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v4k-git-backup/engine/split/v4k_audio.c

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// @fixme: really shutdown audio & related threads before quitting. drwav crashes.
#if is(win32) && !is(gcc)
#include <windows.h>
#include <mmeapi.h> // midi
static HMIDIOUT midi_out_handle = 0;
#elif is(osx)
static AudioUnit midi_out_handle = 0;
#endif
static void midi_init() {
#if is(win32) && !is(gcc)
if( midiOutGetNumDevs() != 0 ) {
midiOutOpen(&midi_out_handle, 0, 0, 0, 0);
}
#elif is(osx)
AUGraph graph;
AUNode outputNode, mixerNode, dlsNode;
NewAUGraph(&graph);
AudioComponentDescription output = {'auou','ahal','appl',0,0};
AUGraphAddNode(graph, &output, &outputNode);
AUGraphOpen(graph);
AUGraphInitialize(graph);
AUGraphStart(graph);
AudioComponentDescription dls = {'aumu','dls ','appl',0,0};
AUGraphAddNode(graph, &dls, &dlsNode);
AUGraphNodeInfo(graph, dlsNode, NULL, &midi_out_handle);
AudioComponentDescription mixer = {'aumx','smxr','appl',0,0};
AUGraphAddNode(graph, &mixer, &mixerNode);
AUGraphConnectNodeInput(graph,mixerNode,0,outputNode,0);
AUGraphConnectNodeInput(graph,dlsNode,0,mixerNode,0);
AUGraphUpdate(graph,NULL);
#endif
}
static void midi_quit() {
#if is(win32) && !is(gcc)
if (midi_out_handle) midiOutClose(midi_out_handle);
#endif
// @fixme: osx
// https://developer.apple.com/library/archive/samplecode/PlaySoftMIDI/Listings/main_cpp.html#//apple_ref/doc/uid/DTS40008635-main_cpp-DontLinkElementID_4
}
void midi_send(unsigned midi_msg) {
#if is(win32) && !is(gcc)
if( midi_out_handle ) {
midiOutShortMsg(midi_out_handle, midi_msg);
}
#elif is(osx)
if( midi_out_handle ) {
MusicDeviceMIDIEvent(midi_out_handle, (midi_msg) & 0xFF, (midi_msg >> 8) & 0xFF, (midi_msg >> 16) & 0xFF, 0);
}
#endif
}
// encapsulate drwav,drmp3,stbvorbis and some buffer with the sts_mixer_stream_t
enum { UNK, WAV, OGG, MP1, MP3 };
typedef struct {
int type;
union {
drwav wav;
stb_vorbis *ogg;
void *opaque;
drmp3 mp3_;
};
sts_mixer_stream_t stream; // mixer stream
union {
int32_t data[4096*2]; // static sample buffer
float dataf[4096*2];
};
bool rewind;
} mystream_t;
static void downsample_to_mono_flt( int channels, float *buffer, int samples ) {
if( channels > 1 ) {
float *output = buffer;
while( samples-- > 0 ) {
float mix = 0;
for( int i = 0; i < channels; ++i ) mix += *buffer++;
*output++ = (float)(mix / channels);
}
}
}
static void downsample_to_mono_s16( int channels, short *buffer, int samples ) {
if( channels > 1 ) {
short *output = buffer;
while( samples-- > 0 ) {
float mix = 0;
for( int i = 0; i < channels; ++i ) mix += *buffer++;
*output++ = (short)(mix / channels);
}
}
}
// the callback to refill the (stereo) stream data
static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
mystream_t* stream = (mystream_t*)userdata;
switch( stream->type ) {
default:
break; case WAV: {
int sl = sample->length / 2; /*sample->channels*/;
if( stream->rewind ) stream->rewind = 0, drwav_seek_to_pcm_frame(&stream->wav, 0);
if (drwav_read_pcm_frames_s16(&stream->wav, sl, (short*)stream->data) < sl) {
drwav_seek_to_pcm_frame(&stream->wav, 0);
}
}
break; case MP3: {
int sl = sample->length / 2; /*sample->channels*/;
if( stream->rewind ) stream->rewind = 0, drmp3_seek_to_pcm_frame(&stream->mp3_, 0);
if (drmp3_read_pcm_frames_f32(&stream->mp3_, sl, stream->dataf) < sl) {
drmp3_seek_to_pcm_frame(&stream->mp3_, 0);
}
}
break; case OGG: {
stb_vorbis *ogg = (stb_vorbis*)stream->ogg;
if( stream->rewind ) stream->rewind = 0, stb_vorbis_seek(stream->ogg, 0);
if( stb_vorbis_get_samples_short_interleaved(ogg, 2, (short*)stream->data, sample->length) == 0 ) {
stb_vorbis_seek(stream->ogg, 0);
}
}
}
}
static void reset_stream(mystream_t* stream) {
if( stream ) memset( stream->data, 0, sizeof(stream->data) ), stream->rewind = 1;
}
// load a (stereo) stream
static bool load_stream(mystream_t* stream, const char *filename) {
int datalen;
char *data = vfs_load(filename, &datalen); if(!data) return false;
int error;
int HZ = 44100;
stream->type = UNK;
if( stream->type == UNK && (stream->ogg = stb_vorbis_open_memory((const unsigned char *)data, datalen, &error, NULL)) ) {
stb_vorbis_info info = stb_vorbis_get_info(stream->ogg);
if( info.channels != 2 ) { puts("cannot stream ogg file. stereo required."); goto end; } // @fixme: upsample
stream->type = OGG;
stream->stream.sample.frequency = info.sample_rate;
stream->stream.sample.audio_format = STS_MIXER_SAMPLE_FORMAT_16;
}
if( stream->type == UNK && drwav_init_memory(&stream->wav, data, datalen, NULL)) {
if( stream->wav.channels != 2 ) { puts("cannot stream wav file. stereo required."); goto end; } // @fixme: upsample
stream->type = WAV;
stream->stream.sample.frequency = stream->wav.sampleRate;
stream->stream.sample.audio_format = STS_MIXER_SAMPLE_FORMAT_16;
}
drmp3_config mp3_cfg = { 2, HZ };
if( stream->type == UNK && (drmp3_init_memory(&stream->mp3_, data, datalen, NULL/*&mp3_cfg*/) != 0) ) {
stream->type = MP3;
stream->stream.sample.frequency = stream->mp3_.sampleRate;
stream->stream.sample.audio_format = STS_MIXER_SAMPLE_FORMAT_FLOAT;
}
if( stream->type == UNK ) {
return false;
}
end:;
stream->stream.userdata = stream;
stream->stream.callback = refill_stream;
stream->stream.sample.length = sizeof(stream->data) / sizeof(stream->data[0]);
stream->stream.sample.data = stream->data;
refill_stream(&stream->stream.sample, stream);
return true;
}
// load a (mono) sample
static bool load_sample(sts_mixer_sample_t* sample, const char *filename) {
int datalen;
char *data = vfs_load(filename, &datalen); if(!data) return false;
int error;
int channels = 0;
if( !channels ) for( drwav w = {0}, *wav = &w; wav && drwav_init_memory(wav, data, datalen, NULL); wav = 0 ) {
channels = wav->channels;
sample->frequency = wav->sampleRate;
sample->audio_format = STS_MIXER_SAMPLE_FORMAT_16;
sample->length = wav->totalPCMFrameCount;
sample->data = REALLOC(0, sample->length * sizeof(short) * channels);
drwav_read_pcm_frames_s16(wav, sample->length, (short*)sample->data);
drwav_uninit(wav);
}
if( !channels ) for( stb_vorbis *ogg = stb_vorbis_open_memory((const unsigned char *)data, datalen, &error, NULL); ogg; ogg = 0 ) {
stb_vorbis_info info = stb_vorbis_get_info(ogg);
channels = info.channels;
sample->frequency = info.sample_rate;
sample->audio_format = STS_MIXER_SAMPLE_FORMAT_16;
sample->length = (int)stb_vorbis_stream_length_in_samples(ogg);
stb_vorbis_close(ogg);
short *buffer;
int sample_rate;
stb_vorbis_decode_memory((const unsigned char *)data, datalen, &channels, &sample_rate, (short **)&buffer);
sample->data = buffer;
}
drmp3_config mp3_cfg = { 2, 44100 };
drmp3_uint64 mp3_fc;
if( !channels ) for( short *fbuf = drmp3_open_memory_and_read_pcm_frames_s16(data, datalen, &mp3_cfg, &mp3_fc, NULL); fbuf ; fbuf = 0 ) {
channels = mp3_cfg.channels;
sample->frequency = mp3_cfg.sampleRate;
sample->audio_format = STS_MIXER_SAMPLE_FORMAT_16;
sample->length = mp3_fc; // / sizeof(float) / mp3_cfg.channels;
sample->data = fbuf;
}
if( !channels ) {
short *output = 0;
int outputSize, hz, mp1channels;
bool ok = jo_read_mp1(data, datalen, &output, &outputSize, &hz, &mp1channels);
if( ok ) {
channels = mp1channels;
sample->frequency = hz;
sample->audio_format = STS_MIXER_SAMPLE_FORMAT_16;
sample->length = outputSize / sizeof(int16_t) / channels;
sample->data = output; // REALLOC(0, sample->length * sizeof(int16_t) * channels );
// memcpy( sample->data, output, outputSize );
}
}
if( !channels ) {
return false;
}
if( channels > 1 ) {
if( sample->audio_format == STS_MIXER_SAMPLE_FORMAT_FLOAT ) {
downsample_to_mono_flt( channels, sample->data, sample->length );
sample->data = REALLOC( sample->data, sample->length * sizeof(float));
}
else
if( sample->audio_format == STS_MIXER_SAMPLE_FORMAT_16 ) {
downsample_to_mono_s16( channels, sample->data, sample->length );
sample->data = REALLOC( sample->data, sample->length * sizeof(short));
}
else {
puts("error!"); // @fixme
}
}
return true;
}
// -----------------------------------------------------------------------------
static ma_device device;
static ma_context context;
static sts_mixer_t mixer;
// This is the function that's used for sending more data to the device for playback.
static ma_uint32 audio_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount) {
int len = frameCount;
sts_mixer_mix_audio(&mixer, pOutput, len / (sizeof(int32_t) / 4));
(void)pDevice; (void)pInput;
return len / (sizeof(int32_t) / 4);
}
void audio_drop(void) {
ma_device_stop(&device);
ma_device_uninit(&device);
ma_context_uninit(&context);
}
int audio_init( int flags ) {
atexit(audio_drop);
// init sts_mixer
sts_mixer_init(&mixer, 44100, STS_MIXER_SAMPLE_FORMAT_32);
// The prioritization of backends can be controlled by the application. You need only specify the backends
// you care about. If the context cannot be initialized for any of the specified backends ma_context_init()
// will fail.
ma_backend backends[] = {
#if 1
ma_backend_wasapi, // Higest priority.
ma_backend_dsound,
ma_backend_winmm,
ma_backend_pulseaudio,
ma_backend_alsa,
ma_backend_oss,
ma_backend_jack,
ma_backend_opensl,
ma_backend_webaudio,
//ma_backend_openal,
//ma_backend_sdl,
ma_backend_null // Lowest priority.
#else
// Highest priority
ma_backend_wasapi, // WASAPI | Windows Vista+
ma_backend_dsound, // DirectSound | Windows XP+
ma_backend_winmm, // WinMM | Windows XP+ (may work on older versions, but untested)
ma_backend_coreaudio, // Core Audio | macOS, iOS
ma_backend_pulseaudio, // PulseAudio | Cross Platform (disabled on Windows, BSD and Android)
ma_backend_alsa, // ALSA | Linux
ma_backend_oss, // OSS | FreeBSD
ma_backend_jack, // JACK | Cross Platform (disabled on BSD and Android)
ma_backend_opensl, // OpenSL ES | Android (API level 16+)
ma_backend_webaudio, // Web Audio | Web (via Emscripten)
ma_backend_sndio, // sndio | OpenBSD
ma_backend_audio4, // audio(4) | NetBSD, OpenBSD
ma_backend_aaudio, // AAudio | Android 8+
ma_backend_custom, // Custom | Cross Platform
ma_backend_null, // Null | Cross Platform (not used on Web)
// Lowest priority
#endif
};
if (ma_context_init(backends, countof(backends), NULL, &context) != MA_SUCCESS) {
PRINTF("%s\n", "Failed to initialize audio context.");
return false;
}
ma_device_config config = ma_device_config_init(ma_device_type_playback); // Or ma_device_type_capture or ma_device_type_duplex.
config.playback.pDeviceID = NULL; // &myPlaybackDeviceID; // Or NULL for the default playback device.
config.playback.format = ma_format_s32;
config.playback.channels = 2;
config.sampleRate = 44100;
config.dataCallback = (void*)audio_callback; //< @r-lyeh add void* cast
config.pUserData = NULL;
if (ma_device_init(NULL, &config, &device) != MA_SUCCESS) {
printf("Failed to open playback device.");
ma_context_uninit(&context);
return false;
}
(void)flags;
ma_device_start(&device);
return true;
}
typedef struct audio_handle {
bool is_clip;
bool is_stream;
union {
sts_mixer_sample_t clip;
mystream_t stream;
};
} audio_handle;
static array(audio_handle*) audio_instances;
audio_t audio_clip( const char *pathfile ) {
audio_handle *a = REALLOC(0, sizeof(audio_handle) );
memset(a, 0, sizeof(audio_handle));
a->is_clip = load_sample( &a->clip, pathfile );
array_push(audio_instances, a);
return a;
}
audio_t audio_stream( const char *pathfile ) {
audio_handle *a = REALLOC(0, sizeof(audio_handle) );
memset(a, 0, sizeof(audio_handle));
a->is_stream = load_stream( &a->stream, pathfile );
array_push(audio_instances, a);
return a;
}
static float volume_clip = 1, volume_stream = 1, volume_master = 1;
float audio_volume_clip(float gain) {
if( gain >= 0 && gain <= 1 ) volume_clip = gain * gain;
// patch all live clips
for(int i = 0, active = 0; i < STS_MIXER_VOICES; ++i) {
if(mixer.voices[i].state != STS_MIXER_VOICE_STOPPED) // is_active?
if( mixer.voices[i].sample ) // is_sample?
mixer.voices[i].gain = volume_clip;
}
return sqrt( volume_clip );
}
float audio_volume_stream(float gain) {
if( gain >= 0 && gain <= 1 ) volume_stream = gain * gain;
// patch all live streams
for(int i = 0, active = 0; i < STS_MIXER_VOICES; ++i) {
if(mixer.voices[i].state != STS_MIXER_VOICE_STOPPED) // is_active?
if( mixer.voices[i].stream ) // is_stream?
mixer.voices[i].gain = volume_stream;
}
return sqrt( volume_stream );
}
float audio_volume_master(float gain) {
if( gain >= 0 && gain <= 1 ) volume_master = gain * gain;
// patch global mixer
mixer.gain = volume_master;
return sqrt( volume_master );
}
int audio_play_gain_pitch_pan( audio_t a, int flags, float gain, float pitch, float pan ) {
static bool muted = 0; do_once muted = flag("--mute") || flag("--muted");
if(muted) return 1;
if( flags & AUDIO_IGNORE_MIXER_GAIN ) {
// do nothing, gain used as-is
} else {
// apply mixer gains on top
gain += a->is_clip ? volume_clip : volume_stream;
}
if( flags & AUDIO_SINGLE_INSTANCE ) {
audio_stop( a );
}
// gain: [0..+1], pitch: (0..N], pan: [-1..+1]
if( a->is_clip ) {
int voice = sts_mixer_play_sample(&mixer, &a->clip, gain, pitch, pan);
if( voice == -1 ) return 0; // all voices busy
}
if( a->is_stream ) {
int voice = sts_mixer_play_stream(&mixer, &a->stream.stream, gain);
if( voice == -1 ) return 0; // all voices busy
}
return 1;
}
int audio_play_gain_pitch( audio_t a, int flags, float gain, float pitch ) {
return audio_play_gain_pitch_pan(a, flags, gain, pitch, 0);
}
int audio_play_gain( audio_t a, int flags, float gain ) {
return audio_play_gain_pitch(a, flags, gain, 1.f);
}
int audio_play( audio_t a, int flags ) {
return audio_play_gain(a, flags & ~AUDIO_IGNORE_MIXER_GAIN, 0.f);
}
int audio_stop( audio_t a ) {
if( a->is_clip ) {
sts_mixer_stop_sample(&mixer, &a->clip);
}
if( a->is_stream ) {
sts_mixer_stop_stream(&mixer, &a->stream.stream);
reset_stream(&a->stream);
}
return 1;
}
// -----------------------------------------------------------------------------
// audio queue
#ifndef AUDIO_QUEUE_BUFFERING_MS
#define AUDIO_QUEUE_BUFFERING_MS 50 // 10 // 100
#endif
#ifndef AUDIO_QUEUE_MAX
#define AUDIO_QUEUE_MAX 2048
#endif
#ifndef AUDIO_QUEUE_TIMEOUT
#define AUDIO_QUEUE_TIMEOUT ifdef(win32, THREAD_QUEUE_WAIT_INFINITE, 500)
#endif
typedef struct audio_queue_t {
int cursor;
int avail;
unsigned flags;
char data[0];
} audio_queue_t;
static thread_queue_t queue_mutex;
static void audio_queue_init() {
static void* audio_queues[AUDIO_QUEUE_MAX] = {0};
do_once thread_queue_init(&queue_mutex, countof(audio_queues), audio_queues, 0);
}
static void audio_queue_callback(sts_mixer_sample_t* sample, void* userdata) {
(void)userdata;
int sl = sample->length / 2; // 2 ch
int bytes = sl * 2 * (sample->audio_format == STS_MIXER_SAMPLE_FORMAT_16 ? 2 : 4);
char *dst = sample->data;
static audio_queue_t *aq = 0;
do {
while( !aq ) aq = (audio_queue_t*)thread_queue_consume(&queue_mutex, THREAD_QUEUE_WAIT_INFINITE);
int len = aq->avail > bytes ? bytes : aq->avail;
memcpy(dst, (char*)aq->data + aq->cursor, len);
dst += len;
bytes -= len;
aq->cursor += len;
aq->avail -= len;
if( aq->avail <= 0 ) {
FREE(aq); // @fixme: mattias' original thread_queue_consume() implementation crashes here on tcc+win because of a double free on same pointer. using mcmp for now
aq = 0;
}
} while( bytes > 0 );
}
static int audio_queue_voice = -1;
void audio_queue_clear() {
do_once audio_queue_init();
sts_mixer_stop_voice(&mixer, audio_queue_voice);
audio_queue_voice = -1;
}
int audio_queue( const void *samples, int num_samples, int flags ) {
do_once audio_queue_init();
float gain = 1; // [0..1]
float pitch = 1; // (0..N]
float pan = 0; // [-1..1]
int bits = flags & AUDIO_8 ? 8 : flags & (AUDIO_32|AUDIO_FLOAT) ? 32 : 16;
int channels = flags & AUDIO_2CH ? 2 : 1;
int bytes_per_sample = channels * (bits / 8);
int bytes = num_samples * bytes_per_sample;
static sts_mixer_stream_t q = { 0 };
if( audio_queue_voice < 0 ) {
void *reuse_ptr = q.sample.data;
q = ((sts_mixer_stream_t){0});
q.sample.data = reuse_ptr;
q.callback = audio_queue_callback;
q.sample.frequency = flags & AUDIO_8KHZ ? 8000 : flags & AUDIO_11KHZ ? 11025 : flags & AUDIO_44KHZ ? 44100 : flags & AUDIO_32KHZ ? 32000 : 22050;
q.sample.audio_format = flags & AUDIO_FLOAT ? STS_MIXER_SAMPLE_FORMAT_FLOAT : STS_MIXER_SAMPLE_FORMAT_16;
q.sample.length = q.sample.frequency / (1000 / AUDIO_QUEUE_BUFFERING_MS); // num_samples;
int bytes = q.sample.length * 2 * (flags & AUDIO_FLOAT ? 4 : 2);
q.sample.data = memset(REALLOC(q.sample.data, bytes), 0, bytes);
audio_queue_voice = sts_mixer_play_stream(&mixer, &q, gain * 1.f);
if( audio_queue_voice < 0 ) return 0;
}
audio_queue_t *aq = MALLOC(sizeof(audio_queue_t) + (bytes << (channels == 1))); // dupe space if going to be converted from mono to stereo
aq->cursor = 0;
aq->avail = bytes;
aq->flags = flags;
if( !samples ) {
memset(aq->data, 0, bytes);
} else {
// @todo: convert from other source formats to target format in here: add AUDIO_8, AUDIO_32
if( channels == 1 ) {
// mixer accepts stereo samples only; so resample mono to stereo if needed
for( int i = 0; i < num_samples; ++i ) {
memcpy((char*)aq->data + (i*2+0) * bytes_per_sample, (char*)samples + i * bytes_per_sample, bytes_per_sample );
memcpy((char*)aq->data + (i*2+1) * bytes_per_sample, (char*)samples + i * bytes_per_sample, bytes_per_sample );
}
} else {
memcpy(aq->data, samples, bytes);
}
}
while( !thread_queue_produce(&queue_mutex, aq, THREAD_QUEUE_WAIT_INFINITE) ) {}
return audio_queue_voice;
}
int ui_audio() {
int changed = 0;
float sfx = sqrt(volume_clip), bgm = sqrt(volume_stream), master = sqrt(volume_master);
if( ui_slider2("BGM volume", &bgm, va("%.2f", bgm))) changed = 1, audio_volume_stream(bgm);
if( ui_slider2("SFX volume", &sfx, va("%.2f", sfx))) changed = 1, audio_volume_clip(sfx);
if( ui_slider2("Master volume", &master, va("%.2f", master))) changed = 1, audio_volume_master(master);
ui_separator();
int num_voices = sts_mixer_get_active_voices(&mixer);
ui_label2("Format", mixer.audio_format == 0 ? "None" : mixer.audio_format == 1 ? "8-bit" : mixer.audio_format == 2 ? "16-bit" : mixer.audio_format == 3 ? "32-bit integer" : "32-bit float");
ui_label2("Frequency", va("%4.1f KHz", mixer.frequency / 1000.0));
ui_label2("Voices", va("%d/%d", num_voices, STS_MIXER_VOICES));
ui_separator();
for( int i = 0; i < STS_MIXER_VOICES; ++i ) {
if( mixer.voices[i].state != STS_MIXER_VOICE_STOPPED ) { // PLAYING || STREAMING
ui_label(va("Voice %d", i+1));
// float mul = mixer.voices[i].state == STS_MIXER_VOICE_STREAMING ? 2 : 1;
// float div = mixer.voices[i].state == STS_MIXER_VOICE_STREAMING ? mixer.voices[i].stream->sample.length : mixer.voices[i].sample->length;
// float pct = mixer.voices[i].position * mul / div;
// if(ui_slider2("Position", &pct, va("%5.2f", pct))) changed = 1;
if(ui_slider2("Gain", &mixer.voices[i].gain, va("%5.2f", mixer.voices[i].gain))) changed = 1;
if(ui_slider2("Pitch", &mixer.voices[i].pitch, va("%5.2f", mixer.voices[i].pitch))) changed = 1;
if(ui_slider2("Pan", &mixer.voices[i].pan, va("%5.2f", mixer.voices[i].pan))) changed = 1;
ui_separator();
}
}
return changed;
}
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