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16-bit sound

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Subversion-branch: /trunk/chocolate-doom
Subversion-revision: 74
This commit is contained in:
Simon Howard 2005-09-05 21:03:43 +00:00
parent 08925d637e
commit b932593c9e
1 changed files with 26 additions and 10 deletions
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36
src/i_sound.c
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@ -1,7 +1,7 @@
// Emacs style mode select -*- C++ -*-
//-----------------------------------------------------------------------------
//
// $Id: i_sound.c 73 2005-09-05 20:32:18Z fraggle $
// $Id: i_sound.c 74 2005-09-05 21:03:43Z fraggle $
//
// Copyright(C) 1993-1996 Id Software, Inc.
// Copyright(C) 2005 Simon Howard
@ -22,6 +22,9 @@
// 02111-1307, USA.
//
// $Log$
// Revision 1.12 2005/09/05 21:03:43 fraggle
// 16-bit sound
//
// Revision 1.11 2005/09/05 20:32:18 fraggle
// Use the system-nonspecific sound code to assign the channel number used
// by SDL. Remove handle tagging stuff.
@ -67,7 +70,7 @@
//-----------------------------------------------------------------------------
static const char
rcsid[] = "$Id: i_sound.c 73 2005-09-05 20:32:18Z fraggle $";
rcsid[] = "$Id: i_sound.c 74 2005-09-05 21:03:43Z fraggle $";
#include <stdio.h>
#include <stdlib.h>
@ -97,25 +100,38 @@ static byte *expand_sound_data(byte *data, int samplerate, int length)
if (samplerate == 11025)
{
// need to expand to 2 channels, and expand 11025->22050
// need to expand to 2 channels, 11025->22050 and 8->16 bit
result = Z_Malloc(length * 4, PU_STATIC, NULL);
result = Z_Malloc(length * 8, PU_STATIC, NULL);
for (i=0; i<length; ++i)
{
result[i * 4] = result[i * 4 + 1]
= result[i * 4 + 2] = result[i * 4 + 3] = data[i];
Uint16 sample;
sample = data[i] | (data[i] << 8);
sample -= 32768;
result[i * 8] = result[i * 8 + 2]
= result[i * 8 + 4] = result[i * 8 + 6] = sample & 0xff;
result[i * 8 + 1] = result[i * 8 + 3]
= result[i * 8 + 5] = result[i * 8 + 7] = (sample >> 8) & 0xff;
}
}
else if (samplerate == 22050)
{
// need to expand to 2 channels (sample rate is already correct)
result = Z_Malloc(length * 2, PU_STATIC, NULL);
result = Z_Malloc(length * 4, PU_STATIC, NULL);
for (i=0; i<length; ++i)
{
result[i * 2] = result[i * 2 + 1] = data[i];
Uint16 sample;
sample = data[i] | (data[i] << 8);
sample -= 32768;
result[i * 4] = result[i * 4 + 2] = sample & 0xff;
result[i * 4 + 1] = result[i * 4 + 3] = (sample >> 8) & 0xff;
}
}
else
@ -145,7 +161,7 @@ static Mix_Chunk *getsfx(int sound)
sound_chunks[sound].allocated = 1;
sound_chunks[sound].abuf = expand_sound_data(data + 8, samplerate, length);
sound_chunks[sound].alen = (length * 2) * (22050 / samplerate);
sound_chunks[sound].alen = (length * 4) * (22050 / samplerate);
sound_chunks[sound].volume = 64;
}
@ -321,7 +337,7 @@ I_InitSound()
return;
}
if (Mix_OpenAudio(22050, AUDIO_U8, 2, 1024) < 0)
if (Mix_OpenAudio(22050, AUDIO_S16LSB, 2, 1024) < 0)
{
fprintf(stderr, "Error initialising SDL_mixer: %s\n", SDL_GetError());
}