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webrtc
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3b6f72889e2ebac50ee5d21836b0b7ca2ba65d26
3b6f728
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
fa2dd22
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
by solenberg@webrtc.org
· 11 years ago
2ffc8bf
Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..."
by wu@webrtc.org
· 11 years ago
365ca40
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
0c0795e
Fix broken audio.
by leozwang@webrtc.org
· 11 years ago
a0bba27
G722-stereo has been missing when creating AudioDecoder.
by turaj@webrtc.org
· 11 years ago
88a7940
NetEq4 fails if the first packets inserted in are out-of-band DTMFs.
by turaj@webrtc.org
· 11 years ago
88f12ab
Fix flakiness in network up/down event tests when running under memcheck.
by stefan@webrtc.org
· 11 years ago
6fc5215
WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14.
by fischman@webrtc.org
· 11 years ago
dca71b2
Add interface to signal a network down event.
by stefan@webrtc.org
· 11 years ago
6e34ceb
Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events).
by henrike@webrtc.org
· 11 years ago
f386e2b
Remove VoE's default call in Trace::SetLevelFilter.
by andrew@webrtc.org
· 11 years ago
aa0fcd7
Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.
by solenberg@webrtc.org
· 11 years ago
31b4448
Alphabetize include order in fake_voe_external_media.h.
by andrew@webrtc.org
· 11 years ago
a7a643e
Restart Android capture after orientation change. Also prevent an NPE on exit.
by fischman@webrtc.org
· 11 years ago
13f66d1
Add some VoE and AudioProcessing mocks.
by andrew@webrtc.org
· 11 years ago
8826e34
Refactor unittest trace printouts to a separate class.
by andrew@webrtc.org
· 11 years ago
0c1f10b
Enable the below APIs for iOS.
by sjlee@webrtc.org
· 11 years ago
9522792
Introduced pause and resume to the pacer
by pwestin@webrtc.org
· 11 years ago
f49577f
Updated WebRTC version to 3.27
by elham@webrtc.org
· 11 years ago
7fd368f
Bugfix for extended RTP/RTCP test
by pwestin@webrtc.org
· 11 years ago
c075e25
Move the VIE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
4c27c03
Add trace printouts to all unit tests.
by andrew@webrtc.org
· 11 years ago
e1198e6
Add min and target bitrate to VideoCodec.
by marpan@webrtc.org
· 11 years ago
aa922de
Move the VoE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
af6aa7b
Creating a copy of Udp transport under webrtc/test
by pwestin@webrtc.org
· 11 years ago
495c563
Cleanup nanosleep -> SleepMs Remove some leftover stuff
by hta@webrtc.org
· 11 years ago
4a48fd6
WebRtc_Word -> stdint in audio_coding/g711/
by pbos@webrtc.org
· 11 years ago
a2df078
Remove incorrect asserts.
by stefan@webrtc.org
· 11 years ago
e49f252
WebRtc_Word -> stdint in audio_coding/cng/
by pbos@webrtc.org
· 11 years ago
0f2782f
Fix -Wstring-conversion warnings.
by wu@webrtc.org
· 11 years ago
5815b7c
Thread safety issue fix in incoming_video_stream.cc. See issue 1465.
by vikasmarwaha@webrtc.org
· 11 years ago
757bf0f
Account for header inside I420Encoder::InitEncode.
by pbos@webrtc.org
· 11 years ago
6313692
Follow-up fix for r3681.
by stefan@webrtc.org
· 11 years ago
46618db
Fixed initialization of SPL in echo_control_mobile.
by kma@webrtc.org
· 11 years ago
9cd73ed
Android: rename android_build_type gyp variable.
by wjia@webrtc.org
· 11 years ago
ffe2ec6
Updated WebRTC version number to 3.26
by elham@webrtc.org
· 11 years ago
692c69d
Fix framerate sent to account for actually sent frames.
by stefan@webrtc.org
· 11 years ago
72e204a
Change VCM interface to take target bitrate in bits per second.
by stefan@webrtc.org
· 11 years ago
e3339fc
Generic video-codec support.
by pbos@webrtc.org
· 11 years ago
0b5c7f1
Revert the deletion of test_api_nack.cc in r3674.
by stefan@webrtc.org
· 11 years ago
87ef38e
Truncated delay quality to avoid negative return values
by bjornv@webrtc.org
· 11 years ago
946d240
Adding RTX on source
by mikhal@webrtc.org
· 11 years ago
cf9ab12
Adding Opus frame length test
by tina.legrand@webrtc.org
· 11 years ago
aecc559
Fixed a crash issue in NSX module.
by kma@webrtc.org
· 11 years ago
912b7f7
Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004
by pwestin@webrtc.org
· 11 years ago
40749c1
Added destructors for tests to control destruct order
by pwestin@webrtc.org
· 11 years ago
b793abe
Increasing size of nack list in buffered mode.
by mikhal@webrtc.org
· 11 years ago
2daec4c
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
2637d61
Refactor webrtc specific Event implementation to an EventFactory.
by stefan@webrtc.org
· 11 years ago
15a03fd
Remove DTMF detection. Talk team has been in the loop and there is no need for
by turaj@webrtc.org
· 11 years ago
df00856
Tool found: pass by value when pass by reference is better in system wrapper unit test.
by henrike@webrtc.org
· 11 years ago
0c46af6
Change intrinsic code in isac fix to let it pass chrome clang compiler.
by kma@webrtc.org
· 11 years ago
160b327
Fixes issue detected by tool.
by henrike@webrtc.org
· 11 years ago
f0f1dc2
Removed redundant VP8 width/height and made sure the generic width/height is set.
by stefan@webrtc.org
· 11 years ago
23a7047
Revert "Internal clean up: removing unused include line."
by dwkang@webrtc.org
· 11 years ago
177ec87
Internal clean up: removing unused include line.
by dwkang@webrtc.org
· 11 years ago
d451969
Fixed issue 1497 in iSAC fixed point.
by kma@webrtc.org
· 11 years ago
9f4ae03
Fix frame_editing_unittest reference file handling.
by kjellander@webrtc.org
· 11 years ago
6b12f97
Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform.
by kma@webrtc.org
· 11 years ago
8623e9b
Lazy capture_device_info acquisition.
by pbos@webrtc.org
· 11 years ago
7b6cbc9
Refactor barcode decoder to use Zxing's C++ version
by kjellander@webrtc.org
· 11 years ago
04b8071
Splitting out video_coding_test executable again.
by kjellander@webrtc.org
· 11 years ago
e4a0d58
Fixed an assembly code error in AECM for ARMv7.
by kma@webrtc.org
· 11 years ago
02c9104
Disable frame dropper for screenshare mode.
by stefan@webrtc.org
· 11 years ago
e33b9de
Move video_coding OWNERS to video_coding/.
by stefan@webrtc.org
· 11 years ago
2ce235e
Fixed a ton of Python lint errors, enabled python lint checking.
by phoglund@webrtc.org
· 11 years ago
fe6b986
Fix debug file buffer bug introduced in r3574.
by andrew@webrtc.org
· 11 years ago
78e450f
Enabling bufffering mode with no sync module or VoE
by mikhal@webrtc.org
· 11 years ago
0f919be
Remove the error return on SetAGC failure introduced by r3605.
by andrew@webrtc.org
· 11 years ago
7f34a88
1. Updated test pages to include Chrome Frame meta tag
by elham@webrtc.org
· 11 years ago
4c2225f
Adds new AEC API to audio_processing.
by bjornv@webrtc.org
· 11 years ago
3c365b2
Fix for build error on android introduced with r3609.
by stefan@webrtc.org
· 11 years ago
15fc445
Split the NACK list into multiple RTCPs if it's too big.
by stefan@webrtc.org
· 11 years ago
8665399
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
b79627b
Expose the capture-side AudioProcessing object and allow it to be injected.
by andrew@webrtc.org
· 11 years ago
d432df4
AEC Refactoring: Removes lint warning
by bjornv@webrtc.org
· 11 years ago
640d1bb
Updated version number to 3.25
by elham@webrtc.org
· 11 years ago
bb78b2c
Refactor NACK list creation to build the NACK list as packets arrive.
by stefan@webrtc.org
· 11 years ago
e294b5f
compile fix for get_nprocs() with uClibc
by phoglund@webrtc.org
· 11 years ago
46e08da
Fixed coverity defects (CID 14657 and 14656).
by phoglund@webrtc.org
· 11 years ago
3d1a637
VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView.
by fischman@webrtc.org
· 11 years ago
4de0a10
Don't upsample the capture signal early.
by andrew@webrtc.org
· 11 years ago
03c41d7
Update integration tests for idempotent RTP header settings.
by bemasc@google.com
· 11 years ago
ea67fe0
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
b563e5e
Properly error check calls to AudioProcessing.
by andrew@webrtc.org
· 11 years ago
1794994
Enable External MediaProcessing on Mobile
by leozwang@webrtc.org
· 11 years ago
d1932fc
Make RtpHeaderExtensionMap::Register and ::Deregister idempotent.
by bemasc@google.com
· 11 years ago
f1de5e9
Return an error when greater than 16 kHz is used with AECM.
by andrew@webrtc.org
· 11 years ago
c6242c9
Destroy VCM and VPM instead of delete.
by mflodman@webrtc.org
· 11 years ago
f653f97
Limit ARM instruction "strheq" to Apple's clang compiler only.
by kma@webrtc.org
· 11 years ago
d836fe5
Turn off error concealment in videoprocessor_integration tests.
by marpan@webrtc.org
· 11 years ago
0e686e6
Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG.
by braveyao@webrtc.org
· 11 years ago
1827258
Rewrite the jitter buffer statistics test and put make it robust under valgrind.
by stefan@webrtc.org
· 11 years ago
840fa03
AEC Refactoring:
by bjornv@webrtc.org
· 11 years ago
6e7945f
Fix to send a full NACK list at least roughly once every 1.5 x RTT.
by stefan@webrtc.org
· 11 years ago
5666f6c
Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform.
by kma@webrtc.org
· 11 years ago
db32d60
Refactor WebRtc_CreateBuffer to return the instance.
by andrew@webrtc.org
· 11 years ago
001d6ed
Use ninja with merge_libs.py on Windows and clean up.
by andrew@webrtc.org
· 11 years ago
ef31f50
Force a memcpy directly from the AEC ring buffer.
by andrew@webrtc.org
· 11 years ago
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