They don't really belong in PeerConnection because the state at that
level can change when a transport channel is removed. That makes almost
any state transition possible.
The asserts are now in P2PTransportChannel (the equivalent to
IceTransport in the spec). Currently it has a reduced set of states,
that don't even take into account writability, but I plan to change
that soon.
BUG=webrtc:4757
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2005573002 .
Cr-Commit-Position: refs/heads/master@{#12937}
The test doesn't really care about the order; the fact that it relied
on a specific order was just an implementation detail.
However, this made the test flaky since race conditions sometimes
determine the order. It also made it frustrating to add new tests.
BUG=webrtc:5930
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2006263004 .
Cr-Commit-Position: refs/heads/master@{#12936}
Due to the experiment in chromium relying on KT_DEFAULT = KT_RSA (bug:
crbug.com/611698) a conditional was introduced. Now that the experiment is
ending and the experiment flag has been removed we can make KT_DEFAULT=KT_ECDSA
unconditionally.
BUG=chromium:611698
Review-Url: https://codereview.webrtc.org/2009533003
Cr-Commit-Position: refs/heads/master@{#12935}
The local endpoint was given 4 seconds to converge but the remote
endpoint was only given 1 second. Now they both collectively
have 4 seconds.
Also fixed the ExpectCandidate2 method (used for verbose failure
logging) and removed a bunch of dead code.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2018703002 .
Cr-Commit-Position: refs/heads/master@{#12934}
Adding a some checks and switching out a few assert for RTC_[D]CHECK.
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Committed: https://crrev.com/60c4e0ae8f124f08372645a95042f4a1246d7aa3
Cr-Commit-Position: refs/heads/master@{#12925}
Committed: https://crrev.com/5771beb265129082d31736259b7dc6ca037cff4d
Cr-Commit-Position: refs/heads/master@{#12926}
Review URL: https://codereview.webrtc.org/2014973002 .
Cr-Commit-Position: refs/heads/master@{#12927}
Adding a some checks and switching out a few assert for RTC_[D]CHECK.
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Committed: https://crrev.com/60c4e0ae8f124f08372645a95042f4a1246d7aa3
Cr-Commit-Position: refs/heads/master@{#12925}
Review URL: https://codereview.webrtc.org/2014973002 .
Cr-Commit-Position: refs/heads/master@{#12926}
Adding a some checks and switching out a few assert for RTC_[D]CHECK.
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
TBR=kjellander@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/2014973002 .
Cr-Commit-Position: refs/heads/master@{#12925}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
This affects the webrtc::VideoFrameBuffer and cricket::VideoFrame
classes.
To make this work, VideoFrameFactory is changed to use an
I420BufferPool rather than a plain VideoFrame to cache allocated
frames.
The I420BufferPool is reorganized to return an I420Buffer,
rather than a proxy object.
BUG=webrtc:5921, webrtc:5682
Review-Url: https://codereview.webrtc.org/2009193002
Cr-Commit-Position: refs/heads/master@{#12919}
A regression happened in https://codereview.webrtc.org/2006723002,
causing neteq_rtpplay not to work. The problem was that when the main
code was moved inside of the webrtc::test namespace, it was no longer
visible to the linker. Meanwhile, the dependency on test_support_main
rather than test_support caused the executable to be a gtest.
In this fix, the gyp dependencies are corrected, and a main method is
added outside of the namespaces.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2018473002
Cr-Commit-Position: refs/heads/master@{#12918}
If deleting files fails, wait a sec and retry. This can help in case
and antivirus software or similar has locked the file, preventing it
from being deleted.
BUG=webrtc:5898
Review-Url: https://codereview.webrtc.org/2014823002
Cr-Commit-Position: refs/heads/master@{#12917}
Also remove the unnecessary code in VideoCapturerAndroid.dispose() and
only log instead.
BUG=webrtc:5519
Review-Url: https://codereview.webrtc.org/2007863005
Cr-Commit-Position: refs/heads/master@{#12913}
Original issue's description:
> Adding a some checks and switching out a few assert for RTC_[D]CHECK.
> These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.
>
> BUG=chromium:613482
> NOTRY=true
> (using notry due to offline android_arm64_rel bot)
>
> Committed: https://crrev.com/d36df89d40bde3c62ee5cbff841933e50b3c007b
> Cr-Commit-Position: refs/heads/master@{#12870}
TBR=henrik.lundin@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:613482
Review-Url: https://codereview.webrtc.org/2009253004
Cr-Commit-Position: refs/heads/master@{#12907}
Also set the Configuration parameters in CreateRtpRtcpModules in the same order as the members are declared.
BUG=webrtc:5917
Review-Url: https://codereview.webrtc.org/2011433002
Cr-Commit-Position: refs/heads/master@{#12905}
This is useful because various pieces of code can then make themselves
more fuzzer-friendly. (For example, checksum verification can always
succeed.) See BORINGSSL_UNSAFE_FUZZER_MODE for an analogous flag.
BUG=chromium:561667
Review-Url: https://codereview.webrtc.org/2000173002
Cr-Commit-Position: refs/heads/master@{#12904}
Run VP8 HW only on M and above devices.
Exynos H.264 HW encoder does not use frame timestamps for
bitrate allocation, so bitrate target need to be adjusted based
on current camera frame rate.
BUG=b/28738766
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/2004973003 .
Cr-Commit-Position: refs/heads/master@{#12900}
Compile fixes for GN on iOS that finally gets our bots green.
Changes to system_wrappers:
* Updated to only use inclusive sources for maintainability
* Add a few missing GN headers.
* Cleanup GYP hack for atomic32_mac.cc
* Renamed changes sources to avoid problems with GYP/GN file
suffix rules:
- atomic32_mac.cc -> atomic32_darwin.cc
- atomic32_posix.cc -> atomic32_non_darwin_unix.cc
See https://code.google.com/p/chromium/codesearch#chromium/src/build/config/BUILDCONFIG.gn&l=325
for details on which extensions can/cannot be used.
BUG=webrtc:5586
NOTRY=True
Review-Url: https://codereview.webrtc.org/1999723002
Cr-Commit-Position: refs/heads/master@{#12897}
Splits VideoCapturer::OnFrameCaptured into helper methods,
which enables use of the VideoAdaptation logic without
using a frame factory.
Refactors AndroidVideoCapturer to make adaptation decision
earlier, so we can crop and rotate using
NV12ToI420Rotate.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1973873003
Cr-Commit-Position: refs/heads/master@{#12895}
The NetEqNetworkStatistics::expand_rate was not incremented during muted
state, which caused under-reporting of that metric. This change fixes
that.
BUG=chromium:613321, webrtc:5608
Review-Url: https://codereview.webrtc.org/2003203004
Cr-Commit-Position: refs/heads/master@{#12894}
Channel's API remains unchanged, but the creation of a BuiltinAudioDecoderFactory is now in Channel. The next step would be to amend Channel's API (through CreateChannel, I believe) to allow an AudioDecoderFactory to be sent along.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1992763002
Cr-Commit-Position: refs/heads/master@{#12893}
Previously, failure codes were ignored, which meant simulcasted codecs
couldn't e.g. trigger software fallback. This stops the simulcasted
Encode call at the first faiulre and returns that code.
Another option is to continue sending the frame to the other encoders
but still return the first failure code. It's not clear that that is any
better than just quitting as soon as a failure happens.
BUG=
Review-Url: https://codereview.webrtc.org/2008723002
Cr-Commit-Position: refs/heads/master@{#12892}
A new API is added which enables detection of support of pro-audio on
Android. This is part of a larger change and the new API is not used yet.
Most likely it will only be used for logging purposes.
BUG=webrtc:5925
Review-Url: https://codereview.webrtc.org/2015483002
Cr-Commit-Position: refs/heads/master@{#12890}
