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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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13790bec6b76753c437acd95d7dc92f820a2ef12
13790be
Add suppression for "uninitialized memory" error in getaddrinfo.
by deadbeef
· 8 years ago
8290ddf
Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )
by deadbeef
· 8 years ago
1095ada
Revert of Injectable Obj-C video codecs (patchset #8 id:140001 of https://codereview.webrtc.org/2966023002/ )
by tkchin
· 8 years ago
aa41f0c
API for periodically regathering ICE candidates
by Steve Anton
· 8 years ago
f1f9889
Roll chromium_revision 2c2aec8ef0..72d8fe0dd3 (485610:485684)
by buildbot
· 8 years ago
b8e7a7d
Use correct value for CFBundleExecutable and CFBundleName.
by Sylvain Defresne
· 8 years ago
950c1c9
TransmitMixer: Check GetSendCodec return value.
by ossu
· 8 years ago
29d0840
Reland of Refactor timing frame logic to work with encoders with internal sources (patchset #1 id:1 of https://codereview.webrtc.org/2980533002/ )
by ilnik
· 8 years ago
2fbd771
Roll chromium_revision dea6441d2d..2c2aec8ef0 (485283:485610)
by buildbot
· 8 years ago
82c5593
Let alr detector use a budged to detect underuse.
by tschumim
· 8 years ago
87b6ddb
Suppress MissingPrefix using tools attribute
by Ingemar Ådahl
· 8 years ago
c024740
Use relative paths in GN files.
by jianjun.zhu
· 8 years ago
c45d6d9
Remove dependency on rtc::Thread and rtc_base from audio_mixer_unittests.
by tommi
· 8 years ago
14c11a4
Add adaptive notch filter to remove narrowband echo components in AEC3
by peah
· 8 years ago
5e6685f
Robustification of the AEC3 echo removal in the first part of the call
by peah
· 8 years ago
c1abde7
Call should allow pass through of keep-alive packets.
by sprang
· 8 years ago
7fc3f15
Now uses CallStaticObjectMethodV to an variable argument list argument
by henrika
· 8 years ago
1e50748
Revert of Make "set_ignore_non_default_routes" actually use its argument. (patchset #1 id:1 of https://codereview.webrtc.org/2974873002/ )
by sprang
· 8 years ago
e5c4a81
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
by sprang
· 8 years ago
2910357
Transparency improvements in the echo canceller 3
by peah
· 8 years ago
863f03b
Fix video_replay tool to respect RTX stream and fix default parameters.
by ilnik
· 8 years ago
d8cf08f
Don't call CreateDtlsTransport_n from non-network thread in WebRtcSession
by deadbeef
· 8 years ago
05314c3
Make "set_ignore_non_default_routes" actually use its argument.
by deadbeef
· 8 years ago
48956a1
Remove unused headers from remote_bitrate_estimator_single_stream.cc
by eladalon
· 8 years ago
0b1e2f3
Revert of Refactor timing frame logic to work with encoders with internal sources (patchset #2 id:20001 of https://codereview.webrtc.org/2968153002/ )
by ilnik
· 8 years ago
bc8ee33
Remove verbose logs from audio_coding_module.cc.
by Noah Richards
· 8 years ago
a7a4535
Refactor timing frame logic to work with encoders with internal sources
by ilnik
· 8 years ago
5ce5a47
Roll chromium_revision b98c4727fc..dea6441d2d (485244:485283)
by buildbot
· 8 years ago
d2702ef
Fix flaky test VideoSendStreamTest.SendsKeepAlive
by sprang
· 8 years ago
fe53355
Fix presubmit check for webrtc/base.
by ehmaldonado
· 8 years ago
ea0e084
Add presubmit check to prevent further changes to webrtc/base.
by ehmaldonado
· 8 years ago
78460a7
Roll chromium_revision 8e79102feb..b98c4727fc (485221:485244)
by buildbot
· 8 years ago
370dd47
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
by ehmaldonado
· 8 years ago
9483b49
Remove remains of webrtc/base
by ehmaldonado
· 8 years ago
ccd7328
Roll chromium_revision a28aeb7be0..8e79102feb (485062:485221)
by buildbot
· 8 years ago
6c9556e
Prevent warnings in PacketRouterTest and PacketRouterRembTest
by eladalon
· 8 years ago
6cc2561
Remove webrtc::VideoEncoderFactory
by magjed
· 8 years ago
c453b08
Adding log in OrtcFactoryIntegrationTest and fix a bug.
by minyue-webrtc
· 8 years ago
0b249c2
Refactor gunit for synergy to gtest.
by minyue-webrtc
· 8 years ago
539d104
Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ )
by mbonadei
· 8 years ago
bffe597
Convert occurrences of deprecated WEBRTC_TRACE logging to LOG style logging in webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc.
by saza
· 8 years ago
5de0680
Add a check in the BlockBuffer of AEC2 to guard for buffer overflows.
by saza
· 8 years ago
3ffa72d
Add AudioFrame::ResetWithoutMuting() to address performance regression.
by Jonathan Yu
· 8 years ago
f1377f7
Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread.
by perkj
· 8 years ago
cf39dd5
Add RTC_FROM_HERE location information to two DCHECKs in ProcessThread.
by tommi
· 8 years ago
f88f72d
Roll chromium_revision 85f2caa791..a28aeb7be0 (484906:485062)
by buildbot
· 8 years ago
4dde3df
Move SrtpSession and tests to their own files.
by zstein
· 8 years ago
7d0a77e
Handle case where UDP packet contains multiple DTLS records.
by jbauch
· 8 years ago
7480da4
Trace loggging: Check for g_event_logger is not null before calling it.
by jtteh
· 8 years ago
fb660ae
Decreased the adaptation rate for the adaptive filter in the echo canceller 3
by peah
· 8 years ago
991521e
Roll chromium_revision 6ddb8d2861..85f2caa791 (484869:484906)
by buildbot
· 8 years ago
de5ff8e
Fix a variable naming typo
by henrik.lundin
· 8 years ago
e67bedb
External APM usage downstream dependency support cleanup
by peah
· 8 years ago
a0349c1
Injectable Obj-C video codecs
by Anders Carlsson
· 8 years ago
c1a58ba
Roll chromium_revision a5ed187842..6ddb8d2861 (484843:484869)
by buildbot
· 8 years ago
eaaae9e
base->rtc_base: Update .c, .mm and .java files.
by ehmaldonado
· 8 years ago
0e9bd8e
Roll chromium_revision 4b357464fd..a5ed187842 (484824:484843)
by buildbot
· 8 years ago
f04afde
Report interframe delay sum in old GetStats
by ilnik
· 8 years ago
5b36173
Support re-entrant calls to MessageQueueManager::Clear.
by jbauch
· 8 years ago
876088a
Roll chromium_revision baaa9eae93..4b357464fd (484696:484824)
by buildbot
· 8 years ago
4a494ff
desktop_capture: crop border in window_capture on Win8/10
by braveyao
· 8 years ago
f07e6b4
Roll chromium_revision 2e0945b687..baaa9eae93 (484611:484696)
by buildbot
· 8 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
ea39dfa
Roll chromium_revision c33c6bfd24..2e0945b687 (484321:484611)
by buildbot
· 8 years ago
9e3f1e4
Fixed a miscalculation of sent bitrate caused by mixup of time units
by Sebastian Jansson
· 8 years ago
d66072b
Moving asm code out of common_audio_c sources list
by mbonadei
· 8 years ago
3b03476
Remove MAIN_NIB_FILE from Info.plist because the substitution is broken
by oprypin
· 8 years ago
a449107
Let NetEq reset the AudioFrame during muted state
by henrik.lundin
· 8 years ago
02569ad
Update screen simulcast config
by sprang
· 8 years ago
168794c
Implement RTP keepalive in native stack.
by sprang
· 8 years ago
5c0d703
Moving asm code out of isac_fix_c sources list
by mbonadei
· 8 years ago
05db21d
Reland of move webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2973493002/ )
by ehmaldonado
· 8 years ago
2edc684
Report timing frames info in GetStats.
by ilnik
· 8 years ago
5b7fc8c
A few simplifications to CodecDatabase and VCMGenericDecoder.
by tommi
· 8 years ago
7025244
Roll chromium_revision f45f1f992e..c33c6bfd24 (484285:484321)
by buildbot
· 8 years ago
6aa9511
Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable
by bdodson
· 8 years ago
e4f63a1
Roll chromium_revision c01b31617b..f45f1f992e (484252:484285)
by buildbot
· 8 years ago
b16a01f
Revert "Reland "Adding ANA config event to debug dump.""
by minyue-webrtc
· 8 years ago
63d146b
NetEq: Rectify the implementation of PacketBuffer::DiscardOldPackets
by henrik.lundin
· 8 years ago
440ea8c
Roll chromium_revision 3fe2409358..c01b31617b (484231:484252)
by buildbot
· 8 years ago
191113a
Added implementation of four functions in the BBR congestion controller:
by gnish
· 8 years ago
bc0c4f5
Roll chromium_revision 6da2ebcead..3fe2409358 (484119:484231)
by buildbot
· 8 years ago
fae474c
Implement packet discard rate in NetEq.
by minyue-webrtc
· 8 years ago
889d965
Fix issue with zero rtt reports when using FlexFEC and add perf test.
by stefan
· 8 years ago
070efc0
Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat
by henrika
· 8 years ago
abee2d8
Roll chromium_revision 2fe6dc66f8..6da2ebcead (484092:484119)
by buildbot
· 8 years ago
0fc6d87
Roll chromium_revision cf58257d56..2fe6dc66f8 (483646:484092)
by buildbot
· 8 years ago
f720704
Added philipel@webrtc.org to webrtc/modules/remote_bitrate_estimator/OWNERS.
by philipel
· 8 years ago
cb576c5
Fixes build issue based on usage of Android O specific API
by henrika
· 8 years ago
c43f68e
Fix do not unregister bluetooth receiver if it was not registered
by Gustavo Garcia
· 8 years ago
cc8856c
Remove unused static VideoEncoder functions
by magjed
· 8 years ago
f612998
Override bots to use libstdc++ on Linux
by oprypin
· 8 years ago
8eadead
Adds support for USB audio devices in AppRTCMobile on Android.
by henrika
· 8 years ago
a9521e2
Reduce send rate to 50% if overusing before we have an acknowledged bitrate.
by terelius
· 8 years ago
2c3161c
Changed default value for the duration of the echo in echocanceller 3
by peah
· 8 years ago
0d7f04d
Reland of Add received audio/video call duration metrics based on packets.
by saza
· 8 years ago
38fecaf
Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ )
by ehmaldonado
· 8 years ago
d3588cf
Improved low-level echo handling in echo canceller 3
by peah
· 8 years ago
ed56680
Remove webrtc/tools
by ehmaldonado
· 8 years ago
382f21c
Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ )
by saza
· 8 years ago
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