Add an attribute to the RTCConfiguration which can be used by specific
mobile devices so that the IPv6 ICE candidates on WiFi will not be collected.
BUG=b/35725283
Review-Url: https://codereview.webrtc.org/2731813002
Cr-Commit-Position: refs/heads/master@{#17100}
for consistency with the WebRTC 1.0 standard as suggested in a TODO.
BUG=None
Review-Url: https://codereview.webrtc.org/2732663004
Cr-Commit-Position: refs/heads/master@{#17077}
It was defined unconditionally and the code for non-HAVE_SRTP was unmaintained
and failed to compile.
BUG=webrtc:7294
Review-Url: https://codereview.webrtc.org/2729373002
Cr-Commit-Position: refs/heads/master@{#17074}
This CL is a reland of https://codereview.webrtc.org/2722423003 which got
reverted due to compile errors when rolling into Chromium.
Original CL description:
Improve testing of SRTP external auth code paths.
Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium
but developed in WebRTC, which made testing rather complicated. This caused
some trouble in the past (e.g. https://crbug.com/628400#c1)
This CL helps in that the external auth code is now compiled with WebRTC
and the srtpfilter integration gets tested.
BUG=chromium:628400
Review-Url: https://codereview.webrtc.org/2735613002
Cr-Commit-Position: refs/heads/master@{#17052}
Reason for revert:
Breaks compilation in FYI bots, e.g. here:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/9314
FAILED: obj/third_party/webrtc/pc/rtc_pc/channel.obj
ninja -t msvc -e environment.x86 -- E:\b\c\goma_client/gomacc.exe "E:\b\depot_tools\win_toolchain\vs_files\d3cb0e37bdd120ad0ac4650b674b09e81be45616\VC\bin\amd64_x86/cl.exe" /nologo /showIncludes /FC @obj/third_party/webrtc/pc/rtc_pc/channel.obj.rsp /c ../../third_party/webrtc/pc/channel.cc /Foobj/third_party/webrtc/pc/rtc_pc/channel.obj /Fd"obj/third_party/webrtc/pc/rtc_pc_cc.pdb"
e:\b\c\b\win_builder\src\third_party\webrtc\pc\channel.cc(176): error C2819: type 'cricket::SrtpFilter' does not have an overloaded member 'operator ->'
e:\b\c\b\win_builder\src\third_party\webrtc\pc\srtpfilter.h(45): note: see declaration of 'cricket::SrtpFilter'
e:\b\c\b\win_builder\src\third_party\webrtc\pc\channel.cc(176): note: did you intend to use '.' instead?
e:\b\c\b\win_builder\src\third_party\webrtc\pc\channel.cc(176): error C2232: '->cricket::SrtpFilter::EnableExternalAuth': left operand has 'class' type, use '.'
Original issue's description:
> Improve testing of SRTP external auth code paths.
>
> Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium
> but developed in WebRTC, which made testing rather complicated. This caused
> some trouble in the past (e.g. https://crbug.com/628400#c1)
>
> This CL helps in that the external auth code is now compiled with WebRTC
> and the srtpfilter integration gets tested.
>
> BUG=chromium:628400
>
> Review-Url: https://codereview.webrtc.org/2722423003
> Cr-Commit-Position: refs/heads/master@{#17030}
> Committed: ac170d5c21TBR=deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:628400
Review-Url: https://codereview.webrtc.org/2734643002
Cr-Commit-Position: refs/heads/master@{#17031}
Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium
but developed in WebRTC, which made testing rather complicated. This caused
some trouble in the past (e.g. https://crbug.com/628400#c1)
This CL helps in that the external auth code is now compiled with WebRTC
and the srtpfilter integration gets tested.
BUG=chromium:628400
Review-Url: https://codereview.webrtc.org/2722423003
Cr-Commit-Position: refs/heads/master@{#17030}
With ENABLE_EXTERNAL_AUTH, external auth will only be used depending
on the selected cipher (allowed for non-GCM, not allowed for GCM).
BUG=webrtc:5222, chromium:628400
Review-Url: https://codereview.webrtc.org/2720663003
Cr-Commit-Position: refs/heads/master@{#16955}
The value is being moved:
https://github.com/w3c/webrtc-stats/pull/167
Stop collecting this value. Our previous value was incorrect, our RTT
value was a smoothed value based on STUN pings but the spec says it
should be based on RTCP timestamps in RTCP Receiver Report (RR) on
inbound streams with isRemote=true (not supported).
Updated some bug references.
BUG=webrtc:7065, webrtc:7066
Review-Url: https://codereview.webrtc.org/2722633005
Cr-Commit-Position: refs/heads/master@{#16931}
We were already OWNERS of these files, but when these files were moved
from webrtc/api/ to webrtc/pc/ and a new OWNERS file created our
ownership was accidentally not moved.
Becoming per-file=rtcstats* OWNER of webrtc/pc/ which includes:
rtcstats_integrationtest.cc
rtcstatscollector.cc
rtcstatscollector.h
rtcstatscollector_unittest.cc
Dropping ownership of webrtc/api/ which no longer includes any
rtcstats* files.
Already OWNER of all of webrtc/api/stats/ which includes:
rtcstats.h
rtcstats_objects.h
rtcstatscollectorcallback.h
rtcstatsreport.h
Already OWNER of all of webrtc/stats/ which includes:
rtcstats.cc
rtcstats_objects.cc
rtcstats_unittest.cc
rtcstatsreport.cc
rtcstatsreport_unittest.cc
BUG=webrtc:7060
TBR=hta@webrtc.org, deadbeef@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2726563002
Cr-Commit-Position: refs/heads/master@{#16906}
Collected in accordance with the spec:
https://w3c.github.io/webrtc-stats/#candidatepair-dict*
totalRoundTripTime is collected as the sum of rtt measurements, it was
previously not collected.
currentRoundTripTime is collected as the latest rtt measurement, it
was previously collected as a smoothed value, which was incorrect.
Connection is updated to collect these values which are surfaced
through ConnectionInfo.
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2719523002
Cr-Commit-Position: refs/heads/master@{#16905}
|transport_overhead_per_packet_| and |rtp_overhead_per_packet_| could
be read from and written to on different threads concurrently. This CL
introduces a lock to GUARD these variables.
NOTRY because master.tryserver.webrtc.linux_ubsan_vptr is broken, all
other tests pass.
BUG=webrtc:7231
NOTRY=True
Review-Url: https://codereview.webrtc.org/2710363003
Cr-Commit-Position: refs/heads/master@{#16900}
Connection::nominated() is updated to mean
(remote_nomination_ || acked_nomination_), which means both a
controlling and controlled agent can be said to be "nominated".
Previously this was (remote_nomination_ > 0) which only applies to the
controlling agent.
PortTest.TestNomination added to test nomination values and nomination
stat.
This value is surfaced through cricket::ConnectionInfo::nominated.
RTCStatsCollector uses this value in its collection of
RTCIceCandidatePairStats.
RTCStatsCollectorTest.CollectRTCIceCandidatePairStats updated to test
that ConnectionInfo::nominated is surfaced using mocks.
rtcstats_integrationtest.cc updated to expect nomination set without
using mocks.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-nominated
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2709293004
Cr-Commit-Position: refs/heads/master@{#16855}
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
The issue was that VideoCapturerTrackSource was adding a reference
to itself, causing it to not be deleted even after no external objects
reference it. The objects underneath it (threads for instance) may
then be destroyed before the object dereferences them.
BUG=webrtc:6487
Review-Url: https://codereview.webrtc.org/2717023002
Cr-Commit-Position: refs/heads/master@{#16841}
I'd like to make a change to rtc::Bind in another CL, and that will
be easier if there are fewer lines of code to modify.
BUG=None
Review-Url: https://codereview.webrtc.org/2719683002
Cr-Commit-Position: refs/heads/master@{#16838}
Where "TRANSPORT attributes" refers to:
https://tools.ietf.org/html/draft-ietf-mmusic-sdp-mux-attributes-16
The BUNDLE draft now says that these attributes can
(in fact, MUST) be omitted when m= sections are bundled
(they only need to go in one of the bundled m= sections),
so we should start accepting that SDP.
This CL doesn't fix "a=rtcp-mux", unfortunately. That will be easier
to fix once we've split apart an "RtpTransport" object from
BaseChannel.
BUG=webrtc:6351
Review-Url: https://codereview.webrtc.org/2647593003
Cr-Commit-Position: refs/heads/master@{#16782}
Running video loopback on https://appr.tc/ revealed that it is possible
to use the same SSRC for a local and remote audio or video track. This
caused a DCHECK crash. The constructor of TrackMediaInfoMap is updated
to support this mapping and the unittest is updated (moved and modified
a test from being a death test to being a non-death test).
I've verified that this fixes the bug.
BUG=chromium:693087
Review-Url: https://codereview.webrtc.org/2703783002
Cr-Commit-Position: refs/heads/master@{#16713}
This change adds a flag, use_sctpmap, to DataContentDescription. The deserialization code sets the flag based on the format of the m= line.
There were already unit tests using SDP in the new format, so I just updated them to check use_sctpmap was set as expected.
The change to mediasession copies use_sctpmap from the offered DataContentDescription to the answer.
I haven't figured out how to test this change yet, but wanted to get feedback before continuing.
BUG=chromium:686212
Review-Url: https://codereview.webrtc.org/2690943011
Cr-Commit-Position: refs/heads/master@{#16686}
This utility class can be used to represent either an error or a
successful return value. Follows the pattern of StatusOr in the protobuf
library.
This will be used by ORTC factory methods; for instance, CreateRtpSender
will either return an RtpSender or an error if the parameters are
invalid or some other failure occurs.
This CL also moves RTCError classes to a separate file, and adds tests
that were missing before.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2692723002
Cr-Commit-Position: refs/heads/master@{#16659}
"bundle_transport_name" is no longer relevant here, and
"rtcp_mux_required" is implied by whether or not an RTCP transport is
passed in.
BUG=None
Review-Url: https://codereview.webrtc.org/2689503002
Cr-Commit-Position: refs/heads/master@{#16551}
It wasn't being used at all, and there's no need to tie LocalAudioSource to
PeerConnection.
BUG=None
Review-Url: https://codereview.webrtc.org/2682253002
Cr-Commit-Position: refs/heads/master@{#16550}
This helps show where ownership is transfered between objects.
Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.
BUG=None
TBR=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
This is the naming scheme we've been using for internal interfaces.
Also, this CL will introduce a PacketTransportInterface in the webrtc namespace,
which would get too easily confused with the rtc:: one:
https://codereview.webrtc.org/2675173003/
BUG=None
Review-Url: https://codereview.webrtc.org/2679103006
Cr-Commit-Position: refs/heads/master@{#16539}
Reason for revert:
Fix the problem.
Original issue's description:
> Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
>
> Reason for revert:
> Breaks downstream build.
>
> Original issue's description:
> > Add QP sum stats for received streams.
> >
> > This is not implemented yet in any of the decoders.
> >
> > BUG=webrtc:6541
> >
> > Review-Url: https://codereview.webrtc.org/2649133005
> > Cr-Commit-Position: refs/heads/master@{#16475}
> > Committed: ff0e72fd16
>
> TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6541
>
> Review-Url: https://codereview.webrtc.org/2680893002 .
> Cr-Commit-Position: refs/heads/master@{#16480}
> Committed: 69fb2cca4dTBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
BUG=webrtc:6541
Review-Url: https://codereview.webrtc.org/2681663005
Cr-Commit-Position: refs/heads/master@{#16511}
Stop the RtcEventLog when the PeerConnection is closed so that Chrome
will not crash because of creating too many threads.
BUG=chromium:687553
Review-Url: https://codereview.webrtc.org/2682433005
Cr-Commit-Position: refs/heads/master@{#16482}
Reason for revert:
Breaks downstream build.
Original issue's description:
> Add QP sum stats for received streams.
>
> This is not implemented yet in any of the decoders.
>
> BUG=webrtc:6541
>
> Review-Url: https://codereview.webrtc.org/2649133005
> Cr-Commit-Position: refs/heads/master@{#16475}
> Committed: ff0e72fd16TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6541
Review-Url: https://codereview.webrtc.org/2680893002 .
Cr-Commit-Position: refs/heads/master@{#16480}
Reason for revert:
Breaks downstream build.
Original issue's description:
> RTCInboundRTPStreamStats.qpSum collected.
>
> This was previously only collected for local tracks
> (RTCOutboundRTPStreamStats.qpSum).
>
> Spec: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
>
> This CL also improves some testing in rtcstatscollector_unittest.cc.
> Default and non-default values are tested in the same unittests,
> removing the test that was specific to default-values, which was
> otherwise code duplication.
>
> BUG=webrtc:7065
>
> Review-Url: https://codereview.webrtc.org/2675943002
> Cr-Commit-Position: refs/heads/master@{#16477}
> Committed: cd195bea5eTBR=sakal@webrtc.org,hta@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7065
Review-Url: https://codereview.webrtc.org/2687483002 .
Cr-Commit-Position: refs/heads/master@{#16479}
This was previously only collected for local tracks
(RTCOutboundRTPStreamStats.qpSum).
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
This CL also improves some testing in rtcstatscollector_unittest.cc.
Default and non-default values are tested in the same unittests,
removing the test that was specific to default-values, which was
otherwise code duplication.
BUG=webrtc:7065
Review-Url: https://codereview.webrtc.org/2675943002
Cr-Commit-Position: refs/heads/master@{#16477}
Replaced by assigning value to a local variable, followed by a DCHECK.
Also deletes dead test code under the always false TEST_DIGEST define.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623473004
Cr-Commit-Position: refs/heads/master@{#16476}
This is not implemented yet in any of the decoders.
BUG=webrtc:6541
Review-Url: https://codereview.webrtc.org/2649133005
Cr-Commit-Position: refs/heads/master@{#16475}
Refactor how |codec_id| is set, remove outdated TODO, update comments
with new bugs IDs.
BUG=webrtc:7061
Review-Url: https://codereview.webrtc.org/2670343002
Cr-Commit-Position: refs/heads/master@{#16467}
