Index: /trunk/include/VBox/vmm/pdmaudioifs.h =================================================================== --- /trunk/include/VBox/vmm/pdmaudioifs.h (revision 61522) +++ /trunk/include/VBox/vmm/pdmaudioifs.h (revision 61523) @@ -173,5 +173,7 @@ /** Interleaved access, where the data can be * mixed together with data of other audio streams. */ - PDMAUDIOSTREAMLAYOUT_INTERLEAVED + PDMAUDIOSTREAMLAYOUT_INTERLEAVED, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOSTREAMLAYOUT_32BIT_HACK = 0x7fffffff } PDMAUDIOSTREAMLAYOUT, *PPDMAUDIOSTREAMLAYOUT; @@ -716,5 +718,5 @@ /** - * Plays (transfers) all available audio samples of a an output stream via the connected host backend. + * Plays (transfers) available audio samples via the host backend. Only works with output streams. * * @returns VBox status code. @@ -723,4 +725,13 @@ */ DECLR3CALLBACKMEMBER(int, pfnStreamPlay, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesPlayed)); + + /** + * Captures (transfers) available audio samples from the host backend. Only works with input streams. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pcSamplesCaptured Number of samples captured. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamCapture, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesCaptured)); #ifdef VBOX_WITH_AUDIO_CALLBACKS @@ -732,5 +743,5 @@ /** PDMIAUDIOCONNECTOR interface ID. */ -#define PDMIAUDIOCONNECTOR_IID "9C097435-3276-4D88-A49A-A4FE671D86F8" +#define PDMIAUDIOCONNECTOR_IID "D1B6465D-E3DD-455B-9E05-FB6D56F7D472" Index: /trunk/src/VBox/Devices/Audio/AudioMixBuffer.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/AudioMixBuffer.cpp (revision 61522) +++ /trunk/src/VBox/Devices/Audio/AudioMixBuffer.cpp (revision 61523) @@ -21,6 +21,6 @@ #ifdef DEBUG_andy /* - * DEBUG_DUMP_PCM_DATA enables dumping the raw PCM data - * to a file on the host. Be sure to adjust DEBUG_DUMP_PCM_DATA_PATH + * AUDIOMIXBUF_DEBUG_DUMP_PCM_DATA enables dumping the raw PCM data + * to a file on the host. Be sure to adjust AUDIOMIXBUF_DEBUG_DUMP_PCM_DATA_PATH * to your needs before using this! */ @@ -243,5 +243,4 @@ pIter->pszName, pIter->cMixed, pIter->cMixed - cSamplesToClear)); - Assert(cSamplesToClear <= pIter->cMixed); pIter->cMixed -= RT_MIN(pIter->cMixed, cSamplesToClear); } Index: /trunk/src/VBox/Devices/Audio/AudioMixer.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/AudioMixer.cpp (revision 61522) +++ /trunk/src/VBox/Devices/Audio/AudioMixer.cpp (revision 61523) @@ -341,8 +341,5 @@ } else - { - AssertMsgFailed(("Direction not implemented\n")); - rc = VERR_NOT_IMPLEMENTED; - } + AssertFailedStmt(rc = VERR_NOT_IMPLEMENTED); #else rc = VINF_SUCCESS; @@ -455,9 +452,9 @@ } -int AudioMixerSinkCtl(PAUDMIXSINK pSink, AUDMIXSINKCMD enmCmd) +int AudioMixerSinkCtl(PAUDMIXSINK pSink, AUDMIXSINKCMD enmSinkCmd) { AssertPtrReturn(pSink, VERR_INVALID_POINTER); - PDMAUDIOSTREAMCMD enmCmdStream = audioMixerSinkToStreamCmd(enmCmd); + PDMAUDIOSTREAMCMD enmCmdStream = audioMixerSinkToStreamCmd(enmSinkCmd); if (enmCmdStream == PDMAUDIOSTREAMCMD_UNKNOWN) return VERR_NOT_SUPPORTED; @@ -474,5 +471,13 @@ } - LogFlowFunc(("[%s]: enmCmd=%ld, rc=%Rrc\n", pSink->pszName, enmCmd, rc)); + /* Remove dirty bit in any case. */ + pSink->fStatus &= ~AUDMIXSINK_STS_DIRTY; + + if (enmSinkCmd == AUDMIXSINKCMD_ENABLE) + pSink->fStatus |= AUDMIXSINK_STS_RUNNING; + else if (enmSinkCmd == AUDMIXSINKCMD_DISABLE) + pSink->fStatus &= ~AUDMIXSINK_STS_RUNNING; + + LogFlowFunc(("[%s]: enmCmd=%ld, fStatus=0x%x, rc=%Rrc\n", pSink->pszName, enmSinkCmd, pSink->fStatus, rc)); return rc; } @@ -557,4 +562,15 @@ } +AUDMIXSINKSTS AudioMixerSinkGetStatus(PAUDMIXSINK pSink) +{ + if (!pSink) + return false; + + LogFlowFunc(("[%s]: fStatus=0x%x\n", pSink->pszName, pSink->fStatus)); + + /* If the dirty flag is set, there is unprocessed data in the sink. */ + return pSink->fStatus; +} + uint8_t AudioMixerSinkGetStreamCount(PAUDMIXSINK pSink) { @@ -563,15 +579,4 @@ return pSink->cStreams; -} - -bool AudioMixerSinkHasData(PAUDMIXSINK pSink) -{ - if (!pSink) - return false; - - LogFlowFunc(("[%s]: %RTbool\n", pSink->pszName, (pSink->fFlags & AUDMIXSINK_FLAG_DIRTY))); - - /* If the dirty flag is set, there is unprocessed data in the sink. */ - return (pSink->fFlags & AUDMIXSINK_FLAG_DIRTY); } @@ -588,11 +593,17 @@ ("Can't read from a sink which is not an input sink\n")); +#ifndef VBOX_AUDIO_MIXER_WITH_MIXBUF uint8_t *pvMixBuf = (uint8_t *)RTMemAlloc(cbBuf); if (!pvMixBuf) return VERR_NO_MEMORY; +#endif int rc = VERR_NOT_FOUND; uint32_t cbRead = 0; + /* Flag indicating whether this sink is in a 'clean' state, + * e.g. there is no more data to read from. */ + bool fClean = true; + PAUDMIXSTREAM pMixStream; RTListForEach(&pSink->lstStreams, pMixStream, AUDMIXSTREAM, Node) @@ -602,5 +613,5 @@ uint32_t cbTotalRead = 0; - uint32_t cbToRead = cbBuf; + uint32_t cbToRead = cbBuf; while (cbToRead) @@ -608,6 +619,8 @@ uint32_t cbReadStrm; AssertPtr(pMixStream->pConn); +#ifndef VBOX_AUDIO_MIXER_WITH_MIXBUF rc = pMixStream->pConn->pfnStreamRead(pMixStream->pConn, pMixStream->pStream, (uint8_t *)pvMixBuf + cbTotalRead, cbToRead, &cbReadStrm); +#endif if ( RT_FAILURE(rc) || !cbReadStrm) @@ -625,17 +638,27 @@ cbRead = RT_MAX(cbRead, cbTotalRead); + + PDMAUDIOSTRMSTS strmSts = pMixStream->pConn->pfnStreamGetStatus(pMixStream->pConn, pMixStream->pStream); + fClean &= !(strmSts & PDMAUDIOSTRMSTS_FLAG_DATA_READABLE); } if (RT_SUCCESS(rc)) { - memcpy(pvBuf, pvMixBuf, cbRead); /* @todo Use an intermediate mixing buffer per sink! */ - + if (fClean) + pSink->fStatus &= ~AUDMIXSINK_STS_DIRTY; + +#ifndef VBOX_AUDIO_MIXER_WITH_MIXBUF + if (cbRead) + memcpy(pvBuf, pvMixBuf, cbRead); +#endif if (pcbRead) *pcbRead = cbRead; } +#ifndef VBOX_AUDIO_MIXER_WITH_MIXBUF RTMemFree(pvMixBuf); - - Log3Func(("[%s]: cbRead=%RU32, rc=%Rrc\n", pSink->pszName, cbRead, rc)); +#endif + + Log3Func(("[%s]: cbRead=%RU32, fStatus=0x%x, rc=%Rrc\n", pSink->pszName, cbRead, pSink->fStatus, rc)); return rc; } @@ -644,9 +667,12 @@ { AssertPtrReturn(pSink, VERR_INVALID_PARAMETER); - if (!pStream) + if ( !pStream + || !pStream->pSink) /* Not part of a sink anymore? */ + { return VERR_NOT_FOUND; + } AssertMsgReturn(pStream->pSink == pSink, ("Stream '%s' is not part of sink '%s'\n", - pStream->pszName, pSink->pszName), VERR_NOT_FOUND); + pStream->pszName, pSink->pszName), VERR_NOT_FOUND); LogFlowFunc(("[%s]: (Stream = %s), cStreams=%RU8\n", @@ -660,6 +686,4 @@ /* Remove stream from sink. */ RTListNodeRemove(&pStream->Node); - Assert(pSink->cStreams); - pSink->cStreams--; /* Set sink to NULL so that we know we're not part of any sink anymore. */ @@ -671,5 +695,10 @@ void AudioMixerSinkRemoveStream(PAUDMIXSINK pSink, PAUDMIXSTREAM pStream) { - audioMixerSinkRemoveStreamInternal(pSink, pStream); + int rc = audioMixerSinkRemoveStreamInternal(pSink, pStream); + if (RT_SUCCESS(rc)) + { + Assert(pSink->cStreams); + pSink->cStreams--; + } } @@ -763,35 +792,14 @@ } -void AudioMixerSinkTimerUpdate(PAUDMIXSINK pSink, uint64_t cTimerTicks, uint64_t cTicksElapsed, uint32_t *pcbData) +void AudioMixerSinkTimerUpdate(PAUDMIXSINK pSink, uint64_t cTimerTicks, uint64_t cTicksElapsed) { AssertPtrReturnVoid(pSink); - /* pcbData is optional. */ /* Note: cTimerTicks / cTicksElapsed = Hz elapsed. */ - LogFlowFunc(("[%s]: cTimerTicks=%RU64, cTicksElapsed=%RU64\n", pSink->pszName, cTimerTicks, cTicksElapsed)); - -// uint32_t cSamplesMin = (uint32_t)((2 * cTicksElapsed * pSink->PCMProps.uHz + cTimerTicks) / cTimerTicks / 2); - uint32_t cSamplesMin = (cTicksElapsed / pSink->PCMProps.uHz) * pSink->PCMProps.cChannels; - // cSamplesMin = (uint32_t)((2 * cTicksElapsed * 44100 + cTimerTicks) / cTimerTicks / 2); - uint32_t cbSamplesMin = _4K; //cSamplesMin << pSink->PCMProps.cShift; - - //Assert((cbSamplesMin % 2 == 0)); - -//LogFlowFunc(("[%s]: cSamplesMin=%RU32 (%RU32 bytes, %RU32Hz)\n", pSink->pszName, cSamplesMin, cbSamplesMin, pSink->PCMProps.uHz)); - - uint32_t cbData = cbSamplesMin; - - if (pSink->enmDir == AUDMIXSINKDIR_OUTPUT) - { -// uint32_t cSinkSamplesLive = AudioMixBufAvail(&pSink->MixBuf); -// if (!cSinkSamplesLive) -// cbData = AUDIOMIXBUF_S2B_RATIO(&pSink->MixBuf, AudioMixBufFree(&pSink->MixBufGuest)); - } + LogFlowFunc(("[%s]: cTimerTicks=%RU64, cTicksElapsed=%RU64 -> %zuHz\n", + pSink->pszName, cTimerTicks, cTicksElapsed, cTimerTicks / cTicksElapsed)); audioMixerSinkUpdateInternal(pSink); - - if (pcbData) - *pcbData = cbData; } @@ -810,26 +818,42 @@ AssertPtr(pStream); - uint32_t cPlayed = 0; - - rc = pMixStream->pConn->pfnStreamIterate(pMixStream->pConn, pStream); - if (RT_SUCCESS(rc)) + uint32_t cPlayed = 0; + uint32_t cCaptured = 0; + + int rc2 = pMixStream->pConn->pfnStreamIterate(pMixStream->pConn, pStream); + if (RT_SUCCESS(rc2)) { if (pStream->enmDir == PDMAUDIODIR_IN) { - /** @todo Implement this! */ -#if 0 - rc = pStream->pConn->pfnStreamCapture(pStream->pConn, NULL /* pcSamplesCaptured */); -#endif + rc = pMixStream->pConn->pfnStreamCapture(pMixStream->pConn, pMixStream->pStream, &cCaptured); + if (RT_FAILURE(rc2)) + { + LogFlowFunc(("%s: Failed capture stream '%s': %Rrc\n", pSink->pszName, pMixStream->pStream->szName, rc2)); + if (RT_SUCCESS(rc)) + rc = rc2; + } + + if (cCaptured) + pSink->fStatus |= AUDMIXSINK_STS_DIRTY; + } + else if (pStream->enmDir == PDMAUDIODIR_OUT) + { + rc2 = pMixStream->pConn->pfnStreamPlay(pMixStream->pConn, pMixStream->pStream, &cPlayed); + if (RT_FAILURE(rc2)) + { + LogFlowFunc(("%s: Failed playing stream '%s': %Rrc\n", pSink->pszName, pMixStream->pStream->szName, rc2)); + if (RT_SUCCESS(rc)) + rc = rc2; + } } else - { - rc = pMixStream->pConn->pfnStreamPlay(pMixStream->pConn, pMixStream->pStream, &cPlayed); - if (RT_FAILURE(rc)) - LogFlowFunc(("%s: Failed playing stream '%s': %Rrc\n", pSink->pszName, pMixStream->pStream->szName, rc)); - } + AssertFailedStmt(rc = VERR_NOT_IMPLEMENTED); } - Log3Func(("\t%s: cPlayed=%RU32, rc=%Rrc\n", pMixStream->pStream->szName, cPlayed, rc)); - } + Log3Func(("\t%s: cPlayed=%RU32, cCaptured=%RU32\n", pMixStream->pStream->szName, cPlayed, cCaptured)); + } + + if (RT_FAILURE(rc)) + LogFlowFunc(("Failed with rc=%Rrc\n", rc)); return rc; @@ -914,6 +938,9 @@ } - /* Set dirty bit. */ - pSink->fFlags |= AUDMIXSINK_FLAG_DIRTY; + if (cbBuf) + { + /* Set dirty bit. */ + pSink->fStatus |= AUDMIXSINK_STS_DIRTY; + } if (pcbWritten) @@ -975,5 +1002,5 @@ int rc; - if (pMixStream->pSink) + if (pMixStream->pSink) /* Is the stream part of a sink? */ { /* Save sink pointer, as after audioMixerSinkRemoveStreamInternal() the @@ -981,8 +1008,7 @@ PAUDMIXSINK pSink = pMixStream->pSink; - rc = audioMixerSinkRemoveStreamInternal(pMixStream->pSink, pMixStream); + rc = audioMixerSinkRemoveStreamInternal(pSink, pMixStream); if (RT_SUCCESS(rc)) { - AssertPtr(pSink); Assert(pSink->cStreams); pSink->cStreams--; @@ -994,4 +1020,6 @@ if (RT_SUCCESS(rc)) audioMixerStreamDestroyInternal(pMixStream); + + LogFlowFunc(("Returning %Rrc\n", rc)); } Index: /trunk/src/VBox/Devices/Audio/AudioMixer.h =================================================================== --- /trunk/src/VBox/Devices/Audio/AudioMixer.h (revision 61522) +++ /trunk/src/VBox/Devices/Audio/AudioMixer.h (revision 61523) @@ -68,8 +68,18 @@ } AUDMIXSTREAM, *PAUDMIXSTREAM; -/** No flags specified. */ -#define AUDMIXSINK_FLAG_NONE 0 -/** Dirty flag. */ -#define AUDMIXSINK_FLAG_DIRTY RT_BIT(0) +/** Defines an audio sink's current status. */ +#define AUDMIXSINKSTS uint32_t + +/** No status specified. */ +#define AUDMIXSINK_STS_NONE 0 +/** The sink is active and running. */ +#define AUDMIXSINK_STS_RUNNING RT_BIT(0) +/** Dirty flag. + * For output sinks this means that there is data in the + * sink which has not been played yet. + * For input sinks this means that there is data in the + * sink which has been recorded but not transferred to the + * destination yet. */ +#define AUDMIXSINK_STS_DIRTY RT_BIT(1) /** @@ -117,6 +127,6 @@ * if this sink handles input or output. */ AUDMIXSINKDIR enmDir; - /** Sink flags of type AUDMIXSINK_FLAG_. */ - uint32_t fFlags; + /** Sink status of type AUDMIXSINK_STS_XXX. */ + AUDMIXSINKSTS fStatus; /** The sink's PCM format. */ PDMPCMPROPS PCMProps; @@ -170,6 +180,6 @@ AUDMIXSINKDIR AudioMixerSinkGetDir(PAUDMIXSINK pSink); PAUDMIXSTREAM AudioMixerSinkGetStream(PAUDMIXSINK pSink, uint8_t uIndex); +AUDMIXSINKSTS AudioMixerSinkGetStatus(PAUDMIXSINK pSink); uint8_t AudioMixerSinkGetStreamCount(PAUDMIXSINK pSink); -bool AudioMixerSinkHasData(PAUDMIXSINK pSink); int AudioMixerSinkRead(PAUDMIXSINK pSink, AUDMIXOP enmOp, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead); void AudioMixerSinkRemoveStream(PAUDMIXSINK pSink, PAUDMIXSTREAM pStream); @@ -177,5 +187,5 @@ int AudioMixerSinkSetFormat(PAUDMIXSINK pSink, PPDMPCMPROPS pPCMProps); int AudioMixerSinkSetVolume(PAUDMIXSINK pSink, PPDMAUDIOVOLUME pVol); -void AudioMixerSinkTimerUpdate(PAUDMIXSINK pSink, uint64_t cTimerTicks, uint64_t cTicksElapsed, uint32_t *pcbData); +void AudioMixerSinkTimerUpdate(PAUDMIXSINK pSink, uint64_t cTimerTicks, uint64_t cTicksElapsed); int AudioMixerSinkWrite(PAUDMIXSINK pSink, AUDMIXOP enmOp, const void *pvBuf, uint32_t cbBuf, uint32_t *pcbWritten); int AudioMixerSinkUpdate(PAUDMIXSINK pSink); Index: /trunk/src/VBox/Devices/Audio/DevIchAc97.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DevIchAc97.cpp (revision 61522) +++ /trunk/src/VBox/Devices/Audio/DevIchAc97.cpp (revision 61523) @@ -46,10 +46,17 @@ *********************************************************************************************************************************/ -#ifdef DEBUG -//#define DEBUG_LUN -# ifdef DEBUG_LUN -# define DEBUG_LUN_NUM 1 +#ifdef DEBUG_andy +/* + * AC97_DEBUG_DUMP_PCM_DATA enables dumping the raw PCM data + * to a file on the host. Be sure to adjust AC97_DEBUG_DUMP_PCM_DATA_PATH + * to your needs before using this! + */ +# define AC97_DEBUG_DUMP_PCM_DATA +# ifdef RT_OS_WINDOWS +# define AC97_DEBUG_DUMP_PCM_DATA_PATH "c:\\temp\\" +# else +# define AC97_DEBUG_DUMP_PCM_DATA_PATH "/tmp/" # endif -#endif /* DEBUG */ +#endif /* DEBUG_andy */ #define AC97_SSM_VERSION 1 @@ -76,7 +83,7 @@ #define CR_IOCE RT_BIT(4) /* rw, Interrupt On Completion Enable. */ #define CR_FEIE RT_BIT(3) /* rw FIFO Error Interrupt Enable. */ -#define CR_LVBIE RT_BIT(2) /* rw */ -#define CR_RR RT_BIT(1) /* rw */ -#define CR_RPBM RT_BIT(0) /* rw */ +#define CR_LVBIE RT_BIT(2) /* rw Last Valid Buffer Interrupt Enable. */ +#define CR_RR RT_BIT(1) /* rw Reset Registers. */ +#define CR_RPBM RT_BIT(0) /* rw Run/Pause Bus Master. */ #define CR_VALID_MASK (RT_BIT(5) - 1) #define CR_DONT_CLEAR_MASK (CR_IOCE | CR_FEIE | CR_LVBIE) @@ -263,5 +270,5 @@ uint8_t lvi; /** rw 0, Last valid index. */ uint16_t sr; /** rw 1, Status register. */ - uint16_t picb; /** ro 0, Position in current buffer. */ + uint16_t picb; /** ro 0, Position in current buffer (in samples). */ uint8_t piv; /** ro 0, Prefetched index value. */ uint8_t cr; /** rw 0, Control register. */ @@ -423,5 +430,5 @@ static DECLCALLBACK(void) ichac97Timer(PPDMDEVINS pDevIns, PTMTIMER pTimer, void *pvUser); #endif -static int ichac97TransferAudio(PAC97STATE pThis, PAC97STREAM pStream, uint32_t cbElapsed); +static int ichac97TransferAudio(PAC97STATE pThis, PAC97STREAM pStream); static void ichac97WarmReset(PAC97STATE pThis) @@ -532,5 +539,5 @@ AssertPtrReturn(pStream, VERR_INVALID_POINTER); - bool fActive = AudioMixerSinkHasData(ichac97IndexToSink(pThis, pStream->u8Strm)); + bool fActive = AudioMixerSinkGetStatus(ichac97IndexToSink(pThis, pStream->u8Strm)); LogFlowFunc(("SD=%RU8, fActive=%RTbool\n", pStream->u8Strm, fActive)); @@ -1145,9 +1152,9 @@ * @param pcbWritten */ -static int ichac97WriteAudio(PAC97STATE pThis, PAC97STREAM pStream, uint32_t cbMax, uint32_t *pcbWritten) +static int ichac97WriteAudio(PAC97STATE pThis, PAC97STREAM pStream, uint32_t cbToWrite, uint32_t *pcbWritten) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); - AssertReturn(cbMax, VERR_INVALID_PARAMETER); + AssertReturn(cbToWrite, VERR_INVALID_PARAMETER); /* pcbWritten is optional. */ @@ -1158,7 +1165,7 @@ uint32_t cbWrittenTotal = 0; - Log3Func(("PICB=%RU16, cbMax=%RU32\n", pRegs->picb, cbMax)); - - uint32_t cbToWrite = RT_MIN((uint32_t)(pRegs->picb << 1), cbMax); /** @todo r=andy Assumes 16bit sample size. */ + Log3Func(("PICB=%RU16, cbMax=%RU32\n", pRegs->picb, cbToWrite)); + + cbToWrite = RT_MIN((uint32_t)(pRegs->picb << 1), cbToWrite); /** @todo r=andy Assumes 16bit sample size. */ if (!cbToWrite) { @@ -1170,5 +1177,5 @@ int rc = VINF_SUCCESS; - LogFlowFunc(("pReg=%p, cbMax=%RU32, cbToWrite=%RU32\n", pRegs, cbMax, cbToWrite)); + LogFlowFunc(("pRegs=%p, cbMax=%RU32, cbToWrite=%RU32\n", pRegs, cbToWrite, cbToWrite)); Assert(pStream->State.offFIFOW <= pStream->State.cbFIFOW); @@ -1182,7 +1189,7 @@ PDMDevHlpPhysRead(pDevIns, addr, pu8FIFOW, cbToRead); /** @todo r=andy Check rc? */ -#if defined (RT_OS_LINUX) && defined(DEBUG_andy) +#ifdef AC97_DEBUG_DUMP_PCM_DATA RTFILE fh; - RTFileOpen(&fh, "/tmp/ac97WriteAudio.pcm", + RTFileOpen(&fh, AC97_DEBUG_DUMP_PCM_DATA_PATH "ac97WriteAudio.pcm", RTFILE_O_OPEN_CREATE | RTFILE_O_APPEND | RTFILE_O_WRITE | RTFILE_O_DENY_NONE); RTFileWrite(fh, pu8FIFOW, cbToRead, NULL); @@ -1234,4 +1241,6 @@ static void ichac97WriteBUP(PAC97STATE pThis, uint32_t cbElapsed) { + LogFlowFunc(("cbElapsed=%RU32\n", cbElapsed)); + if (!(pThis->bup_flag & BUP_SET)) { @@ -1240,5 +1249,5 @@ unsigned int i; uint32_t *p = (uint32_t*)pThis->silence; - for (i = 0; i < sizeof(pThis->silence) / 4; i++) + for (i = 0; i < sizeof(pThis->silence) / 4; i++) /** @todo r=andy Assumes 16-bit samples, stereo. */ *p++ = pThis->last_samp; } @@ -1269,9 +1278,9 @@ } -static int ichac97ReadAudio(PAC97STATE pThis, PAC97STREAM pStream, uint32_t cbMax, uint32_t *pcbRead) +static int ichac97ReadAudio(PAC97STATE pThis, PAC97STREAM pStream, uint32_t cbToRead, uint32_t *pcbRead) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); - AssertReturn(cbMax, VERR_INVALID_PARAMETER); + AssertReturn(cbToRead, VERR_INVALID_PARAMETER); /* pcbRead is optional. */ @@ -1285,6 +1294,7 @@ uint32_t cbRead = 0; - uint32_t cbToRead = RT_MIN((uint32_t)(pRegs->picb << 1), - RT_MIN(pStream->State.cbFIFOW, cbMax)); /** @todo r=andy Assumes 16bit samples. */ + + cbToRead = RT_MIN((uint32_t)(pRegs->picb << 1), + RT_MIN(pStream->State.cbFIFOW, cbToRead)); /** @todo r=andy Assumes 16bit samples. */ if (!cbToRead) @@ -1365,28 +1375,30 @@ uint64_t cTicksPerSec = TMTimerGetFreq(pTimer); + LogFlowFuncEnter(); + /* Update current time timestamp. */ pThis->uTimerTS = cTicksNow; - LogFlowFuncEnter(); - - uint32_t cbLineIn; - AudioMixerSinkTimerUpdate(pThis->pSinkLineIn, pThis->cTimerTicks, cTicksPerSec, &cbLineIn); - if (cbLineIn) - ichac97TransferAudio(pThis, &pThis->StreamLineIn, cbLineIn); - - uint32_t cbMicIn; - AudioMixerSinkTimerUpdate(pThis->pSinkMicIn , pThis->cTimerTicks, cTicksPerSec, &cbMicIn); - if (cbMicIn) - ichac97TransferAudio(pThis, &pThis->StreamMicIn, cbMicIn); - - uint32_t cbOut; - AudioMixerSinkTimerUpdate(pThis->pSinkOutput, pThis->cTimerTicks, cTicksPerSec, &cbOut); - if (cbOut) - ichac97TransferAudio(pThis, &pThis->StreamOut, cbOut); + /* Flag indicating whether to kick the timer again for a + * new data processing round. */ + bool fKickTimer = false; + + AudioMixerSinkTimerUpdate(pThis->pSinkLineIn, pThis->cTimerTicks, cTicksPerSec); + ichac97TransferAudio(pThis, &pThis->StreamLineIn); + + fKickTimer = AudioMixerSinkGetStatus(pThis->pSinkLineIn) & AUDMIXSINK_STS_DIRTY; + + AudioMixerSinkTimerUpdate(pThis->pSinkMicIn , pThis->cTimerTicks, cTicksPerSec); + ichac97TransferAudio(pThis, &pThis->StreamMicIn); + + fKickTimer = AudioMixerSinkGetStatus(pThis->pSinkMicIn) & AUDMIXSINK_STS_DIRTY; + + AudioMixerSinkTimerUpdate(pThis->pSinkOutput, pThis->cTimerTicks, cTicksPerSec); + ichac97TransferAudio(pThis, &pThis->StreamOut); + + fKickTimer = AudioMixerSinkGetStatus(pThis->pSinkOutput) & AUDMIXSINK_STS_DIRTY; if ( ASMAtomicReadBool(&pThis->fTimerActive) - || AudioMixerSinkHasData(pThis->pSinkLineIn) - || AudioMixerSinkHasData(pThis->pSinkMicIn) - || AudioMixerSinkHasData(pThis->pSinkOutput)) + || fKickTimer) { /* Kick the timer again. */ @@ -1401,7 +1413,7 @@ #endif /* !VBOX_WITH_AUDIO_CALLBACKS */ -static int ichac97TransferAudio(PAC97STATE pThis, PAC97STREAM pStream, uint32_t cbElapsed) -{ - Log3Func(("SD=%RU8, cbElapsed=%RU32\n", pStream->u8Strm , cbElapsed)); +static int ichac97TransferAudio(PAC97STATE pThis, PAC97STREAM pStream) +{ + Log3Func(("SD=%RU8\n", pStream->u8Strm)); PAC97BMREGS pRegs = &pStream->Regs; @@ -1409,10 +1421,10 @@ if (pRegs->sr & SR_DCH) /* Controller halted? */ { - if (pRegs->cr & CR_RPBM) + if (pRegs->cr & CR_RPBM) /* Bus master operation starts. */ { switch (pStream->u8Strm) { case PO_INDEX: - ichac97WriteBUP(pThis, cbElapsed); + ichac97WriteBUP(pThis, (uint32_t)(pRegs->picb << 1)); break; @@ -1432,5 +1444,5 @@ uint32_t cbWrittenTotal = 0; - while (cbElapsed >> 1) /** @todo r=andy This assumes (hardcodes) 16bit sample size.*/ + do { if (!pRegs->bd_valid) @@ -1446,5 +1458,5 @@ if (pRegs->civ == pRegs->lvi) { - pRegs->sr |= SR_DCH; /* CELV? */ + pRegs->sr |= SR_DCH; /** @todo r=andy Also set CELV? */ pThis->bup_flag = 0; @@ -1461,17 +1473,17 @@ } - uint32_t cbTransferred; + uint32_t cbToTransfer, cbTransferred; switch (pStream->u8Strm) { case PO_INDEX: { - rc = ichac97WriteAudio(pThis, pStream, cbElapsed, &cbTransferred); + cbToTransfer = (uint32_t)(pRegs->picb << 1); + + rc = ichac97WriteAudio(pThis, pStream, cbToTransfer, &cbTransferred); if ( RT_SUCCESS(rc) && cbTransferred) { cbWrittenTotal += cbTransferred; - Assert(cbElapsed >= cbTransferred); - cbElapsed -= cbTransferred; - Assert((cbTransferred & 1) == 0); /* Else the following shift won't work */ + Assert((cbTransferred & 1) == 0); /* Else the following shift won't work */ pRegs->picb -= (cbTransferred >> 1); /** @todo r=andy Assumes 16bit samples. */ } @@ -1482,11 +1494,11 @@ case MC_INDEX: { - rc = ichac97ReadAudio(pThis, pStream, cbElapsed, &cbTransferred); + cbToTransfer = (uint32_t)(pRegs->picb << 1); + + rc = ichac97ReadAudio(pThis, pStream, cbToTransfer, &cbTransferred); if ( RT_SUCCESS(rc) && cbTransferred) { - Assert(cbElapsed >= cbTransferred); - cbElapsed -= cbTransferred; - Assert((cbTransferred & 1) == 0); /* Else the following shift won't work */ + Assert((cbTransferred & 1) == 0); /* Else the following shift won't work */ pRegs->picb -= (cbTransferred >> 1); /** @todo r=andy Assumes 16bit samples. */ } @@ -1532,10 +1544,8 @@ } - if ( RT_FAILURE(rc) - || rc == VINF_EOF) /* All data processed? */ - { - break; - } - } + if (rc == VINF_EOF) /* All data processed? */ + break; + + } while (RT_SUCCESS(rc)); LogFlowFuncLeaveRC(rc); @@ -1549,5 +1559,5 @@ uint32_t *pu32Val, unsigned cbVal) { - PAC97STATE pThis = (PAC97STATE)pvUser; + PAC97STATE pThis = (PAC97STATE)pvUser; /* Get the index of the NABMBAR port. */ Index: /trunk/src/VBox/Devices/Audio/DevIchHda.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DevIchHda.cpp (revision 61522) +++ /trunk/src/VBox/Devices/Audio/DevIchHda.cpp (revision 61523) @@ -58,4 +58,16 @@ #ifdef DEBUG_andy +/* + * HDA_DEBUG_DUMP_PCM_DATA enables dumping the raw PCM data + * to a file on the host. Be sure to adjust HDA_DEBUG_DUMP_PCM_DATA_PATH + * to your needs before using this! + */ +# define HDA_DEBUG_DUMP_PCM_DATA +# ifdef RT_OS_WINDOWS +# define HDA_DEBUG_DUMP_PCM_DATA_PATH "c:\\temp\\" +# else +# define HDA_DEBUG_DUMP_PCM_DATA_PATH "/tmp/" +# endif + /* Enables experimental support for separate mic-in handling. Do not enable this yet for regular builds, as this needs more testing first! */ @@ -930,5 +942,5 @@ static int hdaStreamStop(PHDASTREAM pStream); static int hdaStreamWaitForStateChange(PHDASTREAM pStream, RTMSINTERVAL msTimeout); -static int hdaTransfer(PHDASTATE pThis, PHDASTREAM pStream, uint32_t cbToProcess, uint32_t *pcbProcessed); +static int hdaTransfer(PHDASTATE pThis, PHDASTREAM pStream); #endif @@ -1756,9 +1768,12 @@ AssertPtrReturn(pStream, VERR_INVALID_POINTER); - bool fActive = false; - - AssertPtr(pStream->pMixSink); - if (pStream->pMixSink->pMixSink) - fActive = AudioMixerSinkHasData(pStream->pMixSink->pMixSink); + bool fActive = pStream->State.fActive; + + if (!fActive) + { + AssertPtr(pStream->pMixSink); + if (pStream->pMixSink->pMixSink) + fActive = AudioMixerSinkGetStatus(pStream->pMixSink->pMixSink) & AUDMIXSINK_STS_DIRTY; + } LogFlowFunc(("SD=%RU8, fActive=%RTbool\n", pStream->u8SD, fActive)); @@ -1780,7 +1795,9 @@ ? AUDMIXSINKCMD_ENABLE : AUDMIXSINKCMD_DISABLE; - /* First, enable or disable the stream's sink, if any. */ + /* First, enable or disable the stream and the stream's sink, if any. */ if (pStream->pMixSink->pMixSink) AudioMixerSinkCtl(pStream->pMixSink->pMixSink, enmCmd); + + pStream->State.fActive = fActive; /* Second, see if we need to start or stop the timer. */ @@ -2723,5 +2740,5 @@ AssertReturn(pCfg->enmDir == PDMAUDIODIR_IN, VERR_INVALID_PARAMETER); - LogFlowFunc(("Stream=%s\n", pCfg->szName)); + LogFlowFunc(("Stream=%s, Source=%ld\n", pCfg->szName, pCfg->DestSource.Source)); int rc; @@ -2731,6 +2748,4 @@ case PDMAUDIORECSOURCE_LINE: { - pCfg->DestSource.Source = PDMAUDIORECSOURCE_LINE; - pCfg->cChannels = 2; rc = hdaCodecRemoveStream(pThis->pCodec, PDMAUDIOMIXERCTL_LINE_IN); if (RT_SUCCESS(rc)) @@ -2741,6 +2756,4 @@ case PDMAUDIORECSOURCE_MIC: { - pCfg->DestSource.Source = PDMAUDIORECSOURCE_MIC; - pCfg->cChannels = 2; rc = hdaCodecRemoveStream(pThis->pCodec, PDMAUDIOMIXERCTL_MIC_IN); if (RT_SUCCESS(rc)) @@ -2785,4 +2798,23 @@ /* Set audio direction. */ strmCfg.enmDir = hdaGetDirFromSD(pStream->u8SD); + switch (strmCfg.enmDir) + { + case PDMAUDIODIR_IN: +#ifdef VBOX_WITH_HDA_MIC_IN +# error "Implement me!" +#else + strmCfg.DestSource.Source = PDMAUDIORECSOURCE_LINE; +#endif + break; + + case PDMAUDIODIR_OUT: + /* Destination(s) will be set in hdaAddStreamOut(), + * based on the channels / stream layout. */ + break; + + default: + rc = VERR_NOT_SUPPORTED; + break; + } /* @@ -2794,6 +2826,9 @@ * number of channels in a single audio stream. */ - rc = hdaStreamMapInit(&pStream->State.Mapping, &strmCfg); - AssertRC(rc); + if (RT_SUCCESS(rc)) + { + rc = hdaStreamMapInit(&pStream->State.Mapping, &strmCfg); + AssertRC(rc); + } if (RT_SUCCESS(rc)) @@ -3169,5 +3204,5 @@ /* Make sure we only copy as much as the stream's FIFO can hold (SDFIFOS, 18.2.39). */ - // cbFree = RT_MIN(cbFree, uint32_t(pStream->u16FIFOS)); + cbFree = RT_MIN(cbFree, uint32_t(pStream->u16FIFOS)); /* Make sure we only transfer as many bytes as requested. */ @@ -3447,9 +3482,9 @@ * but "reports bytes" when all conditions are met (FIFOW). */ -static int hdaReadAudio(PHDASTATE pThis, PHDASTREAM pStream, uint32_t cbMax, uint32_t *pcbRead) +static int hdaReadAudio(PHDASTATE pThis, PHDASTREAM pStream, uint32_t cbToProcess, uint32_t *pcbProcessed) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); - /* pcbRead is optional. */ + /* pcbProcessed is optional. */ int rc; @@ -3460,8 +3495,5 @@ PHDABDLE pBDLE = &pStream->State.BDLE; - uint32_t cbBuf = hdaStreamGetTransferSize(pThis, pStream, cbMax); - Log3Func(("cbBuf=%RU32, %R[bdle]\n", cbBuf, pBDLE)); - - if (!cbBuf) + if (!cbToProcess) { rc = VINF_EOF; @@ -3471,5 +3503,5 @@ AssertPtr(pStream->pMixSink); AssertPtr(pStream->pMixSink->pMixSink); - rc = AudioMixerSinkRead(pStream->pMixSink->pMixSink, AUDMIXOP_BLEND, pBDLE->State.au8FIFO, cbBuf, &cbRead); + rc = AudioMixerSinkRead(pStream->pMixSink->pMixSink, AUDMIXOP_BLEND, pBDLE->State.au8FIFO, cbToProcess, &cbRead); if (RT_FAILURE(rc)) break; @@ -3482,5 +3514,6 @@ /* Sanity checks. */ - Assert(cbRead <= cbBuf); + Assert(cbRead <= cbToProcess); + Assert(cbRead <= sizeof(pBDLE->State.au8FIFO)); Assert(cbRead <= pBDLE->u32BufSize - pBDLE->State.u32BufOff); @@ -3515,17 +3548,18 @@ if (RT_SUCCESS(rc)) { - if (pcbRead) - *pcbRead = cbRead; - } - - Log3Func(("Returning cbRead=%RU32, rc=%Rrc\n", cbRead, rc)); + if (pcbProcessed) + *pcbProcessed = cbRead; + } + + if (RT_FAILURE(rc)) + LogFlowFunc(("Failed with %Rrc\n", rc)); + return rc; } -static int hdaWriteAudio(PHDASTATE pThis, PHDASTREAM pStream, uint32_t cbMax, uint32_t *pcbWritten) +static int hdaWriteAudio(PHDASTATE pThis, PHDASTREAM pStream, uint32_t cbToProcess, uint32_t *pcbProcessed) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); - AssertPtrReturn(pcbWritten, VERR_INVALID_POINTER); /* pcbWritten is optional. */ @@ -3533,7 +3567,4 @@ uint32_t cbWritten = 0; - uint32_t cbToWrite = hdaStreamGetTransferSize(pThis, pStream, cbMax); - - Log3Func(("cbToWrite=%RU32, %R[bdle]\n", cbToWrite, pBDLE)); /* @@ -3542,5 +3573,5 @@ */ int rc; - if (!cbToWrite) + if (!cbToProcess) { rc = VINF_EOF; @@ -3549,6 +3580,6 @@ { void *pvBuf = pBDLE->State.au8FIFO + pBDLE->State.cbBelowFIFOW; - Assert(cbToWrite >= pBDLE->State.cbBelowFIFOW); - uint32_t cbBuf = cbToWrite - pBDLE->State.cbBelowFIFOW; + Assert(cbToProcess >= pBDLE->State.cbBelowFIFOW); + uint32_t cbBuf = cbToProcess - pBDLE->State.cbBelowFIFOW; /* @@ -3559,7 +3590,8 @@ pvBuf, cbBuf); AssertRC(rc); -#if defined (RT_OS_LINUX) && defined(DEBUG_andy) + +#ifdef HDA_DEBUG_DUMP_PCM_DATA RTFILE fh; - RTFileOpen(&fh, "/tmp/hdaWriteAudio-hda.pcm", + RTFileOpen(&fh, HDA_DEBUG_DUMP_PCM_DATA_PATH "hdaWriteAudio-hda.pcm", RTFILE_O_OPEN_CREATE | RTFILE_O_APPEND | RTFILE_O_WRITE | RTFILE_O_DENY_NONE); RTFileWrite(fh, pvBuf, cbBuf, NULL); @@ -3642,7 +3674,7 @@ /* Always report all data as being written; * backends who were not able to catch up have to deal with it themselves. */ - cbWritten = cbToWrite; - - hdaBDLEUpdate(pBDLE, cbToWrite, cbWritten); + cbWritten = cbToProcess; + + hdaBDLEUpdate(pBDLE, cbToProcess, cbWritten); } else @@ -3663,9 +3695,11 @@ if (RT_SUCCESS(rc)) { - if (pcbWritten) - *pcbWritten = cbWritten; - } - - Log3Func(("Returning cbMax=%RU32, cbWritten=%RU32, rc=%Rrc\n", cbMax, cbWritten, rc)); + if (pcbProcessed) + *pcbProcessed = cbWritten; + } + + if (RT_FAILURE(rc)) + LogFlowFunc(("Failed with %Rrc\n", rc)); + return rc; } @@ -3904,9 +3938,8 @@ } - if (pMixStream) - { - AudioMixerSinkRemoveStream(pSink->pMixSink, pMixStream); - AudioMixerStreamDestroy(pMixStream); - } + AssertPtr(pMixStream); + + AudioMixerSinkRemoveStream(pSink->pMixSink, pMixStream); + AudioMixerStreamDestroy(pMixStream); } @@ -4068,6 +4101,12 @@ uint64_t cTicksElapsed = cTicksNow - pThis->uTimerTS; + LogFlowFuncEnter(); + /* Update current time timestamp. */ pThis->uTimerTS = cTicksNow; + + /* Flag indicating whether to kick the timer again for a + * new data processing round. */ + bool fKickTimer = false; PHDASTREAM pStreamLineIn = hdaGetStreamFromSink(pThis, &pThis->SinkLineIn); @@ -4080,25 +4119,31 @@ #endif - uint32_t cbLineIn; - AudioMixerSinkTimerUpdate(pThis->SinkLineIn.pMixSink, pThis->cTimerTicks, cTicksElapsed, &cbLineIn); - if (cbLineIn) - hdaTransfer(pThis, pStreamLineIn, cbLineIn, NULL); /** @todo Add rc! */ + AudioMixerSinkTimerUpdate(pThis->SinkLineIn.pMixSink, pThis->cTimerTicks, cTicksElapsed); + hdaTransfer(pThis, pStreamLineIn); + + if (AudioMixerSinkGetStatus(pThis->SinkLineIn.pMixSink) & AUDMIXSINK_STS_DIRTY) + fKickTimer = true; #ifdef VBOX_WITH_HDA_MIC_IN - uint32_t cbMicIn; - AudioMixerSinkTimerUpdate(pThis->SinkMicIn.pMixSink , pThis->cTimerTicks, cTicksElapsed, &cbMicIn); -#endif - - uint32_t cbFront; - AudioMixerSinkTimerUpdate(pThis->SinkFront.pMixSink, pThis->cTimerTicks, cTicksElapsed, &cbFront); + AudioMixerSinkTimerUpdate(pThis->SinkMicIn.pMixSink, pThis->cTimerTicks, cTicksElapsed); + hdaTransfer(pThis, pStreamMicIn); + + if (AudioMixerSinkGetStatus(pThis->SinkLineIn.pMixSink) & AUDMIXSINK_STS_DIRTY) + fKickTimer = true; +#endif + + AudioMixerSinkTimerUpdate(pThis->SinkFront.pMixSink, pThis->cTimerTicks, cTicksElapsed); + hdaTransfer(pThis, pStreamFront); + + if (AudioMixerSinkGetStatus(pThis->SinkFront.pMixSink) & AUDMIXSINK_STS_DIRTY) + fKickTimer = true; + #ifdef VBOX_WITH_HDA_51_SURROUND - uint32_t cbCenterLFE; - AudioMixerSinkTimerUpdate(pThis->SinkCenterLFE.pMixSink, pThis->cTimerTicks, cTicksElapsed, &cbCenterLFE); - uint32_t cbRear; - AudioMixerSinkTimerUpdate(pThis->SinkRear.pMixSink, pThis->cTimerTicks, cTicksElapsed, &cbRear); -#endif - - if (cbFront) - hdaTransfer(pThis, pStreamFront, cbFront, NULL); /** @todo Add rc! */ + AudioMixerSinkTimerUpdate(pThis->SinkCenterLFE.pMixSink, pThis->cTimerTicks, cTicksElapsed); + AudioMixerSinkTimerUpdate(pThis->SinkRear.pMixSink, pThis->cTimerTicks, cTicksElapsed); + + /** @todo Check for stream interleaving and only call hdaTransfer() if required! */ +#endif + #ifdef VBOX_WITH_HDA_51_SURROUND /* @@ -4110,10 +4155,5 @@ if ( ASMAtomicReadBool(&pThis->fTimerActive) - || AudioMixerSinkHasData(pThis->SinkFront.pMixSink) -#ifdef VBOX_WITH_HDA_51_SURROUND - || AudioMixerSinkHasData(pThis->SinkCenterLFE.pMixSink) - || AudioMixerSinkHasData(pThis->SinkRear.pMixSink) -#endif - || AudioMixerSinkHasData(pThis->SinkLineIn.pMixSink)) + || fKickTimer) { /* Kick the timer again. */ @@ -4122,4 +4162,6 @@ TMTimerSet(pThis->pTimer, cTicksNow + cTicks); } + + LogFlowFuncLeave(); STAM_PROFILE_STOP(&pThis->StatTimer, a); @@ -4176,16 +4218,12 @@ #endif /* VBOX_WITH_AUDIO_CALLBACKS */ -static int hdaTransfer(PHDASTATE pThis, PHDASTREAM pStream, uint32_t cbToProcess, uint32_t *pcbProcessed) +static int hdaTransfer(PHDASTATE pThis, PHDASTREAM pStream) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); - /* pcbProcessed is optional. */ if (ASMAtomicReadBool(&pThis->fInReset)) /* HDA controller in reset mode? Bail out. */ { - LogFlowFunc(("In reset mode, skipping\n")); - - if (pcbProcessed) - *pcbProcessed = 0; + LogFlowFunc(("HDA in reset mode, skipping\n")); return VINF_SUCCESS; } @@ -4196,6 +4234,9 @@ return rc; + Log3Func(("[SD%RU8] fActive=%RTbool\n", pStream->u8SD, pStream->State.fActive)); + /* Stop request received? */ - if (pStream->State.fDoStop) + if ( !pStream->State.fActive + || pStream->State.fDoStop) { pStream->State.fActive = false; @@ -4209,7 +4250,4 @@ else if (!(HDA_STREAM_REG(pThis, CTL, pStream->u8SD) & HDA_REG_FIELD_FLAG_MASK(SDCTL, RUN))) fProceed = false; - /* Nothing to process? */ - else if (!cbToProcess) - fProceed = false; if ((HDA_STREAM_REG(pThis, STS, pStream->u8SD) & HDA_REG_FIELD_FLAG_MASK(SDSTS, BCIS))) @@ -4221,11 +4259,9 @@ if (!fProceed) { - Log3Func(("[SD%RU8] Skipping\n", pStream->u8SD)); + Log3Func(("[SD%RU8]: Skipping\n", pStream->u8SD)); rc = RTSemMutexRelease(pStream->State.hMtx); AssertRC(rc); - if (pcbProcessed) - *pcbProcessed = 0; return VINF_SUCCESS; } @@ -4242,22 +4278,23 @@ Assert(u32LPIB <= pStream->u32CBL); - uint32_t cbLeft = cbToProcess; - uint32_t cbTotal = 0; - bool fInterrupt = false; - Log3Func(("cbLeft=%RU32\n", cbLeft)); - -#define FOO - -#ifdef FOO +#ifdef DEBUG_andy +# define DEBUG_SIMPLE +#endif + +#ifdef DEBUG_SIMPLE uint8_t u8FIFO[_16K+1]; size_t u8FIFOff = 0; #endif + uint32_t cbToProcess = 0; + uint32_t cbProcessed = 0; + uint32_t cbProcessedTotal = 0; + /* Set the FIFORDY bit on the stream while doing the transfer. */ HDA_STREAM_REG(pThis, STS, pStream->u8SD) |= HDA_REG_FIELD_FLAG_MASK(SDSTS, FIFORDY); - while (cbLeft) + do { /* Do we need to fetch the next Buffer Descriptor Entry (BDLE)? */ @@ -4269,13 +4306,15 @@ } - uint32_t cbProcessed = 0; + cbToProcess = hdaStreamGetTransferSize(pThis, pStream, _4K /** @todo Fix this */); + cbProcessed = 0; + if (hdaGetDirFromSD(pStream->u8SD) == PDMAUDIODIR_IN) - rc = hdaReadAudio (pThis, pStream, cbLeft, &cbProcessed); + rc = hdaReadAudio(pThis, pStream, cbToProcess, &cbProcessed); else { -#ifndef FOO - rc = hdaWriteAudio(pThis, pStream, cbLeft, &cbProcessed); +#ifndef DEBUG_SIMPLE + rc = hdaWriteAudio(pThis, pStream, cbToProcess, &cbProcessed); #else - uint32_t cbToWrite = hdaStreamGetTransferSize(pThis, pStream, cbLeft); + uint32_t cbToWrite = hdaStreamGetTransferSize(pThis, pStream, cbToProcess); void *pvBuf = u8FIFO + u8FIFOff; @@ -4290,7 +4329,6 @@ hdaBDLEUpdate(pBDLE, cbToWrite, cbToWrite); - LogFlowFunc(("u8FIFOff=%zu, cbLeft=%RU32, cbToWrite=%RU32\n", u8FIFOff, cbLeft, cbToWrite)); - u8FIFOff += cbToWrite; + Assert((u8FIFOff & 1) == 0); Assert(u8FIFOff <= sizeof(u8FIFO)); @@ -4304,10 +4342,5 @@ hdaStreamTransferUpdate(pThis, pStream, cbProcessed); - Assert(cbLeft >= cbProcessed); - cbLeft -= cbProcessed; - cbTotal += cbProcessed; - - LogFlowFunc(("cbProcessed=%RU32, cbLeft=%RU32, cbTotal=%RU32, rc=%Rrc\n", - cbProcessed, cbLeft, cbTotal, rc)); + cbProcessedTotal += cbProcessed; if (rc == VINF_EOF) @@ -4316,24 +4349,25 @@ if (hdaStreamTransferIsComplete(pThis, pStream, &fInterrupt)) break; - } + + } while (RT_SUCCESS(rc)); /* Remove the FIFORDY bit again. */ HDA_STREAM_REG(pThis, STS, pStream->u8SD) &= ~HDA_REG_FIELD_FLAG_MASK(SDSTS, FIFORDY); -#ifdef FOO - #if defined (RT_OS_LINUX) && defined(DEBUG_andy) + LogFlowFunc(("[SD%RU8]: %RU32 / %RU32, rc=%Rrc\n", pStream->u8SD, cbProcessedTotal, cbToProcess, rc)); + +#ifdef DEBUG_SIMPLE +# ifdef HDA_DEBUG_DUMP_PCM_DATA RTFILE fh; - RTFileOpen(&fh, "/tmp/hdaWriteAudio.pcm", + RTFileOpen(&fh, HDA_DEBUG_DUMP_PCM_DATA_PATH "hdaWriteAudio.pcm", RTFILE_O_OPEN_CREATE | RTFILE_O_APPEND | RTFILE_O_WRITE | RTFILE_O_DENY_NONE); RTFileWrite(fh, u8FIFO, u8FIFOff, NULL); RTFileClose(fh); - #endif -#if 1 +# endif + AudioMixerSinkWrite(pThis->SinkFront.pMixSink, AUDMIXOP_COPY, u8FIFO, u8FIFOff, NULL /* pcbWritten */); -#endif -#endif - -#ifdef FOO +#endif /* DEBUG_SIMPLE */ + if (fInterrupt) { @@ -4352,13 +4386,4 @@ hdaProcessInterrupt(pThis); - } -#endif - - Log3Func(("Written %RU32 / %RU32 (left: %RU32), rc=%Rrc\n", cbTotal, cbToProcess, cbLeft, rc)); - - if (RT_SUCCESS(rc)) - { - if (pcbProcessed) - *pcbProcessed = cbTotal; } Index: /trunk/src/VBox/Devices/Audio/DevIchHdaCodec.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DevIchHdaCodec.cpp (revision 61522) +++ /trunk/src/VBox/Devices/Audio/DevIchHdaCodec.cpp (revision 61523) @@ -267,5 +267,5 @@ #define CODEC_AMP_NUM_STEPS 0x7F /** The initial gain offset (and when doing a node reset). */ -#define CODEC_AMP_OFF_INITIAL 0x40 +#define CODEC_AMP_OFF_INITIAL 0x7F /** The amplifier's gain step size. */ #define CODEC_AMP_STEP_SIZE 0x2 @@ -928,7 +928,7 @@ /* Default input amplifier capabilities. */ pNode->node.au32F00_param[0x0D] = CODEC_MAKE_F00_0D(CODEC_AMP_CAP_MUTE, - 0 /* Step size */, + CODEC_AMP_STEP_SIZE, CODEC_AMP_NUM_STEPS, - 0 /* Initial offset */); + CODEC_AMP_OFF_INITIAL); /* Default output amplifier capabilities. */ pNode->node.au32F00_param[0x12] = CODEC_MAKE_F00_12(CODEC_AMP_CAP_MUTE, @@ -1003,5 +1003,5 @@ pNode->adc.u32F05_param = CODEC_MAKE_F05(0, 0, 0, CODEC_F05_D3, CODEC_F05_D3); /* PS-Act: D3 Set: D3 */ - pNode->adc.node.au32F00_param[0x9] = CODEC_MAKE_F00_09(CODEC_F00_09_TYPE_AUDIO_INPUT, 13, 0) + pNode->adc.node.au32F00_param[0x9] = CODEC_MAKE_F00_09(CODEC_F00_09_TYPE_AUDIO_INPUT, 0xD, 0) | CODEC_F00_09_CAP_POWER_CTRL | CODEC_F00_09_CAP_CONNECTION_LIST @@ -1062,5 +1062,5 @@ case STAC9220_NID_PIN_HEADPHONE0: /* Port A: Headphone in/out (front). */ { - pNode->port.u32F09_param = CODEC_MAKE_F09_ANALOG(false /*fPresent*/, CODEC_F09_ANALOG_NA); + pNode->port.u32F09_param = CODEC_MAKE_F09_ANALOG(0 /*fPresent*/, CODEC_F09_ANALOG_NA); pNode->port.node.au32F00_param[0xC] = CODEC_MAKE_F00_0C(0x17) @@ -1087,5 +1087,5 @@ case STAC9220_NID_PIN_B: /* Port B: Rear CLFE (Center / Subwoofer). */ { - pNode->port.u32F09_param = CODEC_MAKE_F09_ANALOG(true /*fPresent*/, CODEC_F09_ANALOG_NA); + pNode->port.u32F09_param = CODEC_MAKE_F09_ANALOG(1 /*fPresent*/, CODEC_F09_ANALOG_NA); pNode->port.node.au32F00_param[0xC] = CODEC_MAKE_F00_0C(0x17) @@ -1111,5 +1111,5 @@ case STAC9220_NID_PIN_C: /* Rear Speaker. */ { - pNode->port.u32F09_param = CODEC_MAKE_F09_ANALOG(true /*fPresent*/, CODEC_F09_ANALOG_NA); + pNode->port.u32F09_param = CODEC_MAKE_F09_ANALOG(1 /*fPresent*/, CODEC_F09_ANALOG_NA); pNode->port.node.au32F00_param[0xC] = CODEC_MAKE_F00_0C(0x17) @@ -1135,5 +1135,5 @@ case STAC9220_NID_PIN_HEADPHONE1: /* Also known as PIN_D. */ { - pNode->port.u32F09_param = CODEC_MAKE_F09_ANALOG(false /*fPresent*/, CODEC_F09_ANALOG_NA); + pNode->port.u32F09_param = CODEC_MAKE_F09_ANALOG(1 /*fPresent*/, CODEC_F09_ANALOG_NA); pNode->port.node.au32F00_param[0xC] = CODEC_MAKE_F00_0C(0x17) Index: /trunk/src/VBox/Devices/Audio/DrvAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvAudio.cpp (revision 61522) +++ /trunk/src/VBox/Devices/Audio/DrvAudio.cpp (revision 61523) @@ -18,28 +18,4 @@ * hope that it will be useful, but WITHOUT ANY WARRANTY of any kind. * -------------------------------------------------------------------- - * - * This code is based on: audio.c from QEMU AUDIO subsystem. - * - * QEMU Audio subsystem - * - * Copyright (c) 2003-2005 Vassili Karpov (malc) - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. */ #define LOG_GROUP LOG_GROUP_DRV_AUDIO @@ -148,4 +124,11 @@ } +/** + * Returns the host stream part of an audio stream pair, or NULL + * if no host stream has been assigned / is not available. + * + * @returns IPRT status code. + * @param pStream Audio stream to retrieve host stream part for. + */ inline PPDMAUDIOSTREAM drvAudioGetHostStream(PPDMAUDIOSTREAM pStream) { @@ -448,5 +431,4 @@ } -#if 1 /** * Writes VM audio output data from the guest stream into the host stream. @@ -479,4 +461,8 @@ } + AssertMsg(pStream->enmDir == PDMAUDIODIR_OUT, + ("Stream '%s' is not an output stream and therefore cannot be written to (direction is 0x%x)\n", + pStream->szName, pStream->enmDir)); + LogFlowFunc(("[%s]: cbBuf=%RU32\n", pStream->szName, cbBuf)); @@ -516,52 +502,4 @@ } -#if 0 - uint32_t cMixed = 0; - if (RT_SUCCESS(rc)) - { - /* Mix just written guest stream samples to the host immediately. */ - rc = AudioMixBufMixToParent(&pGstStream->MixBuf, cWritten, &cMixed); - } -#endif - -#if 0 - uint32_t cPlayed = 0; - if (RT_SUCCESS(rc)) - { - PDMAUDIOSTRMSTS strmSts = pThis->pHostDrvAudio->pfnStreamGetStatus(pThis->pHostDrvAudio, pHstStream); - if (strmSts & PDMAUDIOSTRMSTS_FLAG_DATA_WRITABLE) - rc = pThis->pHostDrvAudio->pfnStreamPlay(pThis->pHostDrvAudio, pHstStream, &cPlayed); - } -#endif - -#ifdef DEBUG_andy - AssertRC(rc); -#endif - -#if 0 - /* - * Second, mix the guest mixing buffer with the host mixing - * buffer so that the host backend can play the data lateron. - */ - uint32_t cMixed; - if ( RT_SUCCESS(rc) - && cWritten) - { - rc = AudioMixBufMixToParent(&pGstStream->MixBuf, cWritten, &cMixed); - } - else - cMixed = 0; - - if (RT_SUCCESS(rc)) - { - /* - * Return the number of samples which actually have been mixed - * down to the parent, regardless how much samples were written - * into the children buffer. - */ - if (pcbWritten) - *pcbWritten = AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cMixed); - } -#else if (RT_SUCCESS(rc)) { @@ -569,5 +507,4 @@ *pcbWritten = AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cWritten); } -#endif LogFlowFunc(("cWritten=%RU32 (%RU32 bytes), cMixed=%RU32, rc=%Rrc\n", @@ -580,88 +517,4 @@ return rc; } -#else -static DECLCALLBACK(int) drvAudioWrite(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOGSTSTRMOUT pGstStrmOut, - const void *pvBuf, uint32_t cbBuf, uint32_t *pcbWritten) -{ - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - AssertPtrReturn(pThis, VERR_INVALID_POINTER); - - AssertPtrReturn(pGstStrmOut, VERR_INVALID_POINTER); - AssertPtrReturn(pvBuf, VERR_INVALID_POINTER); - AssertReturn(cbBuf, VERR_INVALID_PARAMETER); - /* pcbWritten is optional. */ - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_FAILURE(rc)) - return rc; - - if (!pThis->pHostDrvAudio->pfnIsEnabled(pThis->pHostDrvAudio, PDMAUDIODIR_OUT)) - { - rc = RTCritSectLeave(&pThis->CritSect); - AssertRC(rc); - - return VERR_NOT_AVAILABLE; - } - - PPDMAUDIOHSTSTRMOUT pHstStrmOut = pGstStrmOut->pHstStrmOut; - AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER); - - AssertMsg(pGstStrmOut->pHstStrmOut->fStatus & PDMAUDIOSTRMSTS_FLAG_ENABLED, - ("Writing to disabled host output stream \"%s\" not possible\n", - pHstStrmOut->MixBuf.pszName)); - - if (!AudioMixBufFreeBytes(&pGstStrmOut->MixBuf)) - { - if (pcbWritten) - *pcbWritten = 0; - - return RTCritSectLeave(&pThis->CritSect); - } - - /* - * First, write data from the device emulation into our - * guest mixing buffer. - */ - uint32_t cWritten; - //rc = AudioMixBufWriteAt(&pGstStrmOut->MixBuf, 0 /* Offset in samples */, pvBuf, cbBuf, &cWritten); - rc = AudioMixBufWriteCirc(&pGstStrmOut->MixBuf, pvBuf, cbBuf, &cWritten); - if (rc == VINF_BUFFER_OVERFLOW) - LogRel(("Audio: Lost audio samples from guest, expect stuttering audio output\n")); - - /* - * Second, mix the guest mixing buffer with the host mixing - * buffer so that the host backend can play the data lateron. - */ - uint32_t cMixed; - if ( RT_SUCCESS(rc) - && cWritten) - { - rc = AudioMixBufMixToParent(&pGstStrmOut->MixBuf, cWritten, &cMixed); - } - else - cMixed = 0; - - if (RT_SUCCESS(rc)) - { - /* - * Return the number of samples which actually have been mixed - * down to the parent, regardless how much samples were written - * into the children buffer. - */ - if (pcbWritten) - *pcbWritten = AUDIOMIXBUF_S2B(&pGstStrmOut->MixBuf, cMixed); - } - - LogFlowFunc(("%s -> %s: cbBuf=%RU32, cWritten=%RU32 (%RU32 bytes), cMixed=%RU32, rc=%Rrc\n", - pGstStrmOut->MixBuf.pszName, pHstStrmOut->MixBuf.pszName, cbBuf, cWritten, - AUDIOMIXBUF_S2B(&pGstStrmOut->MixBuf, cWritten), cMixed, rc)); - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - return rc; -} -#endif static DECLCALLBACK(uint32_t) drvAudioStreamAddRef(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream) @@ -688,5 +541,4 @@ } -#if 1 static DECLCALLBACK(int) drvAudioStreamIterate(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream) { @@ -725,5 +577,7 @@ PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); + AssertPtr(pHstStream); PPDMAUDIOSTREAM pGstStream = pHstStream->pPair; + AssertPtr(pGstStream); int rc = VINF_SUCCESS; @@ -731,6 +585,5 @@ do { - /*if (!(pHstStream->fStatus & PDMAUDIOSTRMSTS_FLAG_ENABLED)) - break;*/ + uint32_t cSamplesMixed = 0; PDMAUDIOSTRMSTS hstStrmSts = pThis->pHostDrvAudio->pfnStreamGetStatus(pThis->pHostDrvAudio, pHstStream); @@ -738,16 +591,22 @@ if (pHstStream->enmDir == PDMAUDIODIR_IN) { - /* Call the host backend to capture the audio input data. */ - uint32_t cSamplesCaptured; - rc = pThis->pHostDrvAudio->pfnStreamCapture(pThis->pHostDrvAudio, pHstStream, &cSamplesCaptured); - if (RT_FAILURE(rc)) - break; - - //cbData = AUDIOMIXBUF_S2B(&pStream->MixBuf, AudioMixBufMixed(&pStream->MixBuf))); - - } - else /* PDMAUDIODIR_OUT */ - { - uint32_t cSamplesMixed = 0; + uint32_t cSamplesToCapture = AudioMixBufFree(&pGstStream->MixBuf); + if (cSamplesToCapture) + { + if (hstStrmSts & PDMAUDIOSTRMSTS_FLAG_DATA_READABLE) + { + uint32_t cSamplesCaptured = 0; + + /* Call the host backend to capture the audio input data. */ + rc = pThis->pHostDrvAudio->pfnStreamCapture(pThis->pHostDrvAudio, pHstStream, &cSamplesCaptured); + if (RT_FAILURE(rc)) + break; + + LogFlowFunc(("%s: %RU32 samples captured, rc=%Rrc\n", pHstStream->szName, cSamplesCaptured, rc)); + } + } + } + else if (pHstStream->enmDir == PDMAUDIODIR_OUT) + { uint32_t cSamplesToMix = AudioMixBufUsed(&pGstStream->MixBuf); @@ -757,16 +616,13 @@ if (hstStrmSts & PDMAUDIOSTRMSTS_FLAG_DATA_WRITABLE) { - /* int rc2 = pThis->pHostDrvAudio->pfnStreamPlay(pThis->pHostDrvAudio, pHstStream, &cSamplesPlayed); - if (RT_SUCCESS(rc)) - rc = rc2; - }*/ - rc = AudioMixBufMixToParent(&pGstStream->MixBuf, cSamplesToMix, &cSamplesMixed); - LogFlowFunc(("%s: %RU32/%RU32 samples mixed, rc=%Rrc\n", + LogFlowFunc(("%s: %RU32/%RU32 playback samples mixed, rc=%Rrc\n", pHstStream->szName, cSamplesMixed, cSamplesToMix, rc)); } } } + else + AssertFailedStmt(rc = VERR_NOT_IMPLEMENTED); if (RT_SUCCESS(rc)) @@ -775,175 +631,12 @@ } while (0); + if (RT_FAILURE(rc)) + LogFunc(("Failed with %Rrc\n", rc)); + return rc; } -#else -static DECLCALLBACK(int) drvAudioGetDataIn(PPDMIAUDIOCONNECTOR pInterface, uint32_t *pcbAvailIn) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - /* pcbAvailIn is optional. */ - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_FAILURE(rc)) - return rc; - - uint32_t cbAvailIn = 0; - - PPDMAUDIOHSTSTRMIN pHstStrmIn = NULL; - while ((pHstStrmIn = drvAudioFindAnyHstIn(pThis, pHstStrmIn))) - { - /* Disabled? Skip it! */ - if (!(pHstStrmIn->fStatus & PDMAUDIOSTRMSTS_FLAG_ENABLED)) - continue; - - /* Call the host backend to capture the audio input data. */ - uint32_t cSamplesCaptured; - int rc2 = pThis->pHostDrvAudio->pfnCaptureIn(pThis->pHostDrvAudio, pHstStrmIn, - &cSamplesCaptured); - if (RT_FAILURE(rc2)) - continue; - - PPDMAUDIOGSTSTRMIN pGstStrmIn = pHstStrmIn->pGstStrmIn; - AssertPtrBreak(pGstStrmIn); - - if (pGstStrmIn->State.fActive) - { - cbAvailIn = RT_MAX(cbAvailIn, AUDIOMIXBUF_S2B(&pHstStrmIn->MixBuf, - AudioMixBufMixed(&pHstStrmIn->MixBuf))); -#ifdef DEBUG_andy - LogFlowFunc(("\t[%s] cbAvailIn=%RU32\n", pHstStrmIn->MixBuf.pszName, cbAvailIn)); -#endif - } - } - - if (RT_SUCCESS(rc)) - { - if (pcbAvailIn) - *pcbAvailIn = cbAvailIn; - } - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - if (RT_FAILURE(rc)) - LogFlowFuncLeaveRC(rc); - - return rc; -} - -static DECLCALLBACK(int) drvAudioGetDataOut(PPDMIAUDIOCONNECTOR pInterface, uint32_t *pcbFreeOut, uint32_t *pcSamplesLive) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - /* pcbFreeOut is optional. */ - /* pcSamplesLive is optional. */ - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_FAILURE(rc)) - return rc; - - static uint64_t s_tsLast = 0; - - - uint64_t s_tsDelta = RTTimeNanoTS() - s_tsLast; - LogFlowFunc(("delta=%RU64 (%RU64ms) -> ", s_tsDelta, s_tsDelta / 1000 / 1000)); - - #if 0 - PPDMAUDIOHSTSTRMOUT pHstStrmOut = drvAudioHstFindAnyEnabledOut(pThis, NULL); - uint64_t ns = s_tsDelta / pHstStrmOut->Props.uHz; - - - - uint32_t cSamplesMin = (cTicksElapsed / pSink->PCMProps.uHz) * pSink->PCMProps.cChannels; - #endif - - s_tsLast = RTTimeNanoTS(); - - /* - * Playback. - */ - uint32_t cSamplesLive = 0; - uint32_t cbFreeOut = UINT32_MAX; - - PPDMAUDIOHSTSTRMOUT pHstStrmOut = NULL; - while ((pHstStrmOut = drvAudioHstFindAnyEnabledOut(pThis, pHstStrmOut))) - { - cSamplesLive = AudioMixBufAvail(&pHstStrmOut->MixBuf); - - /* Has this stream marked as disabled but there still were guest streams relying - * on it? Check if this stream now can be closed and do so, if possible. */ - if ( (pHstStrmOut->fStatus & PDMAUDIOSTRMSTS_FLAG_PENDING_DISABLE) - && !cSamplesLive) - { - /* Stop playing the current (pending) stream. */ - int rc2 = drvAudioControlHstOut(pThis, pHstStrmOut, PDMAUDIOSTREAMCMD_DISABLE); - if (RT_SUCCESS(rc2)) - { - pHstStrmOut->fStatus &= ~PDMAUDIOSTRMSTS_FLAG_PENDING_DISABLE; - - LogFunc(("[%s] Disabling stream\n", pHstStrmOut->MixBuf.pszName)); - } - else - LogFunc(("[%s] Backend vetoed against closing output stream, rc=%Rrc\n", pHstStrmOut->MixBuf.pszName, rc2)); - - continue; - } - - LogFlowFunc(("[%s] cSamplesLive=%RU32\n", pHstStrmOut->MixBuf.pszName, cSamplesLive)); - - /* - * No live samples to play at the moment? - * - * Tell the device emulation for each connected guest stream how many - * bytes are free so that the device emulation can continue writing data to - * these streams. - */ - PPDMAUDIOGSTSTRMOUT pGstStrmOut; - uint32_t cbFree2 = UINT32_MAX; - RTListForEach(&pHstStrmOut->lstGstStrmOut, pGstStrmOut, PDMAUDIOGSTSTRMOUT, Node) - { - if (pGstStrmOut->State.fActive) - { - /* Tell the sound device emulation how many samples are free - * so that it can start writing PCM data to us. */ - cbFree2 = RT_MIN(cbFree2, AUDIOMIXBUF_S2B_RATIO(&pGstStrmOut->MixBuf, - AudioMixBufFree(&pGstStrmOut->MixBuf))); -#ifdef DEBUG_andy - LogFlowFunc(("\t[%s] cbFreeOut=%RU32\n", pGstStrmOut->MixBuf.pszName, cbFree2)); -#endif - } - } - - cbFreeOut = RT_MIN(cbFreeOut, cbFree2); - } - - if (RT_SUCCESS(rc)) - { - if (cbFreeOut == UINT32_MAX) - cbFreeOut = 0; - - if (pcbFreeOut) - *pcbFreeOut = cbFreeOut; - - if (pcSamplesLive) - *pcSamplesLive = cSamplesLive; - } - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - if (RT_FAILURE(rc)) - LogFlowFuncLeaveRC(rc); - - return rc; -} -#endif - -#if 1 -static DECLCALLBACK(int) drvAudioStreamPlay(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesPlayed) + +static DECLCALLBACK(int) drvAudioStreamPlay(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, + uint32_t *pcSamplesPlayed) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); @@ -951,4 +644,8 @@ /* pcSamplesPlayed is optional. */ + AssertMsg(pStream->enmDir == PDMAUDIODIR_OUT, + ("Stream '%s' is not an output stream and therefore cannot be played back (direction is 0x%x)\n", + pStream->szName, pStream->enmDir)); + AssertMsg(pStream->fStatus & PDMAUDIOSTRMSTS_FLAG_INITIALIZED, ("Unable to play stream '%s' (status is 0x%x)\n", pStream->szName, pStream->fStatus)); @@ -981,4 +678,5 @@ PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); + AssertPtr(pHstStream); PPDMAUDIOSTREAM pGstStream = pHstStream->pPair; AssertPtr(pGstStream); @@ -1021,29 +719,5 @@ } } - #if 0 - if (pStream->pPair) - { - bool fIsEmpty = AudioMixBufIsEmpty(&pStream->pPair->MixBuf); - - if ( !pPair->State.fActive - && fIsEmpty) - continue; - - if () - { - pGstStrmOut->State.fEmpty = true; - fNeedsCleanup |= !pGstStrmOut->State.fActive; - } - } - - if (fNeedsCleanup) - { - RTListForEach(&pHstStrmOut->lstGstStrmOut, pGstStrmOut, PDMAUDIOGSTSTRMOUT, Node) - { - if (!pGstStrmOut->State.fActive) - drvAudioDestroyGstOut(pThis, pGstStrmOut); - } - } - #endif + } while (0); @@ -1063,9 +737,14 @@ return rc; } -#else -static DECLCALLBACK(int) drvAudioPlayOut(PPDMIAUDIOCONNECTOR pInterface, uint32_t *pcSamplesPlayed) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - /* pcSamplesPlayed is optional. */ + +static DECLCALLBACK(int) drvAudioStreamCapture(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, + uint32_t *pcSamplesCaptured) +{ + AssertMsg(pStream->enmDir == PDMAUDIODIR_IN, + ("Stream '%s' is not an input stream and therefore cannot be captured (direction is 0x%x)\n", + pStream->szName, pStream->enmDir)); + + AssertMsg(pStream->fStatus & PDMAUDIOSTRMSTS_FLAG_INITIALIZED, + ("Unable to capture stream '%s' (status is 0x%x)\n", pStream->szName, pStream->fStatus)); PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); @@ -1075,98 +754,73 @@ return rc; - /* Backend output (temporarily) disabled / unavailable? */ - if (!pThis->pHostDrvAudio->pfnIsEnabled(pThis->pHostDrvAudio, PDMAUDIODIR_OUT)) - { - rc = pThis->pHostDrvAudio->pfnGetConf(pThis->pHostDrvAudio, &pThis->BackendCfg); - AssertRC(rc); - - if ( !pThis->BackendCfg.cSinks - || !pThis->BackendCfg.cMaxStreamsOut) - { - int rc2 = RTCritSectLeave(&pThis->CritSect); - AssertRC(rc2); - - return VERR_NOT_AVAILABLE; - } - } - - /* - * Process all enabled host output streams. - */ - uint32_t cSamplesPlayedMax = 0; - PPDMAUDIOHSTSTRMOUT pHstStrmOut = NULL; - while ((pHstStrmOut = drvAudioHstFindAnyEnabledOut(pThis, pHstStrmOut))) - { -#if 0 - uint32_t cStreamsLive; - uint32_t cSamplesLive = drvAudioHstOutSamplesLive(pHstStrmOut, &cStreamsLive); - if (!cStreamsLive) - cSamplesLive = 0; - - /* Has this stream marked as disabled but there still were guest streams relying - * on it? Check if this stream now can be closed and do so, if possible. */ - if ( pHstStrmOut->fPendingDisable - && !cStreamsLive) - { - /* Stop playing the current (pending) stream. */ - int rc2 = pThis->pHostDrvAudio->pfnControlOut(pThis->pHostDrvAudio, pHstStrmOut, - PDMAUDIOSTREAMCMD_DISABLE); - if (RT_SUCCESS(rc2)) - { - pHstStrmOut->fEnabled = false; - pHstStrmOut->fPendingDisable = false; - - LogFunc(("\t%p: Disabling stream\n", pHstStrmOut)); - } - else - LogFunc(("\t%p: Backend vetoed against closing output stream, rc=%Rrc\n", - pHstStrmOut, rc2)); - - continue; - } -#endif - uint32_t cSamplesPlayed = 0; - int rc2 = pThis->pHostDrvAudio->pfnPlayOut(pThis->pHostDrvAudio, pHstStrmOut, &cSamplesPlayed); - if (RT_FAILURE(rc2)) - { - int rc3 = pThis->pHostDrvAudio->pfnControlOut(pThis->pHostDrvAudio, pHstStrmOut, PDMAUDIOSTREAMCMD_DISABLE); - AssertRC(rc3); - } - else - cSamplesPlayedMax = RT_MAX(cSamplesPlayed, cSamplesPlayedMax); - - LogFlowFunc(("\t[%s] cSamplesPlayed=%RU32, cSamplesPlayedMax=%RU32, rc=%Rrc\n", - pHstStrmOut->MixBuf.pszName, cSamplesPlayed, cSamplesPlayedMax, rc2)); - - bool fNeedsCleanup = false; - - PPDMAUDIOGSTSTRMOUT pGstStrmOut; - RTListForEach(&pHstStrmOut->lstGstStrmOut, pGstStrmOut, PDMAUDIOGSTSTRMOUT, Node) - { - if ( !pGstStrmOut->State.fActive - && pGstStrmOut->State.fEmpty) - continue; - - if (AudioMixBufIsEmpty(&pGstStrmOut->MixBuf)) - { - pGstStrmOut->State.fEmpty = true; - fNeedsCleanup |= !pGstStrmOut->State.fActive; - } - } - - if (fNeedsCleanup) - { - RTListForEach(&pHstStrmOut->lstGstStrmOut, pGstStrmOut, PDMAUDIOGSTSTRMOUT, Node) - { - if (!pGstStrmOut->State.fActive) - drvAudioDestroyGstOut(pThis, pGstStrmOut); - } - } - } - - if (RT_SUCCESS(rc)) - { - if (pcSamplesPlayed) - *pcSamplesPlayed = cSamplesPlayedMax; + LogFlowFunc(("[%s]\n", pStream->szName)); + + uint32_t cSamplesCaptured = 0; + + do + { + /* Backend input (temporarily) disabled / unavailable? */ + if (pThis->pHostDrvAudio->pfnGetStatus(pThis->pHostDrvAudio, PDMAUDIODIR_IN) != PDMAUDIOBACKENDSTS_RUNNING) + { + rc = pThis->pHostDrvAudio->pfnGetConfig(pThis->pHostDrvAudio, &pThis->BackendCfg); + AssertRC(rc); + + if ( !pThis->BackendCfg.cSources + || !pThis->BackendCfg.cMaxStreamsIn) + { + rc = VERR_NOT_AVAILABLE; + break; + } + } + + PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); + AssertPtr(pHstStream); + PPDMAUDIOSTREAM pGstStream = pHstStream->pPair; + AssertPtr(pGstStream); + + uint32_t cSamplesLive = AudioMixBufUsed(&pHstStream->MixBuf); + if (cSamplesLive) + { + PDMAUDIOSTRMSTS strmSts = pThis->pHostDrvAudio->pfnStreamGetStatus(pThis->pHostDrvAudio, pHstStream); + if (strmSts & PDMAUDIOSTRMSTS_FLAG_DATA_READABLE) + { + rc = pThis->pHostDrvAudio->pfnStreamCapture(pThis->pHostDrvAudio, pHstStream, &cSamplesCaptured); + if (RT_FAILURE(rc)) + drvAudioStreamControlInternalBackend(pThis, pHstStream, PDMAUDIOSTREAMCMD_DISABLE); + } + + LogFlowFunc(("[%s] strmSts=0x%x, cSamplesCaptured=%RU32, rc=%Rrc\n", pStream->szName, strmSts, cSamplesCaptured, rc)); + + if (RT_SUCCESS(rc)) + { + rc = drvAudioStreamIterateInternal(pThis, pStream); + if (RT_SUCCESS(rc)) + cSamplesLive = AudioMixBufUsed(&pHstStream->MixBuf); + } + } + + if (!cSamplesLive) + { + /* Has the host stream marked as disabled but there still were guest streams relying + * on it? Check if the stream now can be closed and do so, if possible. */ + if (pHstStream->fStatus & PDMAUDIOSTRMSTS_FLAG_PENDING_DISABLE) + { + LogFunc(("%s: Closing pending stream\n", pHstStream->szName)); + rc = drvAudioStreamControlInternalBackend(pThis, pHstStream, PDMAUDIOSTREAMCMD_DISABLE); + if (RT_SUCCESS(rc)) + { + pHstStream->fStatus &= ~PDMAUDIOSTRMSTS_FLAG_PENDING_DISABLE; + } + else + LogFunc(("%s: Backend vetoed against closing input stream, rc=%Rrc\n", pHstStream->szName, rc)); + } + } + + } while (0); + + if (RT_SUCCESS(rc)) + { + if (pcSamplesCaptured) + *pcSamplesCaptured = cSamplesCaptured; } @@ -1180,5 +834,4 @@ return rc; } -#endif #ifdef VBOX_WITH_AUDIO_CALLBACKS @@ -1378,4 +1031,564 @@ PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); LogFlowFunc(("pThis=%p, pDrvIns=%p\n", pThis, pDrvIns)); + + int rc = RTCritSectInit(&pThis->CritSect); + + /** @todo Add audio driver options. */ + + /* + * If everything went well, initialize the lower driver. + */ + if (RT_SUCCESS(rc)) + rc = drvAudioHostInit(pCfgHandle, pThis); + + LogFlowFuncLeaveRC(rc); + return rc; +} + +static DECLCALLBACK(int) drvAudioStreamRead(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, + void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead) +{ + PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); + AssertPtrReturn(pThis, VERR_INVALID_POINTER); + + if (!pStream) + { + if (pcbRead) + *pcbRead = 0; + return VINF_SUCCESS; + } + + AssertPtrReturn(pvBuf, VERR_INVALID_POINTER); + AssertReturn(cbBuf, VERR_INVALID_PARAMETER); + /* pcbWritten is optional. */ + + int rc = RTCritSectEnter(&pThis->CritSect); + if (RT_FAILURE(rc)) + return rc; + + AssertMsg(pStream->enmDir == PDMAUDIODIR_IN, + ("Stream '%s' is not an input stream and therefore cannot be read from (direction is 0x%x)\n", + pStream->szName, pStream->enmDir)); + + if (pThis->pHostDrvAudio->pfnGetStatus(pThis->pHostDrvAudio, PDMAUDIODIR_IN) != PDMAUDIOBACKENDSTS_RUNNING) + { + if (pcbRead) + *pcbRead = 0; + + return RTCritSectLeave(&pThis->CritSect); + } + + PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); + PPDMAUDIOSTREAM pGstStream = pHstStream->pPair; + + AssertMsg(pHstStream->fStatus & PDMAUDIOSTRMSTS_FLAG_ENABLED, + ("Reading from disabled host input stream '%s' not possible\n", pHstStream->szName)); + + Log3Func(("%s\n", pStream->szName)); + + /* + * Read from the parent buffer (that is, the guest buffer) which + * should have the audio data in the format the guest needs. + */ + uint32_t cRead; + rc = AudioMixBufReadCirc(&pGstStream->MixBuf, pvBuf, cbBuf, &cRead); + if (RT_SUCCESS(rc)) + { + if (cRead) + AudioMixBufFinish(&pGstStream->MixBuf, cRead); + + if (pcbRead) + *pcbRead = AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cRead); + } + + LogFlowFunc(("cRead=%RU32 (%RU32 bytes), rc=%Rrc\n", + cRead, AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cRead), rc)); + + int rc2 = RTCritSectLeave(&pThis->CritSect); + if (RT_SUCCESS(rc)) + rc = rc2; + + return rc; +} + +static DECLCALLBACK(int) drvAudioStreamCreate(PPDMIAUDIOCONNECTOR pInterface, + PPDMAUDIOSTREAMCFG pCfgHost, PPDMAUDIOSTREAMCFG pCfgGuest, + PPDMAUDIOSTREAM *ppStream) +{ + AssertPtrReturn(pInterface, VERR_INVALID_POINTER); + AssertPtrReturn(pCfgHost, VERR_INVALID_POINTER); + AssertPtrReturn(pCfgGuest, VERR_INVALID_POINTER); + AssertPtrReturn(ppStream, VERR_INVALID_POINTER); + + PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); + + int rc = RTCritSectEnter(&pThis->CritSect); + if (RT_FAILURE(rc)) + return rc; + + LogFlowFunc(("Host=%s, Guest=%s\n", pCfgHost->szName, pCfgGuest->szName)); +#ifdef DEBUG + DrvAudioHlpStreamCfgPrint(pCfgHost); + DrvAudioHlpStreamCfgPrint(pCfgGuest); +#endif + + /* + * The guest stream always will get the audio stream configuration told + * by the device emulation (which in turn was/could be set by the guest OS). + */ + PPDMAUDIOSTREAM pGstStrm = NULL; + + /** @todo Docs! */ + PPDMAUDIOSTREAM pHstStrm = NULL; + +#define RC_BREAK(x) { rc = x; break; } + + do + { + if ( !DrvAudioHlpStreamCfgIsValid(pCfgHost) + || !DrvAudioHlpStreamCfgIsValid(pCfgGuest)) + { + RC_BREAK(VERR_INVALID_PARAMETER); + } + + /* Make sure that both configurations actually intend the same thing. */ + if (pCfgHost->enmDir != pCfgGuest->enmDir) + { + AssertMsgFailed(("Stream configuration directions do not match\n")); + RC_BREAK(VERR_INVALID_PARAMETER); + } + + /* Note: cbHstStrm will contain sizeof(PDMAUDIOSTREAM) + additional data + * which the host backend will need. */ + size_t cbHstStrm; + if (pCfgHost->enmDir == PDMAUDIODIR_IN) + { + if (!pThis->cStreamsFreeIn) + { + LogFlowFunc(("No more input streams free to use, bailing out\n")); + RC_BREAK(VERR_AUDIO_NO_FREE_INPUT_STREAMS); + } + + /* Validate backend configuration. */ + if (!pThis->BackendCfg.cbStreamIn) + { + LogFlowFunc(("Backend input configuration not valid, bailing out\n")); + RC_BREAK(VERR_INVALID_PARAMETER); + } + + cbHstStrm = pThis->BackendCfg.cbStreamIn; + } + else /* Out */ + { + if (!pThis->cStreamsFreeOut) + { + LogFlowFunc(("Maximum number of host output streams reached\n")); + RC_BREAK(VERR_AUDIO_NO_FREE_OUTPUT_STREAMS); + } + + /* Validate backend configuration. */ + if (!pThis->BackendCfg.cbStreamOut) + { + LogFlowFunc(("Backend output configuration invalid, bailing out\n")); + RC_BREAK(VERR_INVALID_PARAMETER); + } + + cbHstStrm = pThis->BackendCfg.cbStreamOut; + } + + pHstStrm = (PPDMAUDIOSTREAM)RTMemAllocZ(cbHstStrm); + AssertPtrBreakStmt(pHstStrm, rc = VERR_NO_MEMORY); + + pHstStrm->enmCtx = PDMAUDIOSTREAMCTX_HOST; + pHstStrm->enmDir = pCfgHost->enmDir; + + pGstStrm = (PPDMAUDIOSTREAM)RTMemAllocZ(sizeof(PDMAUDIOSTREAM)); + AssertPtrBreakStmt(pGstStrm, rc = VERR_NO_MEMORY); + + pGstStrm->enmCtx = PDMAUDIOSTREAMCTX_GUEST; + pGstStrm->enmDir = pCfgGuest->enmDir; + + /* + * Create host stream. + */ + + RTStrPrintf(pHstStrm->szName, RT_ELEMENTS(pHstStrm->szName), "%s (Host)", + strlen(pCfgHost->szName) ? pCfgHost->szName : ""); + + /* Note: Direction is always from child -> parent. */ + uint32_t cSamples = 0; + rc = pThis->pHostDrvAudio->pfnStreamCreate(pThis->pHostDrvAudio, pHstStrm, pCfgHost, &cSamples); + if (RT_FAILURE(rc)) + { + LogFlowFunc(("Initializing host backend failed with rc=%Rrc\n", rc)); + break; + } + + pHstStrm->fStatus |= PDMAUDIOSTRMSTS_FLAG_INITIALIZED; + pHstStrm->pPair = pGstStrm; + + rc = DrvAudioHlpStreamCfgToProps(pCfgHost, &pHstStrm->Props); + AssertRCBreak(rc); + + rc = AudioMixBufInit(&pHstStrm->MixBuf, pHstStrm->szName, &pHstStrm->Props, cSamples * 4); + AssertRCBreak(rc); + + /* + * Create guest stream. + */ + + RTStrPrintf(pGstStrm->szName, RT_ELEMENTS(pGstStrm->szName), "%s (Guest)", + strlen(pCfgGuest->szName) ? pCfgGuest->szName : ""); + + rc = DrvAudioHlpStreamCfgToProps(pCfgGuest, &pGstStrm->Props); + AssertRCBreak(rc); + + rc = AudioMixBufInit(&pGstStrm->MixBuf, pGstStrm->szName, &pGstStrm->Props, cSamples * 2); + if (RT_SUCCESS(rc)) + { + if (pCfgGuest->enmDir == PDMAUDIODIR_IN) + { + /* Host (Parent) -> Guest (Child). */ + rc = AudioMixBufLinkTo(&pHstStrm->MixBuf, &pGstStrm->MixBuf); + } + else + { + /* Guest (Parent) -> Host (Child). */ + rc = AudioMixBufLinkTo(&pGstStrm->MixBuf, &pHstStrm->MixBuf); + } + } + + pGstStrm->fStatus |= PDMAUDIOSTRMSTS_FLAG_INITIALIZED; + pGstStrm->pPair = pHstStrm; + + AssertRCBreak(rc); + + } while (0); + +#undef RC_BREAK + + if (RT_FAILURE(rc)) + { + drvAudioStreamDestroyInternal(pThis, pGstStrm); + pGstStrm = NULL; + + drvAudioStreamDestroyInternal(pThis, pHstStrm); + pHstStrm = NULL; + } + else + { + /* Set initial reference counts. */ + RTListAppend(&pThis->lstGstStreams, &pGstStrm->Node); + pGstStrm->cRefs = 1; + + RTListAppend(&pThis->lstHstStreams, &pHstStrm->Node); + pHstStrm->cRefs = 1; + + if (pCfgHost->enmDir == PDMAUDIODIR_IN) + { + Assert(pThis->cStreamsFreeIn); + pThis->cStreamsFreeIn--; + } + else /* Out */ + { + Assert(pThis->cStreamsFreeOut); + pThis->cStreamsFreeOut--; + } + + /* Always return the guest-side part to the device emulation. */ + *ppStream = pGstStrm; + } + + int rc2 = RTCritSectLeave(&pThis->CritSect); + if (RT_SUCCESS(rc)) + rc = rc2; + + LogFlowFuncLeaveRC(rc); + return rc; +} + +#if 1 +static DECLCALLBACK(int) drvAudioGetConfig(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOBACKENDCFG pCfg) +{ + AssertPtrReturn(pInterface, VERR_INVALID_POINTER); + AssertPtrReturn(pCfg, VERR_INVALID_POINTER); + + PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); + + int rc = RTCritSectEnter(&pThis->CritSect); + if (RT_FAILURE(rc)) + return rc; + + rc = pThis->pHostDrvAudio->pfnGetConfig(pThis->pHostDrvAudio, pCfg); + + int rc2 = RTCritSectLeave(&pThis->CritSect); + if (RT_SUCCESS(rc)) + rc = rc2; + + LogFlowFuncLeaveRC(rc); + return rc; +} + +static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvAudioGetStatus(PPDMIAUDIOCONNECTOR pInterface, PDMAUDIODIR enmDir) +{ + AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN); + + PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); + + int rc = RTCritSectEnter(&pThis->CritSect); + if (RT_FAILURE(rc)) + return PDMAUDIOBACKENDSTS_UNKNOWN; + + PDMAUDIOBACKENDSTS backendSts = pThis->pHostDrvAudio->pfnGetStatus(pThis->pHostDrvAudio, enmDir); + + int rc2 = RTCritSectLeave(&pThis->CritSect); + if (RT_SUCCESS(rc)) + rc = rc2; + + LogFlowFuncLeaveRC(rc); + return backendSts; +} + +static DECLCALLBACK(PDMAUDIOSTRMSTS) drvAudioStreamGetStatus(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream) +{ + AssertPtrReturn(pInterface, false); + + if (!pStream) + return PDMAUDIOSTRMSTS_FLAG_NONE; + + PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); + + int rc2 = RTCritSectEnter(&pThis->CritSect); + AssertRC(rc2); + + PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); + PDMAUDIOSTRMSTS strmSts = pHstStream->fStatus; + + LogFlowFunc(("%s: strmSts=0x%x\n", pHstStream->szName, strmSts)); + rc2 = RTCritSectLeave(&pThis->CritSect); + AssertRC(rc2); + + return strmSts; +} + +static DECLCALLBACK(int) drvAudioStreamSetVolume(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, PPDMAUDIOVOLUME pVol) +{ + AssertPtrReturn(pInterface, VERR_INVALID_POINTER); + AssertPtrReturn(pStream, VERR_INVALID_POINTER); + AssertPtrReturn(pVol, VERR_INVALID_POINTER); + + PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); + + LogFlowFunc(("%s: volL=%RU32, volR=%RU32, fMute=%RTbool\n", pStream->szName, pVol->uLeft, pVol->uRight, pVol->fMuted)); + + AudioMixBufSetVolume(&pStream->MixBuf, pVol); + + return VINF_SUCCESS; +} +#endif + +static DECLCALLBACK(int) drvAudioStreamDestroy(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream) +{ + AssertPtrReturn(pInterface, VERR_INVALID_POINTER); + AssertPtrReturn(pStream, VERR_INVALID_POINTER); + + PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); + + int rc = RTCritSectEnter(&pThis->CritSect); + AssertRC(rc); + + PDMAUDIODIR enmDir = pStream->enmDir; + + LogFlowFunc(("%s: cRefs=%RU32\n", pStream->szName, pStream->cRefs)); + if (pStream->cRefs > 1) + rc = VERR_WRONG_ORDER; + + if (RT_SUCCESS(rc)) + { + PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); + PPDMAUDIOSTREAM pGstStream = pHstStream ? pHstStream->pPair : pStream; + + LogFlowFunc(("\tHost : %s\n", pHstStream ? pHstStream->szName : "")); + LogFlowFunc(("\tGuest: %s\n", pGstStream ? pGstStream->szName : "")); + + /* Should prevent double frees. */ + Assert(pHstStream != pGstStream); + + if (pHstStream) + { + pHstStream->pPair = NULL; + RTListNodeRemove(&pHstStream->Node); + } + + if (pGstStream) + { + pGstStream->pPair = NULL; + RTListNodeRemove(&pGstStream->Node); + } + + if (pHstStream) + { + rc = drvAudioStreamDestroyInternal(pThis, pHstStream); + AssertRC(rc); + + pHstStream = NULL; + } + + if (pGstStream) + { + rc = drvAudioStreamDestroyInternal(pThis, pGstStream); + AssertRC(rc); + + pGstStream = NULL; + } + } + + if (RT_SUCCESS(rc)) + { + if (enmDir == PDMAUDIODIR_IN) + { + pThis->cStreamsFreeIn++; + } + else /* Out */ + { + pThis->cStreamsFreeOut++; + } + } + + int rc2 = RTCritSectLeave(&pThis->CritSect); + if (RT_SUCCESS(rc)) + rc = rc2; + + LogFlowFuncLeaveRC(rc); + return rc; +} + +static int drvAudioStreamDestroyInternalBackend(PDRVAUDIO pThis, PPDMAUDIOSTREAM pHstStream) +{ + AssertPtrReturn(pThis, VERR_INVALID_POINTER); + AssertPtrReturn(pHstStream, VERR_INVALID_POINTER); + + AssertMsg(pHstStream->enmCtx == PDMAUDIOSTREAMCTX_HOST, + ("Stream '%s' is not a host stream and therefore has no backend\n", pHstStream->szName)); + + int rc = VINF_SUCCESS; + + LogFlowFunc(("%s: fStatus=0x%x\n", pHstStream->szName, pHstStream->fStatus)); + + if (pHstStream->fStatus & PDMAUDIOSTRMSTS_FLAG_INITIALIZED) + { + if (pThis->pHostDrvAudio) + rc = pThis->pHostDrvAudio->pfnStreamDestroy(pThis->pHostDrvAudio, pHstStream); + if (RT_SUCCESS(rc)) + pHstStream->fStatus &= ~PDMAUDIOSTRMSTS_FLAG_INITIALIZED; + } + + LogFlowFunc(("%s: Returning %Rrc\n", pHstStream->szName, rc)); + return rc; +} + +static int drvAudioStreamDestroyInternal(PDRVAUDIO pThis, PPDMAUDIOSTREAM pStream) +{ + AssertPtrReturn(pThis, VERR_INVALID_POINTER); + AssertPtrReturn(pStream, VERR_INVALID_POINTER); + + LogFlowFunc(("%s: cRefs=%RU32\n", pStream->szName, pStream->cRefs)); + + if (pStream->cRefs > 1) + return VERR_WRONG_ORDER; + + int rc = VINF_SUCCESS; + + if (pStream->enmCtx == PDMAUDIOSTREAMCTX_GUEST) + { + if (pStream->fStatus & PDMAUDIOSTRMSTS_FLAG_INITIALIZED) + { + rc = drvAudioStreamControlInternal(pThis, pStream, PDMAUDIOSTREAMCMD_DISABLE); + if (RT_SUCCESS(rc)) + pStream->fStatus &= ~PDMAUDIOSTRMSTS_FLAG_INITIALIZED; + } + } + else if (pStream->enmCtx == PDMAUDIOSTREAMCTX_HOST) + { + rc = drvAudioStreamDestroyInternalBackend(pThis, pStream); + } + else + AssertFailedReturn(VERR_NOT_IMPLEMENTED); + + if (RT_SUCCESS(rc)) + { + /* Destroy mixing buffer. */ + AudioMixBufDestroy(&pStream->MixBuf); + + if (pStream) + { + RTMemFree(pStream); + pStream = NULL; + } + } + + LogFlowFunc(("Returning %Rrc\n", rc)); + return rc; +} + +/********************************************************************/ + +/** + * @interface_method_impl{PDMIBASE,pfnQueryInterface} + */ +static DECLCALLBACK(void *) drvAudioQueryInterface(PPDMIBASE pInterface, const char *pszIID) +{ + LogFlowFunc(("pInterface=%p, pszIID=%s\n", pInterface, pszIID)); + + PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface); + PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); + + PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase); + PDMIBASE_RETURN_INTERFACE(pszIID, PDMIAUDIOCONNECTOR, &pThis->IAudioConnector); + + return NULL; +} + +/** + * Power Off notification. + * + * @param pDrvIns The driver instance data. + */ +static DECLCALLBACK(void) drvAudioPowerOff(PPDMDRVINS pDrvIns) +{ + PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); + + LogFlowFuncEnter(); + + /* Just destroy the host stream on the backend side. + * The rest will either be destructed by the device emulation or + * in drvAudioDestruct(). */ + PPDMAUDIOSTREAM pStream; + RTListForEach(&pThis->lstHstStreams, pStream, PDMAUDIOSTREAM, Node) + drvAudioStreamDestroyInternalBackend(pThis, pStream); + + /* + * Last call for the driver below us. + * Let it know that we reached end of life. + */ + if (pThis->pHostDrvAudio->pfnShutdown) + pThis->pHostDrvAudio->pfnShutdown(pThis->pHostDrvAudio); + + pThis->pHostDrvAudio = NULL; + + LogFlowFuncLeave(); +} + +/** + * Constructs an audio driver instance. + * + * @copydoc FNPDMDRVCONSTRUCT + */ +static DECLCALLBACK(int) drvAudioConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfgHandle, uint32_t fFlags) +{ + LogFlowFunc(("pDrvIns=%#p, pCfgHandle=%#p, fFlags=%x\n", pDrvIns, pCfgHandle, fFlags)); + + PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); + PDMDRV_CHECK_VERSIONS_RETURN(pDrvIns); RTListInit(&pThis->lstHstStreams); @@ -1385,693 +1598,4 @@ RTListInit(&pThis->lstCBOut); #endif - - int rc = RTCritSectInit(&pThis->CritSect); - - /** @todo Add audio driver options. */ - - /* - * If everything went well, initialize the lower driver. - */ - if (RT_SUCCESS(rc)) - rc = drvAudioHostInit(pCfgHandle, pThis); - - LogFlowFuncLeaveRC(rc); - return rc; -} - -#if 1 -static DECLCALLBACK(int) drvAudioStreamRead(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, - void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead) -{ - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - AssertPtrReturn(pThis, VERR_INVALID_POINTER); - - if (!pStream) - { - if (pcbRead) - *pcbRead = 0; - return VINF_SUCCESS; - } - - AssertPtrReturn(pvBuf, VERR_INVALID_POINTER); - AssertReturn(cbBuf, VERR_INVALID_PARAMETER); - /* pcbWritten is optional. */ - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_FAILURE(rc)) - return rc; - - if (pThis->pHostDrvAudio->pfnGetStatus(pThis->pHostDrvAudio, PDMAUDIODIR_IN) != PDMAUDIOBACKENDSTS_RUNNING) - { - if (pcbRead) - *pcbRead = 0; - - return RTCritSectLeave(&pThis->CritSect); - } - - PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); - - AssertMsg(pHstStream->fStatus & PDMAUDIOSTRMSTS_FLAG_ENABLED, - ("Reading from disabled host input stream '%s' not possible\n", pHstStream->szName)); - - /* - * Read from the parent buffer (that is, the guest buffer) which - * should have the audio data in the format the guest needs. - */ - uint32_t cRead; - rc = AudioMixBufReadCirc(&pStream->MixBuf, pvBuf, cbBuf, &cRead); - if (RT_SUCCESS(rc)) - { - AudioMixBufFinish(&pStream->MixBuf, cRead); - - if (pcbRead) - *pcbRead = AUDIOMIXBUF_S2B(&pStream->MixBuf, cRead); - } - - LogFlowFunc(("cRead=%RU32 (%RU32 bytes), rc=%Rrc\n", - cRead, AUDIOMIXBUF_S2B(&pStream->MixBuf, cRead), rc)); - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - return rc; -} -#else -static DECLCALLBACK(int) drvAudioRead(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOGSTSTRMIN pGstStrmIn, - void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead) -{ - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - AssertPtrReturn(pThis, VERR_INVALID_POINTER); - - if (!pGstStrmIn) - return VERR_NOT_AVAILABLE; - - AssertPtrReturn(pvBuf, VERR_INVALID_POINTER); - AssertReturn(cbBuf, VERR_INVALID_PARAMETER); - /* pcbWritten is optional. */ - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_FAILURE(rc)) - return rc; - - if (!pThis->pHostDrvAudio->pfnIsEnabled(pThis->pHostDrvAudio, PDMAUDIODIR_IN)) - { - if (pcbRead) - *pcbRead = 0; - - return RTCritSectLeave(&pThis->CritSect); - } - - PPDMAUDIOHSTSTRMIN pHstStrmIn = pGstStrmIn->pHstStrmIn; - AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER); - - AssertMsg(pGstStrmIn->pHstStrmIn->fStatus & PDMAUDIOSTRMSTS_FLAG_ENABLED, - ("Reading from disabled host input stream \"%s\" not possible\n", pGstStrmIn->MixBuf.pszName)); - - /* - * Read from the parent buffer (that is, the guest buffer) which - * should have the audio data in the format the guest needs. - */ - uint32_t cRead; - rc = AudioMixBufReadCirc(&pGstStrmIn->MixBuf, pvBuf, cbBuf, &cRead); - if (RT_SUCCESS(rc)) - { - AudioMixBufFinish(&pGstStrmIn->MixBuf, cRead); - - if (pcbRead) - *pcbRead = AUDIOMIXBUF_S2B(&pGstStrmIn->MixBuf, cRead); - } - - LogFlowFunc(("cRead=%RU32 (%RU32 bytes), rc=%Rrc\n", - cRead, AUDIOMIXBUF_S2B(&pGstStrmIn->MixBuf, cRead), rc)); - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - return rc; -} -#endif - -#if 0 -static DECLCALLBACK(int) drvAudioEnableOut(PPDMIAUDIOCONNECTOR pInterface, - PPDMAUDIOGSTSTRMOUT pGstStrmOut, bool fEnable) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - - if (!pGstStrmOut) - return VERR_NOT_AVAILABLE; - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - int rc = VINF_SUCCESS; - - PPDMAUDIOHSTSTRMOUT pHstStrmOut = pGstStrmOut->pHstStrmOut; - AssertPtr(pHstStrmOut); - - if (fEnable) - { - /* Is a pending disable outstanding? Then disable first. */ - if (pHstStrmOut->fStatus & PDMAUDIOSTRMSTS_FLAG_PENDING_DISABLE) - { - rc = drvAudioControlHstOut(pThis, pHstStrmOut, PDMAUDIOSTREAMCMD_DISABLE); - if (RT_SUCCESS(rc)) - pHstStrmOut->fStatus &= ~PDMAUDIOSTRMSTS_FLAG_PENDING_DISABLE; - } - - if (RT_SUCCESS(rc)) - rc = drvAudioControlHstOut(pThis, pHstStrmOut, PDMAUDIOSTREAMCMD_ENABLE); - - if (RT_SUCCESS(rc)) - pGstStrmOut->State.fActive = fEnable; - } - else /* Disable */ - { - if (pHstStrmOut->fStatus & PDMAUDIOSTRMSTS_FLAG_ENABLED) - { - size_t cGstStrmsActive = 0; - - /* - * Check if there are any active guest streams assigned - * to this host stream which still are being marked as active. - * - * In that case we have to defer closing the host stream and - * wait until all guest streams have been finished. - */ - PPDMAUDIOGSTSTRMOUT pIter; - RTListForEach(&pHstStrmOut->lstGstStrmOut, pIter, PDMAUDIOGSTSTRMOUT, Node) - { - if (pIter->State.fActive) - { - cGstStrmsActive++; - break; /* At least one assigned & active guest stream is enough. */ - } - } - - /* Do we need to defer closing the host stream? */ - if (cGstStrmsActive) - { - LogFlowFunc(("Closing stream deferred: %zu guest stream(s) active\n", cGstStrmsActive)); - pHstStrmOut->fStatus |= PDMAUDIOSTRMSTS_FLAG_PENDING_DISABLE; - - rc = VERR_AUDIO_STREAM_PENDING_DISABLE; - } - else - { - rc = drvAudioControlHstOut(pThis, pHstStrmOut, PDMAUDIOSTREAMCMD_DISABLE); - if (RT_SUCCESS(rc)) - pGstStrmOut->State.fActive = fEnable; - } - } - } - - LogFlowFunc(("%s: fEnable=%RTbool, fStatus=0x%x, rc=%Rrc\n", - pGstStrmOut->MixBuf.pszName, fEnable, pHstStrmOut->fStatus, rc)); - - return rc; -} -#endif - -static DECLCALLBACK(int) drvAudioStreamCreate(PPDMIAUDIOCONNECTOR pInterface, - PPDMAUDIOSTREAMCFG pCfgHost, PPDMAUDIOSTREAMCFG pCfgGuest, - PPDMAUDIOSTREAM *ppStream) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pCfgHost, VERR_INVALID_POINTER); - AssertPtrReturn(pCfgGuest, VERR_INVALID_POINTER); - AssertPtrReturn(ppStream, VERR_INVALID_POINTER); - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_FAILURE(rc)) - return rc; - - LogFlowFunc(("Host=%s, Guest=%s\n", pCfgHost->szName, pCfgGuest->szName)); -#ifdef DEBUG - DrvAudioHlpStreamCfgPrint(pCfgHost); - DrvAudioHlpStreamCfgPrint(pCfgGuest); -#endif - - /* - * The guest stream always will get the audio stream configuration told - * by the device emulation (which in turn was/could be set by the guest OS). - */ - PPDMAUDIOSTREAM pGstStrm = NULL; - - /** @todo Docs! */ - PPDMAUDIOSTREAM pHstStrm = NULL; - -#define RC_BREAK(x) { rc = x; break; } - - do - { - if ( !DrvAudioHlpStreamCfgIsValid(pCfgHost) - || !DrvAudioHlpStreamCfgIsValid(pCfgGuest)) - { - RC_BREAK(VERR_INVALID_PARAMETER); - } - - /* Make sure that both configurations actually intend the same thing. */ - if (pCfgHost->enmDir != pCfgGuest->enmDir) - { - AssertMsgFailed(("Stream configuration directions do not match\n")); - RC_BREAK(VERR_INVALID_PARAMETER); - } - - /* Note: cbHstStrm will contain sizeof(PDMAUDIOSTREAM) + additional data - * which the host backend will need. */ - size_t cbHstStrm; - if (pCfgHost->enmDir == PDMAUDIODIR_IN) - { - if (!pThis->cStreamsFreeIn) - { - LogFlowFunc(("No more input streams free to use, bailing out\n")); - RC_BREAK(VERR_AUDIO_NO_FREE_INPUT_STREAMS); - } - - /* Validate backend configuration. */ - if (!pThis->BackendCfg.cbStreamIn) - { - LogFlowFunc(("Backend input configuration not valid, bailing out\n")); - RC_BREAK(VERR_INVALID_PARAMETER); - } - - cbHstStrm = pThis->BackendCfg.cbStreamIn; - } - else /* Out */ - { - if (!pThis->cStreamsFreeOut) - { - LogFlowFunc(("Maximum number of host output streams reached\n")); - RC_BREAK(VERR_AUDIO_NO_FREE_OUTPUT_STREAMS); - } - - /* Validate backend configuration. */ - if (!pThis->BackendCfg.cbStreamOut) - { - LogFlowFunc(("Backend output configuration invalid, bailing out\n")); - RC_BREAK(VERR_INVALID_PARAMETER); - } - - cbHstStrm = pThis->BackendCfg.cbStreamOut; - } - - pHstStrm = (PPDMAUDIOSTREAM)RTMemAllocZ(cbHstStrm); - AssertPtrBreakStmt(pHstStrm, rc = VERR_NO_MEMORY); - - pHstStrm->enmCtx = PDMAUDIOSTREAMCTX_HOST; - pHstStrm->enmDir = pCfgHost->enmDir; - - RTListAppend(&pThis->lstHstStreams, &pHstStrm->Node); - - pGstStrm = (PPDMAUDIOSTREAM)RTMemAllocZ(sizeof(PDMAUDIOSTREAM)); - AssertPtrBreakStmt(pGstStrm, rc = VERR_NO_MEMORY); - - pGstStrm->enmCtx = PDMAUDIOSTREAMCTX_GUEST; - pGstStrm->enmDir = pCfgGuest->enmDir; - - RTListAppend(&pThis->lstGstStreams, &pGstStrm->Node); - - /* - * Create host stream. - */ - - RTStrPrintf(pHstStrm->szName, RT_ELEMENTS(pHstStrm->szName), "%s (Host)", - strlen(pCfgHost->szName) ? pCfgHost->szName : ""); - - /* Note: Direction is always from child -> parent. */ - uint32_t cSamples = 0; - rc = pThis->pHostDrvAudio->pfnStreamCreate(pThis->pHostDrvAudio, pHstStrm, pCfgHost, &cSamples); - if (RT_FAILURE(rc)) - { - LogFlowFunc(("Initializing host backend failed with rc=%Rrc\n", rc)); - break; - } - - pHstStrm->fStatus |= PDMAUDIOSTRMSTS_FLAG_INITIALIZED; - pHstStrm->pPair = pGstStrm; - - rc = DrvAudioHlpStreamCfgToProps(pCfgHost, &pHstStrm->Props); - AssertRCBreak(rc); - - rc = AudioMixBufInit(&pHstStrm->MixBuf, pHstStrm->szName, &pHstStrm->Props, cSamples * 4); - AssertRCBreak(rc); - - /* - * Create guest stream. - */ - - RTStrPrintf(pGstStrm->szName, RT_ELEMENTS(pGstStrm->szName), "%s (Guest)", - strlen(pCfgGuest->szName) ? pCfgGuest->szName : ""); - - rc = DrvAudioHlpStreamCfgToProps(pCfgGuest, &pGstStrm->Props); - AssertRCBreak(rc); - - rc = AudioMixBufInit(&pGstStrm->MixBuf, pGstStrm->szName, &pGstStrm->Props, cSamples * 2); - if (RT_SUCCESS(rc)) - { - if (pCfgGuest->enmDir == PDMAUDIODIR_IN) - { - /* Host (Parent) -> Guest (Child). */ - rc = AudioMixBufLinkTo(&pHstStrm->MixBuf, &pGstStrm->MixBuf); - } - else - { - /* Guest (Parent) -> Host (Child). */ - rc = AudioMixBufLinkTo(&pGstStrm->MixBuf, &pHstStrm->MixBuf); - } - } - - pGstStrm->fStatus |= PDMAUDIOSTRMSTS_FLAG_INITIALIZED; - pGstStrm->pPair = pHstStrm; - - AssertRCBreak(rc); - - } while (0); - -#undef RC_BREAK - - if (RT_FAILURE(rc)) - { - drvAudioStreamDestroyInternal(pThis, pGstStrm); - pGstStrm = NULL; - - drvAudioStreamDestroyInternal(pThis, pHstStrm); - pHstStrm = NULL; - } - else - { - /* Set initial reference counts. */ - pGstStrm->cRefs = 1; - pHstStrm->cRefs = 1; - - if (pCfgHost->enmDir == PDMAUDIODIR_IN) - { - Assert(pThis->cStreamsFreeIn); - pThis->cStreamsFreeIn--; - } - else /* Out */ - { - Assert(pThis->cStreamsFreeOut); - pThis->cStreamsFreeOut--; - } - - /* Always return the guest-side part to the device emulation. */ - *ppStream = pGstStrm; - } - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - LogFlowFuncLeaveRC(rc); - return rc; -} - -#if 1 -static DECLCALLBACK(int) drvAudioGetConfig(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOBACKENDCFG pCfg) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pCfg, VERR_INVALID_POINTER); - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_FAILURE(rc)) - return rc; - - rc = pThis->pHostDrvAudio->pfnGetConfig(pThis->pHostDrvAudio, pCfg); - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - LogFlowFuncLeaveRC(rc); - return rc; -} - -static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvAudioGetStatus(PPDMIAUDIOCONNECTOR pInterface, PDMAUDIODIR enmDir) -{ - AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN); - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - int rc = RTCritSectEnter(&pThis->CritSect); - if (RT_FAILURE(rc)) - return PDMAUDIOBACKENDSTS_UNKNOWN; - - PDMAUDIOBACKENDSTS backendSts = pThis->pHostDrvAudio->pfnGetStatus(pThis->pHostDrvAudio, enmDir); - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - LogFlowFuncLeaveRC(rc); - return backendSts; -} - -static DECLCALLBACK(PDMAUDIOSTRMSTS) drvAudioStreamGetStatus(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream) -{ - AssertPtrReturn(pInterface, false); - - if (!pStream) - return PDMAUDIOSTRMSTS_FLAG_NONE; - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - int rc2 = RTCritSectEnter(&pThis->CritSect); - AssertRC(rc2); - - PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); - PDMAUDIOSTRMSTS strmSts = pHstStream->fStatus; - - LogFlowFunc(("%s: strmSts=0x%x\n", pHstStream->szName, strmSts)); - rc2 = RTCritSectLeave(&pThis->CritSect); - AssertRC(rc2); - - return strmSts; -} - -static DECLCALLBACK(int) drvAudioStreamSetVolume(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, PPDMAUDIOVOLUME pVol) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pStream, VERR_INVALID_POINTER); - AssertPtrReturn(pVol, VERR_INVALID_POINTER); - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - LogFlowFunc(("%s: volL=%RU32, volR=%RU32, fMute=%RTbool\n", pStream->szName, pVol->uLeft, pVol->uRight, pVol->fMuted)); - - AudioMixBufSetVolume(&pStream->MixBuf, pVol); - - return VINF_SUCCESS; -} -#endif - -static DECLCALLBACK(int) drvAudioStreamDestroy(PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream) -{ - AssertPtrReturn(pInterface, VERR_INVALID_POINTER); - AssertPtrReturn(pStream, VERR_INVALID_POINTER); - - PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); - - int rc = RTCritSectEnter(&pThis->CritSect); - AssertRC(rc); - - PDMAUDIODIR enmDir = pStream->enmDir; - - LogFlowFunc(("%s: cRefs=%RU32\n", pStream->szName, pStream->cRefs)); - if (pStream->cRefs > 1) - rc = VERR_WRONG_ORDER; - - if (RT_SUCCESS(rc)) - { - PPDMAUDIOSTREAM pHstStream = drvAudioGetHostStream(pStream); - PPDMAUDIOSTREAM pGstStream = pHstStream ? pHstStream->pPair : pStream; - - LogFlowFunc(("\tHost : %s\n", pHstStream ? pHstStream->szName : "")); - LogFlowFunc(("\tGuest: %s\n", pGstStream ? pGstStream->szName : "")); - - /* Should prevent double frees. */ - Assert(pHstStream != pGstStream); - - if (pHstStream) - { - pHstStream->pPair = NULL; - RTListNodeRemove(&pHstStream->Node); - } - - if (pGstStream) - { - pGstStream->pPair = NULL; - RTListNodeRemove(&pGstStream->Node); - } - - if (pHstStream) - { - rc = drvAudioStreamDestroyInternal(pThis, pHstStream); - AssertRC(rc); - - pHstStream = NULL; - } - - if (pGstStream) - { - rc = drvAudioStreamDestroyInternal(pThis, pGstStream); - AssertRC(rc); - - pGstStream = NULL; - } - } - - if (RT_SUCCESS(rc)) - { - if (enmDir == PDMAUDIODIR_IN) - { - pThis->cStreamsFreeIn++; - } - else /* Out */ - { - pThis->cStreamsFreeOut++; - } - } - - int rc2 = RTCritSectLeave(&pThis->CritSect); - if (RT_SUCCESS(rc)) - rc = rc2; - - LogFlowFuncLeaveRC(rc); - return rc; -} - -static int drvAudioStreamDestroyInternalBackend(PDRVAUDIO pThis, PPDMAUDIOSTREAM pHstStream) -{ - AssertPtrReturn(pThis, VERR_INVALID_POINTER); - AssertPtrReturn(pHstStream, VERR_INVALID_POINTER); - - AssertMsg(pHstStream->enmCtx == PDMAUDIOSTREAMCTX_HOST, - ("Stream '%s' is not a host stream and therefore has no backend\n", pHstStream->szName)); - - int rc = VINF_SUCCESS; - - LogFlowFunc(("%s: fStatus=0x%x\n", pHstStream->szName, pHstStream->fStatus)); - - if (pHstStream->fStatus & PDMAUDIOSTRMSTS_FLAG_INITIALIZED) - { - if (pThis->pHostDrvAudio) - rc = pThis->pHostDrvAudio->pfnStreamDestroy(pThis->pHostDrvAudio, pHstStream); - if (RT_SUCCESS(rc)) - pHstStream->fStatus &= ~PDMAUDIOSTRMSTS_FLAG_INITIALIZED; - } - - LogFlowFunc(("%s: Returning %Rrc\n", pHstStream->szName, rc)); - return rc; -} - -static int drvAudioStreamDestroyInternal(PDRVAUDIO pThis, PPDMAUDIOSTREAM pStream) -{ - AssertPtrReturn(pThis, VERR_INVALID_POINTER); - AssertPtrReturn(pStream, VERR_INVALID_POINTER); - - LogFlowFunc(("%s: cRefs=%RU32\n", pStream->szName, pStream->cRefs)); - - if (pStream->cRefs > 1) - return VERR_WRONG_ORDER; - - int rc = VINF_SUCCESS; - - if (pStream->enmCtx == PDMAUDIOSTREAMCTX_GUEST) - { - if (pStream->fStatus & PDMAUDIOSTRMSTS_FLAG_INITIALIZED) - { - rc = drvAudioStreamControlInternal(pThis, pStream, PDMAUDIOSTREAMCMD_DISABLE); - if (RT_SUCCESS(rc)) - pStream->fStatus &= ~PDMAUDIOSTRMSTS_FLAG_INITIALIZED; - } - } - else if (pStream->enmCtx == PDMAUDIOSTREAMCTX_HOST) - { - rc = drvAudioStreamDestroyInternalBackend(pThis, pStream); - } - else - AssertFailedReturn(VERR_NOT_IMPLEMENTED); - - if (RT_SUCCESS(rc)) - { - /* Destroy mixing buffer. */ - AudioMixBufDestroy(&pStream->MixBuf); - - if (pStream) - { - RTMemFree(pStream); - pStream = NULL; - } - } - - LogFlowFunc(("Returning %Rrc\n", rc)); - return rc; -} - -/********************************************************************/ - -/** - * @interface_method_impl{PDMIBASE,pfnQueryInterface} - */ -static DECLCALLBACK(void *) drvAudioQueryInterface(PPDMIBASE pInterface, const char *pszIID) -{ - LogFlowFunc(("pInterface=%p, pszIID=%s\n", pInterface, pszIID)); - - PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface); - PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); - - PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase); - PDMIBASE_RETURN_INTERFACE(pszIID, PDMIAUDIOCONNECTOR, &pThis->IAudioConnector); - - return NULL; -} - -/** - * Power Off notification. - * - * @param pDrvIns The driver instance data. - */ -static DECLCALLBACK(void) drvAudioPowerOff(PPDMDRVINS pDrvIns) -{ - PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); - - LogFlowFuncEnter(); - - /* Just destroy the host stream on the backend side. - * The rest will either be destructed by the device emulation or - * in drvAudioDestruct(). */ - PPDMAUDIOSTREAM pStream; - RTListForEach(&pThis->lstHstStreams, pStream, PDMAUDIOSTREAM, Node) - drvAudioStreamDestroyInternalBackend(pThis, pStream); - - /* - * Last call for the driver below us. - * Let it know that we reached end of life. - */ - if (pThis->pHostDrvAudio->pfnShutdown) - pThis->pHostDrvAudio->pfnShutdown(pThis->pHostDrvAudio); - - pThis->pHostDrvAudio = NULL; - - LogFlowFuncLeave(); -} - -/** - * Constructs an audio driver instance. - * - * @copydoc FNPDMDRVCONSTRUCT - */ -static DECLCALLBACK(int) drvAudioConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfgHandle, uint32_t fFlags) -{ - LogFlowFunc(("pDrvIns=%#p, pCfgHandle=%#p, fFlags=%x\n", pDrvIns, pCfgHandle, fFlags)); - - PDRVAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVAUDIO); - PDMDRV_CHECK_VERSIONS_RETURN(pDrvIns); /* @@ -2095,4 +1619,5 @@ pThis->IAudioConnector.pfnStreamSetVolume = drvAudioStreamSetVolume; pThis->IAudioConnector.pfnStreamPlay = drvAudioStreamPlay; + pThis->IAudioConnector.pfnStreamCapture = drvAudioStreamCapture; #ifdef VBOX_WITH_AUDIO_CALLBACKS pThis->IAudioConnector.pfnRegisterCallbacks = drvAudioRegisterCallbacks; @@ -2145,4 +1670,11 @@ int rc2 = RTCritSectEnter(&pThis->CritSect); AssertRC(rc2); + + PPDMAUDIOSTREAM pStream, pStreamNext; + RTListForEachSafe(&pThis->lstHstStreams, pStream, pStreamNext, PDMAUDIOSTREAM, Node) + drvAudioStreamDestroyInternal(pThis, pStream); + + RTListForEachSafe(&pThis->lstGstStreams, pStream, pStreamNext, PDMAUDIOSTREAM, Node) + drvAudioStreamDestroyInternal(pThis, pStream); /* Index: /trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp (revision 61522) +++ /trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp (revision 61523) @@ -94,4 +94,6 @@ void *pvBuf; size_t cbBuf; + /** Minimum samples required for ALSA to play data. */ + uint32_t cSamplesMin; } ALSAAUDIOSTREAMOUT, *PALSAAUDIOSTREAMOUT; @@ -171,10 +173,15 @@ typedef struct ALSAAUDIOSTREAMCFG { - unsigned int freq; - snd_pcm_format_t fmt; - int nchannels; - unsigned long buffer_size; - unsigned long period_size; - snd_pcm_uframes_t samples; + unsigned int freq; + /** PCM sound format. */ + snd_pcm_format_t fmt; + /** PCM data access type. */ + snd_pcm_access_t access; + /** Whether resampling should be performed by alsalib or not. */ + int resample; + int nchannels; + unsigned long buffer_size; + unsigned long period_size; + snd_pcm_uframes_t samples; } ALSAAUDIOSTREAMCFG, *PALSAAUDIOSTREAMCFG; @@ -348,4 +355,13 @@ } + err = snd_pcm_sw_params_set_avail_min(phPCM, pSWParms, 512); + if (err < 0) + { + LogRel(("ALSA: Failed to set available minimum to %ld: %s\n", + threshold, snd_strerror(err))); + rc = VERR_ACCESS_DENIED; + break; + } + err = snd_pcm_sw_params(phPCM, pSWParms); if (err < 0) @@ -385,4 +401,195 @@ return rc; } + +#if 0 /* After Beta. */ +static int alsaSetHWParams(snd_pcm_t *phPCM, PALSAAUDIOSTREAMCFG pCfg) +{ + int rc; + snd_pcm_hw_params_t *pParams = NULL; + + do + { + snd_pcm_hw_params_alloca(&pParams); + if (!pParams) + { + rc = VERR_NO_MEMORY; + break; + } + + unsigned int rrate; + snd_pcm_uframes_t size; + int dir; + + /* choose all parameters */ + int err = snd_pcm_hw_params_any(phPCM, pParams); + if (err < 0) + { + LogRel(("ALSA: Broken configuration for playback: no configurations available: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* set hardware resampling */ + err = snd_pcm_hw_params_set_rate_resample(phPCM, pParams, pCfg->resample); + if (err < 0) + { + LogRel(("ALSA: Resampling setup failed for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* set the interleaved read/write format */ + err = snd_pcm_hw_params_set_access(phPCM, pParams, pCfg->access); + if (err < 0) + { + LogRel(("ALSA: Access type not available for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* set the sample format */ + err = snd_pcm_hw_params_set_format(phPCM, pParams, pCfg->fmt); + if (err < 0) + { + LogRel(("ALSA: Sample format not available for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* set the count of channels */ + err = snd_pcm_hw_params_set_channels(phPCM, pParams, pCfg->nchannels); + if (err < 0) + { + LogRel(("ALSA: Channels count (%d) not available for playbacks: %s\n", pCfg->nchannels, snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* set the stream rate */ + rrate = pCfg->freq; + err = snd_pcm_hw_params_set_rate_near(phPCM, pParams, &rrate, 0); + if (err < 0) + { + LogRel(("ALSA: Rate %uHz not available for playback: %s\n", pCfg->freq, snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + if (rrate != pCfg->freq) + { + LogRel(("ALSA: Rate doesn't match (requested %iHz, get %uHz)\n", pCfg->freq, err)); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* set the buffer time */ + err = snd_pcm_hw_params_set_buffer_time_near(phPCM, pParams, &pCfg->buffer_time, &dir); + if (err < 0) + { + LogRel(("ALSA: Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + err = snd_pcm_hw_params_get_buffer_size(pParams, &size); + if (err < 0) + { + LogRel(("ALSA: Unable to get buffer size for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + buffer_size = size; + /* set the period time */ + err = snd_pcm_hw_params_set_period_time_near(phPCM, pParams, &period_time, &dir); + if (err < 0) + { + LogRel(("ALSA: Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + err = snd_pcm_hw_params_get_period_size(pParams, &size, &dir); + if (err < 0) + { + LogRel(("ALSA: Unable to get period size for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + period_size = size; + /* write the parameters to device */ + err = snd_pcm_hw_params(phPCM, pParams); + if (err < 0) + { + LogRel(("ALSA: Unable to set hw params for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + + rc = VINF_SUCCESS; + + } while (0); + + if (pParams) + { + snd_pcm_hw_params_free(pParams); + pParams = NULL; + } + + LogFlowFuncLeaveRC(rc); + return rc; +} +static int alsaSetSWParams(snd_pcm_t *phPCM, PALSAAUDIOCFG pCfg) +{ + int rc; + snd_pcm_sw_params_t *pParams = NULL; + + do + { + snd_pcm_sw_params_alloca(&pParams); + if (!pParams) + { + rc = VERR_NO_MEMORY; + break; + } + /* get the current swparams */ + int err = snd_pcm_sw_params_current(phPCM, pParams); + if (err < 0) + { + LogRel(("ALSA: Unable to determine current swparams for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* start the transfer when the buffer is almost full: */ + /* (buffer_size / avail_min) * avail_min */ + err = snd_pcm_sw_params_set_start_threshold(phPCM, pParams, (buffer_size / period_size) * period_size); + if (err < 0) + { + LogRel(("ALSA: Unable to set start threshold mode for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* allow the transfer when at least period_size samples can be processed */ + /* or disable this mechanism when period event is enabled (aka interrupt like style processing) */ + err = snd_pcm_sw_params_set_avail_min(phPCM, pParams, period_size); + if (err < 0) + { + LogRel(("ALSA: Unable to set avail min for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + /* write the parameters to the playback device */ + err = snd_pcm_sw_params(phPCM, pParams); + if (err < 0) + { + LogRel(("ALSA: Unable to set sw params for playback: %s\n", snd_strerror(err))); + rc = VERR_AUDIO_BACKEND_INIT_FAILED; + break; + } + + rc = VINF_SUCCESS; + + } while (0); + + if (pParams) + { + snd_pcm_sw_params_free(pParams); + pParams = NULL; + } + + LogFlowFuncLeaveRC(rc); + return rc; +} +#endif static int alsaStreamOpen(bool fIn, PALSAAUDIOSTREAMCFG pCfgReq, PALSAAUDIOSTREAMCFG pCfgObt, snd_pcm_t **pphPCM) @@ -614,4 +821,6 @@ } + LogFunc(("Buffer sample size is: %RU32\n", obt_buffer_size)); + snd_pcm_uframes_t obt_period_size; int dir = 0; @@ -983,5 +1192,5 @@ case 0: { - LogFunc(("Failed to write %RI32 frames\n", cRead)); + LogFunc(("Failed to write %RU32 samples\n", cRead)); rc = VERR_ACCESS_DENIED; break; @@ -1149,5 +1358,5 @@ if (pcSamples) - *pcSamples = obt.samples; + *pcSamples = obt.samples * 4; } while (0); @@ -1445,9 +1654,9 @@ snd_pcm_sframes_t cAvail; int rc2 = alsaStreamGetAvail(pStreamOut->phPCM, &cAvail); - if ( RT_SUCCESS(rc2) - && cAvail >= 1024) /** @todo !!! HACK ALERT !!! Use bufsize. */ + if (RT_SUCCESS(rc2)) { LogFlowFunc(("cAvail=%ld\n", cAvail)); - strmSts |= PDMAUDIOSTRMSTS_FLAG_DATA_WRITABLE; + if (cAvail >= pStreamOut->cSamplesMin) + strmSts |= PDMAUDIOSTRMSTS_FLAG_DATA_WRITABLE; } } Index: /trunk/src/VBox/Devices/Audio/DrvHostPulseAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostPulseAudio.cpp (revision 61522) +++ /trunk/src/VBox/Devices/Audio/DrvHostPulseAudio.cpp (revision 61523) @@ -1417,4 +1417,6 @@ | PDMAUDIOSTRMSTS_FLAG_ENABLED; + pa_threaded_mainloop_lock(pThis->pMainLoop); + pa_context_state_t ctxState = pa_context_get_state(pThis->pContext); @@ -1422,19 +1424,25 @@ && pa_stream_get_state(pStrm->pPAStream) == PA_STREAM_READY) { + size_t cbSize; + if (pStream->enmDir == PDMAUDIODIR_IN) { - + cbSize = pa_stream_readable_size(pStrm->pPAStream); + + if (cbSize) + strmSts |= PDMAUDIOSTRMSTS_FLAG_DATA_READABLE; } else { - size_t cbSize = pa_stream_writable_size(pStrm->pPAStream); - LogFlowFunc(("cbSize=%zu\n", cbSize)); + cbSize = pa_stream_writable_size(pStrm->pPAStream); if (cbSize >= pStrm->BufAttr.minreq) - { strmSts |= PDMAUDIOSTRMSTS_FLAG_DATA_WRITABLE; - } - } - } + } + + LogFlowFunc(("cbSize=%zu\n", cbSize)); + } + + pa_threaded_mainloop_unlock(pThis->pMainLoop); return strmSts; Index: /trunk/src/VBox/Devices/Audio/alsa_mangling.h =================================================================== --- /trunk/src/VBox/Devices/Audio/alsa_mangling.h (revision 61522) +++ /trunk/src/VBox/Devices/Audio/alsa_mangling.h (revision 61523) @@ -56,4 +56,5 @@ #define snd_pcm_sw_params_current ALSA_MANGLER(snd_pcm_sw_params_current) #define snd_pcm_sw_params_set_start_threshold ALSA_MANGLER(snd_pcm_sw_params_set_start_threshold) +#define snd_pcm_sw_params_set_avail_min ALSA_MANGLER(snd_pcm_sw_params_set_avail_min) #endif /* !AUDIO_ALSA_MANGLING_H */ Index: /trunk/src/VBox/Devices/Audio/alsa_stubs.c =================================================================== --- /trunk/src/VBox/Devices/Audio/alsa_stubs.c (revision 61522) +++ /trunk/src/VBox/Devices/Audio/alsa_stubs.c (revision 61523) @@ -116,4 +116,7 @@ (pcm, params)) PROXY_STUB(snd_pcm_sw_params_set_start_threshold, int, + (snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val), + (pcm, params, val)) +PROXY_STUB(snd_pcm_sw_params_set_avail_min, int, (snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val), (pcm, params, val)) @@ -162,4 +165,5 @@ ELEMENT(snd_pcm_sw_params_current), ELEMENT(snd_pcm_sw_params_set_start_threshold), + ELEMENT(snd_pcm_sw_params_set_avail_min) }; #undef ELEMENT