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Default sending capture clock offset in abs-capture-time header extension.

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Bug: webrtc:10739
Change-Id: Ieadb6d75122e5988b22509ac14dc528277a7f56f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232906
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35149}
This commit is contained in:
Minyue Li 2021-10-05 14:46:20 +02:00 committed by WebRTC LUCI CQ
parent c9f43f8f81
commit 2bfa5b20fe
4 changed files with 14 additions and 14 deletions
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4
modules/rtp_rtcp/source/rtp_sender_audio.cc
View file

@ -60,8 +60,8 @@ RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
rtp_sender_(rtp_sender),
absolute_capture_time_sender_(clock),
include_capture_clock_offset_(
absl::StartsWith(field_trials_.Lookup(kIncludeCaptureClockOffset),
"Enabled")) {
!absl::StartsWith(field_trials_.Lookup(kIncludeCaptureClockOffset),
"Disabled")) {
RTC_DCHECK(clock_);
}

18
modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
View file

@ -148,7 +148,16 @@ TEST_F(RtpSenderAudioTest, SendAudioWithoutAbsoluteCaptureTime) {
.HasExtension<AbsoluteCaptureTimeExtension>());
}
// Essentially the same test as
// SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset but with a field
// trial. We will remove this test eventually.
TEST_F(RtpSenderAudioTest, SendAudioWithAbsoluteCaptureTime) {
// Recreate rtp_sender_audio_ with new field trial.
test::ScopedFieldTrials field_trial(
"WebRTC-IncludeCaptureClockOffset/Disabled/");
rtp_sender_audio_ =
std::make_unique<RTPSenderAudio>(&fake_clock_, rtp_module_->RtpSender());
rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
kAbsoluteCaptureTimeExtensionId);
constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
@ -174,17 +183,8 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAbsoluteCaptureTime) {
absolute_capture_time->estimated_capture_clock_offset.has_value());
}
// Essentially the same test as SendAudioWithAbsoluteCaptureTime but with a
// field trial. After the field trial is experimented, we will remove
// SendAudioWithAbsoluteCaptureTime.
TEST_F(RtpSenderAudioTest,
SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset) {
// Recreate rtp_sender_audio_ wieh new field trial.
test::ScopedFieldTrials field_trial(
"WebRTC-IncludeCaptureClockOffset/Enabled/");
rtp_sender_audio_ =
std::make_unique<RTPSenderAudio>(&fake_clock_, rtp_module_->RtpSender());
rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
kAbsoluteCaptureTimeExtensionId);
constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;

4
modules/rtp_rtcp/source/rtp_sender_video.cc
View file

@ -175,9 +175,9 @@ RTPSenderVideo::RTPSenderVideo(const Config& config)
rtp_sender_->SSRC(),
config.send_transport_queue)
: nullptr),
include_capture_clock_offset_(absl::StartsWith(
include_capture_clock_offset_(!absl::StartsWith(
config.field_trials->Lookup(kIncludeCaptureClockOffset),
"Enabled")) {
"Disabled")) {
if (frame_transformer_delegate_)
frame_transformer_delegate_->Init();
}

2
modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
View file

@ -159,7 +159,7 @@ class FieldTrials : public WebRtcKeyValueConfig {
if (key == "WebRTC-SendSideBwe-WithOverhead") {
return use_send_side_bwe_with_overhead_ ? "Enabled" : "";
} else if (key == "WebRTC-IncludeCaptureClockOffset") {
return include_capture_clock_offset_ ? "Enabled" : "";
return include_capture_clock_offset_ ? "" : "Disabled";
}
return "";
}