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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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73e1a8bb741e8e3956815f75b4977c99f0bba0db
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
bd0a216
Remove usage of webrtc trace in video processing modules.
by asapersson@webrtc.org
· 10 years ago
ea1b72d
Add ability to control peer connection constraints for the loopback test.
by andresp@webrtc.org
· 10 years ago
284f401
Remove self-assignment hacks that were added to avoid unused variable warnings.
by fischman@webrtc.org
· 10 years ago
9c31dee
Move a chatty creation log in neteq to LS_VERBOSE.
by andrew@webrtc.org
· 10 years ago
303f24f
Make Android-APK compile in release again.
by solenberg@webrtc.org
· 10 years ago
7a06daa
(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds
by henrika@webrtc.org
· 10 years ago
365b4aa
Unbreak android APK buildbots by emptying the video_capture_tests_apk target.
by fischman@webrtc.org
· 10 years ago
4e8afab
VideoCaptureAndroid: support multiple frame-rates per resolution.
by fischman@webrtc.org
· 10 years ago
523753b
Fix DesktopSize::is_empty() for the case when only width or only height is 0.
by sergeyu@chromium.org
· 10 years ago
eb90479
Move output_mixer_unittest.cc to utility_unittest.cc.
by andrew@webrtc.org
· 10 years ago
a67c9a4
VideoCaptureAndroid: stop referencing ViERenderer
by fischman@webrtc.org
· 10 years ago
fc0693b
video_capture(iOS): move stopCapture to background thread
by fischman@webrtc.org
· 10 years ago
d8b4d0f
Implement FEC support in VideoReceiveStream.
by pbos@webrtc.org
· 10 years ago
4b0cd7f
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
by andresp@webrtc.org
· 10 years ago
5406963
New NetEq test to verify correct timestamp propagation
by henrik.lundin@webrtc.org
· 10 years ago
213590d
Removed the disabling of include_tests from r2729.
by henrike@webrtc.org
· 10 years ago
ff46b81
Updated WebRTC version to 3.52 TBR=wu@webrtc.org
by elham@webrtc.org
· 10 years ago
1982636
Clean up traces and logs in RemoteBitrateEstimator.
by stefan@webrtc.org
· 10 years ago
44c9b9a
Log Fixit for parts of video_engine folder.
by mflodman@webrtc.org
· 10 years ago
4fe54a8
Fix logging calls in bitrate_controller module.
by andresp@webrtc.org
· 10 years ago
3aded9d
Remove WEBRTC_TRACE use in common_video/
by pbos@webrtc.org
· 10 years ago
7cb3251
Fix a crash in WindowCapturereMac when capture() fails.
by jiayl@webrtc.org
· 10 years ago
0115a83
Fix the library path for android 64-bit build
by michaelbai@google.com
· 10 years ago
bf4f232
Consolidate audio conversion from Channel and TransmitMixer.
by andrew@webrtc.org
· 10 years ago
0eb8ec6
Delay Estimator: Minor refactoring and added a setter function.
by bjornv@webrtc.org
· 10 years ago
3aa1ac2
Rename RTPanalyze to rtp_analyze and remove old version
by henrik.lundin@webrtc.org
· 10 years ago
c55faad
Remove AudioDevice::{Microphone,Speaker}IsAvailable.
by andrew@webrtc.org
· 10 years ago
acb49e5
This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets.
by minyue@webrtc.org
· 10 years ago
71c9ebd
Add format specification to output file names
by henrik.lundin@webrtc.org
· 10 years ago
0725df6
Extends max sample rate from 96kHz to 192kHz on the input side.
by henrika@webrtc.org
· 10 years ago
a0acb1f
sink_filter_ds.cc: add lock to Receive procedure to Pause().
by braveyao@webrtc.org
· 10 years ago
5ae01bf
Make ACM2 the default in voe_cmd_test.
by andrew@webrtc.org
· 10 years ago
15f109e
Added simulations of capacity variations and wifi recordings.
by stefan@webrtc.org
· 10 years ago
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