diff --git a/.vscode/.server-controller-port.log b/.vscode/.server-controller-port.log index 592365f..0a5cf76 100644 --- a/.vscode/.server-controller-port.log +++ b/.vscode/.server-controller-port.log @@ -1,5 +1,5 @@ { "port": 13452, - "time": 1737738374932, + "time": 1737831115102, "version": "0.0.3" } \ No newline at end of file diff --git a/satire/crosby_stereo_coder.c b/satire/crosby_stereo_coder.c deleted file mode 100644 index 8006eea..0000000 --- a/satire/crosby_stereo_coder.c +++ /dev/null @@ -1,197 +0,0 @@ -#include -#include -#include -#include -#include -#include -#include -#include - -#include "../lib/constants.h" -#include "../lib/oscillator.h" -#include "../lib/filters.h" - -#include "options.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 BUFFER_SIZE 512 -#define CLIPPER_THRESHOLD 0.525 // Adjust this as needed - -#define MONO_VOLUME 0.5f // L+R Signal -#define STEREO_VOLUME_AUDIO 1.0f // L-R signal -#define STEREO_VOLUME_MODULATION 0.5f // L-R signal - -#ifdef PREEMPHASIS -#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america -#endif - -#ifdef LPF -#define LPF_CUTOFF 15000 -#endif - -volatile sig_atomic_t to_run = 1; - -float 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 - } -} - -void uninterleave(const float *input, float *left, float *right, size_t num_samples) { - // For stereo, usually it is like this: LEFT RIGHT LEFT RIGHT LEFT RIGHT so this is used to get LEFT LEFT LEFT and RIGHT RIGHT RIGHT - for (size_t i = 0; i < num_samples/2; i++) { - left[i] = input[i * 2]; - right[i] = input[i * 2 + 1]; - } -} - -static void stop(int signum) { - (void)signum; - printf("\nReceived stop signal. Cleaning up...\n"); - to_run = 0; -} - -int main() { - printf("CrosbySTCode : Stereo encoder (using the crosby system) made by radio95 (with help of ChatGPT and Claude, thanks!)\n"); - // Define formats and buffer atributes - pa_sample_spec stereo_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 = 2, - .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better - }; - pa_sample_spec mono_format = { - .format = PA_SAMPLE_FLOAT32NE, - .channels = 1, - .rate = SAMPLE_RATE - }; - - pa_buffer_attr input_buffer_atr = { - .maxlength = buffer_maxlength, - .fragsize = buffer_tlength_fragsize - }; - pa_buffer_attr output_buffer_atr = { - .maxlength = buffer_maxlength, - .tlength = buffer_tlength_fragsize, - .prebuf = buffer_prebuf - }; - - printf("Connecting to input device... (%s)\n", INPUT_DEVICE); - - pa_simple *input_device = pa_simple_new( - NULL, - "CrosbyStereoEncoder", - PA_STREAM_RECORD, - INPUT_DEVICE, - "Audio Input", - &stereo_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); - - pa_simple *output_device = pa_simple_new( - NULL, - "CrosbyStereoEncoder", - PA_STREAM_PLAYBACK, - OUTPUT_DEVICE, - "MPX", - &mono_format, - NULL, - &output_buffer_atr, - NULL - ); - if (!output_device) { - fprintf(stderr, "Error: cannot open output device.\n"); - pa_simple_free(input_device); - return 1; - } - - Oscillator osc; - init_oscillator(&osc, 50000.0, SAMPLE_RATE); -#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 -#ifdef LPF - ResistorCapacitor lpf_l, lpf_r; - init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE); - init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE); -#endif - - signal(SIGINT, stop); - signal(SIGTERM, stop); - - int pulse_error; - float input[BUFFER_SIZE*2]; // Input from device, interleaved stereo - float left[BUFFER_SIZE+64], right[BUFFER_SIZE+64]; // Audio, same thing as in input but ininterleaved, ai told be there could be a buffer overflow here - float mpx[BUFFER_SIZE]; // MPX, 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; - } - uninterleave(input, left, right, BUFFER_SIZE*2); - - for (int i = 0; i < BUFFER_SIZE; i++) { - float l_in = left[i]; - float r_in = right[i]; - -#ifdef PREEMPHASIS -#ifdef LPF - float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in); - float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in); - float preemphasized_left = apply_pre_emphasis(&preemp_l, lowpassed_left); - float preemphasized_right = apply_pre_emphasis(&preemp_r, lowpassed_right); - float current_left_input = clip(preemphasized_left); - float current_right_input = clip(preemphasized_right); -#else - float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in); - float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in); - float current_left_input = clip(preemphasized_left); - float current_right_input = clip(preemphasized_right); -#endif -#else -#ifdef LPF - float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in); - float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in); - float current_left_input = clip(lowpassed_left); - float current_right_input = clip(lowpassed_right); -#else - float current_left_input = clip(l_in); - float current_right_input = clip(r_in); -#endif -#endif - - float mono = (current_left_input + current_right_input) / 2.0f; // Stereo to Mono - float stereo = (current_left_input - current_right_input) / 2.0f; // Also Stereo to Mono but a bit diffrent - change_oscillator_frequency(&osc, (50000+((stereo*STEREO_VOLUME_AUDIO)*15000))); - mpx[i] = mono * MONO_VOLUME + - get_oscillator_sin_sample(&osc) * STEREO_VOLUME_MODULATION; - } - - if (pa_simple_write(output_device, mpx, sizeof(mpx), &pulse_error) < 0) { - fprintf(stderr, "Error writing to output device: %s\n", pa_strerror(pulse_error)); - to_run = 0; - break; - } - } - printf("Cleaning up...\n"); - pa_simple_free(input_device); - pa_simple_free(output_device); - return 0; -} diff --git a/satire/stereo_sca_mod.c b/satire/stereo_sca_mod.c deleted file mode 100644 index 7b13e93..0000000 --- a/satire/stereo_sca_mod.c +++ /dev/null @@ -1,198 +0,0 @@ -#include -#include -#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 BUFFER_SIZE 512 -#define CLIPPER_THRESHOLD 0.75 // Adjust this as needed, this also limits deviation, so if you set this to 0.5 then the deviation will be limited to half - -#define MONO_VOLUME 0.075f // Mono Volume -#define STEREO_VOLUME 0.025f // Stereo Volume - -#ifdef PREEMPHASIS -#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america -#endif - -#ifdef LPF -#define LPF_CUTOFF 8000 -#endif - -volatile sig_atomic_t to_run = 1; - -void uninterleave(const float *input, float *left, float *right, size_t num_samples) { - // For stereo, usually it is like this: LEFT RIGHT LEFT RIGHT LEFT RIGHT so this is used to get LEFT LEFT LEFT and RIGHT RIGHT RIGHT - for (size_t i = 0; i < num_samples/2; i++) { - left[i] = input[i * 2]; - right[i] = input[i * 2 + 1]; - } -} - -float 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("StereoSCAMod : Stereo SCA Modulator (based on the SCA encoder SCAMod) made by radio95 (with help of ChatGPT and Claude, thanks!)\n"); - - // Define formats and buffer atributes - pa_sample_spec mono_audio_format = { - .format = PA_SAMPLE_FLOAT32LE, - .channels = 1, - .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better - }; - pa_sample_spec stereo_audio_format = { - .format = PA_SAMPLE_FLOAT32LE, - .channels = 2, - .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 - }; - pa_buffer_attr output_buffer_atr = { - .maxlength = buffer_maxlength, - .tlength = buffer_tlength_fragsize, - .prebuf = buffer_prebuf - }; - - printf("Connecting to input device... (%s)\n", INPUT_DEVICE); - - pa_simple *input_device = pa_simple_new( - NULL, - "StereoSCAMod", - PA_STREAM_RECORD, - INPUT_DEVICE, - "Audio Input", - &stereo_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); - - pa_simple *output_device = pa_simple_new( - NULL, - "StereoSCAMod", - PA_STREAM_PLAYBACK, - OUTPUT_DEVICE, - "Signal", - &mono_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; - } - - FMModulator mod_mono, mod_stereo; - init_fm_modulator(&mod_mono, 67000, 6000, SAMPLE_RATE); - init_fm_modulator(&mod_stereo, 80000, 6000, SAMPLE_RATE); -#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 -#ifdef LPF - ResistorCapacitor lpf_l, lpf_r; - init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE); - init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE); -#endif - - signal(SIGINT, stop); - signal(SIGTERM, stop); - - int pulse_error; - float input[BUFFER_SIZE*2]; // Input from device - float left[BUFFER_SIZE+64], right[BUFFER_SIZE+64]; // Audio, same thing as in input but ininterleaved, ai told be there could be a buffer overflow here - 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; - } - uninterleave(input, left, right, BUFFER_SIZE*2); - - for (int i = 0; i < BUFFER_SIZE; i++) { - float l_in = left[i]; - float r_in = right[i]; - -#ifdef PREEMPHASIS -#ifdef LPF - float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in); - float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in); - float preemphasized_left = apply_pre_emphasis(&preemp_l, lowpassed_left); - float preemphasized_right = apply_pre_emphasis(&preemp_r, lowpassed_right); - float current_left_input = clip(preemphasized_left); - float current_right_input = clip(preemphasized_right); -#else - float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in); - float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in); - float current_left_input = clip(preemphasized_left); - float current_right_input = clip(preemphasized_right); -#endif -#else -#ifdef LPF - float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in); - float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in); - float current_left_input = clip(lowpassed_left); - float current_right_input = clip(lowpassed_right); -#else - float current_left_input = clip(l_in); - float current_right_input = clip(r_in); -#endif -#endif - float mono = (current_left_input+current_right_input)/2.0f; - float stereo = (current_left_input-current_right_input)/2.0f; - - signal[i] = modulate_fm(&mod_mono, mono)*MONO_VOLUME+ - modulate_fm(&mod_stereo, stereo)*STEREO_VOLUME; - } - - 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; - } - } - printf("Cleaning up...\n"); - pa_simple_free(input_device); - pa_simple_free(output_device); - return 0; -} diff --git a/src/polar_stereo_coder.c b/src/polar_stereo_coder.c index 783842a..f3799cf 100644 --- a/src/polar_stereo_coder.c +++ b/src/polar_stereo_coder.c @@ -17,6 +17,7 @@ #define INPUT_DEVICE "real_real_tx_audio_input.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 0.525 // Adjust this as needed @@ -75,11 +76,13 @@ int main() { .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); @@ -101,6 +104,7 @@ int main() { printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE); +#ifndef ALSA_OUTPUT pa_simple *output_device = pa_simple_new( NULL, "PolarStereoEncoder", @@ -117,6 +121,33 @@ int main() { 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 Oscillator stereo_osc; init_oscillator(&stereo_osc, 31250.0, SAMPLE_RATE); @@ -186,14 +217,22 @@ int main() { ((stereo+0.2) * stereo_carrier)*STEREO_VOLUME; // the 0.2 add DC, you know what happens then? Carrier wave } +#ifndef ALSA_OUTPUT if (pa_simple_write(output_device, mpx, sizeof(mpx), &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, mpx, sizeof(mpx)); +#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_free(output_handle); return 0; } diff --git a/src/sca_mod.c b/src/sca_mod.c index 8d766b1..a53a6f5 100644 --- a/src/sca_mod.c +++ b/src/sca_mod.c @@ -1,6 +1,4 @@ #include -#include -#include #include #include #include @@ -18,9 +16,16 @@ #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 @@ -66,11 +71,13 @@ int main() { .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); @@ -91,7 +98,7 @@ int main() { } printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE); - + #ifndef ALSA_OUTPUT pa_simple *output_device = pa_simple_new( NULL, "SCAMod", @@ -108,6 +115,33 @@ int main() { 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); @@ -157,14 +191,23 @@ int main() { 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_free(output_handle); + #endif return 0; } diff --git a/src/stereo_coder.c b/src/stereo_coder.c index a93f78c..384e896 100644 --- a/src/stereo_coder.c +++ b/src/stereo_coder.c @@ -23,7 +23,7 @@ #define INPUT_DEVICE "real_real_tx_audio_input.monitor" #define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback" -#define ALSA_OUTPUT // Output, not input or both +// #define ALSA_OUTPUT // Output, not input or both #define BUFFER_SIZE 512 #define CLIPPER_THRESHOLD 0.525 // Adjust this as needed @@ -89,11 +89,13 @@ int main() { .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 int open_pulse_error; diff --git a/wip/tv_encoder.c b/wip/tv_encoder.c deleted file mode 100644 index c0de682..0000000 --- a/wip/tv_encoder.c +++ /dev/null @@ -1,48 +0,0 @@ -// This will encode a black and white TV signal using a luminance value, how does it work? -/* - It encodes the luminance into negative values, so totally white pixel should output -1, a black one should be 0 - - Every new line it sends a 0.5, every frame it is a 1.0 -*/ - -#include "../lib/fm_modulator.h" - -unsigned int rgb_to_luminance(unsigned int r, unsigned int g, unsigned int b) { - return (unsigned int)(0.299 * r + 0.587 * g + 0.114 * b); -} - -typedef struct { - int line; - int pixel; - int lines; - int pixels; -} TVEncoder; - -void init_tv_modulator(TVEncoder* tv, int lines, int pixels) { - tv->pixels = pixels; - tv->lines = lines; - tv->line = 0; - tv->pixel = 0; -} - -float tv_encode(TVEncoder* tv, float luminance) { - float normalized_luminance = luminance / 255.0f; // Normalize luminance to [0, 1] - - if (tv->line < tv->lines) { - if (tv->pixel < tv->pixels) { - // Process pixel within the current line - tv->pixel++; - return -normalized_luminance; - } else { - // End of line: reset pixel counter and move to the next line - tv->pixel = 0; - tv->line++; - return 0.5f; - } - } else { - // End of frame: reset frame counters - tv->line = 0; - tv->pixel = 0; - return 1.0f; - } -} \ No newline at end of file