diff options
Diffstat (limited to 'src/i_sdlsound.c')
| -rw-r--r-- | src/i_sdlsound.c | 72 |
1 files changed, 68 insertions, 4 deletions
diff --git a/src/i_sdlsound.c b/src/i_sdlsound.c index 0b3f8aa3..7deb683d 100644 --- a/src/i_sdlsound.c +++ b/src/i_sdlsound.c @@ -25,7 +25,6 @@ // //----------------------------------------------------------------------------- - #include "config.h" #include <stdio.h> @@ -42,6 +41,7 @@ #include "deh_str.h" #include "i_sound.h" #include "i_system.h" +#include "i_swap.h" #include "m_argv.h" #include "w_wad.h" #include "z_zone.h" @@ -49,6 +49,7 @@ #include "doomtype.h" #define LOW_PASS_FILTER +//#define DEBUG_DUMP_WAVS #define MAX_SOUND_SLICE_TIME 70 /* ms */ #define NUM_CHANNELS 16 @@ -288,6 +289,56 @@ static boolean ConvertibleRatio(int freq1, int freq2) } } +#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. // Returns number of clipped samples (always 0). @@ -313,7 +364,7 @@ static void ExpandSoundData_SDL(sfxinfo_t *sfxinfo, chunk = AllocateChunk(sfxinfo, expanded_length); // If we can, use the standard / optimized SDL conversion routines. - + if (samplerate <= mixer_freq && ConvertibleRatio(samplerate, mixer_freq) && SDL_BuildAudioCVT(&convertor, @@ -379,9 +430,12 @@ static void ExpandSoundData_SDL(sfxinfo_t *sfxinfo, rc = 1.0f / (3.14f * samplerate); alpha = dt / (rc + dt); - for (i=1; i<expanded_length; ++i) + // 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-1]); + expanded[i] = (Sint16) (alpha * expanded[i] + + (1 - alpha) * expanded[i-2]); } } #endif /* #ifdef LOW_PASS_FILTER */ @@ -432,6 +486,16 @@ static boolean CacheSFX(sfxinfo_t *sfxinfo) ExpandSoundData(sfxinfo, data + 8, samplerate, length); +#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); |