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Reland "Represent RtpPacketToSend::capture_time with Timestamp"

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This reverts commit 56db8d09529d5ba92d24954a1d78a90c8ea2cd85.

Reason for revert: downstream problem addressed

Original change's description:
> Revert "Represent RtpPacketToSend::capture_time with Timestamp"
>
> This reverts commit 385eb9714daa80306d2f92d36678c42892dab555.
>
> 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}
This commit is contained in:
Danil Chapovalov
2022-02-27 21:10:55 +00:00
committed by WebRTC LUCI CQ
parent 6ccd74816a
commit 9af4aa7cf4
19 changed files with 140 additions and 89 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
api/video/BUILD.gn
View File

@ -31,6 +31,7 @@ rtc_library("video_rtp_headers") {
"../../rtc_base:rtc_base_approved",
"../../rtc_base/system:rtc_export",
"../units:data_rate",
"../units:time_delta",
]
absl_deps = [
"//third_party/abseil-cpp/absl/container:inlined_vector",

9
api/video/video_timing.cc
View File

@ -11,6 +11,7 @@
#include "api/video/video_timing.h"
#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"
@ -25,6 +26,14 @@ uint16_t VideoSendTiming::GetDeltaCappedMs(int64_t base_ms, int64_t time_ms) {
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),

3
api/video/video_timing.h
View File

@ -16,6 +16,8 @@
#include <limits>
#include <string>
#include "api/units/time_delta.h"
namespace webrtc {
// Video timing timestamps in ms counted from capture_time_ms of a frame.
@ -34,6 +36,7 @@ struct VideoSendTiming {
// https://webrtc.org/experiments/rtp-hdrext/video-timing/ extension stores
// 16-bit deltas of timestamps from packet capture time.
static uint16_t GetDeltaCappedMs(int64_t base_ms, int64_t time_ms);
static uint16_t GetDeltaCappedMs(TimeDelta delta);
uint16_t encode_start_delta_ms;
uint16_t encode_finish_delta_ms;