There exist a number of tunable settings in the Iota ABB (the chip that performs
A-to-D and D-to-A conversion for the voice path) and in the Calypso DSP which
in TI's firmware architecture are meant to be configured through the audio mode
facility of the RiViera Audio Service.  The ABB settings grouped under the audio
mode are as follows:

* The selection of which analog interface pins the downlink audio should be
  sent to: EARN&EARP (earpiece), AUXON&AUXOP (auxiliary) or HSO (headset).

* The selection of which analog interface pins the uplink audio should be taken
  from: MICIN&MICIP (main microphone), AUXI (auxiliary input) or HSMICP
  (headset microphone).

* The selection of AUXI input levels when this analog input is in use for the
  voice uplink.

* Analog gains for the uplink, the downlink and the analog sidetone from the
  uplink input to the downlink output.

* Selection of a special filter bypass mode for the voice downlink.

* The selection of MICBIAS (or HSMICBIAS) voltage between 2.0 V and 2.5 V.

The DSP voice path settings grouped under the audio mode are as follows:

* The selection of the digital voice path as being between GSM and the ABB (the
  default for analog voice interfaces), between GSM and MCSI (the external
  digital voice interface) or between MCSI and the ABB (non-GSM operation).

* FIR filter coefficients for the voice uplink and for the voice downlink.

* Enabling/disabling and configuration of the Acoustic Echo Cancellation (AEC)
  mechanism.

The firmware paradigm for working with all of the above settings is as follows:

* In a lab environment, each of the listed settings can be independently tweaked
  and read back through ETM packets over the RVTMUX debug serial interface; the
  corresponding fc-tmsh commands (matching TI's original Windows-based TMSH)
  are auw for writing individual audio parameters and aur for reading them back.

* In end-use operation, TI's intent as realized in the firmware design is that
  all of the listed audio settings will only be changed as a group, loaded from
  audio mode configuration files in FFS.

Each audio mode configuration needs to be assigned a name between 1 and 9
characters long, and for each named configuration there are two files in FFS:

/aud/modename.cfg is the main configuration file
/aud/modename.vol is the corresponding volume setting file

This paradigm is a good fit for "dumbphone" handsets in which there usually
will be several different voice audio configurations for classic handheld
operation, for the hands-free loudspeaker mode, for operation with a wired
headset, and if the phone uses its hands-free loudspeaker plus the Calypso DSP
to play ringtones (as opposed to using a buzzer on BU/PWT or a ringtone player
chip that drives the speaker bypassing the voice path), there will also need to
be an output-only audio configuration for ringing.

How do the audio mode config files under /aud come into being?  It appears that
TI's original intent was that a configuration would be manually constructed on
a test device via TMSH auw commands, saved in the FFS of that test device with
the aus command, then read out of that test device FFS in binary form and
reuploaded as an opaque blob to all devices on the production line.  One can do
the same procedure with our fc-tmsh and fc-fsio which fully replicate the
relevant functionality of TI's original TMSH (to the best of our knowledge),
but in FreeCalypso we have an alternate way which fits better with our UNIX
philosophy: we have created our own ASCII text format for representing all of
the content in TI's /aud/*.cfg binary files and tiaud-* utilities for compiling
TI's binary cfg files from our ASCII source format, disassembling a *.cfg file
read out of FFS into the same ASCII format, and creating the required *.vol
companion files, which are also binary.

A note about volume settings: the Iota ABB has two variable gain controls in
the voice downlink path: the main "volume" gain in rather coarse 6 dB steps
(the choices being 0 dB, -6 dB, -12 dB, -18 dB, -24 dB and mute) and a finer
"calibration" gain in 1 dB steps between -6 and +6 dB.  It appears that TI's
intent was that only the coarse volume control in 6 dB steps is to be visible
to the user, with just 5 possible non-mute volume levels, and that the finer
gain control be set at the factory in the audio mode config files for each mode
as some form of calibration.  Pirelli DP-L10 significantly deviates from this
model by providing 10 non-mute volume levels to the user with 2 dB or 3 dB steps
between them by changing both VOLCTL and VDLPG fields in the VBDCTRL register,
but at the present time we have no plans to make a similar drastic change in
FreeCalypso.

Another noteworthy feature of the audio mode system with respect to volume
control is that there is a separate *.vol file that stores the current volume
setting for each mode.  In a "dumbphone" handset firmware built according to
TI's paradigm, the /aud/*.cfg files will be written once on the factory
production line and only read afterward, but whenever the user turns the volume
up or down in the UI, the *.vol file _corresponding to the current mode_ will
be updated by the running fw.  Thus the fw would maintain a separate notion of
the current volume for ringing, for the earpiece speaker, for the hands-free
loudspeaker and for the wired headset, something which Pirelli's fw very
notoriously fails to do.

Old vs. new AEC
===============

One of the settings in the audio mode config structure underwent an evolutionary
change within the span of history that is relevant to FreeCalypso - this setting
is the configuration for AEC, the Acoustic Echo Cancellation functional block
of the Calypso DSP.  As TI's GSM DSPs evolved (before, during and after the
Calypso era), their AEC implementation evolved along with the rest, and
different evolutionary versions of AEC require different configuration and
tuning parameters.  When the audio mode facility was first implemented, the AEC
block in TI's GSM DSPs of that time was controlled with a single 16-bit control
word; the people in the SSA group who implemented RiViera Audio Service then
decided to split different bits from this one DSP control word into 5 different
parameter words, and the result was the "old" 5-word AEC config.

But the version of AEC implemented in the DSP ROM in the Calypso silicon version
we work with is slightly newer; this version corresponds to what TI's L1 code
calls L1_NEW_AEC.  However, the waters then got muddied: for reasons which we
(FreeCalypso team) cannot understand (perhaps miscommunication between different
groups at TI), TI's TCS211 reference firmware shipped with L1_NEW_AEC disabled
(C preprocessor symbol set to 0 instead of 1), even though the underlying DSP
AEC block (combination of ROM and official patches) is the "new" kind and not
the "old" one.  There are two fallouts from this software misconfiguration on
TI's part:

1) If one takes stock TCS211 from TI or any derivative version in which this
   aspect is unchanged (all mokoN firmwares, and all FC firmwares up to
   Magnetite) and tries to enable AEC, the result will be a poor AEC
   configuration: the old echo level and long vs short settings do nothing on
   the new DSP, whereas the new tunable parameters will remain at their defaults
   with no way to tweak them.  I (Mother Mychaela) can only guess that this
   situation is what Openmoko must have run into when they tried to get AEC
   working.

2) When someone downstream of TI figures out that L1_NEW_AEC needs to be changed
   from 0 to 1 and actually makes that change, like we did in our Tourmaline fw,
   the format and size of the audio mode binary structure change, and all old
   audio mode config files become invalid.

Our FreeCalypso work is affected by point 2 above: we started working with audio
mode config files in 2017, using the old AEC configuration, and only made the
switch to L1_NEW_AEC in 2021.  We now have two kinds of audio mode config binary
files: the old kind that are 164 bytes long, and the new kind that are 176 bytes
long.  Our Tourmaline firmware has L1_NEW_AEC enabled, while Magnetite (our
legacy backward compatiblity fw) has it disabled.

To prevent loading of garbage into AEC config when an audio mode file of the
wrong kind is loaded, we have implemented the following workaround in both
Tourmaline and Magnetite: if the loaded mode config file has the wrong length,
the AEC config is set to the default disabled state instead of whatever is in
the mode file - loading an AEC config of the wrong format is not possible.

Default audio configuration
===========================

The default audio config set in the Iota ABB registers and in the DSP when no
named audio mode config has been loaded with the audio_mode_load() API call
(accessible via AT@AUL or via fc-tmsh aul command) is as follows, in the syntax
which our tiaud-compile utility accepts as input and which our tiaud-decomp
utility emits as output:

voice-path 0
mic default {
	gain 3
	output-bias 0
	fir  0 0x4000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
	fir  8 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
	fir 16 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
	fir 24 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
}
speaker ear+aux {
	gain 0
	audio-filter 0
	fir  0 0x4000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
	fir  8 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
	fir 16 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
	fir 24 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
}
sidetone -5
aec 0 0 0 0 0

The above version is the one produced by Magnetite and earlier firmwares without
L1_NEW_AEC; in the new Tourmaline version the last line changes to:

aec-new 0 0 0x1 0x7FFF 0x1FFF 0x4000 0x32 0x1000 0x1000 0 0 0

The meaning is as follows:

* voice-path is the DSP digital voice path setting, 0 means the standard
  configuration with the voice channel going between GSM and the local analog
  voice hardware attached to the ABB.

* The default microphone input is used for the voice uplink (MICIN&MICIP pins),
  whereas the voice downlink is presented on both EARN&EARP and AUXON&AUXOP
  pins, i.e., both "ear" and "aux" VDL amplifiers are enabled.

* The microphone gain is 3 dB, the fine gain adjustment in the voice downlink
  path is 0 dB, and the sidetone gain is -5 dB.

* output-bias 0 under mic means that the MICBIAS voltage is set to 2.0 V.

* audio-filter 0 under speaker means that the VFBYP bit in the VBCTRL1 register
  is NOT set, i.e., the normal configuration.

* DSP FIR filters do nothing, as coefficient 0 is set to unity and all other
  coefficients are set to zero.

* The AEC mechanism in the DSP is disabled, although the format of the bits that
  say so is different between old and new AEC versions.  In the new version
  there are a number of tunable settings that only kick in when AEC is enabled;
  when AEC is disabled by default, these tunable knobs still have sensible
  defaults that aren't all zeros.

Creating your own audio mode configurations
===========================================

The input to our tiaud-compile utility can contain every setting shown in the
default case above, or any desired subset thereof.  For any settings not given
in the input, the defaults from the above will be used, except that
tiaud-compile's current default for the speaker mode is just ear rather than
ear+aux.  (It is a default which you should NOT depend on; set it explicitly if
it matters!)  A few notes:

* For all settings given as numbers, the number given in the ASCII input is the
  number that goes into TI's binary structure, without any transformation, even
  in those cases where the result is counter-intuitive, such as "audio-filter 0"
  meaning that the filter is *enabled*.

* The 3 possible mode keywords for the mic mode are default, aux and headset,
  corresponding to MICIN&MICIP, AUXI and HSMICP analog inputs, respectively.

* The 5 possible mode keywords for the speaker mode are ear, aux, headset,
  buzzer and ear+aux.  The buzzer speaker mode exists only on TI's Nausica ABB
  predating Iota, i.e., it won't work on any of the Calypso+Iota+Rita devices
  built or supported by FreeCalypso, but our tiaud-compile and tiaud-decomp
  utilities support it because it is nominally supported by TI's RiViera Audio
  Service and its binary data structure for audio mode configuration.

* When mic is set to aux, an additional mic setting called extra-gain becomes
  available.  If extra-gain is set to 0, the AUXI gain will be set to 28.2 dB,
  if extra-gain is set to 1, the AUXI gain will be set to 4.6 dB; all other
  values will be considered invalid by the firmware.

* Each of the two FIR filters in the DSP (one for uplink, one for downlink) has
  a total of 31 coefficients, numbered 0 through 30, inclusive.  In the ASCII
  input to tiaud-compile you can put each coefficient on its own fir line, put
  all 31 coefficients on the same line, or group them in any other way you like.
  The grouping used in the tiaud-decomp output has been chosen for line length
  reasons.

aec vs aec-new in tiaud-compile input
=====================================

tiaud-compile accepts both aec (old) and aec-new settings; aec must be followed
by 5 numbers, aec-new must be followed by 12 numbers.  Each number is a 16-bit
value, and they go into the binary structure without further interpretation by
tiaud-compile - instead the firmware is the entity that gives them meaning.
Numbers without 0x prefix are interpreted as decimal.

tiaud-compile will generate one type or the other of the binary output file,
following these rules:

* If an aec setting is given, a 164 byte file will be produced, with the 5 AEC
  words being the given ones.

* If an aec-new setting is given, a 176 byte file will be produced, with the 12
  AEC words being the given ones.

* If neither setting is given, a 164 byte file will be produced, with the 5 AEC
  words of the old type being all zeros.  Thanks to the modified audio mode
  loading code in our firmwares, these 164 byte mode files can still be used
  with current Tourmaline fw, with AEC set to its default disabled state.