What is the ideal way to play-through or monitor audio input using arecord, parec or jackd?

Question

I have failed to pipe the recorded audio input on my audio device (an alsa driven hifiberry with an adc hosted on a raspberry pi) despite trying my best to pipe the output into tee after arecord, install and set up pulseaudio and utilize parec, as well as jackd and jack_capture.

Critical diagram of my mission:

Line level audio -> ADC -> PCM/Wav File -> DAC -> line level audio.

The problem is that no output ever plays during recording.

I am recording a line level input and would like to hear it. Latency is not critical but I expect it to be 200ms or less.

I have succeeded in this incredibly hackish solution which is just to "arecord" in one terminal window and "aplay" in another (which works) but this cannot possibly be the solution to my problem.

To compound the issue, my attempts at googling this have failed miserably as google believes I simply must be trying to capture the output of one application to a file. I am not. I want to monitor the sound card's input. In Apple's Logic Pro this is referred to as "software monitoring" -- I figured this would be easy. I have also seen it referred to as "play-through" but maybe this is something else.

My hopes were raised with pulseaudio -- it's just "sources and sinks" they said.

I did succeed to record a pcm file with pulseaudio's parec, alsa's arecord and jackd + jack_capture. I'm clearly missing something obvious.

parec -d alsa_input.platform-soc_sound.stereo-fallback | sox -t raw -b 16 -e signed -c 2 -r 44100 - /mnt/audio/pulsetest.wav

Clearly my os and hardware can "duplex" because I can arecord and aplay at the same time.

Can this be done or should I continue using arecord and aplay?

Answer 1

Please try something like

pacmd load-module module-loopback source="alsa_input.platform-soc_sound.stereo-fallback" sink="alsa_output.whatever"

where "alsa_output.whatever" is the actual name of the line-out sink. You can see the sinks with

pacmd list-sinks | grep name:

I thought "loopback" is a term relegated to capturing system audio

"Loopback" is a generic term for sending some output back to some input. You have a loopback interface for networking (lo), a loopback function on real LAN devices, an ALSA loopback driver, and the Pulseaudio loopback module, and probably many more.

I set default-sample-format and rate in the pulse conf

Don't do this, Pulseaudio does resampling. Don't mess with anything else, just establish a module-loopback connection from your source to your sink.

It should be so simple.

It is simple. I have no idea what you are doing to make it difficult.

I guess i should bust out a sniffer and probe the chips to see what alsa is doing.

This is a fun thing to do (in particular on real PC, with Intel HDA drivers - on a RaspPi it's very boring), but since you are working on the Pulseaudio level, and not ALSA, probably not helpful. As you have a working aplay and arecord, this shouldn't be an issue.

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If you have the additional requirement (which you never mentioned in your question) of having the source and the sink use 96k/24bit, then this is going to be a journey in Pulseaudio: Pulseaudio is designed to run ALSA hardware sources/sinks with reasonable defaults, and then will upsample/downsample when streaming, as required. Fiddling with Pulseaudio internals to change that is tricky.

Have a look at the module-loopback docs: You can set the desired rate and format with extra parameters (and that's what you should do), but there's no guarantee that the source and sink will end up in this mode, and Pulseaudio won't do resampling somewhere.

You also didn't say where this requirement comes from; if you want to use the RaspPi to do professional-grade audio-processing (DAW, digital audio workstation), then JACK is a better choice than Pulseaudio. But all of this depends on your situation.

That said, if you are running module-loopback together with parec --rate=96000 --format=s24le, then of course it cannot change bitrate and format on a running sink, but if you already got the format and bitrate you want, then just leave it alone and be happy.

And yes, with Pulseaudio, you can have multiple streams from a source, and multiple streams into a sink.