ref: df292a6dce27e7b20faa0c829aa0c58a3693f979
dir: /src/i_sdlsound.c/
// Emacs style mode select -*- C++ -*-
//-----------------------------------------------------------------------------
//
// Copyright(C) 1993-1996 Id Software, Inc.
// Copyright(C) 2005 Simon Howard
// Copyright(C) 2008 David Flater
//
// 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.
//
// DESCRIPTION:
// System interface for sound.
//
//-----------------------------------------------------------------------------
#include "config.h"
#include <stdio.h>
#include <stdlib.h>
#include <assert.h>
#include <math.h>
#include "SDL.h"
#include "SDL_mixer.h"
#ifdef HAVE_LIBSAMPLERATE
#include <samplerate.h>
#endif
#include "deh_main.h"
#include "i_system.h"
#include "i_swap.h"
#include "s_sound.h"
#include "m_argv.h"
#include "w_wad.h"
#include "z_zone.h"
#include "doomdef.h"
#define LOW_PASS_FILTER
//#define DEBUG_DUMP_WAVS
#define MAX_SOUND_SLICE_TIME 70 /* ms */
#define NUM_CHANNELS 16
static boolean setpanning_workaround = false;
static boolean sound_initialized = false;
static Mix_Chunk sound_chunks[NUMSFX];
static int channels_playing[NUM_CHANNELS];
static int mixer_freq;
static Uint16 mixer_format;
static int mixer_channels;
int use_libsamplerate = 0;
// When a sound stops, check if it is still playing. If it is not,
// we can mark the sound data as CACHE to be freed back for other
// means.
static void ReleaseSoundOnChannel(int channel)
{
int i;
int id = channels_playing[channel];
if (!id)
{
return;
}
channels_playing[channel] = sfx_None;
#ifdef HAVE_LIBSAMPLERATE
// Don't allow precached sounds to be swapped out.
if (use_libsamplerate)
return;
#endif
for (i=0; i<NUM_CHANNELS; ++i)
{
// Playing on this channel? if so, don't release.
if (channels_playing[i] == id)
return;
}
// Not used on any channel, and can be safely released
Z_ChangeTag(sound_chunks[id].abuf, PU_CACHE);
}
#ifdef HAVE_LIBSAMPLERATE
// Returns the conversion mode for libsamplerate to use.
static int SRC_ConversionMode(void)
{
switch (use_libsamplerate)
{
// 0 = disabled
default:
case 0:
return -1;
// Ascending numbers give higher quality
case 1:
return SRC_LINEAR;
case 2:
return SRC_ZERO_ORDER_HOLD;
case 3:
return SRC_SINC_FASTEST;
case 4:
return SRC_SINC_MEDIUM_QUALITY;
case 5:
return SRC_SINC_BEST_QUALITY;
}
}
#endif
static boolean ConvertibleRatio(int freq1, int freq2)
{
int ratio;
if (freq1 > freq2)
{
return ConvertibleRatio(freq2, freq1);
}
else if ((freq2 % freq1) != 0)
{
// Not in a direct ratio
return false;
}
else
{
// Check the ratio is a power of 2
ratio = freq2 / freq1;
while ((ratio & 1) == 0)
{
ratio = ratio >> 1;
}
return ratio == 1;
}
}
#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)
{
SDL_AudioCVT convertor;
uint32_t expanded_length;
// Calculate the length of the expanded version of the sample.
expanded_length = (uint32_t) ((((uint64_t) length) * mixer_freq) / samplerate);
// Double up twice: 8 -> 16 bit and mono -> stereo
expanded_length *= 4;
destination->alen = expanded_length;
destination->abuf
= 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,
AUDIO_U8, 1, samplerate,
mixer_format, mixer_channels, mixer_freq))
{
convertor.buf = destination->abuf;
convertor.len = length;
memcpy(convertor.buf, data, length);
SDL_ConvertAudio(&convertor);
}
else
{
Sint16 *expanded = (Sint16 *) destination->abuf;
int expanded_length;
int expand_ratio;
int i;
// Generic expansion if conversion does not work:
//
// SDL's audio conversion only works for rate conversions that are
// powers of 2; if the two formats are not in a direct power of 2
// ratio, do this naive conversion instead.
// number of samples in the converted sound
expanded_length = ((uint64_t) length * mixer_freq) / samplerate;
expand_ratio = (length << 8) / expanded_length;
for (i=0; i<expanded_length; ++i)
{
Sint16 sample;
int src;
src = (i * expand_ratio) >> 8;
sample = data[src] | (data[src] << 8);
sample -= 32768;
// expand 8->16 bits, mono->stereo
expanded[i * 2] = expanded[i * 2 + 1] = sample;
}
#ifdef LOW_PASS_FILTER
// Perform a low-pass filter on the upscaled sound to filter
// out high-frequency noise from the conversion process.
{
float rc, dt, alpha;
// Low-pass filter for cutoff frequency f:
//
// For sampling rate r, dt = 1 / r
// rc = 1 / 2*pi*f
// alpha = dt / (rc + dt)
// Filter to the half sample rate of the original sound effect
// (maximum frequency, by nyquist)
dt = 1.0f / mixer_freq;
rc = 1.0f / (3.14f * samplerate);
alpha = dt / (rc + dt);
// Both channels are processed in parallel, hence [i-2]:
for (i=2; i<expanded_length * 2; ++i)
{
expanded[i] = (Sint16) (alpha * expanded[i]
+ (1 - alpha) * expanded[i-2]);
}
}
#endif /* #ifdef LOW_PASS_FILTER */
}
}
// Load and validate a sound effect lump.
// Preconditions:
// S_sfx[sound].lumpnum has been set
// Postconditions if sound is valid:
// returns true
// starred parameters are set, with data_ref pointing to start of sound
// caller is responsible for releasing the identified lump
// Postconditions if sound is invalid:
// returns false
// starred parameters are garbage
// lump already released
static boolean LoadSoundLump(int sound,
int *lumpnum,
int *samplerate,
uint32_t *length,
byte **data_ref)
{
int lumplen;
byte *data;
// Load the sound
*lumpnum = S_sfx[sound].lumpnum;
*data_ref = W_CacheLumpNum(*lumpnum, PU_STATIC);
lumplen = W_LumpLength(*lumpnum);
data = *data_ref;
// Ensure this is a valid sound
if (lumplen < 8 || data[0] != 0x03 || data[1] != 0x00)
{
// Invalid sound
W_ReleaseLumpNum(*lumpnum);
return false;
}
// 16 bit sample rate field, 32 bit length field
*samplerate = (data[3] << 8) | data[2];
*length = (data[7] << 24) | (data[6] << 16) | (data[5] << 8) | data[4];
// If the header specifies that the length of the sound is
// greater than the length of the lump itself, this is an invalid
// sound lump.
// We also discard sound lumps that are less than 49 samples long,
// as this is how DMX behaves - although the actual cut-off length
// seems to vary slightly depending on the sample rate. This needs
// further investigation to better understand the correct
// behavior.
if (*length > lumplen - 8 || *length <= 48)
{
W_ReleaseLumpNum(*lumpnum);
return false;
}
// Prune header
*data_ref += 8;
// The DMX sound library seems to skip the first 16 and last 16
// bytes of the lump - reason unknown.
*data_ref += 16;
*length -= 32;
return true;
}
// Load and convert a sound effect
// Returns true if successful
static boolean CacheSFX_SDL(int sound)
{
int lumpnum;
int samplerate;
uint32_t length;
byte *data;
#ifdef HAVE_LIBSAMPLERATE
assert(!use_libsamplerate); // Should be using I_PrecacheSounds_SRC instead
#endif
if (!LoadSoundLump(sound, &lumpnum, &samplerate, &length, &data))
return false;
// Sample rate conversion
// sound_chunks[sound].alen and abuf are determined by ExpandSoundData.
sound_chunks[sound].allocated = 1;
sound_chunks[sound].volume = MIX_MAX_VOLUME;
ExpandSoundData_SDL(data,
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
W_ReleaseLumpNum(lumpnum);
return true;
}
#ifdef HAVE_LIBSAMPLERATE
// Preload and resample all sound effects with libsamplerate.
static void I_PrecacheSounds_SRC(void)
{
char namebuf[9];
uint32_t sound_i, sample_i;
boolean good_sound[NUMSFX];
float *resampled_sound[NUMSFX];
uint32_t resampled_sound_length[NUMSFX];
float norm_factor;
float max_amp = 0;
unsigned int zone_size;
assert(use_libsamplerate);
zone_size = Z_ZoneSize();
if (zone_size < 32 * 1024 * 1024)
{
fprintf(stderr,
"WARNING: low memory. Heap size is only %d MiB.\n"
"WARNING: use_libsamplerate needs more heap!\n"
"WARNING: put -mb 64 on the command line to avoid "
"\"Error: Z_Malloc: failed on allocation of X bytes\" !\n",
zone_size / (1024 * 1024));
}
printf("I_PrecacheSounds_SRC: Precaching all sound effects..");
// Pass 1: resample all sounds and determine maximum amplitude.
for (sound_i=sfx_pistol; sound_i<NUMSFX; ++sound_i)
{
good_sound[sound_i] = false;
if ((sound_i % 6) == 0)
{
printf(".");
fflush(stdout);
}
sprintf(namebuf, "ds%s", DEH_String(S_sfx[sound_i].name));
S_sfx[sound_i].lumpnum = W_CheckNumForName(namebuf);
if (S_sfx[sound_i].lumpnum != -1)
{
int lumpnum;
int samplerate;
uint32_t length;
byte *data;
double of_temp;
int retn;
float *rsound;
uint32_t rlen;
SRC_DATA src_data;
if (!LoadSoundLump(sound_i, &lumpnum, &samplerate, &length, &data))
continue;
assert(length <= LONG_MAX);
src_data.input_frames = length;
src_data.data_in = malloc(length * sizeof(float));
src_data.src_ratio = (double)mixer_freq / samplerate;
// mixer_freq / 4 adds a quarter-second safety margin.
of_temp = src_data.src_ratio * length + (mixer_freq / 4);
assert(of_temp <= LONG_MAX);
src_data.output_frames = of_temp;
src_data.data_out = malloc(src_data.output_frames * sizeof(float));
assert(src_data.data_in != NULL && src_data.data_out != NULL);
// Convert input data to floats
for (sample_i=0; sample_i<length; ++sample_i)
{
// Unclear whether 128 should be interpreted as "zero" or
// whether a symmetrical range should be assumed. The
// following assumes a symmetrical range.
src_data.data_in[sample_i] = data[sample_i] / 127.5 - 1;
}
// don't need the original lump any more
W_ReleaseLumpNum(lumpnum);
// Resample
retn = src_simple(&src_data, SRC_ConversionMode(), 1);
assert(retn == 0);
assert(src_data.output_frames_gen > 0);
resampled_sound[sound_i] = src_data.data_out;
resampled_sound_length[sound_i] = src_data.output_frames_gen;
free(src_data.data_in);
good_sound[sound_i] = true;
// Track maximum amplitude for later normalization
rsound = resampled_sound[sound_i];
rlen = resampled_sound_length[sound_i];
for (sample_i=0; sample_i<rlen; ++sample_i)
{
float fabs_amp = fabsf(rsound[sample_i]);
if (fabs_amp > max_amp)
max_amp = fabs_amp;
}
}
}
// Pass 2: normalize and convert to signed 16-bit stereo.
if (max_amp <= 0)
max_amp = 1;
norm_factor = INT16_MAX / max_amp;
for (sound_i=sfx_pistol; sound_i<NUMSFX; ++sound_i)
{
if (good_sound[sound_i])
{
uint32_t rlen = resampled_sound_length[sound_i];
int16_t *expanded;
uint32_t abuf_index;
float *rsound;
sound_chunks[sound_i].allocated = 1;
sound_chunks[sound_i].volume = MIX_MAX_VOLUME;
sound_chunks[sound_i].alen = rlen * 4;
sound_chunks[sound_i].abuf = Z_Malloc(sound_chunks[sound_i].alen,
PU_STATIC,
&sound_chunks[sound_i].abuf);
expanded = (int16_t *) sound_chunks[sound_i].abuf;
abuf_index=0;
rsound = resampled_sound[sound_i];
for (sample_i=0; sample_i<rlen; ++sample_i)
{
float cvtval_f = norm_factor * rsound[sample_i];
int16_t cvtval_i = cvtval_f + (cvtval_f < 0 ? -0.5 : 0.5);
// Left and right channels
expanded[abuf_index++] = cvtval_i;
expanded[abuf_index++] = cvtval_i;
}
free(rsound);
}
}
printf(" norm factor = %f\n", norm_factor);
}
#endif
static Mix_Chunk *GetSFXChunk(int sound_id)
{
if (sound_chunks[sound_id].abuf == NULL)
{
#ifdef HAVE_LIBSAMPLERATE
if (use_libsamplerate != 0)
return NULL; /* If valid, it should have been precached */
#endif
if (!CacheSFX_SDL(sound_id))
return NULL;
}
else
{
// don't free the sound while it is playing!
Z_ChangeTag(sound_chunks[sound_id].abuf, PU_STATIC);
}
return &sound_chunks[sound_id];
}
//
// Retrieve the raw data lump index
// for a given SFX name.
//
static int I_SDL_GetSfxLumpNum(sfxinfo_t* sfx)
{
char namebuf[9];
sprintf(namebuf, "ds%s", DEH_String(sfx->name));
return W_GetNumForName(namebuf);
}
static void I_SDL_UpdateSoundParams(int handle, int vol, int sep)
{
int left, right;
if (!sound_initialized)
{
return;
}
left = ((254 - sep) * vol) / 127;
right = ((sep) * vol) / 127;
// SDL_mixer version 1.2.8 and earlier has a bug in the Mix_SetPanning
// function. A workaround is to call Mix_UnregisterAllEffects for
// the channel before calling it. This is undesirable as it may lead
// to the channel volumes resetting briefly.
if (setpanning_workaround)
{
Mix_UnregisterAllEffects(handle);
}
Mix_SetPanning(handle, left, right);
}
//
// 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.
//
static int I_SDL_StartSound(int id, int channel, int vol, int sep)
{
Mix_Chunk *chunk;
if (!sound_initialized)
{
return -1;
}
// Release a sound effect if there is already one playing
// on this channel
ReleaseSoundOnChannel(channel);
// Get the sound data
chunk = GetSFXChunk(id);
if (chunk == NULL)
{
return -1;
}
// play sound
Mix_PlayChannelTimed(channel, chunk, 0, -1);
channels_playing[channel] = id;
// set separation, etc.
I_SDL_UpdateSoundParams(channel, vol, sep);
return channel;
}
static void I_SDL_StopSound (int handle)
{
if (!sound_initialized)
{
return;
}
Mix_HaltChannel(handle);
// Sound data is no longer needed; release the
// sound data being used for this channel
ReleaseSoundOnChannel(handle);
}
static boolean I_SDL_SoundIsPlaying(int handle)
{
if (handle < 0)
{
return false;
}
return Mix_Playing(handle);
}
//
// Periodically called to update the sound system
//
static void I_SDL_UpdateSound(void)
{
int i;
// Check all channels to see if a sound has finished
for (i=0; i<NUM_CHANNELS; ++i)
{
if (channels_playing[i] && !I_SDL_SoundIsPlaying(i))
{
// Sound has finished playing on this channel,
// but sound data has not been released to cache
ReleaseSoundOnChannel(i);
}
}
}
static void I_SDL_ShutdownSound(void)
{
if (!sound_initialized)
{
return;
}
Mix_CloseAudio();
SDL_QuitSubSystem(SDL_INIT_AUDIO);
sound_initialized = false;
}
// Calculate slice size, based on MAX_SOUND_SLICE_TIME.
// The result must be a power of two.
static int GetSliceSize(void)
{
int limit;
int n;
limit = (snd_samplerate * MAX_SOUND_SLICE_TIME) / 1000;
// Try all powers of two, not exceeding the limit.
for (n=0;; ++n)
{
// 2^n <= limit < 2^n+1 ?
if ((1 << (n + 1)) > limit)
{
return (1 << n);
}
}
// Should never happen?
return 1024;
}
static boolean I_SDL_InitSound(void)
{
int i;
// No sounds yet
for (i=0; i<NUMSFX; ++i)
{
sound_chunks[i].abuf = NULL;
}
for (i=0; i<NUM_CHANNELS; ++i)
{
channels_playing[i] = sfx_None;
}
if (SDL_Init(SDL_INIT_AUDIO) < 0)
{
fprintf(stderr, "Unable to set up sound.\n");
return false;
}
if (Mix_OpenAudio(snd_samplerate, AUDIO_S16SYS, 2, GetSliceSize()) < 0)
{
fprintf(stderr, "Error initializing SDL_mixer: %s\n", Mix_GetError());
return false;
}
Mix_QuerySpec(&mixer_freq, &mixer_format, &mixer_channels);
#ifdef HAVE_LIBSAMPLERATE
if (use_libsamplerate != 0)
{
if (SRC_ConversionMode() < 0)
{
I_Error("I_SDL_InitSound: Invalid value for use_libsamplerate: %i",
use_libsamplerate);
}
I_PrecacheSounds_SRC();
}
#else
if (use_libsamplerate != 0)
{
fprintf(stderr, "I_SDL_InitSound: use_libsamplerate=%i, but "
"libsamplerate support not compiled in.\n",
use_libsamplerate);
}
#endif
// SDL_mixer version 1.2.8 and earlier has a bug in the Mix_SetPanning
// function that can cause the game to lock up. If we're using an old
// version, we need to apply a workaround. But the workaround has its
// own drawbacks ...
{
const SDL_version *mixer_version;
int v;
mixer_version = Mix_Linked_Version();
v = SDL_VERSIONNUM(mixer_version->major,
mixer_version->minor,
mixer_version->patch);
if (v <= SDL_VERSIONNUM(1, 2, 8))
{
setpanning_workaround = true;
fprintf(stderr, "\n"
"ATTENTION: You are using an old version of SDL_mixer!\n"
" This version has a bug that may cause "
"your sound to stutter.\n"
" Please upgrade to a newer version!\n"
"\n");
}
}
Mix_AllocateChannels(NUM_CHANNELS);
SDL_PauseAudio(0);
sound_initialized = true;
return true;
}
static snddevice_t sound_sdl_devices[] =
{
SNDDEVICE_SB,
SNDDEVICE_PAS,
SNDDEVICE_GUS,
SNDDEVICE_WAVEBLASTER,
SNDDEVICE_SOUNDCANVAS,
SNDDEVICE_AWE32,
};
sound_module_t sound_sdl_module =
{
sound_sdl_devices,
arrlen(sound_sdl_devices),
I_SDL_InitSound,
I_SDL_ShutdownSound,
I_SDL_GetSfxLumpNum,
I_SDL_UpdateSound,
I_SDL_UpdateSoundParams,
I_SDL_StartSound,
I_SDL_StopSound,
I_SDL_SoundIsPlaying,
};