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-rw-r--r--Android.bp51
-rw-r--r--METADATA3
-rw-r--r--fuzzer/Android.bp82
-rw-r--r--fuzzer/README.md150
-rw-r--r--fuzzer/aac_dec_fuzzer.cpp141
-rw-r--r--fuzzer/aac_enc_fuzzer.cpp479
-rw-r--r--libAACdec/src/usacdec_acelp.cpp4
-rw-r--r--libFDK/include/nlc_dec.h5
-rw-r--r--libFDK/src/autocorr2nd.cpp43
-rw-r--r--libFDK/src/nlc_dec.cpp28
-rw-r--r--libMpegTPDec/src/tpdec_asc.cpp10
-rw-r--r--libPCMutils/src/pcmdmx_lib.cpp56
-rw-r--r--libSACdec/src/sac_bitdec.cpp17
-rw-r--r--libSACdec/src/sac_stp.cpp15
-rw-r--r--libSBRdec/src/arm/lpp_tran_arm.cpp159
-rw-r--r--libSBRdec/src/lpp_tran.cpp94
-rw-r--r--libSBRdec/src/sbr_dec.cpp24
17 files changed, 1059 insertions, 302 deletions
diff --git a/Android.bp b/Android.bp
index 3f42c19..0c67186 100644
--- a/Android.bp
+++ b/Android.bp
@@ -1,6 +1,40 @@
+// *** THIS PACKAGE HAS SPECIAL LICENSING CONDITIONS. PLEASE
+// CONSULT THE OWNERS AND opensource-licensing@google.com BEFORE
+// DEPENDING ON IT IN YOUR PROJECT. ***
+package {
+ default_applicable_licenses: ["external_aac_license"],
+}
+
+// Added automatically by a large-scale-change that took the approach of
+// 'apply every license found to every target'. While this makes sure we respect
+// every license restriction, it may not be entirely correct.
+//
+// e.g. GPL in an MIT project might only apply to the contrib/ directory.
+//
+// Please consider splitting the single license below into multiple licenses,
+// taking care not to lose any license_kind information, and overriding the
+// default license using the 'licenses: [...]' property on targets as needed.
+//
+// For unused files, consider creating a 'fileGroup' with "//visibility:private"
+// to attach the license to, and including a comment whether the files may be
+// used in the current project.
+// See: http://go/android-license-faq
+license {
+ name: "external_aac_license",
+ visibility: [":__subpackages__"],
+ license_kinds: [
+ "SPDX-license-identifier-Apache-2.0",
+ "legacy_by_exception_only", // by exception only
+ ],
+ license_text: [
+ "NOTICE",
+ ],
+}
+
cc_library_static {
name: "libFraunhoferAAC",
vendor_available: true,
+ host_supported: true,
srcs: [
"libAACdec/src/*.cpp",
"libAACenc/src/*.cpp",
@@ -26,13 +60,12 @@ cc_library_static {
"-DSUPPRESS_BUILD_DATE_INFO",
],
sanitize: {
- misc_undefined:[
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- "bounds",
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ "bounds",
],
- // Enable CFI if this becomes a shared library.
- // cfi: true,
+ cfi: true,
},
shared_libs: [
"liblog",
@@ -53,6 +86,12 @@ cc_library_static {
"libSACenc/include",
],
+ target: {
+ darwin: {
+ enabled: false,
+ },
+ },
+
apex_available: [
"//apex_available:platform",
"com.android.bluetooth.updatable",
diff --git a/METADATA b/METADATA
new file mode 100644
index 0000000..5c12860
--- /dev/null
+++ b/METADATA
@@ -0,0 +1,3 @@
+third_party {
+ license_type: BY_EXCEPTION_ONLY
+}
diff --git a/fuzzer/Android.bp b/fuzzer/Android.bp
new file mode 100644
index 0000000..6739798
--- /dev/null
+++ b/fuzzer/Android.bp
@@ -0,0 +1,82 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *****************************************************************************
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore
+ */
+
+package {
+ // See: http://go/android-license-faq
+ // A large-scale-change added 'default_applicable_licenses' to import
+ // all of the 'license_kinds' from "external_aac_license"
+ // to get the below license kinds:
+ // SPDX-license-identifier-Apache-2.0
+ default_applicable_licenses: ["external_aac_license"],
+}
+
+cc_defaults {
+ name: "aac_fuzz_defaults",
+ host_supported: true,
+
+ static_libs: [
+ "libFraunhoferAAC",
+ ],
+
+ target: {
+ darwin: {
+ enabled: false,
+ },
+ },
+
+ fuzz_config: {
+ cc: [
+ "android-media-fuzzing-reports@google.com",
+ ],
+ componentid: 155276,
+ },
+}
+
+cc_fuzz {
+ name: "aac_dec_fuzzer",
+
+ srcs: [
+ "aac_dec_fuzzer.cpp",
+ ],
+
+ static_libs: [
+ "liblog",
+ ],
+
+ defaults: [
+ "aac_fuzz_defaults"
+ ],
+}
+
+cc_fuzz {
+ name: "aac_enc_fuzzer",
+
+ srcs: [
+ "aac_enc_fuzzer.cpp",
+ ],
+
+ defaults: [
+ "aac_fuzz_defaults"
+ ],
+
+ include_dirs: [
+ "external/aac/libAACenc/"
+ ],
+}
diff --git a/fuzzer/README.md b/fuzzer/README.md
new file mode 100644
index 0000000..b8cc260
--- /dev/null
+++ b/fuzzer/README.md
@@ -0,0 +1,150 @@
+# Fuzzer for libFraunhoferAAC decoder
+
+## Plugin Design Considerations
+The fuzzer plugin for aac decoder is designed based on the understanding of the
+codec and tries to achieve the following:
+
+##### Maximize code coverage
+
+This fuzzer makes use of the following config parameters:
+1. Transport type (parameter name: `TRANSPORT_TYPE`)
+
+| Parameter| Valid Values| Configured Value|
+|------------- |-------------| ----- |
+| `TRANSPORT_TYPE` | 0.`TT_UNKNOWN ` 1.`TT_MP4_RAW ` 2.`TT_MP4_ADIF ` 3.`TT_MP4_ADTS ` 4.`TT_MP4_LATM_MCP1 ` 5.`TT_MP4_LATM_MCP0 ` 6.`TT_MP4_LOAS ` 7.`TT_DRM ` | `TT_MP4_ADIF ` |
+
+Note: Value of `TRANSPORT_TYPE` could be set to any of these values.
+It is set to `TT_MP4_ADIF` in the fuzzer plugin.
+
+##### Maximize utilization of input data
+The plugin feeds the entire input data to the codec using a loop.
+ * If the decode operation was successful, the input is advanced by an
+ offset calculated using valid bytes.
+ * If the decode operation was un-successful, the input is advanced by 1 byte
+ till it reaches a valid frame or end of stream.
+
+This ensures that the plugin tolerates any kind of input (empty, huge,
+malformed, etc) and doesnt `exit()` on any input and thereby increasing the
+chance of identifying vulnerabilities.
+
+## Build
+
+This describes steps to build aac_dec_fuzzer binary.
+
+## Android
+
+### Steps to build
+Build the fuzzer
+```
+ $ mm -j$(nproc) aac_dec_fuzzer
+```
+
+### Steps to run
+Create a directory CORPUS_DIR and copy some aac files to that folder.
+Push this directory to device.
+
+To run on device
+```
+ $ adb sync data
+ $ adb shell /data/fuzz/arm64/aac_dec_fuzzer/aac_dec_fuzzer CORPUS_DIR
+```
+To run on host
+```
+ $ $ANDROID_HOST_OUT/fuzz/x86_64/aac_dec_fuzzer/aac_dec_fuzzer CORPUS_DIR
+```
+
+# Fuzzer for libFraunhoferAAC encoder
+
+## Plugin Design Considerations
+The fuzzer plugin for aac encoder is designed based on the understanding of the
+codec and tries to achieve the following:
+
+##### Maximize code coverage
+
+The configuration parameters are not hardcoded, but instead selected based on
+incoming data. This ensures more code paths are reached by the fuzzer.
+
+Following arguments are passed to aacEncoder_SetParam to set the respective AACENC_PARAM parameter:
+
+| AACENC_PARAM param| Valid Values| Configured Value|
+|-------------| ----- |----- |
+|`AACENC_SBR_MODE` | `-1 ` `0 ` `1 ` `2 ` | Calculated using first byte of data |
+|`AACENC_AOT` |`AOT_NONE ` `AOT_NULL_OBJECT ` `AOT_AAC_MAIN ` `AOT_AAC_LC ` `AOT_AAC_SSR ` `AOT_AAC_LTP ` `AOT_SBR ` `AOT_AAC_SCAL ` `AOT_TWIN_VQ ` `AOT_CELP ` `AOT_HVXC ` `AOT_RSVD_10 ` `AOT_RSVD_11 ` `AOT_TTSI ` `AOT_MAIN_SYNTH ` `AOT_WAV_TAB_SYNTH ` `AOT_GEN_MIDI ` `AOT_ALG_SYNTH_AUD_FX ` `AOT_ER_AAC_LC ` `AOT_RSVD_18 ` `AOT_ER_AAC_LTP ` `AOT_ER_AAC_SCAL ` `AOT_ER_TWIN_VQ ` `AOT_ER_BSAC ` `AOT_ER_AAC_LD ` `AOT_ER_CELP ` `AOT_ER_HVXC ` `AOT_ER_HILN ` `AOT_ER_PARA ` `AOT_RSVD_28 ` `AOT_PS ` `AOT_MPEGS ` `AOT_ESCAPE ` `AOT_MP3ONMP4_L1 ` `AOT_MP3ONMP4_L2 ` `AOT_MP3ONMP4_L3 ` `AOT_RSVD_35 ` `AOT_RSVD_36 ` `AOT_AAC_SLS ` `AOT_SLS ` `AOT_ER_AAC_ELD ` `AOT_USAC ` `AOT_SAOC ` `AOT_LD_MPEGS ` `AOT_MP2_AAC_LC ` `AOT_MP2_SBR ` `AOT_DRM_AAC ` `AOT_DRM_SBR ` `AOT_DRM_MPEG_PS ` `AOT_DRM_SURROUND ` `AOT_DRM_USAC ` | Calculated using second byte of data |
+|`AACENC_SAMPLERATE` | `8000 ` `11025 ` `12000 ` `16000 ` `22050 ` `24000 ` `32000 ` `44100 ` `48000 ` `64000 ` `88200 ` `96000 `| Calculated using third byte of data |
+|`AACENC_BITRATE` | In range `8000 ` to `960000 ` | Calculated using fourth, fifth and sixth byte of data |
+|`AACENC_CHANNELMODE` | `MODE_1 ` `MODE_2 ` `MODE_1_2 ` `MODE_1_2_1 ` `MODE_1_2_2 ` `MODE_1_2_2_1 ` `MODE_1_2_2_2_1 ` `MODE_6_1 ` `MODE_7_1_BACK ` `MODE_7_1_TOP_FRONT ` `MODE_7_1_REAR_SURROUND ` `MODE_7_1_FRONT_CENTER ` `MODE_212 ` | Calculated using seventh byte of data |
+|`AACENC_TRANSMUX` | `TT_MP4_RAW ` `TT_MP4_ADIF ` `TT_MP4_ADTS ` `TT_MP4_LATM_MCP1 ` `TT_MP4_LATM_MCP0 ` `TT_MP4_LOAS ` `TT_DRM ` | Calculated using eight byte of data |`AACENC_SBR_RATIO` |`-1 ` `0 ` `1 ` `2 ` | Calculated using ninth byte of data |
+|`AACENC_BITRATEMODE` |`AACENC_BR_MODE_INVALID ` `AACENC_BR_MODE_CBR ` `AACENC_BR_MODE_VBR_1 ` `AACENC_BR_MODE_VBR_2 ` `AACENC_BR_MODE_VBR_3 ` `AACENC_BR_MODE_VBR_4 ` `AACENC_BR_MODE_VBR_5 ` `AACENC_BR_MODE_FF ` `AACENC_BR_MODE_SFR ` | Calculated using thirty-fourth byte of data |
+|`AACENC_GRANULE_LENGTH` |`120 ` `128 ` `240 ` `256 ` `480 ` `512 ` `1024 ` | Calculated using thirty-fifth byte of data |
+|`AACENC_CHANNELORDER` |`CH_ORDER_MPEG ` `CH_ORDER_WAV ` | Calculated using thirty-sixth byte of data |
+|`AACENC_AFTERBURNER` |`0 ` `1 ` | Calculated using thirty-seventh byte of data |
+|`AACENC_BANDWIDTH` |`0 ` `1` | Calculated using thirty-eigth byte of data |
+|` AACENC_IDX_PEAK_BITRATE` | In range `8000 ` to `960000 ` | Calculated using thirty-ninth byte of data |
+|` AACENC_HEADER_PERIOD` |In range `0 ` to `255 ` | Calculated using fortieth byte of data |
+|` AACENC_SIGNALING_MODE` |`-1 ` `0 ` `1 ` `2 ` `3 ` | Calculated using forty-first byte of data |
+|` AACENC_TPSUBFRAMES` |In range `0 ` to `255 ` | Calculated using forty-second byte of data |
+|` AACENC_AUDIOMUXVER` |`-1 ` `0 ` `1 ` `2 ` | Calculated using forty-third byte of data |
+|` AACENC_PROTECTION` |`0 ` `1 ` | Calculated using forty-fourth of data |
+|`AACENC_ANCILLARY_BITRATE` |In range `0 ` to `960000 `| Calculated using forty-fifth byte of data |
+|`AACENC_METADATA_MODE ` |`0 ` `1 ` `2 ` `3 ` | Calculated using forty-sixth byte of data |
+
+Following values are configured to set up the meta data represented by the class variable `mMetaData ` :
+
+| Variable name| Possible Values| Configured Value|
+|------------- | ----- |----- |
+| `drc_profile` | `AACENC_METADATA_DRC_NONE ` `AACENC_METADATA_DRC_FILMSTANDARD ` `AACENC_METADATA_DRC_FILMLIGHT ` `AACENC_METADATA_DRC_MUSICSTANDARD ` `AACENC_METADATA_DRC_MUSICLIGHT ` `AACENC_METADATA_DRC_SPEECH ` `AACENC_METADATA_DRC_NOT_PRESENT ` | Calculated using tenth byte of data |
+| `comp_profile` | `AACENC_METADATA_DRC_NONE ` `AACENC_METADATA_DRC_FILMSTANDARD ` `AACENC_METADATA_DRC_FILMLIGHT ` `AACENC_METADATA_DRC_MUSICSTANDARD ` `AACENC_METADATA_DRC_MUSICLIGHT ` `AACENC_METADATA_DRC_SPEECH ` `AACENC_METADATA_DRC_NOT_PRESENT ` | Calculated using eleventh byte of data |
+| `drc_TargetRefLevel` | In range `0 ` to `255 ` | Calculated using twelfth byte of data |
+| `comp_TargetRefLevel` | In range `0 ` to `255 ` | Calculated using thirteenth byte of data |
+| `prog_ref_level_present` | `0 ` `1 ` | Calculated using fourteenth byte of data |
+| `prog_ref_level` | In range `0 ` to `255 ` | Calculated using fifteenth byte of data |
+| `PCE_mixdown_idx_present` | `0 ` `1 ` | Calculated using sixteenth byte of data |
+| `ETSI_DmxLvl_present` | `0 ` `1 ` | Calculated using seventeenth byte of data |
+| `centerMixLevel` | In range `0 ` to `7 ` | Calculated using eighteenth byte of data |
+| `surroundMixLevel` | In range `0 ` to `7 ` | Calculated using nineteenth byte of data |
+| `dolbySurroundMode` | In range `0 ` to `2 ` | Calculated using twentieth byte of data |
+| `drcPresentationMode` | In range `0 ` to `2 ` | Calculated using twenty-first byte of data |
+| `extAncDataEnable` | `0 ` `1 ` | Calculated using twenty-second byte of data |
+| `extDownmixLevelEnable` | `0 ` `1 ` | Calculated using twenty-third byte of data |
+| `extDownmixLevel_A` | In range `0 ` to `7 ` | Calculated using twenty-fourth byte of data |
+| `extDownmixLevel_B` | In range `0 ` to `7 ` | Calculated using twenty-fifth byte of data |
+| `dmxGainEnable` | `0 ` `1 ` | Calculated using twenty-sixth byte of data |
+| `dmxGain5` | In range `0 ` to `255 ` | Calculated using twenty-seventh byte of data |
+| `dmxGain2` | In range `0 ` to `255 ` | Calculated using twenty-eighth byte of data |
+| `lfeDmxEnable` | `0 ` `1 ` | Calculated using twenty-ninth byte of data |
+| `lfeDmxLevel` | In range `0 ` to `15 ` | Calculated using thirtieth byte of data |
+
+Indexes `mInBufferIdx_1`, `mInBufferIdx_2` and `mInBufferIdx_3`(in range `0 ` to `2`) are calculated using the thirty-first, thirty-second and thirty-third byte respectively.
+
+##### Maximize utilization of input data
+The plugin feeds the entire input data to the codec and continues with the encoding even on a failure. This ensures that the plugin tolerates any kind of input (empty, huge, malformed, etc) and doesnt `exit()` on any input and thereby increasing the chance of identifying vulnerabilities.
+
+## Build
+
+This describes steps to build aac_enc_fuzzer binary.
+
+## Android
+
+### Steps to build
+Build the fuzzer
+```
+ $ mm -j$(nproc) aac_enc_fuzzer
+```
+
+### Steps to run
+Create a directory CORPUS_DIR and copy some raw files to that folder.
+Push this directory to device.
+
+To run on device
+```
+ $ adb sync data
+ $ adb shell /data/fuzz/arm64/aac_enc_fuzzer/aac_enc_fuzzer CORPUS_DIR
+```
+To run on host
+```
+ $ $ANDROID_HOST_OUT/fuzz/x86_64/aac_enc_fuzzer/aac_enc_fuzzer CORPUS_DIR
+```
+
+## References:
+ * http://llvm.org/docs/LibFuzzer.html
+ * https://github.com/google/oss-fuzz
diff --git a/fuzzer/aac_dec_fuzzer.cpp b/fuzzer/aac_dec_fuzzer.cpp
new file mode 100644
index 0000000..c970197
--- /dev/null
+++ b/fuzzer/aac_dec_fuzzer.cpp
@@ -0,0 +1,141 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *****************************************************************************
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore
+ */
+
+#include <stdint.h>
+#include <string.h>
+#include <algorithm>
+#include "aacdecoder_lib.h"
+
+constexpr uint8_t kNumberOfLayers = 1;
+constexpr uint8_t kMaxChannelCount = 8;
+constexpr uint32_t kMaxConfigurationSize = 1024;
+constexpr uint32_t kMaxOutBufferSize = 2048 * kMaxChannelCount;
+
+// Value indicating the start of AAC Header Segment
+constexpr const char *kAacSegStartSeq = "AAC_STRT";
+constexpr uint8_t kAacSegStartSeqLen = sizeof(kAacSegStartSeq);
+// Value indicating the end of AAC Header Segment
+constexpr const char *kAacSegEndSeq = "AAC_ENDS";
+constexpr uint8_t kAacSegEndSeqLen = sizeof(kAacSegEndSeq);
+
+// Number of bytes used to signal the length of the header
+constexpr uint8_t kHeaderLengthBytes = 2;
+// Minimum size of an AAC header is 2
+// Minimum data required is
+// strlen(AAC_STRT) + strlen(AAC_ENDS) + kHeaderLengthBytes + 2;
+constexpr UINT kMinDataSize = kAacSegStartSeqLen + kAacSegEndSeqLen + kHeaderLengthBytes + 2;
+
+UINT getHeaderSize(UCHAR *data, UINT size) {
+ if (size < kMinDataSize) {
+ return 0;
+ }
+
+ int32_t result = memcmp(data, kAacSegStartSeq, kAacSegStartSeqLen);
+ if (result) {
+ return 0;
+ }
+ data += kAacSegStartSeqLen;
+ size -= kAacSegStartSeqLen;
+
+ uint32_t headerLengthInBytes = (data[0] << 8 | data[1]) & 0xFFFF;
+ data += kHeaderLengthBytes;
+ size -= kHeaderLengthBytes;
+
+ if (headerLengthInBytes + kAacSegEndSeqLen > size) {
+ return 0;
+ }
+
+ data += headerLengthInBytes;
+ size -= headerLengthInBytes;
+ result = memcmp(data, kAacSegEndSeq, kAacSegEndSeqLen);
+ if (result) {
+ return 0;
+ }
+
+ return std::min(headerLengthInBytes, kMaxConfigurationSize);
+}
+
+class Codec {
+ public:
+ Codec() = default;
+ ~Codec() { deInitDecoder(); }
+ bool initDecoder();
+ void decodeFrames(UCHAR *data, UINT size);
+ void deInitDecoder();
+
+ private:
+ HANDLE_AACDECODER mAacDecoderHandle = nullptr;
+ AAC_DECODER_ERROR mErrorCode = AAC_DEC_OK;
+};
+
+bool Codec::initDecoder() {
+ mAacDecoderHandle = aacDecoder_Open(TT_MP4_ADIF, kNumberOfLayers);
+ if (!mAacDecoderHandle) {
+ return false;
+ }
+ return true;
+}
+
+void Codec::deInitDecoder() {
+ aacDecoder_Close(mAacDecoderHandle);
+ mAacDecoderHandle = nullptr;
+}
+
+void Codec::decodeFrames(UCHAR *data, UINT size) {
+ UINT headerSize = getHeaderSize(data, size);
+ if (headerSize != 0) {
+ data += kAacSegStartSeqLen + kHeaderLengthBytes;
+ size -= kAacSegStartSeqLen + kHeaderLengthBytes;
+ aacDecoder_ConfigRaw(mAacDecoderHandle, &data, &headerSize);
+ data += headerSize + kAacSegEndSeqLen;
+ size -= headerSize + kAacSegEndSeqLen;
+ }
+ while (size > 0) {
+ UINT inputSize = size;
+ UINT valid = size;
+ mErrorCode = aacDecoder_Fill(mAacDecoderHandle, &data, &inputSize, &valid);
+ if (mErrorCode != AAC_DEC_OK) {
+ ++data;
+ --size;
+ } else {
+ INT_PCM outputBuf[kMaxOutBufferSize];
+ do {
+ mErrorCode =
+ aacDecoder_DecodeFrame(mAacDecoderHandle, outputBuf,
+ kMaxOutBufferSize /*size in number of INT_PCM, not bytes*/, 0);
+ } while (mErrorCode == AAC_DEC_OK);
+ UINT offset = inputSize - valid;
+ data += offset;
+ size = valid;
+ }
+ }
+}
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
+ Codec *codec = new Codec();
+ if (!codec) {
+ return 0;
+ }
+ if (codec->initDecoder()) {
+ codec->decodeFrames((UCHAR *)(data), static_cast<UINT>(size));
+ }
+ delete codec;
+ return 0;
+}
diff --git a/fuzzer/aac_enc_fuzzer.cpp b/fuzzer/aac_enc_fuzzer.cpp
new file mode 100644
index 0000000..5a35d70
--- /dev/null
+++ b/fuzzer/aac_enc_fuzzer.cpp
@@ -0,0 +1,479 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ *****************************************************************************
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore
+ */
+
+#include <string>
+#include "aacenc_lib.h"
+#include "src/aacenc.h"
+
+using namespace std;
+
+// IN_AUDIO_DATA, IN_ANCILLRY_DATA and IN_METADATA_SETUP
+constexpr size_t kMaxBuffers = 3;
+
+constexpr size_t kMaxOutputBufferSize = 8192;
+
+constexpr uint32_t kMinBitRate = 8000;
+constexpr uint32_t kMaxBitRate = 960000;
+
+constexpr int32_t kSampleRates[] = {8000, 11025, 12000, 16000, 22050, 24000,
+ 32000, 44100, 48000, 64000, 88200, 96000};
+constexpr size_t kSampleRatesSize = size(kSampleRates);
+
+constexpr CHANNEL_MODE kChannelModes[] = {MODE_1,
+ MODE_2,
+ MODE_1_2,
+ MODE_1_2_1,
+ MODE_1_2_2,
+ MODE_1_2_2_1,
+ MODE_1_2_2_2_1,
+ MODE_6_1,
+ MODE_7_1_BACK,
+ MODE_7_1_TOP_FRONT,
+ MODE_7_1_REAR_SURROUND,
+ MODE_7_1_FRONT_CENTER,
+ MODE_212};
+constexpr size_t kChannelModesSize = size(kChannelModes);
+
+constexpr TRANSPORT_TYPE kIdentifiers[] = {
+ TT_MP4_RAW, TT_MP4_ADIF, TT_MP4_ADTS, TT_MP4_LATM_MCP1, TT_MP4_LATM_MCP0, TT_MP4_LOAS, TT_DRM};
+constexpr size_t kIdentifiersSize = size(kIdentifiers);
+
+constexpr AUDIO_OBJECT_TYPE kAudioObjectTypes[] = {AOT_NONE, AOT_NULL_OBJECT,
+ AOT_AAC_MAIN, AOT_AAC_LC,
+ AOT_AAC_SSR, AOT_AAC_LTP,
+ AOT_SBR, AOT_AAC_SCAL,
+ AOT_TWIN_VQ, AOT_CELP,
+ AOT_HVXC, AOT_RSVD_10,
+ AOT_RSVD_11, AOT_TTSI,
+ AOT_MAIN_SYNTH, AOT_WAV_TAB_SYNTH,
+ AOT_GEN_MIDI, AOT_ALG_SYNTH_AUD_FX,
+ AOT_ER_AAC_LC, AOT_RSVD_18,
+ AOT_ER_AAC_LTP, AOT_ER_AAC_SCAL,
+ AOT_ER_TWIN_VQ, AOT_ER_BSAC,
+ AOT_ER_AAC_LD, AOT_ER_CELP,
+ AOT_ER_HVXC, AOT_ER_HILN,
+ AOT_ER_PARA, AOT_RSVD_28,
+ AOT_PS, AOT_MPEGS,
+ AOT_ESCAPE, AOT_MP3ONMP4_L1,
+ AOT_MP3ONMP4_L2, AOT_MP3ONMP4_L3,
+ AOT_RSVD_35, AOT_RSVD_36,
+ AOT_AAC_SLS, AOT_SLS,
+ AOT_ER_AAC_ELD, AOT_USAC,
+ AOT_SAOC, AOT_LD_MPEGS,
+ AOT_MP2_AAC_LC, AOT_MP2_SBR,
+ AOT_DRM_AAC, AOT_DRM_SBR,
+ AOT_DRM_MPEG_PS, AOT_DRM_SURROUND,
+ AOT_DRM_USAC};
+
+constexpr size_t kAudioObjectTypesSize = size(kAudioObjectTypes);
+
+constexpr int32_t kSbrRatios[] = {-1, 0, 1, 2};
+constexpr size_t kSbrRatiosSize = size(kSbrRatios);
+
+constexpr int32_t kBitRateModes[] = {
+ AACENC_BR_MODE_INVALID, AACENC_BR_MODE_CBR, AACENC_BR_MODE_VBR_1,
+ AACENC_BR_MODE_VBR_2, AACENC_BR_MODE_VBR_3, AACENC_BR_MODE_VBR_4,
+ AACENC_BR_MODE_VBR_5, AACENC_BR_MODE_FF, AACENC_BR_MODE_SFR};
+constexpr size_t kBitRateModesSize = size(kBitRateModes);
+
+constexpr int32_t kGranuleLengths[] = {120, 128, 240, 256, 480, 512, 1024};
+constexpr size_t kGranuleLengthsSize = size(kGranuleLengths);
+
+constexpr int32_t kChannelOrder[] = {CH_ORDER_MPEG, CH_ORDER_WAV};
+constexpr size_t kChannelOrderSize = size(kChannelOrder);
+
+constexpr int32_t kSignalingModes[] = {-1, 0, 1, 2, 3};
+constexpr size_t kSignalingModesSize = size(kSignalingModes);
+
+constexpr int32_t kAudioMuxVer[] = {-1, 0, 1, 2};
+constexpr size_t kAudioMuxVerSize = size(kAudioMuxVer);
+
+constexpr int32_t kSbrModes[] = {-1, 0, 1, 2};
+constexpr size_t kSbrModesSize = size(kSbrModes);
+
+constexpr AACENC_METADATA_DRC_PROFILE kMetaDataDrcProfiles[] = {
+ AACENC_METADATA_DRC_NONE, AACENC_METADATA_DRC_FILMSTANDARD,
+ AACENC_METADATA_DRC_FILMLIGHT, AACENC_METADATA_DRC_MUSICSTANDARD,
+ AACENC_METADATA_DRC_MUSICLIGHT, AACENC_METADATA_DRC_SPEECH,
+ AACENC_METADATA_DRC_NOT_PRESENT};
+constexpr size_t kMetaDataDrcProfilesSize = size(kMetaDataDrcProfiles);
+
+enum {
+ IDX_SBR_MODE = 0,
+ IDX_AAC_AOT,
+ IDX_SAMPLE_RATE,
+ IDX_BIT_RATE_1,
+ IDX_BIT_RATE_2,
+ IDX_BIT_RATE_3,
+ IDX_CHANNEL,
+ IDX_IDENTIFIER,
+ IDX_SBR_RATIO,
+ IDX_METADATA_DRC_PROFILE,
+ IDX_METADATA_COMP_PROFILE,
+ IDX_METADATA_DRC_TARGET_REF_LEVEL,
+ IDX_METADATA_COMP_TARGET_REF_LEVEL,
+ IDX_METADATA_PROG_LEVEL_PRESENT,
+ IDX_METADATA_PROG_LEVEL,
+ IDX_METADATA_PCE_MIXDOWN_IDX_PRESENT,
+ IDX_METADATA_ETSI_DMXLVL_PRESENT,
+ IDX_METADATA_CENTER_MIX_LEVEL,
+ IDX_METADATA_SURROUND_MIX_LEVEL,
+ IDX_METADATA_DOLBY_SURROUND_MODE,
+ IDX_METADATA_DRC_PRESENTATION_MODE,
+ IDX_METADATA_EXT_ANC_DATA_ENABLE,
+ IDX_METADATA_EXT_DOWNMIX_LEVEL_ENABLE,
+ IDX_METADATA_EXT_DOWNMIX_LEVEL_A,
+ IDX_METADATA_EXT_DOWNMIX_LEVEL_B,
+ IDX_METADATA_DMX_GAIN_ENABLE,
+ IDX_METADATA_DMX_GAIN_5,
+ IDX_METADATA_DMX_GAIN_2,
+ IDX_METADATA_LFE_DMX_ENABLE,
+ IDX_METADATA_LFE_DMX_LEVEL,
+ IDX_IN_BUFFER_INDEX_1,
+ IDX_IN_BUFFER_INDEX_2,
+ IDX_IN_BUFFER_INDEX_3,
+ IDX_BIT_RATE_MODE,
+ IDX_GRANULE_LENGTH,
+ IDX_CHANNELORDER,
+ IDX_AFTERBURNER,
+ IDX_BANDWIDTH,
+ IDX_PEAK_BITRATE,
+ IDX_HEADER_PERIOD,
+ IDX_SIGNALING_MODE,
+ IDX_TPSUBFRAMES,
+ IDX_AUDIOMUXVER,
+ IDX_PROTECTION,
+ IDX_ANCILLARY_BITRATE,
+ IDX_METADATA_MODE,
+ IDX_LAST
+};
+
+template <typename type1, typename type2, typename type3>
+auto generateNumberInRangeFromData(type1 data, type2 min, type3 max) -> decltype(max) {
+ return (data % (1 + max - min)) + min;
+}
+
+class Codec {
+ public:
+ ~Codec() { deInitEncoder(); }
+ bool initEncoder(uint8_t **dataPtr, size_t *sizePtr);
+ void encodeFrames(const uint8_t *data, size_t size);
+ void deInitEncoder();
+
+ private:
+ template <typename type1, typename type2, typename type3>
+ void setAACParam(type1 data, const AACENC_PARAM aacParam, type2 min, type2 max,
+ const type3 *array = nullptr);
+ void setupMetaData(uint8_t *data);
+
+ HANDLE_AACENCODER mEncoder = nullptr;
+ AACENC_MetaData mMetaData = {};
+ uint32_t mInBufferIdx_1 = 0;
+ uint32_t mInBufferIdx_2 = 0;
+ uint32_t mInBufferIdx_3 = 0;
+};
+
+void Codec::setupMetaData(uint8_t *data) {
+ uint32_t drcProfileIndex = generateNumberInRangeFromData(data[IDX_METADATA_DRC_PROFILE], 0,
+ kMetaDataDrcProfilesSize - 1);
+ AACENC_METADATA_DRC_PROFILE drcProfile = kMetaDataDrcProfiles[drcProfileIndex];
+ mMetaData.drc_profile = drcProfile;
+
+ uint32_t compProfileIndex = generateNumberInRangeFromData(data[IDX_METADATA_COMP_PROFILE], 0,
+ kMetaDataDrcProfilesSize - 1);
+ AACENC_METADATA_DRC_PROFILE compProfile = kMetaDataDrcProfiles[compProfileIndex];
+ mMetaData.comp_profile = compProfile;
+
+ INT drcTargetRefLevel =
+ generateNumberInRangeFromData(data[IDX_METADATA_DRC_TARGET_REF_LEVEL], 0, UINT8_MAX);
+ mMetaData.drc_TargetRefLevel = drcTargetRefLevel;
+
+ INT compTargetRefLevel =
+ generateNumberInRangeFromData(data[IDX_METADATA_COMP_TARGET_REF_LEVEL], 0, UINT8_MAX);
+ mMetaData.comp_TargetRefLevel = compTargetRefLevel;
+
+ INT isProgRefLevelPresent =
+ generateNumberInRangeFromData(data[IDX_METADATA_PROG_LEVEL_PRESENT], 0, 1);
+ mMetaData.prog_ref_level_present = isProgRefLevelPresent;
+
+ INT progRefLevel = generateNumberInRangeFromData(data[IDX_METADATA_PROG_LEVEL], 0, UINT8_MAX);
+ mMetaData.prog_ref_level = progRefLevel;
+
+ UCHAR isPCEMixdownIdxPresent =
+ generateNumberInRangeFromData(data[IDX_METADATA_PCE_MIXDOWN_IDX_PRESENT], 0, 1);
+ mMetaData.PCE_mixdown_idx_present = isPCEMixdownIdxPresent;
+
+ UCHAR isETSIDmxLvlPresent =
+ generateNumberInRangeFromData(data[IDX_METADATA_ETSI_DMXLVL_PRESENT], 0, 1);
+ mMetaData.ETSI_DmxLvl_present = isETSIDmxLvlPresent;
+
+ SCHAR centerMixLevel = generateNumberInRangeFromData(data[IDX_METADATA_CENTER_MIX_LEVEL], 0, 7);
+ mMetaData.centerMixLevel = centerMixLevel;
+
+ SCHAR surroundMixLevel =
+ generateNumberInRangeFromData(data[IDX_METADATA_SURROUND_MIX_LEVEL], 0, 7);
+ mMetaData.surroundMixLevel = surroundMixLevel;
+
+ UCHAR dolbySurroundMode =
+ generateNumberInRangeFromData(data[IDX_METADATA_DOLBY_SURROUND_MODE], 0, 2);
+ mMetaData.dolbySurroundMode = dolbySurroundMode;
+
+ UCHAR drcPresentationMode =
+ generateNumberInRangeFromData(data[IDX_METADATA_DRC_PRESENTATION_MODE], 0, 2);
+ mMetaData.drcPresentationMode = drcPresentationMode;
+
+ UCHAR extAncDataEnable =
+ generateNumberInRangeFromData(data[IDX_METADATA_EXT_ANC_DATA_ENABLE], 0, 1);
+ mMetaData.ExtMetaData.extAncDataEnable = extAncDataEnable;
+
+ UCHAR extDownmixLevelEnable =
+ generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_ENABLE], 0, 1);
+ mMetaData.ExtMetaData.extDownmixLevelEnable = extDownmixLevelEnable;
+
+ UCHAR extDownmixLevel_A =
+ generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_A], 0, 7);
+ mMetaData.ExtMetaData.extDownmixLevel_A = extDownmixLevel_A;
+
+ UCHAR extDownmixLevel_B =
+ generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_B], 0, 7);
+ mMetaData.ExtMetaData.extDownmixLevel_B = extDownmixLevel_B;
+
+ UCHAR dmxGainEnable = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_ENABLE], 0, 1);
+ mMetaData.ExtMetaData.dmxGainEnable = dmxGainEnable;
+
+ INT dmxGain5 = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_5], 0, UINT8_MAX);
+ mMetaData.ExtMetaData.dmxGain5 = dmxGain5;
+
+ INT dmxGain2 = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_2], 0, UINT8_MAX);
+ mMetaData.ExtMetaData.dmxGain2 = dmxGain2;
+
+ UCHAR lfeDmxEnable = generateNumberInRangeFromData(data[IDX_METADATA_LFE_DMX_ENABLE], 0, 1);
+ mMetaData.ExtMetaData.lfeDmxEnable = lfeDmxEnable;
+
+ UCHAR lfeDmxLevel = generateNumberInRangeFromData(data[IDX_METADATA_LFE_DMX_LEVEL], 0, 15);
+ mMetaData.ExtMetaData.lfeDmxLevel = lfeDmxLevel;
+}
+
+template <typename type1, typename type2, typename type3>
+void Codec::setAACParam(type1 data, const AACENC_PARAM aacParam, type2 min, type2 max,
+ const type3 *array) {
+ auto value = 0;
+ if (array) {
+ uint32_t index = generateNumberInRangeFromData(data, min, max);
+ value = array[index];
+ } else {
+ value = generateNumberInRangeFromData(data, min, max);
+ }
+ aacEncoder_SetParam(mEncoder, aacParam, value);
+ (void)aacEncoder_GetParam(mEncoder, aacParam);
+}
+
+bool Codec::initEncoder(uint8_t **dataPtr, size_t *sizePtr) {
+ uint8_t *data = *dataPtr;
+
+ if (AACENC_OK != aacEncOpen(&mEncoder, 0, 0)) {
+ return false;
+ }
+
+ setAACParam<uint8_t, size_t, int32_t>(data[IDX_SBR_MODE], AACENC_SBR_MODE, 0, kSbrModesSize - 1,
+ kSbrModes);
+
+ setAACParam<uint8_t, size_t, int32_t>(data[IDX_SBR_RATIO], AACENC_SBR_RATIO, 0,
+ kSbrRatiosSize - 1, kSbrRatios);
+
+ setAACParam<uint8_t, size_t, AUDIO_OBJECT_TYPE>(data[IDX_AAC_AOT], AACENC_AOT, 0,
+ kAudioObjectTypesSize - 1, kAudioObjectTypes);
+
+ setAACParam<uint8_t, size_t, int32_t>(data[IDX_SAMPLE_RATE], AACENC_SAMPLERATE, 0,
+ kSampleRatesSize - 1, kSampleRates);
+
+ uint32_t tempValue =
+ (data[IDX_BIT_RATE_1] << 16) | (data[IDX_BIT_RATE_2] << 8) | data[IDX_BIT_RATE_3];
+ setAACParam<uint8_t, uint32_t, uint32_t>(tempValue, AACENC_BITRATE, kMinBitRate, kMaxBitRate);
+
+ setAACParam<uint8_t, size_t, CHANNEL_MODE>(data[IDX_CHANNEL], AACENC_CHANNELMODE, 0,
+ kChannelModesSize - 1, kChannelModes);
+
+ setAACParam<uint8_t, size_t, TRANSPORT_TYPE>(data[IDX_IDENTIFIER], AACENC_TRANSMUX, 0,
+ kIdentifiersSize - 1, kIdentifiers);
+
+ setAACParam<uint8_t, size_t, int32_t>(data[IDX_BIT_RATE_MODE], AACENC_BITRATEMODE, 0,
+ kBitRateModesSize - 1, kBitRateModes);
+
+ setAACParam<uint8_t, size_t, int32_t>(data[IDX_GRANULE_LENGTH], AACENC_GRANULE_LENGTH, 0,
+ kGranuleLengthsSize - 1, kGranuleLengths);
+
+ setAACParam<uint8_t, size_t, int32_t>(data[IDX_CHANNELORDER], AACENC_CHANNELORDER, 0,
+ kChannelOrderSize - 1, kChannelOrder);
+
+ setAACParam<uint8_t, int32_t, int32_t>(data[IDX_AFTERBURNER], AACENC_AFTERBURNER, 0, 1);
+
+ setAACParam<uint8_t, int32_t, int32_t>(data[IDX_BANDWIDTH], AACENC_BANDWIDTH, 0, 1);
+
+ setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_PEAK_BITRATE], AACENC_PEAK_BITRATE,
+ kMinBitRate, kMinBitRate);
+
+ setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_HEADER_PERIOD], AACENC_HEADER_PERIOD, 0,
+ UINT8_MAX);
+
+ setAACParam<uint8_t, size_t, int32_t>(data[IDX_SIGNALING_MODE], AACENC_SIGNALING_MODE, 0,
+ kSignalingModesSize - 1, kSignalingModes);
+
+ setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_TPSUBFRAMES], AACENC_TPSUBFRAMES, 0,
+ UINT8_MAX);
+
+ setAACParam<uint8_t, size_t, int32_t>(data[IDX_AUDIOMUXVER], AACENC_AUDIOMUXVER, 0,
+ kAudioMuxVerSize - 1, kAudioMuxVer);
+
+ setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_PROTECTION], AACENC_PROTECTION, 0, 1);
+
+ setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_ANCILLARY_BITRATE], AACENC_ANCILLARY_BITRATE,
+ 0, kMaxBitRate);
+
+ setAACParam<uint8_t, uint32_t, uint32_t>(data[IDX_METADATA_MODE], AACENC_METADATA_MODE, 0, 3);
+
+ AACENC_InfoStruct encInfo;
+ aacEncInfo(mEncoder, &encInfo);
+
+ mInBufferIdx_1 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_1], 0, kMaxBuffers - 1);
+ mInBufferIdx_2 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_2], 0, kMaxBuffers - 1);
+ mInBufferIdx_3 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_3], 0, kMaxBuffers - 1);
+
+ setupMetaData(data);
+
+ // Not re-using the data which was used for configuration for encoding
+ *dataPtr += IDX_LAST;
+ *sizePtr -= IDX_LAST;
+
+ return true;
+}
+
+static void deleteBuffers(uint8_t **buffers, size_t size) {
+ for (size_t n = 0; n < size; ++n) {
+ delete[] buffers[n];
+ }
+ delete[] buffers;
+}
+
+void Codec::encodeFrames(const uint8_t *data, size_t size) {
+ uint8_t *audioData = (uint8_t *)data;
+ uint8_t *ancData = (uint8_t *)data;
+ size_t audioSize = size;
+ size_t ancSize = size;
+
+ while ((audioSize > 0) && (ancSize > 0)) {
+ AACENC_InArgs inargs;
+ memset(&inargs, 0, sizeof(inargs));
+ inargs.numInSamples = audioSize / sizeof(int16_t);
+ inargs.numAncBytes = ancSize;
+
+ void *buffers[] = {(void *)audioData, (void *)ancData, &mMetaData};
+ INT bufferIds[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA, IN_METADATA_SETUP};
+ INT bufferSizes[] = {static_cast<INT>(audioSize), static_cast<INT>(ancSize),
+ static_cast<INT>(sizeof(mMetaData))};
+ INT bufferElSizes[] = {sizeof(int16_t), sizeof(UCHAR), sizeof(AACENC_MetaData)};
+
+ void *inBuffer[kMaxBuffers] = {};
+ INT inBufferIds[kMaxBuffers] = {};
+ INT inBufferSize[kMaxBuffers] = {};
+ INT inBufferElSize[kMaxBuffers] = {};
+ for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) {
+ uint32_t Idxs[] = {mInBufferIdx_1, mInBufferIdx_2, mInBufferIdx_3};
+ inBuffer[buffer] = buffers[Idxs[buffer]];
+ inBufferIds[buffer] = bufferIds[Idxs[buffer]];
+ inBufferSize[buffer] = bufferSizes[Idxs[buffer]];
+ inBufferElSize[buffer] = bufferElSizes[Idxs[buffer]];
+ }
+
+ AACENC_BufDesc inBufDesc;
+ inBufDesc.numBufs = kMaxBuffers;
+ inBufDesc.bufs = (void **)&inBuffer;
+ inBufDesc.bufferIdentifiers = inBufferIds;
+ inBufDesc.bufSizes = inBufferSize;
+ inBufDesc.bufElSizes = inBufferElSize;
+
+ uint8_t **outPtrRef = new uint8_t *[kMaxBuffers];
+ for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) {
+ outPtrRef[buffer] = new uint8_t[kMaxOutputBufferSize];
+ }
+
+ void *outBuffer[kMaxBuffers];
+ INT outBufferIds[kMaxBuffers];
+ INT outBufferSize[kMaxBuffers];
+ INT outBufferElSize[kMaxBuffers];
+
+ for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) {
+ outBuffer[buffer] = outPtrRef[buffer];
+ outBufferIds[buffer] = OUT_BITSTREAM_DATA;
+ outBufferSize[buffer] = (INT)kMaxOutputBufferSize;
+ outBufferElSize[buffer] = sizeof(UCHAR);
+ }
+
+ AACENC_BufDesc outBufDesc;
+ outBufDesc.numBufs = kMaxBuffers;
+ outBufDesc.bufs = (void **)&outBuffer;
+ outBufDesc.bufferIdentifiers = outBufferIds;
+ outBufDesc.bufSizes = outBufferSize;
+ outBufDesc.bufElSizes = outBufferElSize;
+
+ AACENC_OutArgs outargs = {};
+ aacEncEncode(mEncoder, &inBufDesc, &outBufDesc, &inargs, &outargs);
+
+ if (outargs.numOutBytes == 0) {
+ if (audioSize > 0) {
+ ++audioData;
+ --audioSize;
+ }
+ if (ancSize > 0) {
+ ++ancData;
+ --ancSize;
+ }
+ } else {
+ size_t audioConsumed = outargs.numInSamples * sizeof(int16_t);
+ audioData += audioConsumed;
+ audioSize -= audioConsumed;
+
+ size_t ancConsumed = outargs.numAncBytes;
+ ancData += ancConsumed;
+ ancSize -= ancConsumed;
+ }
+ deleteBuffers(outPtrRef, kMaxBuffers);
+
+ // break out of loop if only metadata was sent in all the input buffers
+ // as sending it multiple times in a loop is redundant.
+ if ((mInBufferIdx_1 == kMaxBuffers - 1) && (mInBufferIdx_2 == kMaxBuffers - 1) &&
+ (mInBufferIdx_3 == kMaxBuffers - 1)) {
+ break;
+ }
+ }
+}
+
+void Codec::deInitEncoder() { aacEncClose(&mEncoder); }
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
+ if (size < IDX_LAST) {
+ return 0;
+ }
+ Codec encoder;
+ if (encoder.initEncoder(const_cast<uint8_t **>(&data), &size)) {
+ encoder.encodeFrames(data, size);
+ }
+ return 0;
+}
diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp
index a8dadc0..ca1a6a2 100644
--- a/libAACdec/src/usacdec_acelp.cpp
+++ b/libAACdec/src/usacdec_acelp.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -719,7 +719,7 @@ static void ConcealPitchLag(CAcelpStaticMem *acelp_mem, const int PIT_MAX,
UCHAR *pold_T0_frac = &acelp_mem->old_T0_frac;
if ((int)*pold_T0 >= PIT_MAX) {
- *pold_T0 = (UCHAR)(PIT_MAX - 5);
+ *pold_T0 = (USHORT)(PIT_MAX - 5);
}
*pT0 = (int)*pold_T0;
*pT0_frac = (int)*pold_T0_frac;
diff --git a/libFDK/include/nlc_dec.h b/libFDK/include/nlc_dec.h
index cca97f1..aded569 100644
--- a/libFDK/include/nlc_dec.h
+++ b/libFDK/include/nlc_dec.h
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -159,9 +159,6 @@ typedef enum {
#ifndef HUFFDEC_PARAMS
#define HUFFDEC_PARMS
-#define PAIR_SHIFT 4
-#define PAIR_MASK 0xf
-
#define MAX_ENTRIES 168
#define HANDLE_HUFF_NODE const SHORT(*)[MAX_ENTRIES][2]
diff --git a/libFDK/src/autocorr2nd.cpp b/libFDK/src/autocorr2nd.cpp
index 718a555..8c5673c 100644
--- a/libFDK/src/autocorr2nd.cpp
+++ b/libFDK/src/autocorr2nd.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -102,11 +102,6 @@ amm-info@iis.fraunhofer.de
#include "autocorr2nd.h"
-/* If the accumulator does not provide enough overflow bits,
- products have to be shifted down in the autocorrelation below. */
-#define SHIFT_FACTOR (5)
-#define SHIFT >> (SHIFT_FACTOR)
-
/*!
*
* \brief Calculate second order autocorrelation using 2 accumulators
@@ -126,45 +121,49 @@ INT autoCorr2nd_real(
const FIXP_DBL *realBuf = reBuffer;
+ const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)(len / 2)), 1);
/*
r11r,r22r
r01r,r12r
r02r
*/
pReBuf = realBuf - 2;
- accu5 = ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3]))
- SHIFT);
+ accu5 =
+ ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >>
+ len_scale);
pReBuf++;
/* len must be even */
- accu1 = fPow2Div2(pReBuf[0]) SHIFT;
- accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) SHIFT;
+ accu1 = fPow2Div2(pReBuf[0]) >> len_scale;
+ accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) >> len_scale;
pReBuf++;
for (j = (len - 2) >> 1; j != 0; j--, pReBuf += 2) {
- accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) SHIFT);
+ accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) >> len_scale);
- accu3 += ((fMultDiv2(pReBuf[0], pReBuf[1]) +
- fMultDiv2(pReBuf[1], pReBuf[2])) SHIFT);
+ accu3 +=
+ ((fMultDiv2(pReBuf[0], pReBuf[1]) + fMultDiv2(pReBuf[1], pReBuf[2])) >>
+ len_scale);
- accu5 += ((fMultDiv2(pReBuf[0], pReBuf[2]) +
- fMultDiv2(pReBuf[1], pReBuf[3])) SHIFT);
+ accu5 +=
+ ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) >>
+ len_scale);
}
- accu2 = (fPow2Div2(realBuf[-2]) SHIFT);
+ accu2 = (fPow2Div2(realBuf[-2]) >> len_scale);
accu2 += accu1;
- accu1 += (fPow2Div2(realBuf[len - 2]) SHIFT);
+ accu1 += (fPow2Div2(realBuf[len - 2]) >> len_scale);
- accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) SHIFT);
+ accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) >> len_scale);
accu4 += accu3;
- accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) SHIFT);
+ accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) >> len_scale);
mScale = CntLeadingZeros(
(accu1 | accu2 | fAbs(accu3) | fAbs(accu4) | fAbs(accu5))) -
1;
- autoCorrScaling = mScale - 1 - SHIFT_FACTOR; /* -1 because of fMultDiv2*/
+ autoCorrScaling = mScale - 1 - len_scale; /* -1 because of fMultDiv2*/
/* Scale to common scale factor */
ac->r11r = accu1 << mScale;
@@ -190,7 +189,7 @@ INT autoCorr2nd_cplx(
const FIXP_DBL *imBuffer, /*!< Pointer to imag part of input samples */
const int len /*!< Number of input samples (should be smaller than 128) */
) {
- int j, autoCorrScaling, mScale, len_scale;
+ int j, autoCorrScaling, mScale;
FIXP_DBL accu0, accu1, accu2, accu3, accu4, accu5, accu6, accu7, accu8;
@@ -199,7 +198,7 @@ INT autoCorr2nd_cplx(
const FIXP_DBL *realBuf = reBuffer;
const FIXP_DBL *imagBuf = imBuffer;
- (len > 64) ? (len_scale = 6) : (len_scale = 5);
+ const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)len), 1);
/*
r00r,
r11r,r22r
diff --git a/libFDK/src/nlc_dec.cpp b/libFDK/src/nlc_dec.cpp
index 6e98ce0..3733d98 100644
--- a/libFDK/src/nlc_dec.cpp
+++ b/libFDK/src/nlc_dec.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -568,12 +568,12 @@ bail:
static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
SCHAR* out_data_2, DATA_TYPE data_type,
DIFF_TYPE diff_type_1, DIFF_TYPE diff_type_2,
- int num_val, CODING_SCHEME* cdg_scheme, int ldMode) {
+ int num_val, PAIRING* pairing_scheme, int ldMode) {
ERROR_t err = HUFFDEC_OK;
+ CODING_SCHEME coding_scheme = HUFF_1D;
DIFF_TYPE diff_type;
int i = 0;
- ULONG data = 0;
SCHAR pair_vec[28][2];
@@ -596,15 +596,13 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
int hufYY;
/* Coding scheme */
- data = FDKreadBits(strm, 1);
- *cdg_scheme = (CODING_SCHEME)(data << PAIR_SHIFT);
+ coding_scheme = (CODING_SCHEME)FDKreadBits(strm, 1);
- if (*cdg_scheme >> PAIR_SHIFT == HUFF_2D) {
+ if (coding_scheme == HUFF_2D) {
if ((out_data_1 != NULL) && (out_data_2 != NULL) && (ldMode == 0)) {
- data = FDKreadBits(strm, 1);
- *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | data);
+ *pairing_scheme = (PAIRING)FDKreadBits(strm, 1);
} else {
- *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | FREQ_PAIR);
+ *pairing_scheme = FREQ_PAIR;
}
}
@@ -613,7 +611,7 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
hufYY2 = diff_type_2;
}
- switch (*cdg_scheme >> PAIR_SHIFT) {
+ switch (coding_scheme) {
case HUFF_1D:
p0_flag[0] = (diff_type_1 == DIFF_FREQ);
p0_flag[1] = (diff_type_2 == DIFF_FREQ);
@@ -634,7 +632,7 @@ static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
case HUFF_2D:
- switch (*cdg_scheme & PAIR_MASK) {
+ switch (*pairing_scheme) {
case FREQ_PAIR:
if (out_data_1 != NULL) {
@@ -843,7 +841,7 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
SCHAR* pDataVec[2] = {NULL, NULL};
DIFF_TYPE diff_type[2] = {DIFF_FREQ, DIFF_FREQ};
- CODING_SCHEME cdg_scheme = HUFF_1D;
+ PAIRING pairing = FREQ_PAIR;
DIRECTION direction = BACKWARDS;
switch (data_type) {
@@ -959,7 +957,7 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
}
/* Huffman decoding */
err = huff_decode(strm, pDataVec[0], pDataVec[1], data_type, diff_type[0],
- diff_type[1], dataBands, &cdg_scheme,
+ diff_type[1], dataBands, &pairing,
(DECODER == SAOC_DECODER));
if (err != HUFFDEC_OK) {
return HUFFDEC_NOTOK;
@@ -986,8 +984,8 @@ ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm,
}
}
- mixed_time_pair = (diff_type[0] != diff_type[1]) &&
- ((cdg_scheme & PAIR_MASK) == TIME_PAIR);
+ mixed_time_pair =
+ (diff_type[0] != diff_type[1]) && (pairing == TIME_PAIR);
if (direction == BACKWARDS) {
if (diff_type[0] == DIFF_FREQ) {
diff --git a/libMpegTPDec/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp
index e46cb32..8f77017 100644
--- a/libMpegTPDec/src/tpdec_asc.cpp
+++ b/libMpegTPDec/src/tpdec_asc.cpp
@@ -1694,8 +1694,7 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
const AUDIO_OBJECT_TYPE aot) {
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
- USAC_EXT_ELEMENT_TYPE usacExtElementType =
- (USAC_EXT_ELEMENT_TYPE)escapedValue(hBs, 4, 8, 16);
+ UINT usacExtElementType = escapedValue(hBs, 4, 8, 16);
/* recurve extension elements which are invalid for USAC */
if (aot == AOT_USAC) {
@@ -1712,7 +1711,6 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
}
}
- extElement->usacExtElementType = usacExtElementType;
int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16);
extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength;
INT bsAnchor;
@@ -1746,8 +1744,10 @@ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement,
}
} break;
default:
+ usacExtElementType = ID_EXT_ELE_UNKNOWN;
break;
}
+ extElement->usacExtElementType = (USAC_EXT_ELEMENT_TYPE)usacExtElementType;
/* Adjust bit stream position. This is required because of byte alignment and
* unhandled extensions. */
@@ -1776,7 +1776,7 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc,
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
int numConfigExtensions;
- CONFIG_EXT_ID usacConfigExtType;
+ UINT usacConfigExtType;
int usacConfigExtLength;
int loudnessInfoSetIndex =
-1; /* index of loudnessInfoSet config extension. -1 if not contained. */
@@ -1787,7 +1787,7 @@ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc,
for (int confExtIdx = 0; confExtIdx < numConfigExtensions; confExtIdx++) {
INT nbits;
int loudnessInfoSetConfigExtensionPosition = FDKgetValidBits(hBs);
- usacConfigExtType = (CONFIG_EXT_ID)escapedValue(hBs, 4, 8, 16);
+ usacConfigExtType = escapedValue(hBs, 4, 8, 16);
usacConfigExtLength = (int)escapedValue(hBs, 4, 8, 16);
/* Start bit position of config extension */
diff --git a/libPCMutils/src/pcmdmx_lib.cpp b/libPCMutils/src/pcmdmx_lib.cpp
index 2070dbc..fca12ce 100644
--- a/libPCMutils/src/pcmdmx_lib.cpp
+++ b/libPCMutils/src/pcmdmx_lib.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -494,13 +494,40 @@ static PCM_DMX_CHANNEL_MODE getChMode4Plain(
return plainChMode;
}
-static inline UINT getIdxSum(UCHAR numCh) {
- UINT result = 0;
- int i;
- for (i = 1; i < numCh; i += 1) {
- result += i;
+/** Validates the channel indices of all channels present in the bitstream.
+ * The channel indices have to be consecutive and unique for each audio channel
+ *type.
+ * @param [in] The total number of channels of the given configuration.
+ * @param [in] The total number of channels of the current audio channel type of
+ *the given configuration.
+ * @param [in] Audio channel type to be examined.
+ * @param [in] Array holding the corresponding channel types for each channel.
+ * @param [in] Array holding the corresponding channel type indices for each
+ *channel.
+ * @returns Returns 1 on success, returns 0 on error.
+ **/
+static UINT validateIndices(UINT numChannels, UINT numChannelsPlaneAndGrp,
+ AUDIO_CHANNEL_TYPE aChType,
+ const AUDIO_CHANNEL_TYPE channelType[],
+ const UCHAR channelIndices[]) {
+ for (UINT reqValue = 0; reqValue < numChannelsPlaneAndGrp; reqValue++) {
+ int found = FALSE;
+ for (UINT i = 0; i < numChannels; i++) {
+ if (channelType[i] == aChType) {
+ if (channelIndices[i] == reqValue) {
+ if (found == TRUE) {
+ return 0; /* Found channel index a second time */
+ } else {
+ found = TRUE; /* Found channel index */
+ }
+ }
+ }
+ }
+ if (found == FALSE) {
+ return 0; /* Did not find channel index */
+ }
}
- return result;
+ return 1; /* Successfully validated channel indices */
}
/** Evaluate a given channel configuration and extract a packed channel mode. In
@@ -523,7 +550,6 @@ static PCMDMX_ERROR getChannelMode(
UCHAR offsetTable[(8)], /* out */
PCM_DMX_CHANNEL_MODE *chMode /* out */
) {
- UINT idxSum[(3)][(4)];
UCHAR numCh[(3)][(4)];
UCHAR mapped[(8)];
PCM_DMX_SPEAKER_POSITION spkrPos[(8)];
@@ -538,7 +564,6 @@ static PCMDMX_ERROR getChannelMode(
FDK_ASSERT(chMode != NULL);
/* For details see ISO/IEC 13818-7:2005(E), 8.5.3 Channel configuration */
- FDKmemclear(idxSum, (3) * (4) * sizeof(UINT));
FDKmemclear(numCh, (3) * (4) * sizeof(UCHAR));
FDKmemclear(mapped, (8) * sizeof(UCHAR));
FDKmemclear(spkrPos, (8) * sizeof(PCM_DMX_SPEAKER_POSITION));
@@ -552,19 +577,22 @@ static PCMDMX_ERROR getChannelMode(
(channelType[ch] & 0x0F) - 1,
0); /* Assign all undefined channels (ACT_NONE) to front channels. */
numCh[channelType[ch] >> 4][chGrp] += 1;
- idxSum[channelType[ch] >> 4][chGrp] += channelIndices[ch];
}
- if (numChannels > TWO_CHANNEL) {
+
+ {
int chGrp;
/* Sanity check on the indices */
for (chGrp = 0; chGrp < (4); chGrp += 1) {
int plane;
for (plane = 0; plane < (3); plane += 1) {
- if (idxSum[plane][chGrp] != getIdxSum(numCh[plane][chGrp])) {
+ if (numCh[plane][chGrp] == 0) continue;
+ AUDIO_CHANNEL_TYPE aChType =
+ (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF));
+ if (!validateIndices(numChannels, numCh[plane][chGrp], aChType,
+ channelType, channelIndices)) {
unsigned idxCnt = 0;
for (ch = 0; ch < numChannels; ch += 1) {
- if (channelType[ch] ==
- (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF))) {
+ if (channelType[ch] == aChType) {
channelIndices[ch] = idxCnt++;
}
}
diff --git a/libSACdec/src/sac_bitdec.cpp b/libSACdec/src/sac_bitdec.cpp
index 4485ccf..25b3d9e 100644
--- a/libSACdec/src/sac_bitdec.cpp
+++ b/libSACdec/src/sac_bitdec.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -488,12 +488,17 @@ SACDEC_ERROR SpatialDecParseSpecificConfig(
pSpatialSpecificConfig->freqRes =
(SPATIALDEC_FREQ_RES)freqResTable_LD[bsFreqRes];
- pSpatialSpecificConfig->treeConfig =
- (SPATIALDEC_TREE_CONFIG)FDKreadBits(bitstream, 4);
+ {
+ UINT treeConfig = FDKreadBits(bitstream, 4);
- if (pSpatialSpecificConfig->treeConfig != SPATIALDEC_MODE_RSVD7) {
- err = MPS_UNSUPPORTED_CONFIG;
- goto bail;
+ switch (treeConfig) {
+ case SPATIALDEC_MODE_RSVD7:
+ pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG)treeConfig;
+ break;
+ default:
+ err = MPS_UNSUPPORTED_CONFIG;
+ goto bail;
+ }
}
{
diff --git a/libSACdec/src/sac_stp.cpp b/libSACdec/src/sac_stp.cpp
index b328c82..be332c7 100644
--- a/libSACdec/src/sac_stp.cpp
+++ b/libSACdec/src/sac_stp.cpp
@@ -252,12 +252,15 @@ inline void combineSignalCplxScale2(FIXP_DBL *hybOutputRealDry,
int n;
for (n = bands - 1; n >= 0; n--) {
- *hybOutputRealDry =
- *hybOutputRealDry +
- (fMultDiv2(*hybOutputRealWet, scaleX) << (SF_SCALE + 1));
- *hybOutputImagDry =
- *hybOutputImagDry +
- (fMultDiv2(*hybOutputImagWet, scaleX) << (SF_SCALE + 1));
+ *hybOutputRealDry = SATURATE_LEFT_SHIFT(
+ (*hybOutputRealDry >> 1) +
+ (fMultDiv2(*hybOutputRealWet, scaleX) << SF_SCALE),
+ 1, DFRACT_BITS);
+ *hybOutputImagDry = SATURATE_LEFT_SHIFT(
+ (*hybOutputImagDry >> 1) +
+ (fMultDiv2(*hybOutputImagWet, scaleX) << SF_SCALE),
+ 1, DFRACT_BITS);
+ ;
hybOutputRealDry++, hybOutputRealWet++;
hybOutputImagDry++, hybOutputImagWet++;
}
diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp
deleted file mode 100644
index db1948f..0000000
--- a/libSBRdec/src/arm/lpp_tran_arm.cpp
+++ /dev/null
@@ -1,159 +0,0 @@
-/* -----------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
-Forschung e.V. All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
-that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
-scheme for digital audio. This FDK AAC Codec software is intended to be used on
-a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
-general perceptual audio codecs. AAC-ELD is considered the best-performing
-full-bandwidth communications codec by independent studies and is widely
-deployed. AAC has been standardized by ISO and IEC as part of the MPEG
-specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including
-those of Fraunhofer) may be obtained through Via Licensing
-(www.vialicensing.com) or through the respective patent owners individually for
-the purpose of encoding or decoding bit streams in products that are compliant
-with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
-Android devices already license these patent claims through Via Licensing or
-directly from the patent owners, and therefore FDK AAC Codec software may
-already be covered under those patent licenses when it is used for those
-licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions
-with enhanced sound quality, are also available from Fraunhofer. Users are
-encouraged to check the Fraunhofer website for additional applications
-information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification,
-are permitted without payment of copyright license fees provided that you
-satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of
-the FDK AAC Codec or your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation
-and/or other materials provided with redistributions of the FDK AAC Codec or
-your modifications thereto in binary form. You must make available free of
-charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived
-from this library without prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute
-the FDK AAC Codec software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating
-that you changed the software and the date of any change. For modified versions
-of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
-must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
-AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
-limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
-Fraunhofer provides no warranty of patent non-infringement with respect to this
-software.
-
-You may use this FDK AAC Codec software or modifications thereto only for
-purposes that are authorized by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
-holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
-including but not limited to the implied warranties of merchantability and
-fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
-or consequential damages, including but not limited to procurement of substitute
-goods or services; loss of use, data, or profits, or business interruption,
-however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of
-this software, even if advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------ */
-
-/**************************** SBR decoder library ******************************
-
- Author(s): Arthur Tritthart
-
- Description: (ARM optimised) LPP transposer subroutines
-
-*******************************************************************************/
-
-#if defined(__arm__)
-
-#define FUNCTION_LPPTRANSPOSER_func1
-
-#ifdef FUNCTION_LPPTRANSPOSER_func1
-
-/* Note: This code requires only 43 cycles per iteration instead of 61 on
- * ARM926EJ-S */
-static void lppTransposer_func1(FIXP_DBL *lowBandReal, FIXP_DBL *lowBandImag,
- FIXP_DBL **qmfBufferReal,
- FIXP_DBL **qmfBufferImag, int loops, int hiBand,
- int dynamicScale, int descale, FIXP_SGL a0r,
- FIXP_SGL a0i, FIXP_SGL a1r, FIXP_SGL a1i,
- const int fPreWhitening,
- FIXP_DBL preWhiteningGain,
- int preWhiteningGains_sf) {
- FIXP_DBL real1, real2, imag1, imag2, accu1, accu2;
-
- real2 = lowBandReal[-2];
- real1 = lowBandReal[-1];
- imag2 = lowBandImag[-2];
- imag1 = lowBandImag[-1];
- for (int i = 0; i < loops; i++) {
- accu1 = fMultDiv2(a0r, real1);
- accu2 = fMultDiv2(a0i, imag1);
- accu1 = fMultAddDiv2(accu1, a1r, real2);
- accu2 = fMultAddDiv2(accu2, a1i, imag2);
- real2 = fMultDiv2(a1i, real2);
- accu1 = accu1 - accu2;
- accu1 = accu1 >> dynamicScale;
-
- accu2 = fMultAddDiv2(real2, a1r, imag2);
- real2 = real1;
- imag2 = imag1;
- accu2 = fMultAddDiv2(accu2, a0i, real1);
- real1 = lowBandReal[i];
- accu2 = fMultAddDiv2(accu2, a0r, imag1);
- imag1 = lowBandImag[i];
- accu2 = accu2 >> dynamicScale;
-
- accu1 <<= 1;
- accu2 <<= 1;
- accu1 += (real1 >> descale);
- accu2 += (imag1 >> descale);
- if (fPreWhitening) {
- accu1 = scaleValueSaturate(fMultDiv2(accu1, preWhiteningGain),
- preWhiteningGains_sf);
- accu2 = scaleValueSaturate(fMultDiv2(accu2, preWhiteningGain),
- preWhiteningGains_sf);
- }
- qmfBufferReal[i][hiBand] = accu1;
- qmfBufferImag[i][hiBand] = accu2;
- }
-}
-#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */
-
-#endif /* __arm__ */
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
index 93e1158..113b1de 100644
--- a/libSBRdec/src/lpp_tran.cpp
+++ b/libSBRdec/src/lpp_tran.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -132,10 +132,6 @@ amm-info@iis.fraunhofer.de
#include "HFgen_preFlat.h"
-#if defined(__arm__)
-#include "arm/lpp_tran_arm.cpp"
-#endif
-
#define LPC_SCALE_FACTOR 2
/*!
@@ -220,19 +216,21 @@ static inline void calc_qmfBufferReal(FIXP_DBL **qmfBufferReal,
const FIXP_DBL *const lowBandReal,
const int startSample,
const int stopSample, const UCHAR hiBand,
- const int dynamicScale, const int descale,
+ const int dynamicScale,
const FIXP_SGL a0r, const FIXP_SGL a1r) {
- FIXP_DBL accu1, accu2;
- int i;
+ const int dynscale = fixMax(0, dynamicScale - 1) + 1;
+ const int rescale = -fixMin(0, dynamicScale - 1) + 1;
+ const int descale =
+ fixMin(DFRACT_BITS - 1, LPC_SCALE_FACTOR + dynamicScale + rescale);
+
+ for (int i = 0; i < stopSample - startSample; i++) {
+ FIXP_DBL accu;
- for (i = 0; i < stopSample - startSample; i++) {
- accu1 = fMultDiv2(a1r, lowBandReal[i]);
- accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1);
- accu1 = accu1 >> dynamicScale;
+ accu = fMultDiv2(a1r, lowBandReal[i]) + fMultDiv2(a0r, lowBandReal[i + 1]);
+ accu = (lowBandReal[i + 2] >> descale) + (accu >> dynscale);
- accu1 <<= 1;
- accu2 = (lowBandReal[i + 2] >> descale);
- qmfBufferReal[i + startSample][hiBand] = accu1 + accu2;
+ qmfBufferReal[i + startSample][hiBand] =
+ SATURATE_LEFT_SHIFT(accu, rescale, DFRACT_BITS);
}
}
@@ -771,52 +769,50 @@ void lppTransposer(
} else { /* bw <= 0 */
if (!useLP) {
- int descale =
- fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale));
-#ifdef FUNCTION_LPPTRANSPOSER_func1
- lppTransposer_func1(
- lowBandReal + LPC_ORDER + startSample,
- lowBandImag + LPC_ORDER + startSample,
- qmfBufferReal + startSample, qmfBufferImag + startSample,
- stopSample - startSample, (int)hiBand, dynamicScale, descale, a0r,
- a0i, a1r, a1i, fPreWhitening, preWhiteningGains[loBand],
- preWhiteningGains_exp[loBand] + 1);
-#else
+ const int dynscale = fixMax(0, dynamicScale - 2) + 1;
+ const int rescale = -fixMin(0, dynamicScale - 2) + 1;
+ const int descale = fixMin(DFRACT_BITS - 1,
+ LPC_SCALE_FACTOR + dynamicScale + rescale);
+
for (i = startSample; i < stopSample; i++) {
FIXP_DBL accu1, accu2;
- accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) -
- fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) +
- fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) -
- fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >>
- dynamicScale;
- accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) +
- fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) +
- fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) +
- fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >>
- dynamicScale;
-
- accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);
- accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);
+ accu1 = ((fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) -
+ fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1])) >>
+ 1) +
+ ((fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) -
+ fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >>
+ 1);
+ accu2 = ((fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) +
+ fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1])) >>
+ 1) +
+ ((fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) +
+ fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >>
+ 1);
+
+ accu1 =
+ (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 >> dynscale);
+ accu2 =
+ (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 >> dynscale);
if (fPreWhitening) {
- accu1 = scaleValueSaturate(
+ qmfBufferReal[i][hiBand] = scaleValueSaturate(
fMultDiv2(accu1, preWhiteningGains[loBand]),
- preWhiteningGains_exp[loBand] + 1);
- accu2 = scaleValueSaturate(
+ preWhiteningGains_exp[loBand] + 1 + rescale);
+ qmfBufferImag[i][hiBand] = scaleValueSaturate(
fMultDiv2(accu2, preWhiteningGains[loBand]),
- preWhiteningGains_exp[loBand] + 1);
+ preWhiteningGains_exp[loBand] + 1 + rescale);
+ } else {
+ qmfBufferReal[i][hiBand] =
+ SATURATE_LEFT_SHIFT(accu1, rescale, DFRACT_BITS);
+ qmfBufferImag[i][hiBand] =
+ SATURATE_LEFT_SHIFT(accu2, rescale, DFRACT_BITS);
}
- qmfBufferReal[i][hiBand] = accu1;
- qmfBufferImag[i][hiBand] = accu2;
}
-#endif
} else {
FDK_ASSERT(dynamicScale >= 0);
calc_qmfBufferReal(
qmfBufferReal, &(lowBandReal[LPC_ORDER + startSample - 2]),
- startSample, stopSample, hiBand, dynamicScale,
- fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r,
- a1r);
+ startSample, stopSample, hiBand, dynamicScale, a0r, a1r);
}
} /* bw <= 0 */
diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp
index b1fb0da..919e9bb 100644
--- a/libSBRdec/src/sbr_dec.cpp
+++ b/libSBRdec/src/sbr_dec.cpp
@@ -1,7 +1,7 @@
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
-© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
@@ -713,7 +713,8 @@ void sbr_dec(
} else { /* (flags & SBRDEC_PS_DECODED) */
INT sdiff;
- INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
+ INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov,
+ outScalefactor, outScalefactorR, outScalefactorL;
HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->qmfDomainOutCh->fb;
HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->qmfDomainOutCh->fb;
@@ -744,7 +745,7 @@ void sbr_dec(
*/
FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <=
QMF_MAX_SYNTHESIS_BANDS);
- qmfChangeOutScalefactor(synQmfRight, -(8));
+ synQmfRight->outScalefactor = synQmf->outScalefactor;
FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates,
9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis *
sizeof(FIXP_QSS));
@@ -788,9 +789,11 @@ void sbr_dec(
FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel,
sizeof(SBRDEC_DRC_CHANNEL));
- for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
+ outScalefactor = maxShift - (8);
+ outScalefactorL = outScalefactorR =
+ sbrInDataHeadroom + 1; /* +1: psDiffScale! (MPEG-PS) */
- INT outScalefactorR, outScalefactorL;
+ for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
/* qmf timeslot of right channel */
FIXP_DBL *rQmfReal = pWorkBuffer;
@@ -815,27 +818,20 @@ void sbr_dec(
? scaleFactorLowBand_ov
: scaleFactorLowBand_no_ov,
scaleFactorHighBand, synQmf->lsb, synQmf->usb);
-
- outScalefactorL = outScalefactorR =
- 1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */
}
sbrDecoder_drcApplySlot(/* right channel */
&hSbrDecRight->sbrDrcChannel, rQmfReal,
rQmfImag, i, synQmfRight->no_col, maxShift);
- outScalefactorR += maxShift;
-
sbrDecoder_drcApplySlot(/* left channel */
&hSbrDec->sbrDrcChannel, *(pLowBandReal + i),
*(pLowBandImag + i), i, synQmf->no_col,
maxShift);
- outScalefactorL += maxShift;
-
if (!(flags & SBRDEC_SKIP_QMF_SYN)) {
- qmfChangeOutScalefactor(synQmf, -(8));
- qmfChangeOutScalefactor(synQmfRight, -(8));
+ qmfChangeOutScalefactor(synQmf, outScalefactor);
+ qmfChangeOutScalefactor(synQmfRight, outScalefactor);
qmfSynthesisFilteringSlot(
synQmfRight, rQmfReal, /* QMF real buffer */
generated by cgit v1.2.3 (git 2.39.1) at 2025-01-06 00:12:09 +0000