FreeCalypso > hg > freecalypso-sw
view gsm-fw/L1/audio_cust0/l1audio_cust.h @ 978:1ee80bcb0fd0
fc-tmsh: omr (old-style memory read) command implemented
author | Mychaela Falconia <falcon@ivan.Harhan.ORG> |
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date | Thu, 10 Dec 2015 04:05:55 +0000 |
parents | d638de8cc6b8 |
children |
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/************* Revision Controle System Header ************* * GSM Layer 1 software * L1AUDIO_CUST.H * * Filename l1audio_cust.h * Copyright 2003 (C) Texas Instruments * ************* Revision Controle System Header *************/ #ifndef _L1AUDIO_CUST_H_ #define _L1AUDIO_CUST_H_ #if (AUDIO_TASK == 1) #if (OP_RIV_AUDIO == 0) extern void vocoder_mute_dl (BOOL mute); extern void vocoder_mute_ul (BOOL mute); #endif #if (MELODY_E1) //---------------------------------------- // Melody format E1 constant. //---------------------------------------- // Number of oscillators (fixed value) #define SC_NUMBER_OSCILLATOR 8 // Define the unit of the downloading time (fixed value) #define SC_MELO_DOWNLOAD_TIME_UNIT 4 // unit = 20ms #endif // MELODY_E1 #if (VOICE_MEMO) || (SPEECH_RECO) //---------------------------------------- // Voice memo constant. //---------------------------------------- // Word to indicate the end of the speech data (fixed value). #define SC_VM_END_MASK 0xFBFF #endif // VOICE_MEMO || SPEECH_RECO #if (L1_VOICE_MEMO_AMR) //---------------------------------------- // Voice memo amr constant. //---------------------------------------- // Word to indicate the end of the speech data (fixed value). #define SC_VM_AMR_END_MASK_SIZE 1 #define SC_VM_AMR_END_MASK 0xFF #endif // L1_VOICE_MEMO_AMR #if (SPEECH_RECO) //---------------------------------------- // Speech recognition constant. //---------------------------------------- // Error ID (fixed values) #define SC_NO_ERROR 0 // No error #define SC_BAD_ACQUISITION 1 // Bad acquisition of the word. The word is too long or too short #define SC_TIME_OUT 2 // The DSP task to acquire the word takes to much time #define SC_BAD_UPDATE 3 // Bad update of the model. The model from the database is too different // than the model built during the acquisition. #define SC_BAD_RECOGNITION 4 // This word is out of vocabulary or the best words are too close #define SC_CTO_WORD 5 // A word generated by the CTO algorithm is the best word. #define SC_CHECK_ERROR 6 // The best word isn't the word to update. // Time out (fixed values) #define SC_SR_AQUISITION_TIME_OUT 867 // acquisition time out in fn unit (3s). #define SC_SR_UPDATE_TIME_OUT 500 // update time out in fn unit. #define SC_SR_PROCESSING_TIME_OUT 500 // processing time out in fn unit. // CTO algorithm parameters (tuning value) #define SC_SR_MAX_WORDS_FOR_CTO 4 // Threshold to decide when the CTO algorithm is needed: // if the number of model is less 0r equal to this value, the CTO algo. is enabled. // model constant (fixed values) #define SC_SR_MODEL_FRAME_SIZE 16 // size of the model parameters per audio frames (20 ms). #define SC_SR_MODEL_API_SIZE 1041 // size of the model corrsponding to the longest possible word (1,3 second): // 16 words frames* 1,3s/20ms + 1 = 1041 // the header word of the model gives the size of the model in model frame unit. #define SC_SR_MMI_DB_MODEL_SIZE SC_SR_MODEL_API_SIZE // maximum size of the model in the MMI database. // speech constant (fixed values) #define SC_SR_SPEECH_FRAME_SIZE 20 // size of the speech samples per audio frams (20ms) #define SC_SR_SPEECH_WORD_SIZE 65 // maximum size in speech frame size unit of the word to acquire #define SC_SR_SPEECH_WORD_BEGIN_VAD_LATENCY 35 // time in speech frame size unit to detect that the word begins #define SC_SR_SPEECH_WORD_END_VAD_LATENCY 35 // time in speech frame size unit to detect that the word is finished #define SC_SR_SPEECH_WORD_BEGIN_MARGIN 5 // time in speech frame size unit to add a beginning margin of the word #define SC_SR_SPEECH_WORD_END_MARGIN 5 // time in speech frame size unit to add a end margin of the word #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. #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 // size of the speech buffer allocated by MMI to acquire the speech. #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 // "+1" is for the END voice memo mask. // DSP Out-Of-Vocabulary constant (tuning value) #define SC_SR_OOV_SFT_THR 10 // OOV rejection threhold (the lower more rejection) // if this value is equal to 0, ther's no rejection #endif // SPEECH_RECO #if (L1_NEW_AEC) // time interval between 2 AEC debug traces (in TDMA). Must be <= 127 #define SC_AEC_VISIBILITY_INTERVAL 52 #endif #if (FIR) // FIR indication (fixed values) #define DL_FIR 1 // The DL FIR must be updated #define UL_FIR 2 // The UL FIR must be updated #define UL_DL_FIR 3 // The UL&DL FIR must be updated #endif // List of the error returned by the Cust_get_pointer function #define DATA_AVAILABLE 0 // No error is occured #define SESSION_ERROR 1 // Wrong session id #define POINTER_ERROR 2 // Wrong ptr argument #define DATA_AVAIL_ERROR 3 // No more data available #define DATA_LAST 4 // Last buffer, no more data available after #define WAIT_FOR_DATA 6 #if (AUDIO_MODE) #define GSM_ONLY 0 // GSM normal mode #define BT_CORDLESS 1 // Bluetooth cordless mode #define BT_HEADSET 2 // Bluetooth headset mode #endif #if (MELODY_E2) #define SC_AUDIO_MELODY_E2_MAX_NUMBER_OF_INSTRUMENT 8 // Maximum number of instruments allowed to play in thesame time // (Fixed value) #define SC_AUDIO_MELODY_E2_MAX_SIZE_OF_INSTRUMENT (3807 - C_DEBUG_BUFFER_SIZE) // Melody E2 instrument wave table size in the API memory // (fixed value) #define SC_AUDIO_MELODY_E2_MAX_SIZE_OF_DSP_TRACE (C_DEBUG_BUFFER_SIZE + 1) // DSP API buffer trace size (fixed value) // Note :the melody E2 instrument are overlayed with the DSP buffer trace. The size ofthe trace buffer can // change in order to increase the DSP tracability. In all case, the following rules need to be followed // (when melody E2 is activated): // size of the E2 instruments buffer + size of DSP trace buffer = 2049 // size of the E2 instrument buffers > 1 // size of DSP trace buffer > 1 #endif #endif // AUDIO_TASK // Number of coefficient for each FIR (fixed value) #define MAX_FIR_COEF 31 // Triton Audio ON/OFF Changes #if (L1_AUDIO_MCU_ONOFF == 1) // Num of radio frames the audio path is kept on after all // users have requested turn off // 0..255 #define L1_AUDIO_ON2OFF_UL_HOLD_TIME 20 //127 #define L1_AUDIO_ON2OFF_DL_HOLD_TIME 20 //127 #endif #endif // _L1AUDIO_CUST_H_