Add events to Android VideoRendererGui implementation to
optionally report first rendered frame and video frame
dimension changes.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1292293002 .
Cr-Commit-Position: refs/heads/master@{#9715}
There is currently no way to dispose VideoRendererGui or VideoRendererGui.YuvImageRenderer. This CL adds functions to do so.
BUG=webrtc:4892
Review URL: https://codereview.webrtc.org/1273803002
Cr-Commit-Position: refs/heads/master@{#9710}
Reason for revert:
AppRTCDemo often crashes in loopback mode and incorrect layout when connection is established
BUG=webrtc:4909,webrtc:4910
Original issue's description:
> AppRTCDemo: Render each video in a separate SurfaceView
>
> This CL introduces a new org.webrtc.VideoRenderer.Callbacks implementation called SurfaceViewRenderer that renders each video stream in its own SurfaceView. AppRTCDemo is updated to use this new rendering.
>
> This CL also does the following changes:
> * Make the VideoRenderer.Callbacks interface asynchronous and require that renderFrameDone() is called for every renderFrame(). In JNI, this is implemented with cricket::VideoFrame::Copy()/delete.
> * Make public static helper functions: convertScalingTypeToVisibleFraction(), getDisplaySize(), and getTextureMatrix().
> * Introduces new helper functions surfaceWidth()/surfaceHeight() in EGlBase that allows to query the surface size.
> * Introduce PercentFrameLayout that implements the percentage layout that is used by AppRTCDemo.
>
> BUG=webrtc:4742
>
> Committed: https://crrev.com/05bfbe47ef6bcc9ca731c0fa0d5cd15a4f21e93f
> Cr-Commit-Position: refs/heads/master@{#9699}
TBR=glaznev@webrtc.org,wzh@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1286133002
Cr-Commit-Position: refs/heads/master@{#9703}
This CL introduces a new org.webrtc.VideoRenderer.Callbacks implementation called SurfaceViewRenderer that renders each video stream in its own SurfaceView. AppRTCDemo is updated to use this new rendering.
This CL also does the following changes:
* Make the VideoRenderer.Callbacks interface asynchronous and require that renderFrameDone() is called for every renderFrame(). In JNI, this is implemented with cricket::VideoFrame::Copy()/delete.
* Make public static helper functions: convertScalingTypeToVisibleFraction(), getDisplaySize(), and getTextureMatrix().
* Introduces new helper functions surfaceWidth()/surfaceHeight() in EGlBase that allows to query the surface size.
* Introduce PercentFrameLayout that implements the percentage layout that is used by AppRTCDemo.
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1257043004
Cr-Commit-Position: refs/heads/master@{#9699}
onPreviewFrame() might be called with a null data pointer, which is allowed according to the documentation.
BUG=webrtc:4877
Review URL: https://codereview.webrtc.org/1260183004
Cr-Commit-Position: refs/heads/master@{#9674}
For now add only Galaxy S4 to the list, since its H.264 HW encoder
generates two times lower bitrate comparing to target.
Also use VBR mode for H.264 encoder configuration.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1270603007 .
Cr-Commit-Position: refs/heads/master@{#9673}
This CL should not change any visible behaviour. It does the following:
* Extract GLES rendering into separate class GlRectDrawer. This class is also needed for future video encode with OES texture input.
* Clean up current ScalingType -> display size calculation and introduce new SCALE_ASPECT_BALANCED (b/21735609) and remove unused SCALE_FILL.
* Replace current mirror/rotation index juggling with android.opengl.Matrix operations instead.
Review URL: https://codereview.webrtc.org/1191243005
Cr-Commit-Position: refs/heads/master@{#9496}
With this we can write stuff like
assertThat(result.mandatory,
hasItem(new KeyValuePair("minWidth", "1280")));
The above will currently fail because the object falls back to ==.
BUG=None
Review URL: https://codereview.webrtc.org/1193883006
Cr-Commit-Position: refs/heads/master@{#9494}
- Remove an option to use MediaCodec SW decoder from Java layer.
- Better handling Java exceptions in JNI - detect exceptions
and either try to reset the codec or fallback to SW decoder.
- If any error is reported by codec try to fallback to SW
codec for VP8 or reset decoder and continue decoding for H.264.
- Add more logging for error conditions.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1178943007.
Cr-Commit-Position: refs/heads/master@{#9431}
This CL does not make any functional changes. The purpose is to extract some common code that is needed for texture capture and texture encode.
This CL does the following changes:
* Move common EGL functions from org.webrtc.MediaCodecVideoDecoder to org.webrtc.EglBase.
* Move common GL functions from org.webrtc.VideoRendererGui to org.webrtc.GlUtil and org.webrtc.GlShader.
* Remove unused call to surfaceTexture.getTransformMatrix in YuvImageRenderer.
* Add helper functions rotatedWidth()/rotatedHeight() in VideoRenderer.I420Frame.
R=glaznev@webrtc.org, hbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47309005.
Cr-Commit-Position: refs/heads/master@{#9414}
This fixed the warning:
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46: warning: [deprecation] Assert in junit.framework has been deprecated
import static junit.framework.Assert.*;
R=glaznev@webrtc.org, pthatcher@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/50209004
Cr-Commit-Position: refs/heads/master@{#9356}
This CL connects RTCConfiguration::audioJitterBufferFastMode in
PeerConnection.java, through libjingle, down to
NetEq::Config::enable_fast_accelerate in native WebRTC.
When enabled, it will allow NetEq to do faster time-compression when
the buffer level is very high.
BUG=4691
R=henrika@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55479004
Cr-Commit-Position: refs/heads/master@{#9344}
This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54529004
Cr-Commit-Position: refs/heads/master@{#9269}
- Set Camera.ErrorCallback callback when opening camera to
receive camera server error notifications.
- Allow user to provide interface for handling camera errors
happening on camera thread.
- Run camera observer on camera thread and monitor camera fps
and amount of callback buffers, print statistics and report error
if camera stops generating frames.
- Query camera formats starting from front camera instead of back
camera to detect camera failures as fast as possible.
- Change all DCHECK to CHECK in androidvideocapturer.cc to detect
camera error on release builds.
- Plus adding some extra logging.
R=hbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52519004
Cr-Commit-Position: refs/heads/master@{#9221}
This makes the build more flexible when linking against
prebuilt external libraries.
Use existing build_* variables for libyuv and json in talk/
(already in use in webrtc/).
Also make it possible to avoid building the GTK parts of the Linux build.
BUG=4242
R=andrew@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44179005
Cr-Commit-Position: refs/heads/master@{#9087}
