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annotate doc/Audio-mode-config @ 505:7bf0d909c87e

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fc-loadtool flash ID check: change of reset after the check logic This change only affects those flash configurations that have ID checks enabled. The logic for resetting the flash after the ID check has been changed as follows: 1) If the check fails, we return without attempting to reset the flash. 2) If the check is successful, we reset the flash using the configured method (could be AMD or Intel or Intel W30) instead of always doing an AMD flash reset as the original code did.
author Mychaela Falconia <falcon@freecalypso.org>
date Mon, 27 May 2019 19:58:01 +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
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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.