A recent update of Windows 10 blocks IDXGIAdapter::EnumOutputs() in session 0,
so ScreenCapturerWinDirectx::IsSupported() always returns false in session 0. We
should ensure ScreenCapturerWinDirectx can respond correctly in session 0.
Meanwhile, this change looses the requirement of DirectX capturer: it still
works if some of the video adapters do not support DirectX 11 or
IDXGIOutputDuplication. This issue usually happens when handling a virtual video
adapter.
BUG=webrtc:7809
Review-Url: https://codereview.webrtc.org/2937663003
Cr-Commit-Position: refs/heads/master@{#18797}
This change also wires up the rest of the production code in
webrtc/sdk/android to be built when the directory is a dependency.
BUG=webrtc:7613
NOTRY=True
Change-Id: Ideda181970a5a570c3f8148b033e471e926243d1
Reviewed-on: https://chromium-review.googlesource.com/548038
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18791}
It was used to force a codec to the top of a list, but it resulted in
"a < a" being true, which some C++ runtimes complain about.
BUG=None
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2963543002
Cr-Commit-Position: refs/heads/master@{#18786}
Dont do a normal AimdRateControlUpdate update after a probe. Only set result.updated if the bitrate estimate has changed.
BUG=webrtc:7866
Review-Url: https://codereview.webrtc.org/2949203002
Cr-Commit-Position: refs/heads/master@{#18785}
This CL corrects the initialization of the AEC3, as well
as for the other submodules in the whole audio processing module
in the sense that it properly update the submodule states also
for the case when reinitialization is trigger from the render
side of the audio processing module.
Bug: chromium:736889,webrtc:7879
Change-Id: I423e963835d0c3227caa8e186b29031bcb912515
Reviewed-on: https://chromium-review.googlesource.com/549315
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18784}
Reenable hanging tests on Mac.
Deadlock happened because the following locks were grabbed by two threads at the end of a test:
Thread 1:
CapturedFrameForwarder::AddOrUpdateSink() locks CapturedFrameForwarder::crit_ and calls
FrameGeneratorCapturer::AddOrUpdateSink() what tries to lock FrameGeneratorCapturer::lock_.
Thread 2:
FrameGeneratorCapturer::InsertFrame() locks FrameGeneratorCapturer::lock_ and calls
CapturedFrameForwarder::OnFrame() which tries to lock CapturedFrameForwarder::crit_.
So two threads are locking two same locks in different orders which may cause deadlock.
BUG=webrtc:7870
Review-Url: https://codereview.webrtc.org/2955083002
Cr-Commit-Position: refs/heads/master@{#18783}
Patch set 1 is a reland + trivial rebase.
Patch set >= 2 contains bug fixes.
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
BUG=webrtc:7664
Review-Url: https://codereview.webrtc.org/2953053002
Cr-Commit-Position: refs/heads/master@{#18782}
This CL corrects the computation of the delay headroom so that
it is only updated when the delay is updated. This is important
as otherwise a too large headroom will be reported, which then
could cause buffer access issues.
Bug: webrtc:7878, chromium:736893
Change-Id: Ib37cb608b064dd5d4df3f8fc423448ee80ed0106
Reviewed-on: https://chromium-review.googlesource.com/549335
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18781}
The iOS 11 SDK adds nullability annotations to several framework functions
and in this it added the _Nonnull specifier to a protocol method that
we implement. We were passing nil to that method in a test.
The warning is now fixed by passing a mock object instead of nil.
Bug: webrtc:7883
Change-Id: I9f64b0c59750629ca3969400aa725729bb10541b
Reviewed-on: https://chromium-review.googlesource.com/549927
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Kári Tristan Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18780}
Tests that still need to be enabled:
* apprtcmobile_tests
* audio_decoder_unittests
* rtc_media_unittests
* rtc_unittests
* webrtc_nonparallel_tests
Bug: webrtc:7155
Change-Id: Ic3bc91224dff8b32dc71e356a4ba04b386c92a0f
Reviewed-on: https://chromium-review.googlesource.com/541368
Commit-Queue: Oleh Prypin <oprypin@chromium.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18775}
Reason for revert:
About to fix problem and reland.
Original issue's description:
> Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ )
>
> Reason for revert:
> Breaks Chromium FYI bots.
>
> The problem is in the BUILD.gn file.
>
> Sample failure:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829
>
> Sample logs:
> use_goma = true
> """ to /b/c/b/Linux_Builder/src/out/Release/args.gn.
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file.
> "//webrtc/base:rtc_base_approved",
> ^--------------------------------
>
> Original issue's description:
> > Create RtcpDemuxer. Capabilities:
> > 1. Demux RTCP messages according to the sender-SSRC.
> > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
> > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2943693003
> > Cr-Commit-Position: refs/heads/master@{#18763}
> > Committed: cb83bdf01f
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2957763002
> Cr-Commit-Position: refs/heads/master@{#18764}
> Committed: 0e7e7869e7
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2960623002
Cr-Commit-Position: refs/heads/master@{#18768}
Occurrences of WEBRTC_TRACE(...) will in the future be replaced with the preferred logging mechanism LOG(...). That will be done with a script that runs 'git cl format' on diffs, which will break formatting of surrounding code if the file is not already formatted. Hence this CL.
BUG=webrtc:5118
Review-Url: https://codereview.webrtc.org/2953793002
Cr-Commit-Position: refs/heads/master@{#18766}
Reason for revert:
Breaks Chromium FYI bots.
The problem is in the BUILD.gn file.
Sample failure:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829
Sample logs:
use_goma = true
""" to /b/c/b/Linux_Builder/src/out/Release/args.gn.
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file.
"//webrtc/base:rtc_base_approved",
^--------------------------------
Original issue's description:
> Create RtcpDemuxer. Capabilities:
> 1. Demux RTCP messages according to the sender-SSRC.
> 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
> 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2943693003
> Cr-Commit-Position: refs/heads/master@{#18763}
> Committed: cb83bdf01fTBR=stefan@webrtc.org,danilchap@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org,holmer@google.com,eladalon@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2957763002
Cr-Commit-Position: refs/heads/master@{#18764}
1. Demux RTCP messages according to the sender-SSRC.
2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2943693003
Cr-Commit-Position: refs/heads/master@{#18763}
Reason for revert:
Test has been fixed.
Original issue's description:
> Disable RTCVideoRenderFrameCallbackNV12 test
>
> It has been crashing on trybots
>
> BUG=webrtc:7863
> TBR=magjed@webrtc.org
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2951273002
> Cr-Commit-Position: refs/heads/master@{#18720}
> Committed: 4d25a0554aTBR=kjellander@webrtc.org,oprypin@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7863
Review-Url: https://codereview.webrtc.org/2959673002
Cr-Commit-Position: refs/heads/master@{#18758}
For consistency with SafeMin(), SafeMax(), and SafeClamp(). And so that we avoid introducing a namespace.
BUG=webrtc:7459
Review-Url: https://codereview.webrtc.org/2802423002
Cr-Commit-Position: refs/heads/master@{#18756}
After returning from the call the AVAudioSession was configured to
use the receiver instead of the speaker for audio output. The
configuration was only restored if the sound loop was previously
playing, this change makes sure that the configuration is always
reset so the sound can be played audibly after a call has been
finished.
Bug: webrtc:7792
Change-Id: Idabf6fadc8041b18722cb8f5e89e0c8c36b1b74d
Reviewed-on: https://chromium-review.googlesource.com/544819
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18754}
This change preserves rotation through the decoder, rather than requiring
callers to keep track of rotation. The test now uses a non-zero rotation
to ensure it is preserved.
Commit 3814524 inadvertently reverted several changes that weren't merged
properly before submit. In particular, it clobbered some log messages,
comments, and error codes. This change fixes those mistakes.
BUG=webrtc:7760
Change-Id: If529ee59fc56de7937e362dc15591295e2cf9f79
Reviewed-on: https://chromium-review.googlesource.com/546415
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18752}
Update the name of the trybot as well.
BUG=webrtc:7853
NOTRY=True
TBR=ehmaldonado@webrtc.org
Change-Id: I296ee78b0a61b5a335e3bb794cbec72d85c15970
Reviewed-on: https://chromium-review.googlesource.com/547955
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18751}
This test was added in "Add cropping to VIEEncoder to match simulcast streams resolution (https://codereview.webrtc.org/2936393002) and makes
webrtc_perf_tests fail+timeout on Mac:
https://build.chromium.org/p/client.webrtc.perf/builders/Mac%2010.11/builds/2896
[ RUN ] FullStackTest.SimulcastFullHdOveruse
- Uh, I'm-I'm not quite dead, sir.
- Uh, I-I think uh, I could pull through, sir.
../../webrtc/video/video_quality_test.cc:419: Failure
Expected: (frames_processed) > (last_frames_processed), actual: 591 vs 591
Analyzer stalled while waiting for test to finish.
TBR=sprang@webrtc.org,magjed@webrtc.org,ilnik@webrtc.org
BUG=webrtc:7375, webrtc:6958
NOTRY=True
Review-Url: https://codereview.webrtc.org/2960573002
Cr-Commit-Position: refs/heads/master@{#18748}
