mirror of
https://github.com/ddnet/ddnet.git
synced 2024-09-21 10:04:18 +00:00
521 lines
11 KiB
C++
521 lines
11 KiB
C++
#include <baselib/system.h>
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#include <baselib/audio.h>
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#include <baselib/stream/file.h>
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#include <engine/interface.h>
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using namespace baselib;
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static const int NUM_FRAMES_STOP = 512;
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static const float NUM_FRAMES_STOP_INV = 1.0f/(float)NUM_FRAMES_STOP;
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static const int NUM_FRAMES_LERP = 512;
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static const float NUM_FRAMES_LERP_INV = 1.0f/(float)NUM_FRAMES_LERP;
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static const float GLOBAL_VOLUME_SCALE = 0.75f;
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static const int64 GLOBAL_SOUND_DELAY = 1000;
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// --- sound ---
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class sound_data
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{
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public:
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short *data;
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int num_samples;
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int rate;
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int channels;
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int sustain_start;
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int sustain_end;
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int64 last_played;
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};
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inline short clamp(int i)
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{
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if(i > 0x7fff)
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return 0x7fff;
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if(i < -0x7fff)
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return -0x7fff;
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return i;
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}
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class mixer : public audio_stream
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{
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public:
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class channel
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{
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public:
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channel()
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{ data = 0; lerp = -1; stop = -1; }
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sound_data *data;
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int tick;
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int loop;
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float pan;
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float vol;
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float old_vol;
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float new_vol;
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int lerp;
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int stop;
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};
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enum
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{
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MAX_CHANNELS=8,
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};
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channel channels[MAX_CHANNELS];
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virtual void fill(void *output, unsigned long frames)
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{
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//dbg_msg("snd", "mixing!");
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short *out = (short*)output;
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bool clamp_flag = false;
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int active_channels = 0;
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for(unsigned long i = 0; i < frames; i++)
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{
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int left = 0;
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int right = 0;
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for(int c = 0; c < MAX_CHANNELS; c++)
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{
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if(channels[c].data)
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{
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if(channels[c].data->channels == 1)
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{
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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]);
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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++;
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}
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else
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{
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float pl = channels[c].pan<0.0f?-channels[c].pan:1.0f;
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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]);
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right += (int)(pr*channels[c].vol * channels[c].data->data[channels[c].tick + 1]);
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channels[c].tick += 2;
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}
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if(channels[c].loop)
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{
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if(channels[c].data->sustain_start >= 0 && channels[c].tick >= channels[c].data->sustain_end)
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channels[c].tick = channels[c].data->sustain_start;
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else if(channels[c].tick > channels[c].data->num_samples)
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channels[c].tick = 0;
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}
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else if(channels[c].tick > channels[c].data->num_samples)
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channels[c].data = 0;
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if(channels[c].stop == 0)
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{
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channels[c].stop = -1;
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channels[c].data = 0;
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}
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else if(channels[c].stop > 0)
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{
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channels[c].vol = channels[c].old_vol * (float)channels[c].stop * NUM_FRAMES_STOP_INV;
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channels[c].stop--;
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}
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if(channels[c].lerp > 0)
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{
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channels[c].vol = (1.0f - (float)channels[c].lerp * NUM_FRAMES_LERP_INV) * channels[c].new_vol +
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(float)channels[c].lerp * NUM_FRAMES_LERP_INV * channels[c].old_vol;
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channels[c].lerp--;
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}
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active_channels++;
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}
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}
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// TODO: remove these
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*out = clamp(left); // left
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if(*out != left) clamp_flag = true;
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out++;
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*out = clamp(right); // right
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if(*out != right) clamp_flag = true;
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out++;
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}
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if(clamp_flag)
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dbg_msg("snd", "CLAMPED!");
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}
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int play(sound_data *sound, unsigned loop, float vol, float pan)
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{
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if(time_get() - sound->last_played < GLOBAL_SOUND_DELAY)
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return -1;
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for(int c = 0; c < MAX_CHANNELS; c++)
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{
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if(channels[c].data == 0)
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{
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channels[c].data = sound;
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channels[c].tick = 0;
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channels[c].loop = loop;
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channels[c].vol = vol * GLOBAL_VOLUME_SCALE;
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channels[c].pan = pan;
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sound->last_played = time_get();
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return c;
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}
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}
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return -1;
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}
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void stop(int id)
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{
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dbg_assert(id >= 0 && id < MAX_CHANNELS, "id out of bounds");
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channels[id].old_vol = channels[id].vol;
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channels[id].stop = NUM_FRAMES_STOP;
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}
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void set_vol(int id, float vol)
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{
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dbg_assert(id >= 0 && id < MAX_CHANNELS, "id out of bounds");
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channels[id].new_vol = vol * GLOBAL_VOLUME_SCALE;
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channels[id].old_vol = channels[id].vol;
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channels[id].lerp = NUM_FRAMES_LERP;
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}
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};
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static mixer mixer;
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//static sound_data test_sound;
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/*
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extern "C"
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{
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#include "wavpack/wavpack.h"
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}*/
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/*
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static file_stream *read_func_filestream;
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static int32_t read_func(void *buff, int32_t bcount)
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{
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return read_func_filestream->read(buff, bcount);
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}
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static uchar *format_samples(int bps, uchar *dst, int32_t *src, uint32_t samcnt)
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{
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int32_t temp;
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switch (bps) {
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case 1:
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while (samcnt--)
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*dst++ = *src++ + 128;
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break;
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case 2:
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while (samcnt--) {
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*dst++ = (uchar)(temp = *src++);
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*dst++ = (uchar)(temp >> 8);
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}
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break;
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case 3:
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while (samcnt--) {
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*dst++ = (uchar)(temp = *src++);
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*dst++ = (uchar)(temp >> 8);
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*dst++ = (uchar)(temp >> 16);
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}
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break;
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case 4:
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while (samcnt--) {
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*dst++ = (uchar)(temp = *src++);
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*dst++ = (uchar)(temp >> 8);
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*dst++ = (uchar)(temp >> 16);
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*dst++ = (uchar)(temp >> 24);
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}
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break;
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}
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return dst;
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}*/
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/*
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struct sound_holder
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{
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sound_data sound;
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int next;
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};
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static const int MAX_SOUNDS = 256;
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static sound_holder sounds[MAX_SOUNDS];
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static int first_free_sound;
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bool snd_load_wv(const char *filename, sound_data *snd)
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{
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// open file
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file_stream file;
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if(!file.open_r(filename))
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{
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dbg_msg("sound/wv", "failed to open file. filename='%s'", filename);
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return false;
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}
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read_func_filestream = &file;
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// get info
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WavpackContext *wpc;
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char error[128];
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wpc = WavpackOpenFileInput(read_func, error);
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if(!wpc)
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{
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dbg_msg("sound/wv", "failed to open file. err=%s filename='%s'", error, filename);
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return false;
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}
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snd->num_samples = WavpackGetNumSamples(wpc);
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int bps = WavpackGetBytesPerSample(wpc);
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int channels = WavpackGetReducedChannels(wpc);
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snd->rate = WavpackGetSampleRate(wpc);
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int bits = WavpackGetBitsPerSample(wpc);
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(void)bps;
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(void)channels;
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(void)bits;
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// decompress
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int datasize = snd->num_samples*2;
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snd->data = (short*)mem_alloc(datasize, 1);
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int totalsamples = 0;
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while(1)
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{
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int buffer[1024*4];
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int samples_unpacked = WavpackUnpackSamples(wpc, buffer, 1024*4);
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totalsamples += samples_unpacked;
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if(samples_unpacked)
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{
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// convert
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}
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}
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if(snd->num_samples != totalsamples)
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{
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dbg_msg("sound/wv", "wrong amount of samples. filename='%s'", filename);
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mem_free(snd->data);
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return false;;
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}
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return false;
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}*/
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struct sound_holder
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{
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sound_data sound;
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int next;
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};
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static const int MAX_SOUNDS = 1024;
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static sound_holder sounds[MAX_SOUNDS];
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static int first_free_sound;
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bool snd_init()
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{
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first_free_sound = 0;
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for(int i = 0; i < MAX_SOUNDS; i++)
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sounds[i].next = i+1;
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sounds[MAX_SOUNDS-1].next = -1;
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return mixer.create();
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}
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bool snd_shutdown()
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{
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mixer.destroy();
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return true;
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}
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static int snd_alloc_sound()
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{
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if(first_free_sound < 0)
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return -1;
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int id = first_free_sound;
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first_free_sound = sounds[id].next;
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sounds[id].next = -1;
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return id;
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}
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int snd_load_wav(const char *filename)
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{
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sound_data snd;
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// open file for reading
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file_stream file;
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if(!file.open_r(filename))
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{
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dbg_msg("sound/wav", "failed to open file. filename='%s'", filename);
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return -1;
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}
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int id = -1;
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int state = 0;
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while(1)
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{
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// read chunk header
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unsigned char head[8];
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if(file.read(head, sizeof(head)) != 8)
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{
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break;
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}
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int chunk_size = head[4] | (head[5]<<8) | (head[6]<<16) | (head[7]<<24);
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head[4] = 0;
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if(state == 0)
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{
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// read the riff and wave headers
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if(head[0] != 'R' || head[1] != 'I' || head[2] != 'F' || head[3] != 'F')
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{
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dbg_msg("sound/wav", "not a RIFF file. filename='%s'", filename);
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return -1;
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}
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unsigned char type[4];
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file.read(type, 4);
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if(type[0] != 'W' || type[1] != 'A' || type[2] != 'V' || type[3] != 'E')
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{
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dbg_msg("sound/wav", "RIFF file is not a WAVE. filename='%s'", filename);
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return -1;
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}
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state++;
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}
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else if(state == 1)
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{
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// read the format chunk
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if(head[0] == 'f' && head[1] == 'm' && head[2] == 't' && head[3] == ' ')
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{
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unsigned char fmt[16];
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if(file.read(fmt, sizeof(fmt)) != sizeof(fmt))
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{
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dbg_msg("sound/wav", "failed to read format. filename='%s'", filename);
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return -1;
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}
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// decode format
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int compression_code = fmt[0] | (fmt[1]<<8);
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snd.channels = fmt[2] | (fmt[3]<<8);
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snd.rate = fmt[4] | (fmt[5]<<8) | (fmt[6]<<16) | (fmt[7]<<24);
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if(compression_code != 1)
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{
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dbg_msg("sound/wav", "file is not uncompressed. filename='%s'", filename);
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return -1;
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}
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if(snd.channels > 2)
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{
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dbg_msg("sound/wav", "file is not mono or stereo. filename='%s'", filename);
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return -1;
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}
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if(snd.rate != 44100)
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{
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dbg_msg("sound/wav", "file is %d Hz, not 44100 Hz. filename='%s'", snd.rate, filename);
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return -1;
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}
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int bps = fmt[14] | (fmt[15]<<8);
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if(bps != 16)
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{
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dbg_msg("sound/wav", "bps is %d, not 16, filname='%s'", bps, filename);
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return -1;
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}
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// skip extra bytes (not used for uncompressed)
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//int extra_bytes = fmt[14] | (fmt[15]<<8);
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//dbg_msg("sound/wav", "%d", extra_bytes);
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//file.skip(extra_bytes);
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// next state
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state++;
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}
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else
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file.skip(chunk_size);
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}
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else if(state == 2)
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{
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// read the data
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if(head[0] == 'd' && head[1] == 'a' && head[2] == 't' && head[3] == 'a')
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{
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snd.data = (short*)mem_alloc(chunk_size, 1);
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file.read(snd.data, chunk_size);
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snd.num_samples = chunk_size/(2);
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snd.sustain_start = -1;
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snd.sustain_end = -1;
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snd.last_played = 0;
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id = snd_alloc_sound();
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sounds[id].sound = snd;
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state++;
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}
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else
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file.skip(chunk_size);
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}
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else if(state == 3)
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{
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if(head[0] == 's' && head[1] == 'm' && head[2] == 'p' && head[3] == 'l')
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{
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int smpl[9];
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int loop[6];
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file.read(smpl, sizeof(smpl));
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if(smpl[7] > 0)
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{
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file.read(loop, sizeof(loop));
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sounds[id].sound.sustain_start = loop[2] * sounds[id].sound.channels;
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sounds[id].sound.sustain_end = loop[3] * sounds[id].sound.channels;
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}
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if(smpl[7] > 1)
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file.skip((smpl[7]-1) * sizeof(loop));
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file.skip(smpl[8]);
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state++;
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}
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else
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file.skip(chunk_size);
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}
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else
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file.skip(chunk_size);
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}
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if(id >= 0)
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dbg_msg("sound/wav", "loaded %s", filename);
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else
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dbg_msg("sound/wav", "failed to load %s", filename);
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return id;
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}
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int snd_play(int id, int loop, float vol, float pan)
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{
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if(id < 0)
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{
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dbg_msg("snd", "bad sound id");
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return -1;
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}
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dbg_assert(sounds[id].sound.data != 0, "null sound");
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dbg_assert(sounds[id].next == -1, "sound isn't allocated");
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return mixer.play(&sounds[id].sound, loop, vol, pan);
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}
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void snd_stop(int id)
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{
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if(id >= 0)
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mixer.stop(id);
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}
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void snd_set_vol(int id, float vol)
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{
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if(id >= 0)
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mixer.set_vol(id, vol);
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}
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