/* $Id: DrvHostAudioAlsa.cpp 93115 2022-01-01 11:31:46Z vboxsync $ */ /** @file * Host audio driver - Advanced Linux Sound Architecture (ALSA). */ /* * Copyright (C) 2006-2022 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. * -------------------------------------------------------------------- * * This code is based on: alsaaudio.c * * QEMU ALSA audio driver * * Copyright (c) 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. */ /********************************************************************************************************************************* * Header Files * *********************************************************************************************************************************/ #define LOG_GROUP LOG_GROUP_DRV_HOST_AUDIO #include #include #include /* For PDMIBASE_2_PDMDRV. */ #include #include #include #include "DrvHostAudioAlsaStubsMangling.h" #include #include /* For device enumeration. */ #include #include "DrvHostAudioAlsaStubs.h" #include "VBoxDD.h" /********************************************************************************************************************************* * Defined Constants And Macros * *********************************************************************************************************************************/ /** Maximum number of tries to recover a broken pipe. */ #define ALSA_RECOVERY_TRIES_MAX 5 /********************************************************************************************************************************* * Structures * *********************************************************************************************************************************/ /** * ALSA host audio specific stream data. */ typedef struct DRVHSTAUDALSASTREAM { /** Common part. */ PDMAUDIOBACKENDSTREAM Core; /** Handle to the ALSA PCM stream. */ snd_pcm_t *hPCM; /** Internal stream offset (for debugging). */ uint64_t offInternal; /** The stream's acquired configuration. */ PDMAUDIOSTREAMCFG Cfg; } DRVHSTAUDALSASTREAM; /** Pointer to the ALSA host audio specific stream data. */ typedef DRVHSTAUDALSASTREAM *PDRVHSTAUDALSASTREAM; /** * Host Alsa audio driver instance data. * @implements PDMIAUDIOCONNECTOR */ typedef struct DRVHSTAUDALSA { /** Pointer to the driver instance structure. */ PPDMDRVINS pDrvIns; /** Pointer to host audio interface. */ PDMIHOSTAUDIO IHostAudio; /** Error count for not flooding the release log. * UINT32_MAX for unlimited logging. */ uint32_t cLogErrors; /** Critical section protecting the default device strings. */ RTCRITSECT CritSect; /** Default input device name. */ char szInputDev[256]; /** Default output device name. */ char szOutputDev[256]; /** Upwards notification interface. */ PPDMIHOSTAUDIOPORT pIHostAudioPort; } DRVHSTAUDALSA; /** Pointer to the instance data of an ALSA host audio driver. */ typedef DRVHSTAUDALSA *PDRVHSTAUDALSA; /** * Closes an ALSA stream * * @returns VBox status code. * @param phPCM Pointer to the ALSA stream handle to close. Will be set to * NULL. */ static int drvHstAudAlsaStreamClose(snd_pcm_t **phPCM) { if (!phPCM || !*phPCM) return VINF_SUCCESS; LogRelFlowFuncEnter(); int rc; int rc2 = snd_pcm_close(*phPCM); if (rc2 == 0) { *phPCM = NULL; rc = VINF_SUCCESS; } else { rc = RTErrConvertFromErrno(-rc2); LogRel(("ALSA: Closing PCM descriptor failed: %s (%d, %Rrc)\n", snd_strerror(rc2), rc2, rc)); } LogRelFlowFuncLeaveRC(rc); return rc; } #ifdef DEBUG static void drvHstAudAlsaDbgErrorHandler(const char *file, int line, const char *function, int err, const char *fmt, ...) { /** @todo Implement me! */ RT_NOREF(file, line, function, err, fmt); } #endif /** * Tries to recover an ALSA stream. * * @returns VBox status code. * @param hPCM ALSA stream handle. */ static int drvHstAudAlsaStreamRecover(snd_pcm_t *hPCM) { AssertPtrReturn(hPCM, VERR_INVALID_POINTER); int rc = snd_pcm_prepare(hPCM); if (rc >= 0) { LogFlowFunc(("Successfully recovered %p.\n", hPCM)); return VINF_SUCCESS; } LogFunc(("Failed to recover stream %p: %s (%d)\n", hPCM, snd_strerror(rc), rc)); return RTErrConvertFromErrno(-rc); } /** * Resumes an ALSA stream. * * Used by drvHstAudAlsaHA_StreamPlay() and drvHstAudAlsaHA_StreamCapture(). * * @returns VBox status code. * @param hPCM ALSA stream to resume. */ static int drvHstAudAlsaStreamResume(snd_pcm_t *hPCM) { AssertPtrReturn(hPCM, VERR_INVALID_POINTER); int rc = snd_pcm_resume(hPCM); if (rc >= 0) { LogFlowFunc(("Successfuly resumed %p.\n", hPCM)); return VINF_SUCCESS; } LogFunc(("Failed to resume stream %p: %s (%d)\n", hPCM, snd_strerror(rc), rc)); return RTErrConvertFromErrno(-rc); } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnGetConfig} */ static DECLCALLBACK(int) drvHstAudAlsaHA_GetConfig(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pBackendCfg) { RT_NOREF(pInterface); AssertPtrReturn(pBackendCfg, VERR_INVALID_POINTER); /* * Fill in the config structure. */ RTStrCopy(pBackendCfg->szName, sizeof(pBackendCfg->szName), "ALSA"); pBackendCfg->cbStream = sizeof(DRVHSTAUDALSASTREAM); pBackendCfg->fFlags = 0; /* ALSA allows exactly one input and one output used at a time for the selected device(s). */ pBackendCfg->cMaxStreamsIn = 1; pBackendCfg->cMaxStreamsOut = 1; return VINF_SUCCESS; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnGetDevices} */ static DECLCALLBACK(int) drvHstAudAlsaHA_GetDevices(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHOSTENUM pDeviceEnum) { RT_NOREF(pInterface); PDMAudioHostEnumInit(pDeviceEnum); char **papszHints = NULL; int rc = snd_device_name_hint(-1 /* All cards */, "pcm", (void***)&papszHints); if (rc == 0) { rc = VINF_SUCCESS; for (size_t iHint = 0; papszHints[iHint] != NULL && RT_SUCCESS(rc); iHint++) { /* * Retrieve the available info: */ const char * const pszHint = papszHints[iHint]; char * const pszDev = snd_device_name_get_hint(pszHint, "NAME"); char * const pszInOutId = snd_device_name_get_hint(pszHint, "IOID"); char * const pszDesc = snd_device_name_get_hint(pszHint, "DESC"); if (pszDev && RTStrICmpAscii(pszDev, "null") != 0) { /* Detect and log presence of pulse audio plugin. */ if (RTStrIStr("pulse", pszDev) != NULL) LogRel(("ALSA: The ALSAAudio plugin for pulse audio is being used (%s).\n", pszDev)); /* * Add an entry to the enumeration result. * We engage in some trickery here to deal with device names that * are more than 63 characters long. */ size_t const cbId = pszDev ? strlen(pszDev) + 1 : 1; size_t const cbName = pszDesc ? strlen(pszDesc) + 2 + 1 : cbId; PPDMAUDIOHOSTDEV pDev = PDMAudioHostDevAlloc(sizeof(*pDev), cbName, cbId); if (pDev) { RTStrCopy(pDev->pszId, cbId, pszDev); if (pDev->pszId) { pDev->fFlags = PDMAUDIOHOSTDEV_F_NONE; pDev->enmType = PDMAUDIODEVICETYPE_UNKNOWN; if (pszInOutId == NULL) { pDev->enmUsage = PDMAUDIODIR_DUPLEX; pDev->cMaxInputChannels = 2; pDev->cMaxOutputChannels = 2; } else if (RTStrICmpAscii(pszInOutId, "Input") == 0) { pDev->enmUsage = PDMAUDIODIR_IN; pDev->cMaxInputChannels = 2; pDev->cMaxOutputChannels = 0; } else { AssertMsg(RTStrICmpAscii(pszInOutId, "Output") == 0, ("%s (%s)\n", pszInOutId, pszHint)); pDev->enmUsage = PDMAUDIODIR_OUT; pDev->cMaxInputChannels = 0; pDev->cMaxOutputChannels = 2; } if (pszDesc && *pszDesc) { char *pszDesc2 = strchr(pszDesc, '\n'); if (!pszDesc2) RTStrCopy(pDev->pszName, cbName, pszDesc); else { *pszDesc2++ = '\0'; char *psz; while ((psz = strchr(pszDesc2, '\n')) != NULL) *psz = ' '; RTStrPrintf(pDev->pszName, cbName, "%s (%s)", pszDesc2, pszDesc); } } else RTStrCopy(pDev->pszName, cbName, pszDev); PDMAudioHostEnumAppend(pDeviceEnum, pDev); LogRel2(("ALSA: Device #%u: '%s' enmDir=%s: %s\n", iHint, pszDev, PDMAudioDirGetName(pDev->enmUsage), pszDesc)); } else { PDMAudioHostDevFree(pDev); rc = VERR_NO_STR_MEMORY; } } else rc = VERR_NO_MEMORY; } /* * Clean up. */ if (pszInOutId) free(pszInOutId); if (pszDesc) free(pszDesc); if (pszDev) free(pszDev); } snd_device_name_free_hint((void **)papszHints); if (RT_FAILURE(rc)) { PDMAudioHostEnumDelete(pDeviceEnum); PDMAudioHostEnumInit(pDeviceEnum); } } else { int rc2 = RTErrConvertFromErrno(-rc); LogRel2(("ALSA: Error enumerating PCM devices: %Rrc (%d)\n", rc2, rc)); rc = rc2; } return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnSetDevice} */ static DECLCALLBACK(int) drvHstAudAlsaHA_SetDevice(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir, const char *pszId) { PDRVHSTAUDALSA pThis = RT_FROM_MEMBER(pInterface, DRVHSTAUDALSA, IHostAudio); /* * Validate and normalize input. */ AssertReturn(enmDir == PDMAUDIODIR_IN || enmDir == PDMAUDIODIR_OUT || enmDir == PDMAUDIODIR_DUPLEX, VERR_INVALID_PARAMETER); AssertPtrNullReturn(pszId, VERR_INVALID_POINTER); if (!pszId || !*pszId) pszId = "default"; else { size_t cch = strlen(pszId); AssertReturn(cch < sizeof(pThis->szInputDev), VERR_INVALID_NAME); } LogFunc(("enmDir=%d pszId=%s\n", enmDir, pszId)); /* * Update input. */ if (enmDir == PDMAUDIODIR_IN || enmDir == PDMAUDIODIR_DUPLEX) { int rc = RTCritSectEnter(&pThis->CritSect); AssertRCReturn(rc, rc); if (strcmp(pThis->szInputDev, pszId) == 0) RTCritSectLeave(&pThis->CritSect); else { LogRel(("ALSA: Changing input device: '%s' -> '%s'\n", pThis->szInputDev, pszId)); RTStrCopy(pThis->szInputDev, sizeof(pThis->szInputDev), pszId); PPDMIHOSTAUDIOPORT pIHostAudioPort = pThis->pIHostAudioPort; RTCritSectLeave(&pThis->CritSect); if (pIHostAudioPort) { LogFlowFunc(("Notifying parent driver about input device change...\n")); pIHostAudioPort->pfnNotifyDeviceChanged(pIHostAudioPort, PDMAUDIODIR_IN, NULL /*pvUser*/); } } } /* * Update output. */ if (enmDir == PDMAUDIODIR_OUT || enmDir == PDMAUDIODIR_DUPLEX) { int rc = RTCritSectEnter(&pThis->CritSect); AssertRCReturn(rc, rc); if (strcmp(pThis->szOutputDev, pszId) == 0) RTCritSectLeave(&pThis->CritSect); else { LogRel(("ALSA: Changing output device: '%s' -> '%s'\n", pThis->szOutputDev, pszId)); RTStrCopy(pThis->szOutputDev, sizeof(pThis->szOutputDev), pszId); PPDMIHOSTAUDIOPORT pIHostAudioPort = pThis->pIHostAudioPort; RTCritSectLeave(&pThis->CritSect); if (pIHostAudioPort) { LogFlowFunc(("Notifying parent driver about output device change...\n")); pIHostAudioPort->pfnNotifyDeviceChanged(pIHostAudioPort, PDMAUDIODIR_OUT, NULL /*pvUser*/); } } } return VINF_SUCCESS; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnGetStatus} */ static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvHstAudAlsaHA_GetStatus(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir) { RT_NOREF(enmDir); AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN); return PDMAUDIOBACKENDSTS_RUNNING; } /** * Converts internal audio PCM properties to an ALSA PCM format. * * @returns Converted ALSA PCM format. * @param pProps Internal audio PCM configuration to convert. */ static snd_pcm_format_t alsaAudioPropsToALSA(PCPDMAUDIOPCMPROPS pProps) { switch (PDMAudioPropsSampleSize(pProps)) { case 1: return pProps->fSigned ? SND_PCM_FORMAT_S8 : SND_PCM_FORMAT_U8; case 2: if (PDMAudioPropsIsLittleEndian(pProps)) return pProps->fSigned ? SND_PCM_FORMAT_S16_LE : SND_PCM_FORMAT_U16_LE; return pProps->fSigned ? SND_PCM_FORMAT_S16_BE : SND_PCM_FORMAT_U16_BE; case 4: if (PDMAudioPropsIsLittleEndian(pProps)) return pProps->fSigned ? SND_PCM_FORMAT_S32_LE : SND_PCM_FORMAT_U32_LE; return pProps->fSigned ? SND_PCM_FORMAT_S32_BE : SND_PCM_FORMAT_U32_BE; default: AssertLogRelMsgFailed(("%RU8 bytes not supported\n", PDMAudioPropsSampleSize(pProps))); return SND_PCM_FORMAT_UNKNOWN; } } /** * Sets the software parameters of an ALSA stream. * * @returns 0 on success, negative errno on failure. * @param hPCM ALSA stream to set software parameters for. * @param pCfgReq Requested stream configuration (PDM). * @param pCfgAcq The actual stream configuration (PDM). Updated as * needed. */ static int alsaStreamSetSWParams(snd_pcm_t *hPCM, PCPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) { if (pCfgReq->enmDir == PDMAUDIODIR_IN) /* For input streams there's nothing to do in here right now. */ return 0; snd_pcm_sw_params_t *pSWParms = NULL; snd_pcm_sw_params_alloca(&pSWParms); AssertReturn(pSWParms, -ENOMEM); int err = snd_pcm_sw_params_current(hPCM, pSWParms); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to get current software parameters: %s\n", snd_strerror(err)), err); /* Under normal circumstance, we don't need to set a playback threshold because DrvAudio will do the pre-buffering and hand us everything in one continuous chunk when we should start playing. But since it is configurable, we'll set a reasonable minimum of two DMA periods or max 50 milliseconds (the pAlsaCfgReq->threshold value). Of course we also have to make sure the threshold is below the buffer size, or ALSA will never start playing. */ unsigned long const cFramesMax = PDMAudioPropsMilliToFrames(&pCfgAcq->Props, 50); unsigned long cFramesThreshold = RT_MIN(pCfgAcq->Backend.cFramesPeriod * 2, cFramesMax); if (cFramesThreshold >= pCfgAcq->Backend.cFramesBufferSize - pCfgAcq->Backend.cFramesBufferSize / 16) cFramesThreshold = pCfgAcq->Backend.cFramesBufferSize - pCfgAcq->Backend.cFramesBufferSize / 16; err = snd_pcm_sw_params_set_start_threshold(hPCM, pSWParms, cFramesThreshold); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set software threshold to %lu: %s\n", cFramesThreshold, snd_strerror(err)), err); err = snd_pcm_sw_params_set_avail_min(hPCM, pSWParms, pCfgReq->Backend.cFramesPeriod); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set available minimum to %u: %s\n", pCfgReq->Backend.cFramesPeriod, snd_strerror(err)), err); /* Commit the software parameters: */ err = snd_pcm_sw_params(hPCM, pSWParms); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set new software parameters: %s\n", snd_strerror(err)), err); /* Get the actual parameters: */ snd_pcm_uframes_t cFramesThresholdActual = cFramesThreshold; err = snd_pcm_sw_params_get_start_threshold(pSWParms, &cFramesThresholdActual); AssertLogRelMsgStmt(err >= 0, ("ALSA: Failed to get start threshold: %s\n", snd_strerror(err)), cFramesThresholdActual = cFramesThreshold); LogRel2(("ALSA: SW params: %lu frames threshold, %u frames avail minimum\n", cFramesThresholdActual, pCfgAcq->Backend.cFramesPeriod)); return 0; } /** * Maps a PDM channel ID to an ASLA channel map position. */ static unsigned int drvHstAudAlsaPdmChToAlsa(PDMAUDIOCHANNELID enmId, uint8_t cChannels) { switch (enmId) { case PDMAUDIOCHANNELID_UNKNOWN: return SND_CHMAP_UNKNOWN; case PDMAUDIOCHANNELID_UNUSED_ZERO: return SND_CHMAP_NA; case PDMAUDIOCHANNELID_UNUSED_SILENCE: return SND_CHMAP_NA; case PDMAUDIOCHANNELID_FRONT_LEFT: return SND_CHMAP_FL; case PDMAUDIOCHANNELID_FRONT_RIGHT: return SND_CHMAP_FR; case PDMAUDIOCHANNELID_FRONT_CENTER: return cChannels == 1 ? SND_CHMAP_MONO : SND_CHMAP_FC; case PDMAUDIOCHANNELID_LFE: return SND_CHMAP_LFE; case PDMAUDIOCHANNELID_REAR_LEFT: return SND_CHMAP_RL; case PDMAUDIOCHANNELID_REAR_RIGHT: return SND_CHMAP_RR; case PDMAUDIOCHANNELID_FRONT_LEFT_OF_CENTER: return SND_CHMAP_FLC; case PDMAUDIOCHANNELID_FRONT_RIGHT_OF_CENTER: return SND_CHMAP_FRC; case PDMAUDIOCHANNELID_REAR_CENTER: return SND_CHMAP_RC; case PDMAUDIOCHANNELID_SIDE_LEFT: return SND_CHMAP_SL; case PDMAUDIOCHANNELID_SIDE_RIGHT: return SND_CHMAP_SR; case PDMAUDIOCHANNELID_TOP_CENTER: return SND_CHMAP_TC; case PDMAUDIOCHANNELID_FRONT_LEFT_HEIGHT: return SND_CHMAP_TFL; case PDMAUDIOCHANNELID_FRONT_CENTER_HEIGHT: return SND_CHMAP_TFC; case PDMAUDIOCHANNELID_FRONT_RIGHT_HEIGHT: return SND_CHMAP_TFR; case PDMAUDIOCHANNELID_REAR_LEFT_HEIGHT: return SND_CHMAP_TRL; case PDMAUDIOCHANNELID_REAR_CENTER_HEIGHT: return SND_CHMAP_TRC; case PDMAUDIOCHANNELID_REAR_RIGHT_HEIGHT: return SND_CHMAP_TRR; case PDMAUDIOCHANNELID_INVALID: case PDMAUDIOCHANNELID_END: case PDMAUDIOCHANNELID_32BIT_HACK: break; } AssertFailed(); return SND_CHMAP_NA; } /** * Sets the hardware parameters of an ALSA stream. * * @returns 0 on success, negative errno on failure. * @param hPCM ALSA stream to set software parameters for. * @param enmAlsaFmt The ALSA format to use. * @param pCfgReq Requested stream configuration (PDM). * @param pCfgAcq The actual stream configuration (PDM). This is assumed * to be a copy of pCfgReq on input, at least for * properties handled here. On output some of the * properties may be updated to match the actual stream * configuration. */ static int alsaStreamSetHwParams(snd_pcm_t *hPCM, snd_pcm_format_t enmAlsaFmt, PCPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) { /* * Get the current hardware parameters. */ snd_pcm_hw_params_t *pHWParms = NULL; snd_pcm_hw_params_alloca(&pHWParms); AssertReturn(pHWParms, -ENOMEM); int err = snd_pcm_hw_params_any(hPCM, pHWParms); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to initialize hardware parameters: %s\n", snd_strerror(err)), err); /* * Modify them according to pAlsaCfgReq. * We update pAlsaCfgObt as we go for parameters set by "near" methods. */ /* We'll use snd_pcm_writei/snd_pcm_readi: */ err = snd_pcm_hw_params_set_access(hPCM, pHWParms, SND_PCM_ACCESS_RW_INTERLEAVED); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set access type: %s\n", snd_strerror(err)), err); /* Set the format and frequency. */ err = snd_pcm_hw_params_set_format(hPCM, pHWParms, enmAlsaFmt); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set audio format to %d: %s\n", enmAlsaFmt, snd_strerror(err)), err); unsigned int uFreq = PDMAudioPropsHz(&pCfgReq->Props); err = snd_pcm_hw_params_set_rate_near(hPCM, pHWParms, &uFreq, NULL /*dir*/); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set frequency to %uHz: %s\n", PDMAudioPropsHz(&pCfgReq->Props), snd_strerror(err)), err); pCfgAcq->Props.uHz = uFreq; /* Channel count currently does not change with the mapping translations, as ALSA can express both silent and unknown channel positions. */ union { snd_pcm_chmap_t Map; unsigned int padding[1 + PDMAUDIO_MAX_CHANNELS]; } u; uint8_t aidSrcChannels[PDMAUDIO_MAX_CHANNELS]; unsigned int *aidDstChannels = u.Map.pos; unsigned int cChannels = u.Map.channels = PDMAudioPropsChannels(&pCfgReq->Props); unsigned int iDst = 0; for (unsigned int iSrc = 0; iSrc < cChannels; iSrc++) { uint8_t const idSrc = pCfgReq->Props.aidChannels[iSrc]; aidSrcChannels[iDst] = idSrc; aidDstChannels[iDst] = drvHstAudAlsaPdmChToAlsa((PDMAUDIOCHANNELID)idSrc, cChannels); iDst++; } u.Map.channels = cChannels = iDst; for (; iDst < PDMAUDIO_MAX_CHANNELS; iDst++) { aidSrcChannels[iDst] = PDMAUDIOCHANNELID_INVALID; aidDstChannels[iDst] = SND_CHMAP_NA; } err = snd_pcm_hw_params_set_channels_near(hPCM, pHWParms, &cChannels); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set number of channels to %d\n", PDMAudioPropsChannels(&pCfgReq->Props)), err); if (cChannels == PDMAudioPropsChannels(&pCfgReq->Props)) memcpy(pCfgAcq->Props.aidChannels, aidSrcChannels, sizeof(pCfgAcq->Props.aidChannels)); else { LogRel2(("ALSA: Requested %u channels, got %u\n", u.Map.channels, cChannels)); AssertLogRelMsgReturn(cChannels > 0 && cChannels <= PDMAUDIO_MAX_CHANNELS, ("ALSA: Unsupported channel count: %u (requested %d)\n", cChannels, PDMAudioPropsChannels(&pCfgReq->Props)), -ERANGE); PDMAudioPropsSetChannels(&pCfgAcq->Props, (uint8_t)cChannels); /** @todo Can we somehow guess channel IDs? snd_pcm_get_chmap? */ } /* The period size (reportedly frame count per hw interrupt): */ int dir = 0; snd_pcm_uframes_t minval = pCfgReq->Backend.cFramesPeriod; err = snd_pcm_hw_params_get_period_size_min(pHWParms, &minval, &dir); AssertLogRelMsgReturn(err >= 0, ("ALSA: Could not determine minimal period size: %s\n", snd_strerror(err)), err); snd_pcm_uframes_t period_size_f = pCfgReq->Backend.cFramesPeriod; if (period_size_f < minval) period_size_f = minval; err = snd_pcm_hw_params_set_period_size_near(hPCM, pHWParms, &period_size_f, 0); LogRel2(("ALSA: Period size is: %lu frames (min %lu, requested %u)\n", period_size_f, minval, pCfgReq->Backend.cFramesPeriod)); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set period size %d (%s)\n", period_size_f, snd_strerror(err)), err); /* The buffer size: */ minval = pCfgReq->Backend.cFramesBufferSize; err = snd_pcm_hw_params_get_buffer_size_min(pHWParms, &minval); AssertLogRelMsgReturn(err >= 0, ("ALSA: Could not retrieve minimal buffer size: %s\n", snd_strerror(err)), err); snd_pcm_uframes_t buffer_size_f = pCfgReq->Backend.cFramesBufferSize; if (buffer_size_f < minval) buffer_size_f = minval; err = snd_pcm_hw_params_set_buffer_size_near(hPCM, pHWParms, &buffer_size_f); LogRel2(("ALSA: Buffer size is: %lu frames (min %lu, requested %u)\n", buffer_size_f, minval, pCfgReq->Backend.cFramesBufferSize)); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set near buffer size %RU32: %s\n", buffer_size_f, snd_strerror(err)), err); /* * Set the hardware parameters. */ err = snd_pcm_hw_params(hPCM, pHWParms); AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to apply audio parameters: %s\n", snd_strerror(err)), err); /* * Get relevant parameters and put them in the pAlsaCfgObt structure. */ snd_pcm_uframes_t obt_buffer_size = buffer_size_f; err = snd_pcm_hw_params_get_buffer_size(pHWParms, &obt_buffer_size); AssertLogRelMsgStmt(err >= 0, ("ALSA: Failed to get buffer size: %s\n", snd_strerror(err)), obt_buffer_size = buffer_size_f); pCfgAcq->Backend.cFramesBufferSize = obt_buffer_size; snd_pcm_uframes_t obt_period_size = period_size_f; err = snd_pcm_hw_params_get_period_size(pHWParms, &obt_period_size, &dir); AssertLogRelMsgStmt(err >= 0, ("ALSA: Failed to get period size: %s\n", snd_strerror(err)), obt_period_size = period_size_f); pCfgAcq->Backend.cFramesPeriod = obt_period_size; LogRel2(("ALSA: HW params: %u Hz, %u frames period, %u frames buffer, %u channel(s), enmAlsaFmt=%d\n", PDMAudioPropsHz(&pCfgAcq->Props), pCfgAcq->Backend.cFramesPeriod, pCfgAcq->Backend.cFramesBufferSize, PDMAudioPropsChannels(&pCfgAcq->Props), enmAlsaFmt)); #if 0 /* Disabled in the hope to resolve testboxes not being able to drain + crashing when closing the PCM streams. */ /* * Channel config (not fatal). */ if (PDMAudioPropsChannels(&pCfgAcq->Props) == PDMAudioPropsChannels(&pCfgReq->Props)) { err = snd_pcm_set_chmap(hPCM, &u.Map); if (err < 0) { if (err == -ENXIO) LogRel2(("ALSA: Audio device does not support channel maps, skipping\n")); else LogRel2(("ALSA: snd_pcm_set_chmap failed: %s (%d)\n", snd_strerror(err), err)); } } #endif return 0; } /** * Opens (creates) an ALSA stream. * * @returns VBox status code. * @param pThis The alsa driver instance data. * @param enmAlsaFmt The ALSA format to use. * @param pCfgReq Requested configuration to create stream with (PDM). * @param pCfgAcq The actual stream configuration (PDM). This is assumed * to be a copy of pCfgReq on input, at least for * properties handled here. On output some of the * properties may be updated to match the actual stream * configuration. * @param phPCM Where to store the ALSA stream handle on success. */ static int alsaStreamOpen(PDRVHSTAUDALSA pThis, snd_pcm_format_t enmAlsaFmt, PCPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq, snd_pcm_t **phPCM) { /* * Open the stream. */ int rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE; const char * const pszType = pCfgReq->enmDir == PDMAUDIODIR_IN ? "input" : "output"; const char * const pszDev = pCfgReq->enmDir == PDMAUDIODIR_IN ? pThis->szInputDev : pThis->szOutputDev; snd_pcm_stream_t enmType = pCfgReq->enmDir == PDMAUDIODIR_IN ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK; snd_pcm_t *hPCM = NULL; LogRel(("ALSA: Using %s device \"%s\"\n", pszType, pszDev)); int err = snd_pcm_open(&hPCM, pszDev, enmType, SND_PCM_NONBLOCK); if (err >= 0) { err = snd_pcm_nonblock(hPCM, 1); if (err >= 0) { /* * Configure hardware stream parameters. */ err = alsaStreamSetHwParams(hPCM, enmAlsaFmt, pCfgReq, pCfgAcq); if (err >= 0) { /* * Prepare it. */ rc = VERR_AUDIO_BACKEND_INIT_FAILED; err = snd_pcm_prepare(hPCM); if (err >= 0) { /* * Configure software stream parameters. */ rc = alsaStreamSetSWParams(hPCM, pCfgReq, pCfgAcq); if (RT_SUCCESS(rc)) { *phPCM = hPCM; return VINF_SUCCESS; } } else LogRel(("ALSA: snd_pcm_prepare failed: %s\n", snd_strerror(err))); } } else LogRel(("ALSA: Error setting non-blocking mode for %s stream: %s\n", pszType, snd_strerror(err))); drvHstAudAlsaStreamClose(&hPCM); } else LogRel(("ALSA: Failed to open \"%s\" as %s device: %s\n", pszDev, pszType, snd_strerror(err))); *phPCM = NULL; return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCreate} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamCreate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, PCPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) { PDRVHSTAUDALSA pThis = RT_FROM_MEMBER(pInterface, DRVHSTAUDALSA, IHostAudio); AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); AssertPtrReturn(pCfgReq, VERR_INVALID_POINTER); AssertPtrReturn(pCfgAcq, VERR_INVALID_POINTER); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; PDMAudioStrmCfgCopy(&pStreamALSA->Cfg, pCfgReq); int rc; snd_pcm_format_t const enmFmt = alsaAudioPropsToALSA(&pCfgReq->Props); if (enmFmt != SND_PCM_FORMAT_UNKNOWN) { rc = alsaStreamOpen(pThis, enmFmt, pCfgReq, pCfgAcq, &pStreamALSA->hPCM); if (RT_SUCCESS(rc)) { /* We have no objections to the pre-buffering that DrvAudio applies, only we need to adjust it relative to the actual buffer size. */ pCfgAcq->Backend.cFramesPreBuffering = (uint64_t)pCfgReq->Backend.cFramesPreBuffering * pCfgAcq->Backend.cFramesBufferSize / RT_MAX(pCfgReq->Backend.cFramesBufferSize, 1); PDMAudioStrmCfgCopy(&pStreamALSA->Cfg, pCfgAcq); LogFlowFunc(("returns success - hPCM=%p\n", pStreamALSA->hPCM)); return rc; } } else rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE; LogFunc(("returns %Rrc\n", rc)); return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDestroy} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamDestroy(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, bool fImmediate) { RT_NOREF(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; AssertPtrReturn(pStreamALSA, VERR_INVALID_POINTER); RT_NOREF(fImmediate); LogRelFlowFunc(("Stream '%s' state is '%s'\n", pStreamALSA->Cfg.szName, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)))); /** @todo r=bird: It's not like we can do much with a bad status... Check * what the caller does... */ int rc = drvHstAudAlsaStreamClose(&pStreamALSA->hPCM); LogRelFlowFunc(("returns %Rrc\n", rc)); return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamEnable} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamEnable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { RT_NOREF(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; /* * Prepare the stream. */ int rc = snd_pcm_prepare(pStreamALSA->hPCM); if (rc >= 0) { Assert(snd_pcm_state(pStreamALSA->hPCM) == SND_PCM_STATE_PREPARED); /* * Input streams should be started now, whereas output streams must * pre-buffer sufficent data before starting. */ if (pStreamALSA->Cfg.enmDir == PDMAUDIODIR_IN) { rc = snd_pcm_start(pStreamALSA->hPCM); if (rc >= 0) rc = VINF_SUCCESS; else { LogRel(("ALSA: Error starting input stream '%s': %s (%d)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc)); rc = RTErrConvertFromErrno(-rc); } } else rc = VINF_SUCCESS; } else { LogRel(("ALSA: Error preparing stream '%s': %s (%d)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc)); rc = RTErrConvertFromErrno(-rc); } LogFlowFunc(("returns %Rrc (state %s)\n", rc, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)))); return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDisable} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamDisable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { RT_NOREF(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; int rc = snd_pcm_drop(pStreamALSA->hPCM); if (rc >= 0) rc = VINF_SUCCESS; else { LogRel(("ALSA: Error stopping stream '%s': %s (%d)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc)); rc = RTErrConvertFromErrno(-rc); } LogFlowFunc(("returns %Rrc (state %s)\n", rc, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)))); return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPause} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamPause(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { /* Same as disable. */ /** @todo r=bird: Try use pause and fallback on disable/enable if it isn't * supported or doesn't work. */ return drvHstAudAlsaHA_StreamDisable(pInterface, pStream); } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamResume} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamResume(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { /* Same as enable. */ return drvHstAudAlsaHA_StreamEnable(pInterface, pStream); } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDrain} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamDrain(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { RT_NOREF(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; snd_pcm_state_t const enmState = snd_pcm_state(pStreamALSA->hPCM); LogRelFlowFunc(("Stream '%s' input state: %s (%d)\n", pStreamALSA->Cfg.szName, snd_pcm_state_name(enmState), enmState)); /* Only for output streams. */ AssertReturn(pStreamALSA->Cfg.enmDir == PDMAUDIODIR_OUT, VERR_WRONG_ORDER); int rc; switch (enmState) { case SND_PCM_STATE_RUNNING: case SND_PCM_STATE_PREPARED: /* not yet started */ { /* Do not change to blocking here! */ rc = snd_pcm_drain(pStreamALSA->hPCM); if (rc >= 0 || rc == -EAGAIN) rc = VINF_SUCCESS; else { snd_pcm_state_t const enmState2 = snd_pcm_state(pStreamALSA->hPCM); if (rc == -EPIPE && enmState2 == enmState) { /* Not entirely sure, but possibly an underrun, so just disable the stream. */ LogRel2(("ALSA: snd_pcm_drain failed with -EPIPE, stopping stream (%s)\n", pStreamALSA->Cfg.szName)); rc = snd_pcm_drop(pStreamALSA->hPCM); if (rc >= 0) rc = VINF_SUCCESS; else { LogRel(("ALSA: Error draining/stopping stream '%s': %s (%d)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc)); rc = RTErrConvertFromErrno(-rc); } } else { LogRel(("ALSA: Error draining output of '%s': %s (%d; %s -> %s)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc, snd_pcm_state_name(enmState), snd_pcm_state_name(enmState2))); rc = RTErrConvertFromErrno(-rc); } } break; } default: rc = VINF_SUCCESS; break; } LogRelFlowFunc(("returns %Rrc (state %s)\n", rc, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)))); return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetState} */ static DECLCALLBACK(PDMHOSTAUDIOSTREAMSTATE) drvHstAudAlsaHA_StreamGetState(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { RT_NOREF(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; AssertPtrReturn(pStreamALSA, PDMHOSTAUDIOSTREAMSTATE_INVALID); PDMHOSTAUDIOSTREAMSTATE enmStreamState = PDMHOSTAUDIOSTREAMSTATE_OKAY; snd_pcm_state_t enmAlsaState = snd_pcm_state(pStreamALSA->hPCM); if (enmAlsaState == SND_PCM_STATE_DRAINING) { /* We're operating in non-blocking mode, so we must (at least for a demux config) call snd_pcm_drain again to drive it forward. Otherwise we might be stuck in the drain state forever. */ Log5Func(("Calling snd_pcm_drain again...\n")); snd_pcm_drain(pStreamALSA->hPCM); enmAlsaState = snd_pcm_state(pStreamALSA->hPCM); } if (enmAlsaState == SND_PCM_STATE_DRAINING) enmStreamState = PDMHOSTAUDIOSTREAMSTATE_DRAINING; #if (((SND_LIB_MAJOR) << 16) | ((SND_LIB_MAJOR) << 8) | (SND_LIB_SUBMINOR)) >= 0x10002 /* was added in 1.0.2 */ else if (enmAlsaState == SND_PCM_STATE_DISCONNECTED) enmStreamState = PDMHOSTAUDIOSTREAMSTATE_NOT_WORKING; #endif Log5Func(("Stream '%s': ALSA state=%s -> %s\n", pStreamALSA->Cfg.szName, snd_pcm_state_name(enmAlsaState), PDMHostAudioStreamStateGetName(enmStreamState) )); return enmStreamState; } /** * Returns the available audio frames queued. * * @returns VBox status code. * @param hPCM ALSA stream handle. * @param pcFramesAvail Where to store the available frames. */ static int alsaStreamGetAvail(snd_pcm_t *hPCM, snd_pcm_sframes_t *pcFramesAvail) { AssertPtr(hPCM); AssertPtr(pcFramesAvail); int rc; snd_pcm_sframes_t cFramesAvail = snd_pcm_avail_update(hPCM); if (cFramesAvail > 0) { LogFunc(("cFramesAvail=%ld\n", cFramesAvail)); *pcFramesAvail = cFramesAvail; return VINF_SUCCESS; } /* * We can maybe recover from an EPIPE... */ if (cFramesAvail == -EPIPE) { rc = drvHstAudAlsaStreamRecover(hPCM); if (RT_SUCCESS(rc)) { cFramesAvail = snd_pcm_avail_update(hPCM); if (cFramesAvail >= 0) { LogFunc(("cFramesAvail=%ld\n", cFramesAvail)); *pcFramesAvail = cFramesAvail; return VINF_SUCCESS; } } else { *pcFramesAvail = 0; return rc; } } rc = RTErrConvertFromErrno(-(int)cFramesAvail); LogFunc(("failed - cFramesAvail=%ld rc=%Rrc\n", cFramesAvail, rc)); *pcFramesAvail = 0; return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetPending} */ static DECLCALLBACK(uint32_t) drvHstAudAlsaHA_StreamGetPending(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { RT_NOREF(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; AssertPtrReturn(pStreamALSA, 0); /* * This is only relevant to output streams (input streams can't have * any pending, unplayed data). */ uint32_t cbPending = 0; if (pStreamALSA->Cfg.enmDir == PDMAUDIODIR_OUT) { /* * Getting the delay (in audio frames) reports the time it will take * to hear a new sample after all queued samples have been played out. * * We use snd_pcm_avail_delay instead of snd_pcm_delay here as it will * update the buffer positions, and we can use the extra value against * the buffer size to double check since the delay value may include * fixed built-in delays in the processing chain and hardware. */ snd_pcm_sframes_t cFramesAvail = 0; snd_pcm_sframes_t cFramesDelay = 0; int rc = snd_pcm_avail_delay(pStreamALSA->hPCM, &cFramesAvail, &cFramesDelay); /* * We now also get the state as the pending value should be zero when * we're not in a playing state. */ snd_pcm_state_t enmState = snd_pcm_state(pStreamALSA->hPCM); switch (enmState) { case SND_PCM_STATE_RUNNING: case SND_PCM_STATE_DRAINING: if (rc >= 0) { if ((uint32_t)cFramesAvail >= pStreamALSA->Cfg.Backend.cFramesBufferSize) cbPending = 0; else cbPending = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cFramesDelay); } break; default: break; } Log2Func(("returns %u (%#x) - cFramesBufferSize=%RU32 cFramesAvail=%ld cFramesDelay=%ld rc=%d; enmState=%s (%d) \n", cbPending, cbPending, pStreamALSA->Cfg.Backend.cFramesBufferSize, cFramesAvail, cFramesDelay, rc, snd_pcm_state_name(enmState), enmState)); } return cbPending; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetWritable} */ static DECLCALLBACK(uint32_t) drvHstAudAlsaHA_StreamGetWritable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { RT_NOREF(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; uint32_t cbAvail = 0; snd_pcm_sframes_t cFramesAvail = 0; int rc = alsaStreamGetAvail(pStreamALSA->hPCM, &cFramesAvail); if (RT_SUCCESS(rc)) cbAvail = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cFramesAvail); return cbAvail; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPlay} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamPlay(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, const void *pvBuf, uint32_t cbBuf, uint32_t *pcbWritten) { PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); AssertPtrReturn(pcbWritten, VERR_INVALID_POINTER); Log4Func(("@%#RX64: pvBuf=%p cbBuf=%#x (%u) state=%s - %s\n", pStreamALSA->offInternal, pvBuf, cbBuf, cbBuf, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)), pStreamALSA->Cfg.szName)); if (cbBuf) AssertPtrReturn(pvBuf, VERR_INVALID_POINTER); else { /* Fend off draining calls. */ *pcbWritten = 0; return VINF_SUCCESS; } /* * Determine how much we can write (caller actually did this * already, but we repeat it just to be sure or something). */ snd_pcm_sframes_t cFramesAvail; int rc = alsaStreamGetAvail(pStreamALSA->hPCM, &cFramesAvail); if (RT_SUCCESS(rc)) { Assert(cFramesAvail); if (cFramesAvail) { PCPDMAUDIOPCMPROPS pProps = &pStreamALSA->Cfg.Props; uint32_t cbToWrite = PDMAudioPropsFramesToBytes(pProps, (uint32_t)cFramesAvail); if (cbToWrite) { if (cbToWrite > cbBuf) cbToWrite = cbBuf; /* * Try write the data. */ uint32_t cFramesToWrite = PDMAudioPropsBytesToFrames(pProps, cbToWrite); snd_pcm_sframes_t cFramesWritten = snd_pcm_writei(pStreamALSA->hPCM, pvBuf, cFramesToWrite); if (cFramesWritten > 0) { Log4Func(("snd_pcm_writei w/ cbToWrite=%u -> %ld (frames) [cFramesAvail=%ld]\n", cbToWrite, cFramesWritten, cFramesAvail)); *pcbWritten = PDMAudioPropsFramesToBytes(pProps, cFramesWritten); pStreamALSA->offInternal += *pcbWritten; return VINF_SUCCESS; } LogFunc(("snd_pcm_writei w/ cbToWrite=%u -> %ld [cFramesAvail=%ld]\n", cbToWrite, cFramesWritten, cFramesAvail)); /* * There are a couple of error we can recover from, try to do so. * Only don't try too many times. */ for (unsigned iTry = 0; (cFramesWritten == -EPIPE || cFramesWritten == -ESTRPIPE) && iTry < ALSA_RECOVERY_TRIES_MAX; iTry++) { if (cFramesWritten == -EPIPE) { /* Underrun occurred. */ rc = drvHstAudAlsaStreamRecover(pStreamALSA->hPCM); if (RT_FAILURE(rc)) break; LogFlowFunc(("Recovered from playback (iTry=%u)\n", iTry)); } else { /* An suspended event occurred, needs resuming. */ rc = drvHstAudAlsaStreamResume(pStreamALSA->hPCM); if (RT_FAILURE(rc)) { LogRel(("ALSA: Failed to resume output stream (iTry=%u, rc=%Rrc)\n", iTry, rc)); break; } LogFlowFunc(("Resumed suspended output stream (iTry=%u)\n", iTry)); } cFramesWritten = snd_pcm_writei(pStreamALSA->hPCM, pvBuf, cFramesToWrite); if (cFramesWritten > 0) { Log4Func(("snd_pcm_writei w/ cbToWrite=%u -> %ld (frames) [cFramesAvail=%ld]\n", cbToWrite, cFramesWritten, cFramesAvail)); *pcbWritten = PDMAudioPropsFramesToBytes(pProps, cFramesWritten); pStreamALSA->offInternal += *pcbWritten; return VINF_SUCCESS; } LogFunc(("snd_pcm_writei w/ cbToWrite=%u -> %ld [cFramesAvail=%ld, iTry=%d]\n", cbToWrite, cFramesWritten, cFramesAvail, iTry)); } /* Make sure we return with an error status. */ if (RT_SUCCESS_NP(rc)) { if (cFramesWritten == 0) rc = VERR_ACCESS_DENIED; else { rc = RTErrConvertFromErrno(-(int)cFramesWritten); LogFunc(("Failed to write %RU32 bytes: %ld (%Rrc)\n", cbToWrite, cFramesWritten, rc)); } } } } } else LogFunc(("Error getting number of playback frames, rc=%Rrc\n", rc)); *pcbWritten = 0; return rc; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetReadable} */ static DECLCALLBACK(uint32_t) drvHstAudAlsaHA_StreamGetReadable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream) { RT_NOREF(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; uint32_t cbAvail = 0; snd_pcm_sframes_t cFramesAvail = 0; int rc = alsaStreamGetAvail(pStreamALSA->hPCM, &cFramesAvail); if (RT_SUCCESS(rc)) cbAvail = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cFramesAvail); return cbAvail; } /** * @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCapture} */ static DECLCALLBACK(int) drvHstAudAlsaHA_StreamCapture(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead) { RT_NOREF_PV(pInterface); PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream; AssertPtrReturn(pStreamALSA, VERR_INVALID_POINTER); AssertPtrReturn(pvBuf, VERR_INVALID_POINTER); AssertReturn(cbBuf, VERR_INVALID_PARAMETER); AssertPtrReturn(pcbRead, VERR_INVALID_POINTER); Log4Func(("@%#RX64: pvBuf=%p cbBuf=%#x (%u) state=%s - %s\n", pStreamALSA->offInternal, pvBuf, cbBuf, cbBuf, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)), pStreamALSA->Cfg.szName)); /* * Figure out how much we can read without trouble (we're doing * non-blocking reads, but whatever). */ snd_pcm_sframes_t cAvail; int rc = alsaStreamGetAvail(pStreamALSA->hPCM, &cAvail); if (RT_SUCCESS(rc)) { if (!cAvail) /* No data yet? */ { snd_pcm_state_t enmState = snd_pcm_state(pStreamALSA->hPCM); switch (enmState) { case SND_PCM_STATE_PREPARED: /** @todo r=bird: explain the logic here... */ cAvail = PDMAudioPropsBytesToFrames(&pStreamALSA->Cfg.Props, cbBuf); break; case SND_PCM_STATE_SUSPENDED: rc = drvHstAudAlsaStreamResume(pStreamALSA->hPCM); if (RT_SUCCESS(rc)) { LogFlowFunc(("Resumed suspended input stream.\n")); break; } LogFunc(("Failed resuming suspended input stream: %Rrc\n", rc)); return rc; default: LogFlow(("No frames available: state=%s (%d)\n", snd_pcm_state_name(enmState), enmState)); break; } if (!cAvail) { *pcbRead = 0; return VINF_SUCCESS; } } } else { LogFunc(("Error getting number of captured frames, rc=%Rrc\n", rc)); return rc; } size_t cbToRead = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cAvail); cbToRead = RT_MIN(cbToRead, cbBuf); LogFlowFunc(("cbToRead=%zu, cAvail=%RI32\n", cbToRead, cAvail)); /* * Read loop. */ uint32_t cbReadTotal = 0; while (cbToRead > 0) { /* * Do the reading. */ snd_pcm_uframes_t const cFramesToRead = PDMAudioPropsBytesToFrames(&pStreamALSA->Cfg.Props, cbToRead); AssertBreakStmt(cFramesToRead > 0, rc = VERR_NO_DATA); snd_pcm_sframes_t cFramesRead = snd_pcm_readi(pStreamALSA->hPCM, pvBuf, cFramesToRead); if (cFramesRead > 0) { /* * We should not run into a full mixer buffer or we lose samples and * run into an endless loop if ALSA keeps producing samples ("null" * capture device for example). */ uint32_t const cbRead = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cFramesRead); Assert(cbRead <= cbToRead); cbToRead -= cbRead; cbReadTotal += cbRead; pvBuf = (uint8_t *)pvBuf + cbRead; pStreamALSA->offInternal += cbRead; } else { /* * Try recover from overrun and re-try. * Other conditions/errors we cannot and will just quit the loop. */ if (cFramesRead == -EPIPE) { rc = drvHstAudAlsaStreamRecover(pStreamALSA->hPCM); if (RT_SUCCESS(rc)) { LogFlowFunc(("Successfully recovered from overrun\n")); continue; } LogFunc(("Failed to recover from overrun: %Rrc\n", rc)); } else if (cFramesRead == -EAGAIN) LogFunc(("No input frames available (EAGAIN)\n")); else if (cFramesRead == 0) LogFunc(("No input frames available (0)\n")); else { rc = RTErrConvertFromErrno(-(int)cFramesRead); LogFunc(("Failed to read input frames: %s (%ld, %Rrc)\n", snd_strerror(cFramesRead), cFramesRead, rc)); } /* If we've read anything, suppress the error. */ if (RT_FAILURE(rc) && cbReadTotal > 0) { LogFunc(("Suppressing %Rrc because %#x bytes has been read already\n", rc, cbReadTotal)); rc = VINF_SUCCESS; } break; } } LogFlowFunc(("returns %Rrc and %#x (%d) bytes (%u bytes left); state %s\n", rc, cbReadTotal, cbReadTotal, cbToRead, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)))); *pcbRead = cbReadTotal; return rc; } /********************************************************************************************************************************* * PDMIBASE * *********************************************************************************************************************************/ /** * @interface_method_impl{PDMIBASE,pfnQueryInterface} */ static DECLCALLBACK(void *) drvHstAudAlsaQueryInterface(PPDMIBASE pInterface, const char *pszIID) { PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface); PDRVHSTAUDALSA pThis = PDMINS_2_DATA(pDrvIns, PDRVHSTAUDALSA); PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase); PDMIBASE_RETURN_INTERFACE(pszIID, PDMIHOSTAUDIO, &pThis->IHostAudio); return NULL; } /********************************************************************************************************************************* * PDMDRVREG * *********************************************************************************************************************************/ /** * @interface_method_impl{PDMDRVREG,pfnDestruct, * Destructs an ALSA host audio driver instance.} */ static DECLCALLBACK(void) drvHstAudAlsaDestruct(PPDMDRVINS pDrvIns) { PDMDRV_CHECK_VERSIONS_RETURN_VOID(pDrvIns); PDRVHSTAUDALSA pThis = PDMINS_2_DATA(pDrvIns, PDRVHSTAUDALSA); LogFlowFuncEnter(); if (RTCritSectIsInitialized(&pThis->CritSect)) { RTCritSectEnter(&pThis->CritSect); pThis->pIHostAudioPort = NULL; RTCritSectLeave(&pThis->CritSect); RTCritSectDelete(&pThis->CritSect); } LogFlowFuncLeave(); } /** * @interface_method_impl{PDMDRVREG,pfnConstruct, * Construct an ALSA host audio driver instance.} */ static DECLCALLBACK(int) drvHstAudAlsaConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfg, uint32_t fFlags) { RT_NOREF(fFlags); PDMDRV_CHECK_VERSIONS_RETURN(pDrvIns); PDRVHSTAUDALSA pThis = PDMINS_2_DATA(pDrvIns, PDRVHSTAUDALSA); PCPDMDRVHLPR3 pHlp = pDrvIns->pHlpR3; LogRel(("Audio: Initializing ALSA driver\n")); /* * Init the static parts. */ pThis->pDrvIns = pDrvIns; int rc = RTCritSectInit(&pThis->CritSect); AssertRCReturn(rc, rc); /* IBase */ pDrvIns->IBase.pfnQueryInterface = drvHstAudAlsaQueryInterface; /* IHostAudio */ pThis->IHostAudio.pfnGetConfig = drvHstAudAlsaHA_GetConfig; pThis->IHostAudio.pfnGetDevices = drvHstAudAlsaHA_GetDevices; pThis->IHostAudio.pfnSetDevice = drvHstAudAlsaHA_SetDevice; pThis->IHostAudio.pfnGetStatus = drvHstAudAlsaHA_GetStatus; pThis->IHostAudio.pfnDoOnWorkerThread = NULL; pThis->IHostAudio.pfnStreamConfigHint = NULL; pThis->IHostAudio.pfnStreamCreate = drvHstAudAlsaHA_StreamCreate; pThis->IHostAudio.pfnStreamInitAsync = NULL; pThis->IHostAudio.pfnStreamDestroy = drvHstAudAlsaHA_StreamDestroy; pThis->IHostAudio.pfnStreamNotifyDeviceChanged = NULL; pThis->IHostAudio.pfnStreamEnable = drvHstAudAlsaHA_StreamEnable; pThis->IHostAudio.pfnStreamDisable = drvHstAudAlsaHA_StreamDisable; pThis->IHostAudio.pfnStreamPause = drvHstAudAlsaHA_StreamPause; pThis->IHostAudio.pfnStreamResume = drvHstAudAlsaHA_StreamResume; pThis->IHostAudio.pfnStreamDrain = drvHstAudAlsaHA_StreamDrain; pThis->IHostAudio.pfnStreamGetPending = drvHstAudAlsaHA_StreamGetPending; pThis->IHostAudio.pfnStreamGetState = drvHstAudAlsaHA_StreamGetState; pThis->IHostAudio.pfnStreamGetWritable = drvHstAudAlsaHA_StreamGetWritable; pThis->IHostAudio.pfnStreamPlay = drvHstAudAlsaHA_StreamPlay; pThis->IHostAudio.pfnStreamGetReadable = drvHstAudAlsaHA_StreamGetReadable; pThis->IHostAudio.pfnStreamCapture = drvHstAudAlsaHA_StreamCapture; /* * Read configuration. */ PDMDRV_VALIDATE_CONFIG_RETURN(pDrvIns, "OutputDeviceID|InputDeviceID", ""); rc = pHlp->pfnCFGMQueryStringDef(pCfg, "InputDeviceID", pThis->szInputDev, sizeof(pThis->szInputDev), "default"); AssertRCReturn(rc, rc); rc = pHlp->pfnCFGMQueryStringDef(pCfg, "OutputDeviceID", pThis->szOutputDev, sizeof(pThis->szOutputDev), "default"); AssertRCReturn(rc, rc); /* * Init the alsa library. */ rc = audioLoadAlsaLib(); if (RT_FAILURE(rc)) { LogRel(("ALSA: Failed to load the ALSA shared library: %Rrc\n", rc)); return rc; } /* * Query the notification interface from the driver/device above us. */ pThis->pIHostAudioPort = PDMIBASE_QUERY_INTERFACE(pDrvIns->pUpBase, PDMIHOSTAUDIOPORT); AssertReturn(pThis->pIHostAudioPort, VERR_PDM_MISSING_INTERFACE_ABOVE); #ifdef DEBUG /* * Some debug stuff we don't use for anything at all. */ snd_lib_error_set_handler(drvHstAudAlsaDbgErrorHandler); #endif return VINF_SUCCESS; } /** * ALSA audio driver registration record. */ const PDMDRVREG g_DrvHostALSAAudio = { /* u32Version */ PDM_DRVREG_VERSION, /* szName */ "ALSAAudio", /* szRCMod */ "", /* szR0Mod */ "", /* pszDescription */ "ALSA host audio driver", /* fFlags */ PDM_DRVREG_FLAGS_HOST_BITS_DEFAULT, /* fClass. */ PDM_DRVREG_CLASS_AUDIO, /* cMaxInstances */ ~0U, /* cbInstance */ sizeof(DRVHSTAUDALSA), /* pfnConstruct */ drvHstAudAlsaConstruct, /* pfnDestruct */ drvHstAudAlsaDestruct, /* pfnRelocate */ NULL, /* pfnIOCtl */ NULL, /* pfnPowerOn */ NULL, /* pfnReset */ NULL, /* pfnSuspend */ NULL, /* pfnResume */ NULL, /* pfnAttach */ NULL, /* pfnDetach */ NULL, /* pfnPowerOff */ NULL, /* pfnSoftReset */ NULL, /* u32EndVersion */ PDM_DRVREG_VERSION };