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[raudio] ADDED: SetAudioStreamBufferSizeDefault()

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There could be some cases where we need to define the internal audio buffer size, now it's possible.

RENAMED: InitAudioBuffer() to LoadAudioBuffer()
RENAMED: CloseAudioBuffer() to UnloadAudioBuffer()

Renamed some defines and reviewed some comments.
This commit is contained in:
Ray 2020-02-14 17:13:37 +01:00
parent 7459a5c6ef
commit 91b2dc2aa9
2 changed files with 64 additions and 58 deletions
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121
src/raudio.c
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@ -79,7 +79,7 @@
#endif
#if defined(_WIN32)
// @raysan5: To avoid conflicting windows.h symbols with raylib, so flags are defined
// To avoid conflicting windows.h symbols with raylib, some flags are defined
// WARNING: Those flags avoid inclusion of some Win32 headers that could be required
// by user at some point and won't be included...
//-------------------------------------------------------------------------------------
@ -149,7 +149,7 @@ typedef struct tagBITMAPINFOHEADER {
#include <mmreg.h>
#include <mmsystem.h>
// @raysan5: Some required types defined for MSVC/TinyC compiler
// Some required types defined for MSVC/TinyC compiler
#if defined(_MSC_VER) || defined(__TINYC__)
#include "propidl.h"
#endif
@ -157,14 +157,14 @@ typedef struct tagBITMAPINFOHEADER {
#define MA_NO_JACK
#define MINIAUDIO_IMPLEMENTATION
#include "external/miniaudio.h" // miniaudio library
#undef PlaySound // Win32 API: windows.h > mmsystem.h defines PlaySound macro
#include "external/miniaudio.h" // miniaudio library
#undef PlaySound // Win32 API: windows.h > mmsystem.h defines PlaySound macro
#include <stdlib.h> // Required for: malloc(), free()
#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread()
#include <stdlib.h> // Required for: malloc(), free()
#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread()
#if defined(RAUDIO_STANDALONE)
#include <string.h> // Required for: strcmp() [Used in IsFileExtension()]
#include <string.h> // Required for: strcmp() [Used in IsFileExtension()]
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
@ -200,17 +200,9 @@ typedef struct tagBITMAPINFOHEADER {
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
// NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
// 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
#if !defined(AUDIO_BUFFER_SIZE)
#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb)
#endif
#define DEVICE_FORMAT ma_format_f32
#define DEVICE_CHANNELS 2
#define DEVICE_SAMPLE_RATE 44100
#define AUDIO_DEVICE_FORMAT ma_format_f32
#define AUDIO_DEVICE_CHANNELS 2
#define AUDIO_DEVICE_SAMPLE_RATE 44100
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16
@ -286,6 +278,7 @@ typedef struct AudioData {
struct {
AudioBuffer *first; // Pointer to first AudioBuffer in the list
AudioBuffer *last; // Pointer to last AudioBuffer in the list
int defaultSize; // Default audio buffer size for audio streams
} Buffer;
struct {
AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // Multichannel AudioBuffer pointers pool
@ -297,7 +290,14 @@ typedef struct AudioData {
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
static AudioData AUDIO = { 0 }; // Global CORE context
static AudioData AUDIO = { // Global AUDIO context
// NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
// 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 = 4096
};
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
@ -332,8 +332,9 @@ void TRACELOG(int msgType, const char *text, ...); // Show trace log messag
// AudioBuffer management functions declaration
// NOTE: Those functions are not exposed by raylib... for the moment
//----------------------------------------------------------------------------------
AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage);
void CloseAudioBuffer(AudioBuffer *buffer);
AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage);
void UnloadAudioBuffer(AudioBuffer *buffer);
bool IsAudioBufferPlaying(AudioBuffer *buffer);
void PlayAudioBuffer(AudioBuffer *buffer);
void StopAudioBuffer(AudioBuffer *buffer);
@ -367,12 +368,12 @@ void InitAudioDevice(void)
// NOTE: Using the default device. Format is floating point because it simplifies mixing.
ma_device_config config = ma_device_config_init(ma_device_type_playback);
config.playback.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device.
config.playback.format = DEVICE_FORMAT;
config.playback.channels = DEVICE_CHANNELS;
config.playback.format = AUDIO_DEVICE_FORMAT;
config.playback.channels = AUDIO_DEVICE_CHANNELS;
config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device.
config.capture.format = ma_format_s16;
config.capture.channels = 1;
config.sampleRate = DEVICE_SAMPLE_RATE;
config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
config.dataCallback = OnSendAudioDataToDevice;
config.pUserData = NULL;
@ -410,7 +411,7 @@ void InitAudioDevice(void)
TRACELOG(LOG_INFO, "Audio format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
TRACELOG(LOG_INFO, "Audio channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
TRACELOG(LOG_INFO, "Audio sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
TRACELOG(LOG_INFO, "Audio buffer size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
TRACELOG(LOG_INFO, "Audio periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
InitAudioBufferPool();
TRACELOG(LOG_INFO, "Audio multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);
@ -451,27 +452,27 @@ void SetMasterVolume(float volume)
//----------------------------------------------------------------------------------
// Initialize a new audio buffer (filled with silence)
AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
{
AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
if (audioBuffer == NULL)
{
TRACELOG(LOG_ERROR, "InitAudioBuffer() : Failed to allocate memory for audio buffer");
TRACELOG(LOG_ERROR, "LoadAudioBuffer() : Failed to allocate memory for audio buffer");
return NULL;
}
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, DEVICE_FORMAT, channels, DEVICE_CHANNELS, sampleRate, DEVICE_SAMPLE_RATE);
ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO_DEVICE_SAMPLE_RATE);
converterConfig.resampling.allowDynamicSampleRate = true; // Required for pitch shifting
ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter);
if (result != MA_SUCCESS)
{
TRACELOG(LOG_ERROR, "InitAudioBuffer() : Failed to create data conversion pipeline");
TRACELOG(LOG_ERROR, "LoadAudioBuffer() : Failed to create data conversion pipeline");
RL_FREE(audioBuffer);
return NULL;
}
@ -498,7 +499,7 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
}
// Delete an audio buffer
void CloseAudioBuffer(AudioBuffer *buffer)
void UnloadAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
@ -679,19 +680,19 @@ 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, DEVICE_FORMAT, DEVICE_CHANNELS, 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_DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion");
AudioBuffer *audioBuffer = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer");
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, DEVICE_FORMAT, DEVICE_CHANNELS, 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_DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
sound.sampleCount = frameCount*DEVICE_CHANNELS;
sound.stream.sampleRate = DEVICE_SAMPLE_RATE;
sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS;
sound.stream.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
sound.stream.sampleSize = 32;
sound.stream.channels = DEVICE_CHANNELS;
sound.stream.channels = AUDIO_DEVICE_CHANNELS;
sound.stream.buffer = audioBuffer;
}
@ -709,7 +710,7 @@ void UnloadWave(Wave wave)
// Unload sound
void UnloadSound(Sound sound)
{
CloseAudioBuffer(sound.stream.buffer);
UnloadAudioBuffer(sound.stream.buffer);
TRACELOG(LOG_INFO, "Unloaded sound data from RAM");
}
@ -717,16 +718,13 @@ void UnloadSound(Sound sound)
// Update sound buffer with new data
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
AudioBuffer *audioBuffer = sound.stream.buffer;
if (audioBuffer != NULL)
if (sound.stream.buffer != NULL)
{
StopAudioBuffer(audioBuffer);
StopAudioBuffer(sound.stream.buffer);
// TODO: May want to lock/unlock this since this data buffer is read at mixing time
memcpy(audioBuffer->data, data, samplesCount*ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn));
memcpy(sound.stream.buffer->data, data, samplesCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.channelsIn));
}
else TRACELOG(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
}
// Export wave data to file
@ -829,13 +827,12 @@ void PlaySoundMulti(Sound sound)
// If no none playing pool members can be index choose the oldest
if (index == -1)
{
TRACELOG(LOG_WARNING,"pool age %i ended a sound early no room in buffer pool", AUDIO.MultiChannel.poolCounter);
TRACELOG(LOG_WARNING, "Pool age %i ended a sound early no room in buffer pool", AUDIO.MultiChannel.poolCounter);
if (oldIndex == -1)
{
// Shouldn't be able to get here... but just in case something odd happens!
TRACELOG(LOG_ERROR,"sound buffer pool couldn't determine oldest buffer not playing sound");
TRACELOG(LOG_ERROR, "Sound buffer pool could not determine oldest buffer not playing sound");
return;
}
@ -846,7 +843,7 @@ void PlaySoundMulti(Sound sound)
}
// Experimentally mutex lock doesn't seem to be needed this makes sense
// as AUDIO.MultiChannel.pool[index] isn't playing and the only stuff we're copying
// as pool[index] isn't playing and the only stuff we're copying
// shouldn't be changing...
AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter;
@ -930,7 +927,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
{
TRACELOG(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion.");
TRACELOG(LOG_WARNING, "WaveFormat() : Failed to get frame count for format conversion.");
return;
}
@ -939,7 +936,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
{
TRACELOG(LOG_ERROR, "WaveFormat() : Format conversion failed.");
TRACELOG(LOG_WARNING, "WaveFormat() : Format conversion failed.");
return;
}
@ -1397,19 +1394,19 @@ 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_SIZE;
unsigned int subBufferSize = AUDIO.Buffer.defaultSize; // Default buffer size (audio stream)
if (subBufferSize < periodSize) subBufferSize = periodSize;
// Create a double audio buffer of defined AUDIO_BUFFER_SIZE
stream.buffer = InitAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM);
// Create a double audio buffer of defined size
stream.buffer = LoadAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM);
if (stream.buffer != NULL)
{
stream.buffer->looping = true; // Always loop for streaming buffers
TRACELOG(LOG_INFO, "Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
TRACELOG(LOG_INFO, "Audio stream initialized successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
}
else TRACELOG(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer");
else TRACELOG(LOG_ERROR, "Audio stream could not be created: Failed to load audio buffer");
return stream;
}
@ -1417,7 +1414,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
// Close audio stream and free memory
void CloseAudioStream(AudioStream stream)
{
CloseAudioBuffer(stream.buffer);
UnloadAudioBuffer(stream.buffer);
TRACELOG(LOG_INFO, "Unloaded audio stream data");
}
@ -1516,16 +1513,24 @@ void StopAudioStream(AudioStream stream)
StopAudioBuffer(stream.buffer);
}
// Set volume for audio stream (1.0 is max level)
void SetAudioStreamVolume(AudioStream stream, float volume)
{
SetAudioBufferVolume(stream.buffer, volume);
}
// Set pitch for audio stream (1.0 is base level)
void SetAudioStreamPitch(AudioStream stream, float pitch)
{
SetAudioBufferPitch(stream.buffer, pitch);
}
// Default size for new audio streams
void SetAudioStreamBufferSizeDefault(int size)
{
AUDIO.Buffer.defaultSize = size;
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
@ -1713,9 +1718,9 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
float tempBuffer[1024]; // 512 frames for stereo
ma_uint32 framesToReadRightNow = framesToRead;
if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS)
if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS)
{
framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS;
}
ma_uint32 framesJustRead = ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow);
@ -1786,7 +1791,7 @@ static void InitAudioBufferPool(void)
// Dummy buffers
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
AUDIO.MultiChannel.pool[i] = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
}
}

1
src/raylib.h
View File

@ -1428,6 +1428,7 @@ RLAPI bool IsAudioStreamPlaying(AudioStream stream); // Check i
RLAPI void StopAudioStream(AudioStream stream); // Stop audio stream
RLAPI void SetAudioStreamVolume(AudioStream stream, float volume); // Set volume for audio stream (1.0 is max level)
RLAPI void SetAudioStreamPitch(AudioStream stream, float pitch); // Set pitch for audio stream (1.0 is base level)
RLAPI void SetAudioStreamBufferSizeDefault(int size); // Default size for new audio streams
//------------------------------------------------------------------------------------
// Network (Module: network)