7686f0ddda
Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
...
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.
It make the simliar feature's implementation consistent.
R=andrew@webrtc.org , fischman@webrtc.org , fischman@chromium.org
BUG=334447
Review URL: https://webrtc-codereview.appspot.com/7769006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5517 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 17:42:34 +00:00
6f0ca57fb2
Add experiment: SkipEncodingUnusedStreams
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5514 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 09:20:51 +00:00
607c805b87
Roll chromium_revision 245382:249215
...
The find_depot_tools.py is needed to workaround the import
error we get from gyp_chromium when importing it in
webrtc/build/gyp_webrtc (to avoid code duplication).
gyp_chromium introduced a dependency on it in
http://crrev.com/245412 but as we cannot sync all of Chrome's
src/tools (it's quite big), we'll work around this by
adding an empty find_depot_tools module.
The removal of the Cygwin relates to
http://crrev.com/248802 which is a step on the way to remove
Cygwin in Chromium. We seem to already be able to remove it
entirely for WebRTC though.
Changes in the isolate framework required us to update our
copies of the isolate.gypi files.
BUG=none
TEST=trybots passing on all platforms
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5512 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-09 18:38:31 +00:00
ad3035fc9e
Fix WindowCapturerWin to unselect bitmap before destroying DC.
...
BUG=https://code.google.com/p/webrtc/issues/detail?id=2901
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5504 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 21:24:04 +00:00
9510e53cc0
Make VideoReceiveStream::GetStats() const.
...
BUG=
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5501 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 15:32:45 +00:00
09315705b9
Wire up statistics in video receive stream of new API
...
This CL includes Call tests that test both send and receive sides.
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 12:06:29 +00:00
77c917a6ee
Plot the capacity of a trace-based delivery filter.
...
Breaks out the instantaneous rate counters to its own class.
R=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7999005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5494 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-06 16:34:47 +00:00
f928f5c87c
Use system's cpu_features library
...
Remove the copied cpu_featrues.c/h
Use the NDK's cpu_features.a or the one build from android source.
This issue blocked libvpx roll.
BUG=334447
R=andrew@webrtc.org , fischman@webrtc.org , henrike@webrtc.org , wjia@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5492 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 18:43:46 +00:00
c88d3368d5
Add delay and send/receive throughput plots to BWE simulation.
...
R=mflodman@webrtc.org , solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 15:57:14 +00:00
75642fcd9a
Implementing replacement audio support in neteq_rtpplay
...
This CL makes it possible to replace the payload in an RTP stream
with audio from another (PCM) file. The new payload will be encoded as
PCM16b. The RTP headers will be updated to reflect this change of
payload type.
BUG=2834
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 08:49:13 +00:00
e6ab21b9ca
Fixing a bug in DummyRTPpacket
...
This bug caused writing outside allocated memory when RTP header
extensions were used.
BUG=2834
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8009005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5489 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 08:46:46 +00:00
54744918ef
Update AudioProcessing::Create docs.
...
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/8039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 06:30:29 +00:00
20a60ea39d
Fix a cursor capturing issue on Windows.
...
The input position to WindowFromPoint should be relative to the desktop, not
relative to the window; if the result from WindowFromPoint is a child window
of the shared top window, it should be captured.
BUG=
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5487 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 17:49:12 +00:00
0e5a2b5de6
Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered.
...
This can happen when switching between multiple streams and a single while getting feedback from the receiver.
BUG=2881
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5486 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 14:38:25 +00:00
3e6c41c48f
Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents"
...
This reverts commit r5479.
R=henrika@webrtc.org
BUG=2880
Review URL: https://webrtc-codereview.appspot.com/7989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5485 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 10:45:14 +00:00
064b32acbb
Fix locking in LoopBackTransport::StorePacket.
...
The critical section in StorePacket was unnamed and only existed in
expression scope. Added GUARDED_BY annotations (which caught the bug),
then fixed it by naming the variable.
BUG=2880
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5484 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 09:42:02 +00:00
f6a638e001
Trivial rename of non-compile time consts.
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7669006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5482 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 01:31:28 +00:00
f6b8f496ee
Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents
...
Issue: https://code.google.com/p/webrtc/issues/detail?id=2880
R=andrew@webrtc.org
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 21:34:35 +00:00
422fdbf502
Wire up feedback to VideoSender.
...
BUG=
R=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5474 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 16:33:50 +00:00
c9ee412070
Re-enabling audio processing tests
...
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5473 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 14:41:57 +00:00
c1e28038ba
Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-02 15:30:20 +00:00
1af5ea0538
Implement single monitor capture on Mac.
...
BUG=2787, 2824
TESTED=MacBook Pro Retina with an external monitor; verified changing display configuration while capturing; add/remove monitor while capturing; verified cursor position.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5471 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-01 02:03:24 +00:00
83aee8f450
Fixing test name for NetEqPerformanceTest
...
The naming did not follow conventions.
BUG=2859
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5469 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 11:46:34 +00:00
bdc5ed2e7d
Add configuration for cpu overuse detection to video send stream.
...
BUG=2422
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5468 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 10:05:07 +00:00
7d7f08957c
Add gyp_webrtc script to generate projects.
...
The reason for this is that http://crrev.com/245412
introduces a dependency of Chrome's src/build/gyp_chromium
to src/tools/find_depot_tools.py, which we don't have
synced in WebRTC (src/tools is very big).
Offline discussions shows that we cannot rely on syncing
individual subdirectories from Chrome in the future, but
maintaining our own gyp_webrtc file will at least buy us
some time for now, so we can roll past that chromium_revision
in WebRTC DEPS.
Overview of differences between gyp_webrtc and gyp_chromium
(and how we previously used gyp_chromium):
* No .gyp file needs to be passed (defaults to all.gyp)
* CHROMIUM_GYP_FILE is ignored (i.e. cannot be used to
specify an alternate .gyp file to process)
* Ninja is used by default on all platforms unless GYP_GENERATORS
is set.
* Gyp syntax check is always on
* Gyp circular dependency check is always on
* No support for automatic toolchain detection on Windows.
* --depth argument is no longer needed since calculated by
the script.
* Support for a webrtc.gyp_env file sitting next to the
.gclient file in the top dir of checkout, which can be
used to override Gyp variables similar to chromium.gyp_env.
* SKIP_WEBRTC_GYP_ENV can be set to skip reading webrtc.gyp_env.
BUG=2863
TEST=Ran and verified behavior on Linux with:
gclient runhooks
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dextra_gyp_flag=0
. build/android/envsetup.sh && gclient runhooks
SKIP_WEBRTC_GYP_ENV=1 webrtc/build/gyp_webrtc
GYP_GENERATORS=make webrtc/build/gyp_webrtc
The patch also passes runhooks and compile step on all trybots.
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5467 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 09:34:51 +00:00
1dd9b4d98e
Add BWE tools for parsing RTP files.
...
bwe_rtp_play feeds packets from an RTP file into the BWE and prints the estimates.
bwe_rtp_to_text parses an RTP file and outputs the result to a text file.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7689006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5466 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 09:15:48 +00:00
bda5fa77af
Fix the mouse cursor offset issue on Mac.
...
The problem is that MouseCursorMonitor returns coordinates in DIPs, while DisplayAndMouseComposer assumes that they are in physical pixels. The fix is to convert the position to physical pixels in MouseCursorMonitorMac.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7739006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-30 23:27:35 +00:00
c693704cc2
Move out typing detection to its own class.
...
This will allow an embedder to use it directly.
Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)
R=andrew@webrtc.org , niklas.enbom@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-30 09:50:46 +00:00
cf1b51b6fb
Moves the display reconfiguration callback into a separate class,
...
so that it can be shared with the cursor monitor when single monitor capturing
is added (https://webrtc-codereview.appspot.com/4679005/ ).
This Cl should have no functionality change.
BUG=2253
R=henrike@webrtc.org , sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 21:59:12 +00:00
07e5196414
Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc.
...
The callback has to go through VoEBaseImpl since VoEChannel is internal to voice engine.
TEST=compile
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5458 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 13:54:02 +00:00
094ac39b5a
Fix race when deleting video receive streams in Call.
...
BUG=
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5457 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 11:21:58 +00:00
f7c6e743b3
Fix deadlock in video_receiver.cc.
...
In webrtc::vcm::VideoReceiver::ResetDecoder(), the lock order is:
1. take _receiveCritSect,
2. take process_crit_sect_
This conflicts with the follow code path:
1. webrtc::vcm::VideoReceiver::Process(), take process_crit_sect_
call -> webrtc::vcm::VideoReceiver::NackList(),
2. with nackStats=kNackKeyFrameRequest, take _receiveCritSect
BUG=2861
TEST=trybots
R=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5456 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 10:27:51 +00:00
41907748cb
Connect webrtc::Config to WrappingBitrateEstimator
...
This is the second CL for this change. Connection to the ViE API
remains to be done.
BUG=2698
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5455 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 08:47:15 +00:00
c7c7a531f3
Add Config struct for experimental AGC.
...
Disable in the audio mixer.
TBR=aluebs,bjornv
BUG=2844
Review URL: https://webrtc-codereview.appspot.com/7769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5454 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 04:57:25 +00:00
7433a088d2
Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."
...
We reverted the r5421 to allow us roll webrtc to chrome without any modifications
to libjingle. Since webrtc is rolled with r5444, we can add back the original CL
and changes to libjingle will be upstreamed in the next roll.
TBR=andresp@webrtc.org
> Revert 5421 "Fix deadlock on register/unregister observer while ..."
>
> Failure to compile on Chromium Internal bots, because of API changes.
>
> http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
>
> You need to follow the steps mentioned in
> https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
>
> Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
> as mentioned in the doc.
>
> > Fix deadlock on register/unregister observer while there is a an going callback.
> >
> > BUG=2835
> > R=mallinath@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/7119005
>
> TBR=andresp@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7679004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5453 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 00:56:02 +00:00
84eb0e952e
Add clean test to NetEq perf test
...
Add another test to NetEqPerformanceTest with no packet losses or
clock drift. The purpose of this test would be to focus on the
"clean" code path, i.e., the path taken when there are no network
problems. The reason is that this code path is presumably much
lighter in complexity, and regressions could easily drown in the
heavier code involved when combating losses and drift.
BUG=2859
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5452 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-28 21:50:35 +00:00
932b0193e7
VideoCaptureAndroid: stop preview in opposite order of starting.
...
While the SDK documentation doesn't prescribe a required shutdown order, good
hygiene suggests stopping should happen in reverse order of starting. It also
seems to relieve a crash in the system capturer on at least the Galaxy Note 10.
BUG=2793
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5445 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 22:32:05 +00:00
18586d38bc
Revert 5421 "Fix deadlock on register/unregister observer while ..."
...
Failure to compile on Chromium Internal bots, because of API changes.
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
You need to follow the steps mentioned in
https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
as mentioned in the doc.
> Fix deadlock on register/unregister observer while there is a an going callback.
>
> BUG=2835
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7119005
TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 22:00:57 +00:00
a45cac0fb7
Avoid potential dead lock in StreamStatisticianImpl
...
Extract callbacks for rtp/rtcp data, from StreamStatisticianImpl to
ReceiveStatisticsImpl, into separate methods with guards agains having
incorrect lock order.
BUG=2856
R=andresp@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 16:22:08 +00:00
5314e85926
Race condition in RTPSender::UpdateRtpStats
...
The ssrc should not be access directly from the ssrc_ field, without
holding the send_critsect_ lock. A better way is to just use the SSRC()
getter method.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7539006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5439 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 13:20:36 +00:00
d9b9560ee5
Drop early packets when not sending in TransportAdapter.
...
Particularly, suppress periodic RTCP packets before
VideoSendStream.StartSending() or VideoReceiveStream.StartReceiving() have been called, respectively.
RTCP packets are sent periodically, by the Process thread, for every ViE channel even those not sending.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5438 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 13:03:02 +00:00
2397a17c6b
Fix bug introduced during replace of list wrapper with std equivalents in r5378.
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R=henrika@webrtc.org , pbos@webrtc.org , henrike@webrtc.org
TBR=henrike@webrtc.org
BUG=2164
Review URL: https://webrtc-codereview.appspot.com/7639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5437 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 12:33:30 +00:00
c00adbed73
Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket
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StreamStatisticianImpl.ssrc_ is protected by stream_lock_, should
be cached while holding lock to avoid race condition.
Also, rtp_callback_ do not need to be called in GetStatistics() at all
BUG=2853
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5435 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 10:42:48 +00:00
99eab02fb1
Fix "field '_testNo' is uninitialized" warnings.
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BUG=2849
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5434 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 09:30:35 +00:00
c98882dcd3
Always initialize Trace in Call TraceDispatcher.
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Prevents violation of lock order occuring previously when
RegisterCallback called SetTraceCallback while holding its lock, which
called Print back (which acquires the lock).
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5433 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27 09:11:10 +00:00
e84978f3d8
Add a Config parameter to AudioProcessing::Create().
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Also add a parameter-less version; the (int) version is deprecated and
should be removed.
TBR=aluebs,bjornv
BUG=2844
Review URL: https://webrtc-codereview.appspot.com/7609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5431 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-25 02:09:06 +00:00
57f6c10d00
Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.
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BUG=2807(second issue)
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5429 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 18:42:12 +00:00
871d949299
Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules.
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R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5426 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 13:23:49 +00:00
fd0f267bb1
Add new API (webrtc.gyp:webrtc) to merge_libs.gyp.
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Required to be able to link new API code against the merged target.
Replaces old dependency on video_engine_core as the new-API target
depends on it for now, and video_engine_core is being phased out.
R=mflodman@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/7519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5424 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 10:43:47 +00:00
99a8c7e039
Add trace-based delivery filter to BWE test framework.
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R=mflodman@webrtc.org , solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5423 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24 10:00:27 +00:00
