From 2812702e6f77fbd29831b4953ea798d7c1c5c087 Mon Sep 17 00:00:00 2001 From: Jeffery Myers Date: Sun, 21 Mar 2021 13:28:46 -0700 Subject: [PATCH] use device sample rate instead of separate var let config define a device sample rate if it wants to, but let the default be 0 for native rate. --- src/config.h | 1 + src/raudio.c | 26 +++++++++++++------------- 2 files changed, 14 insertions(+), 13 deletions(-) diff --git a/src/config.h b/src/config.h index 0f8884779..882664b32 100644 --- a/src/config.h +++ b/src/config.h @@ -195,6 +195,7 @@ //------------------------------------------------------------------------------------ #define AUDIO_DEVICE_FORMAT ma_format_f32 // Device output format (miniaudio: float-32bit) #define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo +#define AUDIO_DEVICE_SAMPLE_RATE 0 // Device sample rate (device default) #define DEFAULT_AUDIO_BUFFER_SIZE 4096 // Default audio buffer size for streaming #define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Maximum number of audio pool channels diff --git a/src/raudio.c b/src/raudio.c index d1f197061..7bde4d530 100644 --- a/src/raudio.c +++ b/src/raudio.c @@ -256,8 +256,10 @@ typedef struct tagBITMAPINFOHEADER { #ifndef AUDIO_DEVICE_CHANNELS #define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo #endif -int AUDIO_DEVICE_SAMPLE_RATE = 0; +#ifndef AUDIO_DEVICE_SAMPLE_RATE + #define AUDIO_DEVICE_SAMPLE_RATE 0 // Device output channels: stereo +#endif #ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS #define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels #endif @@ -433,7 +435,7 @@ void InitAudioDevice(void) config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device. config.capture.format = ma_format_s16; config.capture.channels = 1; - config.sampleRate = 0; + config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; config.dataCallback = OnSendAudioDataToDevice; config.pUserData = NULL; @@ -456,8 +458,6 @@ void InitAudioDevice(void) return; } - AUDIO_DEVICE_SAMPLE_RATE = AUDIO.System.device.sampleRate; - // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may // want to look at something a bit smarter later on to keep everything real-time, if that's necessary. if (ma_mutex_init(&AUDIO.System.lock) != MA_SUCCESS) @@ -473,7 +473,7 @@ void InitAudioDevice(void) { // WARNING: An empty audioBuffer is created (data = 0) // AudioBuffer data just points to loaded sound data - AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC); + AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC); } TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully"); @@ -546,7 +546,7 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam if (sizeInFrames > 0) audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1); // Audio data runs through a format converter - ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO_DEVICE_SAMPLE_RATE); + ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO.System.device.sampleRate); converterConfig.resampling.allowDynamicSampleRate = true; // Required for pitch shifting ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter); @@ -782,21 +782,21 @@ Sound LoadSoundFromWave(Wave wave) ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32)); ma_uint32 frameCountIn = wave.sampleCount/wave.channels; - ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate); + ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate); if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion"); - AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC); + AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, frameCount, AUDIO_BUFFER_USAGE_STATIC); if (audioBuffer == NULL) { TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer"); return sound; // early return to avoid dereferencing the audioBuffer null pointer } - frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate); + frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate); if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion"); sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS; - sound.stream.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; + sound.stream.sampleRate = AUDIO.System.device.sampleRate; sound.stream.sampleSize = 32; sound.stream.channels = AUDIO_DEVICE_CHANNELS; sound.stream.buffer = audioBuffer; @@ -1220,7 +1220,7 @@ Music LoadMusicStream(const char *fileName) else if (IsFileExtension(fileName, ".xm")) { jar_xm_context_t *ctxXm = NULL; - int result = jar_xm_create_context_from_file(&ctxXm, AUDIO_DEVICE_SAMPLE_RATE, fileName); + int result = jar_xm_create_context_from_file(&ctxXm, AUDIO.System.device.sampleRate, fileName); music.ctxType = MUSIC_MODULE_XM; music.ctxData = ctxXm; @@ -1230,7 +1230,7 @@ Music LoadMusicStream(const char *fileName) jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops // NOTE: Only stereo is supported for XM - music.stream = InitAudioStream(AUDIO_DEVICE_SAMPLE_RATE, 16, AUDIO_DEVICE_CHANNELS); + music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS); music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels music.looping = true; // Looping enabled by default jar_xm_reset(ctxXm); // make sure we start at the beginning of the song @@ -1251,7 +1251,7 @@ Music LoadMusicStream(const char *fileName) if (result > 0) { // NOTE: Only stereo is supported for MOD - music.stream = InitAudioStream(AUDIO_DEVICE_SAMPLE_RATE, 16, AUDIO_DEVICE_CHANNELS); + music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS); music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels music.looping = true; // Looping enabled by default musicLoaded = true;