webrtc/api/media_stream_interface.cc
Ruslan Burakov 493a650b1e Propagate base minimum delay from video jitter buffer to webrtc/api.
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.


Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
2019-02-27 15:08:34 +00:00

39 lines
1 KiB
C++

/*
* Copyright 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/media_stream_interface.h"
namespace webrtc {
const char MediaStreamTrackInterface::kVideoKind[] = "video";
const char MediaStreamTrackInterface::kAudioKind[] = "audio";
VideoTrackInterface::ContentHint VideoTrackInterface::content_hint() const {
return ContentHint::kNone;
}
bool AudioTrackInterface::GetSignalLevel(int* level) {
return false;
}
rtc::scoped_refptr<AudioProcessorInterface>
AudioTrackInterface::GetAudioProcessor() {
return nullptr;
}
const cricket::AudioOptions AudioSourceInterface::options() const {
return {};
}
double MediaSourceInterface::GetLatency() const {
return 0.0;
}
} // namespace webrtc