Commit Graph

22 Commits

Author SHA1 Message Date
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
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
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
92220ffe9f Low-bandwidth audio testing
The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout.

The test cases are meant to try different network and audio configs (more are planned in the future).

The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard.

BUG=webrtc:7229
NOTRY=True

Review-Url: https://codereview.webrtc.org/2694203002
Cr-Commit-Position: refs/heads/master@{#17356}
2017-03-23 10:40:03 +00:00
a514584b9a Add the ability to read/write to WAV files in FakeAudioDevice
BUG=webrtc:7229

Review-Url: https://codereview.webrtc.org/2717623003
Cr-Commit-Position: refs/heads/master@{#17230}
2017-03-14 16:01:47 +00:00
ac61b745df Refactor FakeAudioDevice to have separate methods for starting recording and playout.
Also, change FakeAudioDevice to generate a sine tone instead of using a file.

TBR=henrika@webrtc.org, stefan@webrtc.org

BUG=webrtc:7080

Review-Url: https://codereview.webrtc.org/2652803002
Cr-Commit-Position: refs/heads/master@{#16385}
2017-01-31 21:32:49 +00:00
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
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
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
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
9c6a0c7f6d Added A/V sync tests with drifting clocks.
adding 30% drift to media generator (e.g. audio frame generated every 7ms instead of promised 10ms) works fine
adding 2% drift to video ntp-timestamp-stamper makes A/V sync fail.

BUG=webrtc:5504
R=pbos@webrtc.org,stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1674413004

Cr-Commit-Position: refs/heads/master@{#11556}
2016-02-10 18:54:52 +00:00
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
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
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
dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00
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
64c0366908 Revert "Revert "Split EventWrapper in twain.""
This reverts commit cf3c83e76c273309558c86fda915410f65b7a899.

Reverting EventWrapper split did not fix the issue, re-landing.

BUG=chromium:470013
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49629004

Cr-Commit-Position: refs/heads/master@{#8946}
2015-04-08 09:24:25 +00:00
cf3c83e76c Revert "Split EventWrapper in twain."
This reverts commit 9509fbfc301dd5412804ce5731afedc81480f2f8.

This is to debug a Chromium issue that WebRTC hangs if there is > 1 PeerConnection active in the browser on Win XP.

BUG=

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43019004

Cr-Commit-Position: refs/heads/master@{#8912}
2015-04-01 14:31:45 +00:00
9509fbfc30 Split EventWrapper in twain.
I'm splitting the timer functions in EventWrapper into a separate interface.
- Users of the timer functions have different needs than users of a generic event
- Providing a default implementation for EventWrapper that simply uses rtc::Event.

This means that clients of WebRTC that don't use the relatively few classes, typically rendering classes, that depend on the event timer functionality, also don't pull in dependencies on multimedia timers.

R=mflodman@webrtc.org, mflodman
BUG=

Review URL: https://webrtc-codereview.appspot.com/48599004

Cr-Commit-Position: refs/heads/master@{#8833}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8833 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-23 16:25:46 +00:00
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00
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
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