[Release] 3.1 stable release
The 3.1 release is upon us! Downloads can be found here.
First of all: apologies, this build is still Linux-only for now. While a lot of progress has been made on fixing MacOS and Windows compatibility in the past few months, it's not quite ready yet and we felt it prudent to release some of the fixes and features this release brings earlier rather than later.
This release also marks a new style of release post. We now mention noteworthy features with some text explaining what the feature is and why you'd want to use it, and of course the full changelog is still available as per usual.
What: Support for Apple's LLHLS (Low Latency HTTP Live Streaming) protocol was already included in 3.0. However, that support is CMAF-based (the latest industry standard). HLS also supports an older TS-based segmenting method - this feature adds support for LLHLS in that format as well.
Why: This feature is useful for those that want/need to support LLHLS but are bound to using MPEG2-TS-based playback for some arbitrary reason (e.g. legacy device support, weird TVs they are using, etc).
Forward Error Correction
What: This adds support for ProMPEG Forward Error Correction to our MPEG2-TS-based output over UDP. FEC adds an additional layer of error correction packets on top of the normal packets, to allow for recovering from packet loss using parity data.
Why: Sometimes a connection can only be made over UDP, but there is a small amount of packet loss you want to correct for. This makes that possible. If you're not stuck using UDP, there are likely better solutions than this one.
Input SDP from file / Push output SDP to file
What: RTSP, WebRTC and SIP (VoIP) all use RTP-based transport for their media data. Some users want to bypass those signalling protocols and instead work with raw RTP over the network. The SDP input and output allows you to preconfigure the "handshake" normally done over the signalling protocol, so you can directly send or receive media without needing to do that handshake.
Why: This is especially useful when broadcasting multicast over, say, a company intranet - since the "handshake" should only be done once but there may be potentially infinite receivers in the network. They can all open the SDP file to receive the data needed to connect. Until this feature became available, multicast over the local network was not possible using RTP and only using raw UDP (in an MPEG2-TS transport). The support for SDP is not only multicast usecases, though - that is just the most noteworthy one.
AAC file input support
What: The AAC codec is already supported in Mist, but separate
.aac files are not (they need to be inside some other container, instead). This update adds support for plain
.aac files as a VoD input format.
Why: Because this is the most efficient/best format to store separate AAC files, and it wasn't supported yet. In addition, reading AAC files is needed to be able to replace AAC audio tracks with custom audio when pushing RTMP out, a feature that is also part of this release.
Support for overriding AAC audio in RTMP push output
What: If a stream contains AAC audio, this feature lets you replace the audio in a stream with a looped version of an external AAC audio file. The original audio is not transmitted.
Why: This was built to support the use case that the stream may contain audio material that causes e.g. Youtube or Twitch to block the stream (e.g. copyrighted songs etc), and you want to replace the audio track with a placeholder for sending a "lite" version of the stream to these platforms.
Split config support
What: The ability to split up the config file by section, so that each section can be loaded from a different file on the system.
Why: This makes it easier to set up a generic config that can be re-used and synced across multiple installs, while keeping some of the config stored separately for each install that won't be overwritten. It's also possible to make part of the config read-only and other parts read-write by setting file permissions of their respective files accordingly.
What: RIST is a semi-reliable transport, similar to Haivision SRT, Zixi and BRT. This means that it makes a connection that cannot be controlled (e.g. over the public internet instead of private networks) fairly predictable in behaviour. This adds support for RIST push output and RIST pull input, both in both caller and callee modes. Unfortunately our build system does not yet support building the RIST library itself, so this feature is currently only available if you compile MistServer yourself.
Why: RIST is rapidly gaining popularity, and there is still no clear "winner" in the reliable transport protocols segment today. Mist aspires to connect anything to anything, so we try to support as many relevant protocols as possible.
HEVC/H265 support in browsers
What: Most browsers don't support decoding of H265/HEVC video streams through any method (with or without hardware acceleration). A notable exception is recent versions of iOS, but for the rest mobile and desktop browsers alike generally won't support this codec. It is, however, possible to transpile
Why: H265/HEVC has a much better compression ratio than H264/AVC does, which can be critical in ultra low bandwidth situations. Unfortunately, software decoding of H265/HEVC is fairly slow - so this works best for low framerate or low resolution streams. The player does automatically skip frames to stay roughly at real time speed if/as needed, though - so even full resolution playback should be acceptable quality. There is another downside: the player currently only supports video, no audio.