ref: f291db8f1456b9d1be47eae0ac34bcf0546ce151
parent: 5e9a9a52dbebabda79f221caf318dd8fba4b0781
author: Simon Howard <[email protected]>
date: Sat Aug 7 13:23:09 EDT 2010
Change back filter frequency. Add debug code to dump resampled sound effects to WAV files. Subversion-branch: /trunk/chocolate-doom Subversion-revision: 1952
--- a/src/i_sdlsound.c
+++ b/src/i_sdlsound.c
@@ -25,7 +25,6 @@
//
//-----------------------------------------------------------------------------
-
#include "config.h"
#include <stdio.h>
@@ -41,6 +40,7 @@
#include "deh_main.h"
#include "i_system.h"
+#include "i_swap.h"
#include "s_sound.h"
#include "m_argv.h"
#include "w_wad.h"
@@ -49,6 +49,7 @@
#include "doomdef.h"
#define LOW_PASS_FILTER
+//#define DEBUG_DUMP_WAVS
#define MAX_SOUND_SLICE_TIME 70 /* ms */
#define NUM_CHANNELS 16
@@ -159,12 +160,62 @@
}
}
+#ifdef DEBUG_DUMP_WAVS
+
+// Debug code to dump resampled sound effects to WAV files for analysis.
+
+static void WriteWAV(char *filename, byte *data,
+ uint32_t length, int samplerate)
+{+ FILE *wav;
+ unsigned int i;
+ unsigned short s;
+
+ wav = fopen(filename, "wb");
+
+ // Header
+
+ fwrite("RIFF", 1, 4, wav);+ i = LONG(36 + samplerate);
+ fwrite(&i, 4, 1, wav);
+ fwrite("WAVE", 1, 4, wav);+
+ // Subchunk 1
+
+ fwrite("fmt ", 1, 4, wav);+ i = LONG(16);
+ fwrite(&i, 4, 1, wav); // Length
+ s = SHORT(1);
+ fwrite(&s, 2, 1, wav); // Format (PCM)
+ s = SHORT(2);
+ fwrite(&s, 2, 1, wav); // Channels (2=stereo)
+ i = LONG(samplerate);
+ fwrite(&i, 4, 1, wav); // Sample rate
+ i = LONG(samplerate * 2 * 2);
+ fwrite(&i, 4, 1, wav); // Byte rate (samplerate * stereo * 16 bit)
+ s = SHORT(2 * 2);
+ fwrite(&s, 2, 1, wav); // Block align (stereo * 16 bit)
+ s = SHORT(16);
+ fwrite(&s, 2, 1, wav); // Bits per sample (16 bit)
+
+ // Data subchunk
+
+ fwrite("data", 1, 4, wav);+ i = LONG(length);
+ fwrite(&i, 4, 1, wav); // Data length
+ fwrite(data, 1, length, wav); // Data
+
+ fclose(wav);
+}
+
+#endif
+
// Generic sound expansion function for any sample rate.
static void ExpandSoundData_SDL(byte *data,
- int samplerate,
- uint32_t length,
- Mix_Chunk *destination)
+ int samplerate,
+ uint32_t length,
+ Mix_Chunk *destination)
{SDL_AudioCVT convertor;
uint32_t expanded_length;
@@ -181,7 +232,7 @@
= Z_Malloc(expanded_length, PU_STATIC, &destination->abuf);
// If we can, use the standard / optimized SDL conversion routines.
-
+
if (samplerate <= mixer_freq
&& ConvertibleRatio(samplerate, mixer_freq)
&& SDL_BuildAudioCVT(&convertor,
@@ -244,7 +295,7 @@
// (maximum frequency, by nyquist)
dt = 1.0f / mixer_freq;
- rc = 1.0f / (2 * 3.14f * samplerate);
+ rc = 1.0f / (3.14f * samplerate);
alpha = dt / (rc + dt);
// Both channels are processed in parallel, hence [i-2]:
@@ -346,6 +397,16 @@
samplerate,
length,
&sound_chunks[sound]);
+
+#ifdef DEBUG_DUMP_WAVS
+ {+ char filename[16];
+
+ sprintf(filename, "%s.wav", DEH_String(S_sfx[sound].name));
+ WriteWAV(filename, sound_chunks[sound].abuf,
+ sound_chunks[sound].alen, mixer_freq);
+ }
+#endif
// don't need the original lump any more