This change adds bindings so that native FrameEncryptor and native FrameDecryptor
objects can be set on the objective C RTCRtpSender and RTCRtpReceiver objects.
Bug: webrtc:9681
Change-Id: Iec4006ea020d6ab6adcc0ad068dcd8fb2738063d
Reviewed-on: https://webrtc-review.googlesource.com/c/103020
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24936}
This CL enables tests that were previously disabled and fixes the issues
that made them flaky.
Bug: webrtc:6889, webrtc:7888
Change-Id: I914b59200d7bf2973e8993b04de867cc3355b8a8
Reviewed-on: https://webrtc-review.googlesource.com/98381
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24930}
Symbols under sdk/ are now exported using RTC_OBJC_EXPORT, while
RTC_EXPORT is used for C++ symbols.
Bug: webrtc:9419
Change-Id: Icdf7ee0e7b3faf4d7fec33e9b33a3b13260f45b7
Reviewed-on: https://webrtc-review.googlesource.com/102461
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24886}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
A new version of RTC_EXPORT will be introduced by [1] and it will be
used by WebRTC native code.
This CL renames the current RTC_EXPORT to RTC_OBJC_EXPORT in order
to avoid to mix them. It has been decided to avoid to unify them because
RTC_OBJC_EXPORT always marks symbols with default visibility, while
RTC_EXPORT will do it only when COMPONENT_BUILD is defined.
[1] - https://webrtc-review.googlesource.com/c/src/+/97960 is
Bug: webrtc:9419
Change-Id: I56a3fc6601c72d3ad6a58f9961a00e3761dfb5da
Reviewed-on: https://webrtc-review.googlesource.com/100521
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24754}
This CL removes a set of DCHECKs in AudioDeviceBuffer (ADB) where the goal has been
to ensure that some methods are called on one and the same native I/O thread.
The implementation of the ADB is platform independent but the underlying (driving)
audio components differ between platforms. This combination has shown to generate complex
corner cases such as:
- OS dependent I/O-thread(s) changes while audio is active
- OS dependent audio device changes and it leads to restart of native I/O threads
- Start/Stop of audio has different timing depending on platform and possibly also usage of
JNI and/or emulators.
To summarize: the gain of maintaining the current strict thread checking (in Debug mode)
is not worth all the efforts trying to resolve complex dynamic cases where the native
I/O threads changes ID.
TBR=glaznev
Bug: b/115385789
Change-Id: I681c89adec497a18b97d2a40421c04ea218fd919
Reviewed-on: https://webrtc-review.googlesource.com/100200
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24723}
This reverts commit 4f085434b9.
Reason for revert: breaks downstream projects.
Original change's description:
> Add SSLConfig object to IceServer.
>
> This is a rollforward of https://webrtc-review.googlesource.com/c/src/+/96020,
> with the addition of setting the old tlsCertPolicy, tlsAlpnProtocols and
> tlsEllipticCurves in the RTCIceServer initializer, for backwards compatibility.
>
> Bug: webrtc:9662
> Change-Id: I28706ed4ff5abe3f7f913f105779f0e5412aeac5
> Reviewed-on: https://webrtc-review.googlesource.com/98762
> Commit-Queue: Diogo Real <diogor@google.com>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24696}
TBR=steveanton@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,kthelgason@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org,diogor@google.com
Change-Id: I1cb64b63fec688b4ac90c2fa368eaf0bc11046af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9662
Reviewed-on: https://webrtc-review.googlesource.com/99880
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24701}
This is a rollforward of https://webrtc-review.googlesource.com/c/src/+/96020,
with the addition of setting the old tlsCertPolicy, tlsAlpnProtocols and
tlsEllipticCurves in the RTCIceServer initializer, for backwards compatibility.
Bug: webrtc:9662
Change-Id: I28706ed4ff5abe3f7f913f105779f0e5412aeac5
Reviewed-on: https://webrtc-review.googlesource.com/98762
Commit-Queue: Diogo Real <diogor@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24696}
This component was added to work around an issue in iOS 8, which is
no longer supported by WebRTC. It's removal is made more urgent by
the fact that it prevents WebRTC being used by iOS extensions.
Bug: webrtc:9335
Change-Id: I2a3327534fe6d5014c34a9e908096d825e8149e3
Reviewed-on: https://webrtc-review.googlesource.com/87822
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24688}
With "aggressive" preset the default bundlePolicy is set to "maxBundle" when it shoud be "balanced" according to spec.
Bug: webrtc:9458
Change-Id: Ifbdd76be3a6d9968574cba857f178d5f859dcb87
Reviewed-on: https://webrtc-review.googlesource.com/88567
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24650}
This is a reland of 711a31aead
Changes since original landing:
Rename methods only used by tests, mainly via FakeClock,
MessageQueueManager::ProcessAllMessageQueues
--> ProcessAllMessageQueuesForTesting
MessageQueue::IsProcessingMessages
--> IsProcessingMessagesForTesting
Fix the handling of null rtc::Thread::Current() in
ProcessAllMessageQueuesInternal().
Add override Thread::IsProcessingMessagesForTesting() to return false
for the wrapped main thread, unless it's also the current thread. In
tests, the main thread is typically not processing any messages,
but blocked in an Event::Wait().
Original change's description:
> Explicitly wrap main thread in test_main.cc.
>
> Bug: webrtc:9714
> Change-Id: I6ee234f9a0b88b3656a683f2455c3e4b2acf0d54
> Reviewed-on: https://webrtc-review.googlesource.com/97683
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24560}
Bug: webrtc:9714
Change-Id: I6f022d46aaf1e28f86f09f2d68c1803b69770126
Reviewed-on: https://webrtc-review.googlesource.com/98060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24596}
Similarly to how it is done for iOS.
Bug: webrtc:9627
Change-Id: I7e4e3495d28a0a098531250bfdcf93d272e27b9d
Reviewed-on: https://webrtc-review.googlesource.com/98162
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24594}
This method is allowed to return nil but was not annotated as such.
Bug: webrtc:8560
Change-Id: If54aa94d6ff83b7bdb87b526244616e2627a8999
Reviewed-on: https://webrtc-review.googlesource.com/97380
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24568}
This reverts commit 7f1ffcccce.
Reason for revert: Speculative revert
Original change's description:
> Add SSLConfig object to IceServer.
>
> This is being added to allow greater configurability to TLS connections.
> tlsAlpnProtocols, tlsEllipticCurves and tlsCertPolicy will be removed from IceServer in a
> follow-up CL.
>
> Bug: webrtc:9662
> Change-Id: I33cb804b02c26c662ed2a28c76f9a9dc2df40f36
> Reviewed-on: https://webrtc-review.googlesource.com/96020
> Commit-Queue: Diogo Real <diogor@google.com>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24559}
TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,juberti@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org,diogor@google.com
Change-Id: Iae9fc68b77f743876bda36fc2a04f6d791aae8e2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9662
Reviewed-on: https://webrtc-review.googlesource.com/98000
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24563}
This is being added to allow greater configurability to TLS connections.
tlsAlpnProtocols, tlsEllipticCurves and tlsCertPolicy will be removed from IceServer in a
follow-up CL.
Bug: webrtc:9662
Change-Id: I33cb804b02c26c662ed2a28c76f9a9dc2df40f36
Reviewed-on: https://webrtc-review.googlesource.com/96020
Commit-Queue: Diogo Real <diogor@google.com>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24559}
These headers was lost in the cleanup CL for the Obj-C directories. This
puts them back in the framework headers.
Note that since the protocol and interface was split into two different
headers, and all public framework headers are put into a flat directory
structure, I had to rename the implementation files so they would not collide
in the framework header directory.
Bug: webrtc:9701
Change-Id: I42d4c1e02bdfa4e114575f527c4c42a19be8fb52
Reviewed-on: https://webrtc-review.googlesource.com/97330
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24539}
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}
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}
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}
This code never executes as we always get passed a nil codecSpecificInfo.
Bug: webrtc:9580
Change-Id: I5c5311c20877494978df45d409a53ad5b0e86a9b
Reviewed-on: https://webrtc-review.googlesource.com/92083
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24177}
RTCNativeAudioSessionDelegateAdapter has a raw pointer to AudioDeviceIOS,
and receives callbacks from RTCAudioSession and forwards them to AudioDeviceIOS.
During teardown of these components the situation can occur that the dtor for
AudioDeviceIOS has been called but the ObjC runtime has not yet dealloced
RTCNativeAudioSessionDelegateAdapter, so it's still receiving callbacks while
the pointer it keeps to AudioDeviceIOS has been invalidated.
This occasionally triggers a crash when WebRTC is shutting down.
The fix in this CL is to make sure to deregister the adapter from RTCAudioSession
_before_ the dtor for AudioDeviceIOS returns.
Bug: webrtc:9523
Change-Id: Ica85420d76efc63940472bc43e3ec71d16036ccf
Reviewed-on: https://webrtc-review.googlesource.com/90245
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24173}
After this change, it is only recreated when needed.
This change also clarifies the relation between the compression
session and the pixel buffer pool, and handles invalid sessions
explicitly.
Change-Id: Iae4aa02b60b0d5c153db3ae2d4cd2a0cfa05757b
Bug: webrtc:9562
Reviewed-on: https://webrtc-review.googlesource.com/90403
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24161}
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}
BGRA RTCCVPixelBuffers were cropped and scaled incorrectly. Libyuv’s
`ARGBScale` method is used in RTCCVPixelBuffer to scale and crop the
pixel buffer. To crop by `cropX` and `cropY` pixels, pointer
arithmetic is used to offset the src pointer of the original pixel
buffer bytes. There is a bug in how this offset is calculated.
The offset is done by `src += srcStride * _cropY + _cropX`. Libyuv
expects that the src pointer will point to the start of a new pixel.
However, if _cropX is a not a multiple of 4 (4 bytes for BGRA), the src
pointer will point to a byte in the middle of a pixel and thus libyuv
will incorrectly treat the data as the start of pixel (incorrectly
treating the first byte as red when it is actually green, etc...). To
fix this, the src pointer needs to be offset to always point to the
start of a new pixel.
Before this change:
Original Test Gradient image with a cropX of 2:
https://i.imgur.com/gSIgwGV.jpg
Scaled image (notice the colors are incorrect):
https://i.imgur.com/oPxbTEK.jpg
After this change:
Scaled image (notice the colors are correct):
https://i.imgur.com/dqBsmsH.jpg
A new unit test which tests scaling with cropX and cropY values has been
added. The test fails without this change and now passes with the
correct src pointer offsetting.
Bug: webrtc:9555
Change-Id: I87cbd7b91bc139d51fb4e11cc50ccb014cfa8051
Reviewed-on: https://webrtc-review.googlesource.com/89220
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24076}
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}
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}
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}
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}
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}
This provides an environment for testing out using WebRTC from an iOS
extension. It implements a ReplayKit broadcast extension for live
streaming games and screensharing.
The extension is only supported on iOS 11+ and is guarded by a build
flag.
Bug: webrtc:9335
Change-Id: Id218d6c73ef7599f5953c5a1e0e62e5d0dc4f10b
Reviewed-on: https://webrtc-review.googlesource.com/80000
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23504}
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}
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}
This reverts commit fc4a9c9333.
Reason for revert: Remote video is not showing in a video call.
Original change's description:
> Metal rendering should account for cropping.
>
> Also:
> - added a rotation override to allow ignoring frame rotation
> - fixed a couple of minor issues
> - made it possible to run the MTKView without the DisplayLink
>
> Bug: webrtc:9301
> Change-Id: Ia83c152d9b6d45d56ceb80d287b5d3eacfaebddd
> Reviewed-on: https://webrtc-review.googlesource.com/78282
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23452}
TBR=andersc@webrtc.org,kthelgason@webrtc.org,peterhanspers@webrtc.org
Change-Id: Iddf7793368531d2d7268c1ec138bb3a9874a4ab7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9301
Reviewed-on: https://webrtc-review.googlesource.com/80020
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23455}
Also:
- added a rotation override to allow ignoring frame rotation
- fixed a couple of minor issues
- made it possible to run the MTKView without the DisplayLink
Bug: webrtc:9301
Change-Id: Ia83c152d9b6d45d56ceb80d287b5d3eacfaebddd
Reviewed-on: https://webrtc-review.googlesource.com/78282
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23452}
Made some minor changes to resolve the issue. Only affects Debug builds.
NOTRY=TRUE
Bug: webrtc:9310
Change-Id: Ieeeb57d24b559282b2eefd4d8785f7cfe4f44e40
Reviewed-on: https://webrtc-review.googlesource.com/79624
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23434}
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}
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}
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}
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}
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}
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
Add new team members as owners of sdk/objc.
Bug: None
Change-Id: Id8c40fb018da2ab634bc1117afda555275a8b0f8
Reviewed-on: https://webrtc-review.googlesource.com/74002
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23169}
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}
Intend to delete in a later cl.
Bug: webrtc:6471
Change-Id: Icf0fcd40e0d3287dc59b684fae6552b40b47204a
Reviewed-on: https://webrtc-review.googlesource.com/39511
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23162}
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}
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}
The new RTCMTLRGBRenderer dynamically handles both the kCVPixelFormatType_32BGRA
and the kCVPixelFormatType_32ARGB pixel formats.
Change-Id: I935532f762eff74c4b84fea9b855191f4c321fb7
Bug: webrtc:9200
Reviewed-on: https://webrtc-review.googlesource.com/72482
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23100}
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}
We want to allow the application to set it's own content mode.
Bug: b/73147161
Change-Id: I60fab454353a4c39731e49b7b6066e51d8e9a94d
Reviewed-on: https://webrtc-review.googlesource.com/70501
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22962}
There currently are no Objective-C API's to create a buffer with that data.
This change allows us to create a buffer with yuv data.
Bug: webrtc:9167
Change-Id: I00f1b91b04bbaa013a88137d0f54bef44287c5aa
Reviewed-on: https://webrtc-review.googlesource.com/70563
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Commit-Queue: Peter Slatala <psla@google.com>
Cr-Commit-Position: refs/heads/master@{#22945}
The MTL renderer should also have a way to notify it's delegate
that it's content size changed.
The plan is to introduce this new protocol, move existing clients over
to implementing it in favour of RTCEAGLVideoViewDelegate, and then finally
removing the old protocol.
Bug: b/73147161
Change-Id: I908d7b2667e44e02a58066d701a48efec0e98d14
Reviewed-on: https://webrtc-review.googlesource.com/70243
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22944}
We can then also drop the system_wrappers dependency from the common_video
build target.
Bug: webrtc:6733
Change-Id: I501113d100322d1ebc51b2286970697a24b70a43
Reviewed-on: https://webrtc-review.googlesource.com/70381
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22934}
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}
This reverts commit 4feb2044db.
Reason for revert: Landscape video was not showing as aspect fit as before. .
Original change's description:
> Fix rendering on an iPhone X's tall screen.
>
> Bug: webrtc:8884
> Change-Id: I850e4ea1919837e15a78c90968a4879a1ccbd22c
> Reviewed-on: https://webrtc-review.googlesource.com/52761
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22011}
TBR=magjed@webrtc.org,kthelgason@webrtc.org,jtteh@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8884
Change-Id: I17bcdaf945d74540538162934cd3265240cc9302
Reviewed-on: https://webrtc-review.googlesource.com/68841
Commit-Queue: JT Teh <jtteh@webrtc.org>
Reviewed-by: JT Teh <jtteh@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22828}
Currently there are several checks against _lastDrawnFrame in RTCEAGLVideoView.mm but this variable is not assigned anywhere. Seems like it was missed in 13941912b1 during work on injecting custom shaders.
Bug: webrtc:9133
Change-Id: Ie979a63de343e7253e4b4e70e3b98ffb0880af04
Reviewed-on: https://webrtc-review.googlesource.com/68720
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22819}
This reverts commit a8f13ccad4.
Reason for revert: It's causing no video to be shown after the 1st call.
Original change's description:
> Improve thread-safety of MTL Renderer.
>
> Bug: b/77579859
> Change-Id: I427d0f41593155dc5cbf98a09d7ec826497b803c
> Reviewed-on: https://webrtc-review.googlesource.com/67040
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22795}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
Change-Id: Ia8f33995e087178f1c3be7753f70be8ba18447f8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/77579859
Reviewed-on: https://webrtc-review.googlesource.com/68860
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22800}
Implement debugQuickLookObject for RTCI420Buffers and RTCCVPixelBuffers.
Also draw gradients consistently regardless of endianness in the unit
tests for RTCCVPixelBuffers and ObjCVideoTrackSource.
Bug: webrtc:9007
Change-Id: Ia5a3d0905a763efc190165471983061fc07551f2
Reviewed-on: https://webrtc-review.googlesource.com/64987
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22746}
The settings struct specifies bitrate in kbps, but we are
treating it as bps.
Bug: webrtc:9113
Change-Id: I27745da93aaec68041ea4283b45eccb35d820793
Reviewed-on: https://webrtc-review.googlesource.com/66960
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22743}
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}
This is a reland of 4ea50c2b42
Original change's description:
> Add unit tests for RTCCVPixelBuffer and ObjCVideoTrackSource.
>
> This CL also fixes a couple of bugs found in the toI420 method for
> RTCCVPixelBuffers backed by RGB CVPixelBuffers.
>
> Bug: webrtc:9007
> Change-Id: I19ab8177f4b124a503cfda9f0166bd960f668982
> Reviewed-on: https://webrtc-review.googlesource.com/64940
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22656}
Bug: webrtc:9007
Change-Id: I2a787c64f8d23ffc4ef2419fc258d965f8a9480b
Reviewed-on: https://webrtc-review.googlesource.com/66341
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22706}
This reverts commit 4ea50c2b42.
Reason for revert: This change is causing crashes in video calls.
RTCCVPixelBuffer.mm - line 120
Compare is asserting as 420f is not 420v
Original change's description:
> Add unit tests for RTCCVPixelBuffer and ObjCVideoTrackSource.
>
> This CL also fixes a couple of bugs found in the toI420 method for
> RTCCVPixelBuffers backed by RGB CVPixelBuffers.
>
> Bug: webrtc:9007
> Change-Id: I19ab8177f4b124a503cfda9f0166bd960f668982
> Reviewed-on: https://webrtc-review.googlesource.com/64940
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22656}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9007
Change-Id: I500514ce05dd0555f8c4a05010ad52bd67c2fed3
Reviewed-on: https://webrtc-review.googlesource.com/65561
Commit-Queue: JT Teh <jtteh@webrtc.org>
Reviewed-by: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22686}
IceConfig contains a set of parameters that affect the behavior of ICE.
Inconsistent or conflicting parameters lead to erroneous or
unpredicatble behavior in the network stack. Sanity checks are now added
to validate IceConfig.
TBR=magjed@webrtc.org
Bug: webrtc:8993
Change-Id: I708bc3f1ef970872754a82a47a509bda15061ca6
Reviewed-on: https://webrtc-review.googlesource.com/60847
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22664}
This CL also fixes a couple of bugs found in the toI420 method for
RTCCVPixelBuffers backed by RGB CVPixelBuffers.
Bug: webrtc:9007
Change-Id: I19ab8177f4b124a503cfda9f0166bd960f668982
Reviewed-on: https://webrtc-review.googlesource.com/64940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22656}
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}
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}
Add native api conversions for video frames and video renderer. This
also requires some changes to sdk/BUILD to avoid cyclic dependencies.
Bug: webrtc:8832
Change-Id: Ibf21e63bdcae195dcb61d63f9262e6a8dc4fa790
Reviewed-on: https://webrtc-review.googlesource.com/57142
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22340}
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.
Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
This removes the routing for the deprecated audio control setting
Change-Id: Id83ff548625279d5b34c9e3cadc097c25a00ef05
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/58900
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22279}
This CL adds wrappers for the following PeerConnection native
APIs to the Objective C API:
- SdpSemantics enum added to the RTCConfiguration
- RTCRtpTransceiver
- RTCPeerConnection.addTrack
- RTCPeerConnection.removeTrack
- RTCPeerConnection.addTransceiver
- RTCPeerConnection.transceivers
Bug: webrtc:8870
Change-Id: I9449df9742a59e90894712dc7749ca30b569d94b
Reviewed-on: https://webrtc-review.googlesource.com/54780
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22214}
Summary:
The implementation of H264AnnexBBufferHasVideoFormatDescription was
assuming that the SPS NALU is either the first NALU in the stream, or
the second one, in case an AUD NALU is present in the first location.
This change removes this assumption and instead searches for the SPS
NALU, failing only if we can't find one.
In addition, it cleans up some binary buffer manipulation code, using the
the parsed NALU indices we already have in AnnexBBufferReader instead.
Test Plan: Unit tests
Change-Id: Id9715aa1d751f0ba1a1992def2b690607896df56
bug: webrtc:8922
Change-Id: Id9715aa1d751f0ba1a1992def2b690607896df56
Reviewed-on: https://webrtc-review.googlesource.com/49982
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22205}
Overriding implementations of VideoEncoder::GetScalingSettings that
want to enable quality scaling must now provide the thresholds.
Bug: webrtc:8830
Change-Id: I75c47cb56ac1b9cf77401684980b3167e485f51c
Reviewed-on: https://webrtc-review.googlesource.com/46622
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22172}
After https://webrtc-review.googlesource.com/c/src/+/49060 changed the
gn check config for sdk/.
Add nogncheck for some conditionally imported headers.
Bug: webrtc:7925
Change-Id: I57499e990332636991563c6f550a7c9154e7c2ee
Reviewed-on: https://webrtc-review.googlesource.com/54820
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22083}
It turns out that some headers were not owned by any targets.
These were:
RTCVideoCodec.h
RTCVideoCodecFactory.h
RTCVideoCodecH264.h
RTCVideoEncoderVP8.h
RTCVideoDecoderVP8.h
RTCVideoEncoderVP9.h
RTCVideoDecoderVP9.h
And some were owned by multiple targets, namely:
RTCPeerConnectionFactory+Native.h
RTCPeerConnectionFactory+Private.h
RTCVideoFrameBuffer.h
These have all been moved to their appropriate homes.
This CL also fixes a lot of cyclic interdependencies in the iOS sdk build files.
Bug: webrtc:8855
Change-Id: I1b7ddb6c2a93868d1510ccf0a64bd3dd169ec3e7
Reviewed-on: https://webrtc-review.googlesource.com/49060
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22052}
The reference back to the decoder class in the decode callback
was null. Due to the amazing properties of ObjC this led to the
setError call to silently fail.
Bug: webrtc:8600
Change-Id: I3f70fbe4c9d533c8612d0bc7bc40813252e492fd
Reviewed-on: https://webrtc-review.googlesource.com/52460
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22021}
This reverts commit 6780c51b23.
Reason for revert:
More details in crbug.com/810292
Original change's description:
> Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
>
> A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
> from native apps if really necessary.
>
> R=deadbeef@webrtc.org
>
> Bug: webrtc:7670
> Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
> Reviewed-on: https://webrtc-review.googlesource.com/41420
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21952}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,jbauch@webrtc.org
Change-Id: I643dbe023eca526f2cda4d97df045f2533741dd4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7670
Reviewed-on: https://webrtc-review.googlesource.com/49880
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21961}
A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
from native apps if really necessary.
R=deadbeef@webrtc.org
Bug: webrtc:7670
Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
Reviewed-on: https://webrtc-review.googlesource.com/41420
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21952}
The bug 8432 is caused by trying to connect through a
"link-local" interface (IP address 169.254.0.x/16),
which is listed among the iPhone network interfaces.
The bug is not happening if the link-local network interfaces
are skipped in the ICE candidate gethering process.
To control this behaviour an option - disable_link_local_networks -
is added inside the RTCConfiguration.
It is used to set the new BasicPortAllocatorSession flag -
PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS.
The port allocator flag is added if the configuration option is set.
IPIsLinkLocal IPAddress function and its friends (IPIsLoopback, IPIsPrivate)
are refactored to work on both IPv4 and IPv6.
Unit test IPIsLinkLocal.
Bonus: fix a bug in IPIsLinkLocalV6:
take into account just 10 network mask bits instead of 16.
Bug: webrtc:8432
Change-Id: Ibe8f677a36098057b7fcad5c798380727b23359b
Reviewed-on: https://webrtc-review.googlesource.com/36380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21922}
This target is deprecated and downstream projects have been updated.
This CL replaces https://webrtc-review.googlesource.com/c/src/+/46521
Bug: webrtc:8470
Change-Id: Icf4696c946fd0a1aeeb687c4960586ba0cc52dc0
Reviewed-on: https://webrtc-review.googlesource.com/48362
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21908}
These files were copied to Native/src but were kept around for
downstream projects that included them from their old locations.
Downstream projects have been updated so these can now be removed.
Bug: webrtc:8832
Change-Id: Ic28dc13e4b5bfced4b97ee872068683785d04bb3
Reviewed-on: https://webrtc-review.googlesource.com/47860
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21892}
This can be used to wrap Objective-C components in C++ classes, so users
can use the WebRTC C++ API directly together with the iOS specific
components provided by our SDK.
Bug: webrtc:8832
Change-Id: I6d34f7ec62d51df8d3a5340a2e17d30ae73e13e8
Reviewed-on: https://webrtc-review.googlesource.com/46162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21850}
Building with the newly published cocoapod generated a few warnings,
which looked a bit bad.
Bug: webrtc:8831
Change-Id: I70c06930603b328e4d11c599a5b5dd77b45150c6
Reviewed-on: https://webrtc-review.googlesource.com/46163
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21846}
This CL adds a GN build flag to include builtin software codecs
(enabled by default).
When setting the flag to false, libvpx can also be excluded. The
benefit is that the resulting binary is smaller.
Replaces https://webrtc-review.googlesource.com/c/src/+/29203
Bug: webrtc:7925
Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10
Reviewed-on: https://webrtc-review.googlesource.com/36340
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21818}
This makes it possible to only inject 1 or 0 video codec factories when
consuming the API using the PeerConnectionFactory+Native header.
Bug: webrtc:7925
Change-Id: I671d8dcdbdf2198a31f3890ff6b416441bd32d48
Reviewed-on: https://webrtc-review.googlesource.com/42661
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21715}
This allows a user to only injecting the decoder or encoder factory.
This behavior also matches how it is implemented for Android.
Bug: webrtc:7925
Change-Id: I3dfca6ea2eaeea437b5b78da2373bd6f7cedc8fa
Reviewed-on: https://webrtc-review.googlesource.com/40860
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21695}
Instead of keeping the umbrella header in sync manually and needing
ifdefs to make it include the correct headers depending on platform,
generate it based on the headers we include in the framework target.
Can also be used to only include internal software codec headers when
compiling with support for them.
Bug: webrtc:7925
Change-Id: I63f97af1efc8710cfd62d527fcb343fed05daae2
Reviewed-on: https://webrtc-review.googlesource.com/38702
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21613}
