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64cdcc0792
The features have two safety fallbacks: - multichannel config has a killswitch WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch - stereo detection has a killswitch WebRTC-Aec3StereoContentDetectionKillSwitch Both features are enabled by default in the AEC3 config. Tested: Bitexact on a large number of aecdumps. Bug: chromium:1295710 Change-Id: I340cdc9140dacd4ca22d0911eb9f732b6cf8b226 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258129 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36482}
79 lines
2.9 KiB
C++
79 lines
2.9 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "absl/types/optional.h"
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#include "modules/audio_processing/aec3/echo_canceller3.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "test/fuzzers/fuzz_data_helper.h"
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namespace webrtc {
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namespace {
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using SampleRate = ::webrtc::AudioProcessing::NativeRate;
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void PrepareAudioBuffer(int sample_rate_hz,
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test::FuzzDataHelper* fuzz_data,
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AudioBuffer* buffer) {
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float* const* channels = buffer->channels_f();
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for (size_t i = 0; i < buffer->num_channels(); ++i) {
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for (size_t j = 0; j < buffer->num_frames(); ++j) {
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channels[i][j] =
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static_cast<float>(fuzz_data->ReadOrDefaultValue<int16_t>(0));
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}
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}
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if (sample_rate_hz == 32000 || sample_rate_hz == 48000) {
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buffer->SplitIntoFrequencyBands();
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}
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}
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} // namespace
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void FuzzOneInput(const uint8_t* data, size_t size) {
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if (size > 200000) {
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return;
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}
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test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
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constexpr int kSampleRates[] = {16000, 32000, 48000};
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const int sample_rate_hz =
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static_cast<size_t>(fuzz_data.SelectOneOf(kSampleRates));
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constexpr int kMaxNumChannels = 9;
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const size_t num_render_channels =
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1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
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const size_t num_capture_channels =
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1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
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EchoCanceller3 aec3(EchoCanceller3Config(),
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/*multichannel_config=*/absl::nullopt, sample_rate_hz,
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num_render_channels, num_capture_channels);
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AudioBuffer capture_audio(sample_rate_hz, num_capture_channels,
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sample_rate_hz, num_capture_channels,
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sample_rate_hz, num_capture_channels);
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AudioBuffer render_audio(sample_rate_hz, num_render_channels, sample_rate_hz,
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num_render_channels, sample_rate_hz,
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num_render_channels);
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// Fuzz frames while there is still fuzzer data.
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while (fuzz_data.BytesLeft() > 0) {
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bool is_capture = fuzz_data.ReadOrDefaultValue(true);
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bool level_changed = fuzz_data.ReadOrDefaultValue(true);
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if (is_capture) {
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PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &capture_audio);
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aec3.ProcessCapture(&capture_audio, level_changed);
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} else {
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PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &render_audio);
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aec3.AnalyzeRender(&render_audio);
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}
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}
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}
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} // namespace webrtc
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