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annotate doc/Audio-mode-config @ 421:ab8fb95a28da

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rvinterf/rvtat converted to use libinterf
author Mychaela Falconia <falcon@freecalypso.org>
date Sat, 27 Oct 2018 23:48:20 +0000
parents 0321cd08b19f
children 02d92d49c9f8
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1 There exist a number of tunable settings in the Iota ABB (the chip that performs
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2 A-to-D and D-to-A conversion for the voice path) and in the Calypso DSP which
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3 in TI's firmware architecture are meant to be configured through the audio mode
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4 facility of the RiViera Audio Service. The ABB settings grouped under the audio
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5 mode are as follows:
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6
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7 * The selection of which analog interface pins the downlink audio should be
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8 sent to: EARN&EARP (earpiece), AUXON&AUXOP (auxiliary) or HSO (headset).
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9
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10 * The selection of which analog interface pins the uplink audio should be taken
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11 from: MICIN&MICIP (main microphone), AUXI (auxiliary input) or HSMICP
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12 (headset microphone).
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13
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14 * The selection of AUXI input levels when this analog input is in use for the
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15 voice uplink.
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16
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17 * Analog gains for the uplink, the downlink and the analog sidetone from the
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18 uplink input to the downlink output.
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19
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20 * Selection of a special filter bypass mode for the voice downlink.
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21
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22 * The selection of MICBIAS (or HSMICBIAS) voltage between 2.0 V and 2.5 V.
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23
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24 The DSP voice path settings grouped under the audio mode are as follows:
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25
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26 * The selection of the digital voice path as being between GSM and the ABB (the
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27 default for analog voice interfaces), between GSM and MCSI (the external
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28 digital voice interface) or between MCSI and the ABB (non-GSM operation).
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29
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30 * FIR filter coefficients for the voice uplink and for the voice downlink.
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31
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32 * Enabling/disabling and configuration of the Acoustic Echo Cancellation (AEC)
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33 mechanism.
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34
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35 The firmware paradigm for working with all of the above settings is as follows:
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36
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37 * In a lab environment, each of the listed settings can be independently tweaked
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38 and read back through ETM packets over the RVTMUX debug serial interface; the
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39 corresponding fc-tmsh commands (matching TI's original Windows-based TMSH)
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40 are auw for writing individual audio parameters and aur for reading them back.
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41
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42 * In end-use operation, TI's intent as realized in the firmware design is that
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43 all of the listed audio settings will only be changed as a group, loaded from
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44 audio mode configuration files in FFS.
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45
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46 Each audio mode configuration needs to be assigned a name between 1 and 9
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47 characters long, and for each named configuration there are two files in FFS:
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48
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49 /aud/modename.cfg is the main configuration file
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50 /aud/modename.vol is the corresponding volume setting file
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51
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52 This paradigm is a good fit for "dumbphone" handsets in which there usually
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53 will be several different voice audio configurations for classic handheld
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54 operation, for the hands-free loudspeaker mode, for operation with a wired
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55 headset, and if the phone uses a loudspeaker (as opposed to a piezo buzzer) to
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56 play ringtones and uses the Calypso DSP to generate those ringtone melodies,
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57 there will also need to be an output-only audio configuration for ringing.
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58
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59 How do the audio mode config files under /aud come into being? It appears that
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60 TI's original intent was that a configuration would be manually constructed on
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61 a test device via TMSH auw commands, saved in the FFS of that test device with
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62 the aus command, then read out of that test device FFS in binary form and
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63 reuploaded as an opaque blob to all devices on the production line. One can do
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64 the same procedure with our fc-tmsh and fc-fsio which fully replicate the
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65 relevant functionality of TI's original TMSH (to the best of our knowledge),
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66 but in FreeCalypso we have an alternate way which fits better with our UNIX
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67 philosophy: we have created our own ASCII text format for representing all of
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68 the content in TI's /aud/*.cfg binary files and tiaud-* utilities for compiling
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69 TI's binary cfg files from our ASCII source format, disassembling a *.cfg file
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70 read out of FFS into the same ASCII format, and creating the required *.vol
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71 companion files, which are also binary.
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72
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73 A note about volume settings: the Iota ABB has two variable gain controls in
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74 the voice downlink path: the main "volume" gain in rather coarse 6 dB steps
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75 (the choices being 0 dB, -6 dB, -12 dB, -18 dB, -24 dB and mute) and a finer
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76 "calibration" gain in 1 dB steps between -6 and +6 dB. It appears that TI's
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77 intent was that only the coarse volume control in 6 dB steps is to be visible
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78 to the user, with just 5 possible non-mute volume levels, and that the finer
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79 gain control be set at the factory in the audio mode config files for each mode
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80 as some form of calibration. Pirelli DP-L10 significantly deviates from this
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81 model by providing 10 non-mute volume levels to the user with 2 dB or 3 dB steps
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82 between them by changing both VOLCTL and VDLPG fields in the VBDCTRL register,
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83 but at the present time we have no plans to make a similar drastic change in
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84 FreeCalypso.
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85
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86 Another noteworthy feature of the audio mode system with respect to volume
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87 control is that there is a separate *.vol file that stores the current volume
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88 setting for each mode. In a "dumbphone" handset firmware built according to
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89 TI's paradigm, the /aud/*.cfg files will be written once on the factory
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90 production line and only read afterward, but whenever the user turns the volume
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91 up or down in the UI, the *.vol file _corresponding to the current mode_ will
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92 be updated by the running fw. Thus the fw would maintain a separate notion of
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93 the current volume for ringing, for the earpiece speaker, for the hands-free
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94 loudspeaker and for the wired headset, something which Pirelli's fw very
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95 notoriously fails to do.
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96
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97 Default audio configuration
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98 ===========================
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99
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100 The default audio config set in the Iota ABB registers and in the DSP when no
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101 named audio mode config has been loaded with the audio_mode_load() API call
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102 (accessible via AT@AUL or via fc-tmsh aul command) is as follows, in the syntax
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103 which our tiaud-compile utility accepts as input and which our tiaud-decomp
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104 utility emits as output:
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105
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106 voice-path 0
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107 mic default {
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108 gain 3
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109 output-bias 0
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110 fir 0 0x4000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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111 fir 8 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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112 fir 16 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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113 fir 24 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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114 }
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115 speaker ear+aux {
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116 gain 0
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117 audio-filter 0
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118 fir 0 0x4000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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119 fir 8 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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120 fir 16 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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121 fir 24 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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122 }
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123 sidetone -5
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124 aec 0 0 0 0 0
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125
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126 The meaning is as follows:
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127
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128 * voice-path is the DSP digital voice path setting, 0 means the standard
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129 configuration with the voice channel going between GSM and the local analog
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130 voice hardware attached to the ABB.
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131
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132 * The default microphone input is used for the voice uplink (MICIN&MICIP pins),
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133 whereas the voice downlink is presented on both EARN&EARP and AUXON&AUXOP
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134 pins, i.e., both "ear" and "aux" VDL amplifiers are enabled.
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135
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136 * The microphone gain is 3 dB, the fine gain adjustment in the voice downlink
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137 path is 0 dB, and the sidetone gain is -5 dB.
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138
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139 * output-bias 0 under mic means that the MICBIAS voltage is set to 2.0 V.
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140
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141 * audio-filter 0 under speaker means that the VFBYP bit in the VBCTRL1 register
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142 is NOT set, i.e., the normal configuration.
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143
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144 * DSP FIR filters do nothing, as coefficient 0 is set to unity and all other
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145 coefficients are set to zero.
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146
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147 * The AEC mechanism in the DSP is disabled.
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148
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149 Creating your own audio mode configurations
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150 ===========================================
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151
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152 The input to our tiaud-compile utility can contain every setting shown in the
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153 default case above, or any desired subset thereof. For any settings not given
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154 in the input, the defaults from the above will be used, except that
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155 tiaud-compile's current default for the speaker mode is just ear rather than
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156 ear+aux. (It is a default which you should NOT depend on; set it explicitly if
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157 it matters!) A few notes:
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158
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159 * For all settings given as numbers, the number given in the ASCII input is the
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160 number that goes into TI's binary structure, without any transformation, even
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161 in those cases where the result is counter-intuitive, such as "audio-filter 0"
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162 meaning that the filter is *enabled*.
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163
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164 * The 3 possible mode keywords for the mic mode are default, aux and headset,
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165 corresponding to MICIN&MICIP, AUXI and HSMICP analog inputs, respectively.
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166
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167 * The 5 possible mode keywords for the speaker mode are ear, aux, headset,
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168 buzzer and ear+aux. The buzzer speaker mode exists only on TI's Nausica ABB
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169 predating Iota, i.e., it won't work on any of the Calypso+Iota+Rita devices
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170 built or supported by FreeCalypso, but our tiaud-compile and tiaud-decomp
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171 utilities support it because it is nominally supported by TI's RiViera Audio
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172 Service and its binary data structure for audio mode configuration.
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173
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174 * When mic is set to aux, an additional mic setting called extra-gain becomes
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175 available. If extra-gain is set to 0, the AUXI gain will be set to 28.2 dB,
247
b5b148ef63da doc/Audio-mode-config: fixed logic error in the description of extra-gain
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176 if extra-gain is set to 1, the AUXI gain will be set to 4.6 dB; all other
245
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177 values will be considered invalid by the firmware.
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178
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179 * Each of the two FIR filters in the DSP (one for uplink, one for downlink) has
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180 a total of 31 coefficients, numbered 0 through 30, inclusive. In the ASCII
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181 input to tiaud-compile you can put each coefficient on its own fir line, put
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182 all 31 coefficients on the same line, or group them in any other way you like.
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183 The grouping used in the tiaud-decomp output has been chosen for line length
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184 reasons.
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185
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186 * The numbers given on fir and aec lines are 16-bit values that go directly into
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187 the DSP; the former are FIR coefficients and the latter are bit masks. They
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188 can be given as either decimal or hexadecimal with 0x prefix in the ASCII
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189 input to tiaud-compile.