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

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(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}
This commit is contained in:
Fredrik Solenberg 2017-11-20 22:12:21 +01:00 committed by Commit Bot
parent d6c98c020a
commit 63e6072a43
2 changed files with 5 additions and 0 deletions
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3
audio/audio_state.h
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@ -35,6 +35,9 @@ class AudioState final : public webrtc::AudioState {
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;

2
call/audio_state.h
View File

@ -17,6 +17,7 @@
namespace webrtc {
class AudioProcessing;
class AudioTransport;
class VoiceEngine;
// WORK IN PROGRESS
@ -43,6 +44,7 @@ class AudioState : public rtc::RefCountInterface {
};
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