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annotate doc/Audio-mode-config @ 839:b053e4141a16

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CHANGES: refer to doc/Audio-mode-config for AEC support changes
author Mychaela Falconia <falcon@freecalypso.org>
date Fri, 30 Jul 2021 05:42:21 +0000
parents 02d92d49c9f8
children 6a0fcbca8ac7
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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 its hands-free loudspeaker plus the Calypso DSP
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56 to play ringtones (as opposed to using a buzzer on BU/PWT or a ringtone player
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57 chip that drives the speaker bypassing the voice path), there will also need to
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58 be an output-only audio configuration for ringing.
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59
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60 How do the audio mode config files under /aud come into being? It appears that
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61 TI's original intent was that a configuration would be manually constructed on
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62 a test device via TMSH auw commands, saved in the FFS of that test device with
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63 the aus command, then read out of that test device FFS in binary form and
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64 reuploaded as an opaque blob to all devices on the production line. One can do
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65 the same procedure with our fc-tmsh and fc-fsio which fully replicate the
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66 relevant functionality of TI's original TMSH (to the best of our knowledge),
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67 but in FreeCalypso we have an alternate way which fits better with our UNIX
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68 philosophy: we have created our own ASCII text format for representing all of
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69 the content in TI's /aud/*.cfg binary files and tiaud-* utilities for compiling
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70 TI's binary cfg files from our ASCII source format, disassembling a *.cfg file
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71 read out of FFS into the same ASCII format, and creating the required *.vol
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72 companion files, which are also binary.
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73
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74 A note about volume settings: the Iota ABB has two variable gain controls in
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75 the voice downlink path: the main "volume" gain in rather coarse 6 dB steps
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76 (the choices being 0 dB, -6 dB, -12 dB, -18 dB, -24 dB and mute) and a finer
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77 "calibration" gain in 1 dB steps between -6 and +6 dB. It appears that TI's
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78 intent was that only the coarse volume control in 6 dB steps is to be visible
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79 to the user, with just 5 possible non-mute volume levels, and that the finer
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80 gain control be set at the factory in the audio mode config files for each mode
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81 as some form of calibration. Pirelli DP-L10 significantly deviates from this
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82 model by providing 10 non-mute volume levels to the user with 2 dB or 3 dB steps
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83 between them by changing both VOLCTL and VDLPG fields in the VBDCTRL register,
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84 but at the present time we have no plans to make a similar drastic change in
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85 FreeCalypso.
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86
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87 Another noteworthy feature of the audio mode system with respect to volume
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88 control is that there is a separate *.vol file that stores the current volume
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89 setting for each mode. In a "dumbphone" handset firmware built according to
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90 TI's paradigm, the /aud/*.cfg files will be written once on the factory
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91 production line and only read afterward, but whenever the user turns the volume
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92 up or down in the UI, the *.vol file _corresponding to the current mode_ will
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93 be updated by the running fw. Thus the fw would maintain a separate notion of
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94 the current volume for ringing, for the earpiece speaker, for the hands-free
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95 loudspeaker and for the wired headset, something which Pirelli's fw very
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96 notoriously fails to do.
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97
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98 Old vs. new AEC
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99 ===============
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100
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101 One of the settings in the audio mode config structure underwent an evolutionary
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102 change within the span of history that is relevant to FreeCalypso - this setting
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103 is the configuration for AEC, the Acoustic Echo Cancellation functional block
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104 of the Calypso DSP. As TI's GSM DSPs evolved (before, during and after the
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105 Calypso era), their AEC implementation evolved along with the rest, and
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106 different evolutionary versions of AEC require different configuration and
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107 tuning parameters. When the audio mode facility was first implemented, the AEC
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108 block in TI's GSM DSPs of that time was controlled with a single 16-bit control
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109 word; the people in the SSA group who implemented RiViera Audio Service then
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110 decided to split different bits from this one DSP control word into 5 different
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111 parameter words, and the result was the "old" 5-word AEC config.
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112
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113 But the version of AEC implemented in the DSP ROM in the Calypso silicon version
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114 we work with is slightly newer; this version corresponds to what TI's L1 code
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115 calls L1_NEW_AEC. However, the waters then got muddied: for reasons which we
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116 (FreeCalypso team) cannot understand (perhaps miscommunication between different
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117 groups at TI), TI's TCS211 reference firmware shipped with L1_NEW_AEC disabled
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118 (C preprocessor symbol set to 0 instead of 1), even though the underlying DSP
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119 AEC block (combination of ROM and official patches) is the "new" kind and not
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120 the "old" one. There are two fallouts from this software misconfiguration on
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121 TI's part:
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122
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123 1) If one takes stock TCS211 from TI or any derivative version in which this
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124 aspect is unchanged (all mokoN firmwares, and all FC firmwares up to
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125 Magnetite) and tries to enable AEC, the result will be a poor AEC
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126 configuration: the old echo level and long vs short settings do nothing on
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127 the new DSP, whereas the new tunable parameters will remain at their defaults
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128 with no way to tweak them. I (Mother Mychaela) can only guess that this
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129 situation is what Openmoko must have run into when they tried to get AEC
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130 working.
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131
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132 2) When someone downstream of TI figures out that L1_NEW_AEC needs to be changed
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133 from 0 to 1 and actually makes that change, like we did in our Tourmaline fw,
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134 the format and size of the audio mode binary structure change, and all old
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135 audio mode config files become invalid.
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136
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137 Our FreeCalypso work is affected by point 2 above: we started working with audio
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138 mode config files in 2017, using the old AEC configuration, and only made the
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139 switch to L1_NEW_AEC in 2021. We now have two kinds of audio mode config binary
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140 files: the old kind that are 164 bytes long, and the new kind that are 176 bytes
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141 long. Our Tourmaline firmware has L1_NEW_AEC enabled, while Magnetite (our
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142 legacy backward compatiblity fw) has it disabled.
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143
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144 To prevent loading of garbage into AEC config when an audio mode file of the
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145 wrong kind is loaded, we have implemented the following workaround in both
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146 Tourmaline and Magnetite: if the loaded mode config file has the wrong length,
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147 the AEC config is set to the default disabled state instead of whatever is in
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148 the mode file - loading an AEC config of the wrong format is not possible.
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149
245
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150 Default audio configuration
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151 ===========================
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152
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153 The default audio config set in the Iota ABB registers and in the DSP when no
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154 named audio mode config has been loaded with the audio_mode_load() API call
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155 (accessible via AT@AUL or via fc-tmsh aul command) is as follows, in the syntax
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156 which our tiaud-compile utility accepts as input and which our tiaud-decomp
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157 utility emits as output:
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158
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159 voice-path 0
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160 mic default {
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161 gain 3
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162 output-bias 0
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163 fir 0 0x4000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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164 fir 8 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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165 fir 16 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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166 fir 24 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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167 }
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168 speaker ear+aux {
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169 gain 0
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170 audio-filter 0
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171 fir 0 0x4000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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172 fir 8 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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173 fir 16 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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174 fir 24 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000
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175 }
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176 sidetone -5
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177 aec 0 0 0 0 0
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178
838
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179 The above version is the one produced by Magnetite and earlier firmwares without
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180 L1_NEW_AEC; in the new Tourmaline version the last line changes to:
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181
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182 aec-new 0 0 0x1 0x7FFF 0x1FFF 0x4000 0x32 0x1000 0x1000 0 0 0
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183
245
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184 The meaning is as follows:
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185
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186 * voice-path is the DSP digital voice path setting, 0 means the standard
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187 configuration with the voice channel going between GSM and the local analog
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188 voice hardware attached to the ABB.
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189
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190 * The default microphone input is used for the voice uplink (MICIN&MICIP pins),
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191 whereas the voice downlink is presented on both EARN&EARP and AUXON&AUXOP
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192 pins, i.e., both "ear" and "aux" VDL amplifiers are enabled.
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193
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194 * The microphone gain is 3 dB, the fine gain adjustment in the voice downlink
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195 path is 0 dB, and the sidetone gain is -5 dB.
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196
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197 * output-bias 0 under mic means that the MICBIAS voltage is set to 2.0 V.
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198
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199 * audio-filter 0 under speaker means that the VFBYP bit in the VBCTRL1 register
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200 is NOT set, i.e., the normal configuration.
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201
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202 * DSP FIR filters do nothing, as coefficient 0 is set to unity and all other
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203 coefficients are set to zero.
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204
838
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205 * The AEC mechanism in the DSP is disabled, although the format of the bits that
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206 say so is different between old and new AEC versions. In the new version
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207 there are a number of tunable settings that only kick in when AEC is enabled;
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208 when AEC is disabled by default, these tunable knobs still have sensible
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209 defaults that aren't all zeros.
245
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210
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211 Creating your own audio mode configurations
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212 ===========================================
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213
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214 The input to our tiaud-compile utility can contain every setting shown in the
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215 default case above, or any desired subset thereof. For any settings not given
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216 in the input, the defaults from the above will be used, except that
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217 tiaud-compile's current default for the speaker mode is just ear rather than
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218 ear+aux. (It is a default which you should NOT depend on; set it explicitly if
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219 it matters!) A few notes:
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220
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221 * For all settings given as numbers, the number given in the ASCII input is the
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222 number that goes into TI's binary structure, without any transformation, even
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223 in those cases where the result is counter-intuitive, such as "audio-filter 0"
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224 meaning that the filter is *enabled*.
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225
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226 * The 3 possible mode keywords for the mic mode are default, aux and headset,
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227 corresponding to MICIN&MICIP, AUXI and HSMICP analog inputs, respectively.
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228
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229 * The 5 possible mode keywords for the speaker mode are ear, aux, headset,
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230 buzzer and ear+aux. The buzzer speaker mode exists only on TI's Nausica ABB
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231 predating Iota, i.e., it won't work on any of the Calypso+Iota+Rita devices
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232 built or supported by FreeCalypso, but our tiaud-compile and tiaud-decomp
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233 utilities support it because it is nominally supported by TI's RiViera Audio
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234 Service and its binary data structure for audio mode configuration.
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235
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236 * When mic is set to aux, an additional mic setting called extra-gain becomes
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237 available. If extra-gain is set to 0, the AUXI gain will be set to 28.2 dB,
247
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238 if extra-gain is set to 1, the AUXI gain will be set to 4.6 dB; all other
245
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239 values will be considered invalid by the firmware.
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240
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241 * Each of the two FIR filters in the DSP (one for uplink, one for downlink) has
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242 a total of 31 coefficients, numbered 0 through 30, inclusive. In the ASCII
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243 input to tiaud-compile you can put each coefficient on its own fir line, put
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244 all 31 coefficients on the same line, or group them in any other way you like.
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245 The grouping used in the tiaud-decomp output has been chosen for line length
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246 reasons.
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247
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248 aec vs aec-new in tiaud-compile input
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249 =====================================
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250
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251 tiaud-compile accepts both aec (old) and aec-new settings; aec must be followed
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252 by 5 numbers, aec-new must be followed by 12 numbers. Each number is a 16-bit
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253 value, and they go into the binary structure without further interpretation by
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254 tiaud-compile - instead the firmware is the entity that gives them meaning.
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255 Numbers without 0x prefix are interpreted as decimal.
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256
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257 tiaud-compile will generate one type or the other of the binary output file,
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258 following these rules:
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parents: 261
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259
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260 * If an aec setting is given, a 164 byte file will be produced, with the 5 AEC
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261 words being the given ones.
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parents: 261
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262
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263 * If an aec-new setting is given, a 176 byte file will be produced, with the 12
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264 AEC words being the given ones.
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parents: 261
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265
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266 * If neither setting is given, a 164 byte file will be produced, with the 5 AEC
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parents: 261
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267 words of the old type being all zeros. Thanks to the modified audio mode
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parents: 261
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268 loading code in our firmwares, these 164 byte mode files can still be used
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parents: 261
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269 with current Tourmaline fw, with AEC set to its default disabled state.