Simplification and refactoring of the AudioBuffer code

This CL performs a major refactoring and simplification
of the AudioBuffer code that.
-Removes 7 of the 9 internal buffers of the AudioBuffer.
-Avoids the implicit copying required to keep the
 internal buffers in sync.
-Removes all code relating to handling of fixed-point
 sample data in the AudioBuffer.
-Changes the naming of the class methods to reflect
 that only floating point is handled.
-Corrects some bugs in the code.
-Extends the handling of internal downmixing to be
 more generic.

Bug: webrtc:10882
Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28928}

modules/audio_processing/audio_processing_impl_unittest.cc

@@ -128,7 +128,7 @@ class TestRenderPreProcessor : public CustomProcessing {
   void Initialize(int sample_rate_hz, int num_channels) override {}
   void Process(AudioBuffer* audio) override {
     for (size_t k = 0; k < audio->num_channels(); ++k) {
-      rtc::ArrayView<float> channel_view(audio->channels_f()[k],
+      rtc::ArrayView<float> channel_view(audio->channels()[k],
                                          audio->num_frames());
       std::transform(channel_view.begin(), channel_view.end(),
                      channel_view.begin(), ProcessSample);