Reason for revert:
Breaking webrtc builder.
Original issue's description:
> Adding IceConfig option to assume TURN/TURN candidate pairs will work.
>
> This will allow media to be sent on these pairs before a binding
> response is received, shortening call setup time. However, this is only
> possible if the TURN servers don't require CreatePermission when
> communicating with each other.
>
> R=honghaiz@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/8e6134eae4117a239de67c9a9dae8f5e3235d803
> Cr-Commit-Position: refs/heads/master@{#13263}
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
Review-Url: https://codereview.webrtc.org/2090823002
Cr-Commit-Position: refs/heads/master@{#13264}
This will allow media to be sent on these pairs before a binding
response is received, shortening call setup time. However, this is only
possible if the TURN servers don't require CreatePermission when
communicating with each other.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2063823008 .
Cr-Commit-Position: refs/heads/master@{#13263}
Even if those error happened, the connection may still be able to receive packets for a while.
If we delete the connections, all packets arriving will be dropped.
BUG=webrtc:6007
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2068263003 .
Cr-Commit-Position: refs/heads/master@{#13262}
If an actual peer reflexive candidate was created (and not one that
would just be replaced by a different candidate later), we weren't
setting the generation value. This means that new-generation prflx
candidate pairs weren't being prioritized above the cross-generation
pairs, or above relay<->relay pairs.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2086793002 .
Cr-Commit-Position: refs/heads/master@{#13259}
In some situation, typically when incoming packets were reordered, the
DelayPeakDetector::Update method may be called twice (or more) with
non-zero inter_arrival_time argument, but without the TickTimer object
being updated in between (i.e., packets coming in more or less at the
same time). In these situations, a delay peak may be registered with
zero peak period. This could eventually trigger the DCHECK in
DelayPeakDetector::MaxPeakPeriod().
With this fix, the problem is solved by not registering peaks for which
the TickTimer object has not moved since the last peak.
The problem was originally introduced with
https://codereview.webrtc.org/1921163003.
BUG=webrtc:6021
Review-Url: https://codereview.webrtc.org/2085233002
Cr-Commit-Position: refs/heads/master@{#13257}
- RTP and RTCP corpora for existing fuzzers
- STUN/SDP/pseudotcp for upcoming ones
- STUN/SDP tokens as well
NOTRY=true
Review-Url: https://codereview.webrtc.org/2082943002
Cr-Commit-Position: refs/heads/master@{#13253}
This change is a major refactoring of the neteq_rtpplay tool. It
consists of the following parts:
- NetEqTest class: Breaks out the main simulation loop from
neteq_rtpplay into a separate class with well defined inputs and
outputs.
- NetEqInput: Interface class for the input to NetEqTest.
- NetEqPacketSourceInput: Implementation of NetEqInput that provides a
PacketSource objects with a NetEqInput interface. This has two
subclasses; one for RtpFileSource and one for RtcEventLogSource.
- NetEqReplacementInput: An object that modifies the packets provided by
another NetEqInput object, and replaces the packet payloads with meta
data readable by a FakeDecodeFromFile decoder.
- FakeDecodeFromFile: An AudioDecoder implementation that produces
"decoded" data by reading from an audio file.
BUG=webrtc:2692, webrtc:5447
Review-Url: https://codereview.webrtc.org/2020363003
Cr-Commit-Position: refs/heads/master@{#13252}
VideoDecoderParams contains the id of the receive video
stream. Motivation behind this change is to enable down
stream apps easier map raw non-decoded data to incoming
streams.
BUG=b/28636393
Review-Url: https://codereview.webrtc.org/2052233002
Cr-Commit-Position: refs/heads/master@{#13250}
The method was deprecated and shouldn't be used anywhere now.
BUG=webrtc:5950
Review-Url: https://codereview.webrtc.org/2080573004
Cr-Commit-Position: refs/heads/master@{#13248}
Previously, if the ICE role changed, SetIceRole was only called on
the ports from the most recent ICE generation. However, STUN pings
may still be sent and received by older generation ports, so they
should receive an updated role as well.
This was previously triggering an ASSERT, because a P2PTransportChannel
expects the ICE role of each of its ports to match its own role.
Committed: https://crrev.com/370544594e18deb7f560f961295c8cf3f0a679f1
Review-Url: https://codereview.webrtc.org/2053043003
Cr-Original-Commit-Position: refs/heads/master@{#13226}
Cr-Commit-Position: refs/heads/master@{#13247}
Previously they were only being updated for connections using the
most current "generation" of ports. This results in the older-
generation prflx candidate pair being prioritized above newer-
generation candidate pairs.
Review-Url: https://codereview.webrtc.org/2087713002
Cr-Commit-Position: refs/heads/master@{#13245}
Label less chunks as speech, adapt slower and be more conservative with the maximum gain it can apply.
Review-Url: https://codereview.webrtc.org/2087623003
Cr-Commit-Position: refs/heads/master@{#13242}
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
Committed: https://crrev.com/8f7a5aad55a64f0d81b6436a22ffbdfcdcde91e0
Review-Url: https://codereview.webrtc.org/1944003002
Cr-Original-Commit-Position: refs/heads/master@{#12736}
Cr-Commit-Position: refs/heads/master@{#13241}
Reason for revert:
Speculative revert: breaks video quality tests on Win and Mac (???): https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/31209
Original issue's description:
> Update ICE role on all ports, not just ones used for new connections.
>
> Previously, if the ICE role changed, SetIceRole was only called on
> the ports from the most recent ICE generation. However, STUN pings
> may still be sent and received by older generation ports, so they
> should receive an updated role as well.
>
> This was previously triggering an ASSERT, because a P2PTransportChannel
> expects the ICE role of each of its ports to match its own role.
>
> Committed: https://crrev.com/370544594e18deb7f560f961295c8cf3f0a679f1
> Cr-Commit-Position: refs/heads/master@{#13226}
TBR=pthatcher@webrtc.org,honghaiz@webrtc.org,deadbeef@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/2078423004
Cr-Commit-Position: refs/heads/master@{#13240}
In GN, the libjingle_peerconnection_jni target becomes a part of
'all' implicitly, which surfaced the incompability between it
and the Chromium logging implementation. In the GYP build, the
target is not present due to api.gyp not being depended upon yet.
BUG=webrtc:4256
TBR=perkj@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2082573004
Cr-Commit-Position: refs/heads/master@{#13231}
Reason for revert:
Reverting the revert. This change is not related to the failure on the Windows FYI bots. The cause of the failure has been reverted in Chromium:
https://codereview.chromium.org/2081653004/
Original issue's description:
> Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #5 id:340001 of https://codereview.webrtc.org/2078873002/ )
>
> Reason for revert:
> Breaks chromium.webrtc.fyi
>
> https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4719
> https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3120
>
> Original issue's description:
> > Reland of IncomingVideoStream refactoring.
> > This reland does not contain the non-smoothing part of the original implementation. Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before. This code path is used in Chrome. As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks). Other than that, no change for Chrome.
> >
> > Original issue's description (with non-smoothing references removed):
> >
> > Split IncomingVideoStream into two implementations, with smoothing and without.
> >
> > * Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.
> >
> > * Removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
> >
> > * Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
> >
> > * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
> >
> > * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
> >
> > * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
> >
> > * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
> >
> > * Made the render delay value in VideoRenderFrames, const.
> >
> > BUG=chromium:620232
> > R=mflodman@webrtc.org, nisse@webrtc.org
> >
> > Committed: https://crrev.com/884c336c345d988974c2a69cea402b0fb8b07a63
> > Cr-Commit-Position: refs/heads/master@{#13219}
>
> TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:620232
>
> Committed: https://crrev.com/a536bfe70de38fe877245317a7f0b00bcf69cbd0
> Cr-Commit-Position: refs/heads/master@{#13229}
TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620232
Review-Url: https://codereview.webrtc.org/2089613002
Cr-Commit-Position: refs/heads/master@{#13230}
Reason for revert:
Breaks chromium.webrtc.fyi
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4719https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3120
Original issue's description:
> Reland of IncomingVideoStream refactoring.
> This reland does not contain the non-smoothing part of the original implementation. Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before. This code path is used in Chrome. As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks). Other than that, no change for Chrome.
>
> Original issue's description (with non-smoothing references removed):
>
> Split IncomingVideoStream into two implementations, with smoothing and without.
>
> * Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.
>
> * Removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
>
> * Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
>
> * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
>
> * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
>
> * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
>
> * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
>
> * Made the render delay value in VideoRenderFrames, const.
>
> BUG=chromium:620232
> R=mflodman@webrtc.org, nisse@webrtc.org
>
> Committed: https://crrev.com/884c336c345d988974c2a69cea402b0fb8b07a63
> Cr-Commit-Position: refs/heads/master@{#13219}
TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620232
Review-Url: https://codereview.webrtc.org/2084873002
Cr-Commit-Position: refs/heads/master@{#13229}
Previously, if the ICE role changed, SetIceRole was only called on
the ports from the most recent ICE generation. However, STUN pings
may still be sent and received by older generation ports, so they
should receive an updated role as well.
This was previously triggering an ASSERT, because a P2PTransportChannel
expects the ICE role of each of its ports to match its own role.
Review-Url: https://codereview.webrtc.org/2053043003
Cr-Commit-Position: refs/heads/master@{#13226}
Also added IntelligibilityEnhancer setting to aecdump simulator in audioproc_f
Review-Url: https://codereview.webrtc.org/2075093003
Cr-Commit-Position: refs/heads/master@{#13220}
This reland does not contain the non-smoothing part of the original implementation. Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before. This code path is used in Chrome. As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks). Other than that, no change for Chrome.
Original issue's description (with non-smoothing references removed):
Split IncomingVideoStream into two implementations, with smoothing and without.
* Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.
* Removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
* Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
* Made the render delay value in VideoRenderFrames, const.
BUG=chromium:620232
R=mflodman@webrtc.org, nisse@webrtc.org
Review URL: https://codereview.webrtc.org/2078873002 .
Cr-Commit-Position: refs/heads/master@{#13219}
This lets us use their fancy features, including seed_corpus which is
super handy.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2081683002
Cr-Commit-Position: refs/heads/master@{#13216}
Don't use VideoFrameBuffer::MutableDataY and friends, instead, use
I420Buffer::SetToBlack.
Also introduce static method I420Buffer::Create, to create an object and
return a scoped_refptr.
TBR=marpan@webrtc.org # Trivial change to video_denoiser.cc
BUG=webrtc:5921
Review-Url: https://codereview.webrtc.org/2078943002
Cr-Commit-Position: refs/heads/master@{#13212}
We didn't really want it; it was only necessary because we wanted to
use rtc::Optional<SdpAudioFormat>, and Optional used to require the
contained type to be default constructable. But as of May 9th
(https://codereview.webrtc.org/1896833004), it no longer does.
Review-Url: https://codereview.webrtc.org/2066233002
Cr-Commit-Position: refs/heads/master@{#13211}
Allows detecting large-enough audio packets as part of a probe,
speculative fix for a rampup-time regression in M50. These packets are
accounted on the send side when probing.
BUG=webrtc:5985
R=mflodman@webrtc.org, philipel@webrtc.org
Review URL: https://codereview.webrtc.org/2061193002 .
Cr-Commit-Position: refs/heads/master@{#13210}
In order to switch Chromium to use WebRTC targets instead of
duplicated code listings in src/third_party/libjingle it must
be possible for Chromium to process webrtc/api/api.gyp. This is
currently not possible since it includes build/java.gypi, of which
the path is different in a Chromium checkout. It's not possible
to resolve this in another way since 'includes' processing takes
place early in the GYP cycle, before it's possible to use variables.
They're also processed ignoring conditional statements, resulting
in an error when api.gyp is processed.
BUG=webrtc:4256
TBR=perkj@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2080563002
Cr-Commit-Position: refs/heads/master@{#13208}
