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diff --git a/include/VBox/vmm/pdmaudioifs.h b/include/VBox/vmm/pdmaudioifs.h new file mode 100644 index 00000000..c4b1bc65 --- /dev/null +++ b/include/VBox/vmm/pdmaudioifs.h @@ -0,0 +1,1670 @@ +/** @file + * PDM - Pluggable Device Manager, audio interfaces. + */ + +/* + * Copyright (C) 2006-2019 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. + */ + +/** + * == Audio architecture overview + * + * The audio architecture mainly consists of two PDM interfaces, PDMAUDIOCONNECTOR + * and PDMIHOSTAUDIO. + * + * The PDMAUDIOCONNECTOR interface is responsible of connecting a device emulation, such + * as SB16, AC'97 and HDA to one or multiple audio backend(s). Its API abstracts audio + * stream handling and I/O functions, device enumeration and so on. + * + * The PDMIHOSTAUDIO interface must be implemented by all audio backends to provide an + * abstract and common way of accessing needed functions, such as transferring output audio + * data for playing audio or recording input from the host. + * + * A device emulation can have one or more LUNs attached to it, whereas these LUNs in turn + * then all have their own PDMIAUDIOCONNECTOR, making it possible to connect multiple backends + * to a certain device emulation stream (multiplexing). + * + * An audio backend's job is to record and/or play audio data (depending on its capabilities). + * It highly depends on the host it's running on and needs very specific (host-OS-dependent) code. + * The backend itself only has very limited ways of accessing and/or communicating with the + * PDMIAUDIOCONNECTOR interface via callbacks, but never directly with the device emulation or + * other parts of the audio sub system. + * + * + * == Mixing + * + * The AUDIOMIXER API is optionally available to create and manage virtual audio mixers. + * Such an audio mixer in turn then can be used by the device emulation code to manage all + * the multiplexing to/from the connected LUN audio streams. + * + * Currently only input and output stream are supported. Duplex stream are not supported yet. + * + * This also is handy if certain LUN audio streams should be added or removed during runtime. + * + * To create a group of either input or output streams the AUDMIXSINK API can be used. + * + * For example: The device emulation has one hardware output stream (HW0), and that output + * stream shall be available to all connected LUN backends. For that to happen, + * an AUDMIXSINK sink has to be created and attached to the device's AUDIOMIXER object. + * + * As every LUN has its own AUDMIXSTREAM object, adding all those objects to the + * just created audio mixer sink will do the job. + * + * Note: The AUDIOMIXER API is purely optional and is not used by all currently implemented + * device emulations (e.g. SB16). + * + * + * == Data processing + * + * Audio input / output data gets handed-off to/from the device emulation in an unmodified + * - that is, raw - way. The actual audio frame / sample conversion is done via the PDMAUDIOMIXBUF API. + * + * This concentrates the audio data processing in one place and makes it easier to test / benchmark + * such code. + * + * A PDMAUDIOFRAME is the internal representation of a single audio frame, which consists of a single left + * and right audio sample in time. Only mono (1) and stereo (2) channel(s) currently are supported. + * + * + * == Timing + * + * Handling audio data in a virtual environment is hard, as the human perception is very sensitive + * to the slightest cracks and stutters in the audible data. This can happen if the VM's timing is + * lagging behind or not within the expected time frame. + * + * The two main components which unfortunately contradict each other is a) the audio device emulation + * and b) the audio backend(s) on the host. Those need to be served in a timely manner to function correctly. + * To make e.g. the device emulation rely on the pace the host backend(s) set - or vice versa - will not work, + * as the guest's audio system / drivers then will not be able to compensate this accordingly. + * + * So each component, the device emulation, the audio connector(s) and the backend(s) must do its thing + * *when* it needs to do it, independently of the others. For that we use various (small) ring buffers to + * (hopefully) serve all components with the amount of data *when* they need it. + * + * Additionally, the device emulation can run with a different audio frame size, while the backends(s) may + * require a different frame size (16 bit stereo -> 8 bit mono, for example). + * + * The device emulation can give the audio connector(s) a scheduling hint (optional), e.g. in which interval + * it expects any data processing. + * + * A data transfer for playing audio data from the guest on the host looks like this: + * (RB = Ring Buffer, MB = Mixing Buffer) + * + * (A) Device DMA -> (B) Device RB -> (C) Audio Connector Guest MB -> (D) Audio Connector Host MB -> \ + * (E) Backend RB (optional, up to the backend) > (F) Backend audio framework + * + * For capturing audio data the above chain is similar, just in a different direction, of course. + * + * The audio connector hereby plays a key role when it comes to (pre-) buffering data to minimize any audio stutters + * and/or cracks. The following values, which also can be tweaked via CFGM / extra-data are available: + * + * - The pre-buffering time (in ms): Audio data which needs to be buffered before any playback (or capturing) can happen. + * - The actual buffer size (in ms): How big the mixing buffer (for C and D) will be. + * - The period size (in ms): How big a chunk of audio (often called period or fragment) for F must be to get handled correctly. + * + * The above values can be set on a per-driver level, whereas input and output streams for a driver also can be handled + * set independently. The verbose audio (release) log will tell about the (final) state of each audio stream. + * + * + * == Diagram + * + * +-------------------------+ + * +-------------------------+ +-------------------------+ +-------------------+ + * |PDMAUDIOSTREAM | |PDMAUDIOCONNECTOR | + ++|LUN | + * |-------------------------| |-------------------------| | |||-------------------| + * |PDMAUDIOMIXBUF |+------>|PDMAUDIOSTREAM Host |+---|-|||PDMIAUDIOCONNECTOR | + * |PDMAUDIOSTREAMCFG |+------>|PDMAUDIOSTREAM Guest | | |||AUDMIXSTREAM | + * | | |Device capabilities | | ||| | + * | | |Device configuration | | ||| | + * | | | | | ||| | + * | | +|PDMIHOSTAUDIO | | ||| | + * | | ||+-----------------------+| | ||+-------------------+ + * +-------------------------+ |||Backend storage space || | || + * ||+-----------------------+| | || + * |+-------------------------+ | || + * | | || + * +---------------------+ | | || + * |PDMIHOSTAUDIO | | | || + * |+--------------+ | | +-------------------+ | || +-------------+ + * ||DirectSound | | | |AUDMIXSINK | | || |AUDIOMIXER | + * |+--------------+ | | |-------------------| | || |-------------| + * | | | |AUDMIXSTREAM0 |+---|-||----->|AUDMIXSINK0 | + * |+--------------+ | | |AUDMIXSTREAM1 |+---|-||----->|AUDMIXSINK1 | + * ||PulseAudio | | | |AUDMIXSTREAMn |+---|-||----->|AUDMIXSINKn | + * |+--------------+ |+----------+ +-------------------+ | || +-------------+ + * | | | || + * |+--------------+ | | || + * ||Core Audio | | | || + * |+--------------+ | | || + * | | | || + * | | | ||+----------------------------------+ + * | | | |||Device (SB16 / AC'97 / HDA) | + * | | | |||----------------------------------| + * +---------------------+ | |||AUDIOMIXER (Optional) | + * | |||AUDMIXSINK0 (Optional) | + * | |||AUDMIXSINK1 (Optional) | + * | |||AUDMIXSINKn (Optional) | + * | ||| | + * | |+|LUN0 | + * | ++|LUN1 | + * +--+|LUNn | + * | | + * | | + * | | + * +----------------------------------+ + */ + +#ifndef VBOX_INCLUDED_vmm_pdmaudioifs_h +#define VBOX_INCLUDED_vmm_pdmaudioifs_h +#ifndef RT_WITHOUT_PRAGMA_ONCE +# pragma once +#endif + +#include <iprt/assertcompile.h> +#include <iprt/circbuf.h> +#include <iprt/list.h> +#include <iprt/path.h> + +#include <VBox/types.h> +#ifdef VBOX_WITH_STATISTICS +# include <VBox/vmm/stam.h> +#endif + +/** @defgroup grp_pdm_ifs_audio PDM Audio Interfaces + * @ingroup grp_pdm_interfaces + * @{ + */ + +#ifndef VBOX_AUDIO_DEBUG_DUMP_PCM_DATA_PATH +# ifdef RT_OS_WINDOWS +# define VBOX_AUDIO_DEBUG_DUMP_PCM_DATA_PATH "c:\\temp\\" +# else +# define VBOX_AUDIO_DEBUG_DUMP_PCM_DATA_PATH "/tmp/" +# endif +#endif + +/** PDM audio driver instance flags. */ +typedef uint32_t PDMAUDIODRVFLAGS; + +/** No flags set. */ +#define PDMAUDIODRVFLAGS_NONE 0 +/** Marks a primary audio driver which is critical + * when running the VM. */ +#define PDMAUDIODRVFLAGS_PRIMARY RT_BIT(0) + +/** + * Audio format in signed or unsigned variants. + */ +typedef enum PDMAUDIOFMT +{ + /** Invalid format, do not use. */ + PDMAUDIOFMT_INVALID, + /** 8-bit, unsigned. */ + PDMAUDIOFMT_U8, + /** 8-bit, signed. */ + PDMAUDIOFMT_S8, + /** 16-bit, unsigned. */ + PDMAUDIOFMT_U16, + /** 16-bit, signed. */ + PDMAUDIOFMT_S16, + /** 32-bit, unsigned. */ + PDMAUDIOFMT_U32, + /** 32-bit, signed. */ + PDMAUDIOFMT_S32, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOFMT_32BIT_HACK = 0x7fffffff +} PDMAUDIOFMT; + +/** + * Audio direction. + */ +typedef enum PDMAUDIODIR +{ + /** Unknown direction. */ + PDMAUDIODIR_UNKNOWN = 0, + /** Input. */ + PDMAUDIODIR_IN = 1, + /** Output. */ + PDMAUDIODIR_OUT = 2, + /** Duplex handling. */ + PDMAUDIODIR_ANY = 3, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIODIR_32BIT_HACK = 0x7fffffff +} PDMAUDIODIR; + +/** Device latency spec in milliseconds (ms). */ +typedef uint32_t PDMAUDIODEVLATSPECMS; + +/** Device latency spec in seconds (s). */ +typedef uint32_t PDMAUDIODEVLATSPECSEC; + +/** Audio device flags. Use with PDMAUDIODEV_FLAG_ flags. */ +typedef uint32_t PDMAUDIODEVFLAG; + +/** No flags set. */ +#define PDMAUDIODEV_FLAGS_NONE 0 +/** The device marks the default device within the host OS. */ +#define PDMAUDIODEV_FLAGS_DEFAULT RT_BIT(0) +/** The device can be removed at any time and we have to deal with it. */ +#define PDMAUDIODEV_FLAGS_HOTPLUG RT_BIT(1) +/** The device is known to be buggy and needs special treatment. */ +#define PDMAUDIODEV_FLAGS_BUGGY RT_BIT(2) +/** Ignore the device, no matter what. */ +#define PDMAUDIODEV_FLAGS_IGNORE RT_BIT(3) +/** The device is present but marked as locked by some other application. */ +#define PDMAUDIODEV_FLAGS_LOCKED RT_BIT(4) +/** The device is present but not in an alive state (dead). */ +#define PDMAUDIODEV_FLAGS_DEAD RT_BIT(5) + +/** + * Audio device type. + */ +typedef enum PDMAUDIODEVICETYPE +{ + /** Unknown device type. This is the default. */ + PDMAUDIODEVICETYPE_UNKNOWN = 0, + /** Dummy device; for backends which are not able to report + * actual device information (yet). */ + PDMAUDIODEVICETYPE_DUMMY, + /** The device is built into the host (non-removable). */ + PDMAUDIODEVICETYPE_BUILTIN, + /** The device is an (external) USB device. */ + PDMAUDIODEVICETYPE_USB, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIODEVICETYPE_32BIT_HACK = 0x7fffffff +} PDMAUDIODEVICETYPE; + +/** + * Audio device instance data. + */ +typedef struct PDMAUDIODEVICE +{ + /** List node. */ + RTLISTNODE Node; + /** Friendly name of the device, if any. */ + char szName[64]; + /** The device type. */ + PDMAUDIODEVICETYPE enmType; + /** Reference count indicating how many audio streams currently are relying on this device. */ + uint8_t cRefCount; + /** Usage of the device. */ + PDMAUDIODIR enmUsage; + /** Device flags. */ + PDMAUDIODEVFLAG fFlags; + /** Maximum number of input audio channels the device supports. */ + uint8_t cMaxInputChannels; + /** Maximum number of output audio channels the device supports. */ + uint8_t cMaxOutputChannels; + /** Additional data which might be relevant for the current context. */ + void *pvData; + /** Size of the additional data. */ + size_t cbData; + /** Device type union, based on enmType. */ + union + { + /** USB type specifics. */ + struct + { + /** Vendor ID. */ + int16_t VID; + /** Product ID. */ + int16_t PID; + } USB; + } Type; +} PDMAUDIODEVICE, *PPDMAUDIODEVICE; + +/** + * Structure for keeping an audio device enumeration. + */ +typedef struct PDMAUDIODEVICEENUM +{ + /** Number of audio devices in the list. */ + uint16_t cDevices; + /** List of audio devices. */ + RTLISTANCHOR lstDevices; +} PDMAUDIODEVICEENUM, *PPDMAUDIODEVICEENUM; + +/** + * Audio (static) configuration of an audio host backend. + */ +typedef struct PDMAUDIOBACKENDCFG +{ + /** The backend's friendly name. */ + char szName[32]; + /** Size (in bytes) of the host backend's audio output stream structure. */ + size_t cbStreamOut; + /** Size (in bytes) of the host backend's audio input stream structure. */ + size_t cbStreamIn; + /** Number of concurrent output (playback) streams supported on the host. + * UINT32_MAX for unlimited concurrent streams, 0 if no concurrent input streams are supported. */ + uint32_t cMaxStreamsOut; + /** Number of concurrent input (recording) streams supported on the host. + * UINT32_MAX for unlimited concurrent streams, 0 if no concurrent input streams are supported. */ + uint32_t cMaxStreamsIn; +} PDMAUDIOBACKENDCFG, *PPDMAUDIOBACKENDCFG; + +/** + * A single audio frame. + * + * Currently only two (2) channels, left and right, are supported. + * + * Note: When changing this structure, make sure to also handle + * VRDP's input / output processing in DrvAudioVRDE, as VRDP + * expects audio data in st_sample_t format (historical reasons) + * which happens to be the same as PDMAUDIOFRAME for now. + */ +typedef struct PDMAUDIOFRAME +{ + /** Left channel. */ + int64_t i64LSample; + /** Right channel. */ + int64_t i64RSample; +} PDMAUDIOFRAME; +/** Pointer to a single (stereo) audio frame. */ +typedef PDMAUDIOFRAME *PPDMAUDIOFRAME; +/** Pointer to a const single (stereo) audio frame. */ +typedef PDMAUDIOFRAME const *PCPDMAUDIOFRAME; + +typedef enum PDMAUDIOENDIANNESS +{ + /** The usual invalid endian. */ + PDMAUDIOENDIANNESS_INVALID, + /** Little endian. */ + PDMAUDIOENDIANNESS_LITTLE, + /** Bit endian. */ + PDMAUDIOENDIANNESS_BIG, + /** Endianness doesn't have a meaning in the context. */ + PDMAUDIOENDIANNESS_NA, + /** The end of the valid endian values (exclusive). */ + PDMAUDIOENDIANNESS_END, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOENDIANNESS_32BIT_HACK = 0x7fffffff +} PDMAUDIOENDIANNESS; + +/** + * Audio playback destinations. + */ +typedef enum PDMAUDIOPLAYBACKDEST +{ + /** Unknown destination. */ + PDMAUDIOPLAYBACKDEST_UNKNOWN = 0, + /** Front channel. */ + PDMAUDIOPLAYBACKDEST_FRONT, + /** Center / LFE (Subwoofer) channel. */ + PDMAUDIOPLAYBACKDEST_CENTER_LFE, + /** Rear channel. */ + PDMAUDIOPLAYBACKDEST_REAR, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOPLAYBACKDEST_32BIT_HACK = 0x7fffffff +} PDMAUDIOPLAYBACKDEST; + +/** + * Audio recording sources. + */ +typedef enum PDMAUDIORECSOURCE +{ + /** Unknown recording source. */ + PDMAUDIORECSOURCE_UNKNOWN = 0, + /** Microphone-In. */ + PDMAUDIORECSOURCE_MIC, + /** CD. */ + PDMAUDIORECSOURCE_CD, + /** Video-In. */ + PDMAUDIORECSOURCE_VIDEO, + /** AUX. */ + PDMAUDIORECSOURCE_AUX, + /** Line-In. */ + PDMAUDIORECSOURCE_LINE, + /** Phone-In. */ + PDMAUDIORECSOURCE_PHONE, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIORECSOURCE_32BIT_HACK = 0x7fffffff +} PDMAUDIORECSOURCE; + +/** + * Audio stream (data) layout. + */ +typedef enum PDMAUDIOSTREAMLAYOUT +{ + /** Unknown access type; do not use. */ + PDMAUDIOSTREAMLAYOUT_UNKNOWN = 0, + /** Non-interleaved access, that is, consecutive + * access to the data. */ + PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED, + /** Interleaved access, where the data can be + * mixed together with data of other audio streams. */ + PDMAUDIOSTREAMLAYOUT_INTERLEAVED, + /** Complex layout, which does not fit into the + * interleaved / non-interleaved layouts. */ + PDMAUDIOSTREAMLAYOUT_COMPLEX, + /** Raw (pass through) data, with no data layout processing done. + * + * This means that this stream will operate on PDMAUDIOFRAME data + * directly. Don't use this if you don't have to. */ + PDMAUDIOSTREAMLAYOUT_RAW, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOSTREAMLAYOUT_32BIT_HACK = 0x7fffffff +} PDMAUDIOSTREAMLAYOUT, *PPDMAUDIOSTREAMLAYOUT; + +/** No stream channel data flags defined. */ +#define PDMAUDIOSTREAMCHANNELDATA_FLAG_NONE 0 + +/** + * Structure for keeping a stream channel data block around. + */ +typedef struct PDMAUDIOSTREAMCHANNELDATA +{ + /** Circular buffer for the channel data. */ + PRTCIRCBUF pCircBuf; + /** Amount of audio data (in bytes) acquired for reading. */ + size_t cbAcq; + /** Channel data flags. */ + uint32_t fFlags; +} PDMAUDIOSTREAMCHANNELDATA, *PPDMAUDIOSTREAMCHANNELDATA; + +/** + * Enumeration for standard speaker channel IDs. + * This can cover up to 11.0 surround sound. + * + * Note: Any of those channels can be marked / used as the LFE channel (played through the subwoofer). + */ +typedef enum PDMAUDIOSTREAMCHANNELID +{ + /** Unknown / not set channel ID. */ + PDMAUDIOSTREAMCHANNELID_UNKNOWN = 0, + /** Front left channel. */ + PDMAUDIOSTREAMCHANNELID_FRONT_LEFT, + /** Front right channel. */ + PDMAUDIOSTREAMCHANNELID_FRONT_RIGHT, + /** Front center channel. */ + PDMAUDIOSTREAMCHANNELID_FRONT_CENTER, + /** Low frequency effects (subwoofer) channel. */ + PDMAUDIOSTREAMCHANNELID_LFE, + /** Rear left channel. */ + PDMAUDIOSTREAMCHANNELID_REAR_LEFT, + /** Rear right channel. */ + PDMAUDIOSTREAMCHANNELID_REAR_RIGHT, + /** Front left of center channel. */ + PDMAUDIOSTREAMCHANNELID_FRONT_LEFT_OF_CENTER, + /** Front right of center channel. */ + PDMAUDIOSTREAMCHANNELID_FRONT_RIGHT_OF_CENTER, + /** Rear center channel. */ + PDMAUDIOSTREAMCHANNELID_REAR_CENTER, + /** Side left channel. */ + PDMAUDIOSTREAMCHANNELID_SIDE_LEFT, + /** Side right channel. */ + PDMAUDIOSTREAMCHANNELID_SIDE_RIGHT, + /** Left height channel. */ + PDMAUDIOSTREAMCHANNELID_LEFT_HEIGHT, + /** Right height channel. */ + PDMAUDIOSTREAMCHANNELID_RIGHT_HEIGHT, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOSTREAMCHANNELID_32BIT_HACK = 0x7fffffff +} PDMAUDIOSTREAMCHANNELID; + +/** + * Structure for mapping a single (mono) channel or dual (stereo) channels of an audio stream (aka stream profile). + * + * An audio stream consists of one or multiple channels (e.g. 1 for mono, 2 for stereo), + * depending on the configuration. + */ +typedef struct PDMAUDIOSTREAMMAP +{ + /** Array of channel IDs being handled. + * Note: The first (zero-based) index specifies the leftmost channel. */ + PDMAUDIOSTREAMCHANNELID aID[2]; + /** Step size (in bytes) to the channel's next frame. */ + size_t cbSize; + /** Frame size (in bytes) of this channel. */ + size_t cbFrame; + /** Offset (in bytes) to first frame in the data block. */ + size_t cbFirst; + /** Offset (in bytes) to the next frame in the data block. */ + size_t cbOff; + /** Associated data buffer. */ + PDMAUDIOSTREAMCHANNELDATA Data; +} PDMAUDIOSTREAMMAP, *PPDMAUDIOSTREAMMAP; + +/** + * Union for keeping an audio stream destination or source. + */ +typedef union PDMAUDIODESTSOURCE +{ + /** Desired playback destination (for an output stream). */ + PDMAUDIOPLAYBACKDEST Dest; + /** Desired recording source (for an input stream). */ + PDMAUDIORECSOURCE Source; +} PDMAUDIODESTSOURCE, *PPDMAUDIODESTSOURCE; + +/** + * Properties of audio streams for host/guest for in or out directions. + */ +typedef struct PDMAUDIOPCMPROPS +{ + /** Sample width (in bytes). */ + uint8_t cBytes; + /** Number of audio channels. */ + uint8_t cChannels; + /** Shift count used for faster calculation of various + * values, such as the alignment, bytes to frames and so on. + * Depends on number of stream channels and the stream format + * being used. + * + ** @todo Use some RTAsmXXX functions instead? + */ + uint8_t cShift; + /** Signed or unsigned sample. */ + bool fSigned : 1; + /** Whether the endianness is swapped or not. */ + bool fSwapEndian : 1; + /** Sample frequency in Hertz (Hz). */ + uint32_t uHz; +} PDMAUDIOPCMPROPS; +AssertCompileSizeAlignment(PDMAUDIOPCMPROPS, 8); +/** Pointer to audio stream properties. */ +typedef PDMAUDIOPCMPROPS *PPDMAUDIOPCMPROPS; + +/** Initializor for PDMAUDIOPCMPROPS. */ +#define PDMAUDIOPCMPROPS_INITIALIZOR(a_cBytes, a_fSigned, a_cCannels, a_uHz, a_cShift, a_fSwapEndian) \ + { a_cBytes, a_cCannels, a_cShift, a_fSigned, a_fSwapEndian, a_uHz } +/** Calculates the cShift value of given sample bits and audio channels. + * Note: Does only support mono/stereo channels for now. */ +#define PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(cBytes, cChannels) ((cChannels == 2) + (cBytes / 2)) +/** Calculates the cShift value of a PDMAUDIOPCMPROPS structure. */ +#define PDMAUDIOPCMPROPS_MAKE_SHIFT(pProps) PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS((pProps)->cBytes, (pProps)->cChannels) +/** Converts (audio) frames to bytes. + * Needs the cShift value set correctly, using PDMAUDIOPCMPROPS_MAKE_SHIFT. */ +#define PDMAUDIOPCMPROPS_F2B(pProps, frames) ((frames) << (pProps)->cShift) +/** Converts bytes to (audio) frames. + * Needs the cShift value set correctly, using PDMAUDIOPCMPROPS_MAKE_SHIFT. */ +#define PDMAUDIOPCMPROPS_B2F(pProps, cb) (cb >> (pProps)->cShift) + +/** + * Structure for keeping an audio stream configuration. + */ +typedef struct PDMAUDIOSTREAMCFG +{ + /** Friendly name of the stream. */ + char szName[64]; + /** Direction of the stream. */ + PDMAUDIODIR enmDir; + /** Destination / source indicator, depending on enmDir. */ + PDMAUDIODESTSOURCE DestSource; + /** The stream's PCM properties. */ + PDMAUDIOPCMPROPS Props; + /** The stream's audio data layout. + * This indicates how the audio data buffers to/from the backend is being layouted. + * + * Currently, the following layouts are supported by the audio connector: + * + * PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED: + * One stream at once. The consecutive audio data is exactly in the format and frame width + * like defined in the PCM properties. This is the default. + * + * PDMAUDIOSTREAMLAYOUT_RAW: + * Can be one or many streams at once, depending on the stream's mixing buffer setup. + * The audio data will get handled as PDMAUDIOFRAME frames without any modification done. */ + PDMAUDIOSTREAMLAYOUT enmLayout; + /** Device emulation-specific data needed for the audio connector. */ + struct + { + /** Scheduling hint set by the device emulation about when this stream is being served on average (in ms). + * Can be 0 if not hint given or some other mechanism (e.g. callbacks) is being used. */ + uint32_t uSchedulingHintMs; + } Device; + /** + * Backend-specific data for the stream. + * On input (requested configuration) those values are set by the audio connector to let the backend know what we expect. + * On output (acquired configuration) those values reflect the values set and used by the backend. + * Set by the backend on return. Not all backends support all values / features. + */ + struct + { + /** Period size of the stream (in audio frames). + * This value reflects the number of audio frames in between each hardware interrupt on the + * backend (host) side. 0 if not set / available by the backend. */ + uint32_t cfPeriod; + /** (Ring) buffer size (in audio frames). Often is a multiple of cfPeriod. + * 0 if not set / available by the backend. */ + uint32_t cfBufferSize; + /** Pre-buffering size (in audio frames). Frames needed in buffer before the stream becomes active (pre buffering). + * The bigger this value is, the more latency for the stream will occur. + * 0 if not set / available by the backend. UINT32_MAX if not defined (yet). */ + uint32_t cfPreBuf; + } Backend; +} PDMAUDIOSTREAMCFG; +AssertCompileSizeAlignment(PDMAUDIOPCMPROPS, 8); +/** Pointer to audio stream configuration keeper. */ +typedef PDMAUDIOSTREAMCFG *PPDMAUDIOSTREAMCFG; + + +/** Converts (audio) frames to bytes. */ +#define PDMAUDIOSTREAMCFG_F2B(pCfg, frames) ((frames) << (pCfg->Props).cShift) +/** Converts bytes to (audio) frames. */ +#define PDMAUDIOSTREAMCFG_B2F(pCfg, cb) (cb >> (pCfg->Props).cShift) + +#if defined(RT_LITTLE_ENDIAN) +# define PDMAUDIOHOSTENDIANNESS PDMAUDIOENDIANNESS_LITTLE +#elif defined(RT_BIG_ENDIAN) +# define PDMAUDIOHOSTENDIANNESS PDMAUDIOENDIANNESS_BIG +#else +# error "Port me!" +#endif + +/** + * Audio mixer controls. + */ +typedef enum PDMAUDIOMIXERCTL +{ + /** Unknown mixer control. */ + PDMAUDIOMIXERCTL_UNKNOWN = 0, + /** Master volume. */ + PDMAUDIOMIXERCTL_VOLUME_MASTER, + /** Front. */ + PDMAUDIOMIXERCTL_FRONT, + /** Center / LFE (Subwoofer). */ + PDMAUDIOMIXERCTL_CENTER_LFE, + /** Rear. */ + PDMAUDIOMIXERCTL_REAR, + /** Line-In. */ + PDMAUDIOMIXERCTL_LINE_IN, + /** Microphone-In. */ + PDMAUDIOMIXERCTL_MIC_IN, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOMIXERCTL_32BIT_HACK = 0x7fffffff +} PDMAUDIOMIXERCTL; + +/** + * Audio stream commands. Used in the audio connector + * as well as in the actual host backends. + */ +typedef enum PDMAUDIOSTREAMCMD +{ + /** Unknown command, do not use. */ + PDMAUDIOSTREAMCMD_UNKNOWN = 0, + /** Enables the stream. */ + PDMAUDIOSTREAMCMD_ENABLE, + /** Disables the stream. + * For output streams this stops the stream after playing the remaining (buffered) audio data. + * For input streams this will deliver the remaining (captured) audio data and not accepting + * any new audio input data afterwards. */ + PDMAUDIOSTREAMCMD_DISABLE, + /** Pauses the stream. */ + PDMAUDIOSTREAMCMD_PAUSE, + /** Resumes the stream. */ + PDMAUDIOSTREAMCMD_RESUME, + /** Tells the stream to drain itself. + * For output streams this plays all remaining (buffered) audio frames, + * for input streams this permits receiving any new audio frames. + * No supported by all backends. */ + PDMAUDIOSTREAMCMD_DRAIN, + /** Tells the stream to drop all (buffered) audio data immediately. + * No supported by all backends. */ + PDMAUDIOSTREAMCMD_DROP, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOSTREAMCMD_32BIT_HACK = 0x7fffffff +} PDMAUDIOSTREAMCMD; + +/** + * Audio volume parameters. + */ +typedef struct PDMAUDIOVOLUME +{ + /** Set to @c true if this stream is muted, @c false if not. */ + bool fMuted; + /** Left channel volume. + * Range is from [0 ... 255], whereas 0 specifies + * the most silent and 255 the loudest value. */ + uint8_t uLeft; + /** Right channel volume. + * Range is from [0 ... 255], whereas 0 specifies + * the most silent and 255 the loudest value. */ + uint8_t uRight; +} PDMAUDIOVOLUME, *PPDMAUDIOVOLUME; + +/** Defines the minimum volume allowed. */ +#define PDMAUDIO_VOLUME_MIN (0) +/** Defines the maximum volume allowed. */ +#define PDMAUDIO_VOLUME_MAX (255) + +/** + * Structure for holding rate processing information + * of a source + destination audio stream. This is needed + * because both streams can differ regarding their rates + * and therefore need to be treated accordingly. + */ +typedef struct PDMAUDIOSTREAMRATE +{ + /** Current (absolute) offset in the output + * (destination) stream. */ + uint64_t dstOffset; + /** Increment for moving dstOffset for the + * destination stream. This is needed because the + * source <-> destination rate might be different. */ + uint64_t dstInc; + /** Current (absolute) offset in the input + * stream. */ + uint32_t srcOffset; + /** Last processed frame of the input stream. + * Needed for interpolation. */ + PDMAUDIOFRAME srcFrameLast; +} PDMAUDIOSTREAMRATE, *PPDMAUDIOSTREAMRATE; + +/** + * Structure for holding mixing buffer volume parameters. + * The volume values are in fixed point style and must + * be converted to/from before using with e.g. PDMAUDIOVOLUME. + */ +typedef struct PDMAUDMIXBUFVOL +{ + /** Set to @c true if this stream is muted, @c false if not. */ + bool fMuted; + /** Left volume to apply during conversion. Pass 0 + * to convert the original values. May not apply to + * all conversion functions. */ + uint32_t uLeft; + /** Right volume to apply during conversion. Pass 0 + * to convert the original values. May not apply to + * all conversion functions. */ + uint32_t uRight; +} PDMAUDMIXBUFVOL, *PPDMAUDMIXBUFVOL; + +/** + * Structure for holding frame conversion parameters for + * the audioMixBufConvFromXXX / audioMixBufConvToXXX macros. + */ +typedef struct PDMAUDMIXBUFCONVOPTS +{ + /** Number of audio frames to convert. */ + uint32_t cFrames; + union + { + struct + { + /** Volume to use for conversion. */ + PDMAUDMIXBUFVOL Volume; + } From; + } RT_UNION_NM(u); +} PDMAUDMIXBUFCONVOPTS; +/** Pointer to conversion parameters for the audio mixer. */ +typedef PDMAUDMIXBUFCONVOPTS *PPDMAUDMIXBUFCONVOPTS; +/** Pointer to const conversion parameters for the audio mixer. */ +typedef PDMAUDMIXBUFCONVOPTS const *PCPDMAUDMIXBUFCONVOPTS; + +/** + * Note: All internal handling is done in audio frames, + * not in bytes! + */ +typedef uint32_t PDMAUDIOMIXBUFFMT; +typedef PDMAUDIOMIXBUFFMT *PPDMAUDIOMIXBUFFMT; + +/** + * Convertion-from function used by the PDM audio buffer mixer. + * + * @returns Number of audio frames returned. + * @param paDst Where to return the converted frames. + * @param pvSrc The source frame bytes. + * @param cbSrc Number of bytes to convert. + * @param pOpts Conversion options. + */ +typedef DECLCALLBACK(uint32_t) FNPDMAUDIOMIXBUFCONVFROM(PPDMAUDIOFRAME paDst, const void *pvSrc, uint32_t cbSrc, + PCPDMAUDMIXBUFCONVOPTS pOpts); +/** Pointer to a convertion-from function used by the PDM audio buffer mixer. */ +typedef FNPDMAUDIOMIXBUFCONVFROM *PFNPDMAUDIOMIXBUFCONVFROM; + +/** + * Convertion-to function used by the PDM audio buffer mixer. + * + * @param pvDst Output buffer. + * @param paSrc The input frames. + * @param pOpts Conversion options. + */ +typedef DECLCALLBACK(void) FNPDMAUDIOMIXBUFCONVTO(void *pvDst, PCPDMAUDIOFRAME paSrc, PCPDMAUDMIXBUFCONVOPTS pOpts); +/** Pointer to a convertion-to function used by the PDM audio buffer mixer. */ +typedef FNPDMAUDIOMIXBUFCONVTO *PFNPDMAUDIOMIXBUFCONVTO; + +typedef struct PDMAUDIOMIXBUF *PPDMAUDIOMIXBUF; +typedef struct PDMAUDIOMIXBUF +{ + RTLISTNODE Node; + /** Name of the buffer. */ + char *pszName; + /** Frame buffer. */ + PPDMAUDIOFRAME pFrames; + /** Size of the frame buffer (in audio frames). */ + uint32_t cFrames; + /** The current read position (in frames). */ + uint32_t offRead; + /** The current write position (in frames). */ + uint32_t offWrite; + /** + * Total frames already mixed down to the parent buffer (if any). Always starting at + * the parent's offRead position. + * + * Note: Count always is specified in parent frames, as the sample count can differ between parent + * and child. + */ + uint32_t cMixed; + /** How much audio frames are currently being used + * in this buffer. + * Note: This also is known as the distance in ring buffer terms. */ + uint32_t cUsed; + /** Pointer to parent buffer (if any). */ + PPDMAUDIOMIXBUF pParent; + /** List of children mix buffers to keep in sync with (if being a parent buffer). */ + RTLISTANCHOR lstChildren; + /** Number of children mix buffers kept in lstChildren. */ + uint32_t cChildren; + /** Intermediate structure for buffer conversion tasks. */ + PPDMAUDIOSTREAMRATE pRate; + /** Internal representation of current volume used for mixing. */ + PDMAUDMIXBUFVOL Volume; + /** This buffer's audio format. */ + PDMAUDIOMIXBUFFMT AudioFmt; + /** Standard conversion-to function for set AudioFmt. */ + PFNPDMAUDIOMIXBUFCONVTO pfnConvTo; + /** Standard conversion-from function for set AudioFmt. */ + PFNPDMAUDIOMIXBUFCONVFROM pfnConvFrom; + /** + * Ratio of the associated parent stream's frequency by this stream's + * frequency (1<<32), represented as a signed 64 bit integer. + * + * For example, if the parent stream has a frequency of 44 khZ, and this + * stream has a frequency of 11 kHz, the ration then would be + * (44/11 * (1 << 32)). + * + * Currently this does not get changed once assigned. + */ + int64_t iFreqRatio; + /** For quickly converting frames <-> bytes and vice versa. */ + uint8_t cShift; +} PDMAUDIOMIXBUF; + +typedef uint32_t PDMAUDIOFILEFLAGS; + +/** No flags defined. */ +#define PDMAUDIOFILE_FLAG_NONE 0 +/** Keep the audio file even if it contains no audio data. */ +#define PDMAUDIOFILE_FLAG_KEEP_IF_EMPTY RT_BIT(0) +/** Audio file flag validation mask. */ +#define PDMAUDIOFILE_FLAG_VALID_MASK 0x1 + +/** Audio file default open flags. */ +#define PDMAUDIOFILE_DEFAULT_OPEN_FLAGS (RTFILE_O_OPEN_CREATE | RTFILE_O_APPEND | RTFILE_O_WRITE | RTFILE_O_DENY_WRITE) + +/** + * Audio file types. + */ +typedef enum PDMAUDIOFILETYPE +{ + /** Unknown type, do not use. */ + PDMAUDIOFILETYPE_UNKNOWN = 0, + /** Raw (PCM) file. */ + PDMAUDIOFILETYPE_RAW, + /** Wave (.WAV) file. */ + PDMAUDIOFILETYPE_WAV, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOFILETYPE_32BIT_HACK = 0x7fffffff +} PDMAUDIOFILETYPE; + +typedef uint32_t PDMAUDIOFILENAMEFLAGS; + +/** No flags defined. */ +#define PDMAUDIOFILENAME_FLAG_NONE 0 +/** Adds an ISO timestamp to the file name. */ +#define PDMAUDIOFILENAME_FLAG_TS RT_BIT(0) + +/** + * Structure for an audio file handle. + */ +typedef struct PDMAUDIOFILE +{ + /** Type of the audio file. */ + PDMAUDIOFILETYPE enmType; + /** Audio file flags. */ + PDMAUDIOFILEFLAGS fFlags; + /** File name and path. */ + char szName[RTPATH_MAX + 1]; + /** Actual file handle. */ + RTFILE hFile; + /** Data needed for the specific audio file type implemented. + * Optional, can be NULL. */ + void *pvData; + /** Data size (in bytes). */ + size_t cbData; +} PDMAUDIOFILE, *PPDMAUDIOFILE; + +/** Stream status flag. To be used with PDMAUDIOSTRMSTS_FLAG_ flags. */ +typedef uint32_t PDMAUDIOSTREAMSTS; + +/** No flags being set. */ +#define PDMAUDIOSTREAMSTS_FLAG_NONE 0 +/** Whether this stream has been initialized by the + * backend or not. */ +#define PDMAUDIOSTREAMSTS_FLAG_INITIALIZED RT_BIT_32(0) +/** Whether this stream is enabled or disabled. */ +#define PDMAUDIOSTREAMSTS_FLAG_ENABLED RT_BIT_32(1) +/** Whether this stream has been paused or not. This also implies + * that this is an enabled stream! */ +#define PDMAUDIOSTREAMSTS_FLAG_PAUSED RT_BIT_32(2) +/** Whether this stream was marked as being disabled + * but there are still associated guest output streams + * which rely on its data. */ +#define PDMAUDIOSTREAMSTS_FLAG_PENDING_DISABLE RT_BIT_32(3) +/** Whether this stream is in re-initialization phase. + * All other bits remain untouched to be able to restore + * the stream's state after the re-initialization bas been + * finished. */ +#define PDMAUDIOSTREAMSTS_FLAG_PENDING_REINIT RT_BIT_32(4) +/** Validation mask. */ +#define PDMAUDIOSTREAMSTS_VALID_MASK UINT32_C(0x0000001F) + +/** + * Enumeration presenting a backend's current status. + */ +typedef enum PDMAUDIOBACKENDSTS +{ + /** Unknown/invalid status. */ + PDMAUDIOBACKENDSTS_UNKNOWN = 0, + /** No backend attached. */ + PDMAUDIOBACKENDSTS_NOT_ATTACHED, + /** The backend is in its initialization phase. + * Not all backends support this status. */ + PDMAUDIOBACKENDSTS_INITIALIZING, + /** The backend has stopped its operation. */ + PDMAUDIOBACKENDSTS_STOPPED, + /** The backend is up and running. */ + PDMAUDIOBACKENDSTS_RUNNING, + /** The backend ran into an error and is unable to recover. + * A manual re-initialization might help. */ + PDMAUDIOBACKENDSTS_ERROR, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOBACKENDSTS_32BIT_HACK = 0x7fffffff +} PDMAUDIOBACKENDSTS; + +/** + * Structure for keeping audio input stream specifics. + * Do not use directly. Instead, use PDMAUDIOSTREAM. + */ +typedef struct PDMAUDIOSTREAMIN +{ +#ifdef VBOX_WITH_STATISTICS + struct + { + STAMCOUNTER TotalFramesCaptured; + STAMCOUNTER AvgFramesCaptured; + STAMCOUNTER TotalTimesCaptured; + STAMCOUNTER TotalFramesRead; + STAMCOUNTER AvgFramesRead; + STAMCOUNTER TotalTimesRead; + } Stats; +#endif + struct + { + /** File for writing stream reads. */ + PPDMAUDIOFILE pFileStreamRead; + /** File for writing non-interleaved captures. */ + PPDMAUDIOFILE pFileCaptureNonInterleaved; + } Dbg; +} PDMAUDIOSTREAMIN, *PPDMAUDIOSTREAMIN; + +/** + * Structure for keeping audio output stream specifics. + * Do not use directly. Instead, use PDMAUDIOSTREAM. + */ +typedef struct PDMAUDIOSTREAMOUT +{ +#ifdef VBOX_WITH_STATISTICS + struct + { + STAMCOUNTER TotalFramesPlayed; + STAMCOUNTER AvgFramesPlayed; + STAMCOUNTER TotalTimesPlayed; + STAMCOUNTER TotalFramesWritten; + STAMCOUNTER AvgFramesWritten; + STAMCOUNTER TotalTimesWritten; + } Stats; +#endif + struct + { + /** File for writing stream writes. */ + PPDMAUDIOFILE pFileStreamWrite; + /** File for writing stream playback. */ + PPDMAUDIOFILE pFilePlayNonInterleaved; + } Dbg; +} PDMAUDIOSTREAMOUT, *PPDMAUDIOSTREAMOUT; + +/** Pointer to an audio stream. */ +typedef struct PDMAUDIOSTREAM *PPDMAUDIOSTREAM; + +/** + * Audio stream context. + * Needed for separating data from the guest and host side (per stream). + */ +typedef struct PDMAUDIOSTREAMCTX +{ + /** The stream's audio configuration. */ + PDMAUDIOSTREAMCFG Cfg; + /** This stream's mixing buffer. */ + PDMAUDIOMIXBUF MixBuf; +} PDMAUDIOSTREAMCTX; + +/** Pointer to an audio stream context. */ +typedef struct PDMAUDIOSTREAM *PPDMAUDIOSTREAMCTX; + +/** + * Structure for maintaining an input/output audio stream. + */ +typedef struct PDMAUDIOSTREAM +{ + /** List node. */ + RTLISTNODE Node; + /** Name of this stream. */ + char szName[64]; + /** Number of references to this stream. Only can be + * destroyed if the reference count is reaching 0. */ + uint32_t cRefs; + /** Stream status flag. */ + PDMAUDIOSTREAMSTS fStatus; + /** Audio direction of this stream. */ + PDMAUDIODIR enmDir; + /** The guest side of the stream. */ + PDMAUDIOSTREAMCTX Guest; + /** The host side of the stream. */ + PDMAUDIOSTREAMCTX Host; + /** Union for input/output specifics (based on enmDir). */ + union + { + PDMAUDIOSTREAMIN In; + PDMAUDIOSTREAMOUT Out; + } RT_UNION_NM(u); + /** Timestamp (in ns) since last iteration. */ + uint64_t tsLastIteratedNs; + /** Timestamp (in ns) since last playback / capture. */ + uint64_t tsLastPlayedCapturedNs; + /** Timestamp (in ns) since last read (input streams) or + * write (output streams). */ + uint64_t tsLastReadWrittenNs; + /** For output streams this indicates whether the stream has reached + * its playback threshold, e.g. is playing audio. + * For input streams this indicates whether the stream has enough input + * data to actually start reading audio. */ + bool fThresholdReached; + /** Data to backend-specific stream data. + * This data block will be casted by the backend to access its backend-dependent data. + * + * That way the backends do not have access to the audio connector's data. */ + void *pvBackend; + /** Size (in bytes) of the backend-specific stream data. */ + size_t cbBackend; +} PDMAUDIOSTREAM; + +/** Pointer to a audio connector interface. */ +typedef struct PDMIAUDIOCONNECTOR *PPDMIAUDIOCONNECTOR; + +/** + * Enumeration for an audio callback source. + */ +typedef enum PDMAUDIOCBSOURCE +{ + /** Invalid, do not use. */ + PDMAUDIOCBSOURCE_INVALID = 0, + /** Device emulation. */ + PDMAUDIOCBSOURCE_DEVICE = 1, + /** Audio connector interface. */ + PDMAUDIOCBSOURCE_CONNECTOR = 2, + /** Backend (lower). */ + PDMAUDIOCBSOURCE_BACKEND = 3, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOCBSOURCE_32BIT_HACK = 0x7fffffff +} PDMAUDIOCBSOURCE; + +/** + * Audio device callback types. + * Those callbacks are being sent from the audio connector -> device emulation. + */ +typedef enum PDMAUDIODEVICECBTYPE +{ + /** Invalid, do not use. */ + PDMAUDIODEVICECBTYPE_INVALID = 0, + /** Data is availabe as input for passing to the device emulation. */ + PDMAUDIODEVICECBTYPE_DATA_INPUT, + /** Free data for the device emulation to write to the backend. */ + PDMAUDIODEVICECBTYPE_DATA_OUTPUT, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIODEVICECBTYPE_32BIT_HACK = 0x7fffffff +} PDMAUDIODEVICECBTYPE; + +/** + * Device callback data for audio input. + */ +typedef struct PDMAUDIODEVICECBDATA_DATA_INPUT +{ + /** Input: How many bytes are availabe as input for passing + * to the device emulation. */ + uint32_t cbInAvail; + /** Output: How many bytes have been read. */ + uint32_t cbOutRead; +} PDMAUDIODEVICECBDATA_DATA_INPUT, *PPDMAUDIODEVICECBDATA_DATA_INPUT; + +/** + * Device callback data for audio output. + */ +typedef struct PDMAUDIODEVICECBDATA_DATA_OUTPUT +{ + /** Input: How many bytes are free for the device emulation to write. */ + uint32_t cbInFree; + /** Output: How many bytes were written by the device emulation. */ + uint32_t cbOutWritten; +} PDMAUDIODEVICECBDATA_DATA_OUTPUT, *PPDMAUDIODEVICECBDATA_DATA_OUTPUT; + +/** + * Audio backend callback types. + * Those callbacks are being sent from the backend -> audio connector. + */ +typedef enum PDMAUDIOBACKENDCBTYPE +{ + /** Invalid, do not use. */ + PDMAUDIOBACKENDCBTYPE_INVALID = 0, + /** The backend's status has changed. */ + PDMAUDIOBACKENDCBTYPE_STATUS, + /** One or more host audio devices have changed. */ + PDMAUDIOBACKENDCBTYPE_DEVICES_CHANGED, + /** Hack to blow the type up to 32-bit. */ + PDMAUDIOBACKENDCBTYPE_32BIT_HACK = 0x7fffffff +} PDMAUDIOBACKENDCBTYPE; + +/** Pointer to a host audio interface. */ +typedef struct PDMIHOSTAUDIO *PPDMIHOSTAUDIO; + +/** + * Host audio callback function. + * This function will be called from a backend to communicate with the host audio interface. + * + * @returns IPRT status code. + * @param pDrvIns Pointer to driver instance which called us. + * @param enmType Callback type. + * @param pvUser User argument. + * @param cbUser Size (in bytes) of user argument. + */ +typedef DECLCALLBACK(int) FNPDMHOSTAUDIOCALLBACK(PPDMDRVINS pDrvIns, PDMAUDIOBACKENDCBTYPE enmType, void *pvUser, size_t cbUser); +/** Pointer to a FNPDMHOSTAUDIOCALLBACK(). */ +typedef FNPDMHOSTAUDIOCALLBACK *PFNPDMHOSTAUDIOCALLBACK; + +/** + * Audio callback registration record. + */ +typedef struct PDMAUDIOCBRECORD +{ + /** List node. */ + RTLISTANCHOR Node; + /** Callback source. */ + PDMAUDIOCBSOURCE enmSource; + /** Callback type, based on the given source. */ + union + { + /** Device callback stuff. */ + struct + { + PDMAUDIODEVICECBTYPE enmType; + } Device; + } RT_UNION_NM(u); + /** Pointer to context data. Optional. */ + void *pvCtx; + /** Size (in bytes) of context data. + * Must be 0 if pvCtx is NULL. */ + size_t cbCtx; +} PDMAUDIOCBRECORD, *PPDMAUDIOCBRECORD; + +#define PPDMAUDIOBACKENDSTREAM void * + +/** + * Audio connector interface (up). + */ +typedef struct PDMIAUDIOCONNECTOR +{ + /** + * Enables or disables the given audio direction for this driver. + * + * When disabled, assiociated output streams consume written audio without passing them further down to the backends. + * Associated input streams then return silence when read from those. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param enmDir Audio direction to enable or disable driver for. + * @param fEnable Whether to enable or disable the specified audio direction. + */ + DECLR3CALLBACKMEMBER(int, pfnEnable, (PPDMIAUDIOCONNECTOR pInterface, PDMAUDIODIR enmDir, bool fEnable)); + + /** + * Returns whether the given audio direction for this driver is enabled or not. + * + * @returns True if audio is enabled for the given direction, false if not. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param enmDir Audio direction to retrieve enabled status for. + */ + DECLR3CALLBACKMEMBER(bool, pfnIsEnabled, (PPDMIAUDIOCONNECTOR pInterface, PDMAUDIODIR enmDir)); + + /** + * Retrieves the current configuration of the host audio backend. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pCfg Where to store the host audio backend configuration data. + */ + DECLR3CALLBACKMEMBER(int, pfnGetConfig, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOBACKENDCFG pCfg)); + + /** + * Retrieves the current status of the host audio backend. + * + * @returns Status of the host audio backend. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param enmDir Audio direction to check host audio backend for. Specify PDMAUDIODIR_ANY for the overall + * backend status. + */ + DECLR3CALLBACKMEMBER(PDMAUDIOBACKENDSTS, pfnGetStatus, (PPDMIAUDIOCONNECTOR pInterface, PDMAUDIODIR enmDir)); + + /** + * Creates an audio stream. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pCfgHost Stream configuration for host side. + * @param pCfgGuest Stream configuration for guest side. + * @param ppStream Pointer where to return the created audio stream on success. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamCreate, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAMCFG pCfgHost, PPDMAUDIOSTREAMCFG pCfgGuest, PPDMAUDIOSTREAM *ppStream)); + + /** + * Destroys an audio stream. + * + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamDestroy, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream)); + + /** + * Adds a reference to the specified audio stream. + * + * @returns New reference count. UINT32_MAX on error. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream adding the reference to. + */ + DECLR3CALLBACKMEMBER(uint32_t, pfnStreamRetain, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream)); + + /** + * Releases a reference from the specified stream. + * + * @returns New reference count. UINT32_MAX on error. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream releasing a reference from. + */ + DECLR3CALLBACKMEMBER(uint32_t, pfnStreamRelease, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream)); + + /** + * Reads PCM audio data from the host (input). + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream to write to. + * @param pvBuf Where to store the read data. + * @param cbBuf Number of bytes to read. + * @param pcbRead Bytes of audio data read. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamRead, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead)); + + /** + * Writes PCM audio data to the host (output). + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream to read from. + * @param pvBuf Audio data to be written. + * @param cbBuf Number of bytes to be written. + * @param pcbWritten Bytes of audio data written. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamWrite, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, const void *pvBuf, uint32_t cbBuf, uint32_t *pcbWritten)); + + /** + * Controls a specific audio stream. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + * @param enmStreamCmd The stream command to issue. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamControl, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd)); + + /** + * Processes stream data. + * + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamIterate, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream)); + + /** + * Returns the number of readable data (in bytes) of a specific audio input stream. + * + * @returns Number of readable data (in bytes). + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(uint32_t, pfnStreamGetReadable, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream)); + + /** + * Returns the number of writable data (in bytes) of a specific audio output stream. + * + * @returns Number of writable data (in bytes). + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(uint32_t, pfnStreamGetWritable, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream)); + + /** + * Returns the status of a specific audio stream. + * + * @returns Audio stream status + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(PDMAUDIOSTREAMSTS, pfnStreamGetStatus, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream)); + + /** + * Sets the audio volume of a specific audio stream. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + * @param pVol Pointer to audio volume structure to set the stream's audio volume to. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamSetVolume, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, PPDMAUDIOVOLUME pVol)); + + /** + * Plays (transfers) available audio frames to the host backend. Only works with output streams. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + * @param pcFramesPlayed Number of frames played. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamPlay, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcFramesPlayed)); + + /** + * Captures (transfers) available audio frames from the host backend. Only works with input streams. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + * @param pcFramesCaptured Number of frames captured. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamCapture, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcFramesCaptured)); + + /** + * Registers (device) callbacks. + * This is handy for letting the device emulation know of certain events, e.g. processing input / output data + * or configuration changes. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param paCallbacks Pointer to array of callbacks to register. + * @param cCallbacks Number of callbacks to register. + */ + DECLR3CALLBACKMEMBER(int, pfnRegisterCallbacks, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOCBRECORD paCallbacks, size_t cCallbacks)); + +} PDMIAUDIOCONNECTOR; + +/** PDMIAUDIOCONNECTOR interface ID. */ +#define PDMIAUDIOCONNECTOR_IID "A643B40C-733F-4307-9549-070AF0EE0ED6" + +/** + * Assigns all needed interface callbacks for an audio backend. + * + * @param a_Prefix The function name prefix. + */ +#define PDMAUDIO_IHOSTAUDIO_CALLBACKS(a_Prefix) \ + do { \ + pThis->IHostAudio.pfnInit = RT_CONCAT(a_Prefix,Init); \ + pThis->IHostAudio.pfnShutdown = RT_CONCAT(a_Prefix,Shutdown); \ + pThis->IHostAudio.pfnGetConfig = RT_CONCAT(a_Prefix,GetConfig); \ + /** @todo Add pfnGetDevices here as soon as supported by all backends. */ \ + pThis->IHostAudio.pfnGetStatus = RT_CONCAT(a_Prefix,GetStatus); \ + /** @todo Ditto for pfnSetCallback. */ \ + pThis->IHostAudio.pfnStreamCreate = RT_CONCAT(a_Prefix,StreamCreate); \ + pThis->IHostAudio.pfnStreamDestroy = RT_CONCAT(a_Prefix,StreamDestroy); \ + pThis->IHostAudio.pfnStreamControl = RT_CONCAT(a_Prefix,StreamControl); \ + pThis->IHostAudio.pfnStreamGetReadable = RT_CONCAT(a_Prefix,StreamGetReadable); \ + pThis->IHostAudio.pfnStreamGetWritable = RT_CONCAT(a_Prefix,StreamGetWritable); \ + pThis->IHostAudio.pfnStreamGetStatus = RT_CONCAT(a_Prefix,StreamGetStatus); \ + pThis->IHostAudio.pfnStreamIterate = RT_CONCAT(a_Prefix,StreamIterate); \ + pThis->IHostAudio.pfnStreamPlay = RT_CONCAT(a_Prefix,StreamPlay); \ + pThis->IHostAudio.pfnStreamCapture = RT_CONCAT(a_Prefix,StreamCapture); \ + } while (0) + +/** + * PDM host audio interface. + */ +typedef struct PDMIHOSTAUDIO +{ + /** + * Initializes the host backend (driver). + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + */ + DECLR3CALLBACKMEMBER(int, pfnInit, (PPDMIHOSTAUDIO pInterface)); + + /** + * Shuts down the host backend (driver). + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + */ + DECLR3CALLBACKMEMBER(void, pfnShutdown, (PPDMIHOSTAUDIO pInterface)); + + /** + * Returns the host backend's configuration (backend). + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pBackendCfg Where to store the backend audio configuration to. + */ + DECLR3CALLBACKMEMBER(int, pfnGetConfig, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pBackendCfg)); + + /** + * Returns (enumerates) host audio device information. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pDeviceEnum Where to return the enumerated audio devices. + */ + DECLR3CALLBACKMEMBER(int, pfnGetDevices, (PPDMIHOSTAUDIO pInterface, PPDMAUDIODEVICEENUM pDeviceEnum)); + + /** + * Returns the current status from the audio backend. + * + * @returns PDMAUDIOBACKENDSTS enum. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param enmDir Audio direction to get status for. Pass PDMAUDIODIR_ANY for overall status. + */ + DECLR3CALLBACKMEMBER(PDMAUDIOBACKENDSTS, pfnGetStatus, (PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir)); + + /** + * Sets a callback the audio backend can call. Optional. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pfnCallback The callback function to use, or NULL when unregistering. + */ + DECLR3CALLBACKMEMBER(int, pfnSetCallback, (PPDMIHOSTAUDIO pInterface, PFNPDMHOSTAUDIOCALLBACK pfnCallback)); + + /** + * Creates an audio stream using the requested stream configuration. + * If a backend is not able to create this configuration, it will return its best match in the acquired configuration + * structure on success. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + * @param pCfgReq Pointer to requested stream configuration. + * @param pCfgAcq Pointer to acquired stream configuration. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamCreate, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq)); + + /** + * Destroys an audio stream. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamDestroy, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Controls an audio stream. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + * @param enmStreamCmd The stream command to issue. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamControl, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd)); + + /** + * Returns the amount which is readable from the audio (input) stream. + * + * @returns For non-raw layout streams: Number of readable bytes. + * for raw layout streams : Number of readable audio frames. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(uint32_t, pfnStreamGetReadable, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Returns the amount which is writable to the audio (output) stream. + * + * @returns For non-raw layout streams: Number of writable bytes. + * for raw layout streams : Number of writable audio frames. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(uint32_t, pfnStreamGetWritable, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Returns the amount which is pending (in other words has not yet been processed) by/from the backend yet. + * Optional. + * + * For input streams this is read audio data by the backend which has not been processed by the host yet. + * For output streams this is written audio data to the backend which has not been processed by the backend yet. + * + * @returns For non-raw layout streams: Number of pending bytes. + * for raw layout streams : Number of pending audio frames. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(uint32_t, pfnStreamGetPending, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Returns the current status of the given backend stream. + * + * @returns PDMAUDIOSTREAMSTS + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(PDMAUDIOSTREAMSTS, pfnStreamGetStatus, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Gives the host backend the chance to do some (necessary) iteration work. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamIterate, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Signals the backend that the host wants to begin playing for this iteration. Optional. + * + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(void, pfnStreamPlayBegin, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Plays (writes to) an audio (output) stream. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + * @param pvBuf Pointer to audio data buffer to play. + * @param cxBuf For non-raw layout streams: Size (in bytes) of audio data buffer, + * for raw layout streams : Size (in audio frames) of audio data buffer. + * @param pcxWritten For non-raw layout streams: Returns number of bytes written. Optional. + * for raw layout streams : Returns number of frames written. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamPlay, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, const void *pvBuf, uint32_t cxBuf, uint32_t *pcxWritten)); + + /** + * Signals the backend that the host finished playing for this iteration. Optional. + * + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(void, pfnStreamPlayEnd, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Signals the backend that the host wants to begin capturing for this iteration. Optional. + * + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(void, pfnStreamCaptureBegin, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + + /** + * Captures (reads from) an audio (input) stream. + * + * @returns VBox status code. + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + * @param pvBuf Buffer where to store read audio data. + * @param cxBuf For non-raw layout streams: Size (in bytes) of audio data buffer, + * for raw layout streams : Size (in audio frames) of audio data buffer. + * @param pcxRead For non-raw layout streams: Returns number of bytes read. Optional. + * for raw layout streams : Returns number of frames read. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamCapture, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, void *pvBuf, uint32_t cxBuf, uint32_t *pcxRead)); + + /** + * Signals the backend that the host finished capturing for this iteration. Optional. + * + * @param pInterface Pointer to the interface structure containing the called function pointer. + * @param pStream Pointer to audio stream. + */ + DECLR3CALLBACKMEMBER(void, pfnStreamCaptureEnd, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)); + +} PDMIHOSTAUDIO; + +/** PDMIHOSTAUDIO interface ID. */ +#define PDMIHOSTAUDIO_IID "640F5A31-8245-491C-538F-29A0F9D08881" + +/** @} */ + +#endif /* !VBOX_INCLUDED_vmm_pdmaudioifs_h */ + |