Bump to WebRTC M120 release

Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.

webrtc/modules/audio_processing/ns/BUILD.gn

@@ -57,7 +57,6 @@ rtc_static_library("ns") {
     "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../common_audio/third_party/ooura:fft_size_256",
     "../../../rtc_base:checks",
-    "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:arch",
     "../../../system_wrappers",
@@ -80,12 +79,11 @@ if (rtc_include_tests) {
       "..:apm_logging",
       "..:audio_buffer",
       "..:audio_processing",
-      "..:audio_processing_unittests",
       "..:high_pass_filter",
       "../../../api:array_view",
       "../../../rtc_base:checks",
-      "../../../rtc_base:rtc_base_approved",
       "../../../rtc_base:safe_minmax",
+      "../../../rtc_base:stringutils",
       "../../../rtc_base/system:arch",
       "../../../system_wrappers",
       "../../../test:test_support",
@@ -98,5 +96,9 @@ if (rtc_include_tests) {
     if (rtc_enable_protobuf) {
       sources += []
     }
+
+    if (!build_with_chromium) {
+      deps += [ "..:audio_processing_unittests" ]
+    }
   }
 }

webrtc/modules/audio_processing/ns/noise_suppressor.cc

@@ -13,6 +13,7 @@
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
+
 #include <algorithm>
 
 #include "modules/audio_processing/ns/fast_math.h"
@@ -448,6 +449,12 @@ void NoiseSuppressor::Process(AudioBuffer* audio) {
     }
   }
 
+  // Only do the below processing if the output of the audio processing module
+  // is used.
+  if (!capture_output_used_) {
+    return;
+  }
+
   // Aggregate the Wiener filters for all channels.
   std::array<float, kFftSizeBy2Plus1> filter_data;
   rtc::ArrayView<const float, kFftSizeBy2Plus1> filter = filter_data;

webrtc/modules/audio_processing/ns/noise_suppressor.h

@@ -41,12 +41,21 @@ class NoiseSuppressor {
   // Applies noise suppression.
   void Process(AudioBuffer* audio);
 
+  // Specifies whether the capture output will be used. The purpose of this is
+  // to allow the noise suppressor to deactivate some of the processing when the
+  // resulting output is anyway not used, for instance when the endpoint is
+  // muted.
+  void SetCaptureOutputUsage(bool capture_output_used) {
+    capture_output_used_ = capture_output_used;
+  }
+
  private:
   const size_t num_bands_;
   const size_t num_channels_;
   const SuppressionParams suppression_params_;
   int32_t num_analyzed_frames_ = -1;
   NrFft fft_;
+  bool capture_output_used_ = true;
 
   struct ChannelState {
     ChannelState(const SuppressionParams& suppression_params, size_t num_bands);

webrtc/modules/audio_processing/ns/prior_signal_model_estimator.cc

@@ -11,6 +11,7 @@
 #include "modules/audio_processing/ns/prior_signal_model_estimator.h"
 
 #include <math.h>
+
 #include <algorithm>
 
 #include "modules/audio_processing/ns/fast_math.h"

webrtc/modules/audio_processing/ns/quantile_noise_estimator.h

@@ -12,6 +12,7 @@
 #define MODULES_AUDIO_PROCESSING_NS_QUANTILE_NOISE_ESTIMATOR_H_
 
 #include <math.h>
+
 #include <array>
 
 #include "api/array_view.h"

webrtc/modules/audio_processing/ns/speech_probability_estimator.cc

@@ -11,6 +11,7 @@
 #include "modules/audio_processing/ns/speech_probability_estimator.h"
 
 #include <math.h>
+
 #include <algorithm>
 
 #include "modules/audio_processing/ns/fast_math.h"

webrtc/modules/audio_processing/ns/suppression_params.cc

@@ -42,7 +42,7 @@ SuppressionParams::SuppressionParams(
       use_attenuation_adjustment = true;
       break;
     default:
-      RTC_NOTREACHED();
+      RTC_DCHECK_NOTREACHED();
   }
 }
 

webrtc/modules/audio_processing/ns/wiener_filter.cc

@@ -13,6 +13,7 @@
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
+
 #include <algorithm>
 
 #include "modules/audio_processing/ns/fast_math.h"