blob: c9506b3c59f60f7c372d4a9f0374615de4c0256e [file] [log] [blame]
Ruslan Burakov428dcb22019-04-18 17:49:49 +02001/*
2 * Copyright 2019 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include "pc/jitter_buffer_delay.h"
12
13#include "rtc_base/checks.h"
14#include "rtc_base/location.h"
15#include "rtc_base/logging.h"
16#include "rtc_base/numerics/safe_conversions.h"
17#include "rtc_base/numerics/safe_minmax.h"
18#include "rtc_base/thread.h"
19#include "rtc_base/thread_checker.h"
20
21namespace {
22constexpr int kDefaultDelay = 0;
23constexpr int kMaximumDelayMs = 10000;
24} // namespace
25
26namespace webrtc {
27
28JitterBufferDelay::JitterBufferDelay(rtc::Thread* worker_thread)
29 : signaling_thread_(rtc::Thread::Current()), worker_thread_(worker_thread) {
30 RTC_DCHECK(worker_thread_);
31}
32
33void JitterBufferDelay::OnStart(cricket::Delayable* media_channel,
34 uint32_t ssrc) {
35 RTC_DCHECK_RUN_ON(signaling_thread_);
36
37 media_channel_ = media_channel;
38 ssrc_ = ssrc;
39
40 // Trying to apply cached delay for the audio stream.
41 if (cached_delay_seconds_) {
42 Set(cached_delay_seconds_.value());
43 }
44}
45
46void JitterBufferDelay::OnStop() {
47 RTC_DCHECK_RUN_ON(signaling_thread_);
48 // Assume that audio stream is no longer present.
49 media_channel_ = nullptr;
50 ssrc_ = absl::nullopt;
51}
52
53void JitterBufferDelay::Set(absl::optional<double> delay_seconds) {
54 RTC_DCHECK_RUN_ON(worker_thread_);
55
56 // TODO(kuddai) propagate absl::optional deeper down as default preference.
57 int delay_ms =
58 rtc::saturated_cast<int>(delay_seconds.value_or(kDefaultDelay) * 1000);
59 delay_ms = rtc::SafeClamp(delay_ms, 0, kMaximumDelayMs);
60
61 cached_delay_seconds_ = delay_seconds;
62 if (media_channel_ && ssrc_) {
63 media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms);
64 }
65}
66
67} // namespace webrtc