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Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.

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- 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@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2014-03-06 23:49:08 +00:00
parent 79047f99c1
commit ebdb0e3ad0
18 changed files with 490 additions and 300 deletions
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7
webrtc/voice_engine/channel.h
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@ -253,8 +253,9 @@ public:
int GetLocalSSRC(unsigned int& ssrc);
int GetRemoteSSRC(unsigned int& ssrc);
int GetRemoteCSRCs(unsigned int arrCSRC[15]);
int SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID);
int GetRTPAudioLevelIndicationStatus(bool& enable, unsigned char& ID);
int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
int SetRTCPStatus(bool enable);
int GetRTCPStatus(bool& enabled);
int SetRTCP_CNAME(const char cName[256]);
@ -438,6 +439,8 @@ private:
void RegisterReceiveCodecsToRTPModule();
int SetRedPayloadType(int red_payload_type);
int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type,
unsigned char id);
CriticalSectionWrapper& _fileCritSect;
CriticalSectionWrapper& _callbackCritSect;