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Squashed commit of the following:
commit 833e0173f24b12a12ff80f9a35fb43107d355cc3 Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 26 22:51:02 2024 -0400 Fix Slot Preference Depending on Build; commit 716577ab09c0cee68bca3bcdcbaaa7cfa3be4f97 Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 26 22:27:24 2024 -0400 M17 - Voice Playback Only If Enabled; Misc; commit c0f9818cf38ce7f127370f19ec4afa2457cef19a Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 26 17:39:34 2024 -0400 Add DMR/P25p2 Stereo Channel Patch; commit 1a0baf0a17ccf8cc0eb0a01a809269354480d1d1 Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 26 17:37:46 2024 -0400 Add DMR Cap+ Faster Rest Patch; commit 9f5b52aa8f20bfe3eab5413f2c3fa41003d53b3c Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 26 17:36:24 2024 -0400 Disable EDACS Analog Debug; commit 7dd0de67b602f0acd111200f838fbb1a0541a8bf Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 26 17:36:04 2024 -0400 Squashed commit of the following: commit 95d46fb35d6cb6fb0a9825084f4c2accf8edad4a Author: lwvmobile <lwvmobile@gmail.com> Date: Mon Mar 25 17:48:57 2024 -0400 Enable HPF_D by default; Increase manual analog gain to 50; commit 1f37837664e16760b5cbf5653fc4d829ee6cce83 Author: lwvmobile <lwvmobile@gmail.com> Date: Mon Mar 25 08:43:17 2024 -0400 Enable HPF_D on M17 Voice; commit9bbe4de7fdAuthor: lwvmobile <lwvmobile@gmail.com> Date: Sun Mar 24 16:08:16 2024 -0400 EDACS: Digitize Analog For Analysis of 'Data Burst' Like Behavior; commit9249b7a943Author: lwvmobile <lwvmobile@gmail.com> Date: Sun Mar 24 12:28:26 2024 -0400 Simply Mono Short Playback; commit89ad0f17e3Author: lwvmobile <lwvmobile@gmail.com> Date: Sat Mar 23 21:12:44 2024 -0400 Add HPF to SS4, SS3, and SS Audio Methods; commite8184c7fbdAuthor: lwvmobile <lwvmobile@gmail.com> Date: Sat Mar 23 17:13:44 2024 -0400 RTL - Add The Sample / Volume Multiplier as a user configurable option; commitee04d11116Author: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 23:15:04 2024 -0400 Fix HPF_D so it actually works correctly; commite549b21497Author: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 20:11:34 2024 -0400 Make Analog Auto Gain Control Mostly Tolerable; commit2f9e0525feAuthor: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 18:20:38 2024 -0400 User Toggle HPF on Digital; commitff58d3a6faAuthor: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 18:10:15 2024 -0400 User Toggles and Displays for Filters; commitb6fc17c506Author: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 15:55:09 2024 -0400 More Analog Manual Audio Gain Tweaks; commit076b9fbf04Author: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 06:26:10 2024 -0400 Add SQL Hit on EDACS if Colors Disabled; commit59fe47ef05Author: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 02:23:08 2024 -0400 Disable HPF Test on Digital Voice; commit4a63068ce1Author: lwvmobile <lwvmobile@gmail.com> Date: Thu Mar 21 23:39:43 2024 -0400 Misc Code Organizing; audio_gainA Tweaks; commit3585e7e594Author: lwvmobile <lwvmobile@gmail.com> Date: Thu Mar 21 13:18:31 2024 -0400 Test Tweak to RTL Sample Multiplier; commitfa5b25283fAuthor: lwvmobile <lwvmobile@gmail.com> Date: Thu Mar 21 13:18:07 2024 -0400 Adjust Manual Mic Gain on M17 Encoder; Ncurses Display; commitc985fea035Author: lwvmobile <lwvmobile@gmail.com> Date: Thu Mar 21 12:34:01 2024 -0400 Misc Notes on using raw_rms values during testing; commitb273e5b7beAuthor: lwvmobile <lwvmobile@gmail.com> Date: Thu Mar 21 12:31:33 2024 -0400 Bugfix for EDACS Analog Segfault when using the digitize function for an extended period; commiteadf3fdf23Author: lwvmobile <lwvmobile@gmail.com> Date: Thu Mar 21 10:12:04 2024 -0400 Analog Gain Has Own Variable; Ncurses Keys; Misc; commit0debcc647eAuthor: lwvmobile <lwvmobile@gmail.com> Date: Thu Mar 21 08:49:51 2024 -0400 Filtering Misc Cleanup; commit98e10b2627Author: lwvmobile <lwvmobile@gmail.com> Date: Thu Mar 21 00:02:21 2024 -0400 Automatic Gain On Analog WIP; Manual Gain Control; commit89736ae543Author: lwvmobile <lwvmobile@gmail.com> Date: Wed Mar 20 10:22:20 2024 -0400 Passband Filter Testing on Raw Analog / EDACS / M17 Encoder; commit43f6218129Author: lwvmobile <lwvmobile@gmail.com> Date: Wed Mar 20 09:03:02 2024 -0400 Passband and Notch Filter Setup; Debug Output; commit35a1b53867Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 19 22:35:36 2024 -0400 HPF: Only Use On RTL Input when M17 Encoder Uses It; commitb9949f0d80Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 19 17:31:59 2024 -0400 Disable LFP on EDACS Analog Out; commit3666740674Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 19 13:42:07 2024 -0400 Test HPF On Digital Voice Out; commitef13343925Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 19 12:46:27 2024 -0400 RTL_FM Input for Analog/M17 Encoder; commit2b7de90474Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 19 11:03:55 2024 -0400 New Analog Audio Filters WIP; commit 3b9f9f5a3befb295da284379262323fad7d69c7a Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 26 17:33:34 2024 -0400 Squashed commit of the following: commit a5c5be3b955a82cc72f1486ff87d7c1b4893295f Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Tue Mar 26 17:28:35 2024 -0400 EDACS: Standard display rewrite, parsing/RE fixes, tune i-calls and all-calls (#246) * EDACS EA: Fix reversed TDMA/data calls * EDACS: deduplicate ncurses printing logic * EDACS: pass call state via new state struct field * EDACS: RE some differences from Standard spec * EDACS: show state for data calls * EDACS: tune individual calls on Standard * EDACS: tune all-calls on Standard * EDACS: properly identify emergency on channel updates * EDACS: fix build break * EDACS: fix conditions for refreshing the display source * EDACS EA: Identify calls as voice for display * EDACS: voice channel update emergency refactor * EDACS: Show interconnect calls in call log * EDACS EA: show unknown data channel assignments in log * EDACS: check LCN is not 0 before assigning it * EDACS: C operator precedence is dumb * Initialize edacs_vc_call_type * use j and not i in call_matrix for call history; --------- Co-authored-by: lwvmobile <lwvmobile@gmail.com> commitc8d00846b1Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sun Mar 24 17:48:46 2024 -0400 EDACS: Display cached call source on Standard, tune group calls on assignment (#241) * Trim trailing whitespace * EDACS: Display cached call source on Standard The logic in `edacs-fme.c` didn't work, in retrospect, because it would have depended on getting back-to-back messages for the same call. In practice, that is unlikely even on an idle system as system state gets displayed. Instead, cache on the display end, for a "better" user experience. * EDACS: Pad channel freq to "000.000000 MHz" * EDACS: Process/tune group calls on assignment commit8d432e8b3eAuthor: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sun Mar 24 12:29:22 2024 -0400 EDACS: ESK fixes, RE of standard channel assignments, cleanups (#238) * EDACS: Make i-calls very obvious on standard * EDACS: abstract target for Interconnect Channel Assignment * EDACS: little shortcut example updates * EDACS: Apply ESK to both FR1 and FR4 * EDACS: Group channel assignment by mode * EDACS: Merge MT-A 0/1 and 2/3 cases commit0440dea1deAuthor: lwvmobile <lwvmobile@gmail.com> Date: Sun Mar 24 02:47:09 2024 -0400 EDACS: Fix Call History For Standard Maybe; commit208f188e55Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sat Mar 23 21:35:42 2024 -0400 EDACS: Speculative RE of MT-A value 0, pretty colours (#231) * EDACS: fix log formatting * EDACS: Speculative RE of MT-A value 0 * EDACS: CC message pretty colours * EDACS: little formatting * EDACS: eh make interconnect magenta commit36f76a7e21Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sat Mar 23 17:35:45 2024 -0400 EDACS: Fix copy/paste error (#230) I went through by hand and I'm confident this is the last of the `get_lcn_status_string()` errors. commit4bc73d8c68Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sat Mar 23 17:24:32 2024 -0400 EDACS: Rewrite Standard/Networked parsing per TIA/EIA TSB 69.3 (#229) * EDACS EA: refactor log messages for consistency * EDACS: Log LCN status values * EDACS EA: channel assignment vs update * EDACS: Implement TIA/EIA TSB 69.3 * EDACS: clean up Site ID message * EDACS: Actually process call for Channel Update * EDACS: Update display for Standard calls * EDACS: Format of Voice Group Channel Assignment * EDACS Standard: Log LCN status values * EDACS: Rebase and fix some issues * EDACS: Fix copy/paste error commit03507055a4Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sat Mar 23 16:05:09 2024 -0400 EDACS: add EA messages / start mass Standard refactor (#226) * EDACS EA: refactor log messages for consistency * EDACS EA: channel assignment command (unknown data) * EDACS EA: status/message command * EDACS Standard: Refactor variable declarations * EDACS Standard: log raw payloads for debugging * EDACS: Fix typing * EDACS EA: update status/message CC command * EDACS: Log LCN status values * EDACS: String format * EDACS EA: channel assignment vs update * EDACS: fix syntax error * EDACS: change string for reserved LCN status commitfbe5fd2093Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Fri Mar 22 15:26:50 2024 -0400 EDACS EA: emergency calls, MT1 fixes, logging improvements (#221) * EDACS EA: Fix i-call MT1 after ESK changes * EDACS EA: Fix all-call ncurses display * EDACS EA: Make detailed log slightly more readable * EDACS EA: Improve colours in logs * EDACS EA: Space in logs for consistency * EDACS EA: Fix MT1 for analog group voice after ESK change * EDACS EA: add parsing for emergency calls commit413e5f76a2Merge:de9127c388a57dAuthor: lwvmobile <59371473+lwvmobile@users.noreply.github.com> Date: Fri Mar 22 06:59:31 2024 -0400 Merge pull request #220 from lwvmobile/edacs_manual_modes Edacs manual modes commit388a57d8fcAuthor: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 05:28:11 2024 -0400 EDACS: Re-enable the -9 switch to maintain user compatability; Fix Missing Bracket on non color mode; commit058099f1aaAuthor: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 03:24:57 2024 -0400 EDACS: Tweak Mode Toggle Display; Add Frame Dump on No Mode Set; commit68aa6b018eAuthor: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 02:19:38 2024 -0400 EDACS: Update Docs to Reflect Changes; commitc404f3c48cAuthor: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 02:01:09 2024 -0400 EDACS: Ncurses Display Working Mode Tweaks; commitb2d1608eb7Author: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 01:08:28 2024 -0400 EDACS: Toggle Keys A and S for Mode Selection; commita4553a0a0aAuthor: lwvmobile <lwvmobile@gmail.com> Date: Fri Mar 22 00:45:37 2024 -0400 EDACS: Remove Auto Detect Code; Make CLI Switches; commitde9127cfb4Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Thu Mar 21 21:02:53 2024 -0400 EDACS: support for EA system all-calls (#214) * Refactor debug printing a bit * Clean up group call logic * Minor changes to debug printing * EDACS EA: support for system all-calls * EDACS: split out MT1 for TDMA call since we can't decode it anyway * Refactor identifying call mode (digital/analog) * Maintain consistent indentation for all MT1 * EDACS: Fix Compiler Error. Had an extra right bracket on line 383 --------- Co-authored-by: lwvmobile <59371473+lwvmobile@users.noreply.github.com> commitc85dd60399Author: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 19 17:38:30 2024 -0400 EDACS: NET/STD LCN > 0 Check; commitbed766a86eAuthor: lwvmobile <lwvmobile@gmail.com> Date: Tue Mar 19 09:34:10 2024 -0400 EDACS: STD/NET Don't Tune Unless CC LCN Set; commit69f532a022Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Mon Mar 18 11:41:58 2024 -0400 EDACS: EA I-Call decoding improvements (#205) * EDACS: formatting nit * EDACS: add name to credits * EDACS: Handle initial grant vs grant update for i-call * EDACS: consistency for group vs i-call * EDACS: file naming for i-call recordings * EDACS: EA refactor when MT1 is 0x1F * EDACS: check we know CC LCN before leaving it * EDACS: fix breaking change for Standard Revert mistake inf68367c4f3* Fix patent number * EDACS: Fix Compiler Error 1; Fix Rogue Copy Paste * EDACS: Add MT1, MT2, and FR1 and FR4 Dumps on Payload; --------- Co-authored-by: lwvmobile <59371473+lwvmobile@users.noreply.github.com> commita7de24c47bAuthor: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Mon Mar 18 07:47:34 2024 -0400 EDACS: fixes and improvements to EA analog and i-call parsing (#204) * Fix MT1 math given esk_mask value is 0xA0 * EDACS: Clean up formatting for I-Calls * EDACS: typo, fix the TGID for group calls Fix copy-pasta * EDACS: Fix MT2 for I-Calls Incorrectly copied decimal values as hex - obviously MT2 is only 4 bits so it cannot be greater than 15. * EDACS: check LCN is > 0 before trying to tune An EA CC sometimes includes commands that parse as LCN 0 - but otherwise valid - analog call voice grants for TG 1. Since LCN 0 is not valid, ensure that we don't try to tune to them while we try to figure out what they're about. * EDACS: Process "login" message on CC Log "login" messages sent on the CC. Verified behaviour against UniTrunker. commit10eacd99ccAuthor: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sun Mar 17 18:56:41 2024 -0400 EDACS: Clean up formatting for I-Calls (#203) * Fix MT1 math given esk_mask value is 0xA0 * EDACS: Clean up formatting for I-Calls commitebe6e8fc65Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sun Mar 17 18:02:12 2024 -0400 Fix MT1 math given esk_mask value is 0xA0 (#202) commit8619612b76Author: lwvmobile <lwvmobile@gmail.com> Date: Sat Mar 16 22:42:44 2024 -0400 EDACS: Setup Analog Calls and WAVs on EA and ICALL; commit9ac1573f0bAuthor: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sat Mar 16 17:17:14 2024 -0400 EDACS: Add MT1 value / logging for analog group calls (#198) * Add MT1 value / logging for analog group calls * Respect user preference for tuning to group calls * Revert "Respect user preference for tuning to group calls" This reverts commit83ac8feab7. commit0aa01a45efAuthor: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sat Mar 16 17:16:52 2024 -0400 EDACS: Respect user preference for tuning to group calls (#201) commitf68367c4f3Author: ilyacodes <33097525+ilyacodes@users.noreply.github.com> Date: Sat Mar 16 17:16:20 2024 -0400 EDACS: Add parsing for individual calls (#200) * EDACS: Add parsing for individual calls This change adds parsing for I-Calls on EA systems. Both analog and ProVoice modes are parsed and processed. See related issue for discussion. * Respect user preference for tuning to private calls * EDACS: Respect user preference for tuning to private calls * EDACS: Handle i-call target in ProVoice logging * EDACS: Show i-calls distinctly from group calls
This commit is contained in:
104
include/dsd.h
104
include/dsd.h
@@ -86,6 +86,35 @@
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extern volatile uint8_t exitflag; //fix for issue #136
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//new audio filter stuff from: https://github.com/NedSimao/FilteringLibrary
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typedef struct {
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float coef[2];
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float v_out[2];
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}LPFilter;
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typedef struct {
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float coef;
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float v_out[2];
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float v_in[2];
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}HPFilter;
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typedef struct {
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LPFilter lpf;
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HPFilter hpf;
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float out_in;
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}PBFilter;
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typedef struct {
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float alpha;
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float beta;
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float vin[3];
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float vout[3];
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}NOTCHFilter;
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//end new filters
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//group csv import struct
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typedef struct
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{
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@@ -210,8 +239,8 @@ typedef struct
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SF_INFO *audio_in_file_info;
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uint32_t rtlsdr_center_freq;
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int rtlsdr_ppm_error;
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int audio_in_type;
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int rtlsdr_ppm_error;
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int audio_in_type;
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char audio_out_dev[1024];
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int audio_out_fd;
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int audio_out_fdR; //right channel audio for OSS hack
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@@ -231,6 +260,7 @@ typedef struct
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FILE *symbol_out_f;
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float audio_gain;
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float audio_gainR;
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float audio_gainA;
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int audio_out;
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int dmr_stereo_wav;
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char wav_out_dir[512];
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@@ -285,6 +315,7 @@ typedef struct
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int rtl_started;
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long int rtl_rms;
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int monitor_input_audio;
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int analog_only;
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int pulse_raw_rate_in;
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int pulse_raw_rate_out;
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int pulse_digi_rate_in;
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@@ -367,7 +398,7 @@ typedef struct
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char group_in_file[1024];
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char lcn_in_file[1024];
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char chan_in_file[1024];
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char key_in_file[1024];
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char key_in_file[1024];
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//end import filenames
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//reverse mute
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@@ -405,6 +436,12 @@ typedef struct
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int slot1_on;
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int slot2_on;
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//enable filter options
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int use_lpf;
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int use_hpf;
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int use_pbf;
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int use_hpf_d;
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//'DSP' Format Output
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uint8_t use_dsp_output;
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char dsp_out_file[2048];
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@@ -447,14 +484,14 @@ typedef struct
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//new stereo short sample storage
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short s_l[160]; //single sample left
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short s_r[160]; //single sample right
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short s_l4[4][160]; //quad sample for up to a P25p2 4V
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short s_r4[4][160]; //quad sample for up to a P25p2 4V
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short s_l4[18][160]; //quad sample for up to a P25p2 4V
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short s_r4[18][160]; //quad sample for up to a P25p2 4V
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//new stereo short sample storage tapped from 48_k internal upsampling
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short s_lu[160*6]; //single sample left
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short s_ru[160*6]; //single sample right
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short s_l4u[4][160*6]; //quad sample for up to a P25p2 4V
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short s_r4u[4][160*6]; //quad sample for up to a P25p2 4V
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//end
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//end
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int audio_out_idx;
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int audio_out_idx2;
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int audio_out_idxR;
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@@ -507,6 +544,7 @@ typedef struct
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char slot0light[8];
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float aout_gain;
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float aout_gainR;
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float aout_gainA;
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float aout_max_buf[200];
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float aout_max_bufR[200];
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float *aout_max_buf_p;
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@@ -616,7 +654,7 @@ typedef struct
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char dmr_cach_fragment[4][17]; //unsure of size, will need to check/verify
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int dmr_cach_counter; //counter for dmr_cach_fragments 0-3; not sure if needed yet.
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//dmr talker alias new/fixed stuff
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uint8_t dmr_alias_format[2]; //per slot
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uint8_t dmr_alias_len[2]; //per slot
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@@ -630,6 +668,16 @@ typedef struct
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dPMRVoiceFS2Frame_t dPMRVoiceFS2Frame;
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//new audio filter structs
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LPFilter RCFilter;
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HPFilter HRCFilter;
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PBFilter PBF;
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NOTCHFilter NF;
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LPFilter RCFilterL;
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HPFilter HRCFilterL;
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LPFilter RCFilterR;
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HPFilter HRCFilterR;
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char dpmr_caller_id[20];
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char dpmr_target_id[20];
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@@ -664,6 +712,7 @@ typedef struct
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int p2_vch_chan_num; //vch channel number (0 or 1, not the 0-11 TS)
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int ess_b[2][96]; //external storage for ESS_B fragments
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int fourv_counter[2]; //external reference counter for ESS_B fragment collection
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int voice_counter[2]; //external reference counter for 18V x 2 P25p2 Superframe
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int p2_is_lcch; //flag to tell us when a frame is lcch and not sacch
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//iden freq storage for frequency calculations
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@@ -695,13 +744,23 @@ typedef struct
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//edacs
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int ea_mode;
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int esk_mode;
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unsigned short esk_mask;
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unsigned long long int edacs_site_id;
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int edacs_lcn_count; //running tally of lcn's observed on edacs system
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int edacs_cc_lcn; //current lcn for the edacs control channel
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int edacs_vc_lcn; //current lcn for any active vc (not the one we are tuned/tuning to)
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int edacs_tuned_lcn; //the vc we are currently tuned to...above variable is for updating all in the matrix
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int edacs_vc_call_type; //the type of call on the given VC - see defines below
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//flags for EDACS call type
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#define EDACS_IS_VOICE 0x01
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#define EDACS_IS_DIGITAL 0x02
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#define EDACS_IS_EMERGENCY 0x04
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#define EDACS_IS_GROUP 0x08
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#define EDACS_IS_INDIVIDUAL 0x10
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#define EDACS_IS_ALL_CALL 0x20
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#define EDACS_IS_INTERCONNECT 0x40
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//trunking group and lcn freq list
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long int trunk_lcn_freq[26]; //max number on an EDACS system, should be enough on DMR too hopefully
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@@ -943,10 +1002,15 @@ void playSynthesizedVoiceFS3 (dsd_opts * opts, dsd_state * state); //float stere
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void playSynthesizedVoiceFS4 (dsd_opts * opts, dsd_state * state); //float stereo mix 4v2 P25p2
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void playSynthesizedVoiceFM (dsd_opts * opts, dsd_state * state); //float mono
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void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot); //float gain control
|
||||
void agsm (dsd_opts * opts, dsd_state * state, short * input, int len); //short gain control for analog things
|
||||
void analog_gain (dsd_opts * opts, dsd_state * state, short * input, int len); //manual gain handling for analong things
|
||||
//new short stuff
|
||||
void playSynthesizedVoiceMS (dsd_opts * opts, dsd_state * state); //short mono mix
|
||||
void playSynthesizedVoiceMSR (dsd_opts * opts, dsd_state * state); //short mono mix R (needed for OSS 48k input/output)
|
||||
void playSynthesizedVoiceSS (dsd_opts * opts, dsd_state * state); //short stereo mix
|
||||
void playSynthesizedVoiceSS3 (dsd_opts * opts, dsd_state * state); //short stereo mix 3v2 DMR
|
||||
void playSynthesizedVoiceSS4 (dsd_opts * opts, dsd_state * state); //short stereo mix 4v2 P25p2
|
||||
void playSynthesizedVoiceSS18 (dsd_opts * opts, dsd_state * state); //short stereo mix 18V Superframe
|
||||
void upsampleS (short invalue, short prev, short outbuf[6]); //upsample 8k to 48k short
|
||||
//
|
||||
void openAudioOutDevice (dsd_opts * opts, int speed);
|
||||
@@ -1188,7 +1252,7 @@ bool crc8_ok(uint8_t bits[], unsigned int len);
|
||||
//modified CRC functions for SB/RC
|
||||
uint8_t crc7(uint8_t bits[], unsigned int len);
|
||||
uint8_t crc3(uint8_t bits[], unsigned int len);
|
||||
uint8_t crc4(uint8_t bits[], unsigned int len);
|
||||
uint8_t crc4(uint8_t bits[], unsigned int len);
|
||||
|
||||
//LFSR and LFSRP code courtesy of https://github.com/mattames/LFSR/
|
||||
void LFSR(dsd_state * state);
|
||||
@@ -1285,12 +1349,24 @@ void eot_cc(dsd_opts * opts, dsd_state * state); //end of TX return to CC
|
||||
//Generic Tuning Functions
|
||||
void return_to_cc (dsd_opts * opts, dsd_state * state);
|
||||
|
||||
//misc generic audio filtering for analog at 48k/1
|
||||
//misc audio filtering for analog
|
||||
long int raw_rms(short *samples, int len, int step);
|
||||
void analog_deemph_filter(short * input, int len);
|
||||
void analog_preemph_filter(short * input, int len);
|
||||
void analog_dc_block_filter(short * input, int len);
|
||||
void analog_clipping_filter(short * input, int len);
|
||||
void init_audio_filters(dsd_state * state);
|
||||
void lpf(dsd_state * state, short * input, int len);
|
||||
void hpf(dsd_state * state, short * input, int len);
|
||||
void pbf(dsd_state * state, short * input, int len);
|
||||
void nf(dsd_state * state, short * input, int len);
|
||||
void hpf_dL(dsd_state * state, short * input, int len);
|
||||
void hpf_dR(dsd_state * state, short * input, int len);
|
||||
//from: https://github.com/NedSimao/FilteringLibrary
|
||||
void LPFilter_Init(LPFilter *filter, float cutoffFreqHz, float sampleTimeS);
|
||||
float LPFilter_Update(LPFilter *filter, float v_in);
|
||||
void HPFilter_Init(HPFilter *filter, float cutoffFreqHz, float sampleTimeS);
|
||||
float HPFilter_Update(HPFilter *filter, float v_in);
|
||||
void PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS);
|
||||
float PBFilter_Update(PBFilter *filter, float v_in);
|
||||
void NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS);
|
||||
float NOTCHFilter_Update(NOTCHFilter *filter, float vin);
|
||||
|
||||
//csv imports
|
||||
int csvGroupImport(dsd_opts * opts, dsd_state * state);
|
||||
|
||||
Reference in New Issue
Block a user