This change adds a new statistic for logging how many calls to
NetEq::GetAudio resulted in a "muted output". A muted output happens
if the packet stream has been dead for some time (and the last decoded
packet was not comfort noise).
BUG=webrtc:5606
BUG=b/31256483
Review-Url: https://codereview.webrtc.org/2341293002
Cr-Commit-Position: refs/heads/master@{#14302}
Also enforce a minimum inter-frame interval of 1 ms,
fix a bug in the clipping logic, and improve comments.
BUG=webrtc:5740
Review-Url: https://codereview.webrtc.org/2325563002
Cr-Commit-Position: refs/heads/master@{#14206}
Users are updated to call libyuv functions directly. Also delete
related unit tests.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2287233002
Cr-Commit-Position: refs/heads/master@{#14013}
This test has been failing on TSan lately:
../../webrtc/media/base/videoengine_unittest.h:519: Failure
Value of: GetReceiverStats(i).frame_width
Actual: 0
Expected: DefaultCodec().width
Which is: 640
The root cause for the failure appears to be that the stats update
(https://cs.chromium.org/chromium/src/third_party/webrtc/video/video_receive_stream.cc?rcl=1472584967&l=353)
happens to be after the frame is passed to the renderer - while the test
is only waiting for the former.
The fix is to give it some extra time using EXPECT_EQ_WAIT instead.
Review-Url: https://codereview.webrtc.org/2299483002
Cr-Commit-Position: refs/heads/master@{#13991}
GetCopyWithRotationApplied is not yet deleted; downstream projects
must be updated first.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2285693002
Cr-Commit-Position: refs/heads/master@{#13973}
Pass incoming frames directly to VideoCapturer::OnFrame (after
conversion to cricket::VideoFrame), without using SignalFrameCaptured
or WebRtcCapturedFrame.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2258933003
Cr-Commit-Position: refs/heads/master@{#13861}
New files, classes moved from statscollector_unittest.cc:
+webrtc/api/test/mock_peerconnection.h
for MockPeerConnectionFactory and MockPeerConnection
+webrtc/api/test/mock_webrtcsession.h
for MockWebRtcSession
+webrtc/media/base/test/mock_mediachannel.h
for MockVideoMediaChannel and MockVoiceMediaChannel
The webrtc/media/base/test folder is new.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2238933002
Cr-Commit-Position: refs/heads/master@{#13769}
To allow end-to-end QuicDataChannel usage with a
PeerConnection, RTCConfiguration has been modified to
include a boolean for whether to do QUIC, since negotiation of
QUIC is not implemented. If one peer does QUIC, then it will be
assumed that the other peer must do QUIC or the connection
will fail.
PeerConnection has been modified to create data channels of type
QuicDataChannel when the peer wants to do QUIC.
WebRtcSession has ben modified to use a QuicDataTransport
instead of a DtlsTransportChannelWrapper/DataChannel
when QUIC should be used
QuicDataTransport implements the generic functions of
BaseChannel to manage the QuicTransportChannel.
Committed: https://crrev.com/34b54c36a533dadb6ceb70795119194e6f530ef5
Review-Url: https://codereview.webrtc.org/2166873002
Cr-Original-Commit-Position: refs/heads/master@{#13645}
Cr-Commit-Position: refs/heads/master@{#13657}
Reason for revert:
Reverting because it broke an RTP data channel test on the FYI bots.
Original issue's description:
> Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
>
> To allow end-to-end QuicDataChannel usage with a
> PeerConnection, RTCConfiguration has been modified to
> include a boolean for whether to do QUIC, since negotiation of
> QUIC is not implemented. If one peer does QUIC, then it will be
> assumed that the other peer must do QUIC or the connection
> will fail.
>
> PeerConnection has been modified to create data channels of type
> QuicDataChannel when the peer wants to do QUIC.
>
> WebRtcSession has ben modified to use a QuicDataTransport
> instead of a DtlsTransportChannelWrapper/DataChannel
> when QUIC should be used
>
> QuicDataTransport implements the generic functions of
> BaseChannel to manage the QuicTransportChannel.
>
> Committed: https://crrev.com/34b54c36a533dadb6ceb70795119194e6f530ef5
> Cr-Commit-Position: refs/heads/master@{#13645}
TBR=pthatcher@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2206793007
Cr-Commit-Position: refs/heads/master@{#13647}
To allow end-to-end QuicDataChannel usage with a
PeerConnection, RTCConfiguration has been modified to
include a boolean for whether to do QUIC, since negotiation of
QUIC is not implemented. If one peer does QUIC, then it will be
assumed that the other peer must do QUIC or the connection
will fail.
PeerConnection has been modified to create data channels of type
QuicDataChannel when the peer wants to do QUIC.
WebRtcSession has ben modified to use a QuicDataTransport
instead of a DtlsTransportChannelWrapper/DataChannel
when QUIC should be used
QuicDataTransport implements the generic functions of
BaseChannel to manage the QuicTransportChannel.
Review-Url: https://codereview.webrtc.org/2166873002
Cr-Commit-Position: refs/heads/master@{#13645}
This is part of rewriting the ConferenceMixer and OutputMixer.
Calls are instead routed through AudioReceiveStream::Start/Stop.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2206223002
Cr-Commit-Position: refs/heads/master@{#13636}
Passing transport_frame_id() to VideoSink will allow to identify incoming video
frames, which will make it possible to correlate video frames on the
sender and on the receiver.
BUG=chromium:621691
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2088953002 .
Cr-Commit-Position: refs/heads/master@{#13596}
webrtc doesnt include the header that the function is prototyped in.
This CL makes the convert_from.h include those headers to allow webrtc to
update to the head libyuv.
R=marpan@webrtc.org,pbos@webrtc.org
BUG=libyuv:620,webrtc:6094
TESTED=local build and try bots
Review-Url: https://codereview.webrtc.org/2139853002
Cr-Commit-Position: refs/heads/master@{#13436}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749
Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
functionality and exposes the functionality using the
MediaConstraints.
The exposing of the feature through the MediaConstraints
was done similarly to what was done for the intelligibility
enhancer in the CL
https://codereview.webrtc.org/1952123003
This CL is dependent on the CL https://codereview.webrtc.org/2090583002/ which contains
the level control functionality.
NOTRY=true
BUG=webrtc:5920
Review-Url: https://codereview.webrtc.org/2095563002
Cr-Commit-Position: refs/heads/master@{#13336}
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
Relanding again after fixing issue with RTC_DCHECKs.
This CL eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13305}
Reason for revert:
Broke video sending for iOS AppRTCDemo. To repro, run iOS AppRTCDemo in Release in loopback mode. The revision prior to this change worked.
Original issue's description:
> Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: 2d5491783aTBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2092273003
Cr-Commit-Position: refs/heads/master@{#13289}
This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13287}
Reason for revert:
Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland.
Original issue's description:
> Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c
> Cr-Commit-Position: refs/heads/master@{#13285}
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2099843003
Cr-Commit-Position: refs/heads/master@{#13286}
This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13285}
Reason for revert:
Breaking Chrome FYI bots.
Original issue's description:
> Cleanups in cricket::VideoFrame and cricket::WebRtcVideoFrame.
>
> Removed some protected virtual methods from VideoFrame that no longer
> need to exist. Some minor cleanups in the tests.
>
> BUG=webrtc:5682
>
> Committed: https://crrev.com/742d7b10b9720ec43de26e0faef52e5cb9c0daa8
> Cr-Commit-Position: refs/heads/master@{#13275}
TBR=pbos@webrtc.org,nisse@webrtc.org,deadbeef@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2091983002
Cr-Commit-Position: refs/heads/master@{#13277}
Removed some protected virtual methods from VideoFrame that no longer
need to exist. Some minor cleanups in the tests.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2075983003
Cr-Commit-Position: refs/heads/master@{#13275}
