Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API"
This reverts commit 90bace095806a635411edd40fb8490a144e59e63. Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it. Original change's description: > Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API > > (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201) > > This SetAudioPlayout method lets applications disable audio playout while > still processing incoming audio data and generating statistics on the > received audio. > > This may be useful if the application wants to set up media flows as > soon as possible, but isn't ready to play audio yet. Currently, native > applications don't have any API point to control this, unless they > completely implement their own AudioDeviceModule. > > The SetAudioRecording works in a similar fashion but for the recorded > audio. One difference is that calling SetAudioRecording(false) does not > keep any audio processing alive. > > TBR=solenberg > > Bug: webrtc:7313 > Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa > Reviewed-on: https://webrtc-review.googlesource.com/16180 > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Henrik Andreassson <henrika@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20499} TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7313 Reviewed-on: https://webrtc-review.googlesource.com/17701 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20512}
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@ -44,17 +44,6 @@ class AudioState : public rtc::RefCountInterface {
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virtual AudioProcessing* audio_processing() = 0;
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// Enable/disable playout of the audio channels. Enabled by default.
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// This will stop playout of the underlying audio device but start a task
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// which will poll for audio data every 10ms to ensure that audio processing
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// happens and the audio stats are updated.
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virtual void SetPlayout(bool enabled) = 0;
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// Enable/disable recording of the audio channels. Enabled by default.
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// This will stop recording of the underlying audio device and no audio
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// packets will be encoded or transmitted.
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virtual void SetRecording(bool enabled) = 0;
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// TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
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static rtc::scoped_refptr<AudioState> Create(
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const AudioState::Config& config);
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