Avoid render resampling when there is no need for render signal analysis.

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 900a8ec437abdf069ec29e02afe678ea7fb939c5..10344a4c358db7b388216685f288ac27ae13ee02 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -598,7 +598,13 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
 
   // Always downmix the render stream to mono for analysis. This has been
   // demonstrated to work well for AEC in most practical scenarios.
-  formats_.render_processing_format = StreamConfig(render_processing_rate, 1);
+  if (submodule_states_.RenderMultiBandSubModulesActive()) {
+    formats_.render_processing_format = StreamConfig(render_processing_rate, 1);
+  } else {
+    formats_.render_processing_format = StreamConfig(
+        formats_.api_format.reverse_input_stream().sample_rate_hz(),
+        formats_.api_format.output_stream().num_channels());
+  }
 
   if (capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
           kSampleRate32kHz ||