/* $Id: pdmaudioinline.h 89846 2021-06-23 08:39:03Z vboxsync $ */ /** @file * PDM - Audio Helpers, Inlined Code. (DEV,++) * * This is all inlined because it's too tedious to create a couple libraries to * contain it all (same bad excuse as for intnetinline.h & pdmnetinline.h). */ /* * Copyright (C) 2006-2020 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. * * The contents of this file may alternatively be used under the terms * of the Common Development and Distribution License Version 1.0 * (CDDL) only, as it comes in the "COPYING.CDDL" file of the * VirtualBox OSE distribution, in which case the provisions of the * CDDL are applicable instead of those of the GPL. * * You may elect to license modified versions of this file under the * terms and conditions of either the GPL or the CDDL or both. */ #ifndef VBOX_INCLUDED_vmm_pdmaudioinline_h #define VBOX_INCLUDED_vmm_pdmaudioinline_h #ifndef RT_WITHOUT_PRAGMA_ONCE # pragma once #endif /********************************************************************************************************************************* * Header Files * *********************************************************************************************************************************/ #include #include #include #include #include #include #include #include /** @defgroup grp_pdm_audio_inline The PDM Audio Helper APIs * @ingroup grp_pdm * @{ */ /** * Gets the name of an audio direction enum value. * * @returns Pointer to read-only name string on success, "bad" if passed an * invalid enum value. * @param enmDir The audio direction value to name. */ DECLINLINE(const char *) PDMAudioDirGetName(PDMAUDIODIR enmDir) { switch (enmDir) { case PDMAUDIODIR_INVALID: return "invalid"; case PDMAUDIODIR_UNKNOWN: return "unknown"; case PDMAUDIODIR_IN: return "input"; case PDMAUDIODIR_OUT: return "output"; case PDMAUDIODIR_DUPLEX: return "duplex"; /* no default */ case PDMAUDIODIR_END: case PDMAUDIODIR_32BIT_HACK: break; } AssertMsgFailedReturn(("Invalid audio direction %d\n", enmDir), "bad"); } /** * Gets the name of an audio mixer control enum value. * * @returns Pointer to read-only name, "bad" if invalid input. * @param enmMixerCtl The audio mixer control value. */ DECLINLINE(const char *) PDMAudioMixerCtlGetName(PDMAUDIOMIXERCTL enmMixerCtl) { switch (enmMixerCtl) { case PDMAUDIOMIXERCTL_INVALID: return "Invalid"; case PDMAUDIOMIXERCTL_UNKNOWN: return "Unknown"; case PDMAUDIOMIXERCTL_VOLUME_MASTER: return "Master Volume"; case PDMAUDIOMIXERCTL_FRONT: return "Front"; case PDMAUDIOMIXERCTL_CENTER_LFE: return "Center / LFE"; case PDMAUDIOMIXERCTL_REAR: return "Rear"; case PDMAUDIOMIXERCTL_LINE_IN: return "Line-In"; case PDMAUDIOMIXERCTL_MIC_IN: return "Microphone-In"; /* no default */ case PDMAUDIOMIXERCTL_END: case PDMAUDIOMIXERCTL_32BIT_HACK: break; } AssertMsgFailedReturn(("Invalid mixer control %ld\n", enmMixerCtl), "bad"); } /** * Gets the name of a path enum value. * * @returns Pointer to read-only name, "bad" if invalid input. * @param enmPath The path value to name. */ DECLINLINE(const char *) PDMAudioPathGetName(PDMAUDIOPATH enmPath) { switch (enmPath) { case PDMAUDIOPATH_INVALID: return "invalid"; case PDMAUDIOPATH_UNKNOWN: return "unknown"; case PDMAUDIOPATH_OUT_FRONT: return "front"; case PDMAUDIOPATH_OUT_CENTER_LFE: return "center-lfe"; case PDMAUDIOPATH_OUT_REAR: return "rear"; case PDMAUDIOPATH_IN_MIC: return "mic"; case PDMAUDIOPATH_IN_CD: return "cd"; case PDMAUDIOPATH_IN_VIDEO: return "video-in"; case PDMAUDIOPATH_IN_AUX: return "aux-in"; case PDMAUDIOPATH_IN_LINE: return "line-in"; case PDMAUDIOPATH_IN_PHONE: return "phone"; /* no default */ case PDMAUDIOPATH_END: case PDMAUDIOPATH_32BIT_HACK: break; } AssertMsgFailedReturn(("Unknown enmPath=%d\n", enmPath), "bad"); } /** * Gets the name of a channel. * * @returns Pointer to read-only name, "bad" if invalid input. * @param enmChannelId The channel ID to name. */ DECLINLINE(const char *) PDMAudioChannelIdGetName(PDMAUDIOCHANNELID enmChannelId) { switch (enmChannelId) { case PDMAUDIOCHANNELID_INVALID: return "invalid"; case PDMAUDIOCHANNELID_UNUSED_ZERO: return "unused-zero"; case PDMAUDIOCHANNELID_UNUSED_SILENCE: return "unused-silence"; case PDMAUDIOCHANNELID_UNKNOWN: return "unknown"; case PDMAUDIOCHANNELID_FRONT_LEFT: return "FL"; case PDMAUDIOCHANNELID_FRONT_RIGHT: return "FR"; case PDMAUDIOCHANNELID_FRONT_CENTER: return "FC"; case PDMAUDIOCHANNELID_LFE: return "LFE"; case PDMAUDIOCHANNELID_REAR_LEFT: return "BL"; case PDMAUDIOCHANNELID_REAR_RIGHT: return "BR"; case PDMAUDIOCHANNELID_FRONT_LEFT_OF_CENTER: return "FLC"; case PDMAUDIOCHANNELID_FRONT_RIGHT_OF_CENTER: return "FRC"; case PDMAUDIOCHANNELID_REAR_CENTER: return "BC"; case PDMAUDIOCHANNELID_SIDE_LEFT: return "SL"; case PDMAUDIOCHANNELID_SIDE_RIGHT: return "SR"; case PDMAUDIOCHANNELID_TOP_CENTER: return "TC"; case PDMAUDIOCHANNELID_FRONT_LEFT_HEIGHT: return "TFL"; case PDMAUDIOCHANNELID_FRONT_CENTER_HEIGHT: return "TFC"; case PDMAUDIOCHANNELID_FRONT_RIGHT_HEIGHT: return "TFR"; case PDMAUDIOCHANNELID_REAR_LEFT_HEIGHT: return "TBL"; case PDMAUDIOCHANNELID_REAR_CENTER_HEIGHT: return "TBC"; case PDMAUDIOCHANNELID_REAR_RIGHT_HEIGHT: return "TBR"; /* no default */ case PDMAUDIOCHANNELID_END: case PDMAUDIOCHANNELID_32BIT_HACK: break; } AssertMsgFailedReturn(("Unknown enmChannelId=%d\n", enmChannelId), "bad"); } /********************************************************************************************************************************* * Volume Helpers * *********************************************************************************************************************************/ /** * Initializes a PDMAUDIOVOLUME structure to max. * * @param pVol The structure to initialize. */ DECLINLINE(void) PDMAudioVolumeInitMax(PPDMAUDIOVOLUME pVol) { pVol->fMuted = false; for (uintptr_t i = 0; i < RT_ELEMENTS(pVol->auChannels); i++) pVol->auChannels[i] = PDMAUDIO_VOLUME_MAX; } /** * Initializes a PDMAUDIOVOLUME structure from a simple stereo setting. * * The additional channels will simply be assigned the higer of the two. * * @param pVol The structure to initialize. * @param fMuted Muted. * @param bLeft The left channel volume. * @param bRight The right channel volume. */ DECLINLINE(void) PDMAudioVolumeInitFromStereo(PPDMAUDIOVOLUME pVol, bool fMuted, uint8_t bLeft, uint8_t bRight) { pVol->fMuted = fMuted; pVol->auChannels[0] = bLeft; pVol->auChannels[1] = bRight; uint8_t const bOther = RT_MAX(bLeft, bRight); for (uintptr_t i = 2; i < RT_ELEMENTS(pVol->auChannels); i++) pVol->auChannels[i] = bOther; } /** * Combines two volume settings (typically master and sink). * * @param pVol Where to return the combined volume * @param pVol1 The first volume settings to combine. * @param pVol2 The second volume settings. */ DECLINLINE(void) PDMAudioVolumeCombine(PPDMAUDIOVOLUME pVol, PCPDMAUDIOVOLUME pVol1, PCPDMAUDIOVOLUME pVol2) { if (pVol1->fMuted || pVol2->fMuted) { pVol->fMuted = true; for (uintptr_t i = 0; i < RT_ELEMENTS(pVol->auChannels); i++) pVol->auChannels[i] = 0; } else { pVol->fMuted = false; /** @todo Very crude implementation for now -- needs more work! (At least * when used in audioMixerSinkUpdateVolume it was considered as such.) */ for (uintptr_t i = 0; i < RT_ELEMENTS(pVol->auChannels); i++) { #if 0 /* bird: I think the shift variant should produce the exact same result, w/o two conditionals per iteration. */ /* 255 * 255 / 255 = 0xFF (255) */ /* 17 * 127 / 255 = 8 */ /* 39 * 39 / 255 = 5 */ pVol->auChannels[i] = (uint8_t)( (RT_MAX(pVol1->auChannels[i], 1U) * RT_MAX(pVol2->auChannels[i], 1U)) / PDMAUDIO_VOLUME_MAX); #else /* (((255 + 1) * (255 + 1)) >> 8) - 1 = 0xFF (255) */ /* ((( 17 + 1) * (127 + 1)) >> 8) - 1 = 0x8 (8) */ /* ((( 39 + 1) * ( 39 + 1)) >> 8) - 1 = 0x5 (5) */ pVol->auChannels[i] = (uint8_t)((((1U + pVol1->auChannels[i]) * (1U + pVol2->auChannels[i])) >> 8) - 1U); #endif } } } /********************************************************************************************************************************* * PCM Property Helpers * *********************************************************************************************************************************/ /** * Assigns default channel IDs according to the channel count. * * The assignments are taken from the standard speaker channel layouts table * in the wikipedia article on surround sound: * https://en.wikipedia.org/wiki/Surround_sound#Standard_speaker_channels */ DECLINLINE(void) PDMAudioPropsSetDefaultChannelIds(PPDMAUDIOPCMPROPS pProps) { unsigned cChannels = pProps->cChannelsX; switch (cChannels) { case 1: pProps->aidChannels[0] = PDMAUDIOCHANNELID_MONO; break; case 2: pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; break; case 3: /* 2.1 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_LFE; break; case 4: /* 4.0 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_REAR_LEFT; pProps->aidChannels[3] = PDMAUDIOCHANNELID_REAR_RIGHT; break; case 5: /* 4.1 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_FRONT_CENTER; pProps->aidChannels[3] = PDMAUDIOCHANNELID_LFE; pProps->aidChannels[4] = PDMAUDIOCHANNELID_REAR_CENTER; break; case 6: /* 5.1 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_FRONT_CENTER; pProps->aidChannels[3] = PDMAUDIOCHANNELID_LFE; pProps->aidChannels[4] = PDMAUDIOCHANNELID_REAR_LEFT; pProps->aidChannels[5] = PDMAUDIOCHANNELID_REAR_RIGHT; break; case 7: /* 6.1 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_FRONT_CENTER; pProps->aidChannels[3] = PDMAUDIOCHANNELID_LFE; pProps->aidChannels[4] = PDMAUDIOCHANNELID_REAR_LEFT; pProps->aidChannels[5] = PDMAUDIOCHANNELID_REAR_RIGHT; pProps->aidChannels[6] = PDMAUDIOCHANNELID_REAR_CENTER; break; case 8: /* 7.1 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_FRONT_CENTER; pProps->aidChannels[3] = PDMAUDIOCHANNELID_LFE; pProps->aidChannels[4] = PDMAUDIOCHANNELID_REAR_LEFT; pProps->aidChannels[5] = PDMAUDIOCHANNELID_REAR_RIGHT; pProps->aidChannels[6] = PDMAUDIOCHANNELID_FRONT_LEFT_OF_CENTER; pProps->aidChannels[7] = PDMAUDIOCHANNELID_FRONT_RIGHT_OF_CENTER; break; case 9: /* 9.0 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_FRONT_CENTER; pProps->aidChannels[3] = PDMAUDIOCHANNELID_REAR_LEFT; pProps->aidChannels[4] = PDMAUDIOCHANNELID_REAR_RIGHT; pProps->aidChannels[5] = PDMAUDIOCHANNELID_SIDE_LEFT; pProps->aidChannels[6] = PDMAUDIOCHANNELID_SIDE_RIGHT; pProps->aidChannels[7] = PDMAUDIOCHANNELID_FRONT_LEFT_HEIGHT; pProps->aidChannels[8] = PDMAUDIOCHANNELID_FRONT_RIGHT_HEIGHT; break; case 10: /* 9.1 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_FRONT_CENTER; pProps->aidChannels[3] = PDMAUDIOCHANNELID_LFE; pProps->aidChannels[4] = PDMAUDIOCHANNELID_REAR_LEFT; pProps->aidChannels[5] = PDMAUDIOCHANNELID_REAR_RIGHT; pProps->aidChannels[6] = PDMAUDIOCHANNELID_SIDE_LEFT; pProps->aidChannels[7] = PDMAUDIOCHANNELID_SIDE_RIGHT; pProps->aidChannels[8] = PDMAUDIOCHANNELID_FRONT_LEFT_HEIGHT; pProps->aidChannels[9] = PDMAUDIOCHANNELID_FRONT_RIGHT_HEIGHT; break; case 11: /* 11.0 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_FRONT_CENTER; pProps->aidChannels[3] = PDMAUDIOCHANNELID_REAR_LEFT; pProps->aidChannels[4] = PDMAUDIOCHANNELID_REAR_RIGHT; pProps->aidChannels[5] = PDMAUDIOCHANNELID_FRONT_LEFT_OF_CENTER; pProps->aidChannels[6] = PDMAUDIOCHANNELID_FRONT_RIGHT_OF_CENTER; pProps->aidChannels[7] = PDMAUDIOCHANNELID_SIDE_LEFT; pProps->aidChannels[8] = PDMAUDIOCHANNELID_SIDE_RIGHT; pProps->aidChannels[9] = PDMAUDIOCHANNELID_FRONT_LEFT_HEIGHT; pProps->aidChannels[10]= PDMAUDIOCHANNELID_FRONT_RIGHT_HEIGHT; break; default: AssertFailed(); cChannels = 12; RT_FALL_THROUGH(); case 12: /* 11.1 */ pProps->aidChannels[0] = PDMAUDIOCHANNELID_FRONT_LEFT; pProps->aidChannels[1] = PDMAUDIOCHANNELID_FRONT_RIGHT; pProps->aidChannels[2] = PDMAUDIOCHANNELID_FRONT_CENTER; pProps->aidChannels[3] = PDMAUDIOCHANNELID_LFE; pProps->aidChannels[4] = PDMAUDIOCHANNELID_REAR_LEFT; pProps->aidChannels[5] = PDMAUDIOCHANNELID_REAR_RIGHT; pProps->aidChannels[6] = PDMAUDIOCHANNELID_FRONT_LEFT_OF_CENTER; pProps->aidChannels[7] = PDMAUDIOCHANNELID_FRONT_RIGHT_OF_CENTER; pProps->aidChannels[8] = PDMAUDIOCHANNELID_SIDE_LEFT; pProps->aidChannels[9] = PDMAUDIOCHANNELID_SIDE_RIGHT; pProps->aidChannels[10]= PDMAUDIOCHANNELID_FRONT_LEFT_HEIGHT; pProps->aidChannels[11]= PDMAUDIOCHANNELID_FRONT_RIGHT_HEIGHT; break; case 0: break; } AssertCompile(RT_ELEMENTS(pProps->aidChannels) >= 12); while (cChannels < RT_ELEMENTS(pProps->aidChannels)) pProps->aidChannels[cChannels++] = PDMAUDIOCHANNELID_INVALID; } /** * Initialize PCM audio properties. */ DECLINLINE(void) PDMAudioPropsInit(PPDMAUDIOPCMPROPS pProps, uint8_t cbSample, bool fSigned, uint8_t cChannels, uint32_t uHz) { pProps->cbFrame = cbSample * cChannels; pProps->cbSampleX = cbSample; pProps->cChannelsX = cChannels; pProps->cShiftX = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(cbSample, cChannels); pProps->fSigned = fSigned; pProps->fSwapEndian = false; pProps->fRaw = false; pProps->uHz = uHz; Assert(pProps->cbFrame == (uint32_t)cbSample * cChannels); Assert(pProps->cbSampleX == cbSample); Assert(pProps->cChannelsX == cChannels); PDMAudioPropsSetDefaultChannelIds(pProps); } /** * Initialize PCM audio properties, extended version. */ DECLINLINE(void) PDMAudioPropsInitEx(PPDMAUDIOPCMPROPS pProps, uint8_t cbSample, bool fSigned, uint8_t cChannels, uint32_t uHz, bool fLittleEndian, bool fRaw) { Assert(!fRaw || cbSample == sizeof(int64_t)); pProps->cbFrame = cbSample * cChannels; pProps->cbSampleX = cbSample; pProps->cChannelsX = cChannels; pProps->cShiftX = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(cbSample, cChannels); pProps->fSigned = fSigned; #ifdef RT_LITTLE_ENDIAN pProps->fSwapEndian = !fLittleEndian; #else pProps->fSwapEndian = fLittleEndian; #endif pProps->fRaw = fRaw; pProps->uHz = uHz; Assert(pProps->cbFrame == (uint32_t)cbSample * cChannels); Assert(pProps->cbSampleX == cbSample); Assert(pProps->cChannelsX == cChannels); PDMAudioPropsSetDefaultChannelIds(pProps); } /** * Modifies the channel count. * * @note This will reset the channel IDs to defaults. * * @param pProps The PCM properties to update. * @param cChannels The new channel count. */ DECLINLINE(void) PDMAudioPropsSetChannels(PPDMAUDIOPCMPROPS pProps, uint8_t cChannels) { Assert(cChannels > 0); Assert(cChannels < 16); pProps->cChannelsX = cChannels; pProps->cbFrame = pProps->cbSampleX * cChannels; pProps->cShiftX = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pProps->cbSampleX, cChannels); PDMAudioPropsSetDefaultChannelIds(pProps); } /** * Modifies the sample size. * * @param pProps The PCM properties to update. * @param cbSample The new sample size (in bytes). */ DECLINLINE(void) PDMAudioPropsSetSampleSize(PPDMAUDIOPCMPROPS pProps, uint8_t cbSample) { Assert(cbSample == 1 || cbSample == 2 || cbSample == 4 || cbSample == 8); pProps->cbSampleX = cbSample; pProps->cbFrame = cbSample * pProps->cChannelsX; pProps->cShiftX = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(cbSample, pProps->cChannelsX); } /** * Gets the bitrate. * * Divide the result by 8 to get the byte rate. * * @returns Bit rate. * @param pProps PCM properties to calculate bitrate for. */ DECLINLINE(uint32_t) PDMAudioPropsGetBitrate(PCPDMAUDIOPCMPROPS pProps) { Assert(pProps->cbFrame == pProps->cbSampleX * pProps->cChannelsX); return pProps->cbFrame * pProps->uHz * 8; } /** * Gets the number of channels. * @returns The channel count. * @param pProps The PCM properties. */ DECL_FORCE_INLINE(uint8_t) PDMAudioPropsChannels(PCPDMAUDIOPCMPROPS pProps) { return pProps->cChannelsX; } /** * Gets the sample size in bytes. * @returns Number of bytes per sample. * @param pProps The PCM properties. */ DECL_FORCE_INLINE(uint8_t) PDMAudioPropsSampleSize(PCPDMAUDIOPCMPROPS pProps) { return pProps->cbSampleX; } /** * Gets the sample size in bits. * @returns Number of bits per sample. * @param pProps The PCM properties. */ DECLINLINE(uint8_t) PDMAudioPropsSampleBits(PCPDMAUDIOPCMPROPS pProps) { return pProps->cbSampleX * 8; } /** * Gets the frame size in bytes. * @returns Number of bytes per frame. * @param pProps The PCM properties. */ DECL_FORCE_INLINE(uint8_t) PDMAudioPropsFrameSize(PCPDMAUDIOPCMPROPS pProps) { return pProps->cbFrame; } /** * Gets the frequency. * @returns Frequency. * @param pProps The PCM properties. */ DECL_FORCE_INLINE(uint32_t) PDMAudioPropsHz(PCPDMAUDIOPCMPROPS pProps) { return pProps->uHz; } /** * Checks if the format is signed or unsigned. * @returns true if signed, false if unsigned. * @param pProps The PCM properties. */ DECL_FORCE_INLINE(bool) PDMAudioPropsIsSigned(PCPDMAUDIOPCMPROPS pProps) { return pProps->fSigned; } /** * Checks if the format is little-endian or not. * @returns true if little-endian (or if 8-bit), false if big-endian. * @param pProps The PCM properties. */ DECL_FORCE_INLINE(bool) PDMAudioPropsIsLittleEndian(PCPDMAUDIOPCMPROPS pProps) { #ifdef RT_LITTLE_ENDIAN return !pProps->fSwapEndian || pProps->cbSampleX < 2; #else return pProps->fSwapEndian || pProps->cbSampleX < 2; #endif } /** * Checks if the format is big-endian or not. * @returns true if big-endian (or if 8-bit), false if little-endian. * @param pProps The PCM properties. */ DECL_FORCE_INLINE(bool) PDMAudioPropsIsBigEndian(PCPDMAUDIOPCMPROPS pProps) { #ifdef RT_LITTLE_ENDIAN return pProps->fSwapEndian || pProps->cbSampleX < 2; #else return !pProps->fSwapEndian || pProps->cbSampleX < 2; #endif } /** * Rounds down the given byte amount to the nearest frame boundrary. * * @returns Rounded byte amount. * @param pProps PCM properties to use. * @param cb The size (in bytes) to round. */ DECLINLINE(uint32_t) PDMAudioPropsFloorBytesToFrame(PCPDMAUDIOPCMPROPS pProps, uint32_t cb) { AssertPtrReturn(pProps, 0); return PDMAUDIOPCMPROPS_F2B(pProps, PDMAUDIOPCMPROPS_B2F(pProps, cb)); } /** * Rounds up the given byte amount to the nearest frame boundrary. * * @returns Rounded byte amount. * @param pProps PCM properties to use. * @param cb The size (in bytes) to round. */ DECLINLINE(uint32_t) PDMAudioPropsRoundUpBytesToFrame(PCPDMAUDIOPCMPROPS pProps, uint32_t cb) { AssertPtrReturn(pProps, 0); uint32_t const cbFrame = PDMAudioPropsFrameSize(pProps); AssertReturn(cbFrame, 0); return PDMAUDIOPCMPROPS_F2B(pProps, PDMAUDIOPCMPROPS_B2F(pProps, cb + cbFrame - 1)); } /** * Checks if the given size is aligned on a frame boundrary. * * @returns @c true if properly aligned, @c false if not. * @param pProps PCM properties to use. * @param cb The size (in bytes) to check. */ DECLINLINE(bool) PDMAudioPropsIsSizeAligned(PCPDMAUDIOPCMPROPS pProps, uint32_t cb) { AssertPtrReturn(pProps, false); uint32_t const cbFrame = PDMAudioPropsFrameSize(pProps); AssertReturn(cbFrame, false); return cb % cbFrame == 0; } /** * Converts bytes to frames (rounding down of course). * * @returns Number of frames. * @param pProps PCM properties to use. * @param cb The number of bytes to convert. */ DECLINLINE(uint32_t) PDMAudioPropsBytesToFrames(PCPDMAUDIOPCMPROPS pProps, uint32_t cb) { AssertPtrReturn(pProps, 0); return PDMAUDIOPCMPROPS_B2F(pProps, cb); } /** * Converts bytes to milliseconds. * * @return Number milliseconds @a cb takes to play or record. * @param pProps PCM properties to use. * @param cb The number of bytes to convert. * * @note Rounds up the result. */ DECLINLINE(uint64_t) PDMAudioPropsBytesToMilli(PCPDMAUDIOPCMPROPS pProps, uint32_t cb) { AssertPtrReturn(pProps, 0); /* Check parameters to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) { const unsigned cbFrame = PDMAudioPropsFrameSize(pProps); if (cbFrame) { /* Round cb up to closest frame size: */ cb = (cb + cbFrame - 1) / cbFrame; /* Convert to milliseconds. */ return (cb * (uint64_t)RT_MS_1SEC + uHz - 1) / uHz; } } return 0; } /** * Converts bytes to microseconds. * * @return Number microseconds @a cb takes to play or record. * @param pProps PCM properties to use. * @param cb The number of bytes to convert. * * @note Rounds up the result. */ DECLINLINE(uint64_t) PDMAudioPropsBytesToMicro(PCPDMAUDIOPCMPROPS pProps, uint32_t cb) { AssertPtrReturn(pProps, 0); /* Check parameters to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) { const unsigned cbFrame = PDMAudioPropsFrameSize(pProps); if (cbFrame) { /* Round cb up to closest frame size: */ cb = (cb + cbFrame - 1) / cbFrame; /* Convert to microseconds. */ return (cb * (uint64_t)RT_US_1SEC + uHz - 1) / uHz; } } return 0; } /** * Converts bytes to nanoseconds. * * @return Number nanoseconds @a cb takes to play or record. * @param pProps PCM properties to use. * @param cb The number of bytes to convert. * * @note Rounds up the result. */ DECLINLINE(uint64_t) PDMAudioPropsBytesToNano(PCPDMAUDIOPCMPROPS pProps, uint32_t cb) { AssertPtrReturn(pProps, 0); /* Check parameters to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) { const unsigned cbFrame = PDMAudioPropsFrameSize(pProps); if (cbFrame) { /* Round cb up to closest frame size: */ cb = (cb + cbFrame - 1) / cbFrame; /* Convert to nanoseconds. */ return (cb * (uint64_t)RT_NS_1SEC + uHz - 1) / uHz; } } return 0; } /** * Converts bytes to nanoseconds, 64-bit version. * * @return Number nanoseconds @a cb takes to play or record. * @param pProps PCM properties to use. * @param cb The number of bytes to convert (64-bit). * * @note Rounds up the result. */ DECLINLINE(uint64_t) PDMAudioPropsBytesToNano64(PCPDMAUDIOPCMPROPS pProps, uint64_t cb) { AssertPtrReturn(pProps, 0); /* Check parameters to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) { const unsigned cbFrame = PDMAudioPropsFrameSize(pProps); if (cbFrame) { /* Round cb up to closest frame size: */ cb = (cb + cbFrame - 1) / cbFrame; /* Convert to nanoseconds. */ return (cb * RT_NS_1SEC + uHz - 1) / uHz; } } return 0; } /** * Converts frames to bytes. * * @returns Number of bytes. * @param pProps The PCM properties to use. * @param cFrames Number of audio frames to convert. * @sa PDMAUDIOPCMPROPS_F2B */ DECLINLINE(uint32_t) PDMAudioPropsFramesToBytes(PCPDMAUDIOPCMPROPS pProps, uint32_t cFrames) { AssertPtrReturn(pProps, 0); return PDMAUDIOPCMPROPS_F2B(pProps, cFrames); } /** * Converts frames to milliseconds. * * @returns milliseconds. * @param pProps The PCM properties to use. * @param cFrames Number of audio frames to convert. * @note No rounding here, result is floored. */ DECLINLINE(uint64_t) PDMAudioPropsFramesToMilli(PCPDMAUDIOPCMPROPS pProps, uint32_t cFrames) { AssertPtrReturn(pProps, 0); /* Check input to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) return ASMMultU32ByU32DivByU32(cFrames, RT_MS_1SEC, uHz); return 0; } /** * Converts frames to milliseconds, but not returning more than @a cMsMax * * This is a convenience for logging and such. * * @returns milliseconds (32-bit). * @param pProps The PCM properties to use. * @param cFrames Number of audio frames to convert. * @param cMsMax Max return value (32-bit). * @note No rounding here, result is floored. */ DECLINLINE(uint32_t) PDMAudioPropsFramesToMilliMax(PCPDMAUDIOPCMPROPS pProps, uint32_t cFrames, uint32_t cMsMax) { AssertPtrReturn(pProps, 0); /* Check input to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) { uint32_t const cMsResult = ASMMultU32ByU32DivByU32(cFrames, RT_MS_1SEC, uHz); return RT_MIN(cMsResult, cMsMax); } return 0; } /** * Converts frames to microseconds. * * @returns microseconds. * @param pProps The PCM properties to use. * @param cFrames Number of audio frames to convert. * @note No rounding here, result is floored. */ DECLINLINE(uint64_t) PDMAudioPropsFramesToMicro(PCPDMAUDIOPCMPROPS pProps, uint32_t cFrames) { AssertPtrReturn(pProps, 0); /* Check input to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) return ASMMultU32ByU32DivByU32(cFrames, RT_US_1SEC, uHz); return 0; } /** * Converts frames to nanoseconds. * * @returns Nanoseconds. * @param pProps The PCM properties to use. * @param cFrames Number of audio frames to convert. * @note No rounding here, result is floored. */ DECLINLINE(uint64_t) PDMAudioPropsFramesToNano(PCPDMAUDIOPCMPROPS pProps, uint32_t cFrames) { AssertPtrReturn(pProps, 0); /* Check input to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) return ASMMultU32ByU32DivByU32(cFrames, RT_NS_1SEC, uHz); return 0; } /** * Converts frames to NT ticks (100 ns units). * * @returns NT ticks. * @param pProps The PCM properties to use. * @param cFrames Number of audio frames to convert. * @note No rounding here, result is floored. */ DECLINLINE(uint64_t) PDMAudioPropsFramesToNtTicks(PCPDMAUDIOPCMPROPS pProps, uint32_t cFrames) { AssertPtrReturn(pProps, 0); /* Check input to prevent division by chainsaw: */ uint32_t const uHz = pProps->uHz; if (uHz) return ASMMultU32ByU32DivByU32(cFrames, RT_NS_1SEC / 100, uHz); return 0; } /** * Converts milliseconds to frames. * * @returns Number of frames * @param pProps The PCM properties to use. * @param cMs The number of milliseconds to convert. * * @note The result is rounded rather than floored (hysterical raisins). */ DECLINLINE(uint32_t) PDMAudioPropsMilliToFrames(PCPDMAUDIOPCMPROPS pProps, uint64_t cMs) { AssertPtrReturn(pProps, 0); uint32_t const uHz = pProps->uHz; uint32_t cFrames; if (cMs < RT_MS_1SEC) cFrames = 0; else { cFrames = cMs / RT_MS_1SEC * uHz; cMs %= RT_MS_1SEC; } cFrames += (ASMMult2xU32RetU64(uHz, (uint32_t)cMs) + RT_MS_1SEC - 1) / RT_MS_1SEC; return cFrames; } /** * Converts milliseconds to bytes. * * @returns Number of bytes (frame aligned). * @param pProps The PCM properties to use. * @param cMs The number of milliseconds to convert. * * @note The result is rounded rather than floored (hysterical raisins). */ DECLINLINE(uint32_t) PDMAudioPropsMilliToBytes(PCPDMAUDIOPCMPROPS pProps, uint64_t cMs) { return PDMAUDIOPCMPROPS_F2B(pProps, PDMAudioPropsMilliToFrames(pProps, cMs)); } /** * Converts nanoseconds to frames. * * @returns Number of frames. * @param pProps The PCM properties to use. * @param cNs The number of nanoseconds to convert. * * @note The result is rounded rather than floored (hysterical raisins). */ DECLINLINE(uint32_t) PDMAudioPropsNanoToFrames(PCPDMAUDIOPCMPROPS pProps, uint64_t cNs) { AssertPtrReturn(pProps, 0); uint32_t const uHz = pProps->uHz; uint32_t cFrames; if (cNs < RT_NS_1SEC) cFrames = 0; else { cFrames = cNs / RT_NS_1SEC * uHz; cNs %= RT_NS_1SEC; } cFrames += (ASMMult2xU32RetU64(uHz, (uint32_t)cNs) + RT_NS_1SEC - 1) / RT_NS_1SEC; return cFrames; } /** * Converts nanoseconds to frames, 64-bit return. * * @returns Number of frames (64-bit). * @param pProps The PCM properties to use. * @param cNs The number of nanoseconds to convert. * * @note The result is floored! */ DECLINLINE(uint64_t) PDMAudioPropsNanoToFrames64(PCPDMAUDIOPCMPROPS pProps, uint64_t cNs) { AssertPtrReturn(pProps, 0); uint32_t const uHz = pProps->uHz; uint64_t cFrames; if (cNs < RT_NS_1SEC) cFrames = 0; else { cFrames = cNs / RT_NS_1SEC * uHz; cNs %= RT_NS_1SEC; } cFrames += ASMMult2xU32RetU64(uHz, (uint32_t)cNs) / RT_NS_1SEC; return cFrames; } /** * Converts nanoseconds to bytes. * * @returns Number of bytes (frame aligned). * @param pProps The PCM properties to use. * @param cNs The number of nanoseconds to convert. * * @note The result is rounded rather than floored (hysterical raisins). */ DECLINLINE(uint32_t) PDMAudioPropsNanoToBytes(PCPDMAUDIOPCMPROPS pProps, uint64_t cNs) { return PDMAUDIOPCMPROPS_F2B(pProps, PDMAudioPropsNanoToFrames(pProps, cNs)); } /** * Converts nanoseconds to bytes, 64-bit version. * * @returns Number of bytes (frame aligned), 64-bit. * @param pProps The PCM properties to use. * @param cNs The number of nanoseconds to convert. * * @note The result is floored. */ DECLINLINE(uint64_t) PDMAudioPropsNanoToBytes64(PCPDMAUDIOPCMPROPS pProps, uint64_t cNs) { return PDMAUDIOPCMPROPS_F2B(pProps, PDMAudioPropsNanoToFrames(pProps, cNs)); } /** * Clears a sample buffer by the given amount of audio frames with silence (according to the format * given by the PCM properties). * * @param pProps The PCM properties to apply. * @param pvBuf The buffer to clear. * @param cbBuf The buffer size in bytes. * @param cFrames The number of audio frames to clear. Capped at @a cbBuf * if exceeding the buffer. If the size is an unaligned * number of frames, the extra bytes may be left * uninitialized in some configurations. */ DECLINLINE(void) PDMAudioPropsClearBuffer(PCPDMAUDIOPCMPROPS pProps, void *pvBuf, size_t cbBuf, uint32_t cFrames) { /* * Validate input */ AssertPtrReturnVoid(pProps); Assert(pProps->cbSampleX); if (!cbBuf || !cFrames) return; AssertPtrReturnVoid(pvBuf); /* * Decide how much needs clearing. */ size_t cbToClear = PDMAudioPropsFramesToBytes(pProps, cFrames); AssertStmt(cbToClear <= cbBuf, cbToClear = cbBuf); Log2Func(("pProps=%p, pvBuf=%p, cFrames=%RU32, fSigned=%RTbool, cbSample=%RU8\n", pProps, pvBuf, cFrames, pProps->fSigned, pProps->cbSampleX)); /* * Do the job. */ if (pProps->fSigned) RT_BZERO(pvBuf, cbToClear); else /* Unsigned formats. */ { switch (pProps->cbSampleX) { case 1: /* 8 bit */ memset(pvBuf, 0x80, cbToClear); break; case 2: /* 16 bit */ { uint16_t *pu16Dst = (uint16_t *)pvBuf; uint16_t const u16Offset = !pProps->fSwapEndian ? UINT16_C(0x8000) : UINT16_C(0x80); cbBuf /= sizeof(*pu16Dst); while (cbBuf-- > 0) *pu16Dst++ = u16Offset; break; } case 4: /* 32 bit */ ASMMemFill32(pvBuf, cbToClear & ~(size_t)(sizeof(uint32_t) - 1), !pProps->fSwapEndian ? UINT32_C(0x80000000) : UINT32_C(0x80)); break; default: AssertMsgFailed(("Invalid bytes per sample: %RU8\n", pProps->cbSampleX)); } } } /** * Checks if the given buffer is silence. * * @param pProps The PCM properties to use checking the buffer. * @param pvBuf The buffer to check. * @param cbBuf The number of bytes to check (must be frame aligned). */ DECLINLINE(bool) PDMAudioPropsIsBufferSilence(PCPDMAUDIOPCMPROPS pProps, void const *pvBuf, size_t cbBuf) { /* * Validate input */ AssertPtrReturn(pProps, false); if (!cbBuf) return false; AssertPtrReturn(pvBuf, false); /* * Do the job. */ if (pProps->fSigned) return ASMMemIsZero(pvBuf, cbBuf); switch (pProps->cbSampleX) { case 1: /* 8 bit */ return ASMMemIsAllU8(pvBuf, cbBuf, 0x80); case 2: /* 16 bit */ { uint16_t const *pu16 = (uint16_t const *)pvBuf; uint16_t const u16Offset = !pProps->fSwapEndian ? UINT16_C(0x8000) : UINT16_C(0x80); cbBuf /= sizeof(*pu16); while (cbBuf-- > 0) if (*pu16 != u16Offset) return false; return true; } case 4: /* 32 bit */ { uint32_t const *pu32 = (uint32_t const *)pvBuf; uint32_t const u32Offset = !pProps->fSwapEndian ? UINT32_C(0x80000000) : UINT32_C(0x80); cbBuf /= sizeof(*pu32); while (cbBuf-- > 0) if (*pu32 != u32Offset) return false; return true; } default: AssertMsgFailed(("Invalid bytes per sample: %RU8\n", pProps->cbSampleX)); return false; } } /** * Compares two sets of PCM properties. * * @returns @c true if the same, @c false if not. * @param pProps1 The first set of properties to compare. * @param pProps2 The second set of properties to compare. */ DECLINLINE(bool) PDMAudioPropsAreEqual(PCPDMAUDIOPCMPROPS pProps1, PCPDMAUDIOPCMPROPS pProps2) { uintptr_t idxCh; AssertPtrReturn(pProps1, false); AssertPtrReturn(pProps2, false); if (pProps1 == pProps2) /* If the pointers match, take a shortcut. */ return true; if (pProps1->uHz != pProps2->uHz) return false; if (pProps1->cChannelsX != pProps2->cChannelsX) return false; if (pProps1->cbSampleX != pProps2->cbSampleX) return false; if (pProps1->fSigned != pProps2->fSigned) return false; if (pProps1->fSwapEndian != pProps2->fSwapEndian) return false; if (pProps1->fRaw != pProps2->fRaw) return false; idxCh = pProps1->cChannelsX; while (idxCh-- > 0) if (pProps1->aidChannels[idxCh] != pProps2->aidChannels[idxCh]) return false; return true; } /** * Checks whether the given PCM properties are valid or not. * * @returns true/false accordingly. * @param pProps The PCM properties to check. * * @remarks This just performs a generic check of value ranges. Further, it * will assert if the input is invalid. * * @sa PDMAudioStrmCfgIsValid */ DECLINLINE(bool) PDMAudioPropsAreValid(PCPDMAUDIOPCMPROPS pProps) { AssertPtrReturn(pProps, false); AssertReturn(pProps->cChannelsX != 0, false); AssertReturn(pProps->cChannelsX <= PDMAUDIO_MAX_CHANNELS, false); AssertMsgReturn( pProps->cbSampleX == 1 || pProps->cbSampleX == 2 || pProps->cbSampleX == 4 || (pProps->cbSampleX == 8 && pProps->fRaw), ("%u\n", pProps->cbSampleX), false); AssertMsgReturn(pProps->cbFrame == pProps->cbSampleX * pProps->cChannelsX, ("cbFrame=%u cbSample=%u cChannels=%u\n", pProps->cbFrame, pProps->cbSampleX, pProps->cChannelsX), false); AssertMsgReturn(pProps->uHz >= 1000 && pProps->uHz < 1000000, ("%u\n", pProps->uHz), false); AssertMsgReturn(pProps->cShiftX == PDMAUDIOPCMPROPS_MAKE_SHIFT(pProps), ("cShift=%u cbSample=%u cChannels=%u\n", pProps->cShiftX, pProps->cbSampleX, pProps->cChannelsX), false); AssertReturn(!pProps->fRaw || (pProps->fSigned && pProps->cbSampleX == sizeof(int64_t)), false); return true; } /** * Get number of bytes per frame. * * @returns Number of bytes per audio frame. * @param pProps PCM properties to use. * @sa PDMAUDIOPCMPROPS_F2B */ DECLINLINE(uint32_t) PDMAudioPropsBytesPerFrame(PCPDMAUDIOPCMPROPS pProps) { return PDMAUDIOPCMPROPS_F2B(pProps, 1 /*cFrames*/); } /** * Prints PCM properties to the debug log. * * @param pProps PCM properties to use. */ DECLINLINE(void) PDMAudioPropsLog(PCPDMAUDIOPCMPROPS pProps) { AssertPtrReturnVoid(pProps); Log(("uHz=%RU32, cChannels=%RU8, cBits=%RU8%s", pProps->uHz, pProps->cChannelsX, pProps->cbSampleX * 8, pProps->fSigned ? "S" : "U")); } /** Max necessary buffer space for PDMAudioPropsToString */ #define PDMAUDIOPROPSTOSTRING_MAX sizeof("16ch S64 4294967296Hz swap raw") /** * Formats the PCM audio properties into a string buffer. * * @returns pszDst * @param pProps PCM properties to use. * @param pszDst The destination buffer. * @param cchDst The size of the destination buffer. Recommended to be at * least PDMAUDIOPROPSTOSTRING_MAX bytes. */ DECLINLINE(char *) PDMAudioPropsToString(PCPDMAUDIOPCMPROPS pProps, char *pszDst, size_t cchDst) { /* 2ch S64 44100Hz swap raw */ RTStrPrintf(pszDst, cchDst, "%uch %c%u %RU32Hz%s%s", PDMAudioPropsChannels(pProps), PDMAudioPropsIsSigned(pProps) ? 'S' : 'U', PDMAudioPropsSampleBits(pProps), PDMAudioPropsHz(pProps), pProps->fSwapEndian ? " swap" : "", pProps->fRaw ? " raw" : ""); return pszDst; } /********************************************************************************************************************************* * Stream Configuration Helpers * *********************************************************************************************************************************/ /** * Initializes a stream configuration from PCM properties. * * @returns VBox status code. * @param pCfg The stream configuration to initialize. * @param pProps The PCM properties to use. */ DECLINLINE(int) PDMAudioStrmCfgInitWithProps(PPDMAUDIOSTREAMCFG pCfg, PCPDMAUDIOPCMPROPS pProps) { AssertPtrReturn(pProps, VERR_INVALID_POINTER); AssertPtrReturn(pCfg, VERR_INVALID_POINTER); RT_ZERO(*pCfg); pCfg->Backend.cFramesPreBuffering = UINT32_MAX; /* Explicitly set to "undefined". */ memcpy(&pCfg->Props, pProps, sizeof(PDMAUDIOPCMPROPS)); return VINF_SUCCESS; } /** * Checks whether stream configuration matches the given PCM properties. * * @returns @c true if equal, @c false if not. * @param pCfg The stream configuration. * @param pProps The PCM properties to match with. */ DECLINLINE(bool) PDMAudioStrmCfgMatchesProps(PCPDMAUDIOSTREAMCFG pCfg, PCPDMAUDIOPCMPROPS pProps) { AssertPtrReturn(pCfg, false); return PDMAudioPropsAreEqual(pProps, &pCfg->Props); } /** * Checks whether two stream configuration matches. * * @returns @c true if equal, @c false if not. * @param pCfg1 The first stream configuration. * @param pCfg2 The second stream configuration. */ DECLINLINE(bool) PDMAudioStrmCfgEquals(PCPDMAUDIOSTREAMCFG pCfg1, PCPDMAUDIOSTREAMCFG pCfg2) { if (!pCfg1 || !pCfg2) return false; if (pCfg1 == pCfg2) return pCfg1 != NULL; if (PDMAudioPropsAreEqual(&pCfg1->Props, &pCfg2->Props)) return pCfg1->enmDir == pCfg2->enmDir && pCfg1->enmPath == pCfg2->enmPath && pCfg1->Device.cMsSchedulingHint == pCfg2->Device.cMsSchedulingHint && pCfg1->Backend.cFramesPeriod == pCfg2->Backend.cFramesPeriod && pCfg1->Backend.cFramesBufferSize == pCfg2->Backend.cFramesBufferSize && pCfg1->Backend.cFramesPreBuffering == pCfg2->Backend.cFramesPreBuffering && strcmp(pCfg1->szName, pCfg2->szName) == 0; return false; } /** * Frees an audio stream allocated by PDMAudioStrmCfgDup(). * * @param pCfg The stream configuration to free. */ DECLINLINE(void) PDMAudioStrmCfgFree(PPDMAUDIOSTREAMCFG pCfg) { if (pCfg) RTMemFree(pCfg); } /** * Checks whether the given stream configuration is valid or not. * * @returns true/false accordingly. * @param pCfg Stream configuration to check. * * @remarks This just performs a generic check of value ranges. Further, it * will assert if the input is invalid. * * @sa PDMAudioPropsAreValid */ DECLINLINE(bool) PDMAudioStrmCfgIsValid(PCPDMAUDIOSTREAMCFG pCfg) { AssertPtrReturn(pCfg, false); AssertMsgReturn(pCfg->enmDir >= PDMAUDIODIR_UNKNOWN && pCfg->enmDir < PDMAUDIODIR_END, ("%d\n", pCfg->enmDir), false); return PDMAudioPropsAreValid(&pCfg->Props); } /** * Copies one stream configuration to another. * * @returns VBox status code. * @param pDstCfg The destination stream configuration. * @param pSrcCfg The source stream configuration. */ DECLINLINE(int) PDMAudioStrmCfgCopy(PPDMAUDIOSTREAMCFG pDstCfg, PCPDMAUDIOSTREAMCFG pSrcCfg) { AssertPtrReturn(pDstCfg, VERR_INVALID_POINTER); AssertPtrReturn(pSrcCfg, VERR_INVALID_POINTER); /* This used to be VBOX_STRICT only and return VERR_INVALID_PARAMETER, but that's making release builds work differently from debug & strict builds, which is a terrible idea: */ Assert(PDMAudioStrmCfgIsValid(pSrcCfg)); memcpy(pDstCfg, pSrcCfg, sizeof(PDMAUDIOSTREAMCFG)); return VINF_SUCCESS; } /** * Duplicates an audio stream configuration. * * @returns Pointer to duplicate on success, NULL on failure. Must be freed * using PDMAudioStrmCfgFree(). * * @param pCfg The audio stream configuration to duplicate. */ DECLINLINE(PPDMAUDIOSTREAMCFG) PDMAudioStrmCfgDup(PCPDMAUDIOSTREAMCFG pCfg) { AssertPtrReturn(pCfg, NULL); PPDMAUDIOSTREAMCFG pDst = (PPDMAUDIOSTREAMCFG)RTMemAllocZ(sizeof(PDMAUDIOSTREAMCFG)); if (pDst) { int rc = PDMAudioStrmCfgCopy(pDst, pCfg); if (RT_SUCCESS(rc)) return pDst; PDMAudioStrmCfgFree(pDst); } return NULL; } /** * Logs an audio stream configuration. * * @param pCfg The stream configuration to log. */ DECLINLINE(void) PDMAudioStrmCfgLog(PCPDMAUDIOSTREAMCFG pCfg) { if (pCfg) LogFunc(("szName=%s enmDir=%RU32 uHz=%RU32 cBits=%RU8%s cChannels=%RU8\n", pCfg->szName, pCfg->enmDir, pCfg->Props.uHz, pCfg->Props.cbSampleX * 8, pCfg->Props.fSigned ? "S" : "U", pCfg->Props.cChannelsX)); } /** * Converts a stream command enum value to a string. * * @returns Pointer to read-only stream command name on success, * "bad" if invalid command value. * @param enmCmd The stream command to name. */ DECLINLINE(const char *) PDMAudioStrmCmdGetName(PDMAUDIOSTREAMCMD enmCmd) { switch (enmCmd) { case PDMAUDIOSTREAMCMD_INVALID: return "Invalid"; case PDMAUDIOSTREAMCMD_ENABLE: return "Enable"; case PDMAUDIOSTREAMCMD_DISABLE: return "Disable"; case PDMAUDIOSTREAMCMD_PAUSE: return "Pause"; case PDMAUDIOSTREAMCMD_RESUME: return "Resume"; case PDMAUDIOSTREAMCMD_DRAIN: return "Drain"; case PDMAUDIOSTREAMCMD_END: case PDMAUDIOSTREAMCMD_32BIT_HACK: break; /* no default! */ } AssertMsgFailedReturn(("Invalid stream command %d\n", enmCmd), "bad"); } /** Max necessary buffer space for PDMAudioStrmCfgToString */ #define PDMAUDIOSTRMCFGTOSTRING_MAX \ sizeof("'01234567890123456789012345678901234567890123456789012345678901234' unknown 16ch S64 4294967295Hz swap raw, 9999999ms buffer, 9999999ms period, 9999999ms pre-buffer, 4294967295ms sched, center-lfe") /** * Formats an audio stream configuration. * * @param pCfg The stream configuration to stringify. * @param pszDst The destination buffer. * @param cbDst The size of the destination buffer. Recommend this be * at least PDMAUDIOSTRMCFGTOSTRING_MAX bytes. */ DECLINLINE(const char *) PDMAudioStrmCfgToString(PCPDMAUDIOSTREAMCFG pCfg, char *pszDst, size_t cbDst) { /* 'front' output 2ch 44100Hz raw, 300ms buffer, 75ms period, 150ms pre-buffer, 10ms sched */ RTStrPrintf(pszDst, cbDst, "'%s' %s %uch %c%u %RU32Hz%s%s, %RU32ms buffer, %RU32ms period, %RU32ms pre-buffer, %RU32ms sched%s%s", pCfg->szName, PDMAudioDirGetName(pCfg->enmDir), PDMAudioPropsChannels(&pCfg->Props), PDMAudioPropsIsSigned(&pCfg->Props) ? 'S' : 'U', PDMAudioPropsSampleBits(&pCfg->Props), PDMAudioPropsHz(&pCfg->Props), pCfg->Props.fSwapEndian ? " swap" : "", pCfg->Props.fRaw ? " raw" : "", PDMAudioPropsFramesToMilliMax(&pCfg->Props, pCfg->Backend.cFramesBufferSize, 9999999), PDMAudioPropsFramesToMilliMax(&pCfg->Props, pCfg->Backend.cFramesPeriod, 9999999), PDMAudioPropsFramesToMilliMax(&pCfg->Props, pCfg->Backend.cFramesPreBuffering, 9999999), pCfg->Device.cMsSchedulingHint, pCfg->enmPath == PDMAUDIOPATH_UNKNOWN ? "" : ", ", pCfg->enmPath == PDMAUDIOPATH_UNKNOWN ? "" : PDMAudioPathGetName(pCfg->enmPath) ); return pszDst; } /********************************************************************************************************************************* * Stream Status Helpers * *********************************************************************************************************************************/ /** * Converts a audio stream state enum value to a string. * * @returns Pointer to read-only audio stream state string on success, * "illegal" if invalid command value. * @param enmStreamState The state to convert. */ DECLINLINE(const char *) PDMAudioStreamStateGetName(PDMAUDIOSTREAMSTATE enmStreamState) { switch (enmStreamState) { case PDMAUDIOSTREAMSTATE_INVALID: return "invalid"; case PDMAUDIOSTREAMSTATE_NOT_WORKING: return "not-working"; case PDMAUDIOSTREAMSTATE_NEED_REINIT: return "need-reinit"; case PDMAUDIOSTREAMSTATE_INACTIVE: return "inactive"; case PDMAUDIOSTREAMSTATE_ENABLED: return "enabled"; case PDMAUDIOSTREAMSTATE_ENABLED_READABLE: return "enabled-readable"; case PDMAUDIOSTREAMSTATE_ENABLED_WRITABLE: return "enabled-writable"; /* no default: */ case PDMAUDIOSTREAMSTATE_END: case PDMAUDIOSTREAMSTATE_32BIT_HACK: break; } AssertMsgFailedReturn(("Invalid audio stream state: %d\n", enmStreamState), "illegal"); } /** * Converts a host audio (backend) stream state enum value to a string. * * @returns Pointer to read-only host audio stream state string on success, * "illegal" if invalid command value. * @param enmHostAudioStreamState The state to convert. */ DECLINLINE(const char *) PDMHostAudioStreamStateGetName(PDMHOSTAUDIOSTREAMSTATE enmHostAudioStreamState) { switch (enmHostAudioStreamState) { case PDMHOSTAUDIOSTREAMSTATE_INVALID: return "invalid"; case PDMHOSTAUDIOSTREAMSTATE_INITIALIZING: return "initializing"; case PDMHOSTAUDIOSTREAMSTATE_NOT_WORKING: return "not-working"; case PDMHOSTAUDIOSTREAMSTATE_OKAY: return "okay"; case PDMHOSTAUDIOSTREAMSTATE_DRAINING: return "draining"; case PDMHOSTAUDIOSTREAMSTATE_INACTIVE: return "inactive"; /* no default: */ case PDMHOSTAUDIOSTREAMSTATE_END: case PDMHOSTAUDIOSTREAMSTATE_32BIT_HACK: break; } AssertMsgFailedReturn(("Invalid host audio stream state: %d\n", enmHostAudioStreamState), "illegal"); } /** @} */ #endif /* !VBOX_INCLUDED_vmm_pdmaudioinline_h */