Commit Graph

12 Commits

Author SHA1 Message Date
a59dcc3de2 Use Abseil container algorithms in api/
Bug: None
Change-Id: I87439a234d7018757eb61e99d5c6f9c7be4ab357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128825
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27272}
2019-03-25 22:11:06 +00:00
b1ae10b172 Add x-mt line to the offer.
We already support decoding of the x-mt line. This change adds the
a=x-mt line to the SDP offer. This is not a backward compatible change
for media transport (because of the changes in pre-shared key handling)

1) if media transport is enabled, and SDES is enabled, generate the
media transport offer.
2) if media transport generated the offer, add that offer to the x-mt
line.
3) in order to create media transport, require an x-mt line (backward incompatible).

The way it works is that
1) PeerConnection, on the offerer, asks jsep transport for the
configuration of the media transport.
2) Tentative media transport is created in JsepTransportController when
that happens.
3) SessionDescription will include configuration from this tentative
media transport.
4) When the LocalDescription is set on the offerer, the tentative media
transport is promoted to the real media transport.

Caveats:
- now we really only support MaxBundle. In the previous implementations,
two media transports were briefly created in some tests, and the second
one was destroyed shortly after instantiation.
- we, for now, enforce SDES. In the future, whether SDES is used will be
refactored out of the peer connection.

In the future (on the callee) we should ignore 'is_media_transport' setting. If
Offer contains x-mt, media transport should be used (if the factory is
present). However, we need to decide how to negotiate media transport
for data channels vs data transport for media (x-mt line at this point
doesn't differentiate the two, so we still need to use app setting).

This change also removes the negotation of pre-shared key from the
a=crypto line. Instead, media transport will have its own, 256bit key.
Such key should be transported in the x-mt line. This makes the code
much simpler, and simplifies the dependency / a=crypto lines parsing.

Also, adds a proper test for the connection re-offer (on both sides: callee and caller).
Before, it was possible that media transport could get recreated, based on the offer.
The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test.
This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even
when there is a re-offer.

Bug: webrtc:9719
Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01
Reviewed-on: https://webrtc-review.googlesource.com/c/125040
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26933}
2019-03-01 20:32:16 +00:00
9ded485caa Implement OpenChannel() on test media transports and make it pure virtual.
Bug: webrtc:9719
Change-Id: I9ec89fca7d4555f31b5192980f193b58d99e3b71
Reviewed-on: https://webrtc-review.googlesource.com/c/125100
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26910}
2019-03-01 00:24:07 +00:00
e049eba27c Revert "Add Sender and Receiver interfaces for MediaTransport audio"
This reverts commit 0d8eed6ac77fadf7f9bcf70c671710d60b1ee62d.

Reason for revert: crashes of unit tests.

Original change's description:
> Add Sender and Receiver interfaces for MediaTransport audio
> 
> Implement in LoopbackMediaTransport.
> 
> Bug: webrtc:9719
> Change-Id: I429ac3f78d99b8ea4f9ac85b9a3600b215b61a55
> Reviewed-on: https://webrtc-review.googlesource.com/c/121957
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26731}

TBR=solenberg@webrtc.org,nisse@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com

Change-Id: I02e409e1bbe2b2dea8a7b1aa08fa44d4146bda8f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/123232
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26733}
2019-02-18 09:52:40 +00:00
0d8eed6ac7 Add Sender and Receiver interfaces for MediaTransport audio
Implement in LoopbackMediaTransport.

Bug: webrtc:9719
Change-Id: I429ac3f78d99b8ea4f9ac85b9a3600b215b61a55
Reviewed-on: https://webrtc-review.googlesource.com/c/121957
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26731}
2019-02-18 08:51:26 +00:00
80a8687082 [clang-tidy] Apply performance-move-const-arg fixes (mutable lambdas).
This CL is a manual spin-off of [1], which tried to apply clang-tidy's
performance-move-const-arg [1] to the WebRTC codebase.

Since there were some wrong fixes to correct, this CL lands all the
manual fixes where std::move was actually fine but the lambda was not
mutable.

[1] - https://webrtc-review.googlesource.com/c/src/+/120350
[2] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html

Bug: webrtc:10252
Change-Id: I4602e3d4a63d2637dd389e775ffbf80fe95f40fc
Reviewed-on: https://webrtc-review.googlesource.com/c/120927
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26532}
2019-02-04 14:47:56 +00:00
05cf6be726 [clang-tidy] Apply performance-move-const-arg fixes.
This CL is a manual spin-off of [1], which tried to apply clang-tidy's
performance-move-const-arg [1] to the WebRTC codebase.

Since there are some wrong fixes to correct, this CL collects all the
fixes that could be landed as is.

[1] - https://webrtc-review.googlesource.com/c/src/+/120350
[2] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html

Bug: webrtc:10252
Change-Id: Ic4882213556344e65c66e27415e91ff6f89134d7
Reviewed-on: https://webrtc-review.googlesource.com/c/120814
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26515}
2019-02-01 15:02:36 +00:00
48c5493393 Add 'UpdateAllocationLimits' in media transport.
Bug: webrtc:9719
Change-Id: I90bd1d9858c259d7339420c574ad83d6fb18299c
Reviewed-on: https://webrtc-review.googlesource.com/c/118946
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26426}
2019-01-28 18:20:47 +00:00
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
4687915495 Enable use of MediaTransportInterface for video streams.
Bug: webrtc:9719
Change-Id: I8c6027b4b15ed641e42fd210b3ea87d121508a69
Reviewed-on: https://webrtc-review.googlesource.com/c/111751
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26219}
2019-01-11 14:06:15 +00:00
1c7f5f63d1 Add SetKeyFrameRequestCallback to MediaTransportInterface
And implemented in LoopbackMediaTransport.

Bug: webrtc:9719
Change-Id: I68b16c2b6ed5583ffe9a5266e3d4cb1d94afbb97
Reviewed-on: https://webrtc-review.googlesource.com/c/113523
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25948}
2018-12-10 14:01:31 +00:00
e0446cb80c Move implementation of LoopbackMediaTransport to .cc file
Needed for coming cls to be able to use rtc_base/timeutils.h, which
shouldn't be included by api/ headers.

Bug: webrtc:9719
Change-Id: Ia36c0a9218ad505e1eb4f2d9c26d44d5673c2632
Reviewed-on: https://webrtc-review.googlesource.com/c/112580
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25855}
2018-11-30 10:39:26 +00:00