Sometimes Android OS provides unknown connection type,
causing a DCHECK failure. This CL temporarily removes that checking.
BUG=
TBR=glaznev@webrtc.org
NOTRY=true
Review URL: https://codereview.webrtc.org/1667503002
Cr-Commit-Position: refs/heads/master@{#11466}
- Add extra logging for Android HW codec corner cases
when frames are dropped or resolution is changed.
- Use presentation timestamps for decoded frame logging.
- Enable key frame sending on long frame gap for
H.264 codec.
BUG=b/26504665
Review URL: https://codereview.webrtc.org/1653523003
Cr-Commit-Position: refs/heads/master@{#11452}
including options related to experimental constraints which are
recognized but never applied.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1642513002
Cr-Commit-Position: refs/heads/master@{#11424}
It's never used anywhere, so it only causes confusion between
itself and SessionDescriptionInterface::candidates.
Review URL: https://codereview.webrtc.org/1642733002
Cr-Commit-Position: refs/heads/master@{#11420}
The plan is that this interface should be used by all classes which receive a stream of video frames, and replace the two generic classes webrtc::VideoRendererInterface and cricket::VideoRenderer.
And the list goes on, there's a dozen of different classes which act as video frame sinks.
At some point, we will likely add some method to handle sink properties like, e.g, maximum useful width and height. But hopefully this can be done while keeping the interface very simple.
BUG=webrtc:5426
R=perkj@webrtc.org, pthatcher@webrtc.org
Committed: https://crrev.com/a862d4563fbc26e52bed442de784094ae1dfe5ee
Cr-Commit-Position: refs/heads/master@{#11396}
Review URL: https://codereview.webrtc.org/1594973006
Cr-Commit-Position: refs/heads/master@{#11414}
This is somewhat easier than looking up the media type by iterating pc.getLocalStreams / pc.getRemoteStreams and all tracks. Temporary until stats get revamped to conform to the spec obviously.
BUG=webrtc:4117
Review URL: https://codereview.webrtc.org/1307633007
Cr-Commit-Position: refs/heads/master@{#11412}
Expose disableBuffering method on underlying log sink.
This will make every write to the stream immediately write to the disk.
Useful in crash situations so that buffered output is not lost.
BUG=
Review URL: https://codereview.webrtc.org/1638283003
Cr-Commit-Position: refs/heads/master@{#11407}
Before HW VP8 downscaling was triggered by frame drops only.
Also reset the encoder when it drops large amount of frames in a row.
BUG=b/26504665
Review URL: https://codereview.webrtc.org/1592883004
Cr-Commit-Position: refs/heads/master@{#11406}
Before this fix, it was required that the EGL context used as an argument was kept open until all PeerConnections using the context had been closed. With this fix, that is no longer required.
Also, if a released EGLContext (EGL_NO_CONTEXT) is used, harware codecs will fallback to use byte buffers for encoding and decoding.
BUG=b/26583522
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1615153002 .
Cr-Commit-Position: refs/heads/master@{#11398}
Reason for revert:
Broke chrome build. Investigating.
First error relating to AddSink method in mock_peer_connection_dependency_factory.h
Original issue's description:
> New rtc::VideoSinkInterface.
>
> The plan is that this interface should be used by all classes which receive a stream of video frames, and replace the two generic classes webrtc::VideoRendererInterface and cricket::VideoRenderer.
>
> And the list goes on, there's a dozen of different classes which act as video frame sinks.
>
> At some point, we will likely add some method to handle sink properties like, e.g, maximum useful width and height. But hopefully this can be done while keeping the interface very simple.
>
> BUG=webrtc:5426
> R=perkj@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/a862d4563fbc26e52bed442de784094ae1dfe5ee
> Cr-Commit-Position: refs/heads/master@{#11396}
TBR=pthatcher@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1646463002
Cr-Commit-Position: refs/heads/master@{#11397}
The plan is that this interface should be used by all classes which receive a stream of video frames, and replace the two generic classes webrtc::VideoRendererInterface and cricket::VideoRenderer.
And the list goes on, there's a dozen of different classes which act as video frame sinks.
At some point, we will likely add some method to handle sink properties like, e.g, maximum useful width and height. But hopefully this can be done while keeping the interface very simple.
BUG=webrtc:5426
R=perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1594973006 .
Cr-Commit-Position: refs/heads/master@{#11396}
We can't assume that the audio source implementation will be our own internal one and we shouldn't apply local audio options to a remote audio track this way either.
BUG=5423
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1576913002 .
Cr-Commit-Position: refs/heads/master@{#11341}
This type is used for GL contexts in android hardware encode/decode, so
it seems correct/useful to make it publicly constructable.
Review URL: https://codereview.webrtc.org/1604403002
Cr-Commit-Position: refs/heads/master@{#11331}
This ensures the DtmfSender won't try to access it after it's
destroyed.
BUG=webrtc:5403
Review URL: https://codereview.webrtc.org/1590333004
Cr-Commit-Position: refs/heads/master@{#11327}
webrtc::VideoRendererInterface::SetSize was completely unused.
cricket::VideoRenderer::SetSize only had dummy implementations
returning true and doing nothing.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1574963002
Cr-Commit-Position: refs/heads/master@{#11298}
If the track is disabled, replace incoming frames by black frames. Affects local rendering of disabled tracks.
Also intended to replace the similar logic in
WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame, once
VideoRtpSender is hooked up as a renderer for the track.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1575223003
Cr-Commit-Position: refs/heads/master@{#11297}
Plus, in stunport, turnport and allocation sequence, create a socket using the new interface.
BUG=
Review URL: https://codereview.webrtc.org/1556743002
Cr-Commit-Position: refs/heads/master@{#11279}
- Similar to HW decoder log timing for first 5 frames
of HW encoder input and output to track correct encoder
start.
- Increment input timestamp when frame is dropped.
- Better calculate average encoding time.
- Use different tags for encoder and decoder modules.
BUG=b/26504665
Review URL: https://codereview.webrtc.org/1587943004
Cr-Commit-Position: refs/heads/master@{#11277}
Also, now creating the RtpSender/RtpReceiver proxy objects immediately,
rather than waiting until when GetSenders/GetReceivers is called.
Review URL: https://codereview.webrtc.org/1563403002
Cr-Commit-Position: refs/heads/master@{#11259}
Reason for revert:
These changes broke chrome.
Need to temporarily keep methods InitToEmptyBuffer, InitToBlack, CreateEmptyFrame with old but ignored arguments for pixel_width and pixel_height. Then update chrome, and delete the old methods in a separate cl.
Original issue's description:
> Delete remnants of non-square pixel support from cricket::VideoFrame.
>
> If ever needed, add some aspect ratio parameter, without pixel_width
> and pixel_height arguments cluttering commonly used functions.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/709513d4133107d5c02aed34a5ee99444c4d4e25
> Cr-Commit-Position: refs/heads/master@{#11243}
TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1583223002
Cr-Commit-Position: refs/heads/master@{#11246}
If ever needed, add some aspect ratio parameter, without pixel_width
and pixel_height arguments cluttering commonly used functions.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1586613002
Cr-Commit-Position: refs/heads/master@{#11243}
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.
Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}
TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1588693002
Cr-Commit-Position: refs/heads/master@{#11241}
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.
BUG=webrtc:4525
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516993002 .
Cr-Commit-Position: refs/heads/master@{#11192}
