Looking for plugin which parses stream for sample rate/bits per sample

[Mnyb]

Thats a feature Squeezeboxes always lacked to be able to route analog inputs trough the house , quite usefull , for exqample LP playback (but that wont need 24/192 ) .

If LMS is running on one of RPi you get exaclty that if you suceed

[bpa]

A bit of background would help here I'm sure.

This info has helped greatly. Since this is a analog source - why did you mention multi channel multiplexing in the other thread - I had assumed you were recording from the digital input.

If the recording is fixed at 192/14/ch and it never changes then using Flac as the output stream means the Wavin plugin conf file can be hardcoded to your application and no changes are need to plugin.

Thats a feature Squeezeboxes always lacked to be able to route analog inputs trough the house , quite usefull , for exqample LP playback (but that wont need 24/192 ) .

The WaveInput plugin requires LMS to be running on the device. The use-case I saw was remote device with no LMS such as Touch player - and user wants to route audio form remote player to LMS which then can then be played on any or all SB players. For personal use, I did a rough version of plugin to do this where the player used netcat to create a Flac stream from audio input streamed over a specific tcp/ip socket and the modded version wavinput which would create a netcat to connect to player when remote audio streaming was required. Using netcat means remote audio port can be on a Windows, OSX or Linux box (e.g. Touch).

[mike_b16]

why did you mention multi channel multiplexing in the other thread

Apologies here, this was due to my idiocy in the first two thirds of the last thread where I read into your description of the plugin too much and thought I literally needed local playback on the card in order to "copy" that using arecord, which would have required multiple instances of arecord for me.

If the recording is fixed at 192/14/ch and it never changes then using Flac as the output stream means the Wavin plugin conf file can be hardcoded to your application and no changes are need to plugin.

So can I simply remove the lines that deal with PCM and mp3 in your custom-convert.conf file and leave the two lines that deal with converting the stream to FLAC? Or is there a better way to modify it?

I have hope, a solution is near!

Thanks,

Michael

[bpa]

So can I simply remove the lines that deal with PCM and mp3 in your custom-convert.conf file and leave the two lines that deal with converting the stream to FLAC? Or is there a better way to modify it?

Remove the PCM line as there is no easy way this is going to work.
Remove the MP3 because even though it can be made to work - why waste time working out the correct lame option settings.

The flac command line to encode the stream has to have option telling flac that the input is pcm 192 24bit 2 ch (or whatever you are using) and then the output will have a header. If you wish to support downsampling - you should look at main convert.conf file to how to do downsampling using flac.

[philippe_44]

A bit of background would help here I'm sure.

I have 3 Raspberry Pi units. One has a Wolfson audio card on it, and the other two have HiFiBerry DACs attached to them. I have a high quality audio source (capable of outputting 24/192/2ch analog audio) which I want to be able to pipe to the rest of my home, similar to a Sonos system. I wanted to do this using the RPi with the Wolfson card as a server which can pull this HQ audio stream in at 24/192/2ch and send it over the network to my 2 RPi units with the HiFiBerry DACs. The Wolfson card is capable of 24/192 recording/playback and the HiFiBerry DACs are capable of 24/192/2ch playback. After researching several solutions based on my requirements (playback sync, reliable stream, don't care too much about latency as long as it's within 5 seconds), I decided on using LMS on my Wolfson Pi with bpa's WaveInput plugin. Now, it appears that the plugin requires some modification if I want to allow this 24/192/2ch capability.

As for what I want, I don't care whether the stream is sent as PCM or FLAC. As long as I get that 24/192/2ch playback at the sources. If I understand it, the format of the custom-convert.conf file in the WaveInput plugin needs to be written using a very specific format (including sections for PCM, mp3, and FLAC). However, if this can be modified to just send FLAC and I can skip this problem with PCM not containing format information in the headers, that would be great.

Cheers,

Michael

Sorry for the dumb question, but the audio source is only available in analogue format ? Other than that, would creating a source stream using (eg) vlc work ? that source stream could then be a radio feed fed back for LMS - if you are not sensitive to delay, that would probably work, I think

[Mnyb]

This info has helped greatly. Since this is a analog source - why did you mention multi channel multiplexing in the other thread - I had assumed you were recording from the digital input.

If the recording is fixed at 192/14/ch and it never changes then using Flac as the output stream means the Wavin plugin conf file can be hardcoded to your application and no changes are need to plugin.



The WaveInput plugin requires LMS to be running on the device. The use-case I saw was remote device with no LMS such as Touch player - and user wants to route audio form remote player to LMS which then can then be played on any or all SB players. For personal use, I did a rough version of plugin to do this where the player used netcat to create a Flac stream from audio input streamed over a specific tcp/ip socket and the modded version wavinput which would create a netcat to connect to player when remote audio streaming was required. Using netcat means remote audio port can be on a Windows, OSX or Linux box (e.g. Touch).

Yes op has LMS a running on the RPi he intends to stream from , but your idea with netcat sounds good indeed . But you need inputs analog or digital , so some USB add on or this special board for RPi is needed . Personally I have reduced all my source components to digital ones. And afaik no ones has yet provided a low latency way of doing it so TV a sound would not be a really good idea , but terrestrial radio and other audio only sources

[bpa]

Sorry for the dumb question, but the audio source is only available in analogue format ? Other than that, would creating a source stream using (eg) vlc work ? that source stream could then be a radio feed fed back for LMS - if you are not sensitive to delay, that would probably work, I think

User wants lossless 192Khz/24 bit stream. I have never been able to create a http/flac stream using vlc btu perhaps vlc has been updated recently. MP3 stream can be created easily.

If you have a vlc command line for creating a http/flac stream - please share.

[mike_b16]

If you wish to support downsampling - you should look at main convert.conf file to how to do downsampling using flac.

Downsampling is not needed in my case; all of the players support 24/192/2ch.

Sorry for the dumb question, but the audio source is only available in analogue format ? Other than that, would creating a source stream using (eg) vlc work ? that source stream could then be a radio feed fed back for LMS - if you are not sensitive to delay, that would probably work, I think

Yes, the audio from my HQ audio sources is only available in analog format. While the streaming with vlc would also (hopefully) work too, I wouldn't be the only one using this solution, and I wanted to have a nice interface and an easy-to-control method as well as synchronization of the playback streams. I have a feeling that would be complex with that direct streaming method.

Remove the PCM line as there is no easy way this is going to work.
Remove the MP3 because even though it can be made to work - why waste time working out the correct lame option settings.

After this suggestion, I wanted to make sure that whatever command I have in the custom-convert.conf file would be more or less usable from the command line. I did several tests with this, and ran into problems. I tested playback of the .flac files I created with mplayer, since arecord/aplay doesn't support flac files. I downloaded a 24/192/2ch flac test file from Linn Recordings to ensure that I could playback a 24/192/2ch file locally on my server unit, which I was able to do through mplayer with no problems. However, when I tried to create a line close to what I'd be using in the custom-convert.conf file using "arecord -d0 -c2 -f S24_LE -r 192000 -traw -D hw:0 | flac -cs --endian=little --sign=signed --channels=2 --bps=24 --sample-rate=192000 --compression-level-0 - -o flac_raw24.flac", I simply got static with warped music underneath. When I change the bits per sample (in both the arecord portion and the flac portion) though, mplayer plays back the file fine. These two behaviors also occur if I change the -t field to wav. Changing the sample rate doesn't change the behavior. The mplayer states that the file is s32le format (which I thought might be a problem), but it also states that the working flac file from Linn is s32le. Also, I tried to simply record 24 bit and 16 bit audio to a wav file (not using the flac conversion) and play it back with mplayer. mplayer will sound the same as the 24 bit flac files I created (static, warped music), but when I play it through aplay, it plays fine.

Does anyone know why the flac conversion is crapping out like this at 24 bits? The Wolfson card can definitely support 24/192/2ch capture/playback. I know this thread is running a bit off topic, but I feel like this is the last hurdle before a working solution. I've attached screenshots of all the flac conversions/wav captures/playbacks I attempted and the mplayer results when I play them. test192.flac is the Linn test file:







Thanks for all the responses so far!

Cheers,

Michael

[bpa]

Yes op has LMS a running on the RPi he intends to stream from , but your idea with netcat sounds good indeed . But you need inputs analog or digital , so some USB add on or this special board for RPi is needed . Personally I have reduced all my source components to digital ones. And afaik no ones has yet provided a low latency way of doing it so TV a sound would not be a really good idea , but terrestrial radio and other audio only sources

1. A plugin that is able to play from a netcat TCP/IP stream won't care where the stream comes from so the plugin is general purpose and source system agnostic. It is up to user to create the source in the case of RPI if recording audio from an external source then some extra hardware is needed. This RPI user has a Wolfson card which has both an analogue and digital input. This sort of plugin would have clean netcat interface and so user will never have to fiddle with LMS plugins or code.

2. LMS based solutions will always have latency. If user doesn't want latency - then don't use LMS. I have never offered low latency.

[bpa]

Does anyone know why the flac conversion is crapping out like this at 24 bits? The Wolfson card can definitely support 24/192/2ch capture/playback.

IIRC Alsa can do conversion if application and driver don't match.

Did you run the hw_params program I suggested in http://forums.slimdevices.com/showth...l=1#post798887 to find out exactly what the Wolfson driver supports natively ?