Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

BUG=webrtc:5565
Committed: https://crrev.com/e5b4141746f31754c27f90f5e2eb70406288d51a
Cr-Commit-Position: refs/heads/master@{#13842}

[danilchap, 2016-08-18 16:04:03]

Description was changed from

==========
Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData
==========

to

==========
Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

BUG=webrtc:5565
==========

[danilchap, 2016-08-18 16:09:08]

danilchap@webrtc.org changed reviewers:
+ stefan@webrtc.org

[danilchap, 2016-08-18 16:09:08]

This cleanup remove one critsect and will simplify propagating RtpPacket class
on the sender side.

No functional changes expected except some tracing would now switch to rtp
timestamp as written in the network packet instead of as created near capture
device.

ptal.

[Sergey Ulanov, 2016-08-19 18:19:13]

sergeyu@chromium.org changed reviewers:
+ sergeyu@chromium.org

[Sergey Ulanov, 2016-08-19 18:19:13]

Thanks for the cleanup!

https://codereview.webrtc.org/2249223005/diff/1/webrtc/modules/rtp_rtcp/sourc...
File webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc (right):

https://codereview.webrtc.org/2249223005/diff/1/webrtc/modules/rtp_rtcp/sourc...
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc:286: rtp_timestamp, "seqnum",
Is it necessary to pass the same timestamp twice here?

[Sergey Ulanov, 2016-08-19 18:19:55]

LGTM, except that one question about tracing.

[danilchap, 2016-08-20 11:50:47]

https://codereview.webrtc.org/2249223005/diff/1/webrtc/modules/rtp_rtcp/sourc...
File webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc (right):

https://codereview.webrtc.org/2249223005/diff/1/webrtc/modules/rtp_rtcp/sourc...
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc:286: rtp_timestamp, "seqnum",
On 2016/08/19 18:19:13, Sergey Ulanov wrote:
> Is it necessary to pass the same timestamp twice here?

They have different meaning: first timestamp is a key to find matching
TRACE_EVENT_ASYNC_BEGIN (it is in rtp_sender.cc)
2nd timestamp is part of key-value pair with key="timestamp", something to
plot/use in stats.

[danilchap, 2016-08-22 10:34:23]

The CQ bit was checked by danilchap@webrtc.org

[commit-bot: I haz the power, 2016-08-22 10:34:28]

CQ is trying da patch. Follow status at
 
https://chromium-cq-status.appspot.com/v2/patch-status/codereview.webrtc.org/...

[commit-bot: I haz the power, 2016-08-22 10:39:26]

Description was changed from

==========
Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

BUG=webrtc:5565
==========

to

==========
Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

BUG=webrtc:5565
==========

[commit-bot: I haz the power, 2016-08-22 10:39:27]

Committed patchset #1 (id:1)

[commit-bot: I haz the power, 2016-08-22 10:39:35]

Description was changed from

==========
Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

BUG=webrtc:5565
==========

to

==========
Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

BUG=webrtc:5565

Committed: https://crrev.com/e5b4141746f31754c27f90f5e2eb70406288d51a
Cr-Commit-Position: refs/heads/master@{#13842}
==========

[commit-bot: I haz the power, 2016-08-22 10:39:36]

Patchset 1 (id:??) landed as
https://crrev.com/e5b4141746f31754c27f90f5e2eb70406288d51a
Cr-Commit-Position: refs/heads/master@{#13842}