This CL turns nativeConfiguration into createNativeConfiguration returning a
pointer or nil on failure. This method's certificate generation is updated to
use the new API and reports failure (nil) if unsuccessful instead of relying on
the default certificate. We also remove the implicit assumption (now incorrect)
that RSA is the default. This is the same type of changes as was done in
https://codereview.webrtc.org/1965313002 but this file
(RTCPeerConnectionInterface.mm) was forgotten.
With no more usages of kIdentityName it and dtlsidentitystore.cc is removed.
Also removes unnecessary #include in peerconnectioninterface.h that was still
remnant due to an indirect include of kIdentityName.
RTCConfiguration+Private.h now lists method nativeEncryptionKeyTypeForKeyType
which was added in the above mentioned prior CL.
BUG=webrtc:5707, webrtc:5708
Review-Url: https://codereview.webrtc.org/2035473004
Cr-Commit-Position: refs/heads/master@{#13089}
Sync the GYP and GN targets and update the name of the GN one
to 'remote_bitrate_estimator'.
Move the GYP variable 'enable_bwe_test_logging' into the local scope.
Remove redundant entries in modules.gyp.
These are preparations related to the GN migration.
BUG=webrtc:5949
TESTED=Ran GYP with the default variables and with
-Denable_bwe_test_logging=1. Compiled remote_bitrate_estimator
and verified that bwe_test_logging.cc is compiled only when
set.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2040313004
Cr-Commit-Position: refs/heads/master@{#13087}
In org.webrtc.VideoCapturerAndroidTest#startWhileCameraIsAlreadyOpenAndCloseCamera,
use a video renderer instead of a capture observer. The video renderer
automatically returns the texture buffers, which resolves the bug.
There shouldn't be any changes to the effectiveness of the test.
BUG=webrtc:5982
Review-Url: https://codereview.webrtc.org/2042283004
Cr-Commit-Position: refs/heads/master@{#13085}
Reason for revert:
Plan to reland with InitToBlack kept, to be able to update Chrome to use the new I420Buffer::SetToBlack method.
Original issue's description:
> Revert of New static method I420Buffer::SetToBlack. (patchset #4 id:60001 of https://codereview.webrtc.org/2029273004/ )
>
> Reason for revert:
> Breaks chrome, in particular, the tests in
>
> media_stream_remote_video_source_unittest.cc
>
> use the InitToBlack method which is being deleted.
>
> Original issue's description:
> > New static method I420Buffer::SetToBlack.
> >
> > Replaces cricket::VideoFrame::SetToBlack and
> > cricket::WebRtcVideoFrame::InitToBlack, which are deleted.
> >
> > Refactors the black frame logic in VideoBroadcaster, and a few of the
> > tests.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/663f9e2ddc86e813f6db04ba2cf5ac1ed9e7ef67
> > Cr-Commit-Position: refs/heads/master@{#13063}
>
> TBR=perkj@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/271d74078894bb24f454eb31b77e4ce38097a2fa
> Cr-Commit-Position: refs/heads/master@{#13065}
TBR=perkj@webrtc.org,pbos@webrtc.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/2049513005
Cr-Commit-Position: refs/heads/master@{#13083}
The AppRTCDemo app on Mac OSX does not show or send local video streams,
as ACFoundation capture session is not compiled in or implemented in
the appropriate places. This is the first part of a two-part patch
that implements local capture on the Mac for AppRTCDemo
P.S. This is my first patch to WebRTC. I didn't see any relevant tests, but I could write some if you can point me at a location. Also, I don't have access to the automated tests (I don't think)
BUG=webrtc:3417
Review-Url: https://codereview.webrtc.org/2046863004
Cr-Commit-Position: refs/heads/master@{#13080}
Previously RefCountedObject was passing all parameters by value.
This meant that it was hard to use it with movable types, such
as unique_ptr<>. Now there is a constructor that takes r-value,
which means that RefCountedObject<std::unique_ptr<foo>> can be
initialized by passing std::unique_ptr<foo> to the constructor.
Review-Url: https://codereview.webrtc.org/2036123002
Cr-Commit-Position: refs/heads/master@{#13079}
This only causes an increased likelihood of role conflicts as each peer
is picking a new role. It also means that if ICE is restarted for only
one media stream, roles can be different across media streams (which isn't
even allowed).
For more explanation of why this is unnecessary and should be changed,
see this discussion:
https://mailarchive.ietf.org/arch/msg/ice/C0_QRCTNcwtvUF12y28jQicPR10
Review-Url: https://codereview.webrtc.org/2046123003
Cr-Commit-Position: refs/heads/master@{#13077}
For hiding xcworkspace files, along with xcodeproj files, both of which
are created by the ninja xcode generator
BUG=NONE
NOTRY=True
Review-Url: https://codereview.webrtc.org/2045733005
Cr-Commit-Position: refs/heads/master@{#13075}
to fix projects that has own copy of base/thread_annotation.h
R=åsapersson
NOTRY=true
Review-Url: https://codereview.webrtc.org/2048113002
Cr-Commit-Position: refs/heads/master@{#13073}
This reduces its complexity by a factor of 2.7x total.
The mean error introduced by this is in the 6 different noise scenarios and 6 different speech signals tested is below -52dB.
Review-Url: https://codereview.webrtc.org/2035213002
Cr-Commit-Position: refs/heads/master@{#13072}
The changes are done in several patches in order to make
the review easier.
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2051443002
Cr-Commit-Position: refs/heads/master@{#13068}
WebRtcSpl_CrossCorrelation and WebRtcSpl_DotProductWithScale compute
the int32 sum of pairwise products from two int16 arrays. So as to
avoid overflow (which could otherwise happen when as little as two
products were summed), the products are right-shifted by an amount
specified by the caller.
This CL changes WebRtcIlbcfix_MyCorr and WebRtcIlbcfix_Smooth to give
sufficient right-shift amounts, instead of ones that may be too small
and cause overflow.
BUG=chromium:601787
Review-Url: https://codereview.webrtc.org/2014033002
Cr-Commit-Position: refs/heads/master@{#13066}
Reason for revert:
Breaks chrome, in particular, the tests in
media_stream_remote_video_source_unittest.cc
use the InitToBlack method which is being deleted.
Original issue's description:
> New static method I420Buffer::SetToBlack.
>
> Replaces cricket::VideoFrame::SetToBlack and
> cricket::WebRtcVideoFrame::InitToBlack, which are deleted.
>
> Refactors the black frame logic in VideoBroadcaster, and a few of the
> tests.
>
> BUG=webrtc:5682
>
> Committed: https://crrev.com/663f9e2ddc86e813f6db04ba2cf5ac1ed9e7ef67
> Cr-Commit-Position: refs/heads/master@{#13063}
TBR=perkj@webrtc.org,pbos@webrtc.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/2049023002
Cr-Commit-Position: refs/heads/master@{#13065}
Replaces cricket::VideoFrame::SetToBlack and
cricket::WebRtcVideoFrame::InitToBlack, which are deleted.
Refactors the black frame logic in VideoBroadcaster, and a few of the
tests.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2029273004
Cr-Commit-Position: refs/heads/master@{#13063}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2040073002
Cr-Commit-Position: refs/heads/master@{#13061}
In https://codereview.webrtc.org/2034923003 it was discovered
that a test binary rtc_sdk_peerconnection_objc_tests was
a dependency to rtc_unittests. Unfortunately gtest doesn't
include dependent executables into the same test executable;
only libraries (so theses tests weren't run).
This CL incorporates those tests into rtc_unittests and
does the same changes to the GN build.
BUG=webrtc:5949
TESTED=Built and ran rtc_unittests locally on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2041743003
Cr-Commit-Position: refs/heads/master@{#13060}
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.
The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.
There are no tests at this time and most of testing is done with chromium
webrtc prototype.
On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.
BUG=webrtc:5895
Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
Previously raw pointers were used for owned DesktopFrame instances.
Updated all screen and window capturer implementations to use
std::unique_ptr<>.
Also includes some other cleanups in the capturers:
- s/NULL/nullptr
- moved default initializers to class definition.
BUG=webrtc:5950
Review-Url: https://codereview.webrtc.org/1988783003
Cr-Commit-Position: refs/heads/master@{#13058}
This moves the implementation specific methods to separate classes
(RtpSenderInternal/RtpReceiverInternal) so that the interface classes
represent the interface that external applications can rely on.
The reason this wasn't done earlier was that PeerConnection needed
to store proxy pointers, but also needed to access implementation-
specific methods on the underlying objects. This is now possible
by using "RtpSenderProxyWithInternal<RtpSenderInternal>", which is a proxy
that implements RtpSenderInterface but also provides direct access
to an RtpSenderInternal.
Review-Url: https://codereview.webrtc.org/2023373002
Cr-Commit-Position: refs/heads/master@{#13056}
When the fake clock's time is advanced, it now ensures all pending
queued messages have been dispatched. This allows us to write a
"SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
until the target time.
Useful in this case, where we know the STUN timeout should take a total
of 9500ms, but it would be overly complex to write test code that waits
for each individual timeout, ensures a STUN packet has been
retransmited, etc.
(The test described above *should* be written, but it belongs in
p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).
Review-Url: https://codereview.webrtc.org/2024813004
Cr-Commit-Position: refs/heads/master@{#13052}
The STUN fuzzer is split into two parts: validation and parsing. The
latter should be able to handle invalid packets instead of assuming
the validation deals with them, since an adversary could set a valid
HMAC on an invalid packet.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2044523002
Cr-Commit-Position: refs/heads/master@{#13050}
The iOS GN bots are the only ones using MB, which seems to enable
Goma by default. Since Goma has started supporting autostart, disable
it for now until a goma start step is added to the bots.
BUG=chromium:617541
TBR=phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/2045583002 .
Cr-Commit-Position: refs/heads/master@{#13049}
Reason for revert:
There seems to be a TSan warning that wasn't caught by the trybot: https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/6732/steps/peerconnection_unittests/logs/stdio
Apparently a thread is still alive even after destroying WebRTCSession. Need to think of a way to fix this, without adding a critical section around g_clock (since that would hurt performance).
Original issue's description:
> Improving the fake clock and using it to fix a flaky STUN timeout test.
>
> When the fake clock's time is advanced, it now ensures all pending
> queued messages have been dispatched. This allows us to write a
> "SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
> until the target time.
>
> Useful in this case, where we know the STUN timeout should take a total
> of 9500ms, but it would be overly complex to write test code that waits
> for each individual timeout, ensures a STUN packet has been
> retransmited, etc.
>
> (The test described above *should* be written, but it belongs in
> p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).
>
> Committed: https://crrev.com/ffbe0e17e2c9b7fe101023acf40574dc0c95631a
> Cr-Commit-Position: refs/heads/master@{#13043}
TBR=pthatcher@webrtc.org,tommi@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/2038213002
Cr-Commit-Position: refs/heads/master@{#13045}
When the fake clock's time is advanced, it now ensures all pending
queued messages have been dispatched. This allows us to write a
"SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up
until the target time.
Useful in this case, where we know the STUN timeout should take a total
of 9500ms, but it would be overly complex to write test code that waits
for each individual timeout, ensures a STUN packet has been
retransmited, etc.
(The test described above *should* be written, but it belongs in
p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc).
Review-Url: https://codereview.webrtc.org/2024813004
Cr-Commit-Position: refs/heads/master@{#13043}
Using an "EXPECT_WAIT" pattern with a long timeout rather than calling
"SleepMs" with a margin of 250ms.
BUG=webrtc:5953
Review-Url: https://codereview.webrtc.org/2029853002
Cr-Commit-Position: refs/heads/master@{#13040}
