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fp2-dev
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platform
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chromium_org
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third_party
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webrtc
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f7795df4261877af9eaf982cccfafc3fc52aeb4a
f7795df
Adds a modified copy of talk/base to webrtc/base. It is the first step in
by henrike@webrtc.org
· 10 years ago
7ea3607
Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255
by bjornv@webrtc.org
· 10 years ago
fc3a6c0
Revert "FieldTrial implementation for webrtc." (rev 6089)
by andresp@webrtc.org
· 10 years ago
b16a722
Reduced kMaxSampleDiffMs (limit to 22fps).
by asapersson@webrtc.org
· 10 years ago
838c9da
Move gflags usage to video_loopback.
by pbos@webrtc.org
· 10 years ago
2efcf70
Deleting all NetEq3 files
by henrik.lundin@webrtc.org
· 10 years ago
cf1f0b0
The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.
by henrik.lundin@webrtc.org
· 10 years ago
2661819
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
by perkj@webrtc.org
· 10 years ago
4a792f0
Deleting all ACM1 files
by henrik.lundin@webrtc.org
· 10 years ago
a229768
Fix failing test introduced with r6111.
by stefan@webrtc.org
· 10 years ago
305fd94
Fixes log spam introduced with r6041.
by stefan@webrtc.org
· 10 years ago
b9c8d1a
Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
by henrike@webrtc.org
· 10 years ago
bb1e3ff
Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
8773fa6
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
9c8f347
Echo cancellation functions docs: Follow style guide w.r.t. placement of *
by kwiberg@webrtc.org
· 10 years ago
b0295bf
Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
aa169d2
One of the NetEq methods needs to be virtual.
by turaj@webrtc.org
· 10 years ago
da9b404
Modifying neteq.gyp
by turaj@webrtc.org
· 10 years ago
3a87cff
Removes parts of the webrtc::VoEHardware sub API (relanding)
by henrika@webrtc.org
· 10 years ago
28e9b66
Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..."
by henrika@webrtc.org
· 10 years ago
5c6f3fd
Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
d12d7b6
FieldTrial implementation for webrtc.
by andresp@webrtc.org
· 10 years ago
0d47fe1
Raise kViEMaxNumberOfChannels from 32 to 64
by wu@webrtc.org
· 10 years ago
569487d
Updated WebRTC version to 3.53 TBR=wu@webrtc.org
by elham@webrtc.org
· 10 years ago
8b4f539
AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_
by kwiberg@webrtc.org
· 10 years ago
60f1422
Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
by wu@webrtc.org
· 10 years ago
8539c4a
Fix odd codes in video_capture on Mac.
by braveyao@webrtc.org
· 10 years ago
4fb1a55
video_render.gypi: clean up some libraries directives to be more specific.
by fischman@webrtc.org
· 10 years ago
73c2412
Remove timestamp_extrapolator's dependency to Clock and vcm defines.
by wu@webrtc.org
· 10 years ago
8ec46c6
Allow the RTP level indicator computation to work at any sample rate.
by andrew@webrtc.org
· 10 years ago
96cccf7
Remove ALLOW_UNUSED.
by andrew@webrtc.org
· 10 years ago
5e44f56
* Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter.
by wu@webrtc.org
· 10 years ago
ebb4b94
Implement the Windows screen capturer using the Magnification API.
by jiayl@webrtc.org
· 10 years ago
7c434be
Revert 6048 "Implement the Windows screen capturer using the Mag..."
by tina.legrand@webrtc.org
· 10 years ago
6ccd081
WebRTCDemo: correct set trace filter operation.
by braveyao@webrtc.org
· 10 years ago
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
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