The implementation here has a number of changes that force the callers
that called the "channel" functions into specific interfaces rather than
just letting C++ take care of it; this should go away once there stops
being a common implementation class for those interfaces.
Bug: webrtc:13931
Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38888}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
This is an error that puts the PC into an inconsistent state, so
causing a crash is the right thing to do.
Bug: chromium:1341043
Change-Id: Ie1eb89400ad87f0c83634b7073236b07e92ec7ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267281
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37391}
This is a preparatory step in deleting the ChannelManager class.
Also delete some declarations whose implementation was previously removed.
Bug: webrtc:13931
Change-Id: I8764c00fa696932e79fcfe17550ef2490d6a1ed1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262804
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36923}
This makes the channel manager object into a factory, not a manager.
Bug: webrtc:13931
Change-Id: I59f7d818a739797a7c0a7a32e6583450834df122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260467
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36718}
This better reflects the ownership passing of AddTrack, and is more
consistent for RemoveTrack.
Bug: webrtc:13980
Change-Id: Ide5baccf15fc687a4e092f8831ce8c0fea46604e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259740
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36603}
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.
Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
This makes SetChannel() consistently make 2 invokes instead of a
multiple of senders+receivers (previous minimum was 4 but could be
larger).
* Stop() doesn't hop to the worker thread.
* SetMediaChannel(), an already-required step on the worker thread for
senders and *sometimes* for receivers[1], is now consistently required
for both. This simplifies transceiver teardown and enables the next
bullet.
* Transceiver stops all senders and receivers in one go rather than
ping ponging between threads.
[1] When not required, it was done implicitly inside of Stop().
See changes in `RtpTransceiver::SetChannel`
Bug: webrtc:13540
Change-Id: Ied61636c8ef09d782bf519524fff2a31e15219a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249797
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36057}
The interface is implemented by the ChannelManager and contains methods
to create and destroy media channel objects as used by a transceiver.
This will subsequently allow us to delete the channel objects from
the transceiver class where ownership really lies rather than from
the outside - which is currently required by some tests that keep
channel objects on the stack. We'll furthermore be able to do the
destruction asynchronously without additional Invoke()s as we do now
which will remove an Invoke when making sdp changes.
With introducing the interface, the following simplifications were made:
* ChannelManager constructed on the signaling thread.
Before, there was an Invoke in the context class, which existed
for the purposes of calling MediaEngine::Init() (which in turn is
only needed for the VoiceEngine). This Invoke has now been moved
into the CM (more tbd).
* The CM now has a pointer to the signaling thread (since that's the
construction thread). That allows us to remove the signaling thread
parameter from the CreateFooChannel methods.
* The ssrc_generator (UniqueRandomIdGenerator) instance for SSRCs moved
from SdpOfferAnswerHandler to the CM, as it's always used in
combination with the CM. This simplifies the CreateFooChannel methods
as well as a couple of other classes that have a CM dependency.
* Removed DestroyFooChannel related code from SdpOfferAnswerHandler since
the channel type detail can be taken care of by the CM.
Bug: webrtc:11992, webrtc:13540
Change-Id: I04938a803734de8489ba31e6212d9eaecc244126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247904
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35766}
...and one when destroying a channel object.
This CL removes Init_n() and Deinit_n() from the BaseChannel class.
Channel classes now use SetRtpTransport to do initialization and
uninitialization on the network thread.
Notably if an implementation has called SetRtpTransport() with a valid
transport pointer, it is required that SetRtpTransport be called again
with a nullptr before the channel object can be deleted.
In situations where multiple channels are created, this can mean
a substantial reduction in thread hops. We still hop to the worker
in order to construct the objects - this can probably be avoided
and SetChannel() is still a synchronous operation for the transceivers.
Furthermore, teardown of channel objects also still happens
synchronously and across network/worker/signaling threads.
Bug: webrtc:11992
Change-Id: I68ca7596e181fc82996e3e290733d97381aa5e78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35738}
This is a necessary precondition for upcoming changes that will remove
calls to Invoke() that currently occur during construction/destruction
of media channel objects.
Subsequently fix RtpSenderReceiverTest to destroy channels that were
created in the constructor, in the destructor.
Bug: webrtc:11992
Change-Id: I526908d997d27495659805d84113c84c48568712
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246680
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35702}
Setting different number of temporal layers is supported by SimulcastEncodeAdapter and LibvpxVp8Encoder will fallback to SimulcastEncoderAdapter if InitEncode fails.
Bug: none
Change-Id: I8a09ee1e6c70a0006317957c0802d019a0d28ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228642
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34785}
This reverts commit 37ee0f5e59.
Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
Original change's description:
> Use backticks not vertical bars to denote variables in comments for /pc
>
> Bug: webrtc:12338
> Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34575}
TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34577}
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}
Classes associated with the Call instance, need access to these threads
and/or awareness, for checking for thread correctness.
Bug: webrtc:11993
Change-Id: I93bcee0657875f211be2ec959b96f818fa9fd8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215584
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33772}
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}
* 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}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
This reverts commit 15be5282e9.
Reason for revert: crbug.com/1028937
Original change's description:
> Add support for RtpEncodingParameters::max_framerate
>
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}
TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.
Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.
TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.
Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
This is a reland of df5731e44d with fixes
to avoid existing chromium tests to fail.
Instead of replacing the existing RtpSender::set_stream_ids() to
also fire OnRenegotiationNeeded(), this CL keeps the old
set_stream_ids() and adds the new RtpSender::SetStreams() which sets
the stream IDs and fires the callback.
This allows existing callsites to maintain behavior, and reserve
SetStreams() for the cases when we want OnRenegotiationNeeded() to fire.
Using the SetStreams() name instead of SetStreamIDs() to match the W3C
spec and to make it more different that the existing set_stream_ids().
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}
Bug: webrtc:10129
Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27992}
This reverts commit df5731e44d.
Reason for revert: Breaks WebRTC in Chrome FYI for all platforms.
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/2966
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}
TBR=steveanton@webrtc.org,hbos@webrtc.org,guidou@webrtc.org
# Passing all bots except for flaky webrtc_perf_tests
NOTRY=True
Bug: webrtc:10129
Change-Id: If97317f7a01b34465685fcebbeea0d7576ed7328
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137431
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27988}
This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
event if needed and exposes the method on RtpSenderInterface.
This is a spec-compliance change.
Bug: webrtc:10129
Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27974}
This patch makes VideoBitrateAllocatorFactory injectable
by adding to PeerConnectionDependencies instead of allowing it to be
overridden using MediaEngine (on PeerConnectionFactory).
With this patch VideoBitrateAllocatorFactory is owned
by the PeerConnection.
WANT_LGTM (examples) : sakal@
WANT_LGTM (api/pc) : steveanton@
Bug: webrtc:10547
Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27654}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
RtpSender uses a transactional model when getting and setting RtpParameters.
One must call GetParameters() and then can use the returned object in a
subsequent call to SetParameters().
PeerConnection was calling GetParameters() and SetParameters() during
negotiation in SetLocalDescription and SetRemoteDescription effectively
invalidating any parameters that the client previously held.
This change introduces an internal way for the platform to modify
parameters without invalidating the transactional model, provided that
the modification is not severe.
Ex. removing simulcast layers is a severe modification and will
invalidate any outstanding parameters.
Bug: webrtc:10339
Change-Id: I362e8ca4d9556e04a1aa7a3e74e2c275f8d16fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/124504
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26864}
Most of the implementation in rtp_sender.cc is a copy paste for both
Audio & Video RTP senders. This change moves all the common behavior
into the base RtpSenderInternal class.
Template method pattern is used to accomodate for the very slight differences
between audio and video senders.
Bug: None
Change-Id: I6d4e93cd32fbb0fb361fd0e1883791019bde9a92
Reviewed-on: https://webrtc-review.googlesource.com/c/123411
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26758}
