Logging the OpenGL shader source code makes it easier to debug problems.
Bug: None
Change-Id: Ie4724b1353511eae3806e98270b04e5daa4c11fc
Reviewed-on: https://webrtc-review.googlesource.com/69322
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22900}
This reverts commit 64051d4975.
Reason for revert: Fix applied.
Original change's description:
> Revert "Android: Generalize and make TextureBufferImpl public"
>
> This reverts commit 28111d7fa0.
>
> Reason for revert: Crashes video_quality_loopback_test.
>
> Original change's description:
> > Android: Generalize and make TextureBufferImpl public
> >
> > This CL generalizes TextureBufferImpl so it's useful from other contexts than
> > from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
> > the class in the api so that clients don't have to duplicate the logic.
> >
> > Bug: None
> > Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
> > Reviewed-on: https://webrtc-review.googlesource.com/69819
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22875}
>
> TBR=magjed@webrtc.org,sakal@webrtc.org
>
> Change-Id: Ica7fc181fec70b8b79f39f0e114eef81a03aa116
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/70240
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22878}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: I173d1ccfe0baa80460f796ebaedc51731233108f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/70183
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22883}
This reverts commit 28111d7fa0.
Reason for revert: Crashes video_quality_loopback_test.
Original change's description:
> Android: Generalize and make TextureBufferImpl public
>
> This CL generalizes TextureBufferImpl so it's useful from other contexts than
> from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
> the class in the api so that clients don't have to duplicate the logic.
>
> Bug: None
> Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
> Reviewed-on: https://webrtc-review.googlesource.com/69819
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22875}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: Ica7fc181fec70b8b79f39f0e114eef81a03aa116
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/70240
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22878}
This CL generalizes TextureBufferImpl so it's useful from other contexts than
from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
the class in the api so that clients don't have to duplicate the logic.
Bug: None
Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
Reviewed-on: https://webrtc-review.googlesource.com/69819
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22875}
This CL makes it possible to create a GlTextureFrameBuffer from any
thread. The actual GL resources will be allocated the first time
setSize() is called. The purpose is to be able to use 'final' variables
more often for this class and avoid @Nullable annotations.
Bug: None
Change-Id: I350304bcd33fd674990254df37a615995972f322
Reviewed-on: https://webrtc-review.googlesource.com/69241
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22835}
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}
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}
