FreeCalypso > hg > freecalypso-tools
view doc/Audio-mode-config @ 492:ac48ed111d6a
loadtools/scripts/compal.init: updated comments for new understanding
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Fri, 24 May 2019 06:15:02 +0000 |
parents | 0321cd08b19f |
children | 02d92d49c9f8 |
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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 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. 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 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. * 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.