Removes VP8::Encode trace in favor of VCMGenericEncoder ones and adds
one to InitEncode. Also adds an instant event to ::Encoded since this
can be done on a different thread.
Also adds the corresponding traces to VCMGenericDecoder.
BUG=webrtc:5167
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1412573010
Cr-Commit-Position: refs/heads/master@{#10674}
Disable OpenSL ES by default.
Plus remove no longer used CPU overuse detection option.
Review URL: https://codereview.webrtc.org/1449083002
Cr-Commit-Position: refs/heads/master@{#10670}
-Moved the initialize function
-Moved api_format into the shared state
BUG=
Review URL: https://codereview.webrtc.org/1413093002
Cr-Commit-Position: refs/heads/master@{#10668}
Move logic into cc file
Simplify interval calculation
Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
Remove double divide on each ::Now() invocation on mac
Move TickTime and TickInterval funcitons to cc file in prep for refactoring.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1415923010 .
Cr-Commit-Position: refs/heads/master@{#10661}
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1418913006
Cr-Commit-Position: refs/heads/master@{#10659}
Chromium only checks out the webrtc directory so it misses the root
level .gitignore file which leads to messy "git status" reports inside
third_party/webrtc. This copies the root level .gitignore so that
.vcxproj files, the OSX equivalent, and other files will be ignored.
Some of the entries are irrelevant, but it is better too have a few
irrelevant entries than to be missing some, and the simplicity of
copying is valuable.
NOTRY=True
TBR=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1432413002
Cr-Commit-Position: refs/heads/master@{#10656}
This is to prevent size_t from undefined. This does not happen in current WebRTC since the sources that opus_inst.h gets used have proper definitions. But it would be good to add the definition in itself.
Review URL: https://codereview.webrtc.org/1446093003
Cr-Commit-Position: refs/heads/master@{#10653}
This leaves CodecOwner without a job, so we eliminate it.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1443653004
Cr-Commit-Position: refs/heads/master@{#10650}
Reason for revert:
I have to revert this unfortunately because it adds a dependency on AsyncInvoker, which is not included when building libjingle_nacl in Chromium.
AsyncInvoker needs to first be added to the list of sources in Chromium.
Original issue's description:
> Do not delete the turn port entry right away when the respective connection is deleted.
> BUG=webrtc:5120
>
> Committed: https://crrev.com/e58fe8ef0e6d959f54adee3ed77764927d3845cc
> Cr-Commit-Position: refs/heads/master@{#10641}
TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5120
Review URL: https://codereview.webrtc.org/1449863002
Cr-Commit-Position: refs/heads/master@{#10649}
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}
