Bump to WebRTC M131 release

Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42

webrtc/modules/audio_processing/agc/BUILD.gn

@@ -21,11 +21,11 @@ rtc_library("agc") {
   deps = [
     ":gain_control_interface",
     ":level_estimation",
-    "..:api",
     "..:apm_logging",
     "..:audio_buffer",
     "..:audio_frame_view",
     "../../../api:array_view",
+    "../../../api/audio:audio_processing",
     "../../../common_audio",
     "../../../common_audio:common_audio_c",
     "../../../rtc_base:checks",
@@ -39,7 +39,6 @@ rtc_library("agc") {
     "../agc2:input_volume_stats_reporter",
     "../vad",
   ]
-  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("level_estimation") {
@@ -117,10 +116,7 @@ if (rtc_include_tests) {
       "../../../test:fileutils",
       "../../../test:test_support",
       "//testing/gtest",
-    ]
-    absl_deps = [
-      "//third_party/abseil-cpp/absl/strings",
-      "//third_party/abseil-cpp/absl/types:optional",
+      "//third_party/abseil-cpp/absl/strings:string_view",
     ]
   }
 }

webrtc/modules/audio_processing/agc/agc_manager_direct.cc

@@ -69,11 +69,11 @@ using AnalogAgcConfig =
 // string. Returns an unspecified value if the field trial is not specified, if
 // disabled or if it cannot be parsed. Example:
 // 'WebRTC-Audio-2ndAgcMinMicLevelExperiment/Enabled-80' => returns 80.
-absl::optional<int> GetMinMicLevelOverride() {
+std::optional<int> GetMinMicLevelOverride() {
   constexpr char kMinMicLevelFieldTrial[] =
       "WebRTC-Audio-2ndAgcMinMicLevelExperiment";
   if (!webrtc::field_trial::IsEnabled(kMinMicLevelFieldTrial)) {
-    return absl::nullopt;
+    return std::nullopt;
   }
   const auto field_trial_string =
       webrtc::field_trial::FindFullName(kMinMicLevelFieldTrial);
@@ -84,7 +84,7 @@ absl::optional<int> GetMinMicLevelOverride() {
   } else {
     RTC_LOG(LS_WARNING) << "[agc] Invalid parameter for "
                         << kMinMicLevelFieldTrial << ", ignored.";
-    return absl::nullopt;
+    return std::nullopt;
   }
 }
 
@@ -189,8 +189,8 @@ void MonoAgc::Initialize() {
 }
 
 void MonoAgc::Process(rtc::ArrayView<const int16_t> audio,
-                      absl::optional<int> rms_error_override) {
-  new_compression_to_set_ = absl::nullopt;
+                      std::optional<int> rms_error_override) {
+  new_compression_to_set_ = std::nullopt;
 
   if (check_volume_on_next_process_) {
     check_volume_on_next_process_ = false;
@@ -617,13 +617,13 @@ void AgcManagerDirect::AnalyzePreProcess(const AudioBuffer& audio_buffer) {
 }
 
 void AgcManagerDirect::Process(const AudioBuffer& audio_buffer) {
-  Process(audio_buffer, /*speech_probability=*/absl::nullopt,
-          /*speech_level_dbfs=*/absl::nullopt);
+  Process(audio_buffer, /*speech_probability=*/std::nullopt,
+          /*speech_level_dbfs=*/std::nullopt);
 }
 
 void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
-                               absl::optional<float> speech_probability,
-                               absl::optional<float> speech_level_dbfs) {
+                               std::optional<float> speech_probability,
+                               std::optional<float> speech_level_dbfs) {
   AggregateChannelLevels();
   const int volume_after_clipping_handling = recommended_input_volume_;
 
@@ -632,7 +632,7 @@ void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
   }
 
   const size_t num_frames_per_band = audio_buffer.num_frames_per_band();
-  absl::optional<int> rms_error_override = absl::nullopt;
+  std::optional<int> rms_error_override = std::nullopt;
   if (speech_probability.has_value() && speech_level_dbfs.has_value()) {
     rms_error_override =
         GetSpeechLevelErrorDb(*speech_level_dbfs, *speech_probability);
@@ -656,7 +656,7 @@ void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
   }
 }
 
-absl::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
+std::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
   return new_compressions_to_set_[channel_controlling_gain_];
 }
 

webrtc/modules/audio_processing/agc/agc_manager_direct.h

@@ -13,13 +13,13 @@
 
 #include <atomic>
 #include <memory>
+#include <optional>
 
-#include "absl/types/optional.h"
 #include "api/array_view.h"
+#include "api/audio/audio_processing.h"
 #include "modules/audio_processing/agc/agc.h"
 #include "modules/audio_processing/agc2/clipping_predictor.h"
 #include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/gtest_prod_util.h"
 
@@ -75,8 +75,8 @@ class AgcManagerDirect final {
   // TODO(webrtc:7494): This signature is needed for testing purposes, unify
   // the signatures when the clean-up is done.
   void Process(const AudioBuffer& audio_buffer,
-               absl::optional<float> speech_probability,
-               absl::optional<float> speech_level_dbfs);
+               std::optional<float> speech_probability,
+               std::optional<float> speech_level_dbfs);
 
   // Processes `audio_buffer`. Chooses a digital compression gain and the new
   // input volume to recommend. Must be called after `AnalyzePreProcess()`.
@@ -100,7 +100,7 @@ class AgcManagerDirect final {
 
   // If available, returns the latest digital compression gain that has been
   // chosen.
-  absl::optional<int> GetDigitalComressionGain();
+  std::optional<int> GetDigitalComressionGain();
 
   // Returns true if clipping prediction is enabled.
   bool clipping_predictor_enabled() const { return !!clipping_predictor_; }
@@ -150,7 +150,7 @@ class AgcManagerDirect final {
 
   const bool analog_controller_enabled_;
 
-  const absl::optional<int> min_mic_level_override_;
+  const std::optional<int> min_mic_level_override_;
   std::unique_ptr<ApmDataDumper> data_dumper_;
   static std::atomic<int> instance_counter_;
   const int num_capture_channels_;
@@ -176,7 +176,7 @@ class AgcManagerDirect final {
   const int clipped_wait_frames_;
 
   std::vector<std::unique_ptr<MonoAgc>> channel_agcs_;
-  std::vector<absl::optional<int>> new_compressions_to_set_;
+  std::vector<std::optional<int>> new_compressions_to_set_;
 
   const std::unique_ptr<ClippingPredictor> clipping_predictor_;
   const bool use_clipping_predictor_step_;
@@ -213,16 +213,14 @@ class MonoAgc {
   // after `HandleClipping()`. If `rms_error_override` has a value, RMS error
   // from AGC is overridden by it.
   void Process(rtc::ArrayView<const int16_t> audio,
-               absl::optional<int> rms_error_override);
+               std::optional<int> rms_error_override);
 
   // Returns the recommended input volume. Must be called after `Process()`.
   int recommended_analog_level() const { return recommended_input_volume_; }
 
   float voice_probability() const { return agc_->voice_probability(); }
   void ActivateLogging() { log_to_histograms_ = true; }
-  absl::optional<int> new_compression() const {
-    return new_compression_to_set_;
-  }
+  std::optional<int> new_compression() const { return new_compression_to_set_; }
 
   // Only used for testing.
   void set_agc(Agc* agc) { agc_.reset(agc); }
@@ -263,7 +261,7 @@ class MonoAgc {
   // recommended input volume.
   int recommended_input_volume_ = 0;
 
-  absl::optional<int> new_compression_to_set_;
+  std::optional<int> new_compression_to_set_;
   bool log_to_histograms_ = false;
   const int clipped_level_min_;