f9e1b922ef
Ignore Camera and Flip bits in CVO when parsing video rotation
...
Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
The Camera and Flip bit is still unimplemented and will just be ignored
though.
BUG=webrtc:6120
R=danilchap@webrtc.org , pthatcher@webrtc.org , tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2280703002 .
Cr-Commit-Position: refs/heads/master@{#14027}
2016-09-01 17:58:28 +00:00
2800d74fcf
Change RtpSender::OnReceiveNACK name and signature
...
Name changed to follow style.
list replaced with vector to decrease number of included headers.
R=philipel@webrtc.org
Review URL: https://codereview.webrtc.org/2276833003 .
Cr-Commit-Position: refs/heads/master@{#13938}
2016-08-26 16:49:05 +00:00
a246cfb8b5
Don't include RTP headers in send-side BWE.
...
When they are included there will be a mismatch between what the BWE says and
what the encoder is allowed to use, causing us to send more than the network
can handle.
BUG=webrtc:6247
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/2269923003 .
Cr-Commit-Position: refs/heads/master@{#13866}
2016-08-23 15:51:57 +00:00
71fead2146
Reland of StartTimestamp generated randomly in RtpSender constructor (patchset #1 id:1 of https://codereview.webrtc.org/2248413002/ )
...
Reason for revert:
Reland: downstream code expectation about rtp_sender timestamp adjusted.
Original issue's description:
> Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
>
> Reason for revert:
> Breaks downstream code.
>
> Original issue's description:
> > StartTimestamp generated randomly in RtpSender constructor
> > instead of not-randomly at SetSendingState(true)
> > Renamed to timestamp_offset_ to better match meaning of the variable.
> >
> > R=asapersson@webrtc.org , terelius@webrtc.org
> >
> > Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> > Cr-Commit-Position: refs/heads/master@{#13796}
>
> TBR=asapersson@webrtc.org ,terelius@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/86c96948e340cf8b879bddb0c7293f3b5ad4dad4
> Cr-Commit-Position: refs/heads/master@{#13798}
TBR=asapersson@webrtc.org ,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2257083002
Cr-Commit-Position: refs/heads/master@{#13811}
2016-08-18 09:02:16 +00:00
86c96948e3
Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
...
Reason for revert:
Breaks downstream code.
Original issue's description:
> StartTimestamp generated randomly in RtpSender constructor
> instead of not-randomly at SetSendingState(true)
> Renamed to timestamp_offset_ to better match meaning of the variable.
>
> R=asapersson@webrtc.org , terelius@webrtc.org
>
> Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> Cr-Commit-Position: refs/heads/master@{#13796}
TBR=asapersson@webrtc.org ,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2248413002
Cr-Commit-Position: refs/heads/master@{#13798}
2016-08-17 15:12:27 +00:00
4466782ae4
StartTimestamp generated randomly in RtpSender constructor
...
instead of not-randomly at SetSendingState(true)
Renamed to timestamp_offset_ to better match meaning of the variable.
R=asapersson@webrtc.org , terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2241193002 .
Cr-Commit-Position: refs/heads/master@{#13796}
2016-08-17 13:07:49 +00:00
31e4e806b1
RtpPacketHistory rewritten to use RtpPacket class.
...
RtpSender updated to use new version of RtpPacketHistory.
BUG=webrtc:5261
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1945773002 .
Cr-Commit-Position: refs/heads/master@{#13626}
2016-08-03 16:27:50 +00:00
525df3ffd1
Add EncodedImageCallback::OnEncodedImage().
...
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.
BUG=chromium:621691
R=niklas.enbom@webrtc.org , sprang@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2089773002 .
Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Original-Commit-Position: refs/heads/master@{#13615}
Cr-Commit-Position: refs/heads/master@{#13617}
2016-08-03 00:46:47 +00:00
51db4dd1bd
Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #14 id:300001 of https://codereview.chromium.org/2089773002/ )
...
Reason for revert:
broke browser_tests
Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org , sprang@webrtc.org , stefan@webrtc.org
>
> Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Original-Commit-Position: refs/heads/master@{#13613}
> Cr-Commit-Position: refs/heads/master@{#13615}
TBR=pbos@webrtc.org ,mflodman@webrtc.org ,sprang@webrtc.org ,stefan@webrtc.org ,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2203233002
Cr-Commit-Position: refs/heads/master@{#13616}
2016-08-03 00:33:47 +00:00
4c7f4cd2ef
Add EncodedImageCallback::OnEncodedImage().
...
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.
BUG=chromium:621691
R=niklas.enbom@webrtc.org , sprang@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2089773002 .
Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Commit-Position: refs/heads/master@{#13615}
2016-08-02 22:14:51 +00:00
ac4dc2cefe
Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #13 id:280001 of https://codereview.webrtc.org/2089773002/ )
...
Reason for revert:
broke internal tests
Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org , sprang@webrtc.org , stefan@webrtc.org
>
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Commit-Position: refs/heads/master@{#13613}
TBR=pbos@webrtc.org ,mflodman@webrtc.org ,sprang@webrtc.org ,stefan@webrtc.org ,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2206743002
Cr-Commit-Position: refs/heads/master@{#13614}
2016-08-02 21:33:21 +00:00
ad34dbe934
Add EncodedImageCallback::OnEncodedImage().
...
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.
BUG=chromium:621691
R=niklas.enbom@webrtc.org , sprang@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2089773002 .
Cr-Commit-Position: refs/heads/master@{#13613}
2016-08-02 20:44:25 +00:00
737336d37a
Add NACK rate throttling for audio channels.
...
Not really used for audio today (already in place for video), but should
still function anyway.
BUG=
R=henrika@webrtc.org , minyue@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2181383002 .
Cr-Commit-Position: refs/heads/master@{#13571}
2016-07-29 10:59:49 +00:00
ec4f068bcd
Style cleanups in RtpSender.
...
- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h
R=danilchap@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2067673004 .
Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 22:19:18 +00:00
a23fc626a2
Fix bug where transport sequence numbers are allocated for packets without the header extension registered.
...
This is an issue if the sequence numbers are to be used to compute packet loss statistics since it introduces gaps which are not related to loss.
Also making sure that the header extensions are properly guarded by the send crit sect.
Review-Url: https://codereview.webrtc.org/2190913002
Cr-Commit-Position: refs/heads/master@{#13557}
2016-07-28 14:56:45 +00:00
cd349d9743
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
...
Reason for revert:
Upstream fixes in place, should be OK now.
Original issue's description:
> Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
>
> Reason for revert:
> Breaks upstream code.
>
> Original issue's description:
> > Refactor NACK bitrate allocation
> >
> > Nack bitrate allocation should not be done on a per-rtp-module basis,
> > but rather shared bitrate pool per call. This CL moves allocation to the
> > pacer and cleans up a bunch if bitrate stats handling.
> >
> > BUG=
> > R=danilchap@webrtc.org , stefan@webrtc.org , tommi@webrtc.org
> >
> > Committed: 5fc59e810bTBR=tommi@webrtc.org ,danilchap@webrtc.org ,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e5dd44101eca485f5ad12e5f7ce6f6b0d204116b
> Cr-Commit-Position: refs/heads/master@{#13417}
TBR=tommi@webrtc.org ,danilchap@webrtc.org ,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=
Review-Url: https://codereview.webrtc.org/2146013002
Cr-Commit-Position: refs/heads/master@{#13465}
2016-07-13 16:11:38 +00:00
a49f1105eb
Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ )
...
Reason for revert:
It keeps breaking upstream.
Original issue's description:
> Reland Issue 2061423003: Refactor NACK bitrate allocation
>
> This is a reland of https://codereview.webrtc.org/2061423003/
> Which was reverted in https://codereview.webrtc.org/2131913003/
>
> The reason for the revert was that some upstream code used
> RtpSender::SetTargetBitrate(). I've added that back as a no-op until we
> it's been brought up to date.
>
> TBR=tommi@webrtc.org
>
> Committed: 05ce4ae31fTBR=tommi@webrtc.org ,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2130423002
Cr-Commit-Position: refs/heads/master@{#13419}
2016-07-08 18:02:02 +00:00
05ce4ae31f
Reland Issue 2061423003: Refactor NACK bitrate allocation
...
This is a reland of https://codereview.webrtc.org/2061423003/
Which was reverted in https://codereview.webrtc.org/2131913003/
The reason for the revert was that some upstream code used
RtpSender::SetTargetBitrate(). I've added that back as a no-op until we
it's been brought up to date.
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2131313002 .
Cr-Commit-Position: refs/heads/master@{#13418}
2016-07-08 17:11:23 +00:00
e5dd44101e
Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
...
Reason for revert:
Breaks upstream code.
Original issue's description:
> Refactor NACK bitrate allocation
>
> Nack bitrate allocation should not be done on a per-rtp-module basis,
> but rather shared bitrate pool per call. This CL moves allocation to the
> pacer and cleans up a bunch if bitrate stats handling.
>
> BUG=
> R=danilchap@webrtc.org , stefan@webrtc.org , tommi@webrtc.org
>
> Committed: 5fc59e810bTBR=tommi@webrtc.org ,danilchap@webrtc.org ,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2131913003
Cr-Commit-Position: refs/heads/master@{#13417}
2016-07-08 16:39:02 +00:00
5fc59e810b
Refactor NACK bitrate allocation
...
Nack bitrate allocation should not be done on a per-rtp-module basis,
but rather shared bitrate pool per call. This CL moves allocation to the
pacer and cleans up a bunch if bitrate stats handling.
BUG=
R=danilchap@webrtc.org , stefan@webrtc.org , tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2061423003 .
Cr-Commit-Position: refs/heads/master@{#13416}
2016-07-08 16:15:29 +00:00
6b4b5f3770
Add sender controlled playout delay limits
...
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.
The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.
There are no tests at this time and most of testing is done with chromium
webrtc prototype.
On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.
BUG=webrtc:5895
Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
2016-06-08 07:24:30 +00:00
a1ed0b3241
Revert "Revert of Propagate probing cluster id to SendTimeHistory. (patchset #5 id:80001 of https://codereview.webrtc.org/2005313003/ )"
...
This reverts commit 46948c17fd09e4957bebc8ea61f0a8e77ff84b48.
TBR=mflodman@webrtc.org
BUG=webrtc:5859
Review-Url: https://codereview.webrtc.org/2032473002
Cr-Commit-Position: refs/heads/master@{#12992}
2016-06-01 13:31:22 +00:00
46948c17fd
Revert of Propagate probing cluster id to SendTimeHistory. (patchset #5 id:80001 of https://codereview.webrtc.org/2005313003/ )
...
Reason for revert:
Breaks google3 buildbot: http://webrtc-buildbot-master.mtv.corp.google.com:21000/builders/WebRTC%20google3%20Importer/builds/8640
Original issue's description:
> Propagate probing cluster id to SendTimeHistory, both for packets and padding.
>
> BUG=webrtc:5859
>
> Committed: https://crrev.com/5be28c848b91bc6e4800eac07a3f5ac09a32ad70
> Cr-Commit-Position: refs/heads/master@{#12985}
TBR=danilchap@webrtc.org ,stefan@webrtc.org ,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5859
Review-Url: https://codereview.webrtc.org/2032463003
Cr-Commit-Position: refs/heads/master@{#12987}
2016-06-01 11:04:49 +00:00
5be28c848b
Propagate probing cluster id to SendTimeHistory, both for packets and padding.
...
BUG=webrtc:5859
Review-Url: https://codereview.webrtc.org/2005313003
Cr-Commit-Position: refs/heads/master@{#12985}
2016-06-01 09:49:29 +00:00
35151f35ec
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
...
- "WebRTC.Video.SendDelayInMs"
Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.
BUG=webrtc:5215
Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
2016-05-03 06:44:11 +00:00
c04305200e
Reland of move VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1935753002/ )
...
Reason for revert:
Not root cause for perf regression (regression still ongoing).
Original issue's description:
> Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ )
>
> Reason for revert:
> Speculative revert for perf regression.
>
> Original issue's description:
> > Remove VCMQmRobustness.
> >
> > Class contained a lot of not-really-wired-up functionality that ended up
> > being complicated ways of saying return 1; or return false;. This
> > removes this dependency that complicates code readability significantly.
> >
> > BUG=webrtc:5066
> > R=marpan@google.com , marpan@webrtc.org
> > TBR=stefan@webrtc.org
> >
> > Committed: https://crrev.com/73894369791cb5eedc8788baf918ec07d11d351d
> > Cr-Commit-Position: refs/heads/master@{#12516}
>
> TBR=marpan@webrtc.org ,stefan@webrtc.org ,marpan@google.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5066, chromium:607838
>
> Committed: https://crrev.com/602316c3cd8556cc78d44f3ea4cd5fc8e70d9417
> Cr-Commit-Position: refs/heads/master@{#12572}
TBR=marpan@webrtc.org ,stefan@webrtc.org ,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5066, chromium:607838
Review-Url: https://codereview.webrtc.org/1941643002
Cr-Commit-Position: refs/heads/master@{#12583}
2016-05-02 00:19:13 +00:00
602316c3cd
Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ )
...
Reason for revert:
Speculative revert for perf regression.
Original issue's description:
> Remove VCMQmRobustness.
>
> Class contained a lot of not-really-wired-up functionality that ended up
> being complicated ways of saying return 1; or return false;. This
> removes this dependency that complicates code readability significantly.
>
> BUG=webrtc:5066
> R=marpan@google.com , marpan@webrtc.org
> TBR=stefan@webrtc.org
>
> Committed: https://crrev.com/73894369791cb5eedc8788baf918ec07d11d351d
> Cr-Commit-Position: refs/heads/master@{#12516}
TBR=marpan@webrtc.org ,stefan@webrtc.org ,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5066, chromium:607838
Review-Url: https://codereview.webrtc.org/1935753002
Cr-Commit-Position: refs/heads/master@{#12572}
2016-04-29 23:10:36 +00:00
84be511ac0
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
...
(This is a re-land of https://codereview.webrtc.org/1921233002 , which
got reverted for breaking Chromium.)
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1923133002
Cr-Commit-Position: refs/heads/master@{#12522}
2016-04-27 08:20:08 +00:00
7389436979
Remove VCMQmRobustness.
...
Class contained a lot of not-really-wired-up functionality that ended up
being complicated ways of saying return 1; or return false;. This
removes this dependency that complicates code readability significantly.
BUG=webrtc:5066
R=marpan@google.com , marpan@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1917083003 .
Cr-Commit-Position: refs/heads/master@{#12516}
2016-04-26 21:54:41 +00:00
52d4e6bf5e
Revert of Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/ (patchset #1 id:40001 of https://codereview.webrtc.org/1921233002/ )
...
Reason for revert:
Fails on Chromium FYI bots.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/5392/
Original issue's description:
> Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
>
> BUG=webrtc:5520
>
> Committed: https://crrev.com/2c27a062ee46258abe9facc2cceee74f09bf6a99
> Cr-Commit-Position: refs/heads/master@{#12511}
TBR=tommi@webrtc.org ,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1924443002
Cr-Commit-Position: refs/heads/master@{#12513}
2016-04-26 16:32:09 +00:00
2c27a062ee
Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
...
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1921233002
Cr-Commit-Position: refs/heads/master@{#12511}
2016-04-26 15:38:03 +00:00
d98d457134
Remove "This file includes unit tests" comments.
...
Superfluous, neteq_unittests.cc conveys that the file contains unittests
for NetEq for instance.
BUG=
TBR=henrik.lundin@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1922103002 .
Cr-Commit-Position: refs/heads/master@{#12504}
2016-04-26 11:44:02 +00:00
6021fe2b1e
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
...
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1803923003
Cr-Commit-Position: refs/heads/master@{#12003}
2016-03-15 18:41:58 +00:00
f5dca48dc0
Add transport sequence number on the non-pacer path of the rtp sender.
...
BUG=4173
R=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1635093002 .
Cr-Commit-Position: refs/heads/master@{#11395}
2016-01-27 11:59:05 +00:00
429c345b02
Fixes a bug which incorrectly logs incoming RTCP as outgoing.
...
Adds logging to RTPSender and RTCPSender, pushing an event log pointer from Channel through ModuleRtpRtcpImpl to the Sender objects.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1571283002
Cr-Commit-Position: refs/heads/master@{#11336}
2016-01-21 13:42:10 +00:00
d9e62f5837
Fixed sending Rtp packets with non zero csrcs and certain extensions.
...
Added test that fails because of given issue.
BUG=webrtc:5413
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1586523003
Cr-Commit-Position: refs/heads/master@{#11258}
2016-01-14 22:55:23 +00:00
5d332ac8ff
Fix expectation bug in the RTPSender unit test.
...
The current expectation for InsertPacket(...) uses WillRepeatedly, which accepts if the function is called zero or more times. This CL changes this to either a fixed number of calls, or at least a positive number of calls.
Review URL: https://codereview.webrtc.org/1585783003
Cr-Commit-Position: refs/heads/master@{#11256}
2016-01-14 22:37:43 +00:00
f6975f4613
[rtp_rtcp] Lint errors cleaned from rtp_utility
...
R=åsapersson
BUG=webrtc:5277
Review URL: https://codereview.webrtc.org/1539423003
Cr-Commit-Position: refs/heads/master@{#11131}
2015-12-28 18:18:52 +00:00
6db6cdc604
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
...
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1513303003
Cr-Commit-Position: refs/heads/master@{#11025}
2015-12-15 10:54:50 +00:00
b8b6fbb7a5
lint build/include errors fixed in rtp_rtcp module
...
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1505993003
Cr-Commit-Position: refs/heads/master@{#10971}
2015-12-10 13:05:35 +00:00
162abd3562
lint whitespace warning removed from most rtp_rtcp/source/ files
...
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.
BUG=webrtc:5277
R=pbos@webrtc.org , mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512493002
Cr-Commit-Position: refs/heads/master@{#10966}
2015-12-10 10:39:45 +00:00
ff761fba82
modules: more interface -> include renames
...
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
98f53510b2
system_wrappers: rename interface -> include
...
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
0a87ffcaad
Fix bug in how send timestamps are converted to 24 bits.
...
BUG=webrtc:4173
R=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1412683004 .
Cr-Commit-Position: refs/heads/master@{#10356}
2015-10-21 11:42:09 +00:00
22993e1a0c
Unify FrameType and VideoFrameType.
...
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.
BUG=webrtc:5042
R=magjed@webrtc.org , mflodman@webrtc.org , henrik.lundin@webrtc.org , solenberg@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1371043003
Cr-Commit-Position: refs/heads/master@{#10320}
2015-10-19 09:39:15 +00:00
e23e737177
Disable pacer disabling.
...
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
2015-10-08 09:44:29 +00:00
1d8a506405
Add a PacketOptions struct to webrtc::Transport.
...
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
2015-10-02 10:39:40 +00:00
38778b046f
Add unit test for nack bandwidth constraint.
...
BUG=
Review URL: https://codereview.webrtc.org/1341743002
Cr-Commit-Position: refs/heads/master@{#10111}
2015-09-29 16:48:30 +00:00
2d566686a2
Unify Transport and newapi::Transport interfaces.
...
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1369263002
Cr-Commit-Position: refs/heads/master@{#10096}
2015-09-28 16:59:36 +00:00
ebbf8a805b
Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it.
...
BUG=
Review URL: https://codereview.webrtc.org/1350163005
Cr-Commit-Position: refs/heads/master@{#10005}
2015-09-21 22:11:18 +00:00
