Neo 1973 audio subsystem
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Phase0 Quick Start
In my experience this works but what comes out is not stereo. The wolfson driver has to be loaded late at boot time so it works to put it in /etc/modutils/zound :)
wget http://opensource.wolfsonmicro.com/~gg/neo1973/stereoout.state wget http://www-public.tu-bs.de:8080/~y0019680/tmp/thereisnophone.mp3 alsactl -f stereoout.state restore madplay thereisnophone.mp3
Voice Calls
using phone-internal microphone and speaker
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]
- 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
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
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
The current proposal is for bluez to provide a libalsa plugin which either sends audio to the real audio device or encodes and transmits it over bluetooth. The plugin would watch for headset connect/disconnect events generated by a bluez audio daemon.
Driver Status
UNKNOWN
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
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
FIXME
Driver Status
UNKNOWN
playback
FIXME
Driver Status
UNKNOWN
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, do we want to mix the ring tones only into the headset audio, or actually interrupt and play it on the speaker?
More details with using a bluetooth headset
There doesn't seem to be a capturebluetooth.state file so one could listen to system audio, for example internet radio, on a bluetooth headset.
The bluetooth_pcm.c example above for using a bluetooth headset does not seem to change the codec mode.