Use ffmpeg to read audio files, instead of assuming raw format.

[c64tapwav] / decode.cpp

diff --git a/decode.cpp b/decode.cpp
index fdc1347e0e124a7297ab4c6ef4a71e644c58f0fd..e85f57934e685f214f4cda1cf02fc247a9a332fa 100644 (file)
--- a/decode.cpp
+++ b/decode.cpp
@@ -1,32 +1,25 @@
 #include <stdio.h>
 #include <string.h>
 #include <math.h>
-#include <unistd.h>
 #include <assert.h>
 #include <limits.h>
 #include <vector>
 #include <algorithm>
 
+#include "audioreader.h"
 #include "interpolate.h"
+#include "tap.h"
 
 #define BUFSIZE 4096
 #define HYSTERESIS_LIMIT 3000
 #define SAMPLE_RATE 44100
 #define C64_FREQUENCY 985248
-#define TAP_RESOLUTION 8
 
 #define SYNC_PULSE_START 1000
 #define SYNC_PULSE_END 15000
 #define SYNC_PULSE_LENGTH 378.0
 #define SYNC_TEST_TOLERANCE 1.10
 
-struct tap_header {
-       char identifier[12];
-       char version;
-       char reserved[3];
-       unsigned int data_len;
-};
-
 // between [x,x+1]
 double find_zerocrossing(const std::vector<short> &pcm, int x)
 {
@@ -42,7 +35,7 @@ double find_zerocrossing(const std::vector<short> &pcm, int x)
 
        double upper = x;
        double lower = x + 1;
-       while (upper - lower > 1e-6) {
+       while (lower - upper > 1e-3) {
                double mid = 0.5f * (upper + lower);
                if (lanczos_interpolate(pcm, mid) > 0) {
                        upper = mid;
@@ -61,15 +54,11 @@ struct pulse {
        
 int main(int argc, char **argv)
 {
+       make_lanczos_weight_table();
        std::vector<short> pcm;
-
-       while (!feof(stdin)) {
-               short buf[BUFSIZE];
-               ssize_t ret = fread(buf, 2, BUFSIZE, stdin);
-               if (ret >= 0) {
-                       pcm.insert(pcm.end(), buf, buf + ret);
-               }
-       }       
+       if (!read_audio_file(argv[1], &pcm)) {
+               exit(1);
+       }
 
 #if 0
        for (int i = 0; i < LEN; ++i) {