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gerrit-public.fairphone.software
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webrtc
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08b0351ddd0469c8b0aa7c028ac93878a350e593
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fcada90
Fixing timestamp comparison assert.
by deadbeef
· 8 years ago
36a06a9
Increase QP threshold for H.264 encoder QP based scaling.
by glaznev
· 8 years ago
1184025
Restart capture session if needed on active.
by tkchin
· 8 years ago
5fac3f0
NetEq: Don't check sample rate and frame size upon error
by henrik.lundin
· 8 years ago
d1a10a0
Make FakeDecodeFromFile handle codec-internal CNG
by henrik.lundin
· 8 years ago
f02207d
MB: Flip Mac bots to GN by default.
by kjellander
· 8 years ago
b0b0edb
Roll chromium_revision e3860bd297..938114be1e (412289:414059)
by ehmaldonado
· 8 years ago
28a0ffd
GN: Synchronize resources between Android and iOS.
by kjellander
· 8 years ago
2df32a3
GN: Override lsan and tsan suppression files.
by ehmaldonado
· 8 years ago
5f2e7c4
Added more targets to .gn.
by aleloi
· 8 years ago
2ec45b9
Make dependency of audio_device of ApplicationServices explicit.
by maxmorin
· 8 years ago
4e7e8d7
Now probe for x3 and x6 of the initial start bitrate and allow for faster receive bitrates when calculating probing estimates.
by philipel
· 8 years ago
2c670db
Added GN target for webrtc_opus_fec_test.
by ivoc
· 8 years ago
7a0ff2f
Disable examples for GYP Android bots.
by ehmaldonado
· 8 years ago
98468bb
Revert of GN build rules for four audio processing test executables (patchset #3 id:40001 of https://codereview.webrtc.org/2267403003/ )
by sakal
· 8 years ago
538b560
GN build rules for four audio processing test executables
by kwiberg
· 8 years ago
0561bdf
Only use payload size within the know send/receive interval for probing calculations.
by philipel
· 8 years ago
619a211
iLBC: Handle a case of bad input data
by kwiberg
· 8 years ago
0aa9d18
Set send side bitrate estimate on successful probing attempt.
by philipel
· 8 years ago
cd8ae61
Add missing dependencies to setup_links.
by ehmaldonado
· 8 years ago
f944c35
GN: Add resources for webrtc_perf_tests on Android
by kjellander
· 8 years ago
e51b41a
Added GN target for isac_test.
by ivoc
· 8 years ago
5d167d6
Removals and renamings in the new audio mixer.
by aleloi
· 8 years ago
76f91cd
Add ThreadChecker to the TimestampAligner class.
by nisse
· 8 years ago
665d181
Increased column width for python tool rtp_analyzer.py.
by aleloi
· 8 years ago
30be5d7
Updated mixer unittests and fixed a related bug in the new mixer.
by aleloi
· 8 years ago
615d301
RTCStats and RTCStatsReport added (webrtc/stats).
by hbos
· 8 years ago
616df1e
Added a level indicator to new mixer.
by aleloi
· 8 years ago
1f77905
Remove outdated symlink
by kthelgason
· 8 years ago
a53fa0a
Fix AppRTC Android Demo GN build when is_component_build=true.
by sakal
· 8 years ago
4c8adb1
MB: Flip Android bots to GN by default.
by kjellander
· 8 years ago
24ee050
CQ: Remove android_arm64_rel trybot
by kjellander
· 8 years ago
b246a29
Define a protobuf format for representing plots. Add code to convert the C-representation generated by the RtcEventLog analysis tool, to the new protobuf format.
by terelius
· 8 years ago
6addf49
Adds function for computing moving average to visualization tool.
by terelius
· 8 years ago
5048f57
Add logs and small change in BasicPortAllocator.
by Honghai Zhang
· 8 years ago
f99a9de
ProbingEstimator: Erase history based on time threshold
by Irfan Sheriff
· 8 years ago
185ba29
Extract library from the RTCEventLog visualizer
by skvlad
· 8 years ago
5bed20f
Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video.
by Per
· 8 years ago
b37c45c
GN: Add libjingle_peerconnection_java to peerconnection_unittests.
by kjellander
· 8 years ago
a246cfb
Don't include RTP headers in send-side BWE.
by Stefan Holmer
· 8 years ago
9a11784
Migrated GN target :g722_test
by aleloi
· 8 years ago
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
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