1 /* 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 "api/sequence_checker.h" 14 #include "rtc_base/checks.h" 15 #include "rtc_base/numerics/safe_conversions.h" 16 #include "rtc_base/numerics/safe_minmax.h" 17 #include "rtc_base/thread.h" 18 19 namespace { 20 constexpr int kDefaultDelay = 0; 21 constexpr int kMaximumDelayMs = 10000; 22 } // namespace 23 24 namespace webrtc { 25 JitterBufferDelay(rtc::Thread * worker_thread)26JitterBufferDelay::JitterBufferDelay(rtc::Thread* worker_thread) 27 : signaling_thread_(rtc::Thread::Current()), worker_thread_(worker_thread) { 28 RTC_DCHECK(worker_thread_); 29 } 30 OnStart(cricket::Delayable * media_channel,uint32_t ssrc)31void JitterBufferDelay::OnStart(cricket::Delayable* media_channel, 32 uint32_t ssrc) { 33 RTC_DCHECK_RUN_ON(signaling_thread_); 34 35 media_channel_ = media_channel; 36 ssrc_ = ssrc; 37 38 // Trying to apply cached delay for the audio stream. 39 if (cached_delay_seconds_) { 40 Set(cached_delay_seconds_.value()); 41 } 42 } 43 OnStop()44void JitterBufferDelay::OnStop() { 45 RTC_DCHECK_RUN_ON(signaling_thread_); 46 // Assume that audio stream is no longer present. 47 media_channel_ = nullptr; 48 ssrc_ = absl::nullopt; 49 } 50 Set(absl::optional<double> delay_seconds)51void JitterBufferDelay::Set(absl::optional<double> delay_seconds) { 52 RTC_DCHECK_RUN_ON(worker_thread_); 53 54 // TODO(kuddai) propagate absl::optional deeper down as default preference. 55 int delay_ms = 56 rtc::saturated_cast<int>(delay_seconds.value_or(kDefaultDelay) * 1000); 57 delay_ms = rtc::SafeClamp(delay_ms, 0, kMaximumDelayMs); 58 59 cached_delay_seconds_ = delay_seconds; 60 if (media_channel_ && ssrc_) { 61 media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms); 62 } 63 } 64 65 } // namespace webrtc 66