Commit graph

111 commits

Author SHA1 Message Date
Niels Möller
cce51b628a Delete unused header file sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h
Bug: webrtc:9627
Change-Id: Iac95406dc512480788f222261db5c9b578b36bfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185810
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32221}
2020-09-29 08:42:02 +00:00
Jonas Olsson
0182a0300f Reland "Remove the injectable bitrate allocation strategy API."
This is a reland of 80cb3f6db6

Original change's description:
> Remove the injectable bitrate allocation strategy API.
>
> This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
> plus a ton of now-dead code.
>
> Bug: webrtc:10556
> Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28523}

TBR=kwiberg@webrtc.org

Bug: webrtc:10556
Change-Id: Ic17a7a7cc447292306876ee9582ad62fd2499765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145900
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28585}
2019-07-17 10:20:45 +00:00
Mirko Bonadei
e95b57cdfc Revert "Remove the injectable bitrate allocation strategy API."
This reverts commit 80cb3f6db6.

Reason for revert: Performance regression on downstream project.

Original change's description:
> Remove the injectable bitrate allocation strategy API.
> 
> This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
> plus a ton of now-dead code.
> 
> Bug: webrtc:10556
> Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28523}

TBR=henrika@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,srte@webrtc.org,alexnarest@webrtc.org,jonasolsson@webrtc.org

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

Bug: webrtc:10556
Change-Id: Ife905d661e7b1a227662395c729a9336c62fd2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145338
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28560}
2019-07-12 15:27:19 +00:00
Jonas Olsson
80cb3f6db6 Remove the injectable bitrate allocation strategy API.
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.

Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
2019-07-10 13:13:25 +00:00
Anders Carlsson
7bca8ca4e2 Obj-C SDK Cleanup
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.

A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.

The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.

The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.

Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
2018-08-30 10:42:41 +00:00
Niels Möller
f06f923ef0 Delete almost all use of MediaConstraintsInterface in the PeerConnection API
Bug: webrtc:9239
Change-Id: I04f4370f624346bf72c7e4e090b57987b558213b
Reviewed-on: https://webrtc-review.googlesource.com/74420
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24396}
2018-08-23 07:14:37 +00:00
Zeke Chin
8de502ba11 Add didRemoveReceiver delegate callback.
Bug: None
Change-Id: I7d3badc9005f51a641febd359d037ed37a205101
Reviewed-on: https://webrtc-review.googlesource.com/95241
Commit-Queue: Zeke Chin <tkchin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24391}
2018-08-22 17:51:03 +00:00
Niels Möller
2377588c82 Add accessor methods for RTP timestamp of EncodedImage.
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.

Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
2018-08-21 09:15:51 +00:00
Michael Iedema
ccee56beee Add certificate generate/set functionality to bring iOS closer to JS API
The JS API supports two operations which have never been implemented in
the iOS counterpart:
 - generate a new certificate
 - use this certificate when creating a new PeerConnection

Both functions are illustrated in the generateCertificate example code:
 - https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/generateCertificate

Currently, on iOS, a new certificate is automatically generated for
every PeerConnection with no programmatic way to set a specific
certificate.

Work sponsored by |pipe|

Bug: webrtc:9498
Change-Id: Ic1936c3de8b8bd18aef67c784727b72f90e7157c
Reviewed-on: https://webrtc-review.googlesource.com/87303
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24276}
2018-08-13 22:25:15 +00:00
Mirko Bonadei
17aff35e1d Enable clang::find_bad_constructs for sdk/ (part 1).
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I6f03c46e772ccf4d15951a4b9d4e12015d539e58
Reviewed-on: https://webrtc-review.googlesource.com/90408
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24113}
2018-07-26 12:16:31 +00:00
Benjamin Wright
d0136b8afb Added API to Objective-C PeerConnectionFactoryOptions to enable GCM Ciphers.
This changeset adds the ability for API users to enable or disable GCM Cipher
suites from objective-c.

Bug: chromium:713701
Change-Id: I0ac7b60f55dd56bebbcfb315a542ef4843099802
Reviewed-on: https://webrtc-review.googlesource.com/89263
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24028}
2018-07-18 18:10:26 +00:00
Yura Yaroshevich
01cee079dc Fixed crash when PCF is destroyed before MediaSource/Track in ObjC
Bug: webrtc:9231
Change-Id: I31b86aa560f4ad230c9a94fedebebf320e0370a4
Reviewed-on: https://webrtc-review.googlesource.com/88221
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23981}
2018-07-16 12:03:16 +00:00
Yura Yaroshevich
7a16c54571 Fixed crash when PCF is destroyed before RTCRtpReceiver in ObjC
Bug: webrtc:9231
Change-Id: Ic532b7661bb8765f0fc2309d2ad530f664ccfd14
Reviewed-on: https://webrtc-review.googlesource.com/87840
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23931}
2018-07-11 11:16:56 +00:00
Yura Yaroshevich
ef43aafcf5 Fixed crash when PCF is destroyed before RTCRtpSender in ObjC
Bug: webrtc:9231
Change-Id: I3b90400bf619938817d7a04a7a1130ba86ad65df
Reviewed-on: https://webrtc-review.googlesource.com/87623
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23896}
2018-07-10 07:35:35 +00:00
Yura Yaroshevich
08f14dd388 Fixed crash when PCF is destroyed before RTCRtpTranceiver in ObjC
Bug: webrtc:9231
Change-Id: Icecc319eaf6edd2c4b7b05fda984660412cb0d40
Reviewed-on: https://webrtc-review.googlesource.com/87439
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23884}
2018-07-09 12:14:50 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Taylor Brandstetter
dc99e244ca Removing deadbeef@ from OWNERS files.
Since I'm leaving Google.

Bug: None
Notry: True
Change-Id: Ibb5c3e09fce007d149200dcb6cac74be53084764
Reviewed-on: https://webrtc-review.googlesource.com/86461
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23795}
2018-07-02 00:40:38 +00:00
Yura Yaroshevich
c75b35ab40 Fixed crash when PCF is destroyed before DataChannel in ObjC
Bug: webrtc:9231
Change-Id: Ifad698b366be61d33ffca81cf4f8ca8aba2988a2
Reviewed-on: https://webrtc-review.googlesource.com/86040
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23771}
2018-06-28 12:54:22 +00:00
Yura Yaroshevich
c806c1d337 Fixed crash when PCF is destroyed before MediaStream in ObjC
Bug: webrtc:9231
Change-Id: I04e76172dd0d5ee5e9040e773e63fd4df0c797ce
Reviewed-on: https://webrtc-review.googlesource.com/84580
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23699}
2018-06-21 11:12:30 +00:00
Danil Chapovalov
196100efa6 Replace rtc::Optional with absl::optional
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script passing top level directories except rtc_base and api

find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \
-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: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71
Reviewed-on: https://webrtc-review.googlesource.com/84584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23697}
2018-06-21 09:32:56 +00:00
Jiawei Ou
ae810c10b4 Create a peer connection factory builder
Similar to the builder on android: https://cs.chromium.org/chromium/src/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java?rcl=b90e63c620877712e45ee320cfa25cb825bf5373&l=134

1. A builder will allow us to choose what module factories to provide and use default for the others.
2. A helper category is added to provide helpers functions for creating common builders.

Bug: None
Change-Id: I5889bdd7dc2a2aeded62ef5f2c2381edd07089b3
Reviewed-on: https://webrtc-review.googlesource.com/83280
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23696}
2018-06-21 09:22:50 +00:00
Yura Yaroshevich
5297bd21b1 Fixed crash when PCF is destroyed before PC in ObjC
Bug: webrtc:9231
Change-Id: Iaf18257b8f38fa786d462bca5f860f9a7b1cc2d0
Reviewed-on: https://webrtc-review.googlesource.com/78800
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23674}
2018-06-20 06:45:17 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
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}
2018-06-19 14:00:39 +00:00
Zhi Huang
b57e169f3c Add a flag to actively reset the SRTP parameters
Add a new flag to RtcConfiguration. By setting that flag to true, the
SRTP parameters will be reset whenever the DTLS transports are reset
after every offer/answer negotiation.

The flag is added to Android and Objc wrapper as well.

This should only be used as a workaround for the linked bug, if the
application knows that the other party is affected (for instance,
using a version number).

TBR=sakal@webrtc.org, denicija@webrtc.org

Bug: chromium:835958
Change-Id: I6db025e1c69bf83e1b1908f7df4627430db9920c
Reviewed-on: https://webrtc-review.googlesource.com/83101
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23587}
2018-06-12 20:32:00 +00:00
Florent Castelli
abe301fe6c Add HeaderExtensions to RtpParameters
Bug: webrtc:7580
Change-Id: I4fcf3e8bc4975a6b2baa6f24a17c254d2bf521d9
Reviewed-on: https://webrtc-review.googlesource.com/78288
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23584}
2018-06-12 17:01:40 +00:00
Rasmus Brandt
a3e69e6c74 Add min_bitrate_bps to RTCRtpEncodingParameters.
This is an ObjC followup to https://webrtc-review.googlesource.com/c/src/+/78741.

This CL only adds the field to the API, but does not wire it up.

Bug: webrtc:9341
Change-Id: Id6b1ac681324120bc90158029da7a80bf99aa512
Reviewed-on: https://webrtc-review.googlesource.com/81182
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23524}
2018-06-07 07:26:07 +00:00
Niels Möller
2d02e085de Delete deprecated CreateAudioSource method, with constraints.
Bug: webrtc:9239
Change-Id: I5025b7fd103247e0426ceabedc1216a4f0f0ab34
Reviewed-on: https://webrtc-review.googlesource.com/76560
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23501}
2018-06-04 08:19:30 +00:00
Anders Carlsson
79ce820a13 Obj-C SDK for parsing and generating H264 ProfileLevelIds.
Expose this functionality in the Obj-C SDK to make it nicer to use for
Obj-C clients.

Bug: None
Change-Id: I5cb511af8799ac0fda15153d16f2550b848b93b2
Reviewed-on: https://webrtc-review.googlesource.com/80481
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23488}
2018-06-01 11:23:31 +00:00
Kári Tristan Helgason
ccac98861f iOS SDK 10.0 compatability.
This CL adds support targeting iOS 10 as a min version.

Bug: None
Change-Id: I353a9884eb907e97387553fd73427fd7cb0dbfc2
Reviewed-on: https://webrtc-review.googlesource.com/79921
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23461}
2018-05-31 07:28:34 +00:00
Harald Alvestrand
73771a893f Prepare to remove old OnFailure implementations
This removes usage of the old OnFailure methods on CreateSessionDescriptionObserver
and SetSessionDescriptionObserver, so that WebRTC will continue to compile
once all the default implementations are removed.

Bug: chromium:589455
Change-Id: Id67295b3ad0c30d24d79589c2041acdd507a19f3
Reviewed-on: https://webrtc-review.googlesource.com/78480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23427}
2018-05-29 10:34:14 +00:00
Florent Castelli
dacec71b16 Add Rtcp parameters for PeerConnection senders
Bug: webrtc:7580
Change-Id: Ibcf5e849a1f11f21fa75f6d006fecf1cd54f8552
Reviewed-on: https://webrtc-review.googlesource.com/78063
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23407}
2018-05-28 09:28:59 +00:00
Florent Castelli
b7d9d8346f Implement RtpCodecParameters::parameters
This will return all the fmtp parameters for the codecs, except for
DTMF codes that don't fit the key=value pattern.

Bug: webrtc:7112
Change-Id: I06a203ff64df2c3bc9bc2082cd0f374718b23510
Reviewed-on: https://webrtc-review.googlesource.com/71801
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23250}
2018-05-15 17:12:02 +00:00
Florent Castelli
cebf50ff75 Reland "Implement RtpParameters.transaction_id for PC RtpSenderInterface"
This is a reland of 5faf36ef3c
The issue in Chrome has been fixed and this should be safe to reland.

TBR=deadbeef

Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}

Bug: webrtc:7580
Change-Id: Iabd41fb21afdf452c039d5513824ae334f8d1d3f
Reviewed-on: https://webrtc-review.googlesource.com/76980
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23247}
2018-05-15 15:51:02 +00:00
Peter Hanspers
8d95e3b211 Moving iOS Audio Device to sdk.
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

(was: https://webrtc-review.googlesource.com/c/src/+/67300)

Bug: webrtc:9120
Change-Id: I46c09900246f75ca5285aeb38f7b8b295784ffac
Reviewed-on: https://webrtc-review.googlesource.com/76741
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23238}
2018-05-15 10:33:01 +00:00
Peter Hanspers
43619a4f4a Revert "Moving iOS Audio Device to sdk."
This reverts commit 08da28dd60.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Moving iOS Audio Device to sdk.
> 
> This change forks the existing iOS audio device module and audio device
> from modules/audio_device/ into sdk/objc/Framework. It also updates
> RTCPeerConnectionFactory to use the forked implementation.
> 
> The unit tests are re-implemented as XCTests.
> 
> (was: https://webrtc-review.googlesource.com/c/src/+/67300)
> 
> Bug: webrtc:9120
> Change-Id: I07340505137b16c2dd487569ad0112f984557bba
> Reviewed-on: https://webrtc-review.googlesource.com/75125
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23208}

TBR=andersc@webrtc.org,kthelgason@webrtc.org,peterhanspers@webrtc.org

Change-Id: Ibbf8d53eaef386bc3033dc71e9490d5e48911fc9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9120
Reviewed-on: https://webrtc-review.googlesource.com/76460
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23211}
2018-05-14 10:41:20 +00:00
Peter Hanspers
08da28dd60 Moving iOS Audio Device to sdk.
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

(was: https://webrtc-review.googlesource.com/c/src/+/67300)

Bug: webrtc:9120
Change-Id: I07340505137b16c2dd487569ad0112f984557bba
Reviewed-on: https://webrtc-review.googlesource.com/75125
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23208}
2018-05-14 09:25:49 +00:00
Max Morin
826738b78c Revert "Moving iOS Audio Device to sdk."
This reverts commit a167212657.

Reason for revert: Breaks Chromium build.
Log:
https://ci.chromium.org/buildbot/chromium.webrtc.fyi/ios-device/
Writing """\
additional_target_cpus = [ "arm64" ]
goma_dir = "/b/c/goma_client"
ios_enable_code_signing = false
is_component_build = false
is_debug = false
target_cpu = "arm"
target_os = "ios"
use_goma = true
""" to /b/c/b/ios_device/src/out/Release-iphoneos/args.gn.
/b/c/b/ios_device/src/buildtools/mac/gn gen //out/Release-iphoneos --check
  -> returned 1
ERROR at //third_party/webrtc/sdk/BUILD.gn:108:9: Can't load input file.
        "../../rtc_base:checks",
        ^----------------------
Unable to load:
  /b/c/b/ios_device/src/third_party/rtc_base/BUILD.gn
I also checked in the secondary tree for:
  /b/c/b/ios_device/src/build/secondary/third_party/rtc_base/BUILD.gn

Original change's description:
> Moving iOS Audio Device to sdk.
> 
> This change forks the existing iOS audio device module and audio device
> from modules/audio_device/ into sdk/objc/Framework. It also updates
> RTCPeerConnectionFactory to use the forked implementation.
> 
> The unit tests are re-implemented as XCTests.
> 
> Bug: webrtc:9120
> Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
> Reviewed-on: https://webrtc-review.googlesource.com/67300
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23163}

TBR=andersc@webrtc.org,kthelgason@webrtc.org,peterhanspers@webrtc.org

Change-Id: Iebe52e9775409a3bdd6d5e44f4f985d56b859cbe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9120
Reviewed-on: https://webrtc-review.googlesource.com/75220
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23166}
2018-05-08 11:00:37 +00:00
Peter Hanspers
a167212657 Moving iOS Audio Device to sdk.
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

Bug: webrtc:9120
Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
Reviewed-on: https://webrtc-review.googlesource.com/67300
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23163}
2018-05-08 08:46:25 +00:00
Niels Möller
c56ff11984 Delete deprecated decode:...fragmentationHeader:... objc method.
Next step after cl https://webrtc-review.googlesource.com/72442.

Bug: webrtc:6471
Change-Id: I2cbb8cef37dbb0762bf5ef57f68d690a21f341de
Reviewed-on: https://webrtc-review.googlesource.com/73820
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23143}
2018-05-07 13:27:08 +00:00
Max Morin
909338b027 Revert "Implement RtpParameters.transaction_id for PC RtpSenderInterface"
This reverts commit 5faf36ef3c.

Reason for revert: fast/peerconnection/RTCRtpSender-setParameters.html
 failing in webrtc roll, probably this CL? https://chromium-review.googlesource.com/c/chromium/src/+/1045889.

Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
> 
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
> 
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,orphis@webrtc.org

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

Bug: webrtc:7580
Change-Id: I86da108227f8fc8d235bb2e9559377c800595b8c
Reviewed-on: https://webrtc-review.googlesource.com/74740
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23134}
2018-05-07 08:02:34 +00:00
Florent Castelli
5faf36ef3c Implement RtpParameters.transaction_id for PC RtpSenderInterface
The transaction_id field should be refreshed for every getParameters()
call and checked at each setParameters() call.
This also checks that getParameters() was ever called to return a proper
error code.

Bug: webrtc:7580
Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
Reviewed-on: https://webrtc-review.googlesource.com/70820
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23120}
2018-05-04 13:07:25 +00:00
Yura Yaroshevich
cef0650781 Set name for threads created in ObjC SDK
Bug: webrtc:9216
Change-Id: I89ee671409db5c227ba1f9fd0a583be6ee4df63b
Reviewed-on: https://webrtc-review.googlesource.com/73560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23081}
2018-05-02 17:12:57 +00:00
Kári Tristan Helgason
cad94449dd Remove H264 CHP field trial code.
Bug: webrtc:8317
Change-Id: I2da3cc6578dd8ff6e88052bc33cd38cb92af46dc
Reviewed-on: https://webrtc-review.googlesource.com/73242
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23077}
2018-05-02 13:42:37 +00:00
Magnus Jedvert
8b4e92d0a5 ObjC SDK: Stop using built-in SW video codecs
This CL removes the use of default built-in SW in the ObjC layer. If a
client want to depend on the video SW codecs, they must inject them
explicitly.

Bug: webrtc:7925
Change-Id: If752e7f02109ff768dc5ec38d935203de85987c2
Reviewed-on: https://webrtc-review.googlesource.com/69800
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23073}
2018-05-02 10:15:56 +00:00
Niels Möller
c199fae89f Deprecate RTCRtpFragmentationHeader argument for objc decoders.
Bug: webrtc:6471
Change-Id: Id542360c470ed0ea13b7e963f11bcd50d52c1d43
Reviewed-on: https://webrtc-review.googlesource.com/72442
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23036}
2018-04-26 15:28:17 +00:00
Yura Yaroshevich
0f77feae6d Init max supported H.264 profile at runtime on iOS
Bug: webrtc:9134, webrtc:7992
Change-Id: Id24c570bf3296298901f61ee817a3d7c3f8c6347
Reviewed-on: https://webrtc-review.googlesource.com/71560
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23034}
2018-04-26 15:01:07 +00:00
Steve Anton
3acffc3b16 Remove SdpSemantics::kDefault
This adds confusion to the native API and is only needed for
Chromium UMA metrics, so the appropriate metrics have been moved
upstream and kDefault option removed.

Bug: chromium:811683
Change-Id: I666d7f7793765b8d6edcd99416c8b6c957766f00
Reviewed-on: https://webrtc-review.googlesource.com/59261
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22864}
2018-04-13 17:03:08 +00:00
Anders Carlsson
2a1bbc3422 ObjC: Deprecate codec settings parameter in startDecode method.
This parameter is being removed from the C++ API, remove it from the
ObjC API also. It was never used for anything by the H264 decoder.

Bug: webrtc:9107
Change-Id: I5222eac932a4e7d4129d803f8126b5e8d0b027b6
Reviewed-on: https://webrtc-review.googlesource.com/66740
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22730}
2018-04-04 12:29:30 +00:00
Taylor Brandstetter
5e55fe845e Adding flag to enable/disable use of SRTP_AES128_CM_SHA1_32 crypto suite.
This flag (added to CryptoOptions) will allow applications to opt-in to
use of this suite, before it's disabled by default later. See bug for
more details.

TBR=magjed@webrtc.org

Bug: webrtc:7670
Change-Id: I800bedd4b26d807b6b7ac66b505d419c3323e454
Reviewed-on: https://webrtc-review.googlesource.com/64390
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22586}
2018-03-23 19:26:55 +00:00
Anders Carlsson
7311918269 Add an example app for iOS native API.
Demonstrates how to use the iOS native API to wrap components into
C++ classes.

This CL also introduces a native API wrapper for the capturer.

The C++ code is forked from the corresponding CL for Android at
https://webrtc-review.googlesource.com/c/src/+/60540

Bug: webrtc:8832
Change-Id: I12d9f30e701c0222628e329218f6d5bfca26e6e0
Reviewed-on: https://webrtc-review.googlesource.com/61422
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22484}
2018-03-19 09:31:06 +00:00