SEAL Synthetic Audio Library

Programmer's Guide

Copyright Information

The Synthetic Audio Library Software Developer Kit
Copyright (C) 1995, 1996 Carlos Hasan. All Rights Reserved.

If you have questions, problems or comments about the SEAL audio system you can contact the author at the following Internet email address:

Carlos Hasan
chasan@dcc.uchile.cl

About the Synthetic Audio Library

The SEAL API allows you, as an application developer, access the audio hardware while insulating you from the specific details of that hardware. The design goal of SEAL is speed. Instead of providing a high-level set of functions, it provides a device-independant interface, allowing applications to take full adventage of the capabilities of the audio hardware.

Supported Platforms and Hardware

• WATCOM C/C++32 10.0 Compiler for DOS4GW and Win32
• DJGPP 2.0 Compiler port of GNU CC for DOS/DPMI32
• GNU CC 2.7.0 Compiler for Linux (a.out and ELF) operating system
• Visual C++ 4.1 Compiler for Windows 95 and Windows NT
• Borland Delphi 2.0 Compiler for Windows 95 and Windows NT
• Metrowerks CodeWarrior C++ for Intel BeOS Release 3
• EMX port of GNU CC 2.7.x Compiler for OS/2 operating system

Under the DOS operating system the following hardware audio devices are natively supported by the audio system:

• Sound Blaster 1.0
• Sound Blaster 1.5
• Sound Blaster 2.0
• Sound Blaster Pro I and II
• Sound Blaster 16 / ASP
• Sound Blaster 32
• Sound Blaster AWE32
• Creative PCI64 cards
• Windows Sound System
• Ensoniq Soundscape
• Ensoniq Soundscape Elite
• Ensoniq AudioPCI cards
• Gravis Ultrasound
• Gravis Ultrasound Daughterboard
• Gravis Ultrasound MAX
• AMD InterWave-based boards
• Pro Audio Spectrum
• Pro Audio Spectrum Plus
• Pro Audio Spectrum 16
• Aria sound cards

Data Types

Features

• Mixing The most used feature of the audio system is the low-latency mixing of digital audio channels. Your application can create up to 32 voices to play concurrently audio data through your sound device. The audio system mixes all the active voices and writes the result to the audio hardware.
Low-latency mixing allows the user to experience no perceptible delay between the time that the application start playing a waveform through the audio hardware and the time that the speakers reproduct the sound.
• Hardware Acceleration The SEAL audio system automatically takes advantage of accelerated sound hardware, including hardware mixing and hardware local memory. Your application does not need to query the hardware to specifically use hardware acceleration.

Audio Device Interface

Initialization

Device Capabilities

Your application can also call the routine APingAudio to determine what kind of audio hardware is present on the system. If there is more than one device present on the system, the first detected audio device will be used.

Opening the Audio Device

While the audio system is active, your application must call the routine AUpdateAudio to update and send the output audio buffers to the audio hardware. It's not required for Windows 95 and Windows NT applications.

Closing the Audio Device

Timer Interrupt Services

Since the audio system plays music sequences using the virtual audio timer interrupt services, your applications won't be able to play music modules and use the timer services simultaneously. You must use your own timer services if you want to play music in your applications. For applications running in DOS, the hardware timer interrupt vector can be used for such purposes.

Audio Voice Interface

Creating Audio Voices

Every allocated audio voice must be released by calling ADestroyAudioVoice whenever your application finishes using it. It's important to stop the audio voices before releasing them.

The ACloseVoices routine must be called to release all the available audio voices when your application is going to close the audio system. Your application must stop all the voices and module files before calling it.

Play Management

To retrieve or set the current playing position in the waveform of an audio voice, call the AGetVoicePosition or ASetVoicePosition routines. The playing position of the voices is specified as an offset from the start of the waveform data measured in samples.

Environment Management

Similarly, by calling the AGetVoiceFrequency and ASetVoiceFrequency routines, you can retrieve and set the sampling frequency at which the audio waveform is being resampled and played by a given audio voice.

To retrieve and set the stereo pan position of an audio voice, your application can call the AGetVoicePanning and ASetVoicePanning routines. These routines have no effect when the system is playing in monophonic format.

Retrieving State Information

Audio Waveform Interface

Creating Waveforms

Playing Waveforms

Audio Music Module Interface

Creating Modules

Play Management

Environment Management

Similarly, the AGetModuleVolume and ASetModuleVolume routines can be used by your application to change the global volume at which the audio system is playing the music sequence. These routines can only be called while a music module is active.

Reference Guide

Functions

Initializes the Synthetic Audio Library.

Returns zero on success.

This routine must be called to initialize the audio system, after initialization your application can use the audio system API routines. It's not required to initialize the audio system for applications running in Windows 95 or Windows NT.

See also AOpenAudio, ACloseAudio

Retrieves the Synthetic Audio Library version.

Returns the audio system major and minor version numbers.

This routine can be used by your application to get the audio system version number, the major version number is encoded in the most significant byte of the returned value and the minor version number in the least significant byte.

Retrieves number of installed audio device drivers.

Returns the number of device drivers installed on the system.

This routine can be used by your application to enumerate the installed audio device drivers on the system. The AGetAudioDevCaps routine can be used to retrieve information from a particular audio device driver.

See also AGetAudioDevCaps

Retrieves information about audio device drivers.

Return zero on success.

• nDeviceId Audio device driver identifier.
• lpCaps Address of an AUDIOCAPS structure that will hold the capabitilies information of the audio device driver referenced by the nDeviceId device driver identifier.

    VOID PrintAudioDevs(VOID)
    {
        AUDIOCAPS caps;
        UINT nDeviceId;

        /* show all the installed device drivers */
        for (nDeviceId = 0; nDeviceId < AGetAudioNumDevs(); nDeviceId++) {
            AGetAudioDevCaps(nDeviceId, &caps);
            printf("nDeviceId=%d wProductId=%d szProductName=%s\n",
                   nDeviceId, caps.wProductId, caps.szProductName);
        }
    }

Return readable text description for a given error return code.

Return zero on success.

• nErrorCode Numeric error return code to be translated.
• lpText Address of the null-terminated text description buffer.
• nSize Number of bytes of the text description buffer.

    VOID InitializeAudio(VOID)
    {
        AUDIOINFO info;
        CHAR szText[128];
        UINT rc;

        info.nDeviceId = AUDIO_DEVICE_MAPPER;
        info.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_STEREO;
        info.nSampleRate = 44100;
        if ((rc = AOpenAudio(&info)) != AUDIO_ERROR_NONE) {
            AGetErrorText(rc, szText, sizeof(szText) - 1);
            printf("ERROR: %s\n", szText);
            exit(1);
       }
    }

Automatically detect the installed audio device.

Returns zero on success.

• lpnDeviceId Address of an unsigned integer that will hold the detect device identifier.

    VOID DetectAudioDevice(VOID)
    {
        AUDIOCAPS caps;
        UINT nDeviceId;

        if (APingAudio(&nDeviceId) != AUDIO_ERROR_NONE) {
            printf("no audio device found.\n");
        }
        else {
            AGetAudioDevCaps(nDeviceId, &caps);
            printf("%s device found.\n", caps.szProductName);
        }
    }

Open audio device for playback.

Returns zero on success.

• lpInfo Address of an AUDIOINFO structure that holds the playback configuration.

    VOID InitializeAudio(VOID)
    {
        AUDIOINFO info;

        info.nDeviceId = AUDIO_DEVICE_MAPPER;
        info.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_STEREO;
        info.nSampleRate = 44100;
        if (AOpenAudio(&info) != AUDIO_ERROR_NONE) {
            printf("Audio initialization failed.\n");
            exit(1);
        }
        else {
            printf("Audio device initialized at %d bits %s %u Hz\n",
                   info.wFormat & AUDIO_FORMAT_16BITS ? 16 : 8,
                   info.wFormat & AUDIO_FORMAT_STEREO ? 
                   "stereo": "mono", info.nSampleRate);
        }
    }

Close the audio device.

Returns zero on success.

This routine must be called to close the audio device before leaving from your application. Your application must release all the resources allocated before closing the audio system.

See Also AOpenAudio, AUpdateAudio

Updates the audio system output buffers.

Returns zero on success.

This routine must be called periodically by your application to update the audio buffers which are sent to the hardware for playback. If your application fails to call this routine enough times per second, the audio output will sound like a broken disc. It's not required to call this routine for Windows 95 and Windows NT applications.

See Also AOpenAudio, ACloseAudio

Install audio wave callback routine. Internal use only.

Returns zero on success.

This routine is used internally by the audio system to install the audio wave callback routine used by the software wavetable synthesizer to fill the hardware DMA audio buffer with samples.

Install virtual audio timer callback routine.

Returns zero on success.

• lpfnAudioTimer Address of the timer handler routine.

    volatile UINT nTickCounter = 0;

    VOID AIAPI TimerHandler(VOID)
    {
        nTickCounter++;
    }

    VOID InitTimerHandler(VOID)
    {
        /* the handler will be called at 125 BPM (50 times per second) */
        ASetAudioTimerProc(TimerHandler);
        ASetAudioTimerRate(125);
    }

    VOID DoneTimerHandler(VOID)
    {
        ASetAudioTimerProc(NULL);
    }

Changes the virtual audio timer rate.

Returns zero on success.

• nTimerRate The timer rate given in beats per minute (32-255).

See Also ASetAudioTimerProc

Open audio voices.

Returns zero on success.

• nVoices Maximum number of allocatable audio voices (1-32).

See Also ACloseVoices, ACreateAudioVoice, ADestroyAudioVoice

Close all the audio voices.

Returns zero on success.

This routine closes all the active audio voices. Before calling this routine, all the allocated voices, including those allocated indirectly by calling the APlayModule routine, must be released.

See Also AOpenVoices, ACreateAudioVoice, ADestroyAudioVoice

Allocates a single audio voice.

Returns zero on success.

• lphVoice Address of a voice handle that will hold the allocated voice.

    VOID PlayWaveform(LPAUDIOWAVE lpWave)
    {
        HAC hVoice;
        BOOL stopped;

        /* allocate an audio voice or channel */                
        if (ACreateAudioVoice(&hVoice) == AUDIO_ERROR_NONE) {
            /* play the waveform through the voice */
            APlayVoice(hVoice, lpWave);
            ASetVoiceVolume(hVoice, 64);
            ASetVoicePanning(hVoice, 128);
            /* wait until the wave finishes or a key is pressed */
            while (!kbhit()) {
                AUpdateAudio();
                AGetVoiceStatus(hVoice, &stopped);
                if (stopped) break;
            }
            /* stop the voice (if already playing) */
            AStopVoice(hVoice);
            /* release the allocated voice */
            ADestroyAudioVoice(hVoice);
        }
    }

Releases an allocated audio voice.

Returns zero on success.

• hVoice Handle of the audio voice which will be released.

See Also ACreateAudioVoice

Start playing a waveform through a voice.

Returns zero on success.

• hVoice Handle of the audio voice that will play the waveform.
• lpWave Address of the waveform structure to be played.

See Also APrimeVoice, AStartVoice, AStopVoice

Prepares a waveform on an audio voice.

Returns zero on success.

• hVoice Handle of the audio voice that will be prepared.
• lpWave Address of the waveform structure that will be played.

    VOID PlayChord(HAC aVoices[3], LPAUDIOWAVE lpWave, LONG aFreqs[3])
    {
        UINT n;

        /* first prepare the voices to play the waveform */
        for (n = 0 ; n < 3 ; n++) {
            APrimeVoice(aVoices[n], lpWave);
            ASetVoiceFrequency(aVoices[n], aFreqs[n]);
            ASetVoiceVolume(aVoices[n], 64);
        }
        /* now start playing all them simultaneously */
        for (n = 0 ; n < 3 ; n++) {
            AStartVoice(aVoices[n]);
        }
    }

Start playing an audio voice.

Returns zero on success.

• hVoice Handle of the audio voice to be started.

See Also APrimeVoice, APlayVoice, AStopVoice

Stop playing an audio voice.

Returns zero on success.

• hVoice Handle of the audio voice to be stopped.

See Also APrimeVoice, APlayVoice, AStartVoice

Changes the voice playing position.

Returns zero on success.

• hVoice Handle of an audio voice.
• dwPosition Playing position measured in samples.

    VOID PlayEchoVoices(HAC aVoices[2], LPAUDIOWAVE lpWave, LONG dwDelay)
    {
        /* prepare two voices with the same waveform */
        APrimeVoice(aVoices[0], lpWave);
        APrimeVoice(aVoices[1], lpWave);
        ASetVoiceFrequency(aVoices[0], lpWave->nSampleRate);
        ASetVoiceFrequency(aVoices[1], lpWave->nSampleRate);

        /* set voice volumes (the "echo" voice has a lower volume) */
        ASetVoiceVolume(aVoices[0], 64);
        ASetVoiceVolume(aVoices[1], 32);

        /* set the delay (measured in samples) for the "echo" voice */
        ASetVoicePosition(aVoices[1], dwDelay);

        /* start both voices as simultenously as possible */                
        AStartVoice(aVoices[0]);
        AStartVoice(aVoices[1]);
    }

Changes the voice sampling frequency.

Returns zero on success.

• hVoice Handle of the audio voice to be changed.
• dwFrequency Sampling frequency measured in Hertz (cycles per second).

    VOID PlayVoiceStereo(HAC aVoices[2], LPAUDIOWAVE lpWave, LONG dwPitchShift)
    {
        /* prepare two voices to play a waveform */
        APrimeVoice(aVoices[0], lpWave);
        APrimeVoice(aVoices[1], lpWave);
        ASetVoiceVolume(aVoices[0], 64);
        ASetVoiceVolume(aVoices[1], 64);

        /* slightly change the pitch of one of the voices */
        ASetVoiceFrequency(aVoices[0], lpWave->nSampleRate);
        ASetVoiceFrequency(aVoices[1], lpWave->nSampleRate + dwPitchShift);

        /* set the pan position of the voices to left and right */
        ASetVoicePanning(aVoices[0], 0);
        ASetVoicePanning(aVoices[1], 255);

        /* start both voices simultaneously */
        AStartVoice(aVoices[0]);
        AStartVoice(aVoices[1]);
    }

Changes the voice volume level.

Returns zero on success.

• hVoice Handle of the audio voice to be changed.
• nVolume Linear volume level (0-64).

See Also AGetVoiceVolume

Changes the voice stereo pan position.

Returns zero on success.

• hVoice Handle of the audio voice to be changed.
• nPanning Stereo pan position value (0-255).

See Also AGetVoicePanning

Retrieves the playing position of a voice.

Returns zero on success.

• hVoice Handle of the audio voice to be queried.
• lpdwPosition Address of a long variable which will hold the playing position.

Retrieves the sampling frequency of a voice.

Returns zero on success.

• hVoice Handle of the audio voice to be queried.
• lpdwFrequency Address of a long variable that will hold the frequency.

See Also ASetVoiceFrequency

Retrieves the volume level of a voice.

Returns zero on success.

• hVoice Handle of the audio voice to be queried.
• lpnVolume Address of an unsigned integer variable that will hold the volume.

See Also ASetVoiceVolume

Retrieves the stereo pan position of a voice.

Returns zero on success.

• hVoice Handle of the audio voice which will be queried.
• lpnPanning Address of an unsigned integer variable that will hold the pan position.

See Also ASetVoicePanning

Retrieves the current status of a voice.

Returns zero on success.

• hVoice Handle of the audio voice that will be queried.
• lpbStatus Address of a boolean variable that will hold the voice status.

    VOID CheckVoice(HAC hVoice)
    {
        BOOL stopped;
        AGetVoiceStatus(hVoice, &stopped);
        if (stopped)
            printf("voice is stopped.\n");
    }

Retrieves the amount of available DRAM memory.

Returns amount of available DRAM memory (in bytes).

This routine retrieves the number of free bytes on the hardware audio device when applicable. It can be used to determine if there is enough space to hold a set of samples or waveforms simultaneously. It returns zero when the audio system is using system memory to hold the audio waveforms data.

    VOID ShowFreeMemory(VOID)
    {
        LONG dwMemFree;
        dwMemFree = AGetAudioDataAvail();
        printf("%ld bytes free.\n", dwMemFree);
    }

Creates a waveform object.

Returns zero on success.

• lpWave Address of the waveform structure.

    LPAUDIOWAVE CreateAudio8BitMono(WORD nSampleRate,
                                    LPBYTE lpData, DWORD dwLength)
    {
        LPAUDIOWAVE lpWave;
        /* first allocate structure to hold the waveform object */
        if ((lpWave = (LPAUDIOWAVE) malloc(sizeof(AUDIOWAVE))) != NULL) {
            /* create a 8-bit mono one-shot waveform object */
            lpWave->wFormat = AUDIO_FORMAT_8BITS | AUDIO_FORMAT_MONO;
            lpWave->nSampleRate = nSampleRate;
            lpWave->dwLength = dwLength;
            lpWave->dwLoopStart = lpWave->dwLoopEnd = 0;
            if (ACreateAudioData(lpWave) != AUDIO_ERROR_NONE) {
                free(lpWave);
                return NULL;
            }
            /* copy the data into the waveform object */
            memcpy(lpWave->lpData, lpData, dwLength);
            /* upload the data to the audio DRAM local memory */
            AWriteAudioData(lpWave, 0L, dwLength);
        }
        return lpWave;
    }

See Also ADestroyAudioData, AWriteAudioData

Releases a waveform object.

Returns zero on success.

• lpWave Address of the waveform object to be released.

See Also ACreateAudioData, AWriteAudioData

Upload waveform audio data to DRAM local memory.

Returns zero on success.

• lpWave Address of the waveform object.
• dwOffset Starting offset of the block of samples to be uploaded.
• nCount Number of bytes in the block of samples to be transferred.

    VOID StreamData8BitMono(FILE *stream, HAC hVoice, LPAUDIOWAVE lpWave)
    {
        BYTE aBuffer[1024];
        LPBYTE lpChunk;
        UINT nLength, nChunkSize;
        DWORD dwOffset, dwVoicePosition;

        /* check if the waveform is large enough for double buffering */
        if (2*sizeof(aBuffer) > lpWave->dwLength) {
            printf("the waveform is too small\n");
            return;
        }

        /* first clean up the audio waveform buffer */
        memset(lpWave->lpData, 0x80, lpWave->dwLength);
        AWriteAudioData(lpWave, 0L, lpWave->dwLength);

        /* start playing the waveform with forward looping enabled */
        lpWave->wFormat |= AUDIO_FORMAT_LOOP;
        lpWave->dwLoopStart = 0L;
        lpWave->dwLoopEnd = lpWave->dwLength;
        APlayVoice(hVoice, lpWave);
        ASetVoiceVolume(hVoice, 64);

        /* play the whole 8-bit mono stream of audio through the voice */
        dwOffset = 0;
        while ((nLength = fread(aBuffer, 1, nLength, stream)) != 0) {
            /* write the waveform audio data in chunks */
            lpChunk = aBuffer;
            while (nLength > 0) {
                /* determine size of chunk of data */
                nChunkSize = nLength;
                if (dwOffset + nChunkSize >= lpWave->dwLength)
                    nChunkSize = lpWave->dwLength - dwOffset;

                /* wait until we can write the chunk of data */
                for (;;) {
                    AUpdateAudio();
                    AGetVoicePosition(hVoice, &dwVoicePosition);
                    if (dwOffset + nChunkSize > lpWave->dwLength) {
                        if (dwVoicePosition < dwOffset &&
                            dwVoicePosition > dwOffset +
                            nChunkSize - lpWave->dwLength)
                            break;
                    }
                    else {
                        if (dwVoicePosition < dwOffset ||
                            dwVoicePosition > dwOffset + nChunkSize)
                            break;
                    }
                }

                /* upload the chunk of waveform data */
                memcpy(lpWave->lpData + dwOffset, lpChunk, nChunkSize);
                AWriteAudioData(lpWave, dwOffset, nChunkSize);
                if ((dwOffset += nChunkSize) >= lpWave->dwLength)
                    dwOffset = 0;
                lpChunk += nChunkSize;
                nLength -= nChunkSize;
            }
        }
    }

Load a RIFF/WAVE waveform file from disk.

Returns zero on success.

• lpszFileName Filename of the .WAV waveform file on disk.
• lplpWave Address of a pointer to a waveform structure.
• dwOffset Starting file offset of the waveform file.

    VOID PlayWaveFile(HAC hVoice, LPSTR lpszFileName)
    {
        LPAUDIOWAVE lpWave;
        BOOL stopped;

        if (ALoadWaveFile(lpszFileName, &lpWave, 0L) == AUDIO_ERROR_NONE) {
            APlayVoice(hVoice, lpWave);
            ASetVoiceVolume(hVoice, 64);
            while (!kbhit()) {
                AUpdateAudio();
                AGetVoiceStatus(hVoice, &stopped);
                if (stopped) break;
            }
            AFreeWaveFile(lpWave);
        }
    }

Releases a loaded RIFF/WAVE waveform file.

Returns zero on success.

• lpWave Address of the waveform structure to be released.

See Also ALoadWaveFile

Start playing a music module file.

Returns zero on success.

• lpModule Address of the module to be played.

See Also AStopModule

Stop playing a music module file.

Returns zero on success.

This routine stop and releases the audio system module player. After calling this routine, your application can select and start another module by calling the APlayModule routine.

See Also APlayModule

Stop playing a music module file.

Returns zero on success.

This routine stop the audio system module player. You can later continue playing the music module by calling the AResumeModule routine. Your application must use AStopModule when the currently select module file is going to be released, or if another music module must be played.

See Also APlayModule, AStopModule, AResumeModule

Restart playing a music module file.

Returns zero on success.

This routine is used to continue playing a music module that was previously paused by the APauseModule routine. Your application can use AGetModuleStatus to retrieve the current state of the audio system module player.

See Also APlayModule, AStopModule, APauseModule, AGetModuleStatus

Changes the music module global volume.

Returns zero on success.

• nVolume The new volume level for the music module (0-64).

See Also APlayModule, AStopModule, AGetModuleVolume

Changes the music module playing position.

Returns zero on success.

• nOrder The new order position.
• nRow The new pattern row number.

See Also APlayModule, AStopModule, AGetModulePosition

Retrieves the music module global volume.

Returns zero on success.

• lpnVolume Address of an unsigned integer that will hold the module volume.

See Also APlayModule, AStopModule, ASetModuleVolume

Retrieves the music module playing position.

Returns zero on success.

• lpnOrder Address of an unsigned integer that will hold the order position.
• lpnRow Address of an unsigned integer that will hold the pattern row number.

See Also APlayModule, AStopModule, ASetModulePosition

Retrieves the current state of the music module player.

Returns zero on success.

• lpbStatus Address of an unsigned integer that will hold the status.

See Also APlayModule, AStopModule, APauseModule, AResumeModule

Changes the music module synchronization callback routine.

Returns zero on success.

• lpfnAudioCallback Address of the synchronization callback routine.

    #define START_INTRO_SEQUENCE     0x01        /* Z01 */
    #define START_VECTOR_GRAPHICS    0x02        /* Z02 */

    VOID AIAPI SyncCallback(BYTE nParms, UINT nOrder, UINT nRow)
    {
        switch (nParms) {
        case START_INTRO_SEQUENCE:
            StartIntroSequence();
            break;
        case START_VECTOR_GRAPHICS:
            StartVectorGraphics();
            break;
        default:
            /* unknown Znn command found at position (nOrder, nRow) */
             break;
        }
    }

    VOID InitSyncCallback(VOID)
    {
        ASetModuleCallback(SyncCallback);
    }

    VOID DoneSyncCallback(VOID)
    {
        ASetModuleCallback(NULL);
    }

Load MOD/MTM/S3M/XM module files from disk.

Returns zero on success.

• lpszFileName File name of the music module file.
• lplpModule Address of a music module reference variable.
• dwOffset Starting file offset of the music module file.

See Also AFreeModuleFile

Releases a music module file from memory.

Returns zero on success.

• lpModule Address of a music module file in memory.

See Also ALoadModuleFile

Structures

    typedef struct {
        WORD    wProductId;
        CHAR    szProductName[30];
        DWORD   dwFormats;

    } AUDIOCAPS, *LPAUDIOCAPS;

Members:

• wProductId Specifies the hardware audio device product identifier. This member can hold any of the following values:

  Value	Description
  AUDIO_PRODUCT_NONE	Virtual silence device
  AUDIO_PRODUCT_SB	Sound Blaster 1.0 device
  AUDIO_PRODUCT_SB15	Sound Blaster 1.5 device
  AUDIO_PRODUCT_SB20	Sound Blaster 2.0 device
  AUDIO_PRODUCT_SBPRO	Sound Blaster Pro I or II device
  AUDIO_PRODUCT_SB16	Sound Blaster 16/ASP device
  AUDIO_PRODUCT_AWE32	Sound Blaster AWE32 device
  AUDIO_PRODUCT_WSS	Windows Sound System device
  AUDIO_PRODUCT_ESS	Ensoniq Soundscape (Elite) device
  AUDIO_PRODUCT_GUS	Gravis Ultrasound (GF1) device
  AUDIO_PRODUCT_GUSDB	Gravis Ultrasound Daughterboard device
  AUDIO_PRODUCT_GUSMAX	Gravis Ultrasound Max (CS4231) device
  AUDIO_PRODUCT_IWAVE	AMD Interwave-based device
  AUDIO_PRODUCT_PAS	Pro Audio Spectrum (Plus) device
  AUDIO_PRODUCT_PAS16	Pro Audio Spectrum 16 device
  AUDIO_PRODUCT_ARIA	Sierra Semiconductors Aria Sound device
  AUDIO_PRODUCT_WINDOWS	Microsoft Windows WAVE device
  AUDIO_PRODUCT_DSOUND	Microsoft Windows DirectSound device
  AUDIO_PRODUCT_LINUX	UNIX Sound System for Linux device
  AUDIO_PRODUCT_SPARC	SPARC workstation 8-bit ulaw or dbri device
  AUDIO_PRODUCT_SGI	Silicon Graphics Indigo Audio Ports device
  AUDIO_PRODUCT_OS2MMPM	OS/2 Multimedia Presentation Manager device
  AUDIO_PRODUCT_BEOSR3	BeOS Release 3 OLD MediaKit device
  AUDIO_PRODUCT_BEOS	BeOS Release 4 New MediaKit device

• szProductName Null terminated readable text string describing the hardware audio device product.
• dwFormats Specifies the playback formats supported by the hardware audio device product. This member can be built using any combination of the following values:

  Value	Description
  AUDIO_FORMAT_1M08	8-bit mono 11025 Hz
  AUDIO_FORMAT_1S08	8-bit stereo 11025 Hz
  AUDIO_FORMAT_1M16	16-bit mono 11025 Hz
  AUDIO_FORMAT_1S16	16-bit stereo 11025 Hz
  AUDIO_FORMAT_2M08	8-bit mono 22050 Hz
  AUDIO_FORMAT_2S08	8-bit stereo 22050 Hz
  AUDIO_FORMAT_2M16	16-bit mono 22050 Hz
  AUDIO_FORMAT_2S16	16-bit stereo 22050 Hz
  AUDIO_FORMAT_4M08	8-bit mono 44100 Hz
  AUDIO_FORMAT_4S08	8-bit stereo 44100 Hz
  AUDIO_FORMAT_4M16	16-bit mono 44100 Hz
  AUDIO_FORMAT_4S16	16-bit stereo 44100 Hz

    typedef struct {
        UINT    nDeviceId;
        WORD    wFormat;
        WORD    nSampleRate;
    } AUDIOINFO, *LPAUDIOINFO;

Members:

• nDeviceId Specifies the audio device we want to use for playback. The following virtual audio device identifiers are always defined:

  Value	Description
  AUDIO_DEVICE_NONE	Silence device driver
  AUDIO_DEVICE_MAPPER	Audio mapper driver

• wFormat Specifies the audio output format for playback. This member can be built using the following values:

  Value	Description
  AUDIO_FORMAT_8BITS	8 bits per sample
  AUDIO_FORMAT_16BITS	16 bits per sample
  AUDIO_FORMAT_MONO	Monophonic output
  AUDIO_FORMAT_STEREO	Stereo output
  AUDIO_FORMAT_FILTER	Enable filtering

• nSampleRate Specifies the sampling frequency used for playback. It's recommended to use the standard frequencies 11025, 22050 or 44100 Hz for better compatibility.

    typedef struct {
        LPBYTE  lpData;
        DWORD   dwHandle; 
        DWORD   dwLength;
        DWORD   dwLoopStart; 
        DWORD   dwLoopEnd;
        WORD    nSampleRate;
        WORD    wFormat;
    } AUDIOWAVE, *LPAUDIOWAVE;

• lpData Read Only. Address of the a buffer in system memory used to hold the waveform data.
• dwHandle Read Only. Handle used internally by the audio device drivers to reference the waveform data.
• dwLength Specifies the length in bytes of the waveform data.
• dwLoopStart Specifies the loop start point measured in bytes, only used for looped samples.
• dwLoopEnd Specifies the loop end point measured in bytes, only used for looped samples.
• nSampleEate The default sampling frequency of the waveform data.
• wFormat Specifies the format and the looping type of the waveform data. This member can be built using the following defines:

  Value	Description
  AUDIO_FORMAT_8BITS	8 bits per sample
  AUDIO_FORMAT_16BITS	16 bits per sample
  AUDIO_FORMAT_MONO	Monophonic output
  AUDIO_FORMAT_STEREO	Stereo output
  AUDIO_FORMAT_LOOP	Forward looping
  AUDIO_FORMAT_BIDILOOP	Bidirectional looping
  AUDIO_FORMAT_REVERSE	Reverse playing

    typedef struct {
        CHAR    szModuleName[32];
        WORD    wFlags;
        WORD    nOrders;

        WORD    nRestart;
        WORD    nTracks;

        WORD    nPatterns;
        WORD    nPatches;
        WORD    nTempo;
        WORD    nBPM;
        BYTE    aOrderTable[AUDIO_MAX_ORDERS];
        BYTE    aPanningTable[AUDIO_MAX_VOICES];
        LPAUDIOPATTERN aPatternTable;
        LPAUDIOPATCH aPatchTable;
    } AUDIOMODULE, *LPAUDIOMODULE;

Members:

• szModuleName Specifies the music module name as a null-terminated text string.
• wFlags Specifies some intial flags for the music module. This member can be built using the following values:

  Value	Description
  AUDIO_MODULE_LINEAR	The module uses linear frequencies
  AUDIO_MODULE_AMIGA	Use Amiga logarithmic frequencies
  AUDIO_MODULE_PANNING	Use pan position from instruments

• nOrders Number of entries in the sequence order table.
• nRestart Specifies the restart order position for looped modules.
• nTracks Number of tracks used in the music module.
• nPatterns Number of patterns used in the music module.
• nPatches Number of patches or instruments used in the music module.
• nTempo Initial speed or tempo value.
• nBPM Initial BPM or beats per minute value.
• aOrderTable Specifies the sequence order in which the patterns are played.
• aPanningTable Initial stereo pan position for all the music module tracks.
• aPatternTable Array of pattern structures.
• aPatchTable Array of patches structures.

Value	Description
AUDIO_ERROR_NONE	No error
AUDIO_ERROR_INVALHANDLE	Invalid voice or waveform handle
AUDIO_ERROR_INVALPARAM	Invalid parameter passed
AUDIO_ERROR_NOTSUPPORTED	Invalid audio system call
AUDIO_ERROR_BADDEVICEID	Invalid device identifier
AUDIO_ERROR_NODEVICE	No audio device found
AUDIO_ERROR_DEVICEBUSY	Audio device is busy
AUDIO_ERROR_BADFORMAT	Bad or unsupported audio playback format
AUDIO_ERROR_NOMEMORY	Not enough system memory
AUDIO_ERROR_NODRAMMEMORY	Not enough onboard DRAM memory
AUDIO_ERROR_FILENOTFOUND	Module or waveform file not found
AUDIO_ERROR_BADFILEFORMAT	Invalid module or waveform file format