Important: This change does not in any way affect echo cancellation or standardized stats. The user audio experience is unchanged. Only non-standard stats are affected. Echo return loss metrics are unchanged. Residual echo likelihood {recent max} will no longer be computed by default.
Important: The echo detector is no longer enabled by default.
API change, PSA: https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
This CL removes the default usage of the residual echo detector in APM.
It can now only be used via injection and the helper function webrtc::CreateEchoDetector. See how the function audio_processing_unittest.cc:CreateApm() changed, for an example.
The echo detector implementation is marked poisonous, to avoid accidental dependencies.
Some cleanup is done:
- EchoDetector::PackRenderAudioBuffer is declared in one target but is defined in another target. It is not necessary to keep in the API. It is made an implementation detail, and the echo detector input is documented in the API.
- The internal state of APM is large and difficult to track. Submodule pointers that are set permanently on construction are now appropriately marked const.
Tested:
- existing + new unit tests
- audioproc_f is bitexact on a large number of aecdumps
Bug: webrtc:11539
Change-Id: I00cc2ee112fedb06451a533409311605220064d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239652
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35550}
159 lines
6.3 KiB
C++
159 lines
6.3 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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bool use_vad = 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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apm_config.voice_detection.enabled = use_vad;
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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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