The flag used in thread_unittest.cc:FunctorB is subject to a (mostly
harmless) data race. In a tsan build, reproduce using
out/Release/rtc_unittests --gtest_filter=AsyncInvokeTest.FireAndForget
There are additional tsan warnings, not all deterministic, when
running all the rtc_unittets: Some data races related to destructors,
and a locking-order-inversion warning. Hence applying this patch does
not make the unit tests tsan-clean.
I should also add that this is my very first cl, so I'm not at all
familiar with the process.
Review URL: https://codereview.webrtc.org/1439613004
Cr-Commit-Position: refs/heads/master@{#10645}
The ARRAY_SIZE macro it defines is not used anymore, as all the usages
were converted to arraysize macro from arraysize.h.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1443273002
Cr-Commit-Position: refs/heads/master@{#10640}
The original purpose with keeping one pending frame in SurfaceViewRenderer was to reduce latency for the first rendered frame when we are waiting for the Surface to be created. However, it is very dangerous to hold a pending frame indefinitely when used with a SurfaceTexture, because the SurfaceTexture only has one frame and thus holding a frame in the renderer will freeze everything and typically cause timeout crashes.
Review URL: https://codereview.webrtc.org/1435413006
Cr-Commit-Position: refs/heads/master@{#10638}
Reason for revert:
Causes fallback to SW decoder if a renderer is put in the background.
Original issue's description:
> Patchset 1 is a pure
> revert of "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/
>
> Following patchsets move the responsibility of calculating the decode time to Java.
>
> TESTED= Apprtc loopback using H264 and VP8 on N5, N6, N7, S5
>
> Committed: https://crrev.com/9cb8982e64f08d3d630bf7c3d2bcc78c10db88e2
> Cr-Commit-Position: refs/heads/master@{#10597}
TBR=magjed@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
Review URL: https://codereview.webrtc.org/1441363002 .
Cr-Commit-Position: refs/heads/master@{#10637}
to match name given in the RFC5450
private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
to make class usable for parsing packet
Review URL: https://codereview.webrtc.org/1434213004
Cr-Commit-Position: refs/heads/master@{#10636}
Reason for revert:
Unfortunately this breaks an internal downstream project since we have an ancient libsrtp. Reverting until we can figure out how to update our libsrtp.
Original issue's description:
> Remove global list of SRTP sessions.
> Instead save a reference to the SrtpSession inside the srtp_ctx_t.
>
> BUG=webrtc:5133
>
> Committed: https://crrev.com/9cafd972779ed7b25886ab276e0ede7b7a8b76a1
> Cr-Commit-Position: refs/heads/master@{#10591}
TBR=juberti@google.com,juberti@webrtc.org,jbauch@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5133
Review URL: https://codereview.webrtc.org/1442863003
Cr-Commit-Position: refs/heads/master@{#10635}
Put VideoSender/VideoReceiver flat within the object, not as
scoped_ptrs, giving fewer allocations and looking a bit nicer.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1443613002
Cr-Commit-Position: refs/heads/master@{#10634}
I have another CL moving REMB from CongestonController to Call, then
I'll remove CongestoinController from this class too.
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1442003002 .
Cr-Commit-Position: refs/heads/master@{#10632}
Renamed the test to reflect what is actually tested. What the old test
did I don't know since there has never been possible to use screenshare
with two temporal layers in VP9.
BUG=chromium:554515
Review URL: https://codereview.webrtc.org/1441693002
Cr-Commit-Position: refs/heads/master@{#10631}
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1441673002
Cr-Commit-Position: refs/heads/master@{#10630}
When this is detected, we'll now "reconfigure" the senders and
receivers, which will reconnect the capturers/renderers to the
underlying streams which have been recreated.
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1428243005
Cr-Commit-Position: refs/heads/master@{#10628}
This CL attempts to annotate accesses on >16 API levels using as
small scopes as possible. The TargetApi notations mean "yes, I know
I'm accessing a higher API and I take responsibility for gating the
call on Android API level". The Encoder/Decoder classes are annotated
on the whole class, but they're only accessed through JNI; we should
annotate on method level otherwise and preferably on private methods.
This patch also fixes some compiler-level deprecation warnings (i.e.
-Xlint:deprecation), but probably not all of them.
BUG=webrtc:5063
R=henrika@webrtc.org, kjellander@webrtc.org, magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1412673008 .
Cr-Commit-Position: refs/heads/master@{#10624}
This reverts commit 12f680214e28dc5f0a13ac8afc0d1445f89e67e6.
Original cl in https://codereview.webrtc.org/1396073003/
Prepare MediaCodecVideoEncoder for surface textures.
This refactors MediaVideoEncoder to prepare for adding support to encode from textures. The C++ layer does not have any functional changes.
- Moves ResetEncoder to always work on the codec thread
- Adds use of ThreadChecker.
- Change Java MediaEncoder.Init to return true or false and introduce method getInputBuffers.
- Add simple unit test for Java MediaCodecVideoEncoder.
The pure revert of the revert is in patchset 1.
Patchset 2, moves getting the input buffer to before storing pending timestamps etc to fix b/24984012.
BUG=webrtc:4993 b/24984012
Review URL: https://codereview.webrtc.org/1406203002
Cr-Commit-Position: refs/heads/master@{#10622}
This CL does two things:
1) Improves stability in the existing OpenSL ES implementation for devices that
supports OpenSL ES. The cost is a slight increase in latency since the focus here
has been on avoiding audio glitches.
2) Adds a new Java API to exclude usage of OpenSL ES to enable comparisons between
OpenSL ES and Java based audio backends.
BUG=b/22452539
Review URL: https://codereview.webrtc.org/1440623002
Cr-Commit-Position: refs/heads/master@{#10618}
1. allow situation where all ping is lost
2. use the raw count to calculate the interval.
Since we now send 1 request per IP, the chance of losing all of them is higher and we shouldn't just quit if we don't have any response.
BUG=
Review URL: https://codereview.webrtc.org/1406223011
Cr-Commit-Position: refs/heads/master@{#10613}
Reason for revert:
Relanding with compile warning fixed.
Original issue's description:
> Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
>
> Reason for revert:
> Caused compiler warning, breaking Chrome FYI bots.
>
> Original issue's description:
> > Adding the ability to change ICE servers through SetConfiguration.
> >
> > Added a SetIceServers method to PortAllocator. Also added a new
> > PeerConnection Initialize method that takes a PortAllocator, in the
> > hope that we can get rid of PortAllocatorFactoryInterface, since the
> > only substantial thing a factory does is convert the webrtc:: ICE
> > servers to cricket:: versions.
> >
> > Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> > Cr-Commit-Position: refs/heads/master@{#10420}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/18a944bf0ac9eed872dc009bd58e6bc12c946303
> Cr-Commit-Position: refs/heads/master@{#10421}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1414313003
Cr-Commit-Position: refs/heads/master@{#10609}
The JNI code for VoiceEngine is not maintained and VoiceEngine is being
refactored. This is not a supported Java interface, use AppRTCDemo as a
starting point instead.
Also renames webrtc/libjingle_examples.gyp webrtc/webrtc_examples.gyp to
replace the previous file (that only contained media_demo).
BUG=
R=henrika@webrtc.org, kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1439593002 .
Cr-Commit-Position: refs/heads/master@{#10599}
