commit 1aec594018f29bda5a8fbe69d5d8cf6e635b7490
author Alex Loiko <aleloi@webrtc.org> Tue May 15 13:13:22 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Tue May 15 14:22:52 2018 +0000
tree d7f37897f911e9a34ae059e4e472c34275598915
parent 28a325b52314d31a2e3ac4e3c8f46761784183f5

Merge :audio_processing and :aec_dump_interface.

Merges the two targets in modules/audio_processing
and removes some redundant code. This enables not writing
a bunch of redundant code in
https://webrtc-review.googlesource.com/c/src/+/70502

':audio_processing' did depend on ':aec_dump_interface'.
'modules/audio_processing/aec_dump' did depend on
'aec_dump_interface' but not ':audio_processing'.

Having the AecDump implementation not depending on
'audio_processing' allows to have faster compilation time and
reduces the dependencies. However, maintaining such a decoupling
makes APM and AecDump client code more complex.

NOTRY=true # want this in and 'ios_api_framework' seems stuck.

Bug: webrtc:7404
Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c
Reviewed-on: https://webrtc-review.googlesource.com/76603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23244}

modules/audio_processing/audio_processing_impl.cc

diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index e3b3857..6d46490 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -148,24 +148,6 @@
   AudioProcessingImpl* apm_;
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
 };
-
-webrtc::InternalAPMStreamsConfig ToStreamsConfig(
-    const ProcessingConfig& api_format) {
-  webrtc::InternalAPMStreamsConfig result;
-  result.input_sample_rate = api_format.input_stream().sample_rate_hz();
-  result.input_num_channels = api_format.input_stream().num_channels();
-  result.output_num_channels = api_format.output_stream().num_channels();
-  result.render_input_num_channels =
-      api_format.reverse_input_stream().num_channels();
-  result.render_input_sample_rate =
-      api_format.reverse_input_stream().sample_rate_hz();
-  result.output_sample_rate = api_format.output_stream().sample_rate_hz();
-  result.render_output_sample_rate =
-      api_format.reverse_output_stream().sample_rate_hz();
-  result.render_output_num_channels =
-      api_format.reverse_output_stream().num_channels();
-  return result;
-}
 }  // namespace
 
 // Throughout webrtc, it's assumed that success is represented by zero.
@@ -608,7 +590,7 @@
   InitializePreProcessor();
 
   if (aec_dump_) {
-    aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
+    aec_dump_->WriteInitMessage(formats_.api_format);
   }
   return kNoError;
 }
@@ -1638,7 +1620,7 @@
   // 'aec_dump' parameter, which is after locks are released.
   aec_dump_.swap(aec_dump);
   WriteAecDumpConfigMessage(true);
-  aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
+  aec_dump_->WriteInitMessage(formats_.api_format);
 }
 
 void AudioProcessingImpl::DetachAecDump() {