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Commit 76327e6a authored by Treehugger Robot's avatar Treehugger Robot Committed by Automerger Merge Worker
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Merge "Add a fuzzer for RecordBufferConverter." am: 4c22a5dd am: d9003e98... 

Merge "Add a fuzzer for RecordBufferConverter." am: 4c22a5dd am: d9003e98 am: 9e54b915 am: 2d60aa43

Original change: https://android-review.googlesource.com/c/platform/frameworks/av/+/1489021

Change-Id: Iae1ffde99ce064b85f7b8a19a9f6cdbeb3d10229
parents c14604e3 2d60aa43

media/libaudioprocessing/tests/fuzzer/Android.bp

+11 −0
Original line number Diff line number Diff line
@@ -8,3 +8,14 @@ cc_fuzz { 
    "libsndfile",

  ],

}



cc_fuzz {

  name: "libaudioprocessing_record_buffer_converter_fuzzer",

  srcs: [

    "libaudioprocessing_record_buffer_converter_fuzzer.cpp",

  ],

  defaults: ["libaudioprocessing_test_defaults"],

  static_libs: [

    "libsndfile",

  ],

}

media/libaudioprocessing/tests/fuzzer/libaudioprocessing_fuzz_utils.h

0 → 100644
+67 −0
Original line number Diff line number Diff line 
/*

 * Copyright (C) 2021 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.

 */



#ifndef ANDROID_LIBAUDIOPROCESSING_FUZZ_UTILS_H

#define ANDROID_LIBAUDIOPROCESSING_FUZZ_UTILS_H



#include <media/AudioBufferProvider.h>

#include <system/audio.h>



namespace android {



class Provider : public AudioBufferProvider {

  const void* mAddr;        // base address

  const size_t mNumFrames;  // total frames

  const size_t mFrameSize;  // size of each frame in bytes

  size_t mNextFrame;        // index of next frame to provide

  size_t mUnrel;            // number of frames not yet released

 public:

  Provider(const void* addr, size_t frames, size_t frameSize)

      : mAddr(addr),

        mNumFrames(frames),

        mFrameSize(frameSize),

        mNextFrame(0),

        mUnrel(0) {}

  status_t getNextBuffer(Buffer* buffer) override {

    if (buffer->frameCount > mNumFrames - mNextFrame) {

      buffer->frameCount = mNumFrames - mNextFrame;

    }

    mUnrel = buffer->frameCount;

    if (buffer->frameCount > 0) {

      buffer->raw = (char*)mAddr + mFrameSize * mNextFrame;

      return NO_ERROR;

    } else {

      buffer->raw = nullptr;

      return NOT_ENOUGH_DATA;

    }

  }

  void releaseBuffer(Buffer* buffer) override {

    if (buffer->frameCount > mUnrel) {

      mNextFrame += mUnrel;

      mUnrel = 0;

    } else {

      mNextFrame += buffer->frameCount;

      mUnrel -= buffer->frameCount;

    }

    buffer->frameCount = 0;

    buffer->raw = nullptr;

  }

  void reset() { mNextFrame = 0; }

};



} // namespace android



#endif // ANDROID_LIBAUDIOPROCESSING_FUZZ_UTILS_H

media/libaudioprocessing/tests/fuzzer/libaudioprocessing_record_buffer_converter_fuzzer.cpp

0 → 100644
+177 −0
Original line number Diff line number Diff line 
/*

 * 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.

 */



#include "libaudioprocessing_fuzz_utils.h"

#include "fuzzer/FuzzedDataProvider.h"

#include <media/AudioResampler.h>

#include <media/RecordBufferConverter.h>

#include <stddef.h>

#include <stdint.h>



using namespace android;



constexpr int MAX_FRAMES = 1024;



#define AUDIO_FORMAT_PCM_MAIN 0



// Copied and simplified from audio-hal-enums.h?l=571

constexpr uint32_t FUZZ_AUDIO_FORMATS[] = {

  AUDIO_FORMAT_PCM_MAIN | AUDIO_FORMAT_PCM_SUB_16_BIT,

  AUDIO_FORMAT_PCM_MAIN | AUDIO_FORMAT_PCM_SUB_8_BIT,

  AUDIO_FORMAT_PCM_MAIN | AUDIO_FORMAT_PCM_SUB_32_BIT,

  AUDIO_FORMAT_PCM_MAIN | AUDIO_FORMAT_PCM_SUB_8_24_BIT,

  AUDIO_FORMAT_PCM_MAIN | AUDIO_FORMAT_PCM_SUB_FLOAT,

  AUDIO_FORMAT_PCM_MAIN | AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED,

  0x01000000u,

  0x02000000u,

  0x03000000u,

  0x04000000u,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_MAIN,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_LC,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_SSR,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_LTP,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_HE_V1,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_SCALABLE,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_ERLC,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_LD,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_HE_V2,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_ELD,

  AUDIO_FORMAT_AAC | AUDIO_FORMAT_AAC_SUB_XHE,

  0x05000000u,

  0x06000000u,

  0x07000000u,

  0x08000000u,

  0x09000000u,

  0x0A000000u,

  AUDIO_FORMAT_E_AC3 | AUDIO_FORMAT_E_AC3_SUB_JOC,

  0x0B000000u,

  0x0C000000u,

  0x0D000000u,

  0x0E000000u,

  0x10000000u,

  0x11000000u,

  0x12000000u,

  0x13000000u,

  0x14000000u,

  0x15000000u,

  0x16000000u,

  0x17000000u,

  0x18000000u,

  0x19000000u,

  0x1A000000u,

  0x1B000000u,

  0x1C000000u,

  0x1D000000u,

  0x1E000000u,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_MAIN,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_LC,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_SSR,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_LTP,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_HE_V1,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_SCALABLE,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_ERLC,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_LD,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_HE_V2,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_ELD,

  AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_XHE,

  0x1F000000u,

  0x20000000u,

  0x21000000u,

  0x22000000u,

  0x23000000u,

  0x24000000u,

  AUDIO_FORMAT_MAT | AUDIO_FORMAT_MAT_SUB_1_0,

  AUDIO_FORMAT_MAT | AUDIO_FORMAT_MAT_SUB_2_0,

  AUDIO_FORMAT_MAT | AUDIO_FORMAT_MAT_SUB_2_1,

  0x25000000u,

  AUDIO_FORMAT_AAC_LATM | AUDIO_FORMAT_AAC_SUB_LC,

  AUDIO_FORMAT_AAC_LATM | AUDIO_FORMAT_AAC_SUB_HE_V1,

  AUDIO_FORMAT_AAC_LATM | AUDIO_FORMAT_AAC_SUB_HE_V2,

  0x26000000u,

  0x27000000u,

  0x28000000u,

  0x29000000u,

  0x2A000000u,

  0x2B000000u,

  0xFFFFFFFFu,

  AUDIO_FORMAT_PCM_MAIN,

  AUDIO_FORMAT_PCM,

};

constexpr size_t NUM_AUDIO_FORMATS = std::size(FUZZ_AUDIO_FORMATS);



extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {

  FuzzedDataProvider fdp(data, size);

  fdp.ConsumeIntegral<int>();



  const audio_channel_mask_t srcChannelMask = (audio_channel_mask_t)fdp.ConsumeIntegral<int>();

  const audio_format_t srcFormat =

      (audio_format_t)FUZZ_AUDIO_FORMATS[fdp.ConsumeIntegralInRange<int>(0, NUM_AUDIO_FORMATS - 1)];

  const uint32_t srcSampleRate = fdp.ConsumeIntegralInRange<int>(1, 0x7fffffff);

  const audio_channel_mask_t dstChannelMask = (audio_channel_mask_t)fdp.ConsumeIntegral<int>();

  const audio_format_t dstFormat =

      (audio_format_t)FUZZ_AUDIO_FORMATS[fdp.ConsumeIntegralInRange<int>(0, NUM_AUDIO_FORMATS - 1)];

  const uint32_t dstSampleRate = fdp.ConsumeIntegralInRange<int>(1, 0x7fffffff);



  // Certain formats will result in LOG_ALWAYS_FATAL errors that aren't interesting crashes

  // for fuzzing.  Don't use those ones.

  const uint32_t dstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);

  constexpr android::AudioResampler::src_quality quality =

      android::AudioResampler::DEFAULT_QUALITY;

  const int maxChannels =

      quality < android::AudioResampler::DYN_LOW_QUALITY ? 2 : 8;

  if (dstChannelCount < 1 || dstChannelCount > maxChannels) {

    return 0;

  }



  const uint32_t srcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);

  if (srcChannelCount < 1 || srcChannelCount > maxChannels) {

    return 0;

  }



  RecordBufferConverter converter(srcChannelMask, srcFormat, srcSampleRate,

                                  dstChannelMask, dstFormat, dstSampleRate);

  if (converter.initCheck() != NO_ERROR) {

    return 0;

  }



  const uint32_t srcFrameSize = srcChannelCount * audio_bytes_per_sample(srcFormat);

  const int srcNumFrames = fdp.ConsumeIntegralInRange<int>(0, MAX_FRAMES);

  constexpr size_t metadataSize = 2 + 3 * sizeof(int) + 2 * sizeof(float);

  std::vector<uint8_t> inputData = fdp.ConsumeBytes<uint8_t>(

      metadataSize + (srcFrameSize * srcNumFrames));

  Provider provider(inputData.data(), srcNumFrames, srcFrameSize);



  const uint32_t dstFrameSize = dstChannelCount * audio_bytes_per_sample(dstFormat);

  const size_t frames = fdp.ConsumeIntegralInRange<size_t>(0, MAX_FRAMES + 1);

  int8_t dst[dstFrameSize * frames];

  memset(dst, 0, sizeof(int8_t) * dstFrameSize * frames);



  // Add a small number of loops to see if repeated calls to convert cause

  // any change in behavior.

  const int numLoops = fdp.ConsumeIntegralInRange<int>(1, 3);

  for (int loop = 0; loop < numLoops; ++loop) {

    switch (fdp.ConsumeIntegralInRange<int>(0, 1)) {

      case 0:

        converter.reset();

        FALLTHROUGH_INTENDED;

      case 1:

        converter.convert(dst, &provider, frames);

        break;

    }

  }



  return 0;

}

media/libaudioprocessing/tests/fuzzer/libaudioprocessing_resampler_fuzzer.cpp

+2 −40
Original line number Diff line number Diff line
@@ -34,6 +34,8 @@ 
#include <unistd.h>

#include <utils/Vector.h>



#include "libaudioprocessing_fuzz_utils.h"



#include <memory>



using namespace android;
@@ -53,46 +55,6 @@ const AudioResampler::src_quality qualities[] = { 
    AudioResampler::DYN_HIGH_QUALITY,

};



class Provider : public AudioBufferProvider {

  const void* mAddr;        // base address

  const size_t mNumFrames;  // total frames

  const size_t mFrameSize;  // size of each frame in bytes

  size_t mNextFrame;        // index of next frame to provide

  size_t mUnrel;            // number of frames not yet released

 public:

  Provider(const void* addr, size_t frames, size_t frameSize)

      : mAddr(addr),

        mNumFrames(frames),

        mFrameSize(frameSize),

        mNextFrame(0),

        mUnrel(0) {}

  status_t getNextBuffer(Buffer* buffer) override {

    if (buffer->frameCount > mNumFrames - mNextFrame) {

      buffer->frameCount = mNumFrames - mNextFrame;

    }

    mUnrel = buffer->frameCount;

    if (buffer->frameCount > 0) {

      buffer->raw = (char*)mAddr + mFrameSize * mNextFrame;

      return NO_ERROR;

    } else {

      buffer->raw = nullptr;

      return NOT_ENOUGH_DATA;

    }

  }

  virtual void releaseBuffer(Buffer* buffer) {

    if (buffer->frameCount > mUnrel) {

      mNextFrame += mUnrel;

      mUnrel = 0;

    } else {

      mNextFrame += buffer->frameCount;

      mUnrel -= buffer->frameCount;

    }

    buffer->frameCount = 0;

    buffer->raw = nullptr;

  }

  void reset() { mNextFrame = 0; }

};



audio_format_t chooseFormat(AudioResampler::src_quality quality,

                            uint8_t input_byte) {

  switch (quality) {