The LockManagerOperation global function uses void** which does not work
well with the thread_annotations.h macros and produce compiler warnings
on clang (chromium default). Workarounds to this is hacky and unhelpful
so we disable the analysis for this function, which isn't helpful in
this case anyway.
webrtc_h264_config is no longer needed and is removed.
BUG=8090, 8119
Review-Url: https://codereview.webrtc.org/3000263002
Cr-Commit-Position: refs/heads/master@{#19425}
* Rename some members.
* Shorten visualization file names.
* Make some member functions static, in preparation for moving them
to be helper functions in an anonymous namespace.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3001193002
Cr-Commit-Position: refs/heads/master@{#19424}
* Make ProcessFrame return void.
* Make |encode_callback_| and |decode_callback_| direct members.
* Remove ::EncodedFrameSize() and ::EncodedFrameType()
* Remove unused |timestamp| member from FrameInfo.
* Reorder printf output from PrintCodecSettings.
* Make some member functions const.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2998063002
Cr-Commit-Position: refs/heads/master@{#19421}
Clang is enabled by default and we currently lack coverage for building on Windows
with MSVC (see crbug.com/757293). This should unblock rolling WebRTC into
Chromium DEPS. We need to improve our trybot coverage for standalone WebRTC
to prevent things like this in the future though (crbug.com/756840).
BUG=webrtc:8119,chromium:756840,chromium:757293
TBR=hbos@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/3003473002
Cr-Commit-Position: refs/heads/master@{#19420}
Reason for revert:
Breaks webrtc.linux
Original issue's description:
> Add Jpeg frame writer for test support.
>
> 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/2990563002
> Cr-Commit-Position: refs/heads/master@{#19414}
> Committed: 26e5cbd6bbTBR=stefan@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8030
Review-Url: https://codereview.webrtc.org/2998133002
Cr-Commit-Position: refs/heads/master@{#19419}
1) Entering PROBE_RTT when necessary.
2) Congestion window gain of 0.65 instead of constant 4 packets.
3) {1.1, 0.9} pair instead of {1.25, 0.75}
4) Recovery mode.
5) No reaction to losses due to Recovery mode's implementation.
6) Supports encoder.
7) A new test compiling most of the simulation tests.
8) Bucket for high gain phase, disabled by default.
9) Pacer specific to BBR.
BUG=webrtc:7713
Review-Url: https://codereview.webrtc.org/2999073002
Cr-Commit-Position: refs/heads/master@{#19418}
Converts RtcpDemuxerTest to use a test fixture which creates the
RtcpDemuxer under test and wraps sink adding functions to
automatically remove added sinks at the end of the test case.
Also adds domain-specific expectation macros to improve readability
of test cases.
Bug: None
Change-Id: I2416ce67c4fc8edc9ecc2cc98d191562c3bf81e4
Reviewed-on: https://chromium-review.googlesource.com/601049
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19417}
Extends the RtpDemuxer to do demuxing according to the BUNDLE spec,
using MID and payload types in addition to RSID and SSRC. Also extends
SsrcBindingObserver to receive notification for all types of SSRC
binding that can occur with the new algorithm.
Bug: webrtc:4050
Change-Id: Ie2f347f90d5074ab537fa1162fa7314dd292b68b
Reviewed-on: https://chromium-review.googlesource.com/578628
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19415}
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/2990563002
Cr-Commit-Position: refs/heads/master@{#19414}
Instead explicitly ignore only the flags we know should be ignored.
BUG=webrtc:7568
Review-Url: https://codereview.webrtc.org/2968003003
Cr-Commit-Position: refs/heads/master@{#19412}
parameters are interesting.
Since this is used by users when implementing injectable Obj-C video
codecs, the API where the unused payload is required is unnecessarily
confusing. Also make the codec parameters optional.
BUG=None
Review-Url: https://codereview.webrtc.org/3001183002
Cr-Commit-Position: refs/heads/master@{#19411}
Reason for revert:
iOS workaround.
Original issue's description:
> Revert of quest keyframes more frequently on stream start/decoding error. (patchset #2 id:170001 of https://codereview.webrtc.org/2996823002/ )
>
> Reason for revert:
> Causes iOS H264 calls received in the background to have increased delay before being able to decode stream from sender due to not having a keyframe.
>
> Original issue's description:
> > Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
> >
> > Reason for revert:
> > Create fix CL.
> >
> > Original issue's description:
> > > Revert of Request keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.webrtc.org/2993793002/ )
> > >
> > > Reason for revert:
> > > Broke downstream test that was waiting for 5 keyframes to be received within 10 seconds. Maybe the issue is that "stats_callback_->OnCompleteFrame(frame->num_references == 0, ..." was changed to "frame->is_keyframe()"?
> > >
> > > Original issue's description:
> > > > Request keyframes more frequently on stream start/decoding error.
> > > >
> > > > In this CL:
> > > > - Added FrameObject::is_keyframe() convinience function.
> > > > - Moved logic to request keyframes on decoding error from VideoReceived to
> > > > VideoReceiveStream.
> > > > - Added keyframe_required as a parameter to FrameBuffer::NextFrame.
> > > >
> > > > BUG=webrtc:8074
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2993793002
> > > > Cr-Commit-Position: refs/heads/master@{#19280}
> > > > Committed: 26b4804358
> > >
> > > TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,philipel@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:8074
> > >
> > > Review-Url: https://codereview.webrtc.org/2994043002
> > > Cr-Commit-Position: refs/heads/master@{#19295}
> > > Committed: 77a983185f
> >
> > TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > BUG=webrtc:8074
> >
> > Review-Url: https://codereview.webrtc.org/2996823002
> > Cr-Commit-Position: refs/heads/master@{#19324}
> > Committed: 628ac5964e
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org,philipel@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8074
>
> Review-Url: https://codereview.webrtc.org/2995153002
> Cr-Commit-Position: refs/heads/master@{#19392}
> Committed: 53959fcc2bTBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org,tkchin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
BUG=webrtc:8074
Review-Url: https://codereview.webrtc.org/2996153003
Cr-Commit-Position: refs/heads/master@{#19410}
In preparation of running DirectTransport on a TaskQueue in unit-tests, change the thread-checkers to sequence-checkers. This is necessary for Mac and iOS, where the TaskQueue is guaranteed to run sequentially, but not guaranteed to do so on only one thread.
TODO: Add the relevant BUGs.
BUG=None
Review-Url: https://codereview.webrtc.org/2997853002
Cr-Commit-Position: refs/heads/master@{#19408}
Reason for revert:
Create reland CL to add fix to.
Original issue's description:
> Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
>
> Reason for revert:
> Speculative revet for breaking remoting_unittests in fyi bots.
> https://build.chromium.org/p/chromium.webrtc.fyi/waterfall?builder=Win7%20Tester
>
> Original issue's description:
> > Add a flags field to video timing extension.
> >
> > The rtp header extension for video timing shuold have an additional
> > field for signaling metadata, such as what triggered the extension for
> > this particular frame. This will allow separating frames select because
> > of outlier sizes from regular frames, for more accurate stats.
> >
> > This implementation is backwards compatible in that it can read video
> > timing extensions without the new flag field, but it always sends with
> > it included.
> >
> > BUG=webrtc:7594
> >
> > Review-Url: https://codereview.webrtc.org/3000753002
> > Cr-Commit-Position: refs/heads/master@{#19353}
> > Committed: cf5d485e14
>
> TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7594
>
> Review-Url: https://codereview.webrtc.org/2995953002
> Cr-Commit-Position: refs/heads/master@{#19360}
> Committed: f0f7378b05TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,emircan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2996153002
Cr-Commit-Position: refs/heads/master@{#19405}
In preparation of running DirectTransport on a TaskQueue in unit-tests, change the thread-checkers to sequence-checkers. This is necessary for Mac and iOS, where the TaskQueue is guaranteed to run sequentially, but not guaranteed to do so on only one thread.
TODO: Add the relevant BUGs.
BUG=None
Review-Url: https://codereview.webrtc.org/2999973002
Cr-Commit-Position: refs/heads/master@{#19404}
No longer active with WebRTC, last commit 2014-10-10
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2999183002
Cr-Commit-Position: refs/heads/master@{#19403}
This triggered ASan, saying "object passed to delete has wrong type".
This error is caused by allocating a `struct sockaddr_in6`, casting it
and deleting it as `struct sockaddr*` which has a smaller size.
BUG=None
Review-Url: https://codereview.webrtc.org/2999053002
Cr-Commit-Position: refs/heads/master@{#19401}
Converts RtpDemuxerTest to use a test fixture which creates the
RtpDemuxer under test and wraps sink adding/observer adding
functions to automatically remove them at the end of the test case.
Bug: None
Change-Id: I7e40223f6837caa5443d9850477198c1f7a8d14a
Reviewed-on: https://chromium-review.googlesource.com/608906
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19400}
We are already asserting in tryDeliverTextureFrame() that we are on the
SurfaceTextureHelper handler thread, but this behavior isn't guaranteed
by Android.
Also updated docs for OnTextureFrameAvailableListener, since other
classes (e.g. capturers) were also asserting that they were called on
the SurfaceTextureHelper thread.
Bug: webrtc:8087, b/64609528
Change-Id: I47148c452f66f24477e438b06ef124a96965853d
Reviewed-on: https://chromium-review.googlesource.com/610495
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19398}
This change extends the TurnPort tests to cover connections to
TURN servers over TLS.
As part of this, the TestTurnServer is extended to support
connections from clients over TLS.
Note that this also fixes the remaining bugs in webrtc:7562
Bug: webrtc:7584
Change-Id: If89ceae49d33417625464b5892d20eee4de7c3b5
Reviewed-on: https://chromium-review.googlesource.com/611520
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19397}
This rename prepares SsrcBindingObserver to be used for observing all
the ways a sink can be bound to SSRCs (e.g., MID, payload types).
Bug: webrtc:4050
Change-Id: I16b68481d01f921c326a33f5f2baf79d8b3f12e2
Reviewed-on: https://chromium-review.googlesource.com/590762
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19396}
This commit makes the following changes:
1. Splits TestReconstructedServerUrl into 3 tests that individually
check the reconstructed URL for UDP IPv4, UDP IPv6, and TCP.
2. Factors out common code between protocols for release allocation and
reconstructed URL tests.
3. Provides functions for getting the expected RTT of various operations
based on the protocol used. TurnPort tests use a fake clock in part
to check tight bounds on the number of network round trips it takes
to complete operations like getting TURN candidates and trying
alternate servers. These RTTs are highly dependent on the
characteristics of the transport-layer protocol used, so these
details have been moved to dedicated functions with comments
explaining how the numbers are calculated.
Bug: webrtc:7584
Change-Id: I3b065e25446cb5ecd955f359625a35fb0ad46777
Reviewed-on: https://chromium-review.googlesource.com/611500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19395}
This layer takes in a simplified "options" struct and the current local description,
and generates a new offer/answer. Previously the options struct assumed there would
only be one media description per media type (audio/video), but it now supports
N number of audio/video descriptions.
The |add_legacy_stream| options is removed from the mediasession.cc/.h
in this CL.
The next step is to add the ability for PeerConnection/WebRtcSession to create
"options" to represent multiple RtpTransceivers, and apply the Unified Plan
descriptions correctly. Right now, only Plan B descriptions will be
generated in unit tests.
BUG=chromium:465349
Review-Url: https://codereview.webrtc.org/2991693002
Cr-Original-Commit-Position: refs/heads/master@{#19343}
Committed: a77e6bbd30
Review-Url: https://codereview.webrtc.org/2991693002
Cr-Commit-Position: refs/heads/master@{#19394}
This change renames ScreenDrawerLockLinux into ScreenDrawerLockPosix and shares
it with Mac OSX.
Bug: webrtc:7950
Change-Id: Ib141781d2c35bfda0d6f9458fff235adbb643280
Reviewed-on: https://chromium-review.googlesource.com/607688
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19393}
Reason for revert:
Causes iOS H264 calls received in the background to have increased delay before being able to decode stream from sender due to not having a keyframe.
Original issue's description:
> Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
>
> Reason for revert:
> Create fix CL.
>
> Original issue's description:
> > Revert of Request keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.webrtc.org/2993793002/ )
> >
> > Reason for revert:
> > Broke downstream test that was waiting for 5 keyframes to be received within 10 seconds. Maybe the issue is that "stats_callback_->OnCompleteFrame(frame->num_references == 0, ..." was changed to "frame->is_keyframe()"?
> >
> > Original issue's description:
> > > Request keyframes more frequently on stream start/decoding error.
> > >
> > > In this CL:
> > > - Added FrameObject::is_keyframe() convinience function.
> > > - Moved logic to request keyframes on decoding error from VideoReceived to
> > > VideoReceiveStream.
> > > - Added keyframe_required as a parameter to FrameBuffer::NextFrame.
> > >
> > > BUG=webrtc:8074
> > >
> > > Review-Url: https://codereview.webrtc.org/2993793002
> > > Cr-Commit-Position: refs/heads/master@{#19280}
> > > Committed: 26b4804358
> >
> > TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,philipel@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:8074
> >
> > Review-Url: https://codereview.webrtc.org/2994043002
> > Cr-Commit-Position: refs/heads/master@{#19295}
> > Committed: 77a983185f
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> BUG=webrtc:8074
>
> Review-Url: https://codereview.webrtc.org/2996823002
> Cr-Commit-Position: refs/heads/master@{#19324}
> Committed: 628ac5964eTBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8074
Review-Url: https://codereview.webrtc.org/2995153002
Cr-Commit-Position: refs/heads/master@{#19392}
Make it possible for forced VP8 SW fallback encoder to set min_pixels_per_frame via GetScalingSettings().
Add a min required resolution (in addition to bitrate) before releasing forced SW fallback.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3000693003
Cr-Commit-Position: refs/heads/master@{#19390}
This test is the only remaining one that does not use gtest and that's
blocking some infra cleanup tasks. Ideally this test would use
webrtc/rtc_base/flags.h but that's a lot of unnecessary work.
This also replaces some exit() status codes - the logic behind this is
if you get incorrectly specified command line arguments, exit(1) is
invoked for a failure, because it's not a test failure, and if flag
parsing was done properly, it would not be a gtest failure anyway.
BUG=webrtc:7568
Review-Url: https://codereview.webrtc.org/3000033002
Cr-Commit-Position: refs/heads/master@{#19388}
Reason for revert:
BUG=webrtc:8108: breaks Clang build.
Original issue's description:
> Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
>
> This layer takes in a simplified "options" struct and the current local description,
> and generates a new offer/answer. Previously the options struct assumed there would
> only be one media description per media type (audio/video), but it now supports
> N number of audio/video descriptions.
>
> The |add_legacy_stream| options is removed from the mediasession.cc/.h
> in this CL.
>
> The next step is to add the ability for PeerConnection/WebRtcSession to create
> "options" to represent multiple RtpTransceivers, and apply the Unified Plan
> descriptions correctly. Right now, only Plan B descriptions will be
> generated in unit tests.
>
> BUG=chromium:465349
>
> Review-Url: https://codereview.webrtc.org/2991693002
> Cr-Commit-Position: refs/heads/master@{#19343}
> Committed: a77e6bbd30TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:465349
Review-Url: https://codereview.webrtc.org/3001083002
Cr-Commit-Position: refs/heads/master@{#19384}
As it is now, the first time a TRACE_EVENT... is called, the result from
the current handler is stored in a static const variable, and subsequent
calls will use that value regardless of changes to the handler.
This is a problem if a test wants to use another handler.
BUG=None
Review-Url: https://codereview.webrtc.org/3002663002
Cr-Commit-Position: refs/heads/master@{#19382}
I want to publish an API for iSAC in webrtc/api/, and I want to use
the class names Audio{De,En}coderIsac{Fix,Float}.
BUG=webrtc:7835, webrtc:7841
Review-Url: https://codereview.webrtc.org/2996593002
Cr-Commit-Position: refs/heads/master@{#19381}
RTCAudioSession and RTCAudioSessionConfiguration allow users to handle
audio manually and is used by the AppRTCMobile example.
RTCVideoFrameBuffer exposes a protocol that users can implement to
create their own frame buffer formats, as long as they can be converted
into i420.
RTCVideoCapturer and RTCVideoViewShading are imported by other headers
already included by the umbrella header, so they were always accessible
to users. Added them to the umbrella header to make it explicit.
BUG=webrtc:7351, webrtc:8027
Review-Url: https://codereview.webrtc.org/2994253002
Cr-Commit-Position: refs/heads/master@{#19379}
Reason for revert:
Reland
Original issue's description:
> Revert of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2998753002/ )
>
> Reason for revert:
> Speculative revert to see if this caused regressions in android perf tests.
>
> Original issue's description:
> > Make the acceptable queue in the cwnd experiment configurable.
> >
> > BUG=webrtc:7926
> >
> > Review-Url: https://codereview.webrtc.org/2998753002
> > Cr-Commit-Position: refs/heads/master@{#19320}
> > Committed: 7c83c56b6d
>
> TBR=philipel@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7926
>
> Review-Url: https://codereview.webrtc.org/2999893002
> Cr-Commit-Position: refs/heads/master@{#19337}
> Committed: c5d9e63c2bTBR=philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/2999083002
Cr-Commit-Position: refs/heads/master@{#19377}
WindowUnderPoint have different signatures on different platforms, which should
be abstract as an interface.
So this change adds a WindowFinder interface to replace WindowUnderPoint free
function. Meanwhile, this change also includes the implementation of
WindowFinderX11 for X11.
Bug: webrtc:7950
Change-Id: I897a50d4033e713b339b6b6f48b5dbbe601e8db0
Reviewed-on: https://chromium-review.googlesource.com/611745
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19375}
