HRESULT CBaseVideoFilter::GetDeliveryBuffer(int w, int h, IMediaSample** ppOut)
{
CheckPointer(ppOut, E_POINTER);
HRESULT hr;
if (FAILED(hr = ReconnectOutput(w, h))) {
return hr;
}
if (FAILED(hr = m_pOutput->GetDeliveryBuffer(ppOut, NULL, NULL, 0))) {
return hr;
}
AM_MEDIA_TYPE* pmt;
if (SUCCEEDED((*ppOut)->GetMediaType(&pmt)) && pmt) {
CMediaType mt = *pmt;
m_pOutput->SetMediaType(&mt);
DeleteMediaType(pmt);
}
(*ppOut)->SetDiscontinuity(FALSE);
(*ppOut)->SetSyncPoint(TRUE);
// FIXME: hell knows why but without this the overlay mixer starts very skippy
// (don't enable this for other renderers, the old for example will go crazy if you do)
if (GetCLSID(m_pOutput->GetConnected()) == CLSID_OverlayMixer) {
(*ppOut)->SetDiscontinuity(TRUE);
}
return S_OK;
}
HRESULT CLAVAudio::DeliverBitstream(AVCodecID codec, const BYTE *buffer, DWORD dwSize, DWORD dwFrameSize, REFERENCE_TIME rtStartInput, REFERENCE_TIME rtStopInput)
{
HRESULT hr = S_OK;
CMediaType mt = CreateBitstreamMediaType(codec, m_bsParser.m_dwSampleRate);
WAVEFORMATEX* wfe = (WAVEFORMATEX*)mt.Format();
if(FAILED(hr = ReconnectOutput(dwSize, mt))) {
return hr;
}
IMediaSample *pOut;
BYTE *pDataOut = NULL;
if(FAILED(GetDeliveryBuffer(&pOut, &pDataOut))) {
return E_FAIL;
}
REFERENCE_TIME rtStart = m_rtStart, rtStop = AV_NOPTS_VALUE;
// TrueHD timings
// Since the SPDIF muxer takes 24 frames and puts them into one IEC61937 frame, we use the cached timestamp from before.
if (codec == AV_CODEC_ID_TRUEHD) {
// long-term cache is valid
if (m_rtBitstreamCache != AV_NOPTS_VALUE)
rtStart = m_rtBitstreamCache;
// Duration - stop time of the current frame is valid
if (rtStopInput != AV_NOPTS_VALUE)
rtStop = rtStopInput;
else // no actual time of the current frame, use typical TrueHD frame size, 24 * 0.83333ms
rtStop = rtStart + (REFERENCE_TIME)(200000 / m_dRate);
m_rtStart = rtStop;
} else {
double dDuration = DBL_SECOND_MULT * (double)m_bsParser.m_dwSamples / m_bsParser.m_dwSampleRate / m_dRate;
m_dStartOffset += fmod(dDuration, 1.0);
// Add rounded duration to rtStop
rtStop = rtStart + (REFERENCE_TIME)(dDuration + 0.5);
// and unrounded to m_rtStart..
m_rtStart += (REFERENCE_TIME)dDuration;
// and accumulate error..
if (m_dStartOffset > 0.5) {
m_rtStart++;
m_dStartOffset -= 1.0;
}
}
REFERENCE_TIME rtJitter = rtStart - m_rtBitstreamCache;
m_faJitter.Sample(rtJitter);
REFERENCE_TIME rtJitterMin = m_faJitter.AbsMinimum();
if (m_settings.AutoAVSync && abs(rtJitterMin) > m_JitterLimit && m_bHasVideo) {
DbgLog((LOG_TRACE, 10, L"::Deliver(): corrected A/V sync by %I64d", rtJitterMin));
m_rtStart -= rtJitterMin;
m_faJitter.OffsetValues(-rtJitterMin);
m_bDiscontinuity = TRUE;
}
#ifdef DEBUG
DbgLog((LOG_CUSTOM5, 20, L"Bitstream Delivery, rtStart(calc): %I64d, rtStart(input): %I64d, duration: %I64d, diff: %I64d", rtStart, m_rtBitstreamCache, rtStop-rtStart, rtJitter));
if (m_faJitter.CurrentSample() == 0) {
DbgLog((LOG_TRACE, 20, L"Jitter Stats: min: %I64d - max: %I64d - avg: %I64d", rtJitterMin, m_faJitter.AbsMaximum(), m_faJitter.Average()));
}
#endif
m_rtBitstreamCache = AV_NOPTS_VALUE;
if(m_settings.AudioDelayEnabled) {
REFERENCE_TIME rtDelay = (REFERENCE_TIME)((m_settings.AudioDelay * 10000i64) / m_dRate);
rtStart += rtDelay;
rtStop += rtDelay;
}
pOut->SetTime(&rtStart, &rtStop);
pOut->SetMediaTime(NULL, NULL);
pOut->SetPreroll(FALSE);
pOut->SetDiscontinuity(m_bDiscontinuity);
m_bDiscontinuity = FALSE;
pOut->SetSyncPoint(TRUE);
pOut->SetActualDataLength(dwSize);
memcpy(pDataOut, buffer, dwSize);
if(hr == S_OK) {
hr = m_pOutput->GetConnected()->QueryAccept(&mt);
if (hr == S_FALSE && m_nCodecId == AV_CODEC_ID_DTS && m_bDTSHD) {
DbgLog((LOG_TRACE, 1, L"DTS-HD Media Type failed with %0#.8x, trying fallback to DTS core", hr));
m_bForceDTSCore = TRUE;
UpdateBitstreamContext();
goto done;
}
DbgLog((LOG_TRACE, 1, L"Sending new Media Type (QueryAccept: %0#.8x)", hr));
m_pOutput->SetMediaType(&mt);
pOut->SetMediaType(&mt);
}
hr = m_pOutput->Deliver(pOut);
if (FAILED(hr)) {
DbgLog((LOG_ERROR, 10, L"::DeliverBitstream failed with code: %0#.8x", hr));
}
done:
SafeRelease(&pOut);
return hr;
}
HRESULT CAudioDecFilter::Transform(IMediaSample *pIn, IMediaSample *pOut)
{
// 入力データポインタを取得する
const DWORD InSize = pIn->GetActualDataLength();
BYTE *pInData = NULL;
HRESULT hr = pIn->GetPointer(&pInData);
if (FAILED(hr))
return hr;
{
CAutoLock Lock(&m_cPropLock);
/*
複数の音声フォーマットに対応する場合、この辺りでフォーマットの判定をする
*/
if (!m_pDecoder) {
m_pDecoder = new CAacDecoder();
m_pDecoder->Open();
}
REFERENCE_TIME rtStart, rtEnd;
hr = pIn->GetTime(&rtStart, &rtEnd);
if (FAILED(hr))
rtStart = -1;
if (pIn->IsDiscontinuity() == S_OK) {
m_bDiscontinuity = true;
m_bInputDiscontinuity = true;
} else if (hr == S_OK || hr == VFW_S_NO_STOP_TIME) {
if (!m_bJitterCorrection) {
m_StartTime = rtStart;
} else if (m_StartTime >= 0
&& _abs64(rtStart - m_StartTime) > MAX_JITTER) {
TRACE(TEXT("Resync audio stream time (%lld -> %lld [%f])\n"),
m_StartTime, rtStart,
(double)(rtStart - m_StartTime) / (double)REFERENCE_TIME_SECOND);
m_StartTime = rtStart;
}
}
if (m_StartTime < 0 || m_bDiscontinuity) {
TRACE(TEXT("Initialize audio stream time (%lld)\n"), rtStart);
m_StartTime = rtStart;
}
m_BitRateCalculator.Update(InSize);
}
DWORD InDataPos = 0;
FrameSampleInfo SampleInfo;
SampleInfo.pData = &m_OutData;
hr = S_OK;
while (InDataPos < InSize) {
{
CAutoLock Lock(&m_cPropLock);
CAudioDecoder::DecodeFrameInfo FrameInfo;
const DWORD DataSize = InSize - InDataPos;
DWORD DecodeSize = DataSize;
if (!m_pDecoder->Decode(&pInData[InDataPos], &DecodeSize, &FrameInfo)) {
if (DecodeSize < DataSize) {
InDataPos += DecodeSize;
continue;
}
break;
}
InDataPos += DecodeSize;
if (FrameInfo.bDiscontinuity)
m_bDiscontinuity = true;
SampleInfo.bMediaTypeChanged = false;
hr = OnFrame(FrameInfo.pData, FrameInfo.Samples, FrameInfo.Info,
&SampleInfo);
}
if (SUCCEEDED(hr)) {
if (SampleInfo.bMediaTypeChanged) {
hr = ReconnectOutput(SampleInfo.MediaBufferSize, SampleInfo.MediaType);
if (FAILED(hr))
break;
OutputLog(TEXT("出力メディアタイプを更新します。\r\n"));
hr = m_pOutput->SetMediaType(&SampleInfo.MediaType);
if (FAILED(hr)) {
OutputLog(TEXT("出力メディアタイプを設定できません。(%08x)\r\n"), hr);
break;
}
m_MediaType = SampleInfo.MediaType;
m_bDiscontinuity = true;
m_bInputDiscontinuity = true;
}
IMediaSample *pOutSample = NULL;
hr = m_pOutput->GetDeliveryBuffer(&pOutSample, NULL, NULL, 0);
if (FAILED(hr)) {
OutputLog(TEXT("出力メディアサンプルを取得できません。(%08x)\r\n"), hr);
break;
}
if (SampleInfo.bMediaTypeChanged)
pOutSample->SetMediaType(&m_MediaType);
// 出力ポインタ取得
BYTE *pOutBuff = NULL;
hr = pOutSample->GetPointer(&pOutBuff);
if (FAILED(hr)) {
OutputLog(TEXT("出力サンプルのバッファを取得できません。(%08x)\r\n"), hr);
pOutSample->Release();
break;
}
::CopyMemory(pOutBuff, m_OutData.GetData(), m_OutData.GetSize());
pOutSample->SetActualDataLength(m_OutData.GetSize());
if (m_StartTime >= 0) {
REFERENCE_TIME rtDuration, rtStart, rtEnd;
rtDuration = REFERENCE_TIME_SECOND * (LONGLONG)SampleInfo.Samples / FREQUENCY;
rtStart = m_StartTime;
m_StartTime += rtDuration;
// 音ずれ補正用時間シフト
if (m_DelayAdjustment > 0) {
// 最大2倍まで時間を遅らせる
if (rtDuration >= m_DelayAdjustment) {
rtDuration += m_DelayAdjustment;
m_DelayAdjustment = 0;
} else {
m_DelayAdjustment -= rtDuration;
rtDuration *= 2;
}
} else if (m_DelayAdjustment < 0) {
// 最短1/2まで時間を早める
if (rtDuration >= -m_DelayAdjustment * 2) {
rtDuration += m_DelayAdjustment;
m_DelayAdjustment = 0;
} else {
m_DelayAdjustment += rtDuration;
rtDuration /= 2;
}
} else {
rtStart += m_Delay;
}
rtEnd = rtStart + rtDuration;
pOutSample->SetTime(&rtStart, &rtEnd);
}
pOutSample->SetMediaTime(NULL, NULL);
pOutSample->SetPreroll(FALSE);
#if 0
// Discontinuityを設定すると倍速再生がおかしくなる模様
pOutSample->SetDiscontinuity(m_bDiscontinuity);
#else
pOutSample->SetDiscontinuity(m_bInputDiscontinuity);
#endif
m_bDiscontinuity = false;
m_bInputDiscontinuity = false;
pOutSample->SetSyncPoint(TRUE);
hr = m_pOutput->Deliver(pOutSample);
#ifdef _DEBUG
if (FAILED(hr)) {
OutputLog(TEXT("サンプルを送信できません。(%08x)\r\n"), hr);
if (m_bPassthrough && !m_bPassthroughError) {
m_bPassthroughError = true;
if (m_pEventHandler)
m_pEventHandler->OnSpdifPassthroughError(hr);
}
}
#endif
pOutSample->Release();
if (FAILED(hr))
break;
}
}
return hr;
}