Commit Graph

1344 Commits

Author SHA1 Message Date
tkchin@webrtc.org
64eb2ff0b9 iOS library build script
Script for building iOS fat libraries with armv7/arm64/x86_64.

BUG=4119
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51429004

Cr-Commit-Position: refs/heads/master@{#8834}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8834 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-23 19:08:15 +00:00
henrik.lundin@webrtc.org
82e8ae4ee8 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest
The test has been flaky recently.

BUG=4464
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46689004

Cr-Commit-Position: refs/heads/master@{#8832}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8832 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-23 14:25:50 +00:00
kjellander@webrtc.org
e5e92bd556 Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix)
In https://webrtc-codereview.appspot.com/43899004/ I managed to get some
kind of weird whitespace character in there that completely breaks Goma
and local compilation. This fixes that.

BUG=4452
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43909004

Cr-Commit-Position: refs/heads/master@{#8821}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8821 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-22 16:28:18 +00:00
kjellander@webrtc.org
cfde27eeb3 Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows.
The test is flaky:
http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/4179

BUG=4452
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43899004

Cr-Commit-Position: refs/heads/master@{#8820}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8820 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-22 16:09:47 +00:00
tommi@webrtc.org
b789f6271a Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..."
I've kicked of a roll into Chromium with out the WebRtcVideoEngine2 change, to see if it was causing the roll problems, but re-landing in the meantime.

> Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine."
> content_browsertests started failing around the time the change landed and rolls are failing now.
> I'm going to try rolling this back, start a roll, and then re-land.
> 
> > Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
> > 
> > Removes the experiment launching WebRTC-NewVideoAPI. This field trial
> > has shown no major regressions on Chrome Canary/Dev that haven't been
> > addressed, so enabling it in time before feature freeze.
> > 
> > BUG=1788
> > R=mflodman@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/44759004
> 
> TBR=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/43889004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50459004

Cr-Commit-Position: refs/heads/master@{#8817}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8817 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-22 12:50:44 +00:00
tommi@webrtc.org
0c3400168a Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine."
content_browsertests started failing around the time the change landed and rolls are failing now.
I'm going to try rolling this back, start a roll, and then re-land.

> Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
> 
> Removes the experiment launching WebRTC-NewVideoAPI. This field trial
> has shown no major regressions on Chrome Canary/Dev that haven't been
> addressed, so enabling it in time before feature freeze.
> 
> BUG=1788
> R=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/44759004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43889004

Cr-Commit-Position: refs/heads/master@{#8816}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8816 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-22 12:45:44 +00:00
glaznev@webrtc.org
4ddc9387bd Support VP8 hardware encoding and decoding on IA devices.
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42829004

Cr-Commit-Position: refs/heads/master@{#8812}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8812 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 21:21:17 +00:00
pbos@webrtc.org
b9557a9bb7 Fix code to handle crashes for non-VP8.
Unit tests will be submitted Monday, submitting this part to get the
Android bots green.

BUG=1667, 1788
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44789004

Cr-Commit-Position: refs/heads/master@{#8811}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8811 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 19:53:15 +00:00
pbos@webrtc.org
66df3cf7ab Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
Removes the experiment launching WebRTC-NewVideoAPI. This field trial
has shown no major regressions on Chrome Canary/Dev that haven't been
addressed, so enabling it in time before feature freeze.

BUG=1788
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44759004

Cr-Commit-Position: refs/heads/master@{#8809}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8809 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 15:45:17 +00:00
pbos@webrtc.org
8296ec518b Fix heap-use-after-free in WebRtcVideoEngine2.
Found in libjingle_peerconnection_unittest on asan while trying to
default-enable WebRtcVideoEngine2.

BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44779004

Cr-Commit-Position: refs/heads/master@{#8808}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8808 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 14:28:31 +00:00
perkj@webrtc.org
9f9ea7e5ab Clean up webrtc external capture.
This cl removes the dependency to the external capture module if external capturing is used in webrtc.
It also removes two external capture methods that is not needed.
Further more it adds I420VideoFrame::Create that takes a pointer to packed memory as input.

R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43879004

Cr-Commit-Position: refs/heads/master@{#8804}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8804 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 10:55:39 +00:00
tina.legrand@webrtc.org
0c26299739 Disabling two flaky tests in libjingle_media_unittest.
BUG=4452,4453
R=kjellander@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44739004

Cr-Commit-Position: refs/heads/master@{#8791}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8791 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-19 13:28:20 +00:00
tkchin@webrtc.org
8cc47e926c Objective-C readability review.
BUG=
R=rsesek@chromium.org

Review URL: https://webrtc-codereview.appspot.com/34679004

Cr-Commit-Position: refs/heads/master@{#8784}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8784 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 23:38:45 +00:00
guoweis@webrtc.org
840da7b755 Implement Rotation in Android Renderer.
Make use of rotation information from the frame and rotate it accordingly when we render the frame.

BUG=4145
R=glaznev@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8770

Review URL: https://webrtc-codereview.appspot.com/50369004

Cr-Commit-Position: refs/heads/master@{#8781}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8781 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 16:58:49 +00:00
pbos@webrtc.org
143451d259 Base start bitrate on last observed bitrate.
Instead of setting bitrates based on codec target settings (which may
have previously been capped by a codec max bitrate), fetch the last
bandwidth allocated for this channel. This fixes broken low start bitrates
due to QCIF being set as default codec in WebRtcVideoEngine2 which caps
the max bitrate to 200kbps.

BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43789004

Cr-Commit-Position: refs/heads/master@{#8780}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8780 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 14:40:52 +00:00
perkj@webrtc.org
af612d5e07 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.

With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame

This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.

Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306

Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/

BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47629004

Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 09:51:44 +00:00
magjed@webrtc.org
14ee8cc9c7 WebRtcVideoFrame: Support odd resolutions
We currently truncate the resolution of frames to a multiple of 4. This is unnecessary as everything supports odd resolutions now.

R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43819004

Cr-Commit-Position: refs/heads/master@{#8774}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8774 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 09:22:19 +00:00
guoweis@webrtc.org
3fffd66dfa Revert "Implement Rotation in Android Renderer."
This reverts commit 835ec63d8a.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/51399004

Cr-Commit-Position: refs/heads/master@{#8771}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8771 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 04:20:47 +00:00
guoweis@webrtc.org
835ec63d8a Implement Rotation in Android Renderer.
Make use of rotation information from the frame and rotate it accordingly when we render the frame.

BUG=4145
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50369004

Cr-Commit-Position: refs/heads/master@{#8770}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8770 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 02:44:39 +00:00
pthatcher@webrtc.org
52cd828e17 Allow webrtc external encoder factories to declare encoders have internal camera sources.
This flag is passed to existing VieExternalCodec API (and others) to denote encoders that don't require/expect frames from the normal capture pipeline. This is the simplest way to allow camera->encoder texture support, until textures are supported through the normal camera pipeline and the lifetime issues are all figured out (I hear this is on the backlog, but not there yet).

Ideally, the flag would be on the encoder, but that doesn't work with SimulcastEncoderAdapter, since it doesn't create an encoder right away.

Note that this change only affects WebRtcVideoEngine (not WRVE2), since WRVE2 uses video_send_stream, and my hope is that by the time things have switched to WRVE2, textures will be supported with the normal camera pipeline and the dependency on internal sources can be thrown away.

BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42349004

Cr-Commit-Position: refs/heads/master@{#8769}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8769 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 02:25:18 +00:00
glaznev@webrtc.org
2161234cf6 Add new features to AppRTCDemo from private repo.
- Add HUD fragment with HUD related controls and more
HUD statistics.
- Create and set all peer connection constraints in
PeerConnectionClient class.
- Handle registration request in web socket class internally
once web socket connection is opened.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44669004

Cr-Commit-Position: refs/heads/master@{#8762}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8762 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 18:24:19 +00:00
perkj@webrtc.org
a78a94e838 Fix RateTracker to set an initial reference time when first updated.
BUG=4442
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43829004

Cr-Commit-Position: refs/heads/master@{#8751}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8751 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 12:45:41 +00:00
pbos@webrtc.org
ae222b5be6 Remove dead code in WebRtcVideoEngine2 unittests.
BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43609004

Cr-Commit-Position: refs/heads/master@{#8747}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8747 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 10:48:28 +00:00
magjed@webrtc.org
858024f1d9 WebRtcVideoFrame: Initialize members in empty constructor
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41319004

Cr-Commit-Position: refs/heads/master@{#8746}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8746 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 08:47:17 +00:00
pthatcher@webrtc.org
592470b4ff Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47599004

Cr-Commit-Position: refs/heads/master@{#8743}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8743 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 21:16:23 +00:00
pthatcher@webrtc.org
6ad507ac35 Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE.
Also, remove channel_name.  It's no longer needed.

This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43719004

Cr-Commit-Position: refs/heads/master@{#8741}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8741 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 20:19:42 +00:00
pthatcher@webrtc.org
4eeef584a7 Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47589004

Cr-Commit-Position: refs/heads/master@{#8740}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8740 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 19:34:40 +00:00
pthatcher@webrtc.org
c04a97f054 Move from BaseSession::GetStats to WebRtcSession::GetTransportStats
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

Review URL: https://webrtc-codereview.appspot.com/45639004

Cr-Commit-Position: refs/heads/master@{#8739}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8739 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 19:32:23 +00:00
bjornv@webrtc.org
3f11823a1a Disables SW AEC when built-in AEC is enabled
As of r7849 the built-in AEC on devicing supporting it is enabled by default.
Unfortunately, the SW AEC (AECM) was not disabled, hence running on top of the built-in one. This is not necessary. In fact it reduce double talk performance significantly.

BUG=4431
TESTED=manually
R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49419004

Cr-Commit-Position: refs/heads/master@{#8735}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8735 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 14:22:17 +00:00
magjed@webrtc.org
2056ee3e3c Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
This reverts commit r8731.

Reason for revert: Breakes Chromium FYI bots.

TBR=hbos, tommi

Review URL: https://webrtc-codereview.appspot.com/40359004

Cr-Commit-Position: refs/heads/master@{#8733}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 13:48:18 +00:00
hbos@webrtc.org
93d9d6503e I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments.
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45629004

Cr-Commit-Position: refs/heads/master@{#8732}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8732 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 13:26:41 +00:00
hbos@webrtc.org
2dc5fa69b2 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*.
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40299004

Cr-Commit-Position: refs/heads/master@{#8731}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 13:02:19 +00:00
tommi@webrtc.org
4b89aa03bb Change StatsCollector to use DCHECK instead of ASSERT.
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46579004

Cr-Commit-Position: refs/heads/master@{#8729}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8729 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 09:52:41 +00:00
kjellander@webrtc.org
eed2fcaa76 Roll chromium_revision 00e438c..8d51d96 (320241:320682)
Relevant changes:
* src/third_party/android_tools: fd5a8ec..98a4345
Details: 00e438c..8d51d96/DEPS

This required updating our Android projects to API level 22,
as third_party/android_tools dropped support for API level 21.

Command used:
perl -pi -e "s/android-21/android-22/g" `find . -name project.properties`
Using 'android update project' would also work but that changes the
ANDROID_SDK_ROOT -> ANDROID_HOME, which the Chromium build toolchain
doesn't set properly when build/android/envsetup.sh is sourced.

Clang version was not updated in this roll.

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

Review URL: https://webrtc-codereview.appspot.com/42779004

Cr-Commit-Position: refs/heads/master@{#8728}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8728 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 09:00:41 +00:00
changbin.shao@webrtc.org
2d25b44f47 Check associated payload type when negotiate RTX codecs.
At the moment, only payload name is checked when match two RTX codecs.
This will cause wrong behavior of codec negotiation if multiple RTX codecs
are added.

BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34189004

Cr-Commit-Position: refs/heads/master@{#8727}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8727 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 04:15:23 +00:00
tommi@webrtc.org
c29f7f3a5f Disable assert for nr of threads in PeerConnectionTest.java.
This test is flaky so we need to figure out a better way to do it.
I've documented what we've observed and added a todo for myself to figure out a solution.

R=kjellander@webrtc.org
BUG=4424

Review URL: https://webrtc-codereview.appspot.com/46599004

Cr-Commit-Position: refs/heads/master@{#8725}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8725 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-14 18:15:47 +00:00
glaznev@webrtc.org
f1f558cde8 Fix AppRTCDemo and AppRTCDemoTest builds.
On fresh checkout AppRTCDemo and corresponding tests
fail to build because resource file R.java is not auto generated properly.
On existing tree R.java will be picked up from previous
build leftover at talk/examples/android/gen.
Build bots did not detect this break for some reason.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43749004

Cr-Commit-Position: refs/heads/master@{#8723}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8723 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-14 02:48:47 +00:00
jiayl@webrtc.org
d83f4eff84 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Committed: https://code.google.com/p/webrtc/source/detail?r=8701

Committed: https://code.google.com/p/webrtc/source/detail?r=8706

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8722}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8722 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 21:26:44 +00:00
pthatcher@webrtc.org
b01c707209 Use a NULL session in unit tests that don't actually use the session.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49379004

Cr-Commit-Position: refs/heads/master@{#8721}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8721 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 20:05:46 +00:00
pthatcher@webrtc.org
b4aac13810 Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49399004

Cr-Commit-Position: refs/heads/master@{#8720}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8720 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 18:25:54 +00:00
pthatcher@webrtc.org
990a00c30a Remove unused transport code.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49389004

Cr-Commit-Position: refs/heads/master@{#8719}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8719 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 18:20:48 +00:00
minyue@webrtc.org
9b2e1144df Supporting Opus DTX in Voice Engine.
Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API.

BUG=1014
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43709004

Cr-Commit-Position: refs/heads/master@{#8716}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 09:38:55 +00:00
kjellander@webrtc.org
503a9e822a Make AppRTCDemoTest pass without Internet connection.
The AppRTCDemoTest is failing if the Android device lacks
an Internet connection (e.g. is in flight mode).
This change makes it benefit from the work done in
https://review.webrtc.org/36769004/ to work around that
limitation for loopback tests.

R=phoglund@webrtc.org
TBR=glaznev@webrtc.org
BUG=4421
TESTED=Successful run on Nexus 7 (2013) in flight mode using:
ninja -C out/Release
. build/android/envsetup.sh
adb install -r out/Release/apks/AppRTCDemo.apk
webrtc/build/android/test_runner.py instrumentation --test-apk AppRTCDemoTest --verbose --release

Review URL: https://webrtc-codereview.appspot.com/45649004

Cr-Commit-Position: refs/heads/master@{#8714}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8714 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 09:06:58 +00:00
jiayl@webrtc.org
8372888b07 Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns."
This reverts commit 45bc01a7172402aa4bb8d457474300533c273413.
TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/47559004

Cr-Commit-Position: refs/heads/master@{#8711}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8711 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 00:32:43 +00:00
glaznev@webrtc.org
3d3c005f36 Fix Android peer connection client instrumentation tests.
- Updated Java VideoRenderer removes setSize() from video renderer interface.
Remove no longer valid test, which requires setSize() call before any
frame can be rendered.
- test_runner.py tries to run private member of InstrumentationTestCase class.
Workaround it by renaming private loopback test method.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47549004

Cr-Commit-Position: refs/heads/master@{#8707}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8707 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 23:07:17 +00:00
jiayl@webrtc.org
fde1de93f9 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Committed: https://code.google.com/p/webrtc/source/detail?r=8701

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8706}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8706 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 23:02:13 +00:00
guoweis@webrtc.org
00c509ad1c Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

Tested with peerconnection_client on windows, AppRTCDemo on Mac.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Committed: https://code.google.com/p/webrtc/source/detail?r=8661

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8705}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8705 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 21:38:19 +00:00
jiayl@webrtc.org
04cd69887d Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns."
This reverts commit 93604daf0e.

TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/40329004

Cr-Commit-Position: refs/heads/master@{#8704}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8704 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 21:36:42 +00:00
guoweis@webrtc.org
fdd1057949 Add CVO support to Vie layer.
1. standard plumbing CVO through vie layer.
2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation.

WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420.

BUG=4145
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429007

Cr-Commit-Position: refs/heads/master@{#8703}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 20:51:50 +00:00
guoweis@webrtc.org
4f85288e71 Socket options are only applied when first setting TransportChannelImpl.
Also fixed the issue when we have an TransportChannelImpl, the socket
option is not preserved.

Since this is a code path that will be modified by bundle (which Peter also has a test case already), we don't need a test case here.

BUG=4374
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42699004

Cr-Commit-Position: refs/heads/master@{#8702}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8702 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 20:10:22 +00:00
jiayl@webrtc.org
93604daf0e Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8701}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8701 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 19:05:22 +00:00
tommi@webrtc.org
d3900296ae Use a variant for storing stats values in StatsCollector code.
This cuts down on the amount of string copying we currently do and paves the way for separating the code that fetches the stats from the code that populates the stats reports.  As is, that code is intertwined, so we populate the stats on both signaling and worker thread.

I'm also adding some documentation and TODOs for further improvements.

BUG=2822
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47459004

Cr-Commit-Position: refs/heads/master@{#8700}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8700 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 16:36:15 +00:00
tommi@webrtc.org
75b7f17c29 Temporary interface change to StatsReport::Id.
This change is just to allow rolling into Chromium, update Chromium and then commit the actual change in WebRTC that requires the interface change. It allows using a StatsReport::Id object as a pointer (foo->Bar()), since in an upcoming change, Id objects will be pointers.

R=magjed@webrtc.org
BUG=2822

Review URL: https://webrtc-codereview.appspot.com/43689004

Cr-Commit-Position: refs/heads/master@{#8697}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8697 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 13:31:24 +00:00
magjed@webrtc.org
afdd5dd372 Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame""
This reverts r8683 and is a reland of r8682.

Reason for revert: The thread checker in Chromium that crashed has been fixed now.

BUG=1128
TBR=tommi,pbos,pthatcher

Review URL: https://webrtc-codereview.appspot.com/40319004

Cr-Commit-Position: refs/heads/master@{#8696}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 13:11:43 +00:00
mflodman@webrtc.org
e9413c686e Revert 8689 "Fix an issue in DtlsIdentityStore when the store is..."
> Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
> 
> BUG=crbug/464995
> R=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/42659004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42729004

Cr-Commit-Position: refs/heads/master@{#8690}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8690 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 08:42:18 +00:00
jiayl@webrtc.org
2a3942adc6 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
BUG=crbug/464995
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8689}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8689 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 23:33:42 +00:00
decurtis@webrtc.org
8c5ea8a811 Fix temporal layer log string.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43639004

Patch from Noah Richards <noahric@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8687}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8687 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 19:59:43 +00:00
glaznev@webrtc.org
ae1a078ac4 Convert AppRTCDemo and AppRTCDemoTest to proper GYP target.
Initial CL for converting AppRTCDemo and AppRTCDemoTest to
the Chromium style of APK targets. This would
make it possible to get rid of all the ugly
bash stuff we currently have.

CL will bump minimum SDK to v14, but this is the requirement to use Chrome tools.

Initial work was done by kjellander@
https://webrtc-codereview.appspot.com/44549005/

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43679004

Cr-Commit-Position: refs/heads/master@{#8686}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8686 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 17:52:44 +00:00
magjed@webrtc.org
b218ff5531 Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"
This reverts r8682.

Reason for revert: Fails on Chromium FYI content_browsertests

BUG=1128
TBR=tommi,pbos,pthatcher

Review URL: https://webrtc-codereview.appspot.com/47529004

Cr-Commit-Position: refs/heads/master@{#8683}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8683 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 15:29:59 +00:00
magjed@webrtc.org
370a72cc3f Remove frame copy from cricket::VideoFrame to I420VideoFrame
BUG=1128
R=pbos@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42249004

Cr-Commit-Position: refs/heads/master@{#8682}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8682 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 14:16:16 +00:00
pbos@webrtc.org
e77c9c8df5 Build WebRtcMediaEngine2 outside of Chromium.
Removes #ifdef WEBRTC_CHROMIUM_BUILD from
talk/media/webrtc/webrtcmediaengine.cc. WebRtcVideoEngine2 is built on
all platforms so there's no longer any need to guard this code under
ifdefs.

BUG=1788
R=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42719004

Cr-Commit-Position: refs/heads/master@{#8679}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8679 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 10:50:16 +00:00
braveyao@webrtc.org
9bfa5f0405 In r8605, DTLS is enabled by default for native webrtc. So we have to disable it explicitly in peerconnection example for loopback test.
BUG=4386
TEST=Manual Test
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44599004

Cr-Commit-Position: refs/heads/master@{#8677}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8677 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 03:21:41 +00:00
glaznev@webrtc.org
fc516077ed Fix Android AppRTCDemo failure on devices with one or no camera.
- Disable video call on devices with no camera.
- Open default camera and disable camera switch on
devices with one camera.

BUG=4373
R=braveyao@webrtc.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46539004

Cr-Commit-Position: refs/heads/master@{#8674}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8674 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 18:21:19 +00:00
magjed@webrtc.org
4052d88162 Remove GetLastRenderedFrame
This function is not used.

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

Review URL: https://webrtc-codereview.appspot.com/40269004

Cr-Commit-Position: refs/heads/master@{#8673}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8673 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 16:36:42 +00:00
magjed@webrtc.org
d7452a0168 Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
This reverts commit r8633.

Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.

BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj

Review URL: https://webrtc-codereview.appspot.com/46549004

Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 15:13:13 +00:00
hbos@webrtc.org
aa57702c08 Removed texture_video_frame.h and webrtctexturevideoframe.h
BUG=1128
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45579004

Cr-Commit-Position: refs/heads/master@{#8667}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8667 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 09:04:18 +00:00
guoweis@webrtc.org
f9a75d99b9 Revert "Add concept of whether video renderer supports rotation."
This reverts commit 0ad48935fc.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/41199004

Cr-Commit-Position: refs/heads/master@{#8663}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8663 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 06:37:41 +00:00
guoweis@webrtc.org
60a2aa0652 Revert "Add concept of whether video renderer supports rotation."
This reverts commit 31d16467ac.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/47489004

Cr-Commit-Position: refs/heads/master@{#8662}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8662 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 06:20:18 +00:00
guoweis@webrtc.org
31d16467ac Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8661}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8661 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 06:11:44 +00:00
guoweis@webrtc.org
0ad48935fc Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8660}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8660 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 02:43:40 +00:00
kwiberg@webrtc.org
67186fe00c Fix clang style warnings in webrtc/base
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.

Not inlining virtual functions with simple bodies such as

  { return false; }

strikes me as probably losing more in readability than we gain in
binary size and compilation time, but I guess it's just like any other
case where enabling a generally good warning forces us to write
slightly worse code in a couple of places.

BUG=163
R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47429004

Cr-Commit-Position: refs/heads/master@{#8656}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-09 22:24:25 +00:00
glaznev@webrtc.org
2989204130 Fix instability in peer connection client unit test.
- Add a separate thread to process peer connection ICE messages
to void setting remote ICe candidate in local ICE candidate callback.
- Set proper constraints values.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42279004

Cr-Commit-Position: refs/heads/master@{#8655}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8655 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-09 19:15:32 +00:00
henrika@webrtc.org
474d1eb223 Adds C++/JNI/Java unit test for audio device module on Android.
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.

It also:

- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.

BUG=NONE
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40069004

Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-09 12:40:43 +00:00
glaznev@webrtc.org
dc08a230da Fix H.264 start code position search.
This will address incorrect start code search
in a sequence like 00 00 00 00 00 01.
Thanks Noah.

R=noahric@chromium.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41159004

Cr-Commit-Position: refs/heads/master@{#8639}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8639 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 23:32:42 +00:00
magjed@webrtc.org
1af1391b41 Remove WebRtcTextureVideoFrame
WebRtcTextureVideoFrame is currently an empty shell that only provides a convenience constructor of I420VideoFrame with a texture buffer. This CL moves that constructor, and all unittests, of WebRtcTextureVideoFrame into the base class. Then it's possible to completely remove WebRtcTextureVideoFrame and all its files.

BUG=1128
R=pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48399004

Cr-Commit-Position: refs/heads/master@{#8638}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8638 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 17:17:43 +00:00
magjed@webrtc.org
c2008a0e8c RTCOpenGLVideoRenderer: Add support for padded frames
This CL allows RTCOpenGLVideoRenderer to handle frames with pitch > width by making an intermediate frame copy.

BUG=4381,1128
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46509004

Cr-Commit-Position: refs/heads/master@{#8637}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8637 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 16:59:23 +00:00
jiayl@webrtc.org
b4cd093f41 Change the unintentioal CHECK to DCHECK in DtlsIdentityStore.
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41139004

Cr-Commit-Position: refs/heads/master@{#8636}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8636 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 16:32:57 +00:00
pbos@webrtc.org
a2a6fe66a3 Reconfigure default streams on AddRecvStream.
Makes sure RTX can be used for streams that have received early media
before being properly configured.

BUG=1788
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46499004

Cr-Commit-Position: refs/heads/master@{#8634}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8634 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 15:35:48 +00:00
perkj@webrtc.org
bcead305a2 Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
This removes the none const pointer entry and SwapFrame.

Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.

With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame

BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429004

Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 12:38:22 +00:00
magjed@webrtc.org
45cdcce5f5 Remove TextureVideoFrame
TextureVideoFrame is currently an empty shell that only provides a convenience constructor of I420VideoFrame with a texture buffer. This CL moves that constructor, and all unittests, of TextureVideoFrame into the base class. Then it's possible to completely remove TextureVideoFrame and all its files. Also, there is no point in having I420VideoFrame virtual anymore.

R=pbos@webrtc.org, perkj@webrtc.org, stefan@webrtc.org
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/40229004

Cr-Commit-Position: refs/heads/master@{#8629}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8629 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 10:41:47 +00:00
kjellander@webrtc.org
e41ec818a7 Remove libjingle_root GYP variable
It is no longer needed.

R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44449004

Cr-Commit-Position: refs/heads/master@{#8627}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8627 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 08:03:47 +00:00
pthatcher@webrtc.org
818c4984e4 Modify the simulcast encoder factory adapter to allow external encoder factories that support more than one codec.
Only VP8 encoders will be wrapped in the simulcast adapter; other codec types will be created directly with the real encoder factory and cleaned up appropriately.

BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40169004

Cr-Commit-Position: refs/heads/master@{#8623}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8623 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 02:21:23 +00:00
magjed@webrtc.org
2386d6dd92 Revert 8599 "Revert 8580 "Unify underlying frame buffer in I420VideoFrame and...""
It's possible to build Chrome on Windows with this patch now.

BUG=1128

> This is unfortunately causing build problems in Chrome on Windows.

>> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
>>
>> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
>>
>> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
>>
>> Some additional minor changes are:
>> * Disallow creation of 0x0 texture frames.
>> * Remove the half-implemented ref count functions in I420VideoFrame.
>> * Remove the Alias functionality in WebRtcVideoFrame
>>
>> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
>> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
>> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
>>
>> BUG=1128
>> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
>>
>> Review URL: https://webrtc-codereview.appspot.com/42469004

R=pbos@webrtc.org
TBR=mflodman, pbos, perkj, tommi

Review URL: https://webrtc-codereview.appspot.com/45489004

Cr-Commit-Position: refs/heads/master@{#8616}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8616 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 14:03:51 +00:00
tommi@webrtc.org
5af41aabae Fix uninitialized variable. If FindConstraint() returns false, we check |value| in two places and at that point, it can hold an uninitialized value. Caught by Linux Memcheck builder.
http://chromegw.corp.google.com/i/client.webrtc/builders/Linux%20Memcheck/builds/3351/steps/libjingle_peerconnection_unittest/logs/0A34BA777AB03D08

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/43579004

Cr-Commit-Position: refs/heads/master@{#8611}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8611 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 08:42:13 +00:00
guoweis@webrtc.org
bbce5efaa6 Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8582

Committed: https://code.google.com/p/webrtc/source/detail?r=8607

Review URL: https://webrtc-codereview.appspot.com/43529004

Cr-Commit-Position: refs/heads/master@{#8609}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8609 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 04:39:07 +00:00
guoweis@webrtc.org
d43b2c098d Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
This reverts commit 86c33e3a94.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/47409004

Cr-Commit-Position: refs/heads/master@{#8608}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8608 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 04:03:44 +00:00
guoweis@webrtc.org
86c33e3a94 Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8582

Review URL: https://webrtc-codereview.appspot.com/43529004

Cr-Commit-Position: refs/heads/master@{#8607}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8607 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 03:40:48 +00:00
jiayl@webrtc.org
61e00b0bca Create a in-memory DTLS identity store that keeps a free identity generated in the background.
BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Committed: https://code.google.com/p/webrtc/source/detail?r=8581

Review URL: https://webrtc-codereview.appspot.com/37889004

Cr-Commit-Position: refs/heads/master@{#8605}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8605 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 22:18:18 +00:00
tommi@webrtc.org
f7abb12aa9 Fix OVERRIDE->override again after reverting video frame cl.
TBR=magjed@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/40199004

Cr-Commit-Position: refs/heads/master@{#8600}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8600 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 17:43:49 +00:00
tommi@webrtc.org
1f94407319 Revert 8580 "Unify underlying frame buffer in I420VideoFrame and..."
This is unfortunately causing build problems in Chrome on Windows.

> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
> 
> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
> 
> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
> 
> Some additional minor changes are:
> * Disallow creation of 0x0 texture frames.
> * Remove the half-implemented ref count functions in I420VideoFrame.
> * Remove the Alias functionality in WebRtcVideoFrame
> 
> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
> 
> BUG=1128
> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/42469004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42199005

Cr-Commit-Position: refs/heads/master@{#8599}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8599 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 17:35:00 +00:00
tommi@webrtc.org
92f4018d80 Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do.
Rename "AddValue" methods to AddXxx where Xxx is the type being added. Moving forward, we'll support those types natively without conversion to string.

Normalizing the extraction code to have fewer places that add the same stats and data driven additions to reports instead of multiple call sites.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47369004

Cr-Commit-Position: refs/heads/master@{#8597}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8597 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 15:25:44 +00:00
kjellander@webrtc.org
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
pbos@webrtc.org
058b1f17ac Remove GetReceiveBandwidthEstimatorStats.
Removes unnecessary non-standard stats that we don't really make use of.

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

Review URL: https://webrtc-codereview.appspot.com/47379004

Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 08:55:16 +00:00
guoweis@webrtc.org
fc2f146af2 Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
This reverts commit bbbdeed2bf.

TBR=juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41109004

Cr-Commit-Position: refs/heads/master@{#8585}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8585 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 04:50:41 +00:00
pthatcher@webrtc.org
7bea1ffe77 Expose negotiated ciphers through stats API.
Use the new internal API to expose the negotiated SRTP/SSL ciphers
through the stats API.
This is a follow-up to https://webrtc-codereview.appspot.com/37209004.

BUG=3976
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35169004

Cr-Commit-Position: refs/heads/master@{#8584}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8584 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 01:38:49 +00:00
jiayl@webrtc.org
be77872d2c Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
Breaking Chromium FYI.

TBR=pthatcher@webrtc.org

This reverts commit 369f68255f.

BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Review URL: https://webrtc-codereview.appspot.com/37889004


Review URL: https://webrtc-codereview.appspot.com/47389004

Cr-Commit-Position: refs/heads/master@{#8583}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8583 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 00:19:16 +00:00
guoweis@webrtc.org
bbbdeed2bf Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43529004

Cr-Commit-Position: refs/heads/master@{#8582}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8582 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 23:27:41 +00:00
jiayl@webrtc.org
369f68255f Create a in-memory DTLS identity store that keeps a free identity generated in the background.
BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Review URL: https://webrtc-codereview.appspot.com/37889004

Cr-Commit-Position: refs/heads/master@{#8581}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8581 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 23:14:18 +00:00
magjed@webrtc.org
c8895aa2f3 Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.

This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.

Some additional minor changes are:
* Disallow creation of 0x0 texture frames.
* Remove the half-implemented ref count functions in I420VideoFrame.
* Remove the Alias functionality in WebRtcVideoFrame

The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
* Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
* Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.

BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42469004

Cr-Commit-Position: refs/heads/master@{#8580}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8580 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 21:22:26 +00:00
jiayl@webrtc.org
8ad96605c1 Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
Test failure: http://chromegw/i/client.webrtc/builders/Linux32%20Release/builds/3557

This reverts commit df512cc8b7.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41089004

Cr-Commit-Position: refs/heads/master@{#8579}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8579 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 20:35:34 +00:00