This change allows EnableBundle and PushdownMediaDescription to
work with RtpTransceivers, which means they can be reused in the
Unified Plan version of SetLocalDescription.
Bug: webrtc:8587
Change-Id: I4d862556879c14cea06fdf9d5c7c29cc32e1057a
Reviewed-on: https://webrtc-review.googlesource.com/27762
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21092}
This rewrites UpdateSessionState to better show the logic common
to all description types and the logic specific to
offers/answers/etc. Separating these will allow more code to be
reused with the Unified Plan implementation.
Bug: webrtc:8587
Change-Id: I56e0370dcb8bb4b59af2a5209edcad4606480e1c
Reviewed-on: https://webrtc-review.googlesource.com/27322
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21065}
PeerConnection had an Action enum as a holdover from the
WebRtcSession merge with the same members as
cricket::ContentAction. Since ContentAction is used in more places
outside of PeerConnection, this change removes the Action enum and
replaces its use with cricket::ContentAction.
Bug: webrtc:8587
Change-Id: I3e825fe285dbaf6b3f128eccde0f38864171af13
Reviewed-on: https://webrtc-review.googlesource.com/27321
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21063}
Calls to SetLocalDescription and SetRemoteDescription in
PeerConnection delegate to many different internal helper methods
which can fail. The error ultimately needs to propagate to the
caller and cause the SetXXXDescription to fail. Right now these
methods signal errors by returning false and copying the error
message into an out parameter. This changes these methods to
return RTCError instead and avoid the use of the out parameter.
Bug: webrtc:8587
Change-Id: Ib1d31622be742718b74780110c1bbe273d66444e
Reviewed-on: https://webrtc-review.googlesource.com/27241
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21061}
Also renames methods for interacting with the session error. This
clarifies the scope of this error type and lets methods have a
local variable named |error| without confusing it with the
|error()| getter.
Bug: webrtc:8587
Change-Id: I90e6eed24d961abbce15e56a76a8793ff1a806ea
Reviewed-on: https://webrtc-review.googlesource.com/27124
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21060}
Eventually we want BaseChannel to depend on the RtpTransportInternal
instead of DtlsTransportInternal and share RtpTransport when bundling.
This CL is the first step.
Add SetRtpTransport and Init_w(RtptransportInternal*) to BaseChannel.
These two methods would replace the existing SetTransports and Init_w
methods.
Add new CreateVoice/VideoChannel methods to the ChannelManager which
take RtpTransportInternal instead of Dtls/PacketTransportInternal.
|cotnent_name| is removed from the SrtpTransport to simplify to code
since it is only used for debugging.
InitNetwork_n is removed from BaseChannel in CL as well.
Bug: webrtc:7013
Change-Id: I35b1565958548bd4896854c49e61d3ee160b7634
Reviewed-on: https://webrtc-review.googlesource.com/27840
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21057}
When the initial offer side uses the ICE lite implementation, and
initiates a peer connection with an endpoint with the full
implementation, the offer side assumes the controlled ICE role per
RFC5245 and the remote endpoint MUST take the controlling role.
This logic was partially implemented in SetRemoteTransportDescription in
reflection where the endpoint switches its role to the controlling after
receiving the offer. The bug was caused by the following
SetLocalDescription at the remote endpoint after creating the answer,
which overrides the role to the controlled since it has no initial offer
and the role is not reflected in SetLocalTransportDescription. This
results in no nomination of candidate pairs and timeout of establishing
the peer connection.
The fix adds reflection on one's ICE role in SetLocalTransportDescription.
This fix also takes into account the case when both sides use the lite
implementation of ICE and the initial offer side MUST take the controlling
role per RFC5245 in this case, which is the default behavior in the
current implementation.
Bug: webrtc:8531
Change-Id: I65edd296c155bff51fcdb28709975e6837f302d5
Reviewed-on: https://webrtc-review.googlesource.com/26780
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21053}
This refactoring reduces code duplication in PeerConnection and
will make it easier to use these methods with the Unified Plan
implementation.
Bug: webrtc:8587
Change-Id: I6afd44fff702290903555cbe7703198b6b091da6
Reviewed-on: https://webrtc-review.googlesource.com/26822
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21052}
This changes the CreateVoiceChannel/CreateVideoChannel helper
methods in PeerConnection to return the created channel instead of
setting it directly. That allows the Unified Plan version of
SetLocalDescription to use the same factory methods without the
assumption that there is at most one voice and one video channel.
Also simplifies and deduplicates the logic for determining the
transport name for a given channel in the presence of BUNDLE.
Bug: webrtc:8587
Change-Id: I1f156f45309ce2d08d6d5d5ed3c6e01fbf094b36
Reviewed-on: https://webrtc-review.googlesource.com/26821
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21050}
When enabling SDES in BaseChannel, SrtpTransport::EnableExternalAuth
should be called if the external authenication is enabled which is
not covered by the current test set.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: Ibbf458516a521a488e8e3bb4a5a29fca70a627f5
Reviewed-on: https://webrtc-review.googlesource.com/27761
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20960}
Add a method to DtlsSrtpTransport to cache the RTP Absolute Send Time
extension id. The method would be called when using DTLS-SRTP with
external authentication.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: Ie9edb9382cbb4cf43eea5da3030991a0d20293a5
Reviewed-on: https://webrtc-review.googlesource.com/27260
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20947}
This was only used to ensure compatibility with external projects.
Now that everyone has moved to RtpTransceiverDirection, these
methods are redundant.
Bug: webrtc:8558
Change-Id: Iff5a8d13f9a4300d06902fa0441ceaeebf6809a2
Reviewed-on: https://webrtc-review.googlesource.com/24746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20945}
The DtlsSrtpTransport takes the reponsiblity of setting up DTLS-SRTP from
the BaseChannel.
The BaseChannel doesn't handle the signals from the P2P layer transport anymore.
The RtpTransport handles the signals from the PacketTransportInternal and the
DtlsSrtpTransport handles the DTLS-specific signals and determines when to extract
the keys and setting the parameters.
In channel_unittests.cc, call from DTLS to SDES is expected to fail since the
fallback from DTLS to SDES is not supported.
Bug: webrtc:7013
Change-Id: I0a54e017986f5a8ae9710e79643a4651bef3c38f
Reviewed-on: https://webrtc-review.googlesource.com/24702
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20941}
This is causing compilation issues in a chromium cl because of type conflicts.
BUG=none
TBR=henrikg@webrtc.org
Tbr-ing to fix build issue upstream and because there's no code change.
Change-Id: Ia34ae3844fe3f57f047cb44422fa591f752b7bda
Reviewed-on: https://webrtc-review.googlesource.com/26680
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20921}
This change adds |transceiver_direction()| and
|set_transceiver_direction()| to MediaContentDescription so that
external users can switch off of MediaContentDirection.
This deprecates the use of |direction()| and |set_direction()|
for external users. Once everyone has moved off of those methods,
the signiture will change to return/set RtpTransceiverDirection.
Then external users can move back to these methods with the new
signature.
Bug: webrtc:8558
Change-Id: I7e3ba289d3a0ac738b364b0388621cc3e7bcf5d3
Reviewed-on: https://webrtc-review.googlesource.com/24743
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20900}
Replaces cricket::RtpTransceiverDirection with
webrtc::RtpTransceiverDirection, which is part of the public API.
Bug: webrtc:8558
Change-Id: Ibfc9373e25187e98fb969e7ac937a1371c8fa4c7
Reviewed-on: https://webrtc-review.googlesource.com/24129
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20899}
WebRTC 1.0 has added the transceiver API to PeerConnection. This
is the first step towards exposing this to WebRTC consumers. For
now, transceivers can be added and fetched but there is not yet
support for creating offers/answers or setting local/remote
descriptions. That support ("Unified Plan") will be added in
follow-up CLs.
The transceiver API is currently only available if the application
opts in by specifying the kUnifiedPlan SDP semantics when creating
the PeerConnection.
Bug: webrtc:7600
Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
Reviewed-on: https://webrtc-review.googlesource.com/23880
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20896}
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.
Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}
Description for changes from the original CL:
Calling legacy SRD, implemented using
SetRemoteDescriptionObserverAdapter wrapping the old observer, was
meant to have the exact same behavior as the legacy SRD implementation
which invokes the callbacks in a Post.
However, in the CL that landed and got reverted (PS1), the Adapter had
its own message handler, and callbacks would be invoked even if the PC
was destroyed.
In PS2 I've changed the Adapter to use the PeerConnection's message
handler. If the PC is destroyed, the callback will not be invoked.
This gives identical behavior to before this CL, and the legacy
behavior is covered by a new unittest.
I changed the adapter to be an implementation detail of
peerconnection.cc, therefor some stuff was moved, and the only tests
covering this is now in peerconnection_rtp_unittest.cc.
This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72
Original change's description:
> SetRemoteDescriptionObserverInterface added.
>
> The new observer replaced SetSessionDescriptionObserver for
> SetRemoteDescription. Unlike SetSessionDescriptionObserver,
> SetRemoteDescriptionObserverInterface is invoked synchronously so
> that the you can rely on the state of the PeerConnection to represent
> the result of the SetRemoteDescription call in the callback.
>
> The new observer succeeds or fails with an RTCError.
>
> This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
> and SetSessionDescriptionObserver, with the benefit that all media
> object changes can be processed in a single callback by the application
> in a synchronous callback. This will help Chromium keep objects in-sync
> across layers and threads in a non-racy and straight-forward way, see
> design doc (Proposal 2):
> https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
>
> An adapter for SetSessionDescriptionObserver is added to allow calling
> the old SetRemoteDescription signature and get the old behavior
> (OnSuccess/OnFailure callback in a Post) until third parties switch.
>
> Bug: webrtc:8473
> Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
> Reviewed-on: https://webrtc-review.googlesource.com/17523
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20841}
TBR=pthatcher@webrtc.org
Bug: webrtc:8473
Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5
Reviewed-on: https://webrtc-review.googlesource.com/25640
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20854}
This reverts commit 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72.
Reason for revert: Third party project breaks due to use-after-free
in the callback. I suspect this is because the adapter is processing
the async callback instead of the pc, i.e. callback is called from
SetRemoteDescriptionObserverAdapter::OnMessage instead of from
PeerConnection::OnMessage. This makes it possible for the callback to
be invoked after the PC is destroyed.
I argue this is how it should be done, and that if you're using a raw
pointer in an async callback you're doing it wrong, but I will reland
this CL with the callback processed in PeerConnection::OnMessage
instead as to not change the behavior of the old SRD signature.
Original change's description:
> SetRemoteDescriptionObserverInterface added.
>
> The new observer replaced SetSessionDescriptionObserver for
> SetRemoteDescription. Unlike SetSessionDescriptionObserver,
> SetRemoteDescriptionObserverInterface is invoked synchronously so
> that the you can rely on the state of the PeerConnection to represent
> the result of the SetRemoteDescription call in the callback.
>
> The new observer succeeds or fails with an RTCError.
>
> This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
> and SetSessionDescriptionObserver, with the benefit that all media
> object changes can be processed in a single callback by the application
> in a synchronous callback. This will help Chromium keep objects in-sync
> across layers and threads in a non-racy and straight-forward way, see
> design doc (Proposal 2):
> https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
>
> An adapter for SetSessionDescriptionObserver is added to allow calling
> the old SetRemoteDescription signature and get the old behavior
> (OnSuccess/OnFailure callback in a Post) until third parties switch.
>
> Bug: webrtc:8473
> Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
> Reviewed-on: https://webrtc-review.googlesource.com/17523
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20841}
TBR=hbos@webrtc.org,hta@webrtc.org,pthatcher@webrtc.org,guidou@webrtc.org
Change-Id: I715555e084d9aae49ee2a8831c70dc006dbdb74c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8473
Reviewed-on: https://webrtc-review.googlesource.com/25580
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20850}
Previously when creating an answer, we used the codecs from the remote list
to get correctly mapped codec id, but the side effect is that we also used
the remote codec parameters which is wrong.
With this change, we use the local codecs with local parameters and the
|NegotiateCodecs| method will take care of the codec id mapping.
Bug: webrtc:8550
Change-Id: Ib7b2fe5f6ed3a72c58cf5df8bf2c1d00476c141c
Reviewed-on: https://webrtc-review.googlesource.com/25285
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20843}
The new observer replaced SetSessionDescriptionObserver for
SetRemoteDescription. Unlike SetSessionDescriptionObserver,
SetRemoteDescriptionObserverInterface is invoked synchronously so
that the you can rely on the state of the PeerConnection to represent
the result of the SetRemoteDescription call in the callback.
The new observer succeeds or fails with an RTCError.
This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
and SetSessionDescriptionObserver, with the benefit that all media
object changes can be processed in a single callback by the application
in a synchronous callback. This will help Chromium keep objects in-sync
across layers and threads in a non-racy and straight-forward way, see
design doc (Proposal 2):
https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
An adapter for SetSessionDescriptionObserver is added to allow calling
the old SetRemoteDescription signature and get the old behavior
(OnSuccess/OnFailure callback in a Post) until third parties switch.
Bug: webrtc:8473
Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
Reviewed-on: https://webrtc-review.googlesource.com/17523
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20841}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=hbos@webrtc.org
Bug: None
Change-Id: I2fc923e788b744fc167f02f9fecd19bea512af81
Reviewed-on: https://webrtc-review.googlesource.com/23613
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20834}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=hbos@webrtc.org
Bug: None
Change-Id: I8f22aef3ad016c5714bc09351135ec4c65ff0cbd
Reviewed-on: https://webrtc-review.googlesource.com/23577
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20830}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=hbos@webrtc.org
Bug: None
Change-Id: I57a1ac8c2a05de403ff87b319c7a50fad17c1c96
Reviewed-on: https://webrtc-review.googlesource.com/23571
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20827}
This makes the receiver know about its associated set of streams, the
equivalent of the [[AssociatedRemoteMediaStreams]] slot in the spec,
https://w3c.github.io/webrtc-pc/#dfn-x%5B%5Bassociatedremotemediastreams%5D%5D
This does not change layers below peerconnection.cc. The streams are set
upon the receiver's construction and is not modified for the duration of
its lifetime.
When we support modifying the associated set of streams of a receiver
the receiver needs to know about it. The receiver's streams() should be
used in all places where a receiver's streams need to be known.
Bug: webrtc:8473
Change-Id: I31202973aed98e61fa9b6a78b52e815227b6c17d
Reviewed-on: https://webrtc-review.googlesource.com/22922
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20825}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
TBR=pthatcher@webrtc.org
Bug: None
Change-Id: If41c462dc3ddff664d0b70d249d760e2ca4c8ab3
Reviewed-on: https://webrtc-review.googlesource.com/23576
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20820}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=hbos@webrtc.org
Bug: None
Change-Id: I435258d0b16660bcc58c236158c19507ef60f57b
Reviewed-on: https://webrtc-review.googlesource.com/23609
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20817}
Some targets depend on "api/peerconnectioninterface.h" which is part of
//api:peerconnection_and_implicit_call_api.
Furthermore, peerconnection_and_implicit_call_api depends among other
things of headers in //media:rtc_media_base, so we should add it as a
dependency as well.
Bug: webrtc:7504
Change-Id: Ifab69253d52532876509b3507917b1c93d99a2ac
Reviewed-on: https://webrtc-review.googlesource.com/24660
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20815}
This reverts commit 33c5c7f5e4f018a5103770021328fc530d451d75.
Reason for revert: Fix broken API change.
TBR=sprang@webrtc.org,solenberg@webrtc.org
TBRing because only pc/ and api/ have changed since last LGTMed version.
Original change's description:
> Revert "Encode log events periodically instead of for every event."
>
> This reverts commit b154c27e72fddb6c0d7cac69a9c68fff22154519.
>
> Reason for revert: Broke the internal project.
>
> Original change's description:
> > Encode log events periodically instead of for every event.
> >
> > Updated unit test to take output_period and random seed as parameters.
> > Updated the peerconnection interface to allow passing in an output_period.
> >
> > This is in preparation of some upcoming CLs that will change the format
> > to store batches of delta-encoded values.
> >
> >
> > Bug: webrtc:8111
> > Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
> > Reviewed-on: https://webrtc-review.googlesource.com/22600
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20736}
>
> Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229
> Bug: webrtc:8111
> Reviewed-on: https://webrtc-review.googlesource.com/24160
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20738}
Bug: webrtc:8111
Change-Id: Ie69862cd52d11c1e15adeb6e2caacafe16863c80
Reviewed-on: https://webrtc-review.googlesource.com/24620
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20811}
This setting allows the user of PeerConnection to choose whether
to use Plan B (current) or Unified Plan (future) semantics.
Unified Plan semantics are not yet supported.
Bug: chromium:465349
Change-Id: I77a5c376c83f335f734488e11e619582a314bffe
Reviewed-on: https://webrtc-review.googlesource.com/22766
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20806}
The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel.
DtlsSrtpTransport is responsible for exporting keys from DtlsTransport
and setting up the wrapped SrtpTransport.
The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL.
Bug: webrtc:7013
Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713
Reviewed-on: https://webrtc-review.googlesource.com/6700
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20804}
Moves ownership of the RtpSenders/RtpReceivers/BaseChannels to
RtpTransceiver objects. For now, there can only be one
RtpTransceiver for audio and one for video. Future work to
implement Unified Plan will relax this restriction.
Bug: webrtc:7600
Change-Id: I9dfe324de61e2b363948858da72624396e27fc1a
Reviewed-on: https://webrtc-review.googlesource.com/21461
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20802}
This new parallel GetStatistics function uses Optionals to indicate if stats are valid or not, and no longer relies on default values. It also takes an argument to indicate if receive streams are present, and if not several stats will not be set.
Bug: b/67926135
Change-Id: I175de1c65c414bea6ec9ca8b0b16f07cb2308d9f
Reviewed-on: https://webrtc-review.googlesource.com/17942
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20789}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=hta@webrtc.org
Bug: None
Change-Id: Ie5488de731bbd377d7694c1c26af26168bf6afd3
Reviewed-on: https://webrtc-review.googlesource.com/23606
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20777}
This reverts commit b154c27e72fddb6c0d7cac69a9c68fff22154519.
Reason for revert: Broke the internal project.
Original change's description:
> Encode log events periodically instead of for every event.
>
> Updated unit test to take output_period and random seed as parameters.
> Updated the peerconnection interface to allow passing in an output_period.
>
> This is in preparation of some upcoming CLs that will change the format
> to store batches of delta-encoded values.
>
>
> Bug: webrtc:8111
> Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
> Reviewed-on: https://webrtc-review.googlesource.com/22600
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20736}
TBR=solenberg@webrtc.org,eladalon@webrtc.org,terelius@webrtc.org,tommi@webrtc.org,sprang@webrtc.org,pthatcher@webrtc.org
Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/24160
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20738}
Updated unit test to take output_period and random seed as parameters.
Updated the peerconnection interface to allow passing in an output_period.
This is in preparation of some upcoming CLs that will change the format
to store batches of delta-encoded values.
Bug: webrtc:8111
Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
Reviewed-on: https://webrtc-review.googlesource.com/22600
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20736}
Indicates if the forced sw fallback has had an effect (or would have had an effect if it had been
enabled).
Bug: webrtc:6634
Change-Id: I574b9001a2fae650fb894a1caa0d0f84257658e3
Reviewed-on: https://webrtc-review.googlesource.com/23300
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20729}
int32_t SetLoudspeakerStatus(bool enable)
int32_t GetLoudspeakerStatus(bool* enabled) const
These APIs are only implemented on iOS and they do not belong in the
native audio layer since the client can achieve the same functionality
by using the shared audio session in sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h.
It also gives the client a better flexibility in how the audio routing is done.
Bug: webrtc:7306
Change-Id: I853e2f57e0f5ae0a0f9fc4729ce961d81f92588b
Reviewed-on: https://webrtc-review.googlesource.com/23740
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20721}
