commit def01801f9bc2f3b802dad6572b98509fbe40123
author stefan <stefan@webrtc.org> Wed Nov 02 10:10:09 2016
committer Commit bot <commit-bot@chromium.org> Wed Nov 02 10:10:12 2016
tree 8ef8abe31dd73edd685da1e2aef28f5942bb0c2d
parent 04729690f5dddb1d4a5ab11139e1415b432b2e90

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

Introduced with r14870.

BUG=b/32591921

Review-Url: https://codereview.webrtc.org/2473663002
Cr-Original-Commit-Position: refs/heads/master@{#14883}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 572ae1212b488d874f6afcf87bc536319b122df7

audio/audio_receive_stream.cc

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index c325b9c..cc30939 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -123,6 +123,9 @@
       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.";
     }

audio/audio_send_stream.cc

diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 3906672..ad6366b 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -79,6 +79,9 @@
       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.";
     }