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| author | wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-03-06 23:49:08 +0000 |
|---|---|---|
| committer | wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-03-06 23:49:08 +0000 |
| commit | 9a8232203a0793a3bbd7edc8bba04a4b48e00210 (patch) | |
| tree | 7c2ab0789be2304ee9d5f13cde4f54b86c132393 /voice_engine/voice_engine.gyp | |
| parent | 7a560797ddeafa0472a0bd85aa9ad62b2530ffba (diff) | |
| download | webrtc-9a8232203a0793a3bbd7edc8bba04a4b48e00210.tar.gz | |
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.
This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.
BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'voice_engine/voice_engine.gyp')
| -rw-r--r-- | voice_engine/voice_engine.gyp | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/voice_engine/voice_engine.gyp b/voice_engine/voice_engine.gyp index 21aef153..1ae4e027 100644 --- a/voice_engine/voice_engine.gyp +++ b/voice_engine/voice_engine.gyp @@ -184,6 +184,7 @@ 'test/auto_test/standard/neteq_test.cc', 'test/auto_test/standard/network_test.cc', 'test/auto_test/standard/rtp_rtcp_before_streaming_test.cc', + 'test/auto_test/standard/rtp_rtcp_extensions.cc', 'test/auto_test/standard/rtp_rtcp_test.cc', 'test/auto_test/standard/voe_base_misc_test.cc', 'test/auto_test/standard/video_sync_test.cc', |