ref: 1fcb507e98d51bbf6c03967964d02f3c621daf6c
dir: /src/support/cmixer.c/
/*
Derivative work of cmixer by rxi (https://github.com/rxi/cmixer)
Copyright (c) 2017 rxi
Copyright (c) 2023 Iliyas Jorio
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to
deal in the Software without restriction, including without limitation the
rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
sell copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in
all copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS
IN THE SOFTWARE.
*/
#include "cmixer.h"
#include "ibxm.h"
#include <SDL.h>
#include <stdint.h>
#include <stdbool.h>
#include <limits.h>
#define MAX_CONCURRENT_VOICES 8
#define BUFFER_SIZE 512
#define CM_DIE(message) \
do { \
char buf[256]; \
SDL_snprintf(buf, sizeof(buf), "%s:%d: %s", __func__, __LINE__, (message)); \
SDL_ShowSimpleMessageBox(SDL_MESSAGEBOX_ERROR, "cmixer", buf, NULL); \
abort(); \
} while(0)
#define CM_ASSERT(assertion, message) do { if (!(assertion)) CM_DIE(message); } while(0)
#define FX_BITS (12L)
#define FX_UNIT (1L << FX_BITS)
#define FX_MASK (FX_UNIT - 1)
#define BUFFER_MASK (BUFFER_SIZE - 1)
enum
{
PCMFORMAT_NULL = 0x00,
PCMFORMAT_1CH_8 = 0x11,
PCMFORMAT_2CH_8 = 0x21,
PCMFORMAT_1CH_LE16 = 0x12,
PCMFORMAT_2CH_LE16 = 0x22,
PCMFORMAT_1CH_BE16 = PCMFORMAT_1CH_LE16 | 0x80,
PCMFORMAT_2CH_BE16 = PCMFORMAT_2CH_LE16 | 0x80,
};
struct CMWavStream
{
uint8_t pcmformat;
int idx;
char* data;
size_t dataLength;
bool ownData;
uint32_t cookie;
};
struct CMModStream
{
struct module* module;
struct replay* replay;
char* moduleFileMemory;
int* replayBuffer;
int replayBufferOffset;
int replayBufferSamples;
double playbackSpeedMult;
uint32_t cookie;
};
struct CMVoice
{
int16_t pcmbuf[BUFFER_SIZE]; // Internal buffer with raw stereo PCM
int sampleRate; // Stream's native samplerate
int sampleCount; // Stream's length in frames
int sustainOffset; // Offset of the sustain loop in frames
int end; // End index for the current play-through
int state; // Current state (playing|paused|stopped)
int64_t position; // Current playhead position (fixed point)
int lgain, rgain; // Left and right gain (fixed point)
int rate; // Playback rate (fixed point)
int nextfill; // Next frame idx where the buffer needs to be filled
bool loop; // Whether the voice will loop when `end` is reached
bool rewind; // Whether the voice will rewind before playing
bool active; // Whether the voice is part of `voices` list
bool interpolate; // Interpolated resampling when played back at a non-native rate
double gain; // Gain set by `cm_set_gain()`
double pan; // Pan set by `cm_set_pan()`
struct
{
void (*fillBuffer)(struct CMVoice* voice, int16_t* into, int len);
void (*completed)(struct CMVoice* voice);
void (*rewind)(struct CMVoice* voice);
void (*free)(struct CMVoice* voice);
} callbacks;
uint32_t cookie;
union
{
struct CMWavStream wav;
struct CMModStream mod;
};
};
typedef struct CMVoice CMVoice;
typedef struct CMWavStream CMWavStream;
typedef struct CMModStream CMModStream;
static void CMVoice_RemoveFromMixer(CMVoice* voice);
static void CMVoice_AddToMix(CMVoice* voice, int len, int32_t* dst);
static inline CMWavStream* CMWavStream_Check(CMVoice* voice);
static void StreamWav(CMVoice* voice, int16_t* output, int length);
static void RewindWav(CMVoice* voice);
static void FreeWav(CMVoice* voice);
static inline CMModStream* CMModStream_Check(CMVoice* voice);
static void StreamMod(CMVoice* voice, int16_t* output, int length);
static void FreeMod(CMVoice* voice);
//-----------------------------------------------------------------------------
// Utilities
static inline int DoubleToFixed(double f)
{
return (int) (f * FX_UNIT);
}
static inline double FixedToDouble(int f)
{
return (double) f / FX_UNIT;
}
static inline int FixedLerp(int a, int b, int p)
{
return a + (((b - a) * p) >> FX_BITS);
}
static inline int16_t UnpackI16BE(const void* data)
{
#if __BIG_ENDIAN__
// no-op on big-endian systems
return *(const uint16_t*) data;
#else
const uint8_t* p = (uint8_t*) data;
return ( p[0] << 8 )
| ( p[1] );
#endif
}
static inline int16_t UnpackI16LE(const void* data)
{
#if __BIG_ENDIAN__
const uint8_t* p = (uint8_t*) data;
return ( p[0] )
| ( p[1] << 8 );
#else
// no-op on little-endian systems
return *(const uint16_t*) data;
#endif
}
static inline int MinInt(int a, int b) { return a < b ? a : b; }
static inline int MaxInt(int a, int b) { return a < b ? b : a; }
static inline int ClampInt(int x, int a, int b) { return x < a ? a : x > b ? b : x; }
static inline double ClampDouble(double x, double a, double b) { return x < a ? a : x > b ? b : x; }
static char* LoadFile(const char* filename, size_t* outSize)
{
SDL_RWops* ifs = SDL_RWFromFile(filename, "rb");
if (!ifs)
return NULL;
SDL_RWseek(ifs, 0, RW_SEEK_END);
long filesize = SDL_RWtell(ifs);
SDL_RWseek(ifs, 0, RW_SEEK_SET);
void* bytes = SDL_malloc(filesize);
SDL_RWread(ifs, bytes, 1, filesize);
SDL_RWclose(ifs);
if (outSize)
*outSize = filesize;
return (char*)bytes;
}
static uint8_t BuildPCMFormat(int bitdepth, int channels, bool bigEndian)
{
return ((!!bigEndian) << 7)
| (channels << 4)
| (bitdepth / 8);
}
//-----------------------------------------------------------------------------
// Global mixer
static struct Mixer
{
SDL_mutex* sdlAudioMutex;
CMVoice* voices[MAX_CONCURRENT_VOICES]; // List of active (playing) voices
int32_t pcmmixbuf[BUFFER_SIZE]; // Internal master buffer
int samplerate; // Master samplerate
int gain; // Master gain (fixed point)
} gMixer;
static void Mixer_Init(struct Mixer* mixer, int samplerate);
static void Mixer_Process(struct Mixer* mixer, int16_t* dst, int len);
static void Mixer_Lock(struct Mixer* mixer);
static void Mixer_Unlock(struct Mixer* mixer);
static void Mixer_SetMasterGain(struct Mixer* mixer, double newGain);
//-----------------------------------------------------------------------------
// Global init/shutdown
static bool sdlAudioSubSystemInited = false;
static SDL_AudioDeviceID sdlDeviceID = 0;
static void MixerCallback(void* udata, Uint8* stream, int size)
{
struct Mixer* mixer = (struct Mixer*) udata;
Mixer_Process(mixer, (int16_t*) stream, size / 2);
}
void cmixer_InitWithSDL(void)
{
CM_ASSERT(!sdlAudioSubSystemInited, "SDL audio subsystem already inited");
if (0 != SDL_InitSubSystem(SDL_INIT_AUDIO))
CM_DIE(SDL_GetError());
sdlAudioSubSystemInited = true;
// Init SDL audio
SDL_AudioSpec fmt =
{
.freq = 44100,
.format = AUDIO_S16SYS,
.channels = 2,
.samples = 1024,
.callback = MixerCallback,
.userdata = &gMixer,
};
SDL_AudioSpec got;
sdlDeviceID = SDL_OpenAudioDevice(NULL, 0, &fmt, &got, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE);
CM_ASSERT(sdlDeviceID, SDL_GetError());
// Init library
Mixer_Init(&gMixer, got.freq);
Mixer_SetMasterGain(&gMixer, 0.5);
// Start audio
SDL_PauseAudioDevice(sdlDeviceID, 0);
}
void cmixer_ShutdownWithSDL()
{
if (sdlDeviceID)
{
SDL_CloseAudioDevice(sdlDeviceID);
sdlDeviceID = 0;
}
if (gMixer.sdlAudioMutex)
{
SDL_DestroyMutex(gMixer.sdlAudioMutex);
gMixer.sdlAudioMutex = NULL;
}
if (sdlAudioSubSystemInited)
{
SDL_QuitSubSystem(SDL_INIT_AUDIO);
sdlAudioSubSystemInited = false;
}
}
double cmixer_GetMasterGain()
{
return FixedToDouble(gMixer.gain);
}
void cmixer_SetMasterGain(double newGain)
{
Mixer_SetMasterGain(&gMixer, newGain);
}
//-----------------------------------------------------------------------------
// Global mixer impl
static void Mixer_Init(struct Mixer* mixer, int newSamplerate)
{
SDL_memset(mixer, 0, sizeof(mixer));
mixer->sdlAudioMutex = SDL_CreateMutex();
mixer->samplerate = newSamplerate;
mixer->gain = FX_UNIT;
}
static void Mixer_Lock(struct Mixer* mixer)
{
SDL_LockMutex(mixer->sdlAudioMutex);
}
static void Mixer_Unlock(struct Mixer* mixer)
{
SDL_UnlockMutex(mixer->sdlAudioMutex);
}
static void Mixer_SetMasterGain(struct Mixer* mixer, double newGain)
{
if (newGain < 0)
newGain = 0;
mixer->gain = DoubleToFixed(newGain);
}
static int Mixer_AddVoice(struct Mixer* mixer, CMVoice* voice)
{
CM_ASSERT(voice->callbacks.fillBuffer, "fill buffer callback not set");
// Look for a free slot
for (int i = 0; i < MAX_CONCURRENT_VOICES; i++)
{
CM_ASSERT(mixer->voices[i] != voice, "voice added twice to mixer");
if (!mixer->voices[i])
{
mixer->voices[i] = voice;
return i;
}
}
return -1;
}
static void Mixer_RemoveVoice(struct Mixer* mixer, CMVoice* voice)
{
for (int i = 0; i < MAX_CONCURRENT_VOICES; i++)
{
if (mixer->voices[i] == voice)
{
mixer->voices[i] = NULL;
break;
}
}
}
static void Mixer_Process(struct Mixer* mixer, int16_t* dst, int len)
{
// Process in chunks of BUFFER_SIZE if `len` is larger than BUFFER_SIZE
while (len > BUFFER_SIZE)
{
Mixer_Process(mixer, dst, BUFFER_SIZE);
dst += BUFFER_SIZE;
len -= BUFFER_SIZE;
}
// Zeroset internal buffer
SDL_memset(mixer->pcmmixbuf, 0, len * sizeof(mixer->pcmmixbuf[0]));
// Process active voices
Mixer_Lock(mixer);
for (int i = 0; i < MAX_CONCURRENT_VOICES; i++)
{
CMVoice* voice = mixer->voices[i];
if (!voice)
continue;
CMVoice_AddToMix(voice, len, mixer->pcmmixbuf);
// Remove voice from list if it is no longer playing
if (voice->state != CM_STATE_PLAYING)
{
voice->active = false;
mixer->voices[i] = NULL;
}
}
Mixer_Unlock(mixer);
// Copy internal buffer to destination and clip
for (int i = 0; i < len; i++)
{
int x = (mixer->pcmmixbuf[i] * mixer->gain) >> FX_BITS;
dst[i] = ClampInt(x, -32768, 32767);
}
}
//-----------------------------------------------------------------------------
// Voice implementation
static inline CMVoice* CMVoice_Check(void* ptr)
{
CMVoice* voice = (CMVoice*) ptr;
CM_ASSERT(voice->cookie == 'VOIX', "VOIX cookie not found");
return voice;
}
static CMVoice* CMVoice_New(int sampleRate, int sampleCount)
{
CMVoice* voice = SDL_calloc(1, sizeof(CMVoice));
voice->cookie = 'VOIX';
voice->sampleRate = 0;
voice->sampleCount = 0;
voice->end = 0;
voice->state = CM_STATE_STOPPED;
voice->position = 0;
voice->lgain = 0;
voice->rgain = 0;
voice->rate = 0;
voice->nextfill = 0;
voice->loop = false;
voice->rewind = true;
voice->interpolate = false;
voice->gain = 0;
voice->pan = 0;
voice->active = false;
voice->sampleRate = sampleRate;
voice->sampleCount = sampleCount;
voice->sustainOffset = 0;
CMVoice_SetGain(voice, 1);
CMVoice_SetPan(voice, 0);
CMVoice_SetPitch(voice, 1);
CMVoice_SetLoop(voice, false);
CMVoice_Stop(voice);
return voice;
}
void CMVoice_Free(CMVoice* voice)
{
CMVoice_Check(voice);
CMVoice_RemoveFromMixer(voice);
if (voice->callbacks.free)
voice->callbacks.free(voice);
voice->cookie = 'DEAD';
SDL_free(voice);
}
static void CMVoice_RemoveFromMixer(CMVoice* voice)
{
CMVoice_Check(voice);
Mixer_Lock(&gMixer);
if (voice->active)
{
Mixer_RemoveVoice(&gMixer, voice);
voice->active = false;
}
Mixer_Unlock(&gMixer);
}
void CMVoice_Rewind(CMVoice* voice)
{
if (voice->callbacks.rewind)
voice->callbacks.rewind(voice);
voice->position = 0;
voice->rewind = false;
voice->end = voice->sampleCount;
voice->nextfill = 0;
}
static void CMVoice_AddToMix(CMVoice* voice, int len, int32_t* dst)
{
CMVoice_Check(voice); // check pointer validity
// Do rewind if flag is set
if (voice->rewind)
{
CMVoice_Rewind(voice);
}
// Don't process if not playing
if (voice->state != CM_STATE_PLAYING)
{
return;
}
// Process audio
while (len > 0)
{
// Get current position frame
int frame = (int) (voice->position >> FX_BITS);
// Fill buffer if required
if (frame + 3 >= voice->nextfill)
{
int fillOffset = (voice->nextfill * 2) & BUFFER_MASK;
int fillLength = BUFFER_SIZE / 2;
voice->callbacks.fillBuffer(voice, voice->pcmbuf + fillOffset, fillLength);
voice->nextfill += BUFFER_SIZE / 4;
}
// Handle reaching the end of the playthrough
if (frame >= voice->end)
{
// As streams continuously fill the raw buffer in a loop,
// increment the end idx by one length
// and continue reading from it another playthrough
voice->end = frame + voice->sampleCount;
// Set state and stop processing if we're not set to loop
if (!voice->loop)
{
voice->state = CM_STATE_STOPPED;
if (voice->callbacks.completed)
voice->callbacks.completed(voice);
break;
}
}
// Work out how many frames we should process in the loop
int n = MinInt(voice->nextfill - 2, voice->end) - frame;
int count = (n << FX_BITS) / voice->rate;
count = MaxInt(count, 1);
count = MinInt(count, len / 2);
len -= count * 2;
// Add audio to master buffer
if (voice->rate == FX_UNIT)
{
// Add audio to buffer -- basic
n = frame * 2;
for (int i = 0; i < count; i++)
{
dst[0] += (voice->pcmbuf[(n ) & BUFFER_MASK] * voice->lgain) >> FX_BITS;
dst[1] += (voice->pcmbuf[(n + 1) & BUFFER_MASK] * voice->rgain) >> FX_BITS;
n += 2;
dst += 2;
}
voice->position += count * FX_UNIT;
}
else if (voice->interpolate)
{
// Resample audio (with linear interpolation) and add to buffer
for (int i = 0; i < count; i++)
{
n = (int) (voice->position >> FX_BITS) * 2;
int p = voice->position & FX_MASK;
int a = voice->pcmbuf[(n ) & BUFFER_MASK];
int b = voice->pcmbuf[(n + 2) & BUFFER_MASK];
dst[0] += (FixedLerp(a, b, p) * voice->lgain) >> FX_BITS;
n++;
a = voice->pcmbuf[(n ) & BUFFER_MASK];
b = voice->pcmbuf[(n + 2) & BUFFER_MASK];
dst[1] += (FixedLerp(a, b, p) * voice->rgain) >> FX_BITS;
voice->position += voice->rate;
dst += 2;
}
}
else
{
// Resample audio (without interpolation) and add to buffer
for (int i = 0; i < count; i++)
{
n = (int) (voice->position >> FX_BITS) * 2;
dst[0] += (voice->pcmbuf[(n ) & BUFFER_MASK] * voice->lgain) >> FX_BITS;
dst[1] += (voice->pcmbuf[(n + 1) & BUFFER_MASK] * voice->rgain) >> FX_BITS;
voice->position += voice->rate;
dst += 2;
}
}
}
}
double CMVoice_GetLength(const CMVoice* voice)
{
return voice->sampleCount / (double) voice->sampleRate;
}
double CMVoice_GetPosition(const CMVoice* voice)
{
return ((voice->position >> FX_BITS) % voice->sampleCount) / (double) voice->sampleRate;
}
int CMVoice_GetState(const CMVoice* voice)
{
return voice->state;
}
static void CMVoice_RecalcGains(CMVoice* voice)
{
double l = voice->gain * (voice->pan <= 0. ? 1. : 1. - voice->pan);
double r = voice->gain * (voice->pan >= 0. ? 1. : 1. + voice->pan);
voice->lgain = DoubleToFixed(l);
voice->rgain = DoubleToFixed(r);
}
void CMVoice_SetGain(CMVoice* voice, double newGain)
{
voice->gain = newGain;
CMVoice_RecalcGains(voice);
}
void CMVoice_SetPan(CMVoice* voice, double newPan)
{
voice->pan = ClampDouble(newPan, -1.0, 1.0);
CMVoice_RecalcGains(voice);
}
void CMVoice_SetPitch(CMVoice* voice, double newPitch)
{
double newRate;
if (newPitch > 0.)
{
newRate = (double)voice->sampleRate / (double) gMixer.samplerate * newPitch;
}
else
{
newRate = 0.001;
}
voice->rate = DoubleToFixed(newRate);
}
void CMVoice_SetLoop(CMVoice* voice, int newLoop)
{
voice->loop = newLoop;
}
void CMVoice_SetInterpolation(CMVoice* voice, int newInterpolation)
{
voice->interpolate = newInterpolation;
}
void CMVoice_Play(CMVoice* voice)
{
CMVoice_Check(voice); // check pointer validity
if (voice->sampleCount == 0)
{
// Don't attempt to play an empty voice as this would result
// in instant starvation when filling mixer buffer
return;
}
Mixer_Lock(&gMixer);
if (!voice->active)
{
int rc = Mixer_AddVoice(&gMixer, voice);
if (rc < 0)
{
// couldn't add voice
}
else
{
voice->state = CM_STATE_PLAYING;
voice->active = true;
}
}
Mixer_Unlock(&gMixer);
}
void CMVoice_Pause(CMVoice* voice)
{
voice->state = CM_STATE_PAUSED;
}
void CMVoice_TogglePause(CMVoice* voice)
{
if (voice->state == CM_STATE_PAUSED)
CMVoice_Play(voice);
else if (voice->state == CM_STATE_PLAYING)
CMVoice_Pause(voice);
}
void CMVoice_Stop(CMVoice* voice)
{
voice->state = CM_STATE_STOPPED;
voice->rewind = true;
}
//-----------------------------------------------------------------------------
// WavStream implementation
static inline CMWavStream* CMWavStream_Check(CMVoice* voice)
{
CM_ASSERT(voice->cookie == 'VOIX', "VOIX cookie not found");
CM_ASSERT(voice->wav.cookie == 'WAVS', "WAVS cookie not found");
return &voice->wav;
}
static CMWavStream* InstallWavStream(CMVoice* voice)
{
CMWavStream* wav = &voice->wav;
wav->cookie = 'WAVS';
wav->pcmformat = PCMFORMAT_NULL;
wav->idx = 0;
voice->callbacks.fillBuffer = StreamWav;
voice->callbacks.rewind = RewindWav;
voice->callbacks.free = FreeWav;
return wav;
}
static void FreeWav(CMVoice* voice)
{
CMWavStream* wav = CMWavStream_Check(voice);
if (!wav->data)
{
return;
}
if (wav->ownData)
{
SDL_free(wav->data);
}
wav->data = NULL;
wav->dataLength = 0;
wav->ownData = false;
wav->cookie = 'DEAD';
}
static void RewindWav(CMVoice* voice)
{
CMWavStream* wav = CMWavStream_Check(voice);
wav->idx = 0;
}
static void StreamWav(CMVoice* voice, int16_t* dst, int fillLength)
{
CMWavStream* wav = CMWavStream_Check(voice);
int x, n;
fillLength /= 2;
const int16_t* data16 = (const int16_t*) wav->data;
const uint8_t* data8 = (const uint8_t*) wav->data;
#define WAV_PROCESS_LOOP(X) \
while (n--) \
{ \
X \
dst += 2; \
wav->idx++; \
}
while (fillLength > 0)
{
n = MinInt(fillLength, voice->sampleCount - wav->idx);
fillLength -= n;
switch (wav->pcmformat)
{
case PCMFORMAT_1CH_BE16:
WAV_PROCESS_LOOP({
dst[0] = dst[1] = UnpackI16BE(&data16[wav->idx]);
});
break;
case PCMFORMAT_2CH_BE16:
WAV_PROCESS_LOOP({
x = wav->idx * 2;
dst[0] = UnpackI16BE(&data16[x]);
dst[1] = UnpackI16BE(&data16[x + 1]);
});
break;
case PCMFORMAT_1CH_LE16:
WAV_PROCESS_LOOP({
dst[0] = dst[1] = UnpackI16LE(&data16[wav->idx]);
});
break;
case PCMFORMAT_2CH_LE16:
WAV_PROCESS_LOOP({
x = wav->idx * 2;
dst[0] = UnpackI16LE(&data16[x]);
dst[1] = UnpackI16LE(&data16[x + 1]);
});
break;
case PCMFORMAT_1CH_8:
case PCMFORMAT_1CH_8 | 0x80: // with big-endian flag
WAV_PROCESS_LOOP({
dst[0] = dst[1] = (data8[wav->idx] - 128) << 8;
});
break;
case PCMFORMAT_2CH_8:
case PCMFORMAT_2CH_8 | 0x80: // with big-endian flag
WAV_PROCESS_LOOP({
x = wav->idx * 2;
dst[0] = (data8[x] - 128) << 8;
dst[1] = (data8[x + 1] - 128) << 8;
});
break;
default:
CM_DIE("unknown pcmformat");
break;
}
// Loop back and continue filling buffer if we didn't fill the buffer
if (fillLength > 0)
{
wav->idx = voice->sustainOffset;
}
}
#undef WAV_PROCESS_LOOP
}
//-----------------------------------------------------------------------------
// LoadWAVFromFile
static const char* FindRIFFChunk(const char* data, size_t len, const char* id, int* size)
{
// TODO : Error handling on malformed wav file
size_t idlen = SDL_strlen(id);
const char* p = data + 12;
next:
*size = *((uint32_t*)(p + 4));
if (SDL_memcmp(p, id, idlen))
{
p += 8 + *size;
if (p > data + len)
return NULL;
goto next;
}
return p + 8;
}
CMVoice* CMVoice_LoadWAV(const char* path)
{
int sz;
size_t len = 0;
char *const data = LoadFile(path, &len);
const char* p = (char*)data;
// Check header
if (SDL_memcmp(p, "RIFF", 4) || SDL_memcmp(p + 8, "WAVE", 4))
CM_DIE("not a WAVE file");
// Find fmt subchunk
p = FindRIFFChunk(data, len, "fmt ", &sz);
CM_ASSERT(p, "no fmt chunk in WAVE");
// Load fmt info
int format = *((uint16_t*)(p));
int channels = *((uint16_t*)(p + 2));
int samplerate = *((uint32_t*)(p + 4));
int bitdepth = *((uint16_t*)(p + 14));
CM_ASSERT(format == 1, "unsupported WAVE format");
CM_ASSERT(channels == 1 || channels == 2, "unsupported channel count");
CM_ASSERT(bitdepth == 8 || bitdepth == 16, "unsupported bitdepth");
CM_ASSERT(samplerate != 0, "weird samplerate");
// Find data subchunk
p = FindRIFFChunk(data, len, "data", &sz);
CM_ASSERT(p, "no data chunk in WAVE");
const char* sampleData = p;
int sampleDataLength = sz;
int samplecount = (sampleDataLength / (bitdepth / 8)) / channels;
CMVoice* voice = CMVoice_New(samplerate, samplecount);
CMWavStream* wav = InstallWavStream(voice);
wav->pcmformat = BuildPCMFormat(bitdepth, channels, 0);
wav->data = SDL_malloc(sampleDataLength);
wav->dataLength = sampleDataLength;
wav->ownData = true;
SDL_memcpy(wav->data, sampleData, sampleDataLength);
SDL_free(data);
CM_ASSERT(wav->pcmformat != 0, "weird pcmformat");
return voice;
}
//-----------------------------------------------------------------------------
// ModStream
static inline CMModStream* CMModStream_Check(CMVoice* voice)
{
CM_ASSERT(voice->cookie == 'VOIX', "VOIX cookie not found");
CM_ASSERT(voice->mod.cookie == 'MODS', "MODS cookie not found");
return &voice->mod;
}
CMVoice* CMVoice_LoadMOD(const char* path)
{
char errors[64];
errors[0] = '\0';
size_t moduleFileSize = 0;
char* moduleFile = LoadFile(path, &moduleFileSize);
struct data d = { .buffer = moduleFile, .length = (int)moduleFileSize };
CMVoice* voice = CMVoice_New(gMixer.samplerate, INT_MAX);
voice->callbacks.fillBuffer = StreamMod;
voice->callbacks.free = FreeMod;
voice->mod.cookie = 'MODS';
voice->mod.replayBuffer = SDL_calloc(1, 2048 * 8 * sizeof(voice->mod.replayBuffer[0]));
voice->mod.replayBufferOffset = 0;
voice->mod.replayBufferSamples = 0;
voice->mod.playbackSpeedMult = 1.0;
voice->mod.moduleFileMemory = moduleFile;
voice->mod.module = module_load(&d, errors);
voice->mod.replay = new_replay(voice->mod.module, gMixer.samplerate, 0);
CM_ASSERT(!errors[0], errors);
return voice;
}
static void FreeMod(CMVoice* voice)
{
CMModStream* mod = CMModStream_Check(voice);
dispose_module(mod->module);
SDL_free(mod->moduleFileMemory);
SDL_free(mod->replayBuffer);
mod->cookie = 'DEAD';
}
void CMVoice_SetMODPlaybackSpeed(CMVoice* voice, double speed)
{
CMModStream* mod = CMModStream_Check(voice);
mod->playbackSpeedMult = speed;
}
static void StreamMod(CMVoice* voice, int16_t* output, int length)
{
CMModStream* mod = CMModStream_Check(voice);
length /= 2;
while (length > 0)
{
// refill replay buffer if exhausted
if (mod->replayBufferSamples == 0)
{
mod->replayBufferOffset = 0;
mod->replayBufferSamples = replay_get_audio(mod->replay, mod->replayBuffer, 0, (int)(mod->playbackSpeedMult * 100.0));
}
// number of stereo samples to copy from replay buffer to output buffer
int nToCopy = MinInt(mod->replayBufferSamples, length);
int* input = &mod->replayBuffer[mod->replayBufferOffset * 2];
// Copy samples
for (int i = 0; i < nToCopy * 2; i++)
{
int sample = *(input++);
sample = ClampInt(sample, -32768, 32767);
*(output++) = sample;
}
mod->replayBufferOffset += nToCopy;
mod->replayBufferSamples -= nToCopy;
length -= nToCopy;
}
}