bpa wrote: 
> Assuming the "it" to be the SB Touch in the phrase "is *it* now decoding
> the streams natively?" 
> 
> "Apple HLS" is just a piece of text to indicate that the stream had a
> MIME type that meant it was going to be processed by the PlayHLSv1
> plugin.  There is no other significance.
> 
> HLS streams can be typically have the following audio formats: MP3,
> ADTS/AAC, MPEG2/AAC or MPEG4/AAC. 
> 
> PlayHLSv1 - uses ffmeg to take the HLS stream and produce a MP3, PCM or
> Flac format - depending on the player capability. ffmpeg would transcode
> bit rate and format into whatever player could handle (or 44.1kHz/16bit
> for PCM).  The source bitrate , sample size etc or metadata in the
> stream is not available to LMS from ffmpeg.
> 
> The plugin prcoesses the HLS stream "packets" (e.g. MPEG2, MPEG4) within
> LMS so it has access to bit rates, sample size and metadata (e.g. track,
> artist, station, images) if available and in a known format (metadata
> data format is not standardized and can vary by station/CDN). This data
> can then be presented via LMS to the UI.  
> 
> If the source is AAC wrapped in ADTS/AAC, MPEG2/AAC or MPEG4/AAC, the
> plugin produces an ADTS/AAC byte stream which is the AAC format that can
> be decoded "natively" by Touch, Radio and most SB software players - so
> these players can play the stream without LMS transcoding but still need
> the plugin (i.e. not direct streaming).
> 
> Older player such as SB3, Classic, Boom Transporter, Receiver,
> Softsqueeze and some builds of Squeezeplay cannot play AAC nativley - in
> which case LMS will transcode to a compatible format according to the
> FileTypes settings for AAC.
> 
> If the source format is MP3, then all players can play it natively.
> 
> The main "performance" difference between PlayHLSv1 and PlayHLS v2  
> - in v2 no ffmpeg so no additional process and memory (ffmpeg is big)
> - in v2 AAC is only transcoded as required and then using standard LMS
> faad which is lighter than ffmpeg.
> - v2 plugin is platform agnostic - no need to find & install a suitable
> ffmpeg/avconv 
> - in v1, Because ffmpeg couldn't generate suitable ADTS/AAC streams, AAC
> was transcoded even for AAC capable players.

Thanks again bpa. No wonder it's snappier than before!


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