23 Commits

Author SHA1 Message Date
kwiberg
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
2016-10-01 05:29:53 +00:00
kwiberg
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +00:00
Stefan Holmer
ff2a6351e0 Add ramp-up tests for transport sequence number with and w/o audio.
Also add a perf metric tracking the average network latency.

The audio stream test is disabled for now since audio isn't included in bitrate allocation.

BUG=webrtc:5263
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1582833002 .

Cr-Commit-Position: refs/heads/master@{#11244}
2016-01-14 09:00:34 +00:00
Peter Boström
5811a39f14 Replace EventWrapper in video/, test/ and call/.
Makes use of rtc::Event which is simpler and can be used without
allocating additional objects on the heap.

Does not modify test/channel_transport/.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1487893004 .

Cr-Commit-Position: refs/heads/master@{#10968}
2015-12-10 12:03:00 +00:00
Peter Boström
d3c944755e Nuke TickTime::UseFakeClock.
Removes the global simulated time that affects (or breaks) following
tests in the same binary and replaces it with SimulatedClock.

BUG=webrtc:5318
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1512853002 .

Cr-Commit-Position: refs/heads/master@{#10947}
2015-12-09 10:21:09 +00:00
Peter Boström
8c38e8b9b9 Clean up PlatformThread.
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1476453002 .

Cr-Commit-Position: refs/heads/master@{#10812}
2015-11-26 16:45:57 +00:00
pbos
12411ef40e Move ThreadWrapper to ProcessThread in base.
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).

BUG=webrtc:5158
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1469013002

Cr-Commit-Position: refs/heads/master@{#10760}
2015-11-23 22:48:01 +00:00
Henrik Kjellander
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
stefan
f116bd0d7a Call OnSentPacket for all packets sent in the test framework.
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1419193002

Cr-Commit-Position: refs/heads/master@{#10430}
2015-10-27 15:29:47 +00:00
stefan
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
Peter Boström
f2f828374c Use rtc::CriticalSection in webrtc/video/.
Removes heap allocation from CriticalSection creation.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50839004

Cr-Commit-Position: refs/heads/master@{#9126}
2015-05-01 14:25:53 +00:00
tommi@webrtc.org
38492c5b6f Re-land 8810 "- Add a SetPriority method to ThreadWr..."
> Revert 8810 "- Add a SetPriority method to ThreadWrapper"
> Seeing if this is causing roll issues.
> 
> > - Add a SetPriority method to ThreadWrapper
> > - Remove 'priority' from CreateThread and related member variables from implementations
> > - Make supplying a name for threads, non-optional
> > 
> > BUG=
> > R=magjed@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/44729004
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/48609004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50459005

Cr-Commit-Position: refs/heads/master@{#8819}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-22 14:42:46 +00:00
tommi@webrtc.org
90a1cb4630 Revert 8810 "- Add a SetPriority method to ThreadWrapper"
Seeing if this is causing roll issues.

> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
> 
> BUG=
> R=magjed@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/44729004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48609004

Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-22 14:34:46 +00:00
tommi@webrtc.org
b6817d793f - Add a SetPriority method to ThreadWrapper
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional

BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44729004

Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 15:52:43 +00:00
pbos@webrtc.org
86639737b8 Remove thread id from ThreadWrapper::Start().
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.

BUG=4413
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43699004

Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 00:07:45 +00:00
pbos@webrtc.org
d5ce2e63df Remove EventWrapper::Reset().
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 14:58:38 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
henrik.lundin@webrtc.org
c0e9aebe8f Add SetConfig method to FakeNetworkPipe and to DirectTransport
This method allow the user to change the network configuration
during run-time. This is useful when testing how components react
to changing bandwidth.

BUG=2636
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-26 13:34:52 +00:00
stefan@webrtc.org
faada6e604 Integrate fake_network_pipe into direct_transport.
TEST=trybots
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5321 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 20:28:25 +00:00
pbos@webrtc.org
27326b6a42 Rename newapi::Transport::SendRTP()->SendRtp().
Also fit rampup_tests.cc to use internal::TransportAdapter instead of
implementing its own.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 12:17:04 +00:00
stefan@webrtc.org
b082ade3db Hook up audio/video sync to Call.
Adds an end-to-end audio/video sync test.

BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 11:45:11 +00:00
stefan@webrtc.org
69969e2e2f Improve Call tests for RTX.
Also does some refactoring to reuse RtpRtcpObserver.

BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 12:32:15 +00:00
pbos@webrtc.org
16e03b7bd8 Separate Call API/build files from video_engine/.
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 16:32:01 +00:00