PJSIP/PJSUA with Wolfson audio card

I too have been bitten by the sample rate bug...

I have modified my ~/.asoundrc quite a bit in order to be able to record 1-channel at 8kHz and 16kHz.

I'm using ALSA's built-in re-sampling capabilities.

This is not for those who like to enjoy HD audio! The sound card operates at 48kHz/16bit at all times.

Everything is re-sampled and converted up or down as needed.

--

Ragnar

#######################################################################
#
# Use libsamplerate instead of internal resampler.
# You might have to: sudo apt-get install libasound2-plugins
# Or just comment out the next line if the internal one is good enough for your needs.
defaults.pcm.rate_converter "samplerate"

######################################################################
#
# Wolfson Audio Card for Raspberry Pi
#
pcm.wolfson_pi_soundcard {
    type hw
    card sndrpiwsp
    device 0
}
# Create a Master volume control
pcm.softvol {
    type            softvol
    slave {
        pcm         "pduplex"
    }
    control {
        name        "Master"
        card        0
    }
}

pcm.!default  {
    type           asym
    playback.pcm   "plug:softvol"
    capture.pcm    "pduplex"
}

#####################################################################
#
# Mixing and resampling goodness :-)
#
# Several clients can record and play back simultaneously.
# The dmix and dsnoop plugins operate at 48kHz by default, i.e. the sound card will be kept at 48kHz at all times.
# All playback streams are converted to 2 channels/48kHz/16bit.
# All recording streams are converted from the card's 2 channels/48kHz/16bit
# to the requested number of channels, sample rate and bit depth.

# We define one pcm device called "playmixer" for playback.
pcm.playmixer {
        ipc_key 140324  # Any number will do, as long as it's unique in the system.
        ipc_key_add_uid true
        type dmix
        slave.pcm "wolfson_pi_soundcard"
}

# Define another one called "recmixer" for capture.
# The sound card will run at 48kHz and input is resampled to the requested rate.
pcm.recmixer {
        ipc_key 140325
        ipc_key_add_uid true
        type dsnoop
        slave.pcm "wolfson_pi_soundcard"
}

# Define a duplex device, with both in- and outputs.
pcm.duplex {
        type asym
        playback.pcm "playmixer"
        capture.pcm "recmixer"
}

# Add a "plug" plugin for the above "duplex" device.
# This is where all the converting takes place.
# Sample rate, number of channels, bit depth.
# By the way, who thought that "plug" was a good name for a plugin?
#
# Use this device for all your playback and recording resampling needs.
pcm.pduplex {
        type plug
        slave.pcm "duplex"
}

# A ctl device to keep xmms happy
ctl.pduplex {
        type hw
        card 0
}

# A DSP to keep the alsa-oss layer happy:
pcm.dsp0 {
        type plug
        slave.pcm "duplex"
}

ctl.mixer0 {
        type hw
        card 0
}

Hi Ragnar,

I'm using your ~/.asoundrc file above and get this error

ALSA lib pcm_direct.c:877:(snd1_pcm_direct_initialize_slave) slave plugin does not support mmap interleaved or mmap noninterleaved access

ALSA lib pcm_dmix.c:1030:(snd_pcm_dmix_open) unable to initialize slave

aplay: main:682: audio open error: Invalid argument

Any help would be really appreciate.

Thank you,

Palm

Is that with the 3.12 kernel? It needs a patch to enable memory mapped I/O.

georgpi has made a patched kernel available at https://blog.georgmill.de/2015/02/18/update-for-wolfson-audio-card-on-raspberry-pi/

Full discussion: http://www.element14.com/community/message/141324/l/cirrus-audio-b-and-jack#141324

--

Ragnar

Hi Ragnar,

No, I'm using the 3.18 kernel on raspberry pi with my custom build kernel to include support for WM8731 codec board from Mikroe. The driver is missing from 3.18 kernel so I copied the code from the older kernel. I've loaded these kernel modules (snd_soc_wm8731, snd_soc_rpi_proto) according to koalo's blog: I²S Audio Support for Raspberry Pi. It work perfectly and I can play and record sound but at 48kHz stereo (2ch) only. My task is to perform speech recognition using the HTK toolkit or Julius. These software only support mono sound input and at 16kHz or 8kHz sampling rate. Thus I think of using alsa plugin dsnooped to split sound from stereo to mono and to convert the sample rate in software. I've tried using mamp_emul plugin but it said the dsnooped can't work with mmap emulator. Can I just apply your patch here http://www.element14.com/community/message/141324/l/cirrus-audio-b-and-jack#141324 (http://www.element14.com/community/servlet/JiveServlet/download/141528-167303/cirrus_logic_mmap.patch.zip) or do I need to do other thing too?

Thank you,

Palm

Not a Wolfson/Cirrus Logic card, then...

The patch is meant for the already heavily patched 3.12 kernel sources from Cirrus Logic.

On the other hand, it's a very small patch. It does apply cleanly to the 3.18 source (if using the "patch" command's -l switch) and I have been able build a 3.18 kernel with it.

It was the only thing I needed to do to get mmap going in the 3.12 bcm2708-i2s driver.

But I don't have any hardware to test it on 3.18.

So, the definite answer is: maybe

--

Ragnar

Not a Wolfson/Cirrus Logic card, then...

The patch is meant for the already heavily patched 3.12 kernel sources from Cirrus Logic.

On the other hand, it's a very small patch. It does apply cleanly to the 3.18 source (if using the "patch" command's -l switch) and I have been able build a 3.18 kernel with it.

It was the only thing I needed to do to get mmap going in the 3.12 bcm2708-i2s driver.

But I don't have any hardware to test it on 3.18.

So, the definite answer is: maybe

--

Ragnar