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REVIEWED: Move some functions, made them static

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Ray 2024-04-20 20:30:35 +02:00
parent d80a622972
commit c1943f0f7c
1 changed files with 108 additions and 104 deletions
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212
src/raudio.c
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@ -410,9 +410,18 @@ static AudioData AUDIO = { // Global AUDIO context
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage);
// Reads audio data from an AudioBuffer object in internal/device formats
static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount);
static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount);
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer);
static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer);
static void StopAudioBufferInLockedState(AudioBuffer *buffer);
static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount);
#if defined(RAUDIO_STANDALONE)
static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
static const char *GetFileExtension(const char *fileName); // Get pointer to extension for a filename string (includes the dot: .png)
@ -431,10 +440,8 @@ static bool SaveFileText(const char *fileName, char *text); // Save text
AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage);
void UnloadAudioBuffer(AudioBuffer *buffer);
bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer);
bool IsAudioBufferPlaying(AudioBuffer *buffer);
void PlayAudioBuffer(AudioBuffer *buffer);
void StopAudioBufferInLockedState(AudioBuffer *buffer);
void StopAudioBuffer(AudioBuffer *buffer);
void PauseAudioBuffer(AudioBuffer *buffer);
void ResumeAudioBuffer(AudioBuffer *buffer);
@ -444,10 +451,6 @@ void SetAudioBufferPan(AudioBuffer *buffer, float pan);
void TrackAudioBuffer(AudioBuffer *buffer);
void UntrackAudioBuffer(AudioBuffer *buffer);
//----------------------------------------------------------------------------------
// AudioStream management functions declaration
//----------------------------------------------------------------------------------
void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount);
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
@ -467,12 +470,12 @@ void InitAudioDevice(void)
}
// Init audio device
// NOTE: Using the default device. Format is floating point because it simplifies mixing.
// 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.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device
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.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;
@ -488,7 +491,7 @@ void InitAudioDevice(void)
}
// Mixing happens on a separate 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.
// 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)
{
TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing");
@ -498,7 +501,7 @@ void InitAudioDevice(void)
}
// Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running
// while there's at least one sound being played.
// while there's at least one sound being played
result = ma_device_start(&AUDIO.System.device);
if (result != MA_SUCCESS)
{
@ -626,17 +629,7 @@ void UnloadAudioBuffer(AudioBuffer *buffer)
}
}
// Check if an audio buffer is playing, assuming the audio system mutex has been locked.
bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer)
{
bool result = false;
if (buffer != NULL) result = (buffer->playing && !buffer->paused);
return result;
}
// Check if an audio buffer is playing from a program state without lock.
// Check if an audio buffer is playing from a program state without lock
bool IsAudioBufferPlaying(AudioBuffer *buffer)
{
bool result = false;
@ -647,8 +640,8 @@ bool IsAudioBufferPlaying(AudioBuffer *buffer)
}
// Play an audio buffer
// NOTE: Buffer is restarted to the start.
// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained.
// NOTE: Buffer is restarted to the start
// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained
void PlayAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
@ -661,24 +654,7 @@ void PlayAudioBuffer(AudioBuffer *buffer)
}
}
// Stop an audio buffer, assuming the audio system mutex has been locked.
void StopAudioBufferInLockedState(AudioBuffer *buffer)
{
if (buffer != NULL)
{
if (IsAudioBufferPlayingInLockedState(buffer))
{
buffer->playing = false;
buffer->paused = false;
buffer->frameCursorPos = 0;
buffer->framesProcessed = 0;
buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true;
}
}
}
// Stop an audio buffer from a program state without lock.
// Stop an audio buffer from a program state without lock
void StopAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&AUDIO.System.lock);
@ -725,7 +701,7 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
if ((buffer != NULL) && (pitch > 0.0f))
{
ma_mutex_lock(&AUDIO.System.lock);
// Pitching is just an adjustment of the sample rate.
// Pitching is just an adjustment of the sample rate
// Note that this changes the duration of the sound:
// - higher pitches will make the sound faster
// - lower pitches make it slower
@ -951,15 +927,15 @@ Sound LoadSoundFromWave(Wave wave)
if (wave.data != NULL)
{
// When using miniaudio we need to do our own mixing.
// When using miniaudio we need to do our own mixing
// To simplify this we need convert the format of each sound to be consistent with
// the format used to open the playback AUDIO.System.device. We can do this two ways:
//
// 1) Convert the whole sound in one go at load time (here).
// 2) Convert the audio data in chunks at mixing time.
// 1) Convert the whole sound in one go at load time (here)
// 2) Convert the audio data in chunks at mixing time
//
// First option has been selected, format conversion is done on the loading stage.
// The downside is that it uses more memory if the original sound is u8 or s16.
// First option has been selected, format conversion is done on the loading stage
// The downside is that it uses more memory if the original sound is u8 or s16
ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_uint32 frameCountIn = wave.frameCount;
@ -2167,55 +2143,6 @@ void UpdateAudioStream(AudioStream stream, const void *data, int frameCount)
ma_mutex_unlock(&AUDIO.System.lock);
}
void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount)
{
if (stream.buffer != NULL)
{
if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1])
{
ma_uint32 subBufferToUpdate = 0;
if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1])
{
// Both buffers are available for updating.
// Update the first one and make sure the cursor is moved back to the front.
subBufferToUpdate = 0;
stream.buffer->frameCursorPos = 0;
}
else
{
// Just update whichever sub-buffer is processed.
subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1;
}
ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2;
unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
// Total frames processed in buffer is always the complete size, filled with 0 if required
stream.buffer->framesProcessed += subBufferSizeInFrames;
// Does this API expect a whole buffer to be updated in one go?
// Assuming so, but if not will need to change this logic.
if (subBufferSizeInFrames >= (ma_uint32)frameCount)
{
ma_uint32 framesToWrite = (ma_uint32)frameCount;
ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
memcpy(subBuffer, data, bytesToWrite);
// Any leftover frames should be filled with zeros.
ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false;
}
else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer");
}
else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating");
}
}
// Check if any audio stream buffers requires refill
bool IsAudioStreamProcessed(AudioStream stream)
{
@ -2246,7 +2173,7 @@ void ResumeAudioStream(AudioStream stream)
ResumeAudioBuffer(stream.buffer);
}
// Check if audio stream is playing.
// Check if audio stream is playing
bool IsAudioStreamPlaying(AudioStream stream)
{
return IsAudioBufferPlaying(stream.buffer);
@ -2293,9 +2220,9 @@ void SetAudioStreamCallback(AudioStream stream, AudioCallback callback)
}
}
// Add processor to audio stream. Contrary to buffers, the order of processors is important.
// Add processor to audio stream. Contrary to buffers, the order of processors is important
// The new processor must be added at the end. As there aren't supposed to be a lot of processors attached to
// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element.
// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element
void AttachAudioStreamProcessor(AudioStream stream, AudioCallback process)
{
ma_mutex_lock(&AUDIO.System.lock);
@ -2410,7 +2337,7 @@ static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage)
TRACELOG(LOG_WARNING, "miniaudio: %s", pMessage); // All log messages from miniaudio are errors
}
// Reads audio data from an AudioBuffer object in internal format.
// Reads audio data from an AudioBuffer object in internal format
static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount)
{
// Using audio buffer callback
@ -2498,20 +2425,20 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer,
// For static buffers we can fill the remaining frames with silence for safety, but we don't want
// to report those frames as "read". The reason for this is that the caller uses the return value
// to know whether a non-looping sound has finished playback.
// to know whether a non-looping sound has finished playback
if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
}
return framesRead;
}
// Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing.
// Reads audio data from an AudioBuffer object in device format, returned data will be in a format appropriate for mixing
static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
{
// What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which
// should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important
// detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output
// frames. This can be achieved with ma_data_converter_get_required_input_frame_count().
// frames. This can be achieved with ma_data_converter_get_required_input_frame_count()
ma_uint8 inputBuffer[4096] = { 0 };
ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
@ -2693,6 +2620,83 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr
}
}
// Check if an audio buffer is playing, assuming the audio system mutex has been locked
static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer)
{
bool result = false;
if (buffer != NULL) result = (buffer->playing && !buffer->paused);
return result;
}
// Stop an audio buffer, assuming the audio system mutex has been locked
static void StopAudioBufferInLockedState(AudioBuffer *buffer)
{
if (buffer != NULL)
{
if (IsAudioBufferPlayingInLockedState(buffer))
{
buffer->playing = false;
buffer->paused = false;
buffer->frameCursorPos = 0;
buffer->framesProcessed = 0;
buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true;
}
}
}
// Update audio stream, assuming the audio system mutex has been locked
static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount)
{
if (stream.buffer != NULL)
{
if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1])
{
ma_uint32 subBufferToUpdate = 0;
if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1])
{
// Both buffers are available for updating
// Update the first one and make sure the cursor is moved back to the front
subBufferToUpdate = 0;
stream.buffer->frameCursorPos = 0;
}
else
{
// Just update whichever sub-buffer is processed
subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1;
}
ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2;
unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
// Total frames processed in buffer is always the complete size, filled with 0 if required
stream.buffer->framesProcessed += subBufferSizeInFrames;
// Does this API expect a whole buffer to be updated in one go?
// Assuming so, but if not will need to change this logic
if (subBufferSizeInFrames >= (ma_uint32)frameCount)
{
ma_uint32 framesToWrite = (ma_uint32)frameCount;
ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
memcpy(subBuffer, data, bytesToWrite);
// Any leftover frames should be filled with zeros
ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false;
}
else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer");
}
else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating");
}
}
// Some required functions for audio standalone module version
#if defined(RAUDIO_STANDALONE)
// Check file extension