7086 Commits

Author SHA1 Message Date
philipel
504c47d750 FrameBuffer2 now has Start/Stop methods.
The Stop method is used to signal any thread that is waiting in the
NextFrame function and will cause it to return immediately.

BUG=webrtc:5514
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/2105323002 .

Cr-Commit-Position: refs/heads/master@{#13349}
2016-06-30 15:33:17 +00:00
Sami Kalliomaki
039083aedc Dispose audio source correctly in AppRTC Demo on Android.
Call dispose method on audio source in PeerConnectionClient dispose
method.

R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/2117463002 .

Cr-Commit-Position: refs/heads/master@{#13348}
2016-06-30 14:57:22 +00:00
Peter Boström
c5ad0c81c7 Respect VP8.automaticResizeOn for MediaCodec.
Disables QualityScaler for screenshare-type content and simulcast inside
MediaCodecVideoEncoder.

BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/2107233003 .

Cr-Commit-Position: refs/heads/master@{#13347}
2016-06-30 14:11:43 +00:00
peah
5f6547e89c Disabling the performance unittests on Android for the level controller.
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2108723004
Cr-Commit-Position: refs/heads/master@{#13346}
2016-06-30 12:13:46 +00:00
aleloi
03dd6db5cb Added empty directory with myself as owner for new mixer.
NOTRY=True

Review-Url: https://codereview.webrtc.org/2109133003
Cr-Commit-Position: refs/heads/master@{#13345}
2016-06-30 12:03:17 +00:00
pbos
9ef6785027 Revert of Workaround for clang bug http://llvm.org/PR28348. (patchset #1 id:1 of https://codereview.webrtc.org/2110043003/ )
Reason for revert:
Not needed since https://codereview.chromium.org/2110873002/.

Original issue's description:
> Workaround for clang bug http://llvm.org/PR28348.
>
> Permits rolling chromium further.
>
> BUG=
> TBR=tommi@webrtc.org
>
> Committed: f516585e10

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2106383002
Cr-Commit-Position: refs/heads/master@{#13344}
2016-06-30 09:32:37 +00:00
ossu
ea41694e08 Added an empty member to the union of rtc::Optional, so that it is always initializable.
Added notry due to flaky android_dbg bot.

NOTRY=true
BUG=webrtc:6033

Review-Url: https://codereview.webrtc.org/2090223003
Cr-Commit-Position: refs/heads/master@{#13343}
2016-06-30 09:15:00 +00:00
danilchap
f4e8cf0d5b Revert of Add config to prune TURN ports (patchset #12 id:360001 of https://codereview.webrtc.org/2093623004/ )
Reason for revert:
Breaks Win32/Win64 Debug bots in client.webrtc waterfall

Original issue's description:
> Add config to prune low-priority TURN ports for creating connections
> When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
>
> This effectively reduces the number of TURN candidates and connections created by TURN ports.
>
> BUG=
> R=deadbeef@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
> Cr-Commit-Position: refs/heads/master@{#13335}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2111663003
Cr-Commit-Position: refs/heads/master@{#13342}
2016-06-30 08:55:10 +00:00
Magnus Jedvert
77ed80a7ef AndroidVideoCapturer: Remove unused member variable
The member variable |current_state_| in AndroidVideoCapturer is
unnecessary. All state changes are reported to the base class
cricket::VideoCapturer that already holds the capture state.

R=sakal@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2104813003 .

Cr-Commit-Position: refs/heads/master@{#13341}
2016-06-30 08:05:46 +00:00
ivoc
9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749

Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
2016-06-30 07:59:49 +00:00
ivoc
308c7b0b5a Revert of Fix to make the start/stop functions for the Rtc Eventlog non-virtual. (patchset #2 id:40001 of https://codereview.webrtc.org/2107253002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Fix to make the start/stop functions for the Rtc Eventlog non-virtual.
>
> This is needed to prevent the Chromium import bot from breaking.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/df6ecea8ac7c4c3bddeda089d5fb9eccdf38a0a6
> Cr-Commit-Position: refs/heads/master@{#13324}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2111803002
Cr-Commit-Position: refs/heads/master@{#13339}
2016-06-30 07:57:40 +00:00
ivoc
bf02e2070f Revert of Fix for RtcEventLog ObjC interface (patchset #3 id:40001 of https://codereview.webrtc.org/2106103003/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Fix for RtcEventLog ObjC interface
>
> This moves the RtcEventLog start/stop functions to PeerConnection on the objC interface.
>
> BUG=
> R=tkchin@webrtc.org
>
> Committed: c43bf56ef1

TBR=tkchin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2110373002
Cr-Commit-Position: refs/heads/master@{#13338}
2016-06-30 07:55:07 +00:00
Ivo Creusen
c43bf56ef1 Fix for RtcEventLog ObjC interface
This moves the RtcEventLog start/stop functions to PeerConnection on the objC interface.

BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/2106103003 .

Cr-Commit-Position: refs/heads/master@{#13337}
2016-06-30 07:32:15 +00:00
peah
a3333bfafb This CL adds activation logic of the new APM level control
functionality and exposes the functionality using the
MediaConstraints.

The exposing of the feature through the  MediaConstraints
was done similarly to what was done for the intelligibility
enhancer in the CL
https://codereview.webrtc.org/1952123003

This CL is dependent on the CL https://codereview.webrtc.org/2090583002/ which contains
the level control functionality.

NOTRY=true
BUG=webrtc:5920

Review-Url: https://codereview.webrtc.org/2095563002
Cr-Commit-Position: refs/heads/master@{#13336}
2016-06-30 07:02:41 +00:00
Honghai Zhang
17aac053f5 Add config to prune low-priority TURN ports for creating connections
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).

This effectively reduces the number of TURN candidates and connections created by TURN ports.

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

Review URL: https://codereview.webrtc.org/2093623004 .

Cr-Commit-Position: refs/heads/master@{#13335}
2016-06-30 04:42:05 +00:00
skvlad
4c4cb5b984 Separate the JNI function that controls logging levels into two.
The parameters for Logging.enableTracing() were creating the impression
that they control level and severity of one tracing system and they are
meant to be used together. In fact the "levels" parameter controlled one
tracing system (WEBRTC_TRACE), and the "severity" parameter was
responsible for a completely different one: setting the severity level
above which log messages from LOG() will be directed to the
platform-specific debug output (logcat on Android).

The method signature suggested that the "path" parameter applied to both
systems - while it was only meaningful for the WEBRTC_TRACE; LOG
messages were directed to ADB logcat no matter what the Path value was.
It is possible to redirect LOG messages to a file, but that is done
using a completely different set of APIs
 - PeerConnectionFactory.startInternalTracingCapture().

I've separated these two methods to make it more clear which of the
parameters controls which system.

NOTRY=true

Review-Url: https://codereview.webrtc.org/2110853003
Cr-Commit-Position: refs/heads/master@{#13334}
2016-06-29 22:30:48 +00:00
peah
ca4cac7e74 New module for the adaptive level controlling functionality in the audio processing module
NOTRY=true
TBR=aluebs@webrtc.org
BUG=webrtc:5920

Review-Url: https://codereview.webrtc.org/2090583002
Cr-Commit-Position: refs/heads/master@{#13333}
2016-06-29 22:26:19 +00:00
deadbeef
886815bc36 Add back P2PTransportChannel.Connect, for Chromium.
Will remove it as soon as Chromium does a roll and can stop using it.

TBR=pthatcher@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2110853005
Cr-Commit-Position: refs/heads/master@{#13332}
2016-06-29 22:21:14 +00:00
Taylor Brandstetter
b825aee04a Start ICE connectivity checks as soon as the first pair is pingable.
Previously, we were starting a periodic timer when the local
description was set. The first connection may be created at any
time after this happens, so after creating the first connection, we
need to wait until that timer next fires before sending a ping.

Now we just start that timer (and send the first ping) immediately
after the first connection becomes pingable.

This CL also removes the "Connect" method. The only vestigal
effect of this method was to start the periodic timer, which is
now not needed since it happens automatically.

R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2099563004 .

Cr-Commit-Position: refs/heads/master@{#13331}
2016-06-29 20:07:26 +00:00
Peter Boström
f516585e10 Workaround for clang bug http://llvm.org/PR28348.
Permits rolling chromium further.

BUG=
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2110043003 .

Cr-Commit-Position: refs/heads/master@{#13330}
2016-06-29 18:25:38 +00:00
Taylor Brandstetter
bf2f569b22 Fixing flakiness of CreateDataChannelAfterNegotiate.
Test worked by creating a dummy data channel just to trigger the
SDP generation, then creating two data channels after negotiation.

However the dummy data channel is then racing with the "real" data
channel to get negotiated, so they could be signaled in the reverse
of the expected order.

Fixed this by simply waiting for the dummy data channel to be
signaled before creating the other data channels.

BUG=webrtc:3980
R=pthatcher@webrtc.org, skvlad@webrtc.org

Review URL: https://codereview.webrtc.org/2112593002 .

Cr-Commit-Position: refs/heads/master@{#13329}
2016-06-29 18:25:15 +00:00
Taylor Brandstetter
6c3e788dcf Add RTX codecs for codecs only supported by external encoder.
Previously we were only adding these RTX codecs if the codec was
internally supported.

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

Review URL: https://codereview.webrtc.org/2088233004 .

Cr-Commit-Position: refs/heads/master@{#13328}
2016-06-29 18:14:29 +00:00
Taylor Brandstetter
f3d8d32c5f Fixing P2PTransportChannelTest.TestIceConfigWillPassDownToPort.
Was flaky because it was checking the role of a port, but a role
conflict could occur before this happened which would swap the
role.

BUG=webrtc:6019
R=honghaiz@webrtc.org, pthatcher@webrtc.org
TBR=honghaiz@webrtc.org

Review URL: https://codereview.webrtc.org/2108493003 .

Cr-Commit-Position: refs/heads/master@{#13327}
2016-06-29 18:07:44 +00:00
Taylor Brandstetter
54433e72aa Fixing flakiness of TestDetectUnresolvedProxy.
If the AutoDetectProxy timed out resolving one address, it was
attempting to use the same resolver to resolve the next address,
which would always result in an assertion. This happened recently
a couple times on the Windows DrMemory bot because of its slowness.

TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2104203002 .

Cr-Commit-Position: refs/heads/master@{#13326}
2016-06-29 17:10:38 +00:00
Johan Ahlers
37f93af032 Remove unused constructor from VCMPacket.
BUG=
R=pbos@webrtc.org, sprang@webrtc.org

Review URL: https://codereview.webrtc.org/2110663004 .

Cr-Commit-Position: refs/heads/master@{#13325}
2016-06-29 16:03:20 +00:00
Ivo Creusen
df6ecea8ac Fix to make the start/stop functions for the Rtc Eventlog non-virtual.
This is needed to prevent the Chromium import bot from breaking.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2107253002 .

Cr-Commit-Position: refs/heads/master@{#13324}
2016-06-29 15:31:07 +00:00
Danil Chapovalov
18c65a448f Expanded error message for unexpected packet
in the flaky test EndToEndTest.DecodesRetransmittedFrame*

BUG=webrtc:5540
R=pbos@webrtc.org, pbos

Review URL: https://codereview.webrtc.org/2100793002 .

Cr-Commit-Position: refs/heads/master@{#13323}
2016-06-29 13:29:48 +00:00
Sami Kalliomaki
9c0c75bd6e Add GN targets for AppRTC Demo on Android.
Adds GN equivalents for following targets:
AppRTCDemo          -> //webrtc/examples:AppRTCDemo
AppRTCDemo_apk      -> //webrtc/examples:AppRTCDemo_lib (kind of)
AppRTCDemoJUnitTest -> //webrtc/examples:AppRTCDemoJUnitTest
AppRTCDemoTest      -> //webrtc/examples:AppRTCDemoTest
libjingle_peerconnection_java -> //webrtc/api/libjingle_peerconnection_java
libjingle_peerconnection_so   -> //webrtc/api/libjingle_peerconnection_so

New GN targets:
//webrtc/base:base_java
//webrtc/examples:AppRTCDemo_resources
//webrtc/examples/androidapp/third_party/autobanh:autobanh_java

BUG=webrtc:6035
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2105803002 .

Cr-Commit-Position: refs/heads/master@{#13322}
2016-06-29 12:55:12 +00:00
Ivo Creusen
1895526c61 Move RtcEventLog object from inside VoiceEngine to Call.
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.

BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1748403002 .

Cr-Commit-Position: refs/heads/master@{#13321}
2016-06-29 11:57:01 +00:00
Sami Kalliomaki
8cf2a3a3ad Android: Camera2 implementation and tests for it.
BUG=webrtc:5519
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/2078473002 .

Cr-Commit-Position: refs/heads/master@{#13320}
2016-06-29 11:27:50 +00:00
Magnus Jedvert
f4878e5968 VideoCapturerAndroid: Remove unused function getCameraThreadHandler
R=sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2107803002 .

Cr-Commit-Position: refs/heads/master@{#13319}
2016-06-29 09:43:13 +00:00
Joachim Bauch
37bb54eb20 Reland: Remove global list of SRTP sessions.
Instead save a reference to the SrtpSession inside the srtp_ctx_t.

The original CL was https://codereview.webrtc.org/1416093010 and
should be good to reland now that internal projects are using a
more recent version of libsrtp.

BUG=webrtc:5133
R=mattdr@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2109893002 .

Cr-Commit-Position: refs/heads/master@{#13318}
2016-06-29 08:41:16 +00:00
Taylor Brandstetter
65f47275cb Fixing race condition in IncomingVideoStream.
This really only happens on the memcheck bot. But the issue is that
the render thread may be started before the timer is started on
the main thread, which incorrectly attaches the timer to the render
thread. Then a thread check assertion occurs when the timer is
stopped on the main thread.

Simply starting the timer before starting the render thread fixes this.

BUG=webrtc:6062
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2105013002 .

Cr-Commit-Position: refs/heads/master@{#13317}
2016-06-28 17:46:44 +00:00
Alejandro Luebs
a3c51ea9f4 Revert "Pull out the PostFilter to its own NonlinearBeamformer API"
This reverts commit b983112bc7686ed4276def4c9215c498444c6bdd.

It was breaking dependencies.
TBR=aluebs@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/2110503002 .

Cr-Commit-Position: refs/heads/master@{#13316}
2016-06-28 17:38:41 +00:00
Sami Kalliomaki
9c7a0dbc8a Constructor in Camera1Enumerator should be public.
R=danilchap@webrtc.org
TBR=magjed_webrtc

Review URL: https://codereview.webrtc.org/2106863002 .

Cr-Commit-Position: refs/heads/master@{#13315}
2016-06-28 17:04:06 +00:00
Alejandro Luebs
b983112bc7 Pull out the PostFilter to its own NonlinearBeamformer API
This is done to avoid having a nonlinear component in the AEC path.
Now the linear delay and sum is run before the AEC and the postfilter after it.

R=henrik.lundin@webrtc.org, peah@webrtc.org

Review URL: https://codereview.webrtc.org/1982183002 .

Cr-Commit-Position: refs/heads/master@{#13314}
2016-06-28 17:02:57 +00:00
Honghai Zhang
d00c05788f Fix the turn and udp port type.
The port type was not set if it was created on a shared socket.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2099023002 .

Cr-Commit-Position: refs/heads/master@{#13313}
2016-06-28 16:44:55 +00:00
sakal
70fae2ccc6 Add override annotation to appropriate methods in Camera1Enumerator.
Also move getDeviceNames to a more appropriate location in the file.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2105813002
Cr-Commit-Position: refs/heads/master@{#13312}
2016-06-28 15:36:43 +00:00
noahric
0dacbf5f62 Create a null decoder for h264 when not supported, instead of crashing.
BUG=
NOTRY=true

Review-Url: https://codereview.webrtc.org/2091923002
Cr-Commit-Position: refs/heads/master@{#13311}
2016-06-28 13:39:52 +00:00
Johan Ahlers
31b2ec4e0d Remove unused output parameter in VCMReceiver::FrameForDecoding().
BUG=
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/2104863002 .

Cr-Commit-Position: refs/heads/master@{#13310}
2016-06-28 11:32:57 +00:00
Johan Ahlers
95348f7663 Remove unused parameters from VCMReceiver::InsertPacket().
BUG=
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/2094183004 .

Cr-Commit-Position: refs/heads/master@{#13309}
2016-06-28 09:11:37 +00:00
Max Morin
098e6c5d0a Logging and tracing of audio devices on Andriod.
Replaced invokations of WEBRTC_TRACE with LOG, which is
visible in the android system log.

BUG=NONE
R=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/2091803002 .

Cr-Commit-Position: refs/heads/master@{#13308}
2016-06-28 07:36:39 +00:00
Taylor Brandstetter
6bb1ef2b86 Fixing bug where Connection drops packets when presumed writable.
The "should I simulate EWOULDBLOCK?" determination now happens
solely in P2PTransportChannel. This also fixes a bug where the
"last packet id" was set even if no packet was sent.

R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2099783002 .

Cr-Commit-Position: refs/heads/master@{#13307}
2016-06-28 01:09:10 +00:00
Taylor Brandstetter
f8e65779a7 Add virtual Initialize methods to PortAllocator and NetworkManager.
This will allow PeerConnection to handle hopping to the right thread
and doing thread-specific initialization for the PortAllocator.
This eliminates a required thread-hop for whatever is passing the
PortAllocator into CreatePeerConnection.

BUG=617648
R=pthatcher@webrtc.org, skvlad@webrtc.org

Review URL: https://codereview.webrtc.org/2097653002 .

Committed: https://crrev.com/a6bdb0990a659ff9e7c4374f5033a6bcc4fbfb21
Cr-Original-Commit-Position: refs/heads/master@{#13283}
Cr-Commit-Position: refs/heads/master@{#13306}
2016-06-28 00:20:25 +00:00
Taylor Brandstetter
ba29c6aac7 Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
Relanding again after fixing issue with RTC_DCHECKs.

This CL eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.

It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2046173002 .

Cr-Commit-Position: refs/heads/master@{#13305}
2016-06-27 23:30:45 +00:00
Taylor Brandstetter
716d07a241 Using fake clock for TURN port tests and un-disabling some tests.
The fake clock has a few advantages:
1. It lets use verify that operations take the expected number of
   round trips.
2. It makes the tests faster by letting us remove the equivalent
   of "Sleep(500)" all over the tests.
3. It makes the tests less flaky, because sometimes sleeping for
   500ms or waiting for 1s is not enough.

R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2097793003 .

Cr-Commit-Position: refs/heads/master@{#13304}
2016-06-27 21:07:51 +00:00
deadbeef
1caff88945 Disabling EndToEndTest.RestartingSendStreamPreservesRtpStatesWithRtx.
Was thought to be only flaky on Mac, but just failed on Win SyzyASan.
So, disabling until flakiness is fixed.

BUG=webrtc:4332
TBR=pbos@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2104583002
Cr-Commit-Position: refs/heads/master@{#13303}
2016-06-27 20:10:02 +00:00
Alex Glaznev
d57048433c Decrease the amount of maximum outstanding frames for Android HW H.264 decoder.
BUG=b/28150902
R=pbos@webrtc.org, sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2088353002 .

Cr-Commit-Position: refs/heads/master@{#13302}
2016-06-27 18:51:24 +00:00
Honghai Zhang
56ce49d710 Delete a method that was not used.
This was a mistake from code merging.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2095263002 .

Cr-Commit-Position: refs/heads/master@{#13301}
2016-06-27 18:19:33 +00:00
Peter Boström
3d2e853f86 Replace OS_ANDROID define to disable webrtc test.
Replaces OS_ANDROID with WEBRTC_ANDROID to disable
ProcessNoLossChangeFrameRateFrameDropVP9 on Android.

BUG=webrtc:6057
R=danilchap@webrtc.org
TBR=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/2101673002 .

Cr-Commit-Position: refs/heads/master@{#13300}
2016-06-27 17:47:20 +00:00