JSEP section 5.1.3 states that:
Any profile matching the following patterns MUST be accepted:
"RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"
NOTRY=True
BUG=webrtc:5638
Review URL: https://codereview.webrtc.org/1880913002
Cr-Commit-Position: refs/heads/master@{#12338}
Reason for revert:
This is preventing a WebRTC roll into Chromium. See https://codereview.chromium.org/1877263003/
Original issue's description:
> Fix screen capturers to initialize on the same thread on which Start() is called.
>
> Previously screen capturers were initialized when they are created.
> This means that in the CRD host they were initialized on the thread
> that's different from the thread on which they are used. Because of this
> on Linux the host was using XErrorTrap() on two different threads and
> this is not supported. Now ScreenCapturer implementations always
> initialize themselves on the thread on which Start() is called.
>
> Also added ThreadChecker to make sure the capturers are always called
> from the same thread.
>
> BUG=600432
>
> Committed: https://crrev.com/e8d4b7d8a3d298438a2ebd9ee8d5aa71f42cf033
> Cr-Commit-Position: refs/heads/master@{#12285}
TBR=jamiewalch@chromium.org,sergeyu@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=600432
Review URL: https://codereview.webrtc.org/1882083002
Cr-Commit-Position: refs/heads/master@{#12334}
This CL generates FMTP parameters that allow H.264 interoperation
with Firefox for the default codec list.
BUG=chromium:591971
Review URL: https://codereview.webrtc.org/1880963002
Cr-Commit-Position: refs/heads/master@{#12333}
This will allow us to remove this flag from the "how to run the tests"
instructions, and also the bot recipes.
BUG=599919
NOTRY=True
Review URL: https://codereview.webrtc.org/1880563002
Cr-Commit-Position: refs/heads/master@{#12332}
Reason for revert:
Speculative revert: want to see if this causes the regression in https://crbug.com/602621
Original issue's description:
> Make QualityScaler more responsive to downgrades.
>
> Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
> windows for going up quickly in the beginning of a call (before any
> downscaling happens due to bad quality).
>
> BUG=webrtc:5678
> R=glaznev@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/85829fd90cc4e7a91c9857921b19e8fc126aeb60
> Cr-Commit-Position: refs/heads/master@{#12219}
TBR=glaznev@webrtc.org,stefan@webrtc.org,pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5678
NOTRY=true
Review URL: https://codereview.webrtc.org/1880103002
Cr-Commit-Position: refs/heads/master@{#12331}
Reason for revert:
Broke import. Implementations of the old interface still exists somewhere.
Original issue's description:
> Remove the deprecated EncodeInternal interface from AudioEncoder
>
> Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
>
> BUG=webrtc:5591
>
> Committed: https://crrev.com/5222d315dbea8f3563c100cc9f2451907f70b05f
> Cr-Commit-Position: refs/heads/master@{#12329}
TBR=kwiberg@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5591
Review URL: https://codereview.webrtc.org/1883543002
Cr-Commit-Position: refs/heads/master@{#12330}
Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
BUG=webrtc:5591
Review URL: https://codereview.webrtc.org/1864993002
Cr-Commit-Position: refs/heads/master@{#12329}
If a very large frame is sent (high res slide change) when the available
send bitrate is very low, the it might take many seconds before any new
frames are emitted as the accrued debt will take time to pay off.
Add a bailout, so that if a frame hasn't been sent for 2 seconds, cancel
the debt immediately, even if the target bitrate is then exceeded.
BUG=webrtc:5750
Review URL: https://codereview.webrtc.org/1869003002
Cr-Commit-Position: refs/heads/master@{#12328}
This eliminates some instances rtc:Optional and makes the code
simpler. No changes in defaults or other behaviour are intended.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1818033002
Cr-Commit-Position: refs/heads/master@{#12326}
The QuicTransportChannel now creates outgoing QUIC streams
for sending a message, and incoming QUIC streams for
receiving a message. It also signals when the QUIC connection
closes.
Split from CL https://codereview.webrtc.org/1844803002/.
BUG=
Review URL: https://codereview.webrtc.org/1856513002
Cr-Commit-Position: refs/heads/master@{#12323}
Added packet type RpsiItem to destinguish parsed rpsi header and Rpsi body
preventing handling two half-valid (header-only) rpsi packets as one valid,
making test parser calculate rpsi packet once instead of twice.
Added check padding bits doesn't exceed native bit string length.
Marking rpsi received moved after it is validated.
BUG=600977
Review URL: https://codereview.webrtc.org/1880443002
Cr-Commit-Position: refs/heads/master@{#12318}
This logic currently prevents loopback calls on Nexus 5X when it's
slightly overloaded to maintain input framerate since encoding at ~25fps
with one framedrop results in >70ms between frames naturally.
With this change applied Nexus 5X can maintain ~25fps both in and out
without building excessive latency (>2 frames) (this is now covered by
CPU adaptation outside the codec wrapper).
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1854413004 .
Cr-Commit-Position: refs/heads/master@{#12317}
Support for reading the newly added fields will be added in a another CL.
BUG=webrtc:5759, webrtc:5724
Review URL: https://codereview.webrtc.org/1878533002
Cr-Commit-Position: refs/heads/master@{#12316}
Adds tracing to AudioDeviceModuleImpl::InitRecording and
AudioDeviceModuleImpl::StartRecording to visualize that they are a
significant part of startup time in performance recordings.
BUG=webrtc:5723
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1874983003 .
Cr-Commit-Position: refs/heads/master@{#12315}
FakeVideoRenderer, only registering itself on a
VideoSourceInterface on construction and removing itself on
destruction. Let it inherit FakeVideoRenderer, instead of
proxying all methods.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1828173002
Cr-Commit-Position: refs/heads/master@{#12313}
This is done in order to allow the ringbuffer to be recorded using protobufs.
The actual recording will be added in other CLs.
BUG=webrtc:5724
Review URL: https://codereview.webrtc.org/1862513007
Cr-Commit-Position: refs/heads/master@{#12308}
-changed filenames to *.cc
-fixed issues with implicit casts causing build errors.
All other CL warnings were ignored as the original
code is not changed, merely moved.
BUG=webrtc:5724
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1857153002
Cr-Commit-Position: refs/heads/master@{#12307}
rate that is higher than the specified rate, and not
the closest native rate the need for a name change has been
raised. This CL addresses that.
BUG=webrtc:5298
Review URL: https://codereview.webrtc.org/1863023002
Cr-Commit-Position: refs/heads/master@{#12302}
With this CL, legacy OpenSSL should work again.
BUG=webrtc:5714
Review URL: https://codereview.webrtc.org/1868033005
Cr-Commit-Position: refs/heads/master@{#12300}
This change adds the Objective C API functions to get and set RtpSender's
RtpParameters, which allows setting bitrate limits for audio and video and
turning off RtpSenders to pre-initialize the encoder.
This CL adds only the smallest set of methods required to support bitrate
limiting - there is no way to create an RtpSender, for example, or to set
its track. The only supported functionality is this:
RTCPeerConnection.senders - a read-only property returning the array
of all RTCRtpSenders for the connection.
RTCRtpSender.parameters - a read-only property returning the current
parameters
RTCRtpSender.setParameters: - a method to change the parameters.
RTCRtpSender.track - a read-only property returning the
RTCMediaStreamTrack corresponding to the sender. It is necessary
to be able to identify RTCRtpSenders for video and audio. The
track object is of the base RTCMediaStreamTrack type, not of the
specific subclass for audio and video - just like it is in the
Java API.
BUG=
Review URL: https://codereview.webrtc.org/1854393002
Cr-Commit-Position: refs/heads/master@{#12297}
This is a follow up to https://codereview.webrtc.org/1859933002 to change this test also to use a separate worker thread.
PeerConnectionEndToEndTest currently use the current thread both as a signaling thread and a worker thread. Although convenient while debugging, it can also hide real bugs. An example is https://codereview.webrtc.org/1766653002/#ps420001 where the worker thread is deadlocked in the track proxy due to that the worker thread waits for the signaling thread but the proxy in turns invokes the worker thread..... That bug was only discovered on Android.
BUG= webrtc:5426
Review URL: https://codereview.webrtc.org/1860423002
Cr-Commit-Position: refs/heads/master@{#12295}
This change builds on top of the refactoring in https://codereview.webrtc.org/1841083008/, and enables WebRTC client applications to control the max send bitrate for every audio stream through RtpParameters.
The AudioSendStream now stores the last codec spec, and whenever a global or per-stream bitrate limit changes, the effective limit (smaller of the two) is recomputed and the codec is reconfigured with that bitrate.
TBR=pthatcher
BUG=
Review URL: https://codereview.webrtc.org/1847353004
Cr-Commit-Position: refs/heads/master@{#12290}
