This change injects the FrameEncryptorInterface and the FrameDecryptorInterface
into the RtpSenderInterface and RtpReceiverInterface respectively. This is the
second stage of the injection. In a follow up CL non owning pointers to these
values will be passed down into the media channel.
This change also updates the corresponding mock files.
Bug: webrtc:9681
Change-Id: I964084fc270e10af9d1127979e713493e6fbba7d
Reviewed-on: https://webrtc-review.googlesource.com/96625
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24489}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'pc'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
This also changes RtpReceiver and RemoteAudioSource to have two-step
initialization, since in Unified Plan RtpReceivers are created much
earlier than in Plan B.
Bug: webrtc:7600
Change-Id: Ia135d25eb8bcab22969007b3a825a5a43ce62bf4
Reviewed-on: https://webrtc-review.googlesource.com/39382
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21681}
This is a reland of e357a4dd4e3b015f8281813f246de793589bd537
Original change's description:
> Move stats ID generation from SSRC to local ID
>
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
>
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
>
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}
TBR=solenberg@webrtc.org
Bug: webrtc:8673
Change-Id: I610302efc5393919569b77e3b59aa3384a9b88a5
Reviewed-on: https://webrtc-review.googlesource.com/38842
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21589}
This reverts commit e357a4dd4e3b015f8281813f246de793589bd537.
Reason for revert: Looks like it's breaking some downstream projects.
Original change's description:
> Move stats ID generation from SSRC to local ID
>
> This generates stats IDs for Track stats (which
> represents stats on the attachment of a track to
> a PeerConnection) from being SSRC-based to being
> based on an ID that is allocated when connecting the
> track to the PC.
>
> This is a prerequisite to generating stats before
> the PeerConnection is connected.
>
> Bug: webrtc:8673
> Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
> Reviewed-on: https://webrtc-review.googlesource.com/38360
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21582}
TBR=solenberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
Change-Id: I621c10236c02be01d82f4660168f0323b85e24af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8673
Reviewed-on: https://webrtc-review.googlesource.com/38681
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21586}
This generates stats IDs for Track stats (which
represents stats on the attachment of a track to
a PeerConnection) from being SSRC-based to being
based on an ID that is allocated when connecting the
track to the PC.
This is a prerequisite to generating stats before
the PeerConnection is connected.
Bug: webrtc:8673
Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4
Reviewed-on: https://webrtc-review.googlesource.com/38360
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21582}
When Unified Plan semantics are set, PeerConnection will fire OnAddTrack
according to the WebRTC spec. OnRemoveTrack will never be fired and will
be deprecated in the future.
Bug: webrtc:7600
Change-Id: Idfaada65b795b5fb9fe4844cff036d52ea90da17
Reviewed-on: https://webrtc-review.googlesource.com/38122
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21564}
Currently, the RtpReceivers take a BaseChannel which is (mostly)
just used for proxying calls to the MediaChannel. This change
removes the extra layer and moves the proxying logic to RtpReceiver.
Bug: webrtc:8587
Change-Id: I01b0e3d57b4629e43d9d148cc94d6dd2941d320e
Reviewed-on: https://webrtc-review.googlesource.com/38120
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21562}
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}
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}
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}
