[AUDIO] Use device native sample rates (#1660)

* Init MinAudio to a sample rate of 0 to let the device pick the rate.
Read the rate from the device after it starts up.
Convert AUDIO_DEVICE_SAMPLE_RATE from a #def into an int, that is set from the device's rate
Set all sample systems to use the AUDIO_DEVICE_SAMPLE_RATE as the target rate to minimize resampling.

* 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.

* Don't use fixed sample rates for tracker files.
Remove config default audio buffer size and replace with a function that computes one for a reasonable frame rate at the output sample rate.

src/config.h

@@ -195,9 +195,8 @@
 //------------------------------------------------------------------------------------
 #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       44100    // Device output sample rate
+#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
 
 //------------------------------------------------------------------------------------

src/raudio.c

@@ -256,8 +256,9 @@ typedef struct tagBITMAPINFOHEADER {
 #ifndef AUDIO_DEVICE_CHANNELS
     #define AUDIO_DEVICE_CHANNELS              2    // Device output channels: stereo
 #endif
+
 #ifndef AUDIO_DEVICE_SAMPLE_RATE
-    #define AUDIO_DEVICE_SAMPLE_RATE       44100    // Device output 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
@@ -358,7 +359,7 @@ static AudioData AUDIO = {          // Global AUDIO context
     // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a
     // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough
     // In case of music-stalls, just increase this number
-    .Buffer.defaultSize = DEFAULT_AUDIO_BUFFER_SIZE
+    .Buffer.defaultSize = 0
 };
 
 //----------------------------------------------------------------------------------
@@ -405,6 +406,7 @@ void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
 void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
 void TrackAudioBuffer(AudioBuffer *buffer);
 void UntrackAudioBuffer(AudioBuffer *buffer);
+int GetAudioStreamBufferSizeDefault();
 
 //----------------------------------------------------------------------------------
 // Module Functions Definition - Audio Device initialization and Closing
@@ -472,7 +474,7 @@ 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");
@@ -545,7 +547,7 @@ 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);
@@ -781,21 +783,21 @@ 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;
@@ -1219,7 +1221,7 @@ 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, 48000, fileName);
+        int result = jar_xm_create_context_from_file(&ctxXm, AUDIO.System.device.sampleRate, fileName);
 
         music.ctxType = MUSIC_MODULE_XM;
         music.ctxData = ctxXm;
@@ -1229,7 +1231,7 @@ 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(48000, 16, 2);
+            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
@@ -1250,7 +1252,7 @@ Music LoadMusicStream(const char *fileName)
         if (result > 0)
         {
             // NOTE: Only stereo is supported for MOD
-            music.stream = InitAudioStream(48000, 16, 2);
+            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;
@@ -1390,14 +1392,14 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int d
     else if (TextIsEqual(fileExtLower, ".xm"))
     {
         jar_xm_context_t *ctxXm = NULL;
-        int result = jar_xm_create_context_safe(&ctxXm, (const char*)data, dataSize, 48000);
+        int result = jar_xm_create_context_safe(&ctxXm, (const char*)data, dataSize, AUDIO.System.device.sampleRate);
         if (result == 0)    // XM AUDIO.System.context created successfully
         {
             music.ctxType = MUSIC_MODULE_XM;
             jar_xm_set_max_loop_count(ctxXm, 0);    // Set infinite number of loops
 
             // NOTE: Only stereo is supported for XM
-            music.stream = InitAudioStream(48000, 16, 2);
+            music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, 2);
             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
@@ -1435,7 +1437,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int d
             music.ctxType = MUSIC_MODULE_MOD;
 
             // NOTE: Only stereo is supported for MOD
-            music.stream = InitAudioStream(48000, 16, 2);
+			music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, 2);
             music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2;    // 2 channels
             music.looping = true;   // Looping enabled by default
             musicLoaded = true;
@@ -1734,7 +1736,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
 
     // The size of a streaming buffer must be at least double the size of a period
     unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames;
-    unsigned int subBufferSize = AUDIO.Buffer.defaultSize;     // Default buffer size (audio stream)
+    unsigned int subBufferSize = GetAudioStreamBufferSizeDefault();
 
     if (subBufferSize < periodSize) subBufferSize = periodSize;
 
@@ -1869,6 +1871,14 @@ void SetAudioStreamBufferSizeDefault(int size)
     AUDIO.Buffer.defaultSize = size;
 }
 
+int GetAudioStreamBufferSizeDefault()
+{
+    // if the buffer is not set, compute one that would give us a buffer good enough for a decent frame rate
+    if (AUDIO.Buffer.defaultSize == 0)
+        AUDIO.Buffer.defaultSize = AUDIO.System.device.sampleRate / 30;
+
+	return AUDIO.Buffer.defaultSize;
+}
 //----------------------------------------------------------------------------------
 // Module specific Functions Definition
 //----------------------------------------------------------------------------------