Mark ab/7061308 as merged in stage.
Bug: 180401296
Merged-In: Id04d2590c249cad13da8dd8975d0a12ce96a9597
Change-Id: I4da99b78b01817308d8572d3c0d5b224b7d36a73
diff --git a/Android.bp b/Android.bp
index 555223b..0c67186 100644
--- a/Android.bp
+++ b/Android.bp
@@ -1,3 +1,36 @@
+// *** 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,
@@ -32,8 +65,7 @@
"signed-integer-overflow",
"bounds",
],
- // Enable CFI if this becomes a shared library.
- // cfi: true,
+ cfi: true,
},
shared_libs: [
"liblog",
diff --git a/fuzzer/Android.bp b/fuzzer/Android.bp
index 28a21b9..6739798 100644
--- a/fuzzer/Android.bp
+++ b/fuzzer/Android.bp
@@ -18,17 +18,21 @@
* Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore
*/
-cc_fuzz {
- name: "aac_dec_fuzzer",
- host_supported:true,
+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",
- "liblog",
- ],
-
- srcs: [
- "aac_dec_fuzzer.cpp",
],
target: {
@@ -44,3 +48,35 @@
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
index d99bc75..b8cc260 100644
--- a/fuzzer/README.md
+++ b/fuzzer/README.md
@@ -53,7 +53,98 @@
$ $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_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/libAACenc/src/aacenc.cpp b/libAACenc/src/aacenc.cpp
index b6f733d..1af8a2e 100644
--- a/libAACenc/src/aacenc.cpp
+++ b/libAACenc/src/aacenc.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
@@ -245,6 +245,46 @@
return bitrate;
}
+/*-----------------------------------------------------------------------------
+
+ functionname: FDKaacEnc_AdjustVBRBitrateMode
+ description: Adjust bitrate mode to given bitrate parameter
+ input params: int vbrQuality (VBR0, VBR1, VBR2)
+ bitrate
+ channelMode
+ returns: vbr bitrate mode
+
+ ------------------------------------------------------------------------------*/
+AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode(
+ AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode) {
+ AACENC_BITRATE_MODE newBitrateMode = bitrateMode;
+
+ if (bitrate != -1) {
+ const INT monoStereoMode =
+ (FDKaacEnc_GetMonoStereoMode(channelMode) == EL_MODE_STEREO) ? 1 : 0;
+ const INT nChannelsEff =
+ FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff;
+ newBitrateMode = AACENC_BR_MODE_INVALID;
+
+ for (int idx = (int)(sizeof(configTabVBR) / sizeof(*configTabVBR)) - 1;
+ idx >= 0; idx--) {
+ if (bitrate >=
+ configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff) {
+ if (configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff <
+ FDKaacEnc_GetVBRBitrate(bitrateMode, channelMode)) {
+ newBitrateMode = configTabVBR[idx].bitrateMode;
+ } else {
+ newBitrateMode = bitrateMode;
+ }
+ break;
+ }
+ }
+ }
+
+ return AACENC_BR_MODE_IS_VBR(newBitrateMode) ? newBitrateMode
+ : AACENC_BR_MODE_INVALID;
+}
+
/**
* \brief Convert encoder bitreservoir value for transport library.
*
@@ -397,7 +437,6 @@
FIXP_DBL mbfac, bw_ratio;
QC_INIT qcInit;
INT averageBitsPerFrame = 0;
- int bitresMin = 0; /* the bitreservoir is always big for AAC-LC */
const CHANNEL_MODE prevChannelMode = hAacEnc->encoderMode;
if (config == NULL) return AAC_ENC_INVALID_HANDLE;
@@ -553,7 +592,6 @@
qcInit.minBits = fixMin(qcInit.minBits, averageBitsPerFrame & ~7);
} else {
INT bitreservoir = -1; /* default bitreservoir size*/
- bitresMin = BITRES_MIN;
if (isLowDelay(config->audioObjectType)) {
INT brPerChannel = config->bitRate / config->nChannels;
brPerChannel = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel));
@@ -567,7 +605,6 @@
bitreservoir = fMultI(slope, (INT)(BITRES_MAX_LD - BITRES_MIN_LD)) +
BITRES_MIN_LD; /* interpolate */
bitreservoir = bitreservoir & ~7; /* align to bytes */
- bitresMin = BITRES_MIN_LD;
}
int maxBitres;
@@ -604,7 +641,8 @@
qcInit.nSubFrames = config->nSubFrames;
qcInit.padding.paddingRest = config->sampleRate;
- if (qcInit.maxBits - qcInit.averageBits >= bitresMin * config->nChannels) {
+ if (qcInit.maxBits - qcInit.averageBits >=
+ ((qcInit.isLowDelay) ? BITRES_MIN_LD : BITRES_MIN) * config->nChannels) {
qcInit.bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */
} else if (qcInit.maxBits > qcInit.averageBits) {
qcInit.bitResMode = AACENC_BR_MODE_REDUCED; /* reduced bitreservoir */
diff --git a/libAACenc/src/aacenc.h b/libAACenc/src/aacenc.h
index 0e0d8c1..b7e0ef2 100644
--- a/libAACenc/src/aacenc.h
+++ b/libAACenc/src/aacenc.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
@@ -335,6 +335,19 @@
/*-----------------------------------------------------------------------------
+ functionname: FDKaacEnc_AdjustVBRBitrateMode
+ description: Adjust bitrate mode to given bitrate parameter
+ input params: int vbrQuality (VBR0, VBR1, VBR2)
+ bitrate
+ channelMode
+ returns: vbr bitrate mode
+
+ ------------------------------------------------------------------------------*/
+AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode(
+ AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode);
+
+/*-----------------------------------------------------------------------------
+
functionname: FDKaacEnc_AacInitDefaultConfig
description: gives reasonable default configuration
returns: ---
diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp
index c3977f3..0ae329b 100644
--- a/libAACenc/src/aacenc_lib.cpp
+++ b/libAACenc/src/aacenc_lib.cpp
@@ -1028,6 +1028,13 @@
case AACENC_BR_MODE_VBR_3:
case AACENC_BR_MODE_VBR_4:
case AACENC_BR_MODE_VBR_5:
+ /* Adjust bitrate mode in case given peak bitrate is lower than expected
+ * VBR bitrate. */
+ if ((INT)config->userPeakBitrate != -1) {
+ hAacConfig->bitrateMode = FDKaacEnc_AdjustVBRBitrateMode(
+ hAacConfig->bitrateMode, config->userPeakBitrate,
+ hAacConfig->channelMode);
+ }
/* Get bitrate in VBR configuration */
/* In VBR mode; SBR-modul depends on bitrate, core encoder on bitrateMode.
*/
diff --git a/libAACenc/src/qc_main.cpp b/libAACenc/src/qc_main.cpp
index bcfaa23..9a42550 100644
--- a/libAACenc/src/qc_main.cpp
+++ b/libAACenc/src/qc_main.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
@@ -373,13 +373,8 @@
hQC->invQuant = init->invQuant;
hQC->maxIterations = init->maxIterations;
- if (isConstantBitrateMode(hQC->bitrateMode)) {
- /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir
- */
- hQC->bitResMode = init->bitResMode;
- } else {
- hQC->bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */
- }
+ /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */
+ hQC->bitResMode = init->bitResMode;
hQC->padding.paddingRest = init->padding.paddingRest;
@@ -800,10 +795,15 @@
INT avgTotalDynBits = 0; /* maximal allowed dynamic bits for all frames */
INT totalAvailableBits = 0;
INT nSubFrames = 1;
+ const INT isCBRAdjustment = (isConstantBitrateMode(hQC->bitrateMode) ||
+ (hQC->bitResMode != AACENC_BR_MODE_FULL))
+ ? 1
+ : 0;
/*-------------------------------------------- */
/* redistribute total bitreservoir to elements */
- ErrorStatus = FDKaacEnc_BitResRedistribution(hQC, cm, avgTotalBits);
+ ErrorStatus = FDKaacEnc_BitResRedistribution(
+ hQC, cm, (isCBRAdjustment == 0) ? hQC->maxBitsPerFrame : avgTotalBits);
if (ErrorStatus != AAC_ENC_OK) {
return ErrorStatus;
}
@@ -831,33 +831,22 @@
/*-------------------------------------------- */
/*-------------------------------------------- */
- if (isConstantBitrateMode(hQC->bitrateMode)) {
- /* calc granted dynamic bits for sub frame and
- distribute it to each element */
- ErrorStatus = FDKaacEnc_prepareBitDistribution(
- hQC, psyOut, qcOut, cm, qcElement, avgTotalBits, &totalAvailableBits,
- &avgTotalDynBits);
+ /* calc granted dynamic bits for sub frame and
+ distribute it to each element */
+ ErrorStatus = FDKaacEnc_prepareBitDistribution(
+ hQC, psyOut, qcOut, cm, qcElement,
+ (isCBRAdjustment == 0) ? hQC->maxBitsPerFrame : avgTotalBits,
+ &totalAvailableBits, &avgTotalDynBits);
- if (ErrorStatus != AAC_ENC_OK) {
- return ErrorStatus;
- }
- } else {
- qcOut[0]->grantedDynBits =
- ((hQC->maxBitsPerFrame - (hQC->globHdrBits)) & ~7) -
- (qcOut[0]->globalExtBits + qcOut[0]->staticBits +
- qcOut[0]->elementExtBits);
- qcOut[0]->maxDynBits = qcOut[0]->grantedDynBits;
-
- totalAvailableBits = hQC->maxBitsPerFrame;
- avgTotalDynBits = 0;
+ if (ErrorStatus != AAC_ENC_OK) {
+ return ErrorStatus;
}
/* for ( all sub frames ) ... */
for (c = 0; c < nSubFrames; c++) {
/* for CBR and VBR mode */
FDKaacEnc_AdjustThresholds(hQC->hAdjThr, qcElement[c], qcOut[c],
- psyOut[c]->psyOutElement,
- isConstantBitrateMode(hQC->bitrateMode), cm);
+ psyOut[c]->psyOutElement, isCBRAdjustment, cm);
} /* -end- sub frame counter */
