- 5c2f1f0 Add some missing includes and dependencies. by Bjorn Terelius · 6 years ago
- e7d08df Fix chromium roll into WebRTC. by Artem Titov · 6 years ago
- 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
- 36faf0b Delete setting of unused variable nack_window_ms by Niels Möller · 6 years ago
- 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
- ba50223 Make voiceengine/audio transport stop using voice_detection() interface by Sam Zackrisson · 6 years ago
- 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
- ac63ac7 Update refcounting of AudioState to use rtc::RefCountedObject by Niels Möller · 6 years ago
- 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
- 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
- 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
- 3d2ed19 Remove Transport implementation from ChannelSend by Fredrik Solenberg · 6 years ago
- f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
- 2a977cf For audio receive channel use default max reordering threshold instead of 0 by Danil Chapovalov · 6 years ago
- 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
- 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
- e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
- ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
- 8ce0d2b In ReceiveStatistic require callbacks during construction by Danil Chapovalov · 6 years ago
- e3abb81 Decouple //rtc_base:rtc_base_tests_utils from gunit. by Mirko Bonadei · 6 years ago
- 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
- 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
- 8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
- 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
- c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
- 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
- 985a1f3 Add const or GUARDED_BY on a few ChannelSend members by Niels Möller · 6 years ago
- 26e88b0 Replace RTC_DCHECK by RTC_DCHECK_RUN_ON for worker thread. by Niels Möller · 6 years ago
- eb13484 Remove ChannelSendState by Fredrik Solenberg · 6 years ago
- c5e8be3 Remove ChannelReceiveState by Fredrik Solenberg · 6 years ago
- 78e88fe Move NetworkStatistics and AudioDecodingCallStats from common_types.h by Fredrik Solenberg · 6 years ago
- dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
- 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
- 1eebec9 Fix data race in channel_send.cc by Piotr (Peter) Slatala · 6 years ago
- 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
- 2681523 Tweak ChannelSend interface, to make it closer to ChannelSendProxy by Niels Möller · 6 years ago
- 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
- 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
- 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
- 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
- b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
- 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
- a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
- c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
- 967f7d5 Add audio level to CSRC class by Jonas Oreland · 6 years ago
- fd1a2fb Set RtpRtcp config receive_only in voe::ChannelReceive by Niels Möller · 6 years ago
- 273d029 Implement data channel methods in LoopbackMediaTransport. by Bjorn Mellem · 6 years ago
- 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
- 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
- 21cddff Harmonize paths to dependent targets. by Yves Gerey · 6 years ago
- fcc3981 Revert "Use only first payload timestamp for RTCP SR generation for audio" by Ilya Nikolaevskiy · 6 years ago
- 9a0662a Use only first payload timestamp for RTCP SR generation for audio by Ilya Nikolaevskiy · 6 years ago
- 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
- 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
- 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
- 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
- c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
- 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
- 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
- 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
- 2dfa998 Reland "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
- c538fc7 Revert "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
- 5ccdc13 Prefix flag macros with WEBRTC_. by Mirko Bonadei · 6 years ago
- bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
- 40a7a35 Extract functionality of test_main into separate library. by Artem Titov · 6 years ago
- b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 6 years ago
- 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
- 2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
- 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
- 96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
- be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
- ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
- 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
- 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
- 4a72ba9 Delete RtpReceiver and related code. by Niels Möller · 6 years ago
- b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
- 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
- 1f3206c Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1 by Niels Möller · 6 years ago
- 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
- 7988e5c Remove echo_cancellation() and echo_control_mobile() interface access outside APM by Sam Zackrisson · 6 years ago
- 637b0b5 Make Python-based performance tests output an empty result output.json by Oleh Prypin · 6 years ago
- db12856 Cleanup modules_common_types by Danil Chapovalov · 6 years ago
- 6151828 Delete always true member voe::Channel::pacing_enabled_ by Niels Möller · 6 years ago
- 5304a32 Delete StreamStatistician::IsRetransmitOfOldPacket by Niels Möller · 6 years ago
- 8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
- 2370b08 Revert "Update packetsLost and jitter stats any time a packet is received." by Qingsi Wang · 6 years ago
- 4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
- 46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
- fa2b2d6 Delete use of RtpPayloadRegistry. by Niels Möller · 6 years ago
- 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
- 9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
- fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
- 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
- 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
- 264bee8 Remove memcheck. by Mirko Bonadei · 6 years ago
- ab4a530 Delete telephone-event handling from RTPReceiverAudio. by Niels Möller · 6 years ago
- a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
- 3890262 Reland "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
- a61f7db Revert "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
- 06f66c7 Removing unneeded dependency. by Mirko Bonadei · 6 years ago