diff --git a/.vscode/.server-controller-port.log b/.vscode/.server-controller-port.log index bbc989c..eb02d3c 100644 --- a/.vscode/.server-controller-port.log +++ b/.vscode/.server-controller-port.log @@ -1,5 +1,5 @@ { "port": 13452, - "time": 1737845288156, + "time": 1737886041050, "version": "0.0.3" } \ No newline at end of file diff --git a/README.md b/README.md index d7e67e9..84476d0 100644 --- a/README.md +++ b/README.md @@ -1,6 +1,3 @@ -# FMTools -FMTools is a repository of apps you can use to make your FM broadcast better, pirate or not this will help you if you don't have something, maybe you want a better stereo encoder? SCA? We have what you need, for RDS just use MiniRDS - # fm95 FM95 is a audio processor for FM, it does: - Pre-Emphasis @@ -9,7 +6,7 @@ FM95 is a audio processor for FM, it does: - SSB Stereo - Polar Stereo - Polar SSB Stereo (huh) -- TODO: SCA +- SCA Supports 2 inputs: - Audio (via Pulse) @@ -18,13 +15,6 @@ Supports 2 inputs: and one output: - MPX (via Pulse or ALSA) -Note that i haven't tested it, but i will on monday (29-01-25) - -# SCAMod -SCAMod is a simple FM modulator which can be used to modulate a secondary audio stream, has similiar cpu usage and latency as STCode - -Has a fine quality, but as it goes for 12 khz fm signals - # How to compile? To compile you need `cmake`, `libasound2-dev` and `libpulse-dev`, if you have those then do these commands: ``` diff --git a/src/fm95.c b/src/fm95.c index 25c7f36..08296c2 100644 --- a/src/fm95.c +++ b/src/fm95.c @@ -1,18 +1,22 @@ #include #include -#include -#include -#include -#include #include -#include "options.h" +#define PREEMPHASIS + +#define buffer_maxlength 12288 +#define buffer_tlength_fragsize 8192 +#define buffer_prebuf 16 #define DEFAULT_STEREO 1 #define DEFAULT_STEREO_POLAR 0 #define DEFAULT_STEREO_SSB 0 #define DEFAULT_CLIPPER_THRESHOLD 1.0f #define DEFAULT_ALSA_OUTPUT 0 +#define DEFAULT_SCA_FREQUENCY 67000.0f +#define DEFAULT_SCA_DEVIATION 7000.0f +#define DEFAULT_SCA_CLIPPER_THRESHOLD 1.0f +#define DEFAULT_PREEMPHASIS_TAU 50e-6 //#define USB @@ -20,12 +24,14 @@ #include "../lib/oscillator.h" #include "../lib/filters.h" #include "../lib/hilbert.h" +#include "../lib/fm_modulator.h" #define SAMPLE_RATE 192000 // Don't go lower than 108 KHz, becuase it (53000*2) and (38000+15000) #define INPUT_DEVICE "real_real_tx_audio_input.monitor" #define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback" // #define MPX_DEVICE "" +// #define SCA_DEVICE "" #define BUFFER_SIZE 512 @@ -36,11 +42,9 @@ #define MONO_VOLUME 0.45f // L+R Signal #define PILOT_VOLUME 0.09f // 19 KHz Pilot #define STEREO_VOLUME 0.45f // L-R signal +#define SCA_VOLUME 0.1f #define MPX_VOLUME 1.0f -#ifdef PREEMPHASIS -#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america -#endif volatile sig_atomic_t to_run = 1; @@ -71,64 +75,115 @@ static void stop(int signum) { void show_version() { printf("fm95 (an FM Processor by radio95) version 1.0\n"); } - void show_help(char *name) { printf( "fm95 (an FM Processor by radio95)\n" "Usage: %s\n\n" - " -m,--mono Force Mono\n" - " -s,--stereo Force Stereo\n" - " -i,--input Override input device\n" - " -o,--output Override output device\n" - " -M,--mpx Override MPX input device\n" - " -c,--clipper Override the clipper threshold\n" - " -P,--polar Force Polar Stereo (does not take effect with -m)\n" - " -g,--ge Force Zenith/GE stereo (does not take effect with -m, default)\n" - " -S,--ssb Force SSB\n" - " -D,--dsb Force DSB\n" + " -m,--mono Force Mono [default: %d]\n" + " -s,--stereo Force Stereo [default: %d]\n" + " -i,--input Override input device [default: %s]\n" + " -o,--output Override output device [default: %s]\n" + " -a,--alsa_out Force alsa output [default: %d]\n" + " -p,--pulse_out Force pulse output [default: %d]\n" + " -M,--mpx Override MPX input device [default: %s]\n" + " -C,--sca Override the SCA input device [default: %s]\n" + " -f,--sca_freq Override the SCA frequency [default: %f]\n" + " -F,--sca_dev Override the SCA deviation [default: %f]\n" + " -L,--sca_clip Override the SCA clipper threshold [default: %f]\n" + " -c,--clipper Override the clipper threshold [default: %f]\n" + " -P,--polar Force Polar Stereo (does not take effect with -m%s)\n" + " -g,--ge Force Zenith/GE stereo (does not take effect with -m%s)\n" + " -S,--ssb Force SSB [default: %d]\n" + " -D,--dsb Force DSB [default: %d]\n" + " -R,--preemp Override preemphasis [default: %f]\n" ,name + ,DEFAULT_STEREO^1 + ,DEFAULT_STEREO + ,INPUT_DEVICE + ,OUTPUT_DEVICE + ,DEFAULT_ALSA_OUTPUT + ,DEFAULT_ALSA_OUTPUT^1 + #ifdef MPX_DEVICE + ,MPX_DEVICE + #else + ,"not set" + #endif + #ifdef SCA_DEVICE + ,SCA_DEVICE + #else + ,"not set" + #endif + ,DEFAULT_SCA_FREQUENCY + ,DEFAULT_SCA_DEVIATION + ,DEFAULT_SCA_CLIPPER_THRESHOLD + ,DEFAULT_CLIPPER_THRESHOLD + ,(DEFAULT_STEREO_POLAR == 1) ? ", default" : "" + ,(DEFAULT_STEREO_POLAR == 1) ? "" : ", default" + ,DEFAULT_STEREO_SSB + ,DEFAULT_STEREO_SSB^1 + ,DEFAULT_PREEMPHASIS_TAU ); } int main(int argc, char **argv) { show_version(); - int stereo = DEFAULT_STEREO; - #ifndef MPX_DEVICE - char audio_mpx_device[64] = "\0"; - #else - char audio_mpx_device[64] = MPX_DEVICE; - #endif + pa_simple *mpx_device; + pa_simple *sca_device; pa_simple *output_device; snd_pcm_hw_params_t *output_params; snd_pcm_t *output_handle; + float clipper_threshold = DEFAULT_CLIPPER_THRESHOLD; + int stereo = DEFAULT_STEREO; int polar_stereo = DEFAULT_STEREO_POLAR; int ssb = DEFAULT_STEREO_SSB; + + float sca_frequency = DEFAULT_SCA_FREQUENCY; + float sca_deviation = DEFAULT_SCA_DEVIATION; + float sca_clipper_threshold = DEFAULT_SCA_CLIPPER_THRESHOLD; + char audio_input_device[64] = INPUT_DEVICE; char audio_output_device[64] = OUTPUT_DEVICE; + #ifndef MPX_DEVICE + char audio_mpx_device[64] = "\0"; +#else + char audio_mpx_device[64] = MPX_DEVICE; +#endif +#ifndef SCA_DEVICE + char audio_sca_device[64] = "\0"; +#else + char audio_sca_device[64] = SCA_DEVICE; +#endif int alsa_output = DEFAULT_ALSA_OUTPUT; + float preemphasis_tau = DEFAULT_PREEMPHASIS_TAU; + // #region Parse Arguments int opt; - const char *short_opt = "msi:o:apM:c:PgSDhv"; + const char *short_opt = "msi:o:apM:C:f:F:L:c:PgSDR:hv"; struct option long_opt[] = { - {"mono", no_argument, NULL, 'm'}, - {"stereo", no_argument, NULL, 's'}, - {"input", optional_argument, NULL, 'i'}, - {"output", optional_argument, NULL, 'o'}, - {"alsa_out", no_argument, NULL, 'a'}, - {"pulse_put", no_argument, NULL, 'p'}, - {"mpx", optional_argument, NULL, 'M'}, - {"clipper", optional_argument, NULL, 'c'}, - {"polar", no_argument, NULL, 'P'}, - {"ge", no_argument, NULL, 'g'}, - {"ssb", no_argument, NULL, 'S'}, - {"dsb", no_argument, NULL, 'D'}, - - {"help", no_argument, NULL, 'h'}, - {"version", no_argument, NULL, 'v'}, - { 0, 0, 0, 0 } + {"mono", no_argument, NULL, 'm'}, + {"stereo", no_argument, NULL, 's'}, + {"input", required_argument, NULL, 'i'}, + {"output", required_argument, NULL, 'o'}, + {"alsa_out", no_argument, NULL, 'a'}, + {"pulse_out", no_argument, NULL, 'p'}, + {"mpx", required_argument, NULL, 'M'}, + {"sca", required_argument, NULL, 'C'}, + {"sca_freq", required_argument, NULL, 'f'}, + {"sca_dev", required_argument, NULL, 'F'}, + {"sca_clip", required_argument, NULL, 'L'}, + {"clipper", required_argument, NULL, 'c'}, + {"polar", no_argument, NULL, 'P'}, + {"ge", no_argument, NULL, 'g'}, + {"ssb", no_argument, NULL, 'S'}, + {"dsb", no_argument, NULL, 'D'}, + {"preemp", no_argument, NULL, 'R'}, + + {"help", no_argument, NULL, 'h'}, + {"version", no_argument, NULL, 'v'}, + {0, 0, 0, 0} // No trailing comma here }; while((opt = getopt_long(argc, argv, short_opt, long_opt, NULL)) != -1) { @@ -158,6 +213,21 @@ int main(int argc, char **argv) { case 'M': //MPX in memcpy(audio_mpx_device, optarg, 63); break; + case 'C': //SCA in + memcpy(audio_sca_device, optarg, 63); + break; + case 'f': //SCA freq + sca_frequency = strtof(optarg, NULL); + printf("Running with a SCA frequency of %f\n", sca_frequency); + break; + case 'F': //SCA deviation + sca_deviation = strtof(optarg, NULL); + printf("Running with a SCA deviation of %f\n", sca_deviation); + break; + case 'L': //SCA clip + sca_clipper_threshold = strtof(optarg, NULL); + printf("Running with a SCA clipper threshold of %f\n", sca_clipper_threshold); + break; case 'c': //Clipper clipper_threshold = strtof(optarg, NULL); printf("Running with a clipper threshold of %f\n", clipper_threshold); @@ -178,6 +248,10 @@ int main(int argc, char **argv) { ssb = 0; printf("Using DSB\n"); break; + case 'R': // Preemp + preemphasis_tau = strtof(optarg, NULL)*0.000001; + printf("Running with a premp of %f\n", preemphasis_tau); + break; case 'v': // Version show_version(); return 0; @@ -186,6 +260,9 @@ int main(int argc, char **argv) { return 1; } } + // #endregion + + // #region Setup devices // Define formats and buffer atributes pa_sample_spec stereo_format = { @@ -215,10 +292,10 @@ int main(int argc, char **argv) { pa_simple *input_device = pa_simple_new( NULL, - "StereoEncoder", + "fm95", PA_STREAM_RECORD, audio_input_device, - "Audio Input", + "Main Audio Input", &stereo_format, NULL, &input_buffer_atr, @@ -234,7 +311,7 @@ int main(int argc, char **argv) { mpx_device = pa_simple_new( NULL, - "StereoEncoder", + "fm95", PA_STREAM_RECORD, audio_mpx_device, "MPX Input", @@ -245,6 +322,28 @@ int main(int argc, char **argv) { ); if (!mpx_device) { fprintf(stderr, "Error: cannot open MPX device: %s\n", pa_strerror(opentime_pulse_error)); + pa_simple_free(input_device); + return 1; + } + } + if(strlen(audio_sca_device) != 0) { + printf("Connecting to SCA device... (%s)\n", audio_sca_device); + + sca_device = pa_simple_new( + NULL, + "fm95", + PA_STREAM_RECORD, + audio_sca_device, + "SCA Input", + &mono_format, + NULL, + &input_buffer_atr, + &opentime_pulse_error + ); + if (!sca_device) { + fprintf(stderr, "Error: cannot open SCA device: %s\n", pa_strerror(opentime_pulse_error)); + pa_simple_free(input_device); + if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device); return 1; } } @@ -266,6 +365,8 @@ int main(int argc, char **argv) { if (!output_device) { fprintf(stderr, "Error: cannot open output device: %s\n", pa_strerror(opentime_pulse_error)); pa_simple_free(input_device); + if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device); + if(strlen(audio_sca_device) != 0) pa_simple_free(sca_device); return 1; } } else { @@ -273,6 +374,8 @@ int main(int argc, char **argv) { if(output_error < 0) { fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error)); pa_simple_free(input_device); + if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device); + if(strlen(audio_sca_device) != 0) pa_simple_free(sca_device); return 1; } snd_pcm_hw_params_malloc(&output_params); @@ -294,29 +397,35 @@ int main(int argc, char **argv) { return 1; } } + // #endregion Oscillator pilot_osc; if(polar_stereo == 1) { - init_oscillator(&pilot_osc, 31250.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier + init_oscillator(&pilot_osc, 31250.0, SAMPLE_RATE); // The stereo carrier itself, the stereo carrier in polar is modulated directly on 31.25 khz with a bit of a carrier } else { init_oscillator(&pilot_osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier } + + FMModulator sca_mod; + init_fm_modulator(&sca_mod, sca_frequency, sca_deviation, SAMPLE_RATE); + HilbertTransformer hilbert; // An Hilbert shifts a signal in quadrature, generating the I/Q data init_hilbert(&hilbert); DelayLine monoDelay; // Hilbert introduces a delay of 99 samples, this should be here to sync the mono with stereo to a sample init_delay_line(&monoDelay, 99); -#ifdef PREEMPHASIS + ResistorCapacitor preemp_l, preemp_r; - init_rc_tau(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE); - init_rc_tau(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE); -#endif + init_rc_tau(&preemp_l, preemphasis_tau, SAMPLE_RATE); + init_rc_tau(&preemp_r, preemphasis_tau, SAMPLE_RATE); signal(SIGINT, stop); signal(SIGTERM, stop); int pulse_error; + float audio_stereo_input[BUFFER_SIZE*2]; // Input from device, interleaved stereo float mpx_in[BUFFER_SIZE]; // Input from MPX device + float sca_in[BUFFER_SIZE]; // Input from SCA device float left[BUFFER_SIZE+64], right[BUFFER_SIZE+64]; // Audio, same thing as in input but uninterleaved, ai told be there could be a buffer overflow here float output[BUFFER_SIZE]; // MPX, this goes to the output while (to_run) { @@ -333,25 +442,30 @@ int main(int argc, char **argv) { break; } } + if(strlen(audio_sca_device) != 0) { + if (pa_simple_read(sca_device, sca_in, sizeof(sca_in), &pulse_error) < 0) { + fprintf(stderr, "Error reading from SCA device: %s\n", pa_strerror(pulse_error)); + to_run = 0; + break; + } + } for (int i = 0; i < BUFFER_SIZE; i++) { float l_in = left[i]; float r_in = right[i]; - float multiplex_in = mpx_in[i]; + float current_mpx_in = mpx_in[i]; + float current_sca_in = sca_in[i]; -#ifdef PREEMPHASIS - float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in)*2; - float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in)*2; - float current_left_input = hard_clip(preemphasized_left, clipper_threshold); - float current_right_input = hard_clip(preemphasized_right, clipper_threshold); -#else - float current_left_input = hard_clip(l_in, clipper_threshold); - float current_right_input = hard_clip(r_in, clipper_threshold); -#endif + float ready_l = l_in; + float ready_r = r_in; + ready_l = apply_pre_emphasis(&preemp_l, ready_l)*2; + ready_r = apply_pre_emphasis(&preemp_r, ready_r)*2; + ready_l = hard_clip(ready_l, clipper_threshold); + ready_r = hard_clip(ready_r, clipper_threshold); - float mono = (current_left_input + current_right_input) / 2.0f; // Stereo to Mono + float mono = (ready_l + ready_r) / 2.0f; // Stereo to Mono if(stereo == 1) { - float stereo = (current_left_input - current_right_input) / 2.0f; // Also Stereo to Mono but a bit diffrent + float stereo = (ready_l - ready_r) / 2.0f; // Also Stereo to Mono but a bit diffrent if(polar_stereo == 1) { if(ssb) { float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc, 1); // Get stereo carrier via multiplication @@ -367,12 +481,14 @@ int main(int argc, char **argv) { #endif output[i] = delay_line(&monoDelay, mono)*MONO_VOLUME + signal*STEREO_VOLUME; - if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME; + if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME; + if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME; } else { float stereo_carrier = get_oscillator_sin_sample(&pilot_osc); output[i] = mono*MONO_VOLUME + ((stereo+0.2)*stereo_carrier)*STEREO_VOLUME; - if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME; + if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME; + if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME; } } else { if(ssb) { @@ -390,18 +506,22 @@ int main(int argc, char **argv) { output[i] = delay_line(&monoDelay, mono)*MONO_VOLUME + pilot*PILOT_VOLUME + signal*STEREO_VOLUME; - if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME; + if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME; + if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME; } else { float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc,2); float pilot = get_oscillator_sin_sample(&pilot_osc); output[i] = mono*MONO_VOLUME + pilot*PILOT_VOLUME + (stereo*stereo_carrier)*STEREO_VOLUME; - if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME; + if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME; + if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME; } } } else { output[i] = mono*MONO_VOLUME; // Only Mono + if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME; + if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME; } } @@ -418,6 +538,7 @@ int main(int argc, char **argv) { printf("Cleaning up...\n"); pa_simple_free(input_device); if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device); + if(strlen(audio_sca_device) != 0) pa_simple_free(sca_device); if(alsa_output == 0) { pa_simple_free(output_device); } else { diff --git a/src/options.h b/src/options.h deleted file mode 100644 index 88f02b8..0000000 --- a/src/options.h +++ /dev/null @@ -1,5 +0,0 @@ -#define PREEMPHASIS - -#define buffer_maxlength 12288 -#define buffer_tlength_fragsize 8192 -#define buffer_prebuf 16 \ No newline at end of file diff --git a/src/sca_mod.c b/src/sca_mod.c deleted file mode 100644 index 442aa14..0000000 --- a/src/sca_mod.c +++ /dev/null @@ -1,195 +0,0 @@ -#include -#include -#include -#include -#include -#include - -#include "../lib/constants.h" -#include "../lib/oscillator.h" -#include "../lib/filters.h" -#include "../lib/fm_modulator.h" - -#include "options.h" - -#define SAMPLE_RATE 192000 - -#define INPUT_DEVICE "SCA.monitor" -#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback" -// #define ALSA_OUTPUT // Output, not input or both -#define BUFFER_SIZE 512 -#define CLIPPER_THRESHOLD 1 // Adjust this as needed, this also limits deviation, so if you set this to 0.5 then the deviation will be limited to half - -#include -#include -#ifdef ALSA_OUTPUT -#include -#endif - -#define VOLUME 0.1f // SCA Volume -#define VOLUME_AUDIO 1.0f // SCA Audio volume -#define FREQUENCY 67000 // SCA Frequency -#define DEVIATION 7000 // SCA Deviation - -#ifdef PREEMPHASIS -#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america -#endif - -volatile sig_atomic_t to_run = 1; - -float hard_clip(float sample) { - if (sample > CLIPPER_THRESHOLD) { - return CLIPPER_THRESHOLD; // Clip to the upper threshold - } else if (sample < -CLIPPER_THRESHOLD) { - return -CLIPPER_THRESHOLD; // Clip to the lower threshold - } else { - return sample; // No clipping - } -} - -static void stop(int signum) { - (void)signum; - printf("\nReceived stop signal. Cleaning up...\n"); - to_run = 0; -} - -int main() { - printf("SCAMod : SCA Modulator (based on the Stereo encoder STCode) made by radio95 (with help of ChatGPT and Claude, thanks!)\n"); - - // Define formats and buffer atributes - pa_sample_spec audio_format = { - .format = PA_SAMPLE_FLOAT32NE, //Float32 NE, or Float32 Native Endian, the float in c uses the endianess of your pc, or native endian, and float is float32, and double is float64 - .channels = 1, - .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better - }; - - pa_buffer_attr input_buffer_atr = { - .maxlength = buffer_maxlength, - .fragsize = buffer_tlength_fragsize - }; -#ifndef ALSA_OUTPUT - pa_buffer_attr output_buffer_atr = { - .maxlength = buffer_maxlength, - .tlength = buffer_tlength_fragsize, - .prebuf = buffer_prebuf - }; -#endif - - printf("Connecting to input device... (%s)\n", INPUT_DEVICE); - - pa_simple *input_device = pa_simple_new( - NULL, - "SCAMod", - PA_STREAM_RECORD, - INPUT_DEVICE, - "Audio Input", - &audio_format, - NULL, - &input_buffer_atr, - NULL - ); - if (!input_device) { - fprintf(stderr, "Error: cannot open input device.\n"); - return 1; - } - - printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE); - #ifndef ALSA_OUTPUT - pa_simple *output_device = pa_simple_new( - NULL, - "SCAMod", - PA_STREAM_PLAYBACK, - OUTPUT_DEVICE, - "Signal", - &audio_format, - NULL, - &output_buffer_atr, - NULL - ); - if (!output_device) { - fprintf(stderr, "Error: cannot open output device.\n"); - pa_simple_free(input_device); - return 1; - } - #else - snd_pcm_hw_params_t *output_params; - snd_pcm_t *output_handle; - int output_error = snd_pcm_open(&output_handle, OUTPUT_DEVICE, SND_PCM_STREAM_PLAYBACK, 0); - if(output_error < 0) { - fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error)); - pa_simple_free(input_device); - return 1; - } - snd_pcm_hw_params_malloc(&output_params); - snd_pcm_hw_params_any(output_handle, output_params); - snd_pcm_hw_params_set_access(output_handle, output_params, SND_PCM_ACCESS_RW_INTERLEAVED); - snd_pcm_hw_params_set_format(output_handle, output_params, SND_PCM_FORMAT_FLOAT); // Same as pulse's Float32NE - snd_pcm_hw_params_set_channels(output_handle, output_params, 1); - unsigned int rate = SAMPLE_RATE; - int dir; - snd_pcm_hw_params_set_rate_near(output_handle, output_params, &rate, &dir); - snd_pcm_uframes_t frames = BUFFER_SIZE; - snd_pcm_hw_params_set_period_size_near(output_handle, output_params, &frames, &dir); // i don't have a clue why like this - output_error = snd_pcm_hw_params(output_handle, output_params); - if(output_error < 0) { - fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error)); - snd_pcm_close(output_handle); - pa_simple_free(input_device); - return 1; - } - #endif - - FMModulator mod; - init_fm_modulator(&mod, FREQUENCY, DEVIATION, SAMPLE_RATE); -#ifdef PREEMPHASIS - ResistorCapacitor preemp; - init_rc_tau(&preemp, PREEMPHASIS_TAU, SAMPLE_RATE); -#endif - - signal(SIGINT, stop); - signal(SIGTERM, stop); - - int pulse_error; - float input[BUFFER_SIZE]; // Input from device - float signal[BUFFER_SIZE]; // this goes to the output - while (to_run) { - if (pa_simple_read(input_device, input, sizeof(input), &pulse_error) < 0) { - fprintf(stderr, "Error reading from input device: %s\n", pa_strerror(pulse_error)); - to_run = 0; - break; - } - - for (int i = 0; i < BUFFER_SIZE; i++) { - float in = input[i]; - -#ifdef PREEMPHASIS - float preemphasized = apply_pre_emphasis(&preemp, in)*2; - float current_input = hard_clip(preemphasized); -#else - float current_input = hard_clip(in); -#endif - - signal[i] = modulate_fm(&mod, current_input)*VOLUME; - } - -#ifndef ALSA_OUTPUT - if (pa_simple_write(output_device, signal, sizeof(signal), &pulse_error) < 0) { - fprintf(stderr, "Error writing to output device: %s\n", pa_strerror(pulse_error)); - to_run = 0; - break; - } -#else - snd_pcm_writei(output_handle, signal, sizeof(signal)); -#endif - } - printf("Cleaning up...\n"); - pa_simple_free(input_device); - #ifndef ALSA_OUTPUT - pa_simple_free(output_device); - #else - snd_pcm_drain(output_handle); - snd_pcm_close(output_handle); - snd_pcm_hw_params_free(&output_params); - #endif - return 0; -}