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Init MinAudio to a sample rate of 0 to let the device pick the rate.

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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.
This commit is contained in:
Jeffery Myers 2021-03-20 18:47:28 -07:00
parent b64fe4dfed
commit 74df385b20
2 changed files with 8 additions and 8 deletions
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1
src/config.h
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@ -195,7 +195,6 @@
//------------------------------------------------------------------------------------
#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 DEFAULT_AUDIO_BUFFER_SIZE 4096 // Default audio buffer size for streaming
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Maximum number of audio pool channels

15
src/raudio.c
View File

@ -256,9 +256,8 @@ 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
#endif
int AUDIO_DEVICE_SAMPLE_RATE = 0;
#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels
#endif
@ -434,7 +433,7 @@ 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 = AUDIO_DEVICE_SAMPLE_RATE;
config.sampleRate = 0;
config.dataCallback = OnSendAudioDataToDevice;
config.pUserData = NULL;
@ -457,6 +456,8 @@ 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)
@ -1219,7 +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, 48000, fileName);
int result = jar_xm_create_context_from_file(&ctxXm, AUDIO_DEVICE_SAMPLE_RATE, fileName);
music.ctxType = MUSIC_MODULE_XM;
music.ctxData = ctxXm;
@ -1229,7 +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(48000, 16, 2);
music.stream = InitAudioStream(AUDIO_DEVICE_SAMPLE_RATE, 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 +1251,7 @@ 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_DEVICE_SAMPLE_RATE, 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;