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gerrit-public.fairphone.software
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platform
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external
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webrtc
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bd1bc473958981995258b7cfff50b94c8a99e762
bd1bc47
Restructure decoder registration in ACM
by Karl Wiberg
· 9 years ago
9d8b71e
Remove some dead code in ViEChannel.
by Peter Boström
· 9 years ago
a6e883b
Fix constant in SetCurrentThreadName.
by André Susano Pinto
· 9 years ago
bebc690
Add platform_thread source files and move types from thread_checker_impl to there.
by Tommi
· 9 years ago
24ec128
Roll chromium_revision 5118a5b..1b9c098 (330060:330302)
by Henrik Kjellander
· 9 years ago
a7d03ae
Roll chromium_revision 62a5bb3..5118a5b (329063:330060)
by Henrik Kjellander
· 9 years ago
144d018
fix indent on tokenize_first function signatures
by Donald Curtis
· 9 years ago
42af6ca
Add logging of "use candidate" and when we switch ICE "best" connections.
by Peter Thatcher
· 9 years ago
b2d2623
Don't use rtc::LogCheckLevel, because it breaks Chrome.
by Peter Thatcher
· 9 years ago
1cf6f81
Add logging for sending and receiving STUN binding requests and TURN requests and responses.
by Peter Thatcher
· 9 years ago
37931c4
Stunprober interface, its implementation and a command line driver.
by Guo-wei Shieh
· 9 years ago
0e07f92
Split fmtp on semicolons not spaces as per RFC6871
by Donald Curtis
· 9 years ago
20f3f94
Clear bitrate stats for unused SSRCs.
by Peter Boström
· 9 years ago
4cd6940
Enable -Wformat-security warning and cleanup GYP.
by Henrik Kjellander
· 9 years ago
39f2b0c
Implemented video device info for iOS
by Yuriy Shevchuk
· 9 years ago
a4463b2
Further updates to fix libjingle logging.
by Tommi
· 9 years ago
99eeee3
Fix logging in Chrome.
by Tommi
· 9 years ago
06c577f
Set msvs_error_on_missing_sources=1 in GYP_GENERATOR_FLAGS on Windows.
by Henrik Kjellander
· 9 years ago
2013aec
Propagating RTT from send-only channel to receive-only channel.
by Minyue
· 9 years ago
0703766
Fix issue where receive-side encoders are included in the padding bitrate.
by Stefan Holmer
· 9 years ago
9a63866
Move IncomingVideoFrames to common_video/.
by Peter Boström
· 9 years ago
4feb505
Remove VideoProcessing::ColorEnhancement.
by Peter Boström
· 9 years ago
5ec9985
Windows utility to setTheadName to help debugging.
by André Susano Pinto
· 9 years ago
9b9f1c4
Remove basictypes.h dependency from bitbuffer.
by Noah Richards
· 9 years ago
e235714
Guard new protobuf target with enable_protobuf==1.
by Andrew MacDonald
· 9 years ago
300eeb6
Remove VideoEngine interfaces.
by Peter Boström
· 9 years ago
8171735
Add NetEqIlbcQualityTest
by Henrik Lundin
· 9 years ago
df66453
Remove FPS->kilo-FPS conversion in VideoSender.
by Peter Boström
· 9 years ago
e5ff00a
Add NetEqPcmuQualityTest
by Henrik Lundin
· 9 years ago
fade179
Remove leaking aecdump testfiles.
by Peter Boström
· 9 years ago
075bb8d
Fix race in AudioCodingModuleImpl::Add10MsData()
by Karl Wiberg
· 9 years ago
1b794d5
Switch to use SHA-256 for certificates / fingerprints.
by Joachim Bauch
· 9 years ago
cb3e8fe
Increase the tolerance in NetEq's DelayManagerTest a notch
by Henrik Lundin
· 9 years ago
67c9df7
Base NACK on send codecs.
by Peter Boström
· 9 years ago
126c03e
Base decision to send REMB on send codecs.
by Peter Boström
· 9 years ago
64dad83
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
by Henrik Lundin
· 9 years ago
092041c
Setting OPUS_SIGNAL_VOICE when enable DTX.
by Minyue Li
· 9 years ago
9f7908e
Roll chromium_revision ec5b768..62a5bb3 (328242:329063)
by Henrik Kjellander
· 9 years ago
242e22b
Refactor RTCP sender
by Erik Språng
· 9 years ago
1f62923
Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
by Henrik Lundin
· 9 years ago
fd32f35
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
by Henrik Lundin
· 9 years ago
54adb28
mac: Explicitly redeclare methods only available on 10.7+.
by Jiayang Liu
· 9 years ago
4c277bb
Add basic SCTP packet logging.
by Lally Singh
· 9 years ago
cdb47a4
Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
by Henrik Lundin
· 9 years ago
45553ae
Remove VideoEngine interface usage from new API.
by Peter Boström
· 9 years ago
208a229
Adding a new constraint to set NetEq buffer capacity from peerconnection
by Henrik Lundin
· 9 years ago
83b5c05
Modify NetEqQualityTest
by Henrik Lundin
· 9 years ago
cb05b72
Add WAV and arbitrary geometry support to nlbf test.
by Andrew MacDonald
· 9 years ago
d3ddc1b
Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
by Fredrik Solenberg
· 9 years ago
e444a3d
WebRtcVoiceEngine: Get rid of unnecessary template base class.
by Fredrik Solenberg
· 9 years ago
aaf8ff2
WebRtcVoiceEngine: virtual to override + git cl format.
by Fredrik Solenberg
· 9 years ago
6179b89
Remove unused API on WebRtcVoiceEngine.
by Fredrik Solenberg
· 9 years ago
2ea71c3
Replace ACMGenericCodec with CodecOwner and AudioEncoderMutable
by Karl Wiberg
· 9 years ago
53d0dc3
Wire up RTT to send-side GCC and TCP.
by Stefan Holmer
· 9 years ago
4b60c73
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
by Fredrik Solenberg
· 9 years ago
dcccab3
New interface: AudioEncoderMutable
by Karl Wiberg
· 9 years ago
81ea54e
Remove WebRtcVideoEngine.
by Peter Boström
· 9 years ago
ccfc939
Reinterpret AudioOption delay_agnostic_aec to override HW-AEC
by Bjorn Volcker
· 9 years ago
c81591d
NADA's proposal from Cisco.
by Cesar Magalhaes
· 9 years ago
f353dd5
VoE: cleanup VoENetwork implementation
by Jelena Marusic
· 9 years ago
1ff218f
audio_processing/aec: Do not scale target delay at startup when on Android
by Bjorn Volcker
· 9 years ago
532531b
audio_processing/delay_estimator: Always update robust validation statistics
by Bjorn Volcker
· 9 years ago
40a6d59
audio_processing/tests: Adds a flag to unpack input data to text file
by Bjorn Volcker
· 9 years ago
9695d85
Added VP9FrameBufferPool, a memory pool that is shared between libvpx and webrtc. Using the VP9 codec, the libvpx decoder will obtain its buffers from our memory pool. This lets us reuse the same buffers for our I420VideoFrames and not have to copy a frame for every decode (from libvpx buffers to webrtc/I420VideoFrame buffers).
by Henrik Boström
· 9 years ago
f242e66
Replace asm NEON function by intrinsics implementation on ARMv7
by Zhongwei Yao
· 9 years ago
507a550
Delete auto-roll script since moved into Chromium.
by Henrik Kjellander
· 9 years ago
589699e
Fix bug in transform_neon.c in iSAC codec.
by Zhongwei Yao
· 9 years ago
57cc74e
iOS camera switching video capturer.
by Zeke Chin
· 9 years ago
5cb9ce4
Remove ViECodec usage in VideoSendStream.
by Peter Boström
· 9 years ago
ab00404
VCMEncodedFrame::VerifyAndAllocate: Use size_t instead of uint32_t for size argument
by Magnus Jedvert
· 9 years ago
01b4888
Use padding to achieve bitrate probing if the initial key frame has too few packets.
by Stefan Holmer
· 9 years ago
78c8bbf
Roll chromium_revision 0cb2549..ec5b768 (327252:328242)
by Henrik Kjellander
· 9 years ago
c56ac1e
rtc::Buffer: Remove backwards compatibility band-aids
by Karl Wiberg
· 9 years ago
f75f0cf
Enable GoogleWifiTrace3Mbps simulations.
by Stefan Holmer
· 9 years ago
0d26605
VoE: apply new style guide on VoE interfaces and their implementations
by Jelena Marusic
· 9 years ago
79c1433
Delete VoiceChannelTransport before deleting Channel in voe_cmd_test
by Minyue Li
· 9 years ago
0b15445
VoE: Follow-up to https://webrtc-codereview.appspot.com/49759004/
by Jelena Marusic
· 9 years ago
e433c0e
Restore back verbosity logging for camera captured frame.
by Alex Glaznev
· 9 years ago
f2f8283
Use rtc::CriticalSection in webrtc/video/.
by Peter Boström
· 9 years ago
cac1b38
Expose RTCConfiguration to java JNI and add an option to disable TCP
by Jiayang Liu
· 9 years ago
4eddf18
Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle.
by Peter Thatcher
· 9 years ago
8a6680e
Remove base/move.h (no one uses it anymore)
by Karl Wiberg
· 9 years ago
cbf0927
Revert "rtc::Buffer: Remove backwards compatibility band-aids"
by Karl Wiberg
· 9 years ago
9e1a6d7
rtc::Buffer: Remove backwards compatibility band-aids
by Karl Wiberg
· 9 years ago
ff019b0
Move rtc::AtomicOps to webrtc/base/atomicops.h.
by Peter Boström
· 9 years ago
f16fcbe
Remove ViECapture usage in VideoSendStream.
by Peter Boström
· 9 years ago
46bd31b
VoE: VoENetwork unit test
by Jelena Marusic
· 9 years ago
3cfa756
audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz
by Bjorn Volcker
· 9 years ago
efbde37
Don't use CPU adaptation for screen content in the new API.
by Erik Språng
· 9 years ago
adf89b7
Added SetBitRate function to VoE API to allow changing the audio bitrate.
by Ivo Creusen
· 9 years ago
23fba1f
Add AudioReceiveStream to Call API.
by Fredrik Solenberg
· 9 years ago
10ba3ee
Roll chromium_revision a12e1e1..0cb2549 (326495:327252)
by Henrik Kjellander
· 9 years ago
dea11f9
Add per flow throughput and delay metrics.
by Stefan Holmer
· 9 years ago
94cc1fe
Remove ViEImageProcess usage in VideoSendStream.
by Peter Boström
· 9 years ago
c444de6
Make setup_links.py handle non-link directories during cleanup
by Henrik Kjellander
· 9 years ago
1ba344a
Adds a MediaConstraint for the AudioOption aec_dump
by Bjorn Volcker
· 9 years ago
97f13c5
Fixed incorrect RBSP parsing. The original version would eat 0x3 as an emulation byte in places where it shouldn't, whereas the real parsing is only supposed to eat 0x3 preceded by 0x0 0x0.
by Noah Richards
· 9 years ago
86153c2
Added a BitBufferWriter subclass that contains methods for writing bit and byte-sized data, along with exponential golomb encoded data.
by Noah Richards
· 9 years ago
80154f6
Set correct .type directive for asm functions.
by Wei Zhong
· 9 years ago
faa6d07
Remove a few verbose log messages from webrtcvideoengine2.
by Alex Glaznev
· 9 years ago
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