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audio improvements

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This commit is contained in:
Tuomas Katajisto 2026-10-17 15:07:05 +03:00
parent 4bf23b1196
commit 40f4f596aa
4 changed files with 92 additions and 39 deletions
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4
modules/stb_vorbis/bindings.jai
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@ -1,7 +1,7 @@
// //
// This file was auto-generated using the following command: // This file was auto-generated using the following command:
// //
// jai modules/stb_vorbis/generate.jai // jai /home/matias/trueno//modules/stb_vorbis/generate.jai modules +jails_diagnostics import_dir /home/matias/.vscode/extensions/apparentlystudio.jails-0.2.22/out/metaprogram
// //
@ -265,8 +265,6 @@ STBVorbisError :: enum u32 {
} else #if CPU == .ARM64 { } else #if CPU == .ARM64 {
lib_stb_vorbis :: #library "switch2/stb_vorbis"; lib_stb_vorbis :: #library "switch2/stb_vorbis";
} }
} else #if OS == .WASM {
lib_stb_vorbis :: #library "wasm/stb_vorbis";
} else { } else {
#assert false; #assert false;
} }

2
settings.cfg
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@ -1,4 +1,4 @@
master_volume 0.205032 master_volume 0.205032
music_volume 1 music_volume 1
sfx_volume 1 sfx_volume 1
fullscreen 1 fullscreen 0

124
src/audio/mixer.jai
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@ -15,6 +15,8 @@ Play_Mode :: enum {
Mixer_Play_Task :: struct { Mixer_Play_Task :: struct {
audio : *Audio_Data; audio : *Audio_Data;
audio_proc : (n_samples: s32, audio: *Audio_Data, buffer_out: *float,channel_count: s32,volume: float, state: *void) -> (n_out: s32);
audio_state : *void;
bus : Mixer_Bus; bus : Mixer_Bus;
mode : Play_Mode; mode : Play_Mode;
unique : bool = true; unique : bool = true;
@ -65,11 +67,12 @@ mixer_add_task :: (task: Mixer_Play_Task) {
array_add(*g_mixer.tasks, task); array_add(*g_mixer.tasks, task);
} }
mixer_add_task :: (audio: *Audio_Data, bus: Mixer_Bus, mode: Play_Mode) { mixer_add_task :: (audio: *Audio_Data, bus: Mixer_Bus, mode: Play_Mode, state: *void) {
mixer_add_task(.{ mixer_add_task(.{
audio = audio, audio = audio,
bus = bus, bus = bus,
mode = mode mode = mode,
audio_state = state
}); });
} }
@ -103,6 +106,7 @@ mixer_get_samples :: (buffer: *float, frame_count: s32, channel_count: s32) {
task := *g_mixer.tasks[i]; task := *g_mixer.tasks[i];
if !task.audio { if !task.audio {
free(task.audio_state);
array_unordered_remove_by_index(*g_mixer.tasks, i); array_unordered_remove_by_index(*g_mixer.tasks, i);
continue; continue;
} }
@ -124,43 +128,52 @@ mixer_get_samples :: (buffer: *float, frame_count: s32, channel_count: s32) {
task_finished := false; task_finished := false;
for f : 0..frame_count-1 { assert(src_rate == out_rate); // TODO for now
src_pos := cast(float64)(task.curSample + f) * cast(float64)src_rate / cast(float64)out_rate;
src_frame0 := cast(s64) src_pos;
src_frame1 := src_frame0 + 1;
t := cast(float)(src_pos - cast(float64)src_frame0);
if src_frame0 >= num_src_frames { n_received_samples := task.audio_proc(frame_count,task.audio,buffer,channel_count,vol,task.audio_state);
if task.mode == .ONESHOT { if(n_received_samples < frame_count){
task_finished = true; task_finished = true;
break;
} else {
src_frame0 = src_frame0 % num_src_frames;
src_frame1 = src_frame1 % num_src_frames;
}
} else if src_frame1 >= num_src_frames {
// At the very last source frame: clamp for ONESHOT, wrap for REPEAT.
src_frame1 = ifx task.mode == .ONESHOT then src_frame0 else 0;
}
for ch : 0..channel_count-1 {
out_index := (f * channel_count) + ch;
src_ch := ifx source_channels == 1 then 0 else ch;
s0 := source_data[src_frame0 * source_channels + src_ch];
s1 := source_data[src_frame1 * source_channels + src_ch];
buffer[out_index] += (s0 + (s1 - s0) * t) * vol;
}
} }
task.curSample += frame_count;
// Keep curSample bounded for REPEAT tracks so it never overflows.
if task.mode == .REPEAT && out_rate > 0 && src_rate > 0 { // for f : 0..frame_count-1 {
loop_period := num_src_frames * out_rate / src_rate; // src_pos := cast(float64)(task.curSample + f) * cast(float64)src_rate / cast(float64)out_rate;
if loop_period > 0 then task.curSample = task.curSample % loop_period; // src_frame0 := cast(s64) src_pos;
} // src_frame1 := src_frame0 + 1;
// t := cast(float)(src_pos - cast(float64)src_frame0);
// if src_frame0 >= num_src_frames {
// if task.mode == .ONESHOT {
// task_finished = true;
// break;
// } else {
// src_frame0 = src_frame0 % num_src_frames;
// src_frame1 = src_frame1 % num_src_frames;
// }
// } else if src_frame1 >= num_src_frames {
// // At the very last source frame: clamp for ONESHOT, wrap for REPEAT.
// src_frame1 = ifx task.mode == .ONESHOT then src_frame0 else 0;
// }
// for ch : 0..channel_count-1 {
// out_index := (f * channel_count) + ch;
// src_ch := ifx source_channels == 1 then 0 else ch;
// s0 := source_data[src_frame0 * source_channels + src_ch];
// s1 := source_data[src_frame1 * source_channels + src_ch];
// buffer[out_index] += (s0 + (s1 - s0) * t) * vol;
// }
// }
// task.curSample += frame_count;
// // Keep curSample bounded for REPEAT tracks so it never overflows.
// if task.mode == .REPEAT && out_rate > 0 && src_rate > 0 {
// loop_period := num_src_frames * out_rate / src_rate;
// if loop_period > 0 then task.curSample = task.curSample % loop_period;
// }
if task_finished { if task_finished {
if task.queued_next { if task.queued_next {
@ -168,6 +181,7 @@ mixer_get_samples :: (buffer: *float, frame_count: s32, channel_count: s32) {
free(task.queued_next); free(task.queued_next);
g_mixer.tasks[i] = next; g_mixer.tasks[i] = next;
} else { } else {
free(task.audio_state);
array_unordered_remove_by_index(*g_mixer.tasks, i); array_unordered_remove_by_index(*g_mixer.tasks, i);
} }
} }
@ -190,3 +204,43 @@ mixer_unlock :: (mixer: *Mixer) {
} }
} }
Simple_State :: struct {
cur_sample : s64 = 0;
}
trivial_data_proc_norepeat::(n_samples: s32, audio: *Audio_Data, buffer_out: *float,channel_count: s32,volume: float, state: *void) -> (n_out: s32){
return trivial_data_proc(n_samples,audio,buffer_out,channel_count,volume,state,false);
}
trivial_data_proc_repeat::(n_samples: s32, audio: *Audio_Data, buffer_out: *float,channel_count: s32,volume: float, state: *void) -> (n_out: s32){
return trivial_data_proc(n_samples,audio,buffer_out,channel_count,volume,state,true);
}
trivial_data_proc::(n_samples: s32, audio: *Audio_Data, buffer_out: *float,channel_count: s32,volume: float, state: *void, repeat:bool) -> (n_out: s32){
state_cast : *Simple_State = cast(*Simple_State)state;
source_channels := cast(s64) audio.channels;
source_samples := cast(s64) audio.data.count / source_channels;
samples_read := 0;
for block_sample : 0..n_samples-1{
sample : s64 = block_sample + state_cast.cur_sample;
if(!repeat && sample >= source_samples) {
break;
}
sample %= source_samples;
for ch : 0..channel_count-1{
out_index := (block_sample * channel_count) + ch;
src_ch := ifx source_channels == 1 then 0 else ch;
buffer_out[out_index] += audio.data[sample*source_channels + ch] * volume;
}
samples_read += 1;
}
state_cast.cur_sample += samples_read;
state_cast.cur_sample %= source_samples;
return cast(s32)samples_read;
}

1
src/main.jai
View File

@ -110,6 +110,7 @@ init :: () {
saudio_setup(*(saudio_desc.{ saudio_setup(*(saudio_desc.{
logger = .{ func = slog_func }, logger = .{ func = slog_func },
stream_cb = sokol_audio_callback, stream_cb = sokol_audio_callback,
num_channels = 2,
})); }));
g_audio_sample_rate = saudio_sample_rate(); g_audio_sample_rate = saudio_sample_rate();
stm_setup(); stm_setup();