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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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16f55a10c4e73e46a58eb45587841914e6b0d851
16f55a1
Migrated GN target :g711_test
by aleloi
· 8 years ago
649a21a
Disable RampUpTest.UpDownUpThreeStreams.
by philipel
· 8 years ago
2e48646
RTC_CHECK and RTC_DCHECK macros for C
by kwiberg
· 8 years ago
7924697
Refactor WebRtcVideoCapturer.
by nisse
· 8 years ago
d8dd190
GN: Fix test_support_unittests and MIPS compile issue.
by kjellander
· 8 years ago
84c03ba
Add rtc_media as a direct dependency of rtc_media_unittests.
by nisse
· 8 years ago
0d1ad32
Add histogram for percentage of incoming frames that are limited in resolution due to cpu:
by asapersson
· 8 years ago
14cf12b
Fixing TSan data race warning in video end-to-end tests.
by Taylor Brandstetter
· 8 years ago
23d947d
Some cleanup in BaseChannel RTCP mux code.
by deadbeef
· 8 years ago
b3f1c5d
Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine
by henrik.lundin
· 8 years ago
e131ea5
Adding deadbeef and honghaiz as owners of webrtc/pc.
by deadbeef
· 8 years ago
72a5645
Removed the deactivation of the level controller when there is a built-in AGC available
by peah
· 8 years ago
8c16520
Method to parse event log directly from a string.
by terelius
· 8 years ago
6c46eaa
Add gtest as a dependency for neteq_quality_test_support.
by ehmaldonado
· 8 years ago
d48717b
Fix issue where the number of packets reported in tests/simulations sometimes are negative.
by stefan
· 8 years ago
4ec01d9
Fix trivial lint errors in FileRecorder and FilePlayer
by kwiberg
· 8 years ago
853ecb2
Style cleanup in UpdateTmmbr:
by danilchap
· 8 years ago
7f82fc9
WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left shift of 1 by 31 overflows)
by kwiberg
· 8 years ago
642e3bc
[rtcp] TransportFeedback adjusted to match other rtcp packets:
by danilchap
· 8 years ago
4981051
[Reland] Cleanup of the AudioDeviceBuffer class.
by henrika
· 8 years ago
83d79cd
Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ )
by kjellander
· 8 years ago
4381700
WebRtcVideoFrame constructor without transport_frame_id.
by nisse
· 8 years ago
e5b4141
Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData
by danilchap
· 8 years ago
ff101d6
iOS: add PlistBuddy location to path to avoid build errors
by vopatop.skam
· 8 years ago
94b9199
Add a copy of gyp_flag_compare from Chromium to WebRTC's webrtc/tools.
by ehmaldonado
· 8 years ago
4905f06
Disable the software noise suppressor on iOS devices as that
by peah
· 8 years ago
abcc3de
Add pps id and sps id parsing to the h.264 depacketizer.
by stefan
· 8 years ago
86ccd7b
Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ )
by sakal
· 8 years ago
a7a01df
Add field_trial_default dependency to libjingle_peerconnection
by arlolra
· 8 years ago
8177452
iOS H264VideoToolBoxEncoder: Stop scaling native CVPixelBuffers
by magjed
· 8 years ago
d7a89db
Revert of Cleanup of the AudioDeviceBuffer class (patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
by henrika
· 8 years ago
cf327b4
Cleanup of the AudioDeviceBuffer class.
by henrika
· 8 years ago
da161d7
Reformat rtcp_receiver git cl format --full
by danilchap
· 8 years ago
861da3c
Refactor neteq_test_support.
by ehmaldonado
· 8 years ago
294fb05
Add a timeout for starting the camera on CameraCapturer.
by sakal
· 8 years ago
bcba64a
GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets.
by ehmaldonado
· 8 years ago
4a85abb
Add support for more resolutions on iOS/macOS
by kthelgason
· 8 years ago
ec5c906
GN: Fix errors when some variables are set to non-default values.
by kjellander
· 8 years ago
72333d2
Add kjellander@webrtc.org to more BUILD.gn OWNERS files.
by kjellander
· 8 years ago
96b6b83
iOS: add type to peer connection local streams
by vopatop.skam
· 8 years ago
c21560b
Remove pbos@webrtc.org from autoroll TBRs.
by Peter Boström
· 8 years ago
9b5306c
Adding test for unordered, fragmented SCTP message delivery.
by Taylor Brandstetter
· 8 years ago
b5b3090
Corrected the testvectors for the level controller
by peah
· 8 years ago
8df4d0e
Add playout_delay_oracle_unittest as gn target
by isheriff
· 8 years ago
3a11933
Remove audio_device_test_func.
by maxmorin
· 8 years ago
644fa96
Added recording of the configuration for the AudioFrame API call
by peah
· 8 years ago
7320866
Reland of Adding audio to video_quality_test.
by minyue
· 8 years ago
2b61639
Remove TMMBRSet class
by danilchap
· 8 years ago
e1f5b4a
voice_engine: Removed old variants of Channel constructor and CreateChannel
by ossu
· 8 years ago
38d840c
NetEq: Changing checked_cast to saturated_cast
by henrik.lundin
· 8 years ago
96bbdd5
WebRtcSpl_SynthesisQMF: Fix UBSan fuzzer bug (left shift of negative value)
by kwiberg
· 8 years ago
e9a6acf
Added missing unittest to the modules/BUILD.gn build file
by peah
· 8 years ago
cb2d701
Add kjellander as BUILD.gn OWNER in webrtc/modules
by kjellander
· 8 years ago
71fead2
Reland of StartTimestamp generated randomly in RtpSender constructor (patchset #1 id:1 of https://codereview.webrtc.org/2248413002/ )
by danilchap
· 8 years ago
d4e9f62
Updated AudioDecoderFactory to list AudioCodecSpecs instead of SdpAudioFormats.
by ossu
· 8 years ago
235020d
Roll chromium_revision 915e47250f..e3860bd297 (412201:412289)
by magjed
· 8 years ago
010f092
GN: Add Android support to video_engine_tests.
by sakal
· 8 years ago
fd16da2
Do not switch to a high-cost connection that is not receiving.
by Honghai Zhang
· 8 years ago
41a3287
Nil out EAGLContext explicitly on RTCEAGLVideoView dealloc.
by tkchin
· 8 years ago
869dab7
Disable Intel VP8 HW encoder.
by Alex Glaznev
· 8 years ago
6a35590
Add code for dummy file audio to fallback to dummy audio.
by noahric
· 8 years ago
7c0f8ee
Avoid null pointer exception if Android getCameraInfo fails.
by Alex Glaznev
· 8 years ago
d8a72f0
Close input file in FileAudioDevice::StopRecording.
by noahric
· 8 years ago
78810b6
Expose media constraint string constants as ObjC NSStrings
by magjed
· 8 years ago
d22854b
FilePlayer: Remove unused default values for arguments
by kwiberg
· 8 years ago
4a42900
Removes redundant log warning in WebRtcAudioManager.
by henrika
· 8 years ago
86c9694
Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
by danilchap
· 8 years ago
5a25d95
FileRecorder + FilePlayer: Let Create functions return unique_ptr
by kwiberg
· 8 years ago
4466782
StartTimestamp generated randomly in RtpSender constructor
by Danil Chapovalov
· 8 years ago
2ae1fb6
Fix get_landmines.py script.
by ehmaldonado
· 8 years ago
144dd27
FileRecorderImpl and FilePlayerImpl don't need their own .h and .cc files
by kwiberg
· 8 years ago
c54071d
WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.
by ossu
· 8 years ago
a93d5ac
Don't simulate probing based on rtc event logs since we don't have that info logged.
by stefan
· 8 years ago
eb680ea
CongestionController::SetBweBitrates may now trigger probing.
by philipel
· 8 years ago
c594aa61
Add a gyp/gn option to use dummy audio file devices.
by noahric
· 8 years ago
e05bcc2
Do not switch a connection if the new connection is not ready to send packets.
by Honghai Zhang
· 8 years ago
49c01d7
Currently there is not way to programmically test whether a ScreenCapturer
by zijiehe
· 8 years ago
895e1a9
Change the default backup connection ping interval to 25 seconds.
by Honghai Zhang
· 8 years ago
287e548
Cleanup RtcpReceiver::TMMBRReceived function
by danilchap
· 8 years ago
f095012
Revert of Adding audio to video_quality_test. (patchset #10 id:230001 of https://codereview.webrtc.org/2136573002/ )
by minyue
· 8 years ago
65a6578
Adding audio to video_quality_test.
by minyue
· 8 years ago
75c287e
Fix incorrect example in mod_ops.h
by philipel
· 8 years ago
a06ce49
Run "git cl format" on some files before I start to modify them
by kwiberg
· 8 years ago
b789439
Roll chromium_revision 2b53ee0889..915e47250f (411979:412201)
by buildbot
· 8 years ago
90920d5
GN: Enable msse2 flag in Mac.
by ehmaldonado
· 8 years ago
9d7eb13
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ )
by kwiberg
· 8 years ago
e252d3c
MB: Fix incorrect iOS builder names.
by kjellander
· 8 years ago
427ce3d
Move FilePlayer and FileRecorder to Voice Engine
by kwiberg
· 8 years ago
2f69ce9
Cleaned out candidateSet member from TMMBRHelp class
by danilchap
· 8 years ago
1c814e7
iOS: Update MB and JSON configs + enable Goma
by kjellander
· 8 years ago
8eb37a3
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
by perkj
· 8 years ago
6910537
Add gn target for audio_device_tests.
by maxmorin
· 8 years ago
70f866c
Added new mixer to |check_targets| in .gn and fixed include/depend errors.
by aleloi
· 8 years ago
7522a28
Removed old probe cluster logic and logic related to ssrcs from DelayBasedBwe.
by philipel
· 8 years ago
b7186d0
Migrated GN target :isac_fix_test
by aleloi
· 8 years ago
b24b1ce
Moving mock classes around so that they may be reused in other unittests
by hbos
· 8 years ago
88e31a3
Fix warnings, simplify ADM.
by maxmorin
· 8 years ago
cc16836
- Add task queue to Call with the intent of replacing the use of one of the process threads.
by perkj
· 8 years ago
82dda1a
[WebRTC] Disable DirectX capturer tests if the system does not support it.
by zijiehe
· 8 years ago
e1b4d24
Skip AUD while extracting SPS and PPS on iOS.
by jianjun.zhu
· 8 years ago
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