ref: 2053768a9251cbd4c731681a3e1e04c800ad21f4
dir: /src/i_sdlsound.c/
// // Copyright(C) 1993-1996 Id Software, Inc. // Copyright(C) 2005-2014 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. // // DESCRIPTION: // System interface for sound. // #include "config.h" #include <stdio.h> #include <stdlib.h> #include <string.h> #include <assert.h> #include "SDL.h" #include "SDL_mixer.h" #ifdef HAVE_LIBSAMPLERATE #include <samplerate.h> #endif #include "deh_str.h" #include "i_sound.h" #include "i_system.h" #include "i_swap.h" #include "m_argv.h" #include "m_misc.h" #include "w_wad.h" #include "z_zone.h" #include "doomtype.h" int use_libsamplerate = 0; // Scale factor used when converting libsamplerate floating point numbers // to integers. Too high means the sounds can clip; too low means they // will be too quiet. This is an amount that should avoid clipping most // of the time: with all the Doom IWAD sound effects, at least. If a PWAD // is used, clipping might occur. float libsamplerate_scale = 0.65f; #ifndef DISABLE_SDL2MIXER #define LOW_PASS_FILTER //#define DEBUG_DUMP_WAVS #define NUM_CHANNELS 16 typedef struct allocated_sound_s allocated_sound_t; struct allocated_sound_s { sfxinfo_t *sfxinfo; Mix_Chunk chunk; int use_count; int pitch; allocated_sound_t *prev, *next; }; static boolean sound_initialized = false; static allocated_sound_t *channels_playing[NUM_CHANNELS]; static int mixer_freq; static Uint16 mixer_format; static int mixer_channels; static boolean use_sfx_prefix; static boolean (*ExpandSoundData)(sfxinfo_t *sfxinfo, byte *data, int samplerate, int length) = NULL; // Doubly-linked list of allocated sounds. // When a sound is played, it is moved to the head, so that the oldest // sounds not used recently are at the tail. static allocated_sound_t *allocated_sounds_head = NULL; static allocated_sound_t *allocated_sounds_tail = NULL; static int allocated_sounds_size = 0; // Hook a sound into the linked list at the head. static void AllocatedSoundLink(allocated_sound_t *snd) { snd->prev = NULL; snd->next = allocated_sounds_head; allocated_sounds_head = snd; if (allocated_sounds_tail == NULL) { allocated_sounds_tail = snd; } else { snd->next->prev = snd; } } // Unlink a sound from the linked list. static void AllocatedSoundUnlink(allocated_sound_t *snd) { if (snd->prev == NULL) { allocated_sounds_head = snd->next; } else { snd->prev->next = snd->next; } if (snd->next == NULL) { allocated_sounds_tail = snd->prev; } else { snd->next->prev = snd->prev; } } static void FreeAllocatedSound(allocated_sound_t *snd) { // Unlink from linked list. AllocatedSoundUnlink(snd); // Keep track of the amount of allocated sound data: allocated_sounds_size -= snd->chunk.alen; free(snd); } // Search from the tail backwards along the allocated sounds list, find // and free a sound that is not in use, to free up memory. Return true // for success. static boolean FindAndFreeSound(void) { allocated_sound_t *snd; snd = allocated_sounds_tail; while (snd != NULL) { if (snd->use_count == 0) { FreeAllocatedSound(snd); return true; } snd = snd->prev; } // No available sounds to free... return false; } // Enforce SFX cache size limit. We are just about to allocate "len" // bytes on the heap for a new sound effect, so free up some space // so that we keep allocated_sounds_size < snd_cachesize static void ReserveCacheSpace(size_t len) { if (snd_cachesize <= 0) { return; } // Keep freeing sound effects that aren't currently being played, // until there is enough space for the new sound. while (allocated_sounds_size + len > snd_cachesize) { // Free a sound. If there is nothing more to free, stop. if (!FindAndFreeSound()) { break; } } } // Allocate a block for a new sound effect. static allocated_sound_t *AllocateSound(sfxinfo_t *sfxinfo, size_t len) { allocated_sound_t *snd; // Keep allocated sounds within the cache size. ReserveCacheSpace(len); // Allocate the sound structure and data. The data will immediately // follow the structure, which acts as a header. do { snd = malloc(sizeof(allocated_sound_t) + len); // Out of memory? Try to free an old sound, then loop round // and try again. if (snd == NULL && !FindAndFreeSound()) { return NULL; } } while (snd == NULL); // Skip past the chunk structure for the audio buffer snd->chunk.abuf = (byte *) (snd + 1); snd->chunk.alen = len; snd->chunk.allocated = 1; snd->chunk.volume = MIX_MAX_VOLUME; snd->pitch = NORM_PITCH; snd->sfxinfo = sfxinfo; snd->use_count = 0; // Keep track of how much memory all these cached sounds are using... allocated_sounds_size += len; AllocatedSoundLink(snd); return snd; } // Lock a sound, to indicate that it may not be freed. static void LockAllocatedSound(allocated_sound_t *snd) { // Increase use count, to stop the sound being freed. ++snd->use_count; //printf("++ %s: Use count=%i\n", snd->sfxinfo->name, snd->use_count); // When we use a sound, re-link it into the list at the head, so // that the oldest sounds fall to the end of the list for freeing. AllocatedSoundUnlink(snd); AllocatedSoundLink(snd); } // Unlock a sound to indicate that it may now be freed. static void UnlockAllocatedSound(allocated_sound_t *snd) { if (snd->use_count <= 0) { I_Error("Sound effect released more times than it was locked..."); } --snd->use_count; //printf("-- %s: Use count=%i\n", snd->sfxinfo->name, snd->use_count); } // Search through the list of allocated sounds and return the one that matches // the supplied sfxinfo entry and pitch level. static allocated_sound_t * GetAllocatedSoundBySfxInfoAndPitch(sfxinfo_t *sfxinfo, int pitch) { allocated_sound_t * p = allocated_sounds_head; while (p != NULL) { if (p->sfxinfo == sfxinfo && p->pitch == pitch) { return p; } p = p->next; } return NULL; } // Allocate a new sound chunk and pitch-shift an existing sound up-or-down // into it. static allocated_sound_t * PitchShift(allocated_sound_t *insnd, int pitch) { allocated_sound_t * outsnd; Sint16 *inp, *outp; Sint16 *srcbuf, *dstbuf; Uint32 srclen, dstlen; srcbuf = (Sint16 *)insnd->chunk.abuf; srclen = insnd->chunk.alen; // determine ratio pitch:NORM_PITCH and apply to srclen, then invert. // This is an approximation of vanilla behaviour based on measurements dstlen = (int)((1 + (1 - (float)pitch / NORM_PITCH)) * srclen); // ensure that the new buffer is an even length if ((dstlen % 2) == 0) { dstlen++; } outsnd = AllocateSound(insnd->sfxinfo, dstlen); if (!outsnd) { return NULL; } outsnd->pitch = pitch; dstbuf = (Sint16 *)outsnd->chunk.abuf; // loop over output buffer. find corresponding input cell, copy over for (outp = dstbuf; outp < dstbuf + dstlen/2; ++outp) { inp = srcbuf + (int)((float)(outp - dstbuf) / dstlen * srclen); *outp = *inp; } return outsnd; } // 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) { allocated_sound_t *snd = channels_playing[channel]; Mix_HaltChannel(channel); if (snd == NULL) { return; } channels_playing[channel] = NULL; UnlockAllocatedSound(snd); // if the sound is a pitch-shift and it's not in use, immediately // free it if (snd->pitch != NORM_PITCH && snd->use_count <= 0) { FreeAllocatedSound(snd); } } #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; } } // libsamplerate-based generic sound expansion function for any sample rate // unsigned 8 bits --> signed 16 bits // mono --> stereo // samplerate --> mixer_freq // Returns number of clipped samples. // DWF 2008-02-10 with cleanups by Simon Howard. static boolean ExpandSoundData_SRC(sfxinfo_t *sfxinfo, byte *data, int samplerate, int length) { SRC_DATA src_data; float *data_in; uint32_t i, abuf_index=0, clipped=0; // uint32_t alen; int retn; int16_t *expanded; allocated_sound_t *snd; Mix_Chunk *chunk; src_data.input_frames = length; data_in = malloc(length * sizeof(float)); src_data.data_in = data_in; src_data.src_ratio = (double)mixer_freq / samplerate; // We include some extra space here in case of rounding-up. src_data.output_frames = src_data.src_ratio * length + (mixer_freq / 4); 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 (i=0; i<length; ++i) { // Unclear whether 128 should be interpreted as "zero" or whether a // symmetrical range should be assumed. The following assumes a // symmetrical range. data_in[i] = data[i] / 127.5 - 1; } // Do the sound conversion retn = src_simple(&src_data, SRC_ConversionMode(), 1); assert(retn == 0); // Allocate the new chunk. // alen = src_data.output_frames_gen * 4; snd = AllocateSound(sfxinfo, src_data.output_frames_gen * 4); if (snd == NULL) { return false; } chunk = &snd->chunk; expanded = (int16_t *) chunk->abuf; // Convert the result back into 16-bit integers. for (i=0; i<src_data.output_frames_gen; ++i) { // libsamplerate does not limit itself to the -1.0 .. 1.0 range on // output, so a multiplier less than INT16_MAX (32767) is required // to avoid overflows or clipping. However, the smaller the // multiplier, the quieter the sound effects get, and the more you // have to turn down the music to keep it in balance. // 22265 is the largest multiplier that can be used to resample all // of the Vanilla DOOM sound effects to 48 kHz without clipping // using SRC_SINC_BEST_QUALITY. It is close enough (only slightly // too conservative) for SRC_SINC_MEDIUM_QUALITY and // SRC_SINC_FASTEST. PWADs with interestingly different sound // effects or target rates other than 48 kHz might still result in // clipping--I don't know if there's a limit to it. // As the number of clipped samples increases, the signal is // gradually overtaken by noise, with the loudest parts going first. // However, a moderate amount of clipping is often tolerated in the // quest for the loudest possible sound overall. The results of // using INT16_MAX as the multiplier are not all that bad, but // artifacts are noticeable during the loudest parts. float cvtval_f = src_data.data_out[i] * libsamplerate_scale * INT16_MAX; int32_t cvtval_i = cvtval_f + (cvtval_f < 0 ? -0.5 : 0.5); // Asymmetrical sound worries me, so we won't use -32768. if (cvtval_i < -INT16_MAX) { cvtval_i = -INT16_MAX; ++clipped; } else if (cvtval_i > INT16_MAX) { cvtval_i = INT16_MAX; ++clipped; } // Left and right channels expanded[abuf_index++] = cvtval_i; expanded[abuf_index++] = cvtval_i; } free(data_in); free(src_data.data_out); if (clipped > 0) { fprintf(stderr, "Sound '%s': clipped %u samples (%0.2f %%)\n", sfxinfo->name, clipped, 400.0 * clipped / chunk->alen); } return true; } #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 = M_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. // Returns number of clipped samples (always 0). static boolean ExpandSoundData_SDL(sfxinfo_t *sfxinfo, byte *data, int samplerate, int length) { SDL_AudioCVT convertor; allocated_sound_t *snd; Mix_Chunk *chunk; 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; // Allocate a chunk in which to expand the sound snd = AllocateSound(sfxinfo, expanded_length); if (snd == NULL) { return false; } chunk = &snd->chunk; // 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.len = length; convertor.buf = malloc(convertor.len * convertor.len_mult); assert(convertor.buf != NULL); memcpy(convertor.buf, data, length); SDL_ConvertAudio(&convertor); memcpy(chunk->abuf, convertor.buf, chunk->alen); free(convertor.buf); } else { Sint16 *expanded = (Sint16 *) chunk->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 */ } return true; } // Load and convert a sound effect // Returns true if successful static boolean CacheSFX(sfxinfo_t *sfxinfo) { int lumpnum; unsigned int lumplen; int samplerate; unsigned int length; byte *data; // need to load the sound lumpnum = sfxinfo->lumpnum; data = W_CacheLumpNum(lumpnum, PU_STATIC); lumplen = W_LumpLength(lumpnum); // Check the header, and ensure this is a valid sound if (lumplen < 8 || data[0] != 0x03 || data[1] != 0x00) { // Invalid sound 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) { return false; } // The DMX sound library seems to skip the first 16 and last 16 // bytes of the lump - reason unknown. data += 16; length -= 32; // Sample rate conversion if (!ExpandSoundData(sfxinfo, data + 8, samplerate, length)) { return false; } #ifdef DEBUG_DUMP_WAVS { char filename[16]; allocated_sound_t * snd; M_snprintf(filename, sizeof(filename), "%s.wav", DEH_String(sfxinfo->name)); snd = GetAllocatedSoundBySfxInfoAndPitch(sfxinfo, NORM_PITCH); WriteWAV(filename, snd->chunk.abuf, snd->chunk.alen,mixer_freq); } #endif // don't need the original lump any more W_ReleaseLumpNum(lumpnum); return true; } static void GetSfxLumpName(sfxinfo_t *sfx, char *buf, size_t buf_len) { // Linked sfx lumps? Get the lump number for the sound linked to. if (sfx->link != NULL) { sfx = sfx->link; } // Doom adds a DS* prefix to sound lumps; Heretic and Hexen don't // do this. if (use_sfx_prefix) { M_snprintf(buf, buf_len, "ds%s", DEH_String(sfx->name)); } else { M_StringCopy(buf, DEH_String(sfx->name), buf_len); } } // Preload all the sound effects - stops nasty ingame freezes static void I_SDL_PrecacheSounds(sfxinfo_t *sounds, int num_sounds) { char namebuf[9]; int i; printf("I_SDL_PrecacheSounds: Precaching all sound effects.."); for (i=0; i<num_sounds; ++i) { if ((i % 6) == 0) { printf("."); fflush(stdout); } GetSfxLumpName(&sounds[i], namebuf, sizeof(namebuf)); sounds[i].lumpnum = W_CheckNumForName(namebuf); if (sounds[i].lumpnum != -1) { CacheSFX(&sounds[i]); } } printf("\n"); } // Load a SFX chunk into memory and ensure that it is locked. static boolean LockSound(sfxinfo_t *sfxinfo) { // If the sound isn't loaded, load it now if (GetAllocatedSoundBySfxInfoAndPitch(sfxinfo, NORM_PITCH) == NULL) { if (!CacheSFX(sfxinfo)) { return false; } } LockAllocatedSound(GetAllocatedSoundBySfxInfoAndPitch(sfxinfo, NORM_PITCH)); return true; } // // Retrieve the raw data lump index // for a given SFX name. // static int I_SDL_GetSfxLumpNum(sfxinfo_t *sfx) { char namebuf[9]; GetSfxLumpName(sfx, namebuf, sizeof(namebuf)); return W_GetNumForName(namebuf); } static void I_SDL_UpdateSoundParams(int handle, int vol, int sep) { int left, right; if (!sound_initialized || handle < 0 || handle >= NUM_CHANNELS) { return; } left = ((254 - sep) * vol) / 127; right = ((sep) * vol) / 127; if (left < 0) left = 0; else if ( left > 255) left = 255; if (right < 0) right = 0; else if (right > 255) right = 255; 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(sfxinfo_t *sfxinfo, int channel, int vol, int sep, int pitch) { allocated_sound_t *snd; if (!sound_initialized || channel < 0 || channel >= NUM_CHANNELS) { return -1; } // Release a sound effect if there is already one playing // on this channel ReleaseSoundOnChannel(channel); // Get the sound data if (!LockSound(sfxinfo)) { return -1; } snd = GetAllocatedSoundBySfxInfoAndPitch(sfxinfo, pitch); if (snd == NULL) { allocated_sound_t *newsnd; // fetch the base sound effect, un-pitch-shifted snd = GetAllocatedSoundBySfxInfoAndPitch(sfxinfo, NORM_PITCH); if (snd == NULL) { return -1; } if (snd_pitchshift) { newsnd = PitchShift(snd, pitch); if (newsnd) { LockAllocatedSound(newsnd); UnlockAllocatedSound(snd); snd = newsnd; } } } else { LockAllocatedSound(snd); } // play sound Mix_PlayChannel(channel, &snd->chunk, 0); channels_playing[channel] = snd; // set separation, etc. I_SDL_UpdateSoundParams(channel, vol, sep); return channel; } static void I_SDL_StopSound(int handle) { if (!sound_initialized || handle < 0 || handle >= NUM_CHANNELS) { return; } // 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 (!sound_initialized || handle < 0 || handle >= NUM_CHANNELS) { 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 snd_maxslicetime_ms. // The result must be a power of two. static int GetSliceSize(void) { int limit; int n; limit = (snd_samplerate * snd_maxslicetime_ms) / 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(boolean _use_sfx_prefix) { int i; use_sfx_prefix = _use_sfx_prefix; // No sounds yet for (i=0; i<NUM_CHANNELS; ++i) { channels_playing[i] = NULL; } if (SDL_Init(SDL_INIT_AUDIO) < 0) { fprintf(stderr, "Unable to set up sound.\n"); return false; } if (Mix_OpenAudioDevice(snd_samplerate, AUDIO_S16SYS, 2, GetSliceSize(), NULL, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE) < 0) { fprintf(stderr, "Error initialising SDL_mixer: %s\n", Mix_GetError()); return false; } ExpandSoundData = ExpandSoundData_SDL; 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); } ExpandSoundData = ExpandSoundData_SRC; } #else if (use_libsamplerate != 0) { fprintf(stderr, "I_SDL_InitSound: use_libsamplerate=%i, but " "libsamplerate support not compiled in.\n", use_libsamplerate); } #endif 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, I_SDL_PrecacheSounds, }; #endif // DISABLE_SDL2MIXER