Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
type-based demuxing. RtpTransport will support MID-based demuxing later.
Each BaseChannel has its own RTP demuxing criteria and when connecting
to the RtpTransport, BaseChannel will register itself as a demuxer sink.
The inheritance model is changed. New inheritance chain:
DtlsSrtpTransport->SrtpTransport->RtpTranpsort
The JsepTransport2 is renamed to JsepTransport.
NOTE:
When RTCP packets are received, Call::DeliverRtcp will be called for
multiple times (webrtc:9035) which is an existing issue. With this CL,
it will become more of a problem and should be fixed.
Bug: webrtc:8587
Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62
Reviewed-on: https://webrtc-review.googlesource.com/65786
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22867}
Currently in the SDP we require an a=ssrc line in the m= section in
order for a StreamParams object to be created with that
MediaContentDescription. This change creates a StreamParams object
without ssrcs in the case that a=msid lines are signaled, but ssrcs
are not. When the remote description is set, this allows us to store
the "unsignaled" StreamParams object in the media channel to later
be used when the first packet is received and we create the
receive stream.
Bug: webrtc:7932, webrtc:7933
Change-Id: Ib6734abeee62b8ed688a8208722c402134c074ef
Reviewed-on: https://webrtc-review.googlesource.com/63141
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22712}
The TransportController will be replaced by the JsepTransportController
and JsepTransport will be replace be JsepTransport2.
The JsepTransportController will take the entire SessionDescription
and handle the RtcpMux, Sdes and BUNDLE internally.
The ownership model is also changed. The P2P layer transports are not
ref-counted and will be owned by the JsepTransport2.
In ORTC aspect, RtpTransportAdapter is now a wrapper over RtpTransport
or SrtpTransport and it implements the public and internal interface
by calling the transport underneath.
Bug: webrtc:8587
Change-Id: Ia7fa61288a566f211f8560072ea0eecaf19e48df
Reviewed-on: https://webrtc-review.googlesource.com/59586
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22693}
This is intended to ensure compatibility between Plan B and
Unified Plan endpoints for the single audio - single video case.
If Unified Plan is the offerer, it will add a=msid and a=ssrc MSID
entries to its offer.
If Unified Plan is the answerer, it will use whatever MSID
signaling mechanism was used in the offer (either a=msid or
a=ssrc).
Bug: webrtc:7600
Change-Id: I6192dec19123fbb56f5d04540d2175c7fb30b9b6
Reviewed-on: https://webrtc-review.googlesource.com/44162
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21859}
SDP is a detail of PeerConnection and is not used by anything in p2p, so
it belongs in the pc/ directory. This also allows
MediaContentDescription to be inlined in the future.
Bug: webrtc:8620
Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce
Reviewed-on: https://webrtc-review.googlesource.com/33781
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21376}
This is a reland of 4770fd935ac92400487bddd3b755753572e6d692
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
Bug: webrtc:8636
Change-Id: Ibce42be898b96dd8e0266b595611d2ffc86581a8
Reviewed-on: https://webrtc-review.googlesource.com/34586
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21371}
This reverts commit 4770fd935ac92400487bddd3b755753572e6d692.
Reason for revert: breaks downstream projects
Original change's description:
> Move JsepTransport from p2p/base to pc/.
>
> The JsepTransport class is moved to pc/ and the utility methods and
> enums are moved to where they are used.
>
> With JsepTransport moved to pc/, JsepTransport can depend on objects in
> pc/ including RtpTranport, SrtpTransport etc.
>
> Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7
>
> Bug: webrtc:8636
> Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
> Reviewed-on: https://webrtc-review.googlesource.com/33701
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21333}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,pthatcher@webrtc.org
Change-Id: Ia72c6d7f185a95b21fd0aec90e7fdc00cb1fb423
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8636
Reviewed-on: https://webrtc-review.googlesource.com/34600
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21335}
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}
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}
This will help keep ortc dependencies clean in the future, since
gn --check forces us to depend on components from which we include
headers.
cryptoparams.h moves into api/, but ortc appears to think it
should be there anyway. We could consider moving it into the ortc/ api,
but it doesn't appear to be specific to ortc.
Bug: webrtc:6828
Change-Id: Iddae438d10b5e84b2fbc52565364319e20f90613
Reviewed-on: https://webrtc-review.googlesource.com/22660
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20686}
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.
Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
