本文整理汇总了C++中fmod::Sound::unlock方法的典型用法代码示例。如果您正苦于以下问题：C++ Sound::unlock方法的具体用法？C++ Sound::unlock怎么用？C++ Sound::unlock使用的例子？那么, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在类fmod::Sound的用法示例。
在下文中一共展示了Sound::unlock方法的4个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于系统推荐出更棒的C++代码示例。
示例1:
SPtr<Resource> FMODImporter::import(const Path& filePath, SPtr<const ImportOptions> importOptions)
{
WString extension = filePath.getWExtension();
StringUtil::toLowerCase(extension);
AudioFileInfo info;
FMOD::Sound* sound;
String pathStr = filePath.toString();
if(gFMODAudio()._getFMOD()->createSound(pathStr.c_str(), FMOD_CREATESAMPLE, nullptr, &sound) != FMOD_OK)
{
LOGERR("Failed importing audio file: " + pathStr);
return nullptr;
}
FMOD_SOUND_FORMAT format;
INT32 numChannels = 0;
INT32 numBits = 0;
sound->getFormat(nullptr, &format, &numChannels, &numBits);
if(format != FMOD_SOUND_FORMAT_PCM8 && format != FMOD_SOUND_FORMAT_PCM16 && format != FMOD_SOUND_FORMAT_PCM24
&& format != FMOD_SOUND_FORMAT_PCM32 && format != FMOD_SOUND_FORMAT_PCMFLOAT)
{
LOGERR("Failed importing audio file, invalid imported format: " + pathStr);
return nullptr;
}
float frequency = 0.0f;
sound->getDefaults(&frequency, nullptr);
UINT32 size;
sound->getLength(&size, FMOD_TIMEUNIT_PCMBYTES);
info.bitDepth = numBits;
info.numChannels = numChannels;
info.sampleRate = (UINT32)frequency;
info.numSamples = size / (info.bitDepth / 8);
UINT32 bytesPerSample = info.bitDepth / 8;
UINT32 bufferSize = info.numSamples * bytesPerSample;
UINT8* sampleBuffer = (UINT8*)bs_alloc(bufferSize);
assert(bufferSize == size);
UINT8* startData = nullptr;
UINT8* endData = nullptr;
UINT32 startSize = 0;
UINT32 endSize = 0;
sound->lock(0, size, (void**)&startData, (void**)&endData, &startSize, &endSize);
if(format == FMOD_SOUND_FORMAT_PCMFLOAT)
{
assert(info.bitDepth == 32);
UINT32* output = (UINT32*)sampleBuffer;
for(UINT32 i = 0; i < info.numSamples; i++)
{
float value = *(((float*)startData) + i);
*output = (UINT32)(value * 2147483647.0f);
output++;
}
}
else
{
memcpy(sampleBuffer, startData, bufferSize);
}
sound->unlock((void**)&startData, (void**)&endData, startSize, endSize);
sound->release();
SPtr<const AudioClipImportOptions> clipIO = std::static_pointer_cast<const AudioClipImportOptions>(importOptions);
// If 3D, convert to mono
if (clipIO->getIs3D() && info.numChannels > 1)
{
UINT32 numSamplesPerChannel = info.numSamples / info.numChannels;
UINT32 monoBufferSize = numSamplesPerChannel * bytesPerSample;
UINT8* monoBuffer = (UINT8*)bs_alloc(monoBufferSize);
AudioUtility::convertToMono(sampleBuffer, monoBuffer, info.bitDepth, numSamplesPerChannel, info.numChannels);
info.numSamples = numSamplesPerChannel;
info.numChannels = 1;
bs_free(sampleBuffer);
sampleBuffer = monoBuffer;
bufferSize = monoBufferSize;
}
// Convert bit depth if needed
if (clipIO->getBitDepth() != info.bitDepth)
{
UINT32 outBufferSize = info.numSamples * (clipIO->getBitDepth() / 8);
UINT8* outBuffer = (UINT8*)bs_alloc(outBufferSize);
AudioUtility::convertBitDepth(sampleBuffer, info.bitDepth, outBuffer, clipIO->getBitDepth(), info.numSamples);
info.bitDepth = clipIO->getBitDepth();
//.........这里部分代码省略.........
示例2: UpdateSound
void FMODVoiceAdapter::UpdateSound(bool isRec)
{
FMOD_RESULT fmodErr;
unsigned int soundLength;
const int sampleSize = 2;
FMOD::Sound *snd = (isRec) ? recSound : sound;
unsigned int& lastPos = (isRec) ? lastRecordingPos : lastPlayPos;
// get current Play or recording position
unsigned int currPos;
if (isRec){
fmodErr=fmodSystem->getRecordPosition(0,&currPos);
RakAssert(fmodErr==FMOD_OK);
} else {
fmodErr=channel->getPosition(&currPos, FMOD_TIMEUNIT_PCM);
RakAssert(fmodErr==FMOD_OK);
}
// Get length of sound in samples
fmodErr=snd->getLength(&soundLength, FMOD_TIMEUNIT_PCM);
RakAssert(fmodErr==FMOD_OK);
// calculate some variables we'll need ahead
int bufferSizeBytes = rakVoice->GetBufferSizeBytes();
// Round down the current position to a multiple of buffer size in samples
currPos -= currPos % (bufferSizeBytes/sampleSize);
if ( ((!isRec)||(isRec && !mute)) && (currPos != lastPos) )
{
void *ptr1, *ptr2;
unsigned int len1, len2;
int blockLength;
blockLength = (int)currPos - (int)lastPos;
// Check for wrap around, and adjust
if (blockLength < 0)
{
blockLength += soundLength;
}
// Lock to get access to the raw data
snd->lock(lastPos * sampleSize, blockLength * sampleSize, &ptr1, &ptr2, &len1, &len2);
// Since the length and current position are both a multiple of bufferSizeBytes
// just treat treat one full buffer at a time
int numFrames = len1 / bufferSizeBytes;
while(numFrames--){
if (isRec) {
BroadcastFrame(ptr1);
} else {
rakVoice->ReceiveFrame(ptr1);
}
ptr1 = (char*)ptr1 + bufferSizeBytes;
}
numFrames = len2 / bufferSizeBytes;
while(numFrames--) {
if (isRec){
BroadcastFrame(ptr2);
} else {
rakVoice->ReceiveFrame(ptr2);
}
ptr2 = (char*)ptr2 + bufferSizeBytes;
}
snd->unlock(ptr1, ptr2, len1, len2);
}
lastPos = currPos;
}示例3: main
//.........这里部分代码省略.........
printf("Press 'Esc' to quit\n");
printf("\n");
fp = fopen("record.wav", "wb");
if (!fp)
{
printf("ERROR : could not open record.wav for writing.\n");
return 1;
}
/*
Write out the wav header. As we don't know the length yet it will be 0.
*/
WriteWavHeader(fp, sound, datalength);
result = sound->getLength(&soundlength, FMOD_TIMEUNIT_PCM);
ERRCHECK(result);
/*
Main loop.
*/
do
{
static unsigned int lastrecordpos = 0;
unsigned int recordpos = 0;
if (_kbhit())
{
key = _getch();
}
system->getRecordPosition(recorddriver, &recordpos);
ERRCHECK(result);
if (recordpos != lastrecordpos)
{
void *ptr1, *ptr2;
int blocklength;
unsigned int len1, len2;
blocklength = (int)recordpos - (int)lastrecordpos;
if (blocklength < 0)
{
blocklength += soundlength;
}
/*
Lock the sound to get access to the raw data.
*/
sound->lock(lastrecordpos * exinfo.numchannels * 2, blocklength * exinfo.numchannels * 2, &ptr1, &ptr2, &len1, &len2); /* * exinfo.numchannels * 2 = stereo 16bit. 1 sample = 4 bytes. */
/*
Write it to disk.
*/
if (ptr1 && len1)
{
datalength += fwrite(ptr1, 1, len1, fp);
}
if (ptr2 && len2)
{
datalength += fwrite(ptr2, 1, len2, fp);
}
/*
Unlock the sound to allow FMOD to use it again.
*/
sound->unlock(ptr1, ptr2, len1, len2);
}
lastrecordpos = recordpos;
printf("%-23s. Record buffer pos = %6d : Record time = %02d:%02d\r", (timeGetTime() / 500) & 1 ? "Recording to record.wav" : "", recordpos, datalength / exinfo.defaultfrequency / exinfo.numchannels / 2 / 60, (datalength / exinfo.defaultfrequency / exinfo.numchannels / 2) % 60);
system->update();
Sleep(10);
} while (key != 27);
printf("\n");
/*
Write back the wav header now that we know its length.
*/
WriteWavHeader(fp, sound, datalength);
fclose(fp);
/*
Shut down
*/
result = sound->release();
ERRCHECK(result);
result = system->release();
ERRCHECK(result);
return 0;
}示例4: main
//.........这里部分代码省略.........
if(iscoreaudio)
{
sound->lock(lastrecordpos * 8, blocklength * 8, &ptr1, &ptr2, &len1, &len2); /* *8 = stereo 32bit. 1 sample = 8 bytes. */
}
else
{
sound->lock(lastrecordpos * 4, blocklength * 4, &ptr1, &ptr2, &len1, &len2); /* *4 = stereo 16bit. 1 sample = 4 bytes. */
}
/*
Write it to disk.
*/
if (ptr1 && len1)
{
#ifdef __BIG_ENDIAN__
if (exinfo.format == FMOD_SOUND_FORMAT_PCM16)
{
signed short *wptr = (signed short *)ptr1;
for (int count = 0; count < len1 >> 1; count++)
{
wptr[count] = SWAPENDIAN_WORD(wptr[count]);
}
}
else if (exinfo.format == FMOD_SOUND_FORMAT_PCMFLOAT)
{
float *fptr = (float *)ptr1;
for (int count = 0; count < len1 >> 2; count++)
{
SWAPENDIAN_FLOAT(fptr[count]);
}
}
#endif
datalength += fwrite(ptr1, 1, len1, fp);
}
if (ptr2 && len2)
{
#ifdef __BIG_ENDIAN__
if (exinfo.format == FMOD_SOUND_FORMAT_PCM16)
{
signed short *wptr = (signed short *)ptr2;
for (int count = 0; count < len2 >> 1; count++)
{
wptr[count] = SWAPENDIAN_WORD(wptr[count]);
}
}
else if (exinfo.format == FMOD_SOUND_FORMAT_PCMFLOAT)
{
float *fptr = (float *)ptr2;
for (int count = 0; count < len2 >> 2; count++)
{
SWAPENDIAN_FLOAT(fptr[count]);
}
}
#endif
datalength += fwrite(ptr2, 1, len2, fp);
}
/*
Unlock the sound to allow FMOD to use it again.
*/
sound->unlock(ptr1, ptr2, len1, len2);
}
lastrecordpos = recordpos;
printf("\rRecord buffer pos = %6d : Record time = %02d:%02d", recordpos, datalength / exinfo.defaultfrequency / 4 / 60, (datalength / exinfo.defaultfrequency / 4) % 60);
fflush(stdout);
system->update();
Sleep(10);
} while (key != 27);
printf("\n");
/*
Write back the wav header now that we know its length.
*/
WriteWavHeader(fp, sound, datalength);
fclose(fp);
/*
Shut down
*/
result = sound->release();
ERRCHECK(result);
result = system->release();
ERRCHECK(result);
return 0;
}