jitterBufferDelay and jitterBufferEmittedCount are defined
in RTCMediaStreamStats for both audio and video.
But for video, they were not populated in RTCInboundRtpStreamStats.
Bug: webrtc:12910
Change-Id: I135d473f055ecfb2c39b078ccf18c1bb9bc4f210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224280
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34398}
When the transport is terminated, if an error has occured, it will
be propagated to the channels.
When such errors can happen at the SCTP level (e.g. out of resources),
RTCError may contain an error code matching the definition at
https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24
If the m= line is rejected or removed from SDP, an error will again be sent
to the data channels, signaling their unexpected transition to closed.
Bug: webrtc:12904
Change-Id: Iea3d8aba0a57bbedb5d03f0fb6f7aba292e92fe8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223541
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34386}
Relanding after adding to chromium stats whitelist:
https://chromium-review.googlesource.com/c/chromium/src/+/2983329
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I3633ee428d07b283b0cc503a84d8fa2e79415dfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223761
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34367}
This reverts commit a27cfbffdfa0bf359628d2164db5b9d6321f9c9c.
Reason for revert: WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise failing.
Original change's description:
> Fix echo return loss stats and add to RTCAudioSourceStats.
>
> This solves two problems:
> * Echo return loss stats weren't being gathered in Chrome, because they
> need to be taken from the audio processor attached to the track
> rather than the audio send stream.
> * The standardized location is in RTCAudioSourceStats, not
> RTCMediaStreamTrackStats. For now, will populate the stats in both
> locations.
>
> Bug: webrtc:12770
> Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34344}
TBR=deadbeef@webrtc.org,hbos@webrtc.org,hbos@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I6b2587d762f005adef67c0d5121f1b58c3b76688
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12770
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223068
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34352}
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34344}
If a bundle is established, then per JSEP, the offerer is required to
include the new track in the bundle, and per BUNDLE, the answerer has
to either accept the track as part of the bundle or reject the track;
it cannot move it out of the group. So we will never need the transport.
Bug: webrtc:12837
Change-Id: I41cae74605fecf454900a958776b95607ccf3745
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221601
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34290}
it.second in JsepTransportCollection::IsConsistent() is of type
std::unique_ptr, which has no operator<< in libstdc++. Even if it
would exist, it would just return the pointer value and not the
content.
Bug: chromium:957519
Change-Id: I17c75db47d63f42b33a551ee2b7afbf5c9a3cfde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222401
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34289}
This entails instantly deleting the transport when it is no longer
referenced by any MID.
Also adds consistency checks to JsepTransportCollection.
Bug: webrtc:12837
Change-Id: I85775aeb676aac3a9aee74280cc72ac87a0f49b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221982
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34273}
This is part of the work to make Bundle handling understandable,
so that we can get it to work right.
Bug: webrtc:12837
Change-Id: I77f046b4bac2d9709460b3b956a2edc3df0cdaac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221745
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34261}
As a side effect, break out pc/simulcast_description.
Step 1: Don't move the {h,cc} files; just declare the targets
so that downstream projects can add dependencies on it.
Bug: webtc:11967
Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34254}
We want to turn off PT based demux because SSRC-based endpoints that
send media prematurely (which is a popular non-standard behavior still
heavily in use) can otherwise get incorrect mappings and unsignalled
ssrc issues because of the PT demux path.
This CL disables PT based demuxing when the MID header extension is
present on all m= sections in the SDP for that kind (audio/video), not
caring if it was in the offer or answer. However if PT demuxing has been
used in the past then it is always allowed. This ensures PT is off by
default but that either offer or answer can enable PT and once it has
been on it is also possible to get early media with PT.
- Want PT-based demux? The MID header extension has to be removed in
either the offer or the answer. Follow-up O/As allow PT demuxing if
possible.
- Want to use MID or SSRC demuxing? Great, you don't need PT-based demux
and won't mind that we turned it off for you.
The reason for disabling PT demux at offer time (if MID is present)
instead of waiting for the SDP answer is because by the time the SDP
answer arrives, early media could have triggered PT demux and caused
incorrect mappings. The safe thing is to assume a spec-compliant
endpoint until proven otherwise.
However if PT demux is ever enabled, then from that point on we always
allow PT-based demux in follow-up O/A exchanges. This ensures we don't
drop packets in follow-up exchanges. The fact that PT-based demux is
disabled during the initial offer should not matter because before the
initial O/A exchange we don't have fingerprints.
This change only affects Unified Plan and bundled groups. Existing test
coverage ensuring we do not break legacy endpoints:
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/peer_connection_integrationtest.cc;l=1156
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/protocol/rtp-demuxing.html;l=59
UnsignaledStreamTest is also updated to test the interesting setups.
A kill-switch is added in case we want to disable this change.
Bug: webrtc:12814
Change-Id: I807a82a543325753633aaef698e06cb4c9dfebaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221101
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34251}
This is step 1: Encapsulating the data.
Bug: webrtc:12837
Change-Id: I15df30dc294c90136a90b072608ed4c2e8925dcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221602
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34245}
CandidateStats didn't use an initializer list which caused the
`candidate` member variable to be constructed with a random id
(calling an expensive rng method), only to be overwritten directly
thereafter.
Bug: webrtc:12840
Change-Id: I0366f674281d236896cb9539812dc2d88c1b37ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34244}
documents the various payload type mappers used by WebRTC.
BUG=webrtc:12194,webrtc:12295
No-try: true
Change-Id: I88e2388f54e72db5e5fe0f72fa4fe4c455c99679
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220936
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34227}
It was found from Chrome tracing that worker packet progression in
https://webrtc.github.io/samples/src/content/peerconnection/negotiate-timing/
during renegotiation of 100 transceivers is hindered by a multi-hundred
millisecond Invoke from the signaling to the worker thread. This
causes audio impairment.
Fix this by splitting the single Invoke into a series of Invokes,
allowing packets received during the renegotiation to be processed
between the worker invocations.
Experimental data of negotiation from 1 to 100 video transceivers
WebRtcDistinctWorkerThread OFF, before change:
4415.60 milliseconds, audio impairment 29760
4216.00 milliseconds, audio impairment 25560
4298.40 milliseconds, audio impairment 25440
WebRtcDistinctWorkerThread OFF, after change:
4258.70 milliseconds, audio impairment 26280
4255.50 milliseconds, audio impairment 25920
4363.10 milliseconds, audio impairment 25200
WebRtcDistinctWorkerThread ON, before change:
4407.80 milliseconds, audio impairment 24840
4541.00 milliseconds, audio impairment 26160
4377.80 milliseconds, audio impairment 17040
WebRtcDistinctWorkerThread ON, after change:
4364.80 milliseconds, audio impairment 0
4174.30 milliseconds, audio impairment 0
4309.00 milliseconds, audio impairment 0
We should reconsider this split after lazy decoders and decoder stream
projects have shipped, see
- bugs.webrtc.org/12462
- crbug.com/1157227
- crbug.com/1187289
Bug: webrtc:12840, webrtc:12462, chromium:1157227, chromium:1187289
Change-Id: I8e3b3943bd76f09da74b457690799415335b51f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34202}
This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call.
Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes.
If there is custom factory present in dependencies it will be used, otherwise default factory will be used.
Intention behind this change is to have ability to have custom QoS with custom parameters.
Bug: webrtc:12778
Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34181}
This is part of moving the thread hops from the network thread to
worker (required by Call) into Call itself so that we can eventually
remove them.
Bug: webrtc:11993
Change-Id: Ib3ccdd6c75a3848daae2e3ce6c9a55d9617c2f50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220604
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34160}
This change creates trace events with a single parameter
composed of ClassName::Method.
The change additionally causes the duration of the proxy call to be
traced, not only the occurrence.
Fixed: webrtc:12787
Change-Id: I1689862318d4c6fc1dcef343c3ccf3ae9f7e17df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219788
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34149}
for consistency with the definition in IsRtcpPacket which takes
into account a collision of H261 feedback for payload types 64 + 65:
https://datatracker.ietf.org/doc/html/rfc5761#section-4
BUG=webrtc:12194
Change-Id: I2ebb0456ae2aff1b1735f26221c7c4ae79698ac9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220021
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34131}
Deprecate CreateDataChannel, and make it a simple wrapper function.
Bug: webrtc:12796
Change-Id: I053d75a264596ba87ca734a29df9241de93a80c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219784
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34130}
Reland of commit a743303211b89bbcf4cea438ee797bbbc7b59e80
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
- Mark FindHeaderExtensionByUri without filter argument as deprecated
Bug: webrtc:11713
Change-Id: I52a5ade1b94bc01d1c2a35cb56023684fcaf9982
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34129}
This reverts commit 096ad02c02b4bc6c046282b8793ef84d041dd0d8.
Reason for revert: Including a fix for the test issue.
Original change's description:
> Revert "Fix race between enabled() and set_enabled() in VideoTrack."
>
> This reverts commit 5ffefe9d2d743c66f8a8bcbc5ad9662a3138840a.
>
> Reason for revert: Breaks Chromium Android browser tests on fyi bots.
>
> Original change's description:
> > Fix race between enabled() and set_enabled() in VideoTrack.
> >
> > Along the way I introduced VideoSourceBaseGuarded, which is equivalent
> > to VideoSourceBase except that it applies thread checks. I found that
> > it's easy to use VideoSourceBase incorrectly and in fact there appear
> > to be tests that do this.
> >
> > I made the source object const in VideoTrack, as it already was in
> > AudioTrack, and that allowed for making the GetSource() accessors
> > bypass the proxy thread hop and give the caller direct access.
> >
> > Bug: webrtc:12773, b/188139639, webrtc:12780
> > Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34096}
>
> TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12773
> Bug: b/188139639
> Bug: webrtc:12780
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#34101}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12773
Bug: b/188139639
Bug: webrtc:12780
Change-Id: Ib35fe15a6c43de8f286d60aff02b19df1ab76925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219639
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34104}
This reverts commit 5ffefe9d2d743c66f8a8bcbc5ad9662a3138840a.
Reason for revert: Breaks Chromium Android browser tests on fyi bots.
Original change's description:
> Fix race between enabled() and set_enabled() in VideoTrack.
>
> Along the way I introduced VideoSourceBaseGuarded, which is equivalent
> to VideoSourceBase except that it applies thread checks. I found that
> it's easy to use VideoSourceBase incorrectly and in fact there appear
> to be tests that do this.
>
> I made the source object const in VideoTrack, as it already was in
> AudioTrack, and that allowed for making the GetSource() accessors
> bypass the proxy thread hop and give the caller direct access.
>
> Bug: webrtc:12773, b/188139639, webrtc:12780
> Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34096}
TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12773
Bug: b/188139639
Bug: webrtc:12780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34101}
Along the way I introduced VideoSourceBaseGuarded, which is equivalent
to VideoSourceBase except that it applies thread checks. I found that
it's easy to use VideoSourceBase incorrectly and in fact there appear
to be tests that do this.
I made the source object const in VideoTrack, as it already was in
AudioTrack, and that allowed for making the GetSource() accessors
bypass the proxy thread hop and give the caller direct access.
Bug: webrtc:12773, b/188139639, webrtc:12780
Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34096}
The tests are now all gated behind a singe WEBRTC_HAVE_SCTP flag,
including the invocations, so the issue of configs that vary which
tests are included should have gone away.
This CL also clears out the remains of
DataChannelIntegrationTestWithFakeClock, which is no longer used.
Bug: webrtc:12525
Change-Id: Ie0133e3e84358e9238b8e7465200f000519b8765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219783
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34085}
This change prepares for a later change in Chromium that makes it
stop depending on headers exposed by WebRTC that require inclusion of
api/proxy.h.
No-Try because of lack of infra lack of capacity on macs.
No-Try: True
Bug: webrtc:12787
Change-Id: I628424fe49e873027595b80336be2b821c22245e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219688
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34072}
These tests document existing behavior; there are notes about where
this behavior is either surprising or wrong.
Bug: chromium:1148951
Change-Id: If9875fb744c44c129ff9949d1bab3d3d99f17b81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217520
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34058}
and fix other tests to actually include CN in supported codecs.
BUG=webrtc:12365
Change-Id: I7154dd0046497604a6fc825ce81e85c1b9cadcef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34033}
The number of invokes will vary based on number of receivers/senders,
so we can't have a fixed number there.
Bug: none
Change-Id: Iab3d529a5935c4b6cc95e9da6637acd880614972
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219101
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34027}
For implementations where the signaling and worker threads are not
the same thread, this significantly cuts down on Thread::Invoke()s that
would block the signaling thread while waiting for the worker thread.
For Audio and Video Rtp receivers, the following methods now do not
block the signaling thread:
* GetParameters
* SetJitterBufferMinimumDelay
* GetSources
* SetFrameDecryptor / GetFrameDecryptor
* SetDepacketizerToDecoderFrameTransformer
Importantly this change also makes the track() accessor accessible
directly from the application thread (bypassing the proxy) since
for receiver objects, the track object is const.
Other changes:
* Remove RefCountedObject inheritance, use make_ref_counted instead.
* Every member variable in the rtp receiver classes is now RTC_GUARDED
* Stop() now fully clears up worker thread state, and Stop() is
consistently called before destruction. This means that there's one
thread hop instead of at least 4 before (sometimes more), per receiver.
* OnChanged triggered volume for audio tracks is done asynchronously.
* Deleted most of the JitterBufferDelay implementation. Turns out that
it was largely unnecessary overhead and complexity.
It seems that these two classes are copy/pasted to a large extent
so further refactoring would be good in the future, as to not have to
fix each issue twice.
Bug: chromium:1184611
Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34022}
