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Remove end-line spaces

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Ray
2018-11-06 15:10:50 +01:00
parent e340517a73
commit fc1c9505ba
8 changed files with 431 additions and 430 deletions
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170
src/audio.c
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@ -80,7 +80,7 @@
#endif
#include "external/mini_al.h" // mini_al audio library
// NOTE: Cannot be implement here because it conflicts with
// NOTE: Cannot be implement here because it conflicts with
// Win32 APIs: Rectangle, CloseWindow(), ShowCursor(), PlaySoundA()
#include <stdlib.h> // Required for: malloc(), free()
@ -132,12 +132,12 @@
// Types and Structures Definition
//----------------------------------------------------------------------------------
typedef enum {
MUSIC_AUDIO_OGG = 0,
MUSIC_AUDIO_FLAC,
MUSIC_AUDIO_MP3,
MUSIC_MODULE_XM,
MUSIC_MODULE_MOD
typedef enum {
MUSIC_AUDIO_OGG = 0,
MUSIC_AUDIO_FLAC,
MUSIC_AUDIO_MP3,
MUSIC_MODULE_XM,
MUSIC_MODULE_MOD
} MusicContextType;
// Music type (file streaming from memory)
@ -167,12 +167,12 @@ typedef struct MusicData {
} MusicData;
#if defined(AUDIO_STANDALONE)
typedef enum {
LOG_INFO = 0,
LOG_ERROR,
LOG_WARNING,
LOG_DEBUG,
LOG_OTHER
typedef enum {
LOG_INFO = 0,
LOG_ERROR,
LOG_WARNING,
LOG_DEBUG,
LOG_OTHER
} TraceLogType;
#endif
@ -214,7 +214,7 @@ typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioB
// Audio buffer structure
// NOTE: Slightly different logic is used when feeding data to the playback device depending on whether or not data is streamed
typedef struct AudioBuffer AudioBuffer;
typedef struct AudioBuffer AudioBuffer;
struct AudioBuffer {
mal_dsp dsp; // Required for format conversion
float volume;
@ -239,7 +239,7 @@ static bool isAudioInitialized = MAL_FALSE;
static float masterVolume = 1.0f;
// Audio buffers are tracked in a linked list
static AudioBuffer *firstAudioBuffer = NULL;
static AudioBuffer *firstAudioBuffer = NULL;
static AudioBuffer *lastAudioBuffer = NULL;
// mini_al functions declaration
@ -268,7 +268,7 @@ static void OnLog(mal_context *pContext, mal_device *pDevice, const char *messag
{
(void)pContext;
(void)pDevice;
TraceLog(LOG_ERROR, message); // All log messages from mini_al are errors
}
@ -291,30 +291,30 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameC
if (!audioBuffer->playing || audioBuffer->paused) continue;
mal_uint32 framesRead = 0;
for (;;)
for (;;)
{
if (framesRead > frameCount)
if (framesRead > frameCount)
{
TraceLog(LOG_DEBUG, "Mixed too many frames from audio buffer");
break;
}
if (framesRead == frameCount) break;
// Just read as much data as we can from the stream.
mal_uint32 framesToRead = (frameCount - framesRead);
while (framesToRead > 0)
while (framesToRead > 0)
{
float tempBuffer[1024]; // 512 frames for stereo.
mal_uint32 framesToReadRightNow = framesToRead;
if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS)
if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS)
{
framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
}
mal_uint32 framesJustRead = (mal_uint32)mal_dsp_read(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, audioBuffer->dsp.pUserData);
if (framesJustRead > 0)
if (framesJustRead > 0)
{
float *framesOut = (float *)pFramesOut + (framesRead*device.channels);
float *framesIn = tempBuffer;
@ -325,16 +325,16 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameC
}
// If we weren't able to read all the frames we requested, break.
if (framesJustRead < framesToReadRightNow)
if (framesJustRead < framesToReadRightNow)
{
if (!audioBuffer->looping)
if (!audioBuffer->looping)
{
StopAudioBuffer(audioBuffer);
break;
}
else
}
else
{
// Should never get here, but just for safety,
// Should never get here, but just for safety,
// move the cursor position back to the start and continue the loop.
audioBuffer->frameCursorPos = 0;
continue;
@ -342,13 +342,13 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameC
}
}
// If for some reason we weren't able to read every frame we'll need to break from the loop.
// If for some reason we weren't able to read every frame we'll need to break from the loop.
// Not doing this could theoretically put us into an infinite loop.
if (framesToRead > 0) break;
}
}
}
mal_mutex_unlock(&audioLock);
return frameCount; // We always output the same number of frames that were originally requested.
@ -361,8 +361,8 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
mal_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
if (currentSubBufferIndex > 1)
if (currentSubBufferIndex > 1)
{
TraceLog(LOG_DEBUG, "Frame cursor position moved too far forward in audio stream");
return 0;
@ -381,11 +381,11 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
{
// We break from this loop differently depending on the buffer's usage. For static buffers, we simply fill as much data as we can. For
// streaming buffers we only fill the halves of the buffer that are processed. Unprocessed halves must keep their audio data in-tact.
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
if (framesRead >= frameCount) break;
}
else
}
else
{
if (isSubBufferProcessed[currentSubBufferIndex]) break;
}
@ -394,11 +394,11 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
if (totalFramesRemaining == 0) break;
mal_uint32 framesRemainingInOutputBuffer;
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos;
}
else
}
else
{
mal_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex;
framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
@ -412,7 +412,7 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
framesRead += framesToRead;
// If we've read to the end of the buffer, mark it as processed.
if (framesToRead == framesRemainingInOutputBuffer)
if (framesToRead == framesRemainingInOutputBuffer)
{
audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true;
isSubBufferProcessed[currentSubBufferIndex] = true;
@ -420,7 +420,7 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
currentSubBufferIndex = (currentSubBufferIndex + 1)%2;
// We need to break from this loop if we're not looping.
if (!audioBuffer->looping)
if (!audioBuffer->looping)
{
StopAudioBuffer(audioBuffer);
break;
@ -430,7 +430,7 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
// Zero-fill excess.
mal_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining > 0)
if (totalFramesRemaining > 0)
{
memset((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);
@ -447,9 +447,9 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
// NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function.
static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 frameCount, float localVolume)
{
for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
{
for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel)
for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel)
{
float *frameOut = framesOut + (iFrame*device.channels);
const float *frameIn = framesIn + (iFrame*device.channels);
@ -519,7 +519,7 @@ void InitAudioDevice(void)
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
if (!isAudioInitialized)
if (!isAudioInitialized)
{
TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
return;
@ -543,7 +543,7 @@ void SetMasterVolume(float volume)
{
if (volume < 0.0f) volume = 0.0f;
else if (volume > 1.0f) volume = 1.0f;
masterVolume = volume;
}
@ -574,7 +574,7 @@ AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint3
dspConfig.pUserData = audioBuffer;
dspConfig.allowDynamicSampleRate = MAL_TRUE; // <-- Required for pitch shifting.
mal_result resultMAL = mal_dsp_init(&dspConfig, &audioBuffer->dsp);
if (resultMAL != MAL_SUCCESS)
if (resultMAL != MAL_SUCCESS)
{
TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to create data conversion pipeline");
free(audioBuffer);
@ -716,10 +716,10 @@ void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch)
void TrackAudioBuffer(AudioBuffer *audioBuffer)
{
mal_mutex_lock(&audioLock);
{
if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer;
else
else
{
lastAudioBuffer->next = audioBuffer;
audioBuffer->prev = lastAudioBuffer;
@ -727,7 +727,7 @@ void TrackAudioBuffer(AudioBuffer *audioBuffer)
lastAudioBuffer = audioBuffer;
}
mal_mutex_unlock(&audioLock);
}
@ -735,7 +735,7 @@ void TrackAudioBuffer(AudioBuffer *audioBuffer)
void UntrackAudioBuffer(AudioBuffer *audioBuffer)
{
mal_mutex_lock(&audioLock);
{
if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next;
else audioBuffer->prev->next = audioBuffer->next;
@ -746,7 +746,7 @@ void UntrackAudioBuffer(AudioBuffer *audioBuffer)
audioBuffer->prev = NULL;
audioBuffer->next = NULL;
}
mal_mutex_unlock(&audioLock);
}
@ -816,7 +816,7 @@ Sound LoadSoundFromWave(Wave wave)
{
// When using mini_al 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 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.
//
@ -861,7 +861,7 @@ void UnloadSound(Sound sound)
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
AudioBuffer *audioBuffer = (AudioBuffer *)sound.audioBuffer;
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
@ -878,9 +878,9 @@ void UpdateSound(Sound sound, const void *data, int samplesCount)
void ExportWave(Wave wave, const char *fileName)
{
bool success = false;
if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
else if (IsFileExtension(fileName, ".raw"))
else if (IsFileExtension(fileName, ".raw"))
{
// Export raw sample data (without header)
// NOTE: It's up to the user to track wave parameters
@ -888,7 +888,7 @@ void ExportWave(Wave wave, const char *fileName)
success = fwrite(wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8, 1, rawFile);
fclose(rawFile);
}
if (success) TraceLog(LOG_INFO, "Wave exported successfully: %s", fileName);
else TraceLog(LOG_WARNING, "Wave could not be exported.");
}
@ -897,12 +897,12 @@ void ExportWave(Wave wave, const char *fileName)
void ExportWaveAsCode(Wave wave, const char *fileName)
{
#define BYTES_TEXT_PER_LINE 20
char varFileName[256] = { 0 };
int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
FILE *txtFile = fopen(fileName, "wt");
fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes //\n");
@ -917,7 +917,7 @@ void ExportWaveAsCode(Wave wave, const char *fileName)
// Get file name from path and convert variable name to uppercase
strcpy(varFileName, GetFileNameWithoutExt(fileName));
for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
fprintf(txtFile, "// Wave data information\n");
fprintf(txtFile, "#define %s_SAMPLE_COUNT %i\n", varFileName, wave.sampleCount);
fprintf(txtFile, "#define %s_SAMPLE_RATE %i\n", varFileName, wave.sampleRate);
@ -983,7 +983,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
mal_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
mal_uint32 frameCount = (mal_uint32)mal_convert_frames(NULL, formatOut, channels, sampleRate, NULL, formatIn, wave->channels, wave->sampleRate, frameCountIn);
if (frameCount == 0)
if (frameCount == 0)
{
TraceLog(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion.");
return;
@ -992,7 +992,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
void *data = malloc(frameCount*channels*(sampleSize/8));
frameCount = (mal_uint32)mal_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn);
if (frameCount == 0)
if (frameCount == 0)
{
TraceLog(LOG_ERROR, "WaveFormat() : Format conversion failed.");
return;
@ -1130,16 +1130,16 @@ Music LoadMusicStream(const char *fileName)
TraceLog(LOG_INFO, "[%s] MP3 sample rate: %i", fileName, music->ctxMp3.sampleRate);
TraceLog(LOG_INFO, "[%s] MP3 bits per sample: %i", fileName, 32);
TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, music->ctxMp3.channels);
music->stream = InitAudioStream(music->ctxMp3.sampleRate, 32, music->ctxMp3.channels);
// TODO: There is not an easy way to compute the total number of samples available
// in an MP3, frames size could be variable... we tried with a 60 seconds music... but crashes...
music->totalSamples = drmp3_get_pcm_frame_count(&music->ctxMp3)*music->ctxMp3.channels;
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_MP3;
music->loopCount = -1; // Infinite loop by default
TraceLog(LOG_INFO, "[%s] MP3 total samples: %i", fileName, music->totalSamples);
}
}
@ -1186,7 +1186,7 @@ Music LoadMusicStream(const char *fileName)
}
#endif
else musicLoaded = false;
if (!musicLoaded)
{
if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close(music->ctxOgg);
@ -1238,7 +1238,7 @@ void UnloadMusicStream(Music music)
void PlayMusicStream(Music music)
{
AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.audioBuffer;
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
@ -1250,7 +1250,7 @@ void PlayMusicStream(Music music)
// // just make sure to play again on window restore
// if (IsMusicPlaying(music)) PlayMusicStream(music);
mal_uint32 frameCursorPos = audioBuffer->frameCursorPos;
PlayAudioStream(music->stream); // <-- This resets the cursor position.
audioBuffer->frameCursorPos = frameCursorPos;
@ -1273,7 +1273,7 @@ void ResumeMusicStream(Music music)
void StopMusicStream(Music music)
{
StopAudioStream(music->stream);
// Restart music context
switch (music->ctxType)
{
@ -1332,18 +1332,18 @@ void UpdateMusicStream(Music music)
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
case MUSIC_AUDIO_MP3:
{
// NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
unsigned int numFramesMp3 = (unsigned int)drmp3_read_pcm_frames_f32(&music->ctxMp3, samplesCount/music->stream.channels, (float *)pcm);
drmp3_read_pcm_frames_f32(&music->ctxMp3, samplesCount/music->stream.channels, (float *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM:
case MUSIC_MODULE_XM:
{
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2 --> WEIRD
jar_xm_generate_samples_16bit(music->ctxXm, (short *)pcm, samplesCount/2);
jar_xm_generate_samples_16bit(music->ctxXm, (short *)pcm, samplesCount/2);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
@ -1369,7 +1369,7 @@ void UpdateMusicStream(Music music)
if (streamEnding)
{
StopMusicStream(music); // Stop music (and reset)
// Decrease loopCount to stop when required
if (music->loopCount > 0)
{
@ -1475,7 +1475,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
void CloseAudioStream(AudioStream stream)
{
DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer);
TraceLog(LOG_INFO, "[AUD ID %i] Unloaded audio stream data", stream.source);
}
@ -1494,7 +1494,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
{
mal_uint32 subBufferToUpdate;
if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->isSubBufferProcessed[1])
{
// Both buffers are available for updating. Update the first one and make sure the cursor is moved back to the front.
@ -1514,7 +1514,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
if (subBufferSizeInFrames >= (mal_uint32)samplesCount/stream.channels)
{
mal_uint32 framesToWrite = subBufferSizeInFrames;
if (framesToWrite > ((mal_uint32)samplesCount/stream.channels)) framesToWrite = (mal_uint32)samplesCount/stream.channels;
mal_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
@ -1522,8 +1522,8 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
// Any leftover frames should be filled with zeros.
mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
if (leftoverFrameCount > 0)
if (leftoverFrameCount > 0)
{
memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
}
@ -1723,7 +1723,7 @@ static int SaveWAV(Wave wave, const char *fileName)
{
int success = 0;
int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
// Basic WAV headers structs
typedef struct {
char chunkID[4];
@ -1748,7 +1748,7 @@ static int SaveWAV(Wave wave, const char *fileName)
} WaveData;
FILE *wavFile = fopen(fileName, "wb");
if (wavFile == NULL) TraceLog(LOG_WARNING, "[%s] WAV audio file could not be created", fileName);
else
{
@ -1784,7 +1784,7 @@ static int SaveWAV(Wave wave, const char *fileName)
waveData.subChunkID[2] = 't';
waveData.subChunkID[3] = 'a';
waveData.subChunkSize = dataSize;
success = fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile);
success = fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile);
success = fwrite(&waveData, sizeof(WaveData), 1, wavFile);
@ -1793,7 +1793,7 @@ static int SaveWAV(Wave wave, const char *fileName)
fclose(wavFile);
}
// If all data has been written correctly to file, success = 1
return success;
}
@ -1812,7 +1812,7 @@ static Wave LoadOGG(const char *fileName)
else
{
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
wave.sampleRate = info.sample_rate;
wave.sampleSize = 16; // 16 bit per sample (short)
wave.channels = info.channels;
@ -1872,7 +1872,7 @@ static Wave LoadMP3(const char *fileName)
uint64_t totalFrameCount = 0;
drmp3_config config = { 0 };
wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);
wave.channels = config.outputChannels;
wave.sampleRate = config.outputSampleRate;
wave.sampleCount = (int)totalFrameCount*wave.channels;
@ -1895,7 +1895,7 @@ bool IsFileExtension(const char *fileName, const char *ext)
{
bool result = false;
const char *fileExt;
if ((fileExt = strrchr(fileName, '.')) != NULL)
{
if (strcmp(fileExt, ext) == 0) result = true;