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/**************************************************************************** * * AUDIO.C * * Simple routines to play audio using an AVIStream to get data. * * Uses global variables, so only one instance at a time. * (Usually, there's only one sound card, so this isn't so bad. * **************************************************************************/ /************************************************************************** * * THIS CODE AND INFORMATION IS PROVIDED "AS IS" WITHOUT WARRANTY OF ANY * KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE * IMPLIED WARRANTIES OF MERCHANTABILITY AND/OR FITNESS FOR A PARTICULAR * PURPOSE. * * Copyright (c) 1992, 1993 Microsoft Corporation. All Rights Reserved. * **************************************************************************/ #include "strbasic.h" #ifdef WIN95STUFF #include <windows.h> #include <windowsx.h> #include <mmsystem.h> #include <vfw.h> #include "udmIA64.h" // WIN64 Cleanup: Shuler // Until the Unified Data Model is complete and // available in MSVC++ 7.0, this header file is // required. #include "audio.h" #include "muldived.h" /*--------------------------------------------------------------+ | ****************** AUDIO PLAYING SUPPORT ******************** | +--------------------------------------------------------------*/ static HWAVEOUT shWaveOut = 0; /* Current MCI device ID */ static LONG slBegin; static LONG slWaveBegin; static LONG slCurrent; static LONG slEnd; static LONG slWaveEnd; static BOOL sfLooping; static BOOL sfPlaying = FALSE; static LONG sTotal; static LONG sPlayed; #define MAX_AUDIO_BUFFERS 16 #define MIN_AUDIO_BUFFERS 2 #define AUDIO_BUFFER_SIZE 16384 static WORD swBuffers; // total # buffers static WORD swBuffersOut; // buffers device has static WORD swNextBuffer; // next buffer to fill static LPWAVEHDR salpAudioBuf[MAX_AUDIO_BUFFERS]; static PAVISTREAM spavi; // stream we're playing static LONG slSampleSize; // size of an audio sample static LONG sdwBytesPerSec; static LONG sdwSamplesPerSec; //watje ranges at where the sound will be played //these are percentages based ver the current time segment static float startRange; static float endRange; /*---------------------------------------------------------------+ | aviaudioCloseDevice -- close the open audio device, if any. | +---------------------------------------------------------------*/ void NEAR aviaudioCloseDevice(void) { WORD w; if (shWaveOut) { while (swBuffers > 0) { --swBuffers; waveOutUnprepareHeader(shWaveOut, salpAudioBuf[swBuffers], sizeof(WAVEHDR)); GlobalFreePtr((LPSTR) salpAudioBuf[swBuffers]); } w = waveOutClose(shWaveOut); shWaveOut = NULL; } } /*--------------------------------------------------------------+ | aviaudioOpenDevice -- get ready to play waveform data. | +--------------------------------------------------------------*/ BOOL FAR aviaudioOpenDevice(HWND hwnd, PAVISTREAM pavi) { WORD w; LPVOID lpFormat; LONG cbFormat; AVISTREAMINFO strhdr; if (!pavi) // no wave data to play return FALSE; if (shWaveOut) // already something playing return TRUE; spavi = pavi; AVIStreamInfo(pavi, &strhdr, sizeof(strhdr)); slSampleSize = (LONG) strhdr.dwSampleSize; if (slSampleSize <= 0 || slSampleSize > AUDIO_BUFFER_SIZE) return FALSE; AVIStreamFormatSize(pavi, 0, &cbFormat); lpFormat = GlobalAllocPtr(GHND, cbFormat); if (!lpFormat) return FALSE; AVIStreamReadFormat(pavi, 0, lpFormat, &cbFormat); sdwSamplesPerSec = ((LPWAVEFORMAT) lpFormat)->nSamplesPerSec; sdwBytesPerSec = ((LPWAVEFORMAT) lpFormat)->nAvgBytesPerSec; w = waveOutOpen(&shWaveOut, (UINT)WAVE_MAPPER, lpFormat, (DWORD_PTR) hwnd, 0L, CALLBACK_WINDOW); // WIN64 Cleanup: Shuler // // Maybe we failed because someone is playing sound already. // Shut any sound off, and try once more before giving up. // if (w) { sndPlaySound(NULL, 0); w = waveOutOpen(&shWaveOut, (UINT)WAVE_MAPPER, lpFormat, (DWORD_PTR)hwnd, 0L, CALLBACK_WINDOW); // WIN64 Cleanup: Shuler } if (w != 0) { /* Show error message here? */ return FALSE; } for (swBuffers = 0; swBuffers < MAX_AUDIO_BUFFERS; swBuffers++) { if (!(salpAudioBuf[swBuffers] = (LPWAVEHDR)GlobalAllocPtr(GMEM_MOVEABLE | GMEM_SHARE, (DWORD)(sizeof(WAVEHDR) + AUDIO_BUFFER_SIZE)))) break; salpAudioBuf[swBuffers]->dwFlags = WHDR_DONE; salpAudioBuf[swBuffers]->lpData = (LPSTR) salpAudioBuf[swBuffers] + sizeof(WAVEHDR); salpAudioBuf[swBuffers]->dwBufferLength = AUDIO_BUFFER_SIZE; if (!waveOutPrepareHeader(shWaveOut, salpAudioBuf[swBuffers], sizeof(WAVEHDR))) continue; GlobalFreePtr((LPSTR) salpAudioBuf[swBuffers]); break; } if (swBuffers < MIN_AUDIO_BUFFERS) { aviaudioCloseDevice(); return FALSE; } swBuffersOut = 0; swNextBuffer = 0; sfPlaying = FALSE; return TRUE; } /*--------------------------------------------------------------+ | aviaudioTime - | | Return the time in milliseconds corresponding to the | | currently playing audio sample, or -1 if no audio is playing. | | | | WARNING: Some sound cards are pretty inaccurate! | +--------------------------------------------------------------*/ LONG FAR aviaudioTime(void) { MMTIME mmtime; if (!sfPlaying) return -1; mmtime.wType = TIME_SAMPLES; waveOutGetPosition(shWaveOut, &mmtime, sizeof(mmtime)); if (mmtime.wType == TIME_SAMPLES) return AVIStreamSampleToTime(spavi, slBegin) + muldiv32(mmtime.u.sample, 1000, sdwSamplesPerSec); else if (mmtime.wType == TIME_BYTES) return AVIStreamSampleToTime(spavi, slBegin) + muldiv32(mmtime.u.cb, 1000, sdwBytesPerSec); else return -1; } /*--------------------------------------------------------------+ | aviaudioiFillBuffers - | | Fill up any empty audio buffers and ship them out to the | | device. | +--------------------------------------------------------------*/ BOOL NEAR aviaudioiFillBuffers(void) { LONG lRead; WORD w; LONG lSamplesToPlay; int i, sti; float perStart,perEnd; float per; /* We're not playing, so do nothing. */ if (!sfPlaying) return TRUE; while (swBuffersOut < swBuffers) { if (sPlayed >= sTotal) { if (sfLooping) { /* Looping, so go to the beginning. */ slCurrent = slBegin; sPlayed = 0; } else { break; } } if (slCurrent >= slWaveEnd) { slCurrent = slWaveBegin; } /* Figure out how much data should go in this buffer */ lSamplesToPlay = slWaveEnd - slCurrent; if (lSamplesToPlay + sPlayed > sTotal) { lSamplesToPlay = sTotal - sPlayed; } if (lSamplesToPlay > AUDIO_BUFFER_SIZE / slSampleSize) lSamplesToPlay = AUDIO_BUFFER_SIZE / slSampleSize; AVIStreamRead(spavi, slCurrent, lSamplesToPlay, salpAudioBuf[swNextBuffer]->lpData, AUDIO_BUFFER_SIZE, &salpAudioBuf[swNextBuffer]->dwBufferLength, &lRead); if (lRead != lSamplesToPlay) { return FALSE; } perStart = (float)sPlayed/(float)sTotal; perEnd = (float)(sPlayed+lRead)/(float)sTotal; //watje if the current time segment of the cache falls outside the range then nuke if ( (perEnd < startRange) || (perStart > endRange) ) { for (i=0; i < lRead; i++) salpAudioBuf[swNextBuffer]->lpData[i] = 0; } //watje if the current end segment of the cache falls inside the range then nuke just part of the cache else if ( (perEnd >= startRange