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}
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);