This it to avoid requiring targets that include header files that in turn use SequenceTaskedChecker, to also have to define the macros needed by TaskQueue.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2145393003
Cr-Commit-Position: refs/heads/master@{#13482}
If the currently selected connection becomes not receiving and if a backup connection
becomes strong first, we will not switch the connection until X milliseconds is passed
but the selected connection is still not receiving and the backup connection is still receiving. This will prevent the connection switching from happening too frequently.
BUG=
Review-Url: https://codereview.webrtc.org/2143653005
Cr-Commit-Position: refs/heads/master@{#13480}
It causes an asan initialization-order-fiasco in trying to read the
names of other globally constructed data:
==21449==ERROR: AddressSanitizer: initialization-order-fiasco on address 0x7f6f297bc5e8 at pc 0x7f6f26b332a7 bp 0x7ffd479f8cb0 sp 0x7ffd479f8ca8
READ of size 8 at 0x7f6f297bc5e8 thread T0
#0 0x7f6f26b332a6 in name
webrtc/base/flags.h:83:38
#1 0x7f6f26b332a6 in Lookup
webrtc/base/flags.cc:133
#2 0x7f6f26b332a6 in rtc::FlagList::Register(rtc::Flag*)
webrtc/base/flags.cc:260
#3 0x7f6f2529972b in __cxx_global_var_init.1
BUG=
Review-Url: https://codereview.webrtc.org/2110963004
Cr-Commit-Position: refs/heads/master@{#13479}
This cl clean up unused forward declaration of cricket::AudioFrame and cricket::Renderer.
cricket::VideoFrame is used in the interface and thus the header file is included.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2145613006
Cr-Commit-Position: refs/heads/master@{#13477}
This cl is in preparation for https://codereview.webrtc.org/2060403002/ Add task queue to Call.
In the coming cl the video_sender, and i420_buffer_pool will be used on a task queue and therefore SequencedTaskChecker is needed instead of a ThreadChecker.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2149553002
Cr-Commit-Position: refs/heads/master@{#13474}
Merge with the latest webrtc native code.
Remove deprecated function Connect() in QuicTransportChannel.
Fix the compiling issue and broken unit tests by adding the network thread to QUIC related classes.
Review-Url: https://codereview.webrtc.org/2089553002
Cr-Commit-Position: refs/heads/master@{#13472}
Reason for revert:
Reverting because description was inaccurate. Will reland after updating description.
Original issue's description:
> Modify PeerConnection for end-to-end QuicDataChannel usage
>
> To allow end-to-end QuicDataChannel usage with a
> PeerConnection, RTCConfiguration has been modified to
> include a boolean for whether to do QUIC, since negotiation of
> QUIC is not implemented. If one peer does QUIC, then it will be
> assumed that the other peer must do QUIC or the connection
> will fail.
>
> PeerConnection has been modified to create data channels of type
> QuicDataChannel when the peer wants to do QUIC.
>
> WebRtcSession has been modified to use a QuicTransportChannel
> instead of a DtlsTransportChannelWrapper/DataChannel
> when QUIC should be used.
>
> Modification of previous in-flight CL: https://codereview.chromium.org/1844803002/
>
> Committed: https://crrev.com/36c8d69ce188102ae6fd48c371cf1518f08698fb
> Cr-Commit-Position: refs/heads/master@{#13470}
TBR=pthatcher@webrtc.org,zhihuang@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/2146133002
Cr-Commit-Position: refs/heads/master@{#13471}
To allow end-to-end QuicDataChannel usage with a
PeerConnection, RTCConfiguration has been modified to
include a boolean for whether to do QUIC, since negotiation of
QUIC is not implemented. If one peer does QUIC, then it will be
assumed that the other peer must do QUIC or the connection
will fail.
PeerConnection has been modified to create data channels of type
QuicDataChannel when the peer wants to do QUIC.
WebRtcSession has been modified to use a QuicTransportChannel
instead of a DtlsTransportChannelWrapper/DataChannel
when QUIC should be used.
Modification of previous in-flight CL: https://codereview.chromium.org/1844803002/
Review-Url: https://codereview.webrtc.org/2089553002
Cr-Commit-Position: refs/heads/master@{#13470}
The DtlsTransportChannel wasn't properly handling the
P2PTransportChannel becoming writable before receiving a remote
fingerprint; it wasn't starting the DTLS handshake whenever this
happened.
Review-Url: https://codereview.webrtc.org/2140283002
Cr-Commit-Position: refs/heads/master@{#13469}
This updates how we initialize timer events to be the same way as we initialize non-timer events.
BUG=
Review-Url: https://codereview.webrtc.org/2147973002
Cr-Commit-Position: refs/heads/master@{#13468}
The decode thread should be stopped before triggering shutdown of the
video receiver, so that the decoder doesn't try to insert a new frame
while the jitter buffer is being shut down.
BUG=webrtc:6102
Review-Url: https://codereview.webrtc.org/2146883002
Cr-Commit-Position: refs/heads/master@{#13467}
Plot graphs to python code. Generate packet, playout, sequence number and total bitrate graphs.
Add bitrate and latency plots. Generate multiple figures, and control which ones are used through command line flags.
Committed: https://crrev.com/a478786ef1513790194792010f766324a469db4d
Review-Url: https://codereview.webrtc.org/1995523002
Cr-Original-Commit-Position: refs/heads/master@{#13443}
Cr-Commit-Position: refs/heads/master@{#13463}
Don't update histograms until we're sure process threads won't call into
the instance being destructed, trying to update stats.
BUG=webrtc:6103
Review-Url: https://codereview.webrtc.org/2151433002
Cr-Commit-Position: refs/heads/master@{#13461}
Reason for revert:
For some reason, payload_type_mapper.cc is not being picked up in Chrome builds, leading to undefined references. Reverting while investigating.
Original issue's description:
> WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.
>
> Changed WebRtcVoiceEngine to present receive codecs from the formats
> provided by its decoder factory. Added supported formats to
> BuiltinAudioDecoderFactory. Added helper functions for creating some
> simple decoder factories for mocking.
>
> Created a PayloadTypeMapper for assigning payload types to formats. I
> think we'll eventually want to use this further up, or possibly in
> both the audio and video sides. It would be best if the engines didn't
> have to talk payload types at all, but it might be more difficult to
> get around when payload types depend on each-other, like the RTX
> codecs for video.
>
> This CL also includes some changes to rtc::Optional. I've put them in
> a separate CL that should (or should not) land first, making these
> changes void.
> See: https://codereview.webrtc.org/2072713002/
>
> BUG=webrtc:5805
>
> Committed: https://crrev.com/95eb1ba0db79d8fd134ae61b0a24648598684e8a
> Cr-Commit-Position: refs/heads/master@{#13459}
TBR=ivoc@webrtc.org,tina.legrand@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2151453002
Cr-Commit-Position: refs/heads/master@{#13460}
Changed WebRtcVoiceEngine to present receive codecs from the formats
provided by its decoder factory. Added supported formats to
BuiltinAudioDecoderFactory. Added helper functions for creating some
simple decoder factories for mocking.
Created a PayloadTypeMapper for assigning payload types to formats. I
think we'll eventually want to use this further up, or possibly in
both the audio and video sides. It would be best if the engines didn't
have to talk payload types at all, but it might be more difficult to
get around when payload types depend on each-other, like the RTX
codecs for video.
This CL also includes some changes to rtc::Optional. I've put them in
a separate CL that should (or should not) land first, making these
changes void.
See: https://codereview.webrtc.org/2072713002/
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2072753002
Cr-Commit-Position: refs/heads/master@{#13459}
Reason for revert:
My original patch broke things that are now fixed by https://codereview.webrtc.org/2141193002/.
Hence I am relanding my original change.
Original issue's description:
> Revert of Adds data logging in native AudioDeviceBuffer class (patchset #10 id:180001 of https://codereview.webrtc.org/2132613002/ )
>
> Reason for revert:
> Seems to break things upstream.
>
> Original issue's description:
> > Adds data logging in native AudioDeviceBuffer class.
> >
> > Goal is to provide periodic logging of most essential audio parameters
> > for playout and recording sides. It will allow us to track if the native audio layer is working as intended.
> >
> > BUG=NONE
> >
> > Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae
> > Cr-Commit-Position: refs/heads/master@{#13440}
>
> TBR=stefan@webrtc.org,henrika@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=NONE
>
> Committed: https://crrev.com/025aa94ccb85e4c6fe20a3fecdac5d27ec9ba3da
> Cr-Commit-Position: refs/heads/master@{#13441}
TBR=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=NONE
Review-Url: https://codereview.webrtc.org/2138403003
Cr-Commit-Position: refs/heads/master@{#13455}
event_set() has been deprecated and doesn't allow passing the main loop to be passed to initialize the event. event_assign() is the method to call in Event2 when available. However, we still need to be compatible with Chromium's version of libevent, so there's an #ifdef check in the meantime.
BUG=
Review-Url: https://codereview.webrtc.org/2141193002
Cr-Commit-Position: refs/heads/master@{#13454}
SequencedTaskChecker is a helper class used to help verify that some methods
of a class are called on the same task queue or thread. A
SequencedTaskChecker is bound to a a task queue if the object is
created on a task queue, or a thread otherwise.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2125113003
Cr-Commit-Position: refs/heads/master@{#13452}
If the test TURN server received two allocate requests from the same
address, it was replacing the old allocation but not deleting it.
Also switching to std::unique_ptr to make it less likely for this to
pop up again.
Review-Url: https://codereview.webrtc.org/2114063002
Cr-Commit-Position: refs/heads/master@{#13449}
SurfaceViewRendererOnMeasureTest#testFrame1280x720 is currently flaky
because of a race. This issue was introduced here:
https://codereview.webrtc.org/2111933002/. That CL moved the function
updateFrameDimensionsAndReportEvents() from renderFrame() to
renderFrameOnRenderThread(). The OnMeasureTest currently just calls
renderFrame() with a new frame size and immediately starts testing the
measured size, which might be before renderFrameOnRenderThread() is
executed. This CL waits for the RendererEvents.onFrameResolutionChanged()
callback before starting the test to fix this race.
BUG=webrtc:6089
Review-Url: https://codereview.webrtc.org/2147463002
Cr-Commit-Position: refs/heads/master@{#13448}
Reason for revert:
Reverting while investigating a downstream build failure.
Original issue's description:
> Initial version of the local visualization tool for WebrtcEventLogs.
>
> Plot graphs to python code. Generate packet, playout, sequence number and total bitrate graphs.
>
> Add bitrate and latency plots. Generate multiple figures, and control which ones are used through command line flags.
>
> Committed: https://crrev.com/a478786ef1513790194792010f766324a469db4d
> Cr-Commit-Position: refs/heads/master@{#13443}
TBR=stefan@webrtc.org,aleloi@webrtc.org,phoglund@webrtc.org,henrika@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/2147453002
Cr-Commit-Position: refs/heads/master@{#13445}
With this change, an increase in Unique_frames_count will be marked
as an improvement (rather than a deterioration) in the Chrome Performance
Dashboard.
BUG=chromium:623521
R=phoglund@webrtc.org
Review-Url: https://codereview.webrtc.org/2142813002
Cr-Commit-Position: refs/heads/master@{#13444}
Plot graphs to python code. Generate packet, playout, sequence number and total bitrate graphs.
Add bitrate and latency plots. Generate multiple figures, and control which ones are used through command line flags.
Review-Url: https://codereview.webrtc.org/1995523002
Cr-Commit-Position: refs/heads/master@{#13443}
Reason for revert:
Seems to break things upstream.
Original issue's description:
> Adds data logging in native AudioDeviceBuffer class.
>
> Goal is to provide periodic logging of most essential audio parameters
> for playout and recording sides. It will allow us to track if the native audio layer is working as intended.
>
> BUG=NONE
>
> Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae
> Cr-Commit-Position: refs/heads/master@{#13440}
TBR=stefan@webrtc.org,henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE
Review-Url: https://codereview.webrtc.org/2139233002
Cr-Commit-Position: refs/heads/master@{#13441}
Goal is to provide periodic logging of most essential audio parameters
for playout and recording sides. It will allow us to track if the native audio layer is working as intended.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2132613002
Cr-Commit-Position: refs/heads/master@{#13440}
Support encoding from CVPixelBuffers directly in H264VideoToolboxEncoder.
If the frame needs to be scaled, it will fall back to the previous slow
path:
CVPixelBuffer -> NV12 -> I420 -> (scale) I420 -> NV12 -> CVPixelBuffer.
BUG=webrtc:4081
Review-Url: https://codereview.webrtc.org/2140573002
Cr-Commit-Position: refs/heads/master@{#13439}
The issue was that if the main entry routine of the thread would get called before the first APC, the thread would hang on a GetMessage() call and the APC sent to initialize the thread, would never run.
BUG=
Review-Url: https://codereview.webrtc.org/2139723003
Cr-Commit-Position: refs/heads/master@{#13438}
webrtc doesnt include the header that the function is prototyped in.
This CL makes the convert_from.h include those headers to allow webrtc to
update to the head libyuv.
R=marpan@webrtc.org,pbos@webrtc.org
BUG=libyuv:620,webrtc:6094
TESTED=local build and try bots
Review-Url: https://codereview.webrtc.org/2139853002
Cr-Commit-Position: refs/heads/master@{#13436}
Added various timestamps to the FrameObject class which are needed to calculate
the jitter delay.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2124943002
Cr-Commit-Position: refs/heads/master@{#13434}
This interface and its implementations have been replaced by
rtc::RTCCertificateGeneratorInterface.
Removes dtlsidentitystore.h, updates .gyp/gn and removes old #includes.
BUG=webrtc:5707, webrtc:5708
Review-Url: https://codereview.webrtc.org/2034013003
Cr-Commit-Position: refs/heads/master@{#13432}
These arguments are not really known when calling SetEncodingData. They are still provided as argument to ProtectionBitrateCalculator::SetTargetRates though.
This cl is broken out from https://codereview.webrtc.org/2060403002/
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2121983002
Cr-Commit-Position: refs/heads/master@{#13429}
