Gitiles
Code Review
Sign In
android-review.linaro.org
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
refs/heads/master-soong
c8b2e2b
Update mac makefiles.
by Torne (Richard Coles)
· 9 years ago
master-soong
android-m-preview
3b77881
MIPS: Update Chromium WebView makefiles (webrtc).
by Paul Lind
· 9 years ago
d31cda8
Set default 'mips_arch_variant%' to 'r6'
by Gordana Cmiljanovic
· 9 years ago
bdd5979
Temporarily disable -Werror in Chromium.
by Torne (Richard Coles)
· 10 years ago
5d04ee7
Merge from Chromium at DEPS revision 03655fd3f6d7
by Torne (Richard Coles)
· 10 years ago
361320e
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 60ab669c4c545b328b5c8b0453eb2cdecf851651
by Torne (Richard Coles)
· 10 years ago
60ab669
Remove partially defined WebRtcRTPHeader from Parse().
by pbos@webrtc.org
· 10 years ago
8645a5a
Use uint16s for port numbers in webrtc/p2p/base.
by pkasting@chromium.org
· 10 years ago
a031c17
Fix WebRTC Win64 + BoringSSL build.
by henrike@webrtc.org
· 10 years ago
eb46bb8
Volume buttons in AppRTCDemo should affect output audio volume (part II).
by henrika@webrtc.org
· 10 years ago
77155b0
Merge from Chromium at DEPS revision db3f05efe0f9
by Torne (Richard Coles)
· 10 years ago
98dd0b8
Log formatting fix for VideoEncoderConfig.
by pbos@webrtc.org
· 10 years ago
01a30f2
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at b831a9e3d5f9f0563d249b726cffa8a070e58aee
by Android Chromium Automerger
· 10 years ago
3ebce78
Remove the state_ member from AudioDecoder
by kwiberg@webrtc.org
· 10 years ago
b831a9e
Adjust parameter in vp9 rate control test.
by marpan@webrtc.org
· 10 years ago
10d91f4
Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.
by marpan@webrtc.org
· 10 years ago
b44eb8f
Update makefiles after merge of Chromium at 5a645aa13b82
by Android Chromium Automerger
· 10 years ago
941173a
Remove uses of build date/time.
by pbos@webrtc.org
· 10 years ago
5232267
Wire up bandwidth stats to the new API and webrtcvideoengine2.
by stefan@webrtc.org
· 10 years ago
7f032d2
Update makefiles after merge of Chromium at 2d0da5605d75
by Android Chromium Automerger
· 10 years ago
048258f
Restore old behavior for Android in fileutils.cc
by kjellander@webrtc.org
· 10 years ago
20b2dc6
Fix android_clang build.
by glaznev@webrtc.org
· 10 years ago
d1f71cc
Revert 7623 "Remove the state_ member from AudioDecoder"
by niklas.enbom@webrtc.org
· 10 years ago
1145210
Revert 7625 "Don't use DCHECK when you need the side effects..."
by niklas.enbom@webrtc.org
· 10 years ago
a91b08b
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9a8c28f10f329c5ce91e77057933e60224000627
by Android Chromium Automerger
· 10 years ago
ff6cd0b
Don't use DCHECK when you need the side effects...
by kwiberg@webrtc.org
· 10 years ago
5721e27
Remove the state_ member from AudioDecoder
by kwiberg@webrtc.org
· 10 years ago
385606f
Add support for VP9 in webrtc::Call and video_loopback.
by stefan@webrtc.org
· 10 years ago
c4cd81a
Reduce to 2 probes when probing for initial bandwidth.
by stefan@webrtc.org
· 10 years ago
4706c96
Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged.
by stefan@webrtc.org
· 10 years ago
8f2a7fa
Update makefiles after merge of Chromium at a99b7ad25d02
by Android Chromium Automerger
· 10 years ago
9a8c28f
Reworked paced sender queue
by sprang@webrtc.org
· 10 years ago
83c1dcb
Update makefiles after merge of Chromium at 30ec995cdb2d
by Android Chromium Automerger
· 10 years ago
6651f18
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at a1fd19c12e4efdf4b8a5f92323070443d50dc34e
by Android Chromium Automerger
· 10 years ago
8d28158
Adds support for finch experiments to video_loopback.
by stefan@webrtc.org
· 10 years ago
95e0f61
Fix problem with late packets in NetEq
by henrik.lundin@webrtc.org
· 10 years ago
e765eef
Delete VideoReceiveStream channels in destructor.
by pbos@webrtc.org
· 10 years ago
ab22837
Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16
by kwiberg@webrtc.org
· 10 years ago
14f28eb
Remove the useless dummy state parameter to WebRtcG711_*
by kwiberg@webrtc.org
· 10 years ago
a1fd19c
Remove the codec_type_ member from AudioDecoder
by kwiberg@webrtc.org
· 10 years ago
343d3cf
Enables AIMD control by default.
by stefan@webrtc.org
· 10 years ago
dba94e1
Improving error message from neteq_rtpplay
by henrik.lundin@webrtc.org
· 10 years ago
deb9e49
Add format members to AudioConverter for DCHECKing.
by andrew@webrtc.org
· 10 years ago
afcd610
Update rate control parameter in vp9 test.
by marpan@webrtc.org
· 10 years ago
dd298d8
Roll chromium_revision: 28d1981..d3db2ff
by marpan@webrtc.org
· 10 years ago
1c755a6
Restore the void return type on WriteWavHeader.
by andrew@webrtc.org
· 10 years ago
7e2ad87
replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics.
by andrew@webrtc.org
· 10 years ago
63c5ce8
Add Opus support to neteq_rtpplay
by henrik.lundin@webrtc.org
· 10 years ago
3894237
Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call.
by stefan@webrtc.org
· 10 years ago
ed94bc8
Add stats for video:
by asapersson@webrtc.org
· 10 years ago
9b1042c
Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already.
by braveyao@webrtc.org
· 10 years ago
5f53af3
Adjust/increase rate control thresold for a vp9 test.
by marpan@webrtc.org
· 10 years ago
6637388
Add VP9 codec to VCM and vie_auto_test.
by marpan@webrtc.org
· 10 years ago
0eb4066
Update Android projects to API level 21.
by kjellander@webrtc.org
· 10 years ago
99f0de3
replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics.
by andrew@webrtc.org
· 10 years ago
46c5634
Add a WavReader counterpart to WavWriter.
by andrew@webrtc.org
· 10 years ago
2dc72e3
Update makefiles after merge of Chromium at a41c404b1c7f
by Android Chromium Automerger
· 10 years ago
1d09eed
Update makefiles after merge of Chromium at b210e2d62956
by Android Chromium Automerger
· 10 years ago
796056b
Update all .isolate files for the new format.
by kjellander@webrtc.org
· 10 years ago
4ed1b70
Update Android projects to API level 20.
by kjellander@webrtc.org
· 10 years ago
cf59c6e
Fix N7 camera aspect ratio.
by glaznev@webrtc.org
· 10 years ago
f8586cc
Build fix for MIPS32R6.
by andrew@webrtc.org
· 10 years ago
6aaaf9f
Fix a name collision with Android libc++
by andrew@webrtc.org
· 10 years ago
08b354d
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 1a02faa335e7d8076b5cf8dd9a584e72669b0c8e
by Android Chromium Automerger
· 10 years ago
ddb84aa
Implement conference-mode temporal-layer screencast.
by pbos@webrtc.org
· 10 years ago
fbdea25
Configure A/V sync in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
4ef173b
Simplify bwe tests.
by stefan@webrtc.org
· 10 years ago
a1f6cf7
Revert "Revert part of r7561, "Refactor audio conversion functions.""
by andrew@webrtc.org
· 10 years ago
8a3acf6
arm64 iOS build.
by tkchin@webrtc.org
· 10 years ago
f37dc42
Add 15 fps support for Android devices with missing 15 fps camera mode.
by glaznev@webrtc.org
· 10 years ago
d7062cc
Creating a C++ wrapper class for VAD
by henrik.lundin@webrtc.org
· 10 years ago
78f89f1
Revert part of r7561, "Refactor audio conversion functions."
by kwiberg@webrtc.org
· 10 years ago
bce1329
Refactor audio conversion functions.
by andrew@webrtc.org
· 10 years ago
7b5a896
Use external VideoDecoders in VideoReceiveStream.
by pbos@webrtc.org
· 10 years ago
2ba45ee
Add stats for duplicate sent and received NACK requests.
by asapersson@webrtc.org
· 10 years ago
2d05389
common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32
by bjornv@webrtc.org
· 10 years ago
3c47df6
Remove unused code in overuse detector.
by asapersson@webrtc.org
· 10 years ago
a999336
AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket
by kwiberg@webrtc.org
· 10 years ago
ff8f833
Enable G.722 for Chromium builds
by henrik.lundin@webrtc.org
· 10 years ago
dcfa54a
Make an AudioEncoder subclass for Opus
by kwiberg@webrtc.org
· 10 years ago
06d5119
Make NSinst_t* const and rename to self in ns_core
by aluebs@webrtc.org
· 10 years ago
e77df57
Update makefiles after merge of Chromium at f92f0738e9e0
by Android Chromium Automerger
· 10 years ago
8239efe
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4f2aa0829e4e69972202efb7de2f53cc8858e2c9
by Android Chromium Automerger
· 10 years ago
1a02faa
move xmpp and p2p to webrtc
by henrike@webrtc.org
· 10 years ago
b136cd2
Make local functions static and dropWebRtcNs_ in ns_core
by aluebs@webrtc.org
· 10 years ago
fc14046
Make all comments whole sentences in ns_core
by aluebs@webrtc.org
· 10 years ago
4f2aa08
scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots.
by henrike@webrtc.org
· 10 years ago
bc10410
audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"
by bjornv@webrtc.org
· 10 years ago
88b4afa
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
by bjornv@webrtc.org
· 10 years ago
e2cf507
Use neteq_unittest_tools in audio_decoder_unittests
by henrik.lundin@webrtc.org
· 10 years ago
d18fd94
Fix double backslashes in incoming_video_stream.cc
by perkj@webrtc.org
· 10 years ago
3aa7f6d
Update makefiles after merge of Chromium at 82ca3b654cda
by Android Chromium Automerger
· 10 years ago
af3d97a
Add a simple AudioConverter class.
by andrew@webrtc.org
· 10 years ago
2e49acd
Only configure the SSL library in one place.
by henrike@webrtc.org
· 10 years ago
c91b433
Move (test) RtpFileReader to a lightweight target.
by pbos@webrtc.org
· 10 years ago
b9d18d9
Move scoped_ptr "free" functions into the webrtc namespace.
by andrew@webrtc.org
· 10 years ago
2b687fb
Upgrade our scoped_ptr copy to match Chromium's latest.
by andrew@webrtc.org
· 10 years ago
4011389
Merge from Chromium at DEPS revision 614f7b807940
by Torne (Richard Coles)
· 10 years ago
c3f7292
Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile
by henrik.lundin@webrtc.org
· 10 years ago
185e7f9
isacfix: Refactor big-endian reading and writing
by kwiberg@webrtc.org
· 10 years ago
Next »