1376873 - Update WebRTC code to webrtc.org stable branch 64

Status: RESOLVED FIXED

(Core :: WebRTC, defect, P2)

Description

[Dan Minor [:dminor]]

61 hasn't branched yet, but that will be our target for the next webrtc.org update.

Comment 1

[Bulk Bug Changes for mreavy's org]

Mass change P1->P2 to align with new Mozilla triage process

Comment 2

[Randell Jesup [:jesup] (needinfo me)]

Note: if https://bugs.chromium.org/p/webrtc/issues/detail?id=8536 isn't in webrtc.org 61, and it's applicable (I think it may well be), we'll want to cherry-pick the patch that landed upstream to fix an H264 packet-loss-recovery bug

Comment 3

[Dan Minor [:dminor]]

I think we should target 64 for the next update. That entered beta on Dec. 18th.

Comment 4

[Dan Minor [:dminor]]

Attachment: This is the import of the webrtc.org code itself.

This is the import of the third party webrtc.org code. It is too large for Phabricator (Bug 1492214) so I'm requesting review here instead.

Comment 5

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9010747 [details]
This is the import of the webrtc.org code itself.

LGTM

Comment 6

[Dan Minor [:dminor]]

Try run here: https://treeherder.mozilla.org/#/jobs?repo=try&selectedJob=202870186&revision=288496a4f9417e6cd3cc47c16754bcb61fd7f5c5

Comment 7

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Update build files from tip of webrtc.org; r=ng

This is a cherrypick and rebase of the build system changes in commit
9c6861308021b1f05fbf17538778ee7177e6d312. These landed on webrtc.org after 64
was branched.

MozReview-Commit-ID: JQKeXWdlkq6

Comment 8

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Allow gn to have a non-default target; r=chmanchester

MozReview-Commit-ID: GBQSi6z059y

Depends on D7422

Comment 9

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Build system fixes; r=ng

MozReview-Commit-ID: HPKBZnAMKoI

Depends on D7423

Comment 10

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Rollup of local modifications; r=ng

MozReview-Commit-ID: 2euYzBEvuNb

Depends on D7424

Comment 11

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Add SpeexResampler; r=padenot

In the past we've replaced the contents of resampler.cc with code to use Speex
instead. This moves that code out into a separate class which should be easier
to maintain and potentially upstream.

MozReview-Commit-ID: H8ogaovdGPf

Depends on D7425

Comment 12

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Rollup conflict fixes for audio/video code; r=pehrsons

MozReview-Commit-ID: 1T8mgqdkzq3

Depends on D7426

Comment 13

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Rollup conflict fixes for rtp_rtcp module; r=ng

MozReview-Commit-ID: D09534DOVLj

Depends on D7427

Comment 14

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix up logging in WebrtcLog.cpp; r=ng

The webrtc::Trace code is removed by this update. We already had support for
LOG (now RTC_LOG) in WebrtcLog.cpp. This removes the trace code from
WebRtcLog.cpp and moves the aec logging code from webrtc::Trace to
rtc::LogMessage.

This also disables logging to stderr in rtc_base/logging.cc. We could disable
it using the API, but that happens through peerconnection resulting in some
logging occuring during getusermedia.

The aec logs were testing with --disable-e10s. Rather than trying to
work around sandboxing, I think it makes more sense to fix Bug 1404982 and
store the logs in memory for retrieval from about:webrtc.

Depends on D7428

Comment 15

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix include path in JsepSessionImpl.cpp; r=bwc

Depends on D7429

Comment 16

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix RTCP SenderReport; r=ng

Depends on D7430

Comment 17

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - WebrtcGmpVideoCodec updates; r=pehrsons

Depends on D7431

Comment 18

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - VideoConduit updates; r=pehrsons

Depends on D7432

Comment 19

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - WebrtcMediaDataDecoderCodec updates; r=pehrsons

Depends on D7433

Comment 20

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Update mtransport sigslot.h; r=bwc

This updates the copy of sigslot used by mtransport to match that used by
webrtc.

Depends on D7435

Comment 21

[Byron Campen [:bwc]]

Comment on attachment 9013620 [details]
Bug 1376873 - Fix include path in JsepSessionImpl.cpp; r=bwc

Byron Campen [:bwc] has approved the revision.

Comment 22

[Byron Campen [:bwc]]

Comment on attachment 9013630 [details]
Bug 1376873 - Update mtransport sigslot.h; r=bwc

Byron Campen [:bwc] has approved the revision.

Comment 23

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Mediapipeline updates; r=pehrsons

Support for native_handle() has been removed by upstream.

Depends on D7436

Comment 24

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Updates to dom/media/, dom/media/systemservices and dom/media/webrtc; r=pehrsons

Depends on D7437

Comment 25

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Add RtpPacketQueue; r=pehrsons

We'll need to queue rtp packets in both the AudioConduit and the VideoConduit.
This adds a class to manage the packet queue to reduce duplicated code between
the conduits.

Depends on D7438

Comment 26

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Disable or replace thread check assertions; r=pehrsons

The changes to Call are required because we create the Call object on the main
thread, but deliver packets and query stats from the socket thread.

The changes to ChannelProxy are required because we query stats from the
socket thread rather than the main thread.

For RtpVideoStreamReceiver, this removes the worker_task_checker_ assertions
and replaces them with a critical section. This is how the code worked prior
to this update. We create the Call object (and thus eventually the
RtpVideoStreamReceiver) on the main thread, but we want to deliver packets on
the socket thread. To retain these assertions we'd either have to dispatch calls
to deliver packets from the socket thread to the main, which seems pretty bad
from a performance point of view, or we'd have to refactor the code to create
the Call object on the socket thread, which seems like a major refactoring
best done outside of a branch update. Going back to the previous behaviour
seemed like the least bad alternative.

Comment 27

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Use Call interface in AudioConduit; r=padenot,pehrsons

Depends on D7440

Comment 28

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Disable Mid support in RtpDemuxer; r=mjf

The only use of Mid in the current webrtc.org code is in the unit tests.
RtpStreamReceiverController only allows adding sinks using SSRCs. Because
of this, we'll end up dropping packets in the RtpDemuxer with the current
code as none of our Mids will be recognized.

Tip of webrtc.org fully supports using Mids, so we'll be able to enable this
code again after the next update.

Depends on D7441

Comment 29

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Remove CPULoadStateObserver; r=ng

We went through a lot of trouble to plumb the CPULoadState down to
MediaOptimization, but the value is not actually used for anything, at least
since the Branch 57 update. This removes the plumbing, since it seems we are
getting along ok without it.

Depends on D7442

Comment 30

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix GetRTCPSenderReport; r=ng

Upstream removed the RTCPSenderInfo structure as well as code to store the
packets and bytes received from the RTCP sender report. This adds code to the
rtcp receiver to store those values, adds a method to the RtpRtcp module to
retrieve the data and uses it to populate new fields added to the existing
CallStatistics (for audio) and VideoReceiveStream::Stats (for video) structures.

Depends on D7443

Comment 31

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix audio/video sync; r=bwc

Depends on D7444

Comment 32

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Update videoconduit_unittests gtests; r=pehrsons

Depends on D7445

Comment 33

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix fake h.264 encoder; r=pehrsons

Upstream webrtc.org has added more stringent checks for SPS and PPS identifiers.
This breaks out the SPS and PPS NALUs into separate binary blobs to ensure that
the identifiers are zero. It is also necessary to set pragma pack so that the
structure does not contain values which trick the parser into thinking a new
NALU has begun. The ENCODED_FRAME_MAGIC is changed for a similar reason. All of
the constants were determined by running mochitests with the actual h.264
encoder.

For similar reasons, this also changes things so that only IFrames are sent,
as the upstream code is now checking for previous IFrame identifiers and will
drop frames if they are not sent.

The SPS and PPS NALUs are now embedded into a single frame rather than being sent
as separate frames as was previously done. This is consistent with the real
H.264 plugin, and fixes a problem with intermittent failures due to occasionally
bad timestamps.

Depends on D7446

Comment 34

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Rollup of changes from e61c4485494e to 4f12d77b4f9b; r=ng

Depends on D7447

Comment 35

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fixups for rollup of changes from e61c4485494e to 4f12d77b4f9b; r=ng

Depends on D7448

Comment 36

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Remove Android java code from AudioConduit; r=padenot

This is no longer necessary as we do not use the webrtc.org audio device
implementations anymore.

Depends on D7449

Comment 37

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Update WebrtcMediaCodecVP8VideoCodec; r=pehrsons

Depends on D7450

Comment 38

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Updates to Android video capture; r=pehrsons

Depends on D7451

Comment 39

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Set default agc mode for Android in MediaEngineWebRTCAudio; r=padenot

This matches the current definition for kDefaultAgcMode in
voice_engine_defines.h.

Depends on D7452

Comment 40

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Updates to VideoConduit.cpp for OS X; r=pehrsons

Depends on D7454

Comment 41

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - OS X desktop capture fixes; r=pehrsons

Depends on D7455

Comment 42

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Change OS X logging include to prevent preprocessor multiple defines; r=ng

Depends on D7464

Comment 43

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Cherrypick rev 09133af36fba93691a22151765097c0bd581c1fa; r=ng

Depends on D7465

Comment 44

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Make lock_block explicit; r=bwc

Depends on D7466

Comment 45

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Allow single channel opus; r=padenot

Depends on D7468

Comment 46

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix warning in mean_variance_estimator.cc; r=padenot

Depends on D7469

Comment 47

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix GetRTPStatistics; r=ng

Depends on D7470

Comment 48

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Rollup of changes from cd8671232fd8 to 2a59b432d2bd; r=ng

Depends on D7471

Comment 49

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Setup contributing sources extension in audio_receive_stream.cc; r=ng

Depends on D7472

Comment 50

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Keep track of last codec mode in VideoConduit; r=pehrsons

Upstream has added an assertion in video send stream that prevents using
ReconfigureVideoEncoder if the content type has changed. This adds a
mLastCodecMode member and uses it to prevent calling that method if the
codec mode has changed.

Depends on D7474

Comment 51

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Don't define _USE_MATH_DEFINES in audio_processing.h; r=pehrsons

Depends on D7475

Comment 52

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Add FunctionThatDoesNothing specialized for const; r=ng

Windows ASAN builds complain about attempting to cast away the const qualifier
in FunctionThatDoesNothing, so this adds a version specialized for const.

Depends on D7476

Comment 53

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Fix up rotation in video_capture_impl.cc; r=pehrsons

This fixes a bug in the upstream code introduced when they removed the
ConvertToI420 helper method from webrtc_libyuv.cc. The buffer size is
passed into libyuv::ConvertI420 incorrectly when rotation is applied, which
causes bad rendering and instabilities.

Depends on D7477

Comment 54

[Dan Minor [:dminor]]

Attachment: Update gn generated json files

Comment 55

[Dan Minor [:dminor]]

Attachment: Add generated moz.build files

Comment 56

[Byron Campen [:bwc]]

Comment on attachment 9013674 [details]
Bug 1376873 - Make lock_block explicit; r=bwc

Byron Campen [:bwc] has approved the revision.

Comment 57

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013684 [details]
Bug 1376873 - Setup contributing sources extension in audio_receive_stream.cc; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 58

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Use VP8SimulcastAdapter in VideoConduit; r=ng

The VP8SimulcastAdapter automatically handles cropping to accomodate
constraints on VP8 simulcast streams that each stream have exactly half the
width and height of the higher resolution stream before it.

This also adjusts min_bitrate_estimate in
test_peerConnection_simulcastOddResolution.html to compensate for changes in
the simulcast resolutions chosen by the webrtc.org code.

Comment 59

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013687 [details]
Bug 1376873 - Add FunctionThatDoesNothing specialized for const; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 60

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013681 [details]
Bug 1376873 - Rollup of changes from cd8671232fd8 to 2a59b432d2bd; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 61

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013679 [details]
Bug 1376873 - Fix GetRTPStatistics; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 62

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013672 [details]
Bug 1376873 - Cherrypick rev 09133af36fba93691a22151765097c0bd581c1fa; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 63

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013671 [details]
Bug 1376873 - Change OS X logging include to prevent preprocessor multiple defines; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 64

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013650 [details]
Bug 1376873 - Rollup of changes from e61c4485494e to 4f12d77b4f9b; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 65

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013643 [details]
Bug 1376873 - Fix GetRTCPSenderReport; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 66

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013707 [details]
Bug 1376873 - Use VP8SimulcastAdapter in VideoConduit; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 67

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013620 [details]
Bug 1376873 - Fix include path in JsepSessionImpl.cpp; r=bwc

Nico Grunbaum [:ng] has approved the revision.

Comment 68

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013618 [details]
Bug 1376873 - Fix up logging in WebrtcLog.cpp; r=ng

Nico Grunbaum [:ng] has approved the revision.

Comment 69

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013698 [details]
Add generated moz.build files

LGTM

Comment 70

[Nico Grunbaum [:ng, @chew:mozilla.org]]

Comment on attachment 9013697 [details]
Update gn generated json files

LGTM

Comment 71

[Dan Minor [:dminor]]

Landry and Jan, we're planning on landing this update early in Firefox 65. Unfortunately, this means you'll need to regenerate gn json files for the BSDs based on the changed build configuration. It probably makes sense to hold off until this is reviewed, but I wanted to give you as much notice as possible. Sorry!

Comment 72

[Landry Breuil (:gaston)]

Thanks for the notice ! I'll have to dig out my notes from the previous upgrade, will try to do that in the coming weeks.

Comment 73

[Jan Beich]

Attachment: BSD gn generated json/mozbuild files (compressed)

As error-prone as in bug 1437670 and bug 1464680. Actual bustage fixes will follow later.

- DragonFly amd64 (GCC)
- FreeBSD i386/amd64/aarch64/armv7 (Clang)
- NetBSD i386/amd64/aarch64/armv7 (GCC)
- OpenBSD i386/amd64/aarch64/armv7 (Clang)

Comment 74

[Jan Beich]

Attachment: Unbreak build on BSDs

getUserMedia audio, DataChannels, Canvas demos work. I don't have microphone or camera to test more.

- event-config.h is required on everything but Windows, not sure why macOS is fine
- déjà vu: bug 1326011 also had to restore changes in platform_thread.cc
- alloca() is defined in stdlib.h (where Linux implicitly includes alloca.h)
- BSD macro (in sys/param.h) is an archaic of the (University of California) past

Comment 75

[Jan Beich]

Comment on attachment 9015301 [details] [diff] [review]
Unbreak build on BSDs

Review of attachment 9015301 [details] [diff] [review]:
-----------------------------------------------------------------

::: media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.cc
@@ +18,5 @@
>  #include <sys/stat.h>
>  #include <unistd.h>
>  // v4l includes
> +#if defined(__NetBSD__) || defined(__OpenBSD__)
> +#include <sys/videoio.h>

Landry, do you still need this conditional?

::: media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.cc
@@ +20,5 @@
>  #include <unistd.h>
>  
> +// v4l includes
> +#if defined(__NetBSD__) || defined(__OpenBSD__)
> +#include <sys/videoio.h>

... and this one?

Comment 76

[Jan Beich]

Comment on attachment 9015266 [details]
BSD gn generated json/mozbuild files (compressed)

https://treeherder.mozilla.org/#/jobs?repo=try&revision=30cc49dbdc0c390d0301a8ad55b18be3f22a0a23

Comment 77

[Landry Breuil (:gaston)]

(In reply to Jan Beich from comment #75)
> Comment on attachment 9015301 [details] [diff] [review]
> Unbreak build on FreeBSD
> 
> Review of attachment 9015301 [details] [diff] [review]:
> -----------------------------------------------------------------
> 
> :::
> media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.cc
> @@ +18,5 @@
> >  #include <sys/stat.h>
> >  #include <unistd.h>
> >  // v4l includes
> > +#if defined(__NetBSD__) || defined(__OpenBSD__)
> > +#include <sys/videoio.h>
> 
> Landry, do you still need this conditional?
> 
> :::
> media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.cc
> @@ +20,5 @@
> >  #include <unistd.h>
> >  
> > +// v4l includes
> > +#if defined(__NetBSD__) || defined(__OpenBSD__)
> > +#include <sys/videoio.h>
> 
> ... and this one?

videoio.h has all the V4L2 defines & ioctls, so i suppose yes ? I don't see why it should be removed, unless you have a real reason..

Is there a stash/branch somewhere with all the pending changes for this upcoming webrtc ? a list of patches ?

Why is attachment 9015266 [details] a diff adding the configs as new files, while they exist in the repo ? because the diff is unreadable ?

Comment 78

[Jan Beich]

(In reply to Landry Breuil (:gaston) from comment #77)
> videoio.h has all the V4L2 defines & ioctls, so i suppose yes ? I don't see
> why it should be removed, unless you have a real reason..

I wondered if OpenBSD may have added linux/ symlinks for compatibility.

> Is there a stash/branch somewhere with all the pending changes for this
> upcoming webrtc ? a list of patches ?

Probably, branch64 on https://hg.mozilla.org/users/dminor_mozilla.com/firefox-branchupdate/
Attachment 9013698 [details] points at the commit in user repo.

> Why is attachment 9015266 [details] a diff adding the configs as new files,
> while they exist in the repo ? because the diff is unreadable ?

Probably, removed in branch64 due to being out of date and confusing GnMozbuildWriter.

Comment 79

[Landry Breuil (:gaston)]

(In reply to Jan Beich from comment #78)
> (In reply to Landry Breuil (:gaston) from comment #77)
> > videoio.h has all the V4L2 defines & ioctls, so i suppose yes ? I don't see
> > why it should be removed, unless you have a real reason..
> 
> I wondered if OpenBSD may have added linux/ symlinks for compatibility.

yeaaah very unlikely to happen.

> > Is there a stash/branch somewhere with all the pending changes for this
> > upcoming webrtc ? a list of patches ?
> 
> Probably, branch64 on
> https://hg.mozilla.org/users/dminor_mozilla.com/firefox-branchupdate/
> Attachment 9013698 [details] points at the commit in user repo.

Meh. Need to learn some hg-fu to point at a different branch in a different repo... sigh.

> > Why is attachment 9015266 [details] a diff adding the configs as new files,
> > while they exist in the repo ? because the diff is unreadable ?
> 
> Probably, removed in branch64 due to being out of date and confusing
> GnMozbuildWriter.

Right, https://hg.mozilla.org/users/dminor_mozilla.com/firefox-branchupdate/rev/e66d7b9ae3fd removed them..

Comment 80

[Dan Minor [:dminor]]

Comment on attachment 9015266 [details]
BSD gn generated json/mozbuild files (compressed)

Are there any differences between the json files included here and the ones in your other patch? If so, please update the other patch.

I will apply your patches and regenerate the moz.build files prior to landing my patches, so I don't think this needs separate review.

Comment 81

[Dan Minor [:dminor]]

Comment on attachment 9015301 [details] [diff] [review]
Unbreak build on BSDs

These changes look fine to me. To get these landed, I'd prefer one patch with the changes to non-generated code (*.gn, *.cc, etc.) and another with the gn generated .json files.

If :gaston doesn't request any changes, and these .json files are the up to date versions, I can split the patch for you when I import it into my series. If you end up making changes, it would be great if you could split this up as part of that.

I'll then regenerate the moz.build files as part of my patch, so we don't end up landing two separate patches for the moz.build files.

Thanks! (and sorry for missing the platform_thread.cc changes)

Comment 82

[Landry Breuil (:gaston)]

I managed to build dminor's repo and jan's two patches on top with:

$ hg update -c f6976b1207d8
$ hg qpush && hg qpush
$ hg log -l3

changeset:   440484:b200ca736938
tag:         bug-1376873-refix
tag:         qtip
tag:         tip
user:        Jan Beich <[e-mail]>
date:        Mon Oct 08 10:21:07 2018 +0000
summary:     [PATCH] Bug 1376873 - Unbreak WebRTC 64 on BSDs. r=dminor f=gaston

changeset:   440483:b448bd995cca
tag:         bug-1376873-gnconfigs
tag:         qbase
user:        Jan Beich <[e-mail]>
date:        Sun Mar 11 20:47:57 2018 +0000
summary:     [PATCH] Bug 1376873 - Add WebRTC's gn-configs for BSDs. r=dminor f=gaston

changeset:   440482:f6976b1207d8
bookmark:    branch64
tag:         qparent
user:        Dan Minor <[e-mail]>
date:        Thu Sep 13 07:53:01 2018 -0400
summary:     Bug 1376873 - Add generated moz.build files; r=ng

now to actually test the resulting build...

Comment 83

[Dan Minor [:dminor]]

(In reply to Landry Breuil (:gaston) from comment #77)
> Why is attachment 9015266 [details] a diff adding the configs as new files,
> while they exist in the repo ? because the diff is unreadable ?

I had to remove the old json configs because they messed up moz.build generation when combined with the json configs generated for the update :/

Comment 84

[Landry Breuil (:gaston)]

Comment on attachment 9015301 [details] [diff] [review]
Unbreak build on BSDs

I've been able to record a video with the webcam on https://www.webrtc-experiment.com/RecordRTC/ - recording audio didnt work but i don't think that's a regression, as i tried last week with 63.0b11 and it didn't work either, i'll have to dig further.

As for the diff itself & the generated files, i glanced over it but i have to admit i wont dive into the build system madness again ( i had enough in bug #1437670), so many thanks jan for dealing with this.

Comment 85

[Jan Beich]

Screen/window/application capture demo also works.

(In reply to Dan Minor [:dminor] from comment #80)
> Are there any differences between the json files included here and the ones
> in your other patch? If so, please update the other patch.

Only missed path to bundled libevent config, introduced in WebRTC 64 update. Initially, I didn't notice due to --with-system-libevent which mimics downstream build.

(In reply to Dan Minor [:dminor] from comment #81)
> These changes look fine to me. To get these landed, I'd prefer one patch
> with the changes to non-generated code (*.gn, *.cc, etc.) and another with
> the gn generated .json files.

Sure. Generating *.json files is both error-prone (e.g., out of sync) and time-consuming (e.g., due to clean environment, proper emulation, workarounds for unrelated bustage), so I've focused on it first then tried to fix build later. Once fixed it was easier to patch *.json files than generate them again.

> If :gaston doesn't request any changes, and these .json files are the up to
> date versions, I can split the patch for you when I import it into my
> series.

Maybe squash into related commits to help other reviewers e.g.,
- configs for BSDs with configs for other platforms
- bustage fixes for BSDs with rollup of Mozilla-specific changes

Comment 86

[Jan Beich]

Attachment: Unbreak build on BSDs, v2

Oops, forgot to restore BSD fix to use > 1 thread by VP8/VP9 encoder. Adding "// WEBRTC_BSD" comment to prevent lossy rollup in future.

https://searchfox.org/mozilla-central/rev/80ac71c1c54a/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_info.cc#21-22,43-44
https://searchfox.org/mozilla-central/rev/80ac71c1c54a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc#476-479

Comment 87

[Jan Beich]

Attachment: Update gn generated json files for BSDs (compressed)

As requested by :dminor in comment 81:
- Moved *.json changes from attachment 9015301 [details] [diff] [review]
- Dropped moz.build changes to avoid conflicts in future

Comment 88

[Jan Beich]

Attachment: Unbreak build on BSDs, v3

- Switch to WEBRTC_BSD in stringutils.h like mozilla-central changeset 44fc6711973e to detect loss on WebRTC update in future
- Cherry-pick https://chromium.googlesource.com/external/webrtc/+/7c4dedade158%5E!/ instead of s/GLIBCXX/GLIBC/ to fix GCC build

Comment 89

[Dan Minor [:dminor]]

Comment on attachment 9016049 [details] [diff] [review]
Unbreak build on BSDs, v3

Review of attachment 9016049 [details] [diff] [review]:
-----------------------------------------------------------------

These changes look good to me. We'll try to get BSD support upstreamed this time so hopefully there will be less to unbreak next time, we have a bug on file: Bug 1498302.

Comment 90

[Dan Minor [:dminor]]

Comment on attachment 9015932 [details]
Update gn generated json files for BSDs (compressed)

Merged into my local set of patches and everything seems to be working well. Thanks!

Comment 91

[Dan Minor [:dminor]]

Getting ready to land this.

A try job with all tests for Linux64:

https://treeherder.mozilla.org/#/jobs?repo=try&revision=234c8e458dd0ad82be3ba3a904b9bb82b8c6fc0d&selectedJob=209198552

All platforms with media tests:

https://treeherder.mozilla.org/#/jobs?repo=try&revision=fdc19b32be8abe5d89bb35b969e866031f58a7c0

The only thing I find concerning is:

GECKO(1063) | Assertion failure: JS::ObjectIsNotGray(target), at /builds/worker/workspace/build/src/js/src/proxy/Wrapper.cpp:353

which has happened a few times and does not seem to be an existing intermittent.

Comment 92

[Pulsebot]

Pushed by [e-mail]:
https://hg.mozilla.org/integration/mozilla-inbound/rev/f40ca921b74b
Update webrtc to release 64; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/dda3c784b0c0
Update build files from tip of webrtc.org; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/2e1af73cdd2e
Allow gn to have a non-default target; r=chmanchester
https://hg.mozilla.org/integration/mozilla-inbound/rev/d78c5c50b6b8
Build system fixes; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/1f86d23e4806
Rollup of local modifications; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/bc92eac0522f
Rollup conflict fixes for audio/video code; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/e1ea09e84feb
Rollup conflict fixes for rtp_rtcp module; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/5d7a8864505d
Fix up logging in WebrtcLog.cpp; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/b59e1fca5c13
Fix include path in JsepSessionImpl.cpp; r=bwc
https://hg.mozilla.org/integration/mozilla-inbound/rev/d1f7d44e05a4
WebrtcGmpVideoCodec updates; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/d12ad321047d
VideoConduit updates; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/ffdbeba93fa6
WebrtcMediaDataDecoderCodec updates; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/6a6771656fe5
Update mtransport sigslot.h; r=bwc
https://hg.mozilla.org/integration/mozilla-inbound/rev/6a52b92ff037
Mediapipeline updates; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/19d9a30b300c
Updates to dom/media/, dom/media/systemservices and dom/media/webrtc; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/909f472c2e40
Add RtpPacketQueue; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/6eca45c83131
Disable or replace thread check assertions; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/b94935661622
Use Call interface in AudioConduit; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/9fe47f8ea1b8
Disable Mid support in RtpDemuxer; r=mjf
https://hg.mozilla.org/integration/mozilla-inbound/rev/2951112ade46
Remove CPULoadStateObserver; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/de3f5bd6846b
Fix GetRTCPSenderReport; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/0fc8c65e163e
Use audio/video sync groups; r=bwc
https://hg.mozilla.org/integration/mozilla-inbound/rev/da8d3eb163bd
Update videoconduit_unittests gtests; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/6b2919ed789b
Fix fake h.264 encoder; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/79010785539c
Remove Android java code from AudioConduit; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/086c722131f9
Update WebrtcMediaCodecVP8VideoCodec; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/622f78404b8a
Updates to Android video capture; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/d082eb8d80c6
Set default agc mode for Android in MediaEngineWebRTCAudio; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/58e47be76761
Updates to VideoConduit.cpp for OS X; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/176b82dccf4c
OS X desktop capture fixes; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/60fdf3598932
Change OS X logging include to prevent preprocessor multiple defines; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/acc06acb5342
Cherrypick rev 09133af36fba93691a22151765097c0bd581c1fa; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/381df4c6ae88
Make lock_block explicit; r=bwc
https://hg.mozilla.org/integration/mozilla-inbound/rev/0c3c2fc4646b
Allow single channel opus; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/9b82f16618b9
Fix warning in mean_variance_estimator.cc; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/5a45d2c706d6
Fix GetRTPStatistics; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/9674bbefc68e
Rollup of changes from cd8671232fd8 to 2a59b432d2bd; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/3d11b041baf2
Setup contributing sources extension in audio_receive_stream.cc; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/f15c4be0f19f
Keep track of active codec mode in VideoConduit; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/5d9c7a09616c
Don't define _USE_MATH_DEFINES in audio_processing.h; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/23b5fb1a2577
Make FunctionThatDoesNothing use const; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/afbd3a8108de
Fix up rotation in video_capture_impl.cc; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/26e09a6dee05
Use VP8SimulcastAdapter in VideoConduit; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/4efbe4d0050c
Update gn generated json files; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/7317f657a9a7
Unbreak WebRTC 64 build on BSDs. r=dminor f=gaston
https://hg.mozilla.org/integration/mozilla-inbound/rev/f029e9f3ceaf
Add WebRTC 64 gn-configs for BSDs. r=dminor f=gaston
https://hg.mozilla.org/integration/mozilla-inbound/rev/f9f879bd247c
Add generated moz.build files; r=ng

Comment 93

[Dan Minor [:dminor]]

RyanVM let me know that the assertion failure I saw above is Bug 1503666 and not related to the changes here.

Comment 94

[Byron Campen [:bwc]]

I did some digging into the intermittents on https://treeherder.mozilla.org/#/jobs?repo=mozilla-inbound&selectedJob=209457529&revision=f9f879bd247c8493c5f2236c1ddff739dc2f7c1b

Some OOM on Win7 pgo, and some CreateThread errors from platform_thread.cc on Win7 opt. Perhaps related to one another. That file changed quite a bit in this merge.

Comment 95

[Jan Beich]

Fails to build on FreeBSD. So, gn-configs have changed but no one said so.

$ ./mach build
[...]
 1:17.10 Creating config.status
 1:17.21 Reticulating splines...
 1:17.70 Traceback (most recent call last):
 1:17.70   File "/configure.py", line 123, in <module>
 1:17.70     sys.exit(main(sys.argv))
 1:17.70   File "/configure.py", line 34, in main
 1:17.70     return config_status(config)
 1:17.70   File "/configure.py", line 118, in config_status
 1:17.70     return config_status(args=[], **encode(sanitized_config, encoding))
 1:17.70   File "/python/mozbuild/mozbuild/config_status.py", line 143, in config_status
 1:17.70     definitions = list(definitions)
 1:17.70   File "/python/mozbuild/mozbuild/frontend/emitter.py", line 184, in emit
 1:17.70     objs = list(emitfn(out))
 1:17.70   File "/python/mozbuild/mozbuild/frontend/emitter.py", line 1196, in emit_from_context
 1:17.70     for obj in self._handle_linkables(context, passthru, generated_files):
 1:17.70   File "/python/mozbuild/mozbuild/frontend/emitter.py", line 874, in _handle_linkables
 1:17.70     'exist: \'%s\'' % (symbol, full_path), context)
 1:17.70 mozbuild.frontend.reader.SandboxValidationError:
 1:17.71 ==============================
 1:17.71 FATAL ERROR PROCESSING MOZBUILD FILE
 1:17.71 ==============================
 1:17.71 The error occurred while processing the following file or one of the files it includes:
 1:17.71     /media/webrtc/trunk/webrtc/common_audio/common_audio_gn/moz.build
 1:17.71 The error occurred when validating the result of the execution. The reported error is:
 1:17.71     File listed in UNIFIED_SOURCES does not exist: '/media/webrtc/trunk/webrtc/common_audio/resampler/speex_resampler.cc'
 1:17.86 *** Fix above errors and then restart with\
 1:17.86                "./mach build"
 1:17.86 gmake: *** [client.mk:115: configure] Error 1

Comment 96

[Jan Beich]

Attachment: Sync BSD configs

Fixes comment 95.

Comment 97

[Pulsebot]

Backout by [e-mail]:
https://hg.mozilla.org/integration/mozilla-inbound/rev/63f135d2db22
Backed out 48 changesets (bug 1376873, bug 1502992) for causing windows 7 wpt crashes on several files.

Comment 98

[Cosmin Sabou [:CosminS]]

Backed out for causing windows 7 pgo & opt wpt crashes on several files.

Push with failures: https://treeherder.mozilla.org/#/jobs?repo=mozilla-inbound&resultStatus=pending%2Crunning%2Csuccess%2Ctestfailed%2Cbusted%2Cexception&searchStr=wpt10%2Cwindows%2C7&tochange=63f135d2db22375b51cf0ded745279afe2188022&fromchange=f9f879bd247c8493c5f2236c1ddff739dc2f7c1b&selectedJob=209497996

Failure log: https://treeherder.mozilla.org/logviewer.html#?job_id=209497996&repo=mozilla-inbound&lineNumber=28956

INFO - TEST-UNEXPECTED-CRASH | /webrtc/RTCRtpTransceiver.https.html | expected OK 
This initially failed on this test and after bwc and jib looked over it and commented on the bug but no fix was landed Aryx disabled the test here (https://bugzilla.mozilla.org/show_bug.cgi?id=1502992#c2) so we can get rid on the oranges in the tree.

That test was disabled but then the wpts started failing on other files like:
TEST-UNEXPECTED-CRASH | /webrtc/idlharness.https.window.html | expected OK - https://treeherder.mozilla.org/logviewer.html#?job_id=209560637&repo=mozilla-inbound&lineNumber=29355
TEST-UNEXPECTED-CRASH | /webrtc/RTCTrackEvent-constructor.html | expected OK - https://treeherder.mozilla.org/logviewer.html#?job_id=209568886&repo=mozilla-inbound&lineNumber=28895
TEST-UNEXPECTED-CRASH | /webrtc/no-media-call.html | expected OK - https://treeherder.mozilla.org/logviewer.html#?job_id=209569090&repo=mozilla-inbound&lineNumber=29484

as you can see on this disable push made by Aryx: https://treeherder.mozilla.org/#/jobs?repo=mozilla-inbound&resultStatus=pending%2Crunning%2Csuccess%2Ctestfailed%2Cbusted%2Cexception&searchStr=wpt10%2Cwindows%2C7&selectedJob=209569459&revision=b9db1780bb9c56883527709e75f58c8c6a06bcb6

Decided to backout the whole patch + the disable patch in order to keep the trees green.

Backout link: https://hg.mozilla.org/integration/mozilla-inbound/rev/63f135d2db22375b51cf0ded745279afe2188022

Comment 99

[Dan Minor [:dminor]]

(In reply to Byron Campen [:bwc] from comment #94)
> I did some digging into the intermittents on
> https://treeherder.mozilla.org/#/jobs?repo=mozilla-
> inbound&selectedJob=209457529&revision=f9f879bd247c8493c5f2236c1ddff739dc2f7c
> 1b
> 
> Some OOM on Win7 pgo, and some CreateThread errors from platform_thread.cc
> on Win7 opt. Perhaps related to one another. That file changed quite a bit
> in this merge.

Thank you for having a look. I'm guessing CreateThread is also a OOM, it's probably failing to allocate the stack for the new thread. They pass in STACK_SIZE_PARAM_IS_A_RESERVATION which should mean they are not committing the full 1M; I'm not sure how much is actually committed when the thread is created.

If this were a memory leak I would hope that it would be picked up by one of the ASAN jobs. It seems likely we're just close to OOM on win32 and sometimes we edge over. Perhaps with the branch update we're holding onto resources for longer that we did in the past.

Comment 100

[Joel Maher ( :jmaher ) (UTC -8)]

this bug when landed caused an installer size regression:
== Change summary for alert #17347 (as of Fri, 02 Nov 2018 19:28:53 GMT) ==

Regressions:

  0%  installer size linux32 pgo             66,977,011.42 -> 67,251,562.17
  0%  installer size osx-cross opt           74,041,973.92 -> 74,321,975.08
  0%  installer size windows2012-32 pgo      64,817,045.92 -> 65,054,935.25
  0%  installer size windows2012-64 pgo      67,822,630.00 -> 68,069,479.42
  0%  installer size linux64 pgo             67,171,045.17 -> 67,389,145.83

For up to date results, see: https://treeherder.mozilla.org/perf.html#/alerts?id=17347

Comment 101

[Dan Minor [:dminor]]

I'm cautiously optimistic that reducing the stack size here [1] from 1M to 512kb or 256kb will fix the OOM [2] [3]. For reference, we default to 256kb in threads allocated from gecko [4].

[1] https://hg.mozilla.org/integration/mozilla-inbound/file/f9f879bd247c/media/webrtc/trunk/webrtc/rtc_base/platform_thread.cc#l234
[2] https://treeherder.mozilla.org/#/jobs?repo=try&revision=7f093f39b381836959dd1e01701424e9db84c543
[3] https://treeherder.mozilla.org/#/jobs?repo=try&revision=68910711caa132b6120fd8459cbb067fa2b59cf2
[4] https://searchfox.org/mozilla-central/rev/7c848ac7630df5baf1314b0c03e015683599efb9/xpcom/threads/nsIThreadManager.idl#52

Comment 102

[Dan Minor [:dminor]]

(In reply to Joel Maher ( :jmaher ) (UTC-4) from comment #100)
> this bug when landed caused an installer size regression:
> == Change summary for alert #17347 (as of Fri, 02 Nov 2018 19:28:53 GMT) ==
> 
> Regressions:
> 
>   0%  installer size linux32 pgo             66,977,011.42 -> 67,251,562.17
>   0%  installer size osx-cross opt           74,041,973.92 -> 74,321,975.08
>   0%  installer size windows2012-32 pgo      64,817,045.92 -> 65,054,935.25
>   0%  installer size windows2012-64 pgo      67,822,630.00 -> 68,069,479.42
>   0%  installer size linux64 pgo             67,171,045.17 -> 67,389,145.83
> 
> For up to date results, see:
> https://treeherder.mozilla.org/perf.html#/alerts?id=17347

Hi Joel, are these increases significant enough to warrant investigation? We already modify the build configuration to avoid building portions of the webrtc.org code base we don't use, so these increases most likely reflect additional code added by upstream and/or the fact that we're now using more of the webrtc.org code itself. I'm sure there are further optimizations that can be made to the build configuration but my guess is that it would end up being very time consuming for limited improvements to installer size.

Comment 103

[Joel Maher ( :jmaher ) (UTC -8)]

I don't know if they are concerning or not, it looks like 250K increase in size- :sylvestre- you are the contact for this data point, I will let you weigh in here.

Comment 104

[Dan Minor [:dminor]]

(In reply to Jan Beich from comment #95)
> Fails to build on FreeBSD. So, gn-configs have changed but no one said so.
> 
Yes, we dropped the speex_resampler patch during review, but because review comments are no longer synced back to Bugzilla, it's not at all obvious that that happened. Thank you for your fix. I'll roll it into the existing BSD gn-config patch prior to landing.

Comment 105

[Dan Minor [:dminor]]

Try job with platform_thread.cc stack size reduced for all platforms:

https://treeherder.mozilla.org/#/jobs?repo=try&selectedJob=210064580&revision=6e11e5491872c455ad414e0d8f224a4d548b187e

Comment 106

[Dan Minor [:dminor]]

Attachment: Bug 1376873 - Reduce thread stack size in platform_thread.cc; r=bwc

The current default stack size of 1M results in intermittent OOMs on win32
builds while running web-platform tests. The value of 256k was chosen for
consistency with the default value used elsewhere in Gecko, which is defined in
nsIThreadManager.idl.

Comment 107

[Sylvestre Ledru [:Sylvestre]]

(In reply to Dan Minor [:dminor] from comment #102)
> 
> Hi Joel, are these increases significant enough to warrant investigation?
I would say no here.
However, I think we should spend some time on investigating the webrtc code built and shipped
which is useless.

The zero code coverage report shows webrtc as one of the biggest offender.
https://marco-c.github.io/code-coverage-reports/#media/webrtc/trunk/webrtc
521 files in webrtc upstream have 0 code coverage which either means that we lack tests (even unit test) or that it is useless code for Firefox.

Comment 108

[Dan Minor [:dminor]]

I've rebased on central as of today and things look good on try: https://treeherder.mozilla.org/#/jobs?repo=try&revision=6d2b188f76f5f72f4f98ff723ee907a3ea4b3d2f. I'll be relanding this soon.

Comment 109

[Pulsebot]

Pushed by [e-mail]:
https://hg.mozilla.org/integration/mozilla-inbound/rev/ef84e45f28d9
Update webrtc to release 64; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/8c08a5bb8a99
Update build files from tip of webrtc.org; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/0e2aee1f2033
Allow gn to have a non-default target; r=chmanchester
https://hg.mozilla.org/integration/mozilla-inbound/rev/4989aa9ad7e0
Build system fixes; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/28b57e3ba51d
Rollup of local modifications; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/93eec571640e
Rollup conflict fixes for audio/video code; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/50f89f4e45b0
Rollup conflict fixes for rtp_rtcp module; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/e84c60e2c937
Fix up logging in WebrtcLog.cpp; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/756ab3fad62b
Fix include path in JsepSessionImpl.cpp; r=bwc
https://hg.mozilla.org/integration/mozilla-inbound/rev/8b5e947a1408
WebrtcGmpVideoCodec updates; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/2fb84f9968af
VideoConduit updates; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/d1535acfa6fe
WebrtcMediaDataDecoderCodec updates; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/a6d993a9b3bc
Update mtransport sigslot.h; r=bwc
https://hg.mozilla.org/integration/mozilla-inbound/rev/3b4d25ce432d
Mediapipeline updates; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/950e95f79312
Updates to dom/media/, dom/media/systemservices and dom/media/webrtc; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/941a85e51309
Add RtpPacketQueue; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/df960ea2d29e
Disable or replace thread check assertions; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/5d2899adc305
Use Call interface in AudioConduit; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/b3ba8452e771
Disable Mid support in RtpDemuxer; r=mjf
https://hg.mozilla.org/integration/mozilla-inbound/rev/a61420479a31
Remove CPULoadStateObserver; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/b55b0368d9f2
Fix GetRTCPSenderReport; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/551533329f3f
Use audio/video sync groups; r=bwc
https://hg.mozilla.org/integration/mozilla-inbound/rev/0cd31b95b9de
Update videoconduit_unittests gtests; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/949c043fb183
Fix fake h.264 encoder; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/549eae0865b6
Remove Android java code from AudioConduit; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/334776e8e614
Update WebrtcMediaCodecVP8VideoCodec; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/1e6d72de0587
Updates to Android video capture; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/ae6e3a39494d
Set default agc mode for Android in MediaEngineWebRTCAudio; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/7f5ae8174a2b
Updates to VideoConduit.cpp for OS X; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/02c038eca65c
OS X desktop capture fixes; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/7f5866295b84
Change OS X logging include to prevent preprocessor multiple defines; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/cca0a311c33b
Cherrypick rev 09133af36fba93691a22151765097c0bd581c1fa; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/200849cab86a
Make lock_block explicit; r=bwc
https://hg.mozilla.org/integration/mozilla-inbound/rev/92a7c3eee9f0
Allow single channel opus; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/cb6c555654c0
Fix warning in mean_variance_estimator.cc; r=padenot
https://hg.mozilla.org/integration/mozilla-inbound/rev/0c885324ba6f
Fix GetRTPStatistics; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/bb450c30e765
Rollup of changes from cd8671232fd8 to 2a59b432d2bd; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/e3fdbe8c23f3
Setup contributing sources extension in audio_receive_stream.cc; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/12fbf0e80d9e
Keep track of active codec mode in VideoConduit; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/846c5bd26264
Don't define _USE_MATH_DEFINES in audio_processing.h; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/a9337131657c
Make FunctionThatDoesNothing use const; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/c56cb66f8651
Fix up rotation in video_capture_impl.cc; r=pehrsons
https://hg.mozilla.org/integration/mozilla-inbound/rev/7e0b405f7a1d
Use VP8SimulcastAdapter in VideoConduit; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/98a403f60b8c
Update gn generated json files; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/a636ecdcb91a
Unbreak WebRTC 64 build on BSDs. r=dminor f=gaston
https://hg.mozilla.org/integration/mozilla-inbound/rev/fdadefc1af6e
Add WebRTC 64 gn-configs for BSDs. r=dminor f=gaston
https://hg.mozilla.org/integration/mozilla-inbound/rev/8d68d175a589
Add generated moz.build files; r=ng
https://hg.mozilla.org/integration/mozilla-inbound/rev/e83c311e5293
Reduce thread stack size in platform_thread.cc; r=bwc

Comment 110

[Daniel Varga [:dvarga]]

https://hg.mozilla.org/mozilla-central/rev/ef84e45f28d9
https://hg.mozilla.org/mozilla-central/rev/8c08a5bb8a99
https://hg.mozilla.org/mozilla-central/rev/0e2aee1f2033
https://hg.mozilla.org/mozilla-central/rev/4989aa9ad7e0
https://hg.mozilla.org/mozilla-central/rev/28b57e3ba51d
https://hg.mozilla.org/mozilla-central/rev/93eec571640e
https://hg.mozilla.org/mozilla-central/rev/50f89f4e45b0
https://hg.mozilla.org/mozilla-central/rev/e84c60e2c937
https://hg.mozilla.org/mozilla-central/rev/756ab3fad62b
https://hg.mozilla.org/mozilla-central/rev/8b5e947a1408
https://hg.mozilla.org/mozilla-central/rev/2fb84f9968af
https://hg.mozilla.org/mozilla-central/rev/d1535acfa6fe
https://hg.mozilla.org/mozilla-central/rev/a6d993a9b3bc
https://hg.mozilla.org/mozilla-central/rev/3b4d25ce432d
https://hg.mozilla.org/mozilla-central/rev/950e95f79312
https://hg.mozilla.org/mozilla-central/rev/941a85e51309
https://hg.mozilla.org/mozilla-central/rev/df960ea2d29e
https://hg.mozilla.org/mozilla-central/rev/5d2899adc305
https://hg.mozilla.org/mozilla-central/rev/b3ba8452e771
https://hg.mozilla.org/mozilla-central/rev/a61420479a31
https://hg.mozilla.org/mozilla-central/rev/b55b0368d9f2
https://hg.mozilla.org/mozilla-central/rev/551533329f3f
https://hg.mozilla.org/mozilla-central/rev/0cd31b95b9de
https://hg.mozilla.org/mozilla-central/rev/949c043fb183
https://hg.mozilla.org/mozilla-central/rev/549eae0865b6
https://hg.mozilla.org/mozilla-central/rev/334776e8e614
https://hg.mozilla.org/mozilla-central/rev/1e6d72de0587
https://hg.mozilla.org/mozilla-central/rev/ae6e3a39494d
https://hg.mozilla.org/mozilla-central/rev/7f5ae8174a2b
https://hg.mozilla.org/mozilla-central/rev/02c038eca65c
https://hg.mozilla.org/mozilla-central/rev/7f5866295b84
https://hg.mozilla.org/mozilla-central/rev/cca0a311c33b
https://hg.mozilla.org/mozilla-central/rev/200849cab86a
https://hg.mozilla.org/mozilla-central/rev/92a7c3eee9f0
https://hg.mozilla.org/mozilla-central/rev/cb6c555654c0
https://hg.mozilla.org/mozilla-central/rev/0c885324ba6f
https://hg.mozilla.org/mozilla-central/rev/bb450c30e765
https://hg.mozilla.org/mozilla-central/rev/e3fdbe8c23f3
https://hg.mozilla.org/mozilla-central/rev/12fbf0e80d9e
https://hg.mozilla.org/mozilla-central/rev/846c5bd26264
https://hg.mozilla.org/mozilla-central/rev/a9337131657c
https://hg.mozilla.org/mozilla-central/rev/c56cb66f8651
https://hg.mozilla.org/mozilla-central/rev/7e0b405f7a1d
https://hg.mozilla.org/mozilla-central/rev/98a403f60b8c
https://hg.mozilla.org/mozilla-central/rev/a636ecdcb91a
https://hg.mozilla.org/mozilla-central/rev/fdadefc1af6e
https://hg.mozilla.org/mozilla-central/rev/8d68d175a589
https://hg.mozilla.org/mozilla-central/rev/e83c311e5293

Comment 111

[Dan Minor [:dminor]]

This broke support for opus maxplaybackrate, which I noticed while writing unit tests for AudioConduit over in Bug 1406941. I have a fix there.