Arun Raghavan b5c48b97f6 Bump to WebRTC M131 release
Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
2024-12-26 12:55:16 -05:00

125 lines
3.7 KiB
C++

/*
* Copyright (c) 2017 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 "api/video/video_timing.h"
#include <algorithm>
#include <cstdint>
#include <string>
#include "api/array_view.h"
#include "api/units/time_delta.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
uint16_t VideoSendTiming::GetDeltaCappedMs(int64_t base_ms, int64_t time_ms) {
if (time_ms < base_ms) {
RTC_DLOG(LS_ERROR) << "Delta " << (time_ms - base_ms)
<< "ms expected to be positive";
}
return rtc::saturated_cast<uint16_t>(time_ms - base_ms);
}
uint16_t VideoSendTiming::GetDeltaCappedMs(TimeDelta delta) {
if (delta < TimeDelta::Zero()) {
RTC_DLOG(LS_ERROR) << "Delta " << delta.ms()
<< "ms expected to be positive";
}
return rtc::saturated_cast<uint16_t>(delta.ms());
}
TimingFrameInfo::TimingFrameInfo()
: rtp_timestamp(0),
capture_time_ms(-1),
encode_start_ms(-1),
encode_finish_ms(-1),
packetization_finish_ms(-1),
pacer_exit_ms(-1),
network_timestamp_ms(-1),
network2_timestamp_ms(-1),
receive_start_ms(-1),
receive_finish_ms(-1),
decode_start_ms(-1),
decode_finish_ms(-1),
render_time_ms(-1),
flags(VideoSendTiming::kNotTriggered) {}
int64_t TimingFrameInfo::EndToEndDelay() const {
return capture_time_ms >= 0 ? decode_finish_ms - capture_time_ms : -1;
}
bool TimingFrameInfo::IsLongerThan(const TimingFrameInfo& other) const {
int64_t other_delay = other.EndToEndDelay();
return other_delay == -1 || EndToEndDelay() > other_delay;
}
bool TimingFrameInfo::operator<(const TimingFrameInfo& other) const {
return other.IsLongerThan(*this);
}
bool TimingFrameInfo::operator<=(const TimingFrameInfo& other) const {
return !IsLongerThan(other);
}
bool TimingFrameInfo::IsOutlier() const {
return !IsInvalid() && (flags & VideoSendTiming::kTriggeredBySize);
}
bool TimingFrameInfo::IsTimerTriggered() const {
return !IsInvalid() && (flags & VideoSendTiming::kTriggeredByTimer);
}
bool TimingFrameInfo::IsInvalid() const {
return flags == VideoSendTiming::kInvalid;
}
std::string TimingFrameInfo::ToString() const {
if (IsInvalid()) {
return "";
}
char buf[1024];
rtc::SimpleStringBuilder sb(buf);
sb << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms << ','
<< encode_finish_ms << ',' << packetization_finish_ms << ','
<< pacer_exit_ms << ',' << network_timestamp_ms << ','
<< network2_timestamp_ms << ',' << receive_start_ms << ','
<< receive_finish_ms << ',' << decode_start_ms << ',' << decode_finish_ms
<< ',' << render_time_ms << ',' << IsOutlier() << ','
<< IsTimerTriggered();
return sb.str();
}
VideoPlayoutDelay::VideoPlayoutDelay(TimeDelta min, TimeDelta max)
: min_(std::clamp(min, TimeDelta::Zero(), kMax)),
max_(std::clamp(max, min_, kMax)) {
if (!(TimeDelta::Zero() <= min && min <= max && max <= kMax)) {
RTC_LOG(LS_ERROR) << "Invalid video playout delay: [" << min << "," << max
<< "]. Clamped to [" << this->min() << "," << this->max()
<< "]";
}
}
bool VideoPlayoutDelay::Set(TimeDelta min, TimeDelta max) {
if (TimeDelta::Zero() <= min && min <= max && max <= kMax) {
min_ = min;
max_ = max;
return true;
}
return false;
}
} // namespace webrtc