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SRB2/src/sdl/sdl_sound.c
James R 7060083db5 Update copyright year to 2020
2020-02-19 14:08:45 -08:00

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// Emacs style mode select -*- C++ -*-
//-----------------------------------------------------------------------------
//
// Copyright (C) 1993-1996 by id Software, Inc.
// Copyright (C) 2014-2020 by Sonic Team Junior.
//
// This program is free software; you can redistribute it and/or
// modify it under the terms of the GNU General Public License
// as published by the Free Software Foundation; either version 2
// of the License, or (at your option) any later version.
//
// The source is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//-----------------------------------------------------------------------------
/// \file
/// \brief SDL interface for sound
#include <math.h>
#include "../doomdef.h"
#ifdef _MSC_VER
#pragma warning(disable : 4214 4244)
#endif
#if defined(HAVE_SDL) && SOUND==SOUND_SDL
#include "SDL.h"
#ifdef _MSC_VER
#pragma warning(default : 4214 4244)
#endif
#ifdef HAVE_MIXER
#include <SDL_mixer.h>
/* This is the version number macro for the current SDL_mixer version: */
#ifndef SDL_MIXER_COMPILEDVERSION
#define SDL_MIXER_COMPILEDVERSION \
SDL_VERSIONNUM(MIX_MAJOR_VERSION, MIX_MINOR_VERSION, MIX_PATCHLEVEL)
#endif
/* This macro will evaluate to true if compiled with SDL_mixer at least X.Y.Z */
#ifndef SDL_MIXER_VERSION_ATLEAST
#define SDL_MIXER_VERSION_ATLEAST(X, Y, Z) \
(SDL_MIXER_COMPILEDVERSION >= SDL_VERSIONNUM(X, Y, Z))
#endif
#else
#define MIX_CHANNELS 8
#endif
#ifdef _WIN32
#include <direct.h>
#elif defined (__GNUC__)
#include <unistd.h>
#endif
#include "../z_zone.h"
#include "../m_swap.h"
#include "../i_system.h"
#include "../i_sound.h"
#include "../m_argv.h"
#include "../m_misc.h"
#include "../w_wad.h"
#include "../screen.h" //vid.WndParent
#include "../doomdef.h"
#include "../doomstat.h"
#include "../s_sound.h"
#include "../d_main.h"
#ifdef HW3SOUND
#include "../hardware/hw3dsdrv.h"
#include "../hardware/hw3sound.h"
#include "hwsym_sdl.h"
#endif
#ifdef HAVE_LIBGME
#include "gme/gme.h"
#endif
// The number of internal mixing channels,
// the samples calculated for each mixing step,
// the size of the 16bit, 2 hardware channel (stereo)
// mixing buffer, and the samplerate of the raw data.
// Needed for calling the actual sound output.
#define NUM_CHANNELS MIX_CHANNELS*4
#define INDEXOFSFX(x) ((sfxinfo_t *)x - S_sfx)
static Uint16 samplecount = 1024; //Alam: 1KB samplecount at 22050hz is 46.439909297052154195011337868481ms of buffer
typedef struct chan_struct
{
// The channel data pointers, start and end.
Uint8 *data; //static unsigned char *channels[NUM_CHANNELS];
Uint8 *end; //static unsigned char *channelsend[NUM_CHANNELS];
// pitch
Uint32 realstep; // The channel step amount...
Uint32 step; //static UINT32 channelstep[NUM_CHANNELS];
Uint32 stepremainder; //static UINT32 channelstepremainder[NUM_CHANNELS];
Uint32 samplerate; // ... and a 0.16 bit remainder of last step.
// Time/gametic that the channel started playing,
// used to determine oldest, which automatically
// has lowest priority.
tic_t starttic; //static INT32 channelstart[NUM_CHANNELS];
// The sound handle, determined on registration,
// used to unregister/stop/modify,
INT32 handle; //static INT32 channelhandles[NUM_CHANNELS];
// SFX id of the playing sound effect.
void *id; // Used to catch duplicates (like chainsaw).
sfxenum_t sfxid; //static INT32 channelids[NUM_CHANNELS];
INT32 vol; //the channel volume
INT32 sep; //the channel pan
// Hardware left and right channel volume lookup.
Sint16* leftvol_lookup; //static INT32 *channelleftvol_lookup[NUM_CHANNELS];
Sint16* rightvol_lookup; //static INT32 *channelrightvol_lookup[NUM_CHANNELS];
} chan_t;
static chan_t channels[NUM_CHANNELS];
// Pitch to stepping lookup
static INT32 steptable[256];
// Volume lookups.
static Sint16 vol_lookup[128 * 256];
UINT8 sound_started = false;
static SDL_mutex *Snd_Mutex = NULL;
//SDL's Audio
static SDL_AudioSpec audio;
static SDL_bool musicStarted = SDL_FALSE;
#ifdef HAVE_MIXER
static SDL_mutex *Msc_Mutex = NULL;
/* FIXME: Make this file instance-specific */
#define MIDI_PATH srb2home
#if defined (__unix__) || defined(__APPLE__) || defined (UNIXCOMMON)
#define MIDI_PATH2 "/tmp"
#endif
#define MIDI_TMPFILE "srb2music"
#define MIDI_TMPFILE2 "srb2wav"
static INT32 musicvol = 62;
#if SDL_MIXER_VERSION_ATLEAST(1,2,2)
#define MIXER_POS //Mix_FadeInMusicPos in 1.2.2+
static void SDLCALL I_FinishMusic(void);
static double loopstartDig = 0.0l;
static SDL_bool loopingDig = SDL_FALSE;
static SDL_bool canlooping = SDL_TRUE;
#endif
#if SDL_MIXER_VERSION_ATLEAST(1,2,7)
#define USE_RWOPS // ok, USE_RWOPS is in here
#if 0 // defined(_WIN32)
#undef USE_RWOPS
#endif
#endif
#if SDL_MIXER_VERSION_ATLEAST(1,2,10)
//#define MIXER_INIT
#endif
#ifdef USE_RWOPS
static void * Smidi[2] = { NULL, NULL };
static SDL_bool canuseRW = SDL_TRUE;
#endif
static const char *fmidi[2] = { MIDI_TMPFILE, MIDI_TMPFILE2};
static const INT32 MIDIfade = 500;
static const INT32 Digfade = 0;
static Mix_Music *music[2] = { NULL, NULL };
#endif
typedef struct srb2audio_s {
void *userdata;
#ifdef HAVE_LIBGME
Music_Emu *gme_emu;
UINT8 gme_pause;
UINT8 gme_loop;
#endif
} srb2audio_t;
static srb2audio_t localdata;
static void Snd_LockAudio(void) //Alam: Lock audio data and uninstall audio callback
{
if (Snd_Mutex) SDL_LockMutex(Snd_Mutex);
else if (sound_disabled) return;
else if (midi_disabled && digital_disabled
#ifdef HW3SOUND
&& hws_mode == HWS_DEFAULT_MODE
#endif
) SDL_LockAudio();
#ifdef HAVE_MIXER
else if (musicStarted) Mix_SetPostMix(NULL, NULL);
#endif
}
static void Snd_UnlockAudio(void) //Alam: Unlock audio data and reinstall audio callback
{
if (Snd_Mutex) SDL_UnlockMutex(Snd_Mutex);
else if (sound_disabled) return;
else if (midi_disabled && digital_disabled
#ifdef HW3SOUND
&& hws_mode == HWS_DEFAULT_MODE
#endif
) SDL_UnlockAudio();
#ifdef HAVE_MIXER
else if (musicStarted) Mix_SetPostMix(audio.callback, audio.userdata);
#endif
}
static inline Uint16 Snd_LowerRate(Uint16 sr)
{
if (sr <= audio.freq) // already lowered rate?
return sr; // good then
for (;sr > audio.freq;) // not good?
{ // then let see...
if (sr % 2) // can we div by half?
return sr; // no, just use the currect rate
sr /= 2; // we can? wonderful
} // let start over again
if (sr == audio.freq) // did we drop to the desired rate?
return sr; // perfect! but if not
return sr*2; // just keep it just above the output sample rate
}
#ifdef _MSC_VER
#pragma warning(disable : 4200)
#pragma pack(1)
#endif
typedef struct
{
Uint16 header; // 3?
Uint16 samplerate; // 11025+
Uint16 samples; // number of samples
Uint16 dummy; // 0
Uint8 data[0]; // data;
} ATTRPACK dssfx_t;
#ifdef _MSC_VER
#pragma pack()
#pragma warning(default : 4200)
#endif
//
// This function loads the sound data from the WAD lump,
// for single sound.
//
static void *getsfx(lumpnum_t sfxlump, size_t *len)
{
dssfx_t *sfx, *paddedsfx;
Uint16 sr , csr;
size_t size = *len;
SDL_AudioCVT sfxcvt;
sfx = (dssfx_t *)malloc(size);
if (sfx) W_ReadLump(sfxlump, (void *)sfx);
else return NULL;
sr = SHORT(sfx->samplerate);
csr = Snd_LowerRate(sr);
if (sr > csr && SDL_BuildAudioCVT(&sfxcvt, AUDIO_U8, 1, sr, AUDIO_U8, 1, csr))
{//Alam: Setup the AudioCVT for the SFX
sfxcvt.len = (INT32)size-8; //Alam: Chop off the header
sfxcvt.buf = malloc(sfxcvt.len * sfxcvt.len_mult); //Alam: make room
if (sfxcvt.buf) M_Memcpy(sfxcvt.buf, &(sfx->data), sfxcvt.len); //Alam: copy the sfx sample
if (sfxcvt.buf && SDL_ConvertAudio(&sfxcvt) == 0) //Alam: let convert it!
{
size = sfxcvt.len_cvt + 8;
*len = sfxcvt.len_cvt;
// Allocate from zone memory.
paddedsfx = (dssfx_t *) Z_Malloc(size, PU_SOUND, NULL);
// Now copy and pad.
M_Memcpy(paddedsfx->data, sfxcvt.buf, sfxcvt.len_cvt);
free(sfxcvt.buf);
M_Memcpy(paddedsfx,sfx,8);
paddedsfx->samplerate = SHORT(csr); // new freq
}
else //Alam: the convert failed, not needed or I couldn't malloc the buf
{
if (sfxcvt.buf) free(sfxcvt.buf);
*len = size - 8;
// Allocate from zone memory then copy and pad
paddedsfx = (dssfx_t *)M_Memcpy(Z_Malloc(size, PU_SOUND, NULL), sfx, size);
}
}
else
{
// Pads the sound effect out to the mixing buffer size.
// The original realloc would interfere with zone memory.
*len = size - 8;
// Allocate from zone memory then copy and pad
paddedsfx = (dssfx_t *)M_Memcpy(Z_Malloc(size, PU_SOUND, NULL), sfx, size);
}
// Remove the cached lump.
free(sfx);
// Return allocated padded data.
return paddedsfx;
}
// used to (re)calculate channel params based on vol, sep, pitch
static void I_SetChannelParams(chan_t *c, INT32 vol, INT32 sep, INT32 step)
{
INT32 leftvol;
INT32 rightvol;
c->vol = vol;
c->sep = sep;
c->step = c->realstep = step;
if (step != steptable[128])
c->step += (((c->samplerate<<16)/audio.freq)-65536);
else if (c->samplerate != (unsigned)audio.freq)
c->step = ((c->samplerate<<16)/audio.freq);
// x^2 separation, that is, orientation/stereo.
// range is: 0 (left) - 255 (right)
// Volume arrives in range 0..255 and it must be in 0..cv_soundvolume...
vol = (vol * cv_soundvolume.value) >> 7;
// note: >> 6 would use almost the entire dynamical range, but
// then there would be no "dynamical room" for other sounds :-/
leftvol = vol - ((vol*sep*sep) >> 16); ///(256*256);
sep = 255 - sep;
rightvol = vol - ((vol*sep*sep) >> 16);
// Sanity check, clamp volume.
if (rightvol < 0)
rightvol = 0;
else if (rightvol > 127)
rightvol = 127;
if (leftvol < 0)
leftvol = 0;
else if (leftvol > 127)
leftvol = 127;
// Get the proper lookup table piece
// for this volume level
c->leftvol_lookup = &vol_lookup[leftvol*256];
c->rightvol_lookup = &vol_lookup[rightvol*256];
}
static INT32 FindChannel(INT32 handle)
{
INT32 i;
for (i = 0; i < NUM_CHANNELS; i++)
if (channels[i].handle == handle)
return i;
// not found
return -1;
}
//
// This function adds a sound to the
// list of currently active sounds,
// which is maintained as a given number
// (eight, usually) of internal channels.
// Returns a handle.
//
static INT32 addsfx(sfxenum_t sfxid, INT32 volume, INT32 step, INT32 seperation)
{
static UINT16 handlenums = 0;
INT32 i, slot, oldestnum = 0;
tic_t oldest = gametic;
// Play these sound effects only one at a time.
#if 1
if (
#if 0
sfxid == sfx_stnmov || sfxid == sfx_sawup || sfxid == sfx_sawidl || sfxid == sfx_sawful || sfxid == sfx_sawhit || sfxid == sfx_pistol
#else
( sfx_litng1 <= sfxid && sfxid >= sfx_litng4 )
|| sfxid == sfx_trfire || sfxid == sfx_alarm || sfxid == sfx_spin
|| sfxid == sfx_athun1 || sfxid == sfx_athun2 || sfxid == sfx_rainin
#endif
)
{
// Loop all channels, check.
for (i = 0; i < NUM_CHANNELS; i++)
{
// Active, and using the same SFX?
if ((channels[i].end) && (channels[i].sfxid == sfxid))
{
// Reset.
channels[i].end = NULL;
// We are sure that iff,
// there will only be one.
break;
}
}
}
#endif
// Loop all channels to find oldest SFX.
for (i = 0; (i < NUM_CHANNELS) && (channels[i].end); i++)
{
if (channels[i].starttic < oldest)
{
oldestnum = i;
oldest = channels[i].starttic;
}
}
// Tales from the cryptic.
// If we found a channel, fine.
// If not, we simply overwrite the first one, 0.
// Probably only happens at startup.
if (i == NUM_CHANNELS)
slot = oldestnum;
else
slot = i;
channels[slot].end = NULL;
// Okay, in the less recent channel,
// we will handle the new SFX.
// Set pointer to raw data.
channels[slot].data = (Uint8 *)S_sfx[sfxid].data;
channels[slot].samplerate = (channels[slot].data[3]<<8)+channels[slot].data[2];
channels[slot].data += 8; //Alam: offset of the sound header
while (FindChannel(handlenums)!=-1)
{
handlenums++;
// Reset current handle number, limited to 0..65535.
if (handlenums == UINT16_MAX)
handlenums = 0;
}
// Assign current handle number.
// Preserved so sounds could be stopped.
channels[slot].handle = handlenums;
// Restart steper
channels[slot].stepremainder = 0;
// Should be gametic, I presume.
channels[slot].starttic = gametic;
I_SetChannelParams(&channels[slot], volume, seperation, step);
// Preserve sound SFX id,
// e.g. for avoiding duplicates of chainsaw.
channels[slot].id = S_sfx[sfxid].data;
channels[slot].sfxid = sfxid;
// Set pointer to end of raw data.
channels[slot].end = channels[slot].data + S_sfx[sfxid].length;
// You tell me.
return handlenums;
}
//
// SFX API
// Note: this was called by S_Init.
// However, whatever they did in the
// old DPMS based DOS version, this
// were simply dummies in the Linux
// version.
// See soundserver initdata().
//
// Well... To keep compatibility with legacy doom, I have to call this in
// I_InitSound since it is not called in S_Init... (emanne@absysteme.fr)
static inline void I_SetChannels(void)
{
// Init internal lookups (raw data, mixing buffer, channels).
// This function sets up internal lookups used during
// the mixing process.
INT32 i;
INT32 j;
INT32 *steptablemid = steptable + 128;
if (sound_disabled)
return;
// This table provides step widths for pitch parameters.
for (i = -128; i < 128; i++)
{
const double po = pow((double)(2.0l), (double)(i / 64.0l));
steptablemid[i] = (INT32)(po * 65536.0l);
}
// Generates volume lookup tables
// which also turn the unsigned samples
// into signed samples.
for (i = 0; i < 128; i++)
for (j = 0; j < 256; j++)
{
//From PrDoom
// proff - made this a little bit softer, because with
// full volume the sound clipped badly
vol_lookup[i * 256 + j] = (Sint16)((i * (j - 128) * 256) / 127);
}
}
void I_SetSfxVolume(UINT8 volume)
{
INT32 i;
(void)volume;
//Snd_LockAudio();
for (i = 0; i < NUM_CHANNELS; i++)
if (channels[i].end) I_SetChannelParams(&channels[i], channels[i].vol, channels[i].sep, channels[i].realstep);
//Snd_UnlockAudio();
}
void *I_GetSfx(sfxinfo_t *sfx)
{
if (sfx->lumpnum == LUMPERROR)
sfx->lumpnum = S_GetSfxLumpNum(sfx);
// else if (sfx->lumpnum != S_GetSfxLumpNum(sfx))
// I_FreeSfx(sfx);
#ifdef HW3SOUND
if (hws_mode != HWS_DEFAULT_MODE)
return HW3S_GetSfx(sfx);
#endif
if (sfx->data)
return sfx->data; //Alam: I have it done!
sfx->length = W_LumpLength(sfx->lumpnum);
return getsfx(sfx->lumpnum, &sfx->length);
}
void I_FreeSfx(sfxinfo_t * sfx)
{
// if (sfx->lumpnum<0)
// return;
#ifdef HW3SOUND
if (hws_mode != HWS_DEFAULT_MODE)
{
HW3S_FreeSfx(sfx);
}
else
#endif
{
size_t i;
for (i = 1; i < NUMSFX; i++)
{
// Alias? Example is the chaingun sound linked to pistol.
if (S_sfx[i].data == sfx->data)
{
if (S_sfx+i != sfx) S_sfx[i].data = NULL;
S_sfx[i].lumpnum = LUMPERROR;
S_sfx[i].length = 0;
}
}
//Snd_LockAudio(); //Alam: too much?
// Loop all channels, check.
for (i = 0; i < NUM_CHANNELS; i++)
{
// Active, and using the same SFX?
if (channels[i].end && channels[i].id == sfx->data)
{
channels[i].end = NULL; // Reset.
}
}
//Snd_UnlockAudio(); //Alam: too much?
Z_Free(sfx->data);
}
sfx->data = NULL;
sfx->lumpnum = LUMPERROR;
}
//
// Starting a sound means adding it
// to the current list of active sounds
// in the internal channels.
// As the SFX info struct contains
// e.g. a pointer to the raw data,
// it is ignored.
// As our sound handling does not handle
// priority, it is ignored.
// Pitching (that is, increased speed of playback)
// is set, but currently not used by mixing.
//
INT32 I_StartSound(sfxenum_t id, UINT8 vol, UINT8 sep, UINT8 pitch, UINT8 priority, INT32 channel)
{
(void)priority;
(void)pitch;
(void)channel;
if (sound_disabled)
return 0;
if (S_sfx[id].data == NULL) return -1;
Snd_LockAudio();
id = addsfx(id, vol, steptable[pitch], sep);
Snd_UnlockAudio();
return id; // Returns a handle (not used).
}
void I_StopSound(INT32 handle)
{
// You need the handle returned by StartSound.
// Would be looping all channels,
// tracking down the handle,
// an setting the channel to zero.
INT32 i;
i = FindChannel(handle);
if (i != -1)
{
//Snd_LockAudio(); //Alam: too much?
channels[i].end = NULL;
//Snd_UnlockAudio(); //Alam: too much?
channels[i].handle = -1;
channels[i].starttic = 0;
}
}
boolean I_SoundIsPlaying(INT32 handle)
{
boolean isplaying = false;
int chan = FindChannel(handle);
if (chan != -1)
isplaying = (channels[chan].end != NULL);
return isplaying;
}
FUNCINLINE static ATTRINLINE void I_UpdateStream8S(Uint8 *stream, int len)
{
// Mix current sound data.
// Data, from raw sound
register Sint16 dr; // Right 8bit stream
register Uint8 sample; // Center 8bit sfx
register Sint16 dl; // Left 8bit stream
// Pointers in audio stream
Sint8 *rightout = (Sint8 *)stream; // currect right
Sint8 *leftout = rightout + 1;// currect left
const Uint8 step = 2; // Step in stream, left and right, thus two.
INT32 chan; // Mixing channel index.
// Determine end of the stream
len /= 2; // not 8bit mono samples, 8bit stereo ones
if (Snd_Mutex) SDL_LockMutex(Snd_Mutex);
// Mix sounds into the mixing buffer.
// Loop over len
while (len--)
{
// Reset left/right value.
dl = *leftout;
dr = *rightout;
// Love thy L2 cache - made this a loop.
// Now more channels could be set at compile time
// as well. Thus loop those channels.
for (chan = 0; chan < NUM_CHANNELS; chan++)
{
// Check channel, if active.
if (channels[chan].end)
{
#if 1
// Get the raw data from the channel.
sample = channels[chan].data[0];
#else
// linear filtering from PrDoom
sample = (((Uint32)channels[chan].data[0] *(0x10000 - channels[chan].stepremainder))
+ ((Uint32)channels[chan].data[1]) * (channels[chan].stepremainder))) >> 16;
#endif
// Add left and right part
// for this channel (sound)
// to the current data.
// Adjust volume accordingly.
dl = (Sint16)(dl+(channels[chan].leftvol_lookup[sample]>>8));
dr = (Sint16)(dr+(channels[chan].rightvol_lookup[sample]>>8));
// Increment stepage
channels[chan].stepremainder += channels[chan].step;
// Check whether we are done.
if (channels[chan].data + (channels[chan].stepremainder >> 16) >= channels[chan].end)
channels[chan].end = NULL;
else
{
// step to next sample
channels[chan].data += (channels[chan].stepremainder >> 16);
// Limit to LSB???
channels[chan].stepremainder &= 0xffff;
}
}
}
// Clamp to range. Left hardware channel.
// Has been char instead of short.
if (dl > 0x7f)
*leftout = 0x7f;
else if (dl < -0x80)
*leftout = -0x80;
else
*leftout = (Sint8)dl;
// Same for right hardware channel.
if (dr > 0x7f)
*rightout = 0x7f;
else if (dr < -0x80)
*rightout = -0x80;
else
*rightout = (Sint8)dr;
// Increment current pointers in stream
leftout += step;
rightout += step;
}
if (Snd_Mutex) SDL_UnlockMutex(Snd_Mutex);
}
FUNCINLINE static ATTRINLINE void I_UpdateStream8M(Uint8 *stream, int len)
{
// Mix current sound data.
// Data, from raw sound
register Sint16 d; // Mono 8bit stream
register Uint8 sample; // Center 8bit sfx
// Pointers in audio stream
Sint8 *monoout = (Sint8 *)stream; // currect mono
const Uint8 step = 1; // Step in stream, left and right, thus two.
INT32 chan; // Mixing channel index.
// Determine end of the stream
//len /= 1; // not 8bit mono samples, 8bit mono ones?
if (Snd_Mutex) SDL_LockMutex(Snd_Mutex);
// Mix sounds into the mixing buffer.
// Loop over len
while (len--)
{
// Reset left/right value.
d = *monoout;
// Love thy L2 cache - made this a loop.
// Now more channels could be set at compile time
// as well. Thus loop those channels.
for (chan = 0; chan < NUM_CHANNELS; chan++)
{
// Check channel, if active.
if (channels[chan].end)
{
#if 1
// Get the raw data from the channel.
sample = channels[chan].data[0];
#else
// linear filtering from PrDoom
sample = (((Uint32)channels[chan].data[0] *(0x10000 - channels[chan].stepremainder))
+ ((Uint32)channels[chan].data[1]) * (channels[chan].stepremainder))) >> 16;
#endif
// Add left and right part
// for this channel (sound)
// to the current data.
// Adjust volume accordingly.
d = (Sint16)(d+((channels[chan].leftvol_lookup[sample] + channels[chan].rightvol_lookup[sample])>>9));
// Increment stepage
channels[chan].stepremainder += channels[chan].step;
// Check whether we are done.
if (channels[chan].data + (channels[chan].stepremainder >> 16) >= channels[chan].end)
channels[chan].end = NULL;
else
{
// step to next sample
channels[chan].data += (channels[chan].stepremainder >> 16);
// Limit to LSB???
channels[chan].stepremainder &= 0xffff;
}
}
}
// Clamp to range. Left hardware channel.
// Has been char instead of short.
if (d > 0x7f)
*monoout = 0x7f;
else if (d < -0x80)
*monoout = -0x80;
else
*monoout = (Sint8)d;
// Increment current pointers in stream
monoout += step;
}
if (Snd_Mutex) SDL_UnlockMutex(Snd_Mutex);
}
FUNCINLINE static ATTRINLINE void I_UpdateStream16S(Uint8 *stream, int len)
{
// Mix current sound data.
// Data, from raw sound
register Sint32 dr; // Right 16bit stream
register Uint8 sample; // Center 8bit sfx
register Sint32 dl; // Left 16bit stream
// Pointers in audio stream
Sint16 *rightout = (Sint16 *)(void *)stream; // currect right
Sint16 *leftout = rightout + 1;// currect left
const Uint8 step = 2; // Step in stream, left and right, thus two.
INT32 chan; // Mixing channel index.
// Determine end of the stream
len /= 4; // not 8bit mono samples, 16bit stereo ones
if (Snd_Mutex) SDL_LockMutex(Snd_Mutex);
// Mix sounds into the mixing buffer.
// Loop over len
while (len--)
{
// Reset left/right value.
dl = *leftout;
dr = *rightout;
// Love thy L2 cache - made this a loop.
// Now more channels could be set at compile time
// as well. Thus loop those channels.
for (chan = 0; chan < NUM_CHANNELS; chan++)
{
// Check channel, if active.
if (channels[chan].end)
{
#if 1
// Get the raw data from the channel.
sample = channels[chan].data[0];
#else
// linear filtering from PrDoom
sample = (((Uint32)channels[chan].data[0] *(0x10000 - channels[chan].stepremainder))
+ ((Uint32)channels[chan].data[1]) * (channels[chan].stepremainder))) >> 16;
#endif
// Add left and right part
// for this channel (sound)
// to the current data.
// Adjust volume accordingly.
dl += channels[chan].leftvol_lookup[sample];
dr += channels[chan].rightvol_lookup[sample];
// Increment stepage
channels[chan].stepremainder += channels[chan].step;
// Check whether we are done.
if (channels[chan].data + (channels[chan].stepremainder >> 16) >= channels[chan].end)
channels[chan].end = NULL;
else
{
// step to next sample
channels[chan].data += (channels[chan].stepremainder >> 16);
// Limit to LSB???
channels[chan].stepremainder &= 0xffff;
}
}
}
// Clamp to range. Left hardware channel.
// Has been char instead of short.
if (dl > 0x7fff)
*leftout = 0x7fff;
else if (dl < -0x8000)
*leftout = -0x8000;
else
*leftout = (Sint16)dl;
// Same for right hardware channel.
if (dr > 0x7fff)
*rightout = 0x7fff;
else if (dr < -0x8000)
*rightout = -0x8000;
else
*rightout = (Sint16)dr;
// Increment current pointers in stream
leftout += step;
rightout += step;
}
if (Snd_Mutex) SDL_UnlockMutex(Snd_Mutex);
}
FUNCINLINE static ATTRINLINE void I_UpdateStream16M(Uint8 *stream, int len)
{
// Mix current sound data.
// Data, from raw sound
register Sint32 d; // Mono 16bit stream
register Uint8 sample; // Center 8bit sfx
// Pointers in audio stream
Sint16 *monoout = (Sint16 *)(void *)stream; // currect mono
const Uint8 step = 1; // Step in stream, left and right, thus two.
INT32 chan; // Mixing channel index.
// Determine end of the stream
len /= 2; // not 8bit mono samples, 16bit mono ones
if (Snd_Mutex) SDL_LockMutex(Snd_Mutex);
// Mix sounds into the mixing buffer.
// Loop over len
while (len--)
{
// Reset left/right value.
d = *monoout;
// Love thy L2 cache - made this a loop.
// Now more channels could be set at compile time
// as well. Thus loop those channels.
for (chan = 0; chan < NUM_CHANNELS; chan++)
{
// Check channel, if active.
if (channels[chan].end)
{
#if 1
// Get the raw data from the channel.
sample = channels[chan].data[0];
#else
// linear filtering from PrDoom
sample = (((Uint32)channels[chan].data[0] *(0x10000 - channels[chan].stepremainder))
+ ((Uint32)channels[chan].data[1]) * (channels[chan].stepremainder))) >> 16;
#endif
// Add left and right part
// for this channel (sound)
// to the current data.
// Adjust volume accordingly.
d += (channels[chan].leftvol_lookup[sample] + channels[chan].rightvol_lookup[sample])>>1;
// Increment stepage
channels[chan].stepremainder += channels[chan].step;
// Check whether we are done.
if (channels[chan].data + (channels[chan].stepremainder >> 16) >= channels[chan].end)
channels[chan].end = NULL;
else
{
// step to next sample
channels[chan].data += (channels[chan].stepremainder >> 16);
// Limit to LSB???
channels[chan].stepremainder &= 0xffff;
}
}
}
// Clamp to range. Left hardware channel.
// Has been char instead of short.
if (d > 0x7fff)
*monoout = 0x7fff;
else if (d < -0x8000)
*monoout = -0x8000;
else
*monoout = (Sint16)d;
// Increment current pointers in stream
monoout += step;
}
if (Snd_Mutex) SDL_UnlockMutex(Snd_Mutex);
}
#if 0 //#ifdef HAVE_LIBGME
static void I_UpdateSteamGME(Music_Emu *emu, INT16 *stream, int len, UINT8 looping)
{
#define GME_BUFFER_LEN 44100*2048
// Mix current sound data.
// Data, from raw sound
register Sint32 da;
static short gme_buffer[GME_BUFFER_LEN]; // a large buffer for gme
Sint16 *in = gme_buffer;
do
{
int out = min(GME_BUFFER_LEN, len);
if ( gme_play( emu, len, gme_buffer ) ) { } // ignore error
len -= out;
while (out--)
{
//Left
da = *in;
in++;
da += *stream;
stream++;
//Right
da = *in;
in++;
da += *stream;
stream++;
}
if (gme_track_ended( emu ))
{
if (looping)
gme_seek( emu, 0);
else
break;
}
} while ( len );
#undef GME_BUFFER_LEN
}
#endif
static void SDLCALL I_UpdateStream(void *userdata, Uint8 *stream, int len)
{
if (!sound_started || !userdata)
return;
memset(stream, 0x00, len); // only work in !AUDIO_U8, that needs 0x80
if ((audio.channels != 1 && audio.channels != 2) ||
(audio.format != AUDIO_S8 && audio.format != AUDIO_S16SYS))
; // no function to encode this type of stream
else if (audio.channels == 1 && audio.format == AUDIO_S8)
I_UpdateStream8M(stream, len);
else if (audio.channels == 2 && audio.format == AUDIO_S8)
I_UpdateStream8S(stream, len);
else if (audio.channels == 1 && audio.format == AUDIO_S16SYS)
I_UpdateStream16M(stream, len);
else if (audio.channels == 2 && audio.format == AUDIO_S16SYS)
{
I_UpdateStream16S(stream, len);
// Crashes! But no matter; this build doesn't play music anyway...
// #ifdef HAVE_LIBGME
// if (userdata)
// {
// srb2audio_t *sa_userdata = userdata;
// if (!sa_userdata->gme_pause)
// I_UpdateSteamGME(sa_userdata->gme_emu, (INT16 *)stream, len/4, sa_userdata->gme_loop);
// }
// #endif
}
}
void I_UpdateSoundParams(INT32 handle, UINT8 vol, UINT8 sep, UINT8 pitch)
{
// Would be using the handle to identify
// on which channel the sound might be active,
// and resetting the channel parameters.
INT32 i = FindChannel(handle);
if (i != -1 && channels[i].end)
{
//Snd_LockAudio(); //Alam: too much?
I_SetChannelParams(&channels[i], vol, sep, steptable[pitch]);
//Snd_UnlockAudio(); //Alam: too much?
}
}
#ifdef HW3SOUND
static void *soundso = NULL;
static INT32 Init3DSDriver(const char *soName)
{
if (soName) soundso = hwOpen(soName);
#if defined (_WIN32) && defined (_X86_) && !defined (STATIC3DS)
HW3DS.pfnStartup = hwSym("Startup@8",soundso);
HW3DS.pfnShutdown = hwSym("Shutdown@0",soundso);
HW3DS.pfnAddSfx = hwSym("AddSfx@4",soundso);
HW3DS.pfnAddSource = hwSym("AddSource@8",soundso);
HW3DS.pfnStartSource = hwSym("StartSource@4",soundso);
HW3DS.pfnStopSource = hwSym("StopSource@4",soundso);
HW3DS.pfnGetHW3DSVersion = hwSym("GetHW3DSVersion@0",soundso);
HW3DS.pfnBeginFrameUpdate = hwSym("BeginFrameUpdate@0",soundso);
HW3DS.pfnEndFrameUpdate = hwSym("EndFrameUpdate@0",soundso);
HW3DS.pfnIsPlaying = hwSym("IsPlaying@4",soundso);
HW3DS.pfnUpdateListener = hwSym("UpdateListener@8",soundso);
HW3DS.pfnUpdateSourceParms = hwSym("UpdateSourceParms@12",soundso);
HW3DS.pfnSetCone = hwSym("SetCone@8",soundso);
HW3DS.pfnSetGlobalSfxVolume = hwSym("SetGlobalSfxVolume@4",soundso);
HW3DS.pfnUpdate3DSource = hwSym("Update3DSource@8",soundso);
HW3DS.pfnReloadSource = hwSym("ReloadSource@8",soundso);
HW3DS.pfnKillSource = hwSym("KillSource@4",soundso);
HW3DS.pfnKillSfx = hwSym("KillSfx@4",soundso);
HW3DS.pfnGetHW3DSTitle = hwSym("GetHW3DSTitle@8",soundso);
#else
HW3DS.pfnStartup = hwSym("Startup",soundso);
HW3DS.pfnShutdown = hwSym("Shutdown",soundso);
HW3DS.pfnAddSfx = hwSym("AddSfx",soundso);
HW3DS.pfnAddSource = hwSym("AddSource",soundso);
HW3DS.pfnStartSource = hwSym("StartSource",soundso);
HW3DS.pfnStopSource = hwSym("StopSource",soundso);
HW3DS.pfnGetHW3DSVersion = hwSym("GetHW3DSVersion",soundso);
HW3DS.pfnBeginFrameUpdate = hwSym("BeginFrameUpdate",soundso);
HW3DS.pfnEndFrameUpdate = hwSym("EndFrameUpdate",soundso);
HW3DS.pfnIsPlaying = hwSym("IsPlaying",soundso);
HW3DS.pfnUpdateListener = hwSym("UpdateListener",soundso);
HW3DS.pfnUpdateSourceParms = hwSym("UpdateSourceParms",soundso);
HW3DS.pfnSetCone = hwSym("SetCone",soundso);
HW3DS.pfnSetGlobalSfxVolume = hwSym("SetGlobalSfxVolume",soundso);
HW3DS.pfnUpdate3DSource = hwSym("Update3DSource",soundso);
HW3DS.pfnReloadSource = hwSym("ReloadSource",soundso);
HW3DS.pfnKillSource = hwSym("KillSource",soundso);
HW3DS.pfnKillSfx = hwSym("KillSfx",soundso);
HW3DS.pfnGetHW3DSTitle = hwSym("GetHW3DSTitle",soundso);
#endif
// if (HW3DS.pfnUpdateListener2 && HW3DS.pfnUpdateListener2 != soundso)
return true;
// else
// return false;
}
#endif
void I_ShutdownSound(void)
{
if (sound_disabled || !sound_started)
return;
CONS_Printf("I_ShutdownSound: ");
#ifdef HW3SOUND
if (hws_mode != HWS_DEFAULT_MODE)
{
HW3S_Shutdown();
hwClose(soundso);
return;
}
#endif
if (midi_disabled && digital_disabled)
SDL_CloseAudio();
CONS_Printf("%s", M_GetText("shut down\n"));
sound_started = false;
SDL_QuitSubSystem(SDL_INIT_AUDIO);
if (Snd_Mutex)
SDL_DestroyMutex(Snd_Mutex);
Snd_Mutex = NULL;
}
void I_UpdateSound(void)
{
}
void I_StartupSound(void)
{
#ifdef HW3SOUND
const char *sdrv_name = NULL;
#endif
#ifndef HAVE_MIXER
midi_disabled = digital_disabled = true;
#endif
memset(channels, 0, sizeof (channels)); //Alam: Clean it
audio.format = AUDIO_S16SYS;
audio.channels = 2;
audio.callback = I_UpdateStream;
audio.userdata = &localdata;
// Configure sound device
CONS_Printf("I_StartupSound:\n");
#ifdef _WIN32
// Force DirectSound instead of WASAPI
// SDL 2.0.6+ defaults to the latter and it screws up our sound effects
SDL_setenv("SDL_AUDIODRIVER", "directsound", 1);
#endif
// EE inits audio first so we're following along.
if (SDL_WasInit(SDL_INIT_AUDIO) == SDL_INIT_AUDIO)
CONS_Printf("SDL Audio already started\n");
else if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0)
{
CONS_Alert(CONS_ERROR, "Error initializing SDL Audio: %s\n", SDL_GetError());
// call to start audio failed -- we do not have it
return;
}
// Open the audio device
if (M_CheckParm ("-freq") && M_IsNextParm())
{
audio.freq = atoi(M_GetNextParm());
if (!audio.freq) audio.freq = cv_samplerate.value;
audio.samples = (Uint16)((samplecount/2)*(INT32)(audio.freq/11025)); //Alam: to keep it around the same XX ms
CONS_Printf (M_GetText(" requested frequency of %d hz\n"), audio.freq);
}
else
{
audio.samples = samplecount;
audio.freq = cv_samplerate.value;
}
if (M_CheckParm ("-mono"))
{
audio.channels = 1;
audio.samples /= 2;
}
if (sound_disabled)
return;
#ifdef HW3SOUND
#ifdef STATIC3DS
if (M_CheckParm("-3dsound") || M_CheckParm("-ds3d"))
{
hws_mode = HWS_OPENAL;
}
#elif defined (_WIN32)
if (M_CheckParm("-ds3d"))
{
hws_mode = HWS_DS3D;
sdrv_name = "s_ds3d.dll";
}
else if (M_CheckParm("-fmod3d"))
{
hws_mode = HWS_FMOD3D;
sdrv_name = "s_fmod.dll";
}
else if (M_CheckParm("-openal"))
{
hws_mode = HWS_OPENAL;
sdrv_name = "s_openal.dll";
}
#else
if (M_CheckParm("-fmod3d"))
{
hws_mode = HWS_FMOD3D;
sdrv_name = "./s_fmod.so";
}
else if (M_CheckParm("-openal"))
{
hws_mode = HWS_OPENAL;
sdrv_name = "./s_openal.so";
}
#endif
else if (M_CheckParm("-sounddriver") && M_IsNextParm())
{
hws_mode = HWS_OTHER;
sdrv_name = M_GetNextParm();
}
if (hws_mode != HWS_DEFAULT_MODE)
{
if (Init3DSDriver(sdrv_name))
{
snddev_t snddev;
//sound_disabled = true;
//I_AddExitFunc(I_ShutdownSound);
snddev.bps = 16;
snddev.sample_rate = audio.freq;
snddev.numsfxs = NUMSFX;
#if defined (_WIN32)
snddev.cooplevel = 0x00000002;
snddev.hWnd = vid.WndParent;
#endif
if (HW3S_Init(I_Error, &snddev))
{
audio.userdata = NULL;
CONS_Printf("%s", M_GetText(" Using 3D sound driver\n"));
return;
}
CONS_Printf("%s", M_GetText(" Failed loading 3D sound driver\n"));
// Falls back to default sound system
HW3S_Shutdown();
hwClose(soundso);
}
CONS_Printf("%s", M_GetText(" Failed loading 3D sound driver\n"));
hws_mode = HWS_DEFAULT_MODE;
}
#endif
if (!musicStarted && SDL_OpenAudio(&audio, &audio) < 0)
{
CONS_Printf("%s", M_GetText(" couldn't open audio with desired format\n"));
sound_disabled = true;
return;
}
else
{
//char ad[100];
//CONS_Printf(M_GetText(" Starting up with audio driver : %s\n"), SDL_AudioDriverName(ad, (int)sizeof ad));
}
samplecount = audio.samples;
CV_SetValue(&cv_samplerate, audio.freq);
CONS_Printf(M_GetText(" configured audio device with %d samples/slice at %ikhz(%dms buffer)\n"), samplecount, audio.freq/1000, (INT32) (((float)audio.samples * 1000.0f) / audio.freq));
// Finished initialization.
CONS_Printf("%s", M_GetText(" Sound module ready\n"));
//[segabor]
if (!musicStarted) SDL_PauseAudio(0);
//Mix_Pause(0);
I_SetChannels();
sound_started = true;
Snd_Mutex = SDL_CreateMutex();
}
//
// MUSIC API.
//
/// ------------------------
// MUSIC SYSTEM
/// ------------------------
#if 0 //#ifdef HAVE_LIBGME
static void I_ShutdownGMEMusic(void)
{
Snd_LockAudio();
if (localdata.gme_emu)
gme_delete(localdata.gme_emu);
localdata.gme_emu = NULL;
Snd_UnlockAudio();
}
#endif
void I_InitMusic(void)
{
#if 0 //#ifdef HAVE_LIBGME
I_AddExitFunc(I_ShutdownGMEMusic);
#endif
}
void I_ShutdownMusic(void) { }
/// ------------------------
// MUSIC PROPERTIES
/// ------------------------
musictype_t I_SongType(void)
{
return MU_NONE;
}
boolean I_SongPlaying(void)
{
return false;
}
boolean I_SongPaused(void)
{
return false;
}
/// ------------------------
// MUSIC EFFECTS
/// ------------------------
boolean I_SetSongSpeed(float speed)
{
(void)speed;
return false;
}
/// ------------------------
// MUSIC SEEKING
/// ------------------------
UINT32 I_GetSongLength(void)
{
return 0;
}
boolean I_SetSongLoopPoint(UINT32 looppoint)
{
(void)looppoint;
return false;
}
UINT32 I_GetSongLoopPoint(void)
{
return 0;
}
boolean I_SetSongPosition(UINT32 position)
{
(void)position;
return false;
}
UINT32 I_GetSongPosition(void)
{
return 0;
}
/// ------------------------
// MUSIC PLAYBACK
/// ------------------------
#if 0 //#ifdef HAVE_LIBGME
static void I_StopGME(void)
{
Snd_LockAudio();
gme_seek(localdata.gme_emu, 0);
Snd_UnlockAudio();
}
static void I_PauseGME(void)
{
localdata.gme_pause = true;
}
static void I_ResumeGME(void)
{
localdata.gme_pause = false;
}
#endif
boolean I_LoadSong(char *data, size_t len)
{
(void)data;
(void)len;
return false;
}
void I_UnloadSong(void) { }
boolean I_PlaySong(boolean looping)
{
(void)looping;
return false;
}
void I_StopSong(void)
{
#if 0 //#ifdef HAVE_LIBGME
I_StopGME();
#endif
}
void I_PauseSong(void)
{
#if 0 //#ifdef HAVE_LIBGME
I_PauseGME();
#endif
}
void I_ResumeSong(void)
{
#if 0
I_ResumeGME();
#endif
}
void I_SetMusicVolume(UINT8 volume)
{
(void)volume;
}
boolean I_SetSongTrack(int track)
{
(void)track;
return false;
}
/// ------------------------
/// MUSIC FADING
/// ------------------------
void I_SetInternalMusicVolume(UINT8 volume)
{
(void)volume;
}
void I_StopFadingSong(void)
{
}
boolean I_FadeSongFromVolume(UINT8 target_volume, UINT8 source_volume, UINT32 ms, void (*callback)(void))
{
(void)target_volume;
(void)source_volume;
(void)ms;
(void)callback;
return false;
}
boolean I_FadeSong(UINT8 target_volume, UINT32 ms, void (*callback)(void))
{
(void)target_volume;
(void)ms;
(void)callback;
return false;
}
boolean I_FadeOutStopSong(UINT32 ms)
{
(void)ms;
return false;
}
boolean I_FadeInPlaySong(UINT32 ms, boolean looping)
{
(void)ms;
(void)looping;
return false;
}
/// ------------------------
// MUSIC LOADING AND CLEANUP
// \todo Split logic between loading and playing,
// then move to Playback section
/// ------------------------
#if 0 //#ifdef HAVE_LIBGME
static void I_CleanupGME(void *userdata)
{
Z_Free(userdata);
}
static boolean I_StartGMESong(const char *musicname, boolean looping)
{
char filename[9];
void *data;
lumpnum_t lumpnum;
size_t lumplength;
Music_Emu *emu;
const char* gme_err;
Snd_LockAudio();
if (localdata.gme_emu)
gme_delete(localdata.gme_emu);
localdata.gme_emu = NULL;
Snd_UnlockAudio();
snprintf(filename, sizeof filename, "o_%s", musicname);
lumpnum = W_CheckNumForName(filename);
if (lumpnum == LUMPERROR)
{
return false; // No music found. Oh well!
}
else
lumplength = W_LumpLength(lumpnum);
data = W_CacheLumpNum(lumpnum, PU_MUSIC);
gme_err = gme_open_data(data, (long)lumplength, &emu, audio.freq);
if (gme_err != NULL) {
//I_OutputMsg("I_StartGMESong: error %s\n",gme_err);
return false;
}
gme_set_user_data(emu, data);
gme_set_user_cleanup(emu, I_CleanupGME);
gme_start_track(emu, 0);
#ifdef HAVE_MIXER
gme_set_fade(emu, Digfade);
#endif
Snd_LockAudio();
localdata.gme_emu = emu;
localdata.gme_pause = false;
localdata.gme_loop = (UINT8)looping;
Snd_UnlockAudio();
return true;
}
#endif
#endif //HAVE_SDL
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