Snap for 8426163 from 76104a5431c7753df108e410c805aefefac16cf1 to mainline-tzdata2-release
Change-Id: I4cfc66299eafb434f52b1ae513a51f126b7714de
diff --git a/Android.bp b/Android.bp
index 0c67186..3f42c19 100644
--- a/Android.bp
+++ b/Android.bp
@@ -1,40 +1,6 @@
-// *** 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",
@@ -60,12 +26,13 @@
"-DSUPPRESS_BUILD_DATE_INFO",
],
sanitize: {
- misc_undefined: [
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- "bounds",
+ misc_undefined:[
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ "bounds",
],
- cfi: true,
+ // Enable CFI if this becomes a shared library.
+ // cfi: true,
},
shared_libs: [
"liblog",
@@ -86,12 +53,6 @@
"libSACenc/include",
],
- target: {
- darwin: {
- enabled: false,
- },
- },
-
apex_available: [
"//apex_available:platform",
"com.android.bluetooth.updatable",
diff --git a/METADATA b/METADATA
deleted file mode 100644
index 5c12860..0000000
--- a/METADATA
+++ /dev/null
@@ -1,3 +0,0 @@
-third_party {
- license_type: BY_EXCEPTION_ONLY
-}
diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf
index 3d4699e..cc7cf41 100644
--- a/documentation/aacDecoder.pdf
+++ b/documentation/aacDecoder.pdf
Binary files differ
diff --git a/documentation/aacEncoder.pdf b/documentation/aacEncoder.pdf
index a47708a..77b8f4c 100644
--- a/documentation/aacEncoder.pdf
+++ b/documentation/aacEncoder.pdf
Binary files differ
diff --git a/fuzzer/Android.bp b/fuzzer/Android.bp
deleted file mode 100644
index 6739798..0000000
--- a/fuzzer/Android.bp
+++ /dev/null
@@ -1,82 +0,0 @@
-/******************************************************************************
- *
- * 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
deleted file mode 100644
index b8cc260..0000000
--- a/fuzzer/README.md
+++ /dev/null
@@ -1,150 +0,0 @@
-# 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
deleted file mode 100644
index c970197..0000000
--- a/fuzzer/aac_dec_fuzzer.cpp
+++ /dev/null
@@ -1,141 +0,0 @@
-/******************************************************************************
- *
- * 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
deleted file mode 100644
index 5a35d70..0000000
--- a/fuzzer/aac_enc_fuzzer.cpp
+++ /dev/null
@@ -1,479 +0,0 @@
-/******************************************************************************
- *
- * 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 d7928c0..56f4ec1 100644
--- a/libAACdec/include/aacdecoder_lib.h
+++ b/libAACdec/include/aacdecoder_lib.h
@@ -1032,7 +1032,7 @@
* \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 in PCM samples.
+ * \param timeDataSize Size of external output buffer.
* \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 d5f0cea..c18e5e9 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -494,75 +494,6 @@
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
@@ -1486,7 +1417,11 @@
const CSAudioSpecificConfig *asc,
UINT flags, UINT *elFlags, int streamIndex,
int elementOffset) {
- FDKmemcpy(self->elFlags, elFlags, sizeof(self->elFlags));
+ {
+ FDKmemcpy(
+ self->elFlags, elFlags,
+ sizeof(*elFlags) * (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1));
+ }
self->flags[streamIndex] = flags;
}
@@ -1589,14 +1524,8 @@
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;
@@ -1780,7 +1709,7 @@
asc->m_sc.m_usacConfig.m_usacNumElements;
}
- mpsEnableCurr = 0;
+ self->mpsEnableCurr = 0;
for (int _el = 0;
_el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements;
_el++) {
@@ -1800,7 +1729,7 @@
self->usacStereoConfigIndex[el] =
asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex;
if (self->elements[el] == ID_USAC_CPE) {
- mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0;
+ self->mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0;
}
}
@@ -1936,7 +1865,7 @@
self->useLdQmfTimeAlign =
asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
}
- if (sbrEnabled != asc->m_sbrPresentFlag) {
+ if (self->sbrEnabled != asc->m_sbrPresentFlag) {
ascChanged = 1;
}
}
@@ -1952,13 +1881,13 @@
flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0;
flags |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0;
if (asc->m_sbrPresentFlag) {
- sbrEnabled = 1;
- sbrEnabledPrev = 1;
+ self->sbrEnabled = 1;
+ self->sbrEnabledPrev = 1;
} else {
- sbrEnabled = 0;
- sbrEnabledPrev = 0;
+ self->sbrEnabled = 0;
+ self->sbrEnabledPrev = 0;
}
- if (sbrEnabled && asc->m_extensionSamplingFrequency) {
+ if (self->sbrEnabled && asc->m_extensionSamplingFrequency) {
if (downscaleFactor != 1 && (downscaleFactor)&1) {
return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* SBR needs an even downscale
factor */
@@ -1985,7 +1914,7 @@
flags |= (asc->m_hcrFlag) ? AC_ER_HCR : 0;
if (asc->m_aot == AOT_ER_AAC_ELD) {
- mpsEnableCurr = 0;
+ self->mpsEnableCurr = 0;
flags |= AC_ELD;
flags |= (asc->m_sbrPresentFlag)
? AC_SBR_PRESENT
@@ -1996,7 +1925,7 @@
? AC_MPS_PRESENT
: 0;
if (self->mpsApplicable) {
- mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
+ self->mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
}
}
flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0;
@@ -2077,7 +2006,7 @@
/* set AC_USAC_SCFGI3 globally if any usac element uses */
switch (asc->m_aot) {
case AOT_USAC:
- if (sbrEnabled) {
+ if (self->sbrEnabled) {
for (int _el = 0;
_el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements;
_el++) {
@@ -2114,7 +2043,7 @@
*/
switch (asc->m_aot) {
case AOT_USAC:
- if (sbrEnabled) {
+ if (self->sbrEnabled) {
const UCHAR map_sbrRatio_2_nAnaBands[] = {16, 24, 32};
FDK_ASSERT(asc->m_sc.m_usacConfig.m_sbrRatioIndex > 0);
@@ -2142,11 +2071,11 @@
}
break;
case AOT_ER_AAC_ELD:
- if (mpsEnableCurr &&
+ if (self->mpsEnableCurr &&
asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) {
- SAC_INPUT_CONFIG sac_interface = (sbrEnabled && self->hSbrDecoder)
- ? SAC_INTERFACE_QMF
- : SAC_INTERFACE_TIME;
+ SAC_INPUT_CONFIG sac_interface =
+ (self->sbrEnabled && self->hSbrDecoder) ? SAC_INTERFACE_QMF
+ : SAC_INTERFACE_TIME;
mpegSurroundDecoder_ConfigureQmfDomain(
(CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface,
(UINT)self->streamInfo.aacSampleRate, asc->m_aot);
@@ -2501,9 +2430,6 @@
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];
@@ -3042,24 +2968,6 @@
} /* 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 0c83191..9d36d10 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1626,11 +1626,6 @@
/* 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 0b80364..b7a9be1 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -628,7 +628,7 @@
SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd;
SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc;
- UCHAR escEscCnt = pRvlc->numDecodedEscapeWordsEsc;
+ UCHAR *pEscEscCnt = &(pRvlc->numDecodedEscapeWordsEsc);
UCHAR *pEscBwdCnt = &(pRvlc->numDecodedEscapeWordsBwd);
pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_bwd);
@@ -636,7 +636,7 @@
*pEscBwdCnt = 0;
pRvlc->direction = BWD;
- pScfEsc += escEscCnt - 1; /* set pScfEsc to last entry */
+ pScfEsc += *pEscEscCnt - 1; /* set pScfEsc to last entry */
pRvlc->firstScf = 0;
pRvlc->firstNrg = 0;
pRvlc->firstIs = 0;
@@ -651,7 +651,7 @@
}
dpcm -= TABLE_OFFSET;
if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
- if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
+ if (pRvlc->length_of_rvlc_escapes) {
pRvlc->conceal_min = bnds;
return;
} else {
@@ -694,7 +694,7 @@
}
dpcm -= TABLE_OFFSET;
if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
- if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
+ if (pRvlc->length_of_rvlc_escapes) {
pScfBwd[bnds] = position;
pRvlc->conceal_min = fMax(0, bnds - offset);
return;
@@ -731,8 +731,7 @@
}
dpcm -= TABLE_OFFSET;
if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
- if ((pRvlc->length_of_rvlc_escapes) ||
- (*pEscBwdCnt >= escEscCnt)) {
+ if (pRvlc->length_of_rvlc_escapes) {
pScfBwd[bnds] = noisenrg;
pRvlc->conceal_min = fMax(0, bnds - offset);
return;
@@ -763,7 +762,7 @@
}
dpcm -= TABLE_OFFSET;
if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
- if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
+ if (pRvlc->length_of_rvlc_escapes) {
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 ca1a6a2..a8dadc0 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -719,7 +719,7 @@
UCHAR *pold_T0_frac = &acelp_mem->old_T0_frac;
if ((int)*pold_T0 >= PIT_MAX) {
- *pold_T0 = (USHORT)(PIT_MAX - 5);
+ *pold_T0 = (UCHAR)(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 f0f23b4..71f7556 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1643,7 +1643,7 @@
*
* \return
* - AACENC_OK, on succes.
- * - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure.
+ * - 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 c11db27..caa62c5 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -2521,11 +2521,6 @@
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 b080f50..b3ec187 100644
--- a/libDRCdec/src/drcDec_reader.cpp
+++ b/libDRCdec/src/drcDec_reader.cpp
@@ -917,7 +917,7 @@
firFilterOrder;
int uniqueEqSubbandGainsCount, eqSubbandGainRepresentation,
eqSubbandGainCount;
- int eqSubbandGainFormat;
+ EQ_SUBBAND_GAIN_FORMAT eqSubbandGainFormat;
eqDelayMaxPresent = FDKreadBits(hBs, 1);
if (eqDelayMaxPresent) {
@@ -958,7 +958,7 @@
uniqueEqSubbandGainsCount = FDKreadBits(hBs, 6);
if (uniqueEqSubbandGainsCount > 0) {
eqSubbandGainRepresentation = FDKreadBits(hBs, 1);
- eqSubbandGainFormat = FDKreadBits(hBs, 4);
+ eqSubbandGainFormat = (EQ_SUBBAND_GAIN_FORMAT)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 aded569..cca97f1 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -159,6 +159,9 @@
#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 8c5673c..718a555 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -102,6 +102,11 @@
#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
@@ -121,49 +126,45 @@
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])) >>
- len_scale);
+ accu5 = ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3]))
+ SHIFT);
pReBuf++;
/* len must be even */
- accu1 = fPow2Div2(pReBuf[0]) >> len_scale;
- accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) >> len_scale;
+ accu1 = fPow2Div2(pReBuf[0]) SHIFT;
+ accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) SHIFT;
pReBuf++;
for (j = (len - 2) >> 1; j != 0; j--, pReBuf += 2) {
- accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) >> len_scale);
+ accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) SHIFT);
- accu3 +=
- ((fMultDiv2(pReBuf[0], pReBuf[1]) + fMultDiv2(pReBuf[1], pReBuf[2])) >>
- len_scale);
+ accu3 += ((fMultDiv2(pReBuf[0], pReBuf[1]) +
+ fMultDiv2(pReBuf[1], pReBuf[2])) SHIFT);
- accu5 +=
- ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >>
- len_scale);
+ accu5 += ((fMultDiv2(pReBuf[0], pReBuf[2]) +
+ fMultDiv2(pReBuf[1], pReBuf[3])) SHIFT);
}
- accu2 = (fPow2Div2(realBuf[-2]) >> len_scale);
+ accu2 = (fPow2Div2(realBuf[-2]) SHIFT);
accu2 += accu1;
- accu1 += (fPow2Div2(realBuf[len - 2]) >> len_scale);
+ accu1 += (fPow2Div2(realBuf[len - 2]) SHIFT);
- accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) >> len_scale);
+ accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) SHIFT);
accu4 += accu3;
- accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) >> len_scale);
+ accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) SHIFT);
mScale = CntLeadingZeros(
(accu1 | accu2 | fAbs(accu3) | fAbs(accu4) | fAbs(accu5))) -
1;
- autoCorrScaling = mScale - 1 - len_scale; /* -1 because of fMultDiv2*/
+ autoCorrScaling = mScale - 1 - SHIFT_FACTOR; /* -1 because of fMultDiv2*/
/* Scale to common scale factor */
ac->r11r = accu1 << mScale;
@@ -189,7 +190,7 @@
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;
+ int j, autoCorrScaling, mScale, len_scale;
FIXP_DBL accu0, accu1, accu2, accu3, accu4, accu5, accu6, accu7, accu8;
@@ -198,7 +199,7 @@
const FIXP_DBL *realBuf = reBuffer;
const FIXP_DBL *imagBuf = imBuffer;
- const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)len), 1);
+ (len > 64) ? (len_scale = 6) : (len_scale = 5);
/*
r00r,
r11r,r22r
diff --git a/libFDK/src/nlc_dec.cpp b/libFDK/src/nlc_dec.cpp
index 3733d98..6e98ce0 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -568,12 +568,12 @@
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, PAIRING* pairing_scheme, int ldMode) {
+ int num_val, CODING_SCHEME* cdg_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];
@@ -596,13 +596,15 @@
int hufYY;
/* Coding scheme */
- coding_scheme = (CODING_SCHEME)FDKreadBits(strm, 1);
+ data = FDKreadBits(strm, 1);
+ *cdg_scheme = (CODING_SCHEME)(data << PAIR_SHIFT);
- if (coding_scheme == HUFF_2D) {
+ if (*cdg_scheme >> PAIR_SHIFT == HUFF_2D) {
if ((out_data_1 != NULL) && (out_data_2 != NULL) && (ldMode == 0)) {
- *pairing_scheme = (PAIRING)FDKreadBits(strm, 1);
+ data = FDKreadBits(strm, 1);
+ *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | data);
} else {
- *pairing_scheme = FREQ_PAIR;
+ *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | FREQ_PAIR);
}
}
@@ -611,7 +613,7 @@
hufYY2 = diff_type_2;
}
- switch (coding_scheme) {
+ switch (*cdg_scheme >> PAIR_SHIFT) {
case HUFF_1D:
p0_flag[0] = (diff_type_1 == DIFF_FREQ);
p0_flag[1] = (diff_type_2 == DIFF_FREQ);
@@ -632,7 +634,7 @@
case HUFF_2D:
- switch (*pairing_scheme) {
+ switch (*cdg_scheme & PAIR_MASK) {
case FREQ_PAIR:
if (out_data_1 != NULL) {
@@ -841,7 +843,7 @@
SCHAR* pDataVec[2] = {NULL, NULL};
DIFF_TYPE diff_type[2] = {DIFF_FREQ, DIFF_FREQ};
- PAIRING pairing = FREQ_PAIR;
+ CODING_SCHEME cdg_scheme = HUFF_1D;
DIRECTION direction = BACKWARDS;
switch (data_type) {
@@ -957,7 +959,7 @@
}
/* Huffman decoding */
err = huff_decode(strm, pDataVec[0], pDataVec[1], data_type, diff_type[0],
- diff_type[1], dataBands, &pairing,
+ diff_type[1], dataBands, &cdg_scheme,
(DECODER == SAOC_DECODER));
if (err != HUFFDEC_OK) {
return HUFFDEC_NOTOK;
@@ -984,8 +986,8 @@
}
}
- mixed_time_pair =
- (diff_type[0] != diff_type[1]) && (pairing == TIME_PAIR);
+ mixed_time_pair = (diff_type[0] != diff_type[1]) &&
+ ((cdg_scheme & PAIR_MASK) == 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 8f77017..e46cb32 100644
--- a/libMpegTPDec/src/tpdec_asc.cpp
+++ b/libMpegTPDec/src/tpdec_asc.cpp
@@ -1694,7 +1694,8 @@
const AUDIO_OBJECT_TYPE aot) {
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
- UINT usacExtElementType = escapedValue(hBs, 4, 8, 16);
+ USAC_EXT_ELEMENT_TYPE usacExtElementType =
+ (USAC_EXT_ELEMENT_TYPE)escapedValue(hBs, 4, 8, 16);
/* recurve extension elements which are invalid for USAC */
if (aot == AOT_USAC) {
@@ -1711,6 +1712,7 @@
}
}
+ extElement->usacExtElementType = usacExtElementType;
int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16);
extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength;
INT bsAnchor;
@@ -1744,10 +1746,8 @@
}
} 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 @@
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
int numConfigExtensions;
- UINT usacConfigExtType;
+ CONFIG_EXT_ID usacConfigExtType;
int usacConfigExtLength;
int loudnessInfoSetIndex =
-1; /* index of loudnessInfoSet config extension. -1 if not contained. */
@@ -1787,7 +1787,7 @@
for (int confExtIdx = 0; confExtIdx < numConfigExtensions; confExtIdx++) {
INT nbits;
int loudnessInfoSetConfigExtensionPosition = FDKgetValidBits(hBs);
- usacConfigExtType = escapedValue(hBs, 4, 8, 16);
+ usacConfigExtType = (CONFIG_EXT_ID)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 c32be54..3b71db8 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -591,18 +591,6 @@
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;
@@ -614,17 +602,11 @@
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:
- auChunkLengthInfo = CLatmDemux_ReadAuChunkLengthInfo(bs);
- if (auChunkLengthInfo >= 0) {
- p_linfo->m_frameLengthInBits = (UINT)auChunkLengthInfo;
- totalPayloadBits += p_linfo->m_frameLengthInBits;
- } else {
- return TRANSPORTDEC_PARSE_ERROR;
- }
+ p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs);
+ totalPayloadBits += p_linfo->m_frameLengthInBits;
break;
case 3:
case 5:
@@ -645,6 +627,23 @@
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 8b8c971..6af553d 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -151,6 +151,8 @@
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 fca12ce..2070dbc 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -494,40 +494,13 @@
return plainChMode;
}
-/** 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 */
- }
+static inline UINT getIdxSum(UCHAR numCh) {
+ UINT result = 0;
+ int i;
+ for (i = 1; i < numCh; i += 1) {
+ result += i;
}
- return 1; /* Successfully validated channel indices */
+ return result;
}
/** Evaluate a given channel configuration and extract a packed channel mode. In
@@ -550,6 +523,7 @@
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)];
@@ -564,6 +538,7 @@
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));
@@ -577,22 +552,19 @@
(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 (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)) {
+ if (idxSum[plane][chGrp] != getIdxSum(numCh[plane][chGrp])) {
unsigned idxCnt = 0;
for (ch = 0; ch < numChannels; ch += 1) {
- if (channelType[ch] == aChType) {
+ if (channelType[ch] ==
+ (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF))) {
channelIndices[ch] = idxCnt++;
}
}
diff --git a/libSACdec/src/sac_bitdec.cpp b/libSACdec/src/sac_bitdec.cpp
index 25b3d9e..4485ccf 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -488,17 +488,12 @@
pSpatialSpecificConfig->freqRes =
(SPATIALDEC_FREQ_RES)freqResTable_LD[bsFreqRes];
- {
- UINT treeConfig = FDKreadBits(bitstream, 4);
+ pSpatialSpecificConfig->treeConfig =
+ (SPATIALDEC_TREE_CONFIG)FDKreadBits(bitstream, 4);
- switch (treeConfig) {
- case SPATIALDEC_MODE_RSVD7:
- pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG)treeConfig;
- break;
- default:
- err = MPS_UNSUPPORTED_CONFIG;
- goto bail;
- }
+ if (pSpatialSpecificConfig->treeConfig != SPATIALDEC_MODE_RSVD7) {
+ err = MPS_UNSUPPORTED_CONFIG;
+ goto bail;
}
{
diff --git a/libSACdec/src/sac_process.cpp b/libSACdec/src/sac_process.cpp
index 33a1647..22091a9 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -517,11 +517,12 @@
maxVal = fAbs(iReal0) | fAbs(iImag0);
maxVal |= fAbs(iReal1);
- s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2);
+ s = fMax(CntLeadingZeros(maxVal) - 1, 0);
+ s = fMin(s, scale_param_m2);
- mReal0 = scaleValue(iReal0, s);
- mImag0 = scaleValue(iImag0, s);
- mReal1 = scaleValue(iReal1, s);
+ mReal0 = iReal0 << s;
+ mImag0 = iImag0 << s;
+ mReal1 = iReal1 << s;
s = scale_param_m2 - s;
@@ -561,11 +562,12 @@
maxVal = fAbs(iReal0) | fAbs(iImag0);
maxVal |= fAbs(iReal1);
- s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2);
+ s = fMax(CntLeadingZeros(maxVal) - 1, 0);
+ s = fMin(s, scale_param_m2);
- mReal0 = FX_DBL2FX_SGL(scaleValue(iReal0, s));
- mImag0 = FX_DBL2FX_SGL(scaleValue(iImag0, s));
- mReal1 = FX_DBL2FX_SGL(scaleValue(iReal1, s));
+ mReal0 = FX_DBL2FX_SGL(iReal0 << s);
+ mImag0 = FX_DBL2FX_SGL(iImag0 << s);
+ mReal1 = FX_DBL2FX_SGL(iReal1 << s);
s = scale_param_m2 - s;
diff --git a/libSACdec/src/sac_reshapeBBEnv.cpp b/libSACdec/src/sac_reshapeBBEnv.cpp
index 72f4e58..272d009 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -241,56 +241,29 @@
}
}
-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;
+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;
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++) {
- /* 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);
+ dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)) +
+ fPow2Div2(pHybOutputImagDry[qs] << (1)));
+ wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)) +
+ fPow2Div2(pHybOutputImagWet[qs] << (1)));
}
for (; qs < hybBands; qs++) {
- dry += (fPow2Div2(pHybOutputRealDry[qs] << headroom_dry) >> s1);
- wet += (fPow2Div2(pHybOutputRealWet[qs] << headroom_wet) >> s1);
+ dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)));
+ wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)));
}
-
- /* 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__)
@@ -560,7 +533,6 @@
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];
@@ -622,25 +594,22 @@
dryFacSF = SF_SHAPE + 2 * dryFacSF;
}
- slotAmp_dry_e = slotAmp_wet_e = 0;
-
/* calculate slotAmp_dry and slotAmp_wet */
- slotAmp(&slotAmp_dry, &slotAmp_dry_e, &slotAmp_wet, &slotAmp_wet_e,
- &self->hybOutputRealDry__FDK[ch][6],
+ slotAmp(&slotAmp_dry, &slotAmp_wet, &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)) {
- slotAmp_wet = sqrtFixp(slotAmp_wet);
+ sc = fixMax(0, CntLeadingZeros(slotAmp_wet) - 1);
+ sc = sc - (sc & 1);
+
+ slotAmp_wet = sqrtFixp(slotAmp_wet << sc);
slotAmp_dry = invSqrtNorm2(slotAmp_dry, &slotAmpSF);
slotAmp_ratio = fMult(slotAmp_wet, slotAmp_dry);
- slotAmpSF = slotAmpSF + (slotAmp_wet_e >> 1) - (slotAmp_dry_e >> 1);
+ slotAmpSF = slotAmpSF - (sc >> 1);
}
/* calculate common scale factor */
diff --git a/libSACdec/src/sac_stp.cpp b/libSACdec/src/sac_stp.cpp
index 0e6affa..b328c82 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -229,13 +229,15 @@
int n;
FIXP_DBL scaleY;
for (n = bands - 1; n >= 0; n--) {
- scaleY = fMult(scaleX, *pBP);
+ scaleY = fMultDiv2(scaleX, *pBP);
*hybOutputRealDry = SATURATE_LEFT_SHIFT(
- (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleY),
- SF_SCALE, DFRACT_BITS);
+ (*hybOutputRealDry >> 1) +
+ (fMultDiv2(*hybOutputRealWet, scaleY) << (SF_SCALE + 1)),
+ 1, DFRACT_BITS);
*hybOutputImagDry = SATURATE_LEFT_SHIFT(
- (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleY),
- SF_SCALE, DFRACT_BITS);
+ (*hybOutputImagDry >> 1) +
+ (fMultDiv2(*hybOutputImagWet, scaleY) << (SF_SCALE + 1)),
+ 1, DFRACT_BITS);
hybOutputRealDry++, hybOutputRealWet++;
hybOutputImagDry++, hybOutputImagWet++;
pBP++;
@@ -250,12 +252,12 @@
int n;
for (n = bands - 1; n >= 0; n--) {
- *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 =
+ *hybOutputRealDry +
+ (fMultDiv2(*hybOutputRealWet, scaleX) << (SF_SCALE + 1));
+ *hybOutputImagDry =
+ *hybOutputImagDry +
+ (fMultDiv2(*hybOutputImagWet, scaleX) << (SF_SCALE + 1));
hybOutputRealDry++, hybOutputRealWet++;
hybOutputImagDry++, hybOutputImagWet++;
}
diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp
new file mode 100644
index 0000000..db1948f
--- /dev/null
+++ b/libSBRdec/src/arm/lpp_tran_arm.cpp
@@ -0,0 +1,159 @@
+/* -----------------------------------------------------------------------------
+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 cefa612..ad5edfe 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -664,7 +664,7 @@
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] >>= fMin(DFRACT_BITS - 1, -gain_sf[i]);
+ gain[i] >>= -gain_sf[i];
gain_sf[i] = 0;
}
} else {
@@ -683,6 +683,11 @@
/* 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] */
@@ -750,15 +755,23 @@
int gain_adj_sf = gain_adj_2_sf;
for (i = 0; i < nbSubsample; ++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);
+ 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];
for (j = lowSubband; j < highSubband; j++) {
- qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain_final);
- qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain_final);
+ qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain[i]);
+ qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain[i]);
}
}
} else { /* gamma_idx == 0 */
diff --git a/libSBRdec/src/hbe.cpp b/libSBRdec/src/hbe.cpp
index f2452ea..d210bb6 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -1400,27 +1400,42 @@
if (shift_ov != 0) {
for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) {
- scaleValuesSaturate(&hQmfTransposer->qmfHBEBufReal_F[i][0],
- QMF_SYNTH_CHANNELS, shift_ov);
- scaleValuesSaturate(&hQmfTransposer->qmfHBEBufImag_F[i][0],
- QMF_SYNTH_CHANNELS, shift_ov);
+ 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);
+ }
+ }
+ }
+ }
+
+ 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++) {
- scaleValuesSaturate(&lpcFilterStatesReal[i][0], (64), shift_ov);
- scaleValuesSaturate(&lpcFilterStatesImag[i][0], (64), 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);
+ }
}
}
}
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
index 68a25bf..93e1158 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -132,6 +132,10 @@
#include "HFgen_preFlat.h"
+#if defined(__arm__)
+#include "arm/lpp_tran_arm.cpp"
+#endif
+
#define LPC_SCALE_FACTOR 2
/*!
@@ -216,21 +220,19 @@
const FIXP_DBL *const lowBandReal,
const int startSample,
const int stopSample, const UCHAR hiBand,
- const int dynamicScale,
+ const int dynamicScale, const int descale,
const FIXP_SGL a0r, const FIXP_SGL a1r) {
- 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);
+ FIXP_DBL accu1, accu2;
+ int i;
- 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);
-
- qmfBufferReal[i + startSample][hiBand] =
- SATURATE_LEFT_SHIFT(accu, rescale, DFRACT_BITS);
+ accu1 <<= 1;
+ accu2 = (lowBandReal[i + 2] >> descale);
+ qmfBufferReal[i + startSample][hiBand] = accu1 + accu2;
}
}
@@ -527,7 +529,7 @@
if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
+ alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphar[1] = -alphar[1];
}
@@ -555,7 +557,7 @@
scale)) {
resetLPCCoeffs = 1;
} else {
- alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
+ alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphai[1] = -alphai[1];
}
@@ -594,7 +596,7 @@
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
+ alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphar[0] = -alphar[0];
@@ -614,7 +616,7 @@
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
+ alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphai[0] = -alphai[0];
}
@@ -657,7 +659,7 @@
INT scale;
FIXP_DBL result =
fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale);
- k1 = scaleValueSaturate(result, scale);
+ k1 = scaleValue(result, scale);
if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) {
k1 = -k1;
@@ -769,50 +771,52 @@
} else { /* bw <= 0 */
if (!useLP) {
- 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);
-
+ 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
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])) >>
- 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 = (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 >> dynscale);
- accu2 =
- (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 >> dynscale);
+ accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);
+ accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);
if (fPreWhitening) {
- qmfBufferReal[i][hiBand] = scaleValueSaturate(
+ accu1 = scaleValueSaturate(
fMultDiv2(accu1, preWhiteningGains[loBand]),
- preWhiteningGains_exp[loBand] + 1 + rescale);
- qmfBufferImag[i][hiBand] = scaleValueSaturate(
+ preWhiteningGains_exp[loBand] + 1);
+ accu2 = scaleValueSaturate(
fMultDiv2(accu2, preWhiteningGains[loBand]),
- 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);
+ preWhiteningGains_exp[loBand] + 1);
}
+ 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, a0r, a1r);
+ startSample, stopSample, hiBand, dynamicScale,
+ fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r,
+ a1r);
}
} /* bw <= 0 */
@@ -1062,7 +1066,7 @@
if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphar[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
+ alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphar[1] = -alphar[1];
}
@@ -1088,7 +1092,7 @@
(result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) {
resetLPCCoeffs = 1;
} else {
- alphai[1] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
+ alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale));
if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
alphai[1] = -alphai[1];
}
@@ -1117,7 +1121,7 @@
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphar[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
+ alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
alphar[0] = -alphar[0];
@@ -1136,7 +1140,7 @@
} else {
INT scale;
FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
- alphai[0] = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
+ alphai[0] = FX_DBL2FX_SGL(scaleValue(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 919e9bb..b1fb0da 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 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -713,8 +713,7 @@
} else { /* (flags & SBRDEC_PS_DECODED) */
INT sdiff;
- INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov,
- outScalefactor, outScalefactorR, outScalefactorL;
+ INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->qmfDomainOutCh->fb;
HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->qmfDomainOutCh->fb;
@@ -745,7 +744,7 @@
*/
FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <=
QMF_MAX_SYNTHESIS_BANDS);
- synQmfRight->outScalefactor = synQmf->outScalefactor;
+ qmfChangeOutScalefactor(synQmfRight, -(8));
FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates,
9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis *
sizeof(FIXP_QSS));
@@ -789,12 +788,10 @@
FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel,
sizeof(SBRDEC_DRC_CHANNEL));
- outScalefactor = maxShift - (8);
- outScalefactorL = outScalefactorR =
- sbrInDataHeadroom + 1; /* +1: psDiffScale! (MPEG-PS) */
-
for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
+ INT outScalefactorR, outScalefactorL;
+
/* qmf timeslot of right channel */
FIXP_DBL *rQmfReal = pWorkBuffer;
FIXP_DBL *rQmfImag = pWorkBuffer + synQmf->no_channels;
@@ -818,20 +815,27 @@
? 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, outScalefactor);
- qmfChangeOutScalefactor(synQmfRight, outScalefactor);
+ qmfChangeOutScalefactor(synQmf, -(8));
+ qmfChangeOutScalefactor(synQmfRight, -(8));
qmfSynthesisFilteringSlot(
synQmfRight, rQmfReal, /* QMF real buffer */
diff --git a/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp
index daa3554..e187656 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -765,6 +765,9 @@
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)
@@ -774,9 +777,6 @@
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 @@
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 7718695..b101a4a 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 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -961,10 +961,8 @@
/* Set sync state UPSAMPLING for the corresponding slot.
This switches off bitstream parsing until a new header arrives. */
- if (hSbrHeader->syncState != SBR_NOT_INITIALIZED) {
- hSbrHeader->syncState = UPSAMPLING;
- hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
- }
+ hSbrHeader->syncState = UPSAMPLING;
+ hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
}
}
} break;
@@ -1373,9 +1371,7 @@
}
if (headerStatus == HEADER_ERROR) {
/* Corrupt SBR info data, do not decode and switch to UPSAMPLING */
- hSbrHeader->syncState = hSbrHeader->syncState > UPSAMPLING
- ? UPSAMPLING
- : hSbrHeader->syncState;
+ hSbrHeader->syncState = UPSAMPLING;
fDoDecodeSbrData = 0;
sbrHeaderPresent = 0;
}
@@ -1614,9 +1610,7 @@
/* No valid SBR payload available, hence switch to upsampling (in all
* headers) */
for (hdrIdx = 0; hdrIdx < ((1) + 1); hdrIdx += 1) {
- if (self->sbrHeader[elementIndex][hdrIdx].syncState > UPSAMPLING) {
- 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
