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gerrit-public.fairphone.software
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platform
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external
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webrtc
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354becf4df807917b698a626a3c649c39eb4c763
354becf
Remove Git ignore exclusion of .sha1 files
by kjellander@webrtc.org
· 10 years ago
7cc92aa
Use WebRtcVideoRenderFrame for texture frames.
by pbos@webrtc.org
· 10 years ago
62f6e75
Refactoring WebRTC Java/JNI audio recording in C++ and Java.
by henrika@webrtc.org
· 10 years ago
c2d0473
Switch to using AudioEncoderPcm16B instead of ACMPCM16B
by henrik.lundin@webrtc.org
· 10 years ago
f58fe0a
Rename GYP and GN targets for video capture+render.
by kjellander@webrtc.org
· 10 years ago
2c29c2e
C++ readability review for ajm.
by andrew@webrtc.org
· 10 years ago
5d60895
Fix bug when there are no blocks in a chunk in Beamformer
by aluebs@webrtc.org
· 10 years ago
bc35703
Add a method to remove an existing renderer from the internal list of Android renderers.
by glaznev@webrtc.org
· 10 years ago
bc40324
Merge fixes and changed for Android AppRTCDemo from internal repo.
by glaznev@webrtc.org
· 10 years ago
d35a5c3
Make ChannelBuffer aware of frequency bands
by aluebs@webrtc.org
· 10 years ago
d7472b5
base/arraysize.h: We use size_t, so need to include stddef.h
by kwiberg@webrtc.org
· 10 years ago
91ba79a
Make sure that the norms are positive in Beamformer
by aluebs@webrtc.org
· 10 years ago
b6856d2
Apply mask smoothing in Beamformer
by aluebs@webrtc.org
· 10 years ago
8da96ac
Switch to using AudioEncoderIlbc instead of ACMILBC
by henrik.lundin@webrtc.org
· 10 years ago
1a072f9
Address comments from previous review round for rtc::Event.
by tommi@webrtc.org
· 10 years ago
f4c10d2
Always use DeliverI420Frame in WebRtcVideoEngine.
by pbos@webrtc.org
· 10 years ago
027e113
Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig.
by stefan@webrtc.org
· 10 years ago
30015e3
Fix bug in EventPosix where we'd miss a set event.
by tommi@webrtc.org
· 10 years ago
648f5d6
pcm16b: Make input arrays const and use uint8_t[] for byte arrays
by kwiberg@webrtc.org
· 10 years ago
948d617
Create a separate thread for pacing.
by mflodman@webrtc.org
· 10 years ago
c11348b
Fixing a bug in expand_rate calculation for stereo signal.
by minyue@webrtc.org
· 10 years ago
8e612ab
Remove voice_engine_ member variable and GetVoiceEngine() from ViEChannelManager.
by tommi@webrtc.org
· 10 years ago
5b8f3e0
Roll chromium_revision 598c3e9..601e6f3
by kjellander@webrtc.org
· 10 years ago
44ae4c8
Support using VP9 video codec in AppRTCDemo.
by glaznev@webrtc.org
· 10 years ago
f7e6cfd
Add CHECK to EventWrapper to see if there's a subtle bug there or not.
by tommi@webrtc.org
· 10 years ago
669bc7e
Modify default field trial implementation to allow
by glaznev@webrtc.org
· 10 years ago
11c5db0
Revert 8273 "Temporarily change ThreadPosix to CHECK (crash) if ..."
by tommi@webrtc.org
· 10 years ago
0d852d5
Use VideoReceiveStream as an ExternalRenderer.
by pbos@webrtc.org
· 10 years ago
d6e25a5
Revert r8297 "Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig."
by stefan@webrtc.org
· 10 years ago
03c1c10
Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig.
by stefan@webrtc.org
· 10 years ago
53d9012
Clean kForever from basictypes and move it to the interfaces that actually have it.
by andresp@webrtc.org
· 10 years ago
e01bae2
Fixing a nit
by henrik.lundin@webrtc.org
· 10 years ago
1c6239a
G711: Make input arrays const and use uint8_t[] for byte arrays
by kwiberg@webrtc.org
· 10 years ago
d0165c6
Use a manual reset event in PosixThread.
by tommi@webrtc.org
· 10 years ago
4c0fd96
Move rtc::Event to rtc_base_approved. We need an event implementation in WebRTC that allows us to specify whether it's manually reset or automatically. EventWrapper currently doesn't support it and it adds a heap allocation + vtable, so rtc::Event is the lighter of the two.
by tommi@webrtc.org
· 10 years ago
8cf9bdb
Remove USE_WEBRTC_DEV_BRANCH.
by pbos@webrtc.org
· 10 years ago
2b69eab
Restructure GYP for vp9, opus and direct trace
by kjellander@webrtc.org
· 10 years ago
f31f56d
Remove default arguments in EncodedImageCallback.
by changbin.shao@webrtc.org
· 10 years ago
6c930c7
Cleanup: unify rotation to be enum based instead of int for degree.
by guoweis@webrtc.org
· 10 years ago
7a57f8f
Reland 8203 "Reducing locking in OveruseFrameDetect..."
by tommi@webrtc.org
· 10 years ago
103f328
Fix the binary layout of ProcessThreadImpl.
by tommi@webrtc.org
· 10 years ago
ec499be
Increase testclient timeout from 1 to 5 seconds
by jlmiller@webrtc.org
· 10 years ago
fe19699
Revert 8260 "Base RWLockWrapper on rtc::SharedExclusiveLock."
by tommi@webrtc.org
· 10 years ago
2eb1660
Switch ThreadCheckerImpl over to using PlatformThreadRef.
by tommi@webrtc.org
· 10 years ago
2bf0e90
Revert 8275 "This CL adds an API to the SSL stream adapters and ..."
by tommi@webrtc.org
· 10 years ago
1d4830a
Disable ProcessThread tests that are dependent on timing.
by tommi@webrtc.org
· 10 years ago
95a32ec
Revert 8271 "VirtualSocketServer out-of-order issue with closing..."
by bjornv@webrtc.org
· 10 years ago
2a44be9
Normalize delay-and-sum mask in Beamformer
by aluebs@webrtc.org
· 10 years ago
799e667
Add high frequency correction to Beamformer
by aluebs@webrtc.org
· 10 years ago
0c7ec77
Cleanup: unify rotation to be enum based instead of int for degree.
by guoweis@webrtc.org
· 10 years ago
110443a
Cleanup: unify rotation to be enum based instead of int for degree.
by guoweis@webrtc.org
· 10 years ago
1d11c82
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
by pthatcher@webrtc.org
· 10 years ago
63da1dd
audio_processing: Now records mic volume level also when using new AGC
by bjornv@webrtc.org
· 10 years ago
ccd7e99
Temporarily change ThreadPosix to CHECK (crash) if we ever spend more than 30 seconds waiting for thread shutdown. There are cases on build bots where it looks like we're hitting this problem, but reproducing locally has been a struggle.
by tommi@webrtc.org
· 10 years ago
13a0e18
Temporarily disable a couple of ThreadChecker tests on Mac.
by tommi@webrtc.org
· 10 years ago
4770437
VirtualSocketServer out-of-order issue with closing TCP sockets
by pthatcher@webrtc.org
· 10 years ago
9baa9ca
Add libjingle_peerconnection_so.so to Java test dependencies.
by perkj@webrtc.org
· 10 years ago
b5a1252
Hack to work around the current issues with rolling WebRTC into chromium.
by tommi@webrtc.org
· 10 years ago
751a365
Switch to using AudioEncoderPcmU/A instead of ACMPCMU/A
by henrik.lundin@webrtc.org
· 10 years ago
02270cd
Implementing a packet router class, used to route RTP packets to the
by mflodman@webrtc.org
· 10 years ago
10a9e92
Fix delete of stack allocated object causing test crashes.
by stefan@webrtc.org
· 10 years ago
4b320cf
Revert "Cleanup: unify rotation to be enum based instead of int for degree."
by magjed@webrtc.org
· 10 years ago
fb609a1
Wire up new feedback format by introducing a FeedbackPacket type.
by stefan@webrtc.org
· 10 years ago
353c8b8
audio_processing/agc: Changed to correct include path in agc_unittests
by bjornv@webrtc.org
· 10 years ago
bc3241a
Update ProcessCallAfterXms to better match the performance of our faster bots. Previously I had made sure these tests didn't flake out on our slow trybots, but apparently I need to do the same for the fast bots :)
by tommi@webrtc.org
· 10 years ago
0c3e12b
Revamp the ProcessThreadImpl implementation.
by tommi@webrtc.org
· 10 years ago
7502543
Base RWLockWrapper on rtc::SharedExclusiveLock.
by pbos@webrtc.org
· 10 years ago
5e05731
Roll chromium_revision cd35af6..598c3e9
by kjellander@webrtc.org
· 10 years ago
57ac2c8
Default destination used by c line should be IPv4 only to avoid parsing error in legacy client.
by guoweis@webrtc.org
· 10 years ago
3e733a4
Cleanup: unify rotation to be enum based instead of int for degree.
by guoweis@webrtc.org
· 10 years ago
74d2788
Remove defined(__cplusplus) tests in C++ code.
by jan.skoglund@webrtc.org
· 10 years ago
f45c8ca
Reland r8248 "Introduce ACMGenericCodecWrapper"
by henrik.lundin@webrtc.org
· 10 years ago
ec4521c
Clean up Beamformer initialization
by aluebs@webrtc.org
· 10 years ago
e69220c
Fix the value of the first byte of nal unit generated by fake H.264 encoder.
by glaznev@webrtc.org
· 10 years ago
f693229
Fix Android video renderer to support video frames with stride > width.
by glaznev@webrtc.org
· 10 years ago
cc64a9c
voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric
by bjornv@webrtc.org
· 10 years ago
4b9622f
Roll gtest-parallel.
by pbos@webrtc.org
· 10 years ago
3a87630
Revert r8248 "Introduce ACMGenericCodecWrapper"
by henrik.lundin@webrtc.org
· 10 years ago
af8c13f
Introduce ACMGenericCodecWrapper
by henrik.lundin@webrtc.org
· 10 years ago
5d32f43
Disable CondVarTest.InitFunctionsWork.
by tommi@webrtc.org
· 10 years ago
877ac76
Cleanup and prepare for bundling.
by pthatcher@webrtc.org
· 10 years ago
cf7efeb
Add new AudioEncoderOpusTest
by henrik.lundin@webrtc.org
· 10 years ago
520a69e
Revert 8238 "Add RefCounting for TransportProxies"
by bjornv@webrtc.org
· 10 years ago
875c97e
Remove SetNotAlive method from the thread class.
by tommi@webrtc.org
· 10 years ago
c5f6971
Revert 8237 "Cleanup and prepare for bundling."
by bjornv@webrtc.org
· 10 years ago
dc096f2
system_wrappers: Disabled flaky test CondVarTest.PassBatonMultipleTimes
by bjornv@webrtc.org
· 10 years ago
4414939
Add method for incrementing RtpPacketCounter. Removes duplicate code.
by asapersson@webrtc.org
· 10 years ago
e250667
Add RefCounting for TransportProxies
by decurtis@webrtc.org
· 10 years ago
af01d93
Cleanup and prepare for bundling.
by pthatcher@webrtc.org
· 10 years ago
322a564
Fix datachannel stats id and timestamp.
by decurtis@webrtc.org
· 10 years ago
d43bdf5
Rewrite ThreadPosix.
by tommi@webrtc.org
· 10 years ago
bfdee69
Roll chromium_revision 9070a80..cd35af6 (313233:314322)
by kjellander@webrtc.org
· 10 years ago
0ec50be
Changing include guard in frame_callback.h.
by mflodman@webrtc.org
· 10 years ago
200ac00
Remove temp files in audio_processing_unittest.cc.
by pbos@webrtc.org
· 10 years ago
0e8bf6c
Enable bitrate probing by default.
by stefan@webrtc.org
· 10 years ago
b1786db
audio_processing: Added a new AEC delay metric value that gives the amount of poor delays
by bjornv@webrtc.org
· 10 years ago
0e81fdf
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
by pkasting@chromium.org
· 10 years ago
19f3f71
Fix apparent typo: int -> char.
by pkasting@chromium.org
· 10 years ago
946ad76
Switched lists of packets to lists of packet pointers. Allows Packet polymorphism.
by stefan@webrtc.org
· 10 years ago
c957ffc
Fixed potential crash if rtp packet history is completely full.
by sprang@webrtc.org
· 10 years ago
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