commit ce4d91527a2cfaca3a8581ae4d5883fd63a346f9
author peah <peah@webrtc.org> Fri May 19 01:28:05 2017 -0700
committer Commit bot <commit-bot@chromium.org> Fri May 19 08:28:05 2017 +0000
tree 8093c0a01cb74d103a879eaa049feea8ac733569
parent 7f52f084219fdd8003ff0312b81a2a991ad5f572

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

This CL adjusts the render processing rate such to avoid resampling of the
render signal when that is not needed.
Note that to avoid acquiring more locks than needed, this should be achieved
during initialization.

BUG=webrtc:7667

Review-Url: https://codereview.webrtc.org/2887693002
Cr-Commit-Position: refs/heads/master@{#18207}

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 756cc0c..961d9a4 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -471,6 +471,7 @@
     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(),
@@ -596,7 +597,13 @@
 
   // 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());
+  }
 
   if (capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
           kSampleRate32kHz ||
@@ -1460,7 +1467,10 @@
   }
 #endif
 
-  QueueBandedRenderAudio(render_buffer);
+  if (submodule_states_.RenderMultiBandSubModulesActive()) {
+    QueueBandedRenderAudio(render_buffer);
+  }
+
   // TODO(peah): Perform the queueing ínside QueueRenderAudiuo().
   if (private_submodules_->echo_canceller3) {
     private_submodules_->echo_canceller3->AnalyzeRender(render_buffer);

webrtc/modules/audio_processing/audio_processing_unittest.cc

diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc
index c56a57b..4d97904 100644
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -862,8 +862,6 @@
   for (size_t i = 1; i < 4; i++) {
     TestChangingChannelsInt16Interface(i, kNoErr);
     EXPECT_EQ(i, apm_->num_input_channels());
-    // We always force the number of reverse channels used for processing to 1.
-    EXPECT_EQ(1u, apm_->num_reverse_channels());
   }
 }