This moves them from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I6dc34fe662f5d87b3b5288d33055345bc6bf91db
Reviewed-on: https://webrtc-review.googlesource.com/21164
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22567}
Add support for creating java PeerConnectionFactory from native one
by adding:
1. Constructor from native pointer in java PeerConnectionFactory
2. Method NativeToJavaPeerConnectionFactory in
sdk/android/native/api/peerconnection/peerconnectionfactory.h that
provides ability to convert native factory to java one.
Bug: webrtc:8946
Change-Id: Ibe8b019bd0d45849e2b16d74663d054784526746
Reviewed-on: https://webrtc-review.googlesource.com/62344
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22564}
This CL performs some simplifications and cleanups of the moved audio code.
* All JNI interaction now goes from the C++ audio manager calling into
the Java audio manager. The calls back from the Java code to the C++
audio manager are removed (this was related to caching audio parameters).
It's simpler this way because the Java code is now unaware of the C++
layer and it will be easier to make this into a Java interface.
* A bunch of state was removed that was related to caching the audio parameters.
* Some unused functions from audio manager was removed.
* The Java audio manager no longer depends on ContextUtils, and the context has
to be passed in externally instead. This is done because we want to get rid of
ContextUtils eventually.
* The selection of what AudioDeviceModule to create (AAudio, OpenSLES
input/output is now exposed in the interface. The reason is that client should
decide and create what they need explicitly instead of setting blacklists
in static global WebRTC classes. This will be more modular long term.
* Selection of what audio device module to create (OpenSLES combinations) no
longer requires instantiating a C++ AudioManager and is done with static
enumeration methods instead.
Bug: webrtc:7452
Change-Id: Iba29cf7447a1f6063abd9544d7315e10095167c8
Reviewed-on: https://webrtc-review.googlesource.com/63760
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22542}
After using JNI generation, there is no need to have a separate class
handling JNI interaction.
Bug: webrtc:7452
Change-Id: I25de6007190d826e2790cf6219a6ac861acfb6a8
Reviewed-on: https://webrtc-review.googlesource.com/63800
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22541}
Needs to be added to the array before the array is copied to the
sources and public_headers arrays.
Bug: None
Change-Id: If41fd1c882dd17e4007b62c9c7a49f196849dd12
Reviewed-on: https://webrtc-review.googlesource.com/63640
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22531}
This CL contains some follow-up fixes for
https://webrtc-review.googlesource.com/c/src/+/60541. It removes all use
of the old voiceengine implementation from AppRTCMobile.
Bug: webrtc:7452
Change-Id: Iea21a4b3be1f3cbb5062831164fffb2c8051d858
Reviewed-on: https://webrtc-review.googlesource.com/63480
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22530}
This CL only affects the forked Android audio device code. The old code
at webrtc/modules/audio_device/android/ is unaffected.
Bug: webrtc:8689, webrtc:8278
Change-Id: I696b8297baba9a0f657ea3df808f57ebf259cb06
Reviewed-on: https://webrtc-review.googlesource.com/36502
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22528}
This CL adds a stand-alone Android AudioDeviceModule in the
sdk/android folder. It's forked from modules/audio_device/android/
and then simplified for the Android case. The stand-alone Android
ADM is available both in the native_api and also under a field trial
in the Java API.
Bug: webrtc:7452
Change-Id: If6e558026bd0ccb52f56d78ac833339a5789d300
Reviewed-on: https://webrtc-review.googlesource.com/60541
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22517}
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 configurable parameters in RTCConfiguration with the default value
given by the constants CONNECTION_WRITE_CONNECT_TIME and
CONNECTION_WRITE_CONNECT_FAILURES in the ICE implementation. These two
parameters define the time period for which a candidate pair must wait
for ping response and the minimum number of connectivity checks that
the pair must send without response before its state becomes unreliable
from writable as defined in the current ICE implementation.
Bug: webrtc:8988
Change-Id: I484599b7d776489a87741ffea8926df766095da9
Reviewed-on: https://webrtc-review.googlesource.com/60704
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22411}
It is unnecessary to include the build hooks implementation because we
don't use them. It was also causing errors because the interface the
class implements is not included in the AAR.
Also removes comments about re-enabling build hooks because it has grown
into something very Chromium specific and it is unlikely that we want to
re-enable them.
Bug: webrtc:8964, webrtc:8168
Change-Id: Ia95af13e90a5511554305d2688ced820e9914beb
Reviewed-on: https://webrtc-review.googlesource.com/61302
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22386}
The app is a simple loopback demo demonstrating the usage of Android
native API. This is an initial version and I will add support for
HW codecs etc. in the future.
Bug: webrtc:8769
Change-Id: Ifb6209769dabeb8ca3185b969a1ef8afd6d84390
Reviewed-on: https://webrtc-review.googlesource.com/60540
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22385}
The connectivity check intervals for candidate pairs with strong and
weak connectivity are currently constants in the ICE implementation. A
set of suboptimal value of these constants for a given application may
result in undesirable behavior including excessive network switching
latency. This CL adds these intervals to RTCConfiguration that is
available to applications to configure, while maintaining the original
constants as their default value for compatibility with existing
applications.
Bug: webrtc:8988
Change-Id: I804b0f4cf7881be7d3c8aec2776bc9596de72482
Reviewed-on: https://webrtc-review.googlesource.com/60585
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22351}
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}
It's been unused for a while and starting to be a maintainance burden.
Bug: webrtc:8943
Change-Id: Ie49d6b06bdeb002496007725009ea194b8130f2b
Reviewed-on: https://webrtc-review.googlesource.com/60160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22339}
This file ended up in the wrong place and prevented building without
SW codecs.
Bug: webrtc:7925
Change-Id: I7909561d96051d5653821130c666ac66938e3edc
Reviewed-on: https://webrtc-review.googlesource.com/59640
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22320}
This updates AppRTC to use addTrack instead of addStream, and removes
the use of onAddStream, because we no longer have to wait for this to be
fired to set the remote track's video renderers.
Bug: webrtc:8869
Change-Id: I1ecae684a9bc4b30512e8c5d717e72b52c589831
Reviewed-on: https://webrtc-review.googlesource.com/57840
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22318}
This adds wrappers to the following native APIs:
- SdpSemantics enum added to the RTCConfiguration
- RtpTransceiver
- PeerConnection.addTrack
- PeerConnection.removeTrack
- PeerConnection.addTransceiver
- PeerConnection.getTransceivers
These APIs are used with the new Unified Plan semantics.
Bug: webrtc:8869
Change-Id: I19443f3ff7ffc91a139ad8276331f09e57cec554
Reviewed-on: https://webrtc-review.googlesource.com/57800
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22317}
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}
Updating the gn args to be more accurate and including information about
fetching the appropriate webrtc android checkout. This is so that this
README will include all necessary information for compiling the android
code for future developers.
Bug: webrtc:8869
Change-Id: I641183705370273d4a8cab044f08b2d203a26102
Reviewed-on: https://webrtc-review.googlesource.com/59060
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22260}
This change makes the class thread-safe.
Bug: b/73773043
Change-Id: I1ad13e4f15907e3dd1fef1307f9c654e53b69b22
Reviewed-on: https://webrtc-review.googlesource.com/57040
Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22238}
Allows passing in the application context to NetworkMonitor in
startMonitoring. The audio code will refactored once it is moved under
sdk/android.
Bug: webrtc:8937
Change-Id: I50c917a845fc4f711899a97d34c04813cc68b68c
Reviewed-on: https://webrtc-review.googlesource.com/58091
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22231}
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}
This is a reland of 12dc1842d62ee8df1e462f9b6a617fef9ab8b3b7.
Original change's description:
> Some cleanup for the logging code:
>
> * Only include 'tag' for Android. Before there was an
> extra std::string variable per log statement for all
> platforms.
> * Remove unused logging macro for Windows and 'module' ctor argument.
> * Move httpcommon code out of logging and to where it's used.
>
> Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
> Bug: webrtc:8928
> Reviewed-on: https://webrtc-review.googlesource.com/57183
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22184}
Bug: webrtc:8928
Change-Id: Ib97895aaeb376e19f136d258c0259a340235a5d1
Reviewed-on: https://webrtc-review.googlesource.com/58200
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22208}
This reverts commit 63e83c77ae81730a78ec4d5bf0465f25970f867a.
Reason for revert: JNI generator is not using the heap profiler
anymore.
Original change's description:
> Forward fix jni_generator_helper.h.
>
> In crrev.com/531028, the JNI generator starts to add heap profiler
> events to JNI generated functions.
>
> This will cause a ~80KiB regression and at the moment it is breaking
> the Chromium Roll into WebRTC.
>
> This CL defines a void macro to re-enable the Chromium Roll avoiding
> the size regression.
>
> Bug: chromium:801260
> Change-Id: I9543299199c4e14b6b9b235c5cb98c0d53cf29ea
> Reviewed-on: https://webrtc-review.googlesource.com/43021
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21730}
TBR=mbonadei@webrtc.org,magjed@webrtc.org,sakal@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:801260
Change-Id: I7dac211b89d8206dc461af0a17b6d53cc8661b2a
Reviewed-on: https://webrtc-review.googlesource.com/58040
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22206}
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}
This reverts commit 12dc1842d62ee8df1e462f9b6a617fef9ab8b3b7.
Reason for revert: Some internal tests keeps failing with this change.
Original change's description:
> Reland "Some cleanup for the logging code:"
>
> This is a reland of 9ecdcdf2b527bdb1d097782d92817c9866d6d18b.
>
> Original change's description:
> > Some cleanup for the logging code:
> >
> > * Only include 'tag' for Android. Before there was an
> > extra std::string variable per log statement for all
> > platforms.
> > * Remove unused logging macro for Windows and 'module' ctor argument.
> > * Move httpcommon code out of logging and to where it's used.
> >
> > Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
> > Bug: webrtc:8928
> > Reviewed-on: https://webrtc-review.googlesource.com/57183
> > Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22184}
>
> Bug: webrtc:8928
> Change-Id: Id062a5b61917e66561f6c8441c2defd525e38f16
> Reviewed-on: https://webrtc-review.googlesource.com/57880
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22191}
TBR=kwiberg@webrtc.org,tommi@webrtc.org,jonasolsson@webrtc.org
Change-Id: I2b04e361459926a503552a0e1fcf3d1da3ddb643
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8928
Reviewed-on: https://webrtc-review.googlesource.com/58101
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22195}
This is a reland of 9ecdcdf2b527bdb1d097782d92817c9866d6d18b.
Original change's description:
> Some cleanup for the logging code:
>
> * Only include 'tag' for Android. Before there was an
> extra std::string variable per log statement for all
> platforms.
> * Remove unused logging macro for Windows and 'module' ctor argument.
> * Move httpcommon code out of logging and to where it's used.
>
> Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
> Bug: webrtc:8928
> Reviewed-on: https://webrtc-review.googlesource.com/57183
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22184}
Bug: webrtc:8928
Change-Id: Id062a5b61917e66561f6c8441c2defd525e38f16
Reviewed-on: https://webrtc-review.googlesource.com/57880
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22191}
This reverts commit 9ecdcdf2b527bdb1d097782d92817c9866d6d18b.
Reason for revert: Breaks downstream project.
Original change's description:
> Some cleanup for the logging code:
>
> * Only include 'tag' for Android. Before there was an
> extra std::string variable per log statement for all
> platforms.
> * Remove unused logging macro for Windows and 'module' ctor argument.
> * Move httpcommon code out of logging and to where it's used.
>
> Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
> Bug: webrtc:8928
> Reviewed-on: https://webrtc-review.googlesource.com/57183
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22184}
TBR=kwiberg@webrtc.org,tommi@webrtc.org,jonasolsson@webrtc.org
Change-Id: I37a13d766fbdee2adb7f45231cf8be6b2b456bec
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8928
Reviewed-on: https://webrtc-review.googlesource.com/57720
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22187}
* Only include 'tag' for Android. Before there was an
extra std::string variable per log statement for all
platforms.
* Remove unused logging macro for Windows and 'module' ctor argument.
* Move httpcommon code out of logging and to where it's used.
Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
Bug: webrtc:8928
Reviewed-on: https://webrtc-review.googlesource.com/57183
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22184}
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}
FALLBACK_SOFTWARE is now treated as a critical error and results in
immediate fallback to software coding if available. If ERROR is
returned, codec reset is attempted. If that fails, software fallback
is used.
Bug: b/73498933
Change-Id: I7fe163efd09e6f27c72491e9595954ddc59b1448
Reviewed-on: https://webrtc-review.googlesource.com/54901
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22169}
This ensures memory is released timely and avoids problems with garbage
collection.
Native buffers don't support array operation, so FileVideoCapturer had
to be update to use FileChannel to write ByteBuffers directly.
Bug: None
Change-Id: I3f63d2adc159e9f39f0c68dd0bd6b1747686584e
Reviewed-on: https://webrtc-review.googlesource.com/55262
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22118}
STUN candidates use STUN binding requests to keep NAT bindings open. The
interval at which the STUN keepalive pings are sent is configurable now
via RTCConfiguration.
TBR=sakal@webrtc.org
Bug: None
Change-Id: I5f99ea3fe1e9042fa2bf7dcab0aace78f57739e6
Reviewed-on: https://webrtc-review.googlesource.com/54180
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22109}
This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122.
Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before.
Original change's description:
> Revert "Enables PeerConnectionFactory using external fec controller"
>
> This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3.
>
> Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java
>
> Original change's description:
> > Enables PeerConnectionFactory using external fec controller
> >
> > Bug: webrtc:8799
> > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8
> > Reviewed-on: https://webrtc-review.googlesource.com/43961
> > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22038}
>
> TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org
>
> Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8799
> Reviewed-on: https://webrtc-review.googlesource.com/54080
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22040}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org
Bug: webrtc:8799
Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8
Reviewed-on: https://webrtc-review.googlesource.com/55400
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22100}
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}
