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+++ b/doc/Audio-mode-config	Sat Aug 26 04:50:25 2017 +0000
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+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 a loudspeaker (as opposed to a piezo buzzer) to
+play ringtones and uses the Calypso DSP to generate those ringtone melodies,
+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
+revelant 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.
+
+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 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.
+
+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 0, 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.
+
+* The numbers given on fir and aec lines are 16-bit values that go directly into
+  the DSP; the former are FIR coefficients and the latter are bit masks.  They
+  can be given as either decimal or hexadecimal with 0x prefix in the ASCII
+  input to tiaud-compile.