Support WAV music streaming #1198

Switched custom WAV laoding/saving funtionality to drwav library, it also provides the required mechanisms to stream wav data.
This commit is contained in:
raysan5 2020-05-23 23:19:59 +02:00
parent 427e543d84
commit a6fcd32339

View File

@ -197,14 +197,13 @@ typedef struct tagBITMAPINFOHEADER {
#include "external/jar_mod.h" // MOD loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
#define DRFLAC_MALLOC RL_MALLOC
#define DRFLAC_REALLOC RL_REALLOC
#define DRFLAC_FREE RL_FREE
#if defined(SUPPORT_FILEFORMAT_WAV)
#define DRWAV_MALLOC RL_MALLOC
#define DRWAV_REALLOC RL_REALLOC
#define DRWAV_FREE RL_FREE
#define DR_FLAC_IMPLEMENTATION
#define DR_FLAC_NO_WIN32_IO
#include "external/dr_flac.h" // FLAC loading functions
#define DR_WAV_IMPLEMENTATION
#include "external/dr_wav.h" // WAV loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
@ -216,6 +215,16 @@ typedef struct tagBITMAPINFOHEADER {
#include "external/dr_mp3.h" // MP3 loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
#define DRFLAC_MALLOC RL_MALLOC
#define DRFLAC_REALLOC RL_REALLOC
#define DRFLAC_FREE RL_FREE
#define DR_FLAC_IMPLEMENTATION
#define DR_FLAC_NO_WIN32_IO
#include "external/dr_flac.h" // FLAC loading functions
#endif
#if defined(_MSC_VER)
#undef bool
#endif
@ -437,11 +446,11 @@ void InitAudioDevice(void)
}
TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully");
TRACELOG(LOG_INFO, " > Backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
TRACELOG(LOG_INFO, " > Format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
TRACELOG(LOG_INFO, " > Channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
TRACELOG(LOG_INFO, " > Sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
TRACELOG(LOG_INFO, " > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
TRACELOG(LOG_INFO, " > Backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
TRACELOG(LOG_INFO, " > Format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
TRACELOG(LOG_INFO, " > Channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
TRACELOG(LOG_INFO, " > Sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
TRACELOG(LOG_INFO, " > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
InitAudioBufferPool();
@ -1046,6 +1055,24 @@ Music LoadMusicStream(const char *fileName)
bool musicLoaded = false;
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (IsFileExtension(fileName, ".wav"))
{
drwav *ctxWav = RL_MALLOC(sizeof(drwav));
bool success = drwav_init_file(ctxWav, fileName, NULL);
if (success)
{
music.ctxType = MUSIC_AUDIO_WAV;
music.ctxData = ctxWav;
music.stream = InitAudioStream(ctxWav->sampleRate, ctxWav->bitsPerSample, ctxWav->channels);
music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (IsFileExtension(fileName, ".ogg"))
{
@ -1151,6 +1178,9 @@ Music LoadMusicStream(const char *fileName)
if (!musicLoaded)
{
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
#endif
@ -1188,6 +1218,9 @@ void UnloadMusicStream(Music music)
CloseAudioStream(music.stream);
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
#endif
@ -1239,6 +1272,9 @@ void StopMusicStream(Music music)
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_WAV)
case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
#endif
@ -1283,6 +1319,14 @@ void UpdateMusicStream(Music music)
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_WAV)
case MUSIC_AUDIO_WAV:
{
// NOTE: Returns the number of samples to process (not required)
drwav_read_pcm_frames_s16((drwav *)music.ctxData, samplesCount/music.stream.channels, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG:
{
@ -1816,204 +1860,43 @@ static void CloseAudioBufferPool(void)
// Load WAV file into Wave structure
static Wave LoadWAV(const char *fileName)
{
// Basic WAV headers structs
typedef struct {
char chunkID[4];
int chunkSize;
char format[4];
} WAVRiffHeader;
typedef struct {
char subChunkID[4];
int subChunkSize;
short audioFormat;
short numChannels;
int sampleRate;
int byteRate;
short blockAlign;
short bitsPerSample;
} WAVFormat;
typedef struct {
char subChunkID[4];
int subChunkSize;
} WAVData;
WAVRiffHeader wavRiffHeader = { 0 };
WAVFormat wavFormat = { 0 };
WAVData wavData = { 0 };
Wave wave = { 0 };
FILE *wavFile = NULL;
wavFile = fopen(fileName, "rb");
// Decode an entire FLAC file in one go
unsigned long long int totalPCMFrameCount = 0;
wave.data = drwav_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalPCMFrameCount, NULL);
if (wavFile == NULL)
{
TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open WAV file", fileName);
wave.data = NULL;
}
if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load WAV data", fileName);
else
{
// Read in the first chunk into the struct
fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile);
wave.sampleCount = (unsigned int)totalPCMFrameCount*wave.channels;
wave.sampleSize = 16;
// Check for RIFF and WAVE tags
if ((wavRiffHeader.chunkID[0] != 'R') ||
(wavRiffHeader.chunkID[1] != 'I') ||
(wavRiffHeader.chunkID[2] != 'F') ||
(wavRiffHeader.chunkID[3] != 'F') ||
(wavRiffHeader.format[0] != 'W') ||
(wavRiffHeader.format[1] != 'A') ||
(wavRiffHeader.format[2] != 'V') ||
(wavRiffHeader.format[3] != 'E'))
{
TRACELOG(LOG_WARNING, "WAVE: [%s] RIFF or WAVE header are not valid", fileName);
}
else
{
// Read in the 2nd chunk for the wave info
fread(&wavFormat, sizeof(WAVFormat), 1, wavFile);
// Check for fmt tag
if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') ||
(wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' '))
{
TRACELOG(LOG_WARNING, "WAVE: [%s] Wave format header is not valid", fileName);
}
else
{
// Check for extra parameters;
if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
// Read in the the last byte of data before the sound file
fread(&wavData, sizeof(WAVData), 1, wavFile);
// Check for data tag
if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') ||
(wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a'))
{
TRACELOG(LOG_WARNING, "WAVE: [%s] Data header is not valid", fileName);
}
else
{
// Allocate memory for data
wave.data = RL_MALLOC(wavData.subChunkSize);
// Read in the sound data into the soundData variable
fread(wave.data, wavData.subChunkSize, 1, wavFile);
// Store wave parameters
wave.sampleRate = wavFormat.sampleRate;
wave.sampleSize = wavFormat.bitsPerSample;
wave.channels = wavFormat.numChannels;
// NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
{
TRACELOG(LOG_WARNING, "WAVE: [%s] Sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
WaveFormat(&wave, wave.sampleRate, 16, wave.channels);
}
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2)
{
WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
TRACELOG(LOG_WARNING, "WAVE: [%s] Channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
}
// NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;
TRACELOG(LOG_INFO, "WAVE: [%s] File loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
}
}
fclose(wavFile);
TRACELOG(LOG_INFO, "WAVE: [%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
return wave;
}
// Save wave data as WAV file
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];
int chunkSize;
char format[4];
} RiffHeader;
typedef struct {
char subChunkID[4];
int subChunkSize;
short audioFormat;
short numChannels;
int sampleRate;
int byteRate;
short blockAlign;
short bitsPerSample;
} WaveFormat;
typedef struct {
char subChunkID[4];
int subChunkSize;
} WaveData;
FILE *wavFile = fopen(fileName, "wb");
if (wavFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open audio file", fileName);
else
{
RiffHeader riffHeader;
WaveFormat waveFormat;
WaveData waveData;
// Fill structs with data
riffHeader.chunkID[0] = 'R';
riffHeader.chunkID[1] = 'I';
riffHeader.chunkID[2] = 'F';
riffHeader.chunkID[3] = 'F';
riffHeader.chunkSize = 44 - 4 + wave.sampleCount*wave.sampleSize/8;
riffHeader.format[0] = 'W';
riffHeader.format[1] = 'A';
riffHeader.format[2] = 'V';
riffHeader.format[3] = 'E';
waveFormat.subChunkID[0] = 'f';
waveFormat.subChunkID[1] = 'm';
waveFormat.subChunkID[2] = 't';
waveFormat.subChunkID[3] = ' ';
waveFormat.subChunkSize = 16;
waveFormat.audioFormat = 1;
waveFormat.numChannels = wave.channels;
waveFormat.sampleRate = wave.sampleRate;
waveFormat.byteRate = wave.sampleRate*wave.sampleSize/8;
waveFormat.blockAlign = wave.sampleSize/8;
waveFormat.bitsPerSample = wave.sampleSize;
waveData.subChunkID[0] = 'd';
waveData.subChunkID[1] = 'a';
waveData.subChunkID[2] = 't';
waveData.subChunkID[3] = 'a';
waveData.subChunkSize = dataSize;
fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile);
fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile);
fwrite(&waveData, sizeof(WaveData), 1, wavFile);
success = (int)fwrite(wave.data, dataSize, 1, wavFile);
fclose(wavFile);
}
// If all data has been written correctly to file, success = 1
return success;
drwav wav = { 0 };
drwav_data_format format = { 0 };
format.container = drwav_container_riff; // <-- drwav_container_riff = normal WAV files, drwav_container_w64 = Sony Wave64.
format.format = DR_WAVE_FORMAT_PCM; // <-- Any of the DR_WAVE_FORMAT_* codes.
format.channels = wave.channels;
format.sampleRate = wave.sampleRate;
format.bitsPerSample = wave.sampleSize;
drwav_init_file_write(&wav, fileName, &format, NULL);
drwav_write_pcm_frames(&wav, wave.sampleCount/wave.channels, wave.data);
printf("save!\n");
drwav_uninit(&wav);
return true;
}
#endif
@ -2069,9 +1952,6 @@ static Wave LoadFLAC(const char *fileName)
wave.sampleCount = (unsigned int)totalSampleCount;
wave.sampleSize = 16;
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] FLAC channels number (%i) not supported", fileName, wave.channels);
TRACELOG(LOG_INFO, "WAVE: [%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}