The code that extracts certificate stats from an SSLCertificate and its
certificate chain is moved into SSLCertificate::GetStats. The stats
collector code loops through the resulting SSLCertificateStats and
creates the StatsReports for those stats.
This will allow the new stats collector to reuse GetStats in a future
CL.
BUG=chromium:627816, chromium:629436
Review-Url: https://codereview.webrtc.org/2259283002
Cr-Commit-Position: refs/heads/master@{#13917}
The rotation is currently always applied by AVFoundation by
reconfiguring the capture connection video orientation. This CL sets the
rotation field in the frame instead. This avoids the current flash in
the video when the device is rotated, and also avoids reconfiguring the
local encoder and remote decoder when the device is rotated.
BUG=b/30651939
Review-Url: https://codereview.webrtc.org/2271583003
Cr-Commit-Position: refs/heads/master@{#13916}
These functions operate directly on the packet masks, and are thus not directly
involved in the FEC encoding/decoding operation. The ::internal namespace is used
for packet mask-related functions, and will be renamed later on.
This CL should have no impact on functionality or performance.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2269893002
Cr-Commit-Position: refs/heads/master@{#13914}
This removes the warning printouts about unknown header extensions.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2266403005
Cr-Commit-Position: refs/heads/master@{#13912}
The current_rtp_payload_type_ should only be updated when the packet is
actually inserted into the packet buffer, since then the payload type
has been validated. This CL removes an unvalidated setting of this value
that happened after SSRC change or upon first packet.
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2270793003
Cr-Commit-Position: refs/heads/master@{#13910}
It was being set at the wrong point in time and with the address
parameter missing, so it wasn't having any effect.
Review-Url: https://codereview.webrtc.org/2237073002
Cr-Commit-Position: refs/heads/master@{#13909}
I found that, depending on when it's called, ClearGettingPorts may or
may not signal CandidatesAllocationDone, and may or may not continue
to gather more ports/candidates.
I'm fixing this inconsistency by having it always signal
CandidatesAllocationDone (if needed), and always stop gathering until
the next network change event. This makes it equivalent to
StopGettingPorts, except that it allows gathering to be restarted if
a network change occurs.
I also found that P2PTransportChannel was signaling "gathering
complete" even when continual gathering was enabled. This wasn't caught
by the unit tests due to the inconsistency of ClearGettingPorts as
described above.
Review-Url: https://codereview.webrtc.org/2124283003
Cr-Commit-Position: refs/heads/master@{#13908}
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}
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}
