ref: 8871dfafcd58c3952762d5f0de3a6222e638fba2
dir: /src/support/cmixer.cpp/
// Adapted from cmixer by rxi (https://github.com/rxi/cmixer)
/*
** Copyright (c) 2017 rxi
**
** 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 <SDL.h>
#include <vector>
#include <fstream>
#include <list>
using namespace cmixer;
#define CLAMP(x, a, b) ((x) < (a) ? (a) : (x) > (b) ? (b) : (x))
#define MIN(a, b) ((a) < (b) ? (a) : (b))
#define MAX(a, b) ((a) > (b) ? (a) : (b))
#define FX_BITS (12)
#define FX_UNIT (1 << FX_BITS)
#define FX_MASK (FX_UNIT - 1)
#define FX_FROM_FLOAT(f) ((long)((f) * FX_UNIT))
#define DOUBLE_FROM_FX(f) ((double)f / FX_UNIT)
#define FX_LERP(a, b, p) ((a) + ((((b) - (a)) * (p)) >> FX_BITS))
#define BUFFER_MASK (BUFFER_SIZE - 1)
//-----------------------------------------------------------------------------
// Global mixer
static struct Mixer
{
SDL_mutex* sdlAudioMutex;
std::list<Source*> sources; // Linked list of active (playing) sources
int32_t pcmmixbuf[BUFFER_SIZE]; // Internal master buffer
int samplerate; // Master samplerate
int gain; // Master gain (fixed point)
void Init(int samplerate);
void Process(int16_t* dst, int len);
void Lock();
void Unlock();
void SetMasterGain(double newGain);
} gMixer = {};
//-----------------------------------------------------------------------------
// Global init/shutdown
static bool sdlAudioSubSystemInited = false;
static SDL_AudioDeviceID sdlDeviceID = 0;
void cmixer::InitWithSDL()
{
if (sdlAudioSubSystemInited)
throw std::runtime_error("SDL audio subsystem already inited");
if (0 != SDL_InitSubSystem(SDL_INIT_AUDIO))
throw std::runtime_error("couldn't init SDL audio subsystem");
sdlAudioSubSystemInited = true;
// Init SDL audio
SDL_AudioSpec fmt = {};
fmt.freq = 44100;
fmt.format = AUDIO_S16;
fmt.channels = 2;
fmt.samples = 1024;
fmt.callback = [](void* udata, Uint8* stream, int size)
{
gMixer.Process((int16_t*) stream, size / 2);
};
SDL_AudioSpec got;
sdlDeviceID = SDL_OpenAudioDevice(NULL, 0, &fmt, &got, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE);
if (!sdlDeviceID)
throw std::runtime_error(SDL_GetError());
// Init library
gMixer.Init(got.freq);
gMixer.SetMasterGain(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 = nullptr;
}
if (sdlAudioSubSystemInited)
{
SDL_QuitSubSystem(SDL_INIT_AUDIO);
sdlAudioSubSystemInited = false;
}
}
double cmixer::GetMasterGain()
{
return DOUBLE_FROM_FX(gMixer.gain);
}
void cmixer::SetMasterGain(double newGain)
{
gMixer.SetMasterGain(newGain);
}
//-----------------------------------------------------------------------------
// Global mixer impl
void Mixer::Lock()
{
SDL_LockMutex(sdlAudioMutex);
}
void Mixer::Unlock()
{
SDL_UnlockMutex(sdlAudioMutex);
}
void Mixer::Init(int newSamplerate)
{
sdlAudioMutex = SDL_CreateMutex();
samplerate = newSamplerate;
gain = FX_UNIT;
}
void Mixer::SetMasterGain(double newGain)
{
if (newGain < 0)
newGain = 0;
gain = FX_FROM_FLOAT(newGain);
}
void Mixer::Process(int16_t* dst, int len)
{
// Process in chunks of BUFFER_SIZE if `len` is larger than BUFFER_SIZE
while (len > BUFFER_SIZE)
{
Process(dst, BUFFER_SIZE);
dst += BUFFER_SIZE;
len -= BUFFER_SIZE;
}
// Zeroset internal buffer
memset(pcmmixbuf, 0, len * sizeof(pcmmixbuf[0]));
// Process active sources
Lock();
for (auto si = sources.begin(); si != sources.end();)
{
auto& s = **si;
s.Process(len);
// Remove source from list if it is no longer playing
if (s.state != CM_STATE_PLAYING)
{
s.active = false;
si = sources.erase(si);
}
else
{
++si;
}
}
Unlock();
// Copy internal buffer to destination and clip
for (int i = 0; i < len; i++)
{
int x = (pcmmixbuf[i] * gain) >> FX_BITS;
dst[i] = CLAMP(x, -32768, 32767);
}
}
//-----------------------------------------------------------------------------
// Source implementation
Source::Source()
{
ClearPrivate();
active = false;
}
void Source::ClearPrivate()
{
samplerate = 0;
length = 0;
end = 0;
state = CM_STATE_STOPPED;
position = 0;
lgain = 0;
rgain = 0;
rate = 0;
nextfill = 0;
loop = false;
rewind = true;
interpolate = false;
// DON'T touch active. The source may still be in gMixer!
gain = 0;
pan = 0;
onComplete = nullptr;
}
void Source::Clear()
{
gMixer.Lock();
ClearPrivate();
ClearImplementation();
gMixer.Unlock();
}
void Source::Init(int theSampleRate, int theLength)
{
this->samplerate = theSampleRate;
this->length = theLength;
SetGain(1);
SetPan(0);
SetPitch(1);
SetLoop(false);
Stop();
}
Source::~Source()
{
gMixer.Lock();
if (active)
{
gMixer.sources.remove(this);
}
gMixer.Unlock();
}
void Source::Rewind()
{
RewindImplementation();
position = 0;
rewind = false;
end = length;
nextfill = 0;
}
void Source::FillBuffer(int offset, int fillLength)
{
FillBuffer(pcmbuf + offset, fillLength);
}
void Source::Process(int len)
{
int32_t* dst = gMixer.pcmmixbuf;
// Do rewind if flag is set
if (rewind)
{
Rewind();
}
// Don't process if not playing
if (state != CM_STATE_PLAYING)
{
return;
}
// Process audio
while (len > 0)
{
// Get current position frame
int frame = int(position >> FX_BITS);
// Fill buffer if required
if (frame + 3 >= nextfill)
{
FillBuffer((nextfill * 2) & BUFFER_MASK, BUFFER_SIZE / 2);
nextfill += BUFFER_SIZE / 4;
}
// Handle reaching the end of the playthrough
if (frame >= end)
{
// As streams continiously fill the raw buffer in a loop we simply
// increment the end idx by one length and continue reading from it for
// another play-through
end = frame + this->length;
// Set state and stop processing if we're not set to loop
if (!loop)
{
state = CM_STATE_STOPPED;
if (onComplete != nullptr)
onComplete();
break;
}
}
// Work out how many frames we should process in the loop
int n = MIN(nextfill - 2, end) - frame;
int count = (n << FX_BITS) / rate;
count = MAX(count, 1);
count = MIN(count, len / 2);
len -= count * 2;
// Add audio to master buffer
if (rate == FX_UNIT)
{
// Add audio to buffer -- basic
n = frame * 2;
for (int i = 0; i < count; i++)
{
dst[0] += (pcmbuf[(n ) & BUFFER_MASK] * lgain) >> FX_BITS;
dst[1] += (pcmbuf[(n + 1) & BUFFER_MASK] * rgain) >> FX_BITS;
n += 2;
dst += 2;
}
this->position += count * FX_UNIT;
}
else if (interpolate)
{
// Resample audio (with linear interpolation) and add to buffer
for (int i = 0; i < count; i++)
{
n = int(position >> FX_BITS) * 2;
int p = position & FX_MASK;
int a = pcmbuf[(n ) & BUFFER_MASK];
int b = pcmbuf[(n + 2) & BUFFER_MASK];
dst[0] += (FX_LERP(a, b, p) * lgain) >> FX_BITS;
n++;
a = pcmbuf[(n ) & BUFFER_MASK];
b = pcmbuf[(n + 2) & BUFFER_MASK];
dst[1] += (FX_LERP(a, b, p) * rgain) >> FX_BITS;
position += rate;
dst += 2;
}
}
else
{
// Resample audio (without interpolation) and add to buffer
for (int i = 0; i < count; i++)
{
n = int(position >> FX_BITS) * 2;
dst[0] += (pcmbuf[(n ) & BUFFER_MASK] * lgain) >> FX_BITS;
dst[1] += (pcmbuf[(n + 1) & BUFFER_MASK] * rgain) >> FX_BITS;
position += rate;
dst += 2;
}
}
}
}
double Source::GetLength() const
{
return length / (double) samplerate;
}
double Source::GetPosition() const
{
return ((position >> FX_BITS) % length) / (double) samplerate;
}
int Source::GetState() const
{
return state;
}
void Source::RecalcGains()
{
double l = this->gain * (pan <= 0. ? 1. : 1. - pan);
double r = this->gain * (pan >= 0. ? 1. : 1. + pan);
this->lgain = FX_FROM_FLOAT(l);
this->rgain = FX_FROM_FLOAT(r);
}
void Source::SetGain(double newGain)
{
gain = newGain;
RecalcGains();
}
void Source::SetPan(double newPan)
{
pan = CLAMP(newPan, -1.0, 1.0);
RecalcGains();
}
void Source::SetPitch(double newPitch)
{
double newRate;
if (newPitch > 0.)
{
newRate = samplerate / (double) gMixer.samplerate * newPitch;
}
else
{
newRate = 0.001;
}
rate = FX_FROM_FLOAT(newRate);
}
void Source::SetLoop(bool newLoop)
{
loop = newLoop;
}
void Source::SetInterpolation(bool newInterpolation)
{
interpolate = newInterpolation;
}
void Source::Play()
{
gMixer.Lock();
state = CM_STATE_PLAYING;
if (!active)
{
active = true;
gMixer.sources.push_front(this);
}
gMixer.Unlock();
}
void Source::Pause()
{
state = CM_STATE_PAUSED;
}
void Source::TogglePause()
{
if (state == CM_STATE_PAUSED)
Play();
else if (state == CM_STATE_PLAYING)
Pause();
}
void Source::Stop()
{
state = CM_STATE_STOPPED;
rewind = true;
}
//-----------------------------------------------------------------------------
// WavStream implementation
#define WAV_PROCESS_LOOP(X) \
while (n--) \
{ \
X \
dst += 2; \
idx++; \
}
WavStream::WavStream()
: Source()
{
ClearImplementation();
}
void WavStream::ClearImplementation()
{
bitdepth = 0;
channels = 0;
idx = 0;
userBuffer.clear();
}
void WavStream::RewindImplementation()
{
idx = 0;
}
void WavStream::FillBuffer(int16_t* dst, int len)
{
int x, n;
len /= 2;
while (len > 0)
{
n = MIN(len, length - idx);
len -= n;
if (bitdepth == 16 && channels == 1)
{
WAV_PROCESS_LOOP({
dst[0] = dst[1] = data16()[idx];
});
}
else if (bitdepth == 16 && channels == 2)
{
WAV_PROCESS_LOOP({
x = idx * 2;
dst[0] = data16()[x];
dst[1] = data16()[x + 1];
});
}
else if (bitdepth == 8 && channels == 1)
{
WAV_PROCESS_LOOP({
dst[0] = dst[1] = (data8()[idx] - 128) << 8;
});
}
else if (bitdepth == 8 && channels == 2)
{
WAV_PROCESS_LOOP({
x = idx * 2;
dst[0] = (data8()[x] - 128) << 8;
dst[1] = (data8()[x + 1] - 128) << 8;
});
}
// Loop back and continue filling buffer if we didn't fill the buffer
if (len > 0)
{
idx = 0;
}
}
}
//-----------------------------------------------------------------------------
// LoadWAVFromFile for testing
static std::vector<char> LoadFile(char const* filename)
{
std::ifstream ifs(filename, std::ios::binary | std::ios::ate);
auto pos = ifs.tellg();
std::vector<char> bytes(pos);
ifs.seekg(0, std::ios::beg);
ifs.read(&bytes[0], pos);
return bytes;
}
static const char* FindChunk(const char* data, int len, const char* id, int* size)
{
// TODO : Error handling on malformed wav file
int idlen = strlen(id);
const char* p = data + 12;
next:
*size = *((uint32_t*)(p + 4));
if (memcmp(p, id, idlen)) {
p += 8 + *size;
if (p > data + len) return NULL;
goto next;
}
return p + 8;
}
void cmixer::WavStream::InitFromWAVFile(const char* path)
{
int sz;
auto filebuf = LoadFile(path);
auto len = filebuf.size();
const char* data = filebuf.data();
const char* p = (char*)data;
// Check header
if (memcmp(p, "RIFF", 4) || memcmp(p + 8, "WAVE", 4))
throw std::invalid_argument("bad wav header");
// Find fmt subchunk
p = FindChunk(data, len, "fmt ", &sz);
if (!p)
throw std::invalid_argument("no fmt subchunk");
// 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));
if (format != 1)
throw std::invalid_argument("unsupported format");
if (channels == 0 || samplerate == 0 || bitdepth == 0)
throw std::invalid_argument("bad format");
// Find data subchunk
p = FindChunk(data, len, "data", &sz);
if (!p)
throw std::invalid_argument("no data subchunk");
Clear();
Init(samplerate, sz / (channels * bitdepth / 8));
this->bitdepth = bitdepth;
this->channels = channels;
this->idx = 0;
this->interpolate = true;
userBuffer.reserve(sz);
for (int i = 0; i < sz; i++)
userBuffer.push_back(p[i]);
}