This work is also done as a preparation for adding stereo support to the
FineAudioBuffer.
Review hints:
Actual changes are in modules/audio_device/fine_audio_buffer.h,cc, the rest is
just adaptations to match these changes.
We do have a forked ADM today, hence, some changes are duplicated.
The changes have been verified on all affected platforms.
Bug: webrtc:6560
Change-Id: I413af41c43809f61455c45ad383fc4b1c65e1fa1
Reviewed-on: https://webrtc-review.googlesource.com/70781
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22938}
Mainly remove CHECKinitialized_ macro and AGC functionality. Also make
actual behavior clearer in some functions.
Bug: webrtc:7452
Change-Id: I1eac86f4eaff7b14820d3e4192b15c20ab6acb45
Reviewed-on: https://webrtc-review.googlesource.com/69161
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22820}
This CL introduces sdk/android/api/org/webrtc/audio/AudioDeviceModule.java,
which is the new interface for audio device modules on Android.
This CL also refactors the main AudioDeviceModule implementation, which
is sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java and makes
it conform to the new interface. The old code used global static methods
to configure the audio device code. This CL gets rid of all that and uses
a builder pattern in JavaAudioDeviceModule instead. The only two dynamic
methods left in the interface are setSpeakerMute() and setMicrophoneMute().
Removing the global static methods allowed a significant cleanup, and e.g.
the file sdk/android/src/jni/audio_device/audio_manager.cc has been
completely removed.
The PeerConnectionFactory interface is also updated to allow passing in
an external AudioDeviceModule. The current built-in ADM is encapsulated
under LegacyAudioDeviceModule.java, which is the default for now to
ensure backwards compatibility.
Bug: webrtc:7452
Change-Id: I64d5f4dba9a004da001f1acb2bd0c1b1f2b64f21
Reviewed-on: https://webrtc-review.googlesource.com/65360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22765}
This CL splits out the audio device module Java code into a separate
target, and also splits up the audio device module implementations into
three different build targets, one for OpenSLES, AAudio, and the Java
based implementation.
Bug: webrtc:7452, webrtc:9048
Change-Id: I8ec09c73580b468837223ddd420fb29ca61fdea5
Reviewed-on: https://webrtc-review.googlesource.com/66461
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22727}
This method is only used for logging and is blocking further refactoring
work. Once the refactoring and cleanup of the external AudioDeviceModule
is complete, we can revisit what logging we want and need and add it in
a cleaner way.
Bug: webrtc:7452
Change-Id: If08bcfb37860e9e7b9b5105cb75f748b53775f69
Reviewed-on: https://webrtc-review.googlesource.com/65460
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22678}
The OpenSLES engine is currently managed by the AudioManager which is
a generic class shared between different kinds of audio input/output.
This CL moves the responsibility of the OpenSLES engine to the actual
OpenSLES implementations.
Bug: webrtc:7452
Change-Id: Iecccb03ec5cd12ce2f3fdc44daaedae27aecf88b
Reviewed-on: https://webrtc-review.googlesource.com/64520
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22661}
Our style guide dictates that we should prefer using return values rather
than output parameters when we can. Some of the methods like
MaxSpeakerVolume() are not required to be able to provide a value. In
these cases I changed the return type to an rtc::Optional.
Also, this CL fixes a bug with StereoRecordingIsAvailable() that would
not previously be passed along correctly in the template layer.
Bug: webrtc:7452
Change-Id: I0a1f455093bfe092627118d65a996212a65eeb2b
Reviewed-on: https://webrtc-review.googlesource.com/64401
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22629}
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}
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}
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}
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}
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}
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}
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 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}
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}
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}
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}
Moves AndroidNetworkMonitor out of pc folder. Even clients not using
PeerConnection seem to be using it and it doesn't have any dependencies
to the PeerConnection API.
Bug: webrtc:8769
Change-Id: I2bdeff9f5c9925e13388fbc77aa9b264a7583548
Reviewed-on: https://webrtc-review.googlesource.com/53260
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22051}
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}
This method used to just wrap frame when passed a native frame and
create a new one when passed non-native frame. This caused a memory
leak when a new frame was returned because the caller didn't release
the frame. Now the method always returns a new frame and the caller is
responsible for releasing it.
Bug: webrtc:8892, b/72675429
Change-Id: I06d67a6ed4c059cae1d709c51b0266f9c72fef1a
Reviewed-on: https://webrtc-review.googlesource.com/53840
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22033}
We want api/peerconnectioninterface.h (and corresponding build target)
to not depend on call.h, and generally we treat Call as an internal,
non-api, class. But we need CallFactoryInterface in the api in order to
enable use of PeerConnection with or without support for media.
Making CallConfig a top-level class makes it possible to forward declare
it, together with Call, for use in callfactoryinterface.h and
peerconnectioninterface.h.
Delete the peerconnection_and_implicit_call_api target, replaced by
new target callfactory_api, to link between Call and Peerconnection.
Bug: webrtc:7504
Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1
Reviewed-on: https://webrtc-review.googlesource.com/46201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22020}
Implements JavaToNativeStringMap that is a replacement for
JavaToStdMapStrings. It uses a new template method JavaToNativeMap. Also
adds testing support for native API and a test for JavaToNativeStringMap.
Bug: webrtc:8769
Change-Id: I580d4992a899ebe02da39af450fa51d52ee9b88b
Reviewed-on: https://webrtc-review.googlesource.com/48060
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21967}
This reverts commit 6780c51b23516803dc27173d10ba98d018780447.
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}
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.
Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
Fixes a race condition where frame_extra_infos_ is accessed from
multiple threads by adding a lock.
Adds thread safety idioms to the file to guard agains similar mistakes
in the future.
Bug: b/72979294
Change-Id: I0f2f947282a5b3414f1351e9e8e52ad523f7d2f6
Reviewed-on: https://webrtc-review.googlesource.com/48641
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21926}
VideoEncoderWrapper may be released and reused (Release() followed by
InitEncode()). This often happens back to back when encoders are
reconfigured. Because encoded frames are posted asynchronously to the
encoder queue, they may be processed after the encoder associated with
them has already been released.
In the existing code, if a frame for the new encoder had already been
received, the processing of the frame for the old encoder would clear
out the record for the new encoder's frame. This is now fixed by only
clearing out records that are older than the encoded frame being
processed.
A particularly bad symptom is when the new encoder is used for the same
stream as the old one (but was reconfigured for e.g. a change in
resolution). In that case, the new encoder's initial keyframe gets
dropped, and all subsequent difference frames are based off the last
sent frame from the old encoder. This all renders as garbage until a new
keyframe is sent.
Bug: webrtc:8849
Change-Id: I25094f12b38e03e158dc10ac66e92aa9ebaa5541
Reviewed-on: https://webrtc-review.googlesource.com/47549
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21896}
In MediaCodecVideoEncoder, VideoFrame timestamp was used as a
presentation timestamp. With this change timestamp maintained in C++
code is used instead. This matches the behaviour with old frame
callbacks.
Bug: b/72832862
Change-Id: I1f0543ebe837ccac22c83a81a81f3ea128e2a866
Reviewed-on: https://webrtc-review.googlesource.com/47381
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21872}
The network preference is added to RTCConfiguration and passed to ICE.
ICE considers now the preference set by applications over network
interface types when making decisions in candidate pair switching.
Bug: webrtc:8816
Change-Id: I40d2612705b54c83dd45772ac855808e0a76b1e1
Reviewed-on: https://webrtc-review.googlesource.com/44020
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21855}
