Refactor to clarify band split rate is always 16k

This is a no-op refactor: The code is a leftover from when APM supported
8 kHz processing.

Bug: b/452621275
Change-Id: Ib42176fd8568addfcbb982b1ab2e61a9cf9188de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/418021
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45973}

modules/audio_processing/audio_processing_impl.cc

diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 8241b7d..7b7f497 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -88,6 +88,7 @@
 }
 
 // Identify the native processing rate that best handles a sample rate.
+// Always returns one of 16000, 32000 and 48000 Hz.
 int SuitableProcessRate(int minimum_rate,
                         int max_splitting_rate,
                         bool band_splitting_required) {
@@ -342,6 +343,7 @@
 }
 
 constexpr int kUnspecifiedDataDumpInputVolume = -100;
+constexpr int kBandSplitRate = AudioProcessing::kSampleRate16kHz;
 
 }  // namespace
 
@@ -643,7 +645,9 @@
       max_splitting_rate,
       submodule_states_.CaptureMultiBandSubModulesActive() ||
           submodule_states_.RenderMultiBandSubModulesActive());
-  RTC_DCHECK_NE(8000, capture_processing_rate);
+  RTC_DCHECK(capture_processing_rate == 16000 ||
+             capture_processing_rate == 32000 ||
+             capture_processing_rate == 48000);
 
   capture_nonlocked_.capture_processing_format =
       StreamConfig(capture_processing_rate);
@@ -659,18 +663,9 @@
   } else {
     render_processing_rate = capture_processing_rate;
   }
-
-  // If the forward sample rate is 8 kHz, the render stream is also processed
-  // at this rate.
-  if (capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
-      kSampleRate8kHz) {
-    render_processing_rate = kSampleRate8kHz;
-  } else {
-    render_processing_rate =
-        std::max(render_processing_rate, static_cast<int>(kSampleRate16kHz));
-  }
-
-  RTC_DCHECK_NE(8000, render_processing_rate);
+  RTC_DCHECK(render_processing_rate == 16000 ||
+             render_processing_rate == 32000 ||
+             render_processing_rate == 48000);
 
   if (submodule_states_.RenderMultiBandSubModulesActive()) {
     // By default, downmix the render stream to mono for analysis. This has been
@@ -689,16 +684,6 @@
         formats_.api_format.reverse_input_stream().num_channels());
   }
 
-  if (capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
-          kSampleRate32kHz ||
-      capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
-          kSampleRate48kHz) {
-    capture_nonlocked_.split_rate = kSampleRate16kHz;
-  } else {
-    capture_nonlocked_.split_rate =
-        capture_nonlocked_.capture_processing_format.sample_rate_hz();
-  }
-
   InitializeLocked();
 }
 
@@ -793,8 +778,7 @@
 }
 
 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
-  // Used as callback from submodules, hence locking is not allowed.
-  return capture_nonlocked_.split_rate;
+  return kBandSplitRate;
 }
 
 size_t AudioProcessingImpl::num_reverse_channels() const {
@@ -2304,7 +2288,6 @@
       capture_output_used_last_frame(true),
       key_pressed(false),
       capture_processing_format(kSampleRate16kHz),
-      split_rate(kSampleRate16kHz),
       echo_path_gain_change(false),
       prev_pre_adjustment_gain(-1.0f),
       playout_volume(-1),

modules/audio_processing/audio_processing_impl.h

diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index 3348f66..ee36a8c 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -452,7 +452,6 @@
     // because the capture processing number of channels is mutable and is
     // tracked by the capture_audio_.
     StreamConfig capture_processing_format;
-    int split_rate;
     bool echo_path_gain_change;
     float prev_pre_adjustment_gain;
     int playout_volume;
@@ -471,13 +470,11 @@
   struct ApmCaptureNonLockedState {
     ApmCaptureNonLockedState()
         : capture_processing_format(kSampleRate16kHz),
-          split_rate(kSampleRate16kHz),
           stream_delay_ms(0) {}
     // Only the rate and samples fields of capture_processing_format_ are used
     // because the forward processing number of channels is mutable and is
     // tracked by the capture_audio_.
     StreamConfig capture_processing_format;
-    int split_rate;
     int stream_delay_ms;
   } capture_nonlocked_;