Partial revert of "Don't always downsample to 16kHz in the reverse stream in APM"

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 348 matching lines @@
       num_in_channels != capture_.array_geometry.size()) {
     return kBadNumberChannelsError;
   }
 
   formats_.api_format = config;
 
   capture_nonlocked_.fwd_proc_format = StreamConfig(ClosestNativeRate(std::min(
       formats_.api_format.input_stream().sample_rate_hz(),
       formats_.api_format.output_stream().sample_rate_hz())));
 
-  int rev_proc_rate = ClosestNativeRate(std::min(

[the sun, 2016/04/06 09:34:57]

nit: ClosestNativeRate() should be renamed btw, since it finds not the closest
rate, bug the closest *higher* rate...

[peah-webrtc, 2016/04/06 09:36:31]

True. Good point! Ok if I address that in another CL?

-      formats_.api_format.reverse_input_stream().sample_rate_hz(),
-      formats_.api_format.reverse_output_stream().sample_rate_hz()));
-  // If the forward sample rate is 8 kHz, the reverse stream is also processed
-  // at this rate.
+  // We normally process the reverse stream at 16 kHz. Unless...
+  int rev_proc_rate = kSampleRate16kHz;
   if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
+    // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
-    rev_proc_rate = std::max(rev_proc_rate, static_cast<int>(kSampleRate16kHz));
+    if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
+        kSampleRate32kHz) {
+      // ...or the input is at 32 kHz, in which case we use the splitting
+      // filter rather than the resampler.
+      rev_proc_rate = kSampleRate32kHz;
+    }
   }
 
   // Always downmix the reverse stream to mono for analysis. This has been
   // demonstrated to work well for AEC in most practical scenarios.
   formats_.rev_proc_format = StreamConfig(rev_proc_rate, 1);
 
   if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate32kHz ||
       capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate48kHz) {
     capture_nonlocked_.split_rate = kSampleRate16kHz;
   } else {
@@ skipping 753 matching lines @@
   }
   return false;
 }
 
 bool AudioProcessingImpl::is_rev_processed() const {
   return constants_.intelligibility_enabled;
 }
 
 bool AudioProcessingImpl::rev_synthesis_needed() const {
   return (is_rev_processed() &&
-          is_multi_band(formats_.rev_proc_format.sample_rate_hz()));
+          formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz);
 }
 
 bool AudioProcessingImpl::rev_analysis_needed() const {
-  return is_multi_band(formats_.rev_proc_format.sample_rate_hz()) &&
+  return formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
          (is_rev_processed() ||
           public_submodules_->echo_cancellation->is_enabled() ||
           public_submodules_->echo_control_mobile->is_enabled() ||
           public_submodules_->gain_control->is_enabled());
 }
 
 bool AudioProcessingImpl::render_check_rev_conversion_needed() const {
   return rev_conversion_needed();
 }
 
@@ skipping 278 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc