commit 63e6072a4356a478dee8dce6cc87878dd77295cb
author Fredrik Solenberg <solenberg@webrtc.org> Mon Nov 20 21:12:21 2017
committer Commit Bot <commit-bot@chromium.org> Tue Nov 21 10:51:02 2017
tree 818f0f3777f0b9c696fe92bae289fd3d38c75651
parent d6c98c020aac5f9d07f0218b5bf34b5eb0029f44

Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine.

(See: https://webrtc-review.googlesource.com/c/src/+/23820)

Bug: webrtc:4690
Change-Id: I474a327303aa0c9b5b34c2055ae3a35e466a7d9f
Reviewed-on: https://webrtc-review.googlesource.com/24720
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20810}

audio/audio_state.h

diff --git a/audio/audio_state.h b/audio/audio_state.h
index 023c7b1..f4bddbf 100644
--- a/audio/audio_state.h
+++ b/audio/audio_state.h
@@ -35,6 +35,9 @@
     RTC_DCHECK(config_.audio_processing);
     return config_.audio_processing.get();
   }
+  AudioTransport* audio_transport() override {
+    return &audio_transport_proxy_;
+  }
 
   void SetPlayout(bool enabled) override;
   void SetRecording(bool enabled) override;

call/audio_state.h

diff --git a/call/audio_state.h b/call/audio_state.h
index ad411d1..e4a281a 100644
--- a/call/audio_state.h
+++ b/call/audio_state.h
@@ -17,6 +17,7 @@
 namespace webrtc {
 
 class AudioProcessing;
+class AudioTransport;
 class VoiceEngine;
 
 // WORK IN PROGRESS
@@ -43,6 +44,7 @@
   };
 
   virtual AudioProcessing* audio_processing() = 0;
+  virtual AudioTransport* audio_transport() = 0;
 
   // Enable/disable playout of the audio channels. Enabled by default.
   // This will stop playout of the underlying audio device but start a task