blob: f2132d318d7c85e71fcef13a2f483fd1d6cb80e8 [file] [log] [blame]
* Copyright 2019 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 <stdint.h>
#include "absl/types/optional.h"
#include "media/base/delayable.h"
#include "rtc_base/ref_count.h"
namespace webrtc {
// JitterBufferDelay delivers user's queries to the underlying media channel. It
// can describe either video or audio delay for receiving stream. "Interface"
// suffix in the interface name is required to be compatible with api/
class JitterBufferDelayInterface : public rtc::RefCountInterface {
// OnStart allows to uniqely identify to which receiving stream playout
// delay must correpond through |media_channel| and |ssrc| pair.
virtual void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) = 0;
// Indicates that underlying receiving stream is stopped.
virtual void OnStop() = 0;
virtual void Set(absl::optional<double> delay_seconds) = 0;
} // namespace webrtc