5328 lines
176 KiB
C
Executable File
5328 lines
176 KiB
C
Executable File
#include <Python.h>
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#include <stdlib.h>
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#include <math.h>
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#include <complex.h> // Use native C99 complex type for fftw3
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#include <fftw3.h>
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#include <sys/types.h>
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#ifdef MS_WINDOWS
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#include <Winsock2.h>
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#include <iphlpapi.h>
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static int cleanupWSA = 0; // Must we call WSACleanup() ?
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HWND quisk_mainwin_handle; // Handle of the main window on Windows
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#else
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#include <sys/socket.h>
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#include <arpa/inet.h>
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#include <netinet/in.h>
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#include <ifaddrs.h>
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#endif
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#include "quisk.h"
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#include "filter.h"
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#define DEBUG 0
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// These are used for input/output of radio samples from/to a file. The SAMPLES_FROM_FILE is 0 for
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// normal operation, 1 to record samples to a file, 2 to play samples from a file. Rate must be 48k.
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#define SAMPLES_FROM_FILE 0
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#define FM_FILTER_DEMPH 300.0 // Frequency of FM lowpass de-emphasis filter
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#define AGC_DELAY 15 // Delay in AGC buffer in milliseconds
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#define FFT_ARRAY_SIZE 4 // Number of FFTs
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#define MULTIRX_FFT_MULT 8 // multirx FFT size is a multiple of graph size
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#define MAX_RX_CHANNELS 3 // maximum paths to decode audio
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#define MAX_RX_FILTERS 3 // maximum number of receiver filters
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#define DGT_NARROW_FREQ 3000 // Use 6 ksps rate below this bandwidth
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#define SQUELCH_FFT_SIZE 512
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static int fft_error; // fft error count
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typedef struct fftd {
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fftw_complex * samples; // complex data for fft
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int index; // position of next fft sample
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int filled; // whether the fft is ready to run
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int block; // block number 0, 1, ...
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} fft_data;
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typedef struct mrx_fftd { // FFT data for sub-receivers
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fftw_complex * samples; // complex data for fft of size multirx_fft_width
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int index; // position of next fft sample
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} mrx_fft_data;
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struct AgcState { // Store state information for the AGC
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double max_out; // Must initialize to maximum output level 0.0 to 1.0.
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int sample_rate; // Must initialize this to the sample rate or zero.
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int buf_size; // Must initialize this to zero.
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int index_read;
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int index_start;
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int is_clipping;
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double themax;
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double gain;
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double delta;
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double target_gain;
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double time_release;
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complex double * c_samp;
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};
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static fft_data fft_data_array[FFT_ARRAY_SIZE]; // Data for several FFTs
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static int fft_data_index = 0; // Write the current samples to this FFT
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static fftw_plan quisk_fft_plan;
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static double * fft_window; // Window for FFT data
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static double * current_graph; // current graph data as returned
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static PyObject * QuiskError; // Exception for this module
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static PyObject * pyApp; // Application instance
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static int fft_size; // size of fft, e.g. 1024
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int data_width; // number of points to return as graph data; fft_size * n
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static int graph_width; // width of the graph in pixels
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rx_mode_type rxMode; // 0 to 13: CWL, CWU, LSB, USB, AM, FM, EXT, DGT-U, DGT-L, DGT-IQ, IMD, FDV-U, FDV-L, DGT-FM
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int quisk_noise_blanker; // noise blanker level, 0 for off
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int quiskTxHoldState; // hold Tx until the repeater frequency shift is complete
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int quisk_is_vna; // zero for normal program, one for the VNA program
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static int py_sample_rx_bytes=2; // number of bytes in each I or Q sample: 1, 2, 3, or 4
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static int py_sample_rx_endian; // order of sample array: 0 == little endian; 1 == big endian
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static int py_bscope_bytes;
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static int py_bscope_endian;
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static int quisk_auto_notch; // auto notch control
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PyObject * quisk_pyConfig=NULL; // Configuration module instance
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static int graphX; // Origin of first X value for graph data
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static int graphY; // Origin of 0 dB for graph data
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static double graphScale; // Scale factor for graph
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static complex double testtonePhase; // Phase increment for test tone
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double quisk_audioVolume; // Audio output level, 0.0 to 1.0
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double quisk_ctcss_freq; // CTCSS repeater access tone frequency in Hertz, or zero
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static double cFilterI[MAX_RX_FILTERS][MAX_FILTER_SIZE]; // Digital filter coefficients for receivers
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static double cFilterQ[MAX_RX_FILTERS][MAX_FILTER_SIZE]; // Digital filter coefficients
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static int sizeFilter; // Number of coefficients for filters
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static int isFDX; // Are we in full duplex mode?
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static int filter_bandwidth[MAX_RX_FILTERS]; // Current filter bandwidth in Hertz
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static int filter_start_offset; // Current filter +/- start offset frequency from rx_tune_freq in Hertz for filter zero
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static int quisk_decim_srate; // Sample rate after decimation
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static int quisk_filter_srate=48000; // Frequency for filters
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static int split_rxtx; // Are we in split rx/tx mode?
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static int kill_audio; // Replace radio sound with silence
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static int quisk_transmit_mode; // Are we in transmit mode for semi-break-in CW?
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static int fft_sample_rate; // Sample rate on the graph (not the audio channel) -fft_srate/2 < freq < +fft_srate/2
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static int scan_blocks=0; // Number of FFT blocks for scan; or zero
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static int scan_sample_rate=1; // Sample rate to use while scanning
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static double scan_valid=0.84; // Fraction of each FFT block that is valid
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static int scan_vfo0;
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static int scan_deltaf;
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static int graph_refresh; // Graph refresh rate from the config file
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static int multiple_sample_rates=0; // Hardware supports multiple different sample rates
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static int vfo_screen; // The VFO (center) frequency on the FFT screen
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static int vfo_audio; // VFO frequency for the audio channel
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static int is_PTT_down; // state 0/1 of PTT button
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static int sample_bytes=3; // number of bytes in each I or Q sample
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static complex double PySampleBuf[SAMP_BUFFER_SIZE]; // buffer for samples returned from Python
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static int PySampleCount; // count of samples in buffer
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static int multirx_data_width; // width of graph data to return
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static int multirx_fft_width; // size of FFT samples
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int quisk_multirx_count; // number of additional receivers zero or 1, 2, 3, ...
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static int quisk_multirx_state; // state of hermes receivers
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static mrx_fft_data multirx_fft_data[QUISK_MAX_RECEIVERS]; // FFT data for the sub-receivers
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static int multirx_fft_next_index; // index of the receiver for the next FFT to return
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static double multirx_fft_next_time; // timing interval for multirx FFT
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static int multirx_fft_next_state; // state of multirx FFT: 0 == filling, 1 == ready, 2 == done
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static fftw_plan multirx_fft_next_plan; // fftw3 plan for multirx FFTs
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static fftw_complex * multirx_fft_next_samples; // sample buffer for multirx FFT
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static int multirx_play_method; // 0== both, 1==left, 2==right
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static int multirx_play_channel = -1; // index of the channel to play; or -1
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static int multirx_freq[QUISK_MAX_RECEIVERS]; // tune frequency for channel
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static int multirx_mode[QUISK_MAX_RECEIVERS]; // mode CW, SSB, etc. for channel
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static complex double * multirx_cSamples[QUISK_MAX_RECEIVERS]; // samples for the sub-receivers
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static int multirx_sample_size; // current size of the sub-receiver sample array
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static double sidetoneVolume; // Audio output level of the CW sidetone, 0.0 to 1.0
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int quiskKeyupDelay=0; // key-up delay from config file
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static int keyupDelayCode; // Play silence after sidetone ends, milliseconds
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static complex double sidetonePhase; // Phase increment for sidetone
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int quisk_sidetoneCtrl; // sidetone control value 0 to 1000
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static double agcReleaseGain=80; // AGC maximum gain
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static double agc_release_time = 1.0; // Release time in seconds
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static double squelch_level=-999.0; // setting of FM squelch control
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static int ssb_squelch_enabled;
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static int ssb_squelch_level;
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static int quisk_invert_spectrum = 0; // Invert the input RF spectrum
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static void process_agc(struct AgcState *, complex double *, int, int);
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static double Smeter; // Measured RMS signal strength
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static int rx_tune_freq; // Receive tuning frequency as +/- sample_rate / 2, including RIT
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int quisk_tx_tune_freq; // Transmit tuning frequency as +/- sample_rate / 2
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static int rit_freq; // RIT frequency in Hertz
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#define RX_UDP_SIZE 1442 // Expected size of UDP samples packet
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static SOCKET rx_udp_socket = INVALID_SOCKET; // Socket for receiving ADC samples from UDP
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int quisk_rx_udp_started = 0; // Have we received any data yet?
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int quisk_using_udp = 0; // Are we using rx_udp_socket? No longer used, but provided for backward compatibility.
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static double rx_udp_gain_correct = 0; // Small correction for different decimation rates
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static double rx_udp_clock; // Clock frequency for UDP samples
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int quisk_use_rx_udp; // from the config file
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static int is_little_endian; // Test byte order; is it little-endian?
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unsigned char quisk_pc_to_hermes[17 * 4]; // data to send from PC to Hermes hardware
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unsigned char quisk_hermeslite_writequeue[4 * 5]; // One-time writes to Hermes-Lite
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unsigned int quisk_hermeslite_writepointer = 0;
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unsigned int quisk_hermeslite_writeattempts = 0;
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static unsigned char quisk_hermes_to_pc[5 * 4]; // data received from the Hermes hardware
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static unsigned char quisk_hermeslite_response[5]; // response from Hermes-Lite commands
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unsigned int quisk_hermes_code_version = -1; // code version returned by the Hermes hardware
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unsigned int quisk_hermes_board_id = -1; // board ID returned by the Hermes hardware
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static double hermes_temperature; // average temperature
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static double hermes_fwd_power; // average forward power
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static double hermes_rev_power; // average reverse power
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static double hermes_pa_current; // average power amp current
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static int hermes_count_temperature; // number of temperature samples
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static int hermes_count_current; // number of current samples
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static int hardware_ptt = -1; // hardware PTT switch
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static int hardware_cwkey = -1; // hardware CW key
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enum quisk_rec_state quisk_record_state = IDLE;
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static float * quisk_record_buffer;
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static int quisk_record_bufsize;
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static int quisk_record_index;
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static int quisk_play_index;
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static int quisk_mic_index;
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static int quisk_record_full;
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// These are used to measure the frequency of a continuous RF signal.
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static void measure_freq(complex double *, int, int);
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static double measured_frequency;
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static int measure_freq_mode=0;
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//These are used to measure the demodulated audio voltage
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static double measured_audio;
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static double measure_audio_sum;
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static int measure_audio_count;
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static int measure_audio_time=1;
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// This is used to measure the squelch level
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static struct _MeasureSquelch {
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int squelch_active;
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// These are used for FM squelch
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double rf_sum;
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double squelch;
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int rf_count;
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// These are used for SSB squelch
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double * in_fft;
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int index;
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int sq_open;
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} MeasureSquelch[MAX_RX_CHANNELS];
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// These are used for playback of a WAV file.
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static int wavStart; // Sound data starts at this offset
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// Two wavFp are needed because the same file is used twice on asynchronous streams.
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static FILE * wavFpSound; // File pointer for audio WAV file input
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static FILE * wavFpMic; // File pointer for microphone WAV file input
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// These are used for bandscope data from Hermes
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static int enable_bandscope = 1;
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static fftw_plan bandscopePlan=NULL;
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static int bandscopeState = 0;
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static int bandscopeBlockCount = 4;
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static int bandscope_size = 0;
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static double * bandscopeSamples = NULL; // bandscope samples are normalized to max 1.0 with bandscopeScale
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static double bandscopeScale = 32768; // maximum value of the samples sent to the bandscope
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static double * bandscopeWindow = NULL;
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static double * bandscopeAverage = NULL;
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static complex double * bandscopeFFT = NULL;
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static double hermes_adc_level = 0.0; // maximum bandscope sample from the ADC, 0.0 to 1.0
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#if SAMPLES_FROM_FILE
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static struct QuiskWav hWav;
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int QuiskWavWriteOpen(struct QuiskWav * hWav, char * file_name, short format, short nChan, short bytes, int rate, double scale)
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{
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unsigned int u; // must be 4 bytes
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unsigned short s; // must be 2 bytes
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hWav->format = format;
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hWav->nChan = nChan;
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hWav->bytes_per_sample = bytes;
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hWav->sample_rate = rate;
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hWav->scale = scale;
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hWav->samples = 0; // number of samples written
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hWav->fp = fopen(file_name, "wb");
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if ( ! hWav->fp)
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return 0;
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if (format == 0) // RAW format - no header
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return 1;
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if (fwrite("RIFF", 1, 4, hWav->fp) != 4) {
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fclose(hWav->fp);
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hWav->fp = NULL;
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return 0;
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}
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if (format == 1) // PCM
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u = 36;
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else
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u = 50;
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fwrite(&u, 4, 1, hWav->fp);
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fwrite("WAVE", 1, 4, hWav->fp);
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fwrite("fmt ", 1, 4, hWav->fp);
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if (format == 1) // PCM
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u = 16;
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else
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u = 18;
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fwrite(&u, 4, 1, hWav->fp);
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fwrite(&format, 2, 1, hWav->fp); // format
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fwrite(&nChan, 2, 1, hWav->fp); // number of channels
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fwrite(&rate, 4, 1, hWav->fp); // sample rate
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u = rate * bytes * nChan;
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fwrite(&u, 4, 1, hWav->fp);
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s = bytes * nChan;
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fwrite(&s, 2, 1, hWav->fp);
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s = bytes * 8;
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fwrite(&s, 2, 1, hWav->fp);
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if (format != 1) {
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s = 0;
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fwrite(&s, 2, 1, hWav->fp);
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fwrite("fact", 1, 4, hWav->fp);
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u = 4;
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fwrite(&u, 4, 1, hWav->fp);
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u = 0;
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fwrite(&u, 4, 1, hWav->fp);
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}
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fwrite("data", 1, 4, hWav->fp);
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u = 0;
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fwrite(&u, 4, 1, hWav->fp);
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return 1;
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}
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void QuiskWavWriteC(struct QuiskWav * hWav, complex double * cSamples, int nSamples)
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{ // Record the samples to a WAV file, two float samples I/Q. Always use IEEE format 3.
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int j; // TODO: add other formats
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float samp; // must be 4 bytes
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if ( ! hWav->fp)
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return;
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// append the samples
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hWav->samples += (unsigned int)nSamples;
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fseek(hWav->fp, 0, SEEK_END); // seek to the end
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for (j = 0; j < nSamples; j++) {
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samp = creal(cSamples[j]) * hWav->scale;
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fwrite(&samp, 4, 1, hWav->fp);
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samp = cimag(cSamples[j]) * hWav->scale;
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fwrite(&samp, 4, 1, hWav->fp);
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}
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// write the sizes to the header
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QuiskWavWriteD(hWav, NULL, 0);
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}
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void QuiskWavWriteD(struct QuiskWav * hWav, double * dSamples, int nSamples)
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{ // Record the samples to a file, one channel.
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int j;
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float samp; // must be 4 bytes
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unsigned int u; // must be 4 bytes
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int i; // must be 4 bytes
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char c; // must be 1 byte
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short s; // must be 2 bytes
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if ( ! hWav->fp)
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return;
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// append the samples
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hWav->samples += (unsigned int)nSamples;
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fseek(hWav->fp, 0, SEEK_END); // seek to the end
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if ( ! dSamples) {
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; // Only update the header
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}
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else if (hWav->format == 3) { // float
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for (j = 0; j < nSamples; j++) {
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samp = dSamples[j] * hWav->scale;
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fwrite(&samp, 4, 1, hWav->fp);
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}
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}
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else { // PCM integer
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switch (hWav->bytes_per_sample) {
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case 1:
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for (j = 0; j < nSamples; j++) {
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c = (char)(dSamples[j] * hWav->scale);
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fwrite(&c, 1, 1, hWav->fp);
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}
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break;
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case 2:
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for (j = 0; j < nSamples; j++) {
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s = (short)(dSamples[j] * hWav->scale);
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fwrite(&s, 2, 1, hWav->fp);
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}
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break;
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case 4:
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for (j = 0; j < nSamples; j++) {
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i = (int)(dSamples[j] * hWav->scale);
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fwrite(&i, 4, 1, hWav->fp);
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}
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break;
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}
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}
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// write the sizes to the header
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if (hWav->format == 0) { // RAW format
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;
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}
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else if (hWav->format == 3) { // float
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fseek(hWav->fp, 54, SEEK_SET); // seek from the beginning
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u = hWav->bytes_per_sample * hWav->nChan * hWav->samples;
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fwrite(&u, 4, 1, hWav->fp);
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fseek(hWav->fp, 4, SEEK_SET);
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u += 50 ;
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fwrite(&u, 4, 1, hWav->fp);
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fseek(hWav->fp, 46, SEEK_SET);
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u = hWav->samples * hWav->nChan;
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fwrite(&u, 4, 1, hWav->fp);
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}
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else {
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fseek(hWav->fp, 40, SEEK_SET);
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u = hWav->bytes_per_sample * hWav->nChan * hWav->samples;
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fwrite(&u, 4, 1, hWav->fp);
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fseek(hWav->fp, 4, SEEK_SET);
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u += 36 ;
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fwrite(&u, 4, 1, hWav->fp);
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}
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if (hWav->samples > 536870000) // 2**32 / 8
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QuiskWavClose(hWav);
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}
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int QuiskWavReadOpen(struct QuiskWav * hWav, char * file_name, short format, short nChan, short bytes, int rate, double scale)
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{ // TODO: Get parameters from the WAV file header.
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char name[5];
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int size;
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hWav->format = format;
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hWav->nChan = nChan;
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hWav->bytes_per_sample = bytes;
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hWav->sample_rate = rate;
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hWav->scale = scale;
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hWav->fp = fopen(file_name, "rb");
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if (!hWav->fp)
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return 0;
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if (hWav->format == 0) { // RAW format
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fseek(hWav->fp, 0, SEEK_END); // seek to the end
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hWav->fpEnd = ftell(hWav->fp);
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hWav->fpStart = hWav->fpPos = 0;
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return 1;
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}
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hWav->fpEnd = 0;
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while (1) { // WAV format
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if (fread (name, 4, 1, hWav->fp) != 1)
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return 0;
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if (fread (&size, 4, 1, hWav->fp) != 1)
|
|
return 0;
|
|
name[4] = 0;
|
|
//printf("name %s size %d\n", name, size);
|
|
if (!strncmp(name, "RIFF", 4))
|
|
fseek (hWav->fp, 4, SEEK_CUR); // Skip "WAVE"
|
|
else if (!strncmp(name, "data", 4)) { // sound data starts here
|
|
hWav->fpStart = ftell(hWav->fp);
|
|
hWav->fpEnd = hWav->fpStart + size;
|
|
hWav->fpPos = hWav->fpStart;
|
|
break;
|
|
}
|
|
else // Skip other records
|
|
fseek (hWav->fp, size, SEEK_CUR);
|
|
}
|
|
//printf("start %d end %d\n", hWav->fpStart, hWav->fpEnd);
|
|
if (!hWav->fpEnd) { // Failure to find "data" record
|
|
fclose(hWav->fp);
|
|
hWav->fp = NULL;
|
|
return 0;
|
|
}
|
|
return 1;
|
|
}
|
|
|
|
void QuiskWavReadC(struct QuiskWav * hWav, complex double * cSamples, int nSamples)
|
|
{ // Always uses format 3. TODO: add other formats.
|
|
int i;
|
|
float fi, fq;
|
|
double di, dq;
|
|
|
|
#if 0
|
|
double noise;
|
|
noise = 1.6E6;
|
|
for (i = 0; i < nSamples; i++) {
|
|
di = ((float)random() / RAND_MAX - 0.5) * noise;
|
|
dq = ((float)random() / RAND_MAX - 0.5) * noise;
|
|
cSamples[i] = di + I * dq;
|
|
}
|
|
#endif
|
|
if (hWav->fp && nSamples > 0) {
|
|
fseek (hWav->fp, hWav->fpPos, SEEK_SET);
|
|
for (i = 0; i < nSamples; i++) {
|
|
if (fread(&fi, 4, 1, hWav->fp) != 1)
|
|
break;
|
|
if (fread(&fq, 4, 1, hWav->fp) != 1)
|
|
break;
|
|
di = fi * hWav->scale;
|
|
dq = fq * hWav->scale;
|
|
cSamples[i] += di + I * dq;
|
|
hWav->fpPos += hWav->bytes_per_sample * hWav->nChan;
|
|
if (hWav->fpPos >= hWav->fpEnd)
|
|
hWav->fpPos = hWav->fpStart;
|
|
}
|
|
}
|
|
}
|
|
|
|
void QuiskWavReadD(struct QuiskWav * hWav, double * dSamples, int nSamples)
|
|
{
|
|
int j;
|
|
float samp;
|
|
int i; // must be 4 bytes
|
|
char c; // must be 1 byte
|
|
short s; // must be 2 bytes
|
|
|
|
if (hWav->fp && nSamples > 0) {
|
|
fseek (hWav->fp, hWav->fpPos, SEEK_SET);
|
|
for (j = 0; j < nSamples; j++) {
|
|
if (hWav->format == 3) { // float
|
|
if (fread(&samp, 4, 1, hWav->fp) != 1)
|
|
return;
|
|
}
|
|
else { // PCM integer
|
|
switch (hWav->bytes_per_sample) {
|
|
case 1:
|
|
if (fread(&c, 1, 1, hWav->fp) != 1)
|
|
return;
|
|
samp = c;
|
|
break;
|
|
case 2:
|
|
if (fread(&s, 2, 1, hWav->fp) != 1)
|
|
return;
|
|
samp = s;
|
|
break;
|
|
case 4:
|
|
if (fread(&i, 4, 1, hWav->fp) != 1)
|
|
return;
|
|
samp = i;
|
|
break;
|
|
}
|
|
}
|
|
dSamples[j] = samp * hWav->scale;
|
|
hWav->fpPos += hWav->bytes_per_sample * hWav->nChan;
|
|
if (hWav->fpPos >= hWav->fpEnd)
|
|
hWav->fpPos = hWav->fpStart;
|
|
}
|
|
}
|
|
}
|
|
|
|
void QuiskWavClose(struct QuiskWav * hWav)
|
|
{
|
|
if (hWav->fp) {
|
|
fclose(hWav->fp);
|
|
hWav->fp = NULL;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
// These are used for digital voice codecs
|
|
ty_dvoice_codec_rx pt_quisk_freedv_rx;
|
|
ty_dvoice_codec_tx pt_quisk_freedv_tx;
|
|
|
|
void quisk_dvoice_freedv(ty_dvoice_codec_rx rx, ty_dvoice_codec_tx tx)
|
|
{
|
|
pt_quisk_freedv_rx = rx;
|
|
pt_quisk_freedv_tx = tx;
|
|
}
|
|
#if 0
|
|
static int fFracDecim(double * dSamples, int nSamples, double fdecim)
|
|
{ // fractional decimation by fdecim > 1.0
|
|
int i, nout;
|
|
double xm0, xm1, xm2, xm3;
|
|
static double dindex = 1;
|
|
static double y0=0, y1=0, y2=0, y3=0;
|
|
static int in=0, out=0;
|
|
|
|
in += nSamples;
|
|
nout = 0;
|
|
for (i = 0; i < nSamples; i++) {
|
|
y3 = dSamples[i];
|
|
if (dindex < 1 || dindex >= 2.4)
|
|
printf ("dindex %.5f fdecim %.8f\n", dindex, fdecim);
|
|
if (dindex < 2) {
|
|
#if 0
|
|
dSamples[nout++] = (1 - (dindex - 1)) * y1 + (dindex - 1) * y2;
|
|
#else
|
|
xm0 = dindex - 0;
|
|
xm1 = dindex - 1;
|
|
xm2 = dindex - 2;
|
|
xm3 = dindex - 3;
|
|
dSamples[nout++] = xm1 * xm2 * xm3 * y0 / -6.0 + xm0 * xm2 * xm3 * y1 / 2.0 +
|
|
xm0 * xm1 * xm3 * y2 / -2.0 + xm0 * xm1 * xm2 * y3 / 6.0;
|
|
#endif
|
|
out++;
|
|
dindex += fdecim - 1;
|
|
y0 = y1;
|
|
y1 = y2;
|
|
y2 = y3;
|
|
}
|
|
else {
|
|
if (dindex > 2.5) printf ("Skip at %.2f\n", dindex);
|
|
y0 = y1;
|
|
y1 = y2;
|
|
y2 = y3;
|
|
dindex -= 1;
|
|
}
|
|
}
|
|
//printf ("in %d out %d\n", in, out);
|
|
return nout;
|
|
}
|
|
#endif
|
|
static int cFracDecim(complex double * cSamples, int nSamples, double fdecim)
|
|
{
|
|
// Fractional decimation of I/Q signals works poorly because it introduces aliases and birdies.
|
|
int i, nout;
|
|
double xm0, xm1, xm2, xm3;
|
|
static double dindex = 1;
|
|
static complex double c0=0, c1=0, c2=0, c3=0;
|
|
static int in=0, out=0;
|
|
|
|
in += nSamples;
|
|
nout = 0;
|
|
for (i = 0; i < nSamples; i++) {
|
|
c3 = cSamples[i];
|
|
if (dindex < 1 || dindex >= 2.4)
|
|
printf ("dindex %.5f fdecim %.8f\n", dindex, fdecim);
|
|
if (dindex < 2) {
|
|
#if 0
|
|
cSamples[nout++] = (1 - (dindex - 1)) * c1 + (dindex - 1) * c2;
|
|
#else
|
|
xm0 = dindex - 0;
|
|
xm1 = dindex - 1;
|
|
xm2 = dindex - 2;
|
|
xm3 = dindex - 3;
|
|
cSamples[nout++] =
|
|
(xm1 * xm2 * xm3 * c0 / -6.0 + xm0 * xm2 * xm3 * c1 / 2.0 +
|
|
xm0 * xm1 * xm3 * c2 / -2.0 + xm0 * xm1 * xm2 * c3 / 6.0);
|
|
#endif
|
|
out++;
|
|
dindex += fdecim - 1;
|
|
c0 = c1;
|
|
c1 = c2;
|
|
c2 = c3;
|
|
}
|
|
else {
|
|
if (dindex > 2.5) printf ("Skip at %.2f\n", dindex);
|
|
c0 = c1;
|
|
c1 = c2;
|
|
c2 = c3;
|
|
dindex -= 1;
|
|
}
|
|
}
|
|
//printf ("in %d out %d\n", in, out);
|
|
return nout;
|
|
}
|
|
|
|
#define QUISK_NB_HWINDOW_SECS 500.E-6 // half-size of blanking window in seconds
|
|
static void NoiseBlanker(complex double * cSamples, int nSamples)
|
|
{
|
|
static complex double * cSaved = NULL;
|
|
static double * dSaved = NULL;
|
|
static double save_sum;
|
|
static int save_size, hwindow_size, state, index, win_index;
|
|
static int sample_rate = -1;
|
|
int i, j, k, is_pulse;
|
|
double mag, limit;
|
|
complex double samp;
|
|
#if DEBUG
|
|
static time_t time0 = 0;
|
|
static int debug_count = 0;
|
|
#endif
|
|
|
|
if (quisk_noise_blanker <= 0)
|
|
return;
|
|
if (quisk_sound_state.sample_rate != sample_rate) { // Initialization
|
|
sample_rate = quisk_sound_state.sample_rate;
|
|
state = 0;
|
|
index = 0;
|
|
win_index = 0;
|
|
save_sum = 0.0;
|
|
hwindow_size = (int)(sample_rate * QUISK_NB_HWINDOW_SECS + 0.5);
|
|
save_size = hwindow_size * 3; // number of samples in the average
|
|
i = save_size * sizeof(double);
|
|
dSaved = (double *) realloc(dSaved, i);
|
|
memset (dSaved, 0, i);
|
|
i = save_size * sizeof(complex double);
|
|
cSaved = (complex double *)realloc(cSaved, i);
|
|
memset (cSaved, 0, i);
|
|
#if DEBUG
|
|
printf ("Noise blanker: save_size %d hwindow_size %d\n",
|
|
save_size, hwindow_size);
|
|
#endif
|
|
}
|
|
switch(quisk_noise_blanker) {
|
|
case 1:
|
|
default:
|
|
limit = 6.0;
|
|
break;
|
|
case 2:
|
|
limit = 4.0;
|
|
break;
|
|
case 3:
|
|
limit = 2.5;
|
|
break;
|
|
}
|
|
for (i = 0; i < nSamples; i++) {
|
|
// output oldest sample, save newest
|
|
samp = cSamples[i]; // newest sample
|
|
cSamples[i] = cSaved[index]; // oldest sample
|
|
cSaved[index] = samp;
|
|
// use newest sample
|
|
mag = cabs(samp);
|
|
save_sum -= dSaved[index]; // remove oldest sample magnitude
|
|
dSaved[index] = mag; // save newest sample magnitude
|
|
save_sum += mag; // update sum of samples
|
|
if (mag <= save_sum / save_size * limit) // see if we have a large pulse
|
|
is_pulse = 0;
|
|
else
|
|
is_pulse = 1;
|
|
switch (state) {
|
|
case 0: // Normal state
|
|
if (is_pulse) { // wait for a pulse
|
|
state = 1;
|
|
k = index;
|
|
for (j = 0; j < hwindow_size; j++) { // apply window to prior samples
|
|
cSaved[k--] *= (double)j / hwindow_size;
|
|
if (k < 0)
|
|
k = save_size - 1;
|
|
}
|
|
}
|
|
else if (win_index) { // pulses have stopped, increase window to 1.0
|
|
cSaved[index] *= (double)win_index / hwindow_size;
|
|
if (++win_index >= hwindow_size)
|
|
win_index = 0; // no more window
|
|
}
|
|
break;
|
|
case 1: // we got a pulse
|
|
cSaved[index] = 0; // zero samples until the pulses stop
|
|
if ( ! is_pulse) {
|
|
// start raising the window, but be prepared to window another pulse
|
|
state = 0;
|
|
win_index = 1;
|
|
}
|
|
break;
|
|
}
|
|
#if DEBUG
|
|
if (debug_count) {
|
|
printf ("%d", is_pulse);
|
|
if (--debug_count == 0)
|
|
printf ("\n");
|
|
}
|
|
else if (is_pulse && time(NULL) != time0) {
|
|
time0 = time(NULL);
|
|
debug_count = hwindow_size * 2;
|
|
printf ("%d", is_pulse);
|
|
}
|
|
#endif
|
|
if (++index >= save_size)
|
|
index = 0;
|
|
}
|
|
return;
|
|
}
|
|
|
|
#define NOTCH_DEBUG 0
|
|
#define NOTCH_DATA_SIZE 2048
|
|
#define NOTCH_FILTER_DESIGN_SIZE NOTCH_DATA_SIZE / 4
|
|
#define NOTCH_FILTER_SIZE (NOTCH_FILTER_DESIGN_SIZE - 1)
|
|
#define NOTCH_FILTER_FFT_SIZE (NOTCH_FILTER_SIZE / 2 + 1)
|
|
#define NOTCH_DATA_START_SIZE (NOTCH_FILTER_SIZE - 1)
|
|
#define NOTCH_DATA_OUTPUT_SIZE (NOTCH_DATA_SIZE - NOTCH_DATA_START_SIZE)
|
|
#define NOTCH_FFT_SIZE (NOTCH_DATA_SIZE / 2 + 1)
|
|
static void dAutoNotch(double * dsamples, int nSamples, int sidetone, int rate)
|
|
{
|
|
int i, j, k, i1, i2, inp, signal, delta_sig, delta_i1, half_width;
|
|
double d, d1, d2, avg;
|
|
static int old1, count1, old2, count2;
|
|
static int index;
|
|
static fftw_plan planFwd=NULL;
|
|
static fftw_plan planRev,fltrFwd, fltrRev;
|
|
static double data_in[NOTCH_DATA_SIZE];
|
|
static double data_out[NOTCH_DATA_SIZE];
|
|
static complex double notch_fft[NOTCH_FFT_SIZE];
|
|
static double fft_window[NOTCH_DATA_SIZE];
|
|
static double fltr_in[NOTCH_DATA_SIZE];
|
|
static double fltr_out[NOTCH_FILTER_DESIGN_SIZE];
|
|
static complex double fltr_fft[NOTCH_FFT_SIZE];
|
|
static double average_fft[NOTCH_FFT_SIZE];
|
|
static int fltrSig;
|
|
#if NOTCH_DEBUG
|
|
static char * txt;
|
|
double dmax;
|
|
#endif
|
|
|
|
if ( ! planFwd) { // set up FFT plans
|
|
planFwd = fftw_plan_dft_r2c_1d(NOTCH_DATA_SIZE, data_in, notch_fft, FFTW_MEASURE);
|
|
planRev = fftw_plan_dft_c2r_1d(NOTCH_DATA_SIZE, notch_fft, data_out, FFTW_MEASURE); // destroys notch_fft
|
|
fltrFwd = fftw_plan_dft_r2c_1d(NOTCH_DATA_SIZE, fltr_in, fltr_fft, FFTW_MEASURE);
|
|
fltrRev = fftw_plan_dft_c2r_1d(NOTCH_FILTER_DESIGN_SIZE, fltr_fft, fltr_out, FFTW_MEASURE);
|
|
for (i = 0; i < NOTCH_FILTER_SIZE; i++)
|
|
fft_window[i] = 0.50 - 0.50 * cos(2. * M_PI * i / (NOTCH_FILTER_SIZE)); // Hanning
|
|
//fft_window[i] = 0.54 - 0.46 * cos(2. * M_PI * i / (NOTCH_FILTER_SIZE)); // Hamming
|
|
}
|
|
if ( ! dsamples) { // initialize
|
|
index = NOTCH_DATA_START_SIZE;
|
|
fltrSig = -1;
|
|
old1 = old2 = 0;
|
|
count1 = count2 = -4;
|
|
memset(data_out, 0, sizeof(double) * NOTCH_DATA_SIZE);
|
|
memset(data_in, 0, sizeof(double) * NOTCH_DATA_SIZE);
|
|
memset(average_fft, 0, sizeof(double) * NOTCH_FFT_SIZE);
|
|
return;
|
|
}
|
|
if ( ! quisk_auto_notch)
|
|
return;
|
|
// index into FFT data = frequency * 2 * NOTCH_FFT_SIZE / rate
|
|
// index into filter design = frequency * 2 * NOTCH_FILTER_FFT_SIZE / rate
|
|
for (inp = 0; inp < nSamples; inp++) {
|
|
data_in[index] = dsamples[inp];
|
|
dsamples[inp] = data_out[index];
|
|
if (++index >= NOTCH_DATA_SIZE) { // we have a full FFT of samples
|
|
index = NOTCH_DATA_START_SIZE;
|
|
fftw_execute(planFwd); // Calculate forward FFT
|
|
// Find maximum FFT bins
|
|
delta_sig = (300 * 2 * NOTCH_FFT_SIZE + rate / 2) / rate; // small frequency interval
|
|
delta_i1 = (400 * 2 * NOTCH_FFT_SIZE + rate / 2) / rate; // small frequency interval
|
|
if (sidetone != 0) // For CW, accept a signal at the frequency of the RIT
|
|
signal = (abs(sidetone) * 2 * NOTCH_FFT_SIZE + rate / 2) / rate;
|
|
else
|
|
signal = -999;
|
|
avg = 1;
|
|
#if NOTCH_DEBUG
|
|
dmax = 0;
|
|
#endif
|
|
d1 = 0;
|
|
i1 = 0; // First maximum signal
|
|
for (i = 0; i < NOTCH_FFT_SIZE; i++) {
|
|
d = cabs(notch_fft[i]);
|
|
avg += d;
|
|
//average_fft[i] = 0.9 * average_fft[i] + 0.1 * d;
|
|
average_fft[i] = 0.5 * average_fft[i] + 0.5 * d;
|
|
if (abs(i - signal) > delta_sig && average_fft[i] > d1) {
|
|
d1 = average_fft[i];
|
|
i1 = i;
|
|
#if NOTCH_DEBUG
|
|
dmax = d;
|
|
#endif
|
|
}
|
|
}
|
|
if (abs(i1 - old1) < 3) // See if the maximum bin i1 is changing
|
|
count1++;
|
|
else
|
|
count1--;
|
|
if (count1 > 4)
|
|
count1 = 4;
|
|
else if (count1 < -1)
|
|
count1 = -1;
|
|
if (count1 < 0)
|
|
old1 = i1;
|
|
avg /= NOTCH_FFT_SIZE;
|
|
d2 = 0;
|
|
i2 = 0; // Next maximum signal not near the first
|
|
for (i = 0; i < NOTCH_FFT_SIZE; i++) {
|
|
if (abs(i - signal) > delta_sig && abs(i - i1) > delta_i1 && average_fft[i] > d2) {
|
|
d2 = average_fft[i];
|
|
i2 = i;
|
|
}
|
|
}
|
|
if (abs(i2 - old2) < 3) // See if the maximum bin i2 is changing
|
|
count2++;
|
|
else
|
|
count2--;
|
|
if (count2 > 4)
|
|
count2 = 4;
|
|
else if (count2 < -2)
|
|
count2 = -2;
|
|
if (count2 < 0)
|
|
old2 = i2;
|
|
|
|
if (count1 > 0 && count2 > 0)
|
|
k = i1 + 10000 * i2; // trial filter index
|
|
else if(count1 > 0)
|
|
k = i1;
|
|
else
|
|
k = 0;
|
|
// Make the filter if it is different
|
|
if (fltrSig != k) {
|
|
fltrSig = k;
|
|
half_width = (100 * 2 * NOTCH_FILTER_FFT_SIZE + rate / 2) / rate; // half the width of the notch
|
|
if (half_width < 3)
|
|
half_width = 3;
|
|
for (i = 0; i < NOTCH_FILTER_FFT_SIZE; i++)
|
|
fltr_fft[i] = 1.0;
|
|
k = (i1 + 2) / 4; // Ratio of index values is 4
|
|
#if NOTCH_DEBUG
|
|
txt = "Fxx";
|
|
#endif
|
|
if (count1 > 0) {
|
|
#if NOTCH_DEBUG
|
|
txt = "F1";
|
|
#endif
|
|
for (i = -half_width; i <= half_width; i++) {
|
|
j = k + i;
|
|
if (j >= 0 && j < NOTCH_FILTER_FFT_SIZE)
|
|
fltr_fft[j] = 0.0;
|
|
}
|
|
}
|
|
k = (i2 + 2) / 4; // Ratio of index values is 4
|
|
if (count1 > 0 && count2 > 0) {
|
|
#if NOTCH_DEBUG
|
|
txt = "F12";
|
|
#endif
|
|
for (i = -half_width; i <= half_width; i++) {
|
|
j = k + i;
|
|
if (j >= 0 && j < NOTCH_FILTER_FFT_SIZE)
|
|
fltr_fft[j] = 0.0;
|
|
}
|
|
}
|
|
fftw_execute(fltrRev);
|
|
// center the coefficient zero, make the filter symetric, reduce the size by one
|
|
memmove(fltr_out + NOTCH_FILTER_DESIGN_SIZE / 2 - 1, fltr_out, sizeof(double) * (NOTCH_FILTER_SIZE / 2 - 1));
|
|
for (i = NOTCH_FILTER_DESIGN_SIZE / 2 - 2, j = NOTCH_FILTER_DESIGN_SIZE / 2; i >= 0; i--, j++)
|
|
fltr_out[i] = fltr_out[j];
|
|
for (i = 0; i < NOTCH_FILTER_SIZE; i++)
|
|
fltr_in[i] = fltr_out[i] * fft_window[i] / NOTCH_FILTER_DESIGN_SIZE;
|
|
for (i = NOTCH_FILTER_SIZE; i < NOTCH_DATA_SIZE; i++)
|
|
fltr_in[i] = 0.0;
|
|
fftw_execute(fltrFwd); // The filter is fltr_fft[]
|
|
}
|
|
#if NOTCH_DEBUG
|
|
printf("Max %12.0lf frequency index1 %3d %5d %12.0lf index2 %3d %5d %12.0lf avg %12.0lf %s\n", dmax, count1, i1, d1, count2, i2, d2, avg, txt);
|
|
#endif
|
|
for (i = 0; i < NOTCH_FFT_SIZE; i++) // Apply the filter
|
|
notch_fft[i] *= fltr_fft[i];
|
|
fftw_execute(planRev); // Calculate inverse FFT
|
|
memmove(data_in, data_in + NOTCH_DATA_OUTPUT_SIZE, NOTCH_DATA_START_SIZE * sizeof(double));
|
|
for (i = NOTCH_DATA_START_SIZE; i < NOTCH_DATA_SIZE; i++)
|
|
data_out[i] /= NOTCH_DATA_SIZE / 20; // Empirical
|
|
}
|
|
}
|
|
return;
|
|
}
|
|
|
|
static int audio_fft_ready=0;
|
|
static double * audio_average_fft;
|
|
void quisk_calc_audio_graph(double scale, complex double * csamples, double * dsamples, int nSamples, int real)
|
|
{ // Calculate an FFT for the audio data. Samples are either csamples or dsamples; the other is NULL.
|
|
// The "scale" is the 0 dB reference. If "real", use the real part of csamples.
|
|
int i, k, inp;
|
|
static int index;
|
|
static int count_fft;
|
|
static int audio_fft_size;
|
|
static int audio_fft_count;
|
|
static fftw_plan plan = NULL;
|
|
static double * fft_window;
|
|
static complex double * audio_fft;
|
|
|
|
if ( ! plan) { // malloc new space and initialize
|
|
index = 0;
|
|
count_fft = 0;
|
|
audio_fft_size = data_width;
|
|
//audio_fft_count = 48000 / audio_fft_size / 5; // Display refresh rate.
|
|
audio_fft_count = 8000 / audio_fft_size / 5; // Display refresh rate.
|
|
if (audio_fft_count <= 0)
|
|
audio_fft_count = 1;
|
|
fft_window = (double *)malloc(audio_fft_size * sizeof(double));
|
|
audio_average_fft = (double *)malloc(audio_fft_size * sizeof(double));
|
|
audio_fft = (complex double *)malloc(audio_fft_size * sizeof(complex double));
|
|
plan = fftw_plan_dft_1d(audio_fft_size, audio_fft, audio_fft, FFTW_FORWARD, FFTW_MEASURE);
|
|
for (i = 0; i < audio_fft_size; i++) {
|
|
audio_average_fft[i] = 0;
|
|
fft_window[i] = 0.50 - 0.50 * cos(2. * M_PI * i / audio_fft_size); // Hanning window loss 50%
|
|
}
|
|
return;
|
|
}
|
|
if (audio_fft_ready == 0) { // calculate a new audio FFT
|
|
if (dsamples || real) // Lyons 2Ed p61
|
|
scale *= audio_fft_size / 2.0;
|
|
else
|
|
scale *= audio_fft_size;
|
|
scale *= audio_fft_count;
|
|
scale *= 0.5; // correct for Hanning window loss
|
|
for (inp = 0; inp < nSamples; inp++) {
|
|
if (dsamples)
|
|
audio_fft[index] = dsamples[inp] / scale;
|
|
else if (real)
|
|
audio_fft[index] = creal(csamples[inp]) / scale;
|
|
else
|
|
audio_fft[index] = csamples[inp] / scale;
|
|
if (++index >= audio_fft_size) { // we have a full FFT of samples
|
|
index = 0;
|
|
for (i = 0; i < audio_fft_size; i++)
|
|
audio_fft[i] *= fft_window[i]; // multiply by window
|
|
fftw_execute(plan); // Calculate forward FFT
|
|
count_fft++;
|
|
k = 0;
|
|
for (i = audio_fft_size / 2; i < audio_fft_size; i++) // Negative frequencies
|
|
audio_average_fft[k++] += cabs(audio_fft[i]);
|
|
for (i = 0; i < audio_fft_size / 2; i++) // Positive frequencies
|
|
audio_average_fft[k++] += cabs(audio_fft[i]);
|
|
if (count_fft >= audio_fft_count) {
|
|
audio_fft_ready = 1;
|
|
count_fft = 0;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static PyObject * get_audio_graph(PyObject * self, PyObject * args)
|
|
{
|
|
int i;
|
|
double d2;
|
|
PyObject * tuple2;
|
|
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
|
|
if ( ! audio_fft_ready) { // a new graph is not yet available
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
tuple2 = PyTuple_New(data_width);
|
|
for (i = 0; i < data_width; i++) {
|
|
d2 = audio_average_fft[i];
|
|
if (d2 < 1E-10)
|
|
d2 = 1E-10;
|
|
d2 = 20.0 * log10(d2);
|
|
PyTuple_SetItem(tuple2, i, PyFloat_FromDouble(d2));
|
|
audio_average_fft[i] = 0;
|
|
}
|
|
audio_fft_ready = 0;
|
|
return tuple2;
|
|
}
|
|
|
|
static void d_delay(double * dsamples, int nSamples, int bank, int samp_delay)
|
|
{ // delay line (FIFO) to delay dsamples by samp_delay samples
|
|
int i;
|
|
double sample;
|
|
static struct {
|
|
double * buffer;
|
|
int index;
|
|
int buf_size;
|
|
} delay[MAX_RX_CHANNELS] = {{NULL, 0, 0}};
|
|
|
|
if ( ! delay[0].buffer)
|
|
for (i = 1; i < MAX_RX_CHANNELS; i++)
|
|
delay[i].buffer = NULL;
|
|
if ( ! delay[bank].buffer) {
|
|
delay[bank].buffer = (double *)malloc(samp_delay * sizeof(double));
|
|
delay[bank].index = 0;
|
|
delay[bank].buf_size = samp_delay;
|
|
for (i = 0; i < samp_delay; i++)
|
|
delay[bank].buffer[i] = 0;
|
|
}
|
|
for (i = 0; i < nSamples; i++) {
|
|
sample = delay[bank].buffer[delay[bank].index];
|
|
delay[bank].buffer[delay[bank].index] = dsamples[i];
|
|
dsamples[i] = sample;
|
|
if (++delay[bank].index >= delay[bank].buf_size)
|
|
delay[bank].index = 0;
|
|
}
|
|
}
|
|
|
|
static void ssb_squelch(double * dsamples, int nSamples, int samp_rate, struct _MeasureSquelch * MS)
|
|
{
|
|
int i, bw, bw1, bw2, inp;
|
|
double d, arith_avg, geom_avg, ratio;
|
|
complex double c;
|
|
static fftw_plan plan = NULL;
|
|
static double * fft_window;
|
|
static complex double * out_fft;
|
|
#ifdef QUISK_PRINT_LEVELS
|
|
static int timer = 0;
|
|
timer += nSamples;
|
|
#endif
|
|
|
|
if ( ! MS->in_fft) {
|
|
MS->in_fft = (double *)fftw_malloc(SQUELCH_FFT_SIZE * sizeof(double));
|
|
MS->index = 0;
|
|
MS->sq_open = 0;
|
|
}
|
|
if ( ! plan) { // malloc new space and initialize
|
|
fft_window = (double *)malloc(SQUELCH_FFT_SIZE * sizeof(double));
|
|
out_fft = (complex double *)fftw_malloc((SQUELCH_FFT_SIZE / 2 + 1) * sizeof(complex double));
|
|
// out_fft[0] is DC, then positive frequencies, then out_fft[N/2] is Nyquist.
|
|
plan = fftw_plan_dft_r2c_1d(SQUELCH_FFT_SIZE, MS->in_fft, out_fft, FFTW_MEASURE);
|
|
for (i = 0; i < SQUELCH_FFT_SIZE; i++)
|
|
fft_window[i] = 0.50 - 0.50 * cos(2. * M_PI * i / SQUELCH_FFT_SIZE); // Hanning window
|
|
return;
|
|
}
|
|
for (inp = 0; inp < nSamples; inp++) {
|
|
MS->in_fft[MS->index++] = dsamples[inp];
|
|
if (MS->index >= SQUELCH_FFT_SIZE) { // we have a full FFT of samples
|
|
MS->index = 0;
|
|
for (i = 0; i < SQUELCH_FFT_SIZE; i++)
|
|
MS->in_fft[i] *= fft_window[i]; // multiply by window
|
|
fftw_execute_dft_r2c(plan, MS->in_fft, out_fft); // Calculate forward FFT
|
|
bw = filter_bandwidth[0]; // Calculate the FFT bins within the filter bandwidth
|
|
if (bw > 3000)
|
|
bw = 3000;
|
|
bw1 = 300 * SQUELCH_FFT_SIZE / samp_rate; // start 300 Hz
|
|
bw2 = (bw + 300) * SQUELCH_FFT_SIZE / samp_rate; // end 300 Hz + bw
|
|
arith_avg = 0.0;
|
|
geom_avg = 0.0;
|
|
for (i = bw1; i < bw2; i++) {
|
|
c = out_fft[i] / CLIP16;
|
|
d = creal(c) * creal(c) + cimag(c) * cimag(c);
|
|
if (d > 1E-4) {
|
|
arith_avg += d;
|
|
geom_avg += log(d);
|
|
}
|
|
}
|
|
#ifdef QUISK_PRINT_SPECTRUM
|
|
// Combine FFT into spectral bands
|
|
int j;
|
|
for (i = 0; i < 128; i += 16) {
|
|
d = 0;
|
|
for (j = i; j < i + 16; j++) {
|
|
c = out_fft[i] / CLIP16;
|
|
d += creal(c) * creal(c) + cimag(c) * cimag(c);
|
|
}
|
|
d = log(d / 16);
|
|
printf ("%12.3f", d);
|
|
if (i == 112)
|
|
printf("\n");
|
|
}
|
|
#endif
|
|
if (arith_avg > 1E-4) {
|
|
bw = bw2 - bw1;
|
|
arith_avg = log(arith_avg / bw);
|
|
geom_avg /= bw;
|
|
ratio = arith_avg - geom_avg;
|
|
}
|
|
else {
|
|
ratio = 1.0;
|
|
}
|
|
// For band noise, ratio is 0.57
|
|
if (ratio > ssb_squelch_level * 0.005)
|
|
MS->sq_open = samp_rate; // one second timer
|
|
#ifdef QUISK_PRINT_LEVELS
|
|
if (timer >= samp_rate / 2) {
|
|
timer = 0;
|
|
printf ("squelch %6d A %6.3f G %6.3f A-G %6.3f\n",
|
|
MS->sq_open, arith_avg, geom_avg, ratio);
|
|
#ifdef QUISK_PRINT_SPECTRUM
|
|
for (i = 0; i < 128; i += 16)
|
|
printf ("%5d - %4d", i * samp_rate / SQUELCH_FFT_SIZE, (i + 16) * samp_rate / SQUELCH_FFT_SIZE);
|
|
printf ("\n");
|
|
#endif
|
|
}
|
|
#endif
|
|
}
|
|
}
|
|
MS->sq_open -= nSamples;
|
|
if (MS->sq_open < 0)
|
|
MS->sq_open = 0;
|
|
MS->squelch_active = MS->sq_open == 0;
|
|
}
|
|
|
|
static complex double dRxFilterOut(complex double sample, int bank, int nFilter)
|
|
{ // Rx FIR filter; bank is the static storage index, and must be different for different data streams.
|
|
// Multiple filters are at nFilter.
|
|
complex double cx;
|
|
int j, k;
|
|
static int init = 0;
|
|
static struct stStorage {
|
|
int indexFilter; // current index into sample buffer
|
|
complex double bufFilterC[MAX_FILTER_SIZE]; // Digital filter sample buffer
|
|
} Storage[MAX_RX_CHANNELS];
|
|
struct stStorage * ptBuf = Storage + bank;
|
|
double * filtI;
|
|
|
|
if ( ! init) {
|
|
init = 1;
|
|
for (j = 0; j < MAX_RX_CHANNELS; j++)
|
|
memset(Storage + j, 0, sizeof(struct stStorage));
|
|
}
|
|
|
|
if ( ! sizeFilter)
|
|
return sample;
|
|
if (ptBuf->indexFilter >= sizeFilter)
|
|
ptBuf->indexFilter = 0;
|
|
ptBuf->bufFilterC[ptBuf->indexFilter] = sample;
|
|
cx = 0;
|
|
filtI = cFilterI[nFilter];
|
|
j = ptBuf->indexFilter;
|
|
for (k = 0; k < sizeFilter; k++) {
|
|
cx += ptBuf->bufFilterC[j] * filtI[k];
|
|
if (++j >= sizeFilter)
|
|
j = 0;
|
|
}
|
|
ptBuf->indexFilter++;
|
|
return cx;
|
|
}
|
|
|
|
complex double cRxFilterOut(complex double sample, int bank, int nFilter)
|
|
{ // Rx FIR filter; bank is the static storage index, and must be different for different data streams.
|
|
// Multiple filters are at nFilter.
|
|
double accI, accQ;
|
|
double * filtI, * filtQ;
|
|
int j, k;
|
|
static int init = 0;
|
|
static struct stStorage {
|
|
int indexFilter; // current index into sample buffer
|
|
double bufFilterI[MAX_FILTER_SIZE]; // Digital filter sample buffer
|
|
double bufFilterQ[MAX_FILTER_SIZE]; // Digital filter sample buffer
|
|
} Storage[MAX_RX_CHANNELS];
|
|
struct stStorage * ptBuf = Storage + bank;
|
|
|
|
if ( ! init) {
|
|
init = 1;
|
|
for (j = 0; j < MAX_RX_CHANNELS; j++)
|
|
memset(Storage + j, 0, sizeof(struct stStorage));
|
|
}
|
|
|
|
if ( ! sizeFilter)
|
|
return sample;
|
|
if (ptBuf->indexFilter >= sizeFilter)
|
|
ptBuf->indexFilter = 0;
|
|
ptBuf->bufFilterI[ptBuf->indexFilter] = creal(sample);
|
|
ptBuf->bufFilterQ[ptBuf->indexFilter] = cimag(sample);
|
|
filtI = cFilterI[nFilter];
|
|
filtQ = cFilterQ[nFilter];
|
|
accI = accQ = 0;
|
|
j = ptBuf->indexFilter;
|
|
for (k = 0; k < sizeFilter; k++) {
|
|
accI += ptBuf->bufFilterI[j] * filtI[k];
|
|
accQ += ptBuf->bufFilterQ[j] * filtQ[k];
|
|
if (++j >= sizeFilter)
|
|
j = 0;
|
|
}
|
|
ptBuf->indexFilter++;
|
|
return accI + I * accQ;
|
|
}
|
|
|
|
static void AddTestTone(complex double * cSamples, int nSamples)
|
|
{
|
|
int i;
|
|
static complex double testtoneVector = 21474836.47; // -40 dB
|
|
static complex double audioVector = 1.0;
|
|
complex double audioPhase;
|
|
|
|
switch (rxMode) {
|
|
default:
|
|
//testtonePhase = cexp(I * 2 * M_PI * (quisk_sidetoneCtrl - 500) / 1000.0);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cSamples[i] += testtoneVector;
|
|
testtoneVector *= testtonePhase;
|
|
}
|
|
break;
|
|
case AM: // AM
|
|
//audioPhase = cexp(I * 2 * M_PI * quisk_sidetoneCtrl * 5 / sample_rate);
|
|
audioPhase = cexp(I * 2.0 * M_PI * 1000 / quisk_sound_state.sample_rate);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cSamples[i] += testtoneVector * (1.0 + creal(audioVector));
|
|
testtoneVector *= testtonePhase;
|
|
audioVector *= audioPhase;
|
|
}
|
|
break;
|
|
case FM: // FM
|
|
case DGT_FM:
|
|
//audioPhase = cexp(I * 2 * M_PI * quisk_sidetoneCtrl * 5 / sample_rate);
|
|
audioPhase = cexp(I * 2.0 * M_PI * 1000 / quisk_sound_state.sample_rate);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cSamples[i] += testtoneVector * cexp(I * creal(audioVector));
|
|
testtoneVector *= testtonePhase;
|
|
audioVector *= audioPhase;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
static int IsSquelch(int freq)
|
|
{ // measure the signal level for squelch
|
|
int i, i1, i2, iBandwidth;
|
|
double meter;
|
|
|
|
// This uses current_graph with width data_width
|
|
iBandwidth = 5000 * data_width / fft_sample_rate; // bandwidth determines number of pixels to average
|
|
if (iBandwidth < 1)
|
|
iBandwidth = 1;
|
|
i1 = (int)((double)freq * data_width / fft_sample_rate + data_width / 2.0 - iBandwidth / 2.0 + 0.5);
|
|
i2 = i1 + iBandwidth;
|
|
meter = 0;
|
|
if (i1 >= 0 && i2 < data_width) { // too close to edge?
|
|
for (i = i1; i < i2; i++)
|
|
meter += current_graph[i];
|
|
}
|
|
meter /= iBandwidth;
|
|
if (meter == 0 || meter < squelch_level)
|
|
return 1; // meter == 0 means Rx freq is off-screen so squelch is on
|
|
else
|
|
return 0;
|
|
}
|
|
|
|
static PyObject * set_record_state(PyObject * self, PyObject * args)
|
|
{ // called when a Record or Play button is pressed, or with -1 to poll
|
|
int button;
|
|
|
|
if (!PyArg_ParseTuple (args, "i", &button))
|
|
return NULL;
|
|
switch (button) {
|
|
case 0: // press record radio
|
|
case 4: // press record microphone
|
|
if ( ! quisk_record_buffer) { // initialize
|
|
quisk_record_bufsize = (int)(QuiskGetConfigDouble("max_record_minutes", 0.25) * quisk_sound_state.playback_rate * 60.0 + 0.2);
|
|
quisk_record_buffer = (float *)malloc(sizeof(float) * quisk_record_bufsize);
|
|
}
|
|
quisk_record_index = 0;
|
|
quisk_play_index = 0;
|
|
quisk_mic_index = 0;
|
|
quisk_record_full = 0;
|
|
if (button == 0)
|
|
quisk_record_state = RECORD_RADIO;
|
|
else
|
|
quisk_record_state = RECORD_MIC;
|
|
break;
|
|
case 1: // release record
|
|
quisk_record_state = IDLE;
|
|
break;
|
|
case 2: // press play
|
|
if (quisk_record_full) {
|
|
quisk_play_index = quisk_record_index + 1;
|
|
if (quisk_play_index >= quisk_record_bufsize)
|
|
quisk_play_index = 0;
|
|
}
|
|
else {
|
|
quisk_play_index = 0;
|
|
}
|
|
quisk_mic_index = quisk_play_index;
|
|
quisk_record_state = PLAYBACK;
|
|
break;
|
|
case 3: // release play
|
|
quisk_record_state = IDLE;
|
|
break;
|
|
case 5: // press play file
|
|
fseek (wavFpSound, wavStart, SEEK_SET);
|
|
fseek (wavFpMic, wavStart, SEEK_SET);
|
|
quisk_record_state = PLAY_FILE;
|
|
break;
|
|
case 6: // press play samples file
|
|
fseek (wavFpSound, wavStart, SEEK_SET);
|
|
quisk_record_state = PLAY_SAMPLES;
|
|
break;
|
|
}
|
|
return PyInt_FromLong(quisk_record_state != PLAYBACK && quisk_record_state != PLAY_FILE && quisk_record_state != PLAY_SAMPLES);
|
|
}
|
|
|
|
void quisk_tmp_record(complex double * cSamples, int nSamples, double scale) // save sound
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < nSamples; i++) {
|
|
quisk_record_buffer[quisk_record_index++] = creal(cSamples[i]) * scale;
|
|
if (quisk_record_index >= quisk_record_bufsize) {
|
|
quisk_record_index = 0;
|
|
quisk_record_full = 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
void quisk_tmp_playback(complex double * cSamples, int nSamples, double volume)
|
|
{ // replace radio sound with saved sound
|
|
int i;
|
|
double d;
|
|
|
|
for (i = 0; i < nSamples; i++) {
|
|
d = quisk_record_buffer[quisk_play_index++] * volume;
|
|
cSamples[i] = d + I * d;
|
|
if (quisk_play_index >= quisk_record_bufsize)
|
|
quisk_play_index = 0;
|
|
if (quisk_play_index == quisk_record_index) {
|
|
quisk_record_state = IDLE;
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void quisk_tmp_microphone(complex double * cSamples, int nSamples)
|
|
{ // replace microphone samples with saved sound
|
|
int i;
|
|
double d;
|
|
|
|
for (i = 0; i < nSamples; i++) {
|
|
d = quisk_record_buffer[quisk_mic_index++];
|
|
cSamples[i] = d + I * d;
|
|
if (quisk_mic_index >= quisk_record_bufsize)
|
|
quisk_mic_index = 0;
|
|
if (quisk_mic_index == quisk_record_index) {
|
|
quisk_record_state = IDLE;
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
static PyObject * open_wav_file_play(PyObject * self, PyObject * args)
|
|
{
|
|
// The WAV file must be recorded at 48000 Hertz in S16_LE format monophonic for audio files.
|
|
// The WAV file must be recorded at the sample_rate in IEEE format stereo for the I/Q samples file.
|
|
const char * fname;
|
|
char name[5];
|
|
int size, rate=0;
|
|
|
|
if (!PyArg_ParseTuple (args, "s", &fname))
|
|
return NULL;
|
|
if (wavFpMic)
|
|
fclose(wavFpMic);
|
|
if (wavFpSound)
|
|
fclose(wavFpSound);
|
|
wavFpSound = wavFpMic = NULL;
|
|
wavFpSound = fopen(fname, "rb");
|
|
if (!wavFpSound) {
|
|
printf("open wav file failed\n");
|
|
return PyInt_FromLong(-1);
|
|
}
|
|
wavStart = 0;
|
|
while (1) {
|
|
if (fread (name, 4, 1, wavFpSound) != 1)
|
|
break;
|
|
if (fread (&size, 4, 1, wavFpSound) != 1)
|
|
break;
|
|
name[4] = 0;
|
|
// printf("name %s size %d\n", name, size);
|
|
if (!strncmp(name, "RIFF", 4))
|
|
fseek (wavFpSound, 4, SEEK_CUR); // Skip "WAVE"
|
|
else if (!strncmp(name, "fmt ", 4)) { // format data starts here
|
|
if (fread (&rate, 4, 1, wavFpSound) != 1) // skip these fields
|
|
break;
|
|
if (fread (&rate, 4, 1, wavFpSound) != 1) // sample rate
|
|
break;
|
|
//printf ("rate %d\n", rate);
|
|
fseek (wavFpSound, size - 8, SEEK_CUR); // skip remainder
|
|
}
|
|
else if (!strncmp(name, "data", 4)) { // sound data starts here
|
|
wavStart = ftell(wavFpSound);
|
|
break;
|
|
}
|
|
else // Skip other records
|
|
fseek (wavFpSound, size, SEEK_CUR);
|
|
}
|
|
if (!wavStart) { // Failure to find "data" record
|
|
fclose(wavFpSound);
|
|
wavFpSound = NULL;
|
|
printf("open wav failed to find the data chunk\n");
|
|
return PyInt_FromLong(-2);
|
|
}
|
|
wavFpMic = fopen(fname, "rb");
|
|
if (!wavFpMic) {
|
|
printf("open microphone wav file failed\n");
|
|
wavFpSound = NULL;
|
|
return PyInt_FromLong(-4);
|
|
}
|
|
return PyInt_FromLong(rate);
|
|
}
|
|
|
|
void quisk_file_playback(complex double * cSamples, int nSamples, double volume)
|
|
{
|
|
// Replace radio sound by file samples.
|
|
// The sample rate must equal quisk_sound_state.mic_sample_rate.
|
|
int i;
|
|
short sh;
|
|
double d;
|
|
|
|
if (wavFpSound) {
|
|
for (i = 0; i < nSamples; i++) {
|
|
if (fread(&sh, 2, 1, wavFpSound) != 1) {
|
|
quisk_record_state = IDLE;
|
|
break;
|
|
}
|
|
d = sh * ((double)CLIP32 / CLIP16) * volume;
|
|
cSamples[i] = d + I * d;
|
|
}
|
|
}
|
|
}
|
|
|
|
void quisk_play_samples(complex double * cSamples, int nSamples)
|
|
{
|
|
int i;
|
|
float fre, fim;
|
|
|
|
if (wavFpSound) {
|
|
for (i = 0; i < nSamples; i++) {
|
|
if (fread(&fre, 4, 1, wavFpSound) != 1 || fread(&fim, 4, 1, wavFpSound) != 1) {
|
|
quisk_record_state = IDLE;
|
|
break;
|
|
}
|
|
fre *= CLIP32;
|
|
fim *= CLIP32;
|
|
cSamples[i] = fre + I * fim;
|
|
}
|
|
}
|
|
}
|
|
|
|
#define BUF2CHAN_SIZE 12000
|
|
static int Buffer2Chan(double * samp1, int count1, double * samp2, int count2)
|
|
{ // return the minimum of count1 and count2, buffering as necessary
|
|
int nout;
|
|
static int nbuf1=0, nbuf2=0;
|
|
static double buf1[BUF2CHAN_SIZE], buf2[BUF2CHAN_SIZE];
|
|
|
|
if (samp1 == NULL) { // initialize
|
|
nbuf1 = nbuf2 = 0;
|
|
return 0;
|
|
}
|
|
if (nbuf1 == 0 && nbuf2 == 0 && count1 == count2) // nothing to do
|
|
return count1;
|
|
if (count1 + nbuf1 >= BUF2CHAN_SIZE || count2 + nbuf2 >= BUF2CHAN_SIZE) { // overflow
|
|
if (DEBUG || DEBUG_IO)
|
|
printf("Overflow in Buffer2Chan nbuf1 %d nbuf2 %d size %d\n", nbuf1, nbuf2, BUF2CHAN_SIZE);
|
|
nbuf1 = nbuf2 = 0;
|
|
}
|
|
memcpy(buf1 + nbuf1, samp1, count1 * sizeof(double)); // add samples to buffer
|
|
nbuf1 += count1;
|
|
memcpy(buf2 + nbuf2, samp2, count2 * sizeof(double));
|
|
nbuf2 += count2;
|
|
if (nbuf1 <= nbuf2)
|
|
nout = nbuf1; // number of samples to output
|
|
else
|
|
nout = nbuf2;
|
|
//if (count1 + nbuf1 >= 2000 || count2 + nbuf2 >= 2000)
|
|
// printf("Buffer2Chan nbuf1 %d nbuf2 %d nout %d\n", nbuf1, nbuf2, nout);
|
|
memcpy(samp1, buf1, nout * sizeof(double)); // output samples
|
|
nbuf1 -= nout;
|
|
memmove(buf1, buf1 + nout, nbuf1 * sizeof(double));
|
|
memcpy(samp2, buf2, nout * sizeof(double));
|
|
nbuf2 -= nout;
|
|
memmove(buf2, buf2 + nout, nbuf2 * sizeof(double));
|
|
return nout;
|
|
}
|
|
|
|
void quisk_file_microphone(complex double * cSamples, int nSamples)
|
|
{
|
|
// Replace mic samples by file samples.
|
|
// The sample rate must equal quisk_sound_state.mic_sample_rate.
|
|
int i;
|
|
short sh;
|
|
double d;
|
|
|
|
if (wavFpMic) {
|
|
for (i = 0; i < nSamples; i++) {
|
|
if (fread(&sh, 2, 1, wavFpMic) != 1) {
|
|
quisk_record_state = IDLE;
|
|
break;
|
|
}
|
|
d = sh * ((double)CLIP32 / CLIP16);
|
|
cSamples[i] = d + I * d;
|
|
}
|
|
}
|
|
}
|
|
|
|
int PlanDecimation(int * pt2, int * pt3, int * pt5) // search for a suitable decimation scheme
|
|
{
|
|
int i, best, try, i2, i3, i5, decim2, decim3, decim5;
|
|
|
|
best = quisk_sound_state.sample_rate;
|
|
decim2 = decim3 = decim5 = 0;
|
|
for (i2 = 0; i2 <= 6; i2++) { // limit to number of /2 filters, currently 6
|
|
for (i3 = 0; i3 <= 3; i3++) { // limit to number of /3 filters, currently 3
|
|
for (i5 = 0; i5 <= 3; i5++) { // limit to number of /5 filters, currently 3
|
|
try = quisk_sound_state.sample_rate;
|
|
for (i = 0; i < i2; i++)
|
|
try /= 2;
|
|
for (i = 0; i < i3; i++)
|
|
try /= 3;
|
|
for (i = 0; i < i5; i++)
|
|
try /= 5;
|
|
if (try >= 48000 && try < best) {
|
|
decim2 = i2;
|
|
decim3 = i3;
|
|
decim5 = i5;
|
|
best = try;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
if (best >= 50000) // special rate converter
|
|
best = best * 24 / 25;
|
|
if (DEBUG)
|
|
printf ("Plan Decimation: rate %i, best %i, decim2 %i, decim3 %i, decim5 %i\n",
|
|
quisk_sound_state.sample_rate, best, decim2, decim3, decim5);
|
|
if (best > 72000)
|
|
printf("Failure to plan a suitable decimation in quisk_process_decimate\n");
|
|
if (pt2) { // return decimations
|
|
*pt2 = decim2;
|
|
*pt3 = decim3;
|
|
*pt5 = decim5;
|
|
}
|
|
return best;
|
|
}
|
|
|
|
static int quisk_process_decimate(complex double * cSamples, int nSamples, int bank, rx_mode_type rx_mode)
|
|
{ // Changes here will require changes to get_filter_rate();
|
|
int i, i2, i3, i5;
|
|
static int decim2, decim3, decim5;
|
|
static int old_rate = 0;
|
|
static struct stStorage {
|
|
struct quisk_cHB45Filter HalfBand1;
|
|
struct quisk_cHB45Filter HalfBand2;
|
|
struct quisk_cHB45Filter HalfBand3;
|
|
struct quisk_cHB45Filter HalfBand4;
|
|
struct quisk_cHB45Filter HalfBand5;
|
|
struct quisk_cFilter filtSdriq111;
|
|
struct quisk_cFilter filtSdriq53;
|
|
struct quisk_cFilter filtSdriq133;
|
|
struct quisk_cFilter filtSdriq167;
|
|
struct quisk_cFilter filtSdriq185;
|
|
struct quisk_cFilter filtDecim3;
|
|
struct quisk_cFilter filtDecim3B;
|
|
struct quisk_cFilter filtDecim3C;
|
|
struct quisk_cFilter filtDecim5;
|
|
struct quisk_cFilter filtDecim5B;
|
|
struct quisk_cFilter filtDecim5S;
|
|
struct quisk_cFilter filtDecim48to24;
|
|
struct quisk_cFilter filtI3D25;
|
|
struct quisk_cFilter filt300D5;
|
|
} Storage[MAX_RX_CHANNELS] ;
|
|
|
|
if ( ! cSamples) { // Initialize all filters
|
|
for (i = 0; i < MAX_RX_CHANNELS; i++) {
|
|
memset(&Storage[i].HalfBand1, 0, sizeof(struct quisk_cHB45Filter));
|
|
memset(&Storage[i].HalfBand2, 0, sizeof(struct quisk_cHB45Filter));
|
|
memset(&Storage[i].HalfBand3, 0, sizeof(struct quisk_cHB45Filter));
|
|
memset(&Storage[i].HalfBand4, 0, sizeof(struct quisk_cHB45Filter));
|
|
memset(&Storage[i].HalfBand5, 0, sizeof(struct quisk_cHB45Filter));
|
|
quisk_filt_cInit(&Storage[i].filtSdriq111, quiskFilt111D2Coefs, sizeof(quiskFilt111D2Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtSdriq53, quiskFilt53D1Coefs, sizeof(quiskFilt53D1Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtSdriq133, quiskFilt133D2Coefs, sizeof(quiskFilt133D2Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtSdriq167, quiskFilt167D3Coefs, sizeof(quiskFilt167D3Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtSdriq185, quiskFilt185D3Coefs, sizeof(quiskFilt185D3Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim3, quiskFilt144D3Coefs, sizeof(quiskFilt144D3Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim3B, quiskFilt144D3Coefs, sizeof(quiskFilt144D3Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim3C, quiskFilt144D3Coefs, sizeof(quiskFilt144D3Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim5, quiskFilt240D5CoefsSharp, sizeof(quiskFilt240D5CoefsSharp)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim5B, quiskFilt240D5CoefsSharp, sizeof(quiskFilt240D5CoefsSharp)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim5S, quiskFilt240D5CoefsSharp, sizeof(quiskFilt240D5CoefsSharp)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim48to24, quiskFilt48dec24Coefs, sizeof(quiskFilt48dec24Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtI3D25, quiskFiltI3D25Coefs, sizeof(quiskFiltI3D25Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filt300D5, quiskFilt300D5Coefs, sizeof(quiskFilt300D5Coefs)/sizeof(double));
|
|
}
|
|
return 0;
|
|
}
|
|
if (quisk_sound_state.sample_rate != old_rate) {
|
|
old_rate = quisk_sound_state.sample_rate;
|
|
PlanDecimation(&decim2, &decim3, &decim5);
|
|
}
|
|
// Decimate: Lower the sample rate to 48000 sps (or approx). Filters are designed for
|
|
// a pass bandwidth of 20 kHz and a stop bandwidth of 24 kHz.
|
|
// We use 48 ksps to accommodate wide digital modes.
|
|
switch((quisk_sound_state.sample_rate + 100) / 1000) {
|
|
case 41:
|
|
quisk_decim_srate = 48000;
|
|
break;
|
|
case 53: // SDR-IQ
|
|
quisk_decim_srate = quisk_sound_state.sample_rate;
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtSdriq53, 1);
|
|
break;
|
|
case 111: // SDR-IQ
|
|
quisk_decim_srate = quisk_sound_state.sample_rate / 2;
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtSdriq111, 2);
|
|
break;
|
|
case 133: // SDR-IQ
|
|
quisk_decim_srate = quisk_sound_state.sample_rate / 2;
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtSdriq133, 2);
|
|
break;
|
|
case 185: // SDR-IQ
|
|
quisk_decim_srate = quisk_sound_state.sample_rate / 3;
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtSdriq185, 3);
|
|
break;
|
|
case 370:
|
|
quisk_decim_srate = quisk_sound_state.sample_rate / 6;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand2);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtSdriq185, 3);
|
|
break;
|
|
case 740:
|
|
quisk_decim_srate = quisk_sound_state.sample_rate / 12;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand2);
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand3);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtSdriq185, 3);
|
|
break;
|
|
case 1333:
|
|
quisk_decim_srate = quisk_sound_state.sample_rate / 24;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand1);
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand2);
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand3);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtSdriq167, 3);
|
|
break;
|
|
default:
|
|
quisk_decim_srate = quisk_sound_state.sample_rate;
|
|
i2 = decim2; // decimate by 2 except for the final /2 filter
|
|
if (i2 > 1) {
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand1);
|
|
quisk_decim_srate /= 2;
|
|
i2--;
|
|
}
|
|
if (i2 > 1) {
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand2);
|
|
quisk_decim_srate /= 2;
|
|
i2--;
|
|
}
|
|
if (i2 > 1) {
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand3);
|
|
quisk_decim_srate /= 2;
|
|
i2--;
|
|
}
|
|
if (i2 > 1) {
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand4);
|
|
quisk_decim_srate /= 2;
|
|
i2--;
|
|
}
|
|
if (i2 > 1) {
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
|
|
quisk_decim_srate /= 2;
|
|
i2--;
|
|
}
|
|
i3 = decim3; // decimate by 3
|
|
if (i3 > 0) {
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim3, 3);
|
|
quisk_decim_srate /= 3;
|
|
i3--;
|
|
}
|
|
if (i3 > 0) {
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim3B, 3);
|
|
quisk_decim_srate /= 3;
|
|
i3--;
|
|
}
|
|
if (i3 > 0) {
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim3C, 3);
|
|
quisk_decim_srate /= 3;
|
|
i3--;
|
|
}
|
|
i5 = decim5; // decimate by 5
|
|
if (i5 > 0) {
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim5, 5);
|
|
quisk_decim_srate /= 5;
|
|
i5--;
|
|
}
|
|
if (i5 > 0) {
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim5B, 5);
|
|
quisk_decim_srate /= 5;
|
|
i5--;
|
|
}
|
|
if (i5 > 0) {
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim5S, 5);
|
|
quisk_decim_srate /= 5;
|
|
i5--;
|
|
}
|
|
if (i2 > 0) { // decimate by 2 last - Unnecessary???
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to24, 2);
|
|
quisk_decim_srate /= 2;
|
|
i2--;
|
|
}
|
|
if (quisk_decim_srate >= 50000) {
|
|
quisk_decim_srate = quisk_decim_srate * 24 / 25;
|
|
nSamples = quisk_cInterpDecim(cSamples, nSamples, &Storage[bank].filt300D5, 6, 5); // 60 kSps
|
|
nSamples = quisk_cInterpDecim(cSamples, nSamples, &Storage[bank].filtDecim5S, 4, 5); // 48 kSps
|
|
}
|
|
if (i2 != 0 || i3 != 0 || i5 != 0)
|
|
printf ("Failure in quisk.c in integer decimation for rate %d\n", quisk_sound_state.sample_rate);
|
|
if (DEBUG && quisk_decim_srate != 48000)
|
|
printf("Failure to achieve rate 48000. Rate is %i\n", quisk_decim_srate);
|
|
break;
|
|
}
|
|
return nSamples;
|
|
}
|
|
|
|
static int quisk_process_demodulate(complex double * cSamples, double * dsamples, int nSamples, int bank, int nFilter, rx_mode_type rx_mode)
|
|
{ // Changes here will require changes to get_filter_rate();
|
|
int i;
|
|
complex double cx, cpx;
|
|
double d, di, dd;
|
|
static struct AgcState Agc1 = {0.3, 16000, 0}, Agc2 = {0.3, 16000, 0};
|
|
static struct stStorage {
|
|
complex double fm_1; // Sample delayed by one
|
|
double dc_remove; // DC removal for AM
|
|
double FM_www;
|
|
double FM_nnn, FM_a_0, FM_a_1, FM_b_1, FM_x_1, FM_y_1; // filter for FM
|
|
struct quisk_cHB45Filter HalfBand4;
|
|
struct quisk_cHB45Filter HalfBand5;
|
|
struct quisk_dHB45Filter HalfBand6;
|
|
struct quisk_dHB45Filter HalfBand7;
|
|
struct quisk_dFilter filtAudio24p3;
|
|
struct quisk_dFilter filtAudio24p4;
|
|
struct quisk_dFilter filtAudio12p2;
|
|
struct quisk_dFilter filtAudio24p6;
|
|
struct quisk_dFilter filtAudioFmHp;
|
|
struct quisk_cFilter filtDecim16to8;
|
|
struct quisk_cFilter filtDecim48to24;
|
|
struct quisk_cFilter filtDecim48to16;
|
|
} Storage[MAX_RX_CHANNELS] ;
|
|
|
|
if ( ! cSamples) { // Initialize all filters
|
|
for (i = 0; i < MAX_RX_CHANNELS; i++) {
|
|
memset(&Storage[i].HalfBand4, 0, sizeof(struct quisk_cHB45Filter));
|
|
memset(&Storage[i].HalfBand5, 0, sizeof(struct quisk_cHB45Filter));
|
|
memset(&Storage[i].HalfBand6, 0, sizeof(struct quisk_dHB45Filter));
|
|
memset(&Storage[i].HalfBand7, 0, sizeof(struct quisk_dHB45Filter));
|
|
quisk_filt_dInit(&Storage[i].filtAudio24p3, quiskAudio24p3Coefs, sizeof(quiskAudio24p3Coefs)/sizeof(double));
|
|
quisk_filt_dInit(&Storage[i].filtAudio24p4, quiskAudio24p4Coefs, sizeof(quiskAudio24p4Coefs)/sizeof(double));
|
|
quisk_filt_dInit(&Storage[i].filtAudio12p2, quiskAudio24p4Coefs, sizeof(quiskAudio24p4Coefs)/sizeof(double));
|
|
quisk_filt_dInit(&Storage[i].filtAudio24p6, quiskAudio24p6Coefs, sizeof(quiskAudio24p6Coefs)/sizeof(double));
|
|
quisk_filt_dInit(&Storage[i].filtAudioFmHp, quiskAudioFmHpCoefs, sizeof(quiskAudioFmHpCoefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim16to8, quiskFilt16dec8Coefs, sizeof(quiskFilt16dec8Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim48to24, quiskFilt48dec24Coefs, sizeof(quiskFilt48dec24Coefs)/sizeof(double));
|
|
quisk_filt_cInit(&Storage[i].filtDecim48to16, quiskAudio24p3Coefs, sizeof(quiskAudio24p3Coefs)/sizeof(double));
|
|
Storage[i].fm_1 = 10;
|
|
Storage[i].FM_www = tan(M_PI * FM_FILTER_DEMPH / 48000); // filter for FM at 48 ksps
|
|
Storage[i].FM_nnn = 1.0 / (1.0 + Storage[i].FM_www);
|
|
Storage[i].FM_a_0 = Storage[i].FM_www * Storage[i].FM_nnn;
|
|
Storage[i].FM_a_1 = Storage[i].FM_a_0;
|
|
Storage[i].FM_b_1 = Storage[i].FM_nnn * (Storage[i].FM_www - 1.0);
|
|
//printf ("dsamples[i] = y_1 = di * %12.6lf + x_1 * %12.6lf - y_1 * %12.6lf\n", FM_a_0, FM_a_1, FM_b_1);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
//quisk_calc_audio_graph(pow(2, 31) - 1, cSamples, NULL, nSamples, 0);
|
|
// Filter and demodulate signal, copy capture buffer cSamples to play buffer dsamples.
|
|
// quisk_decim_srate is the sample rate after integer decimation.
|
|
MeasureSquelch[bank].squelch_active = 0;
|
|
switch(rx_mode) {
|
|
case CWL: // lower sideband CW at 6 ksps
|
|
quisk_filter_srate = quisk_decim_srate / 8;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand4);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to24, 2);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cx = cRxFilterOut(cSamples[i], bank, nFilter);
|
|
dsamples[i] = dd = creal(cx) + cimag(cx);
|
|
if(bank == 0) {
|
|
measure_audio_sum += dd * dd;
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, rit_freq, quisk_filter_srate);
|
|
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio12p2, 2);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand6);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
|
|
break;
|
|
case CWU: // upper sideband CW at 6 ksps
|
|
quisk_filter_srate = quisk_decim_srate / 8;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand4);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to24, 2);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cx = cRxFilterOut(cSamples[i], bank, nFilter);
|
|
dsamples[i] = dd = creal(cx) - cimag(cx);
|
|
if(bank == 0) {
|
|
measure_audio_sum += dd * dd;
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, rit_freq, quisk_filter_srate);
|
|
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio12p2, 2);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand6);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
|
|
break;
|
|
case LSB: // lower sideband SSB at 12 ksps
|
|
quisk_filter_srate = quisk_decim_srate / 4;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to24, 2);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cx = cRxFilterOut(cSamples[i], bank, nFilter);
|
|
dsamples[i] = dd = creal(cx) + cimag(cx);
|
|
if(bank == 0) {
|
|
measure_audio_sum += dd * dd;
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, 0, quisk_filter_srate);
|
|
if (ssb_squelch_enabled) {
|
|
ssb_squelch(dsamples, nSamples, quisk_filter_srate, MeasureSquelch + bank);
|
|
d_delay(dsamples, nSamples, bank, SQUELCH_FFT_SIZE);
|
|
}
|
|
quisk_calc_audio_graph(pow(2, 31) - 1, NULL, dsamples, nSamples, 1);
|
|
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p4, 2);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
|
|
//quisk_calc_audio_graph(pow(2, 31) - 1, NULL, dsamples, nSamples, 1);
|
|
break;
|
|
case USB: // upper sideband SSB at 12 ksps
|
|
default:
|
|
quisk_filter_srate = quisk_decim_srate / 4;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to24, 2);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cx = cRxFilterOut(cSamples[i], bank, nFilter);
|
|
dsamples[i] = dd = creal(cx) - cimag(cx);
|
|
if(bank == 0) {
|
|
measure_audio_sum += dd * dd;
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, 0, quisk_filter_srate);
|
|
if (ssb_squelch_enabled) {
|
|
ssb_squelch(dsamples, nSamples, quisk_filter_srate, MeasureSquelch + bank);
|
|
d_delay(dsamples, nSamples, bank, SQUELCH_FFT_SIZE);
|
|
}
|
|
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p4, 2);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
|
|
//quisk_calc_audio_graph(pow(2, 31) - 1, NULL, dsamples, nSamples, 1);
|
|
break;
|
|
case AM: // AM at 24 ksps
|
|
quisk_filter_srate = quisk_decim_srate / 2;
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to24, 2);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cx = dRxFilterOut(cSamples[i], bank, nFilter);
|
|
di = cabs(cx);
|
|
d = di + Storage[bank].dc_remove * 0.99; // DC removal; R.G. Lyons page 553
|
|
di = d - Storage[bank].dc_remove;
|
|
Storage[bank].dc_remove = d;
|
|
dsamples[i] = di;
|
|
if(bank == 0) {
|
|
measure_audio_sum += di * di;
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
nSamples = quisk_dFilter(dsamples, nSamples, &Storage[bank].filtAudio24p6);
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, 0, quisk_filter_srate);
|
|
if (ssb_squelch_enabled) {
|
|
ssb_squelch(dsamples, nSamples, quisk_filter_srate, MeasureSquelch + bank);
|
|
d_delay(dsamples, nSamples, bank, SQUELCH_FFT_SIZE);
|
|
}
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
|
|
break;
|
|
case FM: // FM at 48 ksps
|
|
case DGT_FM:
|
|
quisk_filter_srate = quisk_decim_srate;
|
|
for (i = 0; i < nSamples; i++) {
|
|
cx = dRxFilterOut(cSamples[i], bank, nFilter);
|
|
MeasureSquelch[bank].rf_sum += cabs(cx);
|
|
MeasureSquelch[bank].rf_count += 1;
|
|
cpx = cx * conj(Storage[bank].fm_1);
|
|
Storage[bank].fm_1 = cx;
|
|
di = quisk_filter_srate * carg(cpx);
|
|
// FM de-emphasis
|
|
dsamples[i] = dd = Storage[bank].FM_y_1 = di * Storage[bank].FM_a_0 +
|
|
Storage[bank].FM_x_1 * Storage[bank].FM_a_1 - Storage[bank].FM_y_1 * Storage[bank].FM_b_1;
|
|
Storage[bank].FM_x_1 = di;
|
|
if(bank == 0) {
|
|
measure_audio_sum += dd * dd;
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
nSamples = quisk_dDecimate(dsamples, nSamples, &Storage[bank].filtAudio24p3, 2);
|
|
nSamples = quisk_dFilter(dsamples, nSamples, &Storage[bank].filtAudioFmHp);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand6);
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, 0, quisk_filter_srate);
|
|
if (MeasureSquelch[bank].rf_count >= 2400) {
|
|
MeasureSquelch[bank].squelch = MeasureSquelch[bank].rf_sum / MeasureSquelch[bank].rf_count / CLIP32;
|
|
if (MeasureSquelch[bank].squelch > 1.E-10)
|
|
MeasureSquelch[bank].squelch = 20 * log10(MeasureSquelch[bank].squelch);
|
|
else
|
|
MeasureSquelch[bank].squelch = -200.0;
|
|
MeasureSquelch[bank].rf_sum = MeasureSquelch[bank].rf_count = 0;
|
|
//if (bank == 0) printf("MeasureSquelch %.5f\n", MeasureSquelch[0].squelch);
|
|
}
|
|
MeasureSquelch[bank].squelch_active = MeasureSquelch[bank].squelch < squelch_level;
|
|
break;
|
|
case DGT_U: // digital mode DGT-U at 48 ksps
|
|
if (filter_bandwidth[nFilter] < DGT_NARROW_FREQ) { // filter at 6 ksps
|
|
quisk_filter_srate = quisk_decim_srate / 8;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand4);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to24, 2);
|
|
}
|
|
else { // filter at 48 ksps
|
|
quisk_filter_srate = quisk_decim_srate;
|
|
}
|
|
for (i = 0; i < nSamples; i++) {
|
|
cx = cRxFilterOut(cSamples[i], bank, nFilter);
|
|
dsamples[i] = dd = creal(cx) - cimag(cx);
|
|
if(bank == 0) {
|
|
measure_audio_sum += dd * dd;
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, 0, quisk_filter_srate);
|
|
if (filter_bandwidth[nFilter] < DGT_NARROW_FREQ) {
|
|
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio12p2, 2);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand6);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
|
|
}
|
|
break;
|
|
case DGT_L: // digital mode DGT-L
|
|
if (filter_bandwidth[nFilter] < DGT_NARROW_FREQ) { // filter at 6 ksps
|
|
quisk_filter_srate = quisk_decim_srate / 8;
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
|
|
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand4);
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to24, 2);
|
|
}
|
|
else { // filter at 48 ksps
|
|
quisk_filter_srate = quisk_decim_srate;
|
|
}
|
|
for (i = 0; i < nSamples; i++) {
|
|
cx = cRxFilterOut(cSamples[i], bank, nFilter);
|
|
dsamples[i] = dd = creal(cx) + cimag(cx);
|
|
if(bank == 0) {
|
|
measure_audio_sum += dd * dd;
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, 0, quisk_filter_srate);
|
|
if (filter_bandwidth[nFilter] < DGT_NARROW_FREQ) {
|
|
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio12p2, 2);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand6);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
|
|
}
|
|
break;
|
|
case DGT_IQ: // digital mode DGT-IQ at 48 ksps
|
|
quisk_filter_srate = quisk_decim_srate;
|
|
if (filter_bandwidth[nFilter] < 19000) { // No filtering for wide bandwidth
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dRxFilterOut(cSamples[i], bank, nFilter);
|
|
}
|
|
if(bank == 0) {
|
|
for (i = 0; i < nSamples; i++) {
|
|
measure_audio_sum = measure_audio_sum + cSamples[i] * conj(cSamples[i]);
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
break;
|
|
case FDV_U: // digital voice at 8 ksps
|
|
case FDV_L:
|
|
quisk_check_freedv_mode();
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to16, 3);
|
|
if (bank == 0)
|
|
process_agc(&Agc1, cSamples, nSamples, 1);
|
|
else
|
|
process_agc(&Agc2, cSamples, nSamples, 1);
|
|
if(bank == 0)
|
|
dAutoNotch(dsamples, nSamples, 0, quisk_decim_srate / 3);
|
|
// Perhaps decimate by an additional fraction
|
|
if (quisk_decim_srate != 48000) {
|
|
dd = quisk_decim_srate / 48000.0;
|
|
nSamples = cFracDecim(cSamples, nSamples, dd);
|
|
quisk_decim_srate = 48000;
|
|
}
|
|
quisk_filter_srate =8000;
|
|
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim16to8, 2);
|
|
if (pt_quisk_freedv_rx)
|
|
nSamples = (* pt_quisk_freedv_rx)(cSamples, dsamples, nSamples, bank);
|
|
if(bank == 0) {
|
|
for (i = 0; i < nSamples; i++) {
|
|
measure_audio_sum += dsamples[i] * dsamples[i];
|
|
measure_audio_count += 1;
|
|
}
|
|
}
|
|
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p3, 3);
|
|
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
|
|
break;
|
|
}
|
|
if (bank == 0 && measure_audio_count >= quisk_filter_srate * measure_audio_time) {
|
|
measured_audio = sqrt(measure_audio_sum / measure_audio_count) / CLIP32 * 1e6;
|
|
measure_audio_sum = measure_audio_count = 0;
|
|
}
|
|
return nSamples;
|
|
}
|
|
|
|
static void process_agc(struct AgcState * dat, complex double * csamples, int count, int is_cpx)
|
|
{
|
|
int i;
|
|
double out_magn, buf_magn, dtmp, clip_gain;
|
|
complex double csample;
|
|
#if DEBUG
|
|
static int printit=0;
|
|
static double maxout=1;
|
|
char * clip;
|
|
#endif
|
|
|
|
if ( ! dat->buf_size) { // initialize
|
|
if (dat->sample_rate == 0)
|
|
dat->sample_rate = quisk_sound_state.playback_rate;
|
|
dat->buf_size = dat->sample_rate * AGC_DELAY / 1000; // total delay in samples
|
|
//printf("play rate %d buf_size %d\n", dat->sample_rate, dat->buf_size);
|
|
dat->index_read = 0; // Index to output; and then write a new sample here
|
|
dat->index_start = 0; // Start index for measure of maximum sample
|
|
dat->is_clipping = 0; // Are we decreasing gain to handle a clipping condition?
|
|
dat->themax = 1.0; // Maximum sample in the buffer
|
|
dat->gain = 100; // Current output gain
|
|
dat->delta = 0; // Amount to change dat->gain at each sample
|
|
dat->target_gain = 100; // Move to this gain unless we clip
|
|
dat->time_release = 1.0 - exp( - 1.0 / dat->sample_rate / agc_release_time); // long time constant for AGC release
|
|
dat->c_samp = (complex double *) malloc(dat->buf_size * sizeof(complex double)); // buffer for complex samples
|
|
for (i = 0; i < dat->buf_size; i++)
|
|
dat->c_samp[i] = 0;
|
|
return;
|
|
}
|
|
for (i = 0; i < count; i++) {
|
|
csample = csamples[i];
|
|
csamples[i] = dat->c_samp[dat->index_read] * dat->gain; // FIFO output
|
|
if (is_cpx)
|
|
out_magn = cabs(csamples[i]);
|
|
else
|
|
out_magn = fabs(creal(csamples[i]));
|
|
//if(dat->is_clipping == 1)
|
|
//printf(" index %5d out_magn %.5lf gain %.2lf delta %.5lf\n",dat->index_read, out_magn / CLIP32, dat->gain, dat->delta);
|
|
#if DEBUG
|
|
if (out_magn > maxout)
|
|
maxout = out_magn;
|
|
#endif
|
|
if (out_magn > CLIP32) {
|
|
csamples[i] /= out_magn;
|
|
#if DEBUG
|
|
printf("Clip out_magn %8.5lf is_clipping %d index_read %5d index_start %5d gain %8.5lf\n",
|
|
out_magn / CLIP32, dat->is_clipping, dat->index_read, dat->index_start, dat->gain);
|
|
#endif
|
|
}
|
|
dat->c_samp[dat->index_read] = csample; // write new sample at read index
|
|
if (is_cpx)
|
|
buf_magn = cabs(csample);
|
|
else
|
|
buf_magn = fabs(creal(csample));
|
|
if (dat->is_clipping == 0) {
|
|
if (buf_magn * dat->gain > dat->max_out * CLIP32) {
|
|
dat->target_gain = dat->max_out * CLIP32 / buf_magn;
|
|
dat->delta = (dat->gain - dat->target_gain) / dat->buf_size;
|
|
dat->is_clipping = 1;
|
|
dat->themax = buf_magn;
|
|
// printf("Start index %5d buf_magn %10.8lf target %8.2lf gain %8.2lf delta %8.5lf\n",
|
|
// dat->index_read, buf_magn / CLIP32, dat->target_gain, dat->gain, dat->delta);
|
|
dat->gain -= dat->delta;
|
|
}
|
|
else if (dat->index_read == dat->index_start) {
|
|
clip_gain = dat->max_out * CLIP32 / dat->themax; // clip gain based on the maximum sample in the buffer
|
|
if (rxMode == FM || rxMode == DGT_FM) // mode is FM
|
|
dat->target_gain = clip_gain;
|
|
else if (agcReleaseGain > clip_gain)
|
|
dat->target_gain = clip_gain;
|
|
else
|
|
dat->target_gain = agcReleaseGain;
|
|
dat->themax = buf_magn;
|
|
dat->gain = dat->gain * (1.0 - dat->time_release) + dat->target_gain * dat->time_release;
|
|
// printf("New index %5d themax %7.5lf clip_gain %5.0lf agcReleaseGain %5.0lf\n",
|
|
// dat->index_start, dat->themax / CLIP32, clip_gain, agcReleaseGain);
|
|
}
|
|
else {
|
|
if (dat->themax < buf_magn)
|
|
dat->themax = buf_magn;
|
|
dat->gain = dat->gain * (1.0 - dat->time_release) + dat->target_gain * dat->time_release;
|
|
}
|
|
}
|
|
else { // dat->is_clipping == 1; we are handling a clip condition
|
|
if (buf_magn > dat->themax) {
|
|
dat->themax = buf_magn;
|
|
dat->target_gain = dat->max_out * CLIP32 / buf_magn;
|
|
dtmp = (dat->gain - dat->target_gain) / dat->buf_size; // new value of delta
|
|
if (dtmp > dat->delta) {
|
|
dat->delta = dtmp;
|
|
// printf(" Strt index %5d buf_magn %10.8lf target %8.2lf gain %8.2lf delta %8.5lf\n",
|
|
// dat->index_read, buf_magn / CLIP32, dat->target_gain, dat->gain, dat->delta);
|
|
}
|
|
else {
|
|
// printf(" Plus index %5d buf_magn %10.8lf target %8.2lf gain %8.2lf delta %8.5lf\n",
|
|
// dat->index_read, buf_magn / CLIP32, dat->target_gain, dat->gain, dat->delta);
|
|
}
|
|
}
|
|
dat->gain -= dat->delta;
|
|
if (dat->gain <= dat->target_gain) {
|
|
dat->is_clipping = 0;
|
|
dat->gain = dat->target_gain;
|
|
// printf("End index %5d buf_magn %10.8lf target %8.2lf gain %8.2lf delta %8.5lf themax %10.8lf\n",
|
|
// dat->index_read, buf_magn / CLIP32, dat->target_gain, dat->gain, dat->delta, dat->themax / CLIP32);
|
|
dat->themax = buf_magn;
|
|
dat->index_start = dat->index_read;
|
|
}
|
|
}
|
|
if (++dat->index_read >= dat->buf_size)
|
|
dat->index_read = 0;
|
|
#if DEBUG
|
|
if (printit++ >= dat->sample_rate * 500 / 1000) {
|
|
printit = 0;
|
|
dtmp = 20 * log10(maxout / CLIP32);
|
|
if (dtmp >= 0)
|
|
clip = "Clip";
|
|
else
|
|
clip = "";
|
|
printf("Out agcGain %5.0lf target_gain %9.0lf gain %9.0lf output %7.2lf %s\n",
|
|
agcReleaseGain, dat->target_gain, dat->gain, dtmp, clip);
|
|
maxout = 1;
|
|
}
|
|
#endif
|
|
}
|
|
return;
|
|
}
|
|
|
|
int quisk_process_samples(complex double * cSamples, int nSamples)
|
|
{
|
|
// Called when samples are available.
|
|
// Samples range from about 2^16 to a max of 2^31.
|
|
int i, n, nout, is_key_down, squelch_real=0, squelch_imag=0;
|
|
double d, di, tune;
|
|
double double_filter_decim;
|
|
complex double phase;
|
|
int orig_nSamples;
|
|
fft_data * ptFFT;
|
|
rx_mode_type rx_mode;
|
|
|
|
static int size_dsamples = 0; // Current dimension of dsamples, dsamples2, orig_cSamples, buf_cSamples
|
|
static int old_split_rxtx = 0; // Prior value of split_rxtx
|
|
static int old_multirx_play_channel = 0; // Prior value of multirx_play_channel
|
|
static double * dsamples = NULL;
|
|
static double * dsamples2 = NULL;
|
|
static complex double * orig_cSamples = NULL;
|
|
static complex double * buf_cSamples = NULL;
|
|
static complex double rxTuneVector = 1;
|
|
static complex double txTuneVector = 1;
|
|
static complex double aux1TuneVector = 1;
|
|
static complex double aux2TuneVector = 1;
|
|
static complex double sidetoneVector = BIG_VOLUME;
|
|
static double dOutCounter = 0; // Cumulative net output samples for sidetone etc.
|
|
static int sidetoneIsOn = 0; // The status of the sidetone
|
|
static double sidetoneEnvelope; // Shape the rise and fall times of the sidetone
|
|
static double keyupEnvelope = 1.0; // Shape the rise time on key up
|
|
static int playSilence;
|
|
static struct quisk_cHB45Filter HalfBand7 = {NULL, 0, 0};
|
|
static struct quisk_cHB45Filter HalfBand8 = {NULL, 0, 0};
|
|
static struct quisk_cHB45Filter HalfBand9 = {NULL, 0, 0};
|
|
static struct AgcState Agc1 = {0.7, 0, 0}, Agc2 = {0.7, 0, 0}, Agc3 = {0.7, 0, 0};
|
|
|
|
#if DEBUG
|
|
static int printit;
|
|
static time_t time0;
|
|
static double levelA=0, levelB=0, levelC=0, levelD=0, levelE=0;
|
|
|
|
if (time(NULL) != time0) {
|
|
time0 = time(NULL);
|
|
printit = 1;
|
|
}
|
|
else {
|
|
printit = 0;
|
|
}
|
|
#endif
|
|
if (nSamples <= 0)
|
|
return nSamples;
|
|
if (nSamples > size_dsamples) {
|
|
if (dsamples)
|
|
free(dsamples);
|
|
if (dsamples2)
|
|
free(dsamples2);
|
|
if (orig_cSamples)
|
|
free(orig_cSamples);
|
|
if (buf_cSamples)
|
|
free(buf_cSamples);
|
|
size_dsamples = nSamples * 2;
|
|
dsamples = (double *)malloc(size_dsamples * sizeof(double));
|
|
dsamples2 = (double *)malloc(size_dsamples * sizeof(double));
|
|
orig_cSamples = (complex double *)malloc(size_dsamples * sizeof(complex double));
|
|
buf_cSamples = (complex double *)malloc(size_dsamples * sizeof(complex double));
|
|
}
|
|
|
|
#if SAMPLES_FROM_FILE == 1
|
|
QuiskWavWriteC(&hWav, cSamples, nSamples);
|
|
#elif SAMPLES_FROM_FILE == 2
|
|
QuiskWavReadC(&hWav, cSamples, nSamples);
|
|
#endif
|
|
|
|
if (rxMode == CWL || rxMode == CWU) // Thanks to Robert, DM4RW
|
|
is_key_down = quisk_is_key_down();
|
|
else
|
|
is_key_down = quisk_transmit_mode || quisk_is_key_down();
|
|
orig_nSamples = nSamples;
|
|
if (split_rxtx) {
|
|
memcpy(orig_cSamples, cSamples, nSamples * sizeof(complex double));
|
|
if ( ! old_split_rxtx) // start of new split mode
|
|
Buffer2Chan(NULL, 0, NULL, 0);
|
|
}
|
|
if (multirx_play_channel != old_multirx_play_channel) // change in play channel
|
|
Buffer2Chan(NULL, 0, NULL, 0);
|
|
old_split_rxtx = split_rxtx;
|
|
old_multirx_play_channel = multirx_play_channel;
|
|
|
|
if (is_key_down && !isFDX) { // The key is down; replace this data block
|
|
dOutCounter += (double)nSamples * quisk_sound_state.playback_rate /
|
|
quisk_sound_state.sample_rate;
|
|
nout = (int)dOutCounter; // number of samples to output
|
|
dOutCounter -= nout;
|
|
playSilence = keyupDelayCode;
|
|
keyupEnvelope = 0;
|
|
i = 0; // whether to play the sidetone
|
|
if (rxMode == CWL || rxMode == CWU) {
|
|
if (quisk_use_rx_udp == 10) {
|
|
if (hardware_cwkey == 1)
|
|
i = 1;
|
|
}
|
|
else {
|
|
i = 1;
|
|
}
|
|
}
|
|
if (i) { // Play sidetone instead of radio for CW
|
|
if (! sidetoneIsOn) { // turn on sidetone
|
|
sidetoneIsOn = 1;
|
|
sidetoneEnvelope = 0;
|
|
sidetoneVector = BIG_VOLUME;
|
|
}
|
|
for (i = 0 ; i < nout; i++) {
|
|
if (sidetoneEnvelope < 1.0) {
|
|
sidetoneEnvelope += 1. / (quisk_sound_state.playback_rate * 5e-3); // 5 milliseconds
|
|
if (sidetoneEnvelope > 1.0)
|
|
sidetoneEnvelope = 1.0;
|
|
}
|
|
d = creal(sidetoneVector) * sidetoneVolume * sidetoneEnvelope;
|
|
cSamples[i] = d + I * d;
|
|
sidetoneVector *= sidetonePhase;
|
|
}
|
|
}
|
|
else { // Otherwise play silence
|
|
for (i = 0 ; i < nout; i++)
|
|
cSamples[i] = 0;
|
|
}
|
|
return nout;
|
|
}
|
|
// Key is up
|
|
if(sidetoneIsOn) { // decrease sidetone until it is off
|
|
dOutCounter += (double)nSamples * quisk_sound_state.playback_rate /
|
|
quisk_sound_state.sample_rate;
|
|
nout = (int)dOutCounter; // number of samples to output
|
|
dOutCounter -= nout;
|
|
for (i = 0; i < nout; i++) {
|
|
sidetoneEnvelope -= 1. / (quisk_sound_state.playback_rate * 5e-3); // 5 milliseconds
|
|
if (sidetoneEnvelope < 0) {
|
|
sidetoneIsOn = 0;
|
|
sidetoneEnvelope = 0;
|
|
break; // sidetone is zero
|
|
}
|
|
d = creal(sidetoneVector) * sidetoneVolume * sidetoneEnvelope;
|
|
cSamples[i] = d + I * d;
|
|
sidetoneVector *= sidetonePhase;
|
|
}
|
|
for ( ; i < nout; i++) { // continue with playSilence, even if zero
|
|
cSamples[i] = 0;
|
|
playSilence--;
|
|
}
|
|
return nout;
|
|
}
|
|
if (playSilence > 0) { // Continue to play silence after the key is up
|
|
dOutCounter += (double)nSamples * quisk_sound_state.playback_rate /
|
|
quisk_sound_state.sample_rate;
|
|
nout = (int)dOutCounter; // number of samples to output
|
|
dOutCounter -= nout;
|
|
for (i = 0; i < nout; i++)
|
|
cSamples[i] = 0;
|
|
playSilence -= nout;
|
|
return nout;
|
|
}
|
|
// We are done replacing sound with a sidetone or silence. Filter and
|
|
// demodulate the samples as radio sound.
|
|
|
|
// Add a test tone to the data
|
|
if (testtonePhase)
|
|
AddTestTone(cSamples, nSamples);
|
|
|
|
// Invert spectrum
|
|
if (quisk_invert_spectrum) {
|
|
for (i = 0; i < nSamples; i++) {
|
|
cSamples[i] = conj(cSamples[i]);
|
|
}
|
|
}
|
|
|
|
NoiseBlanker(cSamples, nSamples);
|
|
|
|
// Put samples into the fft input array.
|
|
// Thanks to WB4JFI for the code to add a third FFT buffer, July 2010.
|
|
// Changed to multiple FFTs May 2014.
|
|
if (multiple_sample_rates == 0) {
|
|
ptFFT = fft_data_array + fft_data_index;
|
|
for (i = 0; i < nSamples; i++) {
|
|
ptFFT->samples[ptFFT->index] = cSamples[i];
|
|
if (++(ptFFT->index) >= fft_size) { // check sample count
|
|
n = fft_data_index + 1; // next FFT data location
|
|
if (n >= FFT_ARRAY_SIZE)
|
|
n = 0;
|
|
if (fft_data_array[n].filled == 0) { // Is the next buffer empty?
|
|
fft_data_array[n].index = 0;
|
|
fft_data_array[n].block = 0;
|
|
fft_data_array[fft_data_index].filled = 1; // Mark the previous buffer ready.
|
|
fft_data_index = n; // Write samples into the new buffer.
|
|
ptFFT = fft_data_array + fft_data_index;
|
|
}
|
|
else { // no place to write samples
|
|
ptFFT->index = 0;
|
|
fft_error++;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Tune the data to frequency
|
|
if (multiple_sample_rates == 0)
|
|
tune = rx_tune_freq;
|
|
else
|
|
tune = rx_tune_freq + vfo_screen - vfo_audio;
|
|
if (tune != 0) {
|
|
phase = cexp((I * -2.0 * M_PI * tune) / quisk_sound_state.sample_rate);
|
|
for (i = 0; i < nSamples; i++) {
|
|
cSamples[i] *= rxTuneVector;
|
|
rxTuneVector *= phase;
|
|
}
|
|
}
|
|
|
|
if (rxMode == EXT) { // External filter and demodulate
|
|
d = (double)quisk_sound_state.sample_rate / quisk_sound_state.playback_rate; // total decimation needed
|
|
nSamples = quisk_extern_demod(cSamples, nSamples, d);
|
|
goto start_agc;
|
|
}
|
|
|
|
// Perhaps write sample data to the soundcard output without decimation
|
|
if (TEST_AUDIO == 1) { // Copy I channel capture to playback
|
|
di = 1.e4 * quisk_audioVolume;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = creal(cSamples[i]) * di;
|
|
return nSamples;
|
|
}
|
|
else if (TEST_AUDIO == 2) { // Copy Q channel capture to playback
|
|
di = 1.e4 * quisk_audioVolume;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = cimag(cSamples[i]) * di;
|
|
return nSamples;
|
|
}
|
|
#if DEBUG
|
|
for (i = 0; i < nSamples; i++) {
|
|
d = cabs(cSamples[i]);
|
|
if (levelA < d)
|
|
levelA = d;
|
|
}
|
|
#endif
|
|
|
|
nSamples = quisk_process_decimate(cSamples, nSamples, 0, rxMode);
|
|
|
|
#if DEBUG
|
|
for (i = 0; i < nSamples; i++) {
|
|
d = cabs(cSamples[i]);
|
|
if (levelB < d)
|
|
levelB = d;
|
|
}
|
|
#endif
|
|
|
|
if (measure_freq_mode)
|
|
measure_freq(cSamples, nSamples, quisk_decim_srate);
|
|
|
|
nSamples = quisk_process_demodulate(cSamples, dsamples, nSamples, 0, 0, rxMode);
|
|
|
|
squelch_real = 0; // keep track of the squelch for the two play channels
|
|
squelch_imag = 0;
|
|
if (rxMode == DGT_IQ) {
|
|
; // This mode is already stereo
|
|
}
|
|
else if (split_rxtx) { // Demodulate a second channel from the same receiver
|
|
phase = cexp((I * -2.0 * M_PI * (quisk_tx_tune_freq + rit_freq)) / quisk_sound_state.sample_rate);
|
|
// Tune the second channel to frequency
|
|
for (i = 0; i < orig_nSamples; i++) {
|
|
orig_cSamples[i] *= txTuneVector;
|
|
txTuneVector *= phase;
|
|
}
|
|
n = quisk_process_decimate(orig_cSamples, orig_nSamples, 1, rxMode);
|
|
n = quisk_process_demodulate(orig_cSamples, dsamples2, n, 1, 0, rxMode);
|
|
nSamples = Buffer2Chan(dsamples, nSamples, dsamples2, n); // buffer dsamples and dsamples2 so the count is equal
|
|
// dsamples was demodulated on bank 0, dsamples2 on bank 1
|
|
switch(split_rxtx) {
|
|
default:
|
|
case 1: // stereo, higher frequency is real
|
|
if (quisk_tx_tune_freq < rx_tune_freq) {
|
|
squelch_real = MeasureSquelch[0].squelch_active;
|
|
squelch_imag = MeasureSquelch[1].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples[i] + I * dsamples2[i];
|
|
}
|
|
else {
|
|
squelch_real = MeasureSquelch[1].squelch_active;
|
|
squelch_imag = MeasureSquelch[0].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples2[i] + I * dsamples[i];
|
|
}
|
|
break;
|
|
case 2: // stereo, lower frequency is real
|
|
if (quisk_tx_tune_freq >= rx_tune_freq) {
|
|
squelch_real = MeasureSquelch[0].squelch_active;
|
|
squelch_imag = MeasureSquelch[1].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples[i] + I * dsamples2[i];
|
|
}
|
|
else {
|
|
squelch_real = MeasureSquelch[1].squelch_active;
|
|
squelch_imag = MeasureSquelch[0].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples2[i] + I * dsamples[i];
|
|
}
|
|
break;
|
|
case 3: // mono receive channel
|
|
squelch_real = squelch_imag = MeasureSquelch[0].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples[i] + I * dsamples[i];
|
|
break;
|
|
case 4: // mono transmit channel
|
|
squelch_real = squelch_imag = MeasureSquelch[1].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples2[i] + I * dsamples2[i];
|
|
break;
|
|
}
|
|
}
|
|
else if (multirx_play_channel >= 0) { // Demodulate a second channel from a different receiver
|
|
memcpy(buf_cSamples, multirx_cSamples[multirx_play_channel], orig_nSamples * sizeof(complex double));
|
|
phase = cexp((I * -2.0 * M_PI * (multirx_freq[multirx_play_channel])) / quisk_sound_state.sample_rate);
|
|
// Tune the second channel to frequency
|
|
for (i = 0; i < orig_nSamples; i++) {
|
|
buf_cSamples[i] *= aux1TuneVector;
|
|
aux1TuneVector *= phase;
|
|
}
|
|
n = quisk_process_decimate(buf_cSamples, orig_nSamples, 1, multirx_mode[multirx_play_channel]);
|
|
n = quisk_process_demodulate(buf_cSamples, dsamples2, n, 1, 1, multirx_mode[multirx_play_channel]);
|
|
nSamples = Buffer2Chan(dsamples, nSamples, dsamples2, n); // buffer dsamples and dsamples2 so the count is equal
|
|
switch(multirx_play_method) {
|
|
default:
|
|
case 0: // play both
|
|
squelch_real = squelch_imag = MeasureSquelch[1].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples2[i] + I * dsamples2[i];
|
|
break;
|
|
case 1: // play left
|
|
squelch_real = MeasureSquelch[0].squelch_active;
|
|
squelch_imag = MeasureSquelch[1].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples[i] + I * dsamples2[i];
|
|
break;
|
|
case 2: // play right
|
|
squelch_real = MeasureSquelch[1].squelch_active;
|
|
squelch_imag = MeasureSquelch[0].squelch_active;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = dsamples2[i] + I * dsamples[i];
|
|
break;
|
|
}
|
|
}
|
|
else { // monophonic sound played on both channels
|
|
squelch_real = squelch_imag = MeasureSquelch[0].squelch_active;
|
|
for (i = 0; i < nSamples; i++) {
|
|
d = dsamples[i];
|
|
cSamples[i] = d + I * d;
|
|
}
|
|
}
|
|
|
|
// play sub-receiver 1 audio on a digital output device
|
|
rx_mode = multirx_mode[0];
|
|
if (quisk_multirx_count > 0 &&
|
|
(rx_mode == DGT_U || rx_mode == DGT_L || rx_mode == DGT_IQ || rx_mode == DGT_FM) &&
|
|
quisk_DigitalRx1Output.driver) {
|
|
phase = cexp((I * -2.0 * M_PI * (multirx_freq[0])) / quisk_sound_state.sample_rate);
|
|
// Tune the channel to frequency
|
|
for (i = 0; i < orig_nSamples; i++) {
|
|
multirx_cSamples[0][i] *= aux2TuneVector;
|
|
aux2TuneVector *= phase;
|
|
}
|
|
n = quisk_process_decimate(multirx_cSamples[0], orig_nSamples, 2, rx_mode);
|
|
n = quisk_process_demodulate(multirx_cSamples[0], dsamples2, n, 2, 2, rx_mode);
|
|
if (rx_mode == DGT_IQ) { // DGT-IQ
|
|
process_agc(&Agc3, multirx_cSamples[0], n, 1);
|
|
}
|
|
else {
|
|
for (i = 0; i < n; i++)
|
|
multirx_cSamples[0][i] = dsamples2[i] + I * dsamples2[i];
|
|
process_agc(&Agc3, multirx_cSamples[0], n, 0);
|
|
}
|
|
play_sound_interface(&quisk_DigitalRx1Output, n, multirx_cSamples[0], 1, 1.0);
|
|
}
|
|
|
|
// Perhaps decimate by an additional fraction
|
|
if (quisk_decim_srate != 48000) {
|
|
double_filter_decim = quisk_decim_srate / 48000.0;
|
|
nSamples = cFracDecim(cSamples, nSamples, double_filter_decim);
|
|
quisk_decim_srate = 48000;
|
|
}
|
|
|
|
// Interpolate the samples from 48000 sps to the play rate.
|
|
switch (quisk_sound_state.playback_rate / 48000) {
|
|
case 1:
|
|
break;
|
|
case 2:
|
|
nSamples = quisk_cInterp2HB45(cSamples, nSamples, &HalfBand7);
|
|
break;
|
|
case 4:
|
|
nSamples = quisk_cInterp2HB45(cSamples, nSamples, &HalfBand7);
|
|
nSamples = quisk_cInterp2HB45(cSamples, nSamples, &HalfBand8);
|
|
break;
|
|
case 8:
|
|
nSamples = quisk_cInterp2HB45(cSamples, nSamples, &HalfBand7);
|
|
nSamples = quisk_cInterp2HB45(cSamples, nSamples, &HalfBand8);
|
|
nSamples = quisk_cInterp2HB45(cSamples, nSamples, &HalfBand9);
|
|
break;
|
|
default:
|
|
printf ("Failure in quisk.c in integer interpolation %d %d\n", quisk_decim_srate, quisk_sound_state.playback_rate);
|
|
break;
|
|
}
|
|
|
|
// Find the peak signal amplitude
|
|
start_agc:
|
|
if (rxMode == EXT || rxMode == DGT_IQ) { // Ext and DGT-IQ stereo sound
|
|
process_agc(&Agc1, cSamples, nSamples, 1);
|
|
}
|
|
else if (rxMode == FDV_U || rxMode == FDV_L) { // Agc already done
|
|
;
|
|
}
|
|
else if (split_rxtx || multirx_play_channel >= 0) { // separate AGC for left and right channels
|
|
for (i = 0; i < nSamples; i++) {
|
|
orig_cSamples[i] = cimag(cSamples[i]);
|
|
cSamples[i] = creal(cSamples[i]);
|
|
}
|
|
process_agc(&Agc1, cSamples, nSamples, 0);
|
|
process_agc(&Agc2, orig_cSamples, nSamples, 0);
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = creal(cSamples[i]) + I * creal(orig_cSamples[i]);
|
|
}
|
|
else { // monophonic sound
|
|
process_agc(&Agc1, cSamples, nSamples, 0);
|
|
}
|
|
#if DEBUG
|
|
if (printit) {
|
|
d = CLIP32;
|
|
//printf ("Levels: %12.8lf %12.8lf %12.8lf %12.8lf %12.8lf\n",
|
|
// levelA/d, levelB/d, levelC/d, levelD/d, levelE/d);
|
|
levelA = levelB = levelC = levelD = levelE = 0;
|
|
}
|
|
#endif
|
|
|
|
if (kill_audio) {
|
|
squelch_real = squelch_imag = 1;
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = 0;
|
|
}
|
|
else if (squelch_real && squelch_imag) {
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = 0;
|
|
}
|
|
else if (squelch_imag) {
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = creal(cSamples[i]);
|
|
}
|
|
else if (squelch_real) {
|
|
for (i = 0; i < nSamples; i++)
|
|
cSamples[i] = I * cimag(cSamples[i]);
|
|
}
|
|
if (keyupEnvelope < 1.0) { // raise volume slowly after the key goes up
|
|
di = 1. / (quisk_sound_state.playback_rate * 5e-3); // 5 milliseconds
|
|
for (i = 0; i < nSamples; i++) {
|
|
keyupEnvelope += di;
|
|
if (keyupEnvelope > 1.0) {
|
|
keyupEnvelope = 1.0;
|
|
break;
|
|
}
|
|
cSamples[i] *= keyupEnvelope;
|
|
}
|
|
}
|
|
if (quisk_record_state == RECORD_RADIO && ! (squelch_real && squelch_imag))
|
|
quisk_tmp_record(cSamples, nSamples, 1.0); // save radio sound
|
|
return nSamples;
|
|
}
|
|
|
|
static PyObject * get_state(PyObject * self, PyObject * args)
|
|
{
|
|
int unused = 0;
|
|
|
|
if (args && !PyArg_ParseTuple (args, "")) // args=NULL internal call
|
|
return NULL;
|
|
return Py_BuildValue("iiiiiNiNiiiiiiiii",
|
|
quisk_sound_state.rate_min,
|
|
quisk_sound_state.rate_max,
|
|
quisk_sound_state.sample_rate,
|
|
quisk_sound_state.chan_min,
|
|
quisk_sound_state.chan_max,
|
|
PyUnicode_DecodeUTF8(quisk_sound_state.msg1, strlen(quisk_sound_state.msg1), "replace"),
|
|
unused,
|
|
PyUnicode_DecodeUTF8(quisk_sound_state.err_msg, strlen(quisk_sound_state.err_msg), "replace"),
|
|
quisk_sound_state.read_error,
|
|
quisk_sound_state.write_error,
|
|
quisk_sound_state.underrun_error,
|
|
quisk_sound_state.latencyCapt,
|
|
quisk_sound_state.latencyPlay,
|
|
quisk_sound_state.interupts,
|
|
fft_error,
|
|
mic_max_display,
|
|
quisk_sound_state.data_poll_usec
|
|
);
|
|
}
|
|
|
|
static PyObject * get_squelch(PyObject * self, PyObject * args)
|
|
{
|
|
int freq;
|
|
|
|
if (!PyArg_ParseTuple (args, "i", &freq))
|
|
return NULL;
|
|
return PyInt_FromLong(IsSquelch(freq));
|
|
}
|
|
|
|
static PyObject * get_overrange(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
return PyInt_FromLong(quisk_get_overrange());
|
|
}
|
|
|
|
static PyObject * get_filter_rate(PyObject * self, PyObject * args)
|
|
{ // Return the filter sample rate as used by quisk_process_samples.
|
|
// Changes to quisk_process_decimate or quisk_process_demodulate will require changes here.
|
|
int rate, decim_srate, filter_srate, mode, bandwidth;
|
|
// mode is -1 to use the rxMode
|
|
if (!PyArg_ParseTuple (args, "ii", &mode, &bandwidth))
|
|
return NULL;
|
|
rate = quisk_sound_state.sample_rate;
|
|
switch((rate + 100) / 1000) {
|
|
case 41:
|
|
decim_srate = 48000;
|
|
case 53: // SDR-IQ
|
|
decim_srate = rate;
|
|
break;
|
|
case 111: // SDR-IQ
|
|
decim_srate = rate / 2;
|
|
break;
|
|
case 133: // SDR-IQ
|
|
decim_srate = rate / 2;
|
|
break;
|
|
case 185: // SDR-IQ
|
|
decim_srate = rate / 3;
|
|
break;
|
|
case 370:
|
|
decim_srate = rate / 6;
|
|
break;
|
|
case 740:
|
|
decim_srate = rate / 12;
|
|
break;
|
|
case 1333:
|
|
decim_srate = rate / 24;
|
|
break;
|
|
default:
|
|
decim_srate = PlanDecimation(NULL, NULL, NULL);
|
|
break;
|
|
}
|
|
if (mode < 0) {
|
|
mode = rxMode;
|
|
bandwidth = filter_bandwidth[0];
|
|
}
|
|
switch(mode) {
|
|
case CWL: // lower sideband CW at 6 ksps
|
|
case CWU: // upper sideband CW at 6 ksps
|
|
filter_srate = decim_srate / 8;
|
|
break;
|
|
case LSB: // lower sideband SSB at 12 ksps
|
|
case USB: // upper sideband SSB at 12 ksps
|
|
default:
|
|
filter_srate = decim_srate / 4;
|
|
break;
|
|
case AM: // AM at 24 ksps
|
|
filter_srate = decim_srate / 2;
|
|
break;
|
|
case FM: // FM at 48 ksps
|
|
case DGT_FM: // digital FM at 48 ksps
|
|
filter_srate = decim_srate;
|
|
break;
|
|
case DGT_U: // digital modes DGT-*
|
|
case DGT_L:
|
|
if (bandwidth < DGT_NARROW_FREQ)
|
|
filter_srate = decim_srate / 8;
|
|
else
|
|
filter_srate = decim_srate;
|
|
break;
|
|
case DGT_IQ: // digital mode at 48 ksps
|
|
filter_srate = decim_srate;
|
|
break;
|
|
case FDV_U: // digital voice at 8 ksps
|
|
case FDV_L:
|
|
filter_srate = 8000;
|
|
break;
|
|
}
|
|
//printf("Filter rate %d\n", filter_srate);
|
|
return PyInt_FromLong(filter_srate);
|
|
}
|
|
|
|
static PyObject * get_smeter(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
return PyFloat_FromDouble(Smeter);
|
|
}
|
|
|
|
static PyObject * get_hardware_ptt(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
return PyInt_FromLong(hardware_ptt);
|
|
}
|
|
|
|
static PyObject * get_hermes_adc(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
return PyFloat_FromDouble(hermes_adc_level);
|
|
}
|
|
|
|
static void init_bandscope(void)
|
|
{
|
|
int i, j;
|
|
|
|
if (bandscope_size > 0) {
|
|
bandscopeSamples = (double *)malloc(bandscope_size * sizeof(double));
|
|
bandscopeWindow = (double *)malloc(bandscope_size * sizeof(double));
|
|
bandscopeAverage = (double *)malloc((bandscope_size / 2 + 1 + 1) * sizeof(double));
|
|
bandscopeFFT = (complex double *)malloc((bandscope_size / 2 + 1) * sizeof(complex double));
|
|
bandscopePlan = fftw_plan_dft_r2c_1d(bandscope_size, bandscopeSamples, bandscopeFFT, FFTW_MEASURE);
|
|
// Create the fft window
|
|
for (i = 0, j = -bandscope_size / 2; i < bandscope_size; i++, j++)
|
|
bandscopeWindow[i] = 0.5 + 0.5 * cos(2. * M_PI * j / bandscope_size); // Hanning
|
|
// zero the average array
|
|
for (i = 0; i < bandscope_size / 2 + 1; i++)
|
|
bandscopeAverage[i] = 0;
|
|
}
|
|
}
|
|
|
|
static PyObject * add_rx_samples(PyObject * self, PyObject * args)
|
|
{
|
|
int i;
|
|
int ii, qq; // ii, qq must be four bytes
|
|
unsigned char * pt_ii;
|
|
unsigned char * pt_qq;
|
|
Py_buffer view;
|
|
PyObject * samples;
|
|
|
|
if (!PyArg_ParseTuple (args, "O", &samples))
|
|
return NULL;
|
|
if ( ! PyObject_CheckBuffer(samples)) {
|
|
printf("add_rx_samples: Invalid object sent as samples\n");
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
if (PyObject_GetBuffer(samples, &view, PyBUF_SIMPLE) != 0) {
|
|
printf("add_rx_samples: Can not view sample buffer\n");
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
if (view.len % (py_sample_rx_bytes * 2) != 0) {
|
|
printf ("add_rx_samples: Odd number of bytes in sample buffer\n");
|
|
}
|
|
else if (PySampleCount + view.len / py_sample_rx_bytes / 2 > SAMP_BUFFER_SIZE * 8 / 10) {
|
|
printf ("add_rx_samples: buffer is too full\n");
|
|
}
|
|
else if (py_sample_rx_endian == 0) { // byte order of samples is little-endian
|
|
void * buf;
|
|
void * buf_end;
|
|
buf = view.buf;
|
|
buf_end = buf + view.len;
|
|
pt_ii = (unsigned char *)&ii + 4 - py_sample_rx_bytes;
|
|
pt_qq = (unsigned char *)&qq + 4 - py_sample_rx_bytes;
|
|
while (buf < buf_end) {
|
|
ii = qq = 0;
|
|
memcpy(pt_ii, buf, py_sample_rx_bytes);
|
|
buf += py_sample_rx_bytes;
|
|
memcpy(pt_qq, buf, py_sample_rx_bytes);
|
|
buf += py_sample_rx_bytes;
|
|
PySampleBuf[PySampleCount++] = ii + qq * I;
|
|
}
|
|
}
|
|
else { // byte order of samples is big-endian
|
|
unsigned char * buf;
|
|
unsigned char * buf_end;
|
|
buf = view.buf;
|
|
buf_end = buf + view.len;
|
|
while (buf < buf_end) {
|
|
ii = qq = 0;
|
|
pt_ii = (unsigned char *)&ii + 3;
|
|
pt_qq = (unsigned char *)&qq + 3;
|
|
for (i = 0; i < py_sample_rx_bytes; i++)
|
|
*pt_ii-- = *buf++;
|
|
for (i = 0; i < py_sample_rx_bytes; i++)
|
|
*pt_qq-- = *buf++;
|
|
PySampleBuf[PySampleCount++] = ii + qq * I;
|
|
}
|
|
}
|
|
PyBuffer_Release(&view);
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * add_bscope_samples(PyObject * self, PyObject * args)
|
|
{
|
|
int i, count;
|
|
int ii; // ii must be four bytes
|
|
unsigned char * pt_ii;
|
|
Py_buffer view;
|
|
PyObject * samples;
|
|
|
|
if (!PyArg_ParseTuple (args, "O", &samples))
|
|
return NULL;
|
|
if (bandscope_size <= 0) {
|
|
printf("add_bscope_samples: The bandscope was not initialized with InitBscope()\n");
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
if ( ! PyObject_CheckBuffer(samples)) {
|
|
printf("add_bscope_samples: Invalid object sent as samples\n");
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
if (PyObject_GetBuffer(samples, &view, PyBUF_SIMPLE) != 0) {
|
|
printf("add_bscope_samples: Can not view sample buffer\n");
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
count = 0;
|
|
if (view.len != bandscope_size * py_bscope_bytes) {
|
|
printf ("add_bscope_samples: Wrong number of bytes in sample buffer\n");
|
|
}
|
|
else if (py_bscope_endian == 0) { // byte order of samples is little-endian
|
|
void * buf;
|
|
void * buf_end;
|
|
buf = view.buf;
|
|
buf_end = buf + view.len;
|
|
pt_ii = (unsigned char *)&ii + 4 - py_bscope_bytes;
|
|
while (buf < buf_end) {
|
|
ii = 0;
|
|
memcpy(pt_ii, buf, py_bscope_bytes);
|
|
buf += py_bscope_bytes;
|
|
bandscopeSamples[count++] = (double)ii / CLIP32;
|
|
}
|
|
}
|
|
else { // byte order of samples is big-endian
|
|
unsigned char * buf;
|
|
unsigned char * buf_end;
|
|
buf = view.buf;
|
|
buf_end = buf + view.len;
|
|
while (buf < buf_end) {
|
|
ii = 0;
|
|
pt_ii = (unsigned char *)&ii + 3;
|
|
for (i = 0; i < py_bscope_bytes; i++)
|
|
*pt_ii-- = *buf++;
|
|
bandscopeSamples[count++] = (double)ii / CLIP32;
|
|
}
|
|
}
|
|
PyBuffer_Release(&view);
|
|
bandscopeState = 99;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static void py_sample_start(void)
|
|
{
|
|
}
|
|
|
|
static void py_sample_stop(void)
|
|
{
|
|
if (bandscopePlan) {
|
|
fftw_destroy_plan(bandscopePlan);
|
|
bandscopePlan = NULL;
|
|
}
|
|
}
|
|
|
|
static int py_sample_read(complex double * cSamples)
|
|
{
|
|
int n;
|
|
|
|
memcpy(cSamples, PySampleBuf, PySampleCount * sizeof(complex double));
|
|
n = PySampleCount;
|
|
PySampleCount = 0;
|
|
return n;
|
|
}
|
|
|
|
static PyObject * set_params(PyObject * self, PyObject * args, PyObject * keywds)
|
|
{ /* Call with keyword arguments ONLY; change local parameters */
|
|
static char * kwlist[] = {"quisk_is_vna", "rx_bytes", "rx_endian", "read_error", "clip",
|
|
"bscope_bytes", "bscope_endian", "bscope_size", "bandscopeScale", "hermes_pause", NULL} ;
|
|
int i, nbytes, read_error, clip, bscope_size, hermes_pause;
|
|
|
|
nbytes = read_error = clip = bscope_size = hermes_pause = -1;
|
|
if (!PyArg_ParseTupleAndKeywords (args, keywds, "|iiiiiiiidi", kwlist,
|
|
&quisk_is_vna, &nbytes, &py_sample_rx_endian, &read_error, &clip,
|
|
&py_bscope_bytes, &py_bscope_endian, &bscope_size, &bandscopeScale, &hermes_pause))
|
|
return NULL;
|
|
if (nbytes != -1) {
|
|
py_sample_rx_bytes = nbytes;
|
|
quisk_sample_source4(py_sample_start, py_sample_stop, py_sample_read, NULL);
|
|
}
|
|
if (read_error != -1)
|
|
quisk_sound_state.read_error++;
|
|
if (clip != -1)
|
|
quisk_sound_state.overrange++;
|
|
if (bscope_size > 0) {
|
|
if (bandscope_size == 0) {
|
|
bandscope_size = bscope_size;
|
|
init_bandscope();
|
|
}
|
|
else if (bscope_size != bandscope_size) {
|
|
printf ("Illegal attempt to change bscope_size\n");
|
|
}
|
|
}
|
|
if (hermes_pause != -1) {
|
|
i = quisk_multirx_state;
|
|
if (hermes_pause) { // pause the hermes samples
|
|
if (quisk_multirx_state < 20)
|
|
quisk_multirx_state = 20;
|
|
}
|
|
else { // resume the hermes samples
|
|
if (quisk_multirx_state >= 20)
|
|
quisk_multirx_state = 0;
|
|
}
|
|
return PyInt_FromLong(i);
|
|
}
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * get_hermes_TFRC(PyObject * self, PyObject * args)
|
|
{ // return average temperature, forward and reverse power and current
|
|
PyObject * ret;
|
|
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
if (hermes_count_temperature > 0) {
|
|
hermes_temperature /= hermes_count_temperature;
|
|
hermes_fwd_power /= hermes_count_temperature;
|
|
}
|
|
else {
|
|
hermes_temperature = 0.0;
|
|
hermes_fwd_power = 0.0;
|
|
}
|
|
if (hermes_count_current > 0) {
|
|
hermes_rev_power /= hermes_count_current;
|
|
hermes_pa_current /= hermes_count_current;
|
|
}
|
|
else {
|
|
hermes_rev_power = 0.0;
|
|
hermes_pa_current = 0.0;
|
|
}
|
|
ret = Py_BuildValue("dddd", hermes_temperature, hermes_fwd_power, hermes_rev_power, hermes_pa_current);
|
|
hermes_temperature = 0;
|
|
hermes_fwd_power = 0;
|
|
hermes_rev_power = 0;
|
|
hermes_pa_current = 0;
|
|
hermes_count_temperature = 0;
|
|
hermes_count_current = 0;
|
|
return ret;
|
|
}
|
|
|
|
static PyObject * measure_frequency(PyObject * self, PyObject * args)
|
|
{
|
|
int mode;
|
|
|
|
if (!PyArg_ParseTuple (args, "i", &mode))
|
|
return NULL;
|
|
if (mode >= 0) // mode >= 0 set the mode; mode < 0, just return the frequency
|
|
measure_freq_mode = mode;
|
|
return PyFloat_FromDouble(measured_frequency);
|
|
}
|
|
|
|
static PyObject * measure_audio(PyObject * self, PyObject * args)
|
|
{
|
|
int time;
|
|
|
|
if (!PyArg_ParseTuple (args, "i", &time))
|
|
return NULL;
|
|
if (time > 0) // set the average time
|
|
measure_audio_time = time;
|
|
return PyFloat_FromDouble(measured_audio);
|
|
}
|
|
|
|
static PyObject * add_tone(PyObject * self, PyObject * args)
|
|
{ /* Add a test tone to the captured audio data */
|
|
int freq;
|
|
|
|
if (!PyArg_ParseTuple (args, "i", &freq))
|
|
return NULL;
|
|
if (freq && quisk_sound_state.sample_rate)
|
|
testtonePhase = cexp((I * 2.0 * M_PI * freq) / quisk_sound_state.sample_rate);
|
|
else
|
|
testtonePhase = 0;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * open_key(PyObject * self, PyObject * args)
|
|
{
|
|
const char * name;
|
|
|
|
if (!PyArg_ParseTuple (args, "s", &name))
|
|
return NULL;
|
|
|
|
return PyInt_FromLong(quisk_open_key(name));
|
|
}
|
|
|
|
static void close_udp(void)
|
|
{
|
|
short msg = 0x7373; // shutdown
|
|
|
|
quisk_using_udp = 0;
|
|
if (rx_udp_socket != INVALID_SOCKET) {
|
|
shutdown(rx_udp_socket, QUISK_SHUT_RD);
|
|
send(rx_udp_socket, (char *)&msg, 2, 0);
|
|
send(rx_udp_socket, (char *)&msg, 2, 0);
|
|
QuiskSleepMicrosec(3000000);
|
|
close(rx_udp_socket);
|
|
rx_udp_socket = INVALID_SOCKET;
|
|
}
|
|
quisk_rx_udp_started = 0;
|
|
#ifdef MS_WINDOWS
|
|
if (cleanupWSA) {
|
|
cleanupWSA = 0;
|
|
WSACleanup();
|
|
}
|
|
#endif
|
|
}
|
|
|
|
static void close_udp10(void) // Metis-Hermes protocol
|
|
{
|
|
int i;
|
|
unsigned char buf[64];
|
|
|
|
quisk_using_udp = 0;
|
|
if (rx_udp_socket != INVALID_SOCKET) {
|
|
shutdown(rx_udp_socket, QUISK_SHUT_RD);
|
|
buf[0] = 0xEF;
|
|
buf[1] = 0xFE;
|
|
buf[2] = 0x04;
|
|
buf[3] = 0x00;
|
|
for (i = 4; i < 64; i++)
|
|
buf[i] = 0;
|
|
send(rx_udp_socket, (char *)buf, 64, 0);
|
|
QuiskSleepMicrosec(5000);
|
|
send(rx_udp_socket, (char *)buf, 64, 0);
|
|
QuiskSleepMicrosec(2000000);
|
|
close(rx_udp_socket);
|
|
rx_udp_socket = INVALID_SOCKET;
|
|
}
|
|
quisk_rx_udp_started = 0;
|
|
quisk_multirx_state = 0;
|
|
if (bandscopePlan) {
|
|
fftw_destroy_plan(bandscopePlan);
|
|
bandscopePlan = NULL;
|
|
}
|
|
#ifdef MS_WINDOWS
|
|
if (cleanupWSA) {
|
|
cleanupWSA = 0;
|
|
WSACleanup();
|
|
}
|
|
#endif
|
|
}
|
|
|
|
static PyObject * close_rx_udp(PyObject * self, PyObject * args)
|
|
{ // Not necessary to call from Python because close_udp() is called from sound.c
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
//close_udp();
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static int quisk_read_rx_udp(complex double * samp) // Read samples from UDP
|
|
{ // Size of complex sample array is SAMP_BUFFER_SIZE
|
|
ssize_t bytes;
|
|
unsigned char buf[1500]; // Maximum Ethernet is 1500 bytes.
|
|
static unsigned char seq0; // must be 8 bits
|
|
int i, nSamples, xr, xi, index, want_samples;
|
|
unsigned char * ptxr, * ptxi;
|
|
struct timeval tm_wait;
|
|
fd_set fds;
|
|
|
|
// Data from the receiver is little-endian
|
|
if ( ! rx_udp_gain_correct) {
|
|
int dec;
|
|
dec = (int)(rx_udp_clock / quisk_sound_state.sample_rate + 0.5);
|
|
if ((dec / 5) * 5 == dec) // Decimation by a factor of 5
|
|
rx_udp_gain_correct = 1.31072;
|
|
else // Decimation by factors of two
|
|
rx_udp_gain_correct = 1.0;
|
|
}
|
|
if ( ! quisk_rx_udp_started) { // we never received any data
|
|
// send our return address until we receive UDP blocks
|
|
tm_wait.tv_sec = 0;
|
|
tm_wait.tv_usec = 5000;
|
|
FD_ZERO (&fds);
|
|
FD_SET (rx_udp_socket, &fds);
|
|
if (select (rx_udp_socket + 1, &fds, NULL, NULL, &tm_wait) == 1) { // see if data is available
|
|
bytes = recv(rx_udp_socket, (char *)buf, 1500, 0); // throw away the first block
|
|
seq0 = buf[0] + 1; // Next expected sequence number
|
|
quisk_rx_udp_started = 1;
|
|
#if DEBUG_IO
|
|
printf("Udp data started\n");
|
|
#endif
|
|
}
|
|
else { // send our return address to the sample source
|
|
buf[0] = buf[1] = 0x72; // UDP command "register return address"
|
|
send(rx_udp_socket, (char *)buf, 2, 0);
|
|
return 0;
|
|
}
|
|
}
|
|
nSamples = 0;
|
|
want_samples = (int)(quisk_sound_state.data_poll_usec * 1e-6 * quisk_sound_state.sample_rate + 0.5);
|
|
while (nSamples < want_samples) { // read several UDP blocks
|
|
tm_wait.tv_sec = 0;
|
|
tm_wait.tv_usec = 100000; // Linux seems to have problems with very small time intervals
|
|
FD_ZERO (&fds);
|
|
FD_SET (rx_udp_socket, &fds);
|
|
i = select (rx_udp_socket + 1, &fds, NULL, NULL, &tm_wait);
|
|
if (i == 1)
|
|
;
|
|
else if (i == 0) {
|
|
#if DEBUG_IO
|
|
printf("Udp socket timeout\n");
|
|
#endif
|
|
return 0;
|
|
}
|
|
else {
|
|
#if DEBUG_IO
|
|
printf("Udp select error %d\n", i);
|
|
#endif
|
|
return 0;
|
|
}
|
|
bytes = recv(rx_udp_socket, (char *)buf, 1500, 0); // blocking read
|
|
if (bytes != RX_UDP_SIZE) { // Known size of sample block
|
|
quisk_sound_state.read_error++;
|
|
#if DEBUG_IO
|
|
printf("read_rx_udp: Bad block size\n");
|
|
#endif
|
|
continue;
|
|
}
|
|
// buf[0] is the sequence number
|
|
// buf[1] is the status:
|
|
// bit 0: key up/down state
|
|
// bit 1: set for ADC overrange (clip)
|
|
if (buf[0] != seq0) {
|
|
#if DEBUG_IO
|
|
printf("read_rx_udp: Bad sequence want %3d got %3d\n",
|
|
(unsigned int)seq0, (unsigned int)buf[0]);
|
|
#endif
|
|
quisk_sound_state.read_error++;
|
|
}
|
|
seq0 = buf[0] + 1; // Next expected sequence number
|
|
quisk_set_key_down(buf[1] & 0x01); // bit zero is key state
|
|
if (buf[1] & 0x02) // bit one is ADC overrange
|
|
quisk_sound_state.overrange++;
|
|
index = 2;
|
|
ptxr = (unsigned char *)&xr;
|
|
ptxi = (unsigned char *)ξ
|
|
// convert 24-bit samples to 32-bit samples; int must be 32 bits.
|
|
if (is_little_endian) {
|
|
while (index < bytes) { // This works for 3, 2, 1 byte samples
|
|
xr = xi = 0;
|
|
memcpy (ptxr + (4 - sample_bytes), buf + index, sample_bytes);
|
|
index += sample_bytes;
|
|
memcpy (ptxi + (4 - sample_bytes), buf + index, sample_bytes);
|
|
index += sample_bytes;
|
|
samp[nSamples++] = (xr + xi * I) * rx_udp_gain_correct;
|
|
xr = xi = 0;
|
|
memcpy (ptxr + (4 - sample_bytes), buf + index, sample_bytes);
|
|
index += sample_bytes;
|
|
memcpy (ptxi + (4 - sample_bytes), buf + index, sample_bytes);
|
|
index += sample_bytes;
|
|
samp[nSamples++] = (xr + xi * I) * rx_udp_gain_correct;
|
|
}
|
|
}
|
|
else { // big-endian
|
|
while (index < bytes) { // This works for 3-byte samples only
|
|
*(ptxr ) = buf[index + 2];
|
|
*(ptxr + 1) = buf[index + 1];
|
|
*(ptxr + 2) = buf[index ];
|
|
*(ptxr + 3) = 0;
|
|
index += 3;
|
|
*(ptxi ) = buf[index + 2];
|
|
*(ptxi + 1) = buf[index + 1];
|
|
*(ptxi + 2) = buf[index ];
|
|
*(ptxi + 3) = 0;
|
|
index += 3;
|
|
samp[nSamples++] = (xr + xi * I) * rx_udp_gain_correct;;
|
|
*(ptxr ) = buf[index + 2];
|
|
*(ptxr + 1) = buf[index + 1];
|
|
*(ptxr + 2) = buf[index ];
|
|
*(ptxr + 3) = 0;
|
|
index += 3;
|
|
*(ptxi ) = buf[index + 2];
|
|
*(ptxi + 1) = buf[index + 1];
|
|
*(ptxi + 2) = buf[index ];
|
|
*(ptxi + 3) = 0;
|
|
index += 3;
|
|
samp[nSamples++] = (xr + xi * I) * rx_udp_gain_correct;;
|
|
|
|
}
|
|
}
|
|
}
|
|
return nSamples;
|
|
}
|
|
|
|
static int quisk_hermes_is_ready(int rx_udp_socket)
|
|
{ // Start Hermes; return 1 when we are ready to receive data
|
|
unsigned char buf[1500];
|
|
int i, dummy;
|
|
struct timeval tm_wait;
|
|
fd_set fds;
|
|
|
|
if (rx_udp_socket == INVALID_SOCKET)
|
|
return 0;
|
|
switch (quisk_multirx_state) {
|
|
case 0: // Start or restart
|
|
case 20: // Temporary shutdown
|
|
quisk_rx_udp_started = 0;
|
|
buf[0] = 0xEF;
|
|
buf[1] = 0xFE;
|
|
buf[2] = 0x04;
|
|
buf[3] = 0x00;
|
|
for (i = 4; i < 64; i++)
|
|
buf[i] = 0;
|
|
send(rx_udp_socket, (char *)buf, 64, 0); // send Stop
|
|
quisk_multirx_state++;
|
|
QuiskSleepMicrosec(2000);
|
|
return 0;
|
|
case 1:
|
|
case 21:
|
|
buf[0] = 0xEF;
|
|
buf[1] = 0xFE;
|
|
buf[2] = 0x04;
|
|
buf[3] = 0x00;
|
|
for (i = 4; i < 64; i++)
|
|
buf[i] = 0;
|
|
send(rx_udp_socket, (char *)buf, 64, 0); // send Stop
|
|
quisk_multirx_state++;
|
|
QuiskSleepMicrosec(9000);
|
|
return 0;
|
|
case 2:
|
|
case 22:
|
|
while (1) {
|
|
tm_wait.tv_sec = 0; // throw away all pending records
|
|
tm_wait.tv_usec = 0;
|
|
FD_ZERO (&fds);
|
|
FD_SET (rx_udp_socket, &fds);
|
|
if (select (rx_udp_socket + 1, &fds, NULL, NULL, &tm_wait) != 1)
|
|
break;
|
|
recv(rx_udp_socket, (char *)buf, 1500, 0);
|
|
}
|
|
// change to state 3 for startup
|
|
// change to state 23 for temporary shutdown
|
|
quisk_multirx_state++;
|
|
return 0;
|
|
case 3:
|
|
quisk_multirx_count = quisk_pc_to_hermes[3] >> 3 & 0x7; // number of receivers
|
|
for (i = 0; i < quisk_multirx_count; i++)
|
|
if ( ! multirx_fft_data[i].samples) // Check that buffer exists
|
|
multirx_fft_data[i].samples = (fftw_complex *)malloc(multirx_fft_width * sizeof(fftw_complex));
|
|
quisk_hermes_tx_send(0, NULL);
|
|
quisk_multirx_state++;
|
|
return 0;
|
|
case 4:
|
|
case 5:
|
|
case 6:
|
|
case 7:
|
|
dummy = 999999; // enable transmit
|
|
quisk_hermes_tx_send(rx_udp_socket, &dummy); // send packets with number of receivers
|
|
quisk_multirx_state++;
|
|
QuiskSleepMicrosec(2000);
|
|
return 0;
|
|
case 8:
|
|
if (quisk_rx_udp_started) {
|
|
quisk_multirx_state++;
|
|
}
|
|
else {
|
|
// send our return address until we receive UDP blocks
|
|
buf[0] = 0xEF;
|
|
buf[1] = 0xFE;
|
|
buf[2] = 0x04;
|
|
if (enable_bandscope)
|
|
buf[3] = 0x03;
|
|
else
|
|
buf[3] = 0x01;
|
|
for (i = 4; i < 64; i++)
|
|
buf[i] = 0;
|
|
send(rx_udp_socket, (char *)buf, 64, 0);
|
|
QuiskSleepMicrosec(2000);
|
|
}
|
|
return 1;
|
|
case 9: // running state; we have received UDP blocks
|
|
default:
|
|
return 1;
|
|
case 23: // we are in a temporary shutdown
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
static int read_rx_udp10(complex double * samp) // Read samples from UDP using the Hermes protocol.
|
|
{ // Size of complex sample array is SAMP_BUFFER_SIZE. Called from the sound thread.
|
|
ssize_t bytes;
|
|
unsigned char buf[1500];
|
|
unsigned int seq;
|
|
unsigned int hlwp = 0;
|
|
static unsigned int seq0;
|
|
static int tx_records;
|
|
static int max_multirx_count=0;
|
|
int i, j, nSamples, xr, xi, index, start, want_samples, dindex, num_records;
|
|
complex double c;
|
|
struct timeval tm_wait;
|
|
fd_set fds;
|
|
|
|
if ( ! quisk_hermes_is_ready(rx_udp_socket)) {
|
|
seq0 = 0;
|
|
tx_records = 0;
|
|
quisk_rx_udp_started = 0;
|
|
multirx_fft_next_index = 0;
|
|
multirx_fft_next_state = 0;
|
|
for (i = 0; i < QUISK_MAX_RECEIVERS; i++)
|
|
multirx_fft_data[i].index = 0;
|
|
return 0;
|
|
}
|
|
nSamples = 0;
|
|
want_samples = (int)(quisk_sound_state.data_poll_usec * 1e-6 * quisk_sound_state.sample_rate + 0.5);
|
|
num_records = 504 / ((quisk_multirx_count + 1) * 6 + 2); // number of samples in each of two blocks for each receiver
|
|
if (quisk_multirx_count) {
|
|
if (multirx_sample_size < want_samples + 2000) {
|
|
multirx_sample_size = want_samples * 2 + 2000;
|
|
for (i = 0; i < max_multirx_count; i++) {
|
|
free(multirx_cSamples[i]);
|
|
multirx_cSamples[i] = (complex double *)malloc(multirx_sample_size * sizeof(complex double));
|
|
}
|
|
}
|
|
if (quisk_multirx_count > max_multirx_count) {
|
|
for (i = max_multirx_count; i < quisk_multirx_count; i++)
|
|
multirx_cSamples[i] = (complex double *)malloc(multirx_sample_size * sizeof(complex double));
|
|
max_multirx_count = quisk_multirx_count;
|
|
}
|
|
}
|
|
while (nSamples < want_samples) { // read several UDP blocks
|
|
tm_wait.tv_sec = 0;
|
|
tm_wait.tv_usec = 100000; // Linux seems to have problems with very small time intervals
|
|
FD_ZERO (&fds);
|
|
FD_SET (rx_udp_socket, &fds);
|
|
i = select (rx_udp_socket + 1, &fds, NULL, NULL, &tm_wait); // blocking wait
|
|
if (i == 1)
|
|
;
|
|
else if (i == 0) {
|
|
#if DEBUG_IO
|
|
printf("Udp socket timeout\n");
|
|
#endif
|
|
return 0;
|
|
}
|
|
else {
|
|
#if DEBUG_IO
|
|
printf("Udp select error %d\n", i);
|
|
#endif
|
|
return 0;
|
|
}
|
|
bytes = recv(rx_udp_socket, (char *)buf, 1500, 0); // blocking read
|
|
if (bytes != 1032 || buf[0] != 0xEF || buf[1] != 0xFE || buf[2] != 0x01) { // Known size of sample block
|
|
quisk_sound_state.read_error++;
|
|
#if DEBUG_IO
|
|
printf("read_rx_udp10: Bad block size %d or header\n", (int)bytes);
|
|
#endif
|
|
return 0;
|
|
}
|
|
//// Bandscope data - reversed byte order ?????
|
|
if (buf[3] == 0x04 && bandscopeSamples) { // ADC samples for bandscope
|
|
seq = buf[7]; // sequence number
|
|
seq = seq & (bandscopeBlockCount - 1); // 0, 1, 2, ...
|
|
switch (bandscopeState) {
|
|
case 0: // Start - wait for the start of a block and record block one
|
|
if (seq == 0) {
|
|
for (i = 0, j = 8; i < 512; i++, j+= 2)
|
|
bandscopeSamples[i] = ((double)(short)(buf[j + 1] << 8 | buf[j])) / bandscopeScale;
|
|
bandscopeState = 1;
|
|
}
|
|
break;
|
|
default:
|
|
case 1: // Record blocks
|
|
if (seq == bandscopeState) {
|
|
for (i = 0, j = 8; i < 512; i++, j+= 2)
|
|
bandscopeSamples[i + 512 * seq] = ((double)(short)(buf[j + 1] << 8 | buf[j])) / bandscopeScale;
|
|
if (++bandscopeState >= bandscopeBlockCount)
|
|
bandscopeState = 99;
|
|
}
|
|
else {
|
|
bandscopeState = 0; // Error
|
|
}
|
|
break;
|
|
case 99: // wait until the complete block is used
|
|
break;
|
|
}
|
|
continue;
|
|
}
|
|
//// ADC Rx samples
|
|
if (buf[3] != 0x06) // End point 6: I/Q and mic samples
|
|
return 0;
|
|
seq = buf[4] << 24 | buf[5] << 16 | buf[6] << 8 | buf[7]; // sequence number
|
|
quisk_rx_udp_started = 1;
|
|
tx_records += num_records * 2; // total samples for each receiver
|
|
quisk_hermes_tx_send(rx_udp_socket, &tx_records); // send Tx samples, decrement tx_records
|
|
if (seq != seq0) {
|
|
#if DEBUG_IO
|
|
printf("read_rx_udp10: Bad sequence want %d got %d\n", seq0, seq);
|
|
#endif
|
|
quisk_sound_state.read_error++;
|
|
}
|
|
seq0 = seq + 1; // Next expected sequence number
|
|
for (start = 11; start < 1000; start += 512) {
|
|
// check the sync bytes
|
|
if (buf[start - 3] != 0x7F || buf[start - 2] != 0x7F || buf[start - 1] != 0x7F) {
|
|
#if DEBUG_IO
|
|
printf("read_rx_udp10: Bad sync byte\n");
|
|
#endif
|
|
quisk_sound_state.read_error++;
|
|
}
|
|
// read five bytes of control information. start is the index of C0.
|
|
// Changes for HermesLite v2 thanks to Steve, KF7O
|
|
dindex = buf[start] >> 1;
|
|
if (dindex & 0x40) { // the ACK bit C0[7] is set
|
|
if (quisk_hermeslite_writepointer > 0) {
|
|
// Save response
|
|
quisk_hermeslite_response[0] = buf[start];
|
|
quisk_hermeslite_response[1] = buf[start+1];
|
|
quisk_hermeslite_response[2] = buf[start+2];
|
|
quisk_hermeslite_response[3] = buf[start+3];
|
|
quisk_hermeslite_response[4] = buf[start+4];
|
|
// Look for match
|
|
hlwp = 5*(quisk_hermeslite_writepointer-1);
|
|
if (dindex == 0x7f) {
|
|
printf("ERROR: Hermes-Lite did not process command\n");
|
|
} else if (dindex != (quisk_hermeslite_writequeue[hlwp])) {
|
|
printf("ERROR: Nonmatching Hermes-Lite response %d seen\n",dindex);
|
|
} else {
|
|
//printf("Response %d received\n",dindex);
|
|
quisk_hermeslite_writepointer--;
|
|
quisk_hermeslite_writeattempts = 0;
|
|
}
|
|
} else {
|
|
printf("ERROR: Unexpected Hermes-Lite response %d seen\n",dindex);
|
|
}
|
|
} else {
|
|
dindex = dindex >> 2;
|
|
}
|
|
// this does not save data for Hermes-Lite ACK
|
|
if (dindex >= 0 && dindex <= 4) { // Save the data returned by the hardware
|
|
quisk_hermes_to_pc[dindex * 4 ] = buf[start + 1]; // C1 to C4
|
|
quisk_hermes_to_pc[dindex * 4 + 1] = buf[start + 2];
|
|
quisk_hermes_to_pc[dindex * 4 + 2] = buf[start + 3];
|
|
quisk_hermes_to_pc[dindex * 4 + 3] = buf[start + 4];
|
|
}
|
|
if (dindex == 0) { // C0 is 0b00000xxx
|
|
//code_version = quisk_hermes_to_pc[3];
|
|
if ((quisk_hermes_to_pc[0] & 0x01) != 0) // C1
|
|
quisk_sound_state.overrange++;
|
|
hardware_ptt = buf[start] & 0x01; // C0 bit zero is PTT
|
|
hardware_cwkey = (buf[start] & 0x04) >> 2; // C0 bit two is CW key state
|
|
}
|
|
else if(dindex == 1) { // temperature and forward power
|
|
hermes_temperature += quisk_hermes_to_pc[4] << 8 | quisk_hermes_to_pc[5];
|
|
hermes_fwd_power += quisk_hermes_to_pc[6] << 8 | quisk_hermes_to_pc[7];
|
|
hermes_count_temperature++;
|
|
}
|
|
else if (dindex == 2) { // reverse power and current
|
|
hermes_rev_power += quisk_hermes_to_pc[8] << 8 | quisk_hermes_to_pc[9];
|
|
hermes_pa_current += quisk_hermes_to_pc[10] << 8 | quisk_hermes_to_pc[11];
|
|
hermes_count_current++;
|
|
}
|
|
// convert 24-bit samples to 32-bit samples; int must be 32 bits.
|
|
index = start + 5;
|
|
for (i = 0; i < num_records; i++) { // read records
|
|
xi = buf[index ] << 24 | buf[index + 1] << 16 | buf[index + 2] << 8;
|
|
xr = buf[index + 3] << 24 | buf[index + 4] << 16 | buf[index + 5] << 8;
|
|
samp[nSamples] = xr + xi * I; // first receiver
|
|
index += 6;
|
|
for (j = 0; j < quisk_multirx_count; j++) { // multirx receivers
|
|
xi = buf[index ] << 24 | buf[index + 1] << 16 | buf[index + 2] << 8;
|
|
xr = buf[index + 3] << 24 | buf[index + 4] << 16 | buf[index + 5] << 8;
|
|
c = xr + xi * I;
|
|
multirx_cSamples[j][nSamples] = c;
|
|
if (multirx_fft_data[j].index < multirx_fft_width)
|
|
multirx_fft_data[j].samples[multirx_fft_data[j].index++] = c;
|
|
index += 6;
|
|
}
|
|
nSamples++;
|
|
index += 2;
|
|
}
|
|
}
|
|
}
|
|
if ((quisk_pc_to_hermes[3] >> 3 & 0x7) != quisk_multirx_count && // change in number of receivers
|
|
( ! quisk_multirx_count || multirx_fft_next_state == 2)) { // wait until the current FFT is finished
|
|
quisk_multirx_state = 0; // Do not change receiver count without stopping Hermes and restarting
|
|
}
|
|
if (multirx_fft_next_state == 2) { // previous FFT is done
|
|
if (++multirx_fft_next_index >= quisk_multirx_count)
|
|
multirx_fft_next_index = 0;
|
|
multirx_fft_next_state = 0;
|
|
}
|
|
if (quisk_multirx_count && multirx_fft_next_state == 0 && multirx_fft_data[multirx_fft_next_index].index >= multirx_fft_width) { // FFT is read to run
|
|
memcpy(multirx_fft_next_samples, multirx_fft_data[multirx_fft_next_index].samples, multirx_fft_width * sizeof(fftw_complex));
|
|
multirx_fft_data[multirx_fft_next_index].index = 0;
|
|
multirx_fft_next_time = 1.0 / graph_refresh / quisk_multirx_count;
|
|
multirx_fft_next_state = 1; // this FFT is ready to run
|
|
}
|
|
return nSamples;
|
|
}
|
|
|
|
static int read_rx_udp17(complex double * cSamples0) // Read samples from UDP
|
|
{ // Size of complex sample array is SAMP_BUFFER_SIZE
|
|
ssize_t bytes;
|
|
unsigned char buf[1500]; // Maximum Ethernet is 1500 bytes.
|
|
static unsigned char seq0; // must be 8 bits
|
|
int i, n, nSamples0, xr, xi, index, want_samples, key_down;
|
|
complex double sample;
|
|
unsigned char * ptxr, * ptxi;
|
|
struct timeval tm_wait;
|
|
fft_data * ptFFT;
|
|
fd_set fds;
|
|
static int block_number=0;
|
|
// Data from the receiver is little-endian
|
|
|
|
if ( ! rx_udp_gain_correct) { // correct for second stage CIC decimation JIM JIM
|
|
int dec;
|
|
dec = (int)(rx_udp_clock / 30.0 / fft_sample_rate + 0.5);
|
|
if ((dec / 3) * 3 == dec) // Decimation by a factor of 3
|
|
rx_udp_gain_correct = 1.053497942;
|
|
else // Decimation by factors of two
|
|
rx_udp_gain_correct = 1.0;
|
|
//printf ("Gain %d %.8lf\n", dec, rx_udp_gain_correct);
|
|
}
|
|
if ( ! quisk_rx_udp_started) { // we never received any data
|
|
// send our return address until we receive UDP blocks
|
|
tm_wait.tv_sec = 0;
|
|
tm_wait.tv_usec = 5000;
|
|
FD_ZERO (&fds);
|
|
FD_SET (rx_udp_socket, &fds);
|
|
if (select (rx_udp_socket + 1, &fds, NULL, NULL, &tm_wait) == 1) { // see if data is available
|
|
bytes = recv(rx_udp_socket, (char *)buf, 1500, 0); // throw away the first block
|
|
seq0 = buf[0] + 1; // Next expected sequence number
|
|
quisk_rx_udp_started = 1;
|
|
#if DEBUG_IO || DEBUG
|
|
printf("Udp data started\n");
|
|
#endif
|
|
}
|
|
else { // send our return address to the sample source
|
|
buf[0] = buf[1] = 0x72; // UDP command "register return address"
|
|
send(rx_udp_socket, (char *)buf, 2, 0);
|
|
return 0;
|
|
}
|
|
}
|
|
nSamples0 = 0;
|
|
want_samples = (int)(quisk_sound_state.data_poll_usec * 1e-6 * quisk_sound_state.sample_rate + 0.5);
|
|
key_down = quisk_is_key_down();
|
|
while (nSamples0 < want_samples) { // read several UDP blocks
|
|
tm_wait.tv_sec = 0;
|
|
tm_wait.tv_usec = 100000; // Linux seems to have problems with very small time intervals
|
|
FD_ZERO (&fds);
|
|
FD_SET (rx_udp_socket, &fds);
|
|
i = select (rx_udp_socket + 1, &fds, NULL, NULL, &tm_wait);
|
|
if (i == 1)
|
|
;
|
|
else if (i == 0) {
|
|
#if DEBUG_IO || DEBUG
|
|
printf("Udp socket timeout\n");
|
|
#endif
|
|
return 0;
|
|
}
|
|
else {
|
|
#if DEBUG_IO || DEBUG
|
|
printf("Udp select error %d\n", i);
|
|
#endif
|
|
return 0;
|
|
}
|
|
bytes = recv(rx_udp_socket, (char *)buf, 1500, 0); // blocking read
|
|
if (bytes != RX_UDP_SIZE) { // Known size of sample block
|
|
quisk_sound_state.read_error++;
|
|
#if DEBUG_IO || DEBUG
|
|
printf("read_rx_udp: Bad block size\n");
|
|
#endif
|
|
continue;
|
|
}
|
|
// buf[0] is the sequence number
|
|
// buf[1] is the status:
|
|
// bit 0: key up/down state
|
|
// bit 1: set for ADC overrange (clip)
|
|
if (buf[0] != seq0) {
|
|
#if DEBUG_IO || DEBUG
|
|
printf("read_rx_udp: Bad sequence want %3d got %3d\n",
|
|
(unsigned int)seq0, (unsigned int)buf[0]);
|
|
#endif
|
|
quisk_sound_state.read_error++;
|
|
}
|
|
seq0 = buf[0] + 1; // Next expected sequence number
|
|
//quisk_set_key_down(buf[1] & 0x01); // bit zero is key state
|
|
if (buf[1] & 0x02) // bit one is ADC overrange
|
|
quisk_sound_state.overrange++;
|
|
index = 2;
|
|
ptxr = (unsigned char *)&xr;
|
|
ptxi = (unsigned char *)ξ
|
|
// convert 24-bit samples to 32-bit samples; int must be 32 bits.
|
|
while (index < bytes) {
|
|
if (is_little_endian) {
|
|
xr = xi = 0;
|
|
memcpy (ptxr + 1, buf + index, 3);
|
|
index += 3;
|
|
memcpy (ptxi + 1, buf + index, 3);
|
|
index += 3;
|
|
sample = (xr + xi * I) * rx_udp_gain_correct;
|
|
}
|
|
else { // big-endian
|
|
*(ptxr ) = buf[index + 2];
|
|
*(ptxr + 1) = buf[index + 1];
|
|
*(ptxr + 2) = buf[index ];
|
|
*(ptxr + 3) = 0;
|
|
index += 3;
|
|
*(ptxi ) = buf[index + 2];
|
|
*(ptxi + 1) = buf[index + 1];
|
|
*(ptxi + 2) = buf[index ];
|
|
*(ptxi + 3) = 0;
|
|
index += 3;
|
|
sample = (xr + xi * I) * rx_udp_gain_correct;
|
|
}
|
|
if (xr & 0x100) { // channel 1
|
|
if (quisk_invert_spectrum) // Invert spectrum
|
|
sample = conj(sample);
|
|
// Put samples into the fft input array.
|
|
ptFFT = fft_data_array + fft_data_index;
|
|
if ( ! (xi & 0x100)) { // zero marker for start of first block
|
|
if (ptFFT->index != 0) {
|
|
//printf("Resync block\n");
|
|
fft_error++;
|
|
ptFFT->index = 0;
|
|
}
|
|
ptFFT->block = block_number = 0;
|
|
}
|
|
else if (ptFFT->index == 0) {
|
|
if (scan_blocks) {
|
|
if (++block_number < scan_blocks)
|
|
ptFFT->block = block_number;
|
|
else
|
|
ptFFT->block = block_number = 0;
|
|
}
|
|
else {
|
|
ptFFT->block = block_number = 0;
|
|
}
|
|
if (scan_blocks && block_number >= scan_blocks)
|
|
printf("Bad block_number %d\n", block_number);
|
|
}
|
|
ptFFT->samples[ptFFT->index] = sample;
|
|
if ((isFDX || ! key_down) && ++(ptFFT->index) >= fft_size) { // check sample count
|
|
n = fft_data_index + 1; // next FFT data location
|
|
if (n >= FFT_ARRAY_SIZE)
|
|
n = 0;
|
|
if (fft_data_array[n].filled == 0) { // Is the next buffer empty?
|
|
fft_data_array[n].index = 0;
|
|
fft_data_array[n].block = 0;
|
|
fft_data_array[fft_data_index].filled = 1; // Mark the previous buffer ready.
|
|
fft_data_index = n; // Write samples into the new buffer.
|
|
ptFFT = fft_data_array + fft_data_index;
|
|
}
|
|
else { // no place to write samples
|
|
ptFFT->index = 0;
|
|
ptFFT->block = 0;
|
|
fft_error++;
|
|
}
|
|
}
|
|
}
|
|
else { // channel 0
|
|
cSamples0[nSamples0++] = sample;
|
|
}
|
|
}
|
|
}
|
|
return nSamples0;
|
|
}
|
|
|
|
static PyObject * open_rx_udp(PyObject * self, PyObject * args)
|
|
{
|
|
const char * ip;
|
|
int port;
|
|
char buf[128];
|
|
struct sockaddr_in Addr;
|
|
int recvsize;
|
|
|
|
#if DEBUG_IO
|
|
int intbuf;
|
|
#ifdef MS_WINDOWS
|
|
int bufsize = sizeof(int);
|
|
#else
|
|
socklen_t bufsize = sizeof(int);
|
|
#endif
|
|
#endif
|
|
|
|
#ifdef MS_WINDOWS
|
|
WORD wVersionRequested;
|
|
WSADATA wsaData;
|
|
#endif
|
|
|
|
if (!PyArg_ParseTuple (args, "si", &ip, &port))
|
|
return NULL;
|
|
#ifdef MS_WINDOWS
|
|
wVersionRequested = MAKEWORD(2, 2);
|
|
if (WSAStartup(wVersionRequested, &wsaData) != 0) {
|
|
sprintf(buf, "Failed to initialize Winsock (WSAStartup)");
|
|
return PyString_FromString(buf);
|
|
}
|
|
else {
|
|
cleanupWSA = 1;
|
|
}
|
|
#endif
|
|
#if DEBUG_IO
|
|
printf("open_rx_udp to IP %s port 0x%X\n", ip, port);
|
|
#endif
|
|
quisk_using_udp = 1;
|
|
rx_udp_socket = socket(PF_INET, SOCK_DGRAM, 0);
|
|
if (rx_udp_socket != INVALID_SOCKET) {
|
|
recvsize = 256000;
|
|
setsockopt(rx_udp_socket, SOL_SOCKET, SO_RCVBUF, (char *)&recvsize, sizeof(recvsize));
|
|
memset(&Addr, 0, sizeof(Addr));
|
|
Addr.sin_family = AF_INET;
|
|
Addr.sin_port = htons(port);
|
|
#ifdef MS_WINDOWS
|
|
Addr.sin_addr.S_un.S_addr = inet_addr(ip);
|
|
#else
|
|
inet_aton(ip, &Addr.sin_addr);
|
|
#endif
|
|
if (connect(rx_udp_socket, (const struct sockaddr *)&Addr, sizeof(Addr)) != 0) {
|
|
shutdown(rx_udp_socket, QUISK_SHUT_BOTH);
|
|
close(rx_udp_socket);
|
|
rx_udp_socket = INVALID_SOCKET;
|
|
sprintf(buf, "Failed to connect to UDP %s port 0x%X", ip, port);
|
|
}
|
|
else {
|
|
sprintf(buf, "Capture from UDP %s port 0x%X", ip, port);
|
|
if (quisk_use_rx_udp == 17)
|
|
quisk_sample_source(NULL, close_udp, read_rx_udp17);
|
|
else if (quisk_use_rx_udp == 10) {
|
|
quisk_sample_source(NULL, close_udp10, read_rx_udp10);
|
|
init_bandscope();
|
|
}
|
|
else
|
|
quisk_sample_source(NULL, close_udp, quisk_read_rx_udp);
|
|
#if DEBUG_IO
|
|
if (getsockopt(rx_udp_socket, SOL_SOCKET, SO_RCVBUF, (char *)&intbuf, &bufsize) == 0)
|
|
printf("UDP socket receive buffer size %d\n", intbuf);
|
|
else
|
|
printf ("Failure SO_RCVBUF\n");
|
|
#endif
|
|
}
|
|
}
|
|
else {
|
|
sprintf(buf, "Failed to open socket");
|
|
}
|
|
return PyString_FromString(buf);
|
|
}
|
|
|
|
static PyObject * open_sound(PyObject * self, PyObject * args)
|
|
{
|
|
int rate;
|
|
char * capt, * play, * mname, * mip, * mpname;
|
|
const char * utf8 = "utf-8";
|
|
|
|
if (!PyArg_ParseTuple (args, "esesiiiessiiiidesi", utf8, &capt, utf8, &play,
|
|
&rate,
|
|
&quisk_sound_state.data_poll_usec,
|
|
&quisk_sound_state.latency_millisecs,
|
|
utf8, &mname, &mip,
|
|
&quisk_sound_state.tx_audio_port,
|
|
&quisk_sound_state.mic_sample_rate,
|
|
&quisk_sound_state.mic_channel_I,
|
|
&quisk_sound_state.mic_channel_Q,
|
|
&quisk_sound_state.mic_out_volume,
|
|
utf8, &mpname,
|
|
&quisk_sound_state.mic_playback_rate
|
|
))
|
|
return NULL;
|
|
|
|
#if SAMPLES_FROM_FILE == 1
|
|
QuiskWavWriteOpen(&hWav, "band.wav", 3, 2, 4, 48000, 1E3 / CLIP32);
|
|
#elif SAMPLES_FROM_FILE == 2
|
|
QuiskWavReadOpen(&hWav, "band.wav", 3, 2, 4, 48000, CLIP32 / 1E6);
|
|
#endif
|
|
if (quisk_sound_state.mic_out_volume > 0.7) // maximum value must leave headroom for
|
|
quisk_sound_state.mic_out_volume = 0.7; // the amplitude and phase adjustments
|
|
quisk_sound_state.playback_rate = QuiskGetConfigInt("playback_rate", 48000);
|
|
quisk_mic_preemphasis = QuiskGetConfigDouble("mic_preemphasis", 0.6);
|
|
//if (quisk_mic_preemphasis < 0.0 || quisk_mic_preemphasis > 1.0)
|
|
// quisk_mic_preemphasis = 1.0;
|
|
quisk_mic_clip = QuiskGetConfigDouble("mic_clip", 3.0);
|
|
agc_release_time = QuiskGetConfigDouble("agc_release_time", 1.0);
|
|
strncpy(quisk_sound_state.dev_capt_name, capt, QUISK_SC_SIZE);
|
|
strncpy(quisk_sound_state.dev_play_name, play, QUISK_SC_SIZE);
|
|
strncpy(quisk_sound_state.mic_dev_name, mname, QUISK_SC_SIZE);
|
|
strncpy(quisk_sound_state.name_of_mic_play, mpname, QUISK_SC_SIZE);
|
|
strncpy(quisk_sound_state.mic_ip, mip, IP_SIZE);
|
|
strncpy(quisk_sound_state.IQ_server, QuiskGetConfigString("IQ_Server_IP", ""), IP_SIZE);
|
|
quisk_sound_state.verbose_pulse = QuiskGetConfigInt("pulse_audio_verbose_output", 0);
|
|
fft_error = 0;
|
|
PyMem_Free(capt);
|
|
PyMem_Free(play);
|
|
PyMem_Free(mname);
|
|
PyMem_Free(mpname);
|
|
quisk_open_sound();
|
|
quisk_open_mic();
|
|
return get_state(NULL, NULL);
|
|
}
|
|
|
|
static PyObject * close_sound(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
quisk_close_mic();
|
|
quisk_close_sound();
|
|
quisk_close_key();
|
|
#if SAMPLES_FROM_FILE
|
|
QuiskWavClose(&hWav);
|
|
#endif
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * change_scan(PyObject * self, PyObject * args) // Called from GUI thread
|
|
{ // Change to a new FFT rate
|
|
|
|
if (!PyArg_ParseTuple (args, "iidii", &scan_blocks, &scan_sample_rate, &scan_valid, &scan_vfo0, &scan_deltaf))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * change_rates(PyObject * self, PyObject * args) // Called from GUI thread
|
|
{ // Change to new sample rates
|
|
|
|
multiple_sample_rates = 1;
|
|
if (!PyArg_ParseTuple (args, "iiii", &quisk_sound_state.sample_rate, &vfo_audio, &fft_sample_rate, &vfo_screen))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * change_rate(PyObject * self, PyObject * args) // Called from GUI thread
|
|
{ // Change to a new sample rate
|
|
int rate, avg;
|
|
|
|
if (!PyArg_ParseTuple (args, "ii", &rate, &avg))
|
|
return NULL;
|
|
if (multiple_sample_rates) {
|
|
fft_sample_rate = rate;
|
|
}
|
|
else {
|
|
quisk_sound_state.sample_rate = rate;
|
|
fft_sample_rate = rate;
|
|
}
|
|
rx_udp_gain_correct = 0; // re-calculate JIM
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * read_sound(PyObject * self, PyObject * args)
|
|
{
|
|
int n;
|
|
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
Py_BEGIN_ALLOW_THREADS
|
|
n = quisk_read_sound();
|
|
Py_END_ALLOW_THREADS
|
|
return PyInt_FromLong(n);
|
|
}
|
|
|
|
static PyObject * start_sound(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
quisk_start_sound();
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * mixer_set(PyObject * self, PyObject * args)
|
|
{
|
|
char * card_name;
|
|
int numid;
|
|
PyObject * value;
|
|
char err_msg[QUISK_SC_SIZE];
|
|
|
|
if (!PyArg_ParseTuple (args, "siO", &card_name, &numid, &value))
|
|
return NULL;
|
|
|
|
quisk_mixer_set(card_name, numid, value, err_msg, QUISK_SC_SIZE);
|
|
return PyString_FromString(err_msg);
|
|
}
|
|
|
|
static PyObject * pc_to_hermes(PyObject * self, PyObject * args)
|
|
{
|
|
PyObject * byteArray;
|
|
|
|
if (!PyArg_ParseTuple (args, "O", &byteArray))
|
|
return NULL;
|
|
if ( ! PyByteArray_Check(byteArray)) {
|
|
PyErr_SetString (QuiskError, "Object is not a bytearray.");
|
|
return NULL;
|
|
}
|
|
if (PyByteArray_Size(byteArray) != 17 * 4) {
|
|
PyErr_SetString (QuiskError, "Bytearray size must be 17 * 4.");
|
|
return NULL;
|
|
}
|
|
memmove(quisk_pc_to_hermes, PyByteArray_AsString(byteArray), 17 * 4);
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
// Changes for HermesLite v2 thanks to Steve, KF7O
|
|
static PyObject * pc_to_hermeslite_writequeue(PyObject * self, PyObject * args)
|
|
{
|
|
PyObject * byteArray;
|
|
|
|
if (!PyArg_ParseTuple (args, "O", &byteArray))
|
|
return NULL;
|
|
if ( ! PyByteArray_Check(byteArray)) {
|
|
PyErr_SetString (QuiskError, "Object is not a bytearray.");
|
|
return NULL;
|
|
}
|
|
if (PyByteArray_Size(byteArray) != 4 * 5) {
|
|
PyErr_SetString (QuiskError, "Bytearray size must be 4 * 5.");
|
|
return NULL;
|
|
}
|
|
memmove(quisk_hermeslite_writequeue, PyByteArray_AsString(byteArray), 4 * 5);
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_hermeslite_writepointer(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "I", &quisk_hermeslite_writepointer))
|
|
return NULL;
|
|
if (quisk_hermeslite_writepointer > 4 || quisk_hermeslite_writepointer < 0) {
|
|
PyErr_SetString (QuiskError, "Hermeslite writepointer must be >=0 and <=4.");
|
|
return NULL;
|
|
}
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * get_hermeslite_writepointer(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
return Py_BuildValue("I",quisk_hermeslite_writepointer);
|
|
}
|
|
|
|
static PyObject * get_hermeslite_response(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
return PyByteArray_FromStringAndSize((char *)quisk_hermeslite_response, 5);
|
|
}
|
|
|
|
static PyObject * clear_hermeslite_response(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
memset(quisk_hermeslite_response, 0, 5*sizeof(char));
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * hermes_to_pc(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
return PyByteArray_FromStringAndSize((char *)quisk_hermes_to_pc, 5 * 4);
|
|
}
|
|
|
|
static PyObject * set_hermes_id(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "ii", &quisk_hermes_code_version, &quisk_hermes_board_id))
|
|
return NULL;
|
|
switch(quisk_hermes_board_id) {
|
|
default:
|
|
case 3: // Angelia and Odyssey-2
|
|
bandscopeBlockCount = 32;
|
|
break;
|
|
case 6: // Hermes Lite
|
|
bandscopeBlockCount = 4;
|
|
break;
|
|
}
|
|
bandscope_size = bandscopeBlockCount * 512;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
#ifdef MS_WINDOWS
|
|
static const char * Win_NtoA(unsigned long addr)
|
|
{
|
|
static char buf32[32];
|
|
|
|
if (addr > 0)
|
|
snprintf(buf32, 32, "%li.%li.%li.%li", (addr>>24)&0xFF, (addr>>16)&0xFF, (addr>>8)&0xFF, (addr>>0)&0xFF);
|
|
else
|
|
buf32[0] = 0;
|
|
return buf32;
|
|
}
|
|
|
|
#else
|
|
static const char * Lin_NtoA(struct sockaddr * a)
|
|
{
|
|
static char buf32[32];
|
|
unsigned long addr;
|
|
|
|
if (a && (addr = ntohl(((struct sockaddr_in *)a)->sin_addr.s_addr)) > 0)
|
|
snprintf(buf32, 32, "%li.%li.%li.%li", (addr>>24)&0xFF, (addr>>16)&0xFF, (addr>>8)&0xFF, (addr>>0)&0xFF);
|
|
else
|
|
buf32[0] = 0;
|
|
return buf32;
|
|
}
|
|
#endif
|
|
|
|
static PyObject * ip_interfaces(PyObject * self, PyObject * args)
|
|
{
|
|
#ifdef MS_WINDOWS
|
|
int i;
|
|
MIB_IPADDRTABLE * ipTable = NULL;
|
|
IP_ADAPTER_INFO * pAdapterInfo;
|
|
PyObject * pylist, * tup;
|
|
MIB_IPADDRROW row;
|
|
ULONG bufLen;
|
|
DWORD ipRet, apRet;
|
|
const char * name;
|
|
unsigned long ipAddr, netmask, baddr;
|
|
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
pylist = PyList_New(0);
|
|
bufLen = 0;
|
|
for (i=0; i<5; i++) {
|
|
ipRet = GetIpAddrTable(ipTable, &bufLen, 0);
|
|
if (ipRet == ERROR_INSUFFICIENT_BUFFER) {
|
|
free(ipTable); // in case we had previously allocated it
|
|
ipTable = (MIB_IPADDRTABLE *) malloc(bufLen);
|
|
}
|
|
else if (ipRet == NO_ERROR)
|
|
break;
|
|
else {
|
|
free(ipTable);
|
|
ipTable = NULL;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (ipTable) {
|
|
pAdapterInfo = NULL;
|
|
bufLen = 0;
|
|
for (i=0; i<5; i++) {
|
|
apRet = GetAdaptersInfo(pAdapterInfo, &bufLen);
|
|
if (apRet == ERROR_BUFFER_OVERFLOW) {
|
|
free(pAdapterInfo); // in case we had previously allocated it
|
|
pAdapterInfo = (IP_ADAPTER_INFO *) malloc(bufLen);
|
|
}
|
|
else if (apRet == ERROR_SUCCESS)
|
|
break;
|
|
else {
|
|
free(pAdapterInfo);
|
|
pAdapterInfo = NULL;
|
|
break;
|
|
}
|
|
}
|
|
|
|
for (i=0; i<ipTable->dwNumEntries; i++) {
|
|
row = ipTable->table[i];
|
|
// Now lookup the appropriate adaptor-name in the pAdaptorInfos, if we can find it
|
|
name = NULL;
|
|
if (pAdapterInfo) {
|
|
IP_ADAPTER_INFO * next = pAdapterInfo;
|
|
while((next)&&(name==NULL)) {
|
|
IP_ADDR_STRING * ipAddr = &next->IpAddressList;
|
|
while(ipAddr) {
|
|
if (inet_addr(ipAddr->IpAddress.String) == row.dwAddr) {
|
|
name = next->AdapterName;
|
|
break;
|
|
}
|
|
ipAddr = ipAddr->Next;
|
|
}
|
|
next = next->Next;
|
|
}
|
|
}
|
|
ipAddr = ntohl(row.dwAddr);
|
|
netmask = ntohl(row.dwMask);
|
|
baddr = ipAddr & netmask;
|
|
if (row.dwBCastAddr)
|
|
baddr |= ~netmask;
|
|
tup = PyTuple_New(4);
|
|
if (name == NULL)
|
|
PyTuple_SetItem(tup, 0, PyString_FromString("unnamed"));
|
|
else
|
|
PyTuple_SetItem(tup, 0, PyString_FromString(name));
|
|
PyTuple_SetItem(tup, 1, PyString_FromString(Win_NtoA(ipAddr)));
|
|
PyTuple_SetItem(tup, 2, PyString_FromString(Win_NtoA(netmask)));
|
|
PyTuple_SetItem(tup, 3, PyString_FromString(Win_NtoA(baddr)));
|
|
PyList_Append(pylist, tup);
|
|
Py_DECREF(tup);
|
|
}
|
|
free(pAdapterInfo);
|
|
free(ipTable);
|
|
}
|
|
#else
|
|
PyObject * pylist, * tup;
|
|
struct ifaddrs * ifap, * p;
|
|
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
pylist = PyList_New(0);
|
|
if (getifaddrs(&ifap) == 0) {
|
|
p = ifap;
|
|
while(p) {
|
|
if ((p->ifa_addr) && p->ifa_addr->sa_family == AF_INET) {
|
|
tup = PyTuple_New(4);
|
|
PyTuple_SetItem(tup, 0, PyString_FromString(p->ifa_name));
|
|
PyTuple_SetItem(tup, 1, PyString_FromString(Lin_NtoA(p->ifa_addr)));
|
|
PyTuple_SetItem(tup, 2, PyString_FromString(Lin_NtoA(p->ifa_netmask)));
|
|
PyTuple_SetItem(tup, 3, PyString_FromString(Lin_NtoA(p->ifa_broadaddr)));
|
|
PyList_Append(pylist, tup);
|
|
Py_DECREF(tup);
|
|
}
|
|
p = p->ifa_next;
|
|
}
|
|
freeifaddrs(ifap);
|
|
}
|
|
#endif
|
|
return pylist;
|
|
}
|
|
|
|
static PyObject * invert_spectrum(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &quisk_invert_spectrum))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_agc(PyObject * self, PyObject * args)
|
|
{ /* Change the AGC level */
|
|
if (!PyArg_ParseTuple (args, "d", &agcReleaseGain))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_filters(PyObject * self, PyObject * args)
|
|
{ // Enter the coefficients of the I and Q digital filters. The storage for
|
|
// filters is not malloc'd because filters may be changed while being used.
|
|
// Multiple filters are available at nFilter.
|
|
PyObject * filterI, * filterQ;
|
|
int i, size, nFilter, bw, start_offset;
|
|
PyObject * obj;
|
|
char buf98[98];
|
|
|
|
if (!PyArg_ParseTuple (args, "OOiii", &filterI, &filterQ, &bw, &start_offset, &nFilter))
|
|
return NULL;
|
|
if (PySequence_Check(filterI) != 1) {
|
|
PyErr_SetString (QuiskError, "Filter I is not a sequence");
|
|
return NULL;
|
|
}
|
|
if (PySequence_Check(filterQ) != 1) {
|
|
PyErr_SetString (QuiskError, "Filter Q is not a sequence");
|
|
return NULL;
|
|
}
|
|
size = PySequence_Size(filterI);
|
|
if (size != PySequence_Size(filterQ)) {
|
|
PyErr_SetString (QuiskError, "The size of filters I and Q must be equal");
|
|
return NULL;
|
|
}
|
|
if (size >= MAX_FILTER_SIZE) {
|
|
snprintf(buf98, 98, "Filter size must be less than %d", MAX_FILTER_SIZE);
|
|
PyErr_SetString (QuiskError, buf98);
|
|
return NULL;
|
|
}
|
|
filter_bandwidth[nFilter] = bw;
|
|
if (nFilter == 0)
|
|
filter_start_offset = start_offset;
|
|
for (i = 0; i < size; i++) {
|
|
obj = PySequence_GetItem(filterI, i);
|
|
cFilterI[nFilter][i] = PyFloat_AsDouble(obj);
|
|
Py_XDECREF(obj);
|
|
obj = PySequence_GetItem(filterQ, i);
|
|
cFilterQ[nFilter][i] = PyFloat_AsDouble(obj);
|
|
Py_XDECREF(obj);
|
|
}
|
|
sizeFilter = size;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_auto_notch(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &quisk_auto_notch))
|
|
return NULL;
|
|
dAutoNotch(NULL, 0, 0, 0);
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_noise_blanker(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &quisk_noise_blanker))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_enable_bandscope(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &enable_bandscope))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_rx_mode(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &rxMode))
|
|
return NULL;
|
|
quisk_set_tx_mode();
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_spot_level(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &quiskSpotLevel))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_imd_level(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &quiskImdLevel))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_mic_out_volume(PyObject * self, PyObject * args)
|
|
{
|
|
int level;
|
|
|
|
if (!PyArg_ParseTuple (args, "i", &level))
|
|
return NULL;
|
|
quisk_sound_state.mic_out_volume = level / 100.0;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * ImmediateChange(PyObject * self, PyObject * args) // called from the GUI thread
|
|
{
|
|
char * name;
|
|
int keyupDelay;
|
|
|
|
if (!PyArg_ParseTuple (args, "s", &name))
|
|
return NULL;
|
|
if ( ! strcmp(name, "keyupDelay")) {
|
|
keyupDelay = quiskKeyupDelay = QuiskGetConfigInt("keyupDelay", 23);
|
|
if (quisk_use_rx_udp == 10)
|
|
keyupDelay += 30;
|
|
keyupDelayCode = (int)(quisk_sound_state.playback_rate *1e-3 * keyupDelay + 0.5);
|
|
}
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_split_rxtx(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &split_rxtx))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_tune(PyObject * self, PyObject * args)
|
|
{ /* Change the tuning frequency */
|
|
if (!PyArg_ParseTuple (args, "ii", &rx_tune_freq, &quisk_tx_tune_freq))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_sidetone(PyObject * self, PyObject * args)
|
|
{
|
|
int keyupDelay; // Add a silent period after key up to remove reception of CW by the receiver.
|
|
|
|
if (!PyArg_ParseTuple (args, "idii", &quisk_sidetoneCtrl, &sidetoneVolume, &rit_freq, &keyupDelay))
|
|
return NULL;
|
|
quiskKeyupDelay = keyupDelay;
|
|
sidetonePhase = cexp((I * 2.0 * M_PI * abs(rit_freq)) / quisk_sound_state.playback_rate);
|
|
if (quisk_use_rx_udp == 10)
|
|
keyupDelay += 30; // Extra delay needed because the CW waveform is transmitted after key up.
|
|
keyupDelayCode = (int)(quisk_sound_state.playback_rate *1e-3 * keyupDelay + 0.5);
|
|
if (rxMode == CWL || rxMode == CWU)
|
|
dAutoNotch(NULL, 0, 0, 0); // for CW, changing the RIT affects autonotch
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_squelch(PyObject * self, PyObject * args) // Set level for FM squelch
|
|
{
|
|
if (!PyArg_ParseTuple (args, "d", &squelch_level))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_ssb_squelch(PyObject * self, PyObject * args) // Set level for SSB squelch
|
|
{
|
|
if (!PyArg_ParseTuple (args, "ii", &ssb_squelch_enabled, &ssb_squelch_level))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_kill_audio(PyObject * self, PyObject * args)
|
|
{ /* replace radio sound with silence */
|
|
if (!PyArg_ParseTuple (args, "i", &kill_audio))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * tx_hold_state(PyObject * self, PyObject * args)
|
|
{ // Query or set the transmit hold state
|
|
int i;
|
|
|
|
if (!PyArg_ParseTuple (args, "i", &i))
|
|
return NULL;
|
|
if (i >= 0) // arg < 0 is a Query for the current value
|
|
quiskTxHoldState = i;
|
|
return PyInt_FromLong(quiskTxHoldState);
|
|
}
|
|
|
|
static PyObject * set_transmit_mode(PyObject * self, PyObject * args)
|
|
{ /* Set the radio to transmit mode */
|
|
if (!PyArg_ParseTuple (args, "i", &quisk_transmit_mode))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_volume(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "d", &quisk_audioVolume))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_ctcss(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "d", &quisk_ctcss_freq))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_key_down(PyObject * self, PyObject * args)
|
|
{
|
|
int down;
|
|
|
|
if (!PyArg_ParseTuple (args, "i", &down))
|
|
return NULL;
|
|
quisk_set_key_down(down);
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_PTT(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &is_PTT_down))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_multirx_mode(PyObject * self, PyObject * args)
|
|
{
|
|
int index, mode;
|
|
|
|
if (!PyArg_ParseTuple (args, "ii", &index, &mode))
|
|
return NULL;
|
|
if (index < QUISK_MAX_RECEIVERS)
|
|
multirx_mode[index] = mode;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_multirx_freq(PyObject * self, PyObject * args)
|
|
{
|
|
int index, freq;
|
|
|
|
if (!PyArg_ParseTuple (args, "ii", &index, &freq))
|
|
return NULL;
|
|
if (index < QUISK_MAX_RECEIVERS)
|
|
multirx_freq[index] = freq;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_multirx_play_method(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &multirx_play_method))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_multirx_play_channel(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &multirx_play_channel))
|
|
return NULL;
|
|
if (multirx_play_channel >= QUISK_MAX_RECEIVERS)
|
|
multirx_play_channel = -1;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * get_multirx_graph(PyObject * self, PyObject * args) // Called by the GUI thread
|
|
{
|
|
int i, j, k;
|
|
double d1, d2, scale;
|
|
static double * fft_window=NULL; // Window for FFT data
|
|
PyObject * retrn, * data;
|
|
static double time0=0; // time of last graph
|
|
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
if ( ! fft_window) {
|
|
// Create the fft window
|
|
fft_window = (double *) malloc(sizeof(double) * multirx_fft_width);
|
|
for (i = 0, j = -multirx_fft_width / 2; i < multirx_fft_width; i++, j++)
|
|
fft_window[i] = 0.5 + 0.5 * cos(2. * M_PI * j / multirx_fft_width); // Hanning
|
|
}
|
|
retrn = PyTuple_New(2);
|
|
if (multirx_fft_next_state == 1 && QuiskTimeSec() - time0 >= multirx_fft_next_time) {
|
|
time0 = QuiskTimeSec();
|
|
// The FFT is ready to run. Calculate FFT.
|
|
for (i = 0; i < multirx_fft_width; i++) // multiply by window
|
|
multirx_fft_next_samples[i] *= fft_window[i];
|
|
fftw_execute(multirx_fft_next_plan);
|
|
// Average the fft data into the graph in order of frequency
|
|
data = PyTuple_New(multirx_data_width);
|
|
scale = log10(multirx_fft_width) + 31.0 * log10(2.0);
|
|
scale *= 20.0;
|
|
j = MULTIRX_FFT_MULT;
|
|
k = 0;
|
|
d1 = 0;
|
|
for (i = multirx_fft_width / 2; i < multirx_fft_width; i++) { // Negative frequencies
|
|
d1 += cabs(multirx_fft_next_samples[i]);
|
|
if (--j == 0) {
|
|
d2 = 20.0 * log10(d1) - scale;
|
|
if (d2 < -200)
|
|
d2 = -200;
|
|
PyTuple_SetItem(data, k++, PyFloat_FromDouble(d2));
|
|
d1 = 0;
|
|
j = MULTIRX_FFT_MULT;
|
|
}
|
|
}
|
|
for (i = 0; i < multirx_fft_width / 2; i++) { // Positive frequencies
|
|
d1 += cabs(multirx_fft_next_samples[i]);
|
|
if (--j == 0) {
|
|
d2 = 20.0 * log10(d1) - scale;
|
|
if (d2 < -200)
|
|
d2 = -200;
|
|
PyTuple_SetItem(data, k++, PyFloat_FromDouble(d2));
|
|
d1 = 0;
|
|
j = MULTIRX_FFT_MULT;
|
|
}
|
|
}
|
|
PyTuple_SetItem(retrn, 0, data);
|
|
PyTuple_SetItem(retrn, 1, PyInt_FromLong(multirx_fft_next_index));
|
|
multirx_fft_next_state = 2; // This FFT is done.
|
|
}
|
|
else {
|
|
data = PyTuple_New(0);
|
|
PyTuple_SetItem(retrn, 0, data);
|
|
PyTuple_SetItem(retrn, 1, PyInt_FromLong(-1));
|
|
}
|
|
return retrn;
|
|
}
|
|
|
|
static PyObject * get_bandscope(PyObject * self, PyObject * args) // Called by the GUI thread
|
|
{
|
|
int i, j, j1, j2, L, clock;
|
|
double zoom, deltaf, rate, f1;
|
|
static int fft_count = 0;
|
|
static double the_max = 0;
|
|
static double time0=0; // time of last graph
|
|
double d1, d2, sample, frac, scale;
|
|
PyObject * tuple2;
|
|
|
|
if (!PyArg_ParseTuple (args, "idd", &clock, &zoom, &deltaf))
|
|
return NULL;
|
|
|
|
if (bandscopeState == 99 && bandscopePlan) { // bandscope samples are ready
|
|
for (i = 0; i < bandscope_size; i++) {
|
|
d1 = fabs(bandscopeSamples[i]);
|
|
if (d1 > the_max)
|
|
the_max = d1;
|
|
bandscopeSamples[i] *= bandscopeWindow[i]; // multiply by window
|
|
}
|
|
fftw_execute(bandscopePlan); // Calculate forward FFT
|
|
// The return FFT has length bandscope_size / 2 + 1
|
|
L = bandscope_size / 2 + 1;
|
|
for (i = 0; i < L; i++)
|
|
bandscopeAverage[i] += cabs(bandscopeFFT[i]);
|
|
bandscopeState = 0;
|
|
fft_count++;
|
|
if (QuiskTimeSec() - time0 >= 1.0 / graph_refresh) { // return FFT data
|
|
bandscopeAverage[L] = 0.0; // in case we run off the end
|
|
// Average the return FFT into the data width
|
|
tuple2 = PyTuple_New(graph_width);
|
|
frac = (double)L / graph_width;
|
|
scale = 1.0 / frac / fft_count / bandscope_size;
|
|
rate = clock / 2.0;
|
|
for (i = 0; i < graph_width; i++) { // for each pixel
|
|
f1 = deltaf + rate / 2.0 * (1.0 - zoom); // frequency at left of graph
|
|
// freq = f1 + pixel / graph_width + zoom * rate = rate * fft_index / L
|
|
d1 = L / rate * (f1 + (double)i / graph_width * zoom * rate);
|
|
d2 = L / rate * (f1 + (double)(i + 1) / graph_width * zoom * rate);
|
|
j1 = floor(d1);
|
|
j2 = floor(d2);
|
|
if (j1 == j2) {
|
|
sample = (d2 - d1) * bandscopeAverage[j1];
|
|
}
|
|
else {
|
|
sample = (j1 + 1 - d1) * bandscopeAverage[j1];
|
|
for (j = j1 + 1; j < j2; j++)
|
|
sample += bandscopeAverage[j];
|
|
sample += (d2 - j2) * bandscopeAverage[j2];
|
|
}
|
|
sample = sample * scale;
|
|
if (sample <= 1E-10)
|
|
sample = -200.0;
|
|
else
|
|
sample = 20.0 * log10(sample);
|
|
PyTuple_SetItem(tuple2, i, PyFloat_FromDouble(sample));
|
|
}
|
|
fft_count = 0;
|
|
time0 = QuiskTimeSec();
|
|
hermes_adc_level = the_max;
|
|
the_max = 0;
|
|
for (i = 0; i < L; i++)
|
|
bandscopeAverage[i] = 0;
|
|
return tuple2;
|
|
}
|
|
}
|
|
Py_INCREF(Py_None); // No data yet
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * get_graph(PyObject * self, PyObject * args) // Called by the GUI thread
|
|
{
|
|
int i, j, k, m, n, index, ffts, ii, mm, m0, deltam;
|
|
fft_data * ptFft;
|
|
PyObject * tuple2;
|
|
double d1, d2, scale, zoom, deltaf;
|
|
complex double c;
|
|
static double meter = 0; // RMS s-meter
|
|
static int use_fft = 1; // Use the FFT, or return raw data
|
|
static double * fft_avg=NULL; // Array to average the FFT
|
|
static double * fft_tmp;
|
|
static int count_fft=0; // how many fft's have occurred (for average)
|
|
static double time0=0; // time of last graph
|
|
|
|
if (!PyArg_ParseTuple (args, "idd", &k, &zoom, &deltaf))
|
|
return NULL;
|
|
if (k != use_fft) { // change in data return type; re-initialize
|
|
use_fft = k;
|
|
count_fft = 0;
|
|
}
|
|
if ( ! fft_avg) {
|
|
fft_avg = (double *) malloc(sizeof(double) * fft_size);
|
|
fft_tmp = (double *) malloc(sizeof(double) * fft_size);
|
|
for (i = 0; i < fft_size; i++)
|
|
fft_avg[i] = 0;
|
|
}
|
|
// Process all FFTs that are ready to run.
|
|
index = fft_data_index; // oldest data first - FIFO
|
|
for (ffts = 0; ffts < FFT_ARRAY_SIZE; ffts++) {
|
|
if (++index >= FFT_ARRAY_SIZE)
|
|
index = 0;
|
|
if (fft_data_array[index].filled)
|
|
ptFft = fft_data_array + index;
|
|
else
|
|
continue;
|
|
if (scan_blocks && ptFft->block >= scan_blocks) {
|
|
//printf("Reject block %d\n", ptFft->block);
|
|
ptFft->filled = 0;
|
|
continue;
|
|
}
|
|
if ( ! use_fft) { // return raw data, not FFT
|
|
tuple2 = PyTuple_New(data_width);
|
|
for (i = 0; i < data_width; i++)
|
|
PyTuple_SetItem(tuple2, i,
|
|
PyComplex_FromDoubles(creal(ptFft->samples[i]), cimag(ptFft->samples[i])));
|
|
ptFft->filled = 0;
|
|
return tuple2;
|
|
}
|
|
// Continue with FFT calculation
|
|
for (i = 0; i < fft_size; i++) // multiply by window
|
|
ptFft->samples[i] *= fft_window[i];
|
|
fftw_execute_dft(quisk_fft_plan, ptFft->samples, ptFft->samples); // Calculate FFT
|
|
// Create RMS s-meter value at known bandwidth
|
|
// The pass band is (rx_tune_freq + filter_start_offset) to += bandwidth
|
|
// d1 is the tune frequency
|
|
// d2 is the number of FFT bins required for the bandwidth
|
|
// i is the starting bin number from - sample_rate / 2 to + sample_rate / 2
|
|
d2 = (double)filter_bandwidth[0] * fft_size / fft_sample_rate;
|
|
if (scan_blocks) { // Use tx, not rx?? ERROR:
|
|
d1 = ((double)quisk_tx_tune_freq + vfo_screen - scan_vfo0 - scan_deltaf * ptFft->block) * fft_size / scan_sample_rate;
|
|
i = (int)(d1 - d2 / 2 + 0.5);
|
|
}
|
|
else
|
|
i = (int)((double)(rx_tune_freq + filter_start_offset) * fft_size / fft_sample_rate + 0.5);
|
|
n = (int)(floor(d2) + 0.01); // number of whole bins to add
|
|
if (i > - fft_size / 2 && i + n + 1 < fft_size / 2) { // too close to edge?
|
|
for (j = 0; j < n; i++, j++) {
|
|
if (i < 0)
|
|
c = ptFft->samples[fft_size + i]; // negative frequencies
|
|
else
|
|
c = ptFft->samples[i]; // positive frequencies
|
|
meter = meter + c * conj(c); // add square of amplitude
|
|
}
|
|
if (i < 0) // add fractional next bin
|
|
c = ptFft->samples[fft_size + i];
|
|
else
|
|
c = ptFft->samples[i];
|
|
meter = meter + c * conj(c) * (d2 - n); // fractional part of next bin
|
|
}
|
|
// Average the fft data into the graph in order of frequency
|
|
if (scan_blocks) {
|
|
if (ptFft->block == (scan_blocks - 1))
|
|
count_fft++;
|
|
k = 0;
|
|
for (i = fft_size / 2; i < fft_size; i++) // Negative frequencies
|
|
fft_tmp[k++] = cabs(ptFft->samples[i]);
|
|
for (i = 0; i < fft_size / 2; i++) // Positive frequencies
|
|
fft_tmp[k++] = cabs(ptFft->samples[i]);
|
|
// Average this block into its correct position
|
|
m0 = (int)(fft_size * ((1.0 - scan_valid) / 2.0));
|
|
deltam = (int)(fft_size * scan_valid / scan_blocks);
|
|
m = mm = m0 + ptFft->block * deltam; // target position
|
|
i = ii = (int)(fft_size * ((1.0 - scan_valid) / 2.0)); // start of valid data
|
|
for (j = 0; j < deltam; j++) {
|
|
d2 = 0;
|
|
for (n = 0; n < scan_blocks; n++)
|
|
d2 += fft_tmp[i++];
|
|
fft_avg[m++] = d2;
|
|
}
|
|
//printf(" %d %.4lf At %5d to %5d place %5d to %5d for block %d\n", fft_size, scan_valid, mm, m, ii, i, ptFft->block);
|
|
}
|
|
else {
|
|
count_fft++;
|
|
k = 0;
|
|
for (i = fft_size / 2; i < fft_size; i++) // Negative frequencies
|
|
fft_avg[k++] += cabs(ptFft->samples[i]);
|
|
for (i = 0; i < fft_size / 2; i++) // Positive frequencies
|
|
fft_avg[k++] += cabs(ptFft->samples[i]);
|
|
}
|
|
ptFft->filled = 0;
|
|
if (count_fft > 0 && QuiskTimeSec() - time0 >= 1.0 / graph_refresh) {
|
|
// We have averaged enough fft's to return the graph data.
|
|
// Average the fft data of size fft_size into the size of data_width.
|
|
n = (int)(zoom * (double)fft_size / data_width + 0.5);
|
|
if (n < 1)
|
|
n = 1;
|
|
for (i = 0; i < data_width; i++) { // For each graph pixel
|
|
// find k, the starting index into the FFT data
|
|
k = (int)(fft_size * (
|
|
deltaf / fft_sample_rate + zoom * ((double)i / data_width - 0.5) + 0.5) + 0.1);
|
|
d2 = 0.0;
|
|
for (j = 0; j < n; j++, k++)
|
|
if (k >= 0 && k < fft_size)
|
|
d2 += fft_avg[k];
|
|
fft_avg[i] = d2;
|
|
}
|
|
scale = 1.0 / 2147483647.0 / fft_size;
|
|
Smeter = meter * scale * scale / count_fft; // record the new s-meter value
|
|
meter = 0;
|
|
if (Smeter > 0)
|
|
Smeter = 10.0 * log10(Smeter);
|
|
else
|
|
Smeter = -160.0;
|
|
// This correction is for a -40 dB strong signal, and is caused by FFT leakage
|
|
// into adjacent bins. It is the amplitude that is spread out, not the squared amplitude.
|
|
Smeter += 4.25969;
|
|
tuple2 = PyTuple_New(data_width);
|
|
// scale = 1.0 / count_fft / fft_size; // Divide by sample count
|
|
// scale /= pow(2.0, 31); // Normalize to max == 1
|
|
scale = log10(count_fft) + log10(fft_size) + 31.0 * log10(2.0);
|
|
scale *= 20.0;
|
|
for (i = 0; i < data_width; i++) {
|
|
d2 = 20.0 * log10(fft_avg[i]) - scale;
|
|
if (d2 < -200)
|
|
d2 = -200;
|
|
current_graph[i] = d2;
|
|
PyTuple_SetItem(tuple2, i, PyFloat_FromDouble(d2));
|
|
}
|
|
for (i = 0; i < fft_size; i++)
|
|
fft_avg[i] = 0;
|
|
count_fft = 0;
|
|
time0 = QuiskTimeSec();
|
|
return tuple2;
|
|
}
|
|
}
|
|
Py_INCREF(Py_None); // No data yet
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * get_filter(PyObject * self, PyObject * args)
|
|
{
|
|
int i, j, k, n;
|
|
int freq, time;
|
|
PyObject * tuple2;
|
|
complex double cx;
|
|
double d2, scale, accI, accQ;
|
|
double * average, * bufI, * bufQ;
|
|
double phase, delta;
|
|
static fftw_complex * samples;
|
|
static fftw_plan plan;
|
|
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
|
|
// Create space for the fft of size data_width
|
|
samples = (fftw_complex *) fftw_malloc(sizeof(fftw_complex) * data_width);
|
|
plan = fftw_plan_dft_1d(data_width, samples, samples, FFTW_FORWARD, FFTW_MEASURE);
|
|
average = (double *) malloc(sizeof(double) * (data_width + sizeFilter));
|
|
bufI = (double *) malloc(sizeof(double) * sizeFilter);
|
|
bufQ = (double *) malloc(sizeof(double) * sizeFilter);
|
|
|
|
for (i = 0; i < data_width + sizeFilter; i++)
|
|
average[i] = 0.5; // Value for freq == 0
|
|
for (freq = 1; freq < data_width / 2.0 - 10.0; freq++) {
|
|
delta = 2 * M_PI / data_width * freq;
|
|
phase = 0;
|
|
// generate some initial samples to fill the filter pipeline
|
|
for (time = 0; time < data_width + sizeFilter; time++) {
|
|
average[time] += cos(phase); // current sample
|
|
phase += delta;
|
|
if (phase > 2 * M_PI)
|
|
phase -= 2 * M_PI;
|
|
}
|
|
}
|
|
// now filter the signal
|
|
n = 0;
|
|
for (time = 0; time < data_width + sizeFilter; time++) {
|
|
d2 = average[time];
|
|
bufI[n] = d2;
|
|
bufQ[n] = d2;
|
|
accI = accQ = 0;
|
|
j = n;
|
|
for (k = 0; k < sizeFilter; k++) {
|
|
accI += bufI[j] * cFilterI[0][k];
|
|
accQ += bufQ[j] * cFilterQ[0][k];
|
|
if (++j >= sizeFilter)
|
|
j = 0;
|
|
}
|
|
cx = accI + I * accQ; // Filter output
|
|
if (++n >= sizeFilter)
|
|
n = 0;
|
|
if (time >= sizeFilter)
|
|
samples[time - sizeFilter] = cx;
|
|
}
|
|
|
|
for (i = 0; i < data_width; i++) // multiply by window
|
|
samples[i] *= fft_window[i];
|
|
fftw_execute(plan); // Calculate FFT
|
|
// Normalize and convert to log10
|
|
scale = 1. / data_width;
|
|
for (k = 0; k < data_width; k++) {
|
|
cx = samples[k];
|
|
average[k] = cabs(cx) * scale;
|
|
if (average[k] <= 1e-7) // limit to -140 dB
|
|
average[k] = -7;
|
|
else
|
|
average[k] = log10(average[k]);
|
|
}
|
|
// Return the graph data
|
|
tuple2 = PyTuple_New(data_width);
|
|
i = 0;
|
|
// Negative frequencies:
|
|
for (k = data_width / 2; k < data_width; k++, i++)
|
|
PyTuple_SetItem(tuple2, i, PyFloat_FromDouble(20.0 * average[k]));
|
|
|
|
// Positive frequencies:
|
|
for (k = 0; k < data_width / 2; k++, i++)
|
|
PyTuple_SetItem(tuple2, i, PyFloat_FromDouble(20.0 * average[k]));
|
|
|
|
free(bufQ);
|
|
free(bufI);
|
|
free(average);
|
|
fftw_destroy_plan(plan);
|
|
fftw_free(samples);
|
|
|
|
return tuple2;
|
|
}
|
|
|
|
static void measure_freq(complex double * cSamples, int nSamples, int srate)
|
|
{
|
|
int i, k, center, ipeak;
|
|
double dmax, c3, freq;
|
|
complex double cBuffer[SAMP_BUFFER_SIZE];
|
|
static int index = 0; // current index of samples
|
|
static int fft_size=12000; // size of fft data
|
|
static int fft_count=0; // number of ffts for the average
|
|
static fftw_complex * samples; // complex data for fft
|
|
static fftw_plan planA; // fft plan for fft
|
|
static double * fft_window; // window function
|
|
static double * fft_average; // average amplitudes
|
|
static struct quisk_cHB45Filter HalfBand1 = {NULL, 0, 0};
|
|
static struct quisk_cHB45Filter HalfBand2 = {NULL, 0, 0};
|
|
static struct quisk_cHB45Filter HalfBand3 = {NULL, 0, 0};
|
|
|
|
if ( ! cSamples) { // malloc new space and initialize
|
|
samples = (fftw_complex *) fftw_malloc(sizeof(fftw_complex) * fft_size);
|
|
planA = fftw_plan_dft_1d(fft_size, samples, samples, FFTW_FORWARD, FFTW_MEASURE);
|
|
fft_window = (double *) malloc(sizeof(double) * (fft_size + 1));
|
|
fft_average = (double *) malloc(sizeof(double) * fft_size);
|
|
memset(fft_average, 0, sizeof(double) * fft_size);
|
|
for (i = 0; i < fft_size; i++) // Hanning
|
|
fft_window[i] = 0.50 - 0.50 * cos(2. * M_PI * i / (fft_size - 1));
|
|
return;
|
|
}
|
|
memcpy(cBuffer, cSamples, nSamples * sizeof(complex double)); // do not destroy cSamples
|
|
nSamples = quisk_cDecim2HB45(cBuffer, nSamples, &HalfBand1);
|
|
nSamples = quisk_cDecim2HB45(cBuffer, nSamples, &HalfBand2);
|
|
nSamples = quisk_cDecim2HB45(cBuffer, nSamples, &HalfBand3);
|
|
srate /= 8; // sample rate as decimated
|
|
for (i = 0; i < nSamples && index < fft_size; i++, index++)
|
|
samples[index] = cBuffer[i];
|
|
if (index < fft_size)
|
|
return; // wait for a full array of samples
|
|
for (i = 0; i < fft_size; i++) // multiply by window
|
|
samples[i] *= fft_window[i];
|
|
fftw_execute(planA); // Calculate FFT
|
|
index = 0;
|
|
fft_count++;
|
|
// Average the fft data into the graph in order of frequency
|
|
k = 0;
|
|
for (i = fft_size / 2; i < fft_size; i++) // Negative frequencies
|
|
fft_average[k++] += cabs(samples[i]);
|
|
for (i = 0; i < fft_size / 2; i++) // Positive frequencies
|
|
fft_average[k++] += cabs(samples[i]);
|
|
if (fft_count < measure_freq_mode / 2)
|
|
return; // continue with average
|
|
// time for a calculation
|
|
fft_count = 0;
|
|
dmax = 1.e-20;
|
|
ipeak = 0;
|
|
center = fft_size / 2 - rit_freq * fft_size / srate;
|
|
k = 500; // desired +/- half-bandwidth to search for a peak
|
|
k = k * fft_size / srate; // convert to index
|
|
for (i = center - k; i <= center + k; i++) { // search for a peak near the RX freq
|
|
if (fft_average[i] > dmax) {
|
|
dmax = fft_average[i];
|
|
ipeak = i;
|
|
}
|
|
}
|
|
c3 = 1.36 * (fft_average[ipeak+1] - fft_average[ipeak - 1]) / (fft_average[ipeak-1] + fft_average[ipeak] + fft_average[ipeak+1]);
|
|
freq = srate * (2 * (ipeak + c3) - fft_size) / 2 / fft_size;
|
|
freq += rx_tune_freq;
|
|
//printf("freq %.0f rx_tune_freq %d vfo_screen %d vfo_audio %d\n", freq, rx_tune_freq, vfo_screen, vfo_audio);
|
|
// printf("\n%5d %.4lf %.2lf k=%d\n", ipeak, c3, freq, k);
|
|
measured_frequency = freq;
|
|
//for (i = ipeak - 10; i <= ipeak + 10 && i >= 0 && i < fft_size; i++)
|
|
// printf("%4d %12.5f\n", i, fft_average[i] / dmax);
|
|
memset(fft_average, 0, sizeof(double) * fft_size);
|
|
}
|
|
|
|
static PyObject * Xdft(PyObject * pyseq, int inverse, int window)
|
|
{ // Native spectral order is 0 Hz to (Fs - 1). Change this to
|
|
// - (Fs - 1)/2 to + Fs/2. For even Fs==32, there are 15 negative
|
|
// frequencies, a zero, and 16 positive frequencies. For odd Fs==31,
|
|
// there are 15 negative and positive frequencies plus zero frequency.
|
|
// Note that zero frequency is always index (Fs - 1) / 2.
|
|
PyObject * obj;
|
|
int i, j, size;
|
|
static int fft_size = -1; // size of fft data
|
|
static fftw_complex * samples; // complex data for fft
|
|
static fftw_plan planF, planB; // fft plan for fftW
|
|
static double * fft_window; // window function
|
|
Py_complex pycx; // Python C complex value
|
|
|
|
if (PySequence_Check(pyseq) != 1) {
|
|
PyErr_SetString (QuiskError, "DFT input data is not a sequence");
|
|
return NULL;
|
|
}
|
|
size = PySequence_Size(pyseq);
|
|
if (size <= 0)
|
|
return PyTuple_New(0);
|
|
if (size != fft_size) { // Change in previous size; malloc new space
|
|
if (fft_size > 0) {
|
|
fftw_destroy_plan(planF);
|
|
fftw_destroy_plan(planB);
|
|
fftw_free(samples);
|
|
free (fft_window);
|
|
}
|
|
fft_size = size; // Create space for one fft
|
|
samples = (fftw_complex *) fftw_malloc(sizeof(fftw_complex) * fft_size);
|
|
planF = fftw_plan_dft_1d(fft_size, samples, samples, FFTW_FORWARD, FFTW_MEASURE);
|
|
planB = fftw_plan_dft_1d(fft_size, samples, samples, FFTW_BACKWARD, FFTW_MEASURE);
|
|
fft_window = (double *) malloc(sizeof(double) * (fft_size + 1));
|
|
for (i = 0; i <= size/2; i++) {
|
|
if (1) // Blackman window
|
|
fft_window[i] = fft_window[size - i] = 0.42 + 0.50 * cos(2. * M_PI * i / size) +
|
|
0.08 * cos(4. * M_PI * i / size);
|
|
else if (1) // Hamming
|
|
fft_window[i] = fft_window[size - i] = 0.54 + 0.46 * cos(2. * M_PI * i / size);
|
|
else // Hanning
|
|
fft_window[i] = fft_window[size - i] = 0.50 + 0.50 * cos(2. * M_PI * i / size);
|
|
}
|
|
}
|
|
j = (size - 1) / 2; // zero frequency in input
|
|
for (i = 0; i < size; i++) {
|
|
obj = PySequence_GetItem(pyseq, j);
|
|
if (PyComplex_Check(obj)) {
|
|
pycx = PyComplex_AsCComplex(obj);
|
|
}
|
|
else if (PyFloat_Check(obj)) {
|
|
pycx.real = PyFloat_AsDouble(obj);
|
|
pycx.imag = 0;
|
|
}
|
|
else if (PyInt_Check(obj)) {
|
|
pycx.real = PyInt_AsLong(obj);
|
|
pycx.imag = 0;
|
|
}
|
|
else {
|
|
Py_XDECREF(obj);
|
|
PyErr_SetString (QuiskError, "DFT input data is not a complex/float/int number");
|
|
return NULL;
|
|
}
|
|
samples[i] = pycx.real + I * pycx.imag;
|
|
if (++j >= size)
|
|
j = 0;
|
|
Py_XDECREF(obj);
|
|
}
|
|
if (inverse) { // Normalize using 1/N
|
|
fftw_execute(planB); // Calculate inverse FFT / N
|
|
if (window) {
|
|
for (i = 0; i < fft_size; i++) // multiply by window / N
|
|
samples[i] *= fft_window[i] / size;
|
|
}
|
|
else {
|
|
for (i = 0; i < fft_size; i++) // divide by N
|
|
samples[i] /= size;
|
|
}
|
|
}
|
|
else {
|
|
if (window) {
|
|
for (i = 0; i < fft_size; i++) // multiply by window
|
|
samples[i] *= fft_window[i];
|
|
}
|
|
fftw_execute(planF); // Calculate FFT
|
|
}
|
|
pyseq = PyList_New(fft_size);
|
|
j = (size - 1) / 2; // zero frequency in input
|
|
for (i = 0; i < fft_size; i++) {
|
|
pycx.real = creal(samples[i]);
|
|
pycx.imag = cimag(samples[i]);
|
|
PyList_SetItem(pyseq, j, PyComplex_FromCComplex(pycx));
|
|
if (++j >= size)
|
|
j = 0;
|
|
}
|
|
return pyseq;
|
|
}
|
|
|
|
static PyObject * dft(PyObject * self, PyObject * args)
|
|
{
|
|
PyObject * tuple2;
|
|
int window;
|
|
|
|
window = 0;
|
|
if (!PyArg_ParseTuple (args, "O|i", &tuple2, &window))
|
|
return NULL;
|
|
return Xdft(tuple2, 0, window);
|
|
}
|
|
|
|
static PyObject * is_key_down(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
return PyInt_FromLong(quisk_is_key_down());
|
|
}
|
|
|
|
static PyObject * idft(PyObject * self, PyObject * args)
|
|
{
|
|
PyObject * tuple2;
|
|
int window;
|
|
|
|
window = 0;
|
|
if (!PyArg_ParseTuple (args, "O|i", &tuple2, &window))
|
|
return NULL;
|
|
return Xdft(tuple2, 1, window);
|
|
}
|
|
|
|
static PyObject * record_app(PyObject * self, PyObject * args)
|
|
{ // Record the Python object for the application instance, malloc space for fft's.
|
|
int i, j, rate;
|
|
unsigned long handle;
|
|
fftw_complex * pt;
|
|
|
|
if (!PyArg_ParseTuple (args, "OOiiiiik", &pyApp, &quisk_pyConfig, &data_width, &graph_width,
|
|
&fft_size, &multirx_data_width, &rate, &handle))
|
|
return NULL;
|
|
|
|
Py_INCREF(quisk_pyConfig);
|
|
|
|
#ifdef MS_WINDOWS
|
|
#ifdef _WIN64
|
|
quisk_mainwin_handle = (HWND)(unsigned long long)handle;
|
|
#else
|
|
quisk_mainwin_handle = (HWND)handle;
|
|
#endif
|
|
#endif
|
|
rx_udp_clock = QuiskGetConfigDouble("rx_udp_clock", 122.88e6);
|
|
graph_refresh = QuiskGetConfigInt("graph_refresh", 7);
|
|
quisk_use_rx_udp = QuiskGetConfigInt("use_rx_udp", 0);
|
|
quisk_sound_state.sample_rate = rate;
|
|
fft_sample_rate = rate;
|
|
is_little_endian = 1; // Test machine byte order
|
|
if (*(char *)&is_little_endian == 1)
|
|
is_little_endian = 1;
|
|
else
|
|
is_little_endian = 0;
|
|
strncpy (quisk_sound_state.err_msg, CLOSED_TEXT, QUISK_SC_SIZE);
|
|
// Initialize space for the FFTs
|
|
for (i = 0; i < FFT_ARRAY_SIZE; i++) {
|
|
fft_data_array[i].filled = 0;
|
|
fft_data_array[i].index = 0;
|
|
fft_data_array[i].block = 0;
|
|
fft_data_array[i].samples = (fftw_complex *) fftw_malloc(sizeof(fftw_complex) * fft_size);
|
|
}
|
|
pt = fft_data_array[0].samples;
|
|
quisk_fft_plan = fftw_plan_dft_1d(fft_size, pt, pt, FFTW_FORWARD, FFTW_MEASURE);
|
|
// Create space for the fft average and window
|
|
if (fft_window)
|
|
free(fft_window);
|
|
fft_window = (double *) malloc(sizeof(double) * fft_size);
|
|
for (i = 0, j = -fft_size / 2; i < fft_size; i++, j++) {
|
|
if (0) // Hamming
|
|
fft_window[i] = 0.54 + 0.46 * cos(2. * M_PI * j / fft_size);
|
|
else // Hanning
|
|
fft_window[i] = 0.5 + 0.5 * cos(2. * M_PI * j / fft_size);
|
|
}
|
|
// Initialize plan for multirx FFT
|
|
multirx_fft_width = multirx_data_width * MULTIRX_FFT_MULT; // Use larger FFT than graph size
|
|
multirx_fft_next_samples = (fftw_complex *)malloc(multirx_fft_width * sizeof(fftw_complex));
|
|
multirx_fft_next_plan = fftw_plan_dft_1d(multirx_fft_width, multirx_fft_next_samples, multirx_fft_next_samples, FFTW_FORWARD, FFTW_MEASURE);
|
|
if (current_graph)
|
|
free(current_graph);
|
|
current_graph = (double *) malloc(sizeof(double) * data_width);
|
|
measure_freq(NULL, 0, 0);
|
|
dAutoNotch(NULL, 0, 0, 0);
|
|
quisk_process_decimate(NULL, 0, 0, 0);
|
|
quisk_process_demodulate(NULL, NULL, 0, 0, 0, 0);
|
|
#if DEBUG_IO
|
|
QuiskPrintTime(NULL, 0);
|
|
#endif
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * record_graph(PyObject * self, PyObject * args)
|
|
{ /* record the Python object for the application instance */
|
|
if (!PyArg_ParseTuple (args, "iid", &graphX, &graphY, &graphScale))
|
|
return NULL;
|
|
graphScale *= 2;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * test_1(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * test_2(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * test_3(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, ""))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_fdx(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &isFDX))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyObject * set_sample_bytes(PyObject * self, PyObject * args)
|
|
{
|
|
if (!PyArg_ParseTuple (args, "i", &sample_bytes))
|
|
return NULL;
|
|
Py_INCREF (Py_None);
|
|
return Py_None;
|
|
}
|
|
|
|
static PyMethodDef QuiskMethods[] = {
|
|
{"add_tone", add_tone, METH_VARARGS, "Add a test tone to the data."},
|
|
{"dft", dft, METH_VARARGS, "Calculate the discrete Fourier transform."},
|
|
{"idft", idft, METH_VARARGS, "Calculate the inverse discrete Fourier transform."},
|
|
{"is_key_down", is_key_down, METH_VARARGS, "Check whether the key is down; return 0 or 1."},
|
|
{"get_state", get_state, METH_VARARGS, "Return a count of read and write errors."},
|
|
{"get_graph", get_graph, METH_VARARGS, "Return a tuple of graph data."},
|
|
{"get_bandscope", get_bandscope, METH_VARARGS, "Return a tuple of bandscope data."},
|
|
{"set_multirx_mode", set_multirx_mode, METH_VARARGS, "Select demodulation mode for sub-receivers."},
|
|
{"set_multirx_freq", set_multirx_freq, METH_VARARGS, "Select how to play audio from sub-receivers."},
|
|
{"set_multirx_play_method", set_multirx_play_method, METH_VARARGS, "Select how to play audio from sub-receivers."},
|
|
{"set_multirx_play_channel", set_multirx_play_channel, METH_VARARGS, "Select which sub-receiver to play audio."},
|
|
{"get_multirx_graph", get_multirx_graph, METH_VARARGS, "Return a tuple of sub-receiver graph data."},
|
|
{"get_filter", get_filter, METH_VARARGS, "Return the frequency response of the receive filter."},
|
|
{"get_filter_rate", get_filter_rate, METH_VARARGS, "Return the sample rate used for the filters."},
|
|
{"get_tx_filter", quisk_get_tx_filter, METH_VARARGS, "Return the frequency response of the transmit filter."},
|
|
{"get_audio_graph", get_audio_graph, METH_VARARGS, "Return a tuple of the audio graph data."},
|
|
{"measure_frequency", measure_frequency, METH_VARARGS, "Set the method, return the measured frequency."},
|
|
{"measure_audio", measure_audio, METH_VARARGS, "Set the method, return the measured audio voltage."},
|
|
{"get_hardware_ptt", get_hardware_ptt, METH_VARARGS, "Return the state of the hardware PTT switch."},
|
|
{"get_overrange", get_overrange, METH_VARARGS, "Return the count of overrange (clip) for the ADC."},
|
|
{"get_smeter", get_smeter, METH_VARARGS, "Return the S meter reading."},
|
|
{"get_hermes_adc", get_hermes_adc, METH_VARARGS, "Return the ADC peak level."},
|
|
{"get_hermes_TFRC", get_hermes_TFRC, METH_VARARGS, "Return the temperature, forward and reverse power and PA current."},
|
|
{"set_hermes_id", set_hermes_id, METH_VARARGS, "Set the Hermes hardware code version and board ID."},
|
|
{"set_hermes_filters", quisk_set_hermes_filter, METH_VARARGS, "Set the Hermes filter to use for Rx and Tx."},
|
|
{"set_alex_hpf", quisk_set_alex_hpf, METH_VARARGS, "Set the Alex HP filter to use for Rx and Tx."},
|
|
{"set_alex_lpf", quisk_set_alex_lpf, METH_VARARGS, "Set the Alex LP filter to use for Rx and Tx."},
|
|
{"invert_spectrum", invert_spectrum, METH_VARARGS, "Invert the input RF spectrum"},
|
|
{"ip_interfaces", ip_interfaces, METH_VARARGS, "Return a list of interface data"},
|
|
{"pc_to_hermes", pc_to_hermes, METH_VARARGS, "Send this block of control data to the Hermes device"},
|
|
{"pc_to_hermeslite_writequeue", pc_to_hermeslite_writequeue, METH_VARARGS, "Fill Hermes-Lite write queue"},
|
|
{"set_hermeslite_writepointer", set_hermeslite_writepointer, METH_VARARGS, "Set Hermes-Lite write pointer"},
|
|
{"get_hermeslite_writepointer", get_hermeslite_writepointer, METH_VARARGS, "Return Hermes-Lite write pointer"},
|
|
{"clear_hermeslite_response", clear_hermeslite_response, METH_VARARGS, "Clear the Hermes-Lite response array"},
|
|
{"get_hermeslite_response", get_hermeslite_response, METH_VARARGS, "Get the Hermes-Lite response array"},
|
|
{"hermes_to_pc", hermes_to_pc, METH_VARARGS, "Get the block of control data from the Hermes device"},
|
|
{"record_app", record_app, METH_VARARGS, "Save the App instance."},
|
|
{"record_graph", record_graph, METH_VARARGS, "Record graph parameters."},
|
|
{"ImmediateChange", ImmediateChange, METH_VARARGS, "Call this to notify the program of changes."},
|
|
{"set_ampl_phase", quisk_set_ampl_phase, METH_VARARGS, "Set the sound card amplitude and phase corrections."},
|
|
{"set_udp_tx_correct", quisk_set_udp_tx_correct, METH_VARARGS, "Set the UDP transmit corrections."},
|
|
{"set_agc", set_agc, METH_VARARGS, "Set the AGC parameters."},
|
|
{"set_squelch", set_squelch, METH_VARARGS, "Set the FM squelch parameter."},
|
|
{"get_squelch", get_squelch, METH_VARARGS, "Get the FM squelch state, 0 or 1."},
|
|
{"set_ssb_squelch", set_ssb_squelch, METH_VARARGS, "Set the SSB squelch parameters."},
|
|
{"set_ctcss", set_ctcss, METH_VARARGS, "Set the frequency of the repeater access tone."},
|
|
{"set_file_name", (PyCFunction)quisk_set_file_name, METH_VARARGS|METH_KEYWORDS, "Set the names and state of the recording and playback files."},
|
|
{"set_params", (PyCFunction)set_params, METH_VARARGS|METH_KEYWORDS, "Set miscellaneous parameters in quisk.c."},
|
|
{"set_sparams", (PyCFunction)quisk_set_sparams, METH_VARARGS|METH_KEYWORDS, "Set miscellaneous parameters in sound.c."},
|
|
{"set_filters", set_filters, METH_VARARGS, "Set the receive audio I and Q channel filters."},
|
|
{"set_auto_notch", set_auto_notch, METH_VARARGS, "Set the auto notch on or off."},
|
|
{"set_kill_audio", set_kill_audio, METH_VARARGS, "Replace radio sound with silence."},
|
|
{"set_enable_bandscope", set_enable_bandscope, METH_VARARGS, "Enable or disable sending bandscope data."},
|
|
{"set_noise_blanker", set_noise_blanker, METH_VARARGS, "Set the noise blanker level."},
|
|
{"set_record_state", set_record_state, METH_VARARGS, "Set the temp buffer record and playback state."},
|
|
{"set_rx_mode", set_rx_mode, METH_VARARGS, "Set the receive mode: CWL, USB, AM, etc."},
|
|
{"set_mic_out_volume", set_mic_out_volume, METH_VARARGS, "Set the level of the mic output for SoftRock transmit"},
|
|
{"set_spot_level", set_spot_level, METH_VARARGS, "Set the spot level, or -1 for no spot"},
|
|
{"set_imd_level", set_imd_level, METH_VARARGS, "Set the imd level 0 to 1000."},
|
|
{"set_sidetone", set_sidetone, METH_VARARGS, "Set the sidetone volume and frequency."},
|
|
{"set_sample_bytes", set_sample_bytes, METH_VARARGS, "Set the number of bytes for each I or Q sample."},
|
|
{"set_transmit_mode", set_transmit_mode, METH_VARARGS, "Change the radio to transmit mode independent of key_down."},
|
|
{"set_volume", set_volume, METH_VARARGS, "Set the audio output volume."},
|
|
{"set_tx_audio", (PyCFunction)quisk_set_tx_audio, METH_VARARGS|METH_KEYWORDS, "Set the transmit audio parameters."},
|
|
{"is_vox", quisk_is_vox, METH_VARARGS, "return the VOX state zero or one."},
|
|
{"set_split_rxtx", set_split_rxtx, METH_VARARGS, "Set split for rx/tx."},
|
|
{"set_tune", set_tune, METH_VARARGS, "Set the tuning frequency."},
|
|
{"test_1", test_1, METH_VARARGS, "Test 1 function."},
|
|
{"test_2", test_2, METH_VARARGS, "Test 2 function."},
|
|
{"test_3", test_3, METH_VARARGS, "Test 3 function."},
|
|
{"tx_hold_state", tx_hold_state, METH_VARARGS, "Query or set the transmit hold state."},
|
|
{"set_fdx", set_fdx, METH_VARARGS, "Set full duplex mode; ignore the key status."},
|
|
{"sound_devices", quisk_sound_devices, METH_VARARGS, "Return a list of available sound device names."},
|
|
{"pa_sound_devices", quisk_pa_sound_devices, METH_VARARGS, "Return a list of available PulseAudio sound device names."},
|
|
{"sound_errors", quisk_sound_errors, METH_VARARGS, "Return a list of text strings with sound devices and error counts"},
|
|
{"open_sound", open_sound, METH_VARARGS, "Open the soundcard device."},
|
|
{"open_wav_file_play", open_wav_file_play, METH_VARARGS, "Open a WAV file to play instead of the microphone."},
|
|
{"close_sound", close_sound, METH_VARARGS, "Stop the soundcard and release resources."},
|
|
{"capt_channels", quisk_capt_channels, METH_VARARGS, "Set the I and Q capture channel numbers"},
|
|
{"play_channels", quisk_play_channels, METH_VARARGS, "Set the I and Q playback channel numbers"},
|
|
{"micplay_channels", quisk_micplay_channels, METH_VARARGS, "Set the I and Q microphone playback channel numbers"},
|
|
{"change_scan", change_scan, METH_VARARGS, "Change to a new FFT rate and multiplier"},
|
|
{"change_rate", change_rate, METH_VARARGS, "Change to a new sample rate"},
|
|
{"change_rates", change_rates, METH_VARARGS, "Change to multiple new sample rates"},
|
|
{"read_sound", read_sound, METH_VARARGS, "Read from the soundcard."},
|
|
{"start_sound", start_sound, METH_VARARGS, "Start the soundcard."},
|
|
{"mixer_set", mixer_set, METH_VARARGS, "Set microphone mixer parameters such as volume."},
|
|
{"open_key", open_key, METH_VARARGS, "Open access to the state of the key (CW or PTT)."},
|
|
{"open_rx_udp", open_rx_udp, METH_VARARGS, "Open a UDP port for capture."},
|
|
{"close_rx_udp", close_rx_udp, METH_VARARGS, "Close the UDP port used for capture."},
|
|
{"add_rx_samples", add_rx_samples, METH_VARARGS, "Record the Rx samples received by Python code."},
|
|
{"add_bscope_samples", add_bscope_samples, METH_VARARGS, "Record the bandscope samples received by Python code."},
|
|
{"set_key_down", set_key_down, METH_VARARGS, "Change the key up/down state."},
|
|
{"set_PTT", set_PTT, METH_VARARGS, "Change the PTT button state."},
|
|
{"freedv_open", quisk_freedv_open, METH_VARARGS, "Open FreeDV."},
|
|
{"freedv_close", quisk_freedv_close, METH_VARARGS, "Close FreeDV."},
|
|
{"freedv_get_snr", quisk_freedv_get_snr, METH_VARARGS, "Return the signal to noise ratio in dB."},
|
|
{"freedv_get_version", quisk_freedv_get_version, METH_VARARGS, "Return the codec2 API version."},
|
|
{"freedv_get_rx_char", quisk_freedv_get_rx_char, METH_VARARGS, "Get text characters received from freedv."},
|
|
{"freedv_set_options", (PyCFunction)quisk_freedv_set_options, METH_VARARGS|METH_KEYWORDS, "Set the freedv parameters."},
|
|
{NULL, NULL, 0, NULL} /* Sentinel */
|
|
};
|
|
|
|
#if PY_MAJOR_VERSION < 3
|
|
// Python 2.7:
|
|
PyMODINIT_FUNC init_quisk (void)
|
|
{
|
|
PyObject * m;
|
|
PyObject * c_api_object;
|
|
static void * Quisk_API[] = QUISK_API_INIT;
|
|
|
|
m = Py_InitModule ("_quisk", QuiskMethods);
|
|
if (m == NULL) {
|
|
printf("Py_InitModule of _quisk failed!\n");
|
|
return;
|
|
}
|
|
|
|
QuiskError = PyErr_NewException ("quisk.error", NULL, NULL);
|
|
Py_INCREF (QuiskError);
|
|
PyModule_AddObject (m, "error", QuiskError);
|
|
|
|
/* Create Capsules for handing _quisk symbols to C extensions in other Python modules. */
|
|
c_api_object = PyCapsule_New(Quisk_API, "_quisk.QUISK_C_API", NULL);
|
|
if (c_api_object != NULL)
|
|
PyModule_AddObject(m, "QUISK_C_API", c_api_object);
|
|
}
|
|
|
|
// Python 3:
|
|
#else
|
|
static struct PyModuleDef _quiskmodule = {
|
|
PyModuleDef_HEAD_INIT,
|
|
"_quisk",
|
|
NULL,
|
|
-1,
|
|
QuiskMethods
|
|
} ;
|
|
|
|
PyMODINIT_FUNC PyInit__quisk(void)
|
|
{
|
|
PyObject * m;
|
|
PyObject * c_api_object;
|
|
static void * Quisk_API[] = QUISK_API_INIT;
|
|
|
|
m = PyModule_Create(&_quiskmodule);
|
|
if (m == NULL)
|
|
return NULL;
|
|
QuiskError = PyErr_NewException("_quisk.error", NULL, NULL);
|
|
if (QuiskError == NULL) {
|
|
Py_DECREF(m);
|
|
return NULL;
|
|
}
|
|
Py_INCREF (QuiskError);
|
|
PyModule_AddObject (m, "error", QuiskError);
|
|
|
|
/* Create Capsules for handing _quisk symbols to C extensions in other Python modules. */
|
|
c_api_object = PyCapsule_New(Quisk_API, "_quisk.QUISK_C_API", NULL);
|
|
if (c_api_object != NULL)
|
|
PyModule_AddObject(m, "QUISK_C_API", c_api_object);
|
|
return m;
|
|
}
|
|
#endif
|