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2014-10-28
audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"
bjornv@webrtc.org
2014-10-28
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
bjornv@webrtc.org
2014-10-28
Use neteq_unittest_tools in audio_decoder_unittests
henrik.lundin@webrtc.org
2014-10-28
Fix double backslashes in incoming_video_stream.cc
perkj@webrtc.org
2014-10-28
Update makefiles after merge of Chromium at 82ca3b654cda
Android Chromium Automerger
2014-10-27
Add a simple AudioConverter class.
andrew@webrtc.org
2014-10-27
Only configure the SSL library in one place.
henrike@webrtc.org
2014-10-27
Move (test) RtpFileReader to a lightweight target.
pbos@webrtc.org
2014-10-27
Move scoped_ptr "free" functions into the webrtc namespace.
andrew@webrtc.org
2014-10-27
Upgrade our scoped_ptr copy to match Chromium's latest.
andrew@webrtc.org
2014-10-27
Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile
henrik.lundin@webrtc.org
2014-10-27
isacfix: Refactor big-endian reading and writing
kwiberg@webrtc.org
2014-10-27
Increase max trace message size to 1024 characters.
pbos@webrtc.org
2014-10-27
Fix ::~LogMessage to print as a string.
pbos@webrtc.org
2014-10-24
Adding the subtool rtcBot report visualizer
houssainy@google.com
2014-10-24
Move min transmit bitrate to VideoEncoderConfig.
pbos@webrtc.org
2014-10-23
Break out WebRtcNs_ComputeDdUpdate function in ns_core
aluebs@webrtc.org
2014-10-23
Break out WebRtcNs_UpdateNoise function in ns_core
aluebs@webrtc.org
2014-10-23
Break out FFT function in ns_core
aluebs@webrtc.org
2014-10-23
Break out ComputeSnr function in ns_core
aluebs@webrtc.org
2014-10-23
Adding three video conference bots test
houssainy@google.com
2014-10-23
Update makefiles after merge of Chromium at 9ef958e74e13
Android Chromium Automerger
2014-10-23
Adding file from test.webrtc.org domain to be downloaded
houssainy@google.com
2014-10-23
Add macros and APIs for webrtc histograms.
asapersson@webrtc.org
2014-10-23
Adds support for sending first set of packets at increasingly higher bitrates...
stefan@webrtc.org
2014-10-23
Using the Unused turn configuration in two way test
houssainy@google.com
2014-10-23
Let video_loopback use internal VCM capturers.
pbos@webrtc.org
2014-10-22
NOTE: This code review based on the running issue:
houssainy@google.com
2014-10-22
Adding Two way video and audio streaming test to RtcBot
houssainy@google.com
2014-10-22
HTTPS Server used instead of HTTP for loading the bots to avoid the media per...
houssainy@google.com
2014-10-22
Make ReconfigureVideoEncoder use current bitrate.
pbos@webrtc.org
2014-10-22
Disable TestVp8Impl.BaseUnitTest on MSan.
pbos@webrtc.org
2014-10-22
For FIR packet, payload length is zero, so SendToNetwork function is failing.
stefan@webrtc.org
2014-10-21
Break out WebRtcNs_Windowing function in ns_core
aluebs@webrtc.org
2014-10-21
Break out WebRtcNs_Energy function in ns_core
aluebs@webrtc.org
2014-10-21
Break out WebRtcNs_IFFT function in ns_core
aluebs@webrtc.org
2014-10-21
Break out WebRtcNs_UpdateBuffer function in ns_core
aluebs@webrtc.org
2014-10-21
Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
henrik.lundin@webrtc.org
2014-10-21
Update makefiles after merge of Chromium at b03027d23881
Android Chromium Automerger
2014-10-21
Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...
Android Chromium Automerger
2014-10-21
audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org
2014-10-21
Fix for glitches in ACM when switching desired output sample rate
henrik.lundin@webrtc.org
2014-10-20
common_audio: Replaced invalid operand in min_max_operations_neon.S"
bjornv@webrtc.org
2014-10-20
Make avg_{psnr,ssim}_threshold_ const.
pbos@webrtc.org
2014-10-20
audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org
2014-10-20
audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org
2014-10-19
Update makefiles after merge of Chromium at 89b463ddd92b
Android Chromium Automerger
2014-10-17
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle...
henrike@webrtc.org
2014-10-17
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
henrike@webrtc.org
2014-10-17
Moving creating TURN configration to the host machine instead of the bots - r...
houssainy@google.com
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