Reason for revert:
sigh. Have to revert again as there seems to have have been some change made for pnacl and CrOS.
Original issue's description:
> Reland of New task queueing primitive for async tasks: TaskQueue. (patchset #1 id:1 of https://codereview.webrtc.org/1935483002/ )
>
> New task queueing primitive for async tasks: TaskQueue.
> TaskQueue is a new way to asynchronously execute tasks sequentially
> in a thread safe manner with minimal locking. The implementation
> uses OS supported APIs to do this that are compatible with async IO
> notifications from things like sockets and files.
>
> This class is a part of rtc_base_approved, so can be used by both
> the webrtc and libjingle parts of the WebRTC library. Moving forward,
> we can replace rtc::Thread and webrtc::ProcessThread with this implementation.
>
> NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
> run on the same thread. E.g. on Mac and iOS, we use GCD dispatch queues
> which means that tasks might execute on different threads depending on
> what's the most efficient thing to do.
>
> TBR=perkj@webrtc.org
>
> Committed: https://crrev.com/65d1f2aba216d077c6d22488f03e56984aef1c68
> Cr-Commit-Position: refs/heads/master@{#12737}
TBR=perkj@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/1981573002
Cr-Commit-Position: refs/heads/master@{#12738}
New task queueing primitive for async tasks: TaskQueue.
TaskQueue is a new way to asynchronously execute tasks sequentially
in a thread safe manner with minimal locking. The implementation
uses OS supported APIs to do this that are compatible with async IO
notifications from things like sockets and files.
This class is a part of rtc_base_approved, so can be used by both
the webrtc and libjingle parts of the WebRTC library. Moving forward,
we can replace rtc::Thread and webrtc::ProcessThread with this implementation.
NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
run on the same thread. E.g. on Mac and iOS, we use GCD dispatch queues
which means that tasks might execute on different threads depending on
what's the most efficient thing to do.
TBR=perkj@webrtc.org
Review-Url: https://codereview.webrtc.org/1927133004
Cr-Commit-Position: refs/heads/master@{#12737}
Writable connections are pinged at a slower rate.
The function IsPingable will filter out the writable connections.
The interval for slower ping rate by default is WRITABLE_CONNECTION_PING_INTERVAL(2500ms) and can be set with the configuration.
BUG=webrtc:1161
Review-Url: https://codereview.webrtc.org/1944003002
Cr-Commit-Position: refs/heads/master@{#12736}
If we call GetStats in PeerConnection before receiving the remote answer, we will get some variables in the StatsReports which are initially set to be -1.
Several conditions are added when extracting the info for the report in StatsCollector.
Those variables include:
gooRtt,
dataChannelId,
googEchoCancellationEchoDelayMedian,
googEchoCancellationEchoQualityMin,
googEchoCancellationEchoDelayStdDev,
googJitterReceived,
audioInputLevel,
googCaptureStartNtpTimeMs
packetsLost.
BUG=webrtc:3377
Review-Url: https://codereview.webrtc.org/1875873002
Cr-Commit-Position: refs/heads/master@{#12735}
If OnOutputFormatRequest() is called, VideoAdapter will crop to the same
aspect ratio as the requested format. The output from
VideoAdapter.AdaptFrameResolution() now contains both how to crop the
input frame, and how to scale the cropped frame to the final adapted
resolution.
BUG=b/28622232
Review-Url: https://codereview.webrtc.org/1966273002
Cr-Commit-Position: refs/heads/master@{#12732}
Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.
BUG=webrtc:4256
NOTRY=True
TBR=perkj@webrtc.org
Review-Url: https://codereview.webrtc.org/1973313002
Cr-Commit-Position: refs/heads/master@{#12731}
The caller can set a negative or zero file size to avoid using a limit.
BUG=
Review-Url: https://codereview.webrtc.org/1974453002
Cr-Commit-Position: refs/heads/master@{#12730}
Depends on this CL in order to work in Chromium:
https://codereview.chromium.org/1976673002/
It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).
This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.
R=pthatcher@webrtc.org
Committed: 48e9d05f51
Review URL: https://codereview.webrtc.org/1956453003 .
Cr-Commit-Position: refs/heads/master@{#12729}
Reason for revert:
Breaks downstream import.
Original issue's description:
> Android: Make base interface for camera1 and camera2
>
> This CL adds a new interface CameraVideoCapturer that extends VideoCapturer with a switchCamera() function. It also moves moves CameraEventsHandler, CameraStatistics, and CameraSwitchHandler from VideoCapturerAndroid to this new interface. The purpose is to prepare for a camera2 implementation that will use the same interfaces and helper class.
>
> BUG=webrtc:5519
>
> Committed: https://crrev.com/6bdacaddfb18edef1f0cdd778209f6b05a8f9210
> Cr-Commit-Position: refs/heads/master@{#12723}
TBR=perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5519
Review-Url: https://codereview.webrtc.org/1979583002
Cr-Commit-Position: refs/heads/master@{#12727}
Reason for revert:
Breaks GN in Chromium.
Original issue's description:
> GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
>
> Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
> preparation for removing src/third_party/libjingle in Chromium.
>
> BUG=webrtc:4256
> NOTRY=True
>
> Committed: https://crrev.com/4d02a358b4205bd0f7b5f794b6fb8c157e075b9e
> Cr-Commit-Position: refs/heads/master@{#12724}
TBR=perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256
Review-Url: https://codereview.webrtc.org/1977853002
Cr-Commit-Position: refs/heads/master@{#12726}
Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.
BUG=webrtc:4256
NOTRY=True
Review-Url: https://codereview.webrtc.org/1929633002
Cr-Commit-Position: refs/heads/master@{#12724}
This CL adds a new interface CameraVideoCapturer that extends VideoCapturer with a switchCamera() function. It also moves moves CameraEventsHandler, CameraStatistics, and CameraSwitchHandler from VideoCapturerAndroid to this new interface. The purpose is to prepare for a camera2 implementation that will use the same interfaces and helper class.
BUG=webrtc:5519
Review-Url: https://codereview.webrtc.org/1895483002
Cr-Commit-Position: refs/heads/master@{#12723}
Reason for revert:
I plan to reland this change in a week or two, after downstream users are updated.
Original issue's description:
> Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
>
> Reason for revert:
> Breaks chrome FYI bots.
>
> Original issue's description:
> > Delete webrtc::VideoFrame methods buffer and stride.
> >
> > To make the HasOneRef/IsMutable hack work, also had to change the
> > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > to not imply an AddRef.
> >
> > BUG=webrtc:5682
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> Cr-Commit-Position: refs/heads/master@{#12558}
TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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/1963413004
Cr-Commit-Position: refs/heads/master@{#12721}
The new parameter indicates if the output in the AudioFrame is muted. If
so, the output samples are not written, but should be interpreted as all
zero.
A version of AudioCodingModule::PlayoutData10Ms() without the new
parameter is maintained while waiting for downstream dependencies to
conform.
BUG=webrtc:5609
Review-Url: https://codereview.webrtc.org/1976913002
Cr-Commit-Position: refs/heads/master@{#12719}
OnReceivedPacket now return the number of times the packet has been nacked. Also some minor refactoring.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1972123002 .
Cr-Commit-Position: refs/heads/master@{#12717}
This CL contains a few minor changes to names, function signatures and
merges two structs into one.
BUG=5868
Review-Url: https://codereview.webrtc.org/1952923005
Cr-Commit-Position: refs/heads/master@{#12716}
Reason for revert:
Breaks Chrome FYI using H264.
Need to investigate.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4170
Original issue's description:
> Remove ViEEncoder::SetNetworkStatus
>
> This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
>
> BUG=webrtc:5687
> NOTRY=True
>
> Committed: https://crrev.com/50b5c3be844ef571a28b2681c549443a26735d72
> Cr-Commit-Position: refs/heads/master@{#12699}
TBR=stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1978783002
Cr-Commit-Position: refs/heads/master@{#12715}
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt.
Review-Url: https://codereview.webrtc.org/1768773002
Cr-Commit-Position: refs/heads/master@{#12714}
This CL will be followed with other CLs that break apart
the application of the comfort noise from the comfort
noise generation.
The changes in the CL are very close to bitexaxt. The
bitinexactness is caused by differences in numerical
behavior when bundling the spectral band power and the
noise scaling based on the NLP gain.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1958933002
Cr-Commit-Position: refs/heads/master@{#12713}
a separate method.
This CL will be followed by other CLs that simplify this method and break out the state specific to this computation
into a separate substate.
The changes are bitexact.
BUG=webrtc:5201, webrtc:5298
Review-Url: https://codereview.webrtc.org/1963493003
Cr-Commit-Position: refs/heads/master@{#12712}
This CL implements the muted output functionality in NetEq. Tests are
added. The feature is currently off by default, and AcmReceiver makes
sure that the muted state is not engaged.
BUG=webrtc:5608
Review-Url: https://codereview.webrtc.org/1965733002
Cr-Commit-Position: refs/heads/master@{#12711}
Reason for revert:
Breaks remoting_unittests. They defined their own operator== which conflicts with this one.
I'll remove the operator== in a roll CL. But until it's approved, I'm reverting this so the FYI bots will pass.
Original issue's description:
> Implement RTCConfiguration.iceCandidatePoolSize.
>
> It works by creating pooled PortAllocatorSessions which can be picked up
> by a P2PTransportChannel when needed (after a local description is set).
>
> This can optimize candidate gathering time when there is some time between
> creating a PeerConnection and setting a local description.
>
> R=pthatcher@webrtc.org
>
> Committed: 48e9d05f51TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/1972043004
Cr-Commit-Position: refs/heads/master@{#12709}
It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).
This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1956453003 .
Cr-Commit-Position: refs/heads/master@{#12708}
As part of the work enabling OpenH264 in
WebRTC it was discovered that some of its code triggers
an UBSan errors:
third_party/openh264/src/codec/common/inc/golomb_common.h:103:3: runtime error: shift exponent 32 is too large for 32-bit type 'uint32_t' (aka 'unsigned int')
third_party/ffmpeg/libavcodec/h264_cavlc.c:585:54: runtime error: index -1 out of bounds for type 'VLC [15]'
Suppress such errors since this source code is out of our control.
This CL also includes a new NetEq suppression.
BUG=webrtc:5889
TBR=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1975863002 .
Cr-Commit-Position: refs/heads/master@{#12706}
Reason for revert:
The devices are now back at the bots!
Original issue's description:
> CQ: Remove android_dbg trybot.
>
> Its machines are currently lacking Android devices, which
> hopefully will be restored shortly.
>
> BUG=chromium:611380
> TBR=phoglund@webrtc.org
>
> Committed: https://crrev.com/7173ddd59c86a3aaf1461ba66715f7418b800f1f
> Cr-Commit-Position: refs/heads/master@{#12701}
TBR=phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:611380
Review-Url: https://codereview.webrtc.org/1974853002
Cr-Commit-Position: refs/heads/master@{#12705}
Some tests were passing in a local description created from hard-coded
SDP strings, which won't work in the future (since some attributes such
as the fingerprint and ICE ufrag/pwd are non-modifiable). These tests
now do the typical approach of calling CreateOffer and modifying the
result if necessary.
Also added some non-const versions of the SessionDescription accessor
helper functions, since that makes it much easier to modify a
SessionDescription. Previous alternatives were re-implementing the
helper methods from scratch, or converting the description to SDP,
modifying it, and converting it back.
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1966333002 .
Cr-Commit-Position: refs/heads/master@{#12704}
Added conditional updating of the statistics and the delay estimate so that
updates are only done when the farend is non-stationary.
The reason for this is that all the values that go into the updating of the
statistics, and that in turn are also used to update the delay, are frozen
when the farend signal is non-stationary. Therefore, when the farend signal
is silent (stationary), the last estimates present before the silent (stationary)
period began are used to continue to update the statistics. This is a problem as
the updating is done in a manner that assumes that the estimates continue
to be updated.
This CL conditions the updating based on stationarity instead of silence
as both are treated in the same manner in the delay agnostic AEC.
This makes sense theoretically as the delay agnostic AEC operates on
analyzing power deviations (in bands) from a slowly updated average power and
therefore for a stationary signal will have no such deviations to base its analysis
on.
BUG=webrtc:5875, chromium:576624
NOTRY=True
Review-Url: https://codereview.webrtc.org/1967033002
Cr-Commit-Position: refs/heads/master@{#12700}
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
BUG=webrtc:5687
NOTRY=True
Review-Url: https://codereview.webrtc.org/1932683002
Cr-Commit-Position: refs/heads/master@{#12699}
Fix a bug where startCaptureOnCameraThread() is called while the camera is already successfully running. It may happen in the scenario when startCapture() is called, but startCaptureOnCameraThread() fails and posts a retry, then stopCapture() is called and removeCallbacksAndMessages() fails to remove the pending retry, and then startCapture() is called successfully.
BUG=b/28181364
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1967053002 .
Cr-Commit-Position: refs/heads/master@{#12697}
Without this, some toolchains may fail to build base/checks.cc
because errno is undefined.
NOTRY=true
Review-Url: https://codereview.webrtc.org/1971513002
Cr-Commit-Position: refs/heads/master@{#12696}
Fixes bug where QualityScaler would be stuck "way below" QVGA (due to
downscale_shift_) even though it would never scale below QVGA. Also
fixes issue where samples would be cleared when either staying at max
resolution or going below QVGA even though no action happened.
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1971693003 .
Cr-Commit-Position: refs/heads/master@{#12691}
BaseChannel do calls to transport_channel on network_thread,
while keep calls to media_engine on worker_thread.
It still works when network_thread == worker_thread.
BUG=webrtc:5645
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1903393004 .
Cr-Commit-Position: refs/heads/master@{#12690}
