Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
setExif: would create a CFDictionary using NULL for keyCallBacks and
valueCallBacks. This has the effect of comparing the keys of the
dictionary by pointer instead of by value. With ld64, this works
because it always dedupes identical constant CFSTR("foo") literal,
but lld currently doesn't do this.
Using kCFTypeDictionaryKeyCallBacks and kCFTypeDictionaryValueCallBacks
fixes the problem with lld and is "more correct" in general: Now the
dictionary would work with computed CFStrings too, it shows up better
in CFShow() output, etc.
While here, also fix a memory leak in setExif:.
Bug: chromium:1251763
Change-Id: I43c96d2189a4a77fe3bd0dfb3e33623925b0f900
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35067}
With this change, we catch audio unit start errors and pipe them to the
audio session. The audio session notifies its delegate, which can then
take appropriate action based on the error code.
The signal follows the same path as the playout glitch detection.
Bug: webrtc:13119
Change-Id: I8c9f9d2a1e3457447d0ce61ad197f7e1c6392837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230240
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34862}
The simulcast_encoder_adapter expects codecs that specify
supports_native_handle to perform resampling/scaling (through
GetEncoderInfo).
This change adds a method to the RTCVideoEncoder to let objc encoders
specify this rather than relying on the default.
Bug: webrtc:13044
Change-Id: I2efcbd42aa4f2048285f451c7b691fdeca111e62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227641
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34683}
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}
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}
