quakespasm/Quake/snd_dma.c

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/*
Copyright (C) 1996-2001 Id Software, Inc.
Copyright (C) 2002-2009 John Fitzgibbons and others
Copyright (C) 2007-2008 Kristian Duske
Copyright (C) 2010-2011 O. Sezer <sezero@users.sourceforge.net>
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.
This program 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.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
// snd_dma.c -- main control for any streaming sound output device
#include "quakedef.h"
Backported external music files support using decoder libraries and the new raw samples interface from Hammer of Thyrion (uhexen2) : - bgmusic.c, bgmusic.h: New BGM interface for background music handling. Handles streaming music as raw sound samples. - bgmnull.c: BGM source for cases where the engine is configured for no sound. - cl_main.c: Include bgmusic.h. Call BGM_Stop() and CDAudio_Stop() in CL_Disconnect(). - cd_sdl.c: Moved bgmvolume boundary checking to bgmusic.c upon value changes. - gl_vidnt.c, gl_vidsdl.c, cl_parse.c: Include bgmusic.h. Add BGM_Pause() and BGM_Resume() calls along with CDAudio_ counterparts. - cl_parse.c: Replace CDAudio_Play() call by the new BGM_PlayCDtrack() which first tries CDAudio_Play() and then streaming music if it fails. - host.c: Include bgmusic.h. Call BGM_Update() just before S_Update() in Host_Frame(). In Host_Init(), call BGM_Init() after other audio init calls. In Host_Shutdown(), call BGM_Shutdown() before all other audio shutdown calls. - snd_dma.c: Include snd_codec.h and bgmusic.h. Call S_CodecInit() from S_Init(). Call S_CodecShutdown() from S_Shutdown(). - snd_codec.c, snd_codec.h: New public codec interface for streaming music as raw samples. Adapted from quake2 and ioquake3 with changes. Individual codecs are responsible for handling any necessary byte swap operations. - snd_codeci.h: New header for snd_codec internals. - snd_wave.c, snd_wave.h: Codec for WAV format streaming music. Adapted from ioquake3 with changes. - snd_vorbis.c, snd_vorbis.h: Codec for Ogg/Vorbis format streaming music. - snd_mp3.c, snd_mp3.h: Codec for MP3 format streaming music using libmad. Adapted from the SoX project with changes. - Makefile: Adjusted for the new sources. Added switches USE_CODEC_WAVE, USE_CODEC_MP3, USE_CODEC_VORBIS for enabling and disabling individual codecs. - Windows makefiles and project files as well as other CodeBlocks project files will be updated shortly. git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@374 af15c1b1-3010-417e-b628-4374ebc0bcbd
2011-01-05 19:50:43 +00:00
#include "snd_codec.h"
#include "bgmusic.h"
static void S_Play (void);
static void S_PlayVol (void);
static void S_SoundList (void);
static void S_Update_ (void);
void S_StopAllSounds (qboolean clear);
static void S_StopAllSoundsC (void);
// =======================================================================
// Internal sound data & structures
// =======================================================================
channel_t snd_channels[MAX_CHANNELS];
int total_channels;
static int snd_blocked = 0;
static qboolean snd_initialized = false;
static dma_t sn;
volatile dma_t *shm = NULL;
vec3_t listener_origin;
vec3_t listener_forward;
vec3_t listener_right;
vec3_t listener_up;
#define sound_nominal_clip_dist 1000.0
int soundtime; // sample PAIRS
int paintedtime; // sample PAIRS
backports from uhexen2 source, preparing for streaming music support: * snd_mix.c: Increased PAINTBUFFER_SIZE from 512 to 2048. * snd_mix.c: snd_vol is static now. it is calculated in S_PaintChannels and only used in SND_PaintChannelFrom16. all its other uses are removed from Snd_WriteLinearBlastStereo16, S_TransferStereo16, S_TransferPaintBuffer. The way it was, the sound volume was applied to the whole final contents of the paint buffer, but with this new quake2+ way we can add raw samples to the paint buffer with its own volume, such as bgmvolume. However, this makes the snd_scaletable to be recalculated everytime the sfxvolume is, changed, therefore it is adjusted that way to incorporate sfxvolume. * snd_mix.c: In S_PaintChannels, check against s_rawend and copy from the streaming sound source if necessary. * snd_dma.c: Added old_volume to detect sfxvolume changes. Made S_Update to compare it to sfxvolume.value and call SND_InitScaletable() if it changed. * snd_dma.c: Add new globals s_rawsamples and s_rawend. Reset s_rawend to 0 in S_ClearBuffer. Add new function S_RawSamples, adapted from quake2 with its 8 bit stereo playback fixed. * snd_dma.c (S_FileExtension): Add new function which returns the given sound file's extension including the dot, or NULL. * q_sound.h: Add new macro MAX_RAW_SAMPLES, defined as 8192. Add externs for new globals s_rawsamples and s_rawend. Add prototype for the new S_RawSamples and S_FileExtension functions. git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@355 af15c1b1-3010-417e-b628-4374ebc0bcbd
2010-12-30 17:11:28 +00:00
int s_rawend;
portable_samplepair_t s_rawsamples[MAX_RAW_SAMPLES];
#define MAX_SFX 512
static sfx_t *known_sfx = NULL; // hunk allocated [MAX_SFX]
static int num_sfx;
static sfx_t *ambient_sfx[NUM_AMBIENTS];
static qboolean sound_started = false;
cvar_t bgmvolume = {"bgmvolume", "1", CVAR_ARCHIVE};
cvar_t sfxvolume = {"volume", "0.7", CVAR_ARCHIVE};
backports from uhexen2 source, preparing for streaming music support: * snd_mix.c: Increased PAINTBUFFER_SIZE from 512 to 2048. * snd_mix.c: snd_vol is static now. it is calculated in S_PaintChannels and only used in SND_PaintChannelFrom16. all its other uses are removed from Snd_WriteLinearBlastStereo16, S_TransferStereo16, S_TransferPaintBuffer. The way it was, the sound volume was applied to the whole final contents of the paint buffer, but with this new quake2+ way we can add raw samples to the paint buffer with its own volume, such as bgmvolume. However, this makes the snd_scaletable to be recalculated everytime the sfxvolume is, changed, therefore it is adjusted that way to incorporate sfxvolume. * snd_mix.c: In S_PaintChannels, check against s_rawend and copy from the streaming sound source if necessary. * snd_dma.c: Added old_volume to detect sfxvolume changes. Made S_Update to compare it to sfxvolume.value and call SND_InitScaletable() if it changed. * snd_dma.c: Add new globals s_rawsamples and s_rawend. Reset s_rawend to 0 in S_ClearBuffer. Add new function S_RawSamples, adapted from quake2 with its 8 bit stereo playback fixed. * snd_dma.c (S_FileExtension): Add new function which returns the given sound file's extension including the dot, or NULL. * q_sound.h: Add new macro MAX_RAW_SAMPLES, defined as 8192. Add externs for new globals s_rawsamples and s_rawend. Add prototype for the new S_RawSamples and S_FileExtension functions. git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@355 af15c1b1-3010-417e-b628-4374ebc0bcbd
2010-12-30 17:11:28 +00:00
cvar_t precache = {"precache", "1", CVAR_NONE};
cvar_t loadas8bit = {"loadas8bit", "0", CVAR_NONE};
cvar_t sndspeed = {"sndspeed", "11025", CVAR_NONE};
static cvar_t nosound = {"nosound", "0", CVAR_NONE};
static cvar_t ambient_level = {"ambient_level", "0.3", CVAR_NONE};
static cvar_t ambient_fade = {"ambient_fade", "100", CVAR_NONE};
static cvar_t snd_noextraupdate = {"snd_noextraupdate", "0", CVAR_NONE};
static cvar_t snd_show = {"snd_show", "0", CVAR_NONE};
static cvar_t _snd_mixahead = {"_snd_mixahead", "0.1", CVAR_ARCHIVE};
static void S_SoundInfo_f (void)
{
if (!sound_started || !shm)
{
Con_Printf ("sound system not started\n");
return;
}
Con_Printf("%d bit, %s, %d Hz\n", shm->samplebits,
(shm->channels == 2) ? "stereo" : "mono", shm->speed);
Con_Printf("%5d samples\n", shm->samples);
Con_Printf("%5d samplepos\n", shm->samplepos);
Con_Printf("%5d submission_chunk\n", shm->submission_chunk);
Con_Printf("%5d total_channels\n", total_channels);
Con_Printf("%p dma buffer\n", shm->buffer);
}
static void SND_Callback_sfxvolume (cvar_t *var)
{
SND_InitScaletable ();
}
/*
================
S_Startup
================
*/
void S_Startup (void)
{
if (!snd_initialized)
return;
sound_started = SNDDMA_Init(&sn);
if (!sound_started)
{
Con_Printf("Failed initializing sound\n");
}
else
{
Con_Printf("Audio: %d bit, %s, %d Hz\n",
shm->samplebits,
(shm->channels == 2) ? "stereo" : "mono",
shm->speed);
}
}
/*
================
S_Init
================
*/
void S_Init (void)
{
int i;
if (snd_initialized)
{
Con_Printf("Sound is already initialized\n");
return;
}
Cvar_RegisterVariable(&nosound);
Cvar_RegisterVariable(&sfxvolume);
Cvar_RegisterVariable(&precache);
Cvar_RegisterVariable(&loadas8bit);
Cvar_RegisterVariable(&bgmvolume);
Cvar_RegisterVariable(&ambient_level);
Cvar_RegisterVariable(&ambient_fade);
Cvar_RegisterVariable(&snd_noextraupdate);
Cvar_RegisterVariable(&snd_show);
Cvar_RegisterVariable(&_snd_mixahead);
Cvar_RegisterVariable(&sndspeed);
if (safemode || COM_CheckParm("-nosound"))
return;
Con_Printf("\nSound Initialization\n");
Cmd_AddCommand("play", S_Play);
Cmd_AddCommand("playvol", S_PlayVol);
Cmd_AddCommand("stopsound", S_StopAllSoundsC);
Cmd_AddCommand("soundlist", S_SoundList);
Cmd_AddCommand("soundinfo", S_SoundInfo_f);
i = COM_CheckParm("-sndspeed");
if (i && i < com_argc-1)
{
Cvar_Set ("sndspeed", com_argv[i+1]);
}
if (host_parms->memsize < 0x800000)
{
Cvar_Set ("loadas8bit", "1");
Con_Printf ("loading all sounds as 8bit\n");
}
Cvar_SetCallback(&sfxvolume, SND_Callback_sfxvolume);
SND_InitScaletable ();
known_sfx = (sfx_t *) Hunk_AllocName (MAX_SFX*sizeof(sfx_t), "sfx_t");
num_sfx = 0;
snd_initialized = true;
S_Startup ();
if (sound_started == 0)
return;
// provides a tick sound until washed clean
// if (shm->buffer)
// shm->buffer[4] = shm->buffer[5] = 0x7f; // force a pop for debugging
ambient_sfx[AMBIENT_WATER] = S_PrecacheSound ("ambience/water1.wav");
ambient_sfx[AMBIENT_SKY] = S_PrecacheSound ("ambience/wind2.wav");
Backported external music files support using decoder libraries and the new raw samples interface from Hammer of Thyrion (uhexen2) : - bgmusic.c, bgmusic.h: New BGM interface for background music handling. Handles streaming music as raw sound samples. - bgmnull.c: BGM source for cases where the engine is configured for no sound. - cl_main.c: Include bgmusic.h. Call BGM_Stop() and CDAudio_Stop() in CL_Disconnect(). - cd_sdl.c: Moved bgmvolume boundary checking to bgmusic.c upon value changes. - gl_vidnt.c, gl_vidsdl.c, cl_parse.c: Include bgmusic.h. Add BGM_Pause() and BGM_Resume() calls along with CDAudio_ counterparts. - cl_parse.c: Replace CDAudio_Play() call by the new BGM_PlayCDtrack() which first tries CDAudio_Play() and then streaming music if it fails. - host.c: Include bgmusic.h. Call BGM_Update() just before S_Update() in Host_Frame(). In Host_Init(), call BGM_Init() after other audio init calls. In Host_Shutdown(), call BGM_Shutdown() before all other audio shutdown calls. - snd_dma.c: Include snd_codec.h and bgmusic.h. Call S_CodecInit() from S_Init(). Call S_CodecShutdown() from S_Shutdown(). - snd_codec.c, snd_codec.h: New public codec interface for streaming music as raw samples. Adapted from quake2 and ioquake3 with changes. Individual codecs are responsible for handling any necessary byte swap operations. - snd_codeci.h: New header for snd_codec internals. - snd_wave.c, snd_wave.h: Codec for WAV format streaming music. Adapted from ioquake3 with changes. - snd_vorbis.c, snd_vorbis.h: Codec for Ogg/Vorbis format streaming music. - snd_mp3.c, snd_mp3.h: Codec for MP3 format streaming music using libmad. Adapted from the SoX project with changes. - Makefile: Adjusted for the new sources. Added switches USE_CODEC_WAVE, USE_CODEC_MP3, USE_CODEC_VORBIS for enabling and disabling individual codecs. - Windows makefiles and project files as well as other CodeBlocks project files will be updated shortly. git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@374 af15c1b1-3010-417e-b628-4374ebc0bcbd
2011-01-05 19:50:43 +00:00
S_CodecInit ();
S_StopAllSounds (true);
}
// =======================================================================
// Shutdown sound engine
// =======================================================================
void S_Shutdown (void)
{
if (!sound_started)
return;
sound_started = 0;
snd_blocked = 0;
Backported external music files support using decoder libraries and the new raw samples interface from Hammer of Thyrion (uhexen2) : - bgmusic.c, bgmusic.h: New BGM interface for background music handling. Handles streaming music as raw sound samples. - bgmnull.c: BGM source for cases where the engine is configured for no sound. - cl_main.c: Include bgmusic.h. Call BGM_Stop() and CDAudio_Stop() in CL_Disconnect(). - cd_sdl.c: Moved bgmvolume boundary checking to bgmusic.c upon value changes. - gl_vidnt.c, gl_vidsdl.c, cl_parse.c: Include bgmusic.h. Add BGM_Pause() and BGM_Resume() calls along with CDAudio_ counterparts. - cl_parse.c: Replace CDAudio_Play() call by the new BGM_PlayCDtrack() which first tries CDAudio_Play() and then streaming music if it fails. - host.c: Include bgmusic.h. Call BGM_Update() just before S_Update() in Host_Frame(). In Host_Init(), call BGM_Init() after other audio init calls. In Host_Shutdown(), call BGM_Shutdown() before all other audio shutdown calls. - snd_dma.c: Include snd_codec.h and bgmusic.h. Call S_CodecInit() from S_Init(). Call S_CodecShutdown() from S_Shutdown(). - snd_codec.c, snd_codec.h: New public codec interface for streaming music as raw samples. Adapted from quake2 and ioquake3 with changes. Individual codecs are responsible for handling any necessary byte swap operations. - snd_codeci.h: New header for snd_codec internals. - snd_wave.c, snd_wave.h: Codec for WAV format streaming music. Adapted from ioquake3 with changes. - snd_vorbis.c, snd_vorbis.h: Codec for Ogg/Vorbis format streaming music. - snd_mp3.c, snd_mp3.h: Codec for MP3 format streaming music using libmad. Adapted from the SoX project with changes. - Makefile: Adjusted for the new sources. Added switches USE_CODEC_WAVE, USE_CODEC_MP3, USE_CODEC_VORBIS for enabling and disabling individual codecs. - Windows makefiles and project files as well as other CodeBlocks project files will be updated shortly. git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@374 af15c1b1-3010-417e-b628-4374ebc0bcbd
2011-01-05 19:50:43 +00:00
S_CodecShutdown();
SNDDMA_Shutdown();
shm = NULL;
}
// =======================================================================
// Load a sound
// =======================================================================
/*
==================
S_FindName
==================
*/
static sfx_t *S_FindName (const char *name)
{
int i;
sfx_t *sfx;
if (!name)
Sys_Error ("S_FindName: NULL");
if (Q_strlen(name) >= MAX_QPATH)
Sys_Error ("Sound name too long: %s", name);
// see if already loaded
for (i = 0; i < num_sfx; i++)
{
if (!Q_strcmp(known_sfx[i].name, name))
{
return &known_sfx[i];
}
}
if (num_sfx == MAX_SFX)
Sys_Error ("S_FindName: out of sfx_t");
sfx = &known_sfx[i];
q_strlcpy (sfx->name, name, sizeof(sfx->name));
num_sfx++;
return sfx;
}
/*
==================
S_TouchSound
==================
*/
void S_TouchSound (const char *name)
{
sfx_t *sfx;
if (!sound_started)
return;
sfx = S_FindName (name);
Cache_Check (&sfx->cache);
}
/*
==================
S_PrecacheSound
==================
*/
sfx_t *S_PrecacheSound (const char *name)
{
sfx_t *sfx;
if (!sound_started || nosound.value)
return NULL;
sfx = S_FindName (name);
// cache it in
if (precache.value)
S_LoadSound (sfx);
return sfx;
}
//=============================================================================
/*
=================
SND_PickChannel
picks a channel based on priorities, empty slots, number of channels
=================
*/
channel_t *SND_PickChannel (int entnum, int entchannel)
{
int ch_idx;
int first_to_die;
int life_left;
// Check for replacement sound, or find the best one to replace
first_to_die = -1;
life_left = 0x7fffffff;
for (ch_idx = NUM_AMBIENTS; ch_idx < NUM_AMBIENTS + MAX_DYNAMIC_CHANNELS; ch_idx++)
{
if (entchannel != 0 // channel 0 never overrides
&& snd_channels[ch_idx].entnum == entnum
&& (snd_channels[ch_idx].entchannel == entchannel || entchannel == -1) )
{ // always override sound from same entity
first_to_die = ch_idx;
break;
}
// don't let monster sounds override player sounds
if (snd_channels[ch_idx].entnum == cl.viewentity && entnum != cl.viewentity && snd_channels[ch_idx].sfx)
continue;
if (snd_channels[ch_idx].end - paintedtime < life_left)
{
life_left = snd_channels[ch_idx].end - paintedtime;
first_to_die = ch_idx;
}
}
if (first_to_die == -1)
return NULL;
if (snd_channels[first_to_die].sfx)
snd_channels[first_to_die].sfx = NULL;
return &snd_channels[first_to_die];
}
/*
=================
SND_Spatialize
spatializes a channel
=================
*/
void SND_Spatialize (channel_t *ch)
{
vec_t dot;
vec_t dist;
vec_t lscale, rscale, scale;
vec3_t source_vec;
// anything coming from the view entity will always be full volume
if (ch->entnum == cl.viewentity)
{
ch->leftvol = ch->master_vol;
ch->rightvol = ch->master_vol;
return;
}
// calculate stereo seperation and distance attenuation
VectorSubtract(ch->origin, listener_origin, source_vec);
dist = VectorNormalize(source_vec) * ch->dist_mult;
dot = DotProduct(listener_right, source_vec);
if (shm->channels == 1)
{
rscale = 1.0;
lscale = 1.0;
}
else
{
rscale = 1.0 + dot;
lscale = 1.0 - dot;
}
// add in distance effect
scale = (1.0 - dist) * rscale;
ch->rightvol = (int) (ch->master_vol * scale);
if (ch->rightvol < 0)
ch->rightvol = 0;
scale = (1.0 - dist) * lscale;
ch->leftvol = (int) (ch->master_vol * scale);
if (ch->leftvol < 0)
ch->leftvol = 0;
}
// =======================================================================
// Start a sound effect
// =======================================================================
void S_StartSound (int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float fvol, float attenuation)
{
channel_t *target_chan, *check;
sfxcache_t *sc;
int ch_idx;
int skip;
if (!sound_started)
return;
if (!sfx)
return;
if (nosound.value)
return;
// pick a channel to play on
target_chan = SND_PickChannel(entnum, entchannel);
if (!target_chan)
return;
// spatialize
memset (target_chan, 0, sizeof(*target_chan));
VectorCopy(origin, target_chan->origin);
target_chan->dist_mult = attenuation / sound_nominal_clip_dist;
target_chan->master_vol = (int) (fvol * 255);
target_chan->entnum = entnum;
target_chan->entchannel = entchannel;
SND_Spatialize(target_chan);
if (!target_chan->leftvol && !target_chan->rightvol)
return; // not audible at all
// new channel
sc = S_LoadSound (sfx);
if (!sc)
{
target_chan->sfx = NULL;
return; // couldn't load the sound's data
}
target_chan->sfx = sfx;
target_chan->pos = 0.0;
target_chan->end = paintedtime + sc->length;
// if an identical sound has also been started this frame, offset the pos
// a bit to keep it from just making the first one louder
check = &snd_channels[NUM_AMBIENTS];
for (ch_idx = NUM_AMBIENTS; ch_idx < NUM_AMBIENTS + MAX_DYNAMIC_CHANNELS; ch_idx++, check++)
{
if (check == target_chan)
continue;
if (check->sfx == sfx && !check->pos)
{
/*
skip = rand () % (int)(0.1 * shm->speed);
if (skip >= target_chan->end)
skip = target_chan->end - 1;
*/
/* LordHavoc: fixed skip calculations */
skip = 0.1 * shm->speed; /* 0.1 * sc->speed */
if (skip > sc->length)
skip = sc->length;
if (skip > 0)
skip = rand() % skip;
target_chan->pos += skip;
target_chan->end -= skip;
break;
}
}
}
void S_StopSound (int entnum, int entchannel)
{
int i;
for (i = 0; i < MAX_DYNAMIC_CHANNELS; i++)
{
if (snd_channels[i].entnum == entnum
&& snd_channels[i].entchannel == entchannel)
{
snd_channels[i].end = 0;
snd_channels[i].sfx = NULL;
return;
}
}
}
void S_StopAllSounds (qboolean clear)
{
int i;
if (!sound_started)
return;
total_channels = MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS; // no statics
for (i = 0; i < MAX_CHANNELS; i++)
{
if (snd_channels[i].sfx)
snd_channels[i].sfx = NULL;
}
memset(snd_channels, 0, MAX_CHANNELS * sizeof(channel_t));
if (clear)
S_ClearBuffer ();
}
static void S_StopAllSoundsC (void)
{
S_StopAllSounds (true);
}
void S_ClearBuffer (void)
{
int clear;
if (!sound_started || !shm)
return;
SNDDMA_LockBuffer ();
if (! shm->buffer)
return;
backports from uhexen2 source, preparing for streaming music support: * snd_mix.c: Increased PAINTBUFFER_SIZE from 512 to 2048. * snd_mix.c: snd_vol is static now. it is calculated in S_PaintChannels and only used in SND_PaintChannelFrom16. all its other uses are removed from Snd_WriteLinearBlastStereo16, S_TransferStereo16, S_TransferPaintBuffer. The way it was, the sound volume was applied to the whole final contents of the paint buffer, but with this new quake2+ way we can add raw samples to the paint buffer with its own volume, such as bgmvolume. However, this makes the snd_scaletable to be recalculated everytime the sfxvolume is, changed, therefore it is adjusted that way to incorporate sfxvolume. * snd_mix.c: In S_PaintChannels, check against s_rawend and copy from the streaming sound source if necessary. * snd_dma.c: Added old_volume to detect sfxvolume changes. Made S_Update to compare it to sfxvolume.value and call SND_InitScaletable() if it changed. * snd_dma.c: Add new globals s_rawsamples and s_rawend. Reset s_rawend to 0 in S_ClearBuffer. Add new function S_RawSamples, adapted from quake2 with its 8 bit stereo playback fixed. * snd_dma.c (S_FileExtension): Add new function which returns the given sound file's extension including the dot, or NULL. * q_sound.h: Add new macro MAX_RAW_SAMPLES, defined as 8192. Add externs for new globals s_rawsamples and s_rawend. Add prototype for the new S_RawSamples and S_FileExtension functions. git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@355 af15c1b1-3010-417e-b628-4374ebc0bcbd
2010-12-30 17:11:28 +00:00
s_rawend = 0;
if (shm->samplebits == 8)
clear = 0x80;
else
clear = 0;
memset(shm->buffer, clear, shm->samples * shm->samplebits / 8);
SNDDMA_Submit ();
}
/*
=================
S_StaticSound
=================
*/
void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation)
{
channel_t *ss;
sfxcache_t *sc;
if (!sfx)
return;
if (total_channels == MAX_CHANNELS)
{
Con_Printf ("total_channels == MAX_CHANNELS\n");
return;
}
ss = &snd_channels[total_channels];
total_channels++;
sc = S_LoadSound (sfx);
if (!sc)
return;
if (sc->loopstart == -1)
{
Con_Printf ("Sound %s not looped\n", sfx->name);
return;
}
ss->sfx = sfx;
VectorCopy (origin, ss->origin);
ss->master_vol = (int)vol;
ss->dist_mult = (attenuation / 64) / sound_nominal_clip_dist;
ss->end = paintedtime + sc->length;
SND_Spatialize (ss);
}
//=============================================================================
/*
===================
S_UpdateAmbientSounds
===================
*/
static void S_UpdateAmbientSounds (void)
{
mleaf_t *l;
int vol, ambient_channel;
channel_t *chan;
// no ambients when disconnected
if (cls.state != ca_connected)
return;
// calc ambient sound levels
if (!cl.worldmodel)
return;
l = Mod_PointInLeaf (listener_origin, cl.worldmodel);
if (!l || !ambient_level.value)
{
for (ambient_channel = 0; ambient_channel < NUM_AMBIENTS; ambient_channel++)
snd_channels[ambient_channel].sfx = NULL;
return;
}
for (ambient_channel = 0; ambient_channel < NUM_AMBIENTS; ambient_channel++)
{
chan = &snd_channels[ambient_channel];
chan->sfx = ambient_sfx[ambient_channel];
vol = (int) (ambient_level.value * l->ambient_sound_level[ambient_channel]);
if (vol < 8)
vol = 0;
// don't adjust volume too fast
if (chan->master_vol < vol)
{
chan->master_vol += (int) (host_frametime * ambient_fade.value);
if (chan->master_vol > vol)
chan->master_vol = vol;
}
else if (chan->master_vol > vol)
{
chan->master_vol -= (int) (host_frametime * ambient_fade.value);
if (chan->master_vol < vol)
chan->master_vol = vol;
}
chan->leftvol = chan->rightvol = chan->master_vol;
}
}
backports from uhexen2 source, preparing for streaming music support: * snd_mix.c: Increased PAINTBUFFER_SIZE from 512 to 2048. * snd_mix.c: snd_vol is static now. it is calculated in S_PaintChannels and only used in SND_PaintChannelFrom16. all its other uses are removed from Snd_WriteLinearBlastStereo16, S_TransferStereo16, S_TransferPaintBuffer. The way it was, the sound volume was applied to the whole final contents of the paint buffer, but with this new quake2+ way we can add raw samples to the paint buffer with its own volume, such as bgmvolume. However, this makes the snd_scaletable to be recalculated everytime the sfxvolume is, changed, therefore it is adjusted that way to incorporate sfxvolume. * snd_mix.c: In S_PaintChannels, check against s_rawend and copy from the streaming sound source if necessary. * snd_dma.c: Added old_volume to detect sfxvolume changes. Made S_Update to compare it to sfxvolume.value and call SND_InitScaletable() if it changed. * snd_dma.c: Add new globals s_rawsamples and s_rawend. Reset s_rawend to 0 in S_ClearBuffer. Add new function S_RawSamples, adapted from quake2 with its 8 bit stereo playback fixed. * snd_dma.c (S_FileExtension): Add new function which returns the given sound file's extension including the dot, or NULL. * q_sound.h: Add new macro MAX_RAW_SAMPLES, defined as 8192. Add externs for new globals s_rawsamples and s_rawend. Add prototype for the new S_RawSamples and S_FileExtension functions. git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@355 af15c1b1-3010-417e-b628-4374ebc0bcbd
2010-12-30 17:11:28 +00:00
/*
===================
S_RawSamples (from QuakeII)
Streaming music support. Byte swapping
of data must be handled by the codec.
===================
*/
void S_RawSamples (int samples, int rate, int width, int channels, byte *data, float volume)
{
int i;
int src, dst;
float scale;
int intVolume;
if (s_rawend < paintedtime)
{
s_rawend = paintedtime;
}
scale = (float) rate / shm->speed;
intVolume = (int) (256 * volume);
if (channels == 2 && width == 2)
{
for (i = 0; ; i++)
{
src = i * scale;
if (src >= samples)
{
break;
}
dst = s_rawend & (MAX_RAW_SAMPLES - 1);
s_rawend++;
s_rawsamples [dst].left = ((short *) data)[src * 2] * intVolume;
s_rawsamples [dst].right = ((short *) data)[src * 2 + 1] * intVolume;
}
}
else if (channels == 1 && width == 2)
{
for (i = 0; ; i++)
{
src = i * scale;
if (src >= samples)
{
break;
}
dst = s_rawend & (MAX_RAW_SAMPLES - 1);
s_rawend++;
s_rawsamples [dst].left = ((short *) data)[src] * intVolume;
s_rawsamples [dst].right = ((short *) data)[src] * intVolume;
}
}
else if (channels == 2 && width == 1)
{
intVolume *= 256;
for (i = 0; ; i++)
{
src = i * scale;
if (src >= samples)
{
break;
}
dst = s_rawend & (MAX_RAW_SAMPLES - 1);
s_rawend++;
// s_rawsamples [dst].left = ((char *) data)[src * 2] * intVolume;
// s_rawsamples [dst].right = ((char *) data)[src * 2 + 1] * intVolume;
/* the above doesn't work for me with U8, only the unsigned ones below do */
s_rawsamples [dst].left = (((byte *) data)[src * 2] - 128) * intVolume;
s_rawsamples [dst].right = (((byte *) data)[src * 2 + 1] - 128) * intVolume;
}
}
else if (channels == 1 && width == 1)
{
intVolume *= 256;
for (i = 0; ; i++)
{
src = i * scale;
if (src >= samples)
{
break;
}
dst = s_rawend & (MAX_RAW_SAMPLES - 1);
s_rawend++;
s_rawsamples [dst].left = (((byte *) data)[src] - 128) * intVolume;
s_rawsamples [dst].right = (((byte *) data)[src] - 128) * intVolume;
}
}
}
/*
============
S_Update
Called once each time through the main loop
============
*/
void S_Update (vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
{
int i, j;
int total;
channel_t *ch;
channel_t *combine;
if (!sound_started || (snd_blocked > 0))
return;
VectorCopy(origin, listener_origin);
VectorCopy(forward, listener_forward);
VectorCopy(right, listener_right);
VectorCopy(up, listener_up);
// update general area ambient sound sources
S_UpdateAmbientSounds ();
combine = NULL;
// update spatialization for static and dynamic sounds
ch = snd_channels + NUM_AMBIENTS;
for (i = NUM_AMBIENTS; i < total_channels; i++, ch++)
{
if (!ch->sfx)
continue;
SND_Spatialize(ch); // respatialize channel
if (!ch->leftvol && !ch->rightvol)
continue;
// try to combine static sounds with a previous channel of the same
// sound effect so we don't mix five torches every frame
if (i >= MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS)
{
// see if it can just use the last one
if (combine && combine->sfx == ch->sfx)
{
combine->leftvol += ch->leftvol;
combine->rightvol += ch->rightvol;
ch->leftvol = ch->rightvol = 0;
continue;
}
// search for one
combine = snd_channels + MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS;
for (j = MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS; j < i; j++, combine++)
{
if (combine->sfx == ch->sfx)
break;
}
if (j == total_channels)
{
combine = NULL;
}
else
{
if (combine != ch)
{
combine->leftvol += ch->leftvol;
combine->rightvol += ch->rightvol;
ch->leftvol = ch->rightvol = 0;
}
continue;
}
}
}
//
// debugging output
//
if (snd_show.value)
{
total = 0;
ch = snd_channels;
for (i = 0; i < total_channels; i++, ch++)
{
if (ch->sfx && (ch->leftvol || ch->rightvol) )
{
// Con_Printf ("%3i %3i %s\n", ch->leftvol, ch->rightvol, ch->sfx->name);
total++;
}
}
Con_Printf ("----(%i)----\n", total);
}
Backported external music files support using decoder libraries and the new raw samples interface from Hammer of Thyrion (uhexen2) : - bgmusic.c, bgmusic.h: New BGM interface for background music handling. Handles streaming music as raw sound samples. - bgmnull.c: BGM source for cases where the engine is configured for no sound. - cl_main.c: Include bgmusic.h. Call BGM_Stop() and CDAudio_Stop() in CL_Disconnect(). - cd_sdl.c: Moved bgmvolume boundary checking to bgmusic.c upon value changes. - gl_vidnt.c, gl_vidsdl.c, cl_parse.c: Include bgmusic.h. Add BGM_Pause() and BGM_Resume() calls along with CDAudio_ counterparts. - cl_parse.c: Replace CDAudio_Play() call by the new BGM_PlayCDtrack() which first tries CDAudio_Play() and then streaming music if it fails. - host.c: Include bgmusic.h. Call BGM_Update() just before S_Update() in Host_Frame(). In Host_Init(), call BGM_Init() after other audio init calls. In Host_Shutdown(), call BGM_Shutdown() before all other audio shutdown calls. - snd_dma.c: Include snd_codec.h and bgmusic.h. Call S_CodecInit() from S_Init(). Call S_CodecShutdown() from S_Shutdown(). - snd_codec.c, snd_codec.h: New public codec interface for streaming music as raw samples. Adapted from quake2 and ioquake3 with changes. Individual codecs are responsible for handling any necessary byte swap operations. - snd_codeci.h: New header for snd_codec internals. - snd_wave.c, snd_wave.h: Codec for WAV format streaming music. Adapted from ioquake3 with changes. - snd_vorbis.c, snd_vorbis.h: Codec for Ogg/Vorbis format streaming music. - snd_mp3.c, snd_mp3.h: Codec for MP3 format streaming music using libmad. Adapted from the SoX project with changes. - Makefile: Adjusted for the new sources. Added switches USE_CODEC_WAVE, USE_CODEC_MP3, USE_CODEC_VORBIS for enabling and disabling individual codecs. - Windows makefiles and project files as well as other CodeBlocks project files will be updated shortly. git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@374 af15c1b1-3010-417e-b628-4374ebc0bcbd
2011-01-05 19:50:43 +00:00
// add raw data from streamed samples
// BGM_Update(); // moved to the main loop just before S_Update ()
// mix some sound
S_Update_();
}
static void GetSoundtime (void)
{
int samplepos;
static int buffers;
static int oldsamplepos;
int fullsamples;
fullsamples = shm->samples / shm->channels;
// it is possible to miscount buffers if it has wrapped twice between
// calls to S_Update. Oh well.
#ifdef __sun__
soundtime = SNDDMA_GetSamples();
#else
samplepos = SNDDMA_GetDMAPos();
if (samplepos < oldsamplepos)
{
buffers++; // buffer wrapped
if (paintedtime > 0x40000000)
{ // time to chop things off to avoid 32 bit limits
buffers = 0;
paintedtime = fullsamples;
S_StopAllSounds (true);
}
}
oldsamplepos = samplepos;
soundtime = buffers*fullsamples + samplepos/shm->channels;
#endif
}
void S_ExtraUpdate (void)
{
if (snd_noextraupdate.value)
return; // don't pollute timings
S_Update_();
}
static void S_Update_ (void)
{
unsigned int endtime;
int samps;
if (!sound_started || (snd_blocked > 0))
return;
SNDDMA_LockBuffer ();
if (! shm->buffer)
return;
// Updates DMA time
GetSoundtime();
// check to make sure that we haven't overshot
if (paintedtime < soundtime)
{
// Con_Printf ("S_Update_ : overflow\n");
paintedtime = soundtime;
}
// mix ahead of current position
endtime = soundtime + (unsigned int)(_snd_mixahead.value * shm->speed);
samps = shm->samples >> (shm->channels - 1);
endtime = q_min(endtime, (unsigned int)(soundtime + samps));
S_PaintChannels (endtime);
SNDDMA_Submit ();
}
void S_BlockSound (void)
{
/* FIXME: do we really need the blocking at the
* driver level?
*/
if (sound_started && snd_blocked == 0) /* ++snd_blocked == 1 */
{
snd_blocked = 1;
S_ClearBuffer ();
if (shm)
SNDDMA_BlockSound();
}
}
void S_UnblockSound (void)
{
if (!sound_started || !snd_blocked)
return;
if (snd_blocked == 1) /* --snd_blocked == 0 */
{
snd_blocked = 0;
SNDDMA_UnblockSound();
S_ClearBuffer ();
}
}
/*
===============================================================================
console functions
===============================================================================
*/
static void S_Play (void)
{
static int hash = 345;
int i;
char name[256];
sfx_t *sfx;
i = 1;
while (i < Cmd_Argc())
{
q_strlcpy(name, Cmd_Argv(i), sizeof(name));
if (!Q_strrchr(Cmd_Argv(i), '.'))
{
q_strlcat(name, ".wav", sizeof(name));
}
sfx = S_PrecacheSound(name);
S_StartSound(hash++, 0, sfx, listener_origin, 1.0, 1.0);
i++;
}
}
static void S_PlayVol (void)
{
static int hash = 543;
int i;
float vol;
char name[256];
sfx_t *sfx;
i = 1;
while (i < Cmd_Argc())
{
q_strlcpy(name, Cmd_Argv(i), sizeof(name));
if (!Q_strrchr(Cmd_Argv(i), '.'))
{
q_strlcat(name, ".wav", sizeof(name));
}
sfx = S_PrecacheSound(name);
vol = Q_atof(Cmd_Argv(i + 1));
S_StartSound(hash++, 0, sfx, listener_origin, vol, 1.0);
i += 2;
}
}
static void S_SoundList (void)
{
int i;
sfx_t *sfx;
sfxcache_t *sc;
int size, total;
total = 0;
for (sfx = known_sfx, i = 0; i < num_sfx; i++, sfx++)
{
sc = (sfxcache_t *) Cache_Check (&sfx->cache);
if (!sc)
continue;
size = sc->length*sc->width*(sc->stereo + 1);
total += size;
if (sc->loopstart >= 0)
Con_SafePrintf ("L"); //johnfitz -- was Con_Printf
else
Con_SafePrintf (" "); //johnfitz -- was Con_Printf
Con_SafePrintf("(%2db) %6i : %s\n", sc->width*8, size, sfx->name); //johnfitz -- was Con_Printf
}
Con_Printf ("%i sounds, %i bytes\n", num_sfx, total); //johnfitz -- added count
}
void S_LocalSound (const char *name)
{
sfx_t *sfx;
if (nosound.value)
return;
if (!sound_started)
return;
sfx = S_PrecacheSound (name);
if (!sfx)
{
Con_Printf ("S_LocalSound: can't cache %s\n", name);
return;
}
S_StartSound (cl.viewentity, -1, sfx, vec3_origin, 1, 1);
}
void S_ClearPrecache (void)
{
}
void S_BeginPrecaching (void)
{
}
void S_EndPrecaching (void)
{
}