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

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* 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.
This commit is contained in:
Jeffery Myers 2021-03-21 14:07:55 -07:00 committed by GitHub
parent c37f61d31f
commit b573ff3e7a
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GPG Key ID: 4AEE18F83AFDEB23
2 changed files with 26 additions and 17 deletions
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3
src/config.h
View File

@ -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
//------------------------------------------------------------------------------------

40
src/raudio.c
View File

@ -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
//----------------------------------------------------------------------------------