MP3Encoder::~MP3Encoder()
{
if (m_cookie != 0) {
mp3_done(m_cookie);
}
free(m_buffer);
}
static void mixer_channel_free(unsigned channel)
{
switch (mixer_map[channel].type) {
case mixer_raw_file :
fzclose(mixer_map[channel].file);
break;
case mixer_mp3_file :
fzclose(mixer_map[channel].file);
mp3_done(&mixer_map[channel].mp3);
break;
default:
break;
}
mixer_map[channel].type = mixer_none;
/* clear any stored data */
memset(mixer_buffer[channel], 0, sizeof(mixer_buffer[channel]));
}
void *loom_asset_soundDeserializer( void *buffer, size_t bufferLen, LoomAssetCleanupCallback *dtor )
{
loom_asset_sound_t *sound = (loom_asset_sound_t*)lmAlloc(gAssetAllocator, sizeof(loom_asset_sound_t));
memset(sound, 0, sizeof(loom_asset_sound_t));
unsigned char *charBuff = (unsigned char *)buffer;
// Look for magic header in buffer.
if(charBuff[0] == 0x4f
&& charBuff[1] == 0x67
&& charBuff[2] == 0x67
&& charBuff[3] == 0x53)
{
// It's an Ogg, assume vorbis and throw it to stb_vorbis.
int channels = 0;
short *outputBuffer = NULL;
int sampleCount = stb_vorbis_decode_memory(charBuff, (int)bufferLen, &channels, &outputBuffer);
if(sampleCount < 0)
{
lmLogError(gSoundAssetGroup, "Failed to decode Ogg Vorbis");
loom_asset_soundDtor(&sound);
return NULL;
}
sound->channels = channels;
sound->bytesPerSample = 2;
sound->sampleCount = sampleCount;
sound->bufferSize = sampleCount * channels * 2;
sound->sampleRate = 44100; // TODO: This should be variable
// We can skip this if we get clever about allocations in stbv.
sound->buffer = lmAlloc(gAssetAllocator, sound->bufferSize);
memcpy(sound->buffer, outputBuffer, sound->bufferSize);
free(outputBuffer);
}
else if((charBuff[0] == 0x49 // ID3
&& charBuff[1] == 0x44
&& charBuff[2] == 0x33)
|| (charBuff[0] == 0xff // Missing ID3 Tag
&& charBuff[1] == 0xfb))
{
// It's an MP3, y'all!
short *outBuffer = (short*)lmAlloc(gAssetAllocator, MP3_MAX_SAMPLES_PER_FRAME * 2);
mp3_info_t mp3Info;
// Decode once to get total size.
size_t totalBytes = 0;
size_t bytesRead = 0, bytesLeft = bufferLen;
mp3_decoder_t decmp3 = mp3_create();
for(;;)
{
int bytesDecoded = mp3_decode(decmp3, charBuff + bytesRead, (int)bytesLeft, outBuffer, &mp3Info);
bytesRead += bytesDecoded;
bytesLeft -= bytesDecoded;
totalBytes += mp3Info.audio_bytes;
if(bytesDecoded > 0)
continue;
// Clean up.
mp3_done(decmp3);
break;
}
// Great, set up the sound asset.
// TODO: Warn about non 44.1khz mp3s.
sound->channels = mp3Info.channels;
sound->bytesPerSample = 2;
sound->sampleCount = (int)totalBytes / sound->bytesPerSample;
sound->bufferSize = sound->channels * sound->bytesPerSample * sound->sampleCount;
sound->sampleRate = 44100; // TODO: This should be variable
sound->buffer = lmAlloc(gAssetAllocator, sound->bufferSize);
// Decode again to get real samples.
decmp3 = mp3_create();
bytesRead = 0; bytesLeft = bufferLen;
int curBufferOffset = 0;
for(;;)
{
int bytesDecoded = mp3_decode(decmp3, charBuff + bytesRead, (int)bytesLeft, outBuffer, &mp3Info);
bytesRead += bytesDecoded;
bytesLeft -= bytesDecoded;
memcpy(((unsigned char*)sound->buffer) + curBufferOffset, outBuffer, mp3Info.audio_bytes);
curBufferOffset += mp3Info.audio_bytes;
if(bytesDecoded > 0)
continue;
// Clean up.
mp3_done(decmp3);
break;
}
// Awesome, all set!
lmFree(gAssetAllocator, outBuffer);
}
else if(charBuff[0] == 0x52 // 'RIFF'
&& charBuff[1] == 0x49
&& charBuff[2] == 0x46
&& charBuff[3] == 0x46)
{
// We've got a wav file
wav_info wav;
bool wavLoadSuccess = load_wav(charBuff, bufferLen, NULL, &wav);
if (!wavLoadSuccess)
{
lmLogError(gSoundAssetGroup, "Failed to load wav format info");
loom_asset_soundDtor(sound);
return 0;
}
sound->channels = wav.numChannels;
sound->bytesPerSample = wav.sampleSize / 8; // wav sample size is in bits
if (sound->bytesPerSample != 1 && sound->bytesPerSample != 2)
{
lmLogError(gSoundAssetGroup, "Unsupported wav format. Currently only 8-bit or 16-bit PCM are supported");
loom_asset_soundDtor(sound);
return 0;
}
sound->bufferSize = wav.sampleDataSize;
sound->sampleCount = sound->bufferSize / sound->bytesPerSample;
sound->sampleRate = wav.samplesPerSecond;
sound->buffer = lmAlloc(gAssetAllocator, sound->bufferSize);
bool dataCopySuccess = load_wav(charBuff, bufferLen, (uint8_t*)sound->buffer, NULL);
if (!dataCopySuccess)
{
lmLogError(gSoundAssetGroup, "Failed to copy wav data");
loom_asset_soundDtor(sound);
return 0;
}
}
else
{
lmLogError(gSoundAssetGroup, "Failed to identify sound buffer by magic number!");
loom_asset_soundDtor(sound);
return 0;
}
*dtor = loom_asset_soundDtor;
if(!sound->buffer)
{
lmLogError(gSoundAssetGroup, "Sound load failed due to this cryptic reason: %s", "(unknown)");
lmFree(gAssetAllocator, sound);
return 0;
}
lmLogDebug(gSoundAssetGroup, "Sound allocation: %d bytes", sound->bufferSize);
return sound;
}
void *loom_asset_soundDeserializer( void *buffer, size_t bufferLen, LoomAssetCleanupCallback *dtor )
{
loom_asset_sound_t *sound = (loom_asset_sound_t*)lmAlloc(gAssetAllocator, sizeof(loom_asset_sound_t));
unsigned char *charBuff = (unsigned char *)buffer;
// Look for magic header in buffer.
if(charBuff[0] == 0x4f
&& charBuff[1] == 0x67
&& charBuff[2] == 0x67
&& charBuff[3] == 0x53)
{
// It's an Ogg, assume vorbis and throw it to stb_vorbis.
int channels = 0;
short *outputBuffer = NULL;
int sampleCount = stb_vorbis_decode_memory(charBuff, bufferLen, &channels, &outputBuffer);
if(sampleCount < 0)
{
lmLogError(gSoundAssetGroup, "Failed to decode Ogg Vorbis!");
return NULL;
}
sound->channels = channels;
sound->bytesPerSample = 2;
sound->sampleCount = sampleCount;
sound->bufferSize = sampleCount * channels * 2;
// We can skip this if we get clever about allocations in stbv.
sound->buffer = lmAlloc(gAssetAllocator, sound->bufferSize);
memcpy(sound->buffer, outputBuffer, sound->bufferSize);
free(outputBuffer);
}
else if((charBuff[0] == 0x49 // ID3
&& charBuff[1] == 0x44
&& charBuff[2] == 0x33)
|| (charBuff[0] == 0xff // Missing ID3 Tag
&& charBuff[1] == 0xfb))
{
// It's an MP3, y'all!
short *outBuffer = (short*)lmAlloc(gAssetAllocator, MP3_MAX_SAMPLES_PER_FRAME * 2);
mp3_info_t mp3Info;
// Decode once to get total size.
int totalBytes = 0;
int bytesRead = 0, bytesLeft = bufferLen;
mp3_decoder_t decmp3 = mp3_create();
for(;;)
{
int bytesDecoded = mp3_decode(decmp3, charBuff + bytesRead, bytesLeft, outBuffer, &mp3Info);
bytesRead += bytesDecoded;
bytesLeft -= bytesDecoded;
totalBytes += mp3Info.audio_bytes;
if(bytesDecoded > 0)
continue;
// Clean up.
mp3_done(decmp3);
break;
}
// Great, set up the sound asset.
// TODO: Warn about non 44.1khz mp3s.
sound->channels = mp3Info.channels;
sound->bytesPerSample = 2;
sound->sampleCount = totalBytes / sound->bytesPerSample;
sound->bufferSize = sound->channels * sound->bytesPerSample * sound->sampleCount;
sound->buffer = lmAlloc(gAssetAllocator, sound->bufferSize);
// Decode again to get real samples.
decmp3 = mp3_create();
bytesRead = 0; bytesLeft = bufferLen;
int curBufferOffset = 0;
for(;;)
{
int bytesDecoded = mp3_decode(decmp3, charBuff + bytesRead, bytesLeft, outBuffer, &mp3Info);
bytesRead += bytesDecoded;
bytesLeft -= bytesDecoded;
memcpy(((unsigned char*)sound->buffer) + curBufferOffset, outBuffer, mp3Info.audio_bytes);
curBufferOffset += mp3Info.audio_bytes;
if(bytesDecoded > 0)
continue;
// Clean up.
mp3_done(decmp3);
break;
}
// Awesome, all set!
lmFree(gAssetAllocator, outBuffer);
}
else
{
lmLogError(gSoundAssetGroup, "Failed to identify sound buffer by magic number!");
return 0;
}
*dtor = loom_asset_soundDtor;
if(!sound->buffer)
{
lmLogError(gSoundAssetGroup, "Sound load failed due to this cryptic reason: %s", "(unknown)");
lmFree(gAssetAllocator, sound);
return 0;
}
lmLogError(gSoundAssetGroup, "Allocated %d bytes for a sound!", sound->bufferSize);
return sound;
}
