It should work now as the packet limit in the jitter buffer has been increased.
BUG=webrtc:4889
Review URL: https://codereview.webrtc.org/1272153002
Cr-Commit-Position: refs/heads/master@{#9700}
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}
DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands:
https://codereview.webrtc.org/1189583002
The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore.
Where a service was previously passed around, a store is now passed around.
Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur.
For more information about the steps being taken to land this without breaking Chromium, see referenced bug.
BUG=webrtc:4899
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1176383004 .
Cr-Commit-Position: refs/heads/master@{#9696}
This CL makes sure the methods are always called on the correct thread.
Review URL: https://codereview.webrtc.org/1235263003
Cr-Commit-Position: refs/heads/master@{#9688}
Currently, we only return frames if CreateAliasedFrame() is called, which is not the case for dropped frames.
Review URL: https://codereview.webrtc.org/1268333005
Cr-Commit-Position: refs/heads/master@{#9683}
Significant changes:
- move the libjingle_examples.gyp file into webrtc directory.
- rename talk/examples/android to webrtc/examples/androidapp to avoid name conflicts.
- update paths in talk/libjingle_tests.gyp to point to webrtc directory for Objective-C test.
BUG=
R=pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1235563006 .
Cr-Commit-Position: refs/heads/master@{#9681}
New PeerConnectionFactoryInterface::CreatePeerConnection taking both service and store added (old CreatePC signature still exists).
This is CL is part of an effort to land https://codereview.webrtc.org/1176383004 without breaking Chromium.
See bug for more information.
BUG=webrtc:4899
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1268363002 .
Cr-Commit-Position: refs/heads/master@{#9680}
Especially the VP9 codec currently may overshoot bitrate target at sudden picture changes, resulting in frames over 800 packets.
This limit should be reduced again once the codec behaves.
BUG=webrtc:4889
Review URL: https://codereview.webrtc.org/1266353003
Cr-Commit-Position: refs/heads/master@{#9675}
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}
Don't verify increasing sequence numbers after test complesion as this
can be racy with regards to test shutting down send transports.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1269743004 .
Cr-Commit-Position: refs/heads/master@{#9672}
Sometimes the port still try to send stun packet when the connection is disconnected,
causing an assertion error.
BUG=4859
Review URL: https://codereview.webrtc.org/1247573002
Cr-Commit-Position: refs/heads/master@{#9671}
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
This change includes base/logging.h instead of the old and deprecated
system_wrappers/interface/logging.h. This requires some changes of the
actual logging invocations.
For reference the following regexps where used (in Eclipse) for a few
of the replacements:
find: LOG_FERR1\(\s*([^,]*),\s*([^,]*),\s*(.*)\);
replace: LOG($1) << "$2 " << $3;
find: LOG_FERR2\(\s*([^,]*),\s*([^,]*),\s*([^,]*),\s*(.*)\);
replace: LOG($1) << "$2 " << $3 << " " << $4;
BUG=4735
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50229004 .
Cr-Commit-Position: refs/heads/master@{#9669}
PacketSender can now log Pause/Resume events into a MetricRecorder. Solved estimate error and optimal bitrate issue for test 5.7 (multiple short TCP flows).
Added Sending Estimate logging and plotting.
Fixed plotting issue on plot_dynamics.py
Now lines with the same color (in different boxes) correspond to the same flow.
Adjusting plot_dynamics.py font size according to number of variables.
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1270543002 .
Cr-Commit-Position: refs/heads/master@{#9664}
On GetCapabilities() failure, caps.cDestinations is left uninitialized.
Without a protection the following code runs in a random loop
in the worst case up to 0xFFFFFFFF times.
for (destId = 0; destId < caps.cDestinations; destId++)
{
GetDestinationLineInfo(mixId, destId, destLine);
BUG=webrtc:4882
Review URL: https://codereview.webrtc.org/1269563002
Cr-Commit-Position: refs/heads/master@{#9663}
Re-lands "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module."
This reverts commit b933667a7f97697d6390d1eee5f378cedd9ca208.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1259683003 .
Cr-Commit-Position: refs/heads/master@{#9661}
Prevents presubmit failures when touching audio_coding_module.gypi due
to source files being included from outside the gypi directory.
BUG=
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1262333002 .
Cr-Commit-Position: refs/heads/master@{#9659}
Computing all metrics using constant extra memory.
PlotHistogram methods are executed in constant time.
-- Previously throughput and delay were using O(num_packets) extra memory and their associated PlotHistograms took linear time complexity.
Added MetricRecorder unittests
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1257683006 .
Cr-Commit-Position: refs/heads/master@{#9658}
Placed the protobuf structures in the namespace webrtc::rtclog. Removed the message field from the DebugEvent structure, since it was not used.
Added an interface to set config information for VideoReceiveStream and VideoSendStream in the event log.
Added function to log full RTCP packets and changed RTP-logging to only log headers.
Significantly extended the unit tests for RtcEventLog.
R=ivoc@webrtc.org, minyue@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1230973005 .
Cr-Commit-Position: refs/heads/master@{#9656}
