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.

Contents [hide]  1 ALSA Channels 1.1 Diagrams 1.2 List of controls 2 Phase0 Quick Start 3 Voice Calls 3.1 using phone-internal microphone and speaker 3.1.1 Internal Codec Route 3.1.2 Driver Status 3.1.3 asound.state 3.2 using analog (4pin 2.5mm) headset 3.2.1 Internal Codec Route 3.2.2 Driver Status 3.2.3 asound.state 3.3 using Bluetooth headset with GSM 3.3.1 Internal Codec Route 3.3.2 Driver Status 3.3.3 asound.state 4 Bluetooth headset with system audio 5 Multimedia 5.1 sound playback to speakers 5.1.1 Driver Status 5.1.2 asound.state 5.2 sound playback to headphone 5.2.1 Driver Status 5.3 sound playback via A2DP 5.3.1 Driver Status 5.4 voice recording 5.4.1 Driver Status 6 Call recording 6.1 recording 6.1.1 Driver Status 6.2 playback 6.2.1 Driver Status 6.3 Recording and Playback .state 6.4 Recording and maintaining a call .state 7 Userspace Sound Control Daemon 8 audio playback 9 audio event management 10 audio scenario management 11 Important issues/pitfalls 11.1 Ringtone while headset playback

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 <lars@metafoo.de>

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!)

Diagrams

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:

Best of both worlds, Png with the ALSA control names *and* the numid's {in blue}:

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

RFC:

#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:

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 - http://radioparadise.steadyhost.com:8050 | madplay -

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

 wget http://opensource.wolfsonmicro.com/~gg/neo1973/stereoout.state

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 http://people.openmoko.org/joerg/ALSA/doc/WM8753_control_diag_gsmhandset_mic_std.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.

asound.state

https://people.openmoko.org/laforge/gta01/gta01b_v2/alsa/gsmhandset.state

For ASoC 0.13.3 http://opensource.wolfsonmicro.com/~gg/neo1973/gsmhandset.state

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

asound.state

http://opensource.wolfsonmicro.com/~gg/neo1973/gsmheadset.state

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.

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.

http://opensource.wolfsonmicro.com/~gg/bluetooth-pcm/bluetooth_pcm.c

asound.state

http://opensource.wolfsonmicro.com/~gg/neo1973/gsmbluetooth.state

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 http://wiki.bluez.org/wiki/HOWTO/AudioDevices or use my script BtHeadset.py

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

bluetooth_pcm

I wrote a script to stop the headset too BtHeadsetDetach.py

The above did not work for me; for some reason, the hifi DAC interface must be exercised once before playing. I have hacked BtHeadset.py 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]

Multimedia

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

asound.state

https://people.openmoko.org/laforge/gta01/gta01b_v2/alsa/stereoout.state

For ASoC 0.13.3 http://opensource.wolfsonmicro.com/~gg/neo1973/stereoout.state

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

UNKNOWN

http://wiki.openmoko.org/wiki/User:Herp

http://wildsau.enemy.org/~moko/voice-recording.state

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.

recording

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

UNKNOWN

playback

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

UNKNOWN

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.