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List: openjdk-sound-dev
Subject: RE: professional (24-bit) sampled audio support in the Windows native implementation of libjsound
From: <magare31 () gmail ! com>
Date: 2022-10-11 1:36:08
Message-ID: 036c01d8dd11$d637e700$82a7b500$ () gmail ! com
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I understand that my bug submission was not clear. Here are most of the things you \
requested. Let me know if you need more. Thank you again.
* Test: See the attached Java code with two simple tests. The code is very short and \
I think I put sufficient comments to make these clear. But let me know if you need \
more. (See below on regression testing).
* On Windows, with the current jdk, both tests fail. I am testing on an audio device \
that supports 24-bit audio resolution. The code: 1) shows that the audio device does \
NOT supports 24-bit; 2) cannot get a 24-bit line to the audio device.
* With the two minor changes below, the code will succeed in both tests (see more on \
testing below).
* Fix: See the attached C++ code with the fix (the diff is below). In the JDK \
source, this file is in src/java.desktop/windows/native/libjsound. The fix is in the \
native Windows code only.
* Diff: There are changes to two lines only in the Windows native C++ code.
* Line 282 changes from
static INT32 bitsArray[] = { 8, 16};
to
static INT32 bitsArray[] = { 8, 16, 24};
* Line 643 changes from
if (channels <= 2 && bits <= 16) {
to
if (channels <= 2 && bits <= 24) {
* Explanation: The first change allows the native implementation to recognize 24 bit \
as a format that an audio device may support. The second change allows the native \
code to treat the 24-bit signed integer representation of sampled audio data as pulse \
code modulation (PCM) audio data. This is the correct way to treat such data.
* JDK 19 without the fix, treats 24-bit audio data as one that needs the extensible \
wave format. This is not correct (see lines 643 to 653 of the code). In fact, the \
extensible wave format probably does not belong at all in a communication with an \
audio device, but I don't want to venture that far in changing the code.
* Regression testing is limited: First, I admit that I do not have a full range of \
tests. Second, I am testing as part of a larger piece of sound production software \
that I cannot share. Unfortunately, at this point, I can only confirm the following:
* On Windows, a full build of the jdk, with these changes, from a recent clone of the \
repository (from Sep 27) works as intended. It permits 24-bit playback and recording \
at various sampling rates.
* Importantly, on Windows, the native query in the JDK of what audio formats are \
supported by the audio hardware was never fully implemented in JDK 19 or previous. \
Whatever bit resolution is included in line 282 of the C++ code will be shown as \
"supported" (the first test). This is not ideal, but it is the same as the current \
implementation.
* JDK 19 on Mac OS (Mojave) supports 24 bit playback (I have not tested recording \
yet).
* JDK 19 on Ubuntu 20 supports 24 bit playback and 24 bit recording.
From: Aleksei Ivanov <alexey.ivanov@oracle.com>
Sent: Monday, October 10, 2022 5:03 PM
To: magare31@gmail.com <mailto:magare31@gmail.com> ; client-libs-dev@openjdk.org \
<mailto:client-libs-dev@openjdk.org>
Subject: Re: professional (24-bit) sampled audio support in the Windows native \
implementation of libjsound
Hi,
JDK-8294904 [1] was resolved as Incomplete because there's no sample code which \
demonstrates the problem. A comment was added:
Mail to submitter:
=============
Please share standalone test case/ reproducible step/ scenario to analyze the issue \
better.
I should have received an email with request for clarification.
I looked through the description but it's still unclear to me how to reproduce the \
problem and how to incorporate the fix you're proposing and what the fix is. If you \
could provide the test case and better yet the fix along with a regression test, it \
would be greatly appreciated.
The fix in the form of the diff would also help.
--
Regards,
Alexey
[1] https://bugs.openjdk.org/browse/JDK-8294904
On 09/10/2022 15:57, magare31@gmail.com <mailto:magare31@gmail.com> wrote:
Since I am new to this, and apologies for the broad email, could someone explain the \
following for the corresponding bug (JDK-8294904)?
1. This bug is listed as one for a generic OS, whereas I specified Windows (I can \
confirm that it is Windows only). 2. This bug is listed as "resolved" with an \
"incomplete resolution". I did provide the full solution, so I am not sure what this \
means. (Also, Oracle asked for a standalone test, which I also provided over email)
And thanks.
From: Kevin Rushforth <mailto:kevin.rushforth@oracle.com> \
<kevin.rushforth@oracle.com>
Sent: Friday, September 30, 2022 8:29 AM
To: magare31@gmail.com <mailto:magare31@gmail.com> ; client-libs-dev@openjdk.org \
<mailto:client-libs-dev@openjdk.org> ; core-libs-dev@openjdk.org \
<mailto:core-libs-dev@openjdk.org>
Subject: Re: professional (24-bit) sampled audio support in the Windows native \
implementation of libjsound
Java Sound is in the client-libs area. You can file the bug yourself at \
https://bugreport.java.com/ if you like, or ask the sponsor of your bug (when one \
steps forward) to do it.
If you want to contribute your fix, please see the contributing a patch section [1] \
in the JDK Developers Guide for the next steps.
-- Kevin
[1] https://openjdk.org/guide/#i-have-a-patch-what-do-i-do
On 9/30/2022 4:33 AM, magare31@gmail.com <mailto:magare31@gmail.com> wrote:
Would anyone want to sponsor the following simple bug fix?
- The purpose is to enable playback and recording of 24-bit sampled audio on Windows. \
This is already supported on other systems.
- There is no associated bug in the bug database. I noted it as a "bug" as the code \
misunderstands the WAVE RIFF format standards.
- There will be two very small changes to one Windows native cpp file under libjsound
- I have tested the changes on a jdk build of the latest code.
Also, please advise which of these two groups this belongs to: client libs or core \
libs?
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style='word-wrap:break-word'><div class=WordSection1><p class=MsoNormal>I understand \
that my bug submission was not clear. Here are most of the things you requested. \
Let me know if you need more. Thank you again.<o:p></o:p></p><p \
class=MsoNormal><o:p> </o:p></p><ul style='margin-top:0in' type=disc><li \
class=MsoListParagraph style='margin-left:0in;mso-list:l2 level1 lfo4'>Test: See the \
attached Java code with two simple tests. The code is very short and I think I put \
sufficient comments to make these clear. But let me know if you need more. (See \
below on regression testing).<o:p></o:p></li><ul style='margin-top:0in' \
type=circle><li class=MsoListParagraph style='margin-left:0in;mso-list:l2 level2 \
lfo4'>On Windows, with the current jdk, both tests fail. I am testing on an audio \
device that supports 24-bit audio resolution. The code: 1) shows that the audio \
device does NOT supports 24-bit; 2) cannot get a 24-bit line to the audio \
device.<o:p></o:p></li><li class=MsoListParagraph style='margin-left:0in;mso-list:l2 \
level2 lfo4'>With the two minor changes below, the code will succeed in both tests \
(see more on testing below).<o:p></o:p></li></ul><li class=MsoListParagraph \
style='margin-left:0in;mso-list:l2 level1 lfo4'>Fix: See the attached C++ code with \
the fix (the diff is below). In the JDK source, this file is in \
src/java.desktop/windows/native/libjsound. The fix is in the native Windows code \
only.<o:p></o:p></li><li class=MsoListParagraph style='margin-left:0in;mso-list:l2 \
level1 lfo4'>Diff: There are changes to two lines only in the Windows native C++ \
code.<o:p></o:p></li><ul style='margin-top:0in' type=circle><li \
class=MsoListParagraph style='margin-left:0in;mso-list:l2 level2 lfo4'>Line 282 \
changes from <o:p></o:p></li></ul></ul><p class=MsoListParagraph \
style='margin-left:1.0in'><o:p> </o:p></p><p class=MsoNormal \
style='margin-left:.5in;text-indent:.5in'><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:blue'>static</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:black'> </span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:#2B91AF'>INT32</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:black'> bitsArray[] = { 8, \
16};</span><o:p></o:p></p><p class=MsoNormal \
style='margin-left:.5in;text-indent:.5in'><o:p> </o:p></p><p class=MsoNormal \
style='margin-left:.5in;text-indent:.5in'>to<o:p></o:p></p><p class=MsoNormal \
style='margin-left:.5in;text-indent:.5in'><o:p> </o:p></p><p \
class=MsoListParagraph style='margin-left:1.0in'><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:blue'>static</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:black'> </span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:#2B91AF'>INT32</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:black'> bitsArray[] = { 8, \
16, 24};<o:p></o:p></span></p><p class=MsoListParagraph \
style='margin-left:1.0in'><o:p> </o:p></p><ul style='margin-top:0in' \
type=disc><ul style='margin-top:0in' type=circle><li class=MsoListParagraph \
style='margin-left:0in;mso-list:l2 level2 lfo4'>Line 643 changes \
from<o:p></o:p></li></ul></ul><p class=MsoListParagraph \
style='text-autospace:none'><span style='font-size:9.5pt;font-family:"Cascadia \
Mono";color:black'> <o:p></o:p></span></p><p class=MsoNormal \
style='margin-left:.5in;text-indent:.5in;text-autospace:none'><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:blue'>if</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:black'> (</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:gray'>channels</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:black'> <= 2 && \
</span><span style='font-size:9.5pt;font-family:"Cascadia \
Mono";color:gray'>bits</span><span style='font-size:9.5pt;font-family:"Cascadia \
Mono";color:black'> <= 16) {<o:p></o:p></span></p><p class=MsoListParagraph \
style='margin-left:1.0in'><o:p> </o:p></p><p class=MsoListParagraph \
style='margin-left:1.0in'>to<o:p></o:p></p><p class=MsoNormal \
style='text-autospace:none'><span style='font-size:9.5pt;font-family:"Cascadia \
Mono";color:blue'><o:p> </o:p></span></p><p class=MsoNormal \
style='margin-left:.5in;text-indent:.5in;text-autospace:none'><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:blue'>if</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:black'> (</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:gray'>channels</span><span \
style='font-size:9.5pt;font-family:"Cascadia Mono";color:black'> <= 2 && \
</span><span style='font-size:9.5pt;font-family:"Cascadia \
Mono";color:gray'>bits</span><span style='font-size:9.5pt;font-family:"Cascadia \
Mono";color:black'> <= 24) {<o:p></o:p></span></p><p class=MsoListParagraph \
style='margin-left:1.0in'><o:p> </o:p></p><ul style='margin-top:0in' \
type=disc><li class=MsoListParagraph style='margin-left:0in;mso-list:l2 level1 \
lfo4'>Explanation: The first change allows the native implementation to recognize 24 \
bit as a format that an audio device may support. The second change allows the \
native code to treat the 24-bit signed integer representation of sampled audio data \
as pulse code modulation (PCM) audio data. This is the correct way to treat such \
data.<o:p></o:p></li><ul style='margin-top:0in' type=circle><li \
class=MsoListParagraph style='margin-left:0in;mso-list:l2 level2 lfo4'>JDK 19 without \
the fix, treats 24-bit audio data as one that needs the extensible wave format. \
This is not correct (see lines 643 to 653 of the code). In fact, the extensible \
wave format probably does not belong at all in a communication with an audio device, \
but I don't want to venture that far in changing the code.<o:p></o:p></li></ul><li \
class=MsoListParagraph style='margin-left:0in;mso-list:l2 level1 lfo4'>Regression \
testing is limited: First, I admit that I do not have a full range of tests. \
Second, I am testing as part of a larger piece of sound production software that I \
cannot share. Unfortunately, at this point, I can only confirm the \
following:<o:p></o:p></li><ul style='margin-top:0in' type=circle><li \
class=MsoListParagraph style='margin-left:0in;mso-list:l2 level2 lfo4'>On Windows, a \
full build of the jdk, with these changes, from a recent clone of the repository \
(from Sep 27) works as intended. It permits 24-bit playback and recording at \
various sampling rates.<o:p></o:p></li><li class=MsoListParagraph \
style='margin-left:0in;mso-list:l2 level2 lfo4'>Importantly, on Windows, the native \
query in the JDK of what audio formats are supported by the audio hardware was never \
fully implemented in JDK 19 or previous. Whatever bit resolution is included in \
line 282 of the C++ code will be shown as "supported" (the first test). \
This is not ideal, but it is the same as the current implementation. \
<o:p></o:p></li><li class=MsoListParagraph style='margin-left:0in;mso-list:l2 level2 \
lfo4'>JDK 19 on Mac OS (Mojave) supports 24 bit playback (I have not tested recording \
yet).<o:p></o:p></li><li class=MsoListParagraph style='margin-left:0in;mso-list:l2 \
level2 lfo4'>JDK 19 on Ubuntu 20 supports 24 bit playback and 24 bit \
recording.<o:p></o:p></li></ul></ul><p class=MsoNormal><o:p> </o:p></p><p \
class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p \
class=MsoNormal><o:p> </o:p></p><div><div style='border:none;border-top:solid \
#E1E1E1 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b>From:</b> Aleksei \
Ivanov <alexey.ivanov@oracle.com> <br><b>Sent:</b> Monday, October 10, 2022 \
5:03 PM<br><b>To:</b> <a href="mailto:magare31@gmail.com">magare31@gmail.com</a>; <a \
href="mailto:client-libs-dev@openjdk.org">client-libs-dev@openjdk.org</a><br><b>Subject:</b> \
Re: professional (24-bit) sampled audio support in the Windows native implementation \
of libjsound<o:p></o:p></p></div></div><p class=MsoNormal><o:p> </o:p></p><p \
class=MsoNormal style='margin-bottom:12.0pt'>Hi,<br><br>JDK-8294904 [1] was resolved \
as Incomplete because there's no sample code which demonstrates the problem. A \
comment was added:<br><br>Mail to submitter: <br>============= <br>Please share \
standalone test case/ reproducible step/ scenario to analyze the issue \
better.<br><br>I should have received an email with request for \
clarification.<br><br>I looked through the description but it's still unclear to me \
how to reproduce the problem and how to incorporate the fix you're proposing and what \
the fix is. If you could provide the test case and better yet the fix along with a \
regression test, it would be greatly appreciated.<br><br>The fix in the form of the \
diff would also help.<br><br>-- <br>Regards,<br>Alexey<br><br>[1] <a \
href="https://bugs.openjdk.org/browse/JDK-8294904">https://bugs.openjdk.org/browse/JDK-8294904</a><o:p></o:p></p><div><p \
class=MsoNormal>On 09/10/2022 15:57, <a \
href="mailto:magare31@gmail.com">magare31@gmail.com</a> \
wrote:<o:p></o:p></p></div><blockquote \
style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>Since I am new to \
this, and apologies for the broad email, could someone explain the following for the \
corresponding bug (JDK-8294904)?<o:p></o:p></p><p \
class=MsoNormal> <o:p></o:p></p><ol style='margin-top:0in' start=1 type=1><li \
class=MsoListParagraph style='margin-left:0in;mso-list:l0 level1 lfo3'>This bug is \
listed as one for a generic OS, whereas I specified Windows (I can confirm that it is \
Windows only).<o:p></o:p></li><li class=MsoListParagraph \
style='margin-left:0in;mso-list:l0 level1 lfo3'>This bug is listed as \
"resolved" with an "incomplete resolution". I did provide \
the full solution, so I am not sure what this means. (Also, Oracle asked for a \
standalone test, which I also provided over email)<o:p></o:p></li></ol><p \
class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>And thanks.<o:p></o:p></p><p \
class=MsoNormal> <o:p></o:p></p><div><div style='border:none;border-top:solid \
#E1E1E1 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b>From:</b> Kevin \
Rushforth <a href="mailto:kevin.rushforth@oracle.com"><kevin.rushforth@oracle.com></a> \
<br><b>Sent:</b> Friday, September 30, 2022 8:29 AM<br><b>To:</b> <a \
href="mailto:magare31@gmail.com">magare31@gmail.com</a>; <a \
href="mailto:client-libs-dev@openjdk.org">client-libs-dev@openjdk.org</a>; <a \
href="mailto:core-libs-dev@openjdk.org">core-libs-dev@openjdk.org</a><br><b>Subject:</b> \
Re: professional (24-bit) sampled audio support in the Windows native implementation \
of libjsound<o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><p \
class=MsoNormal style='margin-bottom:12.0pt'>Java Sound is in the client-libs area. \
You can file the bug yourself at <a \
href="https://bugreport.java.com/">https://bugreport.java.com/</a> if you like, or \
ask the sponsor of your bug (when one steps forward) to do it.<br><br>If you want to \
contribute your fix, please see the contributing a patch section [1] in the JDK \
Developers Guide for the next steps.<br><br>-- Kevin<br><br>[1] <a \
href="https://openjdk.org/guide/#i-have-a-patch-what-do-i-do">https://openjdk.org/guide/#i-have-a-patch-what-do-i-do</a><br><br><br><o:p></o:p></p><div><p \
class=MsoNormal>On 9/30/2022 4:33 AM, <a \
href="mailto:magare31@gmail.com">magare31@gmail.com</a> \
wrote:<o:p></o:p></p></div><blockquote \
style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>Would anyone want to \
sponsor the following simple bug fix?<o:p></o:p></p><p \
class=MsoNormal> <o:p></o:p></p><p class=MsoNormal style='text-indent:.5in'>- \
The purpose is to enable playback and recording of 24-bit sampled audio on \
Windows. This is already supported on other systems.<o:p></o:p></p><p \
class=MsoNormal style='text-indent:.5in'>- There is no associated bug in the bug \
database. I noted it as a "bug" as the code misunderstands the WAVE \
RIFF format standards.<o:p></o:p></p><p class=MsoNormal style='text-indent:.5in'>- \
There will be two very small changes to one Windows native cpp file under \
libjsound<o:p></o:p></p><p class=MsoNormal style='text-indent:.5in'>- I have tested \
the changes on a jdk build of the latest code.<o:p></o:p></p><p \
class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>Also, please advise which of \
these two groups this belongs to: client libs or core libs?<o:p></o:p></p><p \
class=MsoNormal> <o:p></o:p></p><p \
class=MsoNormal> <o:p></o:p></p></blockquote><p \
class=MsoNormal> <o:p></o:p></p></blockquote><p \
class=MsoNormal><o:p> </o:p></p></div></body></html>
["Main.java" (application/octet-stream)]
["PLATFORM_API_WinOS_DirectSound.cpp" (text/plain)]
/*
* Copyright (c) 2003, 2020, Oracle and/or its affiliates. All rights reserved.
* DO NOT ALTER OR REMOVE COPYRIGHT NOTICES OR THIS FILE HEADER.
*
* This code is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License version 2 only, as
* published by the Free Software Foundation. Oracle designates this
* particular file as subject to the "Classpath" exception as provided
* by Oracle in the LICENSE file that accompanied this code.
*
* This code is distributed in the hope that it will be useful, but WITHOUT
* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
* FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
* version 2 for more details (a copy is included in the LICENSE file that
* accompanied this code).
*
* You should have received a copy of the GNU General Public License version
* 2 along with this work; if not, write to the Free Software Foundation,
* Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA.
*
* Please contact Oracle, 500 Oracle Parkway, Redwood Shores, CA 94065 USA
* or visit www.oracle.com if you need additional information or have any
* questions.
*/
#define USE_ERROR
#define USE_TRACE
/* define this for the silencing/servicing code. Requires USE_TRACE */
//#define USE_DEBUG_SILENCING
#ifndef WIN32_EXTRA_LEAN
#define WIN32_EXTRA_LEAN
#endif
#ifndef WIN32_LEAN_AND_MEAN
#define WIN32_LEAN_AND_MEAN
#endif
#include <windows.h>
#include <mmsystem.h>
#include <string.h>
/* include DirectSound headers */
#include <dsound.h>
/* include Java Sound specific headers as C code */
#ifdef __cplusplus
extern "C" {
#endif
#include "DirectAudio.h"
#ifdef __cplusplus
}
#endif
/* include to prevent charset problem */
#include "PLATFORM_API_WinOS_Charset_Util.h"
#ifdef USE_DEBUG_SILENCING
#define DEBUG_SILENCING0(p) TRACE0(p)
#define DEBUG_SILENCING1(p1,p2) TRACE1(p1,p2)
#define DEBUG_SILENCING2(p1,p2,p3) TRACE2(p1,p2,p3)
#else
#define DEBUG_SILENCING0(p)
#define DEBUG_SILENCING1(p1,p2)
#define DEBUG_SILENCING2(p1,p2,p3)
#endif
//cak #if USE_DAUDIO == TRUE
/* 3 seconds to wait before device list is re-read */
#define WAIT_BETWEEN_CACHE_REFRESH_MILLIS 3000
/* maximum number of supported devices, playback+capture */
#define MAX_DS_DEVICES 60
typedef struct {
INT32 mixerIndex;
BOOL isSource;
/* either LPDIRECTSOUND or LPDIRECTSOUNDCAPTURE */
void* dev;
/* how many instances use the dev */
INT32 refCount;
GUID guid;
} DS_AudioDeviceCache;
static DS_AudioDeviceCache g_audioDeviceCache[MAX_DS_DEVICES];
static INT32 g_cacheCount = 0;
static UINT64 g_lastCacheRefreshTime = 0;
static INT32 g_mixerCount = 0;
BOOL DS_lockCache() {
/* dummy implementation for now, Java does locking */
return TRUE;
}
void DS_unlockCache() {
/* dummy implementation for now */
}
static GUID CLSID_DAUDIO_Zero = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0};
BOOL isEqualGUID(LPGUID lpGuid1, LPGUID lpGuid2) {
if (lpGuid1 == NULL || lpGuid2 == NULL) {
if (lpGuid1 == lpGuid2) {
return TRUE;
}
if (lpGuid1 == NULL) {
lpGuid1 = (LPGUID) (&CLSID_DAUDIO_Zero);
} else {
lpGuid2 = (LPGUID) (&CLSID_DAUDIO_Zero);
}
}
return memcmp(lpGuid1, lpGuid2, sizeof(GUID)) == 0;
}
INT32 findCacheItemByGUID(LPGUID lpGuid, BOOL isSource) {
int i;
for (i = 0; i < g_cacheCount; i++) {
if (isSource == g_audioDeviceCache[i].isSource
&& isEqualGUID(lpGuid, &(g_audioDeviceCache[i].guid))) {
return i;
}
}
return -1;
}
INT32 findCacheItemByMixerIndex(INT32 mixerIndex) {
int i;
for (i = 0; i < g_cacheCount; i++) {
if (g_audioDeviceCache[i].mixerIndex == mixerIndex) {
return i;
}
}
return -1;
}
typedef struct {
INT32 currMixerIndex;
BOOL isSource;
} DS_RefreshCacheStruct;
BOOL CALLBACK DS_RefreshCacheEnum(LPGUID lpGuid,
LPCSTR lpstrDescription,
LPCSTR lpstrModule,
DS_RefreshCacheStruct* rs) {
INT32 cacheIndex = findCacheItemByGUID(lpGuid, rs->isSource);
/*TRACE3("Enumerating %d: %s (%s)\n", cacheIndex, lpstrDescription, \
lpstrModule);*/ if (cacheIndex == -1) {
/* add this device */
if (g_cacheCount < MAX_DS_DEVICES-1) {
g_audioDeviceCache[g_cacheCount].mixerIndex = rs->currMixerIndex;
g_audioDeviceCache[g_cacheCount].isSource = rs->isSource;
g_audioDeviceCache[g_cacheCount].dev = NULL;
g_audioDeviceCache[g_cacheCount].refCount = 0;
if (lpGuid == NULL) {
memset(&(g_audioDeviceCache[g_cacheCount].guid), 0, sizeof(GUID));
} else {
memcpy(&(g_audioDeviceCache[g_cacheCount].guid), lpGuid, \
sizeof(GUID)); }
g_cacheCount++;
rs->currMixerIndex++;
} else {
/* failure case: more than MAX_DS_DEVICES available... */
}
} else {
/* device already exists in cache... update mixer number */
g_audioDeviceCache[cacheIndex].mixerIndex = rs->currMixerIndex;
rs->currMixerIndex++;
}
/* continue enumeration */
return TRUE;
}
///// implemented functions of DirectAudio.h
INT32 DAUDIO_GetDirectAudioDeviceCount() {
DS_RefreshCacheStruct rs;
INT32 oldCount;
INT32 cacheIndex;
if (!DS_lockCache()) {
return 0;
}
if (g_lastCacheRefreshTime == 0
|| (UINT64) timeGetTime() > (UINT64) (g_lastCacheRefreshTime + \
WAIT_BETWEEN_CACHE_REFRESH_MILLIS)) { /* first, initialize any old cache items */
for (cacheIndex = 0; cacheIndex < g_cacheCount; cacheIndex++) {
g_audioDeviceCache[cacheIndex].mixerIndex = -1;
}
/* enumerate all devices and either add them to the device cache,
* or refresh the mixer number
*/
rs.currMixerIndex = 0;
rs.isSource = TRUE;
DirectSoundEnumerate((LPDSENUMCALLBACK) DS_RefreshCacheEnum, &rs);
/* if we only got the Primary Sound Driver (GUID=NULL),
* then there aren't any playback devices installed */
if (rs.currMixerIndex == 1) {
cacheIndex = findCacheItemByGUID(NULL, TRUE);
if (cacheIndex == 0) {
rs.currMixerIndex = 0;
g_audioDeviceCache[0].mixerIndex = -1;
TRACE0("Removing stale Primary Sound Driver from list.\n");
}
}
oldCount = rs.currMixerIndex;
rs.isSource = FALSE;
DirectSoundCaptureEnumerate((LPDSENUMCALLBACK) DS_RefreshCacheEnum, &rs);
/* if we only got the Primary Sound Capture Driver (GUID=NULL),
* then there aren't any capture devices installed */
if ((rs.currMixerIndex - oldCount) == 1) {
cacheIndex = findCacheItemByGUID(NULL, FALSE);
if (cacheIndex != -1) {
rs.currMixerIndex = oldCount;
g_audioDeviceCache[cacheIndex].mixerIndex = -1;
TRACE0("Removing stale Primary Sound Capture Driver from list.\n");
}
}
g_mixerCount = rs.currMixerIndex;
g_lastCacheRefreshTime = (UINT64) timeGetTime();
}
DS_unlockCache();
/*TRACE1("DirectSound: %d installed devices\n", g_mixerCount);*/
return g_mixerCount;
}
BOOL CALLBACK DS_GetDescEnum(LPGUID lpGuid,
LPCWSTR lpstrDescription,
LPCWSTR lpstrModule,
DirectAudioDeviceDescription* desc) {
INT32 cacheIndex = findCacheItemByGUID(lpGuid, \
g_audioDeviceCache[desc->deviceID].isSource); if (cacheIndex == desc->deviceID) {
UnicodeToUTF8AndCopy(desc->name, lpstrDescription, DAUDIO_STRING_LENGTH);
//strncpy(desc->description, lpstrModule, DAUDIO_STRING_LENGTH);
desc->maxSimulLines = -1;
/* do not continue enumeration */
return FALSE;
}
return TRUE;
}
INT32 DAUDIO_GetDirectAudioDeviceDescription(INT32 mixerIndex, \
DirectAudioDeviceDescription* desc) {
if (!DS_lockCache()) {
return FALSE;
}
/* set the deviceID field to the cache index */
desc->deviceID = findCacheItemByMixerIndex(mixerIndex);
if (desc->deviceID < 0) {
DS_unlockCache();
return FALSE;
}
desc->maxSimulLines = 0;
if (g_audioDeviceCache[desc->deviceID].isSource) {
DirectSoundEnumerateW((LPDSENUMCALLBACKW) DS_GetDescEnum, desc);
strncpy(desc->description, "DirectSound Playback", DAUDIO_STRING_LENGTH);
} else {
DirectSoundCaptureEnumerateW((LPDSENUMCALLBACKW) DS_GetDescEnum, desc);
strncpy(desc->description, "DirectSound Capture", DAUDIO_STRING_LENGTH);
}
/*desc->vendor;
desc->version;*/
DS_unlockCache();
return (desc->maxSimulLines == -1)?TRUE:FALSE;
}
/* multi-channel info: http://www.microsoft.com/whdc/hwdev/tech/audio/multichaud.mspx \
*/
//static UINT32 sampleRateArray[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, \
56000, 88000, 96000, 172000, 192000 }; static INT32 sampleRateArray[] = { -1 };
static INT32 channelsArray[] = { 1, 2};
static INT32 bitsArray[] = { 8, 16, 24};
#define SAMPLERATE_COUNT sizeof(sampleRateArray)/sizeof(INT32)
#define CHANNELS_COUNT sizeof(channelsArray)/sizeof(INT32)
#define BITS_COUNT sizeof(bitsArray)/sizeof(INT32)
void DAUDIO_GetFormats(INT32 mixerIndex, INT32 deviceID, int isSource, void* creator) \
{
int rateIndex, channelIndex, bitIndex;
/* no need to lock, since deviceID identifies the device sufficiently */
/* sanity */
if (deviceID >= g_cacheCount) {
return;
}
if ((g_audioDeviceCache[deviceID].isSource && !isSource)
|| (!g_audioDeviceCache[deviceID].isSource && isSource)) {
/* only support Playback or Capture */
return;
}
for (rateIndex = 0; rateIndex < SAMPLERATE_COUNT; rateIndex++) {
for (channelIndex = 0; channelIndex < CHANNELS_COUNT; channelIndex++) {
for (bitIndex = 0; bitIndex < BITS_COUNT; bitIndex++) {
DAUDIO_AddAudioFormat(creator, bitsArray[bitIndex],
((bitsArray[bitIndex] + 7) / 8) * \
channelsArray[channelIndex], channelsArray[channelIndex],
(float) sampleRateArray[rateIndex],
DAUDIO_PCM,
(bitsArray[bitIndex]==8)?FALSE:TRUE, /* signed \
*/ (bitsArray[bitIndex]==8)?FALSE:
#ifndef _LITTLE_ENDIAN
TRUE /* big endian */
#else
FALSE /* little endian */
#endif
);
}
}
}
}
typedef struct {
int deviceID;
/* for convenience */
BOOL isSource;
/* the secondary buffer (Playback) */
LPDIRECTSOUNDBUFFER playBuffer;
/* the secondary buffer (Capture) */
LPDIRECTSOUNDCAPTUREBUFFER captureBuffer;
/* size of the directsound buffer, usually 2 seconds */
int dsBufferSizeInBytes;
/* size of the read/write-ahead, as specified by Java */
int bufferSizeInBytes;
int bitsPerSample;
int frameSize; // storage size in Bytes
UINT64 framePos;
/* where to write into the buffer.
* -1 if at current position (Playback)
* For Capture, this is the read position
*/
int writePos;
/* if start() had been called */
BOOL started;
/* how many bytes there is silence from current write position */
int silencedBytes;
BOOL underrun;
} DS_Info;
LPCSTR TranslateDSError(HRESULT hr) {
switch(hr) {
case DSERR_ALLOCATED:
return "DSERR_ALLOCATED";
case DSERR_CONTROLUNAVAIL:
return "DSERR_CONTROLUNAVAIL";
case DSERR_INVALIDPARAM:
return "DSERR_INVALIDPARAM";
case DSERR_INVALIDCALL:
return "DSERR_INVALIDCALL";
case DSERR_GENERIC:
return "DSERR_GENERIC";
case DSERR_PRIOLEVELNEEDED:
return "DSERR_PRIOLEVELNEEDED";
case DSERR_OUTOFMEMORY:
return "DSERR_OUTOFMEMORY";
case DSERR_BADFORMAT:
return "DSERR_BADFORMAT";
case DSERR_UNSUPPORTED:
return "DSERR_UNSUPPORTED";
case DSERR_NODRIVER:
return "DSERR_NODRIVER";
case DSERR_ALREADYINITIALIZED:
return "DSERR_ALREADYINITIALIZED";
case DSERR_NOAGGREGATION:
return "DSERR_NOAGGREGATION";
case DSERR_BUFFERLOST:
return "DSERR_BUFFERLOST";
case DSERR_OTHERAPPHASPRIO:
return "DSERR_OTHERAPPHASPRIO";
case DSERR_UNINITIALIZED:
return "DSERR_UNINITIALIZED";
default:
return "Unknown HRESULT";
}
}
/*
** data/routines for starting DS buffers by separate thread
** (joint into DS_StartBufferHelper class)
** see cr6372428: playback fails after exiting from thread that has started it
** due IDirectSoundBuffer8::Play() description:
** http://msdn.microsoft.com/archive/default.asp?url=/archive/en-us/directx9_c
** /directx/htm/idirectsoundbuffer8play.asp
** (remark section): If the application is multithreaded, the thread that plays
** the buffer must continue to exist as long as the buffer is playing.
** Buffers created on WDM drivers stop playing when the thread is terminated.
** IDirectSoundCaptureBuffer8::Start() has the same remark:
** http://msdn.microsoft.com/archive/default.asp?url=/archive/en-us/directx9_c
** /directx/htm/idirectsoundcapturebuffer8start.asp
*/
class DS_StartBufferHelper {
public:
/* starts DirectSound buffer (playback or capture) */
static HRESULT StartBuffer(DS_Info* info);
/* checks for initialization success */
static inline BOOL isInitialized() { return data.threadHandle != NULL; }
protected:
DS_StartBufferHelper() {} // no need to create an instance
/* data class */
class Data {
public:
Data();
~Data();
// public data to access from parent class
CRITICAL_SECTION crit_sect;
volatile HANDLE threadHandle;
volatile HANDLE startEvent;
volatile HANDLE startedEvent;
volatile DS_Info* line2Start;
volatile HRESULT startResult;
} static data;
/* StartThread function */
static DWORD WINAPI __stdcall ThreadProc(void *param);
};
/* StartBufferHelper class implementation
*/
DS_StartBufferHelper::Data DS_StartBufferHelper::data;
DS_StartBufferHelper::Data::Data() {
threadHandle = NULL;
::InitializeCriticalSection(&crit_sect);
startEvent = ::CreateEvent(NULL, FALSE, FALSE, NULL);
startedEvent = ::CreateEvent(NULL, FALSE, FALSE, NULL);
if (startEvent != NULL && startedEvent != NULL)
threadHandle = ::CreateThread(NULL, 0, ThreadProc, NULL, 0, NULL);
}
DS_StartBufferHelper::Data::~Data() {
::EnterCriticalSection(&crit_sect);
if (threadHandle != NULL) {
// terminate thread
line2Start = NULL;
::SetEvent(startEvent);
::CloseHandle(threadHandle);
threadHandle = NULL;
}
::LeaveCriticalSection(&crit_sect);
// won't delete startEvent/startedEvent/crit_sect
// - Windows will do during process shutdown
}
DWORD WINAPI __stdcall DS_StartBufferHelper::ThreadProc(void *param)
{
::CoInitialize(NULL);
while (1) {
// wait for something to do
::WaitForSingleObject(data.startEvent, INFINITE);
if (data.line2Start == NULL) {
// (data.line2Start == NULL) is a signal to terminate thread
break;
}
if (data.line2Start->isSource) {
data.startResult =
data.line2Start->playBuffer->Play(0, 0, DSBPLAY_LOOPING);
} else {
data.startResult =
data.line2Start->captureBuffer->Start(DSCBSTART_LOOPING);
}
::SetEvent(data.startedEvent);
}
::CoUninitialize();
return 0;
}
HRESULT DS_StartBufferHelper::StartBuffer(DS_Info* info) {
HRESULT hr;
::EnterCriticalSection(&data.crit_sect);
if (!isInitialized()) {
::LeaveCriticalSection(&data.crit_sect);
return E_FAIL;
}
data.line2Start = info;
::SetEvent(data.startEvent);
::WaitForSingleObject(data.startedEvent, INFINITE);
hr = data.startResult;
::LeaveCriticalSection(&data.crit_sect);
return hr;
}
/* helper routines for DS buffer positions */
/* returns distance from pos1 to pos2
*/
inline int DS_getDistance(DS_Info* info, int pos1, int pos2) {
int distance = pos2 - pos1;
while (distance < 0)
distance += info->dsBufferSizeInBytes;
return distance;
}
/* adds 2 positions
*/
inline int DS_addPos(DS_Info* info, int pos1, int pos2) {
int result = pos1 + pos2;
while (result >= info->dsBufferSizeInBytes)
result -= info->dsBufferSizeInBytes;
return result;
}
BOOL DS_addDeviceRef(INT32 deviceID) {
HWND ownerWindow;
HRESULT res = DS_OK;
LPDIRECTSOUND devPlay;
LPDIRECTSOUNDCAPTURE devCapture;
LPGUID lpGuid = NULL;
if (g_audioDeviceCache[deviceID].dev == NULL) {
/* Create DirectSound */
TRACE1("Creating DirectSound object for device %d\n", deviceID);
lpGuid = &(g_audioDeviceCache[deviceID].guid);
if (isEqualGUID(lpGuid, NULL)) {
lpGuid = NULL;
}
if (g_audioDeviceCache[deviceID].isSource) {
res = DirectSoundCreate(lpGuid, &devPlay, NULL);
g_audioDeviceCache[deviceID].dev = (void*) devPlay;
} else {
res = DirectSoundCaptureCreate(lpGuid, &devCapture, NULL);
g_audioDeviceCache[deviceID].dev = (void*) devCapture;
}
g_audioDeviceCache[deviceID].refCount = 0;
if (FAILED(res)) {
ERROR1("DAUDIO_Open: ERROR: Failed to create DirectSound: %s", \
TranslateDSError(res)); g_audioDeviceCache[deviceID].dev = NULL;
return FALSE;
}
if (g_audioDeviceCache[deviceID].isSource) {
ownerWindow = GetForegroundWindow();
if (ownerWindow == NULL) {
ownerWindow = GetDesktopWindow();
}
TRACE0("DAUDIO_Open: Setting cooperative level\n");
res = devPlay->SetCooperativeLevel(ownerWindow, DSSCL_NORMAL);
if (FAILED(res)) {
ERROR1("DAUDIO_Open: ERROR: Failed to set cooperative level: %s", \
TranslateDSError(res)); return FALSE;
}
}
}
g_audioDeviceCache[deviceID].refCount++;
return TRUE;
}
#define DEV_PLAY(devID) ((LPDIRECTSOUND) g_audioDeviceCache[devID].dev)
#define DEV_CAPTURE(devID) ((LPDIRECTSOUNDCAPTURE) g_audioDeviceCache[devID].dev)
void DS_removeDeviceRef(INT32 deviceID) {
if (g_audioDeviceCache[deviceID].refCount) {
g_audioDeviceCache[deviceID].refCount--;
}
if (g_audioDeviceCache[deviceID].refCount == 0) {
if (g_audioDeviceCache[deviceID].dev != NULL) {
if (g_audioDeviceCache[deviceID].isSource) {
DEV_PLAY(deviceID)->Release();
} else {
DEV_CAPTURE(deviceID)->Release();
}
g_audioDeviceCache[deviceID].dev = NULL;
}
}
}
#ifndef _WAVEFORMATEXTENSIBLE_
#define _WAVEFORMATEXTENSIBLE_
typedef struct {
WAVEFORMATEX Format;
union {
WORD wValidBitsPerSample; /* bits of precision */
WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */
WORD wReserved; /* If neither applies, set to zero. */
} Samples;
DWORD dwChannelMask; /* which channels are */
/* present in stream */
GUID SubFormat;
} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
#endif // !_WAVEFORMATEXTENSIBLE_
#if !defined(WAVE_FORMAT_EXTENSIBLE)
#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
#endif // !defined(WAVE_FORMAT_EXTENSIBLE)
#if !defined(DEFINE_WAVEFORMATEX_GUID)
#define DEFINE_WAVEFORMATEX_GUID(x) (USHORT)(x), 0x0000, 0x0010, 0x80, 0x00, 0x00, \
0xaa, 0x00, 0x38, 0x9b, 0x71 #endif
#ifndef STATIC_KSDATAFORMAT_SUBTYPE_PCM
#define STATIC_KSDATAFORMAT_SUBTYPE_PCM\
DEFINE_WAVEFORMATEX_GUID(WAVE_FORMAT_PCM)
#endif
void createWaveFormat(WAVEFORMATEXTENSIBLE* format,
int sampleRate,
int channels,
int bits,
int significantBits) {
GUID subtypePCM = {STATIC_KSDATAFORMAT_SUBTYPE_PCM};
format->Format.nSamplesPerSec = (DWORD)sampleRate;
format->Format.nChannels = (WORD) channels;
/* do not support useless padding, like 24-bit samples stored in 32-bit \
containers */ format->Format.wBitsPerSample = (WORD) ((bits + 7) & 0xFFF8);
if (channels <= 2 && bits <= 24) {
format->Format.wFormatTag = WAVE_FORMAT_PCM;
format->Format.cbSize = 0;
} else {
format->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
format->Format.cbSize = 22;
format->Samples.wValidBitsPerSample = bits;
/* no way to specify speaker locations */
format->dwChannelMask = 0xFFFFFFFF;
format->SubFormat = subtypePCM;
}
format->Format.nBlockAlign = (WORD)((format->Format.wBitsPerSample * \
format->Format.nChannels) / 8); format->Format.nAvgBytesPerSec = \
format->Format.nSamplesPerSec * format->Format.nBlockAlign; }
/* fill buffer with silence
*/
void DS_clearBuffer(DS_Info* info, BOOL fromWritePos) {
UBYTE* pb1=NULL, *pb2=NULL;
DWORD cb1=0, cb2=0;
DWORD flags = 0;
int start, count;
TRACE1("> DS_clearBuffer for device %d\n", info->deviceID);
if (info->isSource) {
if (fromWritePos) {
DWORD playCursor, writeCursor;
int end;
if (FAILED(info->playBuffer->GetCurrentPosition(&playCursor, \
&writeCursor))) {
ERROR0(" DS_clearBuffer: ERROR: Failed to get current \
position."); TRACE0("< DS_clearbuffer\n");
return;
}
DEBUG_SILENCING2(" DS_clearBuffer: DS playPos=%d myWritePos=%d", \
(int) playCursor, (int) info->writePos); if (info->writePos >= 0) {
start = info->writePos + info->silencedBytes;
} else {
start = writeCursor + info->silencedBytes;
//flags |= DSBLOCK_FROMWRITECURSOR;
}
while (start >= info->dsBufferSizeInBytes) {
start -= info->dsBufferSizeInBytes;
}
// fix for bug 6251460 (REGRESSION: short sounds do not play)
// for unknown reason with hardware DS buffer playCursor sometimes
// jumps back for little interval (mostly 2-8 bytes) (writeCursor \
moves forward as usual)
// The issue happens right after start playing and for short sounds \
only (less then DS buffer,
// when whole sound written into the buffer and remaining space \
filled by silence)
// the case doesn't produce any audible aftifacts so just catch it to \
prevent filling // whole buffer by silence.
if (((int)playCursor <= start && start < (int)writeCursor)
|| (writeCursor < playCursor // buffer bound is between \
playCursor & writeCursor
&& (start < (int)writeCursor || (int)playCursor <= start))) {
return;
}
count = info->dsBufferSizeInBytes - info->silencedBytes;
// why / 4?
//if (count > info->dsBufferSizeInBytes / 4) {
// count = info->dsBufferSizeInBytes / 4;
//}
end = start + count;
if ((int) playCursor < start) {
playCursor += (DWORD) info->dsBufferSizeInBytes;
}
if (start <= (int) playCursor && end > (int) playCursor) {
/* at maximum, silence until play cursor */
count = (int) playCursor - start;
#ifdef USE_TRACE
if ((int) playCursor >= info->dsBufferSizeInBytes) playCursor -= \
(DWORD) info->dsBufferSizeInBytes; TRACE3("\n DS_clearBuffer: Start Writing from \
%d, "
"would overwrite playCursor=%d, so reduce count to %d\n",
start, playCursor, count);
#endif
}
DEBUG_SILENCING2(" clearing buffer from %d, count=%d. ", (int)start, \
(int) count); if (count <= 0) {
DEBUG_SILENCING0("\n");
TRACE1("< DS_clearBuffer: no need to clear, silencedBytes=%d\n", \
info->silencedBytes); return;
}
} else {
start = 0;
count = info->dsBufferSizeInBytes;
flags |= DSBLOCK_ENTIREBUFFER;
}
if (FAILED(info->playBuffer->Lock(start,
count,
(LPVOID*) &pb1, &cb1,
(LPVOID*) &pb2, &cb2, flags))) {
ERROR0("\n DS_clearBuffer: ERROR: Failed to lock sound buffer.\n");
TRACE0("< DS_clearbuffer\n");
return;
}
} else {
if (FAILED(info->captureBuffer->Lock(0,
info->dsBufferSizeInBytes,
(LPVOID*) &pb1, &cb1,
(LPVOID*) &pb2, &cb2, \
DSCBLOCK_ENTIREBUFFER))) {
ERROR0(" DS_clearBuffer: ERROR: Failed to lock sound buffer.\n");
TRACE0("< DS_clearbuffer\n");
return;
}
}
if (pb1!=NULL) {
memset(pb1, (info->bitsPerSample == 8)?128:0, cb1);
}
if (pb2!=NULL) {
memset(pb2, (info->bitsPerSample == 8)?128:0, cb2);
}
if (info->isSource) {
info->playBuffer->Unlock( pb1, cb1, pb2, cb2 );
if (!fromWritePos) {
/* doesn't matter where to start writing next time */
info->writePos = -1;
info->silencedBytes = info->dsBufferSizeInBytes;
} else {
info->silencedBytes += (cb1+cb2);
if (info->silencedBytes > info->dsBufferSizeInBytes) {
ERROR1(" DS_clearbuffer: ERROR: silencedBytes=%d exceeds buffer \
size!\n", info->silencedBytes);
info->silencedBytes = info->dsBufferSizeInBytes;
}
}
DEBUG_SILENCING2(" silencedBytes=%d, my writePos=%d\n", \
(int)info->silencedBytes, (int)info->writePos); } else {
info->captureBuffer->Unlock( pb1, cb1, pb2, cb2 );
}
TRACE0("< DS_clearbuffer\n");
}
/* returns pointer to buffer */
void* DS_createSoundBuffer(DS_Info* info,
float sampleRate,
int sampleSizeInBits,
int channels,
int bufferSizeInBytes) {
DSBUFFERDESC dsbdesc;
DSCBUFFERDESC dscbdesc;
HRESULT res;
WAVEFORMATEXTENSIBLE format;
void* buffer;
TRACE1("Creating secondary buffer for device %d\n", info->deviceID);
createWaveFormat(&format,
(int) sampleRate,
channels,
info->frameSize / channels * 8,
sampleSizeInBits);
/* 2 second secondary buffer */
info->dsBufferSizeInBytes = 2 * ((int) sampleRate) * info->frameSize;
if (bufferSizeInBytes > info->dsBufferSizeInBytes / 2) {
bufferSizeInBytes = info->dsBufferSizeInBytes / 2;
}
bufferSizeInBytes = (bufferSizeInBytes / info->frameSize) * info->frameSize;
info->bufferSizeInBytes = bufferSizeInBytes;
if (info->isSource) {
memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
dsbdesc.dwSize = sizeof(DSBUFFERDESC);
dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2
| DSBCAPS_GLOBALFOCUS;
dsbdesc.dwBufferBytes = info->dsBufferSizeInBytes;
dsbdesc.lpwfxFormat = (WAVEFORMATEX*) &format;
res = DEV_PLAY(info->deviceID)->CreateSoundBuffer
(&dsbdesc, (LPDIRECTSOUNDBUFFER*) &buffer, NULL);
} else {
memset(&dscbdesc, 0, sizeof(DSCBUFFERDESC));
dscbdesc.dwSize = sizeof(DSCBUFFERDESC);
dscbdesc.dwFlags = 0;
dscbdesc.dwBufferBytes = info->dsBufferSizeInBytes;
dscbdesc.lpwfxFormat = (WAVEFORMATEX*) &format;
res = DEV_CAPTURE(info->deviceID)->CreateCaptureBuffer
(&dscbdesc, (LPDIRECTSOUNDCAPTUREBUFFER*) &buffer, NULL);
}
if (FAILED(res)) {
ERROR1("DS_createSoundBuffer: ERROR: Failed to create sound buffer: %s", \
TranslateDSError(res)); return NULL;
}
return buffer;
}
void DS_destroySoundBuffer(DS_Info* info) {
if (info->playBuffer != NULL) {
info->playBuffer->Release();
info->playBuffer = NULL;
}
if (info->captureBuffer != NULL) {
info->captureBuffer->Release();
info->captureBuffer = NULL;
}
}
void* DAUDIO_Open(INT32 mixerIndex, INT32 deviceID, int isSource,
int encoding, float sampleRate, int sampleSizeInBits,
int frameSize, int channels,
int isSigned, int isBigEndian, int bufferSizeInBytes) {
DS_Info* info;
void* buffer;
TRACE0("> DAUDIO_Open\n");
/* some sanity checks */
if (deviceID >= g_cacheCount) {
ERROR1("DAUDIO_Open: ERROR: cannot open the device with deviceID=%d!\n", \
deviceID); return NULL;
}
if ((g_audioDeviceCache[deviceID].isSource && !isSource)
|| (!g_audioDeviceCache[deviceID].isSource && isSource)) {
/* only support Playback or Capture */
ERROR0("DAUDIO_Open: ERROR: Cache is corrupt: cannot open the device in \
specified isSource mode!\n"); return NULL;
}
if (encoding != DAUDIO_PCM) {
ERROR1("DAUDIO_Open: ERROR: cannot open the device with encoding=%d!\n", \
encoding); return NULL;
}
if (channels <= 0) {
ERROR1("DAUDIO_Open: ERROR: Invalid number of channels=%d!\n", channels);
return NULL;
}
if (sampleSizeInBits > 8 &&
#ifdef _LITTLE_ENDIAN
isBigEndian
#else
!isBigEndian
#endif
) {
ERROR1("DAUDIO_Open: ERROR: wrong endianness: isBigEndian==%d!\n", \
isBigEndian); return NULL;
}
if (sampleSizeInBits == 8 && isSigned) {
ERROR0("DAUDIO_Open: ERROR: wrong signed'ness: with 8 bits, data must be \
unsigned!\n"); return NULL;
}
if (!DS_StartBufferHelper::isInitialized()) {
ERROR0("DAUDIO_Open: ERROR: StartBufferHelper initialization was failed!\n");
return NULL;
}
info = (DS_Info*) malloc(sizeof(DS_Info));
if (!info) {
ERROR0("DAUDIO_Open: ERROR: Out of memory\n");
return NULL;
}
memset(info, 0, sizeof(DS_Info));
info->deviceID = deviceID;
info->isSource = isSource;
info->bitsPerSample = sampleSizeInBits;
info->frameSize = frameSize;
info->framePos = 0;
info->started = FALSE;
info->underrun = FALSE;
if (!DS_addDeviceRef(deviceID)) {
DS_removeDeviceRef(deviceID);
free(info);
return NULL;
}
buffer = DS_createSoundBuffer(info,
sampleRate,
sampleSizeInBits,
channels,
bufferSizeInBytes);
if (!buffer) {
DS_removeDeviceRef(deviceID);
free(info);
return NULL;
}
if (info->isSource) {
info->playBuffer = (LPDIRECTSOUNDBUFFER) buffer;
} else {
info->captureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) buffer;
}
DS_clearBuffer(info, FALSE /* entire buffer */);
/* use writepos of device */
if (info->isSource) {
info->writePos = -1;
} else {
info->writePos = 0;
}
TRACE0("< DAUDIO_Open: Opened device successfully.\n");
return (void*) info;
}
int DAUDIO_Start(void* id, int isSource) {
DS_Info* info = (DS_Info*) id;
HRESULT res = DS_OK;
DWORD status;
TRACE0("> DAUDIO_Start\n");
if (info->isSource) {
res = info->playBuffer->GetStatus(&status);
if (res == DS_OK) {
if (status & DSBSTATUS_LOOPING) {
ERROR0("DAUDIO_Start: ERROR: Already started!");
return TRUE;
}
/* only start buffer if already something written to it */
if (info->writePos >= 0) {
res = DS_StartBufferHelper::StartBuffer(info);
if (res == DSERR_BUFFERLOST) {
res = info->playBuffer->Restore();
if (res == DS_OK) {
DS_clearBuffer(info, FALSE /* entire buffer */);
/* write() will trigger actual device start */
}
} else {
/* make sure that we will have silence after
the currently valid audio data */
DS_clearBuffer(info, TRUE /* from write position */);
}
}
}
} else {
if (info->captureBuffer->GetStatus(&status) == DS_OK) {
if (status & DSCBSTATUS_LOOPING) {
ERROR0("DAUDIO_Start: ERROR: Already started!");
return TRUE;
}
}
res = DS_StartBufferHelper::StartBuffer(info);
}
if (FAILED(res)) {
ERROR1("DAUDIO_Start: ERROR: Failed to start: %s", TranslateDSError(res));
return FALSE;
}
info->started = TRUE;
return TRUE;
}
int DAUDIO_Stop(void* id, int isSource) {
DS_Info* info = (DS_Info*) id;
TRACE0("> DAUDIO_Stop\n");
info->started = FALSE;
if (info->isSource) {
info->playBuffer->Stop();
} else {
info->captureBuffer->Stop();
}
TRACE0("< DAUDIO_Stop\n");
return TRUE;
}
void DAUDIO_Close(void* id, int isSource) {
DS_Info* info = (DS_Info*) id;
TRACE0("DAUDIO_Close\n");
if (info != NULL) {
DS_destroySoundBuffer(info);
DS_removeDeviceRef(info->deviceID);
free(info);
}
}
/* Check buffer for underrun
* This method is only meaningful for Output devices (write devices).
*/
void DS_CheckUnderrun(DS_Info* info, DWORD playCursor, DWORD writeCursor) {
TRACE5("DS_CheckUnderrun: playCursor=%d, writeCursor=%d, "
"info->writePos=%d silencedBytes=%d dsBufferSizeInBytes=%d\n",
(int) playCursor, (int) writeCursor, (int) info->writePos,
(int) info->silencedBytes, (int) info->dsBufferSizeInBytes);
if (info->underrun || info->writePos < 0) return;
int writeAhead = DS_getDistance(info, writeCursor, info->writePos);
if (writeAhead > info->bufferSizeInBytes) {
// this may occur after Stop(), when writeCursor decreases (real valid data \
size > bufferSizeInBytes)
// But the case can occur only when we have more then info->bufferSizeInBytes \
valid bytes
// (and less then (info->dsBufferSizeInBytes - info->bufferSizeInBytes) \
silenced bytes)
// If we already have a lot of silencedBytes after valid data (written by
// DAUDIO_StillDraining() or DAUDIO_Service()) then it's underrun
if (info->silencedBytes >= info->dsBufferSizeInBytes - \
info->bufferSizeInBytes) { // underrun!
ERROR0("DS_CheckUnderrun: ERROR: underrun detected!\n");
info->underrun = TRUE;
}
}
}
/* For source (playback) line:
* (a) if (fromPlayCursor == FALSE), returns number of bytes available
* for writing: bufferSize - (info->writePos - writeCursor);
* (b) if (fromPlayCursor == TRUE), playCursor is used instead writeCursor
* and returned value can be used for play position calculation (see also
* note about bufferSize)
* For destination (capture) line:
* (c) if (fromPlayCursor == FALSE), returns number of bytes available
* for reading from the buffer: readCursor - info->writePos;
* (d) if (fromPlayCursor == TRUE), captureCursor is used instead readCursor
* and returned value can be used for capture position calculation (see
* note about bufferSize)
* bufferSize parameter are filled by "actual" buffer size:
* if (fromPlayCursor == FALSE), bufferSize = info->bufferSizeInBytes
* otherwise it increase by number of bytes currently processed by DirectSound
* (writeCursor - playCursor) or (captureCursor - readCursor)
*/
int DS_GetAvailable(DS_Info* info,
DWORD* playCursor, DWORD* writeCursor,
int* bufferSize, BOOL fromPlayCursor) {
int available;
int newReadPos;
TRACE2("DS_GetAvailable: fromPlayCursor=%d, deviceID=%d\n", fromPlayCursor, \
info->deviceID); if (!info->playBuffer && !info->captureBuffer) {
ERROR0("DS_GetAvailable: ERROR: buffer not yet created");
return 0;
}
if (info->isSource) {
if (FAILED(info->playBuffer->GetCurrentPosition(playCursor, writeCursor))) {
ERROR0("DS_GetAvailable: ERROR: Failed to get current position.\n");
return 0;
}
int processing = DS_getDistance(info, (int)*playCursor, (int)*writeCursor);
// workaround: sometimes DirectSound report writeCursor is less (for several \
bytes) then playCursor if (processing > info->dsBufferSizeInBytes / 2) {
*writeCursor = *playCursor;
processing = 0;
}
TRACE3(" playCursor=%d, writeCursor=%d, info->writePos=%d\n",
*playCursor, *writeCursor, info->writePos);
*bufferSize = info->bufferSizeInBytes;
if (fromPlayCursor) {
*bufferSize += processing;
}
DS_CheckUnderrun(info, *playCursor, *writeCursor);
if (info->writePos == -1 || (info->underrun && !fromPlayCursor)) {
/* always full buffer if at beginning */
available = *bufferSize;
} else {
int currWriteAhead = DS_getDistance(info, fromPlayCursor ? \
(int)*playCursor : (int)*writeCursor, info->writePos); if (currWriteAhead > \
*bufferSize) { if (info->underrun) {
// playCursor surpassed writePos - no valid data, whole buffer \
available available = *bufferSize;
} else {
// the case may occur after stop(), when writeCursor jumps back \
to playCursor // so "actual" buffer size has grown
*bufferSize = currWriteAhead;
available = 0;
}
} else {
available = *bufferSize - currWriteAhead;
}
}
} else {
if (FAILED(info->captureBuffer->GetCurrentPosition(playCursor, writeCursor))) \
{
ERROR0("DS_GetAvailable: ERROR: Failed to get current position.\n");
return 0;
}
*bufferSize = info->bufferSizeInBytes;
if (fromPlayCursor) {
*bufferSize += DS_getDistance(info, (int)*playCursor, (int)*writeCursor);
}
TRACE4(" captureCursor=%d, readCursor=%d, info->readPos=%d \
refBufferSize=%d\n",
*playCursor, *writeCursor, info->writePos, *bufferSize);
if (info->writePos == -1) {
/* always empty buffer if at beginning */
info->writePos = (int) (*writeCursor);
}
if (fromPlayCursor) {
available = ((int) (*playCursor) - info->writePos);
} else {
available = ((int) (*writeCursor) - info->writePos);
}
if (available < 0) {
available += info->dsBufferSizeInBytes;
}
if (!fromPlayCursor && available > info->bufferSizeInBytes) {
/* overflow */
ERROR2("DS_GetAvailable: ERROR: overflow detected: "
"DirectSoundBufferSize=%d, bufferSize=%d, ",
info->dsBufferSizeInBytes, info->bufferSizeInBytes);
ERROR3("captureCursor=%d, readCursor=%d, info->readPos=%d\n",
*playCursor, *writeCursor, info->writePos);
/* advance read position, to allow exactly one buffer worth of data */
newReadPos = (int) (*writeCursor) - info->bufferSizeInBytes;
if (newReadPos < 0) {
newReadPos += info->dsBufferSizeInBytes;
}
info->writePos = newReadPos;
available = info->bufferSizeInBytes;
}
}
available = (available / info->frameSize) * info->frameSize;
TRACE1("DS_available: Returning %d available bytes\n", (int) available);
return available;
}
// returns -1 on error, otherwise bytes written
int DAUDIO_Write(void* id, char* data, int byteSize) {
DS_Info* info = (DS_Info*) id;
int available;
int thisWritePos;
DWORD playCursor, writeCursor;
HRESULT res;
void* buffer1, *buffer2;
DWORD buffer1len, buffer2len;
BOOL needRestart = FALSE;
int bufferLostTrials = 2;
int bufferSize;
TRACE1("> DAUDIO_Write %d bytes\n", byteSize);
while (--bufferLostTrials > 0) {
available = DS_GetAvailable(info, &playCursor, &writeCursor, &bufferSize, \
FALSE /* fromPlayCursor */); if (byteSize > available) byteSize = available;
if (byteSize == 0) break;
thisWritePos = info->writePos;
if (thisWritePos == -1 || info->underrun) {
// play from current write cursor after flush, etc.
needRestart = TRUE;
thisWritePos = writeCursor;
info->underrun = FALSE;
}
DEBUG_SILENCING2("DAUDIO_Write: writing from %d, count=%d\n", (int) \
thisWritePos, (int) byteSize); res = info->playBuffer->Lock(thisWritePos, byteSize,
(LPVOID *) &buffer1, &buffer1len,
(LPVOID *) &buffer2, &buffer2len,
0);
if (res != DS_OK) {
/* some DS failure */
if (res == DSERR_BUFFERLOST) {
ERROR0("DAUDIO_write: ERROR: Restoring lost Buffer.");
if (info->playBuffer->Restore() == DS_OK) {
DS_clearBuffer(info, FALSE /* entire buffer */);
info->writePos = -1;
/* try again */
continue;
}
}
/* can't recover from error */
byteSize = 0;
break;
}
/* buffer could be locked successfully */
/* first fill first buffer */
if (buffer1) {
memcpy(buffer1, data, buffer1len);
data = (char*) (((UINT_PTR) data) + buffer1len);
} else buffer1len = 0;
if (buffer2) {
memcpy(buffer2, data, buffer2len);
} else buffer2len = 0;
byteSize = buffer1len + buffer2len;
/* update next write pos */
thisWritePos += byteSize;
while (thisWritePos >= info->dsBufferSizeInBytes) {
thisWritePos -= info->dsBufferSizeInBytes;
}
/* commit data to directsound */
info->playBuffer->Unlock(buffer1, buffer1len, buffer2, buffer2len);
info->writePos = thisWritePos;
/* update position
* must be AFTER updating writePos,
* so that getSvailable doesn't return too little,
* so that getFramePos doesn't jump
*/
info->framePos += (byteSize / info->frameSize);
/* decrease silenced bytes */
if (info->silencedBytes > byteSize) {
info->silencedBytes -= byteSize;
} else {
info->silencedBytes = 0;
}
break;
} /* while */
/* start the device, if necessary */
if (info->started && needRestart && (info->writePos >= 0)) {
DS_StartBufferHelper::StartBuffer(info);
}
TRACE1("< DAUDIO_Write: returning %d bytes.\n", byteSize);
return byteSize;
}
// returns -1 on error
int DAUDIO_Read(void* id, char* data, int byteSize) {
DS_Info* info = (DS_Info*) id;
int available;
int thisReadPos;
DWORD captureCursor, readCursor;
HRESULT res;
void* buffer1, *buffer2;
DWORD buffer1len, buffer2len;
int bufferSize;
TRACE1("> DAUDIO_Read %d bytes\n", byteSize);
available = DS_GetAvailable(info, &captureCursor, &readCursor, &bufferSize, FALSE \
/* fromCaptureCursor? */); if (byteSize > available) byteSize = available;
if (byteSize > 0) {
thisReadPos = info->writePos;
if (thisReadPos == -1) {
/* from beginning */
thisReadPos = 0;
}
res = info->captureBuffer->Lock(thisReadPos, byteSize,
(LPVOID *) &buffer1, &buffer1len,
(LPVOID *) &buffer2, &buffer2len,
0);
if (res != DS_OK) {
/* can't recover from error */
byteSize = 0;
} else {
/* buffer could be locked successfully */
/* first fill first buffer */
if (buffer1) {
memcpy(data, buffer1, buffer1len);
data = (char*) (((UINT_PTR) data) + buffer1len);
} else buffer1len = 0;
if (buffer2) {
memcpy(data, buffer2, buffer2len);
} else buffer2len = 0;
byteSize = buffer1len + buffer2len;
/* update next read pos */
thisReadPos = DS_addPos(info, thisReadPos, byteSize);
/* commit data to directsound */
info->captureBuffer->Unlock(buffer1, buffer1len, buffer2, buffer2len);
/* update position
* must be BEFORE updating readPos,
* so that getAvailable doesn't return too much,
* so that getFramePos doesn't jump
*/
info->framePos += (byteSize / info->frameSize);
info->writePos = thisReadPos;
}
}
TRACE1("< DAUDIO_Read: returning %d bytes.\n", byteSize);
return byteSize;
}
int DAUDIO_GetBufferSize(void* id, int isSource) {
DS_Info* info = (DS_Info*) id;
return info->bufferSizeInBytes;
}
int DAUDIO_StillDraining(void* id, int isSource) {
DS_Info* info = (DS_Info*) id;
BOOL draining = FALSE;
int available, bufferSize;
DWORD playCursor, writeCursor;
DS_clearBuffer(info, TRUE /* from write position */);
available = DS_GetAvailable(info, &playCursor, &writeCursor, &bufferSize, TRUE /* \
fromPlayCursor */); draining = (available < bufferSize);
TRACE3("DAUDIO_StillDraining: available=%d silencedBytes=%d Still draining: \
%s\n", available, info->silencedBytes, draining?"TRUE":"FALSE");
return draining;
}
int DAUDIO_Flush(void* id, int isSource) {
DS_Info* info = (DS_Info*) id;
TRACE0("DAUDIO_Flush\n");
if (info->isSource) {
info->playBuffer->Stop();
DS_clearBuffer(info, FALSE /* entire buffer */);
} else {
DWORD captureCursor, readCursor;
/* set the read pointer to the current read position */
if (FAILED(info->captureBuffer->GetCurrentPosition(&captureCursor, \
&readCursor))) {
ERROR0("DAUDIO_Flush: ERROR: Failed to get current position.");
return FALSE;
}
DS_clearBuffer(info, FALSE /* entire buffer */);
/* SHOULD set to *captureCursor*,
* but that would be detected as overflow
* in a subsequent GetAvailable() call.
*/
info->writePos = (int) readCursor;
}
return TRUE;
}
int DAUDIO_GetAvailable(void* id, int isSource) {
DS_Info* info = (DS_Info*) id;
DWORD playCursor, writeCursor;
int ret, bufferSize;
ret = DS_GetAvailable(info, &playCursor, &writeCursor, &bufferSize, \
/*fromPlayCursor?*/ FALSE);
TRACE1("DAUDIO_GetAvailable returns %d bytes\n", ret);
return ret;
}
INT64 estimatePositionFromAvail(DS_Info* info, INT64 javaBytePos, int bufferSize, int \
availInBytes) { // estimate the current position with the buffer size and
// the available bytes to read or write in the buffer.
// not an elegant solution - bytePos will stop on xruns,
// and in race conditions it may jump backwards
// Advantage is that it is indeed based on the samples that go through
// the system (rather than time-based methods)
if (info->isSource) {
// javaBytePos is the position that is reached when the current
// buffer is played completely
return (INT64) (javaBytePos - bufferSize + availInBytes);
} else {
// javaBytePos is the position that was when the current buffer was empty
return (INT64) (javaBytePos + availInBytes);
}
}
INT64 DAUDIO_GetBytePosition(void* id, int isSource, INT64 javaBytePos) {
DS_Info* info = (DS_Info*) id;
int available, bufferSize;
DWORD playCursor, writeCursor;
INT64 result = javaBytePos;
available = DS_GetAvailable(info, &playCursor, &writeCursor, &bufferSize, \
/*fromPlayCursor?*/ TRUE); result = estimatePositionFromAvail(info, javaBytePos, \
bufferSize, available); return result;
}
void DAUDIO_SetBytePosition(void* id, int isSource, INT64 javaBytePos) {
/* save to ignore, since GetBytePosition
* takes the javaBytePos param into account
*/
}
int DAUDIO_RequiresServicing(void* id, int isSource) {
// need servicing on for SourceDataLines
return isSource?TRUE:FALSE;
}
void DAUDIO_Service(void* id, int isSource) {
DS_Info* info = (DS_Info*) id;
if (isSource) {
if (info->silencedBytes < info->dsBufferSizeInBytes) {
// clear buffer
TRACE0("DAUDIO_Service\n");
DS_clearBuffer(info, TRUE /* from write position */);
}
if (info->writePos >= 0
&& info->started
&& !info->underrun
&& info->silencedBytes >= info->dsBufferSizeInBytes) {
// if we're currently playing, and the entire buffer is silenced...
// then we are underrunning!
info->underrun = TRUE;
ERROR0("DAUDIO_Service: ERROR: DirectSound: underrun detected!\n");
}
}
}
//cak #endif // USE_DAUDIO
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