Commit graph

426 commits

Author SHA1 Message Date
Amit Hilbuch
ea7ef2ad1d Refactoring RtpSenderInternal to share implementation for Audio & Video.
Most of the implementation in rtp_sender.cc is a copy paste for both
Audio & Video RTP senders. This change moves all the common behavior
into the base RtpSenderInternal class.
Template method pattern is used to accomodate for the very slight differences
between audio and video senders.

Bug: None
Change-Id: I6d4e93cd32fbb0fb361fd0e1883791019bde9a92
Reviewed-on: https://webrtc-review.googlesource.com/c/123411
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26758}
2019-02-20 01:23:04 +00:00
Amit Hilbuch
2297d3311a Rejected simulcast layers will no longer appear in GetParameters().
Added a layer in RtpSender that bridges the gap between the layers
that the user sees and the layer that the media engine sees.
Media engine still maintains the invariant that the number of layers
cannot be changed, while RtpSender adds and removes layers between
the user GetParameters and SetParameters calls and the media engine.

Bug: webrtc:10251
Change-Id: I33839c1f9a9052cb6130253e5a582606f2cbe54a
Reviewed-on: https://webrtc-review.googlesource.com/c/122641
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26756}
2019-02-19 22:01:53 +00:00
Karl Wiberg
8d2e228991 Add thread safety annotations for PeerConnection::*_state_
Plus all the annotations that are necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: If7bbd5a468a8c50ac3cfe03cd2ed4f5b5f461195
Reviewed-on: https://webrtc-review.googlesource.com/c/123047
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26727}
2019-02-17 13:02:33 +00:00
Ruslan Burakov
7ea460593c Add latency to remote source api.
Latency corresponds to base minimum delay on NetEq.

Bug: webrtc:10287
Change-Id: I538d202e3e4fe07b779c46bf560e2fde38e0468e
Reviewed-on: https://webrtc-review.googlesource.com/c/121704
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26724}
2019-02-16 02:13:44 +00:00
Karl Wiberg
b03ab7107c Add thread safety annotation for PeerConnection::event_log_
Plus all the annotations that are necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: I68a51bfa3342c6d66d67276d5979144af34692c9
Reviewed-on: https://webrtc-review.googlesource.com/c/123046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26716}
2019-02-15 18:31:50 +00:00
Karl Wiberg
744310fcea Add thread safety annotation for PeerConnection::observer_ and factory_
Plus all the annotations that are necessary to make things compile
again.

(The latter could only be made const, since it is accessed from both
the signal thread and the worker thread.)

Bug: webrtc:9987
Change-Id: I49f1568d26a327f48cd94c2c70bc5939e9fbd59a
Reviewed-on: https://webrtc-review.googlesource.com/c/122842
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26715}
2019-02-15 17:17:51 +00:00
Karl Wiberg
106d92d4c9 Add thread safety annotation for PeerConnection::SignalDataChannelCreated_
Plus all the annotations that are necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: I53caa3c6644c5d0fcb50f7c1e89e853eddd769a5
Reviewed-on: https://webrtc-review.googlesource.com/c/122882
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26705}
2019-02-15 10:49:48 +00:00
Amit Hilbuch
aa58415bfa Reland "Enabling Simulcast use via AddTransceiver."
This is a reland of ce470aab51

Original change's description:
> Enabling Simulcast use via AddTransceiver.
> 
> This change removes the restriction on multiple send encodings when
> calling AddTransceiver. If RIDs are not provided in the
> simulcast scenario, they are auto-generated by the platform.
> 
> This effectively enables the use of spec-compliant simulcast.
> Tests are also added to verify simulcast functionality.
> 
> Bug: webrtc:10075
> Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2
> Reviewed-on: https://webrtc-review.googlesource.com/c/121389
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26590}

Bug: webrtc:10075
Change-Id: Ib4392e36d5d60b966ecff56d4df69bf60d13a73d
Reviewed-on: https://webrtc-review.googlesource.com/c/122962
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26694}
2019-02-14 20:58:34 +00:00
Ruslan Burakov
501bfba0cb Split rtp_receiver for readability.
Bug: webrtc:10304
Change-Id: I85e421060d8560cf36cdb5970ae190efc77e7709
Reviewed-on: https://webrtc-review.googlesource.com/c/122085
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26634}
2019-02-11 12:47:51 +00:00
Emircan Uysaler
78323436cf Revert "Enabling Simulcast use via AddTransceiver."
This reverts commit ce470aab51.

Failing below Layout test.
https://cs.chromium.org/chromium/src/third_party/blink/web_tests/external/wpt/webrtc/RTCRtpReceiver-getParameters-expected.txt?type=cs&sq=package:chromium&g=0

Original change's description:
> Enabling Simulcast use via AddTransceiver.
> 
> This change removes the restriction on multiple send encodings when
> calling AddTransceiver. If RIDs are not provided in the
> simulcast scenario, they are auto-generated by the platform.
> 
> This effectively enables the use of spec-compliant simulcast.
> Tests are also added to verify simulcast functionality.
> 
> Bug: webrtc:10075
> Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2
> Reviewed-on: https://webrtc-review.googlesource.com/c/121389
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26590}

TBR=steveanton@webrtc.org,orphis@webrtc.org,amithi@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10075
Change-Id: Idef5ca735eaef190f83eec8630cd54e23737d813
Reviewed-on: https://webrtc-review.googlesource.com/c/122040
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26618}
2019-02-08 21:40:45 +00:00
Steve Anton
d7180cccc4 Also check the pending remote description when generating MIDs for legacy remote offers
Bug: webrtc:10296
Change-Id: Ia10299177175e57d3f494281310d6c91bed9ebdb
Reviewed-on: https://webrtc-review.googlesource.com/c/121860
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26591}
2019-02-07 20:22:38 +00:00
Amit Hilbuch
ce470aab51 Enabling Simulcast use via AddTransceiver.
This change removes the restriction on multiple send encodings when
calling AddTransceiver. If RIDs are not provided in the
simulcast scenario, they are auto-generated by the platform.

This effectively enables the use of spec-compliant simulcast.
Tests are also added to verify simulcast functionality.

Bug: webrtc:10075
Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2
Reviewed-on: https://webrtc-review.googlesource.com/c/121389
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26590}
2019-02-07 17:29:59 +00:00
Niels Möller
260a71d47c Delete deprecated method PeerConnectionFactory::CreateVideoSource
Bug: webrtc:6353
Change-Id: Icb8847b234e7a844a4dff9ff44861f6967ac7b5b
Reviewed-on: https://webrtc-review.googlesource.com/c/118661
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26587}
2019-02-07 14:24:02 +00:00
Mirko Bonadei
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
Mirko Bonadei
9f3a44f515 Introcuce RTCError(const T&) constructor.
This CL is spawned from [1] and it introduces RTCError(const T&) in
order to remove an unneeded std::move.

[1] - https://webrtc-review.googlesource.com/c/src/+/120350

Bug: webrtc:10252
Change-Id: Ibd5aa1c901fd920549e9437908178c786019a328
Reviewed-on: https://webrtc-review.googlesource.com/c/120560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26468}
2019-01-30 13:43:29 +00:00
Florent Castelli
c1a0bcbe89 Implement the encoding RtpParameter scaleResolutionDownBy
Support varies by codec, especially in the simulcast case, but using
the EncoderSimulcastProxy codec should fix this.

Bug: webrtc:10069
Change-Id: Idb6a5f400ffda1cdb139004f540961a9cf85d224
Reviewed-on: https://webrtc-review.googlesource.com/c/119400
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26449}
2019-01-29 14:32:17 +00:00
Steve Anton
64b626b03f Use Abseil container algorithms in pc/
Bug: None
Change-Id: If784461b54d95bdc6f8a7d4e5d1bbfa52d1a390e
Reviewed-on: https://webrtc-review.googlesource.com/c/119862
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26433}
2019-01-29 02:33:50 +00:00
Amit Hilbuch
b7446ed257 Removing receive RIDs and Simulcast Layers.
In the January 22nd 2019 WebRTC meeting it was agreed that an offer
for sending (or receiving) simulcast should only contain the RIDs
of the layers that are sent by the client.
This change removes the complexity that was added to support sending
and receiving the single layer (and RID) that are sent from the server.

Bug: webrtc:10076
Change-Id: I8bae1336d5cb8ba2f91c5b62332dc69e67ddfd47
Reviewed-on: https://webrtc-review.googlesource.com/c/120242
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26432}
2019-01-29 00:54:26 +00:00
Amit Hilbuch
bcd39d483d Creating Simulcast offer and answer in Peer Connection.
CreateOffer and CreateAnswer will now examine the layers on the
transceiver to determine if multiple layers are requested (Simulcast).
In this scenario RIDs will be used in the layers (instead of SSRCs).
When the offer is created, only RIDs are signalled in the offer.
When the offer is set locally SetLocalDescription() SSRCs will be
generated for each layer by the Channel and sent downstream to the
MediaChannel.
The MediaChannel receives configuration that looks identical to that of
legacy simulcast, and should be able to integrate the streams correctly
regardless of how they were signalled.
Setting multiple layers on the transciever is still not supported
through the API.

Bug: webrtc:10075
Change-Id: Id4ad3637b87b68ef6ca7eec69166fee2d9dfa36f
Reviewed-on: https://webrtc-review.googlesource.com/c/119780
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26428}
2019-01-28 18:56:02 +00:00
Mirko Bonadei
739baf097b [clang-tidy] Apply performance-for-range-copy fixes.
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.

All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html

Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
2019-01-28 09:53:50 +00:00
Piotr (Peter) Slatala
55b91b988f Only create no-op DTLS if media transport is used for both media and data
Currently it's possible that no-op DTLS is created if media transport is only used for data channels.
Changing it so that no-op DTLS is only created when both media & data will flow through media transport.

Bug: webrtc:9719
Change-Id: I87f27fc778ea21b12f2904bad1452d893f66b541
Reviewed-on: https://webrtc-review.googlesource.com/c/119909
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26416}
2019-01-26 00:04:22 +00:00
Amit Hilbuch
dbb49dfb27 Moving UniqueIdGenerator into rtc_base.
UniqueIdGenerator classes are useful outside the pc directory.
This change moves them to the rtc_base directory to enable code
in all directories to reference them.

Bug: None
Change-Id: I1c77da87ea26d9611f37dc1d4d2c16006a6589c6
Reviewed-on: https://webrtc-review.googlesource.com/c/119460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26378}
2019-01-24 00:52:31 +00:00
Yves Gerey
ae6e0b2058 [CodeHealth] Fix use after std::move instances.
The parameter was expanded twice in the macro,
leading to double use and move.
This is both an example of:
 * Issue spotted by clandtidy's bug-prone patterns.
 * Premature optimization.

Bug: webrtc:9855
Change-Id: I1a0cb2c99f95c6aec79ba1eb198aa39743ccbcd9
Reviewed-on: https://webrtc-review.googlesource.com/c/119042
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26367}
2019-01-23 11:04:11 +00:00
Harald Alvestrand
4a7b3acfcf Add DTLSTransport info into sender/receiver state.
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.

Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26289}
2019-01-17 10:21:32 +00:00
Steve Anton
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
Steve Anton
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
Renamed from pc/peerconnection.cc (Browse further)