Add GUARDED_BY annotation to DirectTransport to demonstrate that next_scheduled_task_ is properly protected.
SendRtp(), SendRtcp(), etc., remain as unprotected as they were before the migration from using a thread to using a task-queue.
BUG=webrtc:8113
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/3002163003
Cr-Commit-Position: refs/heads/master@{#19487}
We can't handle no value here anyway and end up setting a default
at each call site. The defaults aren't even the same in each place.
BUG=None
Review-Url: https://codereview.webrtc.org/2998293002
Cr-Commit-Position: refs/heads/master@{#19485}
Testcode updated to use webrtc::test::TempFilename.
Also deletes now unused functions AppleDataDirectory,
AppleTempDirectory and AppleAppName.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2995413002
Cr-Commit-Position: refs/heads/master@{#19483}
also C++ code (see https://bugs.chromium.org/p/webrtc/issues/detail?id=7743
for more information).
We have moved .mm files out of test_support and fileutils (into test_support_objc
and fileutils_objc).
To achieve the goal for run_test and test_renderer (in the next part of the phrase
X is run_test or test_renderer) we have created 2 targets (X_objc and X_generic)
and X will act as a proxy between these targets (this way we can avoid a circular
dependency between X_generic and X_objc).
BUG=webrtc:7743
Review-Url: https://codereview.webrtc.org/2991323003
Cr-Commit-Position: refs/heads/master@{#19479}
Renamed the dtlstransportchannel.h/.cc to dtlstransport.h/.cc which are
the right names for those files because the concept of DtlsTransportChannel
no longer exists.
BUG=None
Review-Url: https://codereview.webrtc.org/3004503002
Cr-Commit-Position: refs/heads/master@{#19478}
Reason for revert:
Fix and reland.
Original issue's description:
> Revert of Modify profiles for H264 encode SW fallback (patchset #2 id:20001 of https://codereview.webrtc.org/2997913003/ )
>
> Reason for revert:
> Breaks the internal bots.
> Root cause: The "public_deps" is defined behind an "if" condition which may not be true.
>
> Original issue's description:
> > Modify profiles for H264 encode SW fallback
> >
> > We have only Constrained Baseline profile available in SW encoder impl
> > so modify the profile to that in case of a fallback
> >
> > BUG=chromium:735959
> >
> > Review-Url: https://codereview.webrtc.org/2997913003
> > Cr-Commit-Position: refs/heads/master@{#19436}
> > Committed: 1fd66656b3
>
> TBR=magjed@webrtc.org,emircan@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:735959
>
> Review-Url: https://codereview.webrtc.org/2995373002
> Cr-Commit-Position: refs/heads/master@{#19438}
> Committed: 296b64eb25TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:735959
Review-Url: https://codereview.webrtc.org/2997423002
Cr-Commit-Position: refs/heads/master@{#19476}
We are still using cropping window capturer even the window is out of the screen.
See the bug for details.
Bug: webrtc:8134
Change-Id: I5161b1a17a3a1f8244697eea5eb78975be6908f9
Reviewed-on: https://chromium-review.googlesource.com/627338
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19474}
This test doesn't actually send/receive any audio; it's only testing
the interface layer. But the fact that it was creating/destroying real
audio devices repeatedly caused problems when tests were run in
parallel. So switching to a fake audio device solves this.
BUG=webrtc:7806
Review-Url: https://codereview.webrtc.org/2997383002
Cr-Commit-Position: refs/heads/master@{#19472}
Patch set 1:
Run a script to replace occurrences of WEBRTC_TRACE logging with the new style,
in webrtc/modules/media_file/.
Patch set 2:
- Manually fix log lines not handled by the script
- Update the included headers
- Remove the now unused object ID variables
Bug: webrtc:5118
Change-Id: I1acbaec3fbbdf1deb7b934624a2f1fd38253c7e9
Reviewed-on: https://chromium-review.googlesource.com/602007
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19470}
Bug: webrtc:8105
Change-Id: I751b89194f3fdb10ea41c6f9e48e38edefcbef1a
Reviewed-on: https://chromium-review.googlesource.com/616724
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19469}
The field trials enables producing new VideoFrames in camera classes.
This field trial should be enabled if VideoSinks are used.
BUG=webrtc:7749, webrtc:7760
Review-Url: https://codereview.webrtc.org/2984633002
Cr-Commit-Position: refs/heads/master@{#19467}
Before this change we could crash in Debug when WebRTC audio was first
interrupted and then resumed again. The reason was that the new audio
stream stems from a new native I/O thread and that triggered thread
checkers. With this change, failing thread checkers are detached when
audio is interrupted to ensure that they don't fail when audio is restarted.
NOTRY=TRUE
Bug: webrtc:8126
Change-Id: Ib36ff6bc942477730aba60066f049ed0c43d3901
Reviewed-on: https://chromium-review.googlesource.com/628716
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19465}
Maximum of interframe delay is calculated over moving window in
ReceiveStatistics proxy now and reported via GetStats. Name of a metric
is also changed.
BUG=none
Review-Url: https://codereview.webrtc.org/2995143002
Cr-Commit-Position: refs/heads/master@{#19463}
no-unused-lambda-capture was suppressed, but it's been decided as desireable to stop suppressing it. This CL fixes places in the code that trigger it.
1. Some unnecessary captures removed.
2. s/constexpr/const when capturing a float by value - this is good enough to stop the error.
3. Complete removal of the constexpr/const-modifier for int-types as a workaround.
BUG=webrtc:7133
Review-Url: https://codereview.webrtc.org/3005433002
Cr-Commit-Position: refs/heads/master@{#19462}
Packets recovered by ULPFEC enter through
RtpVideoStreamReceiver::OnRecoveredPacket, which does RTP
header extension parsing. Packets recovered by FlexFEC, however,
enter through Call::OnRecoveredPacket, which prior to this
CL did not do RTP header extension parsing.
The lack of RTP header extension parsing for FlexFEC packets is a
regression since https://codereview.webrtc.org/2886993005/.
TESTED=Using Android app with FlexFEC field trial enabled.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/3002023002
Cr-Commit-Position: refs/heads/master@{#19460}
On Windows a window may be covered by its own child window. So this change also
detects child windows by using EnumChildWindow().
The tooltip or context menu of the child window still cannot be detected after
this change. See bug for details.
Bug: webrtc:8062
Change-Id: I8455a9206d6a1d9da61013ac9debba4d3edae7d8
Reviewed-on: https://chromium-review.googlesource.com/619728
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19457}
This CL ensures that AEC3 recovers more quickly when capture data is
lost in such a manner that the echo path, as seen by AEC3, becomes
noncausal due to the AEC3 buffer misalignment caused by the data loss.
The CL adds the assumption of a minimum echo path delay of 5 blocks
and makes the hysteresis in the delay selection one-sided.
BUG=chromium:757796, webrtc:8131
Review-Url: https://codereview.webrtc.org/2998223002
Cr-Commit-Position: refs/heads/master@{#19454}
Also, use it to save worst psnr frame in video quality tests. It is indented that these saved frames from perfbots will be uploaded to the cloud and will be available in chrome perf dashboard. Because of that size of the saved frame is somewhat an issue. Also, y4m is not convenient to view.
BUG=webrtc:8030
Review-Url: https://codereview.webrtc.org/2998143002
Cr-Commit-Position: refs/heads/master@{#19450}
Moved the headers video_receive_stream.h and video_send_stream.h from
webrtc/ into webrtc/call/ as part of the Slim and Modular work.
The GN target webrtc:video_stream_api has moved to
webrtc/call:video_stream_api.
There are headers left in webrtc/ with the same name including the
moved headers in webrtc/call/ for not breaking external projects
depending on WebRTC.
At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets.
BUG=webrtc:8107
Review-Url: https://codereview.webrtc.org/3000253002
Cr-Commit-Position: refs/heads/master@{#19448}
The whole point of all the audio codec stuff we've recently published
in api/ is to function as lego bricks so that building stuff like our
builtin audio codec factories will be easy.
BUG=webrtc:7821, webrtc:7822
Review-Url: https://codereview.webrtc.org/2997713002
Cr-Commit-Position: refs/heads/master@{#19446}
DirectTransport has so far used its own thread, which led to a different threading-model for in the unit-tests than is used in actual WebRTC. Because of that, some critical-sections that weren't truly necessary in WebRTC could not be replaced with thread-checks, because those checks failed in unit-tests.
This CL introduces SingleThreadedTaskQueue - a TaskQueue which guarantees to run all of its tasks on the same thread (rtc::TaskQueue doesn't guarantee that on Mac) - and uses that for DirectTransport. CLs based on top of this will uncomment thread-checks which had to be commented out before, and remove unnecessary critical-sections.
Future work would probably replace the thread-checkers by more sophisticated serialized-access checks, allowing us to move from the SingleThreadedTaskQueue to a normal TaskQueue.
Related implementation notes:
* This CL has made DirectTransport::StopSending() superfluous, and so it was deleted.
BUG=webrtc:8113, webrtc:7405, webrtc:8056, webrtc:8116
Review-Url: https://codereview.webrtc.org/2998923002
Cr-Commit-Position: refs/heads/master@{#19445}
This test is and should be independent of RTP, so we don't need the
information provided in this struct.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2995403002
Cr-Commit-Position: refs/heads/master@{#19443}
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: f1e08d0b58TBR=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}
This is done with a script found in issue webrtc:5118.
The related header include is also removed.
Bug: webrtc:5118
Change-Id: I9d5cd7b8d0cbaae292632dd945e23709f0e7f7d7
Reviewed-on: https://chromium-review.googlesource.com/602135
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19441}
VideoSink is not assumed to take ownership of the frame, so it is not
necessary to release it unless it is retained for future use.
BUG=webrtc:7749
Review-Url: https://codereview.webrtc.org/3003563002
Cr-Commit-Position: refs/heads/master@{#19440}
Reason for revert:
Breaks the internal bots.
Root cause: The "public_deps" is defined behind an "if" condition which may not be true.
Original issue's description:
> Modify profiles for H264 encode SW fallback
>
> We have only Constrained Baseline profile available in SW encoder impl
> so modify the profile to that in case of a fallback
>
> BUG=chromium:735959
>
> Review-Url: https://codereview.webrtc.org/2997913003
> Cr-Commit-Position: refs/heads/master@{#19436}
> Committed: 1fd66656b3TBR=magjed@webrtc.org,emircan@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:735959
Review-Url: https://codereview.webrtc.org/2995373002
Cr-Commit-Position: refs/heads/master@{#19438}
