/* $Id: DrvHostPulseAudio.cpp 61523 2016-06-07 09:47:21Z vboxsync $ */ /** @file * VBox audio devices: Pulse Audio audio driver. */ /* * Copyright (C) 2006-2016 Oracle Corporation * * This file is part of VirtualBox Open Source Edition (OSE), as * available from http://www.virtualbox.org. This file is free software; * you can redistribute it and/or modify it under the terms of the GNU * General Public License (GPL) as published by the Free Software * Foundation, in version 2 as it comes in the "COPYING" file of the * VirtualBox OSE distribution. VirtualBox OSE is distributed in the * hope that it will be useful, but WITHOUT ANY WARRANTY of any kind. */ /********************************************************************************************************************************* * Header Files * *********************************************************************************************************************************/ #define LOG_GROUP LOG_GROUP_DRV_HOST_AUDIO #include #include #include #include #include #include RT_C_DECLS_BEGIN #include "pulse_mangling.h" #include "pulse_stubs.h" RT_C_DECLS_END #include #include "DrvAudio.h" #include "AudioMixBuffer.h" #include "VBoxDD.h" /********************************************************************************************************************************* * Defines * *********************************************************************************************************************************/ #define VBOX_PULSEAUDIO_MAX_LOG_REL_ERRORS 32 /** @todo Make this configurable thru driver options. */ #ifndef PA_STREAM_NOFLAGS # define PA_STREAM_NOFLAGS (pa_context_flags_t)0x0000U /* since 0.9.19 */ #endif #ifndef PA_CONTEXT_NOFLAGS # define PA_CONTEXT_NOFLAGS (pa_context_flags_t)0x0000U /* since 0.9.19 */ #endif /** No flags specified. */ #define PULSEAUDIOENUMCBFLAGS_NONE 0 /** (Release) log found devices. */ #define PULSEAUDIOENUMCBFLAGS_LOG RT_BIT(0) /** Makes DRVHOSTPULSEAUDIO out of PDMIHOSTAUDIO. */ #define PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface) \ ( (PDRVHOSTPULSEAUDIO)((uintptr_t)pInterface - RT_OFFSETOF(DRVHOSTPULSEAUDIO, IHostAudio)) ) /********************************************************************************************************************************* * Structures * *********************************************************************************************************************************/ /** * Host Pulse audio driver instance data. * @implements PDMIAUDIOCONNECTOR */ typedef struct DRVHOSTPULSEAUDIO { /** Pointer to the driver instance structure. */ PPDMDRVINS pDrvIns; /** Pointer to PulseAudio's threaded main loop. */ pa_threaded_mainloop *pMainLoop; /** * Pointer to our PulseAudio context. * Note: We use a pMainLoop in a separate thread (pContext). * So either use callback functions or protect these functions * by pa_threaded_mainloop_lock() / pa_threaded_mainloop_unlock(). */ pa_context *pContext; /** Shutdown indicator. */ bool fLoopWait; /** Pointer to host audio interface. */ PDMIHOSTAUDIO IHostAudio; /** Error count for not flooding the release log. * Specify UINT32_MAX for unlimited logging. */ uint32_t cLogErrors; } DRVHOSTPULSEAUDIO, *PDRVHOSTPULSEAUDIO; typedef struct PULSEAUDIOSTREAM { /** Associated host input/output stream. * Note: Always must come first! */ PDMAUDIOSTREAM Stream; /** Pointer to driver instance. */ PDRVHOSTPULSEAUDIO pDrv; /** DAC/ADC buffer. */ void *pvPCMBuf; /** Size (in bytes) of DAC/ADC buffer. */ uint32_t cbPCMBuf; /** Pointer to opaque PulseAudio stream. */ pa_stream *pPAStream; /** Pulse sample format and attribute specification. */ pa_sample_spec SampleSpec; /** Pulse playback and buffer metrics. */ pa_buffer_attr BufAttr; int fOpSuccess; /** Pointer to Pulse sample peeking buffer. */ const uint8_t *pu8PeekBuf; /** Current size (in bytes) of peeking data in * buffer. */ size_t cbPeekBuf; /** Our offset (in bytes) in peeking buffer. */ size_t offPeekBuf; pa_operation *pDrainOp; } PULSEAUDIOSTREAM, *PPULSEAUDIOSTREAM; /* The desired buffer length in milliseconds. Will be the target total stream * latency on newer version of pulse. Apparent latency can be less (or more.) */ typedef struct PULSEAUDIOCFG { RTMSINTERVAL buffer_msecs_out; RTMSINTERVAL buffer_msecs_in; } PULSEAUDIOCFG, *PPULSEAUDIOCFG; static PULSEAUDIOCFG s_pulseCfg = { 100, /* buffer_msecs_out */ 100 /* buffer_msecs_in */ }; /** * Callback context for server enumeration callbacks. */ typedef struct PULSEAUDIOENUMCBCTX { /** Pointer to host backend driver. */ PDRVHOSTPULSEAUDIO pDrv; /** Enumeration flags. */ uint32_t fFlags; /** Number of found input devices. */ uint8_t cDevIn; /** Number of found output devices. */ uint8_t cDevOut; /** Name of default sink being used. Must be free'd using RTStrFree(). */ char *pszDefaultSink; /** Name of default source being used. Must be free'd using RTStrFree(). */ char *pszDefaultSource; } PULSEAUDIOENUMCBCTX, *PPULSEAUDIOENUMCBCTX; /********************************************************************************************************************************* * Prototypes * *********************************************************************************************************************************/ static int paEnumerate(PDRVHOSTPULSEAUDIO pThis, PPDMAUDIOBACKENDCFG pCfg, uint32_t fEnum); static int paError(PDRVHOSTPULSEAUDIO pThis, const char *szMsg); static void paStreamCbSuccess(pa_stream *pStream, int fSuccess, void *pvContext); /** * Signal the main loop to abort. Just signalling isn't sufficient as the * mainloop might not have been entered yet. */ static void paSignalWaiter(PDRVHOSTPULSEAUDIO pThis) { if (!pThis) return; pThis->fLoopWait = true; pa_threaded_mainloop_signal(pThis->pMainLoop, 0); } static pa_sample_format_t paFmtToPulse(PDMAUDIOFMT fmt) { switch (fmt) { case PDMAUDIOFMT_U8: return PA_SAMPLE_U8; case PDMAUDIOFMT_S16: return PA_SAMPLE_S16LE; #ifdef PA_SAMPLE_S32LE case PDMAUDIOFMT_S32: return PA_SAMPLE_S32LE; #endif default: break; } AssertMsgFailed(("Format %ld not supported\n", fmt)); return PA_SAMPLE_U8; } static int paPulseToFmt(pa_sample_format_t pulsefmt, PDMAUDIOFMT *pFmt, PDMAUDIOENDIANNESS *pEndianness) { switch (pulsefmt) { case PA_SAMPLE_U8: *pFmt = PDMAUDIOFMT_U8; *pEndianness = PDMAUDIOENDIANNESS_LITTLE; break; case PA_SAMPLE_S16LE: *pFmt = PDMAUDIOFMT_S16; *pEndianness = PDMAUDIOENDIANNESS_LITTLE; break; case PA_SAMPLE_S16BE: *pFmt = PDMAUDIOFMT_S16; *pEndianness = PDMAUDIOENDIANNESS_BIG; break; #ifdef PA_SAMPLE_S32LE case PA_SAMPLE_S32LE: *pFmt = PDMAUDIOFMT_S32; *pEndianness = PDMAUDIOENDIANNESS_LITTLE; break; #endif #ifdef PA_SAMPLE_S32BE case PA_SAMPLE_S32BE: *pFmt = PDMAUDIOFMT_S32; *pEndianness = PDMAUDIOENDIANNESS_BIG; break; #endif default: AssertMsgFailed(("Format %ld not supported\n", pulsefmt)); return VERR_NOT_SUPPORTED; } return VINF_SUCCESS; } /** * Synchronously wait until an operation completed. */ static int paWaitForEx(PDRVHOSTPULSEAUDIO pThis, pa_operation *pOP, RTMSINTERVAL cMsTimeout) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pOP, VERR_INVALID_POINTER); int rc = VINF_SUCCESS; uint64_t u64StartMs = RTTimeMilliTS(); while (pa_operation_get_state(pOP) == PA_OPERATION_RUNNING) { if (!pThis->fLoopWait) { AssertPtr(pThis->pMainLoop); pa_threaded_mainloop_wait(pThis->pMainLoop); } pThis->fLoopWait = false; uint64_t u64ElapsedMs = RTTimeMilliTS() - u64StartMs; if (u64ElapsedMs >= cMsTimeout) { rc = VERR_TIMEOUT; break; } } pa_operation_unref(pOP); return rc; } static int paWaitFor(PDRVHOSTPULSEAUDIO pThis, pa_operation *pOP) { return paWaitForEx(pThis, pOP, 10 * 1000 /* 10s timeout */); } /** * Context status changed. */ static void paContextCbStateChanged(pa_context *pCtx, void *pvUser) { AssertPtrReturnVoid(pCtx); PDRVHOSTPULSEAUDIO pThis = (PDRVHOSTPULSEAUDIO)pvUser; AssertPtrReturnVoid(pThis); switch (pa_context_get_state(pCtx)) { case PA_CONTEXT_READY: case PA_CONTEXT_TERMINATED: paSignalWaiter(pThis); break; case PA_CONTEXT_FAILED: LogRel(("PulseAudio: Audio context has failed, stopping\n")); paSignalWaiter(pThis); break; default: break; } } /** * Callback called when our pa_stream_drain operation was completed. */ static void paStreamCbDrain(pa_stream *pStream, int fSuccess, void *pvUser) { AssertPtrReturnVoid(pStream); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pvUser; AssertPtrReturnVoid(pStrm); pStrm->fOpSuccess = fSuccess; if (fSuccess) { pa_operation_unref(pa_stream_cork(pStream, 1, paStreamCbSuccess, pvUser)); } else paError(pStrm->pDrv, "Failed to drain stream"); if (pStrm->pDrainOp) { pa_operation_unref(pStrm->pDrainOp); pStrm->pDrainOp = NULL; } } /** * Stream status changed. */ static void paStreamCbStateChanged(pa_stream *pStream, void *pvUser) { AssertPtrReturnVoid(pStream); PDRVHOSTPULSEAUDIO pThis = (PDRVHOSTPULSEAUDIO)pvUser; AssertPtrReturnVoid(pThis); switch (pa_stream_get_state(pStream)) { case PA_STREAM_READY: case PA_STREAM_FAILED: case PA_STREAM_TERMINATED: paSignalWaiter(pThis); break; default: break; } } static void paStreamCbSuccess(pa_stream *pStream, int fSuccess, void *pvUser) { AssertPtrReturnVoid(pStream); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pvUser; AssertPtrReturnVoid(pStrm); pStrm->fOpSuccess = fSuccess; if (fSuccess) paSignalWaiter(pStrm->pDrv); else paError(pStrm->pDrv, "Failed to finish stream operation"); } static int paStreamOpen(PDRVHOSTPULSEAUDIO pThis, bool fIn, const char *pszName, pa_sample_spec *pSampleSpec, pa_buffer_attr *pBufAttr, pa_stream **ppStream) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pszName, VERR_INVALID_POINTER); AssertPtrReturn(pSampleSpec, VERR_INVALID_POINTER); AssertPtrReturn(pBufAttr, VERR_INVALID_POINTER); AssertPtrReturn(ppStream, VERR_INVALID_POINTER); if (!pa_sample_spec_valid(pSampleSpec)) { LogRel(("PulseAudio: Unsupported sample specification for stream \"%s\"\n", pszName)); return VERR_NOT_SUPPORTED; } int rc = VINF_SUCCESS; pa_stream *pStream = NULL; uint32_t flags = PA_STREAM_NOFLAGS; LogFunc(("Opening \"%s\", rate=%dHz, channels=%d, format=%s\n", pszName, pSampleSpec->rate, pSampleSpec->channels, pa_sample_format_to_string(pSampleSpec->format))); pa_threaded_mainloop_lock(pThis->pMainLoop); do { /** @todo r=andy Use pa_stream_new_with_proplist instead. */ if (!(pStream = pa_stream_new(pThis->pContext, pszName, pSampleSpec, NULL /* pa_channel_map */))) { LogRel(("PulseAudio: Could not create stream \"%s\"\n", pszName)); rc = VERR_NO_MEMORY; break; } pa_stream_set_state_callback(pStream, paStreamCbStateChanged, pThis); #if PA_API_VERSION >= 12 /* XXX */ flags |= PA_STREAM_ADJUST_LATENCY; #endif #if 0 /* Not applicable as we don't use pa_stream_get_latency() and pa_stream_get_time(). */ flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE; #endif /* No input/output right away after the stream was started. */ flags |= PA_STREAM_START_CORKED; if (fIn) { LogFunc(("Input stream attributes: maxlength=%d fragsize=%d\n", pBufAttr->maxlength, pBufAttr->fragsize)); if (pa_stream_connect_record(pStream, /*dev=*/NULL, pBufAttr, (pa_stream_flags_t)flags) < 0) { LogRel(("PulseAudio: Could not connect input stream \"%s\": %s\n", pszName, pa_strerror(pa_context_errno(pThis->pContext)))); rc = VERR_AUDIO_BACKEND_INIT_FAILED; break; } } else { LogFunc(("Output buffer attributes: maxlength=%d tlength=%d prebuf=%d minreq=%d\n", pBufAttr->maxlength, pBufAttr->tlength, pBufAttr->prebuf, pBufAttr->minreq)); if (pa_stream_connect_playback(pStream, /*dev=*/NULL, pBufAttr, (pa_stream_flags_t)flags, /*cvolume=*/NULL, /*sync_stream=*/NULL) < 0) { LogRel(("PulseAudio: Could not connect playback stream \"%s\": %s\n", pszName, pa_strerror(pa_context_errno(pThis->pContext)))); rc = VERR_AUDIO_BACKEND_INIT_FAILED; break; } } /* Wait until the stream is ready. */ for (;;) { if (!pThis->fLoopWait) pa_threaded_mainloop_wait(pThis->pMainLoop); pThis->fLoopWait = false; pa_stream_state_t streamSt = pa_stream_get_state(pStream); if (streamSt == PA_STREAM_READY) break; else if ( streamSt == PA_STREAM_FAILED || streamSt == PA_STREAM_TERMINATED) { LogRel(("PulseAudio: Failed to initialize stream \"%s\" (state %ld)\n", pszName, streamSt)); rc = VERR_AUDIO_BACKEND_INIT_FAILED; break; } } if (RT_FAILURE(rc)) break; const pa_buffer_attr *pBufAttrObtained = pa_stream_get_buffer_attr(pStream); AssertPtr(pBufAttrObtained); memcpy(pBufAttr, pBufAttrObtained, sizeof(pa_buffer_attr)); if (fIn) LogFunc(("Obtained record buffer attributes: maxlength=%RU32, fragsize=%RU32\n", pBufAttr->maxlength, pBufAttr->fragsize)); else LogFunc(("Obtained playback buffer attributes: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d\n", pBufAttr->maxlength, pBufAttr->tlength, pBufAttr->prebuf, pBufAttr->minreq)); } while (0); if ( RT_FAILURE(rc) && pStream) pa_stream_disconnect(pStream); pa_threaded_mainloop_unlock(pThis->pMainLoop); if (RT_FAILURE(rc)) { if (pStream) pa_stream_unref(pStream); } else *ppStream = pStream; LogFlowFuncLeaveRC(rc); return rc; } static DECLCALLBACK(int) drvHostPulseAudioInit(PPDMIHOSTAUDIO pInterface) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); LogFlowFuncEnter(); int rc = audioLoadPulseLib(); if (RT_FAILURE(rc)) { LogRel(("PulseAudio: Failed to load the PulseAudio shared library! Error %Rrc\n", rc)); return rc; } pThis->fLoopWait = false; pThis->pMainLoop = NULL; bool fLocked = false; do { if (!(pThis->pMainLoop = pa_threaded_mainloop_new())) { LogRel(("PulseAudio: Failed to allocate main loop: %s\n", pa_strerror(pa_context_errno(pThis->pContext)))); rc = VERR_NO_MEMORY; break; } if (!(pThis->pContext = pa_context_new(pa_threaded_mainloop_get_api(pThis->pMainLoop), "VirtualBox"))) { LogRel(("PulseAudio: Failed to allocate context: %s\n", pa_strerror(pa_context_errno(pThis->pContext)))); rc = VERR_NO_MEMORY; break; } if (pa_threaded_mainloop_start(pThis->pMainLoop) < 0) { LogRel(("PulseAudio: Failed to start threaded mainloop: %s\n", pa_strerror(pa_context_errno(pThis->pContext)))); rc = VERR_AUDIO_BACKEND_INIT_FAILED; break; } /* Install a global callback to known if something happens to our acquired context. */ pa_context_set_state_callback(pThis->pContext, paContextCbStateChanged, pThis /* pvUserData */); pa_threaded_mainloop_lock(pThis->pMainLoop); fLocked = true; if (pa_context_connect(pThis->pContext, NULL /* pszServer */, PA_CONTEXT_NOFLAGS, NULL) < 0) { LogRel(("PulseAudio: Failed to connect to server: %s\n", pa_strerror(pa_context_errno(pThis->pContext)))); rc = VERR_AUDIO_BACKEND_INIT_FAILED; break; } /* Wait until the pThis->pContext is ready. */ for (;;) { if (!pThis->fLoopWait) pa_threaded_mainloop_wait(pThis->pMainLoop); pThis->fLoopWait = false; pa_context_state_t cstate = pa_context_get_state(pThis->pContext); if (cstate == PA_CONTEXT_READY) break; else if ( cstate == PA_CONTEXT_TERMINATED || cstate == PA_CONTEXT_FAILED) { LogRel(("PulseAudio: Failed to initialize context (state %d)\n", cstate)); rc = VERR_AUDIO_BACKEND_INIT_FAILED; break; } } } while (0); if (fLocked) pa_threaded_mainloop_unlock(pThis->pMainLoop); if (RT_FAILURE(rc)) { if (pThis->pMainLoop) pa_threaded_mainloop_stop(pThis->pMainLoop); if (pThis->pContext) { pa_context_disconnect(pThis->pContext); pa_context_unref(pThis->pContext); pThis->pContext = NULL; } if (pThis->pMainLoop) { pa_threaded_mainloop_free(pThis->pMainLoop); pThis->pMainLoop = NULL; } } LogFlowFuncLeaveRC(rc); return rc; } static int paCreateStreamOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream, PPDMAUDIOSTREAMCFG pCfg, uint32_t *pcSamples) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); AssertPtrReturn(pCfg, VERR_INVALID_POINTER); /* pcSamples is optional. */ PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; LogFlowFuncEnter(); pStrm->pDrainOp = NULL; pStrm->SampleSpec.format = paFmtToPulse(pCfg->enmFormat); pStrm->SampleSpec.rate = pCfg->uHz; pStrm->SampleSpec.channels = pCfg->cChannels; /* Note that setting maxlength to -1 does not work on PulseAudio servers * older than 0.9.10. So use the suggested value of 3/2 of tlength */ pStrm->BufAttr.tlength = (pa_bytes_per_second(&pStrm->SampleSpec) * s_pulseCfg.buffer_msecs_out) / 1000; pStrm->BufAttr.maxlength = (pStrm->BufAttr.tlength * 3) / 2; pStrm->BufAttr.prebuf = -1; /* Same as tlength */ pStrm->BufAttr.minreq = -1; /* Note that the struct BufAttr is updated to the obtained values after this call! */ int rc = paStreamOpen(pThis, false /* fIn */, "PulseAudio (Out)", &pStrm->SampleSpec, &pStrm->BufAttr, &pStrm->pPAStream); if (RT_FAILURE(rc)) return rc; PDMAUDIOSTREAMCFG streamCfg; rc = paPulseToFmt(pStrm->SampleSpec.format, &streamCfg.enmFormat, &streamCfg.enmEndianness); if (RT_FAILURE(rc)) { LogRel(("PulseAudio: Cannot find audio output format %ld\n", pStrm->SampleSpec.format)); return rc; } streamCfg.uHz = pStrm->SampleSpec.rate; streamCfg.cChannels = pStrm->SampleSpec.channels; rc = DrvAudioHlpStreamCfgToProps(&streamCfg, &pStream->Props); if (RT_SUCCESS(rc)) { uint32_t cbBuf = RT_MIN(pStrm->BufAttr.tlength * 2, pStrm->BufAttr.maxlength); /** @todo Make this configurable! */ if (cbBuf) { pStrm->pvPCMBuf = RTMemAllocZ(cbBuf); if (pStrm->pvPCMBuf) { pStrm->cbPCMBuf = cbBuf; uint32_t cSamples = cbBuf >> pStream->Props.cShift; if (pcSamples) *pcSamples = cSamples; /* Save pointer to driver instance. */ pStrm->pDrv = pThis; LogFunc(("cbBuf=%RU32, cSamples=%RU32\n", cbBuf, cSamples)); } else rc = VERR_NO_MEMORY; } else rc = VERR_INVALID_PARAMETER; } LogFlowFuncLeaveRC(rc); return rc; } static int paCreateStreamIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream, PPDMAUDIOSTREAMCFG pCfg, uint32_t *pcSamples) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); AssertPtrReturn(pCfg, VERR_INVALID_POINTER); /* pcSamples is optional. */ PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; pStrm->SampleSpec.format = paFmtToPulse(pCfg->enmFormat); pStrm->SampleSpec.rate = pCfg->uHz; pStrm->SampleSpec.channels = pCfg->cChannels; /* XXX check these values */ pStrm->BufAttr.fragsize = (pa_bytes_per_second(&pStrm->SampleSpec) * s_pulseCfg.buffer_msecs_in) / 1000; pStrm->BufAttr.maxlength = (pStrm->BufAttr.fragsize * 3) / 2; /* Note: Other members of BufAttr are ignored for record streams. */ int rc = paStreamOpen(pThis, true /* fIn */, "PulseAudio (In)", &pStrm->SampleSpec, &pStrm->BufAttr, &pStrm->pPAStream); if (RT_FAILURE(rc)) return rc; PDMAUDIOSTREAMCFG streamCfg; rc = paPulseToFmt(pStrm->SampleSpec.format, &streamCfg.enmFormat, &streamCfg.enmEndianness); if (RT_FAILURE(rc)) { LogRel(("PulseAudio: Cannot find audio capture format %ld\n", pStrm->SampleSpec.format)); return rc; } streamCfg.uHz = pStrm->SampleSpec.rate; streamCfg.cChannels = pStrm->SampleSpec.channels; rc = DrvAudioHlpStreamCfgToProps(&streamCfg, &pStream->Props); if (RT_SUCCESS(rc)) { uint32_t cSamples = RT_MIN(pStrm->BufAttr.fragsize * 10, pStrm->BufAttr.maxlength) >> pStream->Props.cShift; LogFunc(("uHz=%RU32, cChannels=%RU8, cShift=%RU8, cSamples=%RU32\n", pStream->Props.uHz, pStream->Props.cChannels, pStream->Props.cShift, cSamples)); /* Save pointer to driver instance. */ pStrm->pDrv = pThis; pStrm->pu8PeekBuf = NULL; if (pcSamples) *pcSamples = cSamples; } LogFlowFuncLeaveRC(rc); return rc; } static DECLCALLBACK(int) drvHostPulseAudioStreamCapture(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesCaptured) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); /* pcSamplesPlayed is optional. */ PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; /* We should only call pa_stream_readable_size() once and trust the first value. */ pa_threaded_mainloop_lock(pThis->pMainLoop); size_t cbAvail = pa_stream_readable_size(pStrm->pPAStream); pa_threaded_mainloop_unlock(pThis->pMainLoop); if (cbAvail == (size_t)-1) return paError(pStrm->pDrv, "Failed to determine input data size"); /* If the buffer was not dropped last call, add what remains. */ if (pStrm->pu8PeekBuf) { Assert(pStrm->cbPeekBuf >= pStrm->offPeekBuf); cbAvail += (pStrm->cbPeekBuf - pStrm->offPeekBuf); } if (!cbAvail) /* No data? Bail out. */ { if (pcSamplesCaptured) *pcSamplesCaptured = 0; return VINF_SUCCESS; } int rc = VINF_SUCCESS; size_t cbToRead = RT_MIN(cbAvail, AudioMixBufFreeBytes(&pStream->MixBuf)); LogFlowFunc(("cbToRead=%zu, cbAvail=%zu, offPeekBuf=%zu, cbPeekBuf=%zu\n", cbToRead, cbAvail, pStrm->offPeekBuf, pStrm->cbPeekBuf)); size_t offWrite = 0; uint32_t cWrittenTotal = 0; while (cbToRead) { /* If there is no data, do another peek. */ if (!pStrm->pu8PeekBuf) { pa_threaded_mainloop_lock(pThis->pMainLoop); pa_stream_peek(pStrm->pPAStream, (const void**)&pStrm->pu8PeekBuf, &pStrm->cbPeekBuf); pa_threaded_mainloop_unlock(pThis->pMainLoop); pStrm->offPeekBuf = 0; /* No data anymore? * Note: If there's a data hole (cbPeekBuf then contains the length of the hole) * we need to drop the stream lateron. */ if ( !pStrm->pu8PeekBuf && !pStrm->cbPeekBuf) { break; } } Assert(pStrm->cbPeekBuf >= pStrm->offPeekBuf); size_t cbToWrite = RT_MIN(pStrm->cbPeekBuf - pStrm->offPeekBuf, cbToRead); LogFlowFunc(("cbToRead=%zu, cbToWrite=%zu, offPeekBuf=%zu, cbPeekBuf=%zu, pu8PeekBuf=%p\n", cbToRead, cbToWrite, pStrm->offPeekBuf, pStrm->cbPeekBuf, pStrm->pu8PeekBuf)); if (cbToWrite) { uint32_t cWritten; rc = AudioMixBufWriteCirc(&pStream->MixBuf, pStrm->pu8PeekBuf + pStrm->offPeekBuf, cbToWrite, &cWritten); if (RT_FAILURE(rc)) break; uint32_t cbWritten = AUDIOMIXBUF_S2B(&pStream->MixBuf, cWritten); Assert(cbToRead >= cbWritten); cbToRead -= cbWritten; cWrittenTotal += cWritten; pStrm->offPeekBuf += cbWritten; } if (/* Nothing to write anymore? Drop the buffer. */ !cbToWrite /* Was there a hole in the peeking buffer? Drop it. */ || !pStrm->pu8PeekBuf /* If the buffer is done, drop it. */ || pStrm->offPeekBuf == pStrm->cbPeekBuf) { pa_threaded_mainloop_lock(pThis->pMainLoop); pa_stream_drop(pStrm->pPAStream); pa_threaded_mainloop_unlock(pThis->pMainLoop); pStrm->pu8PeekBuf = NULL; } } if (RT_SUCCESS(rc)) { uint32_t cProcessed = 0; if (cWrittenTotal) rc = AudioMixBufMixToParent(&pStream->MixBuf, cWrittenTotal, &cProcessed); if (pcSamplesCaptured) *pcSamplesCaptured = cWrittenTotal; LogFlowFunc(("cWrittenTotal=%RU32 (%RU32 processed), rc=%Rrc\n", cWrittenTotal, cProcessed, rc)); } LogFlowFuncLeaveRC(rc); return rc; } static DECLCALLBACK(int) drvHostPulseAudioStreamPlay(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesPlayed) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); /* pcSamplesPlayed is optional. */ PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pPAStream = (PPULSEAUDIOSTREAM)pStream; int rc = VINF_SUCCESS; uint32_t cbReadTotal = 0; uint32_t cLive = AudioMixBufUsed(&pStream->MixBuf); if (!cLive) { LogFlowFunc(("No live samples, skipping\n")); if (pcSamplesPlayed) *pcSamplesPlayed = 0; return VINF_SUCCESS; } pa_threaded_mainloop_lock(pThis->pMainLoop); do { size_t cbWriteable = pa_stream_writable_size(pPAStream->pPAStream); if (cbWriteable == (size_t)-1) { rc = paError(pPAStream->pDrv, "Failed to determine output data size"); break; } size_t cbLive = AUDIOMIXBUF_S2B(&pStream->MixBuf, cLive); size_t cbToRead = RT_MIN(cbWriteable, cbLive); LogFlowFunc(("cbToRead=%zu, cbWriteable=%zu, cbLive=%zu\n", cbToRead, cbWriteable, cbLive)); uint32_t cRead, cbRead; while (cbToRead) { rc = AudioMixBufReadCirc(&pStream->MixBuf, pPAStream->pvPCMBuf, RT_MIN(cbToRead, pPAStream->cbPCMBuf), &cRead); if ( !cRead || RT_FAILURE(rc)) { break; } cbRead = AUDIOMIXBUF_S2B(&pStream->MixBuf, cRead); if (pa_stream_write(pPAStream->pPAStream, pPAStream->pvPCMBuf, cbRead, NULL /* Cleanup callback */, 0, PA_SEEK_RELATIVE) < 0) { rc = paError(pPAStream->pDrv, "Failed to write to output stream"); break; } Assert(cbToRead >= cbRead); cbToRead -= cbRead; cbReadTotal += cbRead; LogFlowFunc(("\tcRead=%RU32 (%zu bytes) cbReadTotal=%RU32, cbToRead=%RU32\n", cRead, AUDIOMIXBUF_S2B(&pStream->MixBuf, cRead), cbReadTotal, cbToRead)); } } while (0); pa_threaded_mainloop_unlock(pThis->pMainLoop); if (RT_SUCCESS(rc)) { uint32_t cReadTotal = AUDIOMIXBUF_B2S(&pStream->MixBuf, cbReadTotal); if (cReadTotal) AudioMixBufFinish(&pStream->MixBuf, cReadTotal); if (pcSamplesPlayed) *pcSamplesPlayed = cReadTotal; LogFlowFunc(("cReadTotal=%RU32 (%RU32 bytes), rc=%Rrc\n", cReadTotal, cbReadTotal, rc)); } LogFlowFuncLeaveRC(rc); return rc; } /** @todo Implement va handling. */ static int paError(PDRVHOSTPULSEAUDIO pThis, const char *szMsg) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(szMsg, VERR_INVALID_POINTER); if (pThis->cLogErrors++ < VBOX_PULSEAUDIO_MAX_LOG_REL_ERRORS) { int rc2 = pa_context_errno(pThis->pContext); LogRel2(("PulseAudio: %s: %s\n", szMsg, pa_strerror(rc2))); } /** @todo Implement some PulseAudio -> IPRT mapping here. */ return VERR_GENERAL_FAILURE; } static void paEnumSinkCb(pa_context *pCtx, const pa_sink_info *pInfo, int eol, void *pvUserData) { if (eol != 0) return; AssertPtrReturnVoid(pCtx); AssertPtrReturnVoid(pInfo); PPULSEAUDIOENUMCBCTX pCbCtx = (PPULSEAUDIOENUMCBCTX)pvUserData; AssertPtrReturnVoid(pCbCtx); AssertPtrReturnVoid(pCbCtx->pDrv); LogRel2(("PulseAudio: Using output sink '%s'\n", pInfo->name)); /** @todo Store sinks + channel mapping in callback context as soon as we have surround support. */ pCbCtx->cDevOut++; pa_threaded_mainloop_signal(pCbCtx->pDrv->pMainLoop, 0); } static void paEnumSourceCb(pa_context *pCtx, const pa_source_info *pInfo, int eol, void *pvUserData) { if (eol != 0) return; AssertPtrReturnVoid(pCtx); AssertPtrReturnVoid(pInfo); PPULSEAUDIOENUMCBCTX pCbCtx = (PPULSEAUDIOENUMCBCTX)pvUserData; AssertPtrReturnVoid(pCbCtx); AssertPtrReturnVoid(pCbCtx->pDrv); LogRel2(("PulseAudio: Using input source '%s'\n", pInfo->name)); /** @todo Store sources + channel mapping in callback context as soon as we have surround support. */ pCbCtx->cDevIn++; pa_threaded_mainloop_signal(pCbCtx->pDrv->pMainLoop, 0); } static void paEnumServerCb(pa_context *pCtx, const pa_server_info *pInfo, void *pvUserData) { AssertPtrReturnVoid(pCtx); AssertPtrReturnVoid(pInfo); PPULSEAUDIOENUMCBCTX pCbCtx = (PPULSEAUDIOENUMCBCTX)pvUserData; AssertPtrReturnVoid(pCbCtx); PDRVHOSTPULSEAUDIO pThis = pCbCtx->pDrv; AssertPtrReturnVoid(pThis); if (pInfo->default_sink_name) { Assert(RTStrIsValidEncoding(pInfo->default_sink_name)); pCbCtx->pszDefaultSink = RTStrDup(pInfo->default_sink_name); } if (pInfo->default_sink_name) { Assert(RTStrIsValidEncoding(pInfo->default_source_name)); pCbCtx->pszDefaultSource = RTStrDup(pInfo->default_source_name); } pa_threaded_mainloop_signal(pThis->pMainLoop, 0); } static int paEnumerate(PDRVHOSTPULSEAUDIO pThis, PPDMAUDIOBACKENDCFG pCfg, uint32_t fEnum) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pCfg, VERR_INVALID_POINTER); PDMAUDIOBACKENDCFG Cfg; RT_ZERO(Cfg); Cfg.cbStreamOut = sizeof(PULSEAUDIOSTREAM); Cfg.cbStreamIn = sizeof(PULSEAUDIOSTREAM); Cfg.cMaxStreamsOut = UINT32_MAX; Cfg.cMaxStreamsIn = UINT32_MAX; PULSEAUDIOENUMCBCTX cbCtx; RT_ZERO(cbCtx); cbCtx.pDrv = pThis; cbCtx.fFlags = fEnum; bool fLog = (fEnum & PULSEAUDIOENUMCBFLAGS_LOG); int rc = paWaitFor(pThis, pa_context_get_server_info(pThis->pContext, paEnumServerCb, &cbCtx)); if (RT_SUCCESS(rc)) { if (cbCtx.pszDefaultSink) { if (fLog) LogRel2(("PulseAudio: Default output sink is '%s'\n", cbCtx.pszDefaultSink)); rc = paWaitFor(pThis, pa_context_get_sink_info_by_name(pThis->pContext, cbCtx.pszDefaultSink, paEnumSinkCb, &cbCtx)); if ( RT_FAILURE(rc) && fLog) { LogRel(("PulseAudio: Error enumerating properties for default output sink '%s'\n", cbCtx.pszDefaultSink)); } } else if (fLog) LogRel2(("PulseAudio: No default output sink found\n")); if (RT_SUCCESS(rc)) { if (cbCtx.pszDefaultSource) { if (fLog) LogRel2(("PulseAudio: Default input source is '%s'\n", cbCtx.pszDefaultSource)); rc = paWaitFor(pThis, pa_context_get_source_info_by_name(pThis->pContext, cbCtx.pszDefaultSource, paEnumSourceCb, &cbCtx)); if ( RT_FAILURE(rc) && fLog) { LogRel(("PulseAudio: Error enumerating properties for default input source '%s'\n", cbCtx.pszDefaultSource)); } } else if (fLog) LogRel2(("PulseAudio: No default input source found\n")); } if (RT_SUCCESS(rc)) { Cfg.cSinks = cbCtx.cDevOut; Cfg.cSources = cbCtx.cDevIn; if (fLog) { LogRel2(("PulseAudio: Found %RU8 host playback device(s)\n", cbCtx.cDevOut)); LogRel2(("PulseAudio: Found %RU8 host capturing device(s)\n", cbCtx.cDevIn)); } if (pCfg) memcpy(pCfg, &Cfg, sizeof(PDMAUDIOBACKENDCFG)); } if (cbCtx.pszDefaultSink) { RTStrFree(cbCtx.pszDefaultSink); cbCtx.pszDefaultSink = NULL; } if (cbCtx.pszDefaultSource) { RTStrFree(cbCtx.pszDefaultSource); cbCtx.pszDefaultSource = NULL; } } else if (fLog) LogRel(("PulseAudio: Error enumerating PulseAudio server properties\n")); LogFlowFuncLeaveRC(rc); return rc; } static int paDestroyStreamIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; LogFlowFuncEnter(); if (pStrm->pPAStream) { pa_threaded_mainloop_lock(pThis->pMainLoop); pa_stream_disconnect(pStrm->pPAStream); pa_stream_unref(pStrm->pPAStream); pStrm->pPAStream = NULL; pa_threaded_mainloop_unlock(pThis->pMainLoop); } return VINF_SUCCESS; } static int paDestroyStreamOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; LogFlowFuncEnter(); if (pStrm->pPAStream) { pa_threaded_mainloop_lock(pThis->pMainLoop); /* Make sure to cancel a pending draining operation, if any. */ if (pStrm->pDrainOp) { pa_operation_cancel(pStrm->pDrainOp); pStrm->pDrainOp = NULL; } pa_stream_disconnect(pStrm->pPAStream); pa_stream_unref(pStrm->pPAStream); pStrm->pPAStream = NULL; pa_threaded_mainloop_unlock(pThis->pMainLoop); } if (pStrm->pvPCMBuf) { RTMemFree(pStrm->pvPCMBuf); pStrm->pvPCMBuf = NULL; pStrm->cbPCMBuf = 0; } return VINF_SUCCESS; } static int paControlStreamOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd) { AssertPtrReturn(pInterface , VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; int rc = VINF_SUCCESS; LogFlowFunc(("enmStreamCmd=%ld\n", enmStreamCmd)); switch (enmStreamCmd) { case PDMAUDIOSTREAMCMD_ENABLE: case PDMAUDIOSTREAMCMD_RESUME: { pa_threaded_mainloop_lock(pThis->pMainLoop); if ( pStrm->pDrainOp && pa_operation_get_state(pStrm->pDrainOp) != PA_OPERATION_DONE) { pa_operation_cancel(pStrm->pDrainOp); pa_operation_unref(pStrm->pDrainOp); pStrm->pDrainOp = NULL; } else { rc = paWaitFor(pThis, pa_stream_cork(pStrm->pPAStream, 0, paStreamCbSuccess, pStrm)); } pa_threaded_mainloop_unlock(pThis->pMainLoop); break; } case PDMAUDIOSTREAMCMD_DISABLE: case PDMAUDIOSTREAMCMD_PAUSE: { /* Pause audio output (the Pause bit of the AC97 x_CR register is set). * Note that we must return immediately from here! */ pa_threaded_mainloop_lock(pThis->pMainLoop); if (!pStrm->pDrainOp) { rc = paWaitFor(pThis, pa_stream_trigger(pStrm->pPAStream, paStreamCbSuccess, pStrm)); if (RT_SUCCESS(rc)) pStrm->pDrainOp = pa_stream_drain(pStrm->pPAStream, paStreamCbDrain, pStrm); } pa_threaded_mainloop_unlock(pThis->pMainLoop); break; } default: AssertMsgFailed(("Invalid command %ld\n", enmStreamCmd)); rc = VERR_INVALID_PARAMETER; break; } LogFlowFuncLeaveRC(rc); return rc; } static int paControlStreamIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; int rc = VINF_SUCCESS; LogFlowFunc(("enmStreamCmd=%ld\n", enmStreamCmd)); switch (enmStreamCmd) { case PDMAUDIOSTREAMCMD_ENABLE: case PDMAUDIOSTREAMCMD_RESUME: { pa_threaded_mainloop_lock(pThis->pMainLoop); rc = paWaitFor(pThis, pa_stream_cork(pStrm->pPAStream, 0 /* Play / resume */, paStreamCbSuccess, pStrm)); pa_threaded_mainloop_unlock(pThis->pMainLoop); break; } case PDMAUDIOSTREAMCMD_DISABLE: case PDMAUDIOSTREAMCMD_PAUSE: { pa_threaded_mainloop_lock(pThis->pMainLoop); if (pStrm->pu8PeekBuf) /* Do we need to drop the peek buffer?*/ { pa_stream_drop(pStrm->pPAStream); pStrm->pu8PeekBuf = NULL; } rc = paWaitFor(pThis, pa_stream_cork(pStrm->pPAStream, 1 /* Stop / pause */, paStreamCbSuccess, pStrm)); pa_threaded_mainloop_unlock(pThis->pMainLoop); break; } default: AssertMsgFailed(("Invalid command %ld\n", enmStreamCmd)); rc = VERR_INVALID_PARAMETER; break; } return rc; } static DECLCALLBACK(void) drvHostPulseAudioShutdown(PPDMIHOSTAUDIO pInterface) { AssertPtrReturnVoid(pInterface); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); LogFlowFuncEnter(); if (pThis->pMainLoop) pa_threaded_mainloop_stop(pThis->pMainLoop); if (pThis->pContext) { pa_context_disconnect(pThis->pContext); pa_context_unref(pThis->pContext); pThis->pContext = NULL; } if (pThis->pMainLoop) { pa_threaded_mainloop_free(pThis->pMainLoop); pThis->pMainLoop = NULL; } LogFlowFuncLeave(); } static DECLCALLBACK(int) drvHostPulseAudioGetConfig(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pCfg) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pCfg, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); return paEnumerate(pThis, pCfg, PULSEAUDIOENUMCBFLAGS_LOG /* fEnum */); } static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvHostPulseAudioGetStatus(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir) { AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN); return PDMAUDIOBACKENDSTS_RUNNING; } static DECLCALLBACK(int) drvHostPulseAudioStreamCreate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream, PPDMAUDIOSTREAMCFG pCfg, uint32_t *pcSamples) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); AssertPtrReturn(pCfg, VERR_INVALID_POINTER); int rc; if (pCfg->enmDir == PDMAUDIODIR_IN) rc = paCreateStreamIn(pInterface, pStream, pCfg, pcSamples); else if (pStream->enmDir == PDMAUDIODIR_OUT) rc = paCreateStreamOut(pInterface, pStream, pCfg, pcSamples); else AssertFailedReturn(VERR_NOT_IMPLEMENTED); LogFlowFunc(("%s: rc=%Rrc\n", pStream->szName, rc)); return rc; } static DECLCALLBACK(int) drvHostPulseAudioStreamDestroy(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); LogFlowFunc(("%s\n", pStream->szName)); int rc; if (pStream->enmDir == PDMAUDIODIR_IN) rc = paDestroyStreamIn(pInterface, pStream); else if (pStream->enmDir == PDMAUDIODIR_OUT) rc = paDestroyStreamOut(pInterface, pStream); else AssertFailedReturn(VERR_NOT_IMPLEMENTED); LogFlowFunc(("%s: rc=%Rrc\n", pStream->szName, rc)); return rc; } static DECLCALLBACK(int) drvHostPulseAudioStreamControl(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); Assert(pStream->enmCtx == PDMAUDIOSTREAMCTX_HOST); int rc; if (pStream->enmDir == PDMAUDIODIR_IN) rc = paControlStreamIn(pInterface, pStream, enmStreamCmd); else if (pStream->enmDir == PDMAUDIODIR_OUT) rc = paControlStreamOut(pInterface, pStream, enmStreamCmd); else AssertFailedReturn(VERR_NOT_IMPLEMENTED); LogFlowFunc(("%s: rc=%Rrc\n", pStream->szName, rc)); return rc; } static DECLCALLBACK(PDMAUDIOSTRMSTS) drvHostPulseAudioStreamGetStatus(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; PDMAUDIOSTRMSTS strmSts = PDMAUDIOSTRMSTS_FLAG_INITIALIZED | PDMAUDIOSTRMSTS_FLAG_ENABLED; pa_threaded_mainloop_lock(pThis->pMainLoop); pa_context_state_t ctxState = pa_context_get_state(pThis->pContext); if ( pa_context_get_state(pThis->pContext) == PA_CONTEXT_READY && 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 { 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; } static DECLCALLBACK(int) drvHostPulseAudioStreamIterate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOSTREAM pStream) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface); PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pStream; LogFlowFuncEnter(); /* Nothing to do here for PulseAudio. */ return VINF_SUCCESS; } /** * @interface_method_impl{PDMIBASE,pfnQueryInterface} */ static DECLCALLBACK(void *) drvHostPulseAudioQueryInterface(PPDMIBASE pInterface, const char *pszIID) { AssertPtrReturn(pInterface, NULL); AssertPtrReturn(pszIID, NULL); PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface); PDRVHOSTPULSEAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTPULSEAUDIO); PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase); PDMIBASE_RETURN_INTERFACE(pszIID, PDMIHOSTAUDIO, &pThis->IHostAudio); return NULL; } /** * Constructs a PulseAudio Audio driver instance. * * @copydoc FNPDMDRVCONSTRUCT */ static DECLCALLBACK(int) drvHostPulseAudioConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfg, uint32_t fFlags) { AssertPtrReturn(pDrvIns, VERR_INVALID_POINTER); PDRVHOSTPULSEAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTPULSEAUDIO); LogRel(("Audio: Initializing PulseAudio driver\n")); pThis->pDrvIns = pDrvIns; /* IBase */ pDrvIns->IBase.pfnQueryInterface = drvHostPulseAudioQueryInterface; /* IHostAudio */ PDMAUDIO_IHOSTAUDIO_CALLBACKS(drvHostPulseAudio); return VINF_SUCCESS; } /** * Destructs a PulseAudio Audio driver instance. * * @copydoc FNPDMDRVCONSTRUCT */ static DECLCALLBACK(void) drvHostPulseAudioDestruct(PPDMDRVINS pDrvIns) { PDRVHOSTPULSEAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTPULSEAUDIO); LogFlowFuncEnter(); } /** * Char driver registration record. */ const PDMDRVREG g_DrvHostPulseAudio = { /* u32Version */ PDM_DRVREG_VERSION, /* szName */ "PulseAudio", /* szRCMod */ "", /* szR0Mod */ "", /* pszDescription */ "Pulse Audio host driver", /* fFlags */ PDM_DRVREG_FLAGS_HOST_BITS_DEFAULT, /* fClass. */ PDM_DRVREG_CLASS_AUDIO, /* cMaxInstances */ ~0U, /* cbInstance */ sizeof(DRVHOSTPULSEAUDIO), /* pfnConstruct */ drvHostPulseAudioConstruct, /* pfnDestruct */ drvHostPulseAudioDestruct, /* pfnRelocate */ NULL, /* pfnIOCtl */ NULL, /* pfnPowerOn */ NULL, /* pfnReset */ NULL, /* pfnSuspend */ NULL, /* pfnResume */ NULL, /* pfnAttach */ NULL, /* pfnDetach */ NULL, /* pfnPowerOff */ NULL, /* pfnSoftReset */ NULL, /* u32EndVersion */ PDM_DRVREG_VERSION }; static struct audio_option pulse_options[] = { {"DAC_MS", AUD_OPT_INT, &s_pulseCfg.buffer_msecs_out, "DAC period size in milliseconds", NULL, 0}, {"ADC_MS", AUD_OPT_INT, &s_pulseCfg.buffer_msecs_in, "ADC period size in milliseconds", NULL, 0} };