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Remove Absolute Send Time from list of supported header extensions for audio streams.

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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}
This commit is contained in:
solenberg 2016-11-17 06:26:52 -08:00 committed by Commit bot
parent fbb374d8ed
commit d4adce4672
4 changed files with 8 additions and 13 deletions
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3
webrtc/audio/audio_receive_stream.cc
View File

@ -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.";
}

3
webrtc/audio/audio_send_stream.cc
View File

@ -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.";
}

3
webrtc/config.cc
View File

@ -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;
}

12
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -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.