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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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64c4a7ecfc86307136e3dd07ac46462b56d60fda
64c4a7e
Refactor webrtc/modules/audio_processing for GN check
by ehmaldonado
· 8 years ago
b0a1111
Decode h264 fmtp sprop-parameter-sets to binary.
by johan
· 8 years ago
c95d810
Roll chromium_revision b4a2bcaef4..bd4fdcd8d3 (437198:437221)
by buildbot
· 8 years ago
623427c
Injectable output rate calculater for AudioMixer.
by aleloi
· 8 years ago
9abd275
Remove unused arguments and variable in MediaOptimization.
by asapersson
· 8 years ago
7722a4c
Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ )
by kthelgason
· 8 years ago
710c335
Add ability to scale to arbitrary factors
by kthelgason
· 8 years ago
b39db84
Refactoring: Declare cricket::Codec constructors protected.
by hta
· 8 years ago
06a6984
Android classreferenceholder.h: Reorder function declaration keywords
by magjed
· 8 years ago
7e1070d
Roll chromium_revision 3cd4e5505f..b4a2bcaef4 (436604:437198)
by kjellander
· 8 years ago
2cd872a
Log BitBlt failure
by zijiehe
· 8 years ago
c4adabf
Create the Java Wrapper of RtpReceiverObserverInterface.
by zhihuang
· 8 years ago
dd33e63
Translate isolated-script-test-output to dump_json_test_results in gtest-parallel-wrapper.py
by ehmaldonado
· 8 years ago
157a34d
Add kwiberg@ to gtest-parallel OWNERS.
by Peter Boström
· 8 years ago
676e08f
Refactor webrtc/{api,audio} and modules/audio_coding for GN check
by kjellander
· 8 years ago
f515ab8
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago
7495c8c
Clean up redundant include of ../webrtc_overrides
by ehmaldonado
· 8 years ago
5a4eb51
Roll gtest-parallel.
by ehmaldonado
· 8 years ago
eca373f
Adding OnReceivedOverhead to AudioEncoder.
by minyue
· 8 years ago
ac382f3
Make ostream<< for enum class H264PacketizationMode
by hta
· 8 years ago
e36c46e
Use SSL_CTX_set_max_proto_version instead of SSL_CTX_set_max_version.
by davidben
· 8 years ago
d3de4ab
Remove deprecated comments
by zijiehe
· 8 years ago
49f34fd
Relanding: Refactoring that removes P2PTransport and DtlsTransport classes.
by deadbeef
· 8 years ago
57fd726
Revert of Refactoring that removes P2PTransport and DtlsTransport classes. (patchset #9 id:150001 of https://codereview.webrtc.org/2517883002/ )
by deadbeef
· 8 years ago
dd87d58
Add File::Open / Create functions to take an rtc::Pathname
by zijiehe
· 8 years ago
bd28681
Refactoring that removes P2PTransport and DtlsTransport classes.
by deadbeef
· 8 years ago
ebbe4f2
Set the preferred DSCP value for Rtp data channel to be DSCP_AF41.
by zhihuang
· 8 years ago
66f99a4
Roll chromium_revision d8bf23963e..3cd4e5505f (436579:436604)
by buildbot
· 8 years ago
6d314c7
Reject XR TargetBitrate items with unsupported layer indices
by sprang
· 8 years ago
99cc10f
Roll chromium_revision 092e72e03b..d8bf23963e (436556:436579)
by buildbot
· 8 years ago
9aa9688
Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
by hta
· 8 years ago
28c2487
Add unit tests for avfoundation format mapper functions.
by denicija
· 8 years ago
768c648
Move /webrtc/api/android files to /webrtc/sdk/android
by magjed
· 8 years ago
45bb513
Reland of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #1 id:1 of https://codereview.webrtc.org/2548333002/ )
by henrik.lundin
· 8 years ago
243a0a7
Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
by hta
· 8 years ago
dbc960c
The level controller complexity tests have lately been
by peah
· 8 years ago
c9badd5
Add comment to metrics.h
by henrik.lundin
· 8 years ago
68d3213
RTPPayloadRegistry: Stop using the rate to keep track of receive codecs
by kwiberg
· 8 years ago
e59647b
This approach passes packetization mode to the encoder as part of
by hta
· 8 years ago
5f0ddf2
Roll chromium_revision 4f26eb8a56..092e72e03b (436542:436556)
by buildbot
· 8 years ago
406616f
Fix spelling mistake in rtp_rtcp.h.
by brandtr
· 8 years ago
6612cb7
Roll chromium_revision c4e9c3c5de..4f26eb8a56 (436513:436542)
by buildbot
· 8 years ago
7439f97
Split targets mixing .c and .cc sources.
by kjellander
· 8 years ago
5f7a9dc
Roll chromium_revision c9600cd059..c4e9c3c5de (436473:436513)
by buildbot
· 8 years ago
ea7a33d
Roll chromium_revision f5e72ebfb7..c9600cd059 (436420:436473)
by buildbot
· 8 years ago
548d321
Roll chromium_revision 1376f2e121..f5e72ebfb7 (436364:436420)
by buildbot
· 8 years ago
c9f9500
Expose audio_jitter_buffer_fast_accelerate config to objc wrapper
by haysc
· 8 years ago
5fe4d49
Remove unsupported mac framework target.
by kthelgason
· 8 years ago
61a2af3
Roll chromium_revision d43cbc46d5..1376f2e121 (436297:436364)
by buildbot
· 8 years ago
bd681b9
AGC: Route clipping parameter from webrtc::Config to AGC
by henrik.lundin
· 8 years ago
db752f9
Revert "Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )"
by stefan
· 8 years ago
53e0980
Roll chromium_revision 49f56a2e7b..d43cbc46d5 (436268:436297)
by buildbot
· 8 years ago
37535bf
Refactor fileutils.cc/h and fileutils_unittests.cc into their own targets.
by ehmaldonado
· 8 years ago
1d08100
Use RTC_DISALLOW_COPY_AND_ASSIGN in webrtc/base/sigslottester.h
by kjellander
· 8 years ago
d654a9b
Reduce number of FlexFEC VideoSendStreamTests and lower packet loss.
by brandtr
· 8 years ago
63407a9
Revert of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #2 id:20001 of https://codereview.webrtc.org/2547593002/ )
by henrik.lundin
· 8 years ago
dbf729c
Roll chromium_revision a253a1bb31..49f56a2e7b (436254:436268)
by buildbot
· 8 years ago
49715fe
APM: Change 3 UMA metrics to fewer but linearly distributed buckets
by henrik.lundin
· 8 years ago
fa07b91
Delete unused spreadsort implementation.
by nisse
· 8 years ago
29c9dda
Roll chromium_revision 60f4f97afb..a253a1bb31 (436236:436254)
by buildbot
· 8 years ago
e545e5d
RtpPacketizer::NextPacket fills RtpPacket instead of just payload.
by danilchap
· 8 years ago
f00082d
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
by henrik.lundin
· 8 years ago
6321b49
Move functionality out from AudioFrame and into AudioFrameOperations.
by aleloi
· 8 years ago
bd9bdf6
Roll chromium_revision b5ed348a58..60f4f97afb (436228:436236)
by buildbot
· 8 years ago
deb1f8a
Roll chromium_revision cfcd3337a6..b5ed348a58 (436209:436228)
by buildbot
· 8 years ago
280be7d
Roll chromium_revision 5f3032e780..cfcd3337a6 (436201:436209)
by buildbot
· 8 years ago
595d15c
Roll chromium_revision dfb4c71778..5f3032e780 (436198:436201)
by buildbot
· 8 years ago
a887f7c
Roll chromium_revision b0dfc121a4..dfb4c71778 (436194:436198)
by buildbot
· 8 years ago
347908f
Roll chromium_revision 0740b0dfd2..b0dfc121a4 (436192:436194)
by buildbot
· 8 years ago
a99e52f
Roll chromium_revision 5bbd80fb56..0740b0dfd2 (436191:436192)
by buildbot
· 8 years ago
d3b79f5
Roll chromium_revision 62ef852f1d..5bbd80fb56 (436188:436191)
by buildbot
· 8 years ago
6b5789c
Roll chromium_revision 4c53995f0b..62ef852f1d (436184:436188)
by buildbot
· 8 years ago
aa5c7b9
Roll chromium_revision 1b019f5fd3..4c53995f0b (436175:436184)
by buildbot
· 8 years ago
c46e0b7
Roll chromium_revision 8cbbe0ffc3..1b019f5fd3 (436165:436175)
by buildbot
· 8 years ago
c191bd3
Roll chromium_revision 0a4f767df7..8cbbe0ffc3 (436162:436165)
by buildbot
· 8 years ago
7ba6e02
Roll chromium_revision 64637ac52e..0a4f767df7 (436108:436162)
by buildbot
· 8 years ago
4c6696c
Revert of Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan (patchset #1 id:1 of https://codereview.webrtc.org/2546913003/ )
by deadbeef
· 8 years ago
a59b012
Roll chromium_revision f857831468..64637ac52e (436038:436108)
by buildbot
· 8 years ago
c23efae
Allow locally rendered video to be downscaled in end-to-end tests.
by deadbeef
· 8 years ago
c63b894
Modify the parameter type of PeerConnectionObserver callback OnAddTrack.
by zhihuang
· 8 years ago
d5c93e0
Roll chromium_revision 5e077354e9..f857831468 (435981:436038)
by buildbot
· 8 years ago
2e59a02
Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )
by deadbeef
· 8 years ago
ef0cd56
Roll chromium_revision 58c06a5f1b..5e077354e9 (435946:435981)
by buildbot
· 8 years ago
9cef11b
Fix exponential probing in ProbeController.
by sergeyu
· 8 years ago
5e6c52c
Roll chromium_revision dd10720676..58c06a5f1b (435924:435946)
by buildbot
· 8 years ago
4eb03c7
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
by asapersson
· 8 years ago
5932149
Use different restrictions of acked bitrate lag depending on operating point.
by stefan
· 8 years ago
a790d83
Wire up rtcp xr target bitrate on receive side.
by sprang
· 8 years ago
9afbfc0
Roll chromium_revision f50152dfc4..dd10720676 (435897:435924)
by buildbot
· 8 years ago
0f01c7f
Added tool for reference less video analysis (go/refless-video-analysis)
by charujain
· 8 years ago
b2250e5
New gn target video_frame_api.
by nisse
· 8 years ago
969b12f
Remove xdisplaycheck
by kjellander
· 8 years ago
df28e47
fix coding and documentary ambiguity in AimdRateControl::TimeToReduceFurther.
by howtofly
· 8 years ago
a28a1b9
Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan
by henrik.lundin
· 8 years ago
5c71166
VP8DecoderImpl: Fix uninitialized memory crash
by magjed
· 8 years ago
00bceb1
Deprecated SetAudioPacketSize from RTPSender and removed calls to it.
by ossu
· 8 years ago
e066b30
Remove API-related #defines from voice_engine_configurations.h
by henrik.lundin
· 8 years ago
15cf9f9
Roll chromium_revision 601e4f48a3..f50152dfc4 (435870:435897)
by buildbot
· 8 years ago
b336392
Sanity check parsed QP values from H264 bitstream
by kthelgason
· 8 years ago
864f58b
Roll chromium_revision 353f713f3d..601e4f48a3 (435846:435870)
by buildbot
· 8 years ago
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