This is a reland of d48a2b14e7545d0a0778df753e062075c044e2a1
The diff of the reland (what caused the tsan error) can be seen
by diffing patch sets 2 and 3. Essentially I missed keeping the calls
to the transport controller on the worker thread. Note to self to add
thread/sequence checks to that code so that we won't have to rely on
tsan :)
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
Bug: webrtc:11993, webrtc:12430
Change-Id: I4fccaa418d22c2087a55bbb3ddbb25fac3b4dfcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33153}
This reverts commit d48a2b14e7545d0a0778df753e062075c044e2a1.
Reason for revert: TSan tests started to fail constantly after this CL (it looks like it is flaky and the CQ was lucky to get green). See https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25042/overview.
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
TBR=nisse@webrtc.org,tommi@webrtc.org
Change-Id: Id87cf9cbcc8ed58e74d755a110f0ef9dd980e298
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205525
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33145}
This moves the thread hop for network events, from BaseChannel and
into Call. The reason for this is to move the control over those hops
(including DeliverPacket[Async]) into the same class where the state
is held that is affected by those hops. Once that's done, we can start
moving the relevant network state over to the network thread and
eventually remove the hops.
I'm also adding several TODOs for tracking future steps and give
developers a heads up.
Bug: webrtc:11993
Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33138}
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 c1ad1ff178f0d0dfcde42843c51ae703005aaca1.
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 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}
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}
This log line is causing test failures due to excessive logging (see
referenced bug); reducing log level.
Bug: chromium:984879
Change-Id: Ic94ba0a39b91b4253a58ad54de0cba1ca49882e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175913
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31803}
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}
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
Reason for revert: Breaks Chromium import due to flaky test in Chromium.
Original change's description:
> Reland "Distinguish between send and receive codecs"
>
> This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
>
> Reason for revert: Fixed negotiation of send-only clients.
>
> Original change's description:
> > Revert "Distinguish between send and receive codecs"
> >
> > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> >
> > Reason for revert: breaks negotiation with send-only clients
> >
> > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> >
> > Original change's description:
> > > Distinguish between send and receive codecs
> > >
> > > Even though send and receive codecs may be the same, they might have
> > > different support in HW. Distinguish between send and receive codecs
> > > to be able to keep track of which codecs have HW support.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30284}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30292}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
>
> Bug: chromium:1029737
> Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30348}
TBR=steveanton@webrtc.org,kron@webrtc.org
Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30360}
This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
Reason for revert: Fixed negotiation of send-only clients.
Original change's description:
> Revert "Distinguish between send and receive codecs"
>
> This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
>
> Reason for revert: breaks negotiation with send-only clients
>
> (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
>
> Original change's description:
> > Distinguish between send and receive codecs
> >
> > Even though send and receive codecs may be the same, they might have
> > different support in HW. Distinguish between send and receive codecs
> > to be able to keep track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30284}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
> Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30292}
TBR=steveanton@webrtc.org,kron@webrtc.org
Bug: chromium:1029737
Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30348}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
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}
This reverts commit add7ef974ee2642a3b55a36ec80be50a615bc60a.
Reason for revert: Cause regression in pc_full_stack_tests.cc
Original change's description:
> Sanitize the codec list before sending it to the media engine
>
> The SDP can assign the same codec to two different payload types
> which gets represented as two separate codecs in the SDP structure.
> The media engine assumes that the client does not pass down
> duplicate codecs. This change adds logic to BaseChannel to filter
> out codecs of the same name with different payload types, picking
> the one which is listed first in the m= line.
>
> Bug: chromium:987598
> Change-Id: I6fa813db1769e572ff7c3f322dc9b1de39817ea2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147602
> Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28726}
TBR=steveanton@webrtc.org,amithi@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:987598
Change-Id: I4ffbfcd90c81c6c6c8ee8f872f7e217d8291c857
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147864
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28744}
The SDP can assign the same codec to two different payload types
which gets represented as two separate codecs in the SDP structure.
The media engine assumes that the client does not pass down
duplicate codecs. This change adds logic to BaseChannel to filter
out codecs of the same name with different payload types, picking
the one which is listed first in the m= line.
Bug: chromium:987598
Change-Id: I6fa813db1769e572ff7c3f322dc9b1de39817ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147602
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28726}
This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where
it's used to improve congestion controller behavior.
The functionality is negotiated using SDP.
It's added with a field trial that allow disabling the functionality in
case there's any issues.
Bug: webrtc:10742
Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28654}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
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 46afbf9481fbcc939c998c898ca1031ce41cc6b1.
Reason for revert: Tightened protocol name handling.
Original change's description:
> Revert "Reland "Version 2 "Refactoring DataContentDescription class"""
>
> This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e.
>
> Reason for revert: fuzzer failures
>
> Original change's description:
> > Reland "Version 2 "Refactoring DataContentDescription class""
> >
> > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
> >
> > 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 37f2b43274a0d718de53a4cfcf02226356edcf6e.
Reason for revert: fuzzer failures
Original change's description:
> Reland "Version 2 "Refactoring DataContentDescription class""
>
> This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
>
> 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 14b2758726879d21671a21291dfed8fb4fd5c21c
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}
