| /* |
| * 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 "pc/jitter_buffer_delay.h" |
| |
| #include "rtc_base/checks.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/numerics/safe_minmax.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/thread_checker.h" |
| |
| namespace { |
| constexpr int kDefaultDelay = 0; |
| constexpr int kMaximumDelayMs = 10000; |
| } // namespace |
| |
| namespace webrtc { |
| |
| JitterBufferDelay::JitterBufferDelay(rtc::Thread* worker_thread) |
| : signaling_thread_(rtc::Thread::Current()), worker_thread_(worker_thread) { |
| RTC_DCHECK(worker_thread_); |
| } |
| |
| void JitterBufferDelay::OnStart(cricket::Delayable* media_channel, |
| uint32_t ssrc) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| |
| media_channel_ = media_channel; |
| ssrc_ = ssrc; |
| |
| // Trying to apply cached delay for the audio stream. |
| if (cached_delay_seconds_) { |
| Set(cached_delay_seconds_.value()); |
| } |
| } |
| |
| void JitterBufferDelay::OnStop() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| // Assume that audio stream is no longer present. |
| media_channel_ = nullptr; |
| ssrc_ = absl::nullopt; |
| } |
| |
| void JitterBufferDelay::Set(absl::optional<double> delay_seconds) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| |
| // TODO(kuddai) propagate absl::optional deeper down as default preference. |
| int delay_ms = |
| rtc::saturated_cast<int>(delay_seconds.value_or(kDefaultDelay) * 1000); |
| delay_ms = rtc::SafeClamp(delay_ms, 0, kMaximumDelayMs); |
| |
| cached_delay_seconds_ = delay_seconds; |
| if (media_channel_ && ssrc_) { |
| media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms); |
| } |
| } |
| |
| } // namespace webrtc |