Revert "Simplification and refactoring of the AudioBuffer code"

This reverts commit 81c0cf287c8514cb1cd6f3baca484d668c6eb128.

Reason for revert: internal test failures

Original change's description:
> 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 Ã…hgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}

TBR=gustaf@webrtc.org,peah@webrtc.org

Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28931}
diff --git a/modules/audio_processing/echo_control_mobile_impl.cc b/modules/audio_processing/echo_control_mobile_impl.cc
index 8057e33..982287b 100644
--- a/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/modules/audio_processing/echo_control_mobile_impl.cc
@@ -142,7 +142,7 @@
   for (size_t i = 0; i < num_output_channels; i++) {
     for (size_t j = 0; j < audio->num_channels(); j++) {
       std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> data_to_buffer;
-      FloatS16ToS16(audio->split_bands_const(render_channel)[kBand0To8kHz],
+      FloatS16ToS16(audio->split_bands_const_f(render_channel)[kBand0To8kHz],
                     audio->num_frames_per_band(), data_to_buffer.data());
 
       // Buffer the samples in the render queue.
@@ -185,8 +185,8 @@
     std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> split_bands_data;
     int16_t* split_bands = split_bands_data.data();
     const int16_t* clean = split_bands_data.data();
-    if (audio->split_bands(capture)[kBand0To8kHz]) {
-      FloatS16ToS16(audio->split_bands(capture)[kBand0To8kHz],
+    if (audio->split_bands_f(capture)[kBand0To8kHz]) {
+      FloatS16ToS16(audio->split_bands_f(capture)[kBand0To8kHz],
                     audio->num_frames_per_band(), split_bands_data.data());
     } else {
       clean = nullptr;
@@ -205,7 +205,7 @@
 
       if (split_bands) {
         S16ToFloatS16(split_bands, audio->num_frames_per_band(),
-                      audio->split_bands(capture)[kBand0To8kHz]);
+                      audio->split_bands_f(capture)[kBand0To8kHz]);
       }
 
       if (err != AudioProcessing::kNoError) {
@@ -227,7 +227,7 @@
   RTC_DCHECK_LE(audio->num_channels(), low_pass_reference_.size());
   reference_copied_ = true;
   for (size_t capture = 0; capture < audio->num_channels(); ++capture) {
-    FloatS16ToS16(audio->split_bands_const(capture)[kBand0To8kHz],
+    FloatS16ToS16(audio->split_bands_const_f(capture)[kBand0To8kHz],
                   audio->num_frames_per_band(),
                   low_pass_reference_[capture].data());
   }