From d4adce4672c5860076eed5268cf50a46370268d6 Mon Sep 17 00:00:00 2001 From: solenberg Date: Thu, 17 Nov 2016 06:26:52 -0800 Subject: [PATCH] Remove Absolute Send Time from list of supported header extensions for audio streams. Follow-up to https://codereview.webrtc.org/2473663002/. BUG=b/32591921 Review-Url: https://codereview.webrtc.org/2501503004 Cr-Commit-Position: refs/heads/master@{#15132} --- webrtc/audio/audio_receive_stream.cc | 3 --- webrtc/audio/audio_send_stream.cc | 3 --- webrtc/config.cc | 3 +-- webrtc/media/engine/webrtcvoiceengine_unittest.cc | 12 +++++++----- 4 files changed, 8 insertions(+), 13 deletions(-) diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc index e2f8f14836..0d75307b4b 100644 --- a/webrtc/audio/audio_receive_stream.cc +++ b/webrtc/audio/audio_receive_stream.cc @@ -124,9 +124,6 @@ AudioReceiveStream::AudioReceiveStream( bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( kRtpExtensionTransportSequenceNumber, extension.id); RTC_DCHECK(registered); - } else if (extension.uri == RtpExtension::kAbsSendTimeUri) { - LOG(LS_WARNING) << RtpExtension::kAbsSendTimeUri - << " is no longer supported for audio."; } else { RTC_NOTREACHED() << "Unsupported RTP extension."; } diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc index cb5fe0905c..8c03dbaa48 100644 --- a/webrtc/audio/audio_send_stream.cc +++ b/webrtc/audio/audio_send_stream.cc @@ -79,9 +79,6 @@ AudioSendStream::AudioSendStream( channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id); } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { channel_proxy_->EnableSendTransportSequenceNumber(extension.id); - } else if (extension.uri == RtpExtension::kAbsSendTimeUri) { - LOG(LS_WARNING) << RtpExtension::kAbsSendTimeUri - << " is no longer supported for audio."; } else { RTC_NOTREACHED() << "Registering unsupported RTP extension."; } diff --git a/webrtc/config.cc b/webrtc/config.cc index 4b437e264d..a1789a7f1d 100644 --- a/webrtc/config.cc +++ b/webrtc/config.cc @@ -86,8 +86,7 @@ const char* RtpExtension::kPlayoutDelayUri = const int RtpExtension::kPlayoutDelayDefaultId = 6; bool RtpExtension::IsSupportedForAudio(const std::string& uri) { - return uri == webrtc::RtpExtension::kAbsSendTimeUri || - uri == webrtc::RtpExtension::kAudioLevelUri || + return uri == webrtc::RtpExtension::kAudioLevelUri || uri == webrtc::RtpExtension::kTransportSequenceNumberUri; } diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc index 3bab92ddc6..3b4670ea3e 100644 --- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc +++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc @@ -2071,12 +2071,14 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) { TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri); } -// Test support for absolute send time header extension. -TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) { - TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri); +// Test support for transport sequence number header extension. +TEST_F(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) { + TestSetSendRtpHeaderExtensions( + webrtc::RtpExtension::kTransportSequenceNumberUri); } -TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) { - TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri); +TEST_F(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) { + TestSetRecvRtpHeaderExtensions( + webrtc::RtpExtension::kTransportSequenceNumberUri); } // Test that we can create a channel and start sending on it.