We detect an unreasonable state (caused by a bad encoded stream)
before it can lead to problems, and handle it by resetting the
decoder.
NOPRESUBMIT=true
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2255203002
Cr-Commit-Position: refs/heads/master@{#13888}
Fixes the following error for Android GN bots when trying
to roll Chromium into WebRTC.
Full logs at:
https://build.chromium.org/p/tryserver.webrtc/builders/android_gn_dbg/builds/13405/steps/generate_build_files/logs/stdio
/b/c/b/android_gn_dbg/src/buildtools/linux64/gn gen //out/Debug --check
-> returned 1
ERROR at //build/config/android/internal_rules.gni:140:23: Can't load input file.
deps += [ "${_target_label}__build_config" ]
^-------------------------------
Unable to load:
/b/c/b/android_gn_dbg/src/third_party/byte_buddy/BUILD.gn
I also checked in the secondary tree for:
/b/c/b/android_gn_dbg/src/build/secondary/third_party/byte_buddy/BUILD.gn
GN gen failed: 1
BUG=522043
NOTRY=True
Review-Url: https://codereview.webrtc.org/2268343002
Cr-Commit-Position: refs/heads/master@{#13886}
Removed the OutputMixer part of the new mixer and renamed the new
mixer from NewAudioConferenceMixer to AudioMixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2249213005
Cr-Commit-Position: refs/heads/master@{#13883}
Changes to the mixer unittests:
Removed the tests related to the former 'OutputMixer', as it's going
to be removed. Removed incorrect comparison tests with the old mixer
because doing identical mixing decisions with the old mixer proved
unviable.
When the new mixer went from kMaximumAmountOfMixedAudioSources in the
last iteration to kMaximumAmountOfMixedAudioSources+1, it could hit an
RTC_NOTREACHED(); Added fix to mixer and test
AudioMixer.RampedOutSourcesShouldNotBeMarkedMixed that covers that
case.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2253153004
Cr-Commit-Position: refs/heads/master@{#13880}
The old and new getStats are very different. This CL proposes rewriting
the new getStats from scratch with a bottom-up approach, starting with
the fundamental stats classes. This will allow cleaner and more
efficient code that is more aligned with the spec.
RTCStats and subclasses are the equivalent to RTCStats and RTCStats-
-derived dictionaries from the specs[1][2]. The dictionary members are
public member variables of type RTCStatsMember<T>, where T is one of the
supported types. All members derive from RTCStatsMemberInterface and
iteration of members is possible with RTCStats::Members().
The members are not stored in a map for performance and readability.
Type checking is supported with static class variables, kType.
Only the supported member types T are specialized and may be
instantiated, and sequences are supported with std::vector<...>. Type
checking is again supported with static class variables, kType.
RTCStatsReport is the equivalent from the spec[3], and maps RTCStats::id
to RTCStats-objects. RTCStatsReport is reference counted. It and its
contained stats may be destroyed on any thread. When the
RTCStatsCollector is added in a follow-up CL, it will return const
references to the RTCStatsReports. This means copies don't have to be
made for multiple stats observers or when jumping threads. In fact, no
copies of any stats will have to be made in surfacing stats to Blink.
[1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstats-dictionary
[2] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html
[3] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object
This adds the new folder webrtc/stats/, with target rtc_stats and binary
rtc_stats_unittests. Public api headers are placed in webrtc/api/ and
.cc files are placed in webrtc/stats/.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2241093002
Cr-Commit-Position: refs/heads/master@{#13879}
Added a level indicator to the new mixer. The level indicator is
webrtc::voe::AudioLevel. It computes the current audio level, which is
used all the way up to peerconnection.
This is part of the project to rewrite the old conference mixer and
output mixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2230823004
Cr-Commit-Position: refs/heads/master@{#13878}
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}
