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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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5cc0d0b1f1c0b684223419696347624da7bb22d8
5cc0d0b
Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest.
by andrew@webrtc.org
· 10 years ago
c2b27b5
Implement the Windows screen capturer using the Magnification API.
by jiayl@webrtc.org
· 10 years ago
ba9daa7
Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test.
by henrika@webrtc.org
· 10 years ago
1a9e6ac
Pointers were not dereferenced in GetRtpStatistics.
by asapersson@webrtc.org
· 10 years ago
42fe6b3
Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels.
by stefan@webrtc.org
· 10 years ago
616cbcd
Only clamp to 16 kHz when AECM is enabled.
by andrew@webrtc.org
· 10 years ago
c2e6438
Fix constness of AudioBuffer accessors.
by andrew@webrtc.org
· 10 years ago
0638464
Fix a data race in ACM1 when audio is pulled.
by turaj@webrtc.org
· 10 years ago
976ce98
Added include of assert.h for files calling assert but missing the include.
by henrike@webrtc.org
· 10 years ago
f13f4a7
Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65
by henrike@webrtc.org
· 10 years ago
40b200b
Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord()
by henrike@webrtc.org
· 10 years ago
3d5905b
Disable failing GoogleWifiTrace3Mbps.
by pbos@webrtc.org
· 10 years ago
d592231
Disable GoogleWifiTrace3Mbps.
by pbos@webrtc.org
· 10 years ago
3848107
Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios.
by stefan@webrtc.org
· 10 years ago
c6cfc5c
Upping start bitrate to min, if set to a lower value i SetSendCodec.
by mflodman@webrtc.org
· 10 years ago
c78232f
Fix iOS assembly compile error.
by kjellander@webrtc.org
· 10 years ago
c523211
Remove neteq_unittests from Android builds
by henrik.lundin@webrtc.org
· 10 years ago
ad4cce6
Roll chromium_revision 260462:266514
by kjellander@webrtc.org
· 10 years ago
3cbb2df
Remove Version method from ACM1
by henrik.lundin@webrtc.org
· 10 years ago
dc37088
Remove ACM1 and NetEq3 related targets from modules.gyp
by henrik.lundin@webrtc.org
· 10 years ago
68a95e1
Remove AudioCodingModuleFactory
by henrik.lundin@webrtc.org
· 10 years ago
a48f3c2
Add clock to ACM config struct
by henrik.lundin@webrtc.org
· 10 years ago
db395e4
AEC: Startup phase only runs if reported_delay_enabled
by bjornv@webrtc.org
· 10 years ago
be039c2
Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash.
by fischman@webrtc.org
· 10 years ago
b4945d1
APM: limit native sample rate to 16kHz on mobile.
by fischman@webrtc.org
· 10 years ago
93d270f
Using realpath instead of android_src in Android webview
by michaelbai@google.com
· 10 years ago
a2d989b
Only download the VS toolchain if DEPOT_TOOLS_WIN_TOOLCHAIN=1.
by andrew@webrtc.org
· 10 years ago
c54ff69
Add thread annotations to Call API.
by pbos@webrtc.org
· 10 years ago
267637b
Disable flaky CaptureNtpTimeWithNetworkJitter.
by pbos@webrtc.org
· 10 years ago
0e098e0
AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h
by bjornv@webrtc.org
· 10 years ago
676638c
Disable capture test for FrameRate on Windows.
by pbos@webrtc.org
· 10 years ago
bb62a93
Introduce a config struct for AudioCoding module
by henrik.lundin@webrtc.org
· 10 years ago
c298835
Disabling flaky CanReceiveFec.
by pbos@webrtc.org
· 10 years ago
73e1a8b
Disable flaky RunsRtpRtcpTestWIthoutErrors.
by pbos@webrtc.org
· 10 years ago
abf78cc
Fix the NetEq build
by henrik.lundin@webrtc.org
· 10 years ago
75d1487
Include buffer size limits in NetEq config struct
by henrik.lundin@webrtc.org
· 10 years ago
a714643
Add henrik.lundin as owner in AudioCoding module
by henrik.lundin@webrtc.org
· 10 years ago
b0079ed
Replace scoped_array<T> with scoped_ptr<T[]>.
by andrew@webrtc.org
· 10 years ago
4820f6b
Fix leak in remote bitrate estimator tests introduced in r5980
by stefan@webrtc.org
· 10 years ago
8c4135e
Support for simulating multiple independent flows in a network.
by stefan@webrtc.org
· 10 years ago
0a5fd54
Casting char to int in logs.
by asapersson@webrtc.org
· 10 years ago
85d90de
Returns a NULL frame on all platforms if the captured window is closed.
by jiayl@webrtc.org
· 10 years ago
b991cd0
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
by wu@webrtc.org
· 10 years ago
0061d86
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
by wu@webrtc.org
· 10 years ago
ee6695b
Add an output capacity parameter to ACMResampler::Resample10Msec()
by henrik.lundin@webrtc.org
· 10 years ago
fbf2568
Add keyboard channel support to AudioBuffer.
by andrew@webrtc.org
· 10 years ago
86e3fa8
Fix the Android compilation (better structure for NetEq test libs)
by henrik.lundin@webrtc.org
· 10 years ago
dbebc39
Remove TraceCallback use from Call.
by pbos@webrtc.org
· 10 years ago
9d0f79f
Rename Start/Stop in Video{Send,Receive}Streams.
by pbos@webrtc.org
· 10 years ago
e846663
Fixing a bug in ACM2 where the output frame energy was incorrectly set
by henrik.lundin@webrtc.org
· 10 years ago
757a92f
Use unique filenames in AudioProcessingTests for parallelization.
by andrew@webrtc.org
· 10 years ago
e1b0595
AEC: Adds a reported_delay_enabled_ flag
by bjornv@webrtc.org
· 10 years ago
110a2d2
Remove ACM1/ACM2 switching from VoiceEngine tests
by henrik.lundin@webrtc.org
· 10 years ago
3ab5093
Restore sample_rate_hz() until Chromium is updated to not use it.
by andrew@webrtc.org
· 10 years ago
2e24460
Support arbitrary input/output rates and downmixing in AudioProcessing.
by andrew@webrtc.org
· 10 years ago
8b4811b
Reland "Stop using ACM factory in VoiceEngine"
by henrik.lundin@webrtc.org
· 10 years ago
79a6030
Remove 44.1 kHz workaround from the iOS AudioDevice.
by andrew@webrtc.org
· 10 years ago
0f437b0
Fix a bug in AcmReceiver::NetworkStatistics
by henrik.lundin@webrtc.org
· 10 years ago
a19bee3
Revert "Stop using ACM factory in VoiceEngine"
by henrik.lundin@webrtc.org
· 10 years ago
a61127d
Stop using ACM factory in VoiceEngine
by henrik.lundin@webrtc.org
· 10 years ago
69b14d5
Let A/V sync test use default AudioCoding module
by henrik.lundin@webrtc.org
· 10 years ago
68bd1f3
Create ACM2 instance when calling AudioCodingModule::Create
by henrik.lundin@webrtc.org
· 10 years ago
13f9d37
Reland "Make VoiceEngine choose ACM2 by default""
by henrik.lundin@webrtc.org
· 10 years ago
17d096a
audio_processing: DestroyHandle() now returns void
by bjornv@webrtc.org
· 10 years ago
fb54df6
common_audio: VADFree() now returns void
by bjornv@webrtc.org
· 10 years ago
cf526f7
Resampler modifications in preparation for arbitrary audioproc rates.
by andrew@webrtc.org
· 10 years ago
11720c2
Fix multi-monitor support in the screen capturer for Mac.
by sergeyu@chromium.org
· 10 years ago
5fd5020
Revert r5937 "Fix multi-monitor support in the screen capturer for Mac."
by sergeyu@chromium.org
· 10 years ago
a4fbfd9
Add Chromium's ScopedVector.
by andrew@webrtc.org
· 10 years ago
a73081a
Fix multi-monitor support in the screen capturer for Mac.
by sergeyu@chromium.org
· 10 years ago
bc6b15d
Fix iSAC/48000 issue with ACM2.
by turaj@webrtc.org
· 10 years ago
499ee5e
WebRtcAecm_Process: Reduce code duplication
by kwiberg@webrtc.org
· 10 years ago
2991a30
StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16
by kwiberg@webrtc.org
· 10 years ago
514abde
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
3ea24b2
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
14c5e8a
Revert "Make VoiceEngine choose ACM2 by default"
by henrik.lundin@webrtc.org
· 10 years ago
722cd19
Removing AudioCoding duplicate tests
by henrik.lundin@webrtc.org
· 10 years ago
4f9c08f
Make VoiceEngine choose ACM2 by default
by henrik.lundin@webrtc.org
· 10 years ago
db4b867
Fix crashes due to dangling external decoder pointer.
by fischman@webrtc.org
· 10 years ago
988e753
Set include_internal_video_capture=1 for video_capture_tests
by kjellander@webrtc.org
· 10 years ago
1857d7e
Re-enable AGC tests:
by aluebs@webrtc.org
· 10 years ago
32e7755
Remove use of tmpnam.
by kjellander@webrtc.org
· 10 years ago
566af28
Replace flooding logs in rtp_sender.cc with a comment.
by andrew@webrtc.org
· 10 years ago
a738ae3
iOS: baby steps to being able to include_tests=1
by fischman@webrtc.org
· 10 years ago
0b559b6
Moved voe_neteq_stats_unittest to audio_coding_module_unittest
by henrik.lundin@webrtc.org
· 10 years ago
a1626fe
Propagate capture ntp timestamp from rtp to renderer.
by wu@webrtc.org
· 10 years ago
aee97d8
Check if a header extension is registered before updating it and fail silently if it's not.
by stefan@webrtc.org
· 10 years ago
f6d791d
Make WebRTC Android examples build without sourcing envsetup.sh
by kjellander@webrtc.org
· 10 years ago
6b1114a
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
by fischman@webrtc.org
· 10 years ago
633c598
Adding a config struct to NetEq
by henrik.lundin@webrtc.org
· 10 years ago
fd59b22
New Packet and PacketSource classes for NetEq tests
by henrik.lundin@webrtc.org
· 10 years ago
966744e
Fix gyp for video_capture/ensure_initialized.cc.
by primiano@chromium.org
· 10 years ago
290c5a5
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
e338cc2
Added a new OnMoreData() interface which will not feed the playout data to APM.
by xians@webrtc.org
· 10 years ago
538aff6
Fix the captured screen rect conversion.
by jiayl@webrtc.org
· 10 years ago
d399a50
NetEq changes.
by turaj@webrtc.org
· 10 years ago
b18bff5
Cleaned up logging in video_coding.
by stefan@webrtc.org
· 10 years ago
28d1b61
Convert WEBRTC_TRACE to LOG in utility.
by asapersson@webrtc.org
· 10 years ago
19ca463
Disable UsesTraceCallback
by pbos@webrtc.org
· 10 years ago
3841668
Fix loopback test for case where no constraint is given.
by andresp@webrtc.org
· 10 years ago
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