This cl change so that VideoSendStream::Start adds the stream as a BitrateObserver and VideoSendStream::Stop removes the stream as observer.
That also means that start will trigger a VideoEncoder::SetRate call with the most recent bitrate estimate.
VideoSendStream::Stop will trigger a VideoEncoder::SetRate with bitrate = 0.
BUG=webrtc:5687 b/28636240
Review-Url: https://codereview.webrtc.org/2070343002
Cr-Commit-Position: refs/heads/master@{#13192}
This is a somewhat involved refactoring of this class. Here's an overview of the changes:
* FileWrapper can now be used as a regular class and instances allocated on the stack.
* The type now has support for move semantics and copy isn't allowed.
* New public ctor with FILE* that can be used instead of OpenFromFileHandle.
* New static Open() method. The intent of this is to allow opening a file and getting back a FileWrapper instance. Using this method instead of Create(), will allow us in the future to make the FILE* member pointer, to be const and simplify threading (get rid of the lock).
* Rename the Open() method to is_open() and make it inline.
* The FileWrapper interface is no longer a pure virtual interface. There's only one implementation so there's no need to go through a vtable for everything.
* Functionality offered by the class, is now reduced. No support for looping (not clear if that was actually useful to users of that flag), no need to implement the 'read_only_' functionality in the class, since file APIs implement that already, no support for *not* managing the file handle (this wasn't used). OpenFromFileHandle always "manages" the file.
* Delete the unused WriteText() method and don't support opening files in text mode. Text mode is only different on Windows and on Windows it translates \n to \r\n, which means that files such as log files, could have a slightly different format on Windows than other platforms. Besides, tools on Windows can handle UNIX line endings.
* Remove FileName(), change Trace code to manage its own path.
* Rename id_ member variable to file_.
* Removed the open_ member variable since the same functionality can be gotten from just checking the file pointer.
* Don't call CloseFile inside of Write. Write shouldn't be changing the state of the class beyond just attempting to write.
* Remove concept of looping from FileWrapper and never close inside of Read()
* Changed stream base classes to inherit from a common base class instead of both defining the Rewind method. Ultimately, Id' like to remove these interfaces and just have FileWrapper.
* Remove read_only param from OpenFromFileHandle
* Renamed size_in_bytes_ to position_, since it gets set to 0 when Rewind() is called (and the size actually does not change).
* Switch out rw lock for CriticalSection. The r/w lock was only used for reading when checking the open_ flag.
BUG=
Review-Url: https://codereview.webrtc.org/2054373002
Cr-Commit-Position: refs/heads/master@{#13155}
1. It moves calculation of the needed padding to VideoSendStream instead of ViEEncoder and only does it once per send Stream instead of every time the network estimate changes.
2. The maximum amount of padding sent was prior to this cl calculated and updated based on network estimate changes. However, it can only change based on encoder configuration changes and if send streams are added or removed. This cl change the VideoSendStream/VieEncoder to notify the BitrateAllocator of changes to the needed padding bitrate and for BitrateAllocator to notify Call of these changes.
3. Fixed an issue in the SendPacer where it could send a padding packet before sending a real packet. This caused the test EndToEndTest.RestartingSendStreamPreservesRtpStatesWithRtx to fail with these refactorings since the pacer suddenly could send a padding packet before the encoder had produced its first frame.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1993113003
Cr-Commit-Position: refs/heads/master@{#13149}
CreatePeerConnectionFactory does not yet expose the ability to set the
factory from the outside.
Added notry due to android_dbg being broken.
NOTRY=True
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1991233004
Cr-Commit-Position: refs/heads/master@{#13112}
Instead of the default copy constructor, the Copy() method has to be used. In this CL, the number of copies has been reduced significantly in production code. One case in the video engine remains, where we need to restart a video stream. Even in that case, I'm sure we could avoid it, but for this particular CL, I decided against it to keep things simple (and it's also an edge case). Most importantly, creating copies is made harder and the interface encourages ownership transfers.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/2042603002 .
Cr-Commit-Position: refs/heads/master@{#13102}
This CL implements auto pausing video streams per stream with logic to
avoid toggling state too often.
Also re-enabling tests disabled for Mac, with the assumption the new
logic removes flakiness.
BUG=webrtc:5868,webrtc:5407
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2035383002 .
Cr-Commit-Position: refs/heads/master@{#13092}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
Also set the Configuration parameters in CreateRtpRtcpModules in the same order as the members are declared.
BUG=webrtc:5917
Review-Url: https://codereview.webrtc.org/2011433002
Cr-Commit-Position: refs/heads/master@{#12905}
Updated tests to use the default implementation and removed the test implementation (webrtc/test/histograms.h).
BUG=
Review-Url: https://codereview.webrtc.org/1915523002
Cr-Commit-Position: refs/heads/master@{#12829}
Remove ViEEncoder::SetNetworkStatus.
Original cl description:
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.
Patchset #1 is a pure reland.
Patchset #2 change the bitrate allocator to always return an initial bitrate >0 regardless of the estimates. The observer will be notified though if the network is down.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1972183004
Cr-Commit-Position: refs/heads/master@{#12743}
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}
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}
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 reverts commit e30c27205148b34ba421184efe65f6a0780b436d (https://codereview.webrtc.org/1958053002/)
Original reverted cl is in patch set #1.
Changes in following patch sets.
The cl now also make sure SendPacer starts with the configured bitrate provided in a call to CongestionController::SetBweBitrates)()
It turns out that the failing tests in 609816 is due to a bug in the current code that runs the proper at 300kbit regardless of configured start bitrate.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=chromium:609816, webrtc:5687
TBR=mflodman@webrtc.org
NOTRY=True // Due to bug in android_x86 cq builder....
Review-Url: https://codereview.webrtc.org/1958113003
Cr-Commit-Position: refs/heads/master@{#12688}
This reverts commit 825eb58d59940a4c3c9837595c4b3b07059c93ca.
This Relands the cl reviewed in https://codereview.webrtc.org/1917793002/
patchset #1 is a pure reland.
patchset #2 fix an overflow in BitrateProber that caused WebRtcVideoChannel2BaseTest.TwoStreamsSendAndReceive to fail.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:5687
Review URL: https://codereview.webrtc.org/1947873002 .
Cr-Commit-Position: refs/heads/master@{#12630}
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1917793002
Cr-Commit-Position: refs/heads/master@{#12620}
- "WebRTC.Video.SendDelayInMs"
Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.
BUG=webrtc:5215
Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.
NOPRESUBMIT=True
BUG=webrtc:3970
Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
VoENetwork is kept for now, but is not really used anylonger.
webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.
BUG=webrtc:5079
TBR=tommi
Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
They're just no-ops now, and will soon go away.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1914153002
Cr-Commit-Position: refs/heads/master@{#12510}
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
The logging thread is always active. The main thread uses SwapQueues to pass events to the logging thread. The logging thread moves the events to either a RingBuffer history in memory, or to a string which is written to disc.
RtcEventLogImpl constructor takes a clock for easier testing.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1687703002
Cr-Commit-Position: refs/heads/master@{#12476}
It's flaky not only for ASan/MSan, but regular Windows/Linux bots.
NOTRY=True
BUG=webrtc:5790
Review URL: https://codereview.webrtc.org/1908663002
Cr-Commit-Position: refs/heads/master@{#12475}
Adds logging of:
- video stats that are recorded when a stream is removed
- bitrate stats that are recorded at the end of a call
- initial bwe rampup stats
BUG=
Review URL: https://codereview.webrtc.org/1788783002
Cr-Commit-Position: refs/heads/master@{#12133}
To replace the SmoothsRenderedFrames method, added a corresponding
flag to VideoReceiveStream::Config instead.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1818023002
Cr-Commit-Position: refs/heads/master@{#12102}
This change enables voice-only calls to keep track of the network state.
This is only a partial fix - the last modality to change state controls
the state for the entire call, so a call with a failed video transport
will also stop sending audio packets. Handling this condition correctly
would require the call to keep track of network state for each media
type separately, and take care of conditions such as a failed video
channel getting removed, while a functioning audio channel remains.
BUG=webrtc:5307
Review URL: https://codereview.webrtc.org/1757683002
Cr-Commit-Position: refs/heads/master@{#12093}
webrtc::VideoRenderer class, replacing it by rtc::VideoSinkInterface.
The next step is to convert all places where a renderer is attached to
rtc::VideoSourceInterface, and at that point, the
SmoothsRenderedFrames method can be replaced by a flag
rtc::VideoSinkWants::smoothed_frames.
Delete unused method IsTextureSupported.
Delete unused time argument to RenderFrame.
Let webrtc::VideoRenderer inherit rtc::VideoSinkInterface. Rename RenderFrame --> OnFrame.
TBR=kjellander@webrtc.org
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1814763002
Cr-Commit-Position: refs/heads/master@{#12070}
Reason for revert:
The openmax_dl include change breaks downstream projects.
Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623
Review URL: https://codereview.webrtc.org/1808573002
Cr-Commit-Position: refs/heads/master@{#12009}
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1796413002 .
Cr-Commit-Position: refs/heads/master@{#12008}
