X-Git-Url: http://git.xonotic.org/?a=blobdiff_plain;f=snd_mem.c;h=11b0a9d9a9a22a394497c627ff8293a9cb8e2824;hb=94a5be97432b0a538c0200cd0c837253b65633bf;hp=5e1ea8f94c0a9bcad721fe72f4132d53883f808e;hpb=50179cc6a7fa538adf02e37077776a556e6ba63e;p=xonotic%2Fdarkplaces.git diff --git a/snd_mem.c b/snd_mem.c index 5e1ea8f9..11b0a9d9 100644 --- a/snd_mem.c +++ b/snd_mem.c @@ -17,549 +17,374 @@ along with this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ -// snd_mem.c: sound caching + #include "quakedef.h" +#include "snd_main.h" #include "snd_ogg.h" +#include "snd_wav.h" +#include "snd_modplug.h" + +unsigned char resampling_buffer [48000 * 2 * 2]; /* -================ -ResampleSfx -================ +==================== +Snd_CreateRingBuffer + +If "buffer" is NULL, the function allocates one buffer of "sampleframes" sample frames itself +(if "sampleframes" is 0, the function chooses the size). +==================== */ -size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t* in_format, qbyte *out_data, const char* sfxname) +snd_ringbuffer_t *Snd_CreateRingBuffer (const snd_format_t* format, unsigned int sampleframes, void* buffer) { - int samplefrac, fracstep; - size_t i, srcsample, srclength, outcount; + snd_ringbuffer_t *ringbuffer; - // this is usually 0.5 (128), 1 (256), or 2 (512) - fracstep = ((double) in_format->speed / (double) shm->format.speed) * 256.0; + // If the caller provides a buffer, it must give us its size + if (sampleframes == 0 && buffer != NULL) + return NULL; - srclength = in_length * in_format->channels; + ringbuffer = (snd_ringbuffer_t*)Mem_Alloc(snd_mempool, sizeof (*ringbuffer)); + memset(ringbuffer, 0, sizeof(*ringbuffer)); + memcpy(&ringbuffer->format, format, sizeof(ringbuffer->format)); - outcount = (double) in_length * (double) shm->format.speed / (double) in_format->speed; - Con_DPrintf("ResampleSfx: resampling sound \"%s\" from %dHz to %dHz (%d samples to %d samples)\n", - sfxname, in_format->speed, shm->format.speed, in_length, outcount); + // If we haven't been given a buffer + if (buffer == NULL) + { + unsigned int maxframes; + size_t memsize; -// resample / decimate to the current source rate + if (sampleframes == 0) + maxframes = (format->speed + 1) / 2; // Make the sound buffer large enough for containing 0.5 sec of sound + else + maxframes = sampleframes; - if (fracstep == 256) - { - // fast case for direct transfer - if (in_format->width == 1) // 8bit - for (i = 0;i < srclength;i++) - ((signed char *)out_data)[i] = ((unsigned char *)in_data)[i] - 128; - else //if (sb->width == 2) // 16bit - for (i = 0;i < srclength;i++) - ((short *)out_data)[i] = ((short *)in_data)[i]; + memsize = maxframes * format->width * format->channels; + ringbuffer->ring = (unsigned char *) Mem_Alloc(snd_mempool, memsize); + ringbuffer->maxframes = maxframes; } else { - // general case - samplefrac = 0; - if ((fracstep & 255) == 0) // skipping points on perfect multiple - { - srcsample = 0; - fracstep >>= 8; - if (in_format->width == 2) - { - short *out = (short*)out_data; - const short *in = (const short*)in_data; - if (in_format->channels == 2) // LordHavoc: stereo sound support - { - fracstep <<= 1; - for (i=0 ; ichannels == 2) - { - fracstep <<= 1; - for (i=0 ; iwidth == 2) - { - short *out = (short*)out_data; - const short *in = (const short*)in_data; - if (in_format->channels == 2) - { - for (i=0 ; i> 8) << 1; - a = in[srcsample ]; - if (srcsample+2 >= srclength) - b = 0; - else - b = in[srcsample+2]; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (short) sample; - a = in[srcsample+1]; - if (srcsample+2 >= srclength) - b = 0; - else - b = in[srcsample+3]; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (short) sample; - samplefrac += fracstep; - } - } - else - { - for (i=0 ; i> 8; - a = in[srcsample ]; - if (srcsample+1 >= srclength) - b = 0; - else - b = in[srcsample+1]; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (short) sample; - samplefrac += fracstep; - } - } - } - else - { - signed char *out = out_data; - const unsigned char *in = in_data; - if (in_format->channels == 2) - { - for (i=0 ; i> 8) << 1; - a = (int) in[srcsample ] - 128; - if (srcsample+2 >= srclength) - b = 0; - else - b = (int) in[srcsample+2] - 128; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (signed char) sample; - a = (int) in[srcsample+1] - 128; - if (srcsample+2 >= srclength) - b = 0; - else - b = (int) in[srcsample+3] - 128; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (signed char) sample; - samplefrac += fracstep; - } - } - else - { - for (i=0 ; i> 8; - a = (int) in[srcsample ] - 128; - if (srcsample+1 >= srclength) - b = 0; - else - b = (int) in[srcsample+1] - 128; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (signed char) sample; - samplefrac += fracstep; - } - } - } - } + ringbuffer->ring = (unsigned char *) buffer; + ringbuffer->maxframes = sampleframes; } - return outcount; + return ringbuffer; } -//============================================================================= - -wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength); /* ==================== -WAV_FetchSound +Snd_CreateSndBuffer ==================== */ -static const sfxbuffer_t* WAV_FetchSound (channel_t* ch, unsigned int start, unsigned int nbsamples) +snd_buffer_t *Snd_CreateSndBuffer (const unsigned char *samples, unsigned int sampleframes, const snd_format_t* in_format, unsigned int sb_speed) { - return ch->sfx->fetcher_data; -} - - -snd_fetcher_t wav_fetcher = { WAV_FetchSound, NULL }; + size_t newsampleframes, memsize; + snd_buffer_t* sb; + newsampleframes = (size_t) ceil((double)sampleframes * (double)sb_speed / (double)in_format->speed); -/* -============== -S_LoadWavFile -============== -*/ -qboolean S_LoadWavFile (const char *filename, sfx_t *s) -{ - qbyte *data; - wavinfo_t info; - int len; - sfxbuffer_t* sb; + memsize = newsampleframes * in_format->channels * in_format->width; + memsize += sizeof (*sb) - sizeof (sb->samples); - Mem_FreePool (&s->mempool); - s->mempool = Mem_AllocPool(s->name); + sb = (snd_buffer_t*)Mem_Alloc (snd_mempool, memsize); + sb->format.channels = in_format->channels; + sb->format.width = in_format->width; + sb->format.speed = sb_speed; + sb->maxframes = newsampleframes; + sb->nbframes = 0; - // Load the file - data = FS_LoadFile(filename, s->mempool, false); - if (!data) + if (!Snd_AppendToSndBuffer (sb, samples, sampleframes, in_format)) { - Mem_FreePool (&s->mempool); - return false; + Mem_Free (sb); + return NULL; } - // Don't try to load it if it's not a WAV file - if (memcmp (data, "RIFF", 4) || memcmp (data + 8, "WAVE", 4)) - { - Mem_FreePool (&s->mempool); - return false; - } + return sb; +} + + +/* +==================== +Snd_AppendToSndBuffer +==================== +*/ +qboolean Snd_AppendToSndBuffer (snd_buffer_t* sb, const unsigned char *samples, unsigned int sampleframes, const snd_format_t* format) +{ + size_t srclength, outcount; + unsigned char *out_data; - Con_DPrintf ("Loading WAV file \"%s\"\n", filename); + //Con_DPrintf("ResampleSfx: %d samples @ %dHz -> %d samples @ %dHz\n", + // sampleframes, format->speed, outcount, sb->format.speed); - info = GetWavinfo (s->name, data, fs_filesize); - // Stereo sounds are allowed (intended for music) - if (info.channels < 1 || info.channels > 2) + // If the formats are incompatible + if (sb->format.channels != format->channels || sb->format.width != format->width) { - Con_Printf("%s has an unsupported number of channels (%i)\n",s->name, info.channels); - Mem_FreePool (&s->mempool); + Con_Print("AppendToSndBuffer: incompatible sound formats!\n"); return false; } - // calculate resampled length - len = (int) ((double) info.samples * (double) shm->format.speed / (double) info.rate); - len = len * info.width * info.channels; + outcount = (size_t) ((double)sampleframes * (double)sb->format.speed / (double)format->speed); - sb = Mem_Alloc (s->mempool, len + sizeof (*sb) - sizeof (sb->data)); - if (sb == NULL) + // If the sound buffer is too short + if (outcount > sb->maxframes - sb->nbframes) { - Con_Printf("failed to allocate memory for sound \"%s\"\n", s->name); - Mem_FreePool(&s->mempool); + Con_Print("AppendToSndBuffer: sound buffer too short!\n"); return false; } - s->fetcher = &wav_fetcher; - s->fetcher_data = sb; - s->format.speed = info.rate; - s->format.width = info.width; - s->format.channels = info.channels; - if (info.loopstart < 0) - s->loopstart = -1; - else - s->loopstart = (double)info.loopstart * (double)shm->format.speed / (double)s->format.speed; + out_data = &sb->samples[sb->nbframes * sb->format.width * sb->format.channels]; + srclength = sampleframes * format->channels; -#if BYTE_ORDER != LITTLE_ENDIAN - // We must convert the WAV data from little endian - // to the machine endianess before resampling it - if (info.width == 2) + // Trivial case (direct transfer) + if (format->speed == sb->format.speed) { - int i; - short* ptr; + if (format->width == 1) + { + size_t i; - len = info.samples * info.channels; - ptr = (short*)(data + info.dataofs); - for (i = 0; i < len; i++) - ptr[i] = LittleShort (ptr[i]); + for (i = 0; i < srclength; i++) + ((signed char*)out_data)[i] = samples[i] - 128; + } + else // if (format->width == 2) + memcpy (out_data, samples, srclength * format->width); } -#endif - - sb->length = ResampleSfx (data + info.dataofs, info.samples, &s->format, sb->data, s->name); - s->format.speed = shm->format.speed; - s->total_length = sb->length; - sb->offset = 0; - - Mem_Free (data); - return true; -} - -/* -============== -S_LoadSound -============== -*/ -qboolean S_LoadSound (sfx_t *s, int complain) -{ - char namebuffer[MAX_QPATH]; - size_t len; - qboolean modified_name = false; - - // see if still in memory - if (!shm || !shm->format.speed) - return false; - if (s->fetcher != NULL) + // General case (linear interpolation with a fixed-point fractional + // step, 18-bit integer part and 14-bit fractional part) + // Can handle up to 2^18 (262144) samples per second (> 96KHz stereo) +# define FRACTIONAL_BITS 14 +# define FRACTIONAL_MASK ((1 << FRACTIONAL_BITS) - 1) +# define INTEGER_BITS (sizeof(samplefrac)*8 - FRACTIONAL_BITS) + else { - if (s->format.speed != shm->format.speed) - Sys_Error ("S_LoadSound: sound %s hasn't been resampled (%uHz instead of %uHz)", s->name); - return true; - } + const unsigned int fracstep = (unsigned int)((double)format->speed / sb->format.speed * (1 << FRACTIONAL_BITS)); + size_t remain_in = srclength, total_out = 0; + unsigned int samplefrac; + const unsigned char *in_ptr = samples; + unsigned char *out_ptr = out_data; + + // Check that we can handle one second of that sound + if (format->speed * format->channels > (1 << INTEGER_BITS)) + { + Con_Printf ("ResampleSfx: sound quality too high for resampling (%uHz, %u channel(s))\n", + format->speed, format->channels); + return 0; + } - len = snprintf (namebuffer, sizeof (namebuffer), "sound/%s", s->name); - if (len >= sizeof (namebuffer)) - return false; + // We work 1 sec at a time to make sure we don't accumulate any + // significant error when adding "fracstep" over several seconds, and + // also to be able to handle very long sounds. + while (total_out < outcount) + { + size_t tmpcount, interpolation_limit, i, j; + unsigned int srcsample; - // Try to load it as a WAV file - if (S_LoadWavFile (namebuffer, s)) - return true; + samplefrac = 0; - // Else, try to load it as an Ogg Vorbis file - if (!strcasecmp (namebuffer + len - 4, ".wav")) - { - strcpy (namebuffer + len - 3, "ogg"); - modified_name = true; - } - if (OGG_LoadVorbisFile (namebuffer, s)) - return true; + // If more than 1 sec of sound remains to be converted + if (outcount - total_out > sb->format.speed) + { + tmpcount = sb->format.speed; + interpolation_limit = tmpcount; // all samples can be interpolated + } + else + { + tmpcount = outcount - total_out; + interpolation_limit = (int)ceil((double)(((remain_in / format->channels) - 1) << FRACTIONAL_BITS) / fracstep); + if (interpolation_limit > tmpcount) + interpolation_limit = tmpcount; + } - // Can't load the sound! - if (!complain) - s->flags |= SFXFLAG_SILENTLYMISSING; - else - s->flags &= ~SFXFLAG_SILENTLYMISSING; - if (complain) - { - if (modified_name) - strcpy (namebuffer + len - 3, "wav"); - Con_Printf("Couldn't load %s\n", namebuffer); - } - return false; -} + // 16 bit samples + if (format->width == 2) + { + const short* in_ptr_short; -void S_UnloadSound(sfx_t *s) -{ - if (s->fetcher != NULL) - { - unsigned int i; + // Interpolated part + for (i = 0; i < interpolation_limit; i++) + { + srcsample = (samplefrac >> FRACTIONAL_BITS) * format->channels; + in_ptr_short = &((const short*)in_ptr)[srcsample]; - s->fetcher = NULL; - s->fetcher_data = NULL; - Mem_FreePool(&s->mempool); + for (j = 0; j < format->channels; j++) + { + int a, b; - // At this point, some per-channel data pointers may point to freed zones. - // Practically, it shouldn't be a problem; but it's wrong, so we fix that - for (i = 0; i < total_channels ; i++) - if (channels[i].sfx == s) - channels[i].fetcher_data = NULL; - } -} + a = *in_ptr_short; + b = *(in_ptr_short + format->channels); + *((short*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a; + in_ptr_short++; + out_ptr += sizeof (short); + } -/* -=============================================================================== + samplefrac += fracstep; + } -WAV loading + // Non-interpolated part + for (/* nothing */; i < tmpcount; i++) + { + srcsample = (samplefrac >> FRACTIONAL_BITS) * format->channels; + in_ptr_short = &((const short*)in_ptr)[srcsample]; -=============================================================================== -*/ + for (j = 0; j < format->channels; j++) + { + *((short*)out_ptr) = *in_ptr_short; + in_ptr_short++; + out_ptr += sizeof (short); + } -static qbyte *data_p; -static qbyte *iff_end; -static qbyte *last_chunk; -static qbyte *iff_data; -static int iff_chunk_len; + samplefrac += fracstep; + } + } + // 8 bit samples + else // if (format->width == 1) + { + const unsigned char* in_ptr_byte; + // Convert up to 1 sec of sound + for (i = 0; i < interpolation_limit; i++) + { + srcsample = (samplefrac >> FRACTIONAL_BITS) * format->channels; + in_ptr_byte = &((const unsigned char*)in_ptr)[srcsample]; -short GetLittleShort(void) -{ - short val; + for (j = 0; j < format->channels; j++) + { + int a, b; - val = BuffLittleShort (data_p); - data_p += 2; + a = *in_ptr_byte - 128; + b = *(in_ptr_byte + format->channels) - 128; + *((signed char*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a; - return val; -} + in_ptr_byte++; + out_ptr += sizeof (signed char); + } -int GetLittleLong(void) -{ - int val = 0; + samplefrac += fracstep; + } - val = BuffLittleLong (data_p); - data_p += 4; + // Non-interpolated part + for (/* nothing */; i < tmpcount; i++) + { + srcsample = (samplefrac >> FRACTIONAL_BITS) * format->channels; + in_ptr_byte = &((const unsigned char*)in_ptr)[srcsample]; - return val; -} + for (j = 0; j < format->channels; j++) + { + *((signed char*)out_ptr) = *in_ptr_byte - 128; -void FindNextChunk(char *name) -{ - while (1) - { - data_p=last_chunk; + in_ptr_byte++; + out_ptr += sizeof (signed char); + } - if (data_p >= iff_end) - { // didn't find the chunk - data_p = NULL; - return; - } + samplefrac += fracstep; + } + } - data_p += 4; - iff_chunk_len = GetLittleLong(); - if (iff_chunk_len < 0) - { - data_p = NULL; - return; + // Update the counters and the buffer position + remain_in -= format->speed * format->channels; + in_ptr += format->speed * format->channels * format->width; + total_out += tmpcount; } - data_p -= 8; - last_chunk = data_p + 8 + ( (iff_chunk_len + 1) & ~1 ); - if (!strncmp(data_p, name, 4)) - return; } -} -void FindChunk(char *name) -{ - last_chunk = iff_data; - FindNextChunk (name); + sb->nbframes += outcount; + return true; } -void DumpChunks(void) -{ - char str[5]; - - str[4] = 0; - data_p=iff_data; - do - { - memcpy (str, data_p, 4); - data_p += 4; - iff_chunk_len = GetLittleLong(); - Con_Printf("0x%x : %s (%d)\n", (int)(data_p - 4), str, iff_chunk_len); - data_p += (iff_chunk_len + 1) & ~1; - } while (data_p < iff_end); -} +//============================================================================= /* -============ -GetWavinfo -============ +============== +S_LoadSound +============== */ -wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength) +qboolean S_LoadSound (sfx_t *sfx, qboolean complain) { - wavinfo_t info; - int i; - int format; - int samples; - - memset (&info, 0, sizeof(info)); + char namebuffer[MAX_QPATH + 16]; + size_t len; - if (!wav) - return info; + // See if already loaded + if (sfx->fetcher != NULL) + return true; - iff_data = wav; - iff_end = wav + wavlength; + // If we weren't able to load it previously, no need to retry + // Note: S_PrecacheSound clears this flag to cause a retry + if (sfx->flags & SFXFLAG_FILEMISSING) + return false; - // find "RIFF" chunk - FindChunk("RIFF"); - if (!(data_p && !strncmp(data_p+8, "WAVE", 4))) - { - Con_Print("Missing RIFF/WAVE chunks\n"); - return info; - } + // No sound? + if (snd_renderbuffer == NULL) + return false; - // get "fmt " chunk - iff_data = data_p + 12; - //DumpChunks (); + // Initialize volume peak to 0; if ReplayGain is supported, the loader will change this away + sfx->volume_peak = 0.0; - FindChunk("fmt "); - if (!data_p) - { - Con_Print("Missing fmt chunk\n"); - return info; - } - data_p += 8; - format = GetLittleShort(); - if (format != 1) - { - Con_Print("Microsoft PCM format only\n"); - return info; - } + if (developer_loading.integer) + Con_Printf("loading sound %s\n", sfx->name); - info.channels = GetLittleShort(); - info.rate = GetLittleLong(); - data_p += 4+2; - info.width = GetLittleShort() / 8; + SCR_PushLoadingScreen(true, sfx->name, 1); - // get cue chunk - FindChunk("cue "); - if (data_p) + // LordHavoc: if the sound filename does not begin with sound/, try adding it + if (strncasecmp(sfx->name, "sound/", 6)) { - data_p += 32; - info.loopstart = GetLittleLong(); - - // if the next chunk is a LIST chunk, look for a cue length marker - FindNextChunk ("LIST"); - if (data_p) + dpsnprintf (namebuffer, sizeof(namebuffer), "sound/%s", sfx->name); + len = strlen(namebuffer); + if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav")) { - if (!strncmp (data_p + 28, "mark", 4)) - { // this is not a proper parse, but it works with cooledit... - data_p += 24; - i = GetLittleLong (); // samples in loop - info.samples = info.loopstart + i; - } + if (S_LoadWavFile (namebuffer, sfx)) + goto loaded; + memcpy (namebuffer + len - 3, "ogg", 4); + } + if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".ogg")) + { + if (OGG_LoadVorbisFile (namebuffer, sfx)) + goto loaded; + } + else + { + if (ModPlug_LoadModPlugFile (namebuffer, sfx)) + goto loaded; } } - else - info.loopstart = -1; - // find data chunk - FindChunk("data"); - if (!data_p) + // LordHavoc: then try without the added sound/ as wav and ogg + dpsnprintf (namebuffer, sizeof(namebuffer), "%s", sfx->name); + len = strlen(namebuffer); + // request foo.wav: tries foo.wav, then foo.ogg + // request foo.ogg: tries foo.ogg only + // request foo.mod: tries foo.mod only + if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav")) { - Con_Print("Missing data chunk\n"); - return info; + if (S_LoadWavFile (namebuffer, sfx)) + goto loaded; + memcpy (namebuffer + len - 3, "ogg", 4); } - - data_p += 4; - samples = GetLittleLong () / info.width / info.channels; - - if (info.samples) + if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".ogg")) { - if (samples < info.samples) - Host_Error ("Sound %s has a bad loop length", name); + if (OGG_LoadVorbisFile (namebuffer, sfx)) + goto loaded; } else - info.samples = samples; + { + if (ModPlug_LoadModPlugFile (namebuffer, sfx)) + goto loaded; + } - info.dataofs = data_p - wav; + // Can't load the sound! + sfx->flags |= SFXFLAG_FILEMISSING; + if (complain) + Con_DPrintf("failed to load sound \"%s\"\n", sfx->name); - return info; -} + SCR_PopLoadingScreen(false); + return false; +loaded: + SCR_PopLoadingScreen(false); + return true; +}