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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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0e9c399746f45ceaf46f12b11ba93c09cca0c2bb
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modules
0e9c399
NetEq4: Removing templatization for AudioMultiVector
by henrik.lundin@webrtc.org
· 11 years ago
24f0702
Support for CELT in NetEq4.
by turaj@webrtc.org
· 11 years ago
d4e1329
Remove include_dirs from video_render.
by pbos@webrtc.org
· 11 years ago
76a6ffb
Remove include_dirs from video_capture.
by pbos@webrtc.org
· 11 years ago
0d4d51b
Revert 4876 "Support for CELT in NetEq4."
by tina.legrand@webrtc.org
· 11 years ago
76238f6
Propagate AutoMuter interface out to VideoCodingModule
by henrik.lundin@webrtc.org
· 11 years ago
cd5c882
Support for CELT in NetEq4.
by turaj@webrtc.org
· 11 years ago
2b35b95
Change the parameters of calculating maximum decode time.
by wuchengli@chromium.org
· 11 years ago
9e035d2
Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from
by wu@webrtc.org
· 11 years ago
b503d1e
Only use -lm on Linux in ISAC.
by andrew@webrtc.org
· 11 years ago
424e0e4
With ACM2 and NetEq4, VoE fuzz test very often fails.
by minyue@webrtc.org
· 11 years ago
44f030c
Implemented AutoMuter in MediaOptimization
by henrik.lundin@webrtc.org
· 11 years ago
dc1f7e9
Remove include_dirs from pacing.
by pbos@webrtc.org
· 11 years ago
ee817d3
Remove include_dirs from remote_bitrate_estimator.
by pbos@webrtc.org
· 11 years ago
fc75214
Remove include_dirs from bitrate_controller.
by pbos@webrtc.org
· 11 years ago
1fc4659
Remove include_dirs from video_coding.
by pbos@webrtc.org
· 11 years ago
85592ad
Remove include_dirs from video_processing.
by pbos@webrtc.org
· 11 years ago
1800406
Remove include_dirs from rtp_rtcp.
by pbos@webrtc.org
· 11 years ago
2f0a942
Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3.
by turaj@webrtc.org
· 11 years ago
d4f6789
Move the Config DelayCorrection struct to audio_processing.h.
by andrew@webrtc.org
· 11 years ago
8ddec2c
Add an extended filter mode to AEC.
by andrew@webrtc.org
· 11 years ago
53fdd3b
Fix WindowCapturerWin to capture window decorations after window size changes.
by sergeyu@chromium.org
· 11 years ago
605daf0
Disable a NetEq unittest on Android. The test tries to register iSAC-swb as send codec and fails.
by turaj@webrtc.org
· 11 years ago
72790c7
Remove unused constants, so chrome can enable a warning for that. Patch from thakis@
by niklas.enbom@webrtc.org
· 11 years ago
7f35836
Re-enable verbose logging in NetEq4.
by turaj@webrtc.org
· 11 years ago
79c884c
Convert DeviceInfoImpl::_captureCapabilities from a map to a vector.
by fischman@webrtc.org
· 11 years ago
99b6d9e
Revert 4837 "Add an extended filter mode to AEC."
by asapersson@webrtc.org
· 11 years ago
83c5f62
Add an extended filter mode to AEC.
by andrew@webrtc.org
· 11 years ago
933267f
Small fixes to run ACM2 tests.
by turaj@webrtc.org
· 11 years ago
6ca9e7d
API add to set background noise mode.
by turaj@webrtc.org
· 11 years ago
08099e0
Fix window capturer not to leak HDC.
by sergeyu@chromium.org
· 11 years ago
82707bf
Fix window capturer to stop capturing when the target is minimized.
by sergeyu@chromium.org
· 11 years ago
4b067da
Disable some VP8 tests on Android.
by andrew@webrtc.org
· 11 years ago
da6d2a2
MediaOptimization: Converting a few members to scoped_ptrs
by henrik.lundin@webrtc.org
· 11 years ago
510ee1b
Remove deprecated AudioCodingModule::Destroy.
by andrew@webrtc.org
· 11 years ago
a6665e7
Reformatting media_optimization.cc and .h
by henrik.lundin@webrtc.org
· 11 years ago
36441e3
Re-enable VideoCaptureTest.CreateDelete
by fischman@webrtc.org
· 11 years ago
84afa19
Adding unit tests for default temporal layer strategy.
by andresp@webrtc.org
· 11 years ago
2529558
Enable SetInitialPlayoutDelay on Android.
by dwkang@webrtc.org
· 11 years ago
d1fe828
Fix bugs in DesktopRegion::Subtract().
by sergeyu@chromium.org
· 11 years ago
717267a
VAD changes ported to ACM2.
by turaj@webrtc.org
· 11 years ago
045e45e
Address Windows 64-bits warnings.
by turaj@webrtc.org
· 11 years ago
54f0246
Disable flaky video capture test.
by stefan@webrtc.org
· 11 years ago
51d53aa
Avoid recursively taking critical section.
by stefan@webrtc.org
· 11 years ago
7ab577d
Use link_settings instead of all_dependent_settings to pacify xcode gyp generator
by fischman@webrtc.org
· 11 years ago
6876512
Roll webrtc's chromium_revision 217707:224141
by fischman@webrtc.org
· 11 years ago
f5013c0
Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct
by tina.legrand@webrtc.org
· 11 years ago
4d08199
Compile ACM2 and ACM1.
by turaj@webrtc.org
· 11 years ago
80142aa
Small refactoring of AudioProcessing use in channel.cc.
by andrew@webrtc.org
· 11 years ago
ab34f11
NetEq4: Making a few more members scoped_ptrs
by henrik.lundin@webrtc.org
· 11 years ago
05dd6c0
Dedicated speed test for NetEq3
by henrik.lundin@webrtc.org
· 11 years ago
ec09fcb
Revert r4772 "Compile ACM1 and ACM2."
by stefan@webrtc.org
· 11 years ago
671d90b
NetEq4: Make some DSP operation classes member variables
by henrik.lundin@webrtc.org
· 11 years ago
c2c8e6a
Fix races in vcm::Process().
by stefan@webrtc.org
· 11 years ago
5b7878f
Changing 'frame' method to 'bounds' method.
by sjlee@webrtc.org
· 11 years ago
7556d2d
Compile ACM1 and ACM2.
by turaj@webrtc.org
· 11 years ago
0c57671
Use the native sample rate for OpenSL recording.
by henrike@webrtc.org
· 11 years ago
0277aa4
Fix typo in r4765.
by pbos@webrtc.org
· 11 years ago
54bc776
Fix dangling pointer _encoder in video_sender.cc.
by pbos@webrtc.org
· 11 years ago
64b5c61
Initialize CodecInst structs in test_api_audio.cc.
by pbos@webrtc.org
· 11 years ago
79d3355
Dedicated speed test for NetEq4
by henrik.lundin@webrtc.org
· 11 years ago
7485573
Add support for multiple report blocks.
by stefan@webrtc.org
· 11 years ago
e9d2898
This is related to https://code.google.com/p/webrtc/issues/detail?id=1341
by sjlee@webrtc.org
· 11 years ago
e3a12da
This is related to https://code.google.com/p/webrtc/issues/detail?id=846
by sjlee@webrtc.org
· 11 years ago
e8fdc9d
Split video coding module unit tests into sender and receiver unit tests.
by andresp@webrtc.org
· 11 years ago
36c3652
Remove use of vcm->ResetDecoder from modules/utility.
by andresp@webrtc.org
· 11 years ago
42a65a2
Split VideoCodingModuleImpl into VideoSender and VideoReceiver.
by andresp@webrtc.org
· 11 years ago
ed0b4fb
Prepare to compile ACM1 and ACM2.
by turaj@webrtc.org
· 11 years ago
26251da
Implement DesktopRegion subtraction.
by sergeyu@chromium.org
· 11 years ago
a26a7f6
Moving test-only code (stream_generator) out of vcm implemention.
by andresp@webrtc.org
· 11 years ago
388d16c
Fix win trybot errors due to r4729.
by andrew@webrtc.org
· 11 years ago
d0737d9
Fix crash in the window capturer on windows
by sergeyu@chromium.org
· 11 years ago
3f39c00
ACM2 integration with NetEq 4.
by turaj@webrtc.org
· 11 years ago
a3351c4
Adding Ami to the video renderer and capturer modules.
by mallinath@webrtc.org
· 11 years ago
bc375b5
The video render module for iOS.
by fischman@webrtc.org
· 11 years ago
4489c51
This issue is related to https://chromereviews.googleplex.com/9908014/
by minyue@webrtc.org
· 11 years ago
5e3379e
Make the destructor of AudioCodingModule public.
by andrew@webrtc.org
· 11 years ago
0fd885e
Fix unsigned/signed comparison error due to r4729.
by andrew@webrtc.org
· 11 years ago
f5556f2
Reduce frequency of high audio delay warning logs.
by andrew@webrtc.org
· 11 years ago
9fea95a
Removes function that is not used anywhere but somehow still causing library load issues on Android Release build.
by henrike@webrtc.org
· 11 years ago
8fdce8e
OpenSl: fixes crashes externally reported in issue 2361 and 2362.
by henrike@webrtc.org
· 11 years ago
66dbbd9
Adding APIs. These APIs are not implemented yet, they are to help developement of ACM.
by turaj@webrtc.org
· 11 years ago
f2982c9
Remove FrameForStorage:Follow up on r4688
by mikhal@webrtc.org
· 11 years ago
f0adedc
Reset jitter buffer and timing if frames are getting too much delay.
by stefan@webrtc.org
· 11 years ago
054bc03
Remove repeated conditions key.
by andrew@webrtc.org
· 11 years ago
f46fff6
OpenSL (not default): Enables low latency audio on Android.
by henrike@webrtc.org
· 11 years ago
b676ac7
Lock RTPSender statistics.
by pbos@webrtc.org
· 11 years ago
5cf83f4
Remove redundant STR_CASE_CMP macro definitions.
by andrew@webrtc.org
· 11 years ago
6b4698e
Lock use of _packetRequestCallback in VCM.
by pbos@webrtc.org
· 11 years ago
4e7777b
Break out RTCPSender dependency on ModuleRtpRtcpImpl.
by pbos@webrtc.org
· 11 years ago
6a79c9f
Re-enable tests for Remote Bitrate Estimator
by solenberg@webrtc.org
· 11 years ago
618a0ec
ExternalVideoDecoder for new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
e97b69f
Handle empty RTP video packets agnostic to codec.
by pbos@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
4014302
Add temporal layer factory.
by andresp@webrtc.org
· 11 years ago
31a8ce7
Removing FrameForStorage
by mikhal@webrtc.org
· 11 years ago
f2ef20c
Pre-multiply images for MouseCursorShape.
by alexeypa@chromium.org
· 11 years ago
6f458ed
Recognize armv7 target_arch for ios support in webrtc common.gyp
by fischman@webrtc.org
· 11 years ago
06eaa54
Restore severity precondition to logging.h.
by andrew@webrtc.org
· 11 years ago
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