493a650b1e
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}
88 lines
2.4 KiB
C++
88 lines
2.4 KiB
C++
/*
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* Copyright 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/playout_latency.h"
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#include "iostream"
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#include "rtc_base/checks.h"
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#include "rtc_base/location.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/thread_checker.h"
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namespace {
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constexpr int kDefaultLatency = 0;
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constexpr int kRoundToZeroThresholdMs = 10;
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} // namespace
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namespace webrtc {
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PlayoutLatency::PlayoutLatency(rtc::Thread* worker_thread)
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: signaling_thread_(rtc::Thread::Current()), worker_thread_(worker_thread) {
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RTC_DCHECK(worker_thread_);
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}
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void PlayoutLatency::OnStart(cricket::Delayable* media_channel, uint32_t ssrc) {
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RTC_DCHECK_RUN_ON(signaling_thread_);
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media_channel_ = media_channel;
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ssrc_ = ssrc;
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// Trying to apply cached latency for the audio stream.
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if (cached_latency_) {
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SetLatency(cached_latency_.value());
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}
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}
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void PlayoutLatency::OnStop() {
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RTC_DCHECK_RUN_ON(signaling_thread_);
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// Assume that audio stream is no longer present for latency calls.
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media_channel_ = nullptr;
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ssrc_ = absl::nullopt;
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}
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void PlayoutLatency::SetLatency(double latency) {
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RTC_DCHECK_RUN_ON(worker_thread_);
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RTC_DCHECK_GE(latency, 0);
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RTC_DCHECK_LE(latency, 10);
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int delay_ms = rtc::dchecked_cast<int>(latency * 1000);
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// In JitterBuffer 0 delay has special meaning of being unconstrained value
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// that is why we round delay to 0 if it is small enough during conversion
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// from latency.
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if (delay_ms <= kRoundToZeroThresholdMs) {
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delay_ms = 0;
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}
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cached_latency_ = latency;
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if (media_channel_ && ssrc_) {
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media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms);
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}
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}
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double PlayoutLatency::GetLatency() const {
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RTC_DCHECK_RUN_ON(worker_thread_);
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absl::optional<int> delay_ms;
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if (media_channel_ && ssrc_) {
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delay_ms = media_channel_->GetBaseMinimumPlayoutDelayMs(ssrc_.value());
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}
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if (delay_ms) {
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return delay_ms.value() / 1000.0;
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} else {
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return cached_latency_.value_or(kDefaultLatency);
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}
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}
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} // namespace webrtc
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