There's no change in functionality, which was verified by adding
an 'else' catch-all clause in the loop with an RTC_NOTREACHED()
statement. See patchset #3.
This is mostly a cosmetic change that modifies the loop such that
it's guaranteed that Remove() is always called for transceivers
whose state is "stopped" and there's just one place where Remove()
is called.
Bug: none
Change-Id: Iffe237bb2f08e5e6ef316a6b76c4b183df671f3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215232
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33765}
BaseChannel adds and removes receive streams on the worker thread
(UpdateRemoteStreams_w) and then posts a task to the network thread to
update the demuxer criteria. Until this happens, OnRtpPacket() keeps
forwarding "recently removed" ssrc packets to the WebRtcVideoChannel.
Furthermore WebRtcVideoChannel::OnPacketReceived() posts task from the
network thread to the worker thread, so even if the demuxer criteria was
instantly updated we would still have an issue of in-flight packets for
old ssrcs arriving late on the worker thread inside WebRtcVideoChannel.
The wrong ssrc could also arrive when the demuxer goes from forwarding
all packets to a single m= section to forwarding to different m=
sections. In this case we get packets with an ssrc for a recently
created m= section and the ssrc was never intended for our channel.
This is a problem because when WebRtcVideoChannel sees an unknown ssrc
it treats it as an unsignalled stream, creating and destroying default
streams which can be very expensive and introduce large delays when lots
of packets are queued up.
This CL addresses the issue with callbacks for when a demuxer criteria
update is pending and when it has completed. During this window of time,
WebRtcVideoChannel will drop packets for unknown ssrcs.
This approach fixes the race without introducing any new locks and
packets belonging to ssrcs that were not removed continue to be
forwarded even if a demuxer criteria update is pending. This should make
a=inactive for 50p receive streams a glitch-free experience.
Bug: webrtc:12258, chromium:1069603
Change-Id: I30d85f53d84e7eddf7d21380fb608631863aad21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214964
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33757}
This denies the ability to request RTP data channels to callers.
Later CLs will rip out the actual code for creating these channels.
Bug: chromium:928706
Change-Id: Ibb54197f192f567984a348f1539c26be120903f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177901
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33740}
When fixing so that RemoteAudioSource does not end the track just
because the audio channel is gone in Unified Plan[1], this made it
possible for ~PeerConnection to delete all objects, including deleting
the MediaStreamTrack and its RemoteAudioSource, when all tracks are not
in an ended state.
In a real application or Chromium, the PeerConnection would not be
destroyed prior to closing and not hit this DCHECK. But in upstream
dependent projects' unit tests, it would be possible for ref counted
tracks to be destroyed when the track are still kLive, and as a
side-effect hit this DCHECK.
sinks_ is just a list of raw pointers, and whether or not we have done
sinks_.clear() prior to destruction is irrelevant going forward.
[1] https://webrtc-review.googlesource.com/c/src/+/214136
Bug: chromium:1121454
Change-Id: If6cf3dffcd3cb47d46694755b5dc45fa381285fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215226
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33739}
Also removing a count check from DestroyTransceiverChannel that's
not useful right now. We can bring it back when we have
DestroyChannelInterface better under control as far as Invokes goes.
Bug: none
Change-Id: I8e9c55a980f8f20e8b996fdc461fd90b0fbd4f3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215201
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33730}
This reverts commit a743303211b89bbcf4cea438ee797bbbc7b59e80.
Reason for revert: Breaks downstream tests that attempt to call FindHeaderExtensionByUri with 2 arguments. Could you keep the old 2-argument method declaration and just forward the call to the new 3-argument method with a suitable no-op filter?
Original change's description:
> Fix RTP header extension encryption
>
> Previously, RTP header extensions with encryption had been filtered
> if the encryption had been activated (not the other way around) which
> was likely an unintended logic inversion.
>
> In addition, it ensures that encrypted RTP header extensions are only
> negotiated if RTP header extension encryption is turned on. Formerly,
> which extensions had been negotiated depended on the order in which
> they were inserted, regardless of whether or not header encryption was
> actually enabled, leading to no extensions being sent on the wire.
>
> Further changes:
>
> - If RTP header encryption enabled, prefer encrypted extensions over
> non-encrypted extensions
> - Add most extensions to list of extensions supported for encryption
> - Discard encrypted extensions in a session description in case encryption
> is not supported for that extension
>
> Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
> into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
> header extensions will prevent any RTP packets being sent/received.
>
> Bug: webrtc:11713
> Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33723}
TBR=deadbeef@webrtc.org,terelius@webrtc.org,hta@webrtc.org,lennart.grahl@gmail.com
Change-Id: I7df6b0fa611c6496dccdfb09a65ff33ae4a52b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33727}
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
header extensions will prevent any RTP packets being sent/received.
Bug: webrtc:11713
Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33723}
simplifying the code and comparing against the value libsrtp expects
and increase verbosity of error logging related to key length mismatches.
BUG=None
Change-Id: Icc0d0121d2983e23c95b0f972a5f6cac1d158fd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213146
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33685}
The SdpOfferAnswerHandler::ssrc_generator_ variable is used from
multiple threads.
Adding thread checks + tests for UniqueNumberGenerator along the way.
Bug: webrtc:12666
Change-Id: Id2973362a27fc1d2c7db60de2ea447d84d18ae3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33668}
StatsCollector::ExtractSessionInfo was run fully on the signaling thread
and several calls were being made to methods that need to run on the
network thread.
Additionally, BaseChannel::transport_name() was being read directly
on the signaling thread (needs to be read on the network thread).
So with shifting the work that needs to happen on the network thread
over to that thread, we now also grab the transport name there and
use the name with the work that still needs to happen on the signaling
thread.
These changes allow us to remove Invoke<>() calls to the network thread from
callback functions implemented in PeerConnection:
* GetPooledCandidateStats
* GetTransportNamesByMid
* GetTransportStatsByNames
* Also adding a correctness thread check to:
* GetLocalCertificate
* GetRemoteSSLCertChain
Because PeerConnection now has a way of knowing when things are
or have been uninitialized on the network thread, all of these
functions can exit early without doing throw away work.
Additionally removing thread hops that aren't needed anymore from
JsepTransportController.
Using the RTC_LOG_THREAD_BLOCK_COUNT() macro in GetStats, the number
of Invokes (when >1), goes down by 3. Typically from 8->5, 7->4, 6->3.
Bug: webrtc:11687
Change-Id: I06ab25eab301e192e99076d7891444bcb61b491f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214135
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33656}
This also deletes unused method has_channels() and moves us closer
to having the ChannelManager just be a factory class. Once we get there
the ownership of the channels themselves can be with the classes that
hold pointers to them. Today the initialization and teardown of those
classes need to be synchronized with ChannelManager. But there's no
real value in keeping the channel pointers owned elsewhere.
Places where we have naked un-owned channel pointers:
* RtpTransceiver for voice and video
* PeerConnection::data_channel_controller_ (rtp data channel)
Bug: webrtc:11994
Change-Id: Id6df27414cc57b6ecf0f7f769fdb9603ed114bfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214440
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33654}
The RTCReceivedRtpStreamStats hierarchy, which inherit from
RTCRtpStreamStats, already contain members ssrc, kind, codec_id and
transport_id so there's no need to list them inside
RTCRemoteInboundrtpStreamStats.
This CL removes duplicates so that we don't DCHECK-crash on Android,
and adds a unit test ensuring we never accidentally list the same
member twice.
Bug: webrtc:12658
Change-Id: I27925eadddc6224bf6d6a91784ed7cafd7a4cfb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214343
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33649}
This reverts commit 5a40b3710545edfd8a634341df3de26f57d79281.
Reason for revert: Fixed the bug and ran layout tests.
Original change's description:
> Revert "Use the new DNS resolver API in PeerConnection"
>
> This reverts commit acf8ccb3c9f001b0ed749aca52b2d436d66f9586.
>
> Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview.
>
> Original change's description:
> > Use the new DNS resolver API in PeerConnection
> >
> > Bug: webrtc:12598
> > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33561}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta@webrtc.org
>
> Bug: webrtc:12598
> Change-Id: Idc9853cb569849c49052f9cbd865614710fff979
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33591}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12598
Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33647}
The RemoteAudioSource has an AudioDataProxy that acts as a sink, passing
along data from AudioRecvStreams to the RemoteAudioSource. If an SSRC is
changed (or other reconfiguration happens) with SDP, the recv stream and
proxy get recreated.
In Plan B, because remote tracks maps 1:1 with SSRCs, it made sense to
end remote track/audio source in response to this. In Plan B, a new
receiver, with a new track and a new proxy would be created for the new
SSRC.
In Unified Plan however, remote tracks correspond to m= sections. The
remote track should only end on port:0 (or RTCP BYE or timeout, etc),
not because the recv stream of an m= section is recreated. The code
already supports changing SSRC and this is working correctly, but
because ~AudioDataProxy() would end the source this would cause the
MediaStreamTrack of the receiver to end (even though the media engine
is still processing the remote audio stream correctly under the hood).
This issue only happened on audio tracks, and because of timing of
PostTasks the track would kEnd in Chromium *after* promise.then().
This CL fixes that issue by not ending the source when the proxy is
destroyed. Destroying a recv stream is a temporary action in Unified
Plan, unless stopped. Tests are added ensuring tracks are kLive.
I have manually verified that this CL fixes the issue and that both
audio and video is flowing through the entire pipeline:
https://jsfiddle.net/henbos/h21xec97/122/
Bug: chromium:1121454
Change-Id: Ic21ac8ea263ccf021b96a14d3e4e3b24eb756c86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214136
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33645}
It's triggering when CreateAnswerWithDifferentSslRoles is run
so marking that test for follow-up in the TODO.
Commenting out the check to make bots go green.
Tbr: hta@webrtc.org
Bug: none
Change-Id: I3fe7b67f12c45aace05e2d7e7c267e10cdf3f8f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214138
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33643}
The testing code prevents the production code from protecting the
member variables properly. The convenience methods for testing
purposes, can be located with the testing code.
Bug: none
Change-Id: Ieda248a199db84336dfafbd66c93c35508ab2582
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213661
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33635}
Also updating SocketOptionsMergedOnSetTransport test code to make the
call to SetRtpTransport from the right context.
Bug: webrtc:12636
Change-Id: I343851bcf8ac663d7559128d12447a9a742786f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33633}
This is useful to understand how often we block in certain parts of the
api and track improvements/regressions.
There are two macros, both are only active for RTC_DCHECK_IS_ON builds:
* RTC_LOG_THREAD_BLOCK_COUNT()
Example:
void MyClass::MyFunction() {
RTC_LOG_THREAD_BLOCK_COUNT();
thread_->Invoke<void>([this](){ DoStuff(); });
}
When executing this function during a test, the output could be:
(my_file.cc:2): Blocking MyFunction: total=1 (actual=1, would=0)
The words 'actual' and 'would' reflect whether an actual thread switch
was made, or if in the case of a test using the same thread for more
than one role (e.g. signaling, worker, network are all the same thread)
that an actual thread switch did not occur but it would have occurred
in the case of having dedicated threads. The 'total' count is the sum.
* RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(x)
Example:
void MyClass::MyFunction() {
RTC_LOG_THREAD_BLOCK_COUNT();
thread_->Invoke<void>([this](){ DoStuff(); });
thread_->Invoke<void>([this](){ MoreStuff(); });
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
}
When a function is known to have blocking calls and we want to not
regress from the currently known number of blocking calls, we can use
this macro to state that at a certain point in a function, below
where RTC_LOG_THREAD_BLOCK_COUNT() is called, there must have occurred
no more than |x| (total) blocking calls. If more occur, a DCHECK will
hit and print out what the actual number of calls was:
# Fatal error in: my_file.cc, line 5
# last system error: 60
# Check failed: blocked_call_count_printer.GetTotalBlockedCallCount() <= 1 (2 vs. 1)
Bug: webrtc:12649
Change-Id: Ibac4f85f00b89680601dba54a651eac95a0f45d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213782
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33632}
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}
