This is needed for the following coming tests: VideoSendStream, end-to-end,
full stack, and video_loopback.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2500943002
Cr-Commit-Position: refs/heads/master@{#15087}
The class VideoEncoderSoftwareFallbackWrapper is an implementation
detail of webrtc/media/engine/webrtcvideoengine2.cc and should not be
directly under webrtc/video_encoder.h. The main purpose is to improve
the dependency graph in WebRTC so that VideoEncoderSoftwareFallbackWrapper
can depend on cricket::VideoCodec.
The test for VideoEncoderSoftwareFallbackWrapper is also moved from
webrtc/video/video_encoder_unittest.cc to
webrtc/media/engine/videoencodersoftwarefallbackwrapper_unittest.cc.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2484863009
Cr-Commit-Position: refs/heads/master@{#15085}
Before this change, the configuration logic in FlexfecReceiveStream
tried to make unsupported configurations work, e.g., by dropping the
protection of some media streams when multiple media streams were
protected by a single FlexFEC stream. This CL makes the configuration logic
return more errors on such unsupported configurations.
This harmonizes the logic with the new configuration logic in
VideoSendStream, for the FlexfecSender.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2499963002
Cr-Commit-Position: refs/heads/master@{#15083}
FlexfecSender is owned and configured by VideoSendStream.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2501503003
Cr-Commit-Position: refs/heads/master@{#15082}
This is undefined behavior, according to specification.
BUG=chromium:661133
Review-Url: https://codereview.webrtc.org/2500953003
Cr-Commit-Position: refs/heads/master@{#15078}
Also extract all iOS sources into a static library configuration
so it's easier to include them in the test target as well.
Also, fix a wrong test that was undiscovered because the
tests were not running
BUG=webrtc:6707
Review-Url: https://codereview.webrtc.org/2502623002
Cr-Commit-Position: refs/heads/master@{#15076}
In DxgiOutputDuplicator, we need to convert between a monitor based coordinate
and a entire screen based coordinate. i.e. Copying an updated area from a
monitor (an output in DirectX API) to the entire screen frame (DesktopFrame).
But DxgiOutputDuplicator always assumes the coordinate is based on screen frame.
So we only need to convert a rectange in updated_region to monitor based
coordinate when copying data from texture_. But in last_frame_, the data are
always based on screen coordinate.
So fixes are both required in line 167 and line 180. In the previous one, we do
not need to convert the DesktopRect, which is already screen based, into screen
based coordinate. In the late one, we do not need to convert the DesktopRect at
all. So after these two changes, DxgiOutputDuplicator::TargetRect() function can
be removed.
Flickers of DirectX capturer can happen on any devices, but a virtual machine
can easily reproduce it. While on a regular high performance machine, it's
harder, but not totally impossible, to reproduce the issue.
BUG=314516
Review-Url: https://codereview.webrtc.org/2495143002
Cr-Commit-Position: refs/heads/master@{#15075}
- Functionality now implemented in AudioReceiveStream and Call.
- Added some missing function to MockChannelProxy.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2461523002
Cr-Commit-Position: refs/heads/master@{#15072}
Previously when BitrateControllerImpl::OnDelayBasedBweResult() is
called as result of a probe it was calling
bandwidth_estimation_.SetSendBitrate(), but not
UpdateDelayBasedEstimate(). As result SendSideBandwidthEstimation was
effectively ignoring probe results as it kept the old
delay_based_bitrate_bps_ value, which caps the resulting bitrate.
BUG=webrtc:6332,webrtc:6710
Review-Url: https://codereview.webrtc.org/2481383002
Cr-Commit-Position: refs/heads/master@{#15071}
This can happen in rare and strange cases.
Also taking the opportunity to replace all asserts with DCHECKs in
that file.
BUG=chromium:659225
Review-Url: https://codereview.webrtc.org/2499013002
Cr-Commit-Position: refs/heads/master@{#15070}
This adds two command-line flags to audioproc_f: -red and -red_graph, which can be used to enable/disable the RED, and to set the output path for the graph. The graph is generated as a python script that depends on matplotlib and numpy to display the graph.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2486763002
Cr-Commit-Position: refs/heads/master@{#15069}
Essentially a copy of https://codereview.chromium.org/2299953002/
plus changes to WebRTC's license, changed OWNERS and additional
MB updates up to Chromium revision http://crrev.com/f1e2718a3ff.
The PRESUBMIT.py check was updated to use the existing
webrtc/build/mb_config.pyl to avoid breaking bots (that have
this path hardcoded).
This replaces the previously symlinked MB, which already
runs validation of the WebRTC configs as part of
webrtc/build/PRESUBMIT.py.
BUG=chromium:664425
NOTRY=True
TESTED=Ran:
tools/mb/mb.py gen -m client.webrtc -b 'Mac64 Release' --config-file webrtc/build/mb_config.pyl --isolate-map-file=webrtc/build/gn_isolate_map.pyl --gyp-script=webrtc/build/gyp_webrtc.py //out/Release
Review-Url: https://codereview.webrtc.org/2306163002
Cr-Commit-Position: refs/heads/master@{#15068}
Prior to this change, FlexFEC packets that were paced would be lost in
the RTPSender, since they were not stored in a packet history. This CL
introduces such a packet history, as well as the needed wireup for
higher layers to be aware that the particular RTPSender is able to
send FlexFEC packets with a particular SSRC.
Updated RTPSender unit test to reflect the fact that paced packets
are now actually sent.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2491293002
Cr-Commit-Position: refs/heads/master@{#15066}
Updates GetStats() function in AudioSendStream to use the new GetStatistics function in APM instead of the corresponding VoE functions.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2463813002
Cr-Commit-Position: refs/heads/master@{#15065}
The only difference is that the F and R bits have changed place.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2495253002
Cr-Commit-Position: refs/heads/master@{#15064}
Right now, the H264 end-to-end tests are not run on the bots.
BUG=None
Review-Url: https://codereview.webrtc.org/2484913007
Cr-Commit-Position: refs/heads/master@{#15062}
After the removal of the RED/RTX workaround, we now need to explicitly
enable RTX for RED. Prior to the removal of the workaround, RED over RTX
was implicitly enabled whenever media over RTX was enabled.
BUG=webrtc:6650
Review-Url: https://codereview.webrtc.org/2493723002
Cr-Commit-Position: refs/heads/master@{#15061}
Payload type -> RtpCodecParameters maps added for sender and receiver
side. It contains information that will be needed for RTCCodecStats[1]
dictionaries.
Video[Sender/Receiver]Info is updated with current codec payload type
for every stream which can be used to look up the codec in
VideoMediaInfo.
A similar change should be made for VoiceMediaInfo and
Voice[Sender/Receiver]Info.
[1] https://w3c.github.io/webrtc-stats/#codec-dict*
BUG=chromium:659117
Review-Url: https://codereview.webrtc.org/2484193002
Cr-Commit-Position: refs/heads/master@{#15060}
DataChannel.SignalOpened and unittests added.
PeerConnection.SignalDataChannelCreated added and wired up to
RTCStatsCollector.OnDataChannelCreated on RTCStatsCollector
construction.
RTCStatsCollector.OnSignalOpened/Closed added and wired up on
OnDataChannelCreated.
rtcstatscollector_unittest.cc updated, faking that channels are opened
and closed.
I did not want to use DataChannelObserver because it is used for more
than state changes and there can only be one observer (unless code is
updated). Since DataChannel already had a SignalClosed it made sense to
add a SignalOpened.
Having OnSignalBlah in RTCStatsCollector is new in this CL but will
likely be needed to correctly handle RTPMediaStreamTracks being added
and detached independently of getStats. This CL establishes this
pattern.
(An integration test will be needed for this and all the other stats to
make sure everything is wired up correctly and test outside of a
mock/fake environment, but this is not news.)
BUG=chromium:636818, chromium:627816
Review-Url: https://codereview.webrtc.org/2472113002
Cr-Commit-Position: refs/heads/master@{#15059}
Use system Xcode on Mac instead of the hermetic toolchain which
is now the default in Chromium. WebRTC needs the 10.12 SDK to
compile successfully, which is not availble in the hermetic
toolchain Chromium is rolling out to Googlers.
Change log: 1a6cf4da7c..f1e2718a3f
Full diff: 1a6cf4da7c..f1e2718a3f
No dependencies changed.
No update to Clang.
TBR=ehmaldonado@webrtc.org
BUG=webrtc:6700
Review-Url: https://codereview.webrtc.org/2496113002
Cr-Commit-Position: refs/heads/master@{#15057}
This CL will start to distinguish H264 profiles during SDP negotiation.
We currently don't look at the H264 profile at all and assume they are
all Constrained Baseline Level 3.1. This CL will start to check profiles
for equality when matching, and will generate the correct answer H264
level.
Each local supported H264 profile needs to be listed explicitly in the
list of local supported codecs, even if they are redundant. For example,
Baseline profile should be listed explicitly even though another profile
that is a superset of Baseline is also listed. The reason for this is to
simplify the code and avoid profile intersection during matching. So
VideoCodec::Matches will check for profile equality, and not check if
one codec is a subset of the other. This also leads to the nice property
that VideoCodec::Matches is symmetric, i.e. iif a.Matches(b) then
b.Matches(a).
BUG=webrtc:6337
TBR=tkchin@webrtc.org
Review-Url: https://codereview.webrtc.org/2483173002
Cr-Commit-Position: refs/heads/master@{#15051}
Reason for revert:
Broke a lot of tests in chromium.webrtc browser_tests. See e.g. https://build.chromium.org/p/chromium.webrtc/builders/Mac%20Tester/builds/62228 and https://build.chromium.org/p/chromium.webrtc/builders/Win8%20Tester/builds/30102.
[ RUN ] WebRtcVideoQualityBrowserTests/WebRtcVideoQualityBrowserTest.MANUAL_TestVideoQualityH264/1
...
#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\modules\rtp_rtcp\source\rtp_format_h264.cc, line 170
# last system error: 0
# Check failed: packetization_mode_ == kH264PacketizationMode1 (0 vs. 2)
#
Original issue's description:
> Implement H.264 packetization mode 0.
>
> This approach extends the H.264 specific information with
> a packetization mode enum.
>
> Status: Parameter is in code. No way to set it yet.
>
> Rebase of CL 2009213002
>
> BUG=600254
>
> Committed: https://crrev.com/3bba101f36483b8030a693dfbc93af736d1dba68
> Cr-Commit-Position: refs/heads/master@{#15032}
TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=600254
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2500743002
Cr-Commit-Position: refs/heads/master@{#15050}
This is a trivial change to remove MockScreenCapturerCallback, and use
MockDesktopCapturerCallback to replace it.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2494013003
Cr-Commit-Position: refs/heads/master@{#15047}
A tiny but critical change to avoid a crash failure in DirectX capturer.
A good news is this failure is caught by ScreenCapturer integration tests.
BUG=314516
Review-Url: https://codereview.webrtc.org/2494893002
Cr-Commit-Position: refs/heads/master@{#15046}
The old CroppingWindowCapturer::Create() function returns a raw pointer, which
cannot explain the ownership.
So this change adds a CreateCapturer() function to return a unique_ptr.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2496993003
Cr-Commit-Position: refs/heads/master@{#15045}
We want to keep these warnings on. They were switched off temporarily
to unblock chromium roll.
BUG=webrtc:6520
Review-Url: https://codereview.webrtc.org/2493873005
Cr-Commit-Position: refs/heads/master@{#15043}
Reason for revert:
Relanding, now that downstream issues have been fixed.
Original issue's description:
> Revert of Declare VideoCodec.codec_specific_info private (patchset #5 id:80001 of https://codereview.webrtc.org/2452963002/ )
>
> Reason for revert:
> Broke a google3 build
>
> Original issue's description:
> > Declare VideoCodec.codec_specific_info private
> >
> > This completes the privatization of the codec specific
> > information in VideoCodec.
> >
> > BUG=webrtc:6603
> >
> > Committed: https://crrev.com/792738640234d81c916ac4458ac72286cb2548a4
> > Cr-Commit-Position: refs/heads/master@{#15013}
>
> TBR=tommi@chromium.org,magjed@chromium.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6603
>
> Committed: https://crrev.com/7fe6db91d99cf6d43874651bcca56092cf869e2f
> Cr-Commit-Position: refs/heads/master@{#15027}
TBR=tommi@chromium.org,magjed@chromium.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6603
Review-Url: https://codereview.webrtc.org/2494683006
Cr-Commit-Position: refs/heads/master@{#15041}
These functions currently copy cricket::Codec classes by value which is
expensive since they contain e.g. std::map<std::string, std::string>
containers with parameters. This CL avoids copying them altogether.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2493733003
Cr-Commit-Position: refs/heads/master@{#15040}
Reason for revert:
Relanding after known downstream breakages have been fixed.
Original issue's description:
> Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
>
> Reason for revert:
> Breaks chrome, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/19019/steps/compile/logs/stdio
>
> Analysis: Chrome uses cricket::VideoFrame, without explicitly including webrtc/media/base/videoframe.h, and breaks when that file is no longer included by any other webrtc headers. Will reland after updating Chrome.
>
> Original issue's description:
> > Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
> >
> > Replaced with webrtc::VideoFrame.
> >
> > TBR=mflodman@webrtc.org
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/45c8b8940042bd2574c39920804ade8343cefdba
> > Cr-Commit-Position: refs/heads/master@{#14885}
>
> TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/7341ab8e2505c9763d208e069bda269018357e7d
> Cr-Commit-Position: refs/heads/master@{#14886}
TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2487633002
Cr-Commit-Position: refs/heads/master@{#15039}
