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

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 453 matching lines @@
           formats_.api_format.reverse_input_stream().num_frames(),
           formats_.api_format.reverse_output_stream().num_channels(),
           formats_.api_format.reverse_output_stream().num_frames());
     } else {
       render_.render_converter.reset(nullptr);
     }
   } else {
     render_.render_audio.reset(nullptr);
     render_.render_converter.reset(nullptr);
   }
+
   capture_.capture_audio.reset(
       new AudioBuffer(formats_.api_format.input_stream().num_frames(),
                       formats_.api_format.input_stream().num_channels(),
                       capture_nonlocked_.capture_processing_format.num_frames(),
                       capture_audiobuffer_num_channels,
                       formats_.api_format.output_stream().num_frames()));
 
   public_submodules_->echo_cancellation->Initialize(
       proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
       num_proc_channels());
@@ skipping 105 matching lines @@
   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));
   }
 
   // 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.reverse_input_stream().num_channels());

[peah-webrtc, 2017/05/19 08:07:46]

This change was needed for this to work properly for some combinations of input
channels/rate. With this, the processing rate/channels is done the same for the
input rate and the number of channels, which means that no implicit resampling
will be done at the de-interleave.

[ivoc, 2017/05/19 08:24:22]

Acknowledged.

+  }
 
   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();
   }
@@ skipping 843 matching lines @@
     render_buffer->SplitIntoFrequencyBands();
   }
 
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   if (capture_nonlocked_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
         render_buffer);
   }
 #endif
 
-  QueueBandedRenderAudio(render_buffer);
+  if (submodule_states_.RenderMultiBandSubModulesActive()) {

[peah-webrtc, 2017/05/19 08:07:46]

This change was needed as with the changes in this CL the render signal is no
longer resampled to a native rate if i no processing is being done.

[ivoc, 2017/05/19 08:24:22]

Acknowledged.

+    QueueBandedRenderAudio(render_buffer);
+  }
+
   // TODO(peah): Perform the queueing ínside QueueRenderAudiuo().
   if (private_submodules_->echo_canceller3) {
     private_submodules_->echo_canceller3->AnalyzeRender(render_buffer);
   }
 
   if (submodule_states_.RenderMultiBandProcessingActive() &&
       SampleRateSupportsMultiBand(
           formats_.render_processing_format.sample_rate_hz())) {
     render_buffer->MergeFrequencyBands();
   }
@@ skipping 538 matching lines @@
       previous_agc_level(0),
       echo_path_gain_change(false) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc