Index: /trunk/src/VBox/Devices/Audio/DrvHostAudioDSound.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostAudioDSound.cpp (revision 88553) +++ /trunk/src/VBox/Devices/Audio/DrvHostAudioDSound.cpp (revision 88554) @@ -47,18 +47,10 @@ *********************************************************************************************************************************/ /* - * IDirectSound* interface uses HRESULT status codes and the driver callbacks use - * the VBox status codes. To minimize HRESULT->VBox conversion most internal functions - * in the driver return HRESULT and conversion is done in the driver callbacks. - * - * Naming convention: - * 'dsound*' functions return VBox status code; - * 'directSound*' - return HRESULT. - */ - -/* * Optional release logging, which a user can turn on with the * 'VBoxManage debugvm' command. * Debug logging still uses the common Log* macros from VBox. * Messages which always should go to the release log use LogRel. + * + * @deprecated Use LogRelMax, LogRel2 and LogRel3 directly. */ /** General code behavior. */ @@ -78,12 +70,10 @@ /** Maximum number of attempts to restore the sound buffer before giving up. */ #define DRV_DSOUND_RESTORE_ATTEMPTS_MAX 3 +#if 0 /** @todo r=bird: What are these for? Nobody is using them... */ /** Default input latency (in ms). */ #define DRV_DSOUND_DEFAULT_LATENCY_MS_IN 50 /** Default output latency (in ms). */ #define DRV_DSOUND_DEFAULT_LATENCY_MS_OUT 50 - -/** Makes DRVHOSTDSOUND out of PDMIHOSTAUDIO. */ -#define PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface) \ - ( (PDRVHOSTDSOUND)((uintptr_t)pInterface - RT_UOFFSETOF(DRVHOSTDSOUND, IHostAudio)) ) +#endif @@ -248,5 +238,4 @@ static int dsoundStreamEnable(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fEnable); -static void dsoundStreamReset(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS); static void dsoundUpdateStatusInternal(PDRVHOSTDSOUND pThis); @@ -407,72 +396,48 @@ /** - * Destroys a DirectSound playback interface. - * - * @param pDS Playback interface to destroy. - */ -static void directSoundPlayInterfaceDestroy(LPDIRECTSOUND8 pDS) -{ - if (pDS) - { - LogFlowFuncEnter(); - - IDirectSound8_Release(pDS); - pDS = NULL; - } -} - - -/** - * Creates a DirectSound playback interface. + * Creates a DirectSound playback instance. * * @return HRESULT - * @param pGUID GUID of device to create the playback interface for. - * @param ppDS Where to store the created interface. Optional. - */ -static HRESULT directSoundPlayInterfaceCreate(LPCGUID pGUID, LPDIRECTSOUND8 *ppDS) -{ - /* pGUID can be NULL, if this is the default device. */ - /* ppDS is optional. */ - + * @param pGUID GUID of device to create the playback interface for. NULL + * for the default device. + * @param ppDS Where to return the interface to the created instance. + */ +static HRESULT drvHostDSoundCreateDSPlaybackInstance(LPCGUID pGUID, LPDIRECTSOUND8 *ppDS) +{ LogFlowFuncEnter(); - LPDIRECTSOUND8 pDS; - HRESULT hr = CoCreateInstance(CLSID_DirectSound8, NULL, CLSCTX_ALL, - IID_IDirectSound8, (void **)&pDS); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Creating playback instance failed with %Rhrc\n", hr)); + LPDIRECTSOUND8 pDS = NULL; + HRESULT hrc = CoCreateInstance(CLSID_DirectSound8, NULL, CLSCTX_ALL, IID_IDirectSound8, (void **)&pDS); + if (SUCCEEDED(hrc)) + { + hrc = IDirectSound8_Initialize(pDS, pGUID); + if (SUCCEEDED(hrc)) + { + HWND hWnd = GetDesktopWindow(); + hrc = IDirectSound8_SetCooperativeLevel(pDS, hWnd, DSSCL_PRIORITY); + if (SUCCEEDED(hrc)) + { + *ppDS = pDS; + LogFlowFunc(("LEAVE S_OK\n")); + return S_OK; + } + LogRelMax(64, ("DSound: Setting cooperative level for (hWnd=%p) failed: %Rhrc\n", hWnd, hrc)); + } + else if (hrc == DSERR_NODRIVER) /* Usually means that no playback devices are attached. */ + LogRelMax(64, ("DSound: DirectSound playback is currently unavailable\n")); + else + LogRelMax(64, ("DSound: DirectSound playback initialization failed: %Rhrc\n", hrc)); + + IDirectSound8_Release(pDS); } else - { - hr = IDirectSound8_Initialize(pDS, pGUID); - if (SUCCEEDED(hr)) - { - HWND hWnd = GetDesktopWindow(); - hr = IDirectSound8_SetCooperativeLevel(pDS, hWnd, DSSCL_PRIORITY); - if (FAILED(hr)) - DSLOGREL(("DSound: Setting cooperative level for window %p failed with %Rhrc\n", hWnd, hr)); - } - - if (FAILED(hr)) - { - if (hr == DSERR_NODRIVER) /* Usually means that no playback devices are attached. */ - DSLOGREL(("DSound: DirectSound playback is currently unavailable\n")); - else - DSLOGREL(("DSound: DirectSound playback initialization failed with %Rhrc\n", hr)); - - directSoundPlayInterfaceDestroy(pDS); - } - else if (ppDS) - { - *ppDS = pDS; - } - } - - LogFlowFunc(("Returning %Rhrc\n", hr)); - return hr; -} - - + LogRelMax(64, ("DSound: Creating playback instance failed: %Rhrc\n", hrc)); + + LogFlowFunc(("LEAVE %Rhrc\n", hrc)); + return hrc; +} + + +#if 0 /* not used */ static HRESULT directSoundPlayGetStatus(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB, DWORD *pdwStatus) { @@ -508,4 +473,5 @@ return hr; } +#endif @@ -571,61 +537,37 @@ /** - * Destroys a DirectSound capture interface. + * Creates a DirectSound capture instance. * - * @param pDSC Capture interface to destroy. - */ -static void directSoundCaptureInterfaceDestroy(LPDIRECTSOUNDCAPTURE8 pDSC) -{ - if (pDSC) - { - LogFlowFuncEnter(); - + * @returns HRESULT + * @param pGUID GUID of device to create the capture interface for. NULL + * for default. + * @param ppDSC Where to return the interface to the created instance. + */ +static HRESULT drvHostDSoundCreateDSCaptureInstance(LPCGUID pGUID, LPDIRECTSOUNDCAPTURE8 *ppDSC) +{ + LogFlowFuncEnter(); + + LPDIRECTSOUNDCAPTURE8 pDSC = NULL; + HRESULT hrc = CoCreateInstance(CLSID_DirectSoundCapture8, NULL, CLSCTX_ALL, IID_IDirectSoundCapture8, (void **)&pDSC); + if (SUCCEEDED(hrc)) + { + hrc = IDirectSoundCapture_Initialize(pDSC, pGUID); + if (SUCCEEDED(hrc)) + { + *ppDSC = pDSC; + LogFlowFunc(("LEAVE S_OK\n")); + return S_OK; + } + if (hrc == DSERR_NODRIVER) /* Usually means that no capture devices are attached. */ + LogRelMax(64, ("DSound: Capture device currently is unavailable\n")); + else + LogRelMax(64, ("DSound: Initializing capturing device failed: %Rhrc\n", hrc)); IDirectSoundCapture_Release(pDSC); - pDSC = NULL; - } -} - - -/** - * Creates a DirectSound capture interface. - * - * @return HRESULT - * @param pGUID GUID of device to create the capture interface for. - * @param ppDSC Where to store the created interface. Optional. - */ -static HRESULT directSoundCaptureInterfaceCreate(LPCGUID pGUID, LPDIRECTSOUNDCAPTURE8 *ppDSC) -{ - /* pGUID can be NULL, if this is the default device. */ - /* ppDSC is optional. */ - - LogFlowFuncEnter(); - - LPDIRECTSOUNDCAPTURE8 pDSC; - HRESULT hr = CoCreateInstance(CLSID_DirectSoundCapture8, NULL, CLSCTX_ALL, - IID_IDirectSoundCapture8, (void **)&pDSC); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Creating capture instance failed with %Rhrc\n", hr)); } else - { - hr = IDirectSoundCapture_Initialize(pDSC, pGUID); - if (FAILED(hr)) - { - if (hr == DSERR_NODRIVER) /* Usually means that no capture devices are attached. */ - DSLOGREL(("DSound: Capture device currently is unavailable\n")); - else - DSLOGREL(("DSound: Initializing capturing device failed with %Rhrc\n", hr)); - - directSoundCaptureInterfaceDestroy(pDSC); - } - else if (ppDSC) - { - *ppDSC = pDSC; - } - } - - LogFlowFunc(("Returning %Rhrc\n", hr)); - return hr; + LogRelMax(64, ("DSound: Creating capture instance failed: %Rhrc\n", hrc)); + + LogFlowFunc(("LEAVE %Rhrc\n", hrc)); + return hrc; } @@ -832,5 +774,5 @@ { LPDIRECTSOUND8 pDS; - HRESULT hr = directSoundPlayInterfaceCreate(&pDev->Guid, &pDS); + HRESULT hr = drvHostDSoundCreateDSPlaybackInstance(&pDev->Guid, &pDS); if (SUCCEEDED(hr)) { @@ -879,5 +821,5 @@ } - directSoundPlayInterfaceDestroy(pDS); + IDirectSound8_Release(pDS); } else @@ -887,5 +829,5 @@ { LPDIRECTSOUNDCAPTURE8 pDSC; - HRESULT hr = directSoundCaptureInterfaceCreate(&pDev->Guid, &pDSC); + HRESULT hr = drvHostDSoundCreateDSCaptureInstance(&pDev->Guid, &pDSC); if (SUCCEEDED(hr)) { @@ -905,5 +847,5 @@ } - directSoundCaptureInterfaceDestroy(pDSC); + IDirectSoundCapture_Release(pDSC); } else @@ -1183,18 +1125,18 @@ static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvHostDSoundHA_GetStatus(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir) { - RT_NOREF(enmDir); - AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN); - + RT_NOREF(pInterface, enmDir); return PDMAUDIOBACKENDSTS_RUNNING; } -static int dsoundWaveFmtFromCfg(PPDMAUDIOSTREAMCFG pCfg, PWAVEFORMATEX pFmt) -{ - AssertPtrReturn(pCfg, VERR_INVALID_POINTER); - AssertPtrReturn(pFmt, VERR_INVALID_POINTER); - - RT_BZERO(pFmt, sizeof(WAVEFORMATEX)); - +/** + * Converts from PDM stream config to windows WAVEFORMATEX struct. + * + * @param pCfg The PDM audio stream config to convert from. + * @param pFmt The windows structure to initialize. + */ +static void dsoundWaveFmtFromCfg(PCPDMAUDIOSTREAMCFG pCfg, PWAVEFORMATEX pFmt) +{ + RT_ZERO(*pFmt); pFmt->wFormatTag = WAVE_FORMAT_PCM; pFmt->nChannels = PDMAudioPropsChannels(&pCfg->Props); @@ -1204,81 +1146,112 @@ pFmt->nAvgBytesPerSec = PDMAudioPropsFramesToBytes(&pCfg->Props, PDMAudioPropsHz(&pCfg->Props)); pFmt->cbSize = 0; /* No extra data specified. */ - - return VINF_SUCCESS; -} - - -/** - * Resets the state of a DirectSound stream. +} + + +/** + * Resets the state of a DirectSound stream, clearing the buffer content. * * @param pThis Host audio driver instance. * @param pStreamDS Stream to reset state for. */ -static void dsoundStreamReset(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) +static void drvHostDSoundStreamReset(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) { RT_NOREF(pThis); - - LogFunc(("Resetting %s\n", - pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN ? "capture" : "playback")); + LogFunc(("Resetting %s\n", pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN ? "capture" : "playback")); if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN) { + /* + * Input streams. + */ + LogFunc(("Resetting capture stream '%s'\n", pStreamDS->Cfg.szName)); + + /* Reset the state: */ +/** @todo r=bird: We set the read position to zero here, but shouldn't we query it + * from the buffer instead given that there isn't any interface for repositioning + * to the start of the buffer as with playback buffers? */ pStreamDS->In.offReadPos = 0; pStreamDS->In.cOverruns = 0; - /* Also reset the DirectSound Capture Buffer (DSCB) by clearing all data to make sure - * not stale audio data is left. */ + /* Clear the buffer content: */ + AssertPtr(pStreamDS->In.pDSCB); if (pStreamDS->In.pDSCB) { - PVOID pv1; PVOID pv2; DWORD cb1; DWORD cb2; - HRESULT hr = directSoundCaptureLock(pStreamDS, 0 /* Offset */, pStreamDS->cbBufSize, &pv1, &pv2, &cb1, &cb2, - 0 /* Flags */); - if (SUCCEEDED(hr)) + PVOID pv1 = NULL; + DWORD cb1 = 0; + PVOID pv2 = NULL; + DWORD cb2 = 0; + HRESULT hrc = IDirectSoundCaptureBuffer8_Lock(pStreamDS->In.pDSCB, 0, pStreamDS->cbBufSize, + &pv1, &cb1, &pv2, &cb2, 0 /*fFlags*/); + if (SUCCEEDED(hrc)) { PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv1, cb1, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb1)); if (pv2 && cb2) PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv2, cb2, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb2)); - directSoundCaptureUnlock(pStreamDS->In.pDSCB, pv1, pv2, cb1, cb2); + hrc = IDirectSoundCaptureBuffer8_Unlock(pStreamDS->In.pDSCB, pv1, cb1, pv2, cb2); + if (FAILED(hrc)) + LogRelMaxFunc(64, ("DSound: Unlocking capture buffer '%s' after reset failed: %Rhrc\n", + pStreamDS->Cfg.szName, hrc)); } - } - } - else if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT) - { - /* If draining was enagaged, make sure dsound has stopped playing. */ + else + LogRelMaxFunc(64, ("DSound: Locking capture buffer '%s' for reset failed: %Rhrc\n", + pStreamDS->Cfg.szName, hrc)); + } + } + else + { + /* + * Output streams. + */ + Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT); + LogFunc(("Resetting playback stream '%s'\n", pStreamDS->Cfg.szName)); + + /* If draining was enagaged, make sure dsound has stopped playing: */ if (pStreamDS->Out.fDrain && pStreamDS->Out.pDSB) pStreamDS->Out.pDSB->Stop(); - pStreamDS->Out.fFirstTransfer = true; - pStreamDS->Out.fDrain = false; - pStreamDS->Out.cUnderruns = 0; - - pStreamDS->Out.cbLastTransferred = 0; - pStreamDS->Out.tsLastTransferredMs = 0; - - pStreamDS->Out.cbTransferred = 0; - pStreamDS->Out.cbWritten = 0; - - pStreamDS->Out.offWritePos = 0; + /* Reset the internal state: */ + pStreamDS->Out.fFirstTransfer = true; + pStreamDS->Out.fDrain = false; + pStreamDS->Out.cUnderruns = 0; + pStreamDS->Out.cbLastTransferred = 0; + pStreamDS->Out.tsLastTransferredMs = 0; + pStreamDS->Out.cbTransferred = 0; + pStreamDS->Out.cbWritten = 0; + pStreamDS->Out.offWritePos = 0; pStreamDS->Out.offPlayCursorLastPending = 0; - pStreamDS->Out.offPlayCursorLastPlayed = 0; - - /* Also reset the DirectSound Buffer (DSB) by setting the position to 0 and clear all data to make sure - * not stale audio data is left. */ + pStreamDS->Out.offPlayCursorLastPlayed = 0; + + /* Reset the buffer content and repositioning the buffer to the start of the buffer. */ + AssertPtr(pStreamDS->Out.pDSB); if (pStreamDS->Out.pDSB) { - HRESULT hr = IDirectSoundBuffer8_SetCurrentPosition(pStreamDS->Out.pDSB, 0); - if (SUCCEEDED(hr)) + HRESULT hrc = IDirectSoundBuffer8_SetCurrentPosition(pStreamDS->Out.pDSB, 0); + if (FAILED(hrc)) + LogRelMaxFunc(64, ("DSound: Failed to set buffer position for '%s': %Rhrc\n", pStreamDS->Cfg.szName, hrc)); + + PVOID pv1 = NULL; + DWORD cb1 = 0; + PVOID pv2 = NULL; + DWORD cb2 = 0; + hrc = IDirectSoundBuffer8_Lock(pStreamDS->Out.pDSB, 0, pStreamDS->cbBufSize, &pv1, &cb1, &pv2, &cb2, 0 /*fFlags*/); + if (hrc == DSERR_BUFFERLOST) { - PVOID pv1; PVOID pv2; DWORD cb1; DWORD cb2; - hr = directSoundPlayLock(pThis, pStreamDS, 0 /* Offset */, pStreamDS->cbBufSize, &pv1, &pv2, &cb1, &cb2, - 0 /* Flags */); - if (SUCCEEDED(hr)) - { - PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv1, cb1, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb1)); - if (pv2 && cb2) - PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv2, cb2, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb2)); - directSoundPlayUnlock(pThis, pStreamDS->Out.pDSB, pv1, pv2, cb1, cb2); - } + directSoundPlayRestore(pThis, pStreamDS->Out.pDSB); + hrc = IDirectSoundBuffer8_Lock(pStreamDS->Out.pDSB, 0, pStreamDS->cbBufSize, &pv1, &cb1, &pv2, &cb2, 0 /*fFlags*/); } + if (SUCCEEDED(hrc)) + { + PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv1, cb1, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb1)); + if (pv2 && cb2) + PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv2, cb2, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb2)); + + hrc = IDirectSoundBuffer8_Unlock(pStreamDS->Out.pDSB, pv1, cb1, pv2, cb2); + if (FAILED(hrc)) + LogRelMaxFunc(64, ("DSound: Unlocking playback buffer '%s' after reset failed: %Rhrc\n", + pStreamDS->Cfg.szName, hrc)); + } + else + LogRelMaxFunc(64, ("DSound: Locking playback buffer '%s' for reset failed: %Rhrc\n", pStreamDS->Cfg.szName, hrc)); } } @@ -1290,4 +1263,380 @@ +/** + * Worker for drvHostDSoundHA_StreamCreate that creates caputre stream. + * + * @returns Windows COM status code. + * @param pThis The DSound instance data. + * @param pStreamDS The stream instance data. + * @param pCfgReq The requested stream config (input). + * @param pCfgAcq Where to return the actual stream config. This is a + * copy of @a *pCfgReq when called. + * @param pWaveFmtX On input the requested stream format. + * Updated to the actual stream format on successful + * return. + */ +static HRESULT drvHostDSoundStreamCreateCapture(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, + PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq, WAVEFORMATEX *pWaveFmtX) +{ + Assert(pStreamDS->In.pDSCB == NULL); + HRESULT hrc; + + /* + * Create, initialize and set up a IDirectSoundCapture instance the first time + * we go thru here. + */ + /** @todo bird: Or should we rather just throw this away after we've gotten the + * capture buffer? Old code would just leak it... */ + if (pThis->pDSC == NULL) + { + hrc = drvHostDSoundCreateDSCaptureInstance(pThis->Cfg.pGuidCapture, &pThis->pDSC); + if (FAILED(hrc)) + return hrc; /* The worker has complained to the release log already. */ + } + + /* + * Create the capture buffer. + */ + DSCBUFFERDESC BufferDesc = + { + /*.dwSize = */ sizeof(BufferDesc), + /*.dwFlags = */ 0, + /*.dwBufferBytes =*/ PDMAudioPropsFramesToBytes(&pCfgReq->Props, pCfgReq->Backend.cFramesBufferSize), + /*.dwReserved = */ 0, + /*.lpwfxFormat = */ pWaveFmtX, + /*.dwFXCount = */ 0, + /*.lpDSCFXDesc = */ NULL + }; + + LogRel2(("DSound: Requested capture buffer is %#x B / %u B / %RU64 ms\n", BufferDesc.dwBufferBytes, BufferDesc.dwBufferBytes, + PDMAudioPropsBytesToMilli(&pCfgReq->Props, BufferDesc.dwBufferBytes))); + + LPDIRECTSOUNDCAPTUREBUFFER pLegacyDSCB = NULL; + hrc = IDirectSoundCapture_CreateCaptureBuffer(pThis->pDSC, &BufferDesc, &pLegacyDSCB, NULL); + if (FAILED(hrc)) + { + LogRelMax(64, ("DSound: Creating capture buffer for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc)); + return hrc; + } + + /* Get the IDirectSoundCaptureBuffer8 version of the interface. */ + hrc = IDirectSoundCaptureBuffer_QueryInterface(pLegacyDSCB, IID_IDirectSoundCaptureBuffer8, (void **)&pStreamDS->In.pDSCB); + IDirectSoundCaptureBuffer_Release(pLegacyDSCB); + if (FAILED(hrc)) + { + LogRelMax(64, ("DSound: Querying IID_IDirectSoundCaptureBuffer8 for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc)); + return hrc; + } + + /* + * Query the actual stream configuration. + */ +#if 0 /** @todo r=bird: WTF was this for? */ + DWORD offByteReadPos = 0; + hrc = IDirectSoundCaptureBuffer8_GetCurrentPosition(pStreamDS->In.pDSCB, NULL, &offByteReadPos); + if (FAILED(hrc)) + { + offByteReadPos = 0; + DSLOGREL(("DSound: Getting capture position failed with %Rhrc\n", hr)); + } +#endif + RT_ZERO(*pWaveFmtX); + hrc = IDirectSoundCaptureBuffer8_GetFormat(pStreamDS->In.pDSCB, pWaveFmtX, sizeof(*pWaveFmtX), NULL); + if (SUCCEEDED(hrc)) + { + /** @todo r=bird: We aren't converting/checking the pWaveFmtX content... */ + + DSCBCAPS BufferCaps = { /*.dwSize = */ sizeof(BufferCaps), 0, 0, 0 }; + hrc = IDirectSoundCaptureBuffer8_GetCaps(pStreamDS->In.pDSCB, &BufferCaps); + if (SUCCEEDED(hrc)) + { + LogRel2(("DSound: Acquired capture buffer capabilities for '%s':\n" + "DSound: dwFlags = %#RX32\n" + "DSound: dwBufferBytes = %#RX32 B / %RU32 B / %RU64 ms\n" + "DSound: dwReserved = %#RX32\n", + pCfgReq->szName, BufferCaps.dwFlags, BufferCaps.dwBufferBytes, BufferCaps.dwBufferBytes, + PDMAudioPropsBytesToMilli(&pCfgReq->Props, BufferCaps.dwBufferBytes), BufferCaps.dwReserved )); + + /* Update buffer related stuff: */ + pStreamDS->In.offReadPos = 0; /** @todo shouldn't we use offBytReadPos here to "read at the initial capture position"? */ + pStreamDS->cbBufSize = BufferCaps.dwBufferBytes; + pCfgAcq->Backend.cFramesBufferSize = PDMAudioPropsBytesToFrames(&pCfgAcq->Props, BufferCaps.dwBufferBytes); + + /** @todo r=bird: WTF is this for? */ + pThis->pDSStrmIn = pStreamDS; + +#if 0 /** @todo r=bird: uAlign isn't set anywhere, so this hasn't been checking anything for a while... */ + if (bc.dwBufferBytes & pStreamDS->uAlign) + DSLOGREL(("DSound: Capture GetCaps returned misaligned buffer: size %RU32, alignment %RU32\n", + bc.dwBufferBytes, pStreamDS->uAlign + 1)); +#endif + LogFlow(("returns S_OK\n")); + return S_OK; + } + LogRelMax(64, ("DSound: Getting capture buffer capabilities for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc)); + } + else + LogRelMax(64, ("DSound: Getting capture format for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc)); + + IDirectSoundCaptureBuffer8_Release(pStreamDS->In.pDSCB); + pStreamDS->In.pDSCB = NULL; + LogFlowFunc(("returns %Rhrc\n", hrc)); + return hrc; +} + + +/** + * Worker for drvHostDSoundHA_StreamCreate that creates playback stream. + * + * @returns Windows COM status code. + * @param pThis The DSound instance data. + * @param pStreamDS The stream instance data. + * @param pCfgReq The requested stream config (input). + * @param pCfgAcq Where to return the actual stream config. This is a + * copy of @a *pCfgReq when called. + * @param pWaveFmtX On input the requested stream format. + * Updated to the actual stream format on successful + * return. + */ +static HRESULT drvHostDSoundStreamCreatePlayback(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, + PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq, WAVEFORMATEX *pWaveFmtX) +{ + Assert(pStreamDS->Out.pDSB == NULL); + HRESULT hrc; + + /* + * Create, initialize and set up a DirectSound8 instance the first time + * we go thru here. + */ + /** @todo bird: Or should we rather just throw this away after we've gotten the + * sound buffer? Old code would just leak it... */ + if (pThis->pDS == NULL) + { + hrc = drvHostDSoundCreateDSPlaybackInstance(pThis->Cfg.pGuidPlay, &pThis->pDS); + if (FAILED(hrc)) + return hrc; /* The worker has complained to the release log already. */ + } + + /* + * As we reuse our (secondary) buffer for playing out data as it comes in, + * we're using this buffer as a so-called streaming buffer. + * + * See https://msdn.microsoft.com/en-us/library/windows/desktop/ee419014(v=vs.85).aspx + * + * However, as we do not want to use memory on the sound device directly + * (as most modern audio hardware on the host doesn't have this anyway), + * we're *not* going to use DSBCAPS_STATIC for that. + * + * Instead we're specifying DSBCAPS_LOCSOFTWARE, as this fits the bill + * of copying own buffer data to our secondary's Direct Sound buffer. + */ + DSBUFFERDESC BufferDesc = + { + /*.dwSize = */ sizeof(BufferDesc), + /*.dwFlags = */ DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_LOCSOFTWARE, + /*.dwBufferBytes = */ PDMAudioPropsFramesToBytes(&pCfgReq->Props, pCfgReq->Backend.cFramesBufferSize), + /*.dwReserved = */ 0, + /*.lpwfxFormat = */ pWaveFmtX + /*.guid3DAlgorithm = {0, 0, 0, {0,0,0,0, 0,0,0,0}} */ + }; + LogRel2(("DSound: Requested playback buffer is %#x B / %u B / %RU64 ms\n", BufferDesc.dwBufferBytes, BufferDesc.dwBufferBytes, + PDMAudioPropsBytesToMilli(&pCfgReq->Props, BufferDesc.dwBufferBytes))); + + LPDIRECTSOUNDBUFFER pLegacyDSB = NULL; + hrc = IDirectSound8_CreateSoundBuffer(pThis->pDS, &BufferDesc, &pLegacyDSB, NULL); + if (FAILED(hrc)) + { + LogRelMax(64, ("DSound: Creating playback sound buffer for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc)); + return hrc; + } + + /* Get the IDirectSoundBuffer8 version of the interface. */ + hrc = IDirectSoundBuffer_QueryInterface(pLegacyDSB, IID_IDirectSoundBuffer8, (PVOID *)&pStreamDS->Out.pDSB); + IDirectSoundBuffer_Release(pLegacyDSB); + if (FAILED(hrc)) + { + LogRelMax(64, ("DSound: Querying IID_IDirectSoundBuffer8 for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc)); + return hrc; + } + + /* + * Query the actual stream parameters, they may differ from what we requested. + */ + RT_ZERO(*pWaveFmtX); + hrc = IDirectSoundBuffer8_GetFormat(pStreamDS->Out.pDSB, pWaveFmtX, sizeof(*pWaveFmtX), NULL); + if (SUCCEEDED(hrc)) + { + /** @todo r=bird: We aren't converting/checking the pWaveFmtX content... */ + + DSBCAPS BufferCaps = { /*.dwSize = */ sizeof(BufferCaps), 0, 0, 0, 0 }; + hrc = IDirectSoundBuffer8_GetCaps(pStreamDS->Out.pDSB, &BufferCaps); + if (SUCCEEDED(hrc)) + { + LogRel2(("DSound: Acquired playback buffer capabilities for '%s':\n" + "DSound: dwFlags = %#RX32\n" + "DSound: dwBufferBytes = %#RX32 B / %RU32 B / %RU64 ms\n" + "DSound: dwUnlockTransferRate = %RU32 KB/s\n" + "DSound: dwPlayCpuOverhead = %RU32%%\n", + pCfgReq->szName, BufferCaps.dwFlags, BufferCaps.dwBufferBytes, BufferCaps.dwBufferBytes, + PDMAudioPropsBytesToMilli(&pCfgReq->Props, BufferCaps.dwBufferBytes), + BufferCaps.dwUnlockTransferRate, BufferCaps.dwPlayCpuOverhead)); + + /* Update buffer related stuff: */ + pStreamDS->cbBufSize = BufferCaps.dwBufferBytes; + pCfgAcq->Backend.cFramesBufferSize = PDMAudioPropsBytesToFrames(&pCfgAcq->Props, BufferCaps.dwBufferBytes); + pCfgAcq->Backend.cFramesPeriod = pCfgAcq->Backend.cFramesBufferSize / 4; /* total fiction */ + pCfgAcq->Backend.cFramesPreBuffering = pCfgReq->Backend.cFramesPreBuffering * pCfgAcq->Backend.cFramesBufferSize + / RT_MAX(pCfgReq->Backend.cFramesBufferSize, 1); + + /** @todo r=bird: WTF is this for? */ + pThis->pDSStrmOut = pStreamDS; + +#if 0 /** @todo r=bird: uAlign isn't set anywhere, so this hasn't been checking anything for a while... */ + if (bc.dwBufferBytes & pStreamDS->uAlign) + DSLOGREL(("DSound: Playback capabilities returned misaligned buffer: size %RU32, alignment %RU32\n", + bc.dwBufferBytes, pStreamDS->uAlign + 1)); +#endif + LogFlow(("returns S_OK\n")); + return S_OK; + } + LogRelMax(64, ("DSound: Getting playback buffer capabilities for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc)); + } + else + LogRelMax(64, ("DSound: Getting playback format for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc)); + + IDirectSoundBuffer8_Release(pStreamDS->Out.pDSB); + pStreamDS->Out.pDSB = NULL; + LogFlowFunc(("returns %Rhrc\n", hrc)); + return hrc; +} + + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCreate} + */ +static DECLCALLBACK(int) drvHostDSoundHA_StreamCreate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, + PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) +{ + PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); + PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + AssertPtrReturn(pStreamDS, VERR_INVALID_POINTER); + AssertPtrReturn(pCfgReq, VERR_INVALID_POINTER); + AssertPtrReturn(pCfgAcq, VERR_INVALID_POINTER); + AssertReturn(pCfgReq->enmDir == PDMAUDIODIR_IN || pCfgReq->enmDir == PDMAUDIODIR_OUT, VERR_INVALID_PARAMETER); + Assert(PDMAudioStrmCfgEquals(pCfgReq, pCfgAcq)); + + const char * const pszStreamType = pCfgReq->enmDir == PDMAUDIODIR_IN ? "capture" : "playback"; RT_NOREF(pszStreamType); + LogFlowFunc(("enmSrc/Dst=%s '%s'\n", + pCfgReq->enmDir == PDMAUDIODIR_IN ? PDMAudioRecSrcGetName(pCfgReq->u.enmSrc) + : PDMAudioPlaybackDstGetName(pCfgReq->u.enmDst), pCfgReq->szName)); + + /* For whatever reason: */ + dsoundUpdateStatusInternal(pThis); + + /* + * DSound has different COM interfaces for working with input and output + * streams, so we'll quickly part ways here after some common format + * specification setup and logging. + */ +#if defined(RTLOG_REL_ENABLED) || defined(LOG_ENABLED) + char szTmp[64]; +#endif + LogRel2(("DSound: Opening %s stream '%s' (%s)\n", pCfgReq->szName, pszStreamType, + PDMAudioPropsToString(&pCfgReq->Props, szTmp, sizeof(szTmp)))); + + WAVEFORMATEX WaveFmtX; + dsoundWaveFmtFromCfg(pCfgReq, &WaveFmtX); + LogRel2(("DSound: Requested %s format for '%s':\n" + "DSound: wFormatTag = %RU16\n" + "DSound: nChannels = %RU16\n" + "DSound: nSamplesPerSec = %RU32\n" + "DSound: nAvgBytesPerSec = %RU32\n" + "DSound: nBlockAlign = %RU16\n" + "DSound: wBitsPerSample = %RU16\n" + "DSound: cbSize = %RU16\n", + pszStreamType, pCfgReq->szName, WaveFmtX.wFormatTag, WaveFmtX.nChannels, WaveFmtX.nSamplesPerSec, + WaveFmtX.nAvgBytesPerSec, WaveFmtX.nBlockAlign, WaveFmtX.wBitsPerSample, WaveFmtX.cbSize)); + + HRESULT hrc; + if (pCfgReq->enmDir == PDMAUDIODIR_IN) + hrc = drvHostDSoundStreamCreateCapture(pThis, pStreamDS, pCfgReq, pCfgAcq, &WaveFmtX); + else + hrc = drvHostDSoundStreamCreatePlayback(pThis, pStreamDS, pCfgReq, pCfgAcq, &WaveFmtX); + int rc; + if (SUCCEEDED(hrc)) + { + LogRel2(("DSound: Acquired %s format for '%s':\n" + "DSound: wFormatTag = %RU16\n" + "DSound: nChannels = %RU16\n" + "DSound: nSamplesPerSec = %RU32\n" + "DSound: nAvgBytesPerSec = %RU32\n" + "DSound: nBlockAlign = %RU16\n" + "DSound: wBitsPerSample = %RU16\n" + "DSound: cbSize = %RU16\n", + pszStreamType, pCfgReq->szName, WaveFmtX.wFormatTag, WaveFmtX.nChannels, WaveFmtX.nSamplesPerSec, + WaveFmtX.nAvgBytesPerSec, WaveFmtX.nBlockAlign, WaveFmtX.wBitsPerSample, WaveFmtX.cbSize)); + + /* + * Copy the acquired config and reset the stream (clears the buffer). + */ + PDMAudioStrmCfgCopy(&pStreamDS->Cfg, pCfgAcq); + drvHostDSoundStreamReset(pThis, pStreamDS); + rc = VINF_SUCCESS; + } + else + rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE; + + LogFlowFunc(("returns %Rrc\n", rc)); + return rc; +} + + +static HRESULT directSoundPlayClose(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) +{ + AssertPtrReturn(pThis, E_POINTER); + AssertPtrReturn(pStreamDS, E_POINTER); + + LogFlowFuncEnter(); + + HRESULT hr = directSoundPlayStop(pThis, pStreamDS, true /* fFlush */); + if (SUCCEEDED(hr)) + { + DSLOG(("DSound: Closing playback stream\n")); + RTCritSectEnter(&pThis->CritSect); + + if (pStreamDS->Out.pDSB) + { + IDirectSoundBuffer8_Release(pStreamDS->Out.pDSB); + pStreamDS->Out.pDSB = NULL; + } + + pThis->pDSStrmOut = NULL; + + RTCritSectLeave(&pThis->CritSect); + } + + if (FAILED(hr)) + DSLOGREL(("DSound: Stopping playback stream %p failed with %Rhrc\n", pStreamDS, hr)); + + return hr; +} + + +static int dsoundDestroyStreamOut(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) +{ + LogFlowFuncEnter(); + + HRESULT hr = directSoundPlayStop(pThis, pStreamDS, true /* fFlush */); + if (SUCCEEDED(hr)) + { + hr = directSoundPlayClose(pThis, pStreamDS); + if (FAILED(hr)) + return VERR_GENERAL_FAILURE; /** @todo Fix. */ + } + + return VINF_SUCCESS; +} + + static HRESULT directSoundCaptureClose(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) { @@ -1315,482 +1664,22 @@ -static HRESULT directSoundCaptureOpen(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, - PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) -{ - AssertPtrReturn(pThis, E_POINTER); - AssertPtrReturn(pStreamDS, E_POINTER); - AssertPtrReturn(pCfgReq, E_POINTER); - AssertPtrReturn(pCfgAcq, E_POINTER); - - /** @todo r=bird: I cannot see any code populating pCfgAcq... */ - +static int dsoundDestroyStreamIn(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) +{ LogFlowFuncEnter(); - Assert(pStreamDS->In.pDSCB == NULL); - - DSLOG(("DSound: Opening capturing stream (uHz=%RU32, cChannels=%RU8, cBits=%u, fSigned=%RTbool)\n", pCfgReq->Props.uHz, - PDMAudioPropsChannels(&pCfgReq->Props), PDMAudioPropsSampleBits(&pCfgReq->Props), pCfgReq->Props.fSigned)); - - WAVEFORMATEX wfx; - int rc = dsoundWaveFmtFromCfg(pCfgReq, &wfx); - if (RT_FAILURE(rc)) - return E_INVALIDARG; - - /** @todo r=bird: Why is this called every time? It triggers a device - * enumeration. Andy claimed on IRC that enumeration was only done once... */ - dsoundUpdateStatusInternal(pThis); - - HRESULT hr = directSoundCaptureInterfaceCreate(pThis->Cfg.pGuidCapture, &pThis->pDSC); - if (FAILED(hr)) - return hr; - - do /* For readability breaks... */ - { - DSCBUFFERDESC bd; - RT_ZERO(bd); - - bd.dwSize = sizeof(bd); - bd.lpwfxFormat = &wfx; - bd.dwBufferBytes = PDMAudioPropsFramesToBytes(&pCfgReq->Props, pCfgReq->Backend.cFramesBufferSize); - - DSLOG(("DSound: Requested capture buffer is %RU64ms (%ld bytes)\n", - PDMAudioPropsBytesToMilli(&pCfgReq->Props, bd.dwBufferBytes), bd.dwBufferBytes)); - - LPDIRECTSOUNDCAPTUREBUFFER pDSCB; - hr = IDirectSoundCapture_CreateCaptureBuffer(pThis->pDSC, &bd, &pDSCB, NULL); - if (FAILED(hr)) - { - if (hr == E_ACCESSDENIED) - { - DSLOGREL(("DSound: Capturing input from host not possible, access denied\n")); - } - else - DSLOGREL(("DSound: Creating capture buffer failed with %Rhrc\n", hr)); - break; - } - - hr = IDirectSoundCaptureBuffer_QueryInterface(pDSCB, IID_IDirectSoundCaptureBuffer8, (void **)&pStreamDS->In.pDSCB); - IDirectSoundCaptureBuffer_Release(pDSCB); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Querying interface for capture buffer failed with %Rhrc\n", hr)); - break; - } - - /* - * Query the actual parameters. - */ - DWORD offByteReadPos = 0; - hr = IDirectSoundCaptureBuffer8_GetCurrentPosition(pStreamDS->In.pDSCB, NULL, &offByteReadPos); - if (FAILED(hr)) - { - offByteReadPos = 0; - DSLOGREL(("DSound: Getting capture position failed with %Rhrc\n", hr)); - } - - RT_ZERO(wfx); - hr = IDirectSoundCaptureBuffer8_GetFormat(pStreamDS->In.pDSCB, &wfx, sizeof(wfx), NULL); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Getting capture format failed with %Rhrc\n", hr)); - break; - } - - DSCBCAPS bc; - RT_ZERO(bc); - bc.dwSize = sizeof(bc); - hr = IDirectSoundCaptureBuffer8_GetCaps(pStreamDS->In.pDSCB, &bc); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Getting capture capabilities failed with %Rhrc\n", hr)); - break; - } - - DSLOG(("DSound: Acquired capture buffer is %RU64ms (%ld bytes)\n", - PDMAudioPropsBytesToMilli(&pCfgReq->Props, bc.dwBufferBytes), bc.dwBufferBytes)); - - DSLOG(("DSound: Capture format:\n" - " dwBufferBytes = %RI32\n" - " dwFlags = 0x%x\n" - " wFormatTag = %RI16\n" - " nChannels = %RI16\n" - " nSamplesPerSec = %RU32\n" - " nAvgBytesPerSec = %RU32\n" - " nBlockAlign = %RI16\n" - " wBitsPerSample = %RI16\n" - " cbSize = %RI16\n", - bc.dwBufferBytes, - bc.dwFlags, - wfx.wFormatTag, - wfx.nChannels, - wfx.nSamplesPerSec, - wfx.nAvgBytesPerSec, - wfx.nBlockAlign, - wfx.wBitsPerSample, - wfx.cbSize)); - - if (bc.dwBufferBytes & pStreamDS->uAlign) - DSLOGREL(("DSound: Capture GetCaps returned misaligned buffer: size %RU32, alignment %RU32\n", - bc.dwBufferBytes, pStreamDS->uAlign + 1)); - - /* Initial state: reading at the initial capture position, no error. */ - pStreamDS->In.offReadPos = 0; - pStreamDS->cbBufSize = bc.dwBufferBytes; - - pThis->pDSStrmIn = pStreamDS; - - pCfgAcq->Backend.cFramesBufferSize = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, pStreamDS->cbBufSize); - - } while (0); - - if (FAILED(hr)) - directSoundCaptureClose(pThis, pStreamDS); - - LogFlowFunc(("Returning %Rhrc\n", hr)); - return hr; -} - - -static HRESULT directSoundPlayClose(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - AssertPtrReturn(pThis, E_POINTER); - AssertPtrReturn(pStreamDS, E_POINTER); - - LogFlowFuncEnter(); - - HRESULT hr = directSoundPlayStop(pThis, pStreamDS, true /* fFlush */); - if (SUCCEEDED(hr)) - { - DSLOG(("DSound: Closing playback stream\n")); - RTCritSectEnter(&pThis->CritSect); - - if (pStreamDS->Out.pDSB) - { - IDirectSoundBuffer8_Release(pStreamDS->Out.pDSB); - pStreamDS->Out.pDSB = NULL; - } - - pThis->pDSStrmOut = NULL; - - RTCritSectLeave(&pThis->CritSect); - } - - if (FAILED(hr)) - DSLOGREL(("DSound: Stopping playback stream %p failed with %Rhrc\n", pStreamDS, hr)); - - return hr; -} - - -static HRESULT directSoundPlayOpen(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, - PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) -{ - AssertPtrReturn(pThis, E_POINTER); - AssertPtrReturn(pStreamDS, E_POINTER); - AssertPtrReturn(pCfgReq, E_POINTER); - AssertPtrReturn(pCfgAcq, E_POINTER); - LogFlowFuncEnter(); - - Assert(pStreamDS->Out.pDSB == NULL); - - DSLOG(("DSound: Opening playback stream (uHz=%RU32, cChannels=%RU8, cBits=%u, fSigned=%RTbool)\n", pCfgReq->Props.uHz, - PDMAudioPropsChannels(&pCfgReq->Props), PDMAudioPropsSampleBits(&pCfgReq->Props), pCfgReq->Props.fSigned)); - - WAVEFORMATEX wfx; - int rc = dsoundWaveFmtFromCfg(pCfgReq, &wfx); - if (RT_FAILURE(rc)) - return E_INVALIDARG; - - DSLOG(("DSound: Requested playback format:\n" - " wFormatTag = %RI16\n" - " nChannels = %RI16\n" - " nSamplesPerSec = %RU32\n" - " nAvgBytesPerSec = %RU32\n" - " nBlockAlign = %RI16\n" - " wBitsPerSample = %RI16\n" - " cbSize = %RI16\n", - wfx.wFormatTag, - wfx.nChannels, - wfx.nSamplesPerSec, - wfx.nAvgBytesPerSec, - wfx.nBlockAlign, - wfx.wBitsPerSample, - wfx.cbSize)); - - /* Waste some time: */ - dsoundUpdateStatusInternal(pThis); - - HRESULT hr = directSoundPlayInterfaceCreate(pThis->Cfg.pGuidPlay, &pThis->pDS); - if (FAILED(hr)) - return hr; - - do /* To use breaks. */ - { - LPDIRECTSOUNDBUFFER pDSB = NULL; - - DSBUFFERDESC bd; - RT_ZERO(bd); - bd.dwSize = sizeof(bd); - bd.lpwfxFormat = &wfx; - - /* - * As we reuse our (secondary) buffer for playing out data as it comes in, - * we're using this buffer as a so-called streaming buffer. - * - * See https://msdn.microsoft.com/en-us/library/windows/desktop/ee419014(v=vs.85).aspx - * - * However, as we do not want to use memory on the sound device directly - * (as most modern audio hardware on the host doesn't have this anyway), - * we're *not* going to use DSBCAPS_STATIC for that. - * - * Instead we're specifying DSBCAPS_LOCSOFTWARE, as this fits the bill - * of copying own buffer data to our secondary's Direct Sound buffer. - */ - bd.dwFlags = DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_LOCSOFTWARE; - bd.dwBufferBytes = PDMAudioPropsFramesToBytes(&pCfgReq->Props, pCfgReq->Backend.cFramesBufferSize); - - DSLOG(("DSound: Requested playback buffer is %RU64ms (%ld bytes)\n", - PDMAudioPropsBytesToMilli(&pCfgReq->Props, bd.dwBufferBytes), bd.dwBufferBytes)); - - hr = IDirectSound8_CreateSoundBuffer(pThis->pDS, &bd, &pDSB, NULL); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Creating playback sound buffer failed with %Rhrc\n", hr)); - break; - } - - /* "Upgrade" to IDirectSoundBuffer8 interface. */ - hr = IDirectSoundBuffer_QueryInterface(pDSB, IID_IDirectSoundBuffer8, (PVOID *)&pStreamDS->Out.pDSB); - IDirectSoundBuffer_Release(pDSB); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Querying playback sound buffer interface failed with %Rhrc\n", hr)); - break; - } - - /* - * Query the actual parameters set for this stream. - * Those might be different than the initially requested parameters. - */ - RT_ZERO(wfx); - hr = IDirectSoundBuffer8_GetFormat(pStreamDS->Out.pDSB, &wfx, sizeof(wfx), NULL); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Getting playback format failed with %Rhrc\n", hr)); - break; - } - - DSBCAPS bc; - RT_ZERO(bc); - bc.dwSize = sizeof(bc); - - hr = IDirectSoundBuffer8_GetCaps(pStreamDS->Out.pDSB, &bc); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Getting playback capabilities failed with %Rhrc\n", hr)); - break; - } - - DSLOG(("DSound: Acquired playback buffer is %RU64ms (%ld bytes)\n", - PDMAudioPropsBytesToMilli(&pCfgReq->Props, bc.dwBufferBytes), bc.dwBufferBytes)); - - DSLOG(("DSound: Acquired playback format:\n" - " dwBufferBytes = %RI32\n" - " dwFlags = 0x%x\n" - " wFormatTag = %RI16\n" - " nChannels = %RI16\n" - " nSamplesPerSec = %RU32\n" - " nAvgBytesPerSec = %RU32\n" - " nBlockAlign = %RI16\n" - " wBitsPerSample = %RI16\n" - " cbSize = %RI16\n", - bc.dwBufferBytes, - bc.dwFlags, - wfx.wFormatTag, - wfx.nChannels, - wfx.nSamplesPerSec, - wfx.nAvgBytesPerSec, - wfx.nBlockAlign, - wfx.wBitsPerSample, - wfx.cbSize)); - - if (bc.dwBufferBytes & pStreamDS->uAlign) - DSLOGREL(("DSound: Playback capabilities returned misaligned buffer: size %RU32, alignment %RU32\n", - bc.dwBufferBytes, pStreamDS->uAlign + 1)); - - /* - * Initial state. - * dsoundPlayStart initializes part of it to make sure that Stop/Start continues with a correct - * playback buffer position. - */ - pStreamDS->cbBufSize = bc.dwBufferBytes; - - pThis->pDSStrmOut = pStreamDS; - - const uint32_t cfBufSize = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, pStreamDS->cbBufSize); - - pCfgAcq->Backend.cFramesBufferSize = cfBufSize; - pCfgAcq->Backend.cFramesPeriod = cfBufSize / 4; - pCfgAcq->Backend.cFramesPreBuffering = pCfgAcq->Backend.cFramesPeriod * 2; - - } while (0); - - if (FAILED(hr)) - directSoundPlayClose(pThis, pStreamDS); - - return hr; -} - - -static void dsoundPlayClearBuffer(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - AssertPtrReturnVoid(pStreamDS); - - PPDMAUDIOPCMPROPS pProps = &pStreamDS->Cfg.Props; - - HRESULT hr = IDirectSoundBuffer_SetCurrentPosition(pStreamDS->Out.pDSB, 0 /* Position */); - if (FAILED(hr)) - DSLOGREL(("DSound: Setting current position to 0 when clearing buffer failed with %Rhrc\n", hr)); - - PVOID pv1; - hr = directSoundPlayLock(pThis, pStreamDS, - 0 /* dwOffset */, pStreamDS->cbBufSize, - &pv1, NULL, 0, 0, DSBLOCK_ENTIREBUFFER); - if (SUCCEEDED(hr)) - { - PDMAudioPropsClearBuffer(pProps, pv1, pStreamDS->cbBufSize, PDMAUDIOPCMPROPS_B2F(pProps, pStreamDS->cbBufSize)); - - directSoundPlayUnlock(pThis, pStreamDS->Out.pDSB, pv1, NULL, 0, 0); - - /* Make sure to get the last playback position and current write position from DirectSound again. - * Those positions in theory could have changed, re-fetch them to be sure. */ - DWORD offPlay = 0; - DWORD offWrite = 0; - hr = IDirectSoundBuffer_GetCurrentPosition(pStreamDS->Out.pDSB, &offPlay, &offWrite); - if (SUCCEEDED(hr)) - { - pStreamDS->Out.offPlayCursorLastPlayed = offPlay; - pStreamDS->Out.offWritePos = offWrite; - } - else - DSLOGREL(("DSound: Re-fetching current position when clearing buffer failed with %Rhrc\n", hr)); - } -} - - -static int dsoundCreateStreamOut(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, - PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) -{ - LogFlowFunc(("pStreamDS=%p, pCfgReq=%p\n", pStreamDS, pCfgReq)); - - int rc; - - /* Try to open playback in case the device is already there. */ - HRESULT hr = directSoundPlayOpen(pThis, pStreamDS, pCfgReq, pCfgAcq); - if (SUCCEEDED(hr)) - { - rc = PDMAudioStrmCfgCopy(&pStreamDS->Cfg, pCfgAcq); - if (RT_SUCCESS(rc)) - dsoundStreamReset(pThis, pStreamDS); - } - else - rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE; - - LogFlowFuncLeaveRC(rc); - return rc; -} - - -static int dsoundCreateStreamIn(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, - PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) -{ - LogFunc(("pStreamDS=%p, pCfgReq=%p, enmRecSource=%s\n", - pStreamDS, pCfgReq, PDMAudioRecSrcGetName(pCfgReq->u.enmSrc))); - - - /* Try to open capture in case the device is already there. */ - int rc; - HRESULT hr = directSoundCaptureOpen(pThis, pStreamDS, pCfgReq, pCfgAcq); - if (SUCCEEDED(hr)) - { - rc = PDMAudioStrmCfgCopy(&pStreamDS->Cfg, pCfgAcq); - if (RT_SUCCESS(rc)) - dsoundStreamReset(pThis, pStreamDS); - } - else - rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE; - - return rc; -} - - -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCreate} - */ -static DECLCALLBACK(int) drvHostDSoundHA_StreamCreate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, - PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pStream, VERR_INVALID_POINTER); - AssertPtrReturn(pCfgReq, VERR_INVALID_POINTER); - AssertPtrReturn(pCfgAcq, VERR_INVALID_POINTER); - - PDRVHOSTDSOUND pThis = PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface); + directSoundCaptureClose(pThis, pStreamDS); + + return VINF_SUCCESS; +} + + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDestroy} + */ +static DECLCALLBACK(int) drvHostDSoundHA_StreamDestroy(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) +{ + PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; - - int rc; - if (pCfgReq->enmDir == PDMAUDIODIR_IN) - rc = dsoundCreateStreamIn(pThis, pStreamDS, pCfgReq, pCfgAcq); - else - rc = dsoundCreateStreamOut(pThis, pStreamDS, pCfgReq, pCfgAcq); - - if (RT_SUCCESS(rc)) - { - /** @todo already copied */ - rc = PDMAudioStrmCfgCopy(&pStreamDS->Cfg, pCfgAcq); - if (RT_SUCCESS(rc)) - rc = RTCritSectInit(&pStreamDS->CritSect); - } - - return rc; -} - - -static int dsoundDestroyStreamOut(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - LogFlowFuncEnter(); - - HRESULT hr = directSoundPlayStop(pThis, pStreamDS, true /* fFlush */); - if (SUCCEEDED(hr)) - { - hr = directSoundPlayClose(pThis, pStreamDS); - if (FAILED(hr)) - return VERR_GENERAL_FAILURE; /** @todo Fix. */ - } - - return VINF_SUCCESS; -} - - -static int dsoundDestroyStreamIn(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - LogFlowFuncEnter(); - - directSoundCaptureClose(pThis, pStreamDS); - - return VINF_SUCCESS; -} - - -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDestroy} - */ -static DECLCALLBACK(int) drvHostDSoundHA_StreamDestroy(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pStream, VERR_INVALID_POINTER); - - PDRVHOSTDSOUND pThis = PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface); - PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + AssertPtrReturn(pStreamDS, VERR_INVALID_POINTER); int rc; @@ -1827,5 +1716,5 @@ if (fEnable) - dsoundStreamReset(pThis, pStreamDS); + drvHostDSoundStreamReset(pThis, pStreamDS); pStreamDS->fEnabled = fEnable; @@ -1885,5 +1774,5 @@ { if (fFlush) - dsoundStreamReset(pThis, pStreamDS); + drvHostDSoundStreamReset(pThis, pStreamDS); } @@ -2046,5 +1935,5 @@ { if (fFlush) - dsoundStreamReset(pThis, pStreamDS); + drvHostDSoundStreamReset(pThis, pStreamDS); } @@ -2077,11 +1966,12 @@ if (SUCCEEDED(hr)) { +/** @todo r=bird: This isn't working. CfgAcq isn't initialized! */ PDMAUDIOSTREAMCFG CfgAcq; - hr = directSoundCaptureOpen(pThis, pStreamDS, &pStreamDS->Cfg /* pCfgReq */, &CfgAcq); + WAVEFORMATEX WaveFmtX; + dsoundWaveFmtFromCfg(&pStreamDS->Cfg, &WaveFmtX); + hr = drvHostDSoundStreamCreateCapture(pThis, pStreamDS, &pStreamDS->Cfg, &CfgAcq, &WaveFmtX); if (SUCCEEDED(hr)) { - rc = PDMAudioStrmCfgCopy(&pStreamDS->Cfg, &CfgAcq); - if (RT_FAILURE(rc)) - break; + PDMAudioStrmCfgCopy(&pStreamDS->Cfg, &CfgAcq); /** @todo What to do if the format has changed? */ @@ -2128,9 +2018,7 @@ PPDMAUDIOBACKENDSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd) { - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pStream, VERR_INVALID_POINTER); - - PDRVHOSTDSOUND pThis = PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface); + PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + AssertPtrReturn(pStreamDS, VERR_INVALID_POINTER); int rc;