- 143cec1 Set correct encoder-specific settings for vpx in the new API. by Erik Språng · 10 years ago
- e8a197b Enable isac NEON building on Aarch64 by Zhongwei Yao · 10 years ago
- d7e5c44 STUN allocation should not be disabled when using shared port and TURN servers are provided. by Jiayang Liu · 10 years ago
- 5a92aa8 Add 3-band filter-bank implementation by Alejandro Luebs · 10 years ago
- 494f209 Move CriticalSection into rtc_base_approved. by Tommi · 10 years ago
- 59d91dc Remove ViERTP_RTCP usage in VideoSendStream. by Peter Boström · 10 years ago
- e6cefb6 GYP variables for building expat, icu, libsrtp, usrsctp by Henrik Kjellander · 10 years ago
- 61be2a4 Clean up RTCPSender. by Erik Språng · 10 years ago
- 3c391cb Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api. by Åsa Persson · 10 years ago
- 52ef9d7 Stop IncomingVideoStream on delete. by Peter Boström · 10 years ago
- 23dc68e Add the rtc_build_openmax_dl variable to the GN build. by Andrew MacDonald · 10 years ago
- 12e0329 Do not use Magnifier if there are multiple screens since it sometimes crashes. by Jiayang Liu · 10 years ago
- c4188fd Use IncomingVideoStream in VideoReceiveStream. by Peter Boström · 10 years ago
- f955b5d Add h.264 AVC SPS parsing for resolution (re-land) by Henrik Kjellander · 10 years ago
- c043afc Cleanup inside IncomingVideoStream. by Peter Boström · 10 years ago
- a96f02b Make sure histograms in jitter buffer are only updated if running. by Åsa Persson · 10 years ago
- affcfb2 Refactor common_audio/signal_processing: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16 by Bjorn Volcker · 10 years ago
- e3827f2 Revert "Add h.264 AVC SPS parsing for resolution." by Noah Richards · 10 years ago
- 5ea8eff Add h.264 AVC SPS parsing for resolution. by Noah Richards · 10 years ago
- 9728241 Record H264 NALU type in the h264 header. by Noah Richards · 10 years ago
- fe7a80c Prevent sender RTCP signals for receive-only channels. by Peter Boström · 10 years ago
- 7f287cc rtc::CriticalSection: Add dummy implementation of IsLocked for release builds by Magnus Jedvert · 10 years ago
- d3e8eda (Re-land) AudioEncoderDecoderIsac: Merge the two config structs by Karl Wiberg · 10 years ago
- 92f9eac g722 and red encoders: Use rtc::Buffer instead of scoped_ptr<uint8_t[]> by Karl Wiberg · 10 years ago
- 6bf1084 rtc::CriticalSection: Add function IsLocked by Magnus Jedvert · 10 years ago
- bd67f66 Restore webrtc/base/move.h, because it's used in Windows Chromium builds by Karl Wiberg · 10 years ago
- 3525954 Use short include paths for icu headers. by Henrik Kjellander · 10 years ago
- 915590e Moved ByteBuffer/BitBuffer into rtc_base_approved. by Noah Richards · 10 years ago
- 01aeaee Fix GetSignatureDigestAlgorithm for openssl to prepare for EC key switch. by JiaYang (佳扬) Liu · 10 years ago
- a8e285d Remove webrtc/base/move.h, and make types move-only manually by Karl Wiberg · 10 years ago
- 96d1d89 Do not register bandwidth observer for receive only channels. by Åsa Persson · 10 years ago
- 5a31780 Reformatting RTPtimeshift.cc file. by Ivo Creusen · 10 years ago
- ac69016 Improve TCP by adding a real timeout to in flight packets. by Stefan Holmer · 10 years ago
- e555b7b Fix CC flags in GN Windows build. by Henrik Kjellander · 10 years ago
- fb49451 Disables mic bump-up level if not built with chromium by Bjorn Volcker · 10 years ago
- 8f85dbc Reduce the number of registers used in MIPS optimizations. by Ljubomir Papuga · 10 years ago
- bbf7c86 Add a new BitBuffer class to webrtc base. by Noah Richards · 10 years ago
- 61b4d51 Dynamic resolution change for VP8 HW encode. by jackychen · 10 years ago
- 5464a6e Remove VideoCodingModule::InitializeReceiver. by Peter Boström · 10 years ago
- 9dbbcfb Remove VideoCodingModule::InitializeSender. by Peter Boström · 10 years ago
- 9570224 Fix broken perf prints. by Stefan Holmer · 10 years ago
- 5f92051 Fix bug in TCP implementation (simulations). by Stefan Holmer · 10 years ago
- e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 10 years ago
- 6cff9cf Revert "Remove simulcast modules from ViEReceiver." by Peter Boström · 10 years ago
- 06b08af VoE: VoEBase unit test by Jelena Marusic · 10 years ago
- 011c00f rtc::Buffer: Accept void* in addition to the byte-sized types by Karl Wiberg · 10 years ago
- 9478437 rtc::Buffer improvements by Karl Wiberg · 10 years ago
- 9154373 Do not define POSIX. by Thiago Farina · 10 years ago
- 599beb8 Revert "AudioEncoderDecoderIsac: Merge the two config structs" by Ted Nakamura · 10 years ago
- a51e8f4 Fix some simulation issues. by Stefan Holmer · 10 years ago
- 14a97f0 Remove simulcast modules from ViEReceiver. by Peter Boström · 10 years ago
- 1d19893 Add TCP fairness test. by Stefan Holmer · 10 years ago
- b0b5425 Let rtp_analyze parse absolute sender time by Henrik Lundin · 10 years ago
- 61c2a6f Remove rtc::Buffer::length(), since no one uses it anymore by Karl Wiberg · 10 years ago
- d4e8014 Fix build errors in r9022 / 09bdc1e5f5a9. by Stefan Holmer · 10 years ago
- 09bdc1e Add a BWE fairness test. by Stefan Holmer · 10 years ago
- 3795937 Adds a simplified Reno-type TCP sender. by Stefan Holmer · 10 years ago
- 3f4eed0 Deliver RTCP packets only once per receive stream. by Peter Boström · 10 years ago
- fb98c40 Register RTP/RTCP modules outside rtp_rtcp_cs_. by Peter Boström · 10 years ago
- 382c58d Move target_subarch from gyp_webrtc to supplement.gypi by Henrik Kjellander · 10 years ago
- f2497cf Fix unknown option '-msse2' warning by Henrik Kjellander · 10 years ago
- 7c324ca AudioEncoderDecoderIsac: Merge the two config structs by Karl Wiberg · 10 years ago
- 7d89f80 Use BoringSSL as default on iOS by Zeke Chin · 10 years ago
- 5d22c00 Add performance tests flag to audioproc_float by Alejandro Luebs · 10 years ago
- 41ee1ea Modified the simulcast encoder adapter to correctly handle encoded frames from sub encoders even if the encoder is unable to (temporarily or permanently) produce frames of the exactly matching resolution. This is done by using a different EncodedImageCallback for each encoder, which remembers which VideoEncoder it is registered to and forwards that on to SimulcastEncoderAdapter::Encoded. by Noah Richards · 10 years ago
- 099323e Have ViE sender also use the last encoded frame timestamp when determining if the video stream is paused/muted, for purposes of padding. by Noah Richards · 10 years ago
- 352b2d7 Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream). by Åsa Persson · 10 years ago
- c317ce5 VoE: move mock directory 1 level up by Jelena Marusic · 10 years ago
- adc46c4 audio_processing/agc: Adds config to set minimum microphone volume at startup by Bjorn Volcker · 10 years ago
- a9c0ae2 Add a sparse FIR filter implementation by Alejandro Luebs · 10 years ago
- fcf54bd Reland "Avoid critsect for protection- and qm setting callbacks in VideoSender." by mflodman · 10 years ago
- 73ba7a6 Remove PORTALLOCATOR_ENABLE_BUNDLE, PortAllocatorSessionProxy, PortAllocatorSessionMuxer, and PortProxy. by Peter Thatcher · 10 years ago
- 74b9769 Deliver RTCP packets only once per send stream. by Peter Boström · 10 years ago
- 2dd6a27 VoE: format VoEBase according to new style guide by Jelena Marusic · 10 years ago
- 0de7bcf Removes use of AudioManager.setSpeakerphoneOn in audio manager by henrika · 10 years ago
- 529921e Explicitly set target_subarch for iOS on ia32/x64 by Henrik Kjellander · 10 years ago
- 6ae2572 Add missing configuration of rtx payload type for rtp/rtcp module. by Åsa Persson · 10 years ago
- 0f911d71 Refactor audio_processing/nsx: Removed usage of macro WEBRTC_SPL_MEMCPY_W16 by Bjorn Volcker · 10 years ago
- 61a4b04 Refactor common_audio/vad: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16(a, b) by Bjorn Volcker · 10 years ago
- 6fc2d2f VoE: revert CHECKs into asserts by Jelena Marusic · 10 years ago
- 9e5e421 VoE: cleanup VoEBaseImpl by Jelena Marusic · 10 years ago
- 93ef1d8 Change ACM's CodecManager to hold one encoder instead of an array by Henrik Lundin · 10 years ago
- b32a5c4 Add more logging around TURN refreshes. by Peter Thatcher · 10 years ago
- 3949e86 Prevent decoder busy loop for send-only channels. by Peter Boström · 10 years ago
- a125d7d Changes default audio mode in AppRTCDemo to MODE_RINGTONE. by henrika · 10 years ago
- e12a667 Remove i420_video_frame.h from common_video.gyp by Bjorn Volcker · 10 years ago
- 9bfe3daf Cleanup: Remove i420_video_frame.h header. by Thiago Farina · 10 years ago
- 9526187 Default enable abs send time bwe for CallTest by Erik Språng · 10 years ago
- 09bf1a1 Delays changing to COMMUNICATION mode until streaming starts. by henrika · 10 years ago
- dcbd3ac Improve BWE plotting and logging to make it possible to use multiple windows/figures. by Stefan Holmer · 10 years ago
- f2822ed Refactor audio_coding/codecs/isac/fix: Removed usage of macro WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
- f6a99e6 Refactor audio_processing: Free functions return void by Bjorn Volcker · 10 years ago
- 0666a9b Remove Transport::Reset, which is never used, and only makes reading the code harder. by Peter Thatcher · 10 years ago
- d417c93 Remove android_webview_build conditions. by Richard Coles · 10 years ago
- 9504b89 Cleanup: Remove unnecessary SHA1Transform() declaration. by Thiago Farina · 10 years ago
- 3a93986 Exit after printing usage message. by Thiago Farina · 10 years ago
- 7f6c4d4 Fix clang style warnings in webrtc/modules/audio_coding/neteq by Karl Wiberg · 10 years ago
- 76c53d3 Remove ViE interface usage from VideoReceiveStream. by Peter Boström · 10 years ago
- 9b3f56e Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection."" by Per · 10 years ago
- 2c37078 Fix crash with CVO turned on for VP9 codec by Guo-wei Shieh · 10 years ago
