Change the previous GN configs to build GYP instead
(since we'll keep GYP around for a while) but exclude tests for
that config from now on, since we're facing errors with GYP.
Add new configs for upcoming rename of those bots to GYP instead
of GN.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2264283003
Cr-Commit-Position: refs/heads/master@{#13875}
Uses generic functions to plot packet sizes, sequence number delta and bitrate per SSRC. Also detects and prints warnings if delay differences seem unrealistic.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2234883002
Cr-Commit-Position: refs/heads/master@{#13872}
The added logs will be helpful for debugging.
If a session has stopped, terminate DoAllocate early.
Session::init always returns true, so there is no need to check the return value.
R=deadbeef@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2267163002 .
Cr-Commit-Position: refs/heads/master@{#13871}
When they are included there will be a mismatch between what the BWE says and
what the encoder is allowed to use, causing us to send more than the network
can handle.
BUG=webrtc:6247
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/2269923003 .
Cr-Commit-Position: refs/heads/master@{#13866}
So that we don't have to use assert(). Includes one sample call site.
NOTRY=true
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2262173002
Cr-Commit-Position: refs/heads/master@{#13862}
Pass incoming frames directly to VideoCapturer::OnFrame (after
conversion to cricket::VideoFrame), without using SignalFrameCaptured
or WebRtcCapturedFrame.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2258933003
Cr-Commit-Position: refs/heads/master@{#13861}
Move the webrtc/test/test_support/metrics sources into
test_support[_unittests] targets.
This is essentially reverting https://webrtc-codereview.appspot.com/5789004
and moving these sources back to the right target.
Add missing foreman_cif.yuv resource needed for these tests.
For MIPS, a compile error was surfacing for logcat_trace_context.h when
flipping bot to GN, which was fixed.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2267113002
Cr-Commit-Position: refs/heads/master@{#13860}
Without this, the rtc_media_unittests target was only an indirect
dependency, and compiled without HAVE_WEBRTC_VIDEO. And some testcases,
in particular, all tests defined by webrtcvideocapturer_unittest.cc,
are excluded from rtc_media_unittests.
BUG=
Review-Url: https://codereview.webrtc.org/2250433008
Cr-Commit-Position: refs/heads/master@{#13859}
Removing a redundant variable used to track whether or not RTCP mux has
been fully negotiated. It's RtcpMuxFilter's job to do that, and it
already had the state, it just wasn't exposed.
Review-Url: https://codereview.webrtc.org/2260963002
Cr-Commit-Position: refs/heads/master@{#13856}
The new method returns the current total delay (packet buffer and sync
buffer) in ms, with smoothing applied to even out short-time
fluctuations due to jitter. The packet buffer part of the delay is not
updated during DTX/CNG periods.
This CL also pipes the new metric through ACM and uses it in
VoiceEngine. It replaces the previous method of estimating the buffer
delay (where an inserted packet's RTP timestamp was compared with the
last played timestamp from NetEq). The new method works better under
periods of DTX/CNG.
Review-Url: https://codereview.webrtc.org/2262203002
Cr-Commit-Position: refs/heads/master@{#13855}
Switches the main parsing function for RtcEventLogs to take an istream instead of a file pointer. Adds wrappers that accept either a string or a filename.
Review-Url: https://codereview.webrtc.org/2253943006
Cr-Commit-Position: refs/heads/master@{#13852}
Mostly, it's about replacing mutable reference arguments with pointer
arguments, and replacing C style casts with C++ style casts.
Review-Url: https://codereview.webrtc.org/2056653002
Cr-Commit-Position: refs/heads/master@{#13849}
function names style updated,
unused return type removed.
Comment style fixed, redundant comments removed.
pass-by-pointer parameter changed to pass-by-value because can't be nullptr any more.
NOTRY=true
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2258523005
Cr-Commit-Position: refs/heads/master@{#13848}
scale1 == 31 if and only if w10 == 0. So even though 1 << scale1
overflows, we know that the result of the multiplication should be 0.
Handle that case.
BUG=chromium:615818
Review-Url: https://codereview.webrtc.org/2258543002
Cr-Commit-Position: refs/heads/master@{#13847}
Derived from rtcp::Rtpfb instead of directly from RtcpPacket
Does not depend on RTCPUtility.
Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/1847973003
Cr-Commit-Position: refs/heads/master@{#13846}
Reason for revert:
Breaks some h264 bitstream tests downstream. Reverting for now.
Original issue's description:
> Add pps id and sps id parsing to the h.264 depacketizer.
>
> BUG=webrtc:6208
>
> Committed: https://crrev.com/abcc3de169d8896ad60e920e5677600fb3d40180
> Cr-Commit-Position: refs/heads/master@{#13838}
TBR=sprang@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6208
Review-Url: https://codereview.webrtc.org/2265023002
Cr-Commit-Position: refs/heads/master@{#13844}
Needed as a substitute when eliminating the Copy method.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2262683002
Cr-Commit-Position: refs/heads/master@{#13843}
On OS X El Capitan, the system location of 'PlistBuddy' is:
"/usr/libexec/PlistBuddy"
and default system path environment variable is:
"PATH=/usr/local/bin:/usr/bin:/bin:/usr/sbin:/sbin"
NOTRY=True
Review-Url: https://codereview.webrtc.org/2262813002
Cr-Commit-Position: refs/heads/master@{#13841}
This should help spot any differences between GN and GYP.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2246203004
Cr-Commit-Position: refs/heads/master@{#13840}
functionality is always present in the hardware.
BUG=webrtc:6231
Review-Url: https://codereview.webrtc.org/2260173002
Cr-Commit-Position: refs/heads/master@{#13839}
This is needed for webrtc::field_trial::FindFullName in peerconnection.cc
NOTRY=True
Review-Url: https://codereview.webrtc.org/2120673004
Cr-Commit-Position: refs/heads/master@{#13836}
If the input to H264VideoToolBoxEncoder is a native CVPixelBuffer and
the quality scaler requests scaling, we fall back to a slow path where
the buffer is converted from NV12 to I420 on the CPU and then uploaded
to a native CVPixelBuffer again. It turns out this scaling is not needed
and that the H264VideoToolBoxEncoder can handle the scaling internally.
BUG=b/30939444
Review-Url: https://codereview.webrtc.org/2258103003
Cr-Commit-Position: refs/heads/master@{#13835}
Reason for revert:
Seems to break an external client.
Original issue's description:
> Cleanup of the AudioDeviceBuffer class.
>
> WebRTC works on 10ms buffer sizes in both directions but this class has contained
> support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.
>
> It also updates the style to follow the Google C++ style guide.
>
> Finally, I remove very old (not tested and not maintained) support for file
> handling since the code is never used. It was more or less dead code.
>
> BUG=NONE
> R=magjed@webrtc.org
>
> Committed: https://crrev.com/cf327b45b9f5738950d4fca2b6a7b6030d508cdf
> Cr-Commit-Position: refs/heads/master@{#13833}
TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE
Review-Url: https://codereview.webrtc.org/2260183002
Cr-Commit-Position: refs/heads/master@{#13834}
WebRTC works on 10ms buffer sizes in both directions but this class has contained
support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.
It also updates the style to follow the Google C++ style guide.
Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.
BUG=NONE
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/2256833003 .
Cr-Commit-Position: refs/heads/master@{#13833}
Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support.
BUG=webrtc:6228
NOTRY=True
Review-Url: https://codereview.webrtc.org/2252413002
Cr-Commit-Position: refs/heads/master@{#13831}
This allows to at least get a camera error back if the camera thread freezes. Application can use this as a signal to restart the program.
R=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2257123002
Cr-Commit-Position: refs/heads/master@{#13830}
When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled.
BUG=webrtc:6215
NOTRY=True
Review-Url: https://codereview.webrtc.org/2250893003
Cr-Commit-Position: refs/heads/master@{#13829}
