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Revert "revert to older miniaudio" 

This reverts commit 8834e1671f.
main
Dominik Madarász 2023-10-11 21:23:13 +02:00
parent 8834e1671f
commit b240bfc031
6 changed files with 44044 additions and 40189 deletions
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2
README.md
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@ -78,7 +78,7 @@ echo win/mingw && gcc hello.c -lws2_32 -lwinmm -ldbghelp -lole32 -luse
echo win/clang && clang hello.c -lws2_32 -lwinmm -ldbghelp -lole32 -luser32 -lgdi32 -lcomdlg32
echo linux && cc hello.c -lm -ldl -lpthread -lX11
echo linux/tcc && tcc hello.c -lm -ldl -lpthread -lX11 -D__STDC_NO_VLA__
echo osx && cc -ObjC hello.c -framework cocoa -framework iokit -framework audiotoolbox -framework coreaudio
echo osx && cc -ObjC hello.c -framework cocoa -framework iokit -framework audiotoolbox
```
## Cook

28077
engine/joint/v4k.h
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File diff suppressed because it is too large Load Diff

27993
engine/split/3rd_miniaudio.h
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File diff suppressed because it is too large Load Diff

84
engine/split/v4k_audio.c
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@ -1,4 +1,4 @@
// @fixme: really shutdown audio & related threads before quitting. drwav crashes.
// @fixme: really shutdown audio & related threads before quitting. ma_dr_wav crashes.
#if is(win32) && !is(gcc)
@ -54,15 +54,15 @@ void midi_send(unsigned midi_msg) {
#endif
}
// encapsulate drwav,drmp3,stbvorbis and some buffer with the sts_mixer_stream_t
// encapsulate ma_dr_wav,ma_dr_mp3,stbvorbis and some buffer with the sts_mixer_stream_t
enum { UNK, WAV, OGG, MP1, MP3 };
typedef struct {
int type;
union {
drwav wav;
ma_dr_wav wav;
stb_vorbis *ogg;
void *opaque;
drmp3 mp3_;
ma_dr_mp3 mp3_;
};
sts_mixer_stream_t stream; // mixer stream
union {
@ -100,16 +100,16 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
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);
if( stream->rewind ) stream->rewind = 0, ma_dr_wav_seek_to_pcm_frame(&stream->wav, 0);
if (ma_dr_wav_read_pcm_frames_s16(&stream->wav, sl, (short*)stream->data) < sl) {
ma_dr_wav_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);
if( stream->rewind ) stream->rewind = 0, ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
if (ma_dr_mp3_read_pcm_frames_f32(&stream->mp3_, sl, stream->dataf) < sl) {
ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
}
}
break; case OGG: {
@ -140,14 +140,14 @@ static bool load_stream(mystream_t* stream, const char *filename) {
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->type == UNK && ma_dr_wav_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) ) {
ma_dr_mp3_config mp3_cfg = { 2, HZ };
if( stream->type == UNK && (ma_dr_mp3_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;
@ -175,14 +175,14 @@ static bool load_sample(sts_mixer_sample_t* sample, const char *filename) {
int error;
int channels = 0;
if( !channels ) for( drwav w = {0}, *wav = &w; wav && drwav_init_memory(wav, data, datalen, NULL); wav = 0 ) {
if( !channels ) for( ma_dr_wav w = {0}, *wav = &w; wav && ma_dr_wav_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);
ma_dr_wav_read_pcm_frames_s16(wav, sample->length, (short*)sample->data);
ma_dr_wav_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);
@ -197,9 +197,9 @@ static bool load_sample(sts_mixer_sample_t* sample, const char *filename) {
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 ) {
ma_dr_mp3_config mp3_cfg = { 2, 44100 };
ma_uint64 mp3_fc;
if( !channels ) for( short *fbuf = ma_dr_mp3_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;
@ -249,11 +249,11 @@ 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) {
static void 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);
// return len / (sizeof(int32_t) / 4);
}
void audio_drop(void) {
@ -277,12 +277,13 @@ int audio_init( int flags ) {
ma_backend_dsound,
ma_backend_winmm,
ma_backend_pulseaudio,
ma_backend_coreaudio,
ma_backend_alsa,
ma_backend_oss,
ma_backend_jack,
ma_backend_opensl,
//ma_backend_webaudio,
//ma_backend_openal,
// ma_backend_webaudio,
// ma_backend_openal,
//ma_backend_sdl,
ma_backend_null // Lowest priority.
#else
@ -386,6 +387,9 @@ float audio_volume_master(float gain) {
}
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 {
@ -543,3 +547,37 @@ int audio_queue( const void *samples, int num_samples, int flags ) {
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;
}

27993
engine/v4k
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84
engine/v4k.c
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@ -1043,7 +1043,7 @@ static void v4k_post_init(float);
#line 0
#line 1 "v4k_audio.c"
// @fixme: really shutdown audio & related threads before quitting. drwav crashes.
// @fixme: really shutdown audio & related threads before quitting. ma_dr_wav crashes.
#if is(win32) && !is(gcc)
@ -1099,15 +1099,15 @@ void midi_send(unsigned midi_msg) {
#endif
}
// encapsulate drwav,drmp3,stbvorbis and some buffer with the sts_mixer_stream_t
// encapsulate ma_dr_wav,ma_dr_mp3,stbvorbis and some buffer with the sts_mixer_stream_t
enum { UNK, WAV, OGG, MP1, MP3 };
typedef struct {
int type;
union {
drwav wav;
ma_dr_wav wav;
stb_vorbis *ogg;
void *opaque;
drmp3 mp3_;
ma_dr_mp3 mp3_;
};
sts_mixer_stream_t stream; // mixer stream
union {
@ -1145,16 +1145,16 @@ static void refill_stream(sts_mixer_sample_t* sample, void* userdata) {
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);
if( stream->rewind ) stream->rewind = 0, ma_dr_wav_seek_to_pcm_frame(&stream->wav, 0);
if (ma_dr_wav_read_pcm_frames_s16(&stream->wav, sl, (short*)stream->data) < sl) {
ma_dr_wav_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);
if( stream->rewind ) stream->rewind = 0, ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
if (ma_dr_mp3_read_pcm_frames_f32(&stream->mp3_, sl, stream->dataf) < sl) {
ma_dr_mp3_seek_to_pcm_frame(&stream->mp3_, 0);
}
}
break; case OGG: {
@ -1185,14 +1185,14 @@ static bool load_stream(mystream_t* stream, const char *filename) {
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->type == UNK && ma_dr_wav_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) ) {
ma_dr_mp3_config mp3_cfg = { 2, HZ };
if( stream->type == UNK && (ma_dr_mp3_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;
@ -1220,14 +1220,14 @@ static bool load_sample(sts_mixer_sample_t* sample, const char *filename) {
int error;
int channels = 0;
if( !channels ) for( drwav w = {0}, *wav = &w; wav && drwav_init_memory(wav, data, datalen, NULL); wav = 0 ) {
if( !channels ) for( ma_dr_wav w = {0}, *wav = &w; wav && ma_dr_wav_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);
ma_dr_wav_read_pcm_frames_s16(wav, sample->length, (short*)sample->data);
ma_dr_wav_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);
@ -1242,9 +1242,9 @@ static bool load_sample(sts_mixer_sample_t* sample, const char *filename) {
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 ) {
ma_dr_mp3_config mp3_cfg = { 2, 44100 };
ma_uint64 mp3_fc;
if( !channels ) for( short *fbuf = ma_dr_mp3_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;
@ -1294,11 +1294,11 @@ 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) {
static void 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);
// return len / (sizeof(int32_t) / 4);
}
void audio_drop(void) {
@ -1322,12 +1322,13 @@ int audio_init( int flags ) {
ma_backend_dsound,
ma_backend_winmm,
ma_backend_pulseaudio,
ma_backend_coreaudio,
ma_backend_alsa,
ma_backend_oss,
ma_backend_jack,
ma_backend_opensl,
//ma_backend_webaudio,
//ma_backend_openal,
// ma_backend_webaudio,
// ma_backend_openal,
//ma_backend_sdl,
ma_backend_null // Lowest priority.
#else
@ -1431,6 +1432,9 @@ float audio_volume_master(float gain) {
}
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 {
@ -1588,6 +1592,40 @@ int audio_queue( const void *samples, int num_samples, int flags ) {
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;
}
#line 0
#line 1 "v4k_buffer.c"