Instead of sending a black frame that's the size of the VideoFormat send
a black frame in the format we're already sending. This prevents
expensive encoder reconfiguration when the sending format is a different
resolution. This speeds up setting a null capturer (removing the
capturer) significantly as it doesn't entail an encoder reconfiguration.
R=mflodman@webrtc.org, pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/39179004
Cr-Commit-Position: refs/heads/master@{#8405}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8405 4adac7df-926f-26a2-2b94-8c16560cd09d
It’s possible to implement cricket::VideoFrame::CopyToBuffer using the virtual interface. This removes the need for subclasses to implement their own versions. This CL also fixes a bug in cricket::VideoFrame::CopyToPlanes which currently assumes that GetUPitch() == GetVPitch(), otherwise it may segfault.
I think this CL should land regardless, but the main purpose is to pave the way for for planned changes to I420VideoFrame. See https://review.webrtc.org/38879004.
R=fbarchard@google.com, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39889004
Cr-Commit-Position: refs/heads/master@{#8403}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8403 4adac7df-926f-26a2-2b94-8c16560cd09d
Modifies WebRtcVideoSendStream to use a default width/height of 16px.
This significantly reduces SetRemoteDescription time under
WebRtcVideoEngine2. Also preventing (expensive) reconfigurations due to
incoming frames when the channel is not sending yet.
Tests have been modified to generate a frame before expecting a certain
encoder size to have been configured.
Also adding tracing to WebRtcVideoSendStream::InputFrame as it can lead
to reconfigurations of the encoder which is expensive and it should show
up in chrome://tracing.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42369004
Cr-Commit-Position: refs/heads/master@{#8381}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8381 4adac7df-926f-26a2-2b94-8c16560cd09d
For now:
java_helpers - JNI convenience functions etc. Can in theory be moved to libjingle / webrtc general one day.
classreferenceholder - app/webrtc specific Java class loader.
androidvideocapturer_jni - the jni part of the video capturer I added.
peerconnection_jni - all the rest.
This also move all jni specifics into ns webrtc_jni to avoid naming collision.
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38099004
Cr-Commit-Position: refs/heads/master@{#8363}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8363 4adac7df-926f-26a2-2b94-8c16560cd09d
PeerConnectionTest.java currently works, but only on a device with
network interfaces up. This is not a problem for desktop, but it is a
problem when running on Android devices since the devices in the lab
generally don't have network (due to the chaotic radio environment in
the device labs, devices are simply kept in flight mode).
The test does work if one modifies this line in the file
webrtc/base/network.cc:
bool ignored = ((cursor->ifa_flags & IFF_LOOPBACK) ||
IsIgnoredNetwork(*network));
If we remove the IFF_LOOPBACK clause, the test starts working on
an Android device in flight mode. This is nice - we're running the
call and packets interact with the OS network stack, which is good
for this end-to-end test. We can't just remove the clause though since
having loopback is undesirable for everyone except the test (right)?
so we need to make this behavior configurable.
This CL takes a stab at a complete solution where we pass a boolean
all the way through the Java PeerConnectionFactory down to the
BasicNetworkManager. This comes as a heavy price in interface
changes though. It's pretty out of proportion, but fundamentally we
need some way of telling the network manager that it is on Android
and in test mode. Passing the boolean all the way through is one way.
Another way might be to put the loopback filter behind an ifdef and
link a custom libjingle_peerconnection.so with the test. That is hacky
but doesn't pollute the interfaces. Not sure how to solve that in GYP
but it could mean some duplication between the production and
test .so files.
It would have been perfect to use flags here, but then we need to
hook up gflags parsing to some main() somewhere to make sure the
flag gets parsed, and make sure to pass that flag in our tests.
I'm not sure how that can be done.
Making the loopback filtering conditional is exactly how we solved the
equivalent problem in content_browsertests in Chrome, and it worked
great.
That's all I could think of.
BUG=4181
R=perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36769004
Cr-Commit-Position: refs/heads/master@{#8344}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8344 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes buffer/texture path separation inside WebRtcVideoEngine and
DeliverTextureFrame(). This unifies frame delivery with
WebRtcVideoEngine2 which is expected to automagically work with texture
frames after this change.
BUG=1788
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38069005
Cr-Commit-Position: refs/heads/master@{#8326}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8326 4adac7df-926f-26a2-2b94-8c16560cd09d
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:
- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup
Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).
BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33969004
Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL performs the following renames of targets to
make GYP and GN more unified and make the targets that
have the same name as the module and include the external
render/capture implementation (the internal one is only
used by WebRTC tests).
This makes it natural to declare dependencies in GN
without having to specify the target.
Summary of the renames:
GYP:
video_render_module_impl -> video_render (new target)
video_capture_module_impl -> video_capture (new target)
GN:
video_capture -> video_capture_module (now identical to the GYP target)
video_capture_impl -> video_capture
video_render -> video_render_module (now identical to the GYP target)
video_render_impl -> video_render
BUG=456815
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35099004
Cr-Commit-Position: refs/heads/master@{#8323}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8323 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes AddRenderCallback from ViERenderer and implements
VideoReceiveStream on top of DeliverI420Frame like WebRtcVideoEngine
currently does today.
Also adds ::IsTextureSupported() to the VideoRenderer interface to
permit querying whether an external renderer supports texture rendering.
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/34169004
Cr-Commit-Position: refs/heads/master@{#8299}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8299 4adac7df-926f-26a2-2b94-8c16560cd09d
with stride > width.
Recent libvpx update generates output video frames with stride
value greater than width, which was not supported by Android OpenGL
video renderer (Android GLES2 doesn't have GL_UNPACK_ROW_LENGTH
to provide stride information for buffer in glTexImage2D call).
Fix it by implementing native frame copying for Java
VideoRenderer.I420Frame implementation.
BUG=4248
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40639004
Cr-Commit-Position: refs/heads/master@{#8252}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8252 4adac7df-926f-26a2-2b94-8c16560cd09d
