ddnet/src/engine/client/snd.cpp

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#include <baselib/system.h>
#include <baselib/audio.h>
#include <baselib/stream/file.h>
#include <engine/interface.h>
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using namespace baselib;
static const int NUM_FRAMES_STOP = 512;
static const float NUM_FRAMES_STOP_INV = 1.0f/(float)NUM_FRAMES_STOP;
static const int NUM_FRAMES_LERP = 512;
static const float NUM_FRAMES_LERP_INV = 1.0f/(float)NUM_FRAMES_LERP;
static const float GLOBAL_VOLUME_SCALE = 0.75f;
static const int64 GLOBAL_SOUND_DELAY = 1000;
// --- sound ---
class sound_data
{
public:
short *data;
int num_samples;
int rate;
int channels;
int sustain_start;
int sustain_end;
int64 last_played;
};
inline short clamp(int i)
{
if(i > 0x7fff)
return 0x7fff;
if(i < -0x7fff)
return -0x7fff;
return i;
}
class mixer : public audio_stream
{
public:
class channel
{
public:
channel()
{ data = 0; lerp = -1; stop = -1; }
sound_data *data;
int tick;
int loop;
float pan;
float vol;
float old_vol;
float new_vol;
int lerp;
int stop;
};
enum
{
MAX_CHANNELS=8,
};
channel channels[MAX_CHANNELS];
virtual void fill(void *output, unsigned long frames)
{
//dbg_msg("snd", "mixing!");
short *out = (short*)output;
bool clamp_flag = false;
int active_channels = 0;
for(unsigned long i = 0; i < frames; i++)
{
int left = 0;
int right = 0;
for(int c = 0; c < MAX_CHANNELS; c++)
{
if(channels[c].data)
{
if(channels[c].data->channels == 1)
{
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left += (int)((1.0f-(channels[c].pan+1.0f)*0.5f) * channels[c].vol * channels[c].data->data[channels[c].tick]);
right += (int)((channels[c].pan+1.0f)*0.5f * channels[c].vol * channels[c].data->data[channels[c].tick]);
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channels[c].tick++;
}
else
{
float pl = channels[c].pan<0.0f?-channels[c].pan:1.0f;
float pr = channels[c].pan>0.0f?1.0f-channels[c].pan:1.0f;
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left += (int)(pl*channels[c].vol * channels[c].data->data[channels[c].tick]);
right += (int)(pr*channels[c].vol * channels[c].data->data[channels[c].tick + 1]);
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channels[c].tick += 2;
}
if(channels[c].loop)
{
if(channels[c].data->sustain_start >= 0 && channels[c].tick >= channels[c].data->sustain_end)
channels[c].tick = channels[c].data->sustain_start;
else if(channels[c].tick > channels[c].data->num_samples)
channels[c].tick = 0;
}
else if(channels[c].tick > channels[c].data->num_samples)
channels[c].data = 0;
if(channels[c].stop == 0)
{
channels[c].stop = -1;
channels[c].data = 0;
}
else if(channels[c].stop > 0)
{
channels[c].vol = channels[c].old_vol * (float)channels[c].stop * NUM_FRAMES_STOP_INV;
channels[c].stop--;
}
if(channels[c].lerp > 0)
{
channels[c].vol = (1.0f - (float)channels[c].lerp * NUM_FRAMES_LERP_INV) * channels[c].new_vol +
(float)channels[c].lerp * NUM_FRAMES_LERP_INV * channels[c].old_vol;
channels[c].lerp--;
}
active_channels++;
}
}
// TODO: remove these
*out = clamp(left); // left
if(*out != left) clamp_flag = true;
out++;
*out = clamp(right); // right
if(*out != right) clamp_flag = true;
out++;
}
if(clamp_flag)
dbg_msg("snd", "CLAMPED!");
}
int play(sound_data *sound, unsigned loop, float vol, float pan)
{
if(time_get() - sound->last_played < GLOBAL_SOUND_DELAY)
return -1;
for(int c = 0; c < MAX_CHANNELS; c++)
{
if(channels[c].data == 0)
{
channels[c].data = sound;
channels[c].tick = 0;
channels[c].loop = loop;
channels[c].vol = vol * GLOBAL_VOLUME_SCALE;
channels[c].pan = pan;
sound->last_played = time_get();
return c;
}
}
return -1;
}
void stop(int id)
{
dbg_assert(id >= 0 && id < MAX_CHANNELS, "id out of bounds");
channels[id].old_vol = channels[id].vol;
channels[id].stop = NUM_FRAMES_STOP;
}
void set_vol(int id, float vol)
{
dbg_assert(id >= 0 && id < MAX_CHANNELS, "id out of bounds");
channels[id].new_vol = vol * GLOBAL_VOLUME_SCALE;
channels[id].old_vol = channels[id].vol;
channels[id].lerp = NUM_FRAMES_LERP;
}
};
static mixer mixer;
//static sound_data test_sound;
/*
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extern "C"
{
#include "wavpack/wavpack.h"
}*/
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/*
static file_stream *read_func_filestream;
static int32_t read_func(void *buff, int32_t bcount)
{
return read_func_filestream->read(buff, bcount);
}
static uchar *format_samples(int bps, uchar *dst, int32_t *src, uint32_t samcnt)
{
int32_t temp;
switch (bps) {
case 1:
while (samcnt--)
*dst++ = *src++ + 128;
break;
case 2:
while (samcnt--) {
*dst++ = (uchar)(temp = *src++);
*dst++ = (uchar)(temp >> 8);
}
break;
case 3:
while (samcnt--) {
*dst++ = (uchar)(temp = *src++);
*dst++ = (uchar)(temp >> 8);
*dst++ = (uchar)(temp >> 16);
}
break;
case 4:
while (samcnt--) {
*dst++ = (uchar)(temp = *src++);
*dst++ = (uchar)(temp >> 8);
*dst++ = (uchar)(temp >> 16);
*dst++ = (uchar)(temp >> 24);
}
break;
}
return dst;
}*/
/*
struct sound_holder
{
sound_data sound;
int next;
};
static const int MAX_SOUNDS = 256;
static sound_holder sounds[MAX_SOUNDS];
static int first_free_sound;
bool snd_load_wv(const char *filename, sound_data *snd)
{
// open file
file_stream file;
if(!file.open_r(filename))
{
dbg_msg("sound/wv", "failed to open file. filename='%s'", filename);
return false;
}
read_func_filestream = &file;
// get info
WavpackContext *wpc;
char error[128];
wpc = WavpackOpenFileInput(read_func, error);
if(!wpc)
{
dbg_msg("sound/wv", "failed to open file. err=%s filename='%s'", error, filename);
return false;
}
snd->num_samples = WavpackGetNumSamples(wpc);
int bps = WavpackGetBytesPerSample(wpc);
int channels = WavpackGetReducedChannels(wpc);
snd->rate = WavpackGetSampleRate(wpc);
int bits = WavpackGetBitsPerSample(wpc);
(void)bps;
(void)channels;
(void)bits;
// decompress
int datasize = snd->num_samples*2;
snd->data = (short*)mem_alloc(datasize, 1);
int totalsamples = 0;
while(1)
{
int buffer[1024*4];
int samples_unpacked = WavpackUnpackSamples(wpc, buffer, 1024*4);
totalsamples += samples_unpacked;
if(samples_unpacked)
{
// convert
}
}
if(snd->num_samples != totalsamples)
{
dbg_msg("sound/wv", "wrong amount of samples. filename='%s'", filename);
mem_free(snd->data);
return false;;
}
return false;
}*/
struct sound_holder
{
sound_data sound;
int next;
};
static const int MAX_SOUNDS = 1024;
static sound_holder sounds[MAX_SOUNDS];
static int first_free_sound;
bool snd_init()
{
first_free_sound = 0;
for(int i = 0; i < MAX_SOUNDS; i++)
sounds[i].next = i+1;
sounds[MAX_SOUNDS-1].next = -1;
return mixer.create();
}
bool snd_shutdown()
{
mixer.destroy();
return true;
}
static int snd_alloc_sound()
{
if(first_free_sound < 0)
return -1;
int id = first_free_sound;
first_free_sound = sounds[id].next;
sounds[id].next = -1;
return id;
}
int snd_load_wav(const char *filename)
{
sound_data snd;
// open file for reading
file_stream file;
if(!file.open_r(filename))
{
dbg_msg("sound/wav", "failed to open file. filename='%s'", filename);
return -1;
}
int id = -1;
int state = 0;
while(1)
{
// read chunk header
unsigned char head[8];
if(file.read(head, sizeof(head)) != 8)
{
break;
}
int chunk_size = head[4] | (head[5]<<8) | (head[6]<<16) | (head[7]<<24);
head[4] = 0;
if(state == 0)
{
// read the riff and wave headers
if(head[0] != 'R' || head[1] != 'I' || head[2] != 'F' || head[3] != 'F')
{
dbg_msg("sound/wav", "not a RIFF file. filename='%s'", filename);
return -1;
}
unsigned char type[4];
file.read(type, 4);
if(type[0] != 'W' || type[1] != 'A' || type[2] != 'V' || type[3] != 'E')
{
dbg_msg("sound/wav", "RIFF file is not a WAVE. filename='%s'", filename);
return -1;
}
state++;
}
else if(state == 1)
{
// read the format chunk
if(head[0] == 'f' && head[1] == 'm' && head[2] == 't' && head[3] == ' ')
{
unsigned char fmt[16];
if(file.read(fmt, sizeof(fmt)) != sizeof(fmt))
{
dbg_msg("sound/wav", "failed to read format. filename='%s'", filename);
return -1;
}
// decode format
int compression_code = fmt[0] | (fmt[1]<<8);
snd.channels = fmt[2] | (fmt[3]<<8);
snd.rate = fmt[4] | (fmt[5]<<8) | (fmt[6]<<16) | (fmt[7]<<24);
if(compression_code != 1)
{
dbg_msg("sound/wav", "file is not uncompressed. filename='%s'", filename);
return -1;
}
if(snd.channels > 2)
{
dbg_msg("sound/wav", "file is not mono or stereo. filename='%s'", filename);
return -1;
}
if(snd.rate != 44100)
{
dbg_msg("sound/wav", "file is %d Hz, not 44100 Hz. filename='%s'", snd.rate, filename);
return -1;
}
int bps = fmt[14] | (fmt[15]<<8);
if(bps != 16)
{
dbg_msg("sound/wav", "bps is %d, not 16, filname='%s'", bps, filename);
return -1;
}
// skip extra bytes (not used for uncompressed)
//int extra_bytes = fmt[14] | (fmt[15]<<8);
//dbg_msg("sound/wav", "%d", extra_bytes);
//file.skip(extra_bytes);
// next state
state++;
}
else
file.skip(chunk_size);
}
else if(state == 2)
{
// read the data
if(head[0] == 'd' && head[1] == 'a' && head[2] == 't' && head[3] == 'a')
{
snd.data = (short*)mem_alloc(chunk_size, 1);
file.read(snd.data, chunk_size);
snd.num_samples = chunk_size/(2);
snd.sustain_start = -1;
snd.sustain_end = -1;
snd.last_played = 0;
id = snd_alloc_sound();
sounds[id].sound = snd;
state++;
}
else
file.skip(chunk_size);
}
else if(state == 3)
{
if(head[0] == 's' && head[1] == 'm' && head[2] == 'p' && head[3] == 'l')
{
int smpl[9];
int loop[6];
file.read(smpl, sizeof(smpl));
if(smpl[7] > 0)
{
file.read(loop, sizeof(loop));
sounds[id].sound.sustain_start = loop[2] * sounds[id].sound.channels;
sounds[id].sound.sustain_end = loop[3] * sounds[id].sound.channels;
}
if(smpl[7] > 1)
file.skip((smpl[7]-1) * sizeof(loop));
file.skip(smpl[8]);
state++;
}
else
file.skip(chunk_size);
}
else
file.skip(chunk_size);
}
if(id >= 0)
dbg_msg("sound/wav", "loaded %s", filename);
else
dbg_msg("sound/wav", "failed to load %s", filename);
return id;
}
int snd_play(int id, int loop, float vol, float pan)
{
if(id < 0)
{
dbg_msg("snd", "bad sound id");
return -1;
}
dbg_assert(sounds[id].sound.data != 0, "null sound");
dbg_assert(sounds[id].next == -1, "sound isn't allocated");
return mixer.play(&sounds[id].sound, loop, vol, pan);
}
void snd_stop(int id)
{
if(id >= 0)
mixer.stop(id);
}
void snd_set_vol(int id, float vol)
{
if(id >= 0)
mixer.set_vol(id, vol);
}