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}
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}
The issue occurred when deserializing and then serializing a rejected
content description, which doesn't have the ICE ufrag/pwd in the first
place.
BUG=webrtc:5105
Review URL: https://codereview.webrtc.org/1534363002
Cr-Commit-Position: refs/heads/master@{#11134}
This will allow an app to create senders with the same stream id,
without SDP munging.
Review URL: https://codereview.webrtc.org/1538673002
Cr-Commit-Position: refs/heads/master@{#11092}
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
This regression was introduced by CL 1505573002 to support remote fingerprint update. What happened is that during PrAnswer, we incorrectly do not apply bundle. However, the channel has become writable at that time. When Answer comes, we still reset the srtp_filter but since the channel has been writable, the new SRTP context has never been applied.
We're making sure that we could always apply SRTP context even when channel has been writable. We'll address the issue that bundle should apply even in PrAnswer in a different CL.
BUG=568734
Review URL: https://codereview.webrtc.org/1532543003
Cr-Commit-Position: refs/heads/master@{#11075}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
On the receiving side, if a candidate arrives with an old ufrag, it will be dropped. If it contains a new frag that has never seen before, it will hold the ufrag and create connections, although those connections are not pingable until the ICE credentials are received.
This could avoid a bunch of ICE generation issues.
BUG=webrtc:5138,webrt:5292
Review URL: https://codereview.webrtc.org/1498993002
Cr-Commit-Position: refs/heads/master@{#11060}
If a MediaStream is added to a PeerConnection, and later a track
is added to the MediaStream, a new RtpSender will now be created for
that track, and it will appear in subsequent offers.
Similarly, removed tracks will remove RtpSenders.
BUG=webrtc:5265
Review URL: https://codereview.webrtc.org/1507973003
Cr-Commit-Position: refs/heads/master@{#11040}
This CL:
* Abstracts the functions in GlRectDrawer to an interface.
* Adds viewport location as argument to the draw() functions, because this information may be needed by some shaders. This also moves the responsibility of calling GLES20.glViewport() to the drawer.
* Moves uploadYuvData() into a separate helper class.
* Adds new SurfaceViewRenderer.init() function and new VideoRendererGui.create() function that takes a custom drawer as argument. Each YuvImageRenderer in VideoRendererGui now has their own drawer instead of a common one.
BUG=b/25694445
R=nisse@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1520243003 .
Cr-Commit-Position: refs/heads/master@{#11031}
Additionally:
* Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack.
* AddSink/RemoveSink are now on all audio sources (like they are for video sources).
While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state.
BUG=chromium:569526
Review URL: https://codereview.webrtc.org/1522903002
Cr-Commit-Position: refs/heads/master@{#11026}
