From 31996f48f4c13a68bd3fe476109b4320fbb9ee41 Mon Sep 17 00:00:00 2001 From: Alessio Bazzica Date: Wed, 14 Sep 2022 15:32:53 +0200 Subject: [PATCH] `RtpSource`: remove deprecated ctor, use designated initializers Bug: webrtc:10739, b/246753278 Change-Id: I215483709e1f415170bc42ea6d523ffad8eb1e76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275561 Commit-Queue: Alessio Bazzica Reviewed-by: Tomas Gunnarsson Reviewed-by: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#38085} --- api/transport/rtp/rtp_source.h | 13 -------- .../source/source_tracker_unittest.cc | 30 +++++++++++-------- 2 files changed, 18 insertions(+), 25 deletions(-) diff --git a/api/transport/rtp/rtp_source.h b/api/transport/rtp/rtp_source.h index 8c543cac0c..c19cfebacb 100644 --- a/api/transport/rtp/rtp_source.h +++ b/api/transport/rtp/rtp_source.h @@ -33,19 +33,6 @@ class RtpSource { RtpSource() = delete; - // TODO(bugs.webrtc.org/10739): Remove this constructor once all clients - // migrate to the version with absolute capture time. - RtpSource(int64_t timestamp_ms, - uint32_t source_id, - RtpSourceType source_type, - absl::optional audio_level, - uint32_t rtp_timestamp) - : RtpSource(timestamp_ms, - source_id, - source_type, - rtp_timestamp, - {audio_level, absl::nullopt}) {} - RtpSource(int64_t timestamp_ms, uint32_t source_id, RtpSourceType source_type, diff --git a/modules/rtp_rtcp/source/source_tracker_unittest.cc b/modules/rtp_rtcp/source/source_tracker_unittest.cc index b64f03c469..4cd8ab88ca 100644 --- a/modules/rtp_rtcp/source/source_tracker_unittest.cc +++ b/modules/rtp_rtcp/source/source_tracker_unittest.cc @@ -266,10 +266,12 @@ TEST(SourceTrackerTest, OnFrameDeliveredRecordsSourcesDistinctSsrcs) { kAbsoluteCaptureTime, kReceiveTime1)})); int64_t timestamp_ms = clock.TimeInMilliseconds(); - constexpr RtpSource::Extensions extensions0 = {kAudioLevel0, - kAbsoluteCaptureTime}; - constexpr RtpSource::Extensions extensions1 = {kAudioLevel1, - kAbsoluteCaptureTime}; + constexpr RtpSource::Extensions extensions0 = { + .audio_level = kAudioLevel0, + .absolute_capture_time = kAbsoluteCaptureTime}; + constexpr RtpSource::Extensions extensions1 = { + .audio_level = kAudioLevel1, + .absolute_capture_time = kAbsoluteCaptureTime}; EXPECT_THAT(tracker.GetSources(), ElementsAre(RtpSource(timestamp_ms, kSsrc2, RtpSourceType::SSRC, @@ -354,12 +356,15 @@ TEST(SourceTrackerTest, OnFrameDeliveredUpdatesSources) { constexpr Timestamp kReceiveTime1 = Timestamp::Millis(61); constexpr Timestamp kReceiveTime2 = Timestamp::Millis(62); - constexpr RtpSource::Extensions extensions0 = {kAudioLevel0, - kAbsoluteCaptureTime0}; - constexpr RtpSource::Extensions extensions1 = {kAudioLevel1, - kAbsoluteCaptureTime1}; - constexpr RtpSource::Extensions extensions2 = {kAudioLevel2, - kAbsoluteCaptureTime2}; + constexpr RtpSource::Extensions extensions0 = { + .audio_level = kAudioLevel0, + .absolute_capture_time = kAbsoluteCaptureTime0}; + constexpr RtpSource::Extensions extensions1 = { + .audio_level = kAudioLevel1, + .absolute_capture_time = kAbsoluteCaptureTime1}; + constexpr RtpSource::Extensions extensions2 = { + .audio_level = kAudioLevel2, + .absolute_capture_time = kAbsoluteCaptureTime2}; SimulatedClock clock(1000000000000ULL); SourceTracker tracker(&clock); @@ -453,8 +458,9 @@ TEST(SourceTrackerTest, TimedOutSourcesAreRemoved) { clock.AdvanceTimeMilliseconds(SourceTracker::kTimeoutMs); - constexpr RtpSource::Extensions extensions1 = {kAudioLevel1, - kAbsoluteCaptureTime1}; + constexpr RtpSource::Extensions extensions1 = { + .audio_level = kAudioLevel1, + .absolute_capture_time = kAbsoluteCaptureTime1}; EXPECT_THAT( tracker.GetSources(),