Sam Zackrisson | b0db98c | 2019-11-26 18:55:02 +0100 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #include "modules/audio_processing/aec3/echo_canceller3.h" |
| 12 | #include "modules/audio_processing/audio_buffer.h" |
| 13 | #include "modules/audio_processing/include/audio_processing.h" |
| 14 | #include "test/fuzzers/fuzz_data_helper.h" |
| 15 | |
| 16 | namespace webrtc { |
| 17 | namespace { |
| 18 | using SampleRate = ::webrtc::AudioProcessing::NativeRate; |
| 19 | |
| 20 | void PrepareAudioBuffer(int sample_rate_hz, |
| 21 | test::FuzzDataHelper* fuzz_data, |
| 22 | AudioBuffer* buffer) { |
| 23 | float* const* channels = buffer->channels_f(); |
| 24 | for (size_t i = 0; i < buffer->num_channels(); ++i) { |
| 25 | for (size_t j = 0; j < buffer->num_frames(); ++j) { |
| 26 | channels[i][j] = |
| 27 | static_cast<float>(fuzz_data->ReadOrDefaultValue<int16_t>(0)); |
| 28 | } |
| 29 | } |
| 30 | if (sample_rate_hz == 32000 || sample_rate_hz == 48000) { |
| 31 | buffer->SplitIntoFrequencyBands(); |
| 32 | } |
| 33 | } |
| 34 | |
| 35 | } // namespace |
| 36 | |
| 37 | void FuzzOneInput(const uint8_t* data, size_t size) { |
| 38 | if (size > 200000) { |
| 39 | return; |
| 40 | } |
| 41 | |
| 42 | test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size)); |
| 43 | |
| 44 | constexpr int kSampleRates[] = {16000, 32000, 48000}; |
| 45 | const int sample_rate_hz = |
| 46 | static_cast<size_t>(fuzz_data.SelectOneOf(kSampleRates)); |
| 47 | |
| 48 | constexpr int kMaxNumChannels = 9; |
| 49 | const size_t num_render_channels = |
| 50 | 1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1); |
| 51 | const size_t num_capture_channels = |
| 52 | 1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1); |
| 53 | |
| 54 | EchoCanceller3 aec3(EchoCanceller3Config(), sample_rate_hz, |
| 55 | num_render_channels, num_capture_channels); |
| 56 | |
| 57 | AudioBuffer capture_audio(sample_rate_hz, num_capture_channels, |
| 58 | sample_rate_hz, num_capture_channels, |
| 59 | sample_rate_hz, num_capture_channels); |
| 60 | AudioBuffer render_audio(sample_rate_hz, num_render_channels, sample_rate_hz, |
| 61 | num_render_channels, sample_rate_hz, |
| 62 | num_render_channels); |
| 63 | |
| 64 | // Fuzz frames while there is still fuzzer data. |
| 65 | while (fuzz_data.BytesLeft() > 0) { |
| 66 | bool is_capture = fuzz_data.ReadOrDefaultValue(true); |
| 67 | bool level_changed = fuzz_data.ReadOrDefaultValue(true); |
| 68 | if (is_capture) { |
| 69 | PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &capture_audio); |
| 70 | aec3.ProcessCapture(&capture_audio, level_changed); |
| 71 | } else { |
| 72 | PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &render_audio); |
| 73 | aec3.AnalyzeRender(&render_audio); |
| 74 | } |
| 75 | } |
| 76 | } |
| 77 | } // namespace webrtc |