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SF.net SVN: jazzplusplus:[906] trunk/jazz/src/mswin/WindowsAudioInterface. cpp
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From: <pst...@us...> - 2011-08-08 01:09:25
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Revision: 906
http://jazzplusplus.svn.sourceforge.net/jazzplusplus/?rev=906&view=rev
Author: pstieber
Date: 2011-08-08 01:09:19 +0000 (Mon, 08 Aug 2011)
Log Message:
-----------
1. Fixed incorrect pointer casts from DWORD to DWORD_PTR in calls to waveInOpen and waveOutOpen.
This fixed a crash in the Windows 64-bit build.
2. Made many cosmetic changes.
Modified Paths:
--------------
trunk/jazz/src/mswin/WindowsAudioInterface.cpp
Modified: trunk/jazz/src/mswin/WindowsAudioInterface.cpp
===================================================================
--- trunk/jazz/src/mswin/WindowsAudioInterface.cpp 2011-08-08 00:53:58 UTC (rev 905)
+++ trunk/jazz/src/mswin/WindowsAudioInterface.cpp 2011-08-08 01:09:19 UTC (rev 906)
@@ -3,7 +3,7 @@
//
// Copyright (C) 1994-2000 Andreas Voss and Per Sigmond, all rights reserved.
// Modifications Copyright (C) 2004 Patrick Earl
-// Modifications Copyright (C) 2008-2010 Peter J. Stieber
+// Modifications Copyright (C) 2008-2011 Peter J. Stieber
//
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
@@ -30,10 +30,9 @@
#include <wx/msgdlg.h>
-// not sure if mutex may cause a dead lock when used to
-// synchronize the wxTimer::Notify() interrupt and the
-// low level midi interrupt, as these are not mswin threads.
-// As far as I can see, the mutex is not needed anyway.
+// I'm not sure if a mutex may cause a dead lock when used to synchronize the
+// wxTimer::Notify() interrupt and the low-level midi interrupt, as these are
+// not mswin threads. As far as I can see, the mutex is not needed anyway.
// IMPORTANT: enabling critical sections freezes NT!
@@ -43,12 +42,12 @@
#define LeaveCriticalSection(a)
#define DeleteCriticalSection(a)
-//-----------------------------------------------------------------------------
-//-----------------------------------------------------------------------------
+//*****************************************************************************
+// Description:
+// Class to play a sample from the piano roll.
+//*****************************************************************************
class JZAudioListener : public wxTimer
{
- // play a sample from piano roll
-
public:
JZAudioListener(JZWindowsAudioPlayer* pPlayer, int key)
@@ -173,7 +172,7 @@
if (OpenDsp() == 0)
{
- // check output device capabilities
+ // Check output device capabilities.
WAVEOUTCAPS ocaps;
MMRESULT res = waveOutGetDevCaps((UINT)hout, &ocaps, sizeof(ocaps));
if (res != MMSYSERR_NOERROR)
@@ -265,7 +264,8 @@
int i;
MMRESULT res;
- mErrorCode = NoError; // everything ok for now.
+ // Everything is OK for now.
+ mErrorCode = NoError;
if (!mAudioEnabled)
{
@@ -285,7 +285,7 @@
fmt.wBitsPerSample = 16;
fmt.cbSize = 0;
- blocks_played = 0;
+ blocks_played = 0;
play_buffers_needed = 0;
record_buffers_needed = 0;
@@ -297,8 +297,8 @@
&hout,
WAVE_MAPPER,
&fmt,
- (DWORD)audioInterrupt,
- (DWORD)this,
+ (DWORD_PTR)audioInterrupt,
+ (DWORD_PTR)this,
CALLBACK_FUNCTION);
if (res != MMSYSERR_NOERROR)
@@ -307,7 +307,7 @@
return 1;
}
- // prepare headers
+ // Prepare the headers.
for (i = 0; i < BUFCOUNT; i++)
{
JZAudioBuffer* buf = mSamples.GetBuffer(i);
@@ -315,9 +315,11 @@
memset(hdr, 0, sizeof(WAVEHDR));
buf->hdr = hdr;
- hdr->lpData = (char *)buf->Data();
- hdr->dwBufferLength = BUFBYTES; // length, in bytes, of the buffer
+ hdr->lpData = (char *)buf->Data();
+ // Length, in bytes, of the buffer.
+ hdr->dwBufferLength = BUFBYTES;
+
res = waveOutPrepareHeader(hout, hdr, sizeof(WAVEHDR));
if (res != MMSYSERR_NOERROR)
{
@@ -333,29 +335,41 @@
hinp = 0;
recbuffers.Clear();
- res = waveInOpen(&hinp, WAVE_MAPPER, &fmt, (DWORD)audioInterrupt, (DWORD)this, CALLBACK_FUNCTION);
+ res = waveInOpen(
+ &hinp,
+ WAVE_MAPPER,
+ &fmt,
+ (DWORD_PTR)audioInterrupt,
+ (DWORD_PTR)this,
+ CALLBACK_FUNCTION);
if (res != MMSYSERR_NOERROR)
{
mErrorCode = ErrInpOpen;
return 1;
}
- // prepare headers and add them to recording device
+ // Prepare headers and add them to recording device.
for (i = 0; i < BUFCOUNT; i++)
{
WAVEHDR *hdr = new WAVEHDR;
memset(hdr, 0, sizeof(WAVEHDR));
JZAudioBuffer* buf = recbuffers.RequestBuffer();
- buf->hdr = hdr;
+ buf->hdr = hdr;
- hdr->lpData = (LPSTR)buf->data;
- hdr->dwBufferLength = BUFBYTES; // length, in bytes, of the buffer
- hdr->dwFlags = 0; // see below
+ hdr->lpData = (LPSTR)buf->data;
+ // length, in bytes, of the buffer
+ hdr->dwBufferLength = BUFBYTES;
+
+ // See below.
+ hdr->dwFlags = 0;
+
res = waveInPrepareHeader(hinp, hdr, sizeof(WAVEHDR));
if (res != MMSYSERR_NOERROR)
+ {
return 1;
+ }
res = waveInAddBuffer(hinp, hdr, sizeof(WAVEHDR));
if (res != MMSYSERR_NOERROR)
@@ -365,7 +379,6 @@
}
}
hinp_open = 1;
-
}
return 0;
@@ -527,7 +540,9 @@
if (hout_open)
{
mSamples.FillBuffers(mOutClock);
- if (play_buffers_needed > 0) // dont trigger start play by accident
+
+ // Don't trigger a start play by accident.
+ if (play_buffers_needed > 0)
{
WriteBuffers();
}
@@ -549,18 +564,18 @@
// low pass filter for time-correction (not really necessary)
const long low = 50;
mpState->time_correction =
- (low * mpState->time_correction + (100 - low) * (audio_now - time_now) ) / 100L;
+ (low * mpState->time_correction +
+ (100 - low) * (audio_now - time_now) ) / 100L;
}
}
}
-
if (hinp_open)
{
while (record_buffers_needed > 0)
{
- // add a new record buffer
+ // Add a new record buffer.
WAVEHDR *hdr = new WAVEHDR;
memset(hdr, 0, sizeof(WAVEHDR));
@@ -568,7 +583,10 @@
buf->hdr = hdr;
hdr->lpData = (LPSTR)buf->data;
- hdr->dwBufferLength = BUFBYTES; // length, in bytes, of the buffer
+
+ // Length, in bytes, of the buffer.
+ hdr->dwBufferLength = BUFBYTES;
+
hdr->dwFlags = 0;
if (waveInPrepareHeader(hinp, hdr, sizeof(WAVEHDR)) == MMSYSERR_NOERROR)
@@ -598,7 +616,9 @@
// Low pass filter for time-correction (not really necessary).
const long low = 50;
- mpState->time_correction = (low * mpState->time_correction + (100 - low) * (audio_now - time_now) ) / 100L;
+ mpState->time_correction =
+ (low * mpState->time_correction +
+ (100 - low) * (audio_now - time_now)) / 100L;
}
}
}
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