<= 1;
-    accu2 = (lowBandReal[i + 2] >> descale);
-    qmfBufferReal[i + startSample][hiBand] = accu1 + accu2;
+    qmfBufferReal[i + startSample][hiBand] =
+        SATURATE_LEFT_SHIFT(accu, rescale, DFRACT_BITS);
   }
 }
 
@@ -529,7 +527,7 @@ void lppTransposer(
         if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
           resetLPCCoeffs = 1;
         } else {
-          alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+          alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
           if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
             alphar[1] = -alphar[1];
           }
@@ -557,7 +555,7 @@ void lppTransposer(
                scale)) {
             resetLPCCoeffs = 1;
           } else {
-            alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+            alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
             if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
               alphai[1] = -alphai[1];
             }
@@ -596,7 +594,7 @@ void lppTransposer(
       } else {
         INT scale;
         FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
-        alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+        alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
 
         if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
           alphar[0] = -alphar[0];
@@ -616,7 +614,7 @@ void lppTransposer(
         } else {
           INT scale;
           FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
-          alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+          alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
           if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
             alphai[0] = -alphai[0];
         }
@@ -659,7 +657,7 @@ void lppTransposer(
           INT scale;
           FIXP_DBL result =
               fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale);
-          k1 = scaleValue(result, scale);
+          k1 = scaleValueSaturate(result, scale);
 
           if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) {
             k1 = -k1;
@@ -771,52 +769,50 @@ void lppTransposer(
       } else { /* bw <= 0 */
 
         if (!useLP) {
-          int descale =
-              fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale));
-#ifdef FUNCTION_LPPTRANSPOSER_func1
-          lppTransposer_func1(
-              lowBandReal + LPC_ORDER + startSample,
-              lowBandImag + LPC_ORDER + startSample,
-              qmfBufferReal + startSample, qmfBufferImag + startSample,
-              stopSample - startSample, (int)hiBand, dynamicScale, descale, a0r,
-              a0i, a1r, a1i, fPreWhitening, preWhiteningGains[loBand],
-              preWhiteningGains_exp[loBand] + 1);
-#else
+          const int dynscale = fixMax(0, dynamicScale - 2) + 1;
+          const int rescale = -fixMin(0, dynamicScale - 2) + 1;
+          const int descale = fixMin(DFRACT_BITS - 1,
+                                     LPC_SCALE_FACTOR + dynamicScale + rescale);
+
           for (i = startSample; i < stopSample; i++) {
             FIXP_DBL accu1, accu2;
 
-            accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) -
-                     fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) +
-                     fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) -
-                     fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >>
-                    dynamicScale;
-            accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) +
-                     fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) +
-                     fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) +
-                     fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >>
-                    dynamicScale;
-
-            accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);
-            accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);
+            accu1 = ((fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) -
+                      fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1])) >>
+                     1) +
+                    ((fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) -
+                      fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >>
+                     1);
+            accu2 = ((fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) +
+                      fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1])) >>
+                     1) +
+                    ((fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) +
+                      fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >>
+                     1);
+
+            accu1 =
+                (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 >> dynscale);
+            accu2 =
+                (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 >> dynscale);
             if (fPreWhitening) {
-              accu1 = scaleValueSaturate(
+              qmfBufferReal[i][hiBand] = scaleValueSaturate(
                   fMultDiv2(accu1, preWhiteningGains[loBand]),
-                  preWhiteningGains_exp[loBand] + 1);
-              accu2 = scaleValueSaturate(
+                  preWhiteningGains_exp[loBand] + 1 + rescale);
+              qmfBufferImag[i][hiBand] = scaleValueSaturate(
                   fMultDiv2(accu2, preWhiteningGains[loBand]),
-                  preWhiteningGains_exp[loBand] + 1);
+                  preWhiteningGains_exp[loBand] + 1 + rescale);
+            } else {
+              qmfBufferReal[i][hiBand] =
+                  SATURATE_LEFT_SHIFT(accu1, rescale, DFRACT_BITS);
+              qmfBufferImag[i][hiBand] =
+                  SATURATE_LEFT_SHIFT(accu2, rescale, DFRACT_BITS);
             }
-            qmfBufferReal[i][hiBand] = accu1;
-            qmfBufferImag[i][hiBand] = accu2;
           }
-#endif
         } else {
           FDK_ASSERT(dynamicScale >= 0);
           calc_qmfBufferReal(
               qmfBufferReal, &(lowBandReal[LPC_ORDER + startSample - 2]),
-              startSample, stopSample, hiBand, dynamicScale,
-              fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r,
-              a1r);
+              startSample, stopSample, hiBand, dynamicScale, a0r, a1r);
         }
       } /* bw <= 0 */
 
@@ -1066,7 +1062,7 @@ void lppTransposerHBE(
         if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
           resetLPCCoeffs = 1;
         } else {
-          alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+          alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
           if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
             alphar[1] = -alphar[1];
           }
@@ -1092,7 +1088,7 @@ void lppTransposerHBE(
             (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) {
           resetLPCCoeffs = 1;
         } else {
-          alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
+          alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
           if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
             alphai[1] = -alphai[1];
           }
@@ -1121,7 +1117,7 @@ void lppTransposerHBE(
       } else {
         INT scale;
         FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
-        alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+        alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
 
         if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
           alphar[0] = -alphar[0];
@@ -1140,7 +1136,7 @@ void lppTransposerHBE(
       } else {
         INT scale;
         FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
-        alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+        alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
         if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) {
           alphai[0] = -alphai[0];
         }
diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp
index b1fb0da..919e9bb 100644
--- a/libSBRdec/src/sbr_dec.cpp
+++ b/libSBRdec/src/sbr_dec.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -713,7 +713,8 @@ void sbr_dec(
 
   } else { /* (flags & SBRDEC_PS_DECODED) */
     INT sdiff;
-    INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
+    INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov,
+        outScalefactor, outScalefactorR, outScalefactorL;
 
     HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->qmfDomainOutCh->fb;
     HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->qmfDomainOutCh->fb;
@@ -744,7 +745,7 @@ void sbr_dec(
       */
       FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <=
                  QMF_MAX_SYNTHESIS_BANDS);
-      qmfChangeOutScalefactor(synQmfRight, -(8));
+      synQmfRight->outScalefactor = synQmf->outScalefactor;
       FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates,
                 9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis *
                     sizeof(FIXP_QSS));
@@ -788,9 +789,11 @@ void sbr_dec(
       FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel,
                 sizeof(SBRDEC_DRC_CHANNEL));
 
-      for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
+      outScalefactor = maxShift - (8);
+      outScalefactorL = outScalefactorR =
+          sbrInDataHeadroom + 1; /* +1: psDiffScale! (MPEG-PS) */
 
-        INT outScalefactorR, outScalefactorL;
+      for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
 
         /* qmf timeslot of right channel */
         FIXP_DBL *rQmfReal = pWorkBuffer;
@@ -815,27 +818,20 @@ void sbr_dec(
                   ? scaleFactorLowBand_ov
                   : scaleFactorLowBand_no_ov,
               scaleFactorHighBand, synQmf->lsb, synQmf->usb);
-
-          outScalefactorL = outScalefactorR =
-              1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */
         }
 
         sbrDecoder_drcApplySlot(/* right channel */
                                 &hSbrDecRight->sbrDrcChannel, rQmfReal,
                                 rQmfImag, i, synQmfRight->no_col, maxShift);
 
-        outScalefactorR += maxShift;
-
         sbrDecoder_drcApplySlot(/* left channel */
                                 &hSbrDec->sbrDrcChannel, *(pLowBandReal + i),
                                 *(pLowBandImag + i), i, synQmf->no_col,
                                 maxShift);
 
-        outScalefactorL += maxShift;
-
         if (!(flags & SBRDEC_SKIP_QMF_SYN)) {
-          qmfChangeOutScalefactor(synQmf, -(8));
-          qmfChangeOutScalefactor(synQmfRight, -(8));
+          qmfChangeOutScalefactor(synQmf, outScalefactor);
+          qmfChangeOutScalefactor(synQmfRight, outScalefactor);
 
           qmfSynthesisFilteringSlot(
               synQmfRight, rQmfReal, /* QMF real buffer */
diff --git a/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp
index e187656..daa3554 100644
--- a/libSBRdec/src/sbrdec_freq_sca.cpp
+++ b/libSBRdec/src/sbrdec_freq_sca.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -765,9 +765,6 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) {
   sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1],
                     nBandsHi);
 
-  hFreq->nSfb[0] = nBandsLo;
-  hFreq->nSfb[1] = nBandsHi;
-
   /* Check index to freqBandTable[0] */
   if (!(nBandsLo > 0) ||
       (nBandsLo > (((hHeaderData->numberOfAnalysisBands == 16)
@@ -777,6 +774,9 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) {
     return SBRDEC_UNSUPPORTED_CONFIG;
   }
 
+  hFreq->nSfb[0] = nBandsLo;
+  hFreq->nSfb[1] = nBandsHi;
+
   lsb = hFreq->freqBandTable[0][0];
   usb = hFreq->freqBandTable[0][nBandsLo];
 
@@ -814,15 +814,15 @@ resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) {
 
     if (intTemp == 0) intTemp = 1;
 
+    if (intTemp > MAX_NOISE_COEFFS) {
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+
     hFreq->nNfb = intTemp;
   }
 
   hFreq->nInvfBands = hFreq->nNfb;
 
-  if (hFreq->nNfb > MAX_NOISE_COEFFS) {
-    return SBRDEC_UNSUPPORTED_CONFIG;
-  }
-
   /* Get noise bands */
   sbrdecDownSampleLoRes(hFreq->freqBandTableNoise, hFreq->nNfb,
                         hFreq->freqBandTable[0], nBandsLo);
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index b101a4a..7718695 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -961,8 +961,10 @@ SBR_ERROR sbrDecoder_SetParam(HANDLE_SBRDECODER self, const SBRDEC_PARAM param,
 
           /* Set sync state UPSAMPLING for the corresponding slot.
              This switches off bitstream parsing until a new header arrives. */
-          hSbrHeader->syncState = UPSAMPLING;
-          hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+          if (hSbrHeader->syncState != SBR_NOT_INITIALIZED) {
+            hSbrHeader->syncState = UPSAMPLING;
+            hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+          }
         }
       }
     } break;
@@ -1371,7 +1373,9 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
       }
       if (headerStatus == HEADER_ERROR) {
         /* Corrupt SBR info data, do not decode and switch to UPSAMPLING */
-        hSbrHeader->syncState = UPSAMPLING;
+        hSbrHeader->syncState = hSbrHeader->syncState > UPSAMPLING
+                                    ? UPSAMPLING
+                                    : hSbrHeader->syncState;
         fDoDecodeSbrData = 0;
         sbrHeaderPresent = 0;
       }
@@ -1610,7 +1614,9 @@ static SBR_ERROR sbrDecoder_DecodeElement(
       /* No valid SBR payload available, hence switch to upsampling (in all
        * headers) */
       for (hdrIdx = 0; hdrIdx < ((1) + 1); hdrIdx += 1) {
-        self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
+        if (self->sbrHeader[elementIndex][hdrIdx].syncState > UPSAMPLING) {
+          self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
+        }
       }
     } else {
       /* Move frame pointer to the next slot which is up to be decoded/applied