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-rw-r--r--src/i_sdlsound.c518
1 files changed, 265 insertions, 253 deletions
diff --git a/src/i_sdlsound.c b/src/i_sdlsound.c
index a26a81dd..1cfafa6f 100644
--- a/src/i_sdlsound.c
+++ b/src/i_sdlsound.c
@@ -2,7 +2,7 @@
//-----------------------------------------------------------------------------
//
// Copyright(C) 1993-1996 Id Software, Inc.
-// Copyright(C) 2005 Simon Howard
+// Copyright(C) 2005-8 Simon Howard
// Copyright(C) 2008 David Flater
//
// This program is free software; you can redistribute it and/or
@@ -29,8 +29,8 @@
#include <stdio.h>
#include <stdlib.h>
+#include <string.h>
#include <assert.h>
-#include <math.h>
#include "SDL.h"
#include "SDL_mixer.h"
@@ -38,15 +38,15 @@
#include <samplerate.h>
#endif
-#include "deh_main.h"
+#include "deh_str.h"
+#include "i_sound.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"
+#include "doomtype.h"
#define LOW_PASS_FILTER
//#define DEBUG_DUMP_WAVS
@@ -57,12 +57,16 @@ static boolean setpanning_workaround = false;
static boolean sound_initialized = false;
-static Mix_Chunk sound_chunks[NUMSFX];
-static int channels_playing[NUM_CHANNELS];
+static sfxinfo_t *channels_playing[NUM_CHANNELS];
static int mixer_freq;
static Uint16 mixer_format;
static int mixer_channels;
+static boolean use_sfx_prefix;
+static void (*ExpandSoundData)(sfxinfo_t *sfxinfo,
+ byte *data,
+ int samplerate,
+ int length) = NULL;
int use_libsamplerate = 0;
@@ -73,34 +77,51 @@ int use_libsamplerate = 0;
static void ReleaseSoundOnChannel(int channel)
{
int i;
- int id = channels_playing[channel];
+ sfxinfo_t *sfxinfo = channels_playing[channel];
- if (!id)
+ if (sfxinfo == NULL)
{
return;
}
- channels_playing[channel] = sfx_None;
+ channels_playing[channel] = NULL;
-#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)
+ if (channels_playing[i] == sfxinfo)
return;
}
// Not used on any channel, and can be safely released
-
- Z_ChangeTag(sound_chunks[id].abuf, PU_CACHE);
+
+ Z_ChangeTag(sfxinfo->driver_data, PU_CACHE);
}
+// Allocate a new Mix_Chunk along with its data, storing
+// the result in the variable pointed to by variable
+
+static Mix_Chunk *AllocateChunk(sfxinfo_t *sfxinfo, uint32_t len)
+{
+ Mix_Chunk *chunk;
+
+ // Allocate the chunk and the audio buffer together
+
+ chunk = Z_Malloc(len + sizeof(Mix_Chunk),
+ PU_STATIC,
+ &sfxinfo->driver_data);
+ sfxinfo->driver_data = chunk;
+
+ // Skip past the chunk structure for the audio buffer
+
+ chunk->abuf = (byte *) (chunk + 1);
+ chunk->alen = len;
+ chunk->allocated = 1;
+ chunk->volume = MIX_MAX_VOLUME;
+
+ return chunk;
+}
#ifdef HAVE_LIBSAMPLERATE
@@ -131,6 +152,114 @@ static int SRC_ConversionMode(void)
}
}
+// 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 void ExpandSoundData_SRC(sfxinfo_t *sfxinfo,
+ byte *data,
+ int samplerate,
+ int length)
+{
+ SRC_DATA src_data;
+ uint32_t i, abuf_index=0, clipped=0;
+ uint32_t alen;
+ int retn;
+ int16_t *expanded;
+ Mix_Chunk *chunk;
+
+ src_data.input_frames = length;
+ src_data.data_in = malloc(length * sizeof(float));
+ 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.
+ src_data.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;
+
+ chunk = AllocateChunk(sfxinfo, src_data.output_frames_gen * 4);
+ 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] * 22265;
+ 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(src_data.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);
+ }
+}
+
#endif
static boolean ConvertibleRatio(int freq1, int freq2)
@@ -213,13 +342,15 @@ static void WriteWAV(char *filename, byte *data,
#endif
// Generic sound expansion function for any sample rate.
+// Returns number of clipped samples (always 0).
-static void ExpandSoundData_SDL(byte *data,
+static void ExpandSoundData_SDL(sfxinfo_t *sfxinfo,
+ byte *data,
int samplerate,
- uint32_t length,
- Mix_Chunk *destination)
+ int length)
{
SDL_AudioCVT convertor;
+ Mix_Chunk *chunk;
uint32_t expanded_length;
// Calculate the length of the expanded version of the sample.
@@ -229,9 +360,10 @@ static void ExpandSoundData_SDL(byte *data,
// 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);
+
+ // Allocate a chunk in which to expand the sound
+
+ chunk = AllocateChunk(sfxinfo, expanded_length);
// If we can, use the standard / optimized SDL conversion routines.
@@ -241,7 +373,7 @@ static void ExpandSoundData_SDL(byte *data,
AUDIO_U8, 1, samplerate,
mixer_format, mixer_channels, mixer_freq))
{
- convertor.buf = destination->abuf;
+ convertor.buf = chunk->abuf;
convertor.len = length;
memcpy(convertor.buf, data, length);
@@ -249,7 +381,7 @@ static void ExpandSoundData_SDL(byte *data,
}
else
{
- Sint16 *expanded = (Sint16 *) destination->abuf;
+ Sint16 *expanded = (Sint16 *) chunk->abuf;
int expanded_length;
int expand_ratio;
int i;
@@ -312,93 +444,49 @@ static void ExpandSoundData_SDL(byte *data,
}
}
+// Load and convert a sound effect
+// Returns true if successful
-// 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)
+static boolean CacheSFX(sfxinfo_t *sfxinfo)
{
- int lumplen;
+ int lumpnum;
+ unsigned int lumplen;
+ int samplerate;
+ unsigned int length;
byte *data;
- // Load the sound
+ // need to load the sound
- *lumpnum = S_sfx[sound].lumpnum;
- *data_ref = W_CacheLumpNum(*lumpnum, PU_STATIC);
- lumplen = W_LumpLength(*lumpnum);
- data = *data_ref;
+ lumpnum = sfxinfo->lumpnum;
+ data = W_CacheLumpNum(lumpnum, PU_STATIC);
+ lumplen = W_LumpLength(lumpnum);
- // Ensure this is a valid sound
+ // Check the header, and ensure this is a valid sound
- if (lumplen < 8 || data[0] != 0x03 || data[1] != 0x00)
+ if (lumplen < 8
+ || data[0] != 0x03 || data[1] != 0x00)
{
- // Invalid sound
- W_ReleaseLumpNum(*lumpnum);
- return false;
+ // 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];
+ 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.
+ // 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
- if (*length > lumplen - 8)
+ if (length > lumplen - 8)
{
- W_ReleaseLumpNum(*lumpnum);
- return false;
- }
-
- // Prune header
- *data_ref += 8;
-
- 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]);
+ ExpandSoundData(sfxinfo, data + 8, samplerate, length);
#ifdef DEBUG_DUMP_WAVS
{
@@ -417,196 +505,115 @@ static boolean CacheSFX_SDL(int sound)
return true;
}
+static void GetSfxLumpName(sfxinfo_t *sfx, char *buf)
+{
+ // 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)
+ {
+ sprintf(buf, "ds%s", DEH_String(sfx->name));
+ }
+ else
+ {
+ strcpy(buf, DEH_String(sfx->name));
+ }
+}
#ifdef HAVE_LIBSAMPLERATE
-// Preload and resample all sound effects with libsamplerate.
+// Preload all the sound effects - stops nasty ingame freezes
-static void I_PrecacheSounds_SRC(void)
+static void I_SDL_PrecacheSounds(sfxinfo_t *sounds, int num_sounds)
{
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);
+ int i;
- zone_size = Z_ZoneSize();
+ // Don't need to precache the sounds unless we are using libsamplerate.
- if (zone_size < 32 * 1024 * 1024)
+ if (use_libsamplerate == 0)
{
- 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));
+ return;
}
- printf("I_PrecacheSounds_SRC: Precaching all sound effects..");
-
- // Pass 1: resample all sounds and determine maximum amplitude.
+ printf("I_SDL_PrecacheSounds: Precaching all sound effects..");
- for (sound_i=sfx_pistol; sound_i<NUMSFX; ++sound_i)
+ for (i=0; i<num_sounds; ++i)
{
- good_sound[sound_i] = false;
-
- if ((sound_i % 6) == 0)
+ if ((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.
+ GetSfxLumpName(&sounds[i], namebuf);
- if (max_amp <= 0)
- max_amp = 1;
- norm_factor = INT16_MAX / max_amp;
+ sounds[i].lumpnum = W_CheckNumForName(namebuf);
- for (sound_i=sfx_pistol; sound_i<NUMSFX; ++sound_i)
- {
- if (good_sound[sound_i])
+ if (sounds[i].lumpnum != -1)
{
- 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);
+ // Try to cache the sound and then release it as cache
- // Left and right channels
-
- expanded[abuf_index++] = cvtval_i;
- expanded[abuf_index++] = cvtval_i;
+ if (CacheSFX(&sounds[i]))
+ {
+ Z_ChangeTag(sounds[i].driver_data, PU_CACHE);
}
- free(rsound);
}
}
- printf(" norm factor = %f\n", norm_factor);
+ printf("\n");
+}
+
+#else
+
+static void I_SDL_PrecacheSounds(sfxinfo_t *sounds, int num_sounds)
+{
+ // no-op
}
#endif
+// Load a SFX chunk into memory and ensure that it is locked.
-static Mix_Chunk *GetSFXChunk(int sound_id)
+static boolean LockSound(sfxinfo_t *sfxinfo)
{
- if (sound_chunks[sound_id].abuf == NULL)
+ // If the sound isn't loaded, load it now
+
+ if (sfxinfo->driver_data == 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;
+ if (!CacheSFX(sfxinfo))
+ {
+ return false;
+ }
}
else
{
- // don't free the sound while it is playing!
+ // Lock the sound effect into memory
- Z_ChangeTag(sound_chunks[sound_id].abuf, PU_STATIC);
+ Z_ChangeTag(sfxinfo->driver_data, PU_STATIC);
}
- return &sound_chunks[sound_id];
+ return true;
}
-
//
// Retrieve the raw data lump index
// for a given SFX name.
//
-static int I_SDL_GetSfxLumpNum(sfxinfo_t* sfx)
+static int I_SDL_GetSfxLumpNum(sfxinfo_t *sfx)
{
char namebuf[9];
- sprintf(namebuf, "ds%s", DEH_String(sfx->name));
-
+ GetSfxLumpName(sfx, namebuf);
+
return W_GetNumForName(namebuf);
}
@@ -622,6 +629,11 @@ static void I_SDL_UpdateSoundParams(int handle, int vol, int sep)
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;
+
// 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
@@ -648,7 +660,7 @@ static void I_SDL_UpdateSoundParams(int handle, int vol, int sep)
// is set, but currently not used by mixing.
//
-static int I_SDL_StartSound(int id, int channel, int vol, int sep)
+static int I_SDL_StartSound(sfxinfo_t *sfxinfo, int channel, int vol, int sep)
{
Mix_Chunk *chunk;
@@ -664,18 +676,18 @@ static int I_SDL_StartSound(int id, int channel, int vol, int sep)
// Get the sound data
- chunk = GetSFXChunk(id);
-
- if (chunk == NULL)
+ if (!LockSound(sfxinfo))
{
- return -1;
+ return -1;
}
+ chunk = (Mix_Chunk *) sfxinfo->driver_data;
+
// play sound
Mix_PlayChannelTimed(channel, chunk, 0, -1);
- channels_playing[channel] = id;
+ channels_playing[channel] = sfxinfo;
// set separation, etc.
@@ -772,20 +784,17 @@ static int GetSliceSize(void)
return 1024;
}
-static boolean I_SDL_InitSound(void)
-{
+static boolean I_SDL_InitSound(boolean _use_sfx_prefix)
+{
int i;
-
- // No sounds yet
- for (i=0; i<NUMSFX; ++i)
- {
- sound_chunks[i].abuf = NULL;
- }
+ use_sfx_prefix = _use_sfx_prefix;
+
+ // No sounds yet
for (i=0; i<NUM_CHANNELS; ++i)
{
- channels_playing[i] = sfx_None;
+ channels_playing[i] = NULL;
}
if (SDL_Init(SDL_INIT_AUDIO) < 0)
@@ -796,10 +805,12 @@ static boolean I_SDL_InitSound(void)
if (Mix_OpenAudio(snd_samplerate, AUDIO_S16SYS, 2, GetSliceSize()) < 0)
{
- fprintf(stderr, "Error initializing SDL_mixer: %s\n", Mix_GetError());
+ 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
@@ -811,7 +822,7 @@ static boolean I_SDL_InitSound(void)
use_libsamplerate);
}
- I_PrecacheSounds_SRC();
+ ExpandSoundData = ExpandSoundData_SRC;
}
#else
if (use_libsamplerate != 0)
@@ -879,5 +890,6 @@ sound_module_t sound_sdl_module =
I_SDL_StartSound,
I_SDL_StopSound,
I_SDL_SoundIsPlaying,
+ I_SDL_PrecacheSounds,
};
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-29 05:26:21 +0000