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577 Commits

Author SHA1 Message Date
stefan@webrtc.org
7af12be781 Thread annotations for vie_encoder.cc/.h 
Review URL: https://webrtc-codereview.appspot.com/8739005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 14:46:31 +00:00
andresp@webrtc.org
d11bec40b2 Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 14:32:58 +00:00
pbos@webrtc.org
62bafae661 Some refactoring inside rtp_rtcp/. 
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

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

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:10:51 +00:00
pbos@webrtc.org
2bb1bdab8d Preserve RTP states for restarted VideoSendStreams. 
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.

Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.

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

Review URL: https://webrtc-codereview.appspot.com/20819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 13:06:48 +00:00
stefan@webrtc.org
b9f5453e29 Add boilerplate code for H.264. 
R=mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 12:42:07 +00:00
pbos@webrtc.org
20c1f56992 Configure RTX send status on new modules. 
Fixes bug where newly-allocated modules wouldn't send payload-based
padding (or probably not send over RTX at all).

As the newly-added test exposed lock-inversions shown on tsan in
VideoReceiver, VideoReceiver was thread-annotated and locks taken less.
BUG=chromium:391085
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 10:58:12 +00:00
stefan@webrtc.org
88e0dda475 Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 09:20:42 +00:00
asapersson@webrtc.org
dfdaeb92d8 Removed old code and default implementations. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 07:35:21 +00:00
andresp@webrtc.org
1295dc6a23 Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. 
I think this does not introduces any contention or new deadlocks. But that is hard to verify at the moment.

BUG=chromium:388191
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 13:23:19 +00:00
pbos@webrtc.org
be9d2a4549 Reserve RTP/RTCP modules in SetSSRC. 
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.

Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.

To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.

BUG=3078
R=kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 13:19:09 +00:00
tnakamura@webrtc.org
a2142caa2f Bump version number to 3.55 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-25 22:04:27 +00:00
kjellander@webrtc.org
1227ab89a7 GN: Add BUILD.gn files + kjellander to OWNERS 
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.

I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.

I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.

BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default

I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc

R=brettw@chromium.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-23 19:21:07 +00:00
wuchengli@chromium.org
f425b55eeb Add tests of texture frames in video_send_stream_test. 
Also fix a bug in ViEFrameProviderBase::DeliverFrame that
a texture frame was only delivered to the first callback.

BUG=chromium:362437
TEST=Run video engine test and webrtc call on CrOS.
R=kjellander@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, wuchengli@google.com

Review URL: https://webrtc-codereview.appspot.com/15789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 12:04:05 +00:00
asapersson@webrtc.org
d980307197 Add max limit of number for overuses. When limit is reached always apply the rampup delay. 
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 14:27:19 +00:00
kjellander@webrtc.org
a1bfc50a72 Pass GYP DEPTH variable to isolate. 
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.

Also update all our .isolate files to use the <(DEPTH)
variable.

BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.

R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:02:15 +00:00
stefan@webrtc.org
cb254aac3b Enable pacing by default and remove the option to disable it from the new API. 
BUG=1672
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 15:12:25 +00:00
asapersson@webrtc.org
2881ab1e36 Increased kMaxRampUpDelayMs (120 to 240s). 
Add support for triggering on encode rsd metric if its thresholds are configured. Added unit tests.

BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 08:46:46 +00:00
stefan@webrtc.org
fbb567dacd Add APIs to enable padding with redundant payloads. 
Also makes a small change to the tests to remove flakiness. We can't do
BWE only based on rtp timestamps if we preemptively resend packets instead
of sending padding packets.

BUG=1812,2992
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-11 13:41:36 +00:00
asapersson@webrtc.org
734a532723 Add additional metric (relative standard deviation of encode time) for overuse detection. 
This code is currently only for testing.

BUG=1577
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 06:35:22 +00:00
kjellander@webrtc.org
7b82c18979 Add kjellander@webrtc.org as OWNER for *.isolate 
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.

BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:42:53 +00:00
fischman@webrtc.org
b273b60154 ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. 
Sure would be nice if the try fleet used both gcc _and_ clang...

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:59:30 +00:00
fischman@webrtc.org
9512719569 AppRTCDemo(android): support app (UI) & capture rotation. 
Now app UI rotates as the device orientation changes, and the captured stream
tries to maintain real-world-up, matching Chrome/Android and Hangouts/Android
behavior.

BUG=2432
R=glaznev@webrtc.org, henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:40:44 +00:00
stefan@webrtc.org
ef92755780 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.

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

Review URL: https://webrtc-codereview.appspot.com/15629005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:25:29 +00:00
henrike@webrtc.org
e6e139159f Android: cleanup gtest_target_type conditions. 
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library

Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).

R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 20:46:50 +00:00
wuchengli@chromium.org
637c55f45b Add support of texture frames for video capturer. 
This is a reland of r6252. The video_capture_tests failure on
builder Android Chromium-APK Tests should be flaky.

- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding, video_engine_core_unittests,
     common_video_unittests and video_capture_tests_apk.
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 07:00:51 +00:00
wuchengli@chromium.org
89e8ffb395 Revert "Add support of texture frames for video capturer." 
This reverts commit 83c89cd003be75d7d06ef9a2b139588f08d280ca.

Reason: The Buildbot has detected a new failure on builder
Android Chromium-APK Tests.

BUG=chromium:362437
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 14:12:58 +00:00
wuchengli@chromium.org
efe15355ee Add support of texture frames for video capturer. 
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
  are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding. Run video_engine_core_unittests and common_video_unittests.
R=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 12:40:27 +00:00
asapersson@webrtc.org
ab6bf4f54c Added api for getting cpu measures using a struct. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 07:43:15 +00:00
asapersson@webrtc.org
1457b4737a First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. 
Receive stats is reset if the payload type changes. Update stats after a possible reset.

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

Review URL: https://webrtc-codereview.appspot.com/12549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-26 13:06:04 +00:00
fischman@webrtc.org
440e1d1053 vie_autotest_android.cc: stop referring to undefined functions. 
The roll in r6240 exposed the fact that vie_autotest_android.cc has been
depending on vie_autotest_network.cc since forever, even though that file isn't
part of the build!  #if'ing the references out to green the build.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 21:40:45 +00:00
pbos@webrtc.org
1566ee2893 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" 
This reverts commit r6230 to re-land r6229.

ViECapturer::SwapFrame now resets timestamps.

BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 13:03:45 +00:00
pbos@webrtc.org
6e98ef4b35 Fix deadlock in RegisterPreDecodeImageCallback. 
Fixes lock-order inversion between ViEChannel::callback_cs_ and
VideoReceiver::_receiveCritSect detected on DrMemory Full which
exhibited different timing behavior.

Also removes most of the suppressions on DrMemory Full as they're able
to run again without deadlocking.

BUG=3336,3375
TEST=Run DrMemory Full trybots.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 09:41:07 +00:00
tnakamura@webrtc.org
0720758f9f Bump WebRTC version number to 3.54 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 17:51:18 +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
andresp@webrtc.org
60015d27ae Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} 
This allows use of webrtc field trials  and opens up the possibility to try the different code paths when running the unit tests by wiring them up to a --force_fieldtrials.

Tested: running a test target that links with the above with a flag --force_fieldtrials=invalid leads the test to crash.

BUG=crbug/367114
R=mflodman@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 09:39:51 +00:00
wu@webrtc.org
54231f0662 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log. 
BUG=crbug/371714
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 23:06:23 +00:00
wu@webrtc.org
88abf11cad Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. 
BUG=3111
TEST=try bots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 16:53:51 +00:00
andresp@webrtc.org
a36ad6929d Add webrtc field trials API. 
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.

Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.

Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 12:24:04 +00:00
pbos@webrtc.org
4e2806d85f Remove WEBRTC_TRACE uses in video_engine/ 
Complements fixes by mflodman@.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 08:02:22 +00:00
kjellander@webrtc.org
98c76a120d Make vie/voe_auto_test accept non-supported flags without error. 
With the switch recipes on the buildbots and the deprecation of
the custom script at
https://code.google.com/p/webrtc/source/browse/trunk/webrtc/test/buildbot_tests.py
these tests will start failing when Chromium's runtest.py is passing
--brave-new-test-launcher --test-launcher-bot-mode
to the test.
A similar change was made for most of WebRTC's tests (that depends on
the test_support_main target) in
https://webrtc-codereview.appspot.com/2222005

BUG=chromium:346198
TEST=Successfully launched the executables on Linux and Mac using:
out/Release/voe_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --test-launcher-summary-output=/tmp/tmpwhx6Zz
out/Release/vie_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --capture_test_ensure_resolution_alignment_in_capture_device=false --test-launcher-summary-output=/tmp/tmpwhx6Zz

R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 06:01:40 +00:00
asapersson@webrtc.org
e41dbee8a6 Reduced kMaxSampleDiffMs (limit to 22fps). 
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 13:45:13 +00:00
stefan@webrtc.org
46e636a3f5 Fix failing test introduced with r6111. 
Test was assuming that getting the receive estimate of a stream which hasn't received packets would return an error, new behavior is to return 0.

TBR=wu@webrtc.org
BUG=crbug/371714

Review URL: https://webrtc-codereview.appspot.com/21419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 23:17:29 +00:00
stefan@webrtc.org
72885d1c91 Fixes log spam introduced with r6041. 
We shouldn't return an error if we don't yet have a valid estimate.

BUG=crbug/371714
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 22:09:27 +00:00
wu@webrtc.org
02b286bfc9 Raise kViEMaxNumberOfChannels from 32 to 64 
Recent testing has shown that on modern desktops and laptops, decoding more than
32 low-resolution realtime video streams simultaneously is both possible and
desirable.

Reviewed:
https://webrtc-codereview.appspot.com/16449004/

TBR=mflodman
BUG=

Review URL: https://webrtc-codereview.appspot.com/17429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 22:22:41 +00:00
elham@webrtc.org
e37951d28f Updated WebRTC version to 3.53 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 17:09:31 +00:00
wu@webrtc.org
66773a032a Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. 
BUG=3111
TEST=try bots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 17:09:44 +00:00
wu@webrtc.org
ed4cb56575 Remove timestamp_extrapolator's dependency to Clock and vcm defines. 
TEST=existing tests
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 04:50:49 +00:00
asapersson@webrtc.org
9205c87820 Pointers were not dereferenced in GetRtpStatistics. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 13:24:42 +00:00