Fix an issue where using setNeedsDisplay on a GLKView which has a frame
with size zero will make GLKView/iOS output the following error:
Failed to bind EAGLDrawable: <CAEAGLLayer: 0x1742282e0> to
GL_RENDERBUFFER 1 Failed to make complete framebuffer object 8cd6
(The error code 8cd6 corresponds to
GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT.)
GLKView will internally setup it's render buffer when the delegate is
about to draw into it. Previously when enableSetNeedsDisplay was set to
YES (default), then GLKView would still attempt to setup it's internal
buffer even if it's frame size is zero and that would cause
GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT.
By using enableSetNeedsDisplay = NO, RTCEAGLVideoView can guard against
calling -[GLKView display] if it's current frame size is empty.
Review URL: https://codereview.webrtc.org/1347013002
Cr-Commit-Position: refs/heads/master@{#10076}
Ensures that we can restart audio recording on Android without hitting
a DCHECK. Also adds a symmetric design for the playout side.
BUG=webrtc:5000
TEST=modules_unittests --gtest_filter=AudioDevice*
Review URL: https://codereview.webrtc.org/1373443003
Cr-Commit-Position: refs/heads/master@{#10072}
Ensure the checkout is updated _before_ reading the current
revision from the DEPS file. This will avoid errors when a checkout
is not up to date, as updating happened after reading the revision
before this change.
BUG=webrtc:4688
R=phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1361923004 .
Cr-Commit-Position: refs/heads/master@{#10071}
Reason for revert:
Breaking Chromium FYI bots.
Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}
TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368933002
Cr-Commit-Position: refs/heads/master@{#10069}
This CL adds a slider that can change capture resolution and fps during a call. The camera will no be reconfigured, but the frames will be downscaled/dropped in software by cricket::VideoAdapter in the cricket::VideoCapturer. This is controlled with VideoCapturerAndroid.onOutputFormatRequest(). The slider is turned off by default and can be enabled with a checkbox under 'WebRTC Video Settings'.
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1361083002 .
Cr-Commit-Position: refs/heads/master@{#10067}
Instead of FATAL on a bad codec specification, log and return an error
code. This is a band-aid until callers are taught to only give it good
specifications.
BUG=webrtc:5033, chromium:526478
Review URL: https://codereview.webrtc.org/1364193002
Cr-Commit-Position: refs/heads/master@{#10066}
This will allow Chromium's build files to be updated, so that when the
real RtpSender CL is submitted, it doesn't break the FYI bots.
Review URL: https://codereview.webrtc.org/1364813004
Cr-Commit-Position: refs/heads/master@{#10065}
Otherwise, we may delete a useful connection because the current best connection may be failing.
BUG=
Review URL: https://codereview.webrtc.org/1364683002
Cr-Commit-Position: refs/heads/master@{#10063}
Was previously using the wrong preprocessor define (DEBUG vs _DEBUG).
Review URL: https://codereview.webrtc.org/1361173002
Cr-Commit-Position: refs/heads/master@{#10061}
It's good hygiene and just generally the right thing to do. And
apparently at least sometimes required by Microsoft's compiler.
Review URL: https://codereview.webrtc.org/1364233002
Cr-Commit-Position: refs/heads/master@{#10060}
This will paves the way for continuous nomination lite and multi-networking.
2. Combined checking and pinging to remove some redundant checking and to make it switch to more frequent ping mode earlier.
Review URL: https://codereview.webrtc.org/1311433009
Cr-Commit-Position: refs/heads/master@{#10057}
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1349393003
Cr-Commit-Position: refs/heads/master@{#10046}
A couple of places were missing handling of UYVY video formats.
BUG=webrtc:4816
Review URL: https://codereview.webrtc.org/1317613003
Cr-Commit-Position: refs/heads/master@{#10044}
Reason for revert:
Broke FYI bots because SetConfiguration is pure virtual and MockPeerConnectionImpl doesn't implement it. Need to reland with SetConfiguration not pure virtual.
Original issue's description:
> Adding PeerConnectionInterface::SetConfiguration method.
>
> Also updated the JNI and Objective-C bindings. Later, will have a CL to
> remove UpdateIce, which this method effectively replaces.
>
> BUG=webrtc:4945
>
> Committed: https://crrev.com/70702afbcb8418fe93747e7ed63bcbf5e56b90e9
> Cr-Commit-Position: refs/heads/master@{#10040}
TBR=guoweis@webrtc.org,pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4945
Review URL: https://codereview.webrtc.org/1361263002
Cr-Commit-Position: refs/heads/master@{#10041}
Also updated the JNI and Objective-C bindings. Later, will have a CL to
remove UpdateIce, which this method effectively replaces.
BUG=webrtc:4945
Review URL: https://codereview.webrtc.org/1317353005
Cr-Commit-Position: refs/heads/master@{#10040}
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/
Which fixes the FYI bots for the original CL, and breaks them for this revert.
Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}
TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1361773005
Cr-Commit-Position: refs/heads/master@{#10036}
No longer lock-order inverting since RTP/RTCP modules are instantiated
on construction and no longer guarded by a separate lock.
BUG=webrtc:2999
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1347283004 .
Cr-Commit-Position: refs/heads/master@{#10034}
The BWE expects arrival timestamps in ms, while the audio path delivered
them in us, causing the BWE to break down under the combined audio/video
BWE experiment. This was introduced in r9892 (68786d2040).
BUG=webrtc:4758
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1360913004 .
Cr-Commit-Position: refs/heads/master@{#10032}
Bug 4985 revealed two flaws
1. Opus duration estimate did not return correct length for DTX packets,
2. NetEq DoCodecInternalCng did not assign enough buffer.
P.S.
Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL.
BUG=webrtc:4985
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1334303005 .
Cr-Commit-Position: refs/heads/master@{#10031}
Removes ShouldIgnoreTrace from WebRtcVoiceEngine and removes the spammy
log instances instead. Also removes trace-style logging from getters
(::GetLocalSSRC() for instance would print what SSRC it got, spamming
the log).
BUG=
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1347353004 .
Cr-Commit-Position: refs/heads/master@{#10028}
