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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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9f28b1d354a2f457032fec5f1a0b3aeac31f7275
9f28b1d
Roll chromium_revision 7720633b71..4b07207284 (463923:463938)
by buildbot
· 8 years ago
0efdf3e
Roll chromium_revision ff1db7c1c9..7720633b71 (463722:463923)
by buildbot
· 8 years ago
1cdd16d
Roll chromium_revision aa2ca13eaa..ff1db7c1c9 (463610:463722)
by buildbot
· 8 years ago
0426222
Modified the rtp_receiver_unittests.
by zhihuang
· 8 years ago
00d802b
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
by ilnik
· 8 years ago
10d095d
Revert of Change NetEq::InsertPacket to take an RTPHeader (patchset #2 id:20001 of https://codereview.webrtc.org/2807273004/ )
by henrik.lundin
· 8 years ago
80ff00c
Improve USB device reset logic
by jansson
· 8 years ago
b213a16
Finalized the SSE2 optimizations for the matched filter in AEC3
by peah
· 8 years ago
c0d74d9
Roll chromium_revision 419b6b1a41..aa2ca13eaa (463589:463610)
by buildbot
· 8 years ago
27c46e2
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )
by ilnik
· 8 years ago
4d02757
Change NetEq::InsertPacket to take an RTPHeader
by henrik.lundin
· 8 years ago
774f6b4
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
by ilnik
· 8 years ago
268862c
Address denicija's comments for AppRTCMobile video codec setting.
by sakal
· 8 years ago
24da37b
ObjC: RTCVideoSource cleanup
by magjed
· 8 years ago
29dbb19
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ )
by ilnik
· 8 years ago
dee5eb1
Android Logging.java: Load native library only when needed
by magjed
· 8 years ago
382f2b2
Fix swarming tests not running in parallel
by kjellander
· 8 years ago
4fa0c4f
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
by ilnik
· 8 years ago
5721866
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
by ilnik
· 8 years ago
cabc25c
Roll chromium_revision 860f7b94d4..419b6b1a41 (463558:463589)
by buildbot
· 8 years ago
cde2528
Enabling 'gn check' on //webrtc/ortc.
by mbonadei
· 8 years ago
10fc0e6
Delay based logging.
by philipel
· 8 years ago
64e739a
Add content type information to Encoded Images and add corresponding RTP extension header.
by ilnik
· 8 years ago
93cda2e
APM-QA tool, renaming noise generators into input-reference generators.
by alessiob
· 8 years ago
9765370
Resolve dependency between rtc_event_log_api and remote_bitrate_estimator
by michaelt
· 8 years ago
810eecf
Roll chromium_revision c57654688e..860f7b94d4 (463520:463558)
by buildbot
· 8 years ago
7fb7bbd
Revert of Add first part of the network_tester functionality. (patchset #13 id:260001 of https://codereview.webrtc.org/2779233002/ )
by michaelt
· 8 years ago
333d0ff
Add first part of the network_tester functionality.
by michaelt
· 8 years ago
e0ab0ad
Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
by kjellander
· 8 years ago
0d4e068
Make safe_cmp::* constexpr
by kwiberg
· 8 years ago
d491109
Roll chromium_revision 1af3c1a4a8..c57654688e (463476:463520)
by buildbot
· 8 years ago
4a9d08f
Roll chromium_revision d3a2a83fbf..1af3c1a4a8 (463418:463476)
by buildbot
· 8 years ago
8c459f9
Restore old (deprecated) signature of initializeAndroidGlobals.
by deadbeef
· 8 years ago
20c84cc
Making FakeNetworkPipe demux audio and video packets.
by minyue
· 8 years ago
d9ce764
Make RtpTransport actually implement RtpTransportInterface
by zstein
· 8 years ago
0f92c79
Roll chromium_revision 5d7042a87c..d3a2a83fbf (463209:463418)
by buildbot
· 8 years ago
b4fc73a
Removing unnecessary parameters from initializeAndroidGlobals.
by deadbeef
· 8 years ago
6799553
Add information about microphone gain changes to AEC3
by peah
· 8 years ago
6d822ad
Added forced zero AEC output after call startup and echo path changes
by peah
· 8 years ago
ca31f17
Remove deprecated RTPPayloadStrategy
by danilchap
· 8 years ago
a1ef71f
Add parser to visualise the ana dump
by michaelt
· 8 years ago
8d609f6
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
by hbos
· 8 years ago
b0f7e39
Move IsIntlike to type_traits.h
by kwiberg
· 8 years ago
37e99fd
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/
by kwiberg
· 8 years ago
2fa97fd
Roll chromium_revision 0a53e4a670..5d7042a87c (463181:463209)
by buildbot
· 8 years ago
0642b32
Remove duplicate entries from AUTHORS file
by henrik.lundin
· 8 years ago
fbcc5cb
Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
by olka
· 8 years ago
925e9d7
Removed workaround for the WARN_UNUSED_RESULT issue.
by peah
· 8 years ago
4fb651d
Event log cleanup in tests.
by philipel
· 8 years ago
fca900a
Fix two invalid DCHECKs related to audio BWE.
by stefan
· 8 years ago
49cad02
Ignore some UBSan errors
by kwiberg
· 8 years ago
9f2c18e
Changed OLA window for neteq. Old code didnt work well with 48khz
by soren
· 8 years ago
c547e84
Allow rtp::Packet::*RawExtension to take 0 as an extension id
by danilchap
· 8 years ago
02465b8
Add some unit tests to vie_encoder.
by asapersson
· 8 years ago
36e6a8f
WavReaderAdaptor is a simple adaptor of the existing class WavReader from webrtc/common_audio/wav_file.h. The adaptor was mainly needed to use dependency injection and easily test the MultiEndCall class (see https://codereview.webrtc.org/2761853002/).
by alessiob
· 8 years ago
2042c16
Revert of Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2808463002/ )
by nisse
· 8 years ago
64c93c3
Roll chromium_revision 1ab7c6059c..0a53e4a670 (463170:463181)
by buildbot
· 8 years ago
188596f
Delete class ScopedPtrCollection. Replaced with vector of unique_ptr.
by nisse
· 8 years ago
bd1a681
Roll chromium_revision c8a0f6b4c5..1ab7c6059c (463161:463170)
by buildbot
· 8 years ago
1b0882d
Roll chromium_revision c30f6366d7..c8a0f6b4c5 (463148:463161)
by buildbot
· 8 years ago
1d9ef47
Roll chromium_revision b0bf8e8ed3..c30f6366d7 (463140:463148)
by buildbot
· 8 years ago
a3d4d94
Roll chromium_revision 5e27d4b8d1..b0bf8e8ed3 (463138:463140)
by buildbot
· 8 years ago
4b37127
Fix compilation issues of std::unique_ptr
by steweg
· 8 years ago
8cc642b
Roll chromium_revision d5df028c94..5e27d4b8d1 (463137:463138)
by buildbot
· 8 years ago
89292d1
Roll chromium_revision 270af5af87..d5df028c94 (463130:463137)
by buildbot
· 8 years ago
5253411
Roll chromium_revision 9d8fbbd04c..270af5af87 (463126:463130)
by buildbot
· 8 years ago
01f2793
Roll chromium_revision a35a1e2ce2..9d8fbbd04c (463121:463126)
by buildbot
· 8 years ago
50ddc63
Roll chromium_revision 98c8321fe9..a35a1e2ce2 (463081:463121)
by buildbot
· 8 years ago
9d8052d
Roll chromium_revision 762665735a..98c8321fe9 (462871:463081)
by buildbot
· 8 years ago
66e9f76
Adjust parameter in vp9 videoprocessor_integration test.
by jianj
· 8 years ago
8d23c05
MultiEndCall::CheckTiming() verifies that a set of audio tracks and timing information is valid to simulate conversational speech. Unordered turns are rejected. Self cross-talk and cross-talk with 3 or more speakers are not permitted since it would require mixing at the simulation step.
by alessiob
· 8 years ago
292084c
Added the GetSources() to the RtpReceiverInterface and implemented
by zhihuang
· 8 years ago
bb16a48
Roll chromium_revision 61577bba5a..762665735a (462822:462871)
by buildbot
· 8 years ago
8942045
Adding support for handling highly reverberant echoes in AEC3.
by peah
· 8 years ago
38415b2
Reland of Adding PRESUBMIT check on google::protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2791583002/ )
by mbonadei
· 8 years ago
423f106
Add support for 64-bit architectures in build_aar.py.
by sakal
· 8 years ago
2ce640f
Fixing sample-rate dependent band-split filter issues in AEC3
by peah
· 8 years ago
ea44ad4
Roll chromium_revision d65c1d7370..61577bba5a (462801:462822)
by buildbot
· 8 years ago
7c2c843
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
by mbonadei
· 8 years ago
0944a80
Update stats for cpu/quality adaptation changes to excluded time when video is suspended.
by asapersson
· 8 years ago
533b7ac
Roll chromium_revision c405301b70..d65c1d7370 (462743:462801)
by buildbot
· 8 years ago
7749286
Make AudioProcessing::GetConfig() pure virtual
by henrik.lundin
· 8 years ago
abd101b
Support multiple connected Android devices in low bandwidth audio test
by oprypin
· 8 years ago
225bfc0
Make PacketTransportInternal inherit from PacketTransportInterface.
by deadbeef
· 8 years ago
81c899d
Roll chromium_revision fbddb8a080..c405301b70 (462718:462743)
by buildbot
· 8 years ago
f49011b
Roll chromium_revision 300bd53cd8..fbddb8a080 (462634:462718)
by buildbot
· 8 years ago
4b572d0
Correction of the AEC3 underrun behavior and minor other corrections
by peah
· 8 years ago
86afe9d
Major updates to the echo removal functionality in AEC3
by peah
· 8 years ago
f51517a
Roll chromium_revision bb330d1b8f..300bd53cd8 (462561:462634)
by buildbot
· 8 years ago
8cd1cce
Roll chromium_revision 86718dcdf2..bb330d1b8f (462494:462561)
by buildbot
· 8 years ago
1ffbd6c
Injectable audio encoders: voice_engine/channel changes.
by ossu
· 8 years ago
5f4aaeb
Re-enable FullStackTest.ScreenshareSlidesVP9_2SL test.
by marpan
· 8 years ago
4b62001
Adding AudioDeviceDataObserver interface
by Lu Liu
· 8 years ago
a1a040a
Injectable audio encoders: BuiltinAudioEncoderFactory
by ossu
· 8 years ago
ac4bbdf
Roll chromium_revision 3014f8b41e..86718dcdf2 (462374:462494)
by buildbot
· 8 years ago
cab8d88
Revert of Enable rtc_unittests on iOS simulator (patchset #2 id:20001 of https://codereview.webrtc.org/2799033004/ )
by kjellander
· 8 years ago
6167b26
Make RtpTransportControllerSend::send_side_cc_ a direct member.
by nisse
· 8 years ago
cde46b7
Resolve cyclic dependency between audio network adaptor and event log api
by michaelt
· 8 years ago
28dc285
Adding cbr support for Opus
by soren
· 8 years ago
388fe42
Make WARN_UNUSED_RESULT a no-op on gcc
by kwiberg
· 8 years ago
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