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/agc2/clipping_predictor.h

@@ -12,11 +12,11 @@
 #define MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
 
 #include <memory>
+#include <optional>
 #include <vector>
 
-#include "absl/types/optional.h"
+#include "api/audio/audio_processing.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
-#include "modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 
@@ -35,12 +35,12 @@ class ClippingPredictor {
 
   // Predicts if clipping is going to occur for the specified `channel` in the
   // near-future and, if so, it returns a recommended analog mic level decrease
-  // step. Returns absl::nullopt if clipping is not predicted.
+  // step. Returns std::nullopt if clipping is not predicted.
   // `level` is the current analog mic level, `default_step` is the amount the
   // mic level is lowered by the analog controller with every clipping event and
   // `min_mic_level` and `max_mic_level` is the range of allowed analog mic
   // levels.
-  virtual absl::optional<int> EstimateClippedLevelStep(
+  virtual std::optional<int> EstimateClippedLevelStep(
       int channel,
       int level,
       int default_step,