Reason for revert:
This CL are breaking some of the Android buildbots in Chromium.
The CL will need be be revised to exclude the Android platform.
Original issue's description:
> Added a bitexactness test for the gain controller in the audio processing module.
>
> BUG=webrtc:5339
>
> Committed: https://crrev.com/a49dc36976da44f3d6d75aed2dcab93fe14fc3a0
> Cr-Commit-Position: refs/heads/master@{#12124}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5339
Review URL: https://codereview.webrtc.org/1829323002
Cr-Commit-Position: refs/heads/master@{#12127}
where the render and capture locks were
acquired in the wrong order.
BUG=webrtc:5339
Review URL: https://codereview.webrtc.org/1827013002
Cr-Commit-Position: refs/heads/master@{#12122}
Muting/unmuting is triggered in the PeerConnection API by calling setEnable() on an audio track.
BUG=webrtc:5671
Review URL: https://codereview.webrtc.org/1810413002
Cr-Commit-Position: refs/heads/master@{#12121}
Reason for revert:
We will make it possible to link to BoringSSL for WebRTC's usages of the crypto APIs and OpenSSL for other usages in the same binary. Once that is completed, we will reland this.
Original issue's description:
> Remove code interfacing legacy openssl.
>
> BUG=webrtc:5664
>
> Committed: https://crrev.com/4cd331beade6de16c073dcdaf89c4e038bdbf73f
> Cr-Commit-Position: refs/heads/master@{#12041}
TBR=tommi@webrtc.org,davidben@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5664
Review URL: https://codereview.webrtc.org/1828773003 .
Cr-Commit-Position: refs/heads/master@{#12117}
The track state should be implicitly set by the underlying source.
This removes the public method and cleans up how AudioRtpReceiver is created. Further more it cleans up how the RtpReceivers are destroyed.
Note that this cl depend on https://codereview.webrtc.org/1790633002.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1816143002
Cr-Commit-Position: refs/heads/master@{#12115}
Also include that in the stun-ping request as part of the
network-info attribute.
Change the network cost to be 16 bits.
BUG=
Review URL: https://codereview.webrtc.org/1815473002
Cr-Commit-Position: refs/heads/master@{#12110}
We shouldn't make external users define this flag to use our file.
BUG=
Review URL: https://codereview.webrtc.org/1825713003
Cr-Commit-Position: refs/heads/master@{#12106}
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}
Also removes unused track states kLive and kFailed.
Since this also required a Video source to exist in all unit tests that create a track, a FakeVideoTrackSource is added and used in tests.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1790633002
Cr-Commit-Position: refs/heads/master@{#12098}
This will allow a sender to stop/start sending media on the
application's demand.
Among other things, this can allow an application to set a track on a
sender while the encoding(s) are inactive, allowing the encoder to be
initialized for that track, then later set the encodings to "active"
to instantly start sending the track.
Review URL: https://codereview.webrtc.org/1822923002
Cr-Commit-Position: refs/heads/master@{#12094}
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}
initWithFactory: is clumsy and makes classes difficult to mock out in
tests. By keeping methods on the factory, we can simply mock out the
factory's methods instead.
We can consider adding regular Obj-C like ctors if we move to making
the factory a singleton, but that requires further discussion.
BUG=
R=haysc@webrtc.org, hjon@webrtc.org
Review URL: https://codereview.webrtc.org/1820193002 .
Cr-Commit-Position: refs/heads/master@{#12089}
Un-breaks peerconnection_client which would instantly use-after-free on
an allocated VCM because it wasn't building a scoped_refptr so all
references to the VCM were dropped.
BUG=webrtc:5229
TBR=tommi@webrtc.org
TEST=Run peerconnection_client locally, verify that there's no crash.
Review URL: https://codereview.webrtc.org/1817953005 .
Cr-Commit-Position: refs/heads/master@{#12088}
This CL is expected to lower goog_max_decode_ms and total_delay_incl_network/receiver_time for screenshare.
Reason for revert:
This CL did not cause the unexpected goog_encode_usage_percent and goog_avg_encode_ms perf changes.
Original issue's description:
> Revert of VCMCodecTimer: Change filter from max to 95th percentile (patchset #5 id:180001 of https://codereview.webrtc.org/1742323002/ )
>
> Reason for revert:
> Caused unexpected perf stats changes, see http://crbug/594575.
>
> Original issue's description:
> > VCMCodecTimer: Change filter from max to 95th percentile
> >
> > The purpose with this change is to make the filter more robust against anomalies. googMaxDecodeMs is expected to drop a litte by this.
> >
> > BUG=b/27306053
> >
> > Committed: https://crrev.com/4bf0c717740d1834e810ea5f32b3c4306c64235f
> > Cr-Commit-Position: refs/heads/master@{#11952}
>
> TBR=stefan@webrtc.org,philipel@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=594575,b/27306053
>
> Committed: https://crrev.com/c4a74e95b545f4752d4e72961ac03c1380d4bc1f
> Cr-Commit-Position: refs/heads/master@{#12018}
TBR=stefan@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=594575,b/27306053
Review URL: https://codereview.webrtc.org/1824763003
Cr-Commit-Position: refs/heads/master@{#12087}
Reason for revert:
Breaks RTCVideoEncoder which has incorrect assumptions on where InitEncode etc. is called from. Temporarily reverting until RTCVideoEncoder has been updated.
Original issue's description:
> Initialize/configure video encoders asychronously.
>
> Greatly speeds up setRemoteDescription() by moving encoder initialization
> off the main worker thread, which is free to move onto gathering ICE
> candidates and other tasks while InitEncode() is performed. It also
> un-blocks PeerConnection GetStats() which is no longer blocked on
> encoder initialization.
>
> BUG=webrtc:5410
> R=stefan@webrtc.org
>
> Committed: fb647a67beR=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:595274, chromium:595308, webrtc:5410
Review URL: https://codereview.webrtc.org/1821983002 .
Cr-Commit-Position: refs/heads/master@{#12086}
rtc::ThreadManager::ThreadManager() calls rtc::InitCocoaMultiThreading()
on iOS so add base/maccocoathreadhelper.mm to the file to build on iOS.
Fixes the following linker error:
Undefined symbols for architecture x86_64:
"rtc::InitCocoaMultiThreading()", referenced from:
rtc::ThreadManager::ThreadManager() in librtc_base.a(thread.o)
BUG=459705
NOTRY=True
Review URL: https://codereview.webrtc.org/1810373003
Cr-Commit-Position: refs/heads/master@{#12085}
From an earlier CL, we start to feed UpdateLevel() with power instead of energy. I found that UpdateLevel() is still taking the input as energy and normalize it. This CL fixes this.
The earlier CL is
https://codereview.webrtc.org/1542573002/
BUG=
Review URL: https://codereview.webrtc.org/1810773003
Cr-Commit-Position: refs/heads/master@{#12084}
The intended signalling from StartTimer() to Process() is that
created_at_.tv_sec is set to 0, and timer_event_->Set() is then called
in order to wake the process thread from timer_event_->Wait(). When this
happens the process thread will return early and the run Process()
again. This time it will pick up created_at_.tv_sec = 0 and run a new
Wait() call with the desired end time.
However if the process thread was NOT blocking on timer_event_->Wait()
when timer_event_->Set() was called from StartTimer() it will mean that
the first call to timer_event_->Wait() from Process(), AFTER the new
time has been configured (count_ = 1), will return early.
If the timer is not periodic it means that Set() will never be called,
and any calls will Wait() will block until the time out.
The solution is to always reset the event in timer_event_ on the first
call to timerEvent_->Wait(), after a timer has started.
Also some general cleanup.
BUG=
Review URL: https://codereview.webrtc.org/1812533002
Cr-Commit-Position: refs/heads/master@{#12082}
Applications can choose to decide when to give up control of the
AVAudioSession to WebRTC. Otherwise, behavior should be
unchanged.
Adds a toggle to AppRTCDemo so developers can see the different
paths.
BUG=
R=haysc@webrtc.org
Review URL: https://codereview.webrtc.org/1822543002 .
Cr-Commit-Position: refs/heads/master@{#12080}
webrtc/base/base.gyp unconditionally set SSL_USE_OPENSSL and
HAVE_OPENSSL_SSL_H, fix webrtc/base/BUILD.gn to do the same.
Better implementation than https://codereview.webrtc.org/1441323002
to fix the same underlying issue (i.e. compilation on iOS).
BUG=459705
Review URL: https://codereview.webrtc.org/1812213002
Cr-Commit-Position: refs/heads/master@{#12078}
ScreenCapturerMac wasn't handling the following two cases properly
which could cause crashes:
1. CGDisplayCreateImage() returns image with depth other than 32-bit
2. CGDisplayCreateImage() returns image with dimensions different
from expected (e.g. when screen resolution is being changed).
I suspect that (2) was causing the linked bug.
BUG=crbug.com/504927
R=jiayl@webrtc.org
Review URL: https://codereview.webrtc.org/1816723002 .
Cr-Commit-Position: refs/heads/master@{#12077}
