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prune logging messages

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This commit is contained in:
Anthony Samms
2025-10-24 11:25:40 -04:00
parent 6b39007a3e
commit e3715fec77
1 changed files with 30 additions and 30 deletions
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60
libs/audio/audio.c
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@@ -268,11 +268,11 @@ void list_host_apis(void)
{
PaHostApiIndex hostApiCount = Pa_GetHostApiCount();
if (hostApiCount < 0) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to get host API count: %s", Pa_GetErrorText(hostApiCount));
TRACELOG(LOG_WARNING, "Failed to get host API count: %s", Pa_GetErrorText(hostApiCount));
return;
}
TRACELOG(LOG_INFO, "AUDIO: Available host APIs:");
TRACELOG(LOG_INFO, "Available host APIs:");
for (PaHostApiIndex i = 0; i < hostApiCount; i++) {
const PaHostApiInfo *info = Pa_GetHostApiInfo(i);
if (info) {
@@ -309,19 +309,19 @@ void init_audio_device(PaHostApiIndex host_api, double sample_rate, unsigned lon
{
PaError err = Pa_Initialize();
if (err != paNoError) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize PortAudio: %s", Pa_GetErrorText(err));
TRACELOG(LOG_WARNING, "Failed to initialize PortAudio: %s", Pa_GetErrorText(err));
return;
}
if (pthread_mutex_init(&AUDIO.System.lock, NULL) != 0) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing");
TRACELOG(LOG_WARNING, "Failed to create mutex for mixing");
Pa_Terminate();
return;
}
AUDIO.System.outputParameters.device = get_best_output_device_for_host_api(host_api);
if (AUDIO.System.outputParameters.device == paNoDevice) {
TRACELOG(LOG_WARNING, "AUDIO: No usable output device found");
TRACELOG(LOG_WARNING, "No usable output device found");
pthread_mutex_destroy(&AUDIO.System.lock);
Pa_Terminate();
return;
@@ -343,7 +343,7 @@ void init_audio_device(PaHostApiIndex host_api, double sample_rate, unsigned lon
&minSize, &maxSize, &preferredSize, &granularity);
if (asioErr == paNoError) {
TRACELOG(LOG_INFO, "AUDIO: ASIO buffer size constraints:");
TRACELOG(LOG_INFO, "ASIO buffer size constraints:");
TRACELOG(LOG_INFO, " > Minimum: %ld samples", minSize);
TRACELOG(LOG_INFO, " > Maximum: %ld samples", maxSize);
TRACELOG(LOG_INFO, " > Preferred: %ld samples", preferredSize);
@@ -357,16 +357,16 @@ void init_audio_device(PaHostApiIndex host_api, double sample_rate, unsigned lon
// Warn if requested buffer size is out of range
if (buffer_size > 0 && buffer_size < minSize) {
TRACELOG(LOG_WARNING, "AUDIO: Requested buffer size (%lu) is below ASIO minimum (%ld)", buffer_size, minSize);
TRACELOG(LOG_WARNING, "AUDIO: Driver will use %ld samples instead", minSize);
TRACELOG(LOG_WARNING, "Requested buffer size (%lu) is below ASIO minimum (%ld)", buffer_size, minSize);
TRACELOG(LOG_WARNING, "Driver will use %ld samples instead", minSize);
} else if (buffer_size > maxSize) {
TRACELOG(LOG_WARNING, "AUDIO: Requested buffer size (%lu) exceeds ASIO maximum (%ld)", buffer_size, maxSize);
TRACELOG(LOG_WARNING, "AUDIO: Driver will use %ld samples instead", maxSize);
TRACELOG(LOG_WARNING, "Requested buffer size (%lu) exceeds ASIO maximum (%ld)", buffer_size, maxSize);
TRACELOG(LOG_WARNING, "Driver will use %ld samples instead", maxSize);
} else if (buffer_size == 0) {
TRACELOG(LOG_INFO, "AUDIO: Buffer size not specified, driver will choose (likely %ld samples)", preferredSize);
TRACELOG(LOG_INFO, "Buffer size not specified, driver will choose (likely %ld samples)", preferredSize);
}
} else {
TRACELOG(LOG_WARNING, "AUDIO: Failed to query ASIO buffer sizes: %s", Pa_GetErrorText(asioErr));
TRACELOG(LOG_WARNING, "Failed to query ASIO buffer sizes: %s", Pa_GetErrorText(asioErr));
}
}
#endif
@@ -381,7 +381,7 @@ void init_audio_device(PaHostApiIndex host_api, double sample_rate, unsigned lon
NULL); // User data
if (err != paNoError) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to open audio stream: %s", Pa_GetErrorText(err));
TRACELOG(LOG_WARNING, "Failed to open audio stream: %s", Pa_GetErrorText(err));
pthread_mutex_destroy(&AUDIO.System.lock);
Pa_Terminate();
return;
@@ -389,7 +389,7 @@ void init_audio_device(PaHostApiIndex host_api, double sample_rate, unsigned lon
err = Pa_StartStream(AUDIO.System.stream);
if (err != paNoError) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to start audio stream: %s", Pa_GetErrorText(err));
TRACELOG(LOG_WARNING, "Failed to start audio stream: %s", Pa_GetErrorText(err));
Pa_CloseStream(AUDIO.System.stream);
pthread_mutex_destroy(&AUDIO.System.lock);
Pa_Terminate();
@@ -398,7 +398,7 @@ void init_audio_device(PaHostApiIndex host_api, double sample_rate, unsigned lon
AUDIO.System.isReady = true;
TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully");
TRACELOG(LOG_INFO, "Device initialized successfully");
TRACELOG(LOG_INFO, " > Backend: PortAudio | %s", hostApiInfo->name);
TRACELOG(LOG_INFO, " > Device: %s", deviceInfo->name);
TRACELOG(LOG_INFO, " > Format: %s", "Float32");
@@ -423,12 +423,12 @@ void close_audio_device(void)
if (AUDIO.System.isReady) {
PaError err = Pa_StopStream(AUDIO.System.stream);
if (err != paNoError) {
TRACELOG(LOG_WARNING, "AUDIO: Error stopping stream: %s", Pa_GetErrorText(err));
TRACELOG(LOG_WARNING, "Error stopping stream: %s", Pa_GetErrorText(err));
}
err = Pa_CloseStream(AUDIO.System.stream);
if (err != paNoError) {
TRACELOG(LOG_WARNING, "AUDIO: Error closing stream: %s", Pa_GetErrorText(err));
TRACELOG(LOG_WARNING, "Error closing stream: %s", Pa_GetErrorText(err));
}
pthread_mutex_destroy(&AUDIO.System.lock);
@@ -439,10 +439,10 @@ void close_audio_device(void)
AUDIO.System.pcmBuffer = NULL;
AUDIO.System.pcmBufferSize = 0;
TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
TRACELOG(LOG_INFO, "Device closed successfully");
}
else {
TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized");
TRACELOG(LOG_WARNING, "Device could not be closed, not currently initialized");
}
}
@@ -471,13 +471,13 @@ struct audio_buffer *load_audio_buffer(uint32_t channels, uint32_t size_in_frame
struct audio_buffer *buffer = (struct audio_buffer*)calloc(1, sizeof(struct audio_buffer));
if (buffer == NULL) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for buffer");
TRACELOG(LOG_WARNING, "Failed to allocate memory for buffer");
return NULL;
}
buffer->data = calloc(size_in_frames*channels*sizeof(float), 1);
if (buffer->data == NULL) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for buffer data");
TRACELOG(LOG_WARNING, "Failed to allocate memory for buffer data");
FREE(buffer);
return NULL;
}
@@ -631,7 +631,7 @@ wave load_wave(const char* filename) {
snd_file = sf_open(filename, SFM_READ, &sf_info);
if (snd_file == NULL) {
TRACELOG(LOG_ERROR, "AUDIO: Failed to open file '%s'\n", filename);
TRACELOG(LOG_ERROR, "Failed to open file '%s'\n", filename);
return wave;
}
wave.frameCount = (unsigned int)sf_info.frames;
@@ -642,7 +642,7 @@ wave load_wave(const char* filename) {
size_t total_samples = sf_info.frames * sf_info.channels;
wave.data = malloc(total_samples * sizeof(float));
if (wave.data == NULL) {
TRACELOG(LOG_ERROR, "AUDIO: Failed to allocate memory for wave data");
TRACELOG(LOG_ERROR, "Failed to allocate memory for wave data");
sf_close(snd_file);
return wave;
}
@@ -674,7 +674,7 @@ sound load_sound_from_wave(wave wave) {
bool is_resampled = false;
if (wave.sampleRate != AUDIO.System.sampleRate) {
TRACELOG(LOG_INFO, "AUDIO: Resampling wave from %d Hz to %f Hz", wave.sampleRate, AUDIO.System.sampleRate);
TRACELOG(LOG_INFO, "Resampling wave from %d Hz to %f Hz", wave.sampleRate, AUDIO.System.sampleRate);
SRC_DATA src_data;
src_data.data_in = wave.data;
@@ -684,14 +684,14 @@ sound load_sound_from_wave(wave wave) {
resampled_wave.data = calloc(src_data.output_frames * wave.channels, sizeof(float));
if (resampled_wave.data == NULL) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for resampling");
TRACELOG(LOG_WARNING, "Failed to allocate memory for resampling");
return sound;
}
src_data.data_out = resampled_wave.data;
int error = src_simple(&src_data, SRC_SINC_BEST_QUALITY, wave.channels);
if (error) {
TRACELOG(LOG_WARNING, "AUDIO: Resampling failed: %s", src_strerror(error));
TRACELOG(LOG_WARNING, "Resampling failed: %s", src_strerror(error));
FREE(resampled_wave.data);
return sound;
}
@@ -887,18 +887,18 @@ music load_music_stream(const char* filename) {
if (snd_file != NULL) {
music_ctx *ctx = calloc(1, sizeof(music_ctx));
if (ctx == NULL) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for music context");
TRACELOG(LOG_WARNING, "Failed to allocate memory for music context");
sf_close(snd_file);
return music;
}
ctx->snd_file = snd_file;
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);
TRACELOG(LOG_INFO, "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) {
TRACELOG(LOG_WARNING, "AUDIO: Failed to create resampler: %s", src_strerror(error));
TRACELOG(LOG_WARNING, "Failed to create resampler: %s", src_strerror(error));
free(ctx);
sf_close(snd_file);
return music;
@@ -1029,7 +1029,7 @@ void update_music_stream(music music) {
int error = src_process(ctx->resampler, &src_data);
if (error) {
TRACELOG(LOG_WARNING, "AUDIO: Resampling failed: %s", src_strerror(error));
TRACELOG(LOG_WARNING, "Resampling failed: %s", src_strerror(error));
}
frames_written = src_data.output_frames_gen;
} else {