It turns out that this check always returns 'true' and is
also not safe to do from this thread.
Bug: webrtc:12635
Change-Id: Iebc0097042020707678f3a1ad9c912b227a4257c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213600
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33626}
Make a few more members const, remove members that aren't used,
set max ssl version number on construction and remove setter.
Bug: none
Change-Id: I6c1a7cabf1e795e027f1bc53b994517e9aef0e93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33622}
This stops pending internal callbacks from performing unnecessary
operations when closed.
Also update tests pc tests to call Close().
This will allow PeerConnection to be able to expect the
normal path to be that IsClosed() be true in the dtor
once all 'normal' paths do that
Bug: webrtc:12633
Change-Id: I3882bedf200feda0d04594adeb0fdac85bfef652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213426
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33617}
Before, the calls went through the signaling thread and
blocked while the operation completed on the worker.
Bug: webrtc:12601
Change-Id: I58991fa98a55d0fa9304a68bd85bb269f1f123d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212619
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33615}
* A ChannelManager instance is now created via ChannelManager::Create()
* Initialization is performed inside Create(), RAII.
* All member variables in CM are now either const or RTC_GUARDED_BY
the worker thread.
* Removed dead code (initialization and capturing states are gone).
* ChannelManager now requires construction/destruction on worker thread.
- one fewer threads that its aware of.
* media_engine_ pointer removed from ConnectionContext.
* Thread policy changes moved from ChannelManager to ConnectionContext.
These changes will make a few other issues easier to fix, so tagging
those bugs with this CL.
Bug: webrtc:12601, webrtc:11988, webrtc:11992, webrtc:11994
Change-Id: I3284cf0a08c773e628af4124e8f52e9faddbe57a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212617
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33614}
Going forward, we'll need to read this value from other threads than
signaling, so I've moved the initialization into the constructor.
Bug: none
Change-Id: I56b00d38c86788cbab9a2055719074ea48f4750f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213185
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33613}
Before the tests were using the current thread for three roles,
signaling, worker and network.
Also, removing redundant test and unnecessary setters for test.
Bug: none
Change-Id: Id132b6290b78765dc075ede9483dd2d12b201130
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212615
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33612}
Deleting obsolete stats. Spec: https://www.w3.org/TR/webrtc-stats/
1. RTCInbound/OutboundRtpStats.isRemote: No longer useful with remote stream stats
2. RTCIceCandidateStats.deleted: This field was obsoleted because if the ICE candidate is deleted it no longer appears in getStats()
I also marked as many other obsoleted stats possible according to spec. I am not as confident to delete them but feel free to comment to let me know if anything is off / can be deleted.
Bug: webrtc:12583
Change-Id: I688d0076270f85caa86256349753e5f0e0a44931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33549}
`RTCInboundRtpStreamStats.lastPacketReceivedTimestamp` must be a time
value in milliseconds with Unix epoch as time origin (see
bugs.webrtc.org/12605#c4).
This change fixes both audio and video `RTCInboundRtpStreamStats` stats.
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12605
Change-Id: I68157fcf01a5933f3d4e5d3918b4a9d3fbd64f16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212865
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33547}
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33545}
In refactoring CL https://webrtc-review.googlesource.com/c/src/+/210340,
the RTCRemoteInboundRtpStreamStats hierarchy was updated to inherit from
RTCReceivedRtpStreamStats but we forgot to update the
WEBRTC_RTCSTATS_IMPL() macro to say that RTCReceivedRtpStreamStats is
the parent. As a consequence, RTCReceivedRtpStreamStats's members
(jitter and packetsLost) were not included when iterating over all
members of RTCRemoteInboundRtpStreamStats, which means these two merics
stopped being exposed to JavaScript in Chromium.
There is sadly no way to safe-guard against this, but the fix is simple.
TBR=hta@webrtc.org,meetwudi@gmail.com
Bug: webrtc:12532
Change-Id: I0179dad6eaa592ee36cfe48978f2fc22133b8f45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212866
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33543}
Both inbound RTP stats `estimatedPlayoutTimestamp` and
`lastPacketReceivedTimestamp` are surfaced to JS land as
`DOMHighResTimeStamp` - i.e., time values in milliseconds.
This CL fixes `lastPacketReceivedTimestamp` which is incorrectly
surfaced as time value in seconds.
Bug: webrtc:12605
Change-Id: I290103071cca3331d2a3066b6b6b9fcb4f4fd0af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212742
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33530}
There's another copy in peer_connection_integrationtest.cc, which is
used by one test.
Bug: None
Change-Id: I4f81e107767253357f8eeb83d318133b8ee86698
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212027
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33480}
This increases the running time of the test, but seems to be needed
to avoid flakiness on Windows.
Bug: webrtc:12587
Change-Id: Id8c49910e276b2754244d977d66241e6e211c720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212023
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33473}
This pair of tests will ensure that the SCTP layer's response to
MTU size changes has not been modified.
Bug: webrtc:12495
Change-Id: If9776ad399871e9f01b38715594b732e156118ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211246
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33459}
Spec: https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict*
According to the spec, |RTCReceivedRtpStreamStats| is the base class for |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. This structure isn't visible in JavaScript but it's important to bring it up to spec for the C++ part. This CL adds the barebone |RTCReceivedRtpStreamStats| with a bunch of TODOs for later migrations.
This commit makes the minimum |RTCReceivedRtpStreamStats| and rebase |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats| to use the new class as the parent class.
This commit also moves |jitter| and |packets_lost| to |RTCReceivedRtpStreamStats|, from |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. Moving these two first because they are the two that exist in both subclasses for now.
Bug: webrtc:12532
Change-Id: I0ec74fd241f16c1e1a6498b6baa621ca0489f279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210340
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33435}
use the helper functions
srtp_crypto_policy_set_from_profile_for_rtp
and
srtp_crypto_policy_set_from_profile_for_rtcp
provided by libsrtp to initialize the rtp and rtcp policies.
BUG=None
Change-Id: Ib1560c0fc1c06d9e79c1f871b028555b3b4d66d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33399}
documents why it is safe to not follow libsrtp's advice
to ensure additional SRTP_MAX_TRAILER_LEN bytes are available
when calling srtp_protect (and similar srtcp functions).
BUG=None
Change-Id: I504645d21553160f06133fd8bb3ee79e178247da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209064
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33396}
This is a reland of a37f2bd9421868e222d591d3371486a6ab939fd6
Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}
Bug: none
Change-Id: If46a6679ac0fc947797dd7be87626ef7702faca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208845
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33349}
This makes the proxy macros less confusing when the secondary thread
is sometimes the worker thread, sometimes the networking thread.
Bug: none
Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33346}
Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!
Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.
Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
