This CL doesn't start *using* a=bundle-only; it just adds support for
parsing it. We need to do this first, because otherwise old versions of
WebRTC will interpret a zero port value as a rejected m= section.
BUG=webrtc:4674
Review-Url: https://codereview.webrtc.org/2562183002
Cr-Commit-Position: refs/heads/master@{#15558}
Reason for revert:
A interface change broke some downstream code in google3.
Original issue's description:
> Support external audio mixer in webrtc.
>
> An external audio mixer will be passed from PeerConnectionFactory to
> AudioTransportProxy.
>
> BUG=webrtc:6457
>
> Committed: https://crrev.com/f6bcac59e88c3be5ffd73bbb1098f2d82ee316a1
> Cr-Commit-Position: refs/heads/master@{#15556}
TBR=solenberg@webrtc.org,aleloi@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2562333003
Cr-Commit-Position: refs/heads/master@{#15557}
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2539213003
Cr-Commit-Position: refs/heads/master@{#15556}
It was using a non-WebRTC-named header guard, which could conflict with
other similarly named/intended headers.
BUG=None
NO_DEPENDENCY_CHECKS=true
Review-Url: https://codereview.webrtc.org/2548113002
Cr-Commit-Position: refs/heads/master@{#15554}
SimulcastEncoderAdapter calls Release() on a failed sub-encoder init,
but Release only knows how to clean up encoders that have registered
stream info. Since failed ones don't register, they aren't currently
cleaned up.
BUG=None
Review-Url: https://codereview.webrtc.org/2544003005
Cr-Commit-Position: refs/heads/master@{#15553}
Was added for video initially, but not for audio.
BUG=webrtc:6862
Review-Url: https://codereview.webrtc.org/2568553002
Cr-Commit-Position: refs/heads/master@{#15552}
The packetization parts of this class are accessed from the
encoder thread, which might change under different occasions.
The use of a sequenced task checker here is thus incorrect, since
that requires the access to always be on the same thread, whenever
a task queue is not used.
The access to the instantiated object of this class, at least when
it comes to the RTP packetization parts, is however synchronized
using the lock in PayloadRouter::OnEncodedImage. We can therefore
safely remove the sequenced task checker.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2562983002
Cr-Commit-Position: refs/heads/master@{#15549}
The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}
adaptReason in webrtcvideoengine2.h only defines NONE=0, CPU=1 and BANDWIDTH=2 so &0x4 can not happen anymore.
This was probably never implemented in videoengine2
BUG=webrtc:6870
Review-Url: https://codereview.webrtc.org/1887773002
Cr-Commit-Position: refs/heads/master@{#15546}
The changes here are the same as in https://codereview.webrtc.org/2523993002/:
- reduce packet loss from 50% to 5%, to allow the BWE to ramp up better.
- Do not wait for 100 packets to be sent before letting the test pass.
BUG=webrtc:6851
Review-Url: https://codereview.webrtc.org/2558303002
Cr-Commit-Position: refs/heads/master@{#15542}
This also deletes unused features of the video_capturer interface, the classes
VideoCaptureFeedBack, VideoCaptureEncodeInterface and related methods,
and the module id which used to be passed as an argument to the
VideoCaptureDataCallback.
In theory the module id could have been used to let a single
VideoCaptureDataCallback serve several capturers, and demultiplex
on the id, but in practice, it was unused. With this change, it is
required to use a separate VideoSinkInterface for each capturer.
BUG=webrtc:6789
Review-Url: https://codereview.webrtc.org/2534553002
Cr-Commit-Position: refs/heads/master@{#15540}
This is to allow application to pass an audio network adaptor config string to WebRTC.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2437803004
Cr-Commit-Position: refs/heads/master@{#15532}
Changing the configuration will cause subsequently generated offers to change
the ufrag/pwd as necessary, so that a new round of gathering is started that
uses the new configuration.
This CL also makes some minor unrelated changes: changing the reference SDP in
the PC tests to more match what we generate, and relaxing the network thread
requirement for JsepTransport (since there's no reason the "needs-ice-restart"
flag can't be accessed from the signaling thread).
BUG=webrtc:6714
Review-Url: https://codereview.webrtc.org/2563153002
Cr-Commit-Position: refs/heads/master@{#15527}
The enum is at about the same level of detail as DOMExceptions, and I
looked through the spec making sure that chromium will be able to perform
the DOMException mapping for each one.
The new enum is called RtcError and is outside the PeerConnectionInterface
scope, because we may want to use this for things not associated with a
PeerConnection in the future.
This CL doesn't yet use the error enum anywhere; that will probably happen
in follow-up CLs for the individual methods.
BUG=webrtc:6855
Review-Url: https://codereview.webrtc.org/2564683002
Cr-Commit-Position: refs/heads/master@{#15526}
ScreenCapturerWinGdi randomly returns black frames in test environment. The root
cause is still unknown, so change ScreenCapturerWinGdi tests into MANUAL mode to
execute in test environment, but unblock other developers. We can eventually get
a failure ratio and more samples for debugging.
BUG=webrtc:6666, webrtc:6843
Review-Url: https://codereview.webrtc.org/2564173002
Cr-Commit-Position: refs/heads/master@{#15518}
This should remove the test flakiness, as before this change there
could be collisions from sequence numbers coming from two sequence
number spaces (the media SSRC and the FlexFEC SSRC). The probability
of collisions was low, and hence the flakes were far between.
This change also reduces the packet loss to 5% (down from ~50%), in
order for the BWE to have an easier time to ramp up.
BUG=webrtc:6825
R=philipel@webrtc.org, mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2554403003
Cr-Commit-Position: refs/heads/master@{#15512}
A decision was recently made to limit downscaling to 320x180 on
Android. This causes the perf tests to fail. This test is no
longer valid on android, as the failure is expected behaviour.
BUG=None
NOTRY=true
TBR=phoglund@webrtc.org
Review-Url: https://codereview.webrtc.org/2563913003
Cr-Commit-Position: refs/heads/master@{#15510}
