blob: 3fdf823d24f26e1a7feedf2c208de78d6138f5a8 [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 "pc/jitter_buffer_delay.h"
#include "api/sequence_checker.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/thread.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