PJSIP/PJSUA with Wolfson audio card

I wonder if it is a bitrate issue. A google for that error suggests it might be, but it was unclear.

16000 is possibly unusual (I'm no expert). 8000 is often more likely. You could check your SIP software config file to see if you can set the bitrates used to something else.

I could be completely wrong, but it is worth a try.

Hi, thank you for the response. I could not configre this so I had to do some hard-coding. Given that I wrote the code correctly it did unfortunately not help.

08:34:18.298    pjsua_app.c !.Turning sound device ON

08:34:18.303    pjsua_aud.c !.Opening sound device PCM@8000/1/20ms

08:34:18.311     alsa_dev.c  ..ALSA lib pcm_hw.c:326:(snd_pcm_hw_hw_params) SNDRV_PCM_IOCTL_HW_PARAMS failed (-22): Invalid argument

Same situation for me. I have a voice app which normally runs in 8k mode. The app supports the rates 8k, 16k and 48k, but only 48k works. There is no advantage for the voice processing from using 48k, but it unnecessarily increases the CPU load.

It would be nice to get some feedback from Wolfson what sampling rates will be supported by future driver releases (on the analogue line in/out connectors).

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 use your .asoundrc, and the wolfson PI Card works, but I Can't reproduce the files this is the error

root@raspberrypi:/home/pi/pjproject-2.2.1/pjsip-apps/bin# aplay PJSYSTEST_TESTREC.WAV               Playing WAVE 'PJSYSTEST_TESTREC.WAV' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono

aplay: set_params:1145: Unable to install hw params:

ACCESS:  RW_INTERLEAVED

FORMAT:  S16_LE

SUBFORMAT:  STD

SAMPLE_BITS: 16

FRAME_BITS: 16

CHANNELS: 1

RATE: 8000

PERIOD_TIME: 125000

PERIOD_SIZE: 1000

PERIOD_BYTES: 2000

PERIODS: 4

BUFFER_TIME: 500000

BUFFER_SIZE: 4000

BUFFER_BYTES: 8000

TICK_TIME: 0

Can the Wolfson PI Audio Card play files with 8000hz, this is a critical point in the asterisk communication because the principal audio codecs run to 8000hz. do you know?

I'm graceful with your answer

The card itself (or rather its drivers) cannot record or play back at 8 kHz.

Using my .asoundrc you should be able to record and play back at any sample rate and bit depth.

ragnar@ob3:~/tmp$ arecord -d 10 -f S16_LE -v testing16.wav

Recording WAVE 'testing16.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono

Plug PCM: Rate conversion PCM (48000, sformat=S16_LE)

Converter: libsamplerate

Protocol version: 10002

Its setup is:

  stream : CAPTURE

  access : RW_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 1

  rate : 8000

  exact rate : 8000 (8000/1)

  msbits : 16

  buffer_size : 3000

  period_size : 1000

  period_time : 125000

  tstamp_mode : NONE

  period_step : 1

  avail_min : 1000

  period_event : 0

  start_threshold : 1

  stop_threshold : 3000

  silence_threshold: 0

  silence_size : 0

  boundary : 196608000

Slave: Route conversion PCM (sformat=S16_LE)

  Transformation table:

  0 <- 0*0.5 + 1*0.5

Its setup is:

  stream : CAPTURE

  access : MMAP_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 1

  rate : 48000

  exact rate : 48000 (48000/1)

  msbits : 16

  buffer_size : 18000

  period_size : 6000

  period_time : 125000

  tstamp_mode : NONE

  period_step : 1

  avail_min : 6000

  period_event : 0

  start_threshold : 6

  stop_threshold : 18000

  silence_threshold: 0

  silence_size : 0

  boundary : 1179648000

Slave: Direct Snoop PCM

Its setup is:

  stream : CAPTURE

  access : MMAP_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 2

  rate : 48000

  exact rate : 48000 (48000/1)

  msbits : 16

  buffer_size : 18000

  period_size : 6000

  period_time : 125000

  tstamp_mode : NONE

  period_step : 1

  avail_min : 6000

  period_event : 0

  start_threshold : 6

  stop_threshold : 18000

  silence_threshold: 0

  silence_size : 0

  boundary : 1179648000

Hardware PCM card 0 'snd_rpi_wsp' device 0 subdevice 0

Its setup is:

  stream : CAPTURE

  access : MMAP_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 2

  rate : 48000

  exact rate : 48000 (48000/1)

  msbits : 16

  buffer_size : 18000

  period_size : 6000

  period_time : 125000

  tstamp_mode : ENABLE

  period_step : 1

  avail_min : 6000

  period_event : 0

  start_threshold : 1

  stop_threshold : 1179648000

  silence_threshold: 0

  silence_size : 0

  boundary : 1179648000

  appl_ptr : 0

  hw_ptr : 0

 

 

 

ragnar@ob3:~/tmp$ file testing16.wav

testing16.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

 

 

 

ragnar@ob3:~/tmp$ aplay -v testing16.wav

Playing WAVE 'testing16.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono

Plug PCM: Soft volume PCM

Control: Master

min_dB: -51

max_dB: 0

resolution: 256

Its setup is:

  stream : PLAYBACK

  access : RW_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 1

  rate : 8000

  exact rate : 8000 (8000/1)

  msbits : 16

  buffer_size : 3000

  period_size : 1000

  period_time : 125000

  tstamp_mode : NONE

  period_step : 1

  avail_min : 1000

  period_event : 0

  start_threshold : 3000

  stop_threshold : 3000

  silence_threshold: 0

  silence_size : 0

  boundary : 196608000

Slave: Plug PCM: Rate conversion PCM (48000, sformat=S16_LE)

Converter: libsamplerate

Protocol version: 10002

Its setup is:

  stream : PLAYBACK

  access : MMAP_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 1

  rate : 8000

  exact rate : 8000 (8000/1)

  msbits : 16

  buffer_size : 3000

  period_size : 1000

  period_time : 125000

  tstamp_mode : NONE

  period_step : 1

  avail_min : 1000

  period_event : 0

  start_threshold : 3000

  stop_threshold : 3000

  silence_threshold: 0

  silence_size : 0

  boundary : 196608000

Slave: Route conversion PCM (sformat=S16_LE)

  Transformation table:

  0 <- 0

  1 <- 0

Its setup is:

  stream : PLAYBACK

  access : MMAP_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 1

  rate : 48000

  exact rate : 48000 (48000/1)

  msbits : 16

  buffer_size : 18000

  period_size : 6000

  period_time : 125000

  tstamp_mode : NONE

  period_step : 1

  avail_min : 6000

  period_event : 0

  start_threshold : 18000

  stop_threshold : 18000

  silence_threshold: 0

  silence_size : 0

  boundary : 1179648000

Slave: Direct Stream Mixing PCM

Its setup is:

  stream : PLAYBACK

  access : MMAP_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 2

  rate : 48000

  exact rate : 48000 (48000/1)

  msbits : 16

  buffer_size : 18000

  period_size : 6000

  period_time : 125000

  tstamp_mode : NONE

  period_step : 1

  avail_min : 6000

  period_event : 0

  start_threshold : 18000

  stop_threshold : 18000

  silence_threshold: 0

  silence_size : 0

  boundary : 1179648000

Hardware PCM card 0 'snd_rpi_wsp' device 0 subdevice 0

Its setup is:

  stream : PLAYBACK

  access : MMAP_INTERLEAVED

  format : S16_LE

  subformat : STD

  channels : 2

  rate : 48000

  exact rate : 48000 (48000/1)

  msbits : 16

  buffer_size : 18000

  period_size : 6000

  period_time : 125000

  tstamp_mode : ENABLE

  period_step : 1

  avail_min : 6000

  period_event : 0

  start_threshold : 1

  stop_threshold : 1179648000

  silence_threshold: 0

  silence_size : 1179648000

  boundary : 1179648000

  appl_ptr : 0

  hw_ptr : 0

ragnar@ob3:~/tmp$

 

I notice that you are running aplay as root in your example.

Have you put a copy of .asoundrc in root's home directory too? Every user needs its own copy.

Or install a global one: Asoundrc - AlsaProject

--

Ragnar

HI Ragnar

Thanks for your answer, you are correct, the fail was the asoundrc doesn`t exist in the root folder

How is the CPU usage compared to a USB sound card?

OK, so has the driver update been completed?  Being without an 8K sample rate makes doing SIP telephone difficult, since much telephony is 8K PCM mulaw or alaw.