RTCIceCandidate.nativeCandidate returns a unique_ptr that
can be null. As with the previous CL, this is used without checking
whether it is null or not, so it should be fixed.
Bug: None
Change-Id: I70a84f7a2a9a9d47b8cefa198204f9b6d63a7c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227620
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34649}
There are two problems with setLocalDescription / setRemoteDescription
in ObjC SDK.
First, RTCSessionDescription.nativeDescription returns a raw
nullableSessionDescriptionInterface pointer, where sLD/sRD are calling
Clone() method unconditionally, so it might crash.
Second, unnecessary sLD/sRD calls Clone() of the raw pointer and
does not delete it, so this pointer will leak.
To solve these problems, I changed the return type of nativeDescription to
std::unique_ptr and removed the call to Clone() method.
Bug: webrtc:13022, webrtc:13035
Change-Id: Icbb87dda62d3a11af47ec74621cf64b8a6c05228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227380
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/master@{#34647}
This is useful when building the .framework which doesn't need to
export C++ symbols.
Bug: webrtc:12408
Change-Id: Ied775811a72a06b9ad678c9fb549bca286dd7f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227089
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34613}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
With this change, RTCVideoEncoder can specify:
- requested_resolution_alignment,
- apply_alignment_to_all_simulcast_layers
in the same way scaling_settings is specified.
Change-Id: I3de79a2eabaae581d6a9f2ef3e39496b9545a4f5
Bug: webrtc:12829
Change-Id: I3de79a2eabaae581d6a9f2ef3e39496b9545a4f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220933
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Abby Yeh <abbyyeh@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34196}
Deprecate CreateDataChannel, and make it a simple wrapper function.
Bug: webrtc:12796
Change-Id: I053d75a264596ba87ca734a29df9241de93a80c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219784
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34130}
This property doesn't have a getter and it is not required anymore.
Bug: None
Change-Id: Ie3f057cd6928d7fdef4e7971476fb1257900ccc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215261
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34125}
Applications should use CreatePeerConnectionOrError instead.
Moved fallback implementations of CreatePeerConnection into the
api/peer_connection_interface.h file, so that we do not have to
declare these methods in the proxy.
Bug: webrtc:12238
Change-Id: I70c56336641c2a108b68446ae41f43409277a584
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33964}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
Before these changes default initialized iOS wrappers
around various RTP*Parameters types had their own
default values of nonnull values, which did not always
matched default values from native code, which then causes
override of default native values, if library user didn't
specified it's own initialization.
After these changes default initialization of iOS wrappers
uses default property values from default initialized
native types.
Bug: None
Change-Id: Ie21a7dc38ddc3862aca8ec424859c776c67b1388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215220
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33731}
Removes use of AsyncInvoker, replaced with PendingTaskSafetyFlag. The
latter is extended to support creation on a different thread than
where it will be used, and to support stop and restart.
Bug: webrtc:12339
Change-Id: I28b6e09b1542f50037e842ef5fe3a47d15704b46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211002
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33432}
Some of the PCF and PC methods are actually return nil, but was by
default annotated as nonnull via NS_ASSUME_NONNULL_BEGIN.
Bug: None
No-Presubmit: True
Change-Id: Ib8b9263452a61241c9e7a16c1807d87bd597c093
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209180
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33384}
It is now possible to use AV1 encoder and decoder on iOS and test
them in apps like AppRTCMobile.
Bug: None
Change-Id: Ifae221020e5abf3809010676862eecd9ffeec5e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208400
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33378}
Some locations in the WebRTC codebase RTC_LOG the value of the
__FUNCTION__ macro which probably is useful in debug mode. Moving
these instances to RTC_DLOG saves ~10 KiB on arm64.
Bug: webrtc:11986
Change-Id: I5d81cc459d2850657a712b9aed80c187edf49a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203981
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33086}
In AudioDeviceIOS, when we call Stop() on the VoiceProcessingAudioUnit,
we do not always detach the I/O thread checker in preparation for a new
start. This means that if we start up the VoiceProcessingAudioUnit - and
subsequently a new AURemoteIO thread to deal with I/O operations - the
DCHECK in OnDeliverRecordedData and OnGetPlayoutData will fail. Note
that we want to detach the I/O thread checker regardless of whether
Stop() returns with a success status or not. The success status is
dictated by the iOS function AudioOutputUnitStop. The documentation of
this function does not guarantee that the audio unit will not stop in
the case the function returns with an error code. That is to say, it is
possible the audio unit stops even if the function Stop() returns false.
Therefore, it is safer to prepare the I/O thread checker for a new start
in either case.
Change-Id: Iee50a2457959aff2e6089e9a664c649dc4dbbbd6
Bug: webrtc:12382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202945
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33063}
While RTC_EXPORT is aware of component builds (selecting "default"
visibility only when WebRTC is built as a shared library),
RTC_OBJC_EXPORT (which predates RTC_EXPORT) was always marking symbols
as "default" visible.
This CL fixes the problem but on the other hand it will require
standalone builds of the WebRTC.framework to set the GN argument
`rtc_enable_symbol_export` to true.
No-Presubmit: True
Bug: chromium:1159620
Change-Id: I4a16f9bd3c1564140a5a30f03b3e77caed1df591
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198082
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32856}
This patch adds support for setting the TURN_LOGGING_ID
in RTCConfig using the ios SDK.
TURN_LOGGING_ID was added to webrtc in
https://webrtc-review.googlesource.com/c/src/+/149829
The intended usage of this attribute is to correlate client and
backend logs.
This change was tested out with duo via wireshark.
Bug: webrtc:10897
Change-Id: Iedbefdc6392c4df203aca08cf750028b450a11ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191340
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Brad Pugh <bradpugh@google.com>
Cr-Commit-Position: refs/heads/master@{#32626}
