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minor fixes

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This commit is contained in:
Anthony Samms
2025-09-16 00:26:06 -04:00
parent 1b31c1c466
commit 02dc2fac95
6 changed files with 26 additions and 23 deletions
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2
.github/workflows/python-app.yml vendored
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@@ -128,7 +128,7 @@ jobs:
- name: Build Executable - name: Build Executable
run: | run: |
uv run nuitka --mode=app --noinclude-setuptools-mode=nofollow --noinclude-IPython-mode=nofollow --assume-yes-for-downloads PyTaiko.py uv run nuitka --lto=yes --mode=app --noinclude-setuptools-mode=nofollow --noinclude-IPython-mode=nofollow --assume-yes-for-downloads PyTaiko.py
- name: Create Release Directory - name: Create Release Directory
run: | run: |

1
config.toml
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@@ -30,7 +30,6 @@ right_kat = [12]
[audio] [audio]
device_type = 0 device_type = 0
buffer_size = 16
sample_rate = -1 sample_rate = -1
exclusive = false exclusive = false

12
libs/audio.py
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@@ -44,7 +44,7 @@ ffi.cdef("""
// Device management // Device management
void list_host_apis(void); void list_host_apis(void);
void init_audio_device(PaHostApiIndex host_api); void init_audio_device(PaHostApiIndex host_api, double sample_rate);
void close_audio_device(void); void close_audio_device(void);
bool is_audio_device_ready(void); bool is_audio_device_ready(void);
void set_master_volume(float volume); void set_master_volume(float volume);
@@ -128,9 +128,11 @@ except OSError as e:
raise raise
class AudioEngine: class AudioEngine:
def __init__(self, device_type: int): def __init__(self, device_type: int, sample_rate: float):
self.device_type = device_type self.device_type = device_type
self.target_sample_rate = 44100 if sample_rate == -1:
sample_rate = 44100
self.target_sample_rate = sample_rate
self.sounds = {} # sound_id -> sound struct self.sounds = {} # sound_id -> sound struct
self.music_streams = {} # music_id -> music struct self.music_streams = {} # music_id -> music struct
self.sound_counter = 0 self.sound_counter = 0
@@ -143,7 +145,7 @@ class AudioEngine:
def init_audio_device(self) -> bool: def init_audio_device(self) -> bool:
"""Initialize the audio device""" """Initialize the audio device"""
try: try:
lib.init_audio_device(self.device_type) lib.init_audio_device(self.device_type, self.target_sample_rate)
self.audio_device_ready = lib.is_audio_device_ready() self.audio_device_ready = lib.is_audio_device_ready()
if self.audio_device_ready: if self.audio_device_ready:
print("Audio device initialized successfully") print("Audio device initialized successfully")
@@ -358,4 +360,4 @@ class AudioEngine:
return 0.0 return 0.0
# Create the global audio instance # Create the global audio instance
audio = AudioEngine(get_config()["audio"]["device_type"]) audio = AudioEngine(get_config()["audio"]["device_type"], get_config()["audio"]["sample_rate"])

29
libs/audio/audio.c
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@@ -32,7 +32,6 @@
#define FREE(ptr) do { if (ptr) { free(ptr); (ptr) = NULL; } } while(0) #define FREE(ptr) do { if (ptr) { free(ptr); (ptr) = NULL; } } while(0)
#define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo #define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo
#define AUDIO_DEVICE_SAMPLE_RATE 44100 // Device output sample rate
// Forward declarations of structures // Forward declarations of structures
struct audio_buffer; struct audio_buffer;
@@ -101,6 +100,7 @@ typedef struct AudioData {
PaStreamParameters outputParameters; // Output stream parameters PaStreamParameters outputParameters; // Output stream parameters
pthread_mutex_t lock; // Mutex lock for thread synchronization pthread_mutex_t lock; // Mutex lock for thread synchronization
bool isReady; // Check if audio device is ready bool isReady; // Check if audio device is ready
double sampleRate;
size_t pcmBufferSize; // Pre-allocated buffer size size_t pcmBufferSize; // Pre-allocated buffer size
void *pcmBuffer; // Pre-allocated buffer to read audio data from file/memory void *pcmBuffer; // Pre-allocated buffer to read audio data from file/memory
float masterVolume; // Master volume control float masterVolume; // Master volume control
@@ -114,7 +114,7 @@ typedef struct AudioData {
// Function declarations // Function declarations
// Device management // Device management
void list_host_apis(void); void list_host_apis(void);
void init_audio_device(PaHostApiIndex host_api); void init_audio_device(PaHostApiIndex host_api, double sample_rate);
void close_audio_device(void); void close_audio_device(void);
bool is_audio_device_ready(void); bool is_audio_device_ready(void);
void set_master_volume(float volume); void set_master_volume(float volume);
@@ -322,7 +322,7 @@ PaDeviceIndex get_best_output_device_for_host_api(PaHostApiIndex hostApi)
} }
// Device management implementations // Device management implementations
void init_audio_device(PaHostApiIndex host_api) void init_audio_device(PaHostApiIndex host_api, double sample_rate)
{ {
// Initialize PortAudio // Initialize PortAudio
PaError err = Pa_Initialize(); PaError err = Pa_Initialize();
@@ -351,12 +351,13 @@ void init_audio_device(PaHostApiIndex host_api)
AUDIO.System.outputParameters.sampleFormat = paFloat32; // Using float format like miniaudio version AUDIO.System.outputParameters.sampleFormat = paFloat32; // Using float format like miniaudio version
AUDIO.System.outputParameters.suggestedLatency = Pa_GetDeviceInfo(AUDIO.System.outputParameters.device)->defaultLowOutputLatency; AUDIO.System.outputParameters.suggestedLatency = Pa_GetDeviceInfo(AUDIO.System.outputParameters.device)->defaultLowOutputLatency;
AUDIO.System.outputParameters.hostApiSpecificStreamInfo = NULL; AUDIO.System.outputParameters.hostApiSpecificStreamInfo = NULL;
AUDIO.System.sampleRate = sample_rate;
// Open the audio stream // Open the audio stream
err = Pa_OpenStream(&AUDIO.System.stream, err = Pa_OpenStream(&AUDIO.System.stream,
NULL, // No input NULL, // No input
&AUDIO.System.outputParameters, // Output parameters &AUDIO.System.outputParameters, // Output parameters
AUDIO_DEVICE_SAMPLE_RATE, // Sample rate sample_rate, // Sample rate
paFramesPerBufferUnspecified, // Frames per buffer (let PortAudio decide) paFramesPerBufferUnspecified, // Frames per buffer (let PortAudio decide)
paClipOff, // No clipping paClipOff, // No clipping
port_audio_callback, // Callback function port_audio_callback, // Callback function
@@ -390,7 +391,7 @@ void init_audio_device(PaHostApiIndex host_api)
TRACELOG(LOG_INFO, " > Device: %s", deviceInfo->name); TRACELOG(LOG_INFO, " > Device: %s", deviceInfo->name);
TRACELOG(LOG_INFO, " > Format: %s", "Float32"); TRACELOG(LOG_INFO, " > Format: %s", "Float32");
TRACELOG(LOG_INFO, " > Channels: %d", AUDIO_DEVICE_CHANNELS); TRACELOG(LOG_INFO, " > Channels: %d", AUDIO_DEVICE_CHANNELS);
TRACELOG(LOG_INFO, " > Sample rate: %d", AUDIO_DEVICE_SAMPLE_RATE); TRACELOG(LOG_INFO, " > Sample rate: %f", AUDIO.System.sampleRate);
TRACELOG(LOG_INFO, " > Latency: %f ms", AUDIO.System.outputParameters.suggestedLatency * 1000.0); TRACELOG(LOG_INFO, " > Latency: %f ms", AUDIO.System.outputParameters.suggestedLatency * 1000.0);
} }
@@ -657,13 +658,13 @@ sound load_sound_from_wave(wave wave) {
struct wave resampled_wave = { 0 }; struct wave resampled_wave = { 0 };
bool is_resampled = false; bool is_resampled = false;
if (wave.sampleRate != AUDIO_DEVICE_SAMPLE_RATE) { if (wave.sampleRate != AUDIO.System.sampleRate) {
TRACELOG(LOG_INFO, "AUDIO: Resampling wave from %d Hz to %d Hz", wave.sampleRate, AUDIO_DEVICE_SAMPLE_RATE); TRACELOG(LOG_INFO, "AUDIO: Resampling wave from %d Hz to %f Hz", wave.sampleRate, AUDIO.System.sampleRate);
SRC_DATA src_data; SRC_DATA src_data;
src_data.data_in = wave.data; src_data.data_in = wave.data;
src_data.input_frames = wave.frameCount; src_data.input_frames = wave.frameCount;
src_data.src_ratio = (double)AUDIO_DEVICE_SAMPLE_RATE / wave.sampleRate; src_data.src_ratio = AUDIO.System.sampleRate / wave.sampleRate;
src_data.output_frames = (sf_count_t)(wave.frameCount * src_data.src_ratio); src_data.output_frames = (sf_count_t)(wave.frameCount * src_data.src_ratio);
resampled_wave.data = calloc(src_data.output_frames * wave.channels, sizeof(float)); resampled_wave.data = calloc(src_data.output_frames * wave.channels, sizeof(float));
@@ -681,7 +682,7 @@ sound load_sound_from_wave(wave wave) {
} }
resampled_wave.frameCount = src_data.output_frames_gen; resampled_wave.frameCount = src_data.output_frames_gen;
resampled_wave.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; resampled_wave.sampleRate = AUDIO.System.sampleRate;
resampled_wave.channels = wave.channels; resampled_wave.channels = wave.channels;
resampled_wave.sampleSize = wave.sampleSize; resampled_wave.sampleSize = wave.sampleSize;
is_resampled = true; is_resampled = true;
@@ -790,7 +791,7 @@ audio_stream load_audio_stream(unsigned int sample_rate, unsigned int sample_siz
stream.channels = channels; stream.channels = channels;
// Pass 1 to indicate this is a streaming buffer // Pass 1 to indicate this is a streaming buffer
stream.buffer = load_audio_buffer(AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 1); stream.buffer = load_audio_buffer(AUDIO_DEVICE_CHANNELS, AUDIO.System.sampleRate, 1);
return stream; return stream;
} }
@@ -884,8 +885,8 @@ music load_music_stream(const char* filename) {
} }
ctx->snd_file = snd_file; ctx->snd_file = snd_file;
if (sf_info.samplerate != AUDIO_DEVICE_SAMPLE_RATE) { if (sf_info.samplerate != AUDIO.System.sampleRate) {
TRACELOG(LOG_INFO, "AUDIO: Resampling music from %d Hz to %d Hz", sf_info.samplerate, AUDIO_DEVICE_SAMPLE_RATE); TRACELOG(LOG_INFO, "AUDIO: Resampling music from %d Hz to %f Hz", sf_info.samplerate, AUDIO.System.sampleRate);
int error; int error;
ctx->resampler = src_new(SRC_SINC_FASTEST, sf_info.channels, &error); ctx->resampler = src_new(SRC_SINC_FASTEST, sf_info.channels, &error);
if (ctx->resampler == NULL) { if (ctx->resampler == NULL) {
@@ -894,7 +895,7 @@ music load_music_stream(const char* filename) {
sf_close(snd_file); sf_close(snd_file);
return music; return music;
} }
ctx->src_ratio = (double)AUDIO_DEVICE_SAMPLE_RATE / sf_info.samplerate; ctx->src_ratio = AUDIO.System.sampleRate / sf_info.samplerate;
} else { } else {
ctx->resampler = NULL; ctx->resampler = NULL;
ctx->src_ratio = 1.0; ctx->src_ratio = 1.0;
@@ -902,7 +903,7 @@ music load_music_stream(const char* filename) {
music.ctxData = ctx; music.ctxData = ctx;
int sample_size = 32; // We will work with floats internally int sample_size = 32; // We will work with floats internally
music.stream = load_audio_stream(AUDIO_DEVICE_SAMPLE_RATE, sample_size, sf_info.channels); music.stream = load_audio_stream(AUDIO.System.sampleRate, sample_size, sf_info.channels);
music.frameCount = (unsigned int)(sf_info.frames * ctx->src_ratio); music.frameCount = (unsigned int)(sf_info.frames * ctx->src_ratio);
music_loaded = true; music_loaded = true;
} }

3
libs/audio/audio.h
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@@ -3,6 +3,7 @@
#include <stdbool.h> #include <stdbool.h>
#include <stdint.h> #include <stdint.h>
#include <portaudio.h>
#ifdef __cplusplus #ifdef __cplusplus
extern "C" { extern "C" {
@@ -78,7 +79,7 @@ void list_host_apis(void);
* Initialize the audio device and system * Initialize the audio device and system
* Must be called before using any other audio functions * Must be called before using any other audio functions
*/ */
void init_audio_device(void); void init_audio_device(PaHostApiIndex host_api, double sample_rate);
/** /**
* Close the audio device and cleanup resources * Close the audio device and cleanup resources

2
scenes/settings.py
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@@ -34,6 +34,7 @@ class SettingsScreen:
global_data.config = self.config global_data.config = self.config
audio.close_audio_device() audio.close_audio_device()
audio.device_type = global_data.config["audio"]["device_type"] audio.device_type = global_data.config["audio"]["device_type"]
audio.target_sample_rate = global_data.config["audio"]["sample_rate"]
audio.init_audio_device() audio.init_audio_device()
return "ENTRY" return "ENTRY"
@@ -56,7 +57,6 @@ class SettingsScreen:
step_sizes = { step_sizes = {
'judge_offset': 1, 'judge_offset': 1,
'visual_offset': 1, 'visual_offset': 1,
'buffer_size': 1,
'sample_rate': 1000, 'sample_rate': 1000,
} }