//if a file is dropped onto the playlist from explorer window or from any other data source, we process the file
//and add it to the playlist. Open all subfolders and add process files in them too
bool FileDropTarget::OnDropFiles(wxCoord x, wxCoord y, const wxArrayString &files)
{
wxArrayString dirFiles;
wxDir myDir;
int unsupportedFiles = 0;
for (size_t i = 0; i < files.GetCount(); ++i){
if(myDir.Exists(files[i])){
myDir.Open(files[i]);
size_t itemCount = myDir.GetAllFiles(files[i], &dirFiles);
for(size_t n = 0; n < itemCount; n++){
if(IsFormatSupported(dirFiles[n])){
m_list.AddToPlayList(dirFiles[n]);
}else{
unsupportedFiles++;
}
}
}
else{
if(IsFormatSupported(files[i])){
m_list.AddToPlayList(files[i]);
} else {
unsupportedFiles++;
}
}
}
if(unsupportedFiles != 0)
wxMessageBox(wxString::Format("%d Files couldn't be added\nReason: Unsupported Format",unsupportedFiles));
return true;
}
HRESULT __stdcall CMyDataObject::GetData(FORMATETC *pFormatEtc, STGMEDIUM *pStgMedium)
{
if (!IsFormatSupported(pFormatEtc))
return DV_E_FORMATETC;
// Copy the storage medium data.
pStgMedium->tymed = m_StgMedium.tymed;
pStgMedium->pUnkForRelease = 0;
pStgMedium->hGlobal =
GlobalAlloc(GMEM_SHARE,sizeof(DROPFILES)+(TotalLength+1)*sizeof(TCHAR) );
// ‘ормирование структуры DROPFILES дл¤ драгндропа файлов
DROPFILES *DP = (LPDROPFILES) GlobalLock(pStgMedium->hGlobal);
ZeroMemory(DP, sizeof(DROPFILES));
DP->fWide = TRUE;
DP->pFiles = sizeof(DROPFILES);
LPTSTR Files = (LPTSTR)((PBYTE)DP+ DP->pFiles);
for(size_t i =0; i<m_FileItems.GetCount(); i++)
{
lstrcpy(Files, m_FileItems[i]);
Files += m_FileItems[i].GetLength()+1;
}
*Files=0;
GlobalUnlock(DP);
return S_OK;
}
PVMFStatus PVMFAvcEncPort::SetFormat(PVMFFormatType aFormat)
{
LOG_STACK_TRACE((0, "PVMFAvcEncPort::SetFormat: aFormat=%d", aFormat));
if (!IsFormatSupported(aFormat))
return PVMFFailure;
iFormat = aFormat;
return PVMFSuccess;
}
// ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus OALBuffer::AddAudioDataStatic(char* inAudioData, UInt32 inAudioDataSize, ALenum format, ALsizei freq)
{
#if LOG_VERBOSE
DebugMessageN5("OALBuffer::AddAudioDataStatic() - OALBuffer:inAudioData:inAudioDataSize:format:freq = %ld:%p:%ld:%d:%d", (long int) mSelfToken, inAudioData, (long int) inAudioDataSize, format, freq);
#endif
#if LOG_EXTRAS
DebugMessage("AddAudioDataStatic called: Converting Data Now");
#endif
CAGuard::Locker bufferLock(mBufferLock);
try {
if (!IsFormatSupported(format))
throw ((OSStatus) AL_INVALID_VALUE); // this is not a valid buffer token or is an invalid format
// don't allow if the buffer is in a queue
if (mAttachedSourceList->Size() > 0)
{
DebugMessage("AddAudioDataStatic ATTACHMENT > 0");
throw ((OSStatus) AL_INVALID_OPERATION);
}
mPreConvertedDataSize = (UInt32) inAudioDataSize;
OSStatus result = noErr;
// if this buffer was using memory created by the library, free it now and initialize mData
if (!mAppOwnsBufferMemory && (mData != NULL))
{
free (mData);
mData = NULL;
}
mData = (UInt8*) inAudioData;
mDataSize = (UInt32) inAudioDataSize;
result = FillInASBD(mDataFormat, format, freq);
THROW_RESULT
mPreConvertedDataFormat.SetFrom(mDataFormat); // make sure they are the same so original format info can be returned to caller
}
catch (OSStatus result) {
mData = NULL;
mAppOwnsBufferMemory = false;
DebugMessageN1("AddAudioDataStatic Failed - err = %ld\n", (long int) result);
alSetError(result);
}
catch (...) {
mData = NULL;
mAppOwnsBufferMemory = false;
DebugMessage("AddAudioDataStatic Failed");
alSetError(AL_INVALID_OPERATION);
}
mAppOwnsBufferMemory = true;
return noErr;
}
STDMETHODIMP CAVIOutputPin::SetTimeFormat(const GUID * pFormat)
{
if (IsFormatSupported(pFormat) != S_OK)
return E_FAIL;
if (m_currentTimeFormat != *pFormat)
m_pSplitter->SetTimeFormatForAllPins(pFormat);
return S_OK;
}
void FVulkanDevice::MapFormatSupport(EPixelFormat UEFormat, VkFormat VulkanFormat)
{
FPixelFormatInfo& FormatInfo = GPixelFormats[UEFormat];
FormatInfo.PlatformFormat = VulkanFormat;
FormatInfo.Supported = IsFormatSupported(VulkanFormat);
if(!FormatInfo.Supported)
{
UE_LOG(LogVulkanRHI, Warning, TEXT("EPixelFormat(%d) is not supported"), (int32)UEFormat);
}
}
HRESULT __stdcall CMyDataObject::QueryGetData(FORMATETC *pFormatEtc)
{
return IsFormatSupported(pFormatEtc) ? S_OK : DV_E_FORMATETC;
}
STDMETHODIMP CStreamSwitcherPassThru::IsFormatSupported(const GUID* pFormat)
{
CallPeerSeeking(IsFormatSupported(pFormat));
}
STDMETHODIMP CAMRSplitter::IsUsingTimeFormat(const GUID* pFormat) {return IsFormatSupported(pFormat);}
STDMETHODIMP CAMRSplitter::SetTimeFormat(const GUID* pFormat) {return S_OK == IsFormatSupported(pFormat) ? S_OK : E_INVALIDARG;}
HRESULT CWASAPIRenderFilter::InitAudioClient()
{
Log("WASAPIRenderFilter::InitAudioClient");
HRESULT hr = S_OK;
if (m_pSettings->m_hnsPeriod == 0 || m_pSettings->m_hnsPeriod == 1)
{
REFERENCE_TIME defaultPeriod(0);
REFERENCE_TIME minimumPeriod(0);
hr = m_pAudioClient->GetDevicePeriod(&defaultPeriod, &minimumPeriod);
if (SUCCEEDED(hr))
{
if (m_pSettings->m_hnsPeriod == 0)
m_pSettings->m_hnsPeriod = defaultPeriod;
else
m_pSettings->m_hnsPeriod = minimumPeriod;
Log("WASAPIRenderFilter::InitAudioClient using device period from driver %I64u ms", m_pSettings->m_hnsPeriod / 10000);
}
else
{
Log("WASAPIRenderFilter::InitAudioClient failed to get device period from driver (0x%08x) - using 50 ms", hr);
m_pSettings->m_hnsPeriod = 500000; //50 ms is the best according to James @Slysoft
}
}
WAVEFORMATEXTENSIBLE* pwfxAccepted = NULL;
hr = IsFormatSupported(m_pInputFormat, &pwfxAccepted);
if (FAILED(hr))
{
SAFE_DELETE_WAVEFORMATEX(pwfxAccepted);
return hr;
}
GetBufferSize((WAVEFORMATEX*)pwfxAccepted, &m_pSettings->m_hnsPeriod);
if (SUCCEEDED(hr))
hr = m_pAudioClient->Initialize(m_pSettings->m_WASAPIShareMode, m_dwStreamFlags,
m_pSettings->m_hnsPeriod, m_pSettings->m_hnsPeriod, (WAVEFORMATEX*)pwfxAccepted, NULL);
if (FAILED(hr) && hr != AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED)
{
Log("WASAPIRenderFilter::InitAudioClient Initialize failed (0x%08x)", hr);
SAFE_DELETE_WAVEFORMATEX(pwfxAccepted);
return hr;
}
if (hr == S_OK)
{
SAFE_RELEASE(m_pAudioClock);
hr = m_pAudioClient->GetService(__uuidof(IAudioClock), (void**)&m_pAudioClock);
if (SUCCEEDED(hr))
m_pAudioClock->GetFrequency(&m_nHWfreq);
else
Log("WASAPIRenderFilter::IAudioClock not found!");
}
if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED)
{
// if the buffer size was not aligned, need to do the alignment dance
Log("WASAPIRenderFilter::InitAudioClient Buffer size not aligned. Realigning");
// get the buffer size, which will be aligned
hr = m_pAudioClient->GetBufferSize(&m_nFramesInBuffer);
// throw away this IAudioClient
SAFE_RELEASE(m_pAudioClient);
// calculate the new aligned periodicity
m_pSettings->m_hnsPeriod = // hns =
(REFERENCE_TIME)(
10000.0 * // (hns / ms) *
1000 * // (ms / s) *
m_nFramesInBuffer / // frames /
m_pInputFormat->Format.nSamplesPerSec // (frames / s)
+ 0.5 // rounding
);
if (SUCCEEDED(hr))
hr = CreateAudioClient();
Log("WASAPIRenderFilter::InitAudioClient Trying again with periodicity of %I64u hundred-nanoseconds, or %u frames", m_pSettings->m_hnsPeriod, m_nFramesInBuffer);
if (SUCCEEDED (hr))
hr = m_pAudioClient->Initialize(m_pSettings->m_WASAPIShareMode, m_dwStreamFlags,
m_pSettings->m_hnsPeriod, m_pSettings->m_hnsPeriod, (WAVEFORMATEX*)pwfxAccepted, NULL);
if (FAILED(hr))
{
Log("WASAPIRenderFilter::InitAudioClient Failed to reinitialize the audio client");
SAFE_DELETE_WAVEFORMATEX(pwfxAccepted);
return hr;
}
else
{
SAFE_RELEASE(m_pAudioClock);
hr = m_pAudioClient->GetService(__uuidof(IAudioClock), (void**)&m_pAudioClock);
if (FAILED(hr))
Log("WASAPIRenderFilter::IAudioClock not found!");
else
m_pAudioClock->GetFrequency(&m_nHWfreq);
}
} // if (AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED == hr)
// get the buffer size, which is aligned
if (SUCCEEDED(hr))
hr = m_pAudioClient->GetBufferSize(&m_nFramesInBuffer);
// calculate the new period
if (SUCCEEDED (hr))
hr = m_pAudioClient->GetService(__uuidof(IAudioRenderClient), (void**)(&m_pRenderClient));
if (FAILED(hr))
Log("WASAPIRenderFilter::InitAudioClient service initialization failed (0x%08x)", hr);
else
Log("WASAPIRenderer::InitAudioClient service initialization success");
if (m_pSettings->m_bWASAPIUseEventMode)
{
hr = m_pAudioClient->SetEventHandle(m_hDataEvent);
if (FAILED(hr))
{
Log("WASAPIRenderFilter::InitAudioClient SetEventHandle failed (0x%08x)", hr);
SAFE_DELETE_WAVEFORMATEX(pwfxAccepted);
return hr;
}
}
REFERENCE_TIME latency(0);
m_pAudioClient->GetStreamLatency(&latency);
Log("WASAPIRenderFilter::InitAudioClient device reported latency %I64u ms - buffer based latency %I64u ms",
latency / 10000, Latency() / 10000);
// Dynamic format change requires restart for the audio client
if (m_state != StateStopped)
StartAudioClient();
m_bDeviceInitialized = true;
SAFE_DELETE_WAVEFORMATEX(pwfxAccepted);
return hr;
}
STDMETHODIMP CStreamDriveThruFilter::SetTimeFormat(const GUID* pFormat)
{
return S_OK == IsFormatSupported(pFormat) ? S_OK : E_INVALIDARG;
}
STDMETHODIMP CBaseGraph::SetTimeFormat(const GUID* pFormat)
{
return S_OK == IsFormatSupported(pFormat) ? S_OK : E_INVALIDARG;
}
STDMETHODIMP CBaseMuxerFilter::IsUsingTimeFormat(const GUID* pFormat)
{
return IsFormatSupported(pFormat);
}
// ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
ALUTAPI ALvoid ALUTAPIENTRY alutLoadWAVFile(ALbyte *file,ALenum *format,ALvoid **data,ALsizei *size,ALsizei *freq)
{
OSStatus err = noErr;
AudioFileID audioFile = 0;
FSRef fsRef;
*data = NULL; // in case of failure, do not return some unitialized value as a bogus address
if (IsRelativePath(file))
{
char absolutePath[256];
// we need to make a full path here so FSPathMakeRef() works properly
MakeAbsolutePath(file, absolutePath, 256);
// create an fsref from the file parameter
err = FSPathMakeRef ((const UInt8 *) absolutePath, &fsRef, NULL);
}
else
err = FSPathMakeRef ((const UInt8 *) file, &fsRef, NULL);
if (err == noErr)
{
err = AudioFileOpen(&fsRef, fsRdPerm, 0, &audioFile);
if (err == noErr)
{
UInt32 dataSize;
CAStreamBasicDescription asbd;
dataSize = sizeof(CAStreamBasicDescription);
AudioFileGetProperty(audioFile, kAudioFilePropertyDataFormat, &dataSize, &asbd);
*format = GetOALFormatFromASBD(asbd);
if (IsFormatSupported(*format))
{
*freq = (UInt32) asbd.mSampleRate;
SInt64 audioDataSize = 0;
dataSize = sizeof(audioDataSize);
err = AudioFileGetProperty(audioFile, kAudioFilePropertyAudioDataByteCount, &dataSize, &audioDataSize);
if (err == noErr)
{
*size = audioDataSize;
*data = NULL;
*data = calloc(1, audioDataSize);
if (*data)
{
dataSize = audioDataSize;
err = AudioFileReadBytes(audioFile, false, 0, &dataSize, *data);
if ((asbd.mFormatID == kAudioFormatLinearPCM) && (asbd.mBitsPerChannel > 8))
{
// we just got 16 bit pcm data out of a WAVE file on a big endian platform, so endian swap the data
AudioConverterRef converter;
CAStreamBasicDescription outFormat = asbd;
void * tempData = NULL;
// ste format to big endian
outFormat.mFormatFlags = kAudioFormatFlagIsBigEndian | kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
// make some place for converted data
tempData = calloc(1 , audioDataSize);
err = AudioConverterNew(&asbd, &outFormat, &converter);
if ((err == noErr) && (tempData != NULL))
{
UInt32 bufferSize = audioDataSize;
err = AudioConverterConvertBuffer(converter, audioDataSize, *data, &bufferSize, tempData);
if (err == noErr)
memcpy(*data, tempData, audioDataSize);
AudioConverterDispose(converter);
}
if (tempData) free (tempData);
}
}
}
}
err = AudioFileClose(audioFile);
}
}
}
// ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
OSStatus OALBuffer::AddAudioData(char* inAudioData, UInt32 inAudioDataSize, ALenum format, ALsizei freq, bool inPreConvertToHalFormat)
{
#if LOG_VERBOSE
DebugMessageN6("OALBuffer::AddAudioData() - OALBuffer:inAudioData:inAudioDataSize:format:freq:inPreConvertToHalFormat = %ld:%p:%ld:%d:%d:%d", (long int) mSelfToken, inAudioData, (long int) inAudioDataSize, format, freq, inPreConvertToHalFormat);
#endif
// creates memory if needed
// reallocs if needed
// returns an error if buffer is in use
CAGuard::Locker bufferLock(mBufferLock);
try {
if (!IsFormatSupported(format))
throw ((OSStatus) AL_INVALID_VALUE); // this is not a valid buffer token or is an invalid format
#if USE_SOURCE_LIST_MUTEX
bool wasLocked = mSourceListGuard.Lock();
#endif
// don't allow if the buffer is in a queue
UInt32 attachedCount = mAttachedSourceList->Size();
#if USE_SOURCE_LIST_MUTEX
if (wasLocked) mSourceListGuard.Unlock();
#endif
if (attachedCount > 0)
{
DebugMessage("WAITING: AddAudioData ---> WaitOneRenderCycle");
// Let a render cycle go by and try again
WaitOneRenderCycle();
#if USE_SOURCE_LIST_MUTEX
wasLocked = mSourceListGuard.Lock();
#endif
attachedCount = mAttachedSourceList->Size();
#if USE_SOURCE_LIST_MUTEX
if (wasLocked) mSourceListGuard.Unlock();
#endif
if (attachedCount > 0){
DebugMessageN2("OALBuffer::AddAudioData: buffer ATTACHMENT > 0 - mSelfToken:mAttachedSourceList->Size() = %ld:%ld", (long int) mSelfToken, (long int) mAttachedSourceList->Size());
throw ((OSStatus) AL_INVALID_OPERATION);
}
}
if (mAppOwnsBufferMemory)
{
mData = NULL; // we were using the apps memory before so just initialize mData incase we fail
mAppOwnsBufferMemory = false;
}
mPreConvertedDataSize = (UInt32) inAudioDataSize;
// do not pre-convert stereo sounds, let the AC do the deinterleaving
OSStatus result = noErr;
if (!inPreConvertToHalFormat || ((format == AL_FORMAT_STEREO16) || (format == AL_FORMAT_STEREO8)))
{
if (mData != NULL)
{
if (mDataSize != (UInt32) inAudioDataSize)
{
mDataSize = (UInt32) inAudioDataSize;
void *newDataPtr = realloc(mData, mDataSize);
mData = (UInt8 *) newDataPtr;
}
}
else
{
mDataSize = (UInt32) inAudioDataSize;
mData = (UInt8 *) malloc (mDataSize);
}
if (mData)
{
result = FillInASBD(mDataFormat, format, freq);
THROW_RESULT
mPreConvertedDataFormat.SetFrom(mDataFormat); // make sure they are the same so original format info can be returned to caller
memcpy (mData, inAudioData, mDataSize);
}
}
else
{
#if LOG_EXTRAS
DebugMessage("alBufferData called: Converting Data Now");
#endif
result = ConvertDataForBuffer(inAudioData, inAudioDataSize, format, freq); // convert the data to the mixer's format and copy to the buffer
THROW_RESULT
}
}
catch (OSStatus result) {
DebugMessageN1("OALBuffer::AddAudioData Failed - err = %ld\n", (long int) result);
alSetError(result);
throw result;
}
catch (...) {
DebugMessage("OALBuffer::AddAudioData Failed");
alSetError(AL_INVALID_OPERATION);
throw -1;
}
return noErr;
}
// Format negotiation
HRESULT CWASAPIRenderFilter::NegotiateFormat(const WAVEFORMATEXTENSIBLE* pwfx, int nApplyChangesDepth, ChannelOrder* pChOrder)
{
if (!pwfx)
return VFW_E_TYPE_NOT_ACCEPTED;
if (FormatsEqual(pwfx, m_pInputFormat))
{
*pChOrder = m_chOrder;
return S_OK;
}
bool bApplyChanges = nApplyChangesDepth != 0;
bool bitDepthForced = (m_pSettings->m_nForceBitDepth != 0 && m_pSettings->m_nForceBitDepth != pwfx->Format.wBitsPerSample);
bool sampleRateForced = (m_pSettings->m_nForceSamplingRate != 0 && m_pSettings->m_nForceSamplingRate != pwfx->Format.nSamplesPerSec);
if ((bitDepthForced || sampleRateForced) &&
pwfx->SubFormat == KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL ||
pwfx->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT && bitDepthForced)
return VFW_E_TYPE_NOT_ACCEPTED;
if (((bitDepthForced && m_pSettings->m_nForceBitDepth != pwfx->Format.wBitsPerSample) ||
(sampleRateForced && m_pSettings->m_nForceSamplingRate != pwfx->Format.nSamplesPerSec)))
return VFW_E_TYPE_NOT_ACCEPTED;
CAutoLock lock(&m_csResources);
HRESULT hr = CreateAudioClient();
if (FAILED(hr))
{
Log("CWASAPIRenderFilter::NegotiateFormat Error, audio client not initialized: (0x%08x)", hr);
return VFW_E_CANNOT_CONNECT;
}
WAVEFORMATEXTENSIBLE* pwfxAccepted = NULL;
hr = IsFormatSupported(pwfx, &pwfxAccepted);
if (FAILED(hr))
{
SAFE_DELETE_WAVEFORMATEX(pwfxAccepted);
return hr;
}
if (bApplyChanges)
{
LogWaveFormat(pwfx, "REN - applying ");
// Stop and discard audio client
StopAudioClient();
SAFE_RELEASE(m_pRenderClient);
SAFE_RELEASE(m_pAudioClock);
SAFE_RELEASE(m_pAudioClient);
// We must use incoming format so the WAVEFORMATEXTENSIBLE to WAVEFORMATEXT difference
// that some audio drivers require is not causing an infonite loop of format changes
SetInputFormat(pwfx);
// Reinitialize audio client
hr = CreateAudioClient(true);
}
else
LogWaveFormat(pwfx, "Input format ");
m_chOrder = *pChOrder = DS_ORDER;
SAFE_DELETE_WAVEFORMATEX(pwfxAccepted);
return hr;
}
STDMETHODIMP CBaseGraph::IsUsingTimeFormat(const GUID* pFormat)
{
return IsFormatSupported(pFormat);
}
STDMETHODIMP StaticSourceFilter::IsUsingTimeFormat(const GUID * pFormat)
{
return IsFormatSupported(pFormat);
}
bool SoundBuffer::Create(AudioFormat format, unsigned int sampleCount, unsigned int sampleRate, const Int16* samples)
{
Destroy();
#if NAZARA_AUDIO_SAFE
if (!IsFormatSupported(format))
{
NazaraError("Audio format is not supported");
return false;
}
if (sampleCount == 0)
{
NazaraError("Sample rate must be different from zero");
return false;
}
if (sampleRate == 0)
{
NazaraError("Sample rate must be different from zero");
return false;
}
if (!samples)
{
NazaraError("Invalid sample source");
return false;
}
#endif
// On vide le stack d'erreurs
while (alGetError() != AL_NO_ERROR);
ALuint buffer;
alGenBuffers(1, &buffer);
if (alGetError() != AL_NO_ERROR)
{
NazaraError("Failed to create OpenAL buffer");
return false;
}
alBufferData(buffer, OpenAL::AudioFormat[format], samples, sampleCount*sizeof(Int16), sampleRate);
if (alGetError() != AL_NO_ERROR)
{
alDeleteBuffers(1, &buffer);
NazaraError("Failed to set OpenAL buffer");
return false;
}
m_impl = new SoundBufferImpl;
m_impl->buffer = buffer;
m_impl->duration = (1000*sampleCount / (format * sampleRate));
m_impl->format = format;
m_impl->sampleCount = sampleCount;
m_impl->sampleRate = sampleRate;
m_impl->samples.reset(new Int16[sampleCount]);
std::memcpy(&m_impl->samples[0], samples, sampleCount*sizeof(Int16));
return true;
}
STDMETHODIMP CStreamDriveThruFilter::IsUsingTimeFormat(const GUID* pFormat)
{
return IsFormatSupported(pFormat);
}