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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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cfee215b23105775ec89f80162695bcb7cf4b073
cfee215
Migrated ILBC and ISAC test targets for GN.
by aleloi
· 8 years ago
09f1560
TimestampScaler no longer depends on NetEqDecoder to figure out scaling.
by ossu
· 8 years ago
3e3b6f2
Remove Chromium Clang warnings only on Windows.
by ehmaldonado
· 8 years ago
1b365a8
Remove the old AndroidVideoCapturer stack code.
by sakal
· 8 years ago
de04f8e
Reland of Delete method cricket::VideoFrame::Copy. (patchset #1 id:1 of https://codereview.webrtc.org/2275313003/ )
by nisse
· 8 years ago
ebb8d04
Delete deprecated and unused method VideoFrame::SetTimeStamp.
by nisse
· 8 years ago
cf157cf
CQ: Update GYP trybot names.
by Henrik Kjellander
· 8 years ago
133edb2
MB: Flip Windows bots to GN by default
by kjellander
· 8 years ago
062ce9f
Combining "SetTransportChannel" and "SetRtcpTransportChannel".
by deadbeef
· 8 years ago
f0bb360
Add parameter to TransportController to not change ICE role on restart.
by Taylor Brandstetter
· 8 years ago
8cf8898
Fix Chromium clang plugin warnings
by kwiberg
· 8 years ago
d02fe4b
GN: Fix windows clang errors. Attempt 2.
by ehmaldonado
· 8 years ago
b4677de
Adding GYP/GN owners to stats/ aligning with all other dirs.
by Henrik Kjellander
· 8 years ago
d82eee0
Log how often DTLS negotiation failed because of incompatible ciphersuites.
by zhihuang
· 8 years ago
3026ee8
Added logging of the level controller metrics.
by peah
· 8 years ago
2800d74
Change RtpSender::OnReceiveNACK name and signature
by Danil Chapovalov
· 8 years ago
1bcfce5
Deactivated the intelligibility enhancement functionality by default
by peah
· 8 years ago
7d67e45
Revert of Added functionality for specifying the initial signal level to use for the gain estimation in the l… (patchset #8 id:160001 of https://codereview.webrtc.org/2254973003/ )
by peah
· 8 years ago
fe1d191
MB: Flip Linux bots to GN by default.
by kjellander
· 8 years ago
a897f26
AbsoluteSendTime rtp header extension publish MsTo24Bit conversion
by danilchap
· 8 years ago
84bc985
Removed virtual from several methods in DecoderDatabase to minimize
by ossu
· 8 years ago
c07c8bb
Suppress memcheck errors on linux bot.
by philipel
· 8 years ago
57fec1d
This CL adds functionality in the level controller to
by peah
· 8 years ago
1e8ed4a
Replace calls to assert() with RTC_DCHECK_*() in .c code
by kwiberg
· 8 years ago
073ece4
Skip unit test if GYP_DEFINES="rtc_use_h264=1" is not set.
by johan
· 8 years ago
c766804
NetEq: Update CNG code to accommodate 48 kHz sample rate
by henrik.lundin
· 8 years ago
5085b0c
Adding AecDump functionality to AppRTCDemo for iOS
by peah
· 8 years ago
bad33bf
Renaming BaseChannel methods and adding comments for added clarity.
by Taylor Brandstetter
· 8 years ago
91b03b0
Revert of Delete method cricket::VideoFrame::Copy. (patchset #3 id:210001 of https://codereview.webrtc.org/2275243002/ )
by philipel
· 8 years ago
f715f98
Reland of Delete method cricket::VideoFrame::Copy. (patchset #1 id:1 of https://codereview.webrtc.org/2087923004/ )
by nisse
· 8 years ago
01e0ee6
Revert of CQ: Remove android_arm64_rel trybot (patchset #1 id:1 of https://codereview.webrtc.org/2270943003/ )
by kjellander
· 8 years ago
6bf62f7
Avoids java.lang.NullPointerException in WebRtcAudioRecord
by henrika
· 8 years ago
4805231
Moved format_macros.h from rtc_base to rtc_base_approved.
by ivoc
· 8 years ago
e575c01
Remove GYP execution from DEPS hooks (gclient {sync,runhooks})
by kjellander
· 8 years ago
4bc4d27
GN: Fix Windows Clang errors
by ehmaldonado
· 8 years ago
3f746ea
Fix error when accumulating floats in an int.
by maxmorin
· 8 years ago
e29352b
Refactor certificate stats collection, added SSLCertificateStats.
by hbos
· 8 years ago
2ab012c
Implement CVO for iOS capturer
by magjed
· 8 years ago
19319a3
Add missing "//build/config/sanitizers:deps" to executable targets.
by ehmaldonado
· 8 years ago
00e45bb
Move InsertZeroColumns and CopyColumn to ::internal.
by brandtr
· 8 years ago
7a770e0
GN build rules for four audio processing test executables
by kwiberg
· 8 years ago
8a6a600
Make neteq_rtpplay parse RTP header extensions
by henrik.lundin
· 8 years ago
5f09980
Removed inline definitions and added destructors to fix chromium-style.
by aleloi
· 8 years ago
549d80b
NetEq: only update current_rtp_payload_type_ when validated
by henrik.lundin
· 8 years ago
fe8f489
Fix setting the MTU for SCTP.
by deadbeef
· 8 years ago
b60a819
Fixing inconsistency with behavior of `ClearGettingPorts`.
by deadbeef
· 8 years ago
824f586
Fixing segfault caused by TurnServer.
by deadbeef
· 8 years ago
1d7a637
Fixing off-by-one error with max SCTP id.
by Taylor Brandstetter
· 8 years ago
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
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