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webrtc
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eb66e800d1f5f74ab366715d2618fbede8cf3e12
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f09e09c
VoE: Remove unused interfaces
by Jelena Marusic
· 9 years ago
32c2023
Attempt at fixing error on the Chrome Windows FYI bots.
by Tommi
· 9 years ago
905495c
Introduce NetEq::Config::ToString and use it in NetEq's constructor
by Henrik Lundin
· 9 years ago
e982a70
PRESUBMIT: Fix typo.
by Henrik Kjellander
· 9 years ago
54be3e0
Remove some WebRtcVideoEngine2 unittest stubs.
by Peter Boström
· 9 years ago
d8399e6
Also provide sample rate when registering decoders
by Karl Wiberg
· 9 years ago
323b132
Protect ACM decoder buffer in stereo.
by Minyue
· 9 years ago
57e5fd2
PRESUBMIT: Improve PyLint check and add GN format check.
by Henrik Kjellander
· 9 years ago
00aac5a
Some cleanup for base/logging and base/stream.h
by Tommi
· 9 years ago
23edcff
Move base/logging.* to rtc_base_approved.
by Tommi
· 9 years ago
ee369e4
Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes
by henrika
· 9 years ago
a26c4e5
Script to generate CL descriptions when rolling chromium_revision.
by Henrik Kjellander
· 9 years ago
0eefb4d
Detach base/logging.* from base/stream.*.
by Tommi
· 9 years ago
469c2c0
Make Config::default_value leak instead of having an exit-time destructor.
by Andrew MacDonald
· 9 years ago
4bf12ea
Revert "Fix sending wrong candidates down to transportchannel."
by Alejandro Luebs
· 9 years ago
f65de84
Fix sending wrong candidates down to transportchannel.
by Donald Curtis
· 9 years ago
67b635a
Fix simulcast_encoder_adapter giving full target_bitrate to the 2nd layer of any simulcast setup during InitEncode.
by Noah Richards
· 9 years ago
e4cb4e9
Fix jitter buffer bug around out-of-order packets and non-RTX padding.
by Noah Richards
· 9 years ago
4774874
Enable AudioProcessing48kHzSupport by default
by Alejandro Luebs
· 9 years ago
3548dd2
Set local SSRCs on receivers added before senders.
by Peter Boström
· 9 years ago
367c868
AudioEncoderCng: Handle case where speech encoder is reset
by Henrik Lundin
· 9 years ago
f761d10
Update NetEq Quality Test.
by Minyue Li
· 9 years ago
915df4f
CaptureManager: Don't stop a capturer at UnregisterVideoCapturer if it did not start in the first place.
by Henrik Boström
· 9 years ago
9a416bd
Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
by Fredrik Solenberg
· 9 years ago
5af6d47
Code style change for quality_scaler.
by jackychen
· 9 years ago
98d8cf5
Hardware VP8 encoding: Use QP as metric for resize.
by jackychen
· 9 years ago
5fdcdf6
Enable ciphers to get ECDHE with NSS.
by Joachim Bauch
· 9 years ago
6f2ef74
Keep track of DTLS packet sizes to prevent partial reads.
by Joachim Bauch
· 9 years ago
a3ba0c7
RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits
by Magnus Jedvert
· 9 years ago
36a1438
Remove ViEFrameProviderBase.
by Peter Boström
· 9 years ago
af55ccc
Add RtcpMuxPolicy support to PeerConnection.
by Peter Thatcher
· 9 years ago
02ff911
Feature merge request: Add support for iOS http proxy detection
by Yuriy Shevchuk
· 9 years ago
523183b
Disables AudioDeviceTest.StartStopPlayout for Nexus 9 only
by henrika
· 9 years ago
280ed11
Roll gtest-parallel.
by Peter Boström
· 9 years ago
848d524
Revert "RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits" https://webrtc-codereview.appspot.com/47229004/
by Magnus Jedvert
· 9 years ago
10022cd
RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits
by Magnus Jedvert
· 9 years ago
78ae00e
Remove default encoder/decoders.
by Peter Boström
· 9 years ago
b302ad4
Remove unused VideoDecoder methods.
by Peter Boström
· 9 years ago
1a07a1e
Solve data race in Pulse audio implementation.
by Brave Yao
· 9 years ago
8602a3d
Cast to avoid char-interpretation of uint8_t in logs.
by mflodman
· 9 years ago
05c7605
Add resampling support in AudioBuffer::DeinterleaveFrom
by Alejandro Luebs
· 9 years ago
76b62ff
Clean up now-unused code that was used for libpeerconnection.[so|dll].
by Tommi
· 9 years ago
fce3242
Remove linphonemediaengine.*
by Fredrik Solenberg
· 9 years ago
8eb76ff
Make SHA1 computation thread-safe.
by Sergey Ulanov
· 9 years ago
5cdd702
Add tools/vim to .gitignore.
by Andrew MacDonald
· 9 years ago
9b2b402
Ensures that RECORD_AUDIO permission is required to start recording.
by henrika
· 9 years ago
5779d14
Avoids crash when StartRecording conflicts with existing recording application
by henrika
· 9 years ago
c3f4dbc
Remove rtp_rtcp/ dump functionality.
by Peter Boström
· 9 years ago
ca667db
Remove VCM debug recordings.
by Peter Boström
· 9 years ago
831c558
Allow setting maximum protocol version for SSL stream adapters.
by Joachim Bauch
· 9 years ago
664cdaf
Replace assert() with static_assert() if the condition is evaluatable at
by André Susano Pinto
· 9 years ago
5ca688b
Enable read-ahead on OpenSSL DTLS stream adapters.
by Joachim Bauch
· 9 years ago
931e658
Remove unnecessary dependencies for voe when building with include_internal_audio_device==0.
by Tommi
· 9 years ago
cb7f8ce
Clear ARM NEON flag
by Andrew MacDonald
· 9 years ago
4d71ede
Add HW fallback option to software encoding.
by Peter Boström
· 9 years ago
97bce58
Disable the EXPECT_DEATH check in bitbuffer on Android
by Donald E Curtis
· 9 years ago
bf560dd
remove filelock which is now unused
by Donald E Curtis
· 9 years ago
17b889b
Issue 4366: Adapted frames have wrong width and height and are cropped.
by Guo-wei Shieh
· 9 years ago
65de7d2
Add a link to tools/vim to use the Chromium YCM config with webrtc.
by Andrew MacDonald
· 9 years ago
5ece00f
remove filelock which is now unused
by Donald E Curtis
· 9 years ago
2f5be9a
Improve Android camera error handling.
by Alex Glaznev
· 9 years ago
68898a2
Remove AudioDeviceUtility.
by Tommi
· 9 years ago
df0c05b
Sort source file list for [rtc_]include_internal_audio_device. No code change.
by Tommi
· 9 years ago
c2b63fe
Adding Sony Xperia Z2 D6503 to HW AEC blacklist
by henrika
· 9 years ago
24e56e3
Fixes Chromium FYI build issue on Android.
by henrika
· 9 years ago
ccb49e7
Remove Soundclip handling from libjingle.
by Fredrik Solenberg
· 9 years ago
1ab67ae
Address the corner cases
by Guo-wei Shieh
· 9 years ago
b92be45
Support 720P in portait as maximum on iOS.
by Weiyong Yao
· 9 years ago
8db8069
Change high frequency correction range
by Alejandro Luebs
· 9 years ago
3e95d3e
Don't log warning for unexpected STUN binding responses.
by Peter Thatcher
· 9 years ago
79b2e06
Make the BlockDifference() functions return DiffInfo as their callers expect.
by Peter Kasting
· 9 years ago
2e7a098
Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc.
by Noah Richards
· 9 years ago
7252a2b
Add HW fallback option to software decoding.
by Peter Boström
· 9 years ago
b261989
Adding support for OpenSL ES output in native WebRTC
by henrika
· 9 years ago
02c9b36
Roll gtest-parallel.
by Peter Boström
· 9 years ago
7e0c7d4
Add support for external encoders in ACM
by Karl Wiberg
· 9 years ago
ea14f0a
Move SetCurrentThreadName to platform_thread.* in rtc_base_approved,
by Tommi
· 9 years ago
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
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