32 Commits

Author SHA1 Message Date
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
Henrik Kjellander
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
2017-07-01 14:48:18 +00:00
kjellander
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.

Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
2017-06-30 20:42:44 +00:00
minyue
20c84ccd48 Making FakeNetworkPipe demux audio and video packets.
BUG=None

Review-Url: https://codereview.webrtc.org/2794243002
Cr-Commit-Position: refs/heads/master@{#17629}
2017-04-10 23:57:57 +00:00
nisse
e5ad5ca06a Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
Reason for revert:
Intend to fix perf failures and reland.

Original issue's description:
> Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
>
> Reason for revert:
> Reverting since this seems to break multiple WebRTC Perf buildbots
>
> Original issue's description:
> > Don't hardcode MediaType::ANY in FakeNetworkPipe.
> >
> > Instead let each test set the appropriate media type. This simplifies
> > demuxing in Call and later in RtpTransportController.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2774463003
> > Cr-Commit-Position: refs/heads/master@{#17418}
> > Committed: 9c47b00e24
>
> TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2784543002
> Cr-Commit-Position: refs/heads/master@{#17427}
> Committed: 3a3bd50610

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2783853002
Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-30 06:57:43 +00:00
lliuu
3a3bd50610 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
Reason for revert:
Reverting since this seems to break multiple WebRTC Perf buildbots

Original issue's description:
> Don't hardcode MediaType::ANY in FakeNetworkPipe.
>
> Instead let each test set the appropriate media type. This simplifies
> demuxing in Call and later in RtpTransportController.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2774463003
> Cr-Commit-Position: refs/heads/master@{#17418}
> Committed: 9c47b00e24

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2784543002
Cr-Commit-Position: refs/heads/master@{#17427}
2017-03-28 16:40:59 +00:00
nisse
9c47b00e24 Don't hardcode MediaType::ANY in FakeNetworkPipe.
Instead let each test set the appropriate media type. This simplifies
demuxing in Call and later in RtpTransportController.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2774463003
Cr-Commit-Position: refs/heads/master@{#17418}
2017-03-28 11:59:41 +00:00
kthelgason
d701dfdeef remove more CriticalSectionWrappers.
BUG=webrtc:7035

Review-Url: https://codereview.webrtc.org/2779623002
Cr-Commit-Position: refs/heads/master@{#17392}
2017-03-27 14:24:57 +00:00
stefan
e9ad271db4 Increase the send-time history to 60 seconds.
This helps us avoid time-outs on really bad networks with long queues.
Also adding periodic logging of the fake network pipe's queue in milliseconds.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2687013005
Cr-Commit-Position: refs/heads/master@{#16532}
2017-02-10 14:09:28 +00:00
mflodman
997684079e Remove a few meaningless TODOs
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/2203363002 .

Cr-Commit-Position: refs/heads/master@{#13624}
2016-08-03 13:38:20 +00:00
philipel
536378bf37 Allow FakeNetworkPipe to drop packets in bursts.
The fake network pipe will still only drop packets at an average rate of
|loss_percent| but in bursts at an average length specified by
|avg_burst_loss_length|.

Also added the flag -avg_burst_loss_length to video loopback.

BUG=

Review-Url: https://codereview.webrtc.org/1995683003
Cr-Commit-Position: refs/heads/master@{#12969}
2016-05-31 10:20:28 +00:00
kwiberg
fd8be3468a Remove webrtc/base/scoped_ptr.h
This is a re-land of https://codereview.webrtc.org/1942823002

TBR=tommi@webrtc.org
BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1966423002
Cr-Commit-Position: refs/heads/master@{#12750}
2016-05-15 02:44:18 +00:00
kwiberg
6ab3db249b Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
Reason for revert:
Breaks user code. Said code needs to stop using scoped_ptr!

Original issue's description:
> Remove webrtc/base/scoped_ptr.h
>
> BUG=webrtc:5520
>
> NOTRY=True
>
> Committed: https://crrev.com/65fc62e9dd8a8716db625aaef76ab92f542ecc5a
> Cr-Commit-Position: refs/heads/master@{#12684}

TBR=tommi@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/1965063003
Cr-Commit-Position: refs/heads/master@{#12686}
2016-05-11 12:07:33 +00:00
kwiberg
65fc62e9dd Remove webrtc/base/scoped_ptr.h
BUG=webrtc:5520

NOTRY=True

Review-Url: https://codereview.webrtc.org/1942823002
Cr-Commit-Position: refs/heads/master@{#12684}
2016-05-11 11:29:38 +00:00
kwiberg
bfefb03ec1 Replace scoped_ptr with unique_ptr everywhere
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1937693002
Cr-Commit-Position: refs/heads/master@{#12581}
2016-05-01 21:53:55 +00:00
philipel
a2c55235ca Allow packets to be reordered in the fake network pipe.
BUG=

Review URL: https://codereview.webrtc.org/1606183002

Cr-Commit-Position: refs/heads/master@{#11384}
2016-01-26 16:42:00 +00:00
pbos
5ad935cb56 Remove mutable from rtc::CriticalSection members.
rtc::CriticalSection is now lockable from const methods and no longer
need to remain mutable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1613643004

Cr-Commit-Position: refs/heads/master@{#11367}
2016-01-25 11:52:53 +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
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
henrikg
3c089d751e Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Related CL: https://codereview.webrtc.org/1335923002/

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1345433002

Cr-Commit-Position: refs/heads/master@{#9953}
2015-09-16 12:37:52 +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
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 14:43:50 +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
stefan@webrtc.org
bfe6e08195 Add simulation of network effects to video_loopback tool.
Also add support for uniform random packet loss to the fake network pipe.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6803 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 12:30:18 +00:00
stefan@webrtc.org
b8e9e44eac Add full stack test cases with a fake network pipe.
R=pbos@webrtc.org
BUG=1872

Review URL: https://webrtc-codereview.appspot.com/20889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 11:29:06 +00:00
henrike@webrtc.org
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
mcasas@webrtc.org
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..."
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
henrike@webrtc.org
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +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