- 589699e Fix bug in transform_neon.c in iSAC codec. by Zhongwei Yao · 10 years ago
- 5cb9ce4 Remove ViECodec usage in VideoSendStream. by Peter Boström · 10 years ago
- ab00404 VCMEncodedFrame::VerifyAndAllocate: Use size_t instead of uint32_t for size argument by Magnus Jedvert · 10 years ago
- 01b4888 Use padding to achieve bitrate probing if the initial key frame has too few packets. by Stefan Holmer · 10 years ago
- c56ac1e rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
- f75f0cf Enable GoogleWifiTrace3Mbps simulations. by Stefan Holmer · 10 years ago
- 0d26605 VoE: apply new style guide on VoE interfaces and their implementations by Jelena Marusic · 10 years ago
- 79c1433 Delete VoiceChannelTransport before deleting Channel in voe_cmd_test by Minyue Li · 10 years ago
- 0b15445 VoE: Follow-up to https://webrtc-codereview.appspot.com/49759004/ by Jelena Marusic · 10 years ago
- f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 10 years ago
- 4eddf18 Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle. by Peter Thatcher · 10 years ago
- 8a6680e Remove base/move.h (no one uses it anymore) by Karl Wiberg · 10 years ago
- cbf0927 Revert "rtc::Buffer: Remove backwards compatibility band-aids" by Karl Wiberg · 10 years ago
- 9e1a6d7 rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
- ff019b0 Move rtc::AtomicOps to webrtc/base/atomicops.h. by Peter Boström · 10 years ago
- f16fcbe Remove ViECapture usage in VideoSendStream. by Peter Boström · 10 years ago
- 46bd31b VoE: VoENetwork unit test by Jelena Marusic · 10 years ago
- 3cfa756 audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz by Bjorn Volcker · 10 years ago
- adf89b7e Added SetBitRate function to VoE API to allow changing the audio bitrate. by Ivo Creusen · 10 years ago
- 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
- dea11f9 Add per flow throughput and delay metrics. by Stefan Holmer · 10 years ago
- 94cc1fe Remove ViEImageProcess usage in VideoSendStream. by Peter Boström · 10 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 · 10 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 · 10 years ago
- 80154f6 Set correct .type directive for asm functions. by Wei Zhong · 10 years ago
- 019087f Add safeguards against signalling peer-reflexive candidates. by Peter Thatcher · 10 years ago
- 31dc737 Platform dependent way of generating the seed for srand for simulations, so that they can be run in parallel. by Stefan Holmer · 10 years ago
- 88de479 AudioEncoderIsac: Print error code if CHECK for successful encoding fails by Karl Wiberg · 10 years ago
- bcbcd84 Improve TCP implementation by adding ssthresh and make it possible to start it with an offset. by Stefan Holmer · 10 years ago
- 9d657cf Fix dangling pointer in screenshare_loopback by Erik Språng · 10 years ago
- beb9798 audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool by Bjorn Volcker · 10 years ago
- ddbddbd Remove ViENetwork usage in VideoSendStream. by Peter Boström · 10 years ago
- 038df3c Remove ViEExternalCodec usage in VideoSendStream. by Peter Boström · 10 years ago
- 4a9cb6b Prevent zero-timestamps in captured_frame_. by Peter Boström · 10 years ago
- 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
