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/gain_controller2_unittest.cc

@@ -28,8 +28,7 @@ namespace {
 void SetAudioBufferSamples(float value, AudioBuffer* ab) {
   // Sets all the samples in |ab| to |value|.
   for (size_t k = 0; k < ab->num_channels(); ++k) {
-    std::fill(ab->channels_f()[k], ab->channels_f()[k] + ab->num_frames(),
-              value);
+    std::fill(ab->channels()[k], ab->channels()[k] + ab->num_frames(), value);
   }
 }
 
@@ -38,7 +37,7 @@ float RunAgc2WithConstantInput(GainController2* agc2,
                                size_t num_frames,
                                int sample_rate) {
   const int num_samples = rtc::CheckedDivExact(sample_rate, 100);
-  AudioBuffer ab(num_samples, 1, num_samples, 1, num_samples);
+  AudioBuffer ab(sample_rate, 1, sample_rate, 1, sample_rate);
 
   // Give time to the level estimator to converge.
   for (size_t i = 0; i < num_frames + 1; ++i) {
@@ -47,7 +46,7 @@ float RunAgc2WithConstantInput(GainController2* agc2,
   }
 
   // Return the last sample from the last processed frame.
-  return ab.channels_f()[0][num_samples - 1];
+  return ab.channels()[0][num_samples - 1];
 }
 
 AudioProcessing::Config::GainController2 CreateAgc2FixedDigitalModeConfig(
@@ -74,9 +73,9 @@ float GainAfterProcessingFile(GainController2* gain_controller) {
   constexpr size_t kStereo = 2u;
   const StreamConfig capture_config(AudioProcessing::kSampleRate48kHz, kStereo,
                                     false);
-  AudioBuffer ab(capture_config.num_frames(), capture_config.num_channels(),
-                 capture_config.num_frames(), capture_config.num_channels(),
-                 capture_config.num_frames());
+  AudioBuffer ab(capture_config.sample_rate_hz(), capture_config.num_channels(),
+                 capture_config.sample_rate_hz(), capture_config.num_channels(),
+                 capture_config.sample_rate_hz());
   test::InputAudioFile capture_file(
       test::GetApmCaptureTestVectorFileName(AudioProcessing::kSampleRate48kHz));
   std::vector<float> capture_input(capture_config.num_frames() *
@@ -99,7 +98,7 @@ float GainAfterProcessingFile(GainController2* gain_controller) {
   constexpr float sample_value = 1.f;
   SetAudioBufferSamples(sample_value, &ab);
   gain_controller->Process(&ab);
-  return ab.channels_f()[0][0];
+  return ab.channels()[0][0];
 }
 
 }  // namespace