Commit Graph

186 Commits

Author SHA1 Message Date
e7e0602a0d ObjC: Notify local video track
The macOS demo add camera preview in didReceiveLocalVideoTrack callback, but this callback is never called.

Bug: webrtc:9276
Change-Id: I60b9cc69672f3654d4e36de0e8140e0bbb957540
Reviewed-on: https://webrtc-review.googlesource.com/77950
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23458}
2018-05-30 22:36:14 +00:00
f8d8d6d00c Use range-based-for instead of std::for_each and std::mem_fun
std::mem_fun is deprecated in C++11, and removed in C++17. Using C++17
option for building libwebrtc causes build failure. This is found during
upgrading WebKit tree from C++14 to C++17.
This patch replaces std::for_each and std::mem_fun with range-based-for.
We also merge loops for streams_ into one.

Bug: webrtc:9277
Change-Id: I44a7e44ea21fc33ffa9a586ddfea570f97dfacb6
Reviewed-on: https://webrtc-review.googlesource.com/77280
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23285}
2018-05-17 13:51:02 +00:00
ebd9abc1a2 Use IFA_LOCAL instead of IFA_ADDRESS over IPv4 network on ANDROID
IFA_ADDRESS gives DESTINATION address in case of point-to-point
connection, which is not able to create ports for candidate gathering.
Use IFA_LOCAL to avoid this problem.

Bug: webrtc:9189
Change-Id: Ifcb1955b1b4011dc69c93d99b4e223b370dc16eb
Reviewed-on: https://webrtc-review.googlesource.com/69620
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23055}
2018-04-27 14:16:01 +00:00
a72b7fc30a ObjC: Add missing _lastDrawnFrame assignments
Currently there are several checks against _lastDrawnFrame in RTCEAGLVideoView.mm but this variable is not assigned anywhere. Seems like it was missed in 13941912b1 during work on injecting custom shaders.

Bug: webrtc:9133
Change-Id: Ie979a63de343e7253e4b4e70e3b98ffb0880af04
Reviewed-on: https://webrtc-review.googlesource.com/68720
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22819}
2018-04-11 12:51:06 +00:00
3cfe9e167e Fixed video capturing on Mac.
On specific Macbooks (no exact pattern, unfortunately),
video from an integrated camera is not captured.
Changed AVCaptureVideoDataOutput pixel format configuration
as in Chromium which solved the problem.
https://chromium.googlesource.com/chromium/src/media/+/master/capture/video/mac/video_capture_device_avfoundation_mac.mm
FourCharCode best_fourcc = kCVPixelFormatType_422YpCbCr8;

Tested with external cameras as well.

Bug: webrtc:8958
Change-Id: Ib99382b38d1914e2963761a33df310024524c9a4
Reviewed-on: https://webrtc-review.googlesource.com/58880
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22709}
2018-04-03 16:23:01 +00:00
2870b0a57e Expose a link-local network interfaces enumeration option
The bug 8432 is caused by trying to connect through a
"link-local" interface (IP address 169.254.0.x/16),
which is listed among the iPhone network interfaces.
The bug is not happening if the link-local network interfaces
are skipped in the ICE candidate gethering process.

To control this behaviour an option - disable_link_local_networks -
is added inside the RTCConfiguration.
It is used to set the new BasicPortAllocatorSession flag -
PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS.
The port allocator flag is added if the configuration option is set.

IPIsLinkLocal IPAddress function and its friends (IPIsLoopback, IPIsPrivate)
are refactored to work on both IPv4 and IPv6.
Unit test IPIsLinkLocal.

Bonus: fix a bug in IPIsLinkLocalV6:
take into account just 10 network mask bits instead of 16.

Bug: webrtc:8432
Change-Id: Ibe8f677a36098057b7fcad5c798380727b23359b
Reviewed-on: https://webrtc-review.googlesource.com/36380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21922}
2018-02-06 19:12:04 +00:00
5e4833cc90 Add missing stdio.h header in files using scanf/sscanf function.
Various files in webrtc codebase use scanf/sscanf function without
including stdio.h header file which is supposed to define it. This
somehow works when using glibc, but fails with uClibc.

Bug: webrtc:8641
Change-Id: Ie4ae17af32b32ed8cea567166b6b0e5193966995
Reviewed-on: https://webrtc-review.googlesource.com/32261
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21775}
2018-01-26 13:15:52 +00:00
b8874356f6 RemoteBitrateEstimatorAbsSendTime: check clock is a valid ref
Bug: webrtc:8607
Change-Id: Idc3b6c0b3896381f0140584d8c2952ee26db1646
Reviewed-on: https://webrtc-review.googlesource.com/31320
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21623}
2018-01-16 01:27:11 +00:00
d3c642bc1f Fix typo in the include path of ooura_fft.h
Bug: None
Change-Id: Iaac4a80f75dcd81ab0d2665cb20f27f0342cb17d
Reviewed-on: https://webrtc-review.googlesource.com/38441
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21565}
2018-01-11 07:57:40 +00:00
6728003bcf Skip H246 scaling lists in SPS packets
This code is originally written by marc@frankensteinmotorworks.com

Bug: webrtc:8275
Change-Id: I35e6d21b12e71199e0209ff91740d95c9df3bd10
Reviewed-on: https://webrtc-review.googlesource.com/36520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21528}
2018-01-09 10:22:30 +00:00
5a7508ab24 Fixed NPE inside org.webrtc.Camera1Session.create
On some devices `android.hardware.Camera.open` returns null
instead of raising exception. It causes `NPE` inside
`Camera1Session.create` when method `setPreviewTexture` is
invoked on local variable `camera`, which is `null`.

Bug: webrtc:8658
Change-Id: Ic65b4aef2c0b8b65735a9db02433b536bfe92ddd
Reviewed-on: https://webrtc-review.googlesource.com/33620
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21352}
2017-12-19 10:01:20 +00:00
1c62ffa530 Normalize main(..) routines for WinUWP
In order to support WinUWP platform, all main(..) routines must be normalized to the formal int main(int argc, char* argv[]) form. A platform wrapper main is auto-created linking against the default main(...). This can only work if the linkage is exactly matching the proper formal definition and not a loosely defined main(...) alternative.

Bug: webrtc:8608
Change-Id: I606663aaea7df1792c7c5636279617b8926fa5cc
Reviewed-on: https://webrtc-review.googlesource.com/28721
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21229}
2017-12-12 14:32:56 +00:00
12e555b715 Delete wrapper API ConvertToI420 for YUV conversion to I420
Directly use the libyuv API for YUV conversion to I420

Bug: None
Change-Id: Iea6e8fa8f7179c800ea850305170002398cb00dc
Reviewed-on: https://webrtc-review.googlesource.com/17260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20681}
2017-11-15 11:10:20 +00:00
149533abd4 Move rendering code in SurfaceViewRenderer to a separate class.
The new SurfaceEglRenderer helper class extends EglRenderer and
implements rendering on a SurfaceView.

Bug: webrtc:8242
Change-Id: Ic532fe487755d3b54c6bd03f239d714e1ecb10ad
Reviewed-on: https://webrtc-review.googlesource.com/2940
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20562}
2017-11-06 13:52:26 +00:00
e21be1db4c Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Reason for revert:
Fixes has landed.

Original issue's description:
> Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
>
> Reason for revert:
> We are not certain this is the behavior we want.
>
> Original issue's description:
> > Fix the video buffer size should take rtt into consideration
> >
> > BUG=webrtc:8010
> >
> > Review-Url: https://codereview.webrtc.org/2980413002
> > Cr-Commit-Position: refs/heads/master@{#19285}
> > Committed: f1e08d0b58
>
> TBR=sprang@webrtc.org,gustavogb@gmail.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/3002033002
> Cr-Commit-Position: refs/heads/master@{#19442}
> Committed: bdbc8895f3

TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/3016633002
Cr-Commit-Position: refs/heads/master@{#19944}
2017-09-25 13:37:12 +00:00
bdbc8895f3 Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
Reason for revert:
We are not certain this is the behavior we want.

Original issue's description:
> Fix the video buffer size should take rtt into consideration
>
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/2980413002
> Cr-Commit-Position: refs/heads/master@{#19285}
> Committed: f1e08d0b58

TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/3002033002
Cr-Commit-Position: refs/heads/master@{#19442}
2017-08-22 09:08:51 +00:00
f1e08d0b58 Fix the video buffer size should take rtt into consideration
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/2980413002
Cr-Commit-Position: refs/heads/master@{#19285}
2017-08-09 12:43:08 +00:00
f3a48ab6dc Delete unused field from AndroidVideoTrackSource
BUG=None

Review-Url: https://codereview.webrtc.org/2974713002
Cr-Commit-Position: refs/heads/master@{#19117}
2017-07-24 08:06:39 +00:00
ff7acb19a1 Reset isFirstFrameRendered on init of SurfaceViewRenderer
If a SurfaceViewRenderer is reinitialized, the onFirstFrameRendered
callback is not fired.

Ensure that we reset the flag when the SurfaceViewRenderer is
initialized.

BUG=webrtc:7985

Review-Url: https://codereview.webrtc.org/2981793002
Cr-Commit-Position: refs/heads/master@{#19016}
2017-07-14 09:35:53 +00:00
c43f68e52c Fix do not unregister bluetooth receiver if it was not registered
Bug: webrtc:7890
Change-Id: Ib46b4a4407fa030500930ed03a093b26c71f8963
Reviewed-on: https://chromium-review.googlesource.com/550617
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18892}
2017-07-04 13:50:15 +00:00
1b2469b878 Fix AVFoundation framework import
When building the WebRTC project for iOS, the build will fail on Xcode 9
because of a missing framework-header (AVFoundation). This pull-request
will add the missing "#import <AVFoundation/AVFoundation.h>" line to the
"RTCCameraVideoCapturer" class.

BUG=webrtc:7846

Review-Url: https://codereview.webrtc.org/2944753002
Cr-Commit-Position: refs/heads/master@{#18698}
2017-06-21 10:44:05 +00:00
8e857d10fd Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device.
BUG=webrtc:7719

Change-Id: Iddc66188341c0c90e96766dff671ac3863bf3f5d
Reviewed-on: https://chromium-review.googlesource.com/517523
Commit-Queue: Peter Boström <pbos@webrtc.org>
Reviewed-by: Peter Boström <pbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18392}
2017-06-01 21:10:29 +00:00
ace5c8836d This CL adds RTCMTLVideoView.h and RTCCameraVideoCapturer.h to WebRTC.h
in order to fix a build issue that comes up when using WebRTC.framework from swift code.

BUG=webrtc:7488

Review-Url: https://codereview.webrtc.org/2832803002
Cr-Commit-Position: refs/heads/master@{#17909}
2017-04-27 13:26:19 +00:00
a1fa491334 Fix invalid output buffer usage
This patch fixes the internal AudioCoder output buffer setting to be set
prior it will be used within callback from ACM

BUG=webrtc:7462

Review-Url: https://codereview.webrtc.org/2806933002
Cr-Commit-Position: refs/heads/master@{#17800}
2017-04-20 22:19:10 +00:00
0d335c7756 Fixed that RTCCameraPreviewView did not rotate the video on device rotation.
BUG=webrtc:6749

Review-Url: https://codereview.webrtc.org/2798993002
Cr-Commit-Position: refs/heads/master@{#17742}
2017-04-18 14:12:05 +00:00
dax
9d65f39d52 Added support for changing the volume of AudioTrack as discussed in BUG=webrtc:6533
This is a short term solution to change the volume of an AudioTrack until applyConstraints for MediaStreamTracks has been implemented.

This CL adds 1 new Java method & the relevant JNI file update:

AudioTrack.java:

public void setVolume(double volume);

BUG=webrtc:6533

Review-Url: https://codereview.webrtc.org/2710683009
Cr-Commit-Position: refs/heads/master@{#17682}
2017-04-12 23:58:48 +00:00
0642b3297d Remove duplicate entries from AUTHORS file
BUG=none
NOTRY=True
TBR=alessiob@webrtc.org

Review-Url: https://codereview.webrtc.org/2813553004
Cr-Commit-Position: refs/heads/master@{#17617}
2017-04-10 11:54:00 +00:00
9f2c18e237 Changed OLA window for neteq. Old code didnt work well with 48khz
fixing white spaces

updated authors file

Changed OLA window to use Q14 as Q5 dosnt work with 48khz. 1 ms @ 48 khz is > 2^5

BUG=webrtc:1361

Review-Url: https://codereview.webrtc.org/2763273003
Cr-Commit-Position: refs/heads/master@{#17611}
2017-04-10 09:22:46 +00:00
4b37127414 Fix compilation issues of std::unique_ptr
This patch fixes compilation issues related to usage of std::unique_ptr
and NULL instead of nullptr. This issue pops up once you would try to
compile whole webrtc with using C++14 and gcc-4.9

BUG=webrtc:7461

Review-Url: https://codereview.webrtc.org/2806693004
Cr-Commit-Position: refs/heads/master@{#17600}
2017-04-09 16:09:06 +00:00
28dc285f22 Adding cbr support for Opus
BUG=webrtc:7394

Review-Url: https://codereview.webrtc.org/2772773002
Cr-Commit-Position: refs/heads/master@{#17564}
2017-04-06 12:48:36 +00:00
0248e7c810 Re-add author accidentally removed in https://codereview.webrtc.org/2534843002.
BUG=None

Review-Url: https://codereview.webrtc.org/2785453002
Cr-Commit-Position: refs/heads/master@{#17422}
2017-03-28 13:05:00 +00:00
846e1be85c Fix iOS8 crash in background mode.
Add system version check functionality in UIDevice+RTCDevice category.
Check for iOS system version when handle capture session interruption.

BUG=webrtc:7201

Review-Url: https://codereview.webrtc.org/2733773003
Cr-Commit-Position: refs/heads/master@{#17079}
2017-03-07 00:42:19 +00:00
228c268065 Support 4 channel mic in Windows Core Audio
BUG=webrtc:7220

Review-Url: https://codereview.webrtc.org/2712743004
Cr-Commit-Position: refs/heads/master@{#16940}
2017-03-01 13:11:22 +00:00
0d1305ee88 Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533
This is a short term solution to change the volume of a RTCAudioTrack (which contains an RTCAudioSource property) until applyConstraints for RTCMediaStreamTracks has been implemented.
This CL adds one new Objective-C method to AudioSourceInterface's wrapper: -(void)setVolume:(double)volume

BUG=webrtc:6533, webrtc:6805

This is my first CL for Chromium/WebRTC, so please let me know if I did something wrong.

Review-Url: https://codereview.webrtc.org/2534843002
Cr-Commit-Position: refs/heads/master@{#16809}
2017-02-23 21:57:00 +00:00
8a855d6916 Allow any unsignalled SSRC changes on default video receive channel.
The first unsignalled SSRC creates a default receive channel.
Any unsignalled SSRC changes after that replace the default SSRC.
Add unit tests for changing unsignalled SSRCs.

BUG=webrtc:5208

Review-Url: https://codereview.webrtc.org/2692993009
Cr-Commit-Position: refs/heads/master@{#16682}
2017-02-17 23:46:43 +00:00
b11fb25c12 Protect APM in webkit builds.
Update libwertc AudioRtpSender::SetAudioSend with WEBRTC_WEBKIT_BUILD

This only introduces the WEBRTC_WEBKIT BUILD, inspired by WEBRTC_CHROMIUM_BUILD
macro. It is only defined by Webkit libwebrtc build system.
https://trac.webkit.org/changeset/210977

BUG=webrtc:7039

Review-Url: https://codereview.webrtc.org/2651273003
Cr-Commit-Position: refs/heads/master@{#16432}
2017-02-03 14:37:05 +00:00
888874f761 Allow GCC 4.9 to compile Chromium
In order to implicit cast an lvalue to an rvalue when returning
from a function, the return type and type of variable in the return
statement previously had to be exactly the same. When this was not
the case, std::move was required. For instance, when returning a
std::unique_ptr<Derived> variable in a function with a
std::unique_ptr<Base> return type, std::move is required.

DR 1579 changed this, and allows for implicitly converting
to the return type, if the return type has a constructor(T&&), where
T is the type of the local variable being returned. DR 1579 was
implemented in GCC 5, but not in GCC 4.9 and below. By explicitly
qualifying the local variable with std::move, we allow for compiling
with GCC 4.9 and incur no performance penalty. The code is still
absolutely correct to the word of C++11.

BUG=chromium:682965

See also:
* https://bugs.gentoo.org/show_bug.cgi?id=600288
* https://stackoverflow.com/questions/22018115/converting-stdunique-ptrderived-to-stdunique-ptrbase#comment33375875_22018521
* http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2014/n3833.html#1579

Review-Url: https://codereview.webrtc.org/2642053003
Cr-Commit-Position: refs/heads/master@{#16175}
2017-01-20 04:20:45 +00:00
e5dc62aeb8 PRESUBMIT: Add authorized-authors check + AUTHORS e-mails.
This check will throw a PRESUBMIT error if the author or
organization is not present in the AUTHORS file.

E-mail wildcard entries were also added to the organizations
in the AUTHORS file.

BUG=webrtc:6852
NOTRY=True

Review-Url: https://codereview.webrtc.org/2564613002
Cr-Commit-Position: refs/heads/master@{#15591}
2016-12-14 08:16:29 +00:00
ba7e71b53a remove googViewLimitedResolution stat
adaptReason in webrtcvideoengine2.h only defines NONE=0, CPU=1 and BANDWIDTH=2 so &0x4 can not happen anymore.
This was probably never implemented in videoengine2

BUG=webrtc:6870

Review-Url: https://codereview.webrtc.org/1887773002
Cr-Commit-Position: refs/heads/master@{#15546}
2016-12-12 12:46:27 +00:00
bbfed52cf2 Set OPENSSL_EC_NAMED_CURVE explicitly on EC key so that certificate has ASN1 OID and NIST curve info. Without this openSSL handshake negotiation fails throwing NO_SHARED_CIPHER error. the change made is along the lines of openssl behavior documented here: https://wiki.openssl.org/index.php/Elliptic_Curve_Diffie_Hellman#ECDH_and_Named_Curves
tested with openssl 1.0.2j

BUG=webrtc:6763

Review-Url: https://codereview.webrtc.org/2534773002
Cr-Commit-Position: refs/heads/master@{#15536}
2016-12-12 02:42:14 +00:00
610c454cf9 Add Datachannel support to Android AppRTCMobile
BUG=webrtc:6647

Review-Url: https://codereview.webrtc.org/2464243002
Cr-Commit-Position: refs/heads/master@{#15145}
2016-11-18 08:11:04 +00:00
bcc5d87f09 Add a GN target for unit tests, get them working again and added a test.
BUG=webrtc:3417

Review-Url: https://codereview.webrtc.org/2050153003
Cr-Commit-Position: refs/heads/master@{#14959}
2016-11-07 22:53:35 +00:00
a264ecc456 Copy RTCAudioSource.h and RTCMediaSource.h with other public header files when building the WebRTC framework for iOS / Mac
NOTRY=True

Review-Url: https://codereview.webrtc.org/2313473002
Cr-Commit-Position: refs/heads/master@{#14117}
2016-09-08 06:11:29 +00:00
86ccd7bfba Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ )
Reason for revert:
Breaks chromium.

Original issue's description:
> Add field_trial_default dependency to libjingle_peerconnection
>
> This is needed for webrtc::field_trial::FindFullName in peerconnection.cc
>
> NOTRY=True
>
> Committed: https://crrev.com/a7a01df2aebe7108afad208ccd0341c2f0bc7b3b
> Cr-Commit-Position: refs/heads/master@{#13836}

TBR=pthatcher@webrtc.org,pthatcher@chromium.org,kjellander@webrtc.org,arlolra@gmail.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2263063002
Cr-Commit-Position: refs/heads/master@{#13837}
2016-08-22 07:26:11 +00:00
a7a01df2ae Add field_trial_default dependency to libjingle_peerconnection
This is needed for webrtc::field_trial::FindFullName in peerconnection.cc

NOTRY=True

Review-Url: https://codereview.webrtc.org/2120673004
Cr-Commit-Position: refs/heads/master@{#13836}
2016-08-22 06:48:14 +00:00
96b6b8336a iOS: add type to peer connection local streams
BUG=

Review-Url: https://codereview.webrtc.org/2249173002
Cr-Commit-Position: refs/heads/master@{#13825}
2016-08-18 21:21:27 +00:00
3f70562bbb Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015).
Visual Studio 2015 balks at the implicit truncation of values. Easily fixed with an explicit cast.

Fixed redefinition of CLOCKS_PER_SEC when using Visual Studio 2015 and the Windows 10 SDK. CLOCKS_PER_SEC is also defined in "<WIN10 SDK DIR>\include\10.0.10240.0\ucrt\time.h" and also has the value of 1000

Hiding snprintf definition if building with Visual Studio 2015

Fixed C4573 compiler complaint in audio_processing_impl_locking_unittest.cc.

BUG=webrtc:5183

Review URL: https://codereview.webrtc.org/1412653006

Cr-Commit-Position: refs/heads/master@{#11434}
2016-01-30 22:40:52 +00:00
bedc17be5c Fixing integer underflow in FileAudioDevice (webrtc issue 4554)
Problem is described here:
https://code.google.com/p/webrtc/issues/detail?id=4554

Review URL: https://codereview.webrtc.org/1295603002

Cr-Commit-Position: refs/heads/master@{#11174}
2016-01-07 20:38:36 +00:00
978244ecb0 Adding a bunch of Agora IO team members to the watch lists
BUG=

Review URL: https://codereview.webrtc.org/1470833002

Cr-Commit-Position: refs/heads/master@{#10765}
2015-11-24 08:22:19 +00:00
f70568c04b So long and thanks for all the code reviews!
- Remove myself from OWNERS.
- Add myself to AUTHORS (I signed a CLA).
- Add minyue to audio_conference_mixer which would otherwise be empty.
- Add missing comma in WATCHLISTS.

Review URL: https://codereview.webrtc.org/1458763002

Cr-Commit-Position: refs/heads/master@{#10686}
2015-11-18 11:07:45 +00:00