FreeCalypso > hg > fc-magnetite
annotate src/cs/layer1/audio_cust0/l1audio_cust.h @ 624:012028896cfb
FFS dev.c, Leonardo target: Fujitsu MB84VF5F5F4J2 #if 0'ed out
The FFS code we got from TI/Openmoko had a stanza for "Fujitsu MB84VF5F5F4J2
stacked device", using a fake device ID code that would need to be patched
manually into cfgffs.c (suppressing and overriding autodetection) and using
an FFS base address in the nCS2 bank, indicating that this FFS config was
probably meant for the MCP version of Leonardo which allows for 16 MiB flash
with a second bank on nCS2.
We previously had this FFS config stanza conditionalized under
CONFIG_TARGET_LEONARDO because the base address contained therein is invalid
for other targets, but now that we actually have a Leonardo build target in
FC Magnetite, I realize that the better approach is to #if 0 out this stanza
altogether: it is already non-functional because it uses a fake device ID
code, thus it is does not add support for more Leonardo board variants,
instead it is just noise.
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Sun, 22 Dec 2019 21:24:29 +0000 |
parents | 945cf7f506b2 |
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rev | line source |
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1 /************* Revision Controle System Header ************* |
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2 * GSM Layer 1 software |
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3 * L1AUDIO_CUST.H |
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4 * |
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5 * Filename l1audio_cust.h |
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6 * Copyright 2003 (C) Texas Instruments |
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7 * |
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8 ************* Revision Controle System Header *************/ |
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9 |
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10 #if (AUDIO_TASK == 1) |
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11 |
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12 #if (OP_RIV_AUDIO == 0) |
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13 extern void vocoder_mute_dl (BOOL mute); |
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14 extern void vocoder_mute_ul (BOOL mute); |
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15 #endif |
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16 |
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17 #if (MELODY_E1) |
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18 //---------------------------------------- |
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19 // Melody format E1 constant. |
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20 //---------------------------------------- |
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21 |
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22 // Number of oscillators (fixed value) |
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23 #define SC_NUMBER_OSCILLATOR 8 |
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24 |
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25 // Define the unit of the downloading time (fixed value) |
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26 #define SC_MELO_DOWNLOAD_TIME_UNIT 4 // unit = 20ms |
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27 |
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28 #endif // MELODY_E1 |
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29 |
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30 #if (VOICE_MEMO) || (SPEECH_RECO) |
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31 //---------------------------------------- |
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32 // Voice memo constant. |
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33 //---------------------------------------- |
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34 |
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35 // Word to indicate the end of the speech data (fixed value). |
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36 #define SC_VM_END_MASK 0xFBFF |
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37 |
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38 #endif // VOICE_MEMO || SPEECH_RECO |
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39 |
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40 #if (L1_VOICE_MEMO_AMR) |
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41 //---------------------------------------- |
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42 // Voice memo amr constant. |
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43 //---------------------------------------- |
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44 |
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45 // Word to indicate the end of the speech data (fixed value). |
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46 #define SC_VM_AMR_END_MASK_SIZE 1 |
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47 #define SC_VM_AMR_END_MASK 0xFF |
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48 |
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49 #endif // L1_VOICE_MEMO_AMR |
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50 |
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51 #if (SPEECH_RECO) |
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52 //---------------------------------------- |
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53 // Speech recognition constant. |
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54 //---------------------------------------- |
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55 |
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56 // Error ID (fixed values) |
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57 #define SC_NO_ERROR 0 // No error |
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58 #define SC_BAD_ACQUISITION 1 // Bad acquisition of the word. The word is too long or too short |
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59 #define SC_TIME_OUT 2 // The DSP task to acquire the word takes to much time |
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60 #define SC_BAD_UPDATE 3 // Bad update of the model. The model from the database is too different |
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61 // than the model built during the acquisition. |
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62 #define SC_BAD_RECOGNITION 4 // This word is out of vocabulary or the best words are too close |
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63 #define SC_CTO_WORD 5 // A word generated by the CTO algorithm is the best word. |
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64 #define SC_CHECK_ERROR 6 // The best word isn't the word to update. |
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65 |
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66 // Time out (fixed values) |
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67 #define SC_SR_AQUISITION_TIME_OUT 867 // acquisition time out in fn unit (3s). |
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68 #define SC_SR_UPDATE_TIME_OUT 500 // update time out in fn unit. |
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69 #define SC_SR_PROCESSING_TIME_OUT 500 // processing time out in fn unit. |
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70 |
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71 |
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72 // CTO algorithm parameters (tuning value) |
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73 #define SC_SR_MAX_WORDS_FOR_CTO 4 // Threshold to decide when the CTO algorithm is needed: |
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74 // if the number of model is less 0r equal to this value, the CTO algo. is enabled. |
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75 |
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76 // model constant (fixed values) |
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77 #define SC_SR_MODEL_FRAME_SIZE 16 // size of the model parameters per audio frames (20 ms). |
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78 #define SC_SR_MODEL_API_SIZE 1041 // size of the model corrsponding to the longest possible word (1,3 second): |
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79 // 16 words frames* 1,3s/20ms + 1 = 1041 |
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80 // the header word of the model gives the size of the model in model frame unit. |
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81 #define SC_SR_MMI_DB_MODEL_SIZE SC_SR_MODEL_API_SIZE // maximum size of the model in the MMI database. |
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82 |
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83 // speech constant (fixed values) |
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84 #define SC_SR_SPEECH_FRAME_SIZE 20 // size of the speech samples per audio frams (20ms) |
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85 |
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86 #define SC_SR_SPEECH_WORD_SIZE 65 // maximum size in speech frame size unit of the word to acquire |
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87 #define SC_SR_SPEECH_WORD_BEGIN_VAD_LATENCY 35 // time in speech frame size unit to detect that the word begins |
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88 #define SC_SR_SPEECH_WORD_END_VAD_LATENCY 35 // time in speech frame size unit to detect that the word is finished |
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89 #define SC_SR_SPEECH_WORD_BEGIN_MARGIN 5 // time in speech frame size unit to add a beginning margin of the word |
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90 #define SC_SR_SPEECH_WORD_END_MARGIN 5 // time in speech frame size unit to add a end margin of the word |
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91 #define SC_SR_SPEECH_ENDING_DONE_MARGING 20 // time in speech frame size unit to have the word done status after the word ending status. |
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92 |
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93 #define SC_SR_MMI_2_L1_SPEECH_SIZE (SC_SR_SPEECH_WORD_BEGIN_MARGIN + SC_SR_SPEECH_WORD_SIZE + SC_SR_SPEECH_WORD_END_VAD_LATENCY + SC_SR_SPEECH_ENDING_DONE_MARGING) * SC_SR_SPEECH_FRAME_SIZE |
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94 // size of the speech buffer allocated by MMI to acquire the speech. |
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95 #define SC_SR_MMI_DB_SPEECH_SIZE (SC_SR_SPEECH_WORD_BEGIN_MARGIN + SC_SR_SPEECH_WORD_SIZE + SC_SR_SPEECH_WORD_END_MARGIN) * SC_SR_SPEECH_FRAME_SIZE + 1 // size of the speech buffer included in a MMI database |
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96 // "+1" is for the END voice memo mask. |
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97 |
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98 // DSP Out-Of-Vocabulary constant (tuning value) |
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99 #define SC_SR_OOV_SFT_THR 10 // OOV rejection threhold (the lower more rejection) |
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100 // if this value is equal to 0, ther's no rejection |
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101 |
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102 #endif // SPEECH_RECO |
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103 |
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104 #if (L1_NEW_AEC) |
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105 // time interval between 2 AEC debug traces (in TDMA). Must be <= 127 |
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106 #define SC_AEC_VISIBILITY_INTERVAL 52 |
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107 #endif |
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108 |
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109 #if (FIR) |
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110 // FIR indication (fixed values) |
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111 #define DL_FIR 1 // The DL FIR must be updated |
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112 #define UL_FIR 2 // The UL FIR must be updated |
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113 #define UL_DL_FIR 3 // The UL&DL FIR must be updated |
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114 |
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115 #endif |
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116 |
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117 // List of the error returned by the Cust_get_pointer function |
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118 #define DATA_AVAILABLE 0 // No error is occured |
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119 #define SESSION_ERROR 1 // Wrong session id |
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120 #define POINTER_ERROR 2 // Wrong ptr argument |
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121 #define DATA_AVAIL_ERROR 3 // No more data available |
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122 |
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123 #if (AUDIO_MODE) |
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124 #define GSM_ONLY 0 // GSM normal mode |
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125 #define BT_CORDLESS 1 // Bluetooth cordless mode |
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126 #define BT_HEADSET 2 // Bluetooth headset mode |
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127 #endif |
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128 |
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129 #if (MELODY_E2) |
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130 #define SC_AUDIO_MELODY_E2_MAX_NUMBER_OF_INSTRUMENT 8 // Maximum number of instruments allowed to play in thesame time |
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131 // (Fixed value) |
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132 #define SC_AUDIO_MELODY_E2_MAX_SIZE_OF_INSTRUMENT (3807 - C_DEBUG_BUFFER_SIZE) |
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133 // Melody E2 instrument wave table size in the API memory |
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134 // (fixed value) |
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135 #define SC_AUDIO_MELODY_E2_MAX_SIZE_OF_DSP_TRACE (C_DEBUG_BUFFER_SIZE + 1) |
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136 // DSP API buffer trace size (fixed value) |
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137 |
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138 // Note :the melody E2 instrument are overlayed with the DSP buffer trace. The size ofthe trace buffer can |
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139 // change in order to increase the DSP tracability. In all case, the following rules need to be followed |
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140 // (when melody E2 is activated): |
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141 // size of the E2 instruments buffer + size of DSP trace buffer = 2049 |
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142 // size of the E2 instrument buffers > 1 |
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143 // size of DSP trace buffer > 1 |
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144 #endif |
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145 #endif // AUDIO_TASK |
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146 |
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147 // Number of coefficient for each FIR (fixed value) |
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148 #define MAX_FIR_COEF 31 |
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149 |