Added timestamp logs, i.e. only tracing.
BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/482001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2030 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -18,6 +18,7 @@
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#include "receiver_fec.h"
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#include "rtp_rtcp_impl.h"
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#include "rtp_utility.h"
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#include "trace.h"
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namespace webrtc {
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WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
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@ -401,6 +402,9 @@ WebRtc_Word32 RTPReceiverVideo::ParseVideoCodecSpecificSwitch(
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if (retVal != 0) {
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return retVal;
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}
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WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "%s(timestamp:%u)",
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__FUNCTION__, rtpHeader->header.timestamp);
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// All receive functions release _criticalSectionReceiverVideo before
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// returning.
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switch (videoType) {
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@ -314,6 +314,8 @@ RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
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{
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return retVal;
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}
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WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "%s(timestamp:%u)",
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__FUNCTION__, captureTimeStamp);
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return 0;
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}
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@ -2073,6 +2073,8 @@ WebRtc_Word32 ViEChannel::FrameToRender(VideoFrame& video_frame) {
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arr_ofCSRC[0] = rtp_rtcp_.RemoteSSRC();
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no_of_csrcs = 1;
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}
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WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s(timestamp:%u)", __FUNCTION__, video_frame.TimeStamp());
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DeliverFrame(video_frame, no_of_csrcs, arr_ofCSRC);
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return 0;
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}
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