59 Commits

Author SHA1 Message Date
pbos@webrtc.org
db6e3f8bc5 Fix root-relative includes for pacing/.
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4329 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 09:50:05 +00:00
hclam@chromium.org
2e402ce873 Enqueue packet in pacer if sending fails
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.

BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1693004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20 20:18:31 +00:00
stefan@webrtc.org
8ccb9f9716 Fixes some pacer/padding issues found while testing.
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.

BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-19 14:13:42 +00:00
stefan@webrtc.org
c3cc375499 Add support for padding in pacer.
This improves pacer-based padding by making sure it limits padding according to:
- Never pad more than 800 kbps.
- Padding + media should not go above a given target bitrate.

Also adds appropriate unittests to make sure we reach the given targets.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1582005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4168 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 09:36:56 +00:00
pwestin@webrtc.org
52b4e8871a Adding trace and changing pacing constants
BUG=1721,1722
R=mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1380005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 19:02:17 +00:00
pwestin@webrtc.org
52aa019e98 Avoid adding duplicates in pacer lists.
Review URL: https://webrtc-codereview.appspot.com/1329007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3899 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-25 17:35:56 +00:00
stefan@webrtc.org
bfacda60be Add interface to signal a network down event.
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-27 16:36:01 +00:00
pwestin@webrtc.org
db4185664c Introduced pause and resume to the pacer
Review URL: https://webrtc-codereview.appspot.com/1217007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3717 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-22 23:39:29 +00:00
pwestin@webrtc.org
b518017e71 Adding pacing module, will replace the transmission_bucket in the RTP module.
TESTED=unittest
Review URL: https://webrtc-codereview.appspot.com/930015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3073 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-09 20:56:23 +00:00