Adds logging of:
- video stats that are recorded when a stream is removed
- bitrate stats that are recorded at the end of a call
- initial bwe rampup stats
BUG=
Review URL: https://codereview.webrtc.org/1788783002
Cr-Commit-Position: refs/heads/master@{#12133}
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}
