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f0d1c03c31
The pointer-to-submodule interfaces are being removed. This CL: 1) introduces AudioProcessing::Config::GainController1 with most config, 2) adds functions to APM for setting and getting analog gain, 3) creates a temporary GainControlConfigProxy to support the transition to the new config. 4) Moves the lock references in GainControlForExperimentalAgc and GainControlImpl into the GainControlConfigProxy, as it becomes the sole AGC object with functionality exposed to the client. Bug: webrtc:9947, webrtc:9878 Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27316}
119 lines
4.6 KiB
C++
119 lines
4.6 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 "absl/memory/memory.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/gain_control_impl.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/thread_annotations.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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void FillAudioBuffer(test::FuzzDataHelper* fuzz_data, 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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}
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// This function calls the GainControl functions that are overriden as private
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// in GainControlInterface.
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void FuzzGainControllerConfig(test::FuzzDataHelper* fuzz_data,
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GainControl* gc) {
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GainControl::Mode modes[] = {GainControl::Mode::kAdaptiveAnalog,
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GainControl::Mode::kAdaptiveDigital,
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GainControl::Mode::kFixedDigital};
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GainControl::Mode mode = fuzz_data->SelectOneOf(modes);
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const bool enable_limiter = fuzz_data->ReadOrDefaultValue(true);
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// The values are capped to comply with the API of webrtc::GainControl.
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const int analog_level_min =
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rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(0), 0, 65534);
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const int analog_level_max =
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rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(65535),
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analog_level_min + 1, 65535);
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const int stream_analog_level =
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rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(30000),
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analog_level_min, analog_level_max);
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const int gain =
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rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(30), -1, 100);
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const int target_level_dbfs =
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rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(15), -1, 35);
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gc->set_mode(mode);
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gc->enable_limiter(enable_limiter);
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if (mode == GainControl::Mode::kAdaptiveAnalog) {
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gc->set_analog_level_limits(analog_level_min, analog_level_max);
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gc->set_stream_analog_level(stream_analog_level);
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}
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gc->set_compression_gain_db(gain);
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gc->set_target_level_dbfs(target_level_dbfs);
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gc->Enable(true);
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static_cast<void>(gc->is_enabled());
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static_cast<void>(gc->mode());
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static_cast<void>(gc->analog_level_minimum());
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static_cast<void>(gc->analog_level_maximum());
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static_cast<void>(gc->stream_analog_level());
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static_cast<void>(gc->compression_gain_db());
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static_cast<void>(gc->stream_is_saturated());
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static_cast<void>(gc->target_level_dbfs());
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static_cast<void>(gc->is_limiter_enabled());
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}
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void FuzzGainController(test::FuzzDataHelper* fuzz_data, GainControlImpl* gci) {
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using Rate = ::webrtc::AudioProcessing::NativeRate;
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const Rate rate_kinds[] = {Rate::kSampleRate8kHz, Rate::kSampleRate16kHz,
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Rate::kSampleRate32kHz, Rate::kSampleRate48kHz};
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const auto sample_rate_hz =
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static_cast<size_t>(fuzz_data->SelectOneOf(rate_kinds));
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const size_t samples_per_frame = sample_rate_hz / 100;
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const bool num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
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gci->Initialize(num_channels, sample_rate_hz);
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FuzzGainControllerConfig(fuzz_data, gci);
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// The audio buffer is used for both capture and render.
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AudioBuffer audio(samples_per_frame, num_channels, samples_per_frame,
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num_channels, samples_per_frame);
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std::vector<int16_t> packed_render_audio(samples_per_frame);
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while (fuzz_data->CanReadBytes(1)) {
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FillAudioBuffer(fuzz_data, &audio);
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const bool stream_has_echo = fuzz_data->ReadOrDefaultValue(true);
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gci->AnalyzeCaptureAudio(&audio);
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gci->ProcessCaptureAudio(&audio, stream_has_echo);
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FillAudioBuffer(fuzz_data, &audio);
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gci->PackRenderAudioBuffer(&audio, &packed_render_audio);
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gci->ProcessRenderAudio(packed_render_audio);
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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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auto gci = absl::make_unique<GainControlImpl>();
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FuzzGainController(&fuzz_data, gci.get());
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}
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} // namespace webrtc
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