Files
platform-external-webrtc/api/video/video_timing.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

79 lines
2.4 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 <sstream>
namespace webrtc {
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(TimingFrameFlags::kDefault) {}
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 & TimingFrameFlags::kTriggeredBySize);
}
bool TimingFrameInfo::IsTimerTriggered() const {
return !IsInvalid() && (flags & TimingFrameFlags::kTriggeredByTimer);
}
bool TimingFrameInfo::IsInvalid() const {
return flags == TimingFrameFlags::kInvalid;
}
std::string TimingFrameInfo::ToString() const {
std::stringstream out;
if (IsInvalid()) {
out << "";
} else {
out << 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 out.str();
}
} // namespace webrtc