This reverts commit 09aaf6f7bcfb4da644bd86c76896a04a41f776e1. Reason for revert: downstream fixed (see https://chromium-review.googlesource.com/c/chromium/src/+/3461371) Original change's description: > Revert "Reland "Remove unused APM voice activity detection sub-module"" > > This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f. > > Reason for revert: Breaks chromium roll, see > https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview > > https://chromium-review.googlesource.com/c/chromium/src/+/3461512 > > Original change's description: > > Reland "Remove unused APM voice activity detection sub-module" > > > > This reverts commit a751f167c68343f76528436defdbc61600a8d7b3. > > > > Reason for revert: dependency in a downstream project removed > > > > Original change's description: > > > Revert "Remove unused APM voice activity detection sub-module" > > > > > > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215. > > > > > > Reason for revert: breaking downstream projects > > > > > > Original change's description: > > > > Remove unused APM voice activity detection sub-module > > > > > > > > API changes: > > > > - webrtc::AudioProcessing::Config::VoiceDetection removed > > > > - webrtc::AudioProcessingStats::voice_detected deprecated > > > > - cricket::AudioOptions::typing_detection deprecated > > > > - webrtc::StatsReport::StatsValueName:: > > > > kStatsValueNameTypingNoiseState deprecated > > > > > > > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0 > > > > > > > > Bug: webrtc:11226,webrtc:11292 > > > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666 > > > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> > > > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > > > Cr-Commit-Position: refs/heads/main@{#35975} > > > > > > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > > > > > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: webrtc:11226,webrtc:11292 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600 > > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/main@{#35977} > > > > # Not skipping CQ checks because this is a reland. > > > > Bug: webrtc:11226,webrtc:11292 > > Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660 > > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35984} > > TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11226,webrtc:11292 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Auto-Submit: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35990} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:11226,webrtc:11292 Change-Id: Idfda6a517027ad323caf44c526a88468e5b52b65 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251762 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36012}
158 lines
6.2 KiB
C++
158 lines
6.2 KiB
C++
/*
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* Copyright (c) 2017 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 <bitset>
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#include <string>
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#include "absl/memory/memory.h"
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#include "api/audio/echo_canceller3_factory.h"
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#include "api/audio/echo_detector_creator.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
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#include "rtc_base/arraysize.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/task_queue.h"
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#include "system_wrappers/include/field_trial.h"
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#include "test/fuzzers/audio_processing_fuzzer_helper.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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const std::string kFieldTrialNames[] = {
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"WebRTC-Audio-Agc2ForceExtraSaturationMargin",
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"WebRTC-Audio-Agc2ForceInitialSaturationMargin",
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"WebRTC-Aec3MinErleDuringOnsetsKillSwitch",
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"WebRTC-Aec3ShortHeadroomKillSwitch",
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};
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rtc::scoped_refptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
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std::string* field_trial_string,
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rtc::TaskQueue* worker_queue) {
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// Parse boolean values for optionally enabling different
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// configurable public components of APM.
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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bool use_ts = fuzz_data->ReadOrDefaultValue(true);
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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bool use_red = fuzz_data->ReadOrDefaultValue(true);
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bool use_hpf = fuzz_data->ReadOrDefaultValue(true);
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bool use_aec3 = fuzz_data->ReadOrDefaultValue(true);
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bool use_aec = fuzz_data->ReadOrDefaultValue(true);
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bool use_aecm = fuzz_data->ReadOrDefaultValue(true);
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bool use_agc = fuzz_data->ReadOrDefaultValue(true);
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bool use_ns = fuzz_data->ReadOrDefaultValue(true);
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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bool use_agc_limiter = fuzz_data->ReadOrDefaultValue(true);
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bool use_agc2 = fuzz_data->ReadOrDefaultValue(true);
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// Read an int8 value, but don't let it be too large or small.
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const float gain_controller2_gain_db =
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rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(0), -40, 40);
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constexpr size_t kNumFieldTrials = arraysize(kFieldTrialNames);
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// Verify that the read data type has enough bits to fuzz the field trials.
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using FieldTrialBitmaskType = uint64_t;
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static_assert(kNumFieldTrials <= sizeof(FieldTrialBitmaskType) * 8,
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"FieldTrialBitmaskType is not large enough.");
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std::bitset<kNumFieldTrials> field_trial_bitmask(
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fuzz_data->ReadOrDefaultValue<FieldTrialBitmaskType>(0));
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for (size_t i = 0; i < kNumFieldTrials; ++i) {
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if (field_trial_bitmask[i]) {
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*field_trial_string += kFieldTrialNames[i] + "/Enabled/";
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}
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}
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field_trial::InitFieldTrialsFromString(field_trial_string->c_str());
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bool use_agc2_adaptive_digital = fuzz_data->ReadOrDefaultValue(true);
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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// Ignore a few bytes. Bytes from this segment will be used for
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// future config flag changes. We assume 40 bytes is enough for
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// configuring the APM.
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constexpr size_t kSizeOfConfigSegment = 40;
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RTC_DCHECK(kSizeOfConfigSegment >= fuzz_data->BytesRead());
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static_cast<void>(
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fuzz_data->ReadByteArray(kSizeOfConfigSegment - fuzz_data->BytesRead()));
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// Filter out incompatible settings that lead to CHECK failures.
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if ((use_aecm && use_aec) || // These settings cause CHECK failure.
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(use_aecm && use_aec3 && use_ns) // These settings trigger webrtc:9489.
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) {
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return nullptr;
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}
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std::unique_ptr<EchoControlFactory> echo_control_factory;
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if (use_aec3) {
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echo_control_factory.reset(new EchoCanceller3Factory());
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}
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webrtc::AudioProcessing::Config apm_config;
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apm_config.pipeline.multi_channel_render = true;
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apm_config.pipeline.multi_channel_capture = true;
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apm_config.echo_canceller.enabled = use_aec || use_aecm;
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apm_config.echo_canceller.mobile_mode = use_aecm;
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apm_config.high_pass_filter.enabled = use_hpf;
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apm_config.gain_controller1.enabled = use_agc;
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apm_config.gain_controller1.enable_limiter = use_agc_limiter;
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apm_config.gain_controller2.enabled = use_agc2;
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apm_config.gain_controller2.fixed_digital.gain_db = gain_controller2_gain_db;
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apm_config.gain_controller2.adaptive_digital.enabled =
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use_agc2_adaptive_digital;
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apm_config.noise_suppression.enabled = use_ns;
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apm_config.transient_suppression.enabled = use_ts;
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rtc::scoped_refptr<AudioProcessing> apm =
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AudioProcessingBuilderForTesting()
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.SetEchoControlFactory(std::move(echo_control_factory))
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.SetEchoDetector(use_red ? CreateEchoDetector() : nullptr)
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.SetConfig(apm_config)
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.Create();
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#ifdef WEBRTC_LINUX
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apm->AttachAecDump(AecDumpFactory::Create("/dev/null", -1, worker_queue));
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#endif
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return apm;
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}
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TaskQueueFactory* GetTaskQueueFactory() {
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static TaskQueueFactory* const factory =
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CreateDefaultTaskQueueFactory().release();
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return factory;
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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 > 400000) {
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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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// This string must be in scope during execution, according to documentation
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// for field_trial.h. Hence it's created here and not in CreateApm.
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std::string field_trial_string = "";
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rtc::TaskQueue worker_queue(GetTaskQueueFactory()->CreateTaskQueue(
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"rtc-low-prio", rtc::TaskQueue::Priority::LOW));
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auto apm = CreateApm(&fuzz_data, &field_trial_string, &worker_queue);
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if (apm) {
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FuzzAudioProcessing(&fuzz_data, std::move(apm));
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}
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}
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} // namespace webrtc
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