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Created with Raphaël 2.2.028Mar272625242219181716151211109865426Feb2524221918171615141310985432131Jan2928272522201513121186430Dec22211615141110987542130Nov292827262523201918171312111093230Oct29282726252423222120191615141398652130Sep292826252322191716151312111098432131Aug28272625212018171413121176432131Jul29282725242220181615141398763230Jun29282726252220191816111098531May292827262524222120151413121176130Apr29282724222017161514131198732131Mar30292726252017161312111096543228Febwebrtc: remove unneeded patchconfigure: remove obsolete macroconfigure.ac: Harmonize indentation.configure.ac: Do not set user variables.configure.ac: Resolve libnatpmp conftest issue.icesocket: move ice transport off the main thread for destructionconfigure.ac: Remove uses of obsolete macros.build: Enable the use of the dbusxx-xml2cpp tool built from contrib.contrib: ffmpeg: Add the --pkg-config-flags="--static" option.getline: use input string as stream, skip emptyaddUserAgentHeader: fix typo, cleanupcontrib: update opendhtsipcall: use threadpool to destroy temp media transportmeson: rename the option from 'webrtc-audio-processing' to 'aec'accel: don't transfer frames with invalid formatcore layer: do not destroy stream when in useportaudio: refactor host api to get default comm devicessip: use correct status code for call refusal and call hangupRevert "jamiaccount: make currentDeviceId return const reference"jamiaccount: make currentDeviceId return const referencebuild: fix meson for webrtc supportvideo/mixer: fit the confInfo layout to the participant frame sizejamiaccount: use jami: instead of ring:message_engine: create directory if necessarysipcall: do not reset medias for ended callstring utils: add string_view split, update base64contrib: update opendhtAudioInput: apply mute/un-mute state when creating a new instanceSipVoipLink: prevent a crash at shutdownopus: re-enable FECupnp: code improvements and bug fixesplugins: enable audio/video plugins when hosting a conferencescripts: Do not hard-code /bin/bash.contrib(win32): use WASAPI as the only PortAudio host apiaec: add initial webrtc-audio-processing implementationportaudio: refactor to support split stream designaec: move aec implementations into EchoCancellercontrib: add webrtc-audio-processing @ v0.3.1opensl: use AEC when relevantaudio_rtp_session: fix setMuted