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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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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 AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo
#define AUDIO_DEVICE_SAMPLE_RATE 44100 // Device output sample rate
// Forward declarations of structures
struct audio_buffer;
@@ -101,6 +100,7 @@ typedef struct AudioData {
PaStreamParameters outputParameters; // Output stream parameters
pthread_mutex_t lock; // Mutex lock for thread synchronization
bool isReady; // Check if audio device is ready
double sampleRate;
size_t pcmBufferSize; // Pre-allocated buffer size
void *pcmBuffer; // Pre-allocated buffer to read audio data from file/memory
float masterVolume; // Master volume control
@@ -114,7 +114,7 @@ typedef struct AudioData {
// Function declarations
// Device management
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);
bool is_audio_device_ready(void);
void set_master_volume(float volume);
@@ -322,7 +322,7 @@ PaDeviceIndex get_best_output_device_for_host_api(PaHostApiIndex hostApi)
}
// Device management implementations
void init_audio_device(PaHostApiIndex host_api)
void init_audio_device(PaHostApiIndex host_api, double sample_rate)
{
// Initialize PortAudio
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.suggestedLatency = Pa_GetDeviceInfo(AUDIO.System.outputParameters.device)->defaultLowOutputLatency;
AUDIO.System.outputParameters.hostApiSpecificStreamInfo = NULL;
AUDIO.System.sampleRate = sample_rate;
// Open the audio stream
err = Pa_OpenStream(&AUDIO.System.stream,
NULL, // No input
&AUDIO.System.outputParameters, // Output parameters
AUDIO_DEVICE_SAMPLE_RATE, // Sample rate
sample_rate, // Sample rate
paFramesPerBufferUnspecified, // Frames per buffer (let PortAudio decide)
paClipOff, // No clipping
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, " > Format: %s", "Float32");
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);
}
@@ -657,13 +658,13 @@ sound load_sound_from_wave(wave wave) {
struct wave resampled_wave = { 0 };
bool is_resampled = false;
if (wave.sampleRate != AUDIO_DEVICE_SAMPLE_RATE) {
TRACELOG(LOG_INFO, "AUDIO: Resampling wave from %d Hz to %d Hz", wave.sampleRate, AUDIO_DEVICE_SAMPLE_RATE);
if (wave.sampleRate != AUDIO.System.sampleRate) {
TRACELOG(LOG_INFO, "AUDIO: Resampling wave from %d Hz to %f Hz", wave.sampleRate, AUDIO.System.sampleRate);
SRC_DATA src_data;
src_data.data_in = wave.data;
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);
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.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
resampled_wave.sampleRate = AUDIO.System.sampleRate;
resampled_wave.channels = wave.channels;
resampled_wave.sampleSize = wave.sampleSize;
is_resampled = true;
@@ -790,7 +791,7 @@ audio_stream load_audio_stream(unsigned int sample_rate, unsigned int sample_siz
stream.channels = channels;
// 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;
}
@@ -884,8 +885,8 @@ music load_music_stream(const char* filename) {
}
ctx->snd_file = snd_file;
if (sf_info.samplerate != AUDIO_DEVICE_SAMPLE_RATE) {
TRACELOG(LOG_INFO, "AUDIO: Resampling music from %d Hz to %d Hz", sf_info.samplerate, AUDIO_DEVICE_SAMPLE_RATE);
if (sf_info.samplerate != AUDIO.System.sampleRate) {
TRACELOG(LOG_INFO, "AUDIO: Resampling music from %d Hz to %f Hz", sf_info.samplerate, AUDIO.System.sampleRate);
int error;
ctx->resampler = src_new(SRC_SINC_FASTEST, sf_info.channels, &error);
if (ctx->resampler == NULL) {
@@ -894,7 +895,7 @@ music load_music_stream(const char* filename) {
sf_close(snd_file);
return music;
}
ctx->src_ratio = (double)AUDIO_DEVICE_SAMPLE_RATE / sf_info.samplerate;
ctx->src_ratio = AUDIO.System.sampleRate / sf_info.samplerate;
} else {
ctx->resampler = NULL;
ctx->src_ratio = 1.0;
@@ -902,7 +903,7 @@ music load_music_stream(const char* filename) {
music.ctxData = ctx;
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_loaded = true;
}