shadermeh.c

@@ -21,47 +21,101 @@ static GLfloat vertex_buffer[] = {
 };
 
 static GLubyte audio_buffer[AUDIO_SAMPLES * AUDIO_CHANNELS];
-static float audio_sample_data[AUDIO_SAMPLES];
-static fftw_complex fftw_in[AUDIO_SAMPLES];
-static fftw_complex fftw_out[AUDIO_SAMPLES];
+size_t sample_pointer = 0;
+size_t sample_data_pointer = 0;
+size_t sample_rate = 0;
+static float *audio_sample_data;
+static float *audio_receive_data;
+static double *fftw_in;
+static fftw_complex *fftw_out;
 static fftw_plan plan;
 
-static int try_fetch_audio(void)
+static int try_fetch_audio(float iTimeDelta)
 {
-	size_t i, count = 0;
-	int ret;
+	/* To avoid generating stale images, we keep our own sample buffer,
+	 * which is then used to move a sliding window of data for the fft and
+	 * wave samples. We need to do this, as otherwise we would set an upper
+	 * limit of fps (20 at 4800kHz sample rate), which would not be good.
+	 * The size of the window is set in the header file. The with our
+	 * approach is that the buffer allows for drifting to occur within the
+	 * buffer limits. If you buffer is 3s long the delay can grow to 3s.
+	 * Choose your buffer size wisely for your application.
+	 */
+	size_t i;
+	ssize_t ret = 0;
+	memset(audio_receive_data, 0, AUDIO_BUFFER_SIZE * 
+		sizeof(*audio_receive_data));
+	sample_pointer += (sample_rate * iTimeDelta);
 
 	for (;;) {
-		ret = read(STDIN_FILENO,
-			   (char *)audio_sample_data + count,
-			   sizeof(audio_sample_data) - count);
+		ret = read(STDIN_FILENO, (char *)audio_receive_data,
+			sizeof(*audio_receive_data)*AUDIO_BUFFER_SIZE);
 
 		if (ret < 0) {
 			if (errno == EINTR)
 				continue;
-			if (errno == EAGAIN)
+			if (errno == EAGAIN || errno == EWOULDBLOCK)
 				break;
 			perror("stdin");
 			return -1;
 		}
 
-		if (ret == 0)
+		if (ret == 0 || ret % sizeof(float) != 0){
 			break;
-
-		count += ret;
 		}
 
-	for (i = 0; i < AUDIO_SAMPLES; ++i)
-		fftw_in[i][0] = audio_sample_data[i];
+		ret /= 4;
+		if((ret + sample_pointer) > AUDIO_BUFFER_SIZE){
+			/* Not enough storage space to store all new audio data,
+			 * will override not output data with new one */
+			 memset(audio_sample_data, 0, 
+				AUDIO_BUFFER_SIZE * sizeof(*audio_sample_data));
+			 memcpy(audio_sample_data, audio_receive_data, 
+				ret * sizeof(*audio_sample_data));
+			 sample_pointer = 0;
+			 sample_data_pointer = ret;
+		}else{
+			memmove(audio_sample_data,
+				&audio_sample_data[sample_pointer], 
+				(AUDIO_BUFFER_SIZE - sample_pointer)*
+				sizeof(*audio_sample_data));
+			if(sample_data_pointer <= sample_pointer){
+				sample_data_pointer = 0;
+
+			}else{
+				sample_data_pointer -= sample_pointer;
+
+			}
+			sample_pointer = 0;
+			size_t len = ret;
+			if((ret + sample_data_pointer) >= AUDIO_BUFFER_SIZE){
+				len = AUDIO_BUFFER_SIZE - sample_data_pointer;
+			}
+			memcpy(&audio_sample_data[sample_data_pointer], 
+				audio_receive_data, len * sizeof(float));
+			sample_data_pointer += len;
+			break;
+		}
+
+	}
+	if((sample_pointer+AUDIO_FFT_SIZE) >= sample_data_pointer){
+		fprintf(stderr, "shadermeh input to slow %zu > %zu! wrapping around!\n", sample_pointer+AUDIO_FFT_SIZE, sample_data_pointer);
+		sample_pointer = 0;
+	}
+
+	memset(fftw_in, 0, sizeof(*fftw_in) * AUDIO_BUFFER_SIZE);
+	memset(fftw_out, 0, sizeof(*fftw_out) * AUDIO_BUFFER_SIZE);
+
+	for (i = 0; i < AUDIO_FFT_SIZE; ++i)
+		fftw_in[i] = audio_sample_data[sample_pointer+i];
 
 	fftw_execute(plan);
 
 	for (i = 0; i < AUDIO_SAMPLES; ++i) {
-		float x = fftw_out[i][0], y = fftw_out[i][1];
-		float a = sqrt(x * x + y * y);
+		float a = cabs(fftw_out[i]);
 
-		audio_buffer[i + AUDIO_SAMPLES] = audio_sample_data[i] * 127.0f	+ 127.0f;
-		audio_buffer[i] = 127.0f + a * 127.0f;
+		audio_buffer[i + AUDIO_SAMPLES] = audio_sample_data[sample_pointer+i] * 127.0f	+ 127.0f;
+		audio_buffer[i] = log(fabsf(a)+1) * 50;
 	}
 
 	return 0;
@@ -159,11 +213,11 @@ static const struct option long_opts[] = {
  	{ "height", required_argument, NULL, 'h' },
 	{ "shader", required_argument, NULL, 's' },
 	{ "to-stdout", no_argument, NULL, 'S' },
-	{ "stdin-audio", no_argument, NULL, 'a' },
+	{ "stdin-audio", required_argument, NULL, 'a' },
 	{ NULL, 0, NULL, 0 },
 };
 
-static const char *short_opts = "w:h:s:Sa";
+static const char *short_opts = "w:a:h:s:S";
 
 static const char *usage_str =
 "shadermeh OPTIONS...\n"
@@ -174,7 +228,7 @@ static const char *usage_str =
 "  --height, -h <pixels>\n"
 "\n"
 "  --to-stdout, -S              Poop raw RGB24 frames to stdout (blocking)\n"
-"  --stdin-audio, -a            Read raw PCM audio from stdin (non-blocking)\n"
+"  --stdin-audio, -a <sample rate>     Read raw PCM audio from stdin (non-blocking)\n"
 "\n"
 "  --shader, -s <shader file>\n"
 "\n";
@@ -188,7 +242,7 @@ int main(int argc, char **argv)
 	void *fb32 = NULL, *fb24 = NULL;
 	const char *shader_file = NULL;
 	GLint major, minor, prog;
-	float iTime, iTimeDelta;
+	float iTime, iTimeDelta = 0;
 	bool have_audio = false;
 	bool to_stdout = false;
 	window *wnd;
@@ -218,6 +272,11 @@ int main(int argc, char **argv)
 			break;
 		case 'a':
 			have_audio = true;
+			sample_rate = strtol(optarg, NULL, 10);
+			audio_sample_data = malloc(AUDIO_BUFFER_SIZE * 
+				sizeof(float));
+			audio_receive_data = malloc(AUDIO_BUFFER_SIZE * 
+				sizeof(float));
 			break;
 		default:
 			fputs(usage_str, stderr);
@@ -341,8 +400,12 @@ int main(int argc, char **argv)
 	glBindSampler(0, sampler_sound);
 
 	if (have_audio) {
-		plan = fftw_plan_dft_1d(AUDIO_SAMPLES, fftw_in, fftw_out,
-					FFTW_FORWARD, FFTW_ESTIMATE);
+		fftw_in = fftw_alloc_real(AUDIO_BUFFER_SIZE);
+		fftw_out = fftw_alloc_complex(AUDIO_BUFFER_SIZE);
+		if(fftw_in == NULL || fftw_out == NULL)
+			goto fail_vao;
+		plan = fftw_plan_dft_r2c_1d(AUDIO_BUFFER_SIZE, fftw_in, fftw_out,
+			 FFTW_MEASURE);
 	}
 
 	/******************** framebuffer object ********************/
@@ -377,7 +440,7 @@ int main(int argc, char **argv)
 		glClear(GL_COLOR_BUFFER_BIT);
 
 		if (have_audio) {
-			if (try_fetch_audio())
+			if (try_fetch_audio(iTimeDelta))
 				break;
 
 			glBindTexture(GL_TEXTURE_2D, sound_tex);
@@ -451,6 +514,8 @@ fail_vao:
 	window_make_current(NULL);
 	free(fb32);
 	free(fb24);
+	fftw_free(fftw_in);
+	fftw_free(fftw_out);
 	window_destroy(wnd);
 	return EXIT_SUCCESS;
 }

shadermeh.h

@@ -27,10 +27,13 @@
 #include <poll.h>
 #include <time.h>
 
-#include <fftw3.h>
 #include <math.h>
+#include <complex.h>
+#include <fftw3.h>
 
-#define AUDIO_SAMPLES (512)
+#define AUDIO_SAMPLES (4096)
+#define AUDIO_BUFFER_SIZE (sample_rate * 3)
+#define AUDIO_FFT_SIZE (AUDIO_SAMPLES * 2)
 #define AUDIO_CHANNELS (2)
 
 typedef struct {