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Fix crash when registering abs-send-time to AudioSend/ReceiveStream.

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Introduced with r14870.

BUG=b/32591921

Review-Url: https://codereview.webrtc.org/2473663002
Cr-Commit-Position: refs/heads/master@{#14883}
This commit is contained in:
stefan 2016-11-02 03:10:09 -07:00 committed by Commit bot
parent ae70876c00
commit 572ae1212b
2 changed files with 6 additions and 0 deletions
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3
webrtc/audio/audio_receive_stream.cc
View File

@ -123,6 +123,9 @@ 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

@ -79,6 +79,9 @@ 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.";
}