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gerrit-public.fairphone.software
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fp2-dev
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platform
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external
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chromium_org
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third_party
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webrtc
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d3d364eb3382e3058f2d26f5e63431b23873101c
d3d364e
Fix compile errors in ViE with latest clang.
by andrew@webrtc.org
· 11 years ago
ae2d248
Update SincResampler with the latest Chromium code.
by andrew@webrtc.org
· 11 years ago
e155626
Clean creation of VideoEngine:
by andresp@webrtc.org
· 11 years ago
6027565
Formatted dtmf_queue.
by phoglund@webrtc.org
· 11 years ago
5187bfa
Add script to ensure virtual webcam is running.
by kjellander@webrtc.org
· 11 years ago
3759823
Disable clang C++11 warnings to permit OVERRIDE keyword.
by pbos@webrtc.org
· 11 years ago
bf8b98a
Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem.
by stefan@webrtc.org
· 11 years ago
db2e80b
Enable protobuf use in Chromium.
by andrew@webrtc.org
· 11 years ago
ee706f6
Update protoc.gypi to match Chromium's latest.
by andrew@webrtc.org
· 11 years ago
0e8ff34
Refactoring for typing detection
by niklas.enbom@webrtc.org
· 11 years ago
89f9266
Trigger a PLI if the duration of non-decodable frames exceeds a threshold.
by stefan@webrtc.org
· 11 years ago
5e0194b
VCM/Receiver: Return null when can't extract frame.
by mikhal@webrtc.org
· 11 years ago
570c6be
Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest
by mikhal@webrtc.org
· 11 years ago
47f874a
Relanding r3952: VCM: Updating receiver logic BUG=r1734 R=stefan@webrtc.org
by mikhal@webrtc.org
· 11 years ago
3acb72d
VCM/JB: Break and skip to key if possible
by mikhal@webrtc.org
· 11 years ago
34e0403
Fix clang errors in non-GYP_DEFINES=clang=1 build
by pbos@webrtc.org
· 11 years ago
5762c22
Fix jitter buffer unittest.
by stefan@webrtc.org
· 11 years ago
6ef6170
Correctly add packets to nack list when sequence number wraps.
by stefan@webrtc.org
· 11 years ago
358846f
Fix crash in pacer.
by pwestin@webrtc.org
· 11 years ago
d51d500
Revert r3952 "VCM: Updating receiver logic"
by stefan@webrtc.org
· 11 years ago
b0fca12
Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest."
by stefan@webrtc.org
· 11 years ago
79971c6
Landing 1399004, Minor clean up on the un-used _measureDelay code
by xians@webrtc.org
· 11 years ago
caba49f
Add an option to override the TestToStderr trace printout time.
by andrew@webrtc.org
· 11 years ago
cff84ec
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
by andrew@webrtc.org
· 11 years ago
e0aad3c
Updated WebRTC version number to 3.30
by elham@webrtc.org
· 11 years ago
25d4818
VCM/JB: Porting jitter_buffer_test to gtest.
by mikhal@webrtc.org
· 11 years ago
92b5ea0
Remove 44.1 kHz workaround from AudioDevice on PulseAudio.
by andrew@webrtc.org
· 11 years ago
6cd8727
Remove 44.1 kHz workaround from AudioDevice on WASAPI.
by andrew@webrtc.org
· 11 years ago
8e35807
Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert().
by sergeyu@chromium.org
· 11 years ago
cb0cb7d
VCM: Updating receiver logic
by mikhal@webrtc.org
· 11 years ago
5398063
Correct and update dir name
by leozwang@webrtc.org
· 11 years ago
c22830f
Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..."
by pbos@webrtc.org
· 11 years ago
a257915
Get rid of some unnecessary copying when sending REMBs.
by solenberg@webrtc.org
· 11 years ago
7ee60d0
Formatting ACM tests
by tina.legrand@webrtc.org
· 11 years ago
3794539
Fix when SetMinimumPlayoutDelay is configured to 0
by pwestin@webrtc.org
· 11 years ago
76318c5
Removing bad code resulting in flaky test.
by pwestin@webrtc.org
· 11 years ago
c06da8c
Adding trace and changing pacing constants
by pwestin@webrtc.org
· 11 years ago
6c98c86
Update third party license file
by niklas.enbom@webrtc.org
· 11 years ago
8f5edba
Bugfix custom call stop.
by pwestin@webrtc.org
· 11 years ago
c77e4da
Allow voe_cmd_test to select Opus mono (now the default).
by andrew@webrtc.org
· 11 years ago
c875f20
Relax VoE's max packet length threshold.
by andrew@webrtc.org
· 11 years ago
86f267b
Disabled flaky test.
by phoglund@webrtc.org
· 11 years ago
2d6a699
Revert 3933 "Remove traces of deprecated WebRtc_Word types."
by pbos@webrtc.org
· 11 years ago
bbb54b3
Remove traces of deprecated WebRtc_Word types.
by pbos@webrtc.org
· 11 years ago
74161fc
WebRTCDemo Android app to route audio to headphone when it's plugged in.
by braveyao@webrtc.org
· 11 years ago
b1c40c5
Replace Resampler with PushResampler in transmit_mixer.
by andrew@webrtc.org
· 11 years ago
a23b051
Consolidate common_audio into a single target.
by andrew@webrtc.org
· 11 years ago
22a3795
Add AEC suppression level option to audioproc.
by andrew@webrtc.org
· 11 years ago
10600ab
Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi .
by sergeyu@chromium.org
· 11 years ago
9e65a61
Fixes two bugs in receive statistics.
by stefan@webrtc.org
· 11 years ago
c5fbd58
Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync.
by pwestin@webrtc.org
· 11 years ago
6677d3c
VCM: Setting buffering delay in timing
by mikhal@webrtc.org
· 11 years ago
faec77d
Adding buffered mode to loopback test
by mikhal@webrtc.org
· 11 years ago
48e6258
Apply Chromium C++ style to RemoteRateControl.
by solenberg@webrtc.org
· 11 years ago
dfffece
Add DesktopCapturer interface for desktop capturers.
by sergeyu@chromium.org
· 11 years ago
78a4a73
Don't reset the last je value and mode
by mikhal@webrtc.org
· 11 years ago
45325fd
Add a wrapper around PushSincResampler and the old Resampler.
by andrew@webrtc.org
· 11 years ago
45115c8
Fix two issues where we might end up busy looping in decoder_render mode.
by stefan@webrtc.org
· 11 years ago
8517f00
Enable Nack pacing.
by pwestin@webrtc.org
· 11 years ago
8a159ad
Removing vie file related code from vie_custom_call
by mikhal@webrtc.org
· 11 years ago
fd7a1b7
Fixed remaining nits from Stefan
by pwestin@webrtc.org
· 11 years ago
8fdc74b
Add a push-based wrapper around SincResampler.
by andrew@webrtc.org
· 11 years ago
b537037
Add comfort noise disabling and routing mode selection to audioproc.
by andrew@webrtc.org
· 11 years ago
c960e32
Removing another instance of file api
by mikhal@webrtc.org
· 11 years ago
71645c8
Fix the encoder pause logic. BUG=1691
by pwestin@webrtc.org
· 11 years ago
1c42057
VCM: Adding API for the size(duration) of the jitter buffer.
by mikhal@webrtc.org
· 11 years ago
96535b0
VCM/JB: Using last decoded state for waiting for key
by mikhal@webrtc.org
· 11 years ago
a4d060f
VCM/JB: FrameForDecoding->IncompleteFrameForDecoding
by mikhal@webrtc.org
· 11 years ago
2423690
Disabling avi file interface
by mikhal@webrtc.org
· 11 years ago
911d567
Avoid adding duplicates in pacer lists.
by pwestin@webrtc.org
· 11 years ago
9ad7cce
Make sure timestamps are monotonically increasing.
by stefan@webrtc.org
· 11 years ago
39ea6be
Revert 3892 "VCM/JB: Using last decoded state for waiting for key"
by andrew@webrtc.org
· 11 years ago
1efb11c
Adding extra options to interact with external encoder/decoder.
by andresp@webrtc.org
· 11 years ago
3d0d20f
VCM/JB: Using last decoded state for waiting for key
by mikhal@webrtc.org
· 11 years ago
b35efcc
Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."
by stefan@webrtc.org
· 11 years ago
a09c0c4
Buf fix for r3883.
by turaj@webrtc.org
· 11 years ago
65e6f91
Add a default RTT to CallStats and use different values for buffered/real-time mode.
by stefan@webrtc.org
· 11 years ago
f97287b
VP8: Avoid copying the codec struct on Reset().
by pbos@webrtc.org
· 11 years ago
38617bc
BUG=1351
by mflodman@webrtc.org
· 11 years ago
18a4532
VCM/JB: Skip to the next complete key frame
by mikhal@webrtc.org
· 11 years ago
7a14b35
Updated the sync module with a slow moving filter
by pwestin@webrtc.org
· 11 years ago
e471f0c
Improve AV-sync when initial delay is set and NetEq has long buffer.
by turaj@webrtc.org
· 11 years ago
7070d8f
emove desktop_capture.gypi from modules.gyp
by kjellander@webrtc.org
· 11 years ago
fe2bce3
Removed unused variable.
by mflodman@webrtc.org
· 11 years ago
fb5b5cb
Fixing Coverity issues.
by mflodman@webrtc.org
· 11 years ago
f06fff3
Update iOS build script to run on bots.
by kjellander@webrtc.org
· 11 years ago
437dfaf
Revert 3876
by mikhal@webrtc.org
· 11 years ago
742f3e9
VCM/Receiver: Only update render time when decoding
by mikhal@webrtc.org
· 11 years ago
1ccedf6
Ensure build_demo.py run subprocesses with bash shell.
by kjellander@webrtc.org
· 11 years ago
1b4af0d
Add the build script of the voice engine for iOS.
by sjlee@webrtc.org
· 11 years ago
64823d1
revert r3871
by mikhal@webrtc.org
· 11 years ago
e8a772d
- Replace the BWE_MIN and BWE_MAX macros with std::min and std::max
by solenberg@webrtc.org
· 11 years ago
342ee85
Apply Chromium C++ style to BitRateStats.
by solenberg@webrtc.org
· 11 years ago
69b0d2c
New ViE interface.
by mflodman@webrtc.org
· 11 years ago
27334a1
Add lock to prevent possible rare race condition in Win coreAudio capture implementation.
by braveyao@webrtc.org
· 11 years ago
5b10324
Add desktop_capture directory for screen and window capturers.
by sergeyu@chromium.org
· 11 years ago
42a1a30
Updating delay for first value
by mikhal@webrtc.org
· 11 years ago
95d5bc6
Remove libvpx pre-processor conditions and conditional compile of default temporal layers files.
by andresp@webrtc.org
· 11 years ago
0a884c0
Revert "Updating test file contents to emmastjernloef"
by kjellander@webrtc.org
· 11 years ago
da3ad08
Updating test file contents to emmastjernloef
by kjellander@webrtc.org
· 11 years ago
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