From 2bfa5b20fe3fcba8fa70476cd40095b9f5f1c1e5 Mon Sep 17 00:00:00 2001 From: Minyue Li Date: Tue, 5 Oct 2021 14:46:20 +0200 Subject: [PATCH] Default sending capture clock offset in abs-capture-time header extension. Bug: webrtc:10739 Change-Id: Ieadb6d75122e5988b22509ac14dc528277a7f56f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232906 Commit-Queue: Minyue Li Reviewed-by: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#35149} --- modules/rtp_rtcp/source/rtp_sender_audio.cc | 4 ++-- .../source/rtp_sender_audio_unittest.cc | 18 +++++++++--------- modules/rtp_rtcp/source/rtp_sender_video.cc | 4 ++-- .../source/rtp_sender_video_unittest.cc | 2 +- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc index 2125295886..207d1ca045 100644 --- a/modules/rtp_rtcp/source/rtp_sender_audio.cc +++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc @@ -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_); } diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc index baa392281b..52880b9a59 100644 --- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc @@ -148,7 +148,16 @@ TEST_F(RtpSenderAudioTest, SendAudioWithoutAbsoluteCaptureTime) { .HasExtension()); } +// 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(&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(&fake_clock_, rtp_module_->RtpSender()); - rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(), kAbsoluteCaptureTimeExtensionId); constexpr uint32_t kAbsoluteCaptureTimestampMs = 521; diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc index b8d9b65546..906f2bca76 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/modules/rtp_rtcp/source/rtp_sender_video.cc @@ -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(); } diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc index ffdd0d8f6c..78e842da7e 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc @@ -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 ""; }