This changes the notification to a single std::function pointer
instead of being a sigslot::signal1<> collection.
Summary:
* Remove SignalFirstPacketReceived_, the last sigslot member variable.
(still inherits from sigslot::has_slots<>)
* BaseChannel doesn't post to the signaling thread anymore. The only
reason that remains for the signaling_thread_ variable, is for
thread checking.
* Remove BaseChannel's reliance on MessageHandlerAutoCleanup
(still inherits from MessageHandler)
RtpTransceiver is the consumer of this event. That class is also the
class that sits between the PC classes and the channel object, holding
a pointer to the channel and managing calls that come in on the
signaling thread, such as SetChannel. The responsibility of delivering
the first packet received on the signaling thread is now with
RtpTransceiver:
* RtpTransceiver always requires a ChannelManager instance. Previously
this variable was sometimes set, but it's now required.
* Updated tests in rtp_transceiver_unittest.cc to include a
ChannelManager as well as fix them to include call expectations for
mock sender and receivers.
Bug: webrtc:11993, webrtc:11988
Change-Id: If49d6be157cd7599fa6fe3a42cd0a363464e3a74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215979
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33853}
There's a bit of copy/pasted code in the channel code, which is
making moving network traffic consistently over to the network thread
a bit trickier than it needs to be, so I'm also updating variable
names used in Set[Local|Remote]Content_w to be more explicitly the same
and make it clear that the code is copy/pasted (and future updates can
consolidate more of it).
Also removing some code from the video/voice media channels that's
effectively dead code (vector + registration methods that aren't needed)
Bug: webrtc:12705
Change-Id: I2e14e69fbc489a64fc1e8899aaf1cfc979fe840b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215978
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33847}
* Changing return value from bool to void since the operation as async
effects anyway.
* Removing the `enabled()` accessor due to potential threading issues
and potential TOCTOU issues. It was only used in one place anyway.
* Applying thread restrictions to member variables.
Bug: none
Change-Id: I51949f5594339952d7b717cfd82f99b532e86b23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216182
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33830}
* Adds a OnPacketSent callback to MediaChannel, which matches with
MediaChannel::NetworkInterface::SendPacket.
* Moves the OnPacketSent handling to the media channel implementations
(video/voice) and removes the PeerConnection/SdpOfferAnswerHandler
layer from the call path.
* Call::OnSentPacket is called directly from the channels on the network
thread. This eliminates a PostTask to the worker thread for every
audio/video network packet.
* Remove sigslot dependency from MediaChannel (and derived).
Bug: webrtc:11993
Change-Id: I1f79a7aa60f05d47e1882f9be1c9323ea8fac5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215403
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33777}
With this change, all production callers of BaseChannel::transport_name()
will be making the call from the right thread and we can safely delegate
the call to the transport itself. Some tests still need to be updated.
This facilitates the main goal of not needing synchronization inside
of the channel classes, being able to apply thread checks and eventually
remove thread hops from the channel classes.
A downside of this particular change is that a blocking call to the
network thread from the signaling thread inside of RTCStatsCollector
needs to be done. This is done once though and fixes a race.
Bug: webrtc:12601, webrtc:11687, webrtc:12644
Change-Id: I85f34f3341a06da9a9efd936b1d36722b10ec487
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213080
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33775}
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 reverts commit a743303211.
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}
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}
By changing was_ever_writable_ to be guarded by the worker thread
instead of the network thread.
Gets rid of one network thread invoke per audio/video m= section per
round of negotiation.
NOTRY=True
Bug: webrtc:12266
Change-Id: Ie913a9f96db3fd8351559e916922c82d2d0337f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203881
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33130}
This reverts commit c1ad1ff178.
Reason for revert: This blocks the worker thread for a longer
contiguous period of time which can lead to delays in processing
packets. And due to other recent changes, the need to speed up
SetLocalDescription/SetRemoteDescription is reduced.
Still plan to reland some of the changes from the CL, just not the
part that groups the Invokes.
Original change's description:
> Do all BaseChannel operations within a single Thread::Invoke.
>
> Instead of doing a separate Invoke for each channel, this CL first
> gathers a list of operations to be performed on the signaling thread,
> then does a single Invoke on the worker thread (and nested Invoke
> on the network thread) to update all channels at once.
>
> This includes the methods:
> * Enable
> * SetLocalContent/SetRemoteContent
> * RegisterRtpDemuxerSink
> * UpdateRtpHeaderExtensionMap
>
> Also, removed the need for a network thread Invoke in
> IsReadyToSendMedia_w by moving ownership of was_ever_writable_ to the
> worker thread.
>
> Bug: webrtc:12266
> Change-Id: I31e61fe0758aeb053b09db84f234deb58dfb3d05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194181
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32817}
TBR=deadbeef@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12266
Change-Id: I40ec519a614dc740133219f775b5638a488529b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203860
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33111}
Functionality wise, there should be no change with this CL, aside
from updating tests to anticipate OnPacketReceived to handle the packet
asynchronously (as already was the case via BaseChannel).
This only removes the network->worker hop out of the BaseChannel
class into the WebRTC MediaChannel implementations. However, it updates
the interface contract between BaseChannel and MediaChannel to align
with how we want things to work down the line, i.e. avoid hopping to
the worker thread for every rtp packet.
The following steps will be to update the video and voice channel
classes to call Call::DeliverPacket on the network thread and only
handle unsignalled SSRCs on the worker (exception case).
Bug: webrtc:11993
Change-Id: If0540874444565dc93773aee89d862f3bfc9c502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33040}
all invokes, as well as BaseChannel constructor and destructor
should run on the same task queue which allow to use
simpler cancellation of pending task on BaseChannel destruction
Bug: webrtc:12339
Change-Id: I311b6de940cc24cf6bb5b49e1bbd132fea2439e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202032
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33009}
SetNegotiatedHeaderExtensions_w queued a task to update the list
of negotiated header extensions on the signal thread from the
worker thread, in belief that a later call to
GetNegotiatedHeaderExtensions() would happen on the WebRTC proxies,
leading to the update happening before the readout. In downstream
project, this is not always the case.
Fix this by synchronously updating the list of negotiated header
extensions.
Bug: chromium:1051821
Change-Id: I3266292e7508bb7a22a3f7d871e82c12f60cfc83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201728
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32977}
Instead of doing a separate Invoke for each channel, this CL first
gathers a list of operations to be performed on the signaling thread,
then does a single Invoke on the worker thread (and nested Invoke
on the network thread) to update all channels at once.
This includes the methods:
* Enable
* SetLocalContent/SetRemoteContent
* RegisterRtpDemuxerSink
* UpdateRtpHeaderExtensionMap
Also, removed the need for a network thread Invoke in
IsReadyToSendMedia_w by moving ownership of was_ever_writable_ to the
worker thread.
Bug: webrtc:12266
Change-Id: I31e61fe0758aeb053b09db84f234deb58dfb3d05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194181
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32817}
This moves the code for threadjumping to get the RTP transport
despite its thread guard from the main function to two functions
marked especially "ForTesting".
Bug: webrtc:12230
Change-Id: I4473ed38e6fdedb05e2fbc97c2521bc1993fdd1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196521
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32792}
This makes it thread-safe to access, but not necessarily to use.
Bug: webrtc:12230
Change-Id: I6b48d86dff24b162d382135abeaf560971fdf614
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196524
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32785}
This CL also adds commentary to member variables that couldn't be guarded
because they're accessed from multiple threads.
Bug: webrtc:12230
Change-Id: I5193a7ef36ab25588c76ee6a1863de6a844be1dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195331
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32705}
This fixes regressions caused by:
https://webrtc-review.googlesource.com/c/src/+/183120
... which disabled payload type demuxing when multiple video tracks are
present, to avoid one channel creating a default track intended for
another channel.
However, this isn't an issue when not bundling, as each track will be
delivered on separate transport.
And it's also not an issue when each track uses a distinct set of
payload types (e.g., VP8 is mapped to PT 96 in one m= section, and PT 97
in another).
This CL addresses both of those cases; PT demuxing is only disabled
when two bundled m= sections have overlapping payload types.
Bug: chromium:1139052, webrtc:12029
Change-Id: Ied844bffac2a5fac29147c11b56a5f83a95ecb36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187560
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32419}
SignalDtlsSrtpSetupFailure is never fired, so the setup code for it,
is dead code. Also removing declarations for methods that have no
implementation.
For other public signals in BaseChannel I've added an accessor which
has revealed a threading problem due to the member variable being public.
Bug: webrtc:11994
Change-Id: Iec6046c6a598066b92c956002ba4160708ae7dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32211}
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.
With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.
Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).
Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).
Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.
Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
This fixes some edge cases where early media could cause default
stream that block the actual signaled media from beind delivered.
Bug: webrtc:11477
Change-Id: I8b26df63a690861bd19f083102d1395e882f8733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32030}
Logging of content name (mid) is valuable to debug issues
in scenarios with multiple m= line sections in SDP.
For example, video conferencing applications which
uses SFU and Unified Plan SDPs will likely to leverage
from more detailed logs when issues need to be debugged.
Bug: webrtc:10139
Change-Id: Id52ba3ad54af5caa0f8c03daaa51bdb0caf9fe67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175115
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31302}
to generate less versions of the function template and FunctorMessageHandler helper
thus producing less binary size
Bug: None
Change-Id: Idbd6fb1e1f23b9b2dc4e4306a74ef11e74ba94cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161044
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29962}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
This enables creation and removal of receive streams with SSRC 0.
Several related methods, for example SetOutputVolume, still use 0 as a
special value.
Bug: webrtc:8694
Change-Id: I341e6bd6c981c9838997510d8d712ad2948f6460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29398}
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.
The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.
Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29226}
This makes it easier to follow the flow in a debugger and reduces
the number of methods.
Bug: webrtc:9883
Change-Id: If485ff08a223a3986ff24b29ebf4d37c325f0f26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152669
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29180}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
Bug: webrtc:10139
Change-Id: I6803f4325e7c34915a9ae79e3360a787a7a9df5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149173
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29105}
RtpTransportInternal does not need to expose these. They are only used
by tests and for setting options. Instead, it can expose a SetRtpOption
and SetRtcpOption to set options relevant to each of its transports.
Also updates tests to work around no longer having access to internals.
This will simplify the composite needed during negotiation of different
RTP transport types, as we no longer need to have composites of both
RtpTransport and PacketTransport.
Bug: webrtc:9719
Change-Id: I91bfa6e95b7aa384d10497f47e7d2483c2e0bef2
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138282
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28066}
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 reverts commit 46afbf9481.
Reason for revert: Tightened protocol name handling.
Original change's description:
> Revert "Reland "Version 2 "Refactoring DataContentDescription class"""
>
> This reverts commit 37f2b43274.
>
> Reason for revert: fuzzer failures
>
> Original change's description:
> > Reland "Version 2 "Refactoring DataContentDescription class""
> >
> > This is a reland of 14b2758726
> >
> > Original change's description:
> > > Version 2 "Refactoring DataContentDescription class"
> > >
> > > (substantial changes since version 1)
> > >
> > > This CL splits the cricket::DataContentDescription class into
> > > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > > and cricket::SctpDataContentDescription (used for SCTP only).
> > >
> > > SctpDataContentDescription no longer inherits from
> > > MediaContentDescriptionImpl, and no longer contains "codecs".
> > >
> > > Due to usage of internal interfaces by consumers, shimming the old
> > > DataContentDescription API is needed.
> > >
> > > A new cricket::DataContentDescription class is defined, which is
> > > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > > It exposes as little functionality as possible, but supports the
> > > concerned consumer's usage
> > >
> > > Design document:
> > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> > >
> > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> > >
Bug: webrtc:10358
Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27933}
This reverts commit 37f2b43274.
Reason for revert: fuzzer failures
Original change's description:
> Reland "Version 2 "Refactoring DataContentDescription class""
>
> This is a reland of 14b2758726
>
> Original change's description:
> > Version 2 "Refactoring DataContentDescription class"
> >
> > (substantial changes since version 1)
> >
> > This CL splits the cricket::DataContentDescription class into
> > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > and cricket::SctpDataContentDescription (used for SCTP only).
> >
> > SctpDataContentDescription no longer inherits from
> > MediaContentDescriptionImpl, and no longer contains "codecs".
> >
> > Due to usage of internal interfaces by consumers, shimming the old
> > DataContentDescription API is needed.
> >
> > A new cricket::DataContentDescription class is defined, which is
> > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > It exposes as little functionality as possible, but supports the
> > concerned consumer's usage
> >
> > Design document:
> > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> >
> > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> >
> > Bug: webrtc:10358
> > Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27853}
>
> Bug: webrtc:10358
> Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27896}
TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org
Change-Id: Ied6d9fb96aafe9c957f2658b34b5331b1f359b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135986
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27917}
This is a reland of 14b2758726
Original change's description:
> Version 2 "Refactoring DataContentDescription class"
>
> (substantial changes since version 1)
>
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
>
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
>
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
>
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
>
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
>
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
>
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}
Bug: webrtc:10358
Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27896}
This reverts commit 14b2758726.
Reason for revert: Internal import failed.
Original change's description:
> Version 2 "Refactoring DataContentDescription class"
>
> (substantial changes since version 1)
>
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
>
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
>
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
>
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
>
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
>
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
>
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}
TBR=danilchap@webrtc.org,steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org
Change-Id: Ibc16ba14c1cbf50345a9b79151b79df140482539
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27855}
(substantial changes since version 1)
This CL splits the cricket::DataContentDescription class into
two classes: cricket::RtpDataContentDescription (used for RTP data)
and cricket::SctpDataContentDescription (used for SCTP only).
SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".
Due to usage of internal interfaces by consumers, shimming the old
DataContentDescription API is needed.
A new cricket::DataContentDescription class is defined, which is
a shim over RtpDataContentDescription and SctpDataContentDescription.
It exposes as little functionality as possible, but supports the
concerned consumer's usage
Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
Bug: webrtc:10358
Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27853}
CreateOffer and CreateAnswer will now examine the layers on the
transceiver to determine if multiple layers are requested (Simulcast).
In this scenario RIDs will be used in the layers (instead of SSRCs).
When the offer is created, only RIDs are signalled in the offer.
When the offer is set locally SetLocalDescription() SSRCs will be
generated for each layer by the Channel and sent downstream to the
MediaChannel.
The MediaChannel receives configuration that looks identical to that of
legacy simulcast, and should be able to integrate the streams correctly
regardless of how they were signalled.
Setting multiple layers on the transciever is still not supported
through the API.
Bug: webrtc:10075
Change-Id: Id4ad3637b87b68ef6ca7eec69166fee2d9dfa36f
Reviewed-on: https://webrtc-review.googlesource.com/c/119780
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@{#26428}
So far, base channel was only notifying about 'first audio packet' when
RTP was used, and it never notified about it when media_transport
interface was used. This change adds a sigslot to notify about a new
media packet to the media transport interface.
Bug: webrtc:9719
Change-Id: Ie9230c407f35b1aaa71ba71008ac34ba8869e2d4
Reviewed-on: https://webrtc-review.googlesource.com/c/117249
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26282}
Bug: webrtc:10198
Change-Id: If510e6f508e34aaa36c9ccbbdc90dd33ad5fef10
Reviewed-on: https://webrtc-review.googlesource.com/c/116991
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26253}
