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9af4aa7cf4
This reverts commit56db8d0952. Reason for revert: downstream problem addressed Original change's description: > Revert "Represent RtpPacketToSend::capture_time with Timestamp" > > This reverts commit385eb9714d. > > Reason for revert: Causes problems downstream: > > # > # Fatal error in: rtc_base/units/unit_base.h, line 122 > # last system error: 0 > # Check failed: value >= 0 (-234 vs. 0) > > Original change's description: > > Represent RtpPacketToSend::capture_time with Timestamp > > > > Bug: webrtc:13757 > > Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36083} > > Bug: webrtc:13757 > Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720 > Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#36087} Bug: webrtc:13757 Change-Id: I1fa852757480116f35deb2b6c3c27800bdf5e197 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252781 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36093}
101 lines
3.1 KiB
C++
101 lines
3.1 KiB
C++
/*
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* Copyright (c) 2017 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 "api/video/video_timing.h"
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#include "api/array_view.h"
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#include "api/units/time_delta.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/strings/string_builder.h"
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namespace webrtc {
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uint16_t VideoSendTiming::GetDeltaCappedMs(int64_t base_ms, int64_t time_ms) {
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if (time_ms < base_ms) {
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RTC_DLOG(LS_ERROR) << "Delta " << (time_ms - base_ms)
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<< "ms expected to be positive";
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}
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return rtc::saturated_cast<uint16_t>(time_ms - base_ms);
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}
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uint16_t VideoSendTiming::GetDeltaCappedMs(TimeDelta delta) {
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if (delta < TimeDelta::Zero()) {
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RTC_DLOG(LS_ERROR) << "Delta " << delta.ms()
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<< "ms expected to be positive";
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}
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return rtc::saturated_cast<uint16_t>(delta.ms());
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}
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TimingFrameInfo::TimingFrameInfo()
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: rtp_timestamp(0),
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capture_time_ms(-1),
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encode_start_ms(-1),
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encode_finish_ms(-1),
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packetization_finish_ms(-1),
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pacer_exit_ms(-1),
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network_timestamp_ms(-1),
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network2_timestamp_ms(-1),
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receive_start_ms(-1),
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receive_finish_ms(-1),
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decode_start_ms(-1),
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decode_finish_ms(-1),
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render_time_ms(-1),
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flags(VideoSendTiming::kNotTriggered) {}
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int64_t TimingFrameInfo::EndToEndDelay() const {
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return capture_time_ms >= 0 ? decode_finish_ms - capture_time_ms : -1;
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}
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bool TimingFrameInfo::IsLongerThan(const TimingFrameInfo& other) const {
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int64_t other_delay = other.EndToEndDelay();
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return other_delay == -1 || EndToEndDelay() > other_delay;
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}
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bool TimingFrameInfo::operator<(const TimingFrameInfo& other) const {
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return other.IsLongerThan(*this);
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}
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bool TimingFrameInfo::operator<=(const TimingFrameInfo& other) const {
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return !IsLongerThan(other);
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}
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bool TimingFrameInfo::IsOutlier() const {
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return !IsInvalid() && (flags & VideoSendTiming::kTriggeredBySize);
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}
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bool TimingFrameInfo::IsTimerTriggered() const {
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return !IsInvalid() && (flags & VideoSendTiming::kTriggeredByTimer);
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}
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bool TimingFrameInfo::IsInvalid() const {
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return flags == VideoSendTiming::kInvalid;
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}
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std::string TimingFrameInfo::ToString() const {
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if (IsInvalid()) {
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return "";
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}
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char buf[1024];
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rtc::SimpleStringBuilder sb(buf);
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sb << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms << ','
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<< encode_finish_ms << ',' << packetization_finish_ms << ','
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<< pacer_exit_ms << ',' << network_timestamp_ms << ','
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<< network2_timestamp_ms << ',' << receive_start_ms << ','
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<< receive_finish_ms << ',' << decode_start_ms << ',' << decode_finish_ms
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<< ',' << render_time_ms << ',' << IsOutlier() << ','
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<< IsTimerTriggered();
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return sb.str();
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}
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} // namespace webrtc
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