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}
Reason for revert:
Compile error on Android needs to be fixed before relanding.
Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}
TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1537213002
Cr-Commit-Position: refs/heads/master@{#11094}
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}
Reason for revert:
Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome.
Original issue's description:
> Added option to specify a maximum file size when recording an AEC dump.
>
> 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
>
> Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87
> Cr-Commit-Position: refs/heads/master@{#11081}
TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1533913004
Cr-Commit-Position: refs/heads/master@{#11087}
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}
Original issue's description:
> Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ )
>
> Reason for revert:
> Breaks WebrtcTransportTest.DataStream, due to different rtc::Thread implementation on Chromium.
>
> Original issue's description:
> > Free SCTP data channels asynchronously in PeerConnection.
> >
> > This is needed so that the data channel isn't deleted while one of its
> > own methods is on the call stack.
> >
> > BUG=565048
> >
> > Committed: https://crrev.com/386869247f28e72a00307a1b5c92465eea343ad2
> > Cr-Commit-Position: refs/heads/master@{#10923}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=565048
>
> Committed: https://crrev.com/a1f567ae9012a8de573b5bde492dd9ca0d17f137
> Cr-Commit-Position: refs/heads/master@{#10977}
BUG=565048
Review URL: https://codereview.webrtc.org/1516943002
Cr-Commit-Position: refs/heads/master@{#11015}
Ie, rotation is applied in C++ in the VideoFrameFactory is apply_rotation_ is set. If not, rotation is sent in RTP.
BUG=webrtc:4993
R=nisse@chromium.org
Review URL: https://codereview.webrtc.org/1493913007 .
Cr-Commit-Position: refs/heads/master@{#10986}
Reason for revert:
Breaks WebrtcTransportTest.DataStream, due to different rtc::Thread implementation on Chromium.
Original issue's description:
> Free SCTP data channels asynchronously in PeerConnection.
>
> This is needed so that the data channel isn't deleted while one of its
> own methods is on the call stack.
>
> BUG=565048
>
> Committed: https://crrev.com/386869247f28e72a00307a1b5c92465eea343ad2
> Cr-Commit-Position: refs/heads/master@{#10923}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=565048
Review URL: https://codereview.webrtc.org/1513143003
Cr-Commit-Position: refs/heads/master@{#10977}
The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed.
Removed unused iSupportsRenderTiming and the settings structs since it was not used.
Added VCMReceiver::FrameForDecoding unit test for the case when PreferDecodeLate is set.
Note that this does not change the current behaviour. We actually currently always decode frames late. This cl is to make sure the behaviour is kept for Android, if the default behaviour is changed.
Review URL: https://codereview.webrtc.org/1428293003
Cr-Commit-Position: refs/heads/master@{#10974}
do the conversion using an opengl fragment shader.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1460703002
Cr-Commit-Position: refs/heads/master@{#10972}
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.
BUG=webrtc:4868
Review URL: https://codereview.webrtc.org/1418123003
Cr-Commit-Position: refs/heads/master@{#10958}
This is needed for Chromium so that we can roll, update libjingle.gyp and then continue.
BUG=chromium:121673
Review URL: https://codereview.webrtc.org/1514573003
Cr-Commit-Position: refs/heads/master@{#10955}
If a description is set that requires making a default stream, and one
already exists, we'll now keep the existing default audio/video tracks,
rather than destroying them and recreating them. Destroying them caused
the blink MediaStream to go to an "ended" state, which is the root cause
of the bug.
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1469833006
Cr-Commit-Position: refs/heads/master@{#10946}
Adds tracing specifically to Close, for creating streams and also moves
tracing for SetLocal/RemoteDescription from WebRtcSession. Also adding
some tracing in ChannelManager to see what's taking time inside Close.
BUG=webrtc:5167
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1509903002 .
Cr-Commit-Position: refs/heads/master@{#10943}
This fix an issue seen on Huawei Y300 where the camera feed is black and white if we capture to textures and setpreviewformat is called.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1502223002
Cr-Commit-Position: refs/heads/master@{#10941}
Adds tracing to WebRtcSession and corresponding BaseChannel calls to see
where time is spent better.
BUG=webrtc:5167
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1505023003 .
Cr-Commit-Position: refs/heads/master@{#10934}
This is needed so that the data channel isn't deleted while one of its
own methods is on the call stack.
BUG=565048
Review URL: https://codereview.webrtc.org/1492383002
Cr-Commit-Position: refs/heads/master@{#10923}
Logs tracing events (TRACE_EVENT0 and friends) to storage in a format
compatible with chrome://tracing which can be used for performance
evaluation, finding lock contention and other sweet things). Tracing is
still basic and doesn't contain thread metadata or logging of tracing
arguments.
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1457383002 .
Cr-Commit-Position: refs/heads/master@{#10921}
