ogg_afh.c: Kill unneccessary inclusion of fd.h

[paraslash.git] / ogg_afh.c

diff --git a/ogg_afh.c b/ogg_afh.c
index ba9ef369b8761e6dac8e4c98a08c1673c7941f64..709e7d4b3c5e7278c72ba6fb2451c8798dcab411 100644 (file)
--- a/ogg_afh.c
+++ b/ogg_afh.c
@@ -28,7 +28,6 @@
 #include "afh.h"
 #include "error.h"
 #include "string.h"
 #include "afh.h"
 #include "error.h"
 #include "string.h"

-#include "fd.h"

 
 /** must be big enough to hold header */
 #define CHUNK_SIZE 32768
 
 /** must be big enough to hold header */
 #define CHUNK_SIZE 32768


@@ -37,7 +36,7 @@ static double chunk_time = 0.25;
 static OggVorbis_File *oggvorbis_file;
 static int header_len;
 static char *header;
 static OggVorbis_File *oggvorbis_file;
 static int header_len;
 static char *header;

-static ssize_t *chunk_table, num_chunks;
+static ssize_t *chunk_table;

 static struct audio_format_handler *af;
 
 static int ogg_compute_header_len(FILE *file)
 static struct audio_format_handler *af;
 
 static int ogg_compute_header_len(FILE *file)


@@ -117,7 +116,7 @@ err1:
        return ret;
 }
 
        return ret;
 }
 

-static void tunetable(void)
+static void tunetable(long unsigned num_chunks)

 {
        int i = 1, j = -1, lp = 1;
        while (i < num_chunks) {
 {
        int i = 1, j = -1, lp = 1;
        while (i < num_chunks) {


@@ -147,10 +146,11 @@ static void tunetable(void)
  * offset in the current input file at which the sample containing time i *
  * CHUNK_TIME begins.
  */
  * offset in the current input file at which the sample containing time i *
  * CHUNK_TIME begins.
  */

-static void ogg_compute_chunk_table(double time_total)
+static long unsigned ogg_compute_chunk_table(double time_total)

 {
        int i, ret, num;
        ssize_t max_chunk_len, pos = 0, min = 0, old_pos;
 {
        int i, ret, num;
        ssize_t max_chunk_len, pos = 0, min = 0, old_pos;

+       long unsigned num_chunks;

 
        old_pos = 0;
        ret = 0;
 
        old_pos = 0;
        ret = 0;


@@ -178,9 +178,10 @@ static void ogg_compute_chunk_table(double time_total)
        }
        num_chunks = i - 1;
        chunk_table[i] = pos;
        }
        num_chunks = i - 1;
        chunk_table[i] = pos;

-       tunetable();
+       tunetable(num_chunks);

        PARA_INFO_LOG("%zu chunks (%fs), max chunk: %zd, min chunk: %zd\n",
                num_chunks, chunk_time, max_chunk_len, min);
        PARA_INFO_LOG("%zu chunks (%fs), max chunk: %zd, min chunk: %zd\n",
                num_chunks, chunk_time, max_chunk_len, min);

+       return num_chunks;

 }
 
 static void ogg_close_audio_file(void)
 }
 
 static void ogg_close_audio_file(void)


@@ -196,7 +197,6 @@ static void ogg_close_audio_file(void)
        header_len = 0;
        free(chunk_table);
        chunk_table = NULL;
        header_len = 0;
        free(chunk_table);
        chunk_table = NULL;

-       num_chunks = 0;

 }
 
 static int ogg_save_header(FILE *file, int len)
 }
 
 static int ogg_save_header(FILE *file, int len)


@@ -249,14 +249,13 @@ static int ogg_get_file_info(FILE *file, char *info_str, long unsigned *frames,
        vi_sampling_rate = vi->rate;
        vi_bitrate = ov_bitrate(oggvorbis_file, 0);
        rewind(file);
        vi_sampling_rate = vi->rate;
        vi_bitrate = ov_bitrate(oggvorbis_file, 0);
        rewind(file);

-       ogg_compute_chunk_table(time_total);
+       *frames = ogg_compute_chunk_table(time_total);

        rewind(file);
        rewind(file);

-       *frames = num_chunks;

        *vss_chunk_table = chunk_table;
        *vss_chunk_table = chunk_table;

-       sprintf(info_str, "audio_file_info1:%zu x %lu, %ldkHz, %d channels, %ldkbps\n"
+       sprintf(info_str, "audio_file_info1:%lu x %lu, %ldkHz, %d channels, %ldkbps\n"

                "audio_file_info2: \n"
                "audio_file_info3: \n",
                "audio_file_info2: \n"
                "audio_file_info3: \n",

-               num_chunks, (long unsigned) (chunk_time * 1000 * 1000),
+               *frames, (long unsigned) (chunk_time * 1000 * 1000),

                vi_sampling_rate / 1000, vi->channels, vi_bitrate / 1000
                );
        rewind(file);
                vi_sampling_rate / 1000, vi->channels, vi_bitrate / 1000
                );
        rewind(file);