Index: /trunk/src/VBox/Devices/Audio/DrvHostAudioDSound.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostAudioDSound.cpp (revision 88513) +++ /trunk/src/VBox/Devices/Audio/DrvHostAudioDSound.cpp (revision 88514) @@ -48,9 +48,9 @@ /* * IDirectSound* interface uses HRESULT status codes and the driver callbacks use - * the IPRT status codes. To minimize HRESULT->IPRT conversion most internal functions + * 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 IPRT status code; + * 'dsound*' functions return VBox status code; * 'directSound*' - return HRESULT. */ @@ -59,5 +59,5 @@ * Optional release logging, which a user can turn on with the * 'VBoxManage debugvm' command. - * Debug logging still uses the common Log* macros from IPRT. + * Debug logging still uses the common Log* macros from VBox. * Messages which always should go to the release log use LogRel. */ @@ -260,105 +260,4 @@ } -static int dsoundWaveFmtFromCfg(PPDMAUDIOSTREAMCFG pCfg, PWAVEFORMATEX pFmt) -{ - AssertPtrReturn(pCfg, VERR_INVALID_POINTER); - AssertPtrReturn(pFmt, VERR_INVALID_POINTER); - - RT_BZERO(pFmt, sizeof(WAVEFORMATEX)); - - pFmt->wFormatTag = WAVE_FORMAT_PCM; - pFmt->nChannels = PDMAudioPropsChannels(&pCfg->Props); - pFmt->wBitsPerSample = PDMAudioPropsSampleBits(&pCfg->Props); - pFmt->nSamplesPerSec = PDMAudioPropsHz(&pCfg->Props); - pFmt->nBlockAlign = PDMAudioPropsFrameSize(&pCfg->Props); - pFmt->nAvgBytesPerSec = PDMAudioPropsFramesToBytes(&pCfg->Props, PDMAudioPropsHz(&pCfg->Props)); - pFmt->cbSize = 0; /* No extra data specified. */ - - return VINF_SUCCESS; -} - -/** - * Retrieves the number of free bytes available for writing to a DirectSound output stream. - * - * @return VBox status code. VERR_NOT_AVAILABLE if unable to determine or the - * buffer was not recoverable. - * @param pThis Host audio driver instance. - * @param pStreamDS DirectSound output stream to retrieve number for. - * @param pdwFree Where to return the free amount on success. - * @param poffPlayCursor Where to return the play cursor offset. - */ -static int dsoundGetFreeOut(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, DWORD *pdwFree, DWORD *poffPlayCursor) -{ - AssertPtrReturn(pThis, VERR_INVALID_POINTER); - AssertPtrReturn(pStreamDS, VERR_INVALID_POINTER); - AssertPtrReturn(pdwFree, VERR_INVALID_POINTER); - AssertPtr(poffPlayCursor); - - Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT); /* Paranoia. */ - - LPDIRECTSOUNDBUFFER8 pDSB = pStreamDS->Out.pDSB; - if (!pDSB) - { - AssertPtr(pDSB); - return VERR_INVALID_POINTER; - } - - HRESULT hr = S_OK; - - /* Get the current play position which is used for calculating the free space in the buffer. */ - for (unsigned i = 0; i < DRV_DSOUND_RESTORE_ATTEMPTS_MAX; i++) - { - DWORD offPlayCursor = 0; - DWORD offWriteCursor = 0; - hr = IDirectSoundBuffer8_GetCurrentPosition(pDSB, &offPlayCursor, &offWriteCursor); - if (SUCCEEDED(hr)) - { - int32_t cbDiff = offWriteCursor - offPlayCursor; - if (cbDiff < 0) - cbDiff += pStreamDS->cbBufSize; - - int32_t cbFree = offPlayCursor - pStreamDS->Out.offWritePos; - if (cbFree < 0) - cbFree += pStreamDS->cbBufSize; - - if (cbFree > (int32_t)pStreamDS->cbBufSize - cbDiff) - { - /** @todo count/log these. */ - pStreamDS->Out.offWritePos = offWriteCursor; - cbFree = pStreamDS->cbBufSize - cbDiff; - } - - /* When starting to use a DirectSound buffer, offPlayCursor and offWriteCursor - * both point at position 0, so we won't be able to detect how many bytes - * are writable that way. - * - * So use our per-stream written indicator to see if we just started a stream. */ - if (pStreamDS->Out.cbWritten == 0) - cbFree = pStreamDS->cbBufSize; - - DSLOGREL(("DSound: offPlayCursor=%RU32, offWriteCursor=%RU32, offWritePos=%RU32 -> cbFree=%RI32\n", - offPlayCursor, offWriteCursor, pStreamDS->Out.offWritePos, cbFree)); - - *pdwFree = cbFree; - *poffPlayCursor = offPlayCursor; - return VINF_SUCCESS; - } - - if (hr != DSERR_BUFFERLOST) /** @todo MSDN doesn't state this error for GetCurrentPosition(). */ - break; - - LogFunc(("Getting playing position failed due to lost buffer, restoring ...\n")); - - directSoundPlayRestore(pThis, pDSB); - } - - if (hr != DSERR_BUFFERLOST) /* Avoid log flooding if the error is still there. */ - DSLOGREL(("DSound: Getting current playback position failed with %Rhrc\n", hr)); - - LogFunc(("Failed with %Rhrc\n", hr)); - - *poffPlayCursor = pStreamDS->cbBufSize; - return VERR_NOT_AVAILABLE; -} static char *dsoundGUIDToUtf8StrA(LPCGUID pGUID) @@ -380,4 +279,5 @@ return RTStrDup("{Default device}"); } + static HRESULT directSoundPlayRestore(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB) @@ -393,4 +293,5 @@ } + static HRESULT directSoundPlayUnlock(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB, PVOID pv1, PVOID pv2, @@ -404,13 +305,4 @@ } -static HRESULT directSoundCaptureUnlock(LPDIRECTSOUNDCAPTUREBUFFER8 pDSCB, - PVOID pv1, PVOID pv2, - DWORD cb1, DWORD cb2) -{ - HRESULT hr = IDirectSoundCaptureBuffer8_Unlock(pDSCB, pv1, cb1, pv2, cb2); - if (FAILED(hr)) - DSLOGREL(("DSound: Unlocking capture buffer failed with %Rhrc\n", hr)); - return hr; -} static HRESULT directSoundPlayLock(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, @@ -461,4 +353,16 @@ } + +static HRESULT directSoundCaptureUnlock(LPDIRECTSOUNDCAPTUREBUFFER8 pDSCB, + PVOID pv1, PVOID pv2, + DWORD cb1, DWORD cb2) +{ + HRESULT hr = IDirectSoundCaptureBuffer8_Unlock(pDSCB, pv1, cb1, pv2, cb2); + if (FAILED(hr)) + DSLOGREL(("DSound: Unlocking capture buffer failed with %Rhrc\n", hr)); + return hr; +} + + static HRESULT directSoundCaptureLock(PDSOUNDSTREAM pStreamDS, DWORD dwOffset, DWORD dwBytes, @@ -497,4 +401,5 @@ } + /* * DirectSound playback @@ -516,4 +421,5 @@ } } + /** @@ -568,233 +474,4 @@ } -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); - -/** @todo r=bird: I cannot see any code populating pCfgAcq... */ - - 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)); - - /** @todo r=bird: Why is this called every time? It triggers a device - * enumeration. Andy claimed on IRC that enumeration was only done once... - * It's generally a 'ing waste of time here too, as we dont really use any of - * the information we gather there. */ - 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 HRESULT directSoundPlayGetStatus(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB, DWORD *pdwStatus) @@ -828,168 +505,4 @@ else DSLOGREL(("DSound: Retrieving playback status failed with %Rhrc\n", hr)); - - return hr; -} - -static HRESULT directSoundPlayStop(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fFlush) -{ - AssertPtrReturn(pThis, E_POINTER); - AssertPtrReturn(pStreamDS, E_POINTER); - - HRESULT hr = S_OK; - - if (pStreamDS->Out.pDSB) - { - DSLOG(("DSound: Stopping playback\n")); - hr = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB); - if (FAILED(hr)) - { - hr = directSoundPlayRestore(pThis, pStreamDS->Out.pDSB); - if (FAILED(hr)) /** @todo shouldn't this be a SUCCEEDED? */ - hr = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB); - } - } - - if (SUCCEEDED(hr)) - { - if (fFlush) - dsoundStreamReset(pThis, pStreamDS); - } - - if (FAILED(hr)) - DSLOGREL(("DSound: %s playback failed with %Rhrc\n", fFlush ? "Stopping" : "Pausing", hr)); - - return hr; -} - -/** - * Enables or disables a stream. - * - * @return IPRT status code. - * @param pThis Host audio driver instance. - * @param pStreamDS Stream to enable / disable. - * @param fEnable Whether to enable or disable the stream. - */ -static int dsoundStreamEnable(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fEnable) -{ - RT_NOREF(pThis); - - LogFunc(("%s %s\n", - fEnable ? "Enabling" : "Disabling", - pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN ? "capture" : "playback")); - - if (fEnable) - dsoundStreamReset(pThis, pStreamDS); - - pStreamDS->fEnabled = fEnable; - - return VINF_SUCCESS; -} - - -/** - * Resets the state of a DirectSound stream. - * - * @param pThis Host audio driver instance. - * @param pStreamDS Stream to reset state for. - */ -static void dsoundStreamReset(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - RT_NOREF(pThis); - - LogFunc(("Resetting %s\n", - pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN ? "capture" : "playback")); - - if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN) - { - 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. */ - 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)) - { - 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); - } - } - } - else if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT) - { - /* 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; - 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. */ - if (pStreamDS->Out.pDSB) - { - HRESULT hr = IDirectSoundBuffer8_SetCurrentPosition(pStreamDS->Out.pDSB, 0); - if (SUCCEEDED(hr)) - { - 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); - } - } - } - } - -#ifdef LOG_ENABLED - pStreamDS->Dbg.tsLastTransferredMs = 0; -#endif -} - - -/** - * Starts playing a DirectSound stream. - * - * @return HRESULT - * @param pThis Host audio driver instance. - * @param pStreamDS Stream to start playing. - */ -static HRESULT directSoundPlayStart(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - HRESULT hr = S_OK; - - DWORD fFlags = DSCBSTART_LOOPING; - - for (unsigned i = 0; i < DRV_DSOUND_RESTORE_ATTEMPTS_MAX; i++) - { - DSLOG(("DSound: Starting playback\n")); - hr = IDirectSoundBuffer8_Play(pStreamDS->Out.pDSB, 0, 0, fFlags); - if ( SUCCEEDED(hr) - || hr != DSERR_BUFFERLOST) - break; - LogFunc(("Restarting playback failed due to lost buffer, restoring ...\n")); - directSoundPlayRestore(pThis, pStreamDS->Out.pDSB); - } return hr; @@ -1056,4 +569,5 @@ #endif + /** * Destroys a DirectSound capture interface. @@ -1071,4 +585,5 @@ } } + /** @@ -1115,228 +630,72 @@ } -static HRESULT directSoundCaptureClose(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - AssertPtrReturn(pThis, E_POINTER); - AssertPtrReturn(pStreamDS, E_POINTER); - + +/** + * Updates this host driver's internal status, according to the global, overall input/output + * state and all connected (native) audio streams. + * + * @todo r=bird: This is a 'ing waste of 'ing time! We're doing this everytime + * an 'ing stream is created and we doesn't 'ing use the information here + * for any darn thing! Given the reported slowness of enumeration and + * issues with the 'ing code the only appropriate response is: + * AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAARG!!!!!!! + * + * @param pThis Host audio driver instance. + */ +static void dsoundUpdateStatusInternal(PDRVHOSTDSOUND pThis) +{ +#if 0 /** @todo r=bird: This isn't doing *ANYTHING* useful. So, I've just disabled it. */ + AssertPtrReturnVoid(pThis); LogFlowFuncEnter(); - HRESULT hr = directSoundCaptureStop(pThis, pStreamDS, true /* fFlush */); - if (FAILED(hr)) - return hr; - - if ( pStreamDS - && pStreamDS->In.pDSCB) - { - DSLOG(("DSound: Closing capturing stream\n")); - - IDirectSoundCaptureBuffer8_Release(pStreamDS->In.pDSCB); - pStreamDS->In.pDSCB = NULL; - } - - LogFlowFunc(("Returning %Rhrc\n", hr)); - return hr; -} - -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... */ - - 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 directSoundCaptureStop(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fFlush) -{ - AssertPtrReturn(pThis, E_POINTER); - AssertPtrReturn(pStreamDS, E_POINTER); - + PDMAudioHostEnumDelete(&pThis->DeviceEnum); + int rc = dsoundDevicesEnumerate(pThis, &pThis->DeviceEnum); + if (RT_SUCCESS(rc)) + { +#if 0 + if ( pThis->fEnabledOut != RT_BOOL(cbCtx.cDevOut) + || pThis->fEnabledIn != RT_BOOL(cbCtx.cDevIn)) + { + /** @todo Use a registered callback to the audio connector (e.g "OnConfigurationChanged") to + * let the connector know that something has changed within the host backend. */ + } +#endif + pThis->fEnabledIn = PDMAudioHostEnumCountMatching(&pThis->DeviceEnum, PDMAUDIODIR_IN) != 0; + pThis->fEnabledOut = PDMAudioHostEnumCountMatching(&pThis->DeviceEnum, PDMAUDIODIR_OUT) != 0; + } + + LogFlowFuncLeaveRC(rc); +#else RT_NOREF(pThis); - - HRESULT hr = S_OK; - - if (pStreamDS->In.pDSCB) - { - DSLOG(("DSound: Stopping capture\n")); - hr = IDirectSoundCaptureBuffer_Stop(pStreamDS->In.pDSCB); - } - - if (SUCCEEDED(hr)) - { - if (fFlush) - dsoundStreamReset(pThis, pStreamDS); - } - - if (FAILED(hr)) - DSLOGREL(("DSound: Stopping capture buffer failed with %Rhrc\n", hr)); - - return hr; -} - -static HRESULT directSoundCaptureStart(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - AssertPtrReturn(pThis, E_POINTER); - AssertPtrReturn(pStreamDS, E_POINTER); - - HRESULT hr; - if (pStreamDS->In.pDSCB) - { - DWORD dwStatus; - hr = IDirectSoundCaptureBuffer8_GetStatus(pStreamDS->In.pDSCB, &dwStatus); - if (FAILED(hr)) - { - DSLOGREL(("DSound: Retrieving capture status failed with %Rhrc\n", hr)); - } - else - { - if (dwStatus & DSCBSTATUS_CAPTURING) - { - DSLOG(("DSound: Already capturing\n")); - } - else - { - const DWORD fFlags = DSCBSTART_LOOPING; - - DSLOG(("DSound: Starting to capture\n")); - hr = IDirectSoundCaptureBuffer8_Start(pStreamDS->In.pDSCB, fFlags); - if (FAILED(hr)) - DSLOGREL(("DSound: Starting to capture failed with %Rhrc\n", hr)); - } - } - } - else - hr = E_UNEXPECTED; - - LogFlowFunc(("Returning %Rhrc\n", hr)); - return hr; -} +#endif +} + + +/********************************************************************************************************************************* +* PDMIHOSTAUDIO * +*********************************************************************************************************************************/ + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnGetConfig} + */ +static DECLCALLBACK(int) drvHostDSoundHA_GetConfig(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pBackendCfg) +{ + RT_NOREF(pInterface); + AssertPtrReturn(pInterface, VERR_INVALID_POINTER); + AssertPtrReturn(pBackendCfg, VERR_INVALID_POINTER); + + RT_BZERO(pBackendCfg, sizeof(PPDMAUDIOBACKENDCFG)); + + pBackendCfg->cbStreamOut = sizeof(DSOUNDSTREAM); + pBackendCfg->cbStreamIn = sizeof(DSOUNDSTREAM); + + RTStrPrintf2(pBackendCfg->szName, sizeof(pBackendCfg->szName), "DirectSound"); + + pBackendCfg->cMaxStreamsIn = UINT32_MAX; + pBackendCfg->cMaxStreamsOut = UINT32_MAX; + + return VINF_SUCCESS; +} + /** @@ -1399,4 +758,5 @@ } + /** * Callback for the capture device enumeration. @@ -1455,6 +815,7 @@ } -/** - * Qqueries information for a given (DirectSound) device. + +/** + * Queries information for a given (DirectSound) device. * * @returns VBox status code. @@ -1657,8 +1018,9 @@ } + /** * Does a (Re-)enumeration of the host's playback + capturing devices. * - * @return IPRT status code. + * @return VBox status code. * @param pThis Host audio driver instance. * @param pDevEnm Where to store the enumerated devices. @@ -1794,43 +1156,538 @@ } -/** - * Updates this host driver's internal status, according to the global, overall input/output - * state and all connected (native) audio streams. - * - * @todo r=bird: This is a 'ing waste of 'ing time! We're doing this everytime - * an 'ing stream is created and we doesn't 'ing use the information here - * for any darn thing! Given the reported slowness of enumeration and - * issues with eh 'ing code the only appropriate response is: - * AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAARG!!!!!!! + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnGetDevices} + */ +static DECLCALLBACK(int) drvHostDSoundHA_GetDevices(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHOSTENUM pDeviceEnum) +{ + AssertPtrReturn(pInterface, VERR_INVALID_POINTER); + AssertPtrReturn(pDeviceEnum, VERR_INVALID_POINTER); + + PDRVHOSTDSOUND pThis = PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface); + + int rc = RTCritSectEnter(&pThis->CritSect); + if (RT_SUCCESS(rc)) + { + PDMAudioHostEnumInit(pDeviceEnum); + rc = dsoundDevicesEnumerate(pThis, pDeviceEnum); + if (RT_FAILURE(rc)) + PDMAudioHostEnumDelete(pDeviceEnum); + + int rc2 = RTCritSectLeave(&pThis->CritSect); + AssertRC(rc2); + } + + LogFlowFunc(("Returning %Rrc\n", rc)); + return rc; +} + + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnGetStatus} + */ +static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvHostDSoundHA_GetStatus(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir) +{ + RT_NOREF(enmDir); + AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN); + + 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)); + + pFmt->wFormatTag = WAVE_FORMAT_PCM; + pFmt->nChannels = PDMAudioPropsChannels(&pCfg->Props); + pFmt->wBitsPerSample = PDMAudioPropsSampleBits(&pCfg->Props); + pFmt->nSamplesPerSec = PDMAudioPropsHz(&pCfg->Props); + pFmt->nBlockAlign = PDMAudioPropsFrameSize(&pCfg->Props); + 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. * * @param pThis Host audio driver instance. - */ -static void dsoundUpdateStatusInternal(PDRVHOSTDSOUND pThis) -{ -#if 0 /** @todo r=bird: This isn't doing *ANYTHING* useful. So, I've just disabled it. */ - AssertPtrReturnVoid(pThis); + * @param pStreamDS Stream to reset state for. + */ +static void dsoundStreamReset(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) +{ + RT_NOREF(pThis); + + LogFunc(("Resetting %s\n", + pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN ? "capture" : "playback")); + + if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN) + { + 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. */ + 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)) + { + 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); + } + } + } + else if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT) + { + /* 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; + 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. */ + if (pStreamDS->Out.pDSB) + { + HRESULT hr = IDirectSoundBuffer8_SetCurrentPosition(pStreamDS->Out.pDSB, 0); + if (SUCCEEDED(hr)) + { + 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); + } + } + } + } + +#ifdef LOG_ENABLED + pStreamDS->Dbg.tsLastTransferredMs = 0; +#endif +} + + +static HRESULT directSoundCaptureClose(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) +{ + AssertPtrReturn(pThis, E_POINTER); + AssertPtrReturn(pStreamDS, E_POINTER); + LogFlowFuncEnter(); - PDMAudioHostEnumDelete(&pThis->DeviceEnum); - int rc = dsoundDevicesEnumerate(pThis, &pThis->DeviceEnum); - if (RT_SUCCESS(rc)) - { -#if 0 - if ( pThis->fEnabledOut != RT_BOOL(cbCtx.cDevOut) - || pThis->fEnabledIn != RT_BOOL(cbCtx.cDevIn)) - { - /** @todo Use a registered callback to the audio connector (e.g "OnConfigurationChanged") to - * let the connector know that something has changed within the host backend. */ - } -#endif - pThis->fEnabledIn = PDMAudioHostEnumCountMatching(&pThis->DeviceEnum, PDMAUDIODIR_IN) != 0; - pThis->fEnabledOut = PDMAudioHostEnumCountMatching(&pThis->DeviceEnum, PDMAUDIODIR_OUT) != 0; - } - - LogFlowFuncLeaveRC(rc); -#else - RT_NOREF(pThis); -#endif -} + HRESULT hr = directSoundCaptureStop(pThis, pStreamDS, true /* fFlush */); + if (FAILED(hr)) + return hr; + + if ( pStreamDS + && pStreamDS->In.pDSCB) + { + DSLOG(("DSound: Closing capturing stream\n")); + + IDirectSoundCaptureBuffer8_Release(pStreamDS->In.pDSCB); + pStreamDS->In.pDSCB = NULL; + } + + LogFlowFunc(("Returning %Rhrc\n", hr)); + return hr; +} + + +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... */ + + 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); + +/** @todo r=bird: I cannot see any code populating pCfgAcq... */ + + 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)); + + /** @todo r=bird: Why is this called every time? It triggers a device + * enumeration. Andy claimed on IRC that enumeration was only done once... + * It's generally a 'ing waste of time here too, as we dont really use any of + * the information we gather there. */ + 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, @@ -1855,4 +1712,198 @@ 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); + 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; + + int rc; + if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN) + rc = dsoundDestroyStreamIn(pThis, pStreamDS); + else + rc = dsoundDestroyStreamOut(pThis, pStreamDS); + + if (RT_SUCCESS(rc)) + { + if (RTCritSectIsInitialized(&pStreamDS->CritSect)) + rc = RTCritSectDelete(&pStreamDS->CritSect); + } + + return rc; +} + + +/** + * Enables or disables a stream. + * + * @return VBox status code. + * @param pThis Host audio driver instance. + * @param pStreamDS Stream to enable / disable. + * @param fEnable Whether to enable or disable the stream. + */ +static int dsoundStreamEnable(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fEnable) +{ + RT_NOREF(pThis); + + LogFunc(("%s %s\n", + fEnable ? "Enabling" : "Disabling", + pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN ? "capture" : "playback")); + + if (fEnable) + dsoundStreamReset(pThis, pStreamDS); + + pStreamDS->fEnabled = fEnable; + + return VINF_SUCCESS; +} + + +/** + * Starts playing a DirectSound stream. + * + * @return HRESULT + * @param pThis Host audio driver instance. + * @param pStreamDS Stream to start playing. + */ +static HRESULT directSoundPlayStart(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) +{ + HRESULT hr = S_OK; + + DWORD fFlags = DSCBSTART_LOOPING; + + for (unsigned i = 0; i < DRV_DSOUND_RESTORE_ATTEMPTS_MAX; i++) + { + DSLOG(("DSound: Starting playback\n")); + hr = IDirectSoundBuffer8_Play(pStreamDS->Out.pDSB, 0, 0, fFlags); + if ( SUCCEEDED(hr) + || hr != DSERR_BUFFERLOST) + break; + LogFunc(("Restarting playback failed due to lost buffer, restoring ...\n")); + directSoundPlayRestore(pThis, pStreamDS->Out.pDSB); + } + + return hr; +} + + +static HRESULT directSoundPlayStop(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fFlush) +{ + AssertPtrReturn(pThis, E_POINTER); + AssertPtrReturn(pStreamDS, E_POINTER); + + HRESULT hr = S_OK; + + if (pStreamDS->Out.pDSB) + { + DSLOG(("DSound: Stopping playback\n")); + hr = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB); + if (FAILED(hr)) + { + hr = directSoundPlayRestore(pThis, pStreamDS->Out.pDSB); + if (FAILED(hr)) /** @todo shouldn't this be a SUCCEEDED? */ + hr = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB); + } + } + + if (SUCCEEDED(hr)) + { + if (fFlush) + dsoundStreamReset(pThis, pStreamDS); + } + + if (FAILED(hr)) + DSLOGREL(("DSound: %s playback failed with %Rhrc\n", fFlush ? "Stopping" : "Pausing", hr)); + + return hr; +} + static int dsoundControlStreamOut(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, PDMAUDIOSTREAMCMD enmStreamCmd) @@ -1949,149 +2000,71 @@ } -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPlay} - */ -static DECLCALLBACK(int) drvHostDSoundHA_StreamPlay(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, - const void *pvBuf, uint32_t cbBuf, uint32_t *pcbWritten) -{ - PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); - PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; - AssertPtrReturn(pStreamDS, 0); - AssertPtrReturn(pvBuf, VERR_INVALID_POINTER); - AssertReturn(cbBuf, VERR_INVALID_PARAMETER); - AssertPtrReturn(pcbWritten, VERR_INVALID_POINTER); - - if (pStreamDS->fEnabled) - AssertReturn(pStreamDS->cbBufSize, VERR_INTERNAL_ERROR_2); + +static HRESULT directSoundCaptureStart(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) +{ + AssertPtrReturn(pThis, E_POINTER); + AssertPtrReturn(pStreamDS, E_POINTER); + + HRESULT hr; + if (pStreamDS->In.pDSCB) + { + DWORD dwStatus; + hr = IDirectSoundCaptureBuffer8_GetStatus(pStreamDS->In.pDSCB, &dwStatus); + if (FAILED(hr)) + { + DSLOGREL(("DSound: Retrieving capture status failed with %Rhrc\n", hr)); + } + else + { + if (dwStatus & DSCBSTATUS_CAPTURING) + { + DSLOG(("DSound: Already capturing\n")); + } + else + { + const DWORD fFlags = DSCBSTART_LOOPING; + + DSLOG(("DSound: Starting to capture\n")); + hr = IDirectSoundCaptureBuffer8_Start(pStreamDS->In.pDSCB, fFlags); + if (FAILED(hr)) + DSLOGREL(("DSound: Starting to capture failed with %Rhrc\n", hr)); + } + } + } else - { - Log2Func(("Stream disabled, skipping\n")); - return VINF_SUCCESS; - } - -/** @todo Any condition under which we should call dsoundUpdateStatusInternal(pThis) here? - * The old code thought it did so in case of failure, only it couldn't ever fails, so it never did. */ - - /* - * Transfer loop. - */ - uint32_t cbWritten = 0; - while (cbBuf > 0) - { - /* - * Figure out how much we can possibly write. - */ - DWORD offPlayCursor = 0; - DWORD cbWritable = 0; - int rc = dsoundGetFreeOut(pThis, pStreamDS, &cbWritable, &offPlayCursor); - AssertRCReturn(rc, rc); - if (cbWritable < pStreamDS->Cfg.Props.cbFrame) - break; - - uint32_t const cbToWrite = RT_MIN(cbWritable, cbBuf); - Log3Func(("offPlay=%#x offWritePos=%#x -> cbWritable=%#x cbToWrite=%#x%s%s\n", offPlayCursor, pStreamDS->Out.offWritePos, - cbWritable, cbToWrite, pStreamDS->Out.fFirstTransfer ? " first" : "", pStreamDS->Out.fDrain ? " drain" : "")); - - /* - * Lock that amount of buffer. - */ - PVOID pv1 = NULL; - DWORD cb1 = 0; - PVOID pv2 = NULL; - DWORD cb2 = 0; - HRESULT hrc = directSoundPlayLock(pThis, pStreamDS, pStreamDS->Out.offWritePos, cbToWrite, - &pv1, &pv2, &cb1, &cb2, 0 /*dwFlags*/); - AssertMsgReturn(SUCCEEDED(hrc), ("%Rhrc\n", hrc), VERR_ACCESS_DENIED); /** @todo translate these status codes already! */ - //AssertMsg(cb1 + cb2 == cbToWrite, ("%#x + %#x vs %#x\n", cb1, cb2, cbToWrite)); - - /* - * Copy over the data. - */ - memcpy(pv1, pvBuf, cb1); - pvBuf = (uint8_t *)pvBuf + cb1; - cbBuf -= cb1; - cbWritten += cb1; - - if (pv2) - { - memcpy(pv2, pvBuf, cb2); - pvBuf = (uint8_t *)pvBuf + cb2; - cbBuf -= cb2; - cbWritten += cb2; - } - - /* - * Unlock and update the write position. - */ - directSoundPlayUnlock(pThis, pStreamDS->Out.pDSB, pv1, pv2, cb1, cb2); /** @todo r=bird: pThis + pDSB parameters here for Unlock, but only pThis for Lock. Why? */ - pStreamDS->Out.offWritePos = (pStreamDS->Out.offWritePos + cb1 + cb2) % pStreamDS->cbBufSize; - - /* - * If this was the first chunk, kick off playing. - */ - if (!pStreamDS->Out.fFirstTransfer) - { /* likely */ } - else - { - *pcbWritten = cbWritten; - hrc = directSoundPlayStart(pThis, pStreamDS); - AssertMsgReturn(SUCCEEDED(hrc), ("%Rhrc\n", hrc), VERR_ACCESS_DENIED); /** @todo translate these status codes already! */ - pStreamDS->Out.fFirstTransfer = false; - } - } - - /* - * Done. - */ - *pcbWritten = cbWritten; - - pStreamDS->Out.cbTransferred += cbWritten; - if (cbWritten) - { - uint64_t const msPrev = pStreamDS->Out.tsLastTransferredMs; - pStreamDS->Out.cbLastTransferred = cbWritten; - pStreamDS->Out.tsLastTransferredMs = RTTimeMilliTS(); - LogFlowFunc(("cbLastTransferred=%RU32, tsLastTransferredMs=%RU64 cMsDelta=%RU64\n", - cbWritten, pStreamDS->Out.tsLastTransferredMs, msPrev ? pStreamDS->Out.tsLastTransferredMs - msPrev : 0)); - } - return VINF_SUCCESS; -} - -static int dsoundDestroyStreamOut(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ - LogFlowFuncEnter(); - - HRESULT hr = directSoundPlayStop(pThis, pStreamDS, true /* fFlush */); + hr = E_UNEXPECTED; + + LogFlowFunc(("Returning %Rhrc\n", hr)); + return hr; +} + + +static HRESULT directSoundCaptureStop(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fFlush) +{ + AssertPtrReturn(pThis, E_POINTER); + AssertPtrReturn(pStreamDS, E_POINTER); + + RT_NOREF(pThis); + + HRESULT hr = S_OK; + + if (pStreamDS->In.pDSCB) + { + DSLOG(("DSound: Stopping capture\n")); + hr = IDirectSoundCaptureBuffer_Stop(pStreamDS->In.pDSCB); + } + if (SUCCEEDED(hr)) { - hr = directSoundPlayClose(pThis, pStreamDS); - if (FAILED(hr)) - return VERR_GENERAL_FAILURE; /** @todo Fix. */ - } - - return VINF_SUCCESS; -} - -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; -} + if (fFlush) + dsoundStreamReset(pThis, pStreamDS); + } + + if (FAILED(hr)) + DSLOGREL(("DSound: Stopping capture buffer failed with %Rhrc\n", hr)); + + return hr; +} + static int dsoundControlStreamIn(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, PDMAUDIOSTREAMCMD enmStreamCmd) @@ -2160,4 +2133,320 @@ } + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamControl} + */ +static DECLCALLBACK(int) drvHostDSoundHA_StreamControl(PPDMIHOSTAUDIO pInterface, + PPDMAUDIOBACKENDSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd) +{ + AssertPtrReturn(pInterface, VERR_INVALID_POINTER); + AssertPtrReturn(pStream, VERR_INVALID_POINTER); + + PDRVHOSTDSOUND pThis = PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface); + PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + + int rc; + if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN) + rc = dsoundControlStreamIn(pThis, pStreamDS, enmStreamCmd); + else + rc = dsoundControlStreamOut(pThis, pStreamDS, enmStreamCmd); + + return rc; +} + + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetReadable} + */ +static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetReadable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) +{ + /*PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); */ RT_NOREF(pInterface); + PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + AssertPtrReturn(pStreamDS, 0); + Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN); + + if (pStreamDS->fEnabled) + { + /* This is the same calculation as for StreamGetPending. */ + AssertPtr(pStreamDS->In.pDSCB); + DWORD offCaptureCursor = 0; + DWORD offReadCursor = 0; + HRESULT hrc = IDirectSoundCaptureBuffer_GetCurrentPosition(pStreamDS->In.pDSCB, &offCaptureCursor, &offReadCursor); + if (SUCCEEDED(hrc)) + { + uint32_t cbPending = dsoundRingDistance(offCaptureCursor, offReadCursor, pStreamDS->cbBufSize); + Log3Func(("cbPending=%RU32\n", cbPending)); + return cbPending; + } + AssertMsgFailed(("hrc=%Rhrc\n", hrc)); + } + + return 0; +} + + +/** + * Retrieves the number of free bytes available for writing to a DirectSound output stream. + * + * @return VBox status code. VERR_NOT_AVAILABLE if unable to determine or the + * buffer was not recoverable. + * @param pThis Host audio driver instance. + * @param pStreamDS DirectSound output stream to retrieve number for. + * @param pdwFree Where to return the free amount on success. + * @param poffPlayCursor Where to return the play cursor offset. + */ +static int dsoundGetFreeOut(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, DWORD *pdwFree, DWORD *poffPlayCursor) +{ + AssertPtrReturn(pThis, VERR_INVALID_POINTER); + AssertPtrReturn(pStreamDS, VERR_INVALID_POINTER); + AssertPtrReturn(pdwFree, VERR_INVALID_POINTER); + AssertPtr(poffPlayCursor); + + Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT); /* Paranoia. */ + + LPDIRECTSOUNDBUFFER8 pDSB = pStreamDS->Out.pDSB; + if (!pDSB) + { + AssertPtr(pDSB); + return VERR_INVALID_POINTER; + } + + HRESULT hr = S_OK; + + /* Get the current play position which is used for calculating the free space in the buffer. */ + for (unsigned i = 0; i < DRV_DSOUND_RESTORE_ATTEMPTS_MAX; i++) + { + DWORD offPlayCursor = 0; + DWORD offWriteCursor = 0; + hr = IDirectSoundBuffer8_GetCurrentPosition(pDSB, &offPlayCursor, &offWriteCursor); + if (SUCCEEDED(hr)) + { + int32_t cbDiff = offWriteCursor - offPlayCursor; + if (cbDiff < 0) + cbDiff += pStreamDS->cbBufSize; + + int32_t cbFree = offPlayCursor - pStreamDS->Out.offWritePos; + if (cbFree < 0) + cbFree += pStreamDS->cbBufSize; + + if (cbFree > (int32_t)pStreamDS->cbBufSize - cbDiff) + { + /** @todo count/log these. */ + pStreamDS->Out.offWritePos = offWriteCursor; + cbFree = pStreamDS->cbBufSize - cbDiff; + } + + /* When starting to use a DirectSound buffer, offPlayCursor and offWriteCursor + * both point at position 0, so we won't be able to detect how many bytes + * are writable that way. + * + * So use our per-stream written indicator to see if we just started a stream. */ + if (pStreamDS->Out.cbWritten == 0) + cbFree = pStreamDS->cbBufSize; + + DSLOGREL(("DSound: offPlayCursor=%RU32, offWriteCursor=%RU32, offWritePos=%RU32 -> cbFree=%RI32\n", + offPlayCursor, offWriteCursor, pStreamDS->Out.offWritePos, cbFree)); + + *pdwFree = cbFree; + *poffPlayCursor = offPlayCursor; + return VINF_SUCCESS; + } + + if (hr != DSERR_BUFFERLOST) /** @todo MSDN doesn't state this error for GetCurrentPosition(). */ + break; + + LogFunc(("Getting playing position failed due to lost buffer, restoring ...\n")); + + directSoundPlayRestore(pThis, pDSB); + } + + if (hr != DSERR_BUFFERLOST) /* Avoid log flooding if the error is still there. */ + DSLOGREL(("DSound: Getting current playback position failed with %Rhrc\n", hr)); + + LogFunc(("Failed with %Rhrc\n", hr)); + + *poffPlayCursor = pStreamDS->cbBufSize; + return VERR_NOT_AVAILABLE; +} + + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetWritable} + */ +static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetWritable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) +{ + PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); + PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + AssertPtrReturn(pStreamDS, 0); + + DWORD cbFree = 0; + DWORD offIgn = 0; + int rc = dsoundGetFreeOut(pThis, pStreamDS, &cbFree, &offIgn); + AssertRCReturn(rc, 0); + + return cbFree; +} + +#if 0 /* This isn't working as the write cursor is more a function of time than what we do. + Previously we only reported the pre-buffering status anyway, so no harm. */ +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetPending} + */ +static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetPending(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) +{ + /*PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); */ RT_NOREF(pInterface); + PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + AssertPtrReturn(pStreamDS, 0); + + if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT) + { + /* This is a similar calculation as for StreamGetReadable, only for an output buffer. */ + AssertPtr(pStreamDS->In.pDSCB); + DWORD offPlayCursor = 0; + DWORD offWriteCursor = 0; + HRESULT hrc = IDirectSoundBuffer8_GetCurrentPosition(pStreamDS->Out.pDSB, &offPlayCursor, &offWriteCursor); + if (SUCCEEDED(hrc)) + { + uint32_t cbPending = dsoundRingDistance(offWriteCursor, offPlayCursor, pStreamDS->cbBufSize); + Log3Func(("cbPending=%RU32\n", cbPending)); + return cbPending; + } + AssertMsgFailed(("hrc=%Rhrc\n", hrc)); + } + /* else: For input streams we never have any pending data. */ + + return 0; +} +#endif + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetStatus} + */ +static DECLCALLBACK(PDMAUDIOSTREAMSTS) drvHostDSoundHA_StreamGetStatus(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) +{ + RT_NOREF(pInterface); + AssertPtrReturn(pStream, PDMAUDIOSTREAMSTS_FLAGS_NONE); + + PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + + PDMAUDIOSTREAMSTS fStrmStatus = PDMAUDIOSTREAMSTS_FLAGS_INITIALIZED; + + if (pStreamDS->fEnabled) + fStrmStatus |= PDMAUDIOSTREAMSTS_FLAGS_ENABLED; + + return fStrmStatus; +} + + +/** + * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPlay} + */ +static DECLCALLBACK(int) drvHostDSoundHA_StreamPlay(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, + const void *pvBuf, uint32_t cbBuf, uint32_t *pcbWritten) +{ + PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); + PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; + AssertPtrReturn(pStreamDS, 0); + AssertPtrReturn(pvBuf, VERR_INVALID_POINTER); + AssertReturn(cbBuf, VERR_INVALID_PARAMETER); + AssertPtrReturn(pcbWritten, VERR_INVALID_POINTER); + + if (pStreamDS->fEnabled) + AssertReturn(pStreamDS->cbBufSize, VERR_INTERNAL_ERROR_2); + else + { + Log2Func(("Stream disabled, skipping\n")); + return VINF_SUCCESS; + } + +/** @todo Any condition under which we should call dsoundUpdateStatusInternal(pThis) here? + * The old code thought it did so in case of failure, only it couldn't ever fails, so it never did. */ + + /* + * Transfer loop. + */ + uint32_t cbWritten = 0; + while (cbBuf > 0) + { + /* + * Figure out how much we can possibly write. + */ + DWORD offPlayCursor = 0; + DWORD cbWritable = 0; + int rc = dsoundGetFreeOut(pThis, pStreamDS, &cbWritable, &offPlayCursor); + AssertRCReturn(rc, rc); + if (cbWritable < pStreamDS->Cfg.Props.cbFrame) + break; + + uint32_t const cbToWrite = RT_MIN(cbWritable, cbBuf); + Log3Func(("offPlay=%#x offWritePos=%#x -> cbWritable=%#x cbToWrite=%#x%s%s\n", offPlayCursor, pStreamDS->Out.offWritePos, + cbWritable, cbToWrite, pStreamDS->Out.fFirstTransfer ? " first" : "", pStreamDS->Out.fDrain ? " drain" : "")); + + /* + * Lock that amount of buffer. + */ + PVOID pv1 = NULL; + DWORD cb1 = 0; + PVOID pv2 = NULL; + DWORD cb2 = 0; + HRESULT hrc = directSoundPlayLock(pThis, pStreamDS, pStreamDS->Out.offWritePos, cbToWrite, + &pv1, &pv2, &cb1, &cb2, 0 /*dwFlags*/); + AssertMsgReturn(SUCCEEDED(hrc), ("%Rhrc\n", hrc), VERR_ACCESS_DENIED); /** @todo translate these status codes already! */ + //AssertMsg(cb1 + cb2 == cbToWrite, ("%#x + %#x vs %#x\n", cb1, cb2, cbToWrite)); + + /* + * Copy over the data. + */ + memcpy(pv1, pvBuf, cb1); + pvBuf = (uint8_t *)pvBuf + cb1; + cbBuf -= cb1; + cbWritten += cb1; + + if (pv2) + { + memcpy(pv2, pvBuf, cb2); + pvBuf = (uint8_t *)pvBuf + cb2; + cbBuf -= cb2; + cbWritten += cb2; + } + + /* + * Unlock and update the write position. + */ + directSoundPlayUnlock(pThis, pStreamDS->Out.pDSB, pv1, pv2, cb1, cb2); /** @todo r=bird: pThis + pDSB parameters here for Unlock, but only pThis for Lock. Why? */ + pStreamDS->Out.offWritePos = (pStreamDS->Out.offWritePos + cb1 + cb2) % pStreamDS->cbBufSize; + + /* + * If this was the first chunk, kick off playing. + */ + if (!pStreamDS->Out.fFirstTransfer) + { /* likely */ } + else + { + *pcbWritten = cbWritten; + hrc = directSoundPlayStart(pThis, pStreamDS); + AssertMsgReturn(SUCCEEDED(hrc), ("%Rhrc\n", hrc), VERR_ACCESS_DENIED); /** @todo translate these status codes already! */ + pStreamDS->Out.fFirstTransfer = false; + } + } + + /* + * Done. + */ + *pcbWritten = cbWritten; + + pStreamDS->Out.cbTransferred += cbWritten; + if (cbWritten) + { + uint64_t const msPrev = pStreamDS->Out.tsLastTransferredMs; + pStreamDS->Out.cbLastTransferred = cbWritten; + pStreamDS->Out.tsLastTransferredMs = RTTimeMilliTS(); + LogFlowFunc(("cbLastTransferred=%RU32, tsLastTransferredMs=%RU64 cMsDelta=%RU64\n", + cbWritten, pStreamDS->Out.tsLastTransferredMs, msPrev ? pStreamDS->Out.tsLastTransferredMs - msPrev : 0)); + } + return VINF_SUCCESS; +} + + /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCapture} @@ -2275,59 +2564,56 @@ } -static int dsoundDestroyStreamIn(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS) -{ + +/********************************************************************************************************************************* +* PDMDRVINS::IBase Interface * +*********************************************************************************************************************************/ + +/** + * @callback_method_impl{PDMIBASE,pfnQueryInterface} + */ +static DECLCALLBACK(void *) drvHostDSoundQueryInterface(PPDMIBASE pInterface, const char *pszIID) +{ + PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface); + PDRVHOSTDSOUND pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTDSOUND); + + PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase); + PDMIBASE_RETURN_INTERFACE(pszIID, PDMIHOSTAUDIO, &pThis->IHostAudio); + return NULL; +} + + +/********************************************************************************************************************************* +* PDMDRVREG Interface * +*********************************************************************************************************************************/ + +/** + * @callback_method_impl{FNPDMDRVDESTRUCT, pfnDestruct} + */ +static DECLCALLBACK(void) drvHostDSoundDestruct(PPDMDRVINS pDrvIns) +{ + PDRVHOSTDSOUND pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTDSOUND); + PDMDRV_CHECK_VERSIONS_RETURN_VOID(pDrvIns); + LogFlowFuncEnter(); - directSoundCaptureClose(pThis, pStreamDS); - - return VINF_SUCCESS; -} - -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnGetConfig} - */ -static DECLCALLBACK(int) drvHostDSoundHA_GetConfig(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pBackendCfg) -{ - RT_NOREF(pInterface); - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pBackendCfg, VERR_INVALID_POINTER); - - RT_BZERO(pBackendCfg, sizeof(PPDMAUDIOBACKENDCFG)); - - pBackendCfg->cbStreamOut = sizeof(DSOUNDSTREAM); - pBackendCfg->cbStreamIn = sizeof(DSOUNDSTREAM); - - RTStrPrintf2(pBackendCfg->szName, sizeof(pBackendCfg->szName), "DirectSound"); - - pBackendCfg->cMaxStreamsIn = UINT32_MAX; - pBackendCfg->cMaxStreamsOut = UINT32_MAX; - - return VINF_SUCCESS; -} - -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnGetDevices} - */ -static DECLCALLBACK(int) drvHostDSoundHA_GetDevices(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHOSTENUM pDeviceEnum) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pDeviceEnum, VERR_INVALID_POINTER); - - PDRVHOSTDSOUND pThis = PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface); - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_SUCCESS(rc)) - { - PDMAudioHostEnumInit(pDeviceEnum); - rc = dsoundDevicesEnumerate(pThis, pDeviceEnum); - if (RT_FAILURE(rc)) - PDMAudioHostEnumDelete(pDeviceEnum); - - int rc2 = RTCritSectLeave(&pThis->CritSect); - AssertRC(rc2); - } - - LogFlowFunc(("Returning %Rrc\n", rc)); - return rc; +#ifdef VBOX_WITH_AUDIO_MMNOTIFICATION_CLIENT + if (pThis->m_pNotificationClient) + { + pThis->m_pNotificationClient->Unregister(); + pThis->m_pNotificationClient->Release(); + + pThis->m_pNotificationClient = NULL; + } +#endif + + PDMAudioHostEnumDelete(&pThis->DeviceEnum); + + if (pThis->pDrvIns) + CoUninitialize(); + + int rc2 = RTCritSectDelete(&pThis->CritSect); + AssertRC(rc2); + + LogFlowFuncLeave(); } @@ -2353,4 +2639,5 @@ } + static void dsoundConfigInit(PDRVHOSTDSOUND pThis, PCFGMNODE pCfg) { @@ -2363,242 +2650,4 @@ } -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnGetStatus} - */ -static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvHostDSoundHA_GetStatus(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir) -{ - RT_NOREF(enmDir); - AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN); - - return PDMAUDIOBACKENDSTS_RUNNING; -} - -/** - * @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); - 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; -} - -/** - * @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; - - int rc; - if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN) - rc = dsoundDestroyStreamIn(pThis, pStreamDS); - else - rc = dsoundDestroyStreamOut(pThis, pStreamDS); - - if (RT_SUCCESS(rc)) - { - if (RTCritSectIsInitialized(&pStreamDS->CritSect)) - rc = RTCritSectDelete(&pStreamDS->CritSect); - } - - return rc; -} - -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamControl} - */ -static DECLCALLBACK(int) drvHostDSoundHA_StreamControl(PPDMIHOSTAUDIO pInterface, - PPDMAUDIOBACKENDSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pStream, VERR_INVALID_POINTER); - - PDRVHOSTDSOUND pThis = PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface); - PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; - - int rc; - if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN) - rc = dsoundControlStreamIn(pThis, pStreamDS, enmStreamCmd); - else - rc = dsoundControlStreamOut(pThis, pStreamDS, enmStreamCmd); - - return rc; -} - -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetReadable} - */ -static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetReadable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) -{ - /*PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); */ RT_NOREF(pInterface); - PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; - AssertPtrReturn(pStreamDS, 0); - Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN); - - if (pStreamDS->fEnabled) - { - /* This is the same calculation as for StreamGetPending. */ - AssertPtr(pStreamDS->In.pDSCB); - DWORD offCaptureCursor = 0; - DWORD offReadCursor = 0; - HRESULT hrc = IDirectSoundCaptureBuffer_GetCurrentPosition(pStreamDS->In.pDSCB, &offCaptureCursor, &offReadCursor); - if (SUCCEEDED(hrc)) - { - uint32_t cbPending = dsoundRingDistance(offCaptureCursor, offReadCursor, pStreamDS->cbBufSize); - Log3Func(("cbPending=%RU32\n", cbPending)); - return cbPending; - } - AssertMsgFailed(("hrc=%Rhrc\n", hrc)); - } - - return 0; -} - -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetWritable} - */ -static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetWritable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) -{ - PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); - PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; - AssertPtrReturn(pStreamDS, 0); - - DWORD cbFree = 0; - DWORD offIgn = 0; - int rc = dsoundGetFreeOut(pThis, pStreamDS, &cbFree, &offIgn); - AssertRCReturn(rc, 0); - - return cbFree; -} - -#if 0 /* This isn't working as the write cursor is more a function of time than what we do. - Previously we only reported the pre-buffering status anyway, so no harm. */ -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetPending} - */ -static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetPending(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) -{ - /*PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); */ RT_NOREF(pInterface); - PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; - AssertPtrReturn(pStreamDS, 0); - - if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT) - { - /* This is a similar calculation as for StreamGetReadable, only for an output buffer. */ - AssertPtr(pStreamDS->In.pDSCB); - DWORD offPlayCursor = 0; - DWORD offWriteCursor = 0; - HRESULT hrc = IDirectSoundBuffer8_GetCurrentPosition(pStreamDS->Out.pDSB, &offPlayCursor, &offWriteCursor); - if (SUCCEEDED(hrc)) - { - uint32_t cbPending = dsoundRingDistance(offWriteCursor, offPlayCursor, pStreamDS->cbBufSize); - Log3Func(("cbPending=%RU32\n", cbPending)); - return cbPending; - } - AssertMsgFailed(("hrc=%Rhrc\n", hrc)); - } - /* else: For input streams we never have any pending data. */ - - return 0; -} -#endif - -/** - * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetStatus} - */ -static DECLCALLBACK(PDMAUDIOSTREAMSTS) drvHostDSoundHA_StreamGetStatus(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) -{ - RT_NOREF(pInterface); - AssertPtrReturn(pStream, PDMAUDIOSTREAMSTS_FLAGS_NONE); - - PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream; - - PDMAUDIOSTREAMSTS fStrmStatus = PDMAUDIOSTREAMSTS_FLAGS_INITIALIZED; - - if (pStreamDS->fEnabled) - fStrmStatus |= PDMAUDIOSTREAMSTS_FLAGS_ENABLED; - - return fStrmStatus; -} - - -/********************************************************************************************************************************* -* PDMDRVINS::IBase Interface * -*********************************************************************************************************************************/ - -/** - * @callback_method_impl{PDMIBASE,pfnQueryInterface} - */ -static DECLCALLBACK(void *) drvHostDSoundQueryInterface(PPDMIBASE pInterface, const char *pszIID) -{ - PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface); - PDRVHOSTDSOUND pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTDSOUND); - - PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase); - PDMIBASE_RETURN_INTERFACE(pszIID, PDMIHOSTAUDIO, &pThis->IHostAudio); - return NULL; -} - - -/********************************************************************************************************************************* -* PDMDRVREG Interface * -*********************************************************************************************************************************/ - -/** - * @callback_method_impl{FNPDMDRVDESTRUCT, pfnDestruct} - */ -static DECLCALLBACK(void) drvHostDSoundDestruct(PPDMDRVINS pDrvIns) -{ - PDRVHOSTDSOUND pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTDSOUND); - PDMDRV_CHECK_VERSIONS_RETURN_VOID(pDrvIns); - - LogFlowFuncEnter(); - -#ifdef VBOX_WITH_AUDIO_MMNOTIFICATION_CLIENT - if (pThis->m_pNotificationClient) - { - pThis->m_pNotificationClient->Unregister(); - pThis->m_pNotificationClient->Release(); - - pThis->m_pNotificationClient = NULL; - } -#endif - - PDMAudioHostEnumDelete(&pThis->DeviceEnum); - - if (pThis->pDrvIns) - CoUninitialize(); - - int rc2 = RTCritSectDelete(&pThis->CritSect); - AssertRC(rc2); - - LogFlowFuncLeave(); -} /** @@ -2621,4 +2670,5 @@ /* IHostAudio */ pThis->IHostAudio.pfnGetConfig = drvHostDSoundHA_GetConfig; + pThis->IHostAudio.pfnGetDevices = drvHostDSoundHA_GetDevices; pThis->IHostAudio.pfnGetStatus = drvHostDSoundHA_GetStatus; pThis->IHostAudio.pfnStreamCreate = drvHostDSoundHA_StreamCreate; @@ -2627,9 +2677,8 @@ pThis->IHostAudio.pfnStreamGetReadable = drvHostDSoundHA_StreamGetReadable; pThis->IHostAudio.pfnStreamGetWritable = drvHostDSoundHA_StreamGetWritable; + pThis->IHostAudio.pfnStreamGetPending = NULL; pThis->IHostAudio.pfnStreamGetStatus = drvHostDSoundHA_StreamGetStatus; pThis->IHostAudio.pfnStreamPlay = drvHostDSoundHA_StreamPlay; pThis->IHostAudio.pfnStreamCapture = drvHostDSoundHA_StreamCapture; - pThis->IHostAudio.pfnGetDevices = drvHostDSoundHA_GetDevices; - pThis->IHostAudio.pfnStreamGetPending = NULL; /*