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c18957e8773e558de869a809df97ec6ca924794b
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c957ffc
Fixed potential crash if rtp packet history is completely full.
by sprang@webrtc.org
· 10 years ago
c420a86
Change name for local CriticalSectionScoped variable
by henrik.lundin@webrtc.org
· 10 years ago
a1dfbf1
WebRtcG722_Decode: Input array should be const uint8_t[]
by kwiberg@webrtc.org
· 10 years ago
026b892
Using << on an int8_t or uint8_t will output a character rather than a number.
by pkasting@chromium.org
· 10 years ago
005b6ff
Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails.
by pkasting@chromium.org
· 10 years ago
5e16161
Remove CPU monitor from WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
aef0779
Rewrite ThreadWindows.
by tommi@webrtc.org
· 10 years ago
f2ec814
Move use of DEPTH into build_with_chromium==1.
by kjellander@webrtc.org
· 10 years ago
f88bee6
Refactor senders into senders and sources in the simulation framework.
by stefan@webrtc.org
· 10 years ago
a671f4b
Fixing a VoE test to set correct rate for iSAC
by henrik.lundin@webrtc.org
· 10 years ago
05db352
Fix a bug in ACM test channel
by henrik.lundin@webrtc.org
· 10 years ago
3154a1c
Reland r8210 "Add a new parameter to ACMGenericCodec constructor""
by henrik.lundin@webrtc.org
· 10 years ago
4455f62
WebRtcIsacfix_Time2SpecNeon and _Spec2TimeNeon: Fix stack alignment
by henrik.lundin@webrtc.org
· 10 years ago
8820ac7
peerconnectin_server: missing comma in sprintfn() in r8128
by braveyao@webrtc.org
· 10 years ago
2bbc35d
Remove unused method, SetAffinity, from the ThreadWrapper class.
by tommi@webrtc.org
· 10 years ago
6752b85
Revert r8210 "Add a new parameter to ACMGenericCodec constructor"
by henrik.lundin@webrtc.org
· 10 years ago
c3643f2
Add a new parameter to ACMGenericCodec constructor
by henrik.lundin@webrtc.org
· 10 years ago
2444d96
Control the max IPv6 Networks used by WebRTC.
by guoweis@webrtc.org
· 10 years ago
4ddde2e
Add arbitrary microphone geometry input to audioproc_f test utility.
by mgraczyk@chromium.org
· 10 years ago
1398025
Add new members to AudioEncoderOpus::Config
by henrik.lundin@webrtc.org
· 10 years ago
7a37bfc
Revert 8203 "Reducing locking in OveruseFrameDetector and increa..."
by tommi@webrtc.org
· 10 years ago
a33f05e
Re-land "Remove <(webrtc_root) from source file entries."
by kjellander@webrtc.org
· 10 years ago
bdebccf
Fix a number of things in AudioEncoderDecoderIsac*
by henrik.lundin@webrtc.org
· 10 years ago
18e7585
Reducing locking in OveruseFrameDetector and increasing constness.
by tommi@webrtc.org
· 10 years ago
50fe359
Add tracing for slow paths in new video API.
by pbos@webrtc.org
· 10 years ago
4161715
Remove ChangeUniqueID.
by tommi@webrtc.org
· 10 years ago
1ece0cb
Revert "Remove <(webrtc_root) from source file entries."
by kjellander@webrtc.org
· 10 years ago
a26f511
Remove frame copy in ViEExternalRendererImpl::RenderFrame
by magjed@webrtc.org
· 10 years ago
a87c398
Move audio_codec_speed_tests into include_tests==1 condition.
by kjellander@webrtc.org
· 10 years ago
2d2a1f9
Remove <(webrtc_root) from source file entries.
by kjellander@webrtc.org
· 10 years ago
73ca194
Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h
by kwiberg@webrtc.org
· 10 years ago
43c8839
Allow rtp packet history to dynamically expand in size.
by sprang@webrtc.org
· 10 years ago
827d7e8
Change AsyncInvoker to store its closure in a scoped_refptr instead of using a raw pointer.
by perkj@webrtc.org
· 10 years ago
a742cb1
Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off.
by braveyao@webrtc.org
· 10 years ago
f17ee9c
Add case to ApmTest.Process to test the extended filter mode
by aluebs@webrtc.org
· 10 years ago
e7a4a12
Add arraysize() macro from Chromium, and make use of it in a few places.
by pkasting@chromium.org
· 10 years ago
035e912
Move channel_buffer.{h,cc} to common_audio.
by kjellander@webrtc.org
· 10 years ago
a67ca1a
Only report the first rtp packet because it indicates the media has started flowing.
by honghaiz@google.com
· 10 years ago
a094cac
Add stats for network merge.
by guoweis@webrtc.org
· 10 years ago
7d2b6a9
Enable Clang warning implicit-fallthrough and annotate the code.
by kjellander@webrtc.org
· 10 years ago
a907e01
Adding constness.
by tommi@webrtc.org
· 10 years ago
664ccb7
Reland r8125: Modify some tests to never use DTX disable mode
by henrik.lundin@webrtc.org
· 10 years ago
37c0559
Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets).
by asapersson@webrtc.org
· 10 years ago
22c2f05
Add "score" unit to SSIM perf score output.
by kjellander@webrtc.org
· 10 years ago
4aecd00
Add support for 40 and 60 ms frames to AudioEncoderIlbc
by henrik.lundin@webrtc.org
· 10 years ago
2a6558c
Make sure ByteReader<T>::Read* is properly constified.
by sprang@webrtc.org
· 10 years ago
7aef80c
GN: Remove webrtc_base target in favor for rtc_base.
by kjellander@webrtc.org
· 10 years ago
9b64a6e
Adjust parameter in videoprocessor_integrationtest for VP9.
by marpan@webrtc.org
· 10 years ago
dc8a9da
Adjust qp-max settinhg in VP9 wrapper.
by marpan@webrtc.org
· 10 years ago
922cfcd
Use non-zero data in AudioRingBufferTest.
by andrew@webrtc.org
· 10 years ago
36401ab
Update GAE API paths for join/leave.
by tkchin@webrtc.org
· 10 years ago
8bb32d6
Minor updates to AudioEncoderCng
by henrik.lundin@webrtc.org
· 10 years ago
db1ebf6
Add jakehilton@gmail.com to AUTHORS
by tnakamura@webrtc.org
· 10 years ago
478cedc
Add new methods to AudioEncoder interface
by henrik.lundin@webrtc.org
· 10 years ago
5614cf1
audio_processing: Use fixed aggregation window in delay metrics
by bjornv@webrtc.org
· 10 years ago
6e25182
Whitespace change after enabling gnumbd
by kjellander@webrtc.org
· 10 years ago
ccd608e
Whitespace change for git updater
by kjellander@webrtc.org
· 10 years ago
0bc73a1
Whitespace change to trigger git updater
by kjellander@webrtc.org
· 10 years ago
f68ffca
Add PRESUBMIT check for GYP files including source files above itself.
by kjellander@webrtc.org
· 10 years ago
76e5e20
Roll chromium_revision 4664fe0..9070a80 (312733:313233)
by kjellander@webrtc.org
· 10 years ago
273fbbb
Update StreamDataCounter with FEC bytes.
by asapersson@webrtc.org
· 10 years ago
70117a8
AEC: Implements a new function for calculating delay metrics
by bjornv@webrtc.org
· 10 years ago
fc5ad95
Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139
by magjed@webrtc.org
· 10 years ago
8501ee6
Support VP8 HW decoding on devices with Exynos codec.
by glaznev@webrtc.org
· 10 years ago
df9a41d
Fix bug in GetREDStatus(): it doesn't actually return the current status.
by pkasting@chromium.org
· 10 years ago
82415e3
Update AppRTCDemo to use renamed GAE messages.
by glaznev@webrtc.org
· 10 years ago
041035b
Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface.
by andrew@webrtc.org
· 10 years ago
4dba2e9
Consolidate anonymous namespace content and file-static methods to all be in the
by pkasting@chromium.org
· 10 years ago
d7e34e1
Make it easier to use external libyuv + cleanup GYP files.
by kjellander@webrtc.org
· 10 years ago
d25c034
Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16()
by bjornv@webrtc.org
· 10 years ago
04cd466
Move ThreadChecker into rtc_base_approved.
by tommi@webrtc.org
· 10 years ago
38d11b8
Enable encoder multi-threading for VP9.
by marpan@webrtc.org
· 10 years ago
6f200b5
Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h")
by kwiberg@webrtc.org
· 10 years ago
b6fab2b
Introduce rtc::CheckedDivExact
by henrik.lundin@webrtc.org
· 10 years ago
19eb4e4
Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h
by kwiberg@webrtc.org
· 10 years ago
995b4c9
Remove win_asan trybot from PRESUBMIT.py
by kjellander@webrtc.org
· 10 years ago
acb8085
Roll chromium_revision c086b4e..4664fe0 (312108:312733)
by kjellander@webrtc.org
· 10 years ago
7519de5
Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..."
by tkchin@webrtc.org
· 10 years ago
0f98844
Revert 8139 "Implement elapsed time and capture start NTP time e..."
by tkchin@webrtc.org
· 10 years ago
dacdd94
Reland r7980:
by jiayl@webrtc.org
· 10 years ago
8919cfe
Change a GYP reference to cpufeatures.gypi
by fdegans@chromium.org
· 10 years ago
ad3ee2c
Implement elapsed time and capture start NTP time estimation.
by pbos@webrtc.org
· 10 years ago
a02d768
Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness
by kjellander@webrtc.org
· 10 years ago
456f014
Re-allowing RED in voice engine.
by minyue@webrtc.org
· 10 years ago
182ea46
Remove frame copy in ViEExternalRendererImpl::RenderFrame
by magjed@webrtc.org
· 10 years ago
73ee453
Switch to use range based loops in the BWE simulation framework.
by stefan@webrtc.org
· 10 years ago
36d5c3c
Leave BIO_METHOD non-const.
by davidben@webrtc.org
· 10 years ago
586f2ed
Change GetStreamBySsrc to not copy StreamParams.
by tommi@webrtc.org
· 10 years ago
7e5b380
Fix a crash in AllocationSequence. Internal bug 19074679.
by jiayl@webrtc.org
· 10 years ago
ff108fe
Revert 8125 "Modify some tests to never use DTX disable mode"
by kjellander@webrtc.org
· 10 years ago
b40c7bb
Change sprintf use in talk samples to snprintf
by jlmiller@webrtc.org
· 10 years ago
ea1c842
Correct GetDriveType error handling.
by jlmiller@webrtc.org
· 10 years ago
043db24
Modify some tests to never use DTX disable mode
by henrik.lundin@webrtc.org
· 10 years ago
e5251ad
Integrate send-side BWE into simulation framework.
by stefan@webrtc.org
· 10 years ago
cfd82df
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
by asapersson@webrtc.org
· 10 years ago
3dd33a6
Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes.
by stefan@webrtc.org
· 10 years ago
fbd37bd
Make iSAC SWB own its decoder
by henrik.lundin@webrtc.org
· 10 years ago
cceb166
Fix a use-after-free when sending queued messages is aborted for blocked channel.
by jiayl@webrtc.org
· 10 years ago
e65d9d9
Fix an unitialized variable warning.
by andrew@webrtc.org
· 10 years ago
c429b82
GN: Prepare to remove webrtc_base target
by kjellander@webrtc.org
· 10 years ago
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