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Author SHA1 Message Date
kjellander@webrtc.org
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
stefan@webrtc.org
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
juberti@webrtc.org
668a23b402 Fix MIME type on new demo pages. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-30 23:42:01 +00:00
juberti@webrtc.org
5db9a3f32a Added new create-offer and ice-servers demos to test the exact output of createOffer and .onicecandidate. 
Updated a few demos to work on Firefox.

R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1581006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-30 23:38:44 +00:00
jiayl@webrtc.org
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
henrikg@webrtc.org
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
jiayl@webrtc.org
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
jiayl@webrtc.org
808b99b111 Disable a test assert which fails due to usrsctp not cleaned up in SctpDataEngine.cc 
BUG=2749
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7739005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 19:44:40 +00:00
jiayl@webrtc.org
a576faf82a Enable SCTP and use OPENSSL on Anroid and NSS on other platforms. 
It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests.
The private key and certificate constant strings used in some tests are updated to be compatible with NSS.
A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs.
It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose.

BUG=2253
R=fischman@webrtc.org, juberti@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 17:45:53 +00:00
xians@webrtc.org
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
solenberg@webrtc.org
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
stefan@webrtc.org
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
henrik.lundin@webrtc.org
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
andrew@webrtc.org
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
mallinath@webrtc.org
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
henrik.lundin@webrtc.org
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
kjellander@webrtc.org
45a60c7fdc Add tools/gn and tools/swarming_client to svn:ignore 
This will avoid them getting cleaned on each sync on the bots.


git-svn-id: http://webrtc.googlecode.com/svn/trunk@5450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-28 19:30:26 +00:00
minyue@webrtc.org
83dd95432e rolling Opus 1.1 
This version contains optimizations needed by WebRTC.

More information about version 1.1 can be found here http://people.xiph.org/~xiphmont/demo/opus/demo3.shtml.

Platform specific optimizations are to be added in a following CL.

TEST=passes all trybots

BUG=
R=kjellander@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-28 08:46:58 +00:00
mallinath@webrtc.org
0dac5378e5 Revert 5447 "Update talk to 60420316." 
> Update talk to 60420316.
> 
> TBR=wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7719005

TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-28 06:58:42 +00:00
mallinath@webrtc.org
752a017809 Update talk to 60420316. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7719005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-28 06:45:52 +00:00
fbarchard@google.com
69ff90e832 libyuv r976 for MJPGToI420 return code. 
BUG=2847
TESTED=libyuv MJPGToI420 unittest added which passes invalid MJPG and expects a failure.
R=andrew@webrtc.org, braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-28 03:58:46 +00:00
fischman@webrtc.org
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
mallinath@webrtc.org
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
vikasmarwaha@webrtc.org
ecc96af15b Expose errors in apprtc demo to div. Currently the errors only show in the console, the CL attempts to expose critical errors on to the div element. 
BUG=2786
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 21:13:54 +00:00
wjia@webrtc.org
776d8df25f Fix hooks in DEPS to allow read-only checkout to succeed. 
The tool download_from_google_storage requires authentication by default.
The test resources doesn't fit in this category. Using "no_auth" also
allows read-only checkout to sync successfully.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 19:55:16 +00:00
sprang@webrtc.org
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
kjellander@webrtc.org
2a260d9fab Enable Android APK trybots by default. 
As the new bots building the WebRTC native tests for Android as APKs
and executing them on a device has now proven to be reasonably stable,
it is time to enable them by default for tryjobs.

TEST=several green builds sent from a WebRTC checkout.
BUG=chromium:312827
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5440 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 16:08:43 +00:00
sprang@webrtc.org
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
sprang@webrtc.org
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
andresp@webrtc.org
2397a17c6b Fix bug introduced during replace of list wrapper with std equivalents in r5378. 
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
minyue@webrtc.org
c8b99a49d1 This is to roll a more recent Chromium, which contains latest Clang, so as to be able to roll Opus 1.1, which will the next step. 
There are uninitializion problem with normal_asyn_test.cc. This is fairly easy to solve and therefore is included in this CL.

The following is a memo on the selection of the version to roll. It may be a reference for similar missions.

How was this version picked?

1. The whole purpose of this work is to update to Clang to be able to compile Opus 1.1. In Chromium, Clang got updated to 198389 at r244540.

2. From r245412, gyp_chromium requires "tools\find_depot_tools.py". However, WebRTC does not sync up the root of folder "tools". An issue has been created to Chromium on this.

... So the version must be a good version between r244540 and r245411 (inclusive)

BUG=

TEST=passes all trybots
R=kjellander@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 12:21:42 +00:00
sprang@webrtc.org
c00adbed73 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket 
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
pbos@webrtc.org
99eab02fb1 Fix "field '_testNo' is uninitialized" warnings. 
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
pbos@webrtc.org
c98882dcd3 Always initialize Trace in Call TraceDispatcher. 
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
braveyao@webrtc.org
37c2976511 Samples, add IPv6 supporting into Apprtc demo. 
BUG=2828
TEST=Manual Test
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/7509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 03:08:16 +00:00
andrew@webrtc.org
e84978f3d8 Add a Config parameter to AudioProcessing::Create(). 
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
wu@webrtc.org
256d0ada35 Remove the check for audio codec num in WebRtcVoiceEngineTest.HasCorrectCodecs. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 18:58:51 +00:00
henrike@webrtc.org
57f6c10d00 Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera. 
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
wu@webrtc.org
98aefcd8fe Update tsan suppressions for libjingle_media_unittest. 
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/7559005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 18:32:43 +00:00
wu@webrtc.org
ca5ff9972e Re-enable webrtcvoice/videoengine unittests. 
TEST=try bots
BUG=
R=mallinath@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=5387

Review URL: https://webrtc-codereview.appspot.com/7149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 17:37:46 +00:00
asapersson@webrtc.org
871d949299 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. 
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
andresp@webrtc.org
24999d44c2 Allow ?audio=false&video=false to be used in combination to instantiate a recv-only client. 
R=braveyao@webrtc.org, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/6819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 12:25:50 +00:00
pbos@webrtc.org
fd0f267bb1 Add new API (webrtc.gyp:webrtc) to merge_libs.gyp. 
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
stefan@webrtc.org
99a8c7e039 Add trace-based delivery filter to BWE test framework. 
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
pbos@webrtc.org
c279a5d72c Wire up RTX in VideoReceiveStream. 
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.

BUG=2399
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 09:30:53 +00:00
andresp@webrtc.org
8d375c95b7 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 23:09:25 +00:00
wu@webrtc.org
a8910d2f88 Update talk to 60094938. 
Review URL: https://webrtc-codereview.appspot.com/7489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 22:12:45 +00:00
andrew@webrtc.org
754de528b7 Fix array declarations in aec_rdft.h. 
Was causing warnings in Chromium such as:
warning C4742: 'rdft_wk2i' has different alignment in
'webrtc\modules\audio_processing\aec\aec_rdft_sse2.c' and
'webrtc\modules\audio_processing\aec\aec_rdft.c': 4 and 16

BUG=chromium:336620
R=cduvivier@google.com

Review URL: https://webrtc-codereview.appspot.com/7489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 20:55:14 +00:00
pbos@webrtc.org
e7223e7795 Set NACKed packet to -1 in TestNackRetransmission. 
Zero is a valid sequence number which may occur even if there are no
retransmissions, this caused the test to flake as an incoming packet
would be mistaken for a retransmission.

BUG=2830
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 16:14:34 +00:00
sprang@webrtc.org
0e93257cee Add callbacks for receive channel RTP statistics 
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.

TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 10:00:39 +00:00