TURN server sockets were being destroyed asynchronously, which could
happen after the TurnServer itself (and even the VirtualSocketServer
used by the sockets) were destroyed.
This is fixed easily by using an AsyncInvoker (to ensure the async
operation doesn't occur after its initiator is destroyed), and keeping
the objects waiting for deletion in a unique_ptr vector.
Review-Url: https://codereview.webrtc.org/2264343002
Cr-Commit-Position: refs/heads/master@{#13907}
Normally, when creating a data channel with an out-of-range ID,
createDataChannel returns nullptr. But due to an off-by-one
error, creating a data channel with ID 1023 returns a data channel
that silently fails later.
This probably occurred because it wasn't clear whether "kMaxSctpSid" was an
inclusive or exclusive maximum, so I changed the value to
"kMaxSctpStreams". This wasn't caught by unit tests because the
off-by-one error persisted to the unit tests as well.
Also getting rid of some dead code. We were adding SCTP streams to the
ContentDescription object but they weren't being used.
BUG=619849
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2254003002 .
Cr-Commit-Position: refs/heads/master@{#13906}
We've seen some cases of nonrecoverable runtime error when entering the foreground. This is a theoretical fix to see if we can restart after willEnterForeground in didBecomeActive instead.
NOTRY=True
BUG=
Review-Url: https://codereview.webrtc.org/2258583004
Cr-Commit-Position: refs/heads/master@{#13903}
If an error happens in the GetAudio call, for instance when corrupt
payloads are inserted, GetAudio wil return an error. In this case, the
audio frame is not always correctly populated, which is to be expected.
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2272963002
Cr-Commit-Position: refs/heads/master@{#13902}
This implementation interprets payloads of size 1 as codec-internal SID
frames, marking the start of a CNG period. Changes were made to other
parts of the test payload chain, since it had to make use of the virtual
payload size in the case of header-only RTP files.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2275903002
Cr-Commit-Position: refs/heads/master@{#13901}
Change the previous GN configs to build GYP instead
(since we'll keep GYP around for a while) but exclude tests
and examples for that config, since we'll only support the production
code for 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/2274713003
Cr-Commit-Position: refs/heads/master@{#13900}
iOS tests packaged into an .app uses the same way of
defining resources (the data attribute). Some iOS
simulator tests are failing due to missing resources, so
let's sync them all.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2277753003
Cr-Commit-Position: refs/heads/master@{#13898}
When rolling Chromium into WebRTC, these fail to compile since chromium
no longer supports GYP.
BUG=webrtc:6252
NOTRY=True
Review-Url: https://codereview.webrtc.org/2275973003
Cr-Commit-Position: refs/heads/master@{#13892}
Reason for revert:
Breaks most of chromium.webrtc.fyi bots.
Original issue's description:
> GN build rules for four audio processing test executables
>
> click_annotate, intelligibility_proc, nonlinear_beamformer_test, and
> transient_suppression_test.
>
> BUG=webrtc:5949
>
> Committed: https://crrev.com/538b5606a3fb6310aab7a7e747aee16eac885f02
> Cr-Commit-Position: refs/heads/master@{#13890}
TBR=kjellander@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2274813004
Cr-Commit-Position: refs/heads/master@{#13891}
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}
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}
