In a handful of places we wait for the old IceConnectionState to reach some value and then we assume that the new connection states have also been updated. However those are updated in response to different events that might not have fired yet, so sometimes these tests will fail.
This change makes us wait explicitly for those states to update.
Bug: webrtc:9983
Change-Id: I5cb6652ee29c0b86c0834174442140a3863e08e4
Reviewed-on: https://webrtc-review.googlesource.com/c/110441
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25579}
This changes PeerConnection to allow sending and receiving data channel
messages over the media transport. If |use_media_transport_for_data_channels|
is set, PeerConnection will use a DCT_MEDIA_TRANSPORT mode for data
channels.
DCT_MEDIA_TRANSPORT acts exactly like DCT_SCTP within the data channel
and peer connection layers. On the transport layer, it uses the media
transport instead of SCTP. It appears as an RTP data channel in SDP
(just as media over media-transport appears as RTP in SDP).
Bug: webrtc:9719
Change-Id: I6a90142bd3f43668479c825ed02689dcd0d58b78
Reviewed-on: https://webrtc-review.googlesource.com/c/109740
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25575}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
This reverts commit 61c6e5643e7ea058e653956980a90e033249c055.
Reason for revert: downstream projects prepared for this change
Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
>
> This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
>
> Reason for revert: breaking downstream
>
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> >
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> >
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> >
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
>
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
>
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
Reason for revert: breaking downstream
Original change's description:
> Isolating APM API build target: making :api an actual target.
>
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
>
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
>
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.
More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.
Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
Adds a field |use_media_transport_for_data_channels| to RTCConfiguration.
PeerConnection requires a media transport factory to be set if this bit
is set. As with |use_media_transport|, the value may not be modified
after setting the local or remote description.
If either |use_media_transport| or |use_media_transport_for_data_channel| is
set, PeerConnection uses its media transport factory when creating a JSEP
transport controller.
PeerConnection stops unconditionally using media transport in
CreateVoiceChannel, as it may be present only for use in data channels. It uses
the media transport if it is present and |use_media_transport| is set.
Bug: webrtc:9719
Change-Id: I59d4ce8f7531fd19d9c17eefe033f063f663ebcc
Reviewed-on: https://webrtc-review.googlesource.com/c/109041
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25507}
Replaced by a int64_t representing time in us. To aid transition of
downstream code, rtc::PacketTime is made an alias for int64_t.
Bug: webrtc:9584
Change-Id: Ic3a5ee87d6de2aad7712894906dab074f1443df9
Reviewed-on: https://webrtc-review.googlesource.com/c/91860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25503}
IceConnected state (transport state) now includes the state of the
MediaTransport.
This is a first change of two. Second change will add state change
signals to the PeerConnectionInterface informing separately about
ice+media transport vs ice+dtls.
Bug: webrtc:9719
Change-Id: I5731530073e8f26dfc8b188778d268b815da7052
Reviewed-on: https://webrtc-review.googlesource.com/c/108901
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25473}
This is a preparation for deleting rtc::PacketTime. Next step, after
downstream code has been updated to not access the |timestamp| member,
is to make rtc::PacketTime an alias for int64_t.
Also delete the unused member rtc::PacketTime::not_before.
Bug: webrtc:9584
Change-Id: Iba9d2d55047d69565ad62b1beb525591fd432ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/108860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25468}
This CL prevents dereferencing potentially null pointer by
setting the pointer in client code.
We can now safely call PeerConnection::Close(), which happens
to trigger OnIceConnectionChange() on the observer.
This is a followup to: https://webrtc-review.googlesource.com/c/src/+/107706
Bug: webrtc:9855
Change-Id: Ieebf8415f0a12fe87d8cd80d1eb06797926005df
Reviewed-on: https://webrtc-review.googlesource.com/c/108785
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25462}
to Mdns.*.
MdnsResponderInterface now explicitly requires the reference counting
of created names to allow the coexistence of multiple users of the same
responder where one user would not remove identical names created by
others.
MDns.* is also renamed to Mdns.* per the style guide.
TBR=aleloi@webrtc.org
Bug: webrtc:9605
Change-Id: I047fc41f34de8d4e97c980409a7f373769c4c252
Reviewed-on: https://webrtc-review.googlesource.com/c/101921
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25458}
This allows to use secure, end to end communication if SDES cryptos are
passed. MediaTransport can use a derived key to secure its own
communication.
Bug: webrtc:9719
Change-Id: If1a20b136b3b4af0cb24f10b52fc5ce1eb31daa2
Reviewed-on: https://webrtc-review.googlesource.com/c/108504
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25452}
This change deals with a race condition if the media channel has been stopped
and is in the process of changing while we get a call to set a FrameDecryptor
or FrameEncryptor.
Bug: webrtc:9926, webrtc:9932
Change-Id: Ie2da2fa1f31f5cb5eb0b481861a7008e635f562d
Reviewed-on: https://webrtc-review.googlesource.com/c/107986
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25398}
With the expanding use cases for webrtc::CryptoOptions it makes more sense for
it to be be available per peer connection instead of only as a factory option.
To support backwards compatability for now this code will support the factory
method of setting crypto options by default. However it will completely
overwrite these settings if an RTCConfiguration.crypto_options is provided.
Got LGTM offline from Sami, adding him to TBR if he has any further comments.
TBR=sakal@webrtc.org
Bug: webrtc:9891
Change-Id: I86914cab69284ad82afd7285fd84ec5f4f2c4986
Reviewed-on: https://webrtc-review.googlesource.com/c/107029
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25375}
This change corrects a potential race condition when updating a FrameEncryptor
for the audio send channel. If a FrameEncryptor is set on an active audio
stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to
the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the
channel.
To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor>
to only allow deallocation when it is actually set on the encoder queue.
ChannelSend is unique in this respect as the Audio Receiver a long with the
Video Sender and Video Receiver streams all recreate themselves when they have
a configuration change. ChannelSend instead reconfigures itself using the
existing channel object.
Added Seth as TBR as this only introduces mocks.
TBR=shampson@webrtc.org
Bug: webrtc:9907
Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4
Reviewed-on: https://webrtc-review.googlesource.com/c/107664
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25374}
Underscore methods in the middle of classes is against the chromium style guide
this change is part of a long series of changes to refactor crypto code in
WebRTC to conform to the chromium standard better.
1. ssl_cert() -> GetSSLCertificate()
2. ssl_cert_chain() -> GetSSLCertificateChain()
3. Small tidying up in rtccertificategenerator.cc
Bug: webrtc:9860
Change-Id: I670f76e31d6d4f873034edb72d958b3c227379cb
Reviewed-on: https://webrtc-review.googlesource.com/c/107802
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25371}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I1081af5ecf7ba55a7415e09e45357b783cf300aa
Reviewed-on: https://webrtc-review.googlesource.com/c/107708
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25358}
Compute these states in jseptransportController and store them. Eventually they should be passed on to the peer connection observer and exposed in the blink layer.
Bug: webrtc:9308
Change-Id: Ifdec39c24a607fcb8211c4acf6b9704eaff371b1
Reviewed-on: https://webrtc-review.googlesource.com/c/103506
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25288}
- Similar to network priority
- Still requires MediaConfig.enable_dscp = true (i.e. googDscp == true to peerconnection)
- Needs followups 1) Specify value in chrome renderer js idl 2) disable audio bwe when value differs from video 3)remove googDscp guard
Bug: webrtc:5008
Change-Id: Ibdcbb1183f0ca2ae85e3bced6d0aedbccae3ced4
Reviewed-on: https://webrtc-review.googlesource.com/c/93560
Commit-Queue: Tim Haloun <thaloun@chromium.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25280}
In the past, it would incorrectly set up a state for 'use_media_transport' (i.e. it could say "use_media_transport" is true, but jseptransportcontroller wouldn't know about that).
Also, removes unnecessary field (unused).
Bug: webrtc:9719
Change-Id: I7e5c0ce81b3b70f63c49d661d95b95b5bcbb0c68
Reviewed-on: https://webrtc-review.googlesource.com/c/106960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25263}
Currently with in Unified Plan an initial offer will include both
"a=ssrc:... msid:..." lines and "a=msid:... ..." lines. The a=ssrc line
is added in order to support signaling to a Plan B endpoint. Although if
no stream is associated to a given track it will only be signaled in the
"a=msid" line with "-". The "a=ssrc msid" line will simply put an empty
string for the msid, which does not interoperate with FF. This change
adds support so that both lines will signal a "-".
Bug: webrtc:9880
Change-Id: I73655ce3c11a924b508616d820555bf24aae1bd3
Reviewed-on: https://webrtc-review.googlesource.com/c/106605
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25241}
This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d
Original change's description:
> Reland "Propagate media transport to media channel."
>
> This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c.
>
> Reason for revert: <INSERT REASONING HERE>
>
> Original change's description:
> > Revert "Propagate media transport to media channel."
> >
> > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4.
> >
> > Reason for revert: Breaks downstream project
> >
> > Original change's description:
> > > Propagate media transport to media channel.
> > >
> > > 1. Pass media transport factory to JSEP transport controller.
> > > 2. Pass media transport to voice media channel.
> > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel.
> > >
> > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71
> > > Bug: webrtc:9719
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Peter Slatala <psla@webrtc.org>
> > > Commit-Queue: Anton Sukhanov <sukhanov@google.com>
> > > Cr-Commit-Position: refs/heads/master@{#25152}
> >
> > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: webrtc:9719
> > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/105840
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25154}
>
> TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com
>
> Change-Id: I505ff3451eae81573531faef155ff35d7f894022
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9719
> Reviewed-on: https://webrtc-review.googlesource.com/c/106500
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25220}
Bug: webrtc:9719
Tbr: Steve Anton <steveanton@webrtc.org>
Tbr: Niels Moller <nisse@webrtc.org>
Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d
Reviewed-on: https://webrtc-review.googlesource.com/c/106561
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25240}
This change adds a new subcategory to the public native webrtc::CryptoOptions
structure: webrtc::CryptoOptions::Frame.
This new structure has a single off by default property:
crypto_options.frame.require_frame_encryption.
This new flag if set prevents RtpSenders from sending outgoing payloads unless
a frame_encryptor_ is attached and prevents RtpReceivers from receiving
incoming payloads unless a frame_decryptor_ is attached.
This option is important to enforce no unencrypted data can ever leave the
device or be received.
I have also attached bindings for Java and Objective-C.
I have implemented this functionality for E2EE audio but not E2EE video
since the changes are still in review.
Bug: webrtc:9681
Change-Id: Ie184711190e0cdf5ac781f69e9489ceec904736f
Reviewed-on: https://webrtc-review.googlesource.com/c/105540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25238}
Downstream clients will be able to use GetConfiguration() and SetConfiguration() to enable MediaTransport.
Bug: webrtc:9719
Change-Id: Ica77b25222732df211dc492dac848342d3f90ff2
Reviewed-on: https://webrtc-review.googlesource.com/c/106423
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25221}
This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Propagate media transport to media channel."
>
> This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4.
>
> Reason for revert: Breaks downstream project
>
> Original change's description:
> > Propagate media transport to media channel.
> >
> > 1. Pass media transport factory to JSEP transport controller.
> > 2. Pass media transport to voice media channel.
> > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel.
> >
> > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71
> > Bug: webrtc:9719
> > Reviewed-on: https://webrtc-review.googlesource.com/c/105542
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Peter Slatala <psla@webrtc.org>
> > Commit-Queue: Anton Sukhanov <sukhanov@google.com>
> > Cr-Commit-Position: refs/heads/master@{#25152}
>
> TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9719
> Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/105840
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25154}
TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com
Change-Id: I505ff3451eae81573531faef155ff35d7f894022
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/106500
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25220}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
Bug: webrtc:9419
Change-Id: I6f27003001548ea9d54412fdf62d5dd7a39cfd46
Reviewed-on: https://webrtc-review.googlesource.com/c/106022
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25187}
