The class called AudioDeviceModule today is an implementation of a
future interface. We want to reserve the name AudioDeviceModule for
the actual interface. The implementation class has been renamed to
JavaAudioDeviceModule. 'Java' here refers to the fact that the
implementation is using android.media.AudioRecord as input and
android.media.AudioTrack as output, and this is opposed to native
AudioDeviceModule implementations such as OpenSLES and AAudio.
Bug: webrtc:7452
Change-Id: Ifc243c2e169b12a50128ee3252f06d574aa7b358
Reviewed-on: https://webrtc-review.googlesource.com/65400
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22673}
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}
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}
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 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 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}
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 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 00733015fa.
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 4f07bdb255.
>
> 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}
This reverts commit 4f07bdb255.
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}
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}
Deprecate old constructors. Intended to make java api consistent with
the changes in https://webrtc-review.googlesource.com/c/src/+/46622.
Bug: webrtc:8830
Change-Id: Iadecb5d033b5de841873905af659d8d234b75c7d
Reviewed-on: https://webrtc-review.googlesource.com/49062
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21956}
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}
Allows assuming that the buffer is not accessed after the call returns.
Bug: b/72675429
No-Try: True
Change-Id: Iff4a05433c6eed6aefec49ce67486966b1ed882f
Reviewed-on: https://webrtc-review.googlesource.com/49161
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21940}
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}
This patch exposes the network_thread so that
a custom PortAllocator can use it instead of e.g
creating own thread.
Bug: webrtc:8640
Change-Id: I705629e4f1a4d0a4fed7d53a774ba9564ba076fe
Reviewed-on: https://webrtc-review.googlesource.com/39925
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21643}
New OwnedPeerConnection takes ownership of the observer. This is done
to allow NativePeerConnectionFactory to return a capsulated object.
Bug: webrtc:8662
Change-Id: Ie876f7b9a1a17ebcfbe51537f712a32ab1a7cbfb
Reviewed-on: https://webrtc-review.googlesource.com/35300
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21610}
This removes the DefaultAudioProcessingFactory, PostProcessingFactory and DefaultAudioProcessingFactoryTest classes and leaves the interface AudioProcessingFactory without any default implementation (as the default APM is already created by the PeerConnectionFactory JNI).
Bug: webrtc:8701
Change-Id: I259108afbc5b24cab5161485f45af4236f775c18
Reviewed-on: https://webrtc-review.googlesource.com/37220
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21577}
We currently use raw jobject in our code mixed with sporadic
ScopedLocalRefFrame. This CL moves every jobject into a scoped object,
either local, global, or a parameter. Also, this CL uses the JNI
generation script to generate declaration stubs for the Java->C++
functions so that it no longer becomes possible to mistype them
without getting compilation errors.
TBR=brandt@webrtc.org
Bug: webrtc:8278,webrtc:6969
Change-Id: Ic7bac74a89c11180177d65041086d7db1cdfb516
Reviewed-on: https://webrtc-review.googlesource.com/34655
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21387}
This CL does the following:
* Split out MediaStream JNI code from peerconnection.cc to mediastream.h/mediastream.cc.
* Split out RtpSender JNI code from peerconnection.cc to rtpsender.h/rtpsender.cc.
* Split out TurnCustomizer JNI code from peerconnection.cc to turncustomizer.h/turncustomizer.cc.
* Add missing instanceof function to WrappedNativeVideoDecoder.java.
* Move some PeerConnectionFactory JNI declarations from pc/video.cc to peerconnectionfactory.cc.
* Add declaration to video.h for the JNI functions that depend on EglBase14_jni.h.
* Use a scoped object to store the global Java MediaStream objects that also call dispose.
Bug: webrtc:8278
Change-Id: I3c56a599b8bcbc8f34e5c5a7b9c9fe1d192ff3f3
Reviewed-on: https://webrtc-review.googlesource.com/34645
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21380}
This allows application to construct PeerConnection object in JNI and
pass that to Android API. API for wrapping Java PeerConnection Observers
is exposed for convenience.
Bug: webrtc:8662
Change-Id: Id110b92e6bb5ab00661cd50616d05c3e18a1697d
Reviewed-on: https://webrtc-review.googlesource.com/34520
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21379}
C++ API allows passing all configuration through RTCConfiguration
object. This adds all values previously passed through PC constraints
to Java RTCConfiguration object and deprecates API that takes PC
contraints.
Using the deprecated API overrides the values in RTCConfigration
object.
Bug: webrtc:8663, webrtc:8662
Change-Id: I128432c3caba74403513fb1347ff58830c643885
Reviewed-on: https://webrtc-review.googlesource.com/33460
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21357}
The problem was that the encoder was feeded with frames that had 0 as
a timestamp. This confused the encoder. H264 high profile support
clause was also wrong and is corrected.
Bug: webrtc:8601
Change-Id: Ic5a893b4b7573e694f865b63620843b2c9aa489f
Reviewed-on: https://webrtc-review.googlesource.com/32300
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21234}
This reverts commit aede67a199.
Reason for revert: Causes error:
JNI ERROR (app bug): local reference table overflow (max=512)'
Original change's description:
> Android: Generate JNI code for stats
>
> This CL also unifies the functions for converting from C++ to Java, and
> generates the boiler plate for converting C++ vectors to Java arrays.
>
> Bug: webrtc:8278
> Change-Id: I262e9162beae8a64ba0e8b6a27e1081207b03961
> Reviewed-on: https://webrtc-review.googlesource.com/26020
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20918}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: Ieb26ed8577bd489a4dd4f7542d16a7d0e11f409f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8278
Reviewed-on: https://webrtc-review.googlesource.com/26900
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20926}
This CL also unifies the functions for converting from C++ to Java, and
generates the boiler plate for converting C++ vectors to Java arrays.
Bug: webrtc:8278
Change-Id: I262e9162beae8a64ba0e8b6a27e1081207b03961
Reviewed-on: https://webrtc-review.googlesource.com/26020
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20918}
Also improves ownership model by using std::unique_ptr in a couple of
places instead of raw pointers.
Bug: webrtc:8278
Change-Id: I0429ec3c416b5baa1ffa21dad71e0d64b004c446
Reviewed-on: https://webrtc-review.googlesource.com/25020
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20863}
This is not the case for fallback codecs.
Bug: webrtc:7925
Change-Id: I5039d8879923a2db1e7c08bb5640763dd20ea8e7
Reviewed-on: https://webrtc-review.googlesource.com/24863
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20861}
It's now possible to generate JNI code for constructors and enums
correctly. This CL cleans that up.
Bug: webrtc:8278,webrtc:8551,webrtc:8556
Change-Id: I2284a30139cbb186c80713eb6113eda5659c16ad
Reviewed-on: https://webrtc-review.googlesource.com/25622
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20860}
Previously, wrapped native codec instances would leak the native object
if it was never used. This change fixes it by changing getNative method
to createNative.
Also fixes "Video codec hardware acceleration" setting in AppRTCMobile.
Bug: webrtc:7925
Change-Id: I53f6dc1dd5e37dea8d14278423122dede17719c5
Reviewed-on: https://webrtc-review.googlesource.com/24881
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20859}
This is a debug feature and should be disabled by default. Any client
that needs this functionality should call setEnableInternalTracer.
Bug: webrtc:8553
Change-Id: I78d718ebb95fc5cb8c464327b5b36e385ccfa9c0
Reviewed-on: https://webrtc-review.googlesource.com/24540
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20814}
To be able to compare VideoCodecInfos in a nice way in Java and still
use the correct criteria for comparing H264 codec infos.
A similar thing was done for Obj-C here:
https://webrtc-review.googlesource.com/c/src/+/4383
Bug: webrtc:7925
Change-Id: I43f532d4efa557fc8fe25a82eebc35072b91e6db
Reviewed-on: https://webrtc-review.googlesource.com/23240
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20716}
This reverts commit 64534fd917.
Reason for revert: no longer necessary.
Original change's description:
> Make HardwareVideo*Factory ctors that take fallbackToSoftware public.
>
> https://webrtc-review.googlesource.com/c/src/+/17480 changed the default behavior of HardwareVideoEncoderFactory and HardwareVideoDecoderFactory without providing any way to maintain the previous behavior. This breaks current users.
>
> Making these constructors public at least gives existing users a way to maintain the previous behavior.
>
> Bug: webrtc:7925
> Change-Id: Id8f0ec25026592f5e9096ac5f39fdda22993ff09
> Reviewed-on: https://webrtc-review.googlesource.com/22763
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20675}
TBR=magjed@webrtc.org,sakal@webrtc.org,andersc@webrtc.org,mellem@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:7925
Change-Id: Ia467337a3916371a619e1d238e1dd03aa0e19261
Reviewed-on: https://webrtc-review.googlesource.com/23380
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20695}
This CL also merged native_handle_impl.cc and videoframe_jni.cc to keep
all JNI code for the same Java class in the same file, and also renames
this file to jni/videoframe.cc.
The classes AndroidVideoBufferFactory and JavaVideoFrameFactory are
now unnecessary since we cache everything and can be simplified to
global static functions instead.
Bug: webrtc:8278
Change-Id: I03d7b0bbde64cfb407cd6210478ddf9d5599cd8c
Reviewed-on: https://webrtc-review.googlesource.com/22923
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20677}
https://webrtc-review.googlesource.com/c/src/+/17480 changed the default behavior of HardwareVideoEncoderFactory and HardwareVideoDecoderFactory without providing any way to maintain the previous behavior. This breaks current users.
Making these constructors public at least gives existing users a way to maintain the previous behavior.
Bug: webrtc:7925
Change-Id: Id8f0ec25026592f5e9096ac5f39fdda22993ff09
Reviewed-on: https://webrtc-review.googlesource.com/22763
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20675}
In the SoftwareVideoCodecFactories, don't try to create VP9 encoder or
decoder if WebRTC was built without support for it.
Bug: None
Change-Id: I09b87fdcf798c763310af4998dbea8011843010d
Reviewed-on: https://webrtc-review.googlesource.com/22924
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20672}
This is similar to https://webrtc-review.googlesource.com/c/src/+/3620
for iOS.
Using the new WebRtcMediaEngineFactory::Create API, the built-in
software video codecs are no longer appended to the injected codecs.
To be able to use the software codecs, they are exposed as Java
classes through SoftwareVideoEncoderFactory etc.
There is also a new DefaultVideoEncoderFactory used by AppRTCMobile.
This factory tries to use hardware implementations where available,
but falls back to using the injected software codecs.
The HardwareVideoEncoderFactory is temporarily also falling back on
the software codecs in its default configuration in order to
maintain backwards compatibility.
Bug: webrtc:7925
Change-Id: I3e8c5ed492ccd160aca968986ad217d7978a951c
Reviewed-on: https://webrtc-review.googlesource.com/17480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20647}
The crash occured if removeFrameListener was called after releasing
the EglRenderer.
Bug: b/69040588
Change-Id: I90acc3b280d2009e5f13bb8836a288eb20c7d1d0
Reviewed-on: https://webrtc-review.googlesource.com/21380
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20620}
The first example CL for generating JNI code
(https://webrtc-review.googlesource.com/c/src/+/4500) seems to stick, so
this CL updates the rest of the VideoEncoder. The JNI code for
Java -> C++ is still done manually.
This CL puts the necessary helper Java methods in a class called
VideoEncoderWrapper.
Bug: webrtc:8278
Change-Id: Ic3a6defe59c094f67ffd8ea86d6c272c676980ae
Reviewed-on: https://webrtc-review.googlesource.com/20871
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20587}
In https://chromium-review.googlesource.com/c/chromium/src/+/750645
Chromium started to use an ErrorProne plugin to discourage synchronized
public methods (an encourage the usage of synchronized blocks).
In order to unblock the Chromium Roll we can suppress these warnings
and decide if we want to align with Chromium on this check or ask
them to make it optional.
More details in the bug.
TBR=magjed@webrtc.org
Bug: webrtc:8491
Change-Id: Ie77a324e54aab44a4f59853959549f1d21f884a0
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/20060
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20569}
The new SurfaceEglRenderer helper class extends EglRenderer and
implements rendering on a SurfaceView.
Bug: webrtc:8242
Change-Id: Ic532fe487755d3b54c6bd03f239d714e1ecb10ad
Reviewed-on: https://webrtc-review.googlesource.com/2940
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20562}
Previously input buffers would be filled incorrectly for sparsely
packed buffers where stride is not equal to the plane width.
Bug: webrtc:8478
Change-Id: I080fa3c354a27982bb996be8c1e41b103384e4bc
Reviewed-on: https://webrtc-review.googlesource.com/17321
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20550}
This is a reland of ba78b5a905
Original change's description:
> Android: Generate JNI code for VideoSink and VideoEncoder
>
> This is the first CL to start generating JNI code. It has updated two of
> the most recent classes to use JNI code generation.
>
> Bug: webrtc:8278
> Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
> Reviewed-on: https://webrtc-review.googlesource.com/3820
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19994}
Bug: webrtc:8278
Change-Id: Id3e6513736eb87d7c234be3b0d13c5d30435201c
Reviewed-on: https://webrtc-review.googlesource.com/4500
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20548}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}
This reverts commit 90bace0958.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.
This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.
The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.
TBR=solenberg
Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
This affects the new injectable codecs.
Bug: webrtc:8459
Change-Id: I484a3ae4c29fd8bae8b13308315758b3689bdd4d
Reviewed-on: https://webrtc-review.googlesource.com/16861
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20478}
This is done in preparation to make all javac warnings into errors for
WebRTC targets.
Bug: webrtc:6597
Change-Id: I402043157bd75943adf0de52111e5a1bb179c6d1
Reviewed-on: https://webrtc-review.googlesource.com/15104
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20450}
This allows injection of a user-defined post processing module from
the Android layer.
Bug: webrtc:8163
Change-Id: If3a6b4726c34c5f82d186b8cf95373c283cbd3f6
Reviewed-on: https://webrtc-review.googlesource.com/7610
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20367}
Previous strict threading requirements we not always followed and this
causes crashes in some scenarios.
Bug: webrtc:7760
Change-Id: I4808edfff19c63a9c4a9585834d46f9c73bb1c4b
Reviewed-on: https://webrtc-review.googlesource.com/13660
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20353}
