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Revert "Add Sender and Receiver interfaces for MediaTransport audio"

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This reverts commit 0d8eed6ac77fadf7f9bcf70c671710d60b1ee62d.

Reason for revert: crashes of unit tests.

Original change's description:
> Add Sender and Receiver interfaces for MediaTransport audio
> 
> Implement in LoopbackMediaTransport.
> 
> Bug: webrtc:9719
> Change-Id: I429ac3f78d99b8ea4f9ac85b9a3600b215b61a55
> Reviewed-on: https://webrtc-review.googlesource.com/c/121957
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26731}

TBR=solenberg@webrtc.org,nisse@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com

Change-Id: I02e409e1bbe2b2dea8a7b1aa08fa44d4146bda8f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/123232
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26733}
This commit is contained in:
Sergey Silkin
2019-02-18 09:52:26 +00:00
committed by Commit Bot
parent d2f04360a6
commit e049eba27c
8 changed files with 12 additions and 145 deletions
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20
api/media_transport_interface.h
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@ -187,28 +187,10 @@ class MediaTransportInterface {
MediaTransportInterface();
virtual ~MediaTransportInterface();
// Creates an object representing the send end-point of a audio stream using
// this transport.
// TODO(bugs.webrtc.org/9719): Make pure virtual after downstream
// implementations are updated.
virtual std::unique_ptr<MediaTransportAudioSender> CreateAudioSender(
uint64_t channel_id);
// Creates an object representing the receive end-point of a audio stream
// using this transport.
// TODO(bugs.webrtc.org/9719): Make pure virtual after downstream
// implementations are updated.
virtual std::unique_ptr<MediaTransportAudioReceiver> CreateAudioReceiver(
uint64_t channel_id,
// TODO(nisse): Add Rtt observer, or route that via Call to the receive
// stream instead?
MediaTransportAudioSinkInterface* sink);
// Start asynchronous send of audio frame. The status returned by this method
// only pertains to the synchronous operations (e.g.
// serialization/packetization), not to the asynchronous operation.
// TODO(nisse): Deprecated, should be deleted when implementations are updated
// to use CreateAudioSender.
virtual RTCError SendAudioFrame(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) = 0;