You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
68 lines
2.0 KiB
68 lines
2.0 KiB
/*
|
|
* 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
|