/
sound.c
218 lines (191 loc) · 6.82 KB
/
sound.c
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
inline UINT32 power(UINT32 base, UINT32 exponent) {
UINT32 result = 1;
while (exponent-- > 0) result *= base;
return result;
}
static void initializeSoundSystem(SoundSystem *sys, float bufferDurationSec, float tickDuration) {
//
// Initialize WASAPI
//
HRESULT hr;
IMMDeviceEnumerator *enumerator;
CoInitialize(NULL);
hr = CoCreateInstance(&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, &IID_IMMDeviceEnumerator, (void**)&enumerator);
assert(SUCCEEDED(hr));
IMMDevice *device;
hr = enumerator->lpVtbl->GetDefaultAudioEndpoint(enumerator, eRender, eConsole, &device);
assert(SUCCEEDED(hr));
IAudioClient *audioClient;
hr = device->lpVtbl->Activate(device, &IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&audioClient);
assert(SUCCEEDED(hr));
WAVEFORMATEX *mixFormat;
hr = audioClient->lpVtbl->GetMixFormat(audioClient, &mixFormat);
assert(SUCCEEDED(hr));
WAVEFORMATEX waveFormat;
memcpy(&waveFormat, mixFormat, sizeof(WAVEFORMATEX));
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.cbSize = 0;
REFERENCE_TIME duration = (REFERENCE_TIME)(bufferDurationSec*REFTIMES_PER_SEC);
hr = audioClient->lpVtbl->Initialize(audioClient, AUDCLNT_SHAREMODE_SHARED, 0, duration, 0, &waveFormat, NULL);
assert(SUCCEEDED(hr));
UINT32 bufferFramesCount;
hr = audioClient->lpVtbl->GetBufferSize(audioClient, &bufferFramesCount);
assert(SUCCEEDED(hr));
IAudioRenderClient *renderClient;
hr = audioClient->lpVtbl->GetService(audioClient, &IID_IAudioRenderClient, (void**)&renderClient);
assert(SUCCEEDED(hr));
sys->audioClient = audioClient;
sys->renderClient = renderClient;
sys->bufferFramesCount = bufferFramesCount;
sys->maxIntegerSampleValue = power(2, waveFormat.wBitsPerSample - 1) - 1;
sys->bytesPerSample = waveFormat.wBitsPerSample / 8;
sys->channelsCount = waveFormat.nChannels;
sys->samplesPerSecond = waveFormat.nSamplesPerSec;
sys->tickDuration = tickDuration;
sys->initialAddingTimeToScoreSoundFrequency = 200.0f;
sys->addingTimeToScoreSoundFrequency = sys->initialAddingTimeToScoreSoundFrequency;
sys->addingTimeToScoreSoundFrequencyStep = 5.0f;
audioClient->lpVtbl->Start(audioClient);
}
static void fillNoiseBuffer(Sound *sound) {
for (int i = 0; i < ARRAY_LENGTH(sound->noise); ++i) {
sound->noise[i] = 2.0f*((float)rand()/(float)RAND_MAX) - 1.0f;
}
}
static void playSound(SoundSystem *sys, SoundID soundId) {
Sound *freeSound = 0;
for (int soundIndex = 0; soundIndex < ARRAY_LENGTH(sys->sounds); ++soundIndex) {
Sound *sound = &sys->sounds[soundIndex];
if (!sound->isPlaying) {
freeSound = sound;
break;
}
}
if (freeSound) {
float toneFrequency = 0.0f;
float baseFrequency = 0.0f;
float freqVariance = 0.0f;
float amplitude = 0.0f;
float soundDurationSec = 0.0f;
float baseDuration = 0.0f;
float durationVariance = 0.0f;
// TODO(slava): More sounds
// TODO(slava): Let specify attack, decay, etc?
switch (soundId) {
case SND_ROCKFORD_MOVE_SPACE:
baseFrequency = 100.0f;
freqVariance = 20.0f;
baseDuration = 0.1f;
amplitude = 0.1f;
break;
case SND_ROCKFORD_MOVE_DIRT:
baseFrequency = 800.0f;
freqVariance = 100.0f;
baseDuration = 0.1f;
amplitude = 0.1f;
break;
case SND_DIAMOND:
baseFrequency = 2500.0f;
freqVariance = 200.0f;
baseDuration = 0.4f;
amplitude = 0.2f;
break;
case SND_BOULDER:
baseFrequency = 500.0f;
freqVariance = 0.0f;
baseDuration = 0.5f;
amplitude = 0.1f;
break;
case SND_ADDING_TIME_TO_SCORE:
baseFrequency = sys->addingTimeToScoreSoundFrequency;
baseDuration = 1.0f;
amplitude = 0.2f;
break;
case SND_UPDATE_CELL_COVER:
baseFrequency = 4000.0f;
freqVariance = 1000.0f;
baseDuration = 0.1f;
amplitude = 0.1f;
break;
case SND_UPDATE_TILE_COVER:
baseFrequency = 3000.0f;
freqVariance = 2500.0f;
baseDuration = 0.12f;
durationVariance = 0.09f;
amplitude = 0.05f;
break;
case SND_ROCKFORD_BIRTH:
baseFrequency = 3000.0f;
baseDuration = 0.2f;
amplitude = 0.2f;
break;
case SND_AMOEBA:
baseFrequency = 4000.0f;
freqVariance = 3500.0f;
baseDuration = 0.4f;
durationVariance = 0.2f;
amplitude = 0.02f;
break;
case SND_MAGIC_WALL:
baseFrequency = 3000.0f;
freqVariance = 1000.0f;
baseDuration = 0.2f;
durationVariance = 0.1f;
amplitude = 0.02f;
break;
default:
assert(!"Unknown sound ID");
}
toneFrequency = baseFrequency + freqVariance*(rand()/(float)RAND_MAX) - freqVariance;
soundDurationSec = sys->tickDuration*(baseDuration + durationVariance*(rand()/(float)RAND_MAX) - durationVariance);
freeSound->isPlaying = true;
freeSound->phase = 0;
freeSound->phaseStep = TWO_PI*toneFrequency / sys->samplesPerSecond;
freeSound->samplesLeftToPlay = (int)(soundDurationSec * sys->samplesPerSecond);
freeSound->amplitude = amplitude;
} else {
// All sound slots are occupied.
}
}
static void outputSound(SoundSystem *sys) {
HRESULT hr;
UINT32 paddingFramesCount;
hr = sys->audioClient->lpVtbl->GetCurrentPadding(sys->audioClient, &paddingFramesCount);
assert(SUCCEEDED(hr));
UINT32 availableFramesCount = sys->bufferFramesCount - paddingFramesCount;
BYTE *buffer;
hr = sys->renderClient->lpVtbl->GetBuffer(sys->renderClient, availableFramesCount, &buffer);
assert(SUCCEEDED(hr));
for (UINT32 frame = 0, b = 0; frame < availableFramesCount; ++frame) {
float fval = 0;
for (int soundIndex = 0; soundIndex < ARRAY_LENGTH(sys->sounds); ++soundIndex) {
Sound *sound = &sys->sounds[soundIndex];
if (sound->isPlaying) {
fval += (sound->phase < PI ? -1.0f : 1.0f) * sound->amplitude;
sound->phase += sound->phaseStep;
if (sound->phase >= TWO_PI) {
sound->phase -= TWO_PI;
}
sound->samplesLeftToPlay--;
if (sound->samplesLeftToPlay == 0) {
sound->isPlaying = false;
}
}
}
if (fval > 1.0f) {
fval = 1.0f;
} else if (fval < -1.0f) {
fval = -1.0f;
}
// If fval is 1.0f, an overflow of INT32 is going to happen when we multiply
// by maxSampleVal. To avoid this, we multiply by amplitude which is less
// than 1.
float amplitude = 0.7f;
INT32 val = (INT32)(fval * amplitude * sys->maxIntegerSampleValue);
for (int channel = 0; channel < sys->channelsCount; ++channel)
for (int byte = 0; byte < sys->bytesPerSample; ++byte)
buffer[b++] = (val >> (byte * 8)) & 0xFF;
}
hr = sys->renderClient->lpVtbl->ReleaseBuffer(sys->renderClient, availableFramesCount, 0);
assert(SUCCEEDED(hr));
}