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}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;
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