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ZRTP
libjitsi
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5db7435adf986487b683802279ce052d845c12fd
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8
deb-src-release-593
protected
firs-plis
hw-decoding
jitsi_android
master
default
protected
reusable-rtp-scheduler
rewriting
shortaudio
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28 results
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Created with Raphaël 2.2.0
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Fixes an audio quality degradation observed with Core Audio on OS X when the Audio tab in Preferences is open during a call.
Removes duplication.
Removes duplication.
Fixes a case which may lead to ignoring non-distinguishing but important format parameters/SDP fmpt such as H.264 sprop-parameter-sets.
Updates FMJ to r41 (which fixes warnings). Exposes a couple of fields.
Updates CoreAudio to be default and echo cancel to be activated for it by default on macosx.
Updates AEC for CoreAudio using webrtc library, recompiled with latest webrtc, was crashing when running from source with unresolved reference "___sincos_stret". Adds some basic instructions for compiling.
Removes tab char.
Enhances the support for received H.264 SDP format parameters such as sprop-parameter-sets.
Filters RTP packets in accord with the MediaStream direction. Fixes problems with new ReceiveStream-s being created while on hold.
Stops all receive streams (same reasoning as in the previous commit for starting them).
Makes sure that the file is copied even if the jitsi repo has a version with a newer date (which could happen after a git pull).
In startReceiveStreams(), start both streams from the RTP manager and streams
Prevent NP dereference on IMediaObject
Prevent WASAPI silence insertion to overflow
Add target to copy libjitsi to jitsi
Init COM on the WASAPI stream processing thread
Fix some possible NPEs
Extracts an internal class into its own file for the sake of readability.
Updates fmj.jar to r40 (Fakes a TimeoutEvent when a ReceiveStream is replaced with a PassiveSSRCInfo (e.g. when an RTCP Receiver Report is received, instead of a Sender Report, for the ReceiveStream).).
Fixes warnings.
Calls stop() and disconnect() on DataSource-s when they are removed from
Makes sure that writeLock is released.
Fixes warnings.
Fixes 'Deadlock between GUI and ReceiveStream' reported by Ingo Bauersachs.
For RTCP FIR messages, use the SSRC from the packet FCI, instead of the 'SSRC of media source' field (see RFC5104 sections 4.3.1.1 and 4.3.1.2).
Set the RTP Marker flag for the last RTP packet of a frame.
Fix a bug in calculating the VP8 Payload Descriptor size
Fixes an IllegalStateException in FileAccessServiceImpl when executing inside Jitsi Videobridge.
Allows getting the complexity set on the Opus encoder (for debugging/informational purposes).
Updates bouncycastle to 1.50.
Fixes the sending of RTCP feedback messages to support Full Intra Refresh (FIR).
Call CoInitialize on AEC init
Return an empty format list instead of null
Fixes a deadlock in the ICE implementation between Agent.free() and StunClientTransaction observed in Jitsi Videobridge.
Fixes an issue with Packet Loss Concealment (PLC) which could cause muting in Jitsi Videobridge conferences to 'repeat' after unmuting.
Uses a single Thread per StunStack to expire StunServerTransactions in order to reduce the total number of utilized Threads.
Defaults to not dropping received RTP packets from muted audio sources and allows the behavior to be enabled through a ConfigurationService property.
Uses a Thread pool for StunClientTransaction in order to improve performance.
Let ZRTP childstreams report the verification status of the main stream
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