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gerrit-public.fairphone.software
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fp2-dev
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platform
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external
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chromium_org
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third_party
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webrtc
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bf6d5729629fc98dbbf8a4965cec1efe93cad895
/
video_engine
bf6d572
Rename VideoCall to Call.
by pbos@webrtc.org
· 11 years ago
618a0ec
ExternalVideoDecoder for new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
ca20f3d
Clamp camera id to legal values.
by fischman@webrtc.org
· 11 years ago
7dc1790
Improving padding rules and breaking out bw allocation to ViEEncoder.
by stefan@webrtc.org
· 11 years ago
db74c61
Adds support for combining RTX and FEC/RED.
by stefan@webrtc.org
· 11 years ago
31a8ce7
Removing FrameForStorage
by mikhal@webrtc.org
· 11 years ago
06eaa54
Restore severity precondition to logging.h.
by andrew@webrtc.org
· 11 years ago
0020858
Remove send and receive streams when destroyed.
by pbos@webrtc.org
· 11 years ago
4998966
Allow unknown flags in test_main.cc.
by pbos@webrtc.org
· 11 years ago
c77dcb0
Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter.
by mflodman@webrtc.org
· 11 years ago
1cd055c
Disable EngineTest.ReceivesPliAndRecoversWithNack.
by mflodman@webrtc.org
· 11 years ago
9e70940
Add FakeEncoder to VideoSendStream tests.
by pbos@webrtc.org
· 11 years ago
324a016
Changed method name.
by mflodman@webrtc.org
· 11 years ago
94ef274
Renamed method.
by mflodman@webrtc.org
· 11 years ago
710d2e1
Function name change.
by mflodman@webrtc.org
· 11 years ago
a594db2
Fixing capture frame race in ViECapturer.
by mflodman@webrtc.org
· 11 years ago
ce9de71
Overuse detection based on capture-input jitter.
by pbos@webrtc.org
· 11 years ago
8c6633c
Add isolate configuration for Android for all tests.
by kjellander@webrtc.org
· 11 years ago
9b7bdee
Revert r4562
by elham@webrtc.org
· 11 years ago
6203090
Updated WebRTC version to 3.40
by elham@webrtc.org
· 11 years ago
e2e033a
Relanding 4597 - Don't force key frame when decoding with errors.
by mikhal@webrtc.org
· 11 years ago
c179706
Remove newapi:: namespace for typenames without overlap.
by pbos@webrtc.org
· 11 years ago
f83a872
Revert 4597 "Don't force key frame when decoding with errors"
by henrike@webrtc.org
· 11 years ago
c5fc6e0
Don't force key frame when decoding with errors
by mikhal@webrtc.org
· 11 years ago
0f911c9
Remove template usage of typeless enum in fake_encoder.
by pbos@webrtc.org
· 11 years ago
206c4a5
Enabling and testing RTCP CNAME in new API.
by pbos@webrtc.org
· 11 years ago
55afdbe
Adds two tests for verifying padding and ramp-up behavior.
by stefan@webrtc.org
· 11 years ago
3540c82
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 11 years ago
a20e2d4
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ..."""
by stefan@webrtc.org
· 11 years ago
3ded8c9
Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots.
by henrike@webrtc.org
· 11 years ago
c0976d2
Revert 4582 "Reverts a second set of reverts caused by a bug in ..."
by henrike@webrtc.org
· 11 years ago
efe1f0f
Reverts a second set of reverts caused by a bug in a dependency.
by stefan@webrtc.org
· 11 years ago
f96e534
Call SetExecutablePath from test_main.cc
by pbos@webrtc.org
· 11 years ago
7deb335
Make FrameGeneratorCapturer own frame_generator.
by pbos@webrtc.org
· 11 years ago
eb7b0c4
Merging video_full_stack_tests and video_engine_tests.
by phoglund@webrtc.org
· 11 years ago
67acd69
VideoSendStream SSRC test.
by pbos@webrtc.org
· 11 years ago
96ff6ab
Added missing static_cast conversion.
by pbos@webrtc.org
· 11 years ago
8ce445e
Implementation and testing of PLI in new API.
by pbos@webrtc.org
· 11 years ago
3207eaa
Made all integration tests use consistent naming.
by phoglund@webrtc.org
· 11 years ago
ece3d35
Added choice of decode error mode to loopback test.
by agalusza@google.com
· 11 years ago
7fc75bb
Update talk to 50918584.
by wu@webrtc.org
· 11 years ago
1e817c3
Roll chromium_revision 214260:217707 and gflags 45:84
by fischman@webrtc.org
· 11 years ago
e155918
Revert 4547 "Isolate GYP target and .isolate files for tests"
by kjellander@webrtc.org
· 11 years ago
298bbdb
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 11 years ago
e416ab2
Remove ViEBase::Init() call from VideoCall.
by pbos@webrtc.org
· 11 years ago
c2014fd
Remove VideoEngine class from new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
eca72bf
Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors".
by marpan@webrtc.org
· 11 years ago
48bcf6f
Disable racy part of RunsRtpRtcpTestWithoutErrors.
by pbos@webrtc.org
· 11 years ago
52c5c70
Replace MapWrapper with std::map<>.
by pbos@webrtc.org
· 11 years ago
8c8c87f
Updated WebRTC version to 3.39
by elham@webrtc.org
· 11 years ago
823a888
Signal when shutting down DirectTransport.
by pbos@webrtc.org
· 11 years ago
d893b3f
Avoid acquiring VCM::_receiveCritSect during decode callback.
by wuchengli@chromium.org
· 11 years ago
fe881f6
Run loopback tests with network thread.
by pbos@webrtc.org
· 11 years ago
f43029b
Revert "Avoid acquiring VCM::_receiveCritSect during decode callback."
by wuchengli@chromium.org
· 11 years ago
b0af417
Avoid acquiring VCM::_receiveCritSect during decode callback.
by wuchengli@chromium.org
· 11 years ago
e915174
Allowing decoding with errors, when disabling nack.
by mikhal@webrtc.org
· 11 years ago
ea7b33e
* Update libjingle to 50389769.
by wu@webrtc.org
· 11 years ago
3ddbca9
Updated WebRTC version number to 3.38
by elham@webrtc.org
· 11 years ago
3f45c2e
Switch C++-style C headers with their C equivalents.
by pbos@webrtc.org
· 11 years ago
043f6a8
Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp.
by pbos@webrtc.org
· 11 years ago
78ab511
Use RtpHeaderParser in VideoCall implementation.
by pbos@webrtc.org
· 11 years ago
ce85109
Glue code and tests for NACK in new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
3a74d40
Fix send times in video_full_stack.
by pbos@webrtc.org
· 11 years ago
8704595
Add back is.FrameProvider() call lost in r4194.
by pbos@webrtc.org
· 11 years ago
acb00f5
Adds all unittests to android NDK-APK framework.
by henrike@webrtc.org
· 11 years ago
aa79e6e
Unbreak clang/android build of webrtc.
by fischman@webrtc.org
· 11 years ago
cb9a72b
Adding possibility to use encoding time when trigger underuse for frame based overuse detection.
by mflodman@webrtc.org
· 11 years ago
45e69ce
Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org
by elham@webrtc.org
· 11 years ago
d69e2f4
Access receiving_ under receive_cs critical section
by braveyao@webrtc.org
· 11 years ago
bf76ae2
Hooking up first simple CPU adaptation version.
by mflodman@webrtc.org
· 11 years ago
237fd31
Correctly rebuild WebRTCDemo after jni/ source file changes
by yujie.mao@webrtc.org
· 11 years ago
0ba496b
Revert r4301
by tnakamura@webrtc.org
· 11 years ago
b89eed3
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on
by elham@webrtc.org
· 11 years ago
5526bfb
Updated WebRTC version to 3.36 TBR=tnakamura@webrtc.org
by elham@webrtc.org
· 11 years ago
e835019
Include files from webrtc/.. paths in video_engine/.
by pbos@webrtc.org
· 11 years ago
778a172
Direct3D renderer for new VideoEngine API tests.
by pbos@webrtc.org
· 11 years ago
46088d2
Support sending multiple report blocks and keeping track of statistics on several SSRCs.
by stefan@webrtc.org
· 11 years ago
73a2ba6
Update version number to 3.35
by tnakamura@webrtc.org
· 11 years ago
44cff54
Update version number to 3.34
by tnakamura@webrtc.org
· 11 years ago
e9bd299
Fixed implicit-int-conversion bugs.
by pbos@webrtc.org
· 11 years ago
57dbdbd
Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android.
by stefan@webrtc.org
· 11 years ago
2f02da8
Initial port of FullStackTest to new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
9c0f14d
Cleanup WebRTC tracing
by hclam@chromium.org
· 11 years ago
7e27a0f
Fixes build breakage when building WebRTC in Chromium and having include_tests=1.
by henrike@webrtc.org
· 11 years ago
b16cb05
Use scoped_ptr<> for loopback.cc
by pbos@webrtc.org
· 11 years ago
a32d18f
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
by stefan@webrtc.org
· 11 years ago
39445b0
Adding Stefan as VideoEngine owner, removing Per.
by mflodman@webrtc.org
· 11 years ago
3b89e10
Proper spacing for end-of-namespace comments.
by pbos@webrtc.org
· 11 years ago
438be80
Remove log of undefined input values in GetCodec.
by pbos@webrtc.org
· 11 years ago
50c1aef
Build all java files into jar for each module on Android
by fischman@webrtc.org
· 11 years ago
5e742a8
WebRTCViEDemo: Use global reference when passing variables across different threads
by yujie.mao@webrtc.org
· 11 years ago
c7eab28
Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target!
by fischman@webrtc.org
· 11 years ago
555f1cd
Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes.
by fischman@webrtc.org
· 11 years ago
f8f91d6
Adding a first simple version of overuse detection, but not hooked up.
by mflodman@webrtc.org
· 11 years ago
0291c80
Removed ViE file API.
by mflodman@webrtc.org
· 11 years ago
20cfda6
Remove unused multi stream bandwidth estimator.
by solenberg@webrtc.org
· 11 years ago
0f6f7cb
Enqueue packet in pacer if sending fails
by hclam@chromium.org
· 11 years ago
7533659
Fixes some pacer/padding issues found while testing.
by stefan@webrtc.org
· 11 years ago
69f7605
Wire up pacer-based padding.
by stefan@webrtc.org
· 11 years ago
b06dd93
Fix AV sync issue
by hclam@chromium.org
· 11 years ago
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