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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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f6582783eb30c682d721ff080bc367c08170dd06
f658278
Refactor unittest trace printouts to a separate class.
by andrew@webrtc.org
· 11 years ago
8cfba7e
Enable the below APIs for iOS.
by sjlee@webrtc.org
· 11 years ago
60c8100
Introduced pause and resume to the pacer
by pwestin@webrtc.org
· 11 years ago
e760243
Updated WebRTC version to 3.27
by elham@webrtc.org
· 11 years ago
d3eb512
Bugfix for extended RTP/RTCP test
by pwestin@webrtc.org
· 11 years ago
9c3b7bd
Move the VIE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
680fbc5
Add trace printouts to all unit tests.
by andrew@webrtc.org
· 11 years ago
90fa4a1
Add min and target bitrate to VideoCodec.
by marpan@webrtc.org
· 11 years ago
9c0b169
Move the VoE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
ce2d125
Creating a copy of Udp transport under webrtc/test
by pwestin@webrtc.org
· 11 years ago
ed6b4c8
Cleanup nanosleep -> SleepMs Remove some leftover stuff
by hta@webrtc.org
· 11 years ago
e3abb18
WebRtc_Word -> stdint in audio_coding/g711/
by pbos@webrtc.org
· 11 years ago
48ec040
Remove incorrect asserts.
by stefan@webrtc.org
· 11 years ago
326becd
WebRtc_Word -> stdint in audio_coding/cng/
by pbos@webrtc.org
· 11 years ago
c226567
Fix -Wstring-conversion warnings.
by wu@webrtc.org
· 11 years ago
f99f63f
Thread safety issue fix in incoming_video_stream.cc. See issue 1465.
by vikasmarwaha@webrtc.org
· 11 years ago
d579a2a
Account for header inside I420Encoder::InitEncode.
by pbos@webrtc.org
· 11 years ago
06d1e8f
Follow-up fix for r3681.
by stefan@webrtc.org
· 11 years ago
6f1f826
Fixed initialization of SPL in echo_control_mobile.
by kma@webrtc.org
· 11 years ago
aef22a7
Android: rename android_build_type gyp variable.
by wjia@webrtc.org
· 11 years ago
035c96a
Updated WebRTC version number to 3.26
by elham@webrtc.org
· 11 years ago
ebdc04d
Fix framerate sent to account for actually sent frames.
by stefan@webrtc.org
· 11 years ago
3be5a98
Change VCM interface to take target bitrate in bits per second.
by stefan@webrtc.org
· 11 years ago
a2e9124
Generic video-codec support.
by pbos@webrtc.org
· 11 years ago
857be46
Revert the deletion of test_api_nack.cc in r3674.
by stefan@webrtc.org
· 11 years ago
1f71c06
Truncated delay quality to avoid negative return values
by bjornv@webrtc.org
· 11 years ago
072c9b6
Adding RTX on source
by mikhal@webrtc.org
· 11 years ago
feaa409
Adding Opus frame length test
by tina.legrand@webrtc.org
· 11 years ago
897e86f
Fixed a crash issue in NSX module.
by kma@webrtc.org
· 11 years ago
9a7b9f7
Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004
by pwestin@webrtc.org
· 11 years ago
a891566
Added destructors for tests to control destruct order
by pwestin@webrtc.org
· 11 years ago
25023aa
Increasing size of nack list in buffered mode.
by mikhal@webrtc.org
· 11 years ago
66ccc6e
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
by pwestin@webrtc.org
· 11 years ago
9fe8f22
Refactor webrtc specific Event implementation to an EventFactory.
by stefan@webrtc.org
· 11 years ago
d4caede
Remove DTMF detection. Talk team has been in the loop and there is no need for
by turaj@webrtc.org
· 11 years ago
b661eae
Tool found: pass by value when pass by reference is better in system wrapper unit test.
by henrike@webrtc.org
· 11 years ago
a284f1d
Change intrinsic code in isac fix to let it pass chrome clang compiler.
by kma@webrtc.org
· 11 years ago
2c62fd9
Fixes issue detected by tool.
by henrike@webrtc.org
· 11 years ago
020e6ad
Removed redundant VP8 width/height and made sure the generic width/height is set.
by stefan@webrtc.org
· 11 years ago
9b78141
Revert "Internal clean up: removing unused include line."
by dwkang@webrtc.org
· 11 years ago
1a8d06e
Internal clean up: removing unused include line.
by dwkang@webrtc.org
· 11 years ago
2d2bfb0
Fixed issue 1497 in iSAC fixed point.
by kma@webrtc.org
· 11 years ago
38a5679
Fix frame_editing_unittest reference file handling.
by kjellander@webrtc.org
· 11 years ago
c51b060
Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform.
by kma@webrtc.org
· 11 years ago
2a3949f
Lazy capture_device_info acquisition.
by pbos@webrtc.org
· 11 years ago
46672bb
Refactor barcode decoder to use Zxing's C++ version
by kjellander@webrtc.org
· 11 years ago
c96125a
Splitting out video_coding_test executable again.
by kjellander@webrtc.org
· 11 years ago
ad807de
Fixed an assembly code error in AECM for ARMv7.
by kma@webrtc.org
· 11 years ago
2dbb66b
Disable frame dropper for screenshare mode.
by stefan@webrtc.org
· 11 years ago
2a070a5
Move video_coding OWNERS to video_coding/.
by stefan@webrtc.org
· 11 years ago
2733e12
Fixed a ton of Python lint errors, enabled python lint checking.
by phoglund@webrtc.org
· 11 years ago
4621446
Fix debug file buffer bug introduced in r3574.
by andrew@webrtc.org
· 11 years ago
ace0823
Enabling bufffering mode with no sync module or VoE
by mikhal@webrtc.org
· 11 years ago
66b0c5d
Remove the error return on SetAGC failure introduced by r3605.
by andrew@webrtc.org
· 11 years ago
ad3fd52
1. Updated test pages to include Chrome Frame meta tag
by elham@webrtc.org
· 11 years ago
333987b
Adds new AEC API to audio_processing.
by bjornv@webrtc.org
· 11 years ago
08b9b59
Fix for build error on android introduced with r3609.
by stefan@webrtc.org
· 11 years ago
2654c43
Split the NACK list into multiple RTCPs if it's too big.
by stefan@webrtc.org
· 11 years ago
df1cfd1
None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise.
by turaj@webrtc.org
· 11 years ago
6316d17
Expose the capture-side AudioProcessing object and allow it to be injected.
by andrew@webrtc.org
· 11 years ago
2c1f9d4
AEC Refactoring: Removes lint warning
by bjornv@webrtc.org
· 11 years ago
87d8f2d
Updated version number to 3.25
by elham@webrtc.org
· 11 years ago
eeaacdb
Refactor NACK list creation to build the NACK list as packets arrive.
by stefan@webrtc.org
· 11 years ago
552f230
compile fix for get_nprocs() with uClibc
by phoglund@webrtc.org
· 11 years ago
89cc166
Fixed coverity defects (CID 14657 and 14656).
by phoglund@webrtc.org
· 11 years ago
4aa2314
VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView.
by fischman@webrtc.org
· 11 years ago
7d2689d
Don't upsample the capture signal early.
by andrew@webrtc.org
· 11 years ago
3da576e
Update integration tests for idempotent RTP header settings.
by bemasc@google.com
· 11 years ago
13a186f
Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools.
by kma@webrtc.org
· 11 years ago
ad179ce
Properly error check calls to AudioProcessing.
by andrew@webrtc.org
· 11 years ago
8e4340d
Enable External MediaProcessing on Mobile
by leozwang@webrtc.org
· 11 years ago
8648aad
Make RtpHeaderExtensionMap::Register and ::Deregister idempotent.
by bemasc@google.com
· 11 years ago
bb2973a
Return an error when greater than 16 kHz is used with AECM.
by andrew@webrtc.org
· 11 years ago
1dcba31
Destroy VCM and VPM instead of delete.
by mflodman@webrtc.org
· 11 years ago
22fc115
Limit ARM instruction "strheq" to Apple's clang compiler only.
by kma@webrtc.org
· 11 years ago
fef10a3
Turn off error concealment in videoprocessor_integration tests.
by marpan@webrtc.org
· 11 years ago
51d5c6d
Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG.
by braveyao@webrtc.org
· 11 years ago
eeb8b8f
Rewrite the jitter buffer statistics test and put make it robust under valgrind.
by stefan@webrtc.org
· 11 years ago
12509cf
AEC Refactoring:
by bjornv@webrtc.org
· 11 years ago
6f93416
Fix to send a full NACK list at least roughly once every 1.5 x RTT.
by stefan@webrtc.org
· 11 years ago
9ea9696
Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform.
by kma@webrtc.org
· 11 years ago
24e40df
Refactor WebRtc_CreateBuffer to return the instance.
by andrew@webrtc.org
· 11 years ago
90aa2fe
Use ninja with merge_libs.py on Windows and clean up.
by andrew@webrtc.org
· 11 years ago
c60d33b
Force a memcpy directly from the AEC ring buffer.
by andrew@webrtc.org
· 11 years ago
86e2e1f
Remove unneeded libvpx path from vp8 include_dirs.
by andrew@webrtc.org
· 11 years ago
101eb2c
Refactor ring_buffer interface, add a feature and a test.
by andrew@webrtc.org
· 11 years ago
4211e6a
New attempt at fixing hard-coded libvpx source.
by phoglund@webrtc.org
· 11 years ago
bf03dd4
Revert "Fixing hard-coded libvpx source path."
by phoglund@webrtc.org
· 11 years ago
7f53b4c
Fixing hard-coded libvpx source path.
by phoglund@webrtc.org
· 11 years ago
933af52
Ported assembly coding in APM from Android to iOS.
by kma@webrtc.org
· 11 years ago
ca65c51
Handle multiple calls to set initial delay
by mikhal@webrtc.org
· 11 years ago
c1f0f68
Remove WEBRTC_TRACE completely when tracing is disabled.
by wjia@webrtc.org
· 11 years ago
fa9a633
Minor bug fix in maxFPS parameter declaration.
by vikasmarwaha@webrtc.org
· 11 years ago
c1c5aad
Fix for WebRTC Issue 1384. Some cameras return 0 fps for all capabilities which causes divide-by-zero.
by vikasmarwaha@webrtc.org
· 11 years ago
85e32df
MIPS optimizations for Signal Processing Library patch01
by andrew@webrtc.org
· 11 years ago
ed1c7f4
AEC refactoring: Moved typedefs to _internal.h
by bjornv@webrtc.org
· 11 years ago
4f33453
Changing non-const reference arguments to pointers, ACM
by tina.legrand@webrtc.org
· 11 years ago
db1733f
Misc cleanups to webrtc/android code:
by fischman@webrtc.org
· 11 years ago
c01c6c3
Refactoring AEC: AecCore struct made private
by bjornv@webrtc.org
· 11 years ago
325931a
Refactor AEC: PowerLevel
by bjornv@webrtc.org
· 11 years ago
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