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/* $Id: pdmaudioinline.h $ */
/** @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) 2006-2025 Oracle and/or its affiliates.
*
* This file is part of VirtualBox base platform packages, as
* available from https://www.virtualbox.org.
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation, in version 3 of the
* License.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, see <https://www.gnu.org/licenses>.
*
* The contents of this file may alternatively be used under the terms
* of the Common Development and Distribution License Version 1.0
* (CDDL), a copy of it is provided in the "COPYING.CDDL" file included
* in the VirtualBox 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.
*
* SPDX-License-Identifier: GPL-3.0-only OR CDDL-1.0
*/
#ifndef VBOX_INCLUDED_vmm_pdmaudioinline_h
#define VBOX_INCLUDED_vmm_pdmaudioinline_h
#ifndef RT_WITHOUT_PRAGMA_ONCE
# pragma once
#endif
/*********************************************************************************************************************************
* Header Files *
*********************************************************************************************************************************/
#include <VBox/err.h>
#include <VBox/log.h>
#include <VBox/vmm/pdmaudioifs.h>
#include <iprt/asm-mem.h>
#include <iprt/asm.h>
#include <iprt/asm-math.h>
#include <iprt/assert.h>
#include <iprt/mem.h>
#include <iprt/string.h>
/** @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)
{
Assert(cChannels <= PDMAUDIO_MAX_CHANNELS);
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(cChannels <= PDMAUDIO_MAX_CHANNELS);
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)
{
AssertReturn(pProps->cChannelsX <= PDMAUDIO_MAX_CHANNELS, PDMAUDIO_MAX_CHANNELS);
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.
*
* @sa PDMAudioStrmCfgIsValid
*/
DECLINLINE(bool) PDMAudioPropsAreValid(PCPDMAUDIOPCMPROPS pProps)
{
AssertPtrReturn(pProps, false);
/* Channels. */
if ( pProps->cChannelsX != 0
&& pProps->cChannelsX <= PDMAUDIO_MAX_CHANNELS
/* Sample size. */
&& ( pProps->cbSampleX == 1
|| pProps->cbSampleX == 2
|| pProps->cbSampleX == 4
|| (pProps->cbSampleX == 8 && pProps->fRaw))
/* Hertz rate. */
&& pProps->uHz >= 1000
&& pProps->uHz < 1000000
/* Raw format: Here we only support int64_t as sample size currently, if enabled. */
&& ( !pProps->fRaw
|| (pProps->fSigned && pProps->cbSampleX == sizeof(int64_t)))
)
{
/* A few more sanity checks to see if the structure has been properly initialized (via PDMAudioPropsInit[Ex]). */
AssertMsgReturn(pProps->cShiftX == PDMAUDIOPCMPROPS_MAKE_SHIFT(pProps),
("cShift=%u cbSample=%u cChannels=%u\n", pProps->cShiftX, pProps->cbSampleX, pProps->cChannelsX),
false);
AssertMsgReturn(pProps->cbFrame == pProps->cbSampleX * pProps->cChannelsX,
("cbFrame=%u cbSample=%u cChannels=%u\n", pProps->cbFrame, pProps->cbSampleX, pProps->cChannelsX),
false);
return true;
}
return false;
}
/**
* 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 */
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