Neo 1973 audio subsystem

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In the center of the Neo1973 audio subsystem is the WM8753 (the "Wolfson Codec"), which implements record (ADCs), playback (DACs), and signal mixing. On the stereo output is the LM4857 amplifier, which drives the stereo speakers, the mono earpiece and the headphones. Sound from and to GSM is received from and sent to the GSM modem via analog connections. There's a digital mono interface for sound from and to the Bluetooth chip.


ALSA Channels

The channel numbers shown here are for the Freerunner, not the 1973.

Note: The ALSA channel numbers changed, probably during this change which is just after the release of 2.6.34:

commit d4b8fdb48cf1596e99f2c5cce58ef37bf94221b2
Author: Lars-Peter Clausen <>
Date:   Sat Oct 17 15:02:22 2009 +0200

    Merge gta01 and gta02 sound soc driver.

See the table below for old & new numbers. At the moment, all of the diagrams reflect the old control numbers. (Most of the diagrams are not SVG so it is difficult to fix this!)


WM8753 ALSA Mapping.png A way more pretty diagram is here

Png version with ALSA control names printed over (these are alsa controls like found in statefiles or amixer commands, alsamixer removes trailing "Playback Volume" and such where it sees fit) Inkscape source: WM8753 routing diagram alsa controls.png

Best of both worlds, Png with the ALSA control names *and* the numid's {in blue}: WM8753 routing diagram alsa controls 20101112.png

List of controls

[[Names for aliases in new driver. "NEW"=recommended, "new"=suggested. No kernel uses the NEW/new names at least as of 2010-08-03. Update if some great renaming really happens.]]

(columns are sortable)

Name Old control number post-2.6.34 control number command or recommended/suggested name
PCM Volume 1 1
ADC Capture Volume 2 2
Headphone Playback Volume 3 3 NEW:'Speaker Playback Volume'
Speaker Playback Volume 4 4 NEW:'Earpiece Playback Volume'
Mono Playback Volume 5 5 NEW:'GSM Mic Capture Volume'
Bypass Playback Volume 6 6 new:'GSM Play Playback Volume'
Sidetone Playback Volume 7 7 new:'Mic Sidetone Playback Volume'
Voice Playback Volume 8 8 new:'BT-Mic PB Playback Volume'
Headphone Playback ZC Switch 9 9 NEW:'Speaker Playback ZC Switch'
Speaker Playback ZC Switch 10 10 NEW:'Earpiece Playback ZC Switch'
Mono Bypass Playback Volume 11 11 new:'GSMout2GSMmic feedback! (deprecated)'
Mono Sidetone Playback Volume 12 12 NEW:GSM Mic Mixer Capture Volume'
Mono Voice Playback Volume 13 13 NEW:'GSM BTmic Capture Volume'
Mono Playback ZC Switch 14 14 NEW:'GSM Mic Capture ZC Switch'
Bass Boost 15 15
Bass Filter 16 16
Bass Volume 17 17
Treble Volume 18 18
Treble Cut-off 19 19
Sidetone Capture Volume 20 20
Voice Sidetone Capture Volume 21 21
Capture Volume 22 22
Capture ZC Switch 23 23
Capture Switch 24 24
Capture Filter Select 25 25
Capture Filter Cut-off 26 26
Capture Filter Switch 27 27
ALC Capture Target Volume 28 28
ALC Capture Max Volume 29 29
ALC Capture Function 30 30
ALC Capture ZC Switch 31 31
ALC Capture Hold Time 32 32
ALC Capture Decay Time 33 33
ALC Capture Attack Time 34 34
ALC Capture NG Threshold 35 35
ALC Capture NG Type 36 36
ALC Capture NG Switch 37 37
3D Function 38 38
3D Upper Cut-off 39 39
3D Lower Cut-off 40 40
3D Volume 41 41
3D Switch 42 42
Capture 6dB Attenuate 43 43
Playback 6dB Attenuate 44 44
De-emphasis 45 45
Playback Mono Mix 46 46
Playback Phase 47 47
Mic2 Capture Volume 48 48 new:'int.MIC Gain Capture Volume'
Mic1 Capture Volume 49 49 new:'HS-Mic Gain Capture Volume'
DAI Mode 50 50
ADC Data Select 51 51
ROUT2 Phase 52 52
Mic Selection Mux 53 60
Rx Mixer 54 61
Line Mixer 55 62
Line Mono Mux 56 63
Line Right Mux 57 64 new:'Line/GSM Right Mux'
Line Left Mux 58 65 new:'Line/GSM Left Mux'
ALC Mixer Line Capture Switch 59 66
ALC Mixer Mic2 Capture Switch 60 67 new:s/Mic2/int.Mic/
ALC Mixer Mic1 Capture Switch 61 68 new:s/Mic1/HS-Mic/
ALC Mixer Rx Capture Switch 62 69 new:s/Rx/GSM-PB/
Mic Sidetone Mux 63 70 new:'Mic Path Mux'
Capture Right Mux 64 71
Capture Left Mux 65 72
Capture Right Mixer 66 73
Capture Left Mixer 67 74
Playback Mixer Voice Capture Switch 68 75
Playback Mixer Left Capture Switch 69 76
Playback Mixer Right Capture Switch 70 77
Out4 Mux 71 78
Out3 Mux 72 79
Mono 2 Mux 73 80 new:'GSM Mic(Mono2) Mux'
Mono Mixer Left Playback Switch 74 81 new:s/Mono/GSM Mic/
Mono Mixer Right Playback Switch 75 82 new:s/Mono/GSM Mic/
Mono Mixer Voice Playback Switch 76 83 new:s/Mono/GSM Mic/
Mono Mixer Sidetone Playback Switch 77 84 new:s/Mono/GSM Mic/
Mono Mixer Bypass Playback Switch 78 85 new:s/Mono/GSM Mic/
Right Mixer Voice Playback Switch 79 86
Right Mixer Sidetone Playback Switch 80 87
Right Mixer Right Playback Switch 81 88
Right Mixer Bypass Playback Switch 82 89
Left Mixer Voice Playback Switch 83 90
Left Mixer Sidetone Playback Switch 84 91
Left Mixer Left Playback Switch 85 92
Left Mixer Bypass Playback Switch 86 93
Stereo Out Switch 87 53
GSM Line Out Switch 88 54
GSM Line In Switch 89 55
Headset Mic Switch 90 56
Handset Mic Switch 91 57 new:s/Handset Mic/Int.Mic/
Handset Spk Switch 92 58 new:'DAPM Earpiece Switch'
Amp State Switch 93 absent
Amp Spk Switch 94 59
*) As LOUT1/ROUT1 drives both speaker/sounder and headset, it is
projected to have two dedicated controls for this.
:numid=3,name='Headphone Playback Volume' will be in effect when 
#94 'Amp Spk Switch'=true (speaker mode)
A new control
:numid=95,name='Headphone Playback Volume' will be in effect 
when #94 'Amp Spk Switch'=false (headphone mode)
Two new controls
:numid=96,name='Headphone Playback Switch' 
:numid=97,name='Speaker Playback Switch' 
act as Mute switches:
#96 off->on implies: #97 ->off, #94 ->false
#97 off->on implies: #96 ->off, #94 ->true

#93 and #87 are defined as (#96==on) or (#97==on), i.e. if both 
are muted, amp is shut down. We need to test if we have to keep 
special timings and sequence on enabling/disabling these 
controls #87 and #93, e.g. to avoid clicking sounds

Four variants of using available Digital Audio Interfaces and DACs/ADC, these correspond to the "DAI Mode" ALSA control values:

Wolfson dai routing.png

Keep in mind: left interface (VXxxx) is connected to BlueTooth, right interface (LRC, BCLK, xxxDAT) is connected to SoC (CPU). So mode "11" at least seems isn't useful at all for the way Neo HW is built. Mode "10" is suited for (Stereo/mono) output and recording for digital world, whereas Mode "00" is needed for GSM<->BT operation (calls via BT-headset) only.

BT-VoIP-calls and BT-stereo-headphones playback are done via direct USB-connection SoC<->BT in a very usual standard-linux-way, and therefor need no statefile or any other setup of mixer.

We are still wondering what use Mode "01" might have, other than analog mixed mono output (which could as well be done at digital side by feeding L/R with same data)

  • --MMlosh 10:15, 5 April 2009 (GMT) Mono signal has separate volume control for stereo (control #8) and mono (control #13) output. This might be useful when mixing PCM (VoIP call / music) output into GSM call and playing it locally (and you can still have different volume levels)

Phase0 Quick Start

In my experience this works but I have to fiddle with the connection a bit before I get stereo output. The audio also comes out both the speaker and headphones.

alsactl -f /etc/stereoout.state restore
madplay myfavoritesong.mp3

Another simple test (assuming you have USB Networking configured) is to listen to a radio stream:

wget -O - | madplay -

If for some reason you're missing stereoout.state, try getting a similar copy (a couple of volume levels are different is all)


Voice Calls

using phone-internal microphone and speaker

Actually the diagram below is incorrect (complexity and noise introduced by needless detour for red mic-path, via ACOP ACIN LPGA, should be direct PreAmp MICMUX [control63="Mic 2"] Marked *) below). See

WM8753 BlockDiagram GSM handset.png

This is the default case.

  • microphone path
    • input: built in microphone attached to wolfson MIC2/MIC2N
    • routed from wolfson MIC2/MIC2N to MONO1/MONO2
    • arrives at GSM Modem input MICIP/MICIN
  • speaker path
    • input: GSM Modem attached to wolfson RXN/RXP
    • routed from wolfson RXN/RXP to ROUT1/LOUT1
    • arrives on LM4857 RIN/LIN
    • routed on LM4856 to EP+/EP-

Internal Codec Route

Neo Mode is GSM Handset Amp Mode is Call Speaker

  • audio path Mic -> GSM
    • MIC2/MIC2N
    • Mic2 Volume
    • ALC Micer Mic2 *)
    • Left PGA *)
    • Mic Sidetone Mux [Left PGA *)"Mic 2"]
    • Mono Sidetone Volume
    • Mono Mixer Sidetone Playback Switch
    • Mono Volume
    • Mono 2 Mux [Inverted Mono 1]
  • audio path GSM -> Speaker
    • RXP/RXN
    • Rx Mixer [RXP - RXN]
    • Line Left Mux [Rx Mix]/Line Right Mux [Rx Mix]
    • Left Mixer Bypass Playback Switch/Right Mixer Bypass Playback Switch
    • Headphone Volume

Driver Status

This should be supported by ASoC 0.13rc3 (-moko7 kernel) on.

ASoC 0.13.3 should have same functionality but has renamed the soundcard to neo1973.


For ASoC 0.13.3

using analog (4pin 2.5mm) headset

This is also a quite common case, since we ship the headset with the phone

Headset Detection is done via GPIO on S3C2410

  • microphone path
    • input: headset mic vial HS_MIC attached to wolfson MIC1
    • routed from wolfson MIC1 to MONO1/MONO2
    • arrives at GSM Modem input MICIP/MICIN
  • speaker path
    • input: GSM Modem attached to wolfson RXN/RXP
    • routed from wolfson RXN/RXP to ROUT1/LOUT1
    • arrives on LM4857 RIN/LIN
    • routed on LM4856 to LHP/RHP

Internal Codec Route

Neo Mode is GSM Headset Amp Mode is Headphones

  • audio path Mic -> GSM
    • MIC1
    • Mic Selection Mux [Mic 1]
    • ALC Mixer Mic1
    • Left PGA
    • Mic Sidetone Mux [Left PGA]
    • Mono Sidetone Volume
    • Mono Mixer Sidetone Playback Switch
    • Mono Volume
    • Mono 2 Mux [Inverted Mono 1]
  • Audio path GSM -> Headphones
    • RXP/RXN
    • Rx Mixer [RXP - RXN]
    • Line Left Mux [Rx Mix]/Line Right Mux [Rx Mix]
    • Left Mixer Bypass Playback Switch/Right Mixer Bypass Playback Switch
    • Headphone Volume

Driver Status

Supported in ASoC 0.13.3


using Bluetooth headset with GSM

NOTE none of this works with GTA02. Neo mode has disappeared, and none of the state files are GTA02 compatible.

WM8753 BlockDiagram GSM Bluetooth.png

Headset detection via software

  • microphone path
    • input: from bluetooth via PCM interface to wolfson
    • wolfson: DAC
    • wolfson routes analog signal to MONO1/MONO2
    • arrives at GSM Modem input MICIP/MICIN
  • speaker path
    • input: GSM Modem attached to wolfson RXN/RXP
    • wolfson: ADC
    • wolfson: routes digital signal to PCM
    • arrives on bluetooth chip via PCM

Internal Codec Route

Neo Mode is GSM Bluetooth Amp Mode is Off

  • audio path BT -> GSM
    • Vx DAC
    • Mono Voice Volume
    • Mono Mixer Voice Playback Switch
    • Mono Volume
    • Mono 2 Mux [Inverted Mono 1]
  • audio path GSM -> BT
    • RXP/RXN
    • Rx Mixer [RXP - RXN]
    • ALC Mixer Rx
    • Left PGA
    • Capture Mixer Mux [PGA]
    • Capture Left Mixer [Analogue Mix Left]
    • Left ADC

Driver Status

Should be support by ASoC 0.13.3

Example of how to setup PCM->BT link.


NOTE this will not work with GTA02, as the control numbers have changed Here [1] is a modified version that is GTA02 compatible, But lacks the Neo Mode settings which disappeared on GTA02, and does not seem to work.

The state file above does not work for me. I created one gsm_headset.txt that is working for audio playback on the freerunner. I will be updating that file as I get the mic routing working.

To use this state file there are a number of steps

Turn on bluetooth

Pair the headset ( this only needs to be done once ).

Start the audio subsystem and connect the headset or use my script

If you don't hear static in your headset at this point you may need to reboot.

Start the phone call

alsactl restore 0 -f gsm_headset.txt


I wrote a script to stop the headset too

The above did not work for me; for some reason, the hifi DAC interface must be exercised once before playing. I have hacked to make FR-BTAudio. When paired with GSMBLUETOOTH.txt I get 2-way high quality audio.

I did a lot of this debugging using w8753_dump which is a quick and dirty hack, but quite useful on a large text window.

Bluetooth headset with system audio

For example, using a voip app on the phone with a bt voice headset. This would also be a good way to work on the bluetooth driver without requiring a working GSM and placing a lot of calls.

See ticket 583 for a state file that should route system audio *out* to the headset. The codec does not allow for duplex system audio connected to a headset, so audio in is still using the mic.

NOTE the state file specified does not work for GTA02, and even when modified to be GTA02 compatible still does not route system sound to a BT headset. Modified state file for GTA02 is here [2]


sound playback to speakers

This is an important mode since it is also required for ringtone playback

  • speaker path
    • input: from S3C2410 via IIS interface to wolfson
    • wolfson: DAC
    • wolfson: route to ROUT1/LOUT1
    • LM4857: arrives on RIN/LIN
    • LM4857: route to LLS+-/RLS+-

Driver Status

This is working since ASoC 0.13rc2 (-moko6 kernel)

This should also work on ASoC 0.13.3


For ASoC 0.13.3

sound playback to headphone

  • speaker path
    • input: from S3C2410 via IIS interface to wolfson
    • wolfson: DAC
    • wolfson: route to ROUT1/LOUT1
    • LM4857: arrives on RIN/LIN
    • routed on LM4856 to LHP/RHP

Driver Status

This is working since ASoC 0.13rc2 (-moko6 kernel)

sound playback via A2DP

One way to do this is to use a pulse plugin for bluetooth audio. Pulse would be routed either to the plugin or the default route to the codec. The plugin would watch for headset connect/disconnect events generated by a bluez audio daemon to keep the list of available output devices current.

Driver Status

There is early work on the bluez daemon to handle this. It has been combined with an alsa plugin in the bluez tree but the alsa plugin probably will not be sufficient for neo.

voice recording

This is mainly used to record notes

  • microphone path
    • can be from built-in mic
    • or from headset
    • or bluetooth headset

Driver Status


Call recording

This is a nice wishlist item. The user should be able to receive the full-duplex audio from the wolfson codec, and record it using the S3C2410 IIS.


It is possible to do PCM recording of a GSM voice call. In fact, it is even possible to record the local microphone (what you speak) and the remote voice (what is spoken on the other end of the call) to separate channels (L and R of the Stereo ADC).

If you want to record a GSM voice call, please adjust your mixer settings as follows

  1. Capture Left Mux: Line or RXP-RXN
    • this routes the analog voice from the GSM modem to the left DAC channel
  2. Capture Right Mux: PGA
    • this routes the microphone input to the right DAC channel

FIXME: test this. There's currently a ASoC kernel driver bug preventing audio capture from working at all.

Driver Status



If you want to play PCM audio into a GSM call (i.e. make your remote partner of a voice call hear your PCM audio, e.g. your mp3 or ogg files.

If you are inside a voice call (e.g. FSO/zhone), open amixer or load a state file with alsactl and change the following mixer controls:

  1. Mono Mixer Left
    • this enables audio routing from the Stereo DAC left channel to the Mono Out (GSM Modem)
  2. Mono Mixer Right Playback Switch
    • this enables audio routing from the Stereo DAC right channel to the Mono Out (GSM Modem)
  3. PCM Level
    • adjust the PCM Level up to the desired playback volume

Driver Status


Recording and Playback .state

Here is a state file that allows both recording and playback from and to a gsm call.
File: File:Callrec.txt

To record just issue:

arecord -D hw:0,0 -r 8000 -f S16_LE -c 2 record.wav

and to inject sound just issue:

aplay -D hw:0,0 record.wav

If you have any problems you can contact me on IRC, TAsn.

P.S There's a bug concerning the Right and Left mux, you have to change the left after you change the right, loading the state file may cause this issue to show, so just in case, I recommend appending:

amixer sset 'Capture Right Mux' 'Line or RXP-RXN'

amixer sset 'Capture Left Mux' 'Line or RXP-RXN'

to the alsactl -f Callrec.txt restore command.

Recording and maintaining a call .state

For the actual recording: arecord -D hw:0,0 -r 8000 -f S16_LE -c 2 record.wav

please note that this is the state file I wrote for my Call Recorder, so if you need anything you might miss here, just go and check it's source.

The actual state for gsmhandset (gsmhandset.state): File:Callrec-gsmhandset.txt

A patch (diff gsmhandset.txt callrec-gsmhandset.txt) to apply on every state file, including gsmheadset.state and gsmspeakerout.state. File:Callrec-gsmhandset-patch.txt

Userspace Sound Control Daemon

The userspace sound control deamon might be a separate process or (more likely) part of some larger general hardware management daemon.

It will provide the following features:

audio playback

In order to provide the desired functionality, the daemon first needs to be capable of doing audio playback.

  • supported formats
    • mp3 (libmad)
    • ogg/vorbis (libtremor)
    • mod (mikmod)
    • sid (sidplay)
  • supported functionality
    • start and stop playback
    • interrupt previous sound to play new sound
    • enqueue new sound at end of previous sound
    • smooth fade-in/fade-out

audio event management

  • manage a set of events (basically just a name for each event)
  • manage a set of audio themes
    • each theme contains list of event->audio_file_name mappings
    • themes stored/managed via gconf
  • manage event sources
    • built-in event sources, e.g. touchscreen/button press
    • external event sources (e.g. gsmd, dbus, ...)

audio scenario management

  • e.g. dialer or even gsmd can request audio subsystem to switch to voicecall mode
  • this mainly affects codec/amplifier analog audio routing
  • integrated with bluetooth in case of BT headset or A2DP use
  • How is this management performed currently?

Important issues/pitfalls

Ringtone while headset playback

If the user is listening to music on the headset, we want to mix the ring tones only into the headset audio, as we must not interrupt and play it on the speaker. Reason: headset can't be switched off during playback via speaker, so to avoid extremely loud headset playback there must NOT be any speaker playback while headset is inserted.

In expression: loading speakerout.state is deprecated while JACK_INSERT is asserted.

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