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/common_audio/resampler/include/push_resampler.h

@@ -14,45 +14,44 @@
 #include <memory>
 #include <vector>
 
+#include "api/audio/audio_view.h"
+
 namespace webrtc {
 
 class PushSincResampler;
 
 // Wraps PushSincResampler to provide stereo support.
-// TODO(ajm): add support for an arbitrary number of channels.
+// Note: This implementation assumes 10ms buffer sizes throughout.
 template <typename T>
-class PushResampler {
+class PushResampler final {
  public:
   PushResampler();
-  virtual ~PushResampler();
-
-  // Must be called whenever the parameters change. Free to be called at any
-  // time as it is a no-op if parameters have not changed since the last call.
-  int InitializeIfNeeded(int src_sample_rate_hz,
-                         int dst_sample_rate_hz,
-                         size_t num_channels);
+  PushResampler(size_t src_samples_per_channel,
+                size_t dst_samples_per_channel,
+                size_t num_channels);
+  ~PushResampler();
 
   // Returns the total number of samples provided in destination (e.g. 32 kHz,
   // 2 channel audio gives 640 samples).
-  int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
+  int Resample(InterleavedView<const T> src, InterleavedView<T> dst);
+  // For when a deinterleaved/mono channel already exists and we can skip the
+  // deinterleaved operation.
+  int Resample(MonoView<const T> src, MonoView<T> dst);
 
  private:
-  int src_sample_rate_hz_;
-  int dst_sample_rate_hz_;
-  size_t num_channels_;
-  // Vector that is needed to provide the proper inputs and outputs to the
-  // interleave/de-interleave methods used in Resample. This needs to be
-  // heap-allocated on the state to support an arbitrary number of channels
-  // without doing run-time heap-allocations in the Resample method.
-  std::vector<T*> channel_data_array_;
+  // Ensures that source and destination buffers for deinterleaving are
+  // correctly configured prior to resampling that requires deinterleaving.
+  void EnsureInitialized(size_t src_samples_per_channel,
+                         size_t dst_samples_per_channel,
+                         size_t num_channels);
 
-  struct ChannelResampler {
-    std::unique_ptr<PushSincResampler> resampler;
-    std::vector<T> source;
-    std::vector<T> destination;
-  };
+  // Buffers used for when a deinterleaving step is necessary.
+  std::unique_ptr<T[]> source_;
+  std::unique_ptr<T[]> destination_;
+  DeinterleavedView<T> source_view_;
+  DeinterleavedView<T> destination_view_;
 
-  std::vector<ChannelResampler> channel_resamplers_;
+  std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
 };
 }  // namespace webrtc