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gerrit-public.fairphone.software
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webrtc
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2740f66d22013edfc82062ab23545db801a5a7b5
2740f66
Remove output files from .gitignore.
by andrew@webrtc.org
· 13 years ago
26b9777
Only trigger one call to OnNetworkChanged for each incoming RTCP packet.
by mflodman@webrtc.org
· 13 years ago
471e83e
Refactored ViESharedData.
by mflodman@webrtc.org
· 13 years ago
9af365d
Fixing VP8 RTP parser bug
by henrik.lundin@webrtc.org
· 13 years ago
6f2c016
Updating to VP8 RTP spec rev -02
by henrik.lundin@webrtc.org
· 13 years ago
517dcb7
Migrate to using the same gyp files and gypi generator script as Chromium.
by stefan@webrtc.org
· 13 years ago
6d26ef7
Refactored ViESender.
by mflodman@webrtc.org
· 13 years ago
d492f72
Added empty unit tests to get code coverage measured.
by kjellander@webrtc.org
· 13 years ago
55d81ea
ViE Custom Call observer now using pointers, fixed protection method and miscellaneous TODO cleanup
by amyfong@webrtc.org
· 13 years ago
ba028a3
Fix sample rate printout in process_test.
by andrew@webrtc.org
· 13 years ago
c4ac357
Fix spelling errors in DEPS.
by andrew@webrtc.org
· 13 years ago
f3d10d3
Fixed release compilation error-warnings.
by phoglund@webrtc.org
· 13 years ago
3379b0c
Fixing resources script on Windows
by kjellander@webrtc.org
· 13 years ago
c4c56ed
Rewrote vie_auto_test to use googletest macros.
by phoglund@webrtc.org
· 13 years ago
0c80c7f
Script for automatic download of resources.
by kjellander@webrtc.org
· 13 years ago
48b68c0
Added support for 96 kHz sampling frequency.
by bjornv@webrtc.org
· 13 years ago
4257790
NetEQ-related bug in ACM
by henrik.lundin@webrtc.org
· 13 years ago
543c3ea
Fixing Release compilation errors
by kjellander@webrtc.org
· 13 years ago
89ab652
Cleaning up NetEQ statistics
by henrik.lundin@webrtc.org
· 13 years ago
df10de4
Removing statistics API from NetEQ
by henrik.lundin@webrtc.org
· 13 years ago
7d3e949
This CL is to support certain audio devices which don't offer volume control. Try to be more compatible to those rare cases.
by braveyao@webrtc.org
· 13 years ago
2b838b4
video_coding: updating the session info unit test following recent changes
by mikhal@webrtc.org
· 13 years ago
425b377
video_coding: Updating internal_defines to resolve latest build error. Refers to JB flush update.
by mikhal@webrtc.org
· 13 years ago
f13388f
video_coding: Requesting a key frame after a JB flush
by mikhal@webrtc.org
· 13 years ago
6b9a7f8
video_coding: Allowing for a decodable state independent of selective nacking
by mikhal@webrtc.org
· 13 years ago
828af1b
Add lookahead to the delay estimator.
by andrew@webrtc.org
· 13 years ago
5a52939
Make DMO init safe when not supported.
by andrew@webrtc.org
· 13 years ago
91617ff
by henrikg@webrtc.org
· 13 years ago
dfe89e3
Move ViE main/test/AutoTest to test/auto_test.
by mflodman@webrtc.org
· 13 years ago
8594f76
Add a gyp variable for AEC debug dumps.
by andrew@webrtc.org
· 13 years ago
a249f35
Correct several makefile errors for Android build.
by kma@webrtc.org
· 13 years ago
6830bdd
Fix xcode build.
by mflodman@webrtc.org
· 13 years ago
94ea32e
Move video_engine/source* to video_engine/. No code changes except paths in gyp-files.
by mflodman@webrtc.org
· 13 years ago
274c2ef
Adding empty test method required to get code coverage
by kjellander@webrtc.org
· 13 years ago
3caa327
VP8 wrapper: Turn on some mild amount of deblocking in post-processing.
by marpan@webrtc.org
· 13 years ago
ce9d89d
Fixes linux build error introduced in r980.
by henrike@webrtc.org
· 13 years ago
ad98a3e
Fixes TEST crash triggered by webrtc-codereview.appspot.com/268014.
by henrike@webrtc.org
· 13 years ago
31d3070
Addressed review comments from http://webrtc-codereview.appspot.com/256004/
by henrike@webrtc.org
· 13 years ago
ced1186
Changed keyword __restrict__ to __restrict.
by kma@webrtc.org
· 13 years ago
3798ecb
Made CPU initialization on Windows lazy to prevent long startup time.
by henrike@webrtc.org
· 13 years ago
543611a
Reverting r972 due to compilation error on Windows Release build.
by kjellander@webrtc.org
· 13 years ago
2f047cc
Removed unnecessary variable to avoid compiler error on Win.
by bjornv@webrtc.org
· 13 years ago
ba74924
Remove use of exceptions in NetEQ test code
by henrik.lundin@webrtc.org
· 13 years ago
6a9835d
Delay estimator structural changes.
by bjornv@webrtc.org
· 13 years ago
fa9b016
Optimized WebRtcIsacfix_AutocorrFix() function for iSAC fix.
by kma@webrtc.org
· 13 years ago
f556b9d
This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend().
by braveyao@webrtc.org
· 13 years ago
3041c62
Updating libvpx to revision bdd35c13: This revision includes a fix for frame rate adaptation - avoid resetting the frame rate to 30 on each update.
by mikhal@webrtc.org
· 13 years ago
917fa6b
ViE Custom Call added SetImageScaleStatus toggle option and other changes.
by amyfong@webrtc.org
· 13 years ago
cd7b57e
Fixing release compilation error
by kjellander@webrtc.org
· 13 years ago
3f1cb8e
Restructuring and adding dummy unit test target.
by kjellander@webrtc.org
· 13 years ago
cc2ecb3
Restructuring and adding dummy unit test target.
by kjellander@webrtc.org
· 13 years ago
b72268e
Restructuring and adding dummy unit test target.
by kjellander@webrtc.org
· 13 years ago
64a897a
Restructuring and adding dummy unit test target.
by kjellander@webrtc.org
· 13 years ago
1310e0b
Adding third_party/cygwin to git ignore
by kjellander@webrtc.org
· 13 years ago
8f89f09
Note: this patch may seem intimidating but it mostly moves code around and renames things. There are quite few actual changes.
by phoglund@webrtc.org
· 13 years ago
c05b56a
Fixing compilation error
by kjellander@webrtc.org
· 13 years ago
9f9af7d
Remove Peerconnection Dev branch.
by perkj@webrtc.org
· 13 years ago
0403ef4
Restructuring and adding unit test targets on project level instead of in common_audio.
by kjellander@webrtc.org
· 13 years ago
7eba346
Remove module targets due to Linux-Release errors.
by andrew@webrtc.org
· 13 years ago
337dc68
Included modules in webrtc.gyp and fixed build errors.
by phoglund@webrtc.org
· 13 years ago
af26f64
Inband DTMF stereo support
by niklas.enbom@webrtc.org
· 13 years ago
e33a102
Resubmitting http://webrtc-codereview.appspot.com/269007/
by niklas.enbom@webrtc.org
· 13 years ago
fcf33eb
Limit number of send-side BWE increases to one per second.
by stefan@webrtc.org
· 13 years ago
3d1ef0c
Add an OWNERS file for python_charts.
by andrew@webrtc.org
· 13 years ago
d0e5b96
Fix Amy's email address.
by andrew@webrtc.org
· 13 years ago
81d4499
Microphone volume on Mac not being printed properly due
by punyabrata@webrtc.org
· 13 years ago
755b04a
Add RMS computation for the RTP level indicator.
by andrew@webrtc.org
· 13 years ago
6a85b17
Potential fix for crash after Mac sleep.
by andrew@webrtc.org
· 13 years ago
85596d5
Setting completeFrame to true for all created encoded images.
by kjellander@webrtc.org
· 13 years ago
cde1e7f
Use a TraceNoop instance when tracing disabled (to be used in Chromium).
by tommi@webrtc.org
· 13 years ago
bc91d5a
NetEQ tests
by henrik.lundin@webrtc.org
· 13 years ago
aedf3a7
Loosening makefile rule so it matches *.target.mk but not Android.mk files.
by kjellander@webrtc.org
· 13 years ago
a02ef1a
Fix broken tree.
by mflodman@webrtc.org
· 13 years ago
1f69c03
Added size sanity check for copying app specific RTCP data.
by mflodman@webrtc.org
· 13 years ago
33df533
Change luminance of all pixels by a specified value.
by henrik.lundin@webrtc.org
· 13 years ago
7de0765
Disables a flaky metric test.
by stefan@webrtc.org
· 13 years ago
ded85f1
Enable WEBRTC_NO_TRACE for Chromium builds.
by tommi@webrtc.org
· 13 years ago
0db7dc6
Add file-playing channels to voe_cmd_test.
by andrew@webrtc.org
· 13 years ago
cd82438
Unpack the full set of audioproc data.
by andrew@webrtc.org
· 13 years ago
d71d480
Fixed a build error of audio conference mixer in Android.
by kma@webrtc.org
· 13 years ago
b351d6a
Reverting rev 929 due to failing assert on Linux.
by stefan@webrtc.org
· 13 years ago
9b18ed6
Removed incorrect dependency.
by phoglund@webrtc.org
· 13 years ago
fd3a0ef
RTP bw estimate fix.
by mflodman@webrtc.org
· 13 years ago
1144ba2
Base and codec tests now run verify output and render to file instead of to screen.
by phoglund@webrtc.org
· 13 years ago
62e48eb
adding owners for test
by niklas.enbom@webrtc.org
· 13 years ago
50b3cbe
First pass. You can now enable a stereo codec and send and receive. This does not include more advances use cases (DTMF etc), but I'd rather keep the CLs manageable.
by niklas.enbom@webrtc.org
· 13 years ago
b61c410
Fixed a couple of Android makefiles to let voe and vie build properly.
by kma@webrtc.org
· 13 years ago
13318ef
(1) Corrected the makefile for testing iLBC in Android, and changed the location of the test makefile to make it consistent with audio_processing.
by kma@webrtc.org
· 13 years ago
7a4eb28
Calculate the available bandwidth before sending a TMMBR
by mflodman@webrtc.org
· 13 years ago
637a59e
jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined.
by mflodman@webrtc.org
· 13 years ago
855a77c
Audio Coding Module: Fixing a bug that prevented the encoder from being re-initialized when changing codec from mono to stereo.
by tina.legrand@webrtc.org
· 13 years ago
c4f129f
Improve the mixing saturation protection scheme.
by andrew@webrtc.org
· 13 years ago
41f3855
Upgrade libvpx to b615a6d4.
by marpan@webrtc.org
· 13 years ago
d30b688
Remove TraceScan executable.
by andrew@webrtc.org
· 13 years ago
4b13fc9
Add delay modification to process_test.
by andrew@webrtc.org
· 13 years ago
2f32b5c
Fixes an issue where file playing could happen at a lower sampling frequency than the file.
by henrike@webrtc.org
· 13 years ago
eb4ef17
Removing vplib include and VideoInterpolator when not needed
by mikhal@webrtc.org
· 13 years ago
488ed92
Removing exceptions since not used
by kjellander@webrtc.org
· 13 years ago
c3a4dcd
Removing exceptions since not used
by kjellander@webrtc.org
· 13 years ago
ad79d6f
Removing exceptions since not used
by kjellander@webrtc.org
· 13 years ago
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