Update to current webrtc library

This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
--- a/webrtc/modules/audio_processing/gain_controller2.cc
+++ b/webrtc/modules/audio_processing/gain_controller2.cc
@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/gain_controller2.h"
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/atomic_ops.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+int GainController2::instance_count_ = 0;
+
+GainController2::GainController2()
+    : data_dumper_(
+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
+      gain_applier_(/*hard_clip_samples=*/false,
+                    /*initial_gain_factor=*/0.f),
+      limiter_(static_cast<size_t>(48000), data_dumper_.get(), "Agc2") {
+  if (config_.adaptive_digital.enabled) {
+    adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get()));
+  }
+}
+
+GainController2::~GainController2() = default;
+
+void GainController2::Initialize(int sample_rate_hz) {
+  RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate48kHz);
+  limiter_.SetSampleRate(sample_rate_hz);
+  data_dumper_->InitiateNewSetOfRecordings();
+  data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz);
+}
+
+void GainController2::Process(AudioBuffer* audio) {
+  AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
+                                    audio->num_frames());
+  // Apply fixed gain first, then the adaptive one.
+  gain_applier_.ApplyGain(float_frame);
+  if (adaptive_agc_) {
+    adaptive_agc_->Process(float_frame, limiter_.LastAudioLevel());
+  }
+  limiter_.Process(float_frame);
+}
+
+void GainController2::NotifyAnalogLevel(int level) {
+  if (analog_level_ != level && adaptive_agc_) {
+    adaptive_agc_->Reset();
+  }
+  analog_level_ = level;
+}
+
+void GainController2::ApplyConfig(
+    const AudioProcessing::Config::GainController2& config) {
+  RTC_DCHECK(Validate(config))
+      << " the invalid config was " << ToString(config);
+
+  config_ = config;
+  if (config.fixed_digital.gain_db != config_.fixed_digital.gain_db) {
+    // Reset the limiter to quickly react on abrupt level changes caused by
+    // large changes of the fixed gain.
+    limiter_.Reset();
+  }
+  gain_applier_.SetGainFactor(DbToRatio(config_.fixed_digital.gain_db));
+  if (config_.adaptive_digital.enabled) {
+    adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get(), config_));
+  } else {
+    adaptive_agc_.reset();
+  }
+}
+
+bool GainController2::Validate(
+    const AudioProcessing::Config::GainController2& config) {
+  return config.fixed_digital.gain_db >= 0.f &&
+         config.fixed_digital.gain_db < 50.f &&
+         config.adaptive_digital.extra_saturation_margin_db >= 0.f &&
+         config.adaptive_digital.extra_saturation_margin_db <= 100.f;
+}
+
+std::string GainController2::ToString(
+    const AudioProcessing::Config::GainController2& config) {
+  rtc::StringBuilder ss;
+  std::string adaptive_digital_level_estimator;
+  using LevelEstimatorType =
+      AudioProcessing::Config::GainController2::LevelEstimator;
+  switch (config.adaptive_digital.level_estimator) {
+    case LevelEstimatorType::kRms:
+      adaptive_digital_level_estimator = "RMS";
+      break;
+    case LevelEstimatorType::kPeak:
+      adaptive_digital_level_estimator = "peak";
+      break;
+  }
+  // clang-format off
+  // clang formatting doesn't respect custom nested style.
+  ss << "{"
+        "enabled: " << (config.enabled ? "true" : "false") << ", "
+        "fixed_digital: {gain_db: " << config.fixed_digital.gain_db << "}, "
+        "adaptive_digital: {"
+          "enabled: "
+            << (config.adaptive_digital.enabled ? "true" : "false") << ", "
+          "level_estimator: {"
+            "type: " << adaptive_digital_level_estimator << ", "
+            "adjacent_speech_frames_threshold: "
+              << config.adaptive_digital
+                  .level_estimator_adjacent_speech_frames_threshold << ", "
+            "initial_saturation_margin_db: "
+              << config.adaptive_digital.initial_saturation_margin_db << ", "
+            "extra_saturation_margin_db: "
+              << config.adaptive_digital.extra_saturation_margin_db << "}, "
+          "gain_applier: {"
+            "adjacent_speech_frames_threshold: "
+              << config.adaptive_digital
+                  .gain_applier_adjacent_speech_frames_threshold << ", "
+            "max_gain_change_db_per_second: "
+              << config.adaptive_digital.max_gain_change_db_per_second << ", "
+            "max_output_noise_level_dbfs: "
+              << config.adaptive_digital.max_output_noise_level_dbfs << "}"
+        "}"
+        "}";
+  // clang-format on
+  return ss.Release();
+}
+
+}  // namespace webrtc