Mirrors_Framework/SDL
1
0
Fork 0
mirror of https://github.com/libsdl-org/SDL.git synced 2026-05-13 03:38:35 +08:00
Code Issues Actions 8 Packages Projects Releases Wiki Activity

Merge audio capture work back into the mainline.

Browse Source
This commit is contained in:
Ryan C. Gordon
2016-08-28 13:36:13 -04:00
parent 06dcdc7d48 b6daf1f60a
commit 2da1ec8354
73 changed files with 2448 additions and 1144 deletions
Download Patch File Download Diff File Expand all files Collapse all files
.hgignore
+1
View File
@@ -120,6 +120,7 @@ test/testbounds
test/torturethread
test/testdisplayinfo
test/testqsort
test/testaudiocapture
test/*.exe
test/*.dSYM
buildbot
BUGS.txt
+1 -1
View File
@@ -1,7 +1,7 @@
Bugs are now managed in the SDL bug tracker, here:
http://bugzilla.libsdl.org/
https://bugzilla.libsdl.org/
You may report bugs there, and search to see if a given issue has already
been reported, discussed, and maybe even fixed.
VisualC-WinRT/UWP_VS2015/SDL-UWP.vcxproj
-1
View File
@@ -81,7 +81,6 @@
<ClInclude Include="..\..\src\audio\disk\SDL_diskaudio.h" />
<ClInclude Include="..\..\src\audio\dummy\SDL_dummyaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\src\audio\SDL_sysaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_wave.h" />
VisualC-WinRT/UWP_VS2015/SDL-UWP.vcxproj.filters
-3
View File
@@ -174,9 +174,6 @@
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audiomem.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audio_c.h">
<Filter>Source Files</Filter>
</ClInclude>
VisualC-WinRT/WinPhone80_VS2012/SDL-WinPhone80.vcxproj
-1
View File
@@ -208,7 +208,6 @@
<ClInclude Include="..\..\src\audio\disk\SDL_diskaudio.h" />
<ClInclude Include="..\..\src\audio\dummy\SDL_dummyaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\src\audio\SDL_sysaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_wave.h" />
VisualC-WinRT/WinPhone80_VS2012/SDL-WinPhone80.vcxproj.filters
-3
View File
@@ -159,9 +159,6 @@
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audiomem.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audio_c.h">
<Filter>Source Files</Filter>
</ClInclude>
VisualC-WinRT/WinPhone81_VS2013/SDL-WinPhone81.vcxproj
-1
View File
@@ -73,7 +73,6 @@
<ClInclude Include="..\..\src\audio\disk\SDL_diskaudio.h" />
<ClInclude Include="..\..\src\audio\dummy\SDL_dummyaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\src\audio\SDL_sysaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_wave.h" />
VisualC-WinRT/WinPhone81_VS2013/SDL-WinPhone81.vcxproj.filters
-3
View File
@@ -174,9 +174,6 @@
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audiomem.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audio_c.h">
<Filter>Source Files</Filter>
</ClInclude>
VisualC-WinRT/WinRT80_VS2012/SDL-WinRT80.vcxproj
-1
View File
@@ -287,7 +287,6 @@
<ClInclude Include="..\..\src\audio\disk\SDL_diskaudio.h" />
<ClInclude Include="..\..\src\audio\dummy\SDL_dummyaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\src\audio\SDL_sysaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_wave.h" />
VisualC-WinRT/WinRT80_VS2012/SDL-WinRT80.vcxproj.filters
-3
View File
@@ -420,9 +420,6 @@
<ClInclude Include="..\..\include\SDL_clipboard.h">
<Filter>Header Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audiomem.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\render\software\SDL_blendfillrect.h">
<Filter>Source Files</Filter>
</ClInclude>
VisualC-WinRT/WinRT81_VS2013/SDL-WinRT81.vcxproj
-1
View File
@@ -81,7 +81,6 @@
<ClInclude Include="..\..\src\audio\disk\SDL_diskaudio.h" />
<ClInclude Include="..\..\src\audio\dummy\SDL_dummyaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\src\audio\SDL_sysaudio.h" />
<ClInclude Include="..\..\src\audio\SDL_wave.h" />
VisualC-WinRT/WinRT81_VS2013/SDL-WinRT81.vcxproj.filters
-3
View File
@@ -174,9 +174,6 @@
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audiomem.h">
<Filter>Source Files</Filter>
</ClInclude>
<ClInclude Include="..\..\src\audio\SDL_audio_c.h">
<Filter>Source Files</Filter>
</ClInclude>
VisualC/SDL/SDL.vcxproj
-1
View File
@@ -294,7 +294,6 @@
<ClInclude Include="resource.h" />
<ClInclude Include="..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\src\render\software\SDL_blendfillrect.h" />
<ClInclude Include="..\..\src\render\software\SDL_blendline.h" />
<ClInclude Include="..\..\src\render\software\SDL_blendpoint.h" />
VisualC/SDL/SDL.vcxproj.filters
-1
View File
@@ -230,7 +230,6 @@
<ClInclude Include="resource.h" />
<ClInclude Include="..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\src\render\software\SDL_blendfillrect.h" />
<ClInclude Include="..\..\src\render\software\SDL_blendline.h" />
<ClInclude Include="..\..\src\render\software\SDL_blendpoint.h" />
VisualC/SDL/SDL_VS2008.vcproj
-4
View File
@@ -771,10 +771,6 @@
RelativePath="..\..\src\audio\SDL_audiodev_c.h"
>
</File>
<File
RelativePath="..\..\src\audio\SDL_audiomem.h"
>
</File>
<File
RelativePath="..\..\src\audio\SDL_audiotypecvt.c"
>
Xcode-iOS/SDL/SDL.xcodeproj/project.pbxproj
-2
View File
@@ -376,7 +376,6 @@
FD99B9440DD52EDC00FB1D6B /* SDL_audio.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_audio.c; sourceTree = "<group>"; };
FD99B9450DD52EDC00FB1D6B /* SDL_audio_c.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_audio_c.h; sourceTree = "<group>"; };
FD99B9460DD52EDC00FB1D6B /* SDL_audiocvt.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_audiocvt.c; sourceTree = "<group>"; };
FD99B9490DD52EDC00FB1D6B /* SDL_audiomem.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_audiomem.h; sourceTree = "<group>"; };
FD99B94A0DD52EDC00FB1D6B /* SDL_audiotypecvt.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_audiotypecvt.c; sourceTree = "<group>"; };
FD99B94B0DD52EDC00FB1D6B /* SDL_mixer.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_mixer.c; sourceTree = "<group>"; };
FD99B9520DD52EDC00FB1D6B /* SDL_sysaudio.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_sysaudio.h; sourceTree = "<group>"; };
@@ -795,7 +794,6 @@
FD99B9440DD52EDC00FB1D6B /* SDL_audio.c */,
FD99B9450DD52EDC00FB1D6B /* SDL_audio_c.h */,
FD99B9460DD52EDC00FB1D6B /* SDL_audiocvt.c */,
FD99B9490DD52EDC00FB1D6B /* SDL_audiomem.h */,
FD99B94A0DD52EDC00FB1D6B /* SDL_audiotypecvt.c */,
FD99B94B0DD52EDC00FB1D6B /* SDL_mixer.c */,
FD99B9520DD52EDC00FB1D6B /* SDL_sysaudio.h */,
Xcode/SDL/SDL.xcodeproj/project.pbxproj
-8
View File
@@ -64,7 +64,6 @@
04BD002812E6671800899322 /* SDL_audiocvt.c in Sources */ = {isa = PBXBuildFile; fileRef = 04BDFDB612E6671700899322 /* SDL_audiocvt.c */; };
04BD002912E6671800899322 /* SDL_audiodev.c in Sources */ = {isa = PBXBuildFile; fileRef = 04BDFDB712E6671700899322 /* SDL_audiodev.c */; };
04BD002A12E6671800899322 /* SDL_audiodev_c.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDB812E6671700899322 /* SDL_audiodev_c.h */; };
04BD002B12E6671800899322 /* SDL_audiomem.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDB912E6671700899322 /* SDL_audiomem.h */; };
04BD002C12E6671800899322 /* SDL_audiotypecvt.c in Sources */ = {isa = PBXBuildFile; fileRef = 04BDFDBA12E6671700899322 /* SDL_audiotypecvt.c */; };
04BD002D12E6671800899322 /* SDL_mixer.c in Sources */ = {isa = PBXBuildFile; fileRef = 04BDFDBB12E6671700899322 /* SDL_mixer.c */; };
04BD003412E6671800899322 /* SDL_sysaudio.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDC212E6671700899322 /* SDL_sysaudio.h */; };
@@ -218,7 +217,6 @@
04BD024412E6671800899322 /* SDL_audiocvt.c in Sources */ = {isa = PBXBuildFile; fileRef = 04BDFDB612E6671700899322 /* SDL_audiocvt.c */; };
04BD024512E6671800899322 /* SDL_audiodev.c in Sources */ = {isa = PBXBuildFile; fileRef = 04BDFDB712E6671700899322 /* SDL_audiodev.c */; };
04BD024612E6671800899322 /* SDL_audiodev_c.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDB812E6671700899322 /* SDL_audiodev_c.h */; };
04BD024712E6671800899322 /* SDL_audiomem.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDB912E6671700899322 /* SDL_audiomem.h */; };
04BD024812E6671800899322 /* SDL_audiotypecvt.c in Sources */ = {isa = PBXBuildFile; fileRef = 04BDFDBA12E6671700899322 /* SDL_audiotypecvt.c */; };
04BD024912E6671800899322 /* SDL_mixer.c in Sources */ = {isa = PBXBuildFile; fileRef = 04BDFDBB12E6671700899322 /* SDL_mixer.c */; };
04BD025012E6671800899322 /* SDL_sysaudio.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDC212E6671700899322 /* SDL_sysaudio.h */; };
@@ -563,7 +561,6 @@
DB313F7617554B71006C0E22 /* SDL_coreaudio.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDA112E6671700899322 /* SDL_coreaudio.h */; };
DB313F7717554B71006C0E22 /* SDL_audio_c.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDB512E6671700899322 /* SDL_audio_c.h */; };
DB313F7817554B71006C0E22 /* SDL_audiodev_c.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDB812E6671700899322 /* SDL_audiodev_c.h */; };
DB313F7917554B71006C0E22 /* SDL_audiomem.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDB912E6671700899322 /* SDL_audiomem.h */; };
DB313F7A17554B71006C0E22 /* SDL_sysaudio.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDC212E6671700899322 /* SDL_sysaudio.h */; };
DB313F7B17554B71006C0E22 /* SDL_wave.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDC412E6671700899322 /* SDL_wave.h */; };
DB313F7C17554B71006C0E22 /* blank_cursor.h in Headers */ = {isa = PBXBuildFile; fileRef = 04BDFDD612E6671700899322 /* blank_cursor.h */; };
@@ -864,7 +861,6 @@
04BDFDB612E6671700899322 /* SDL_audiocvt.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_audiocvt.c; sourceTree = "<group>"; };
04BDFDB712E6671700899322 /* SDL_audiodev.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_audiodev.c; sourceTree = "<group>"; };
04BDFDB812E6671700899322 /* SDL_audiodev_c.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_audiodev_c.h; sourceTree = "<group>"; };
04BDFDB912E6671700899322 /* SDL_audiomem.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_audiomem.h; sourceTree = "<group>"; };
04BDFDBA12E6671700899322 /* SDL_audiotypecvt.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_audiotypecvt.c; sourceTree = "<group>"; };
04BDFDBB12E6671700899322 /* SDL_mixer.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = SDL_mixer.c; sourceTree = "<group>"; };
04BDFDC212E6671700899322 /* SDL_sysaudio.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SDL_sysaudio.h; sourceTree = "<group>"; };
@@ -1307,7 +1303,6 @@
04BDFDB612E6671700899322 /* SDL_audiocvt.c */,
04BDFDB712E6671700899322 /* SDL_audiodev.c */,
04BDFDB812E6671700899322 /* SDL_audiodev_c.h */,
04BDFDB912E6671700899322 /* SDL_audiomem.h */,
04BDFDBA12E6671700899322 /* SDL_audiotypecvt.c */,
04BDFDBB12E6671700899322 /* SDL_mixer.c */,
04BDFDC212E6671700899322 /* SDL_sysaudio.h */,
@@ -1840,7 +1835,6 @@
04BD001912E6671800899322 /* SDL_coreaudio.h in Headers */,
04BD002712E6671800899322 /* SDL_audio_c.h in Headers */,
04BD002A12E6671800899322 /* SDL_audiodev_c.h in Headers */,
04BD002B12E6671800899322 /* SDL_audiomem.h in Headers */,
04BD003412E6671800899322 /* SDL_sysaudio.h in Headers */,
04BD003612E6671800899322 /* SDL_wave.h in Headers */,
04BD004212E6671800899322 /* blank_cursor.h in Headers */,
@@ -1996,7 +1990,6 @@
04BD024312E6671800899322 /* SDL_audio_c.h in Headers */,
04BD024612E6671800899322 /* SDL_audiodev_c.h in Headers */,
AAC070FD195606770073DCDF /* SDL_opengles2_gl2.h in Headers */,
04BD024712E6671800899322 /* SDL_audiomem.h in Headers */,
04BD025012E6671800899322 /* SDL_sysaudio.h in Headers */,
04BD025212E6671800899322 /* SDL_wave.h in Headers */,
04BD025D12E6671800899322 /* blank_cursor.h in Headers */,
@@ -2151,7 +2144,6 @@
DB313F7717554B71006C0E22 /* SDL_audio_c.h in Headers */,
DB313F7817554B71006C0E22 /* SDL_audiodev_c.h in Headers */,
AAC070FE195606770073DCDF /* SDL_opengles2_gl2.h in Headers */,
DB313F7917554B71006C0E22 /* SDL_audiomem.h in Headers */,
DB313F7A17554B71006C0E22 /* SDL_sysaudio.h in Headers */,
DB313F7B17554B71006C0E22 /* SDL_wave.h in Headers */,
DB313F7C17554B71006C0E22 /* blank_cursor.h in Headers */,
android-project/AndroidManifest.xml
+3
View File
@@ -17,6 +17,9 @@
<!-- Allow writing to external storage -->
<uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE" />
<!-- if you want to capture audio, uncomment this. -->
<!-- <uses-permission android:name="android.permission.RECORD_AUDIO" /> -->
<!-- Create a Java class extending SDLActivity and place it in a
directory under src matching the package, e.g.
src/com/gamemaker/game/MyGame.java
android-project/src/org/libsdl/app/SDLActivity.java
+63 -2
View File
@@ -59,6 +59,7 @@ public class SDLActivity extends Activity {
// Audio
protected static AudioTrack mAudioTrack;
protected static AudioRecord mAudioRecord;
/**
* This method is called by SDL before loading the native shared libraries.
@@ -106,6 +107,7 @@ public class SDLActivity extends Activity {
mJoystickHandler = null;
mSDLThread = null;
mAudioTrack = null;
mAudioRecord = null;
mExitCalledFromJava = false;
mBrokenLibraries = false;
mIsPaused = false;
@@ -544,7 +546,7 @@ public class SDLActivity extends Activity {
/**
* This method is called by SDL using JNI.
*/
public static int audioInit(int sampleRate, boolean is16Bit, boolean isStereo, int desiredFrames) {
public static int audioOpen(int sampleRate, boolean is16Bit, boolean isStereo, int desiredFrames) {
int channelConfig = isStereo ? AudioFormat.CHANNEL_CONFIGURATION_STEREO : AudioFormat.CHANNEL_CONFIGURATION_MONO;
int audioFormat = is16Bit ? AudioFormat.ENCODING_PCM_16BIT : AudioFormat.ENCODING_PCM_8BIT;
int frameSize = (isStereo ? 2 : 1) * (is16Bit ? 2 : 1);
@@ -623,13 +625,72 @@ public class SDLActivity extends Activity {
/**
* This method is called by SDL using JNI.
*/
public static void audioQuit() {
public static int captureOpen(int sampleRate, boolean is16Bit, boolean isStereo, int desiredFrames) {
int channelConfig = isStereo ? AudioFormat.CHANNEL_CONFIGURATION_STEREO : AudioFormat.CHANNEL_CONFIGURATION_MONO;
int audioFormat = is16Bit ? AudioFormat.ENCODING_PCM_16BIT : AudioFormat.ENCODING_PCM_8BIT;
int frameSize = (isStereo ? 2 : 1) * (is16Bit ? 2 : 1);
Log.v(TAG, "SDL capture: wanted " + (isStereo ? "stereo" : "mono") + " " + (is16Bit ? "16-bit" : "8-bit") + " " + (sampleRate / 1000f) + "kHz, " + desiredFrames + " frames buffer");
// Let the user pick a larger buffer if they really want -- but ye
// gods they probably shouldn't, the minimums are horrifyingly high
// latency already
desiredFrames = Math.max(desiredFrames, (AudioRecord.getMinBufferSize(sampleRate, channelConfig, audioFormat) + frameSize - 1) / frameSize);
if (mAudioRecord == null) {
mAudioRecord = new AudioRecord(MediaRecorder.AudioSource.DEFAULT, sampleRate,
channelConfig, audioFormat, desiredFrames * frameSize);
// see notes about AudioTrack state in audioOpen(), above. Probably also applies here.
if (mAudioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
Log.e(TAG, "Failed during initialization of AudioRecord");
mAudioRecord.release();
mAudioRecord = null;
return -1;
}
mAudioRecord.startRecording();
}
Log.v(TAG, "SDL capture: got " + ((mAudioRecord.getChannelCount() >= 2) ? "stereo" : "mono") + " " + ((mAudioRecord.getAudioFormat() == AudioFormat.ENCODING_PCM_16BIT) ? "16-bit" : "8-bit") + " " + (mAudioRecord.getSampleRate() / 1000f) + "kHz, " + desiredFrames + " frames buffer");
return 0;
}
/** This method is called by SDL using JNI. */
public static int captureReadShortBuffer(short[] buffer, boolean blocking) {
// !!! FIXME: this is available in API Level 23. Until then, we always block. :(
//return mAudioRecord.read(buffer, 0, buffer.length, blocking ? AudioRecord.READ_BLOCKING : AudioRecord.READ_NON_BLOCKING);
return mAudioRecord.read(buffer, 0, buffer.length);
}
/** This method is called by SDL using JNI. */
public static int captureReadByteBuffer(byte[] buffer, boolean blocking) {
// !!! FIXME: this is available in API Level 23. Until then, we always block. :(
//return mAudioRecord.read(buffer, 0, buffer.length, blocking ? AudioRecord.READ_BLOCKING : AudioRecord.READ_NON_BLOCKING);
return mAudioRecord.read(buffer, 0, buffer.length);
}
/** This method is called by SDL using JNI. */
public static void audioClose() {
if (mAudioTrack != null) {
mAudioTrack.stop();
mAudioTrack.release();
mAudioTrack = null;
}
}
/** This method is called by SDL using JNI. */
public static void captureClose() {
if (mAudioRecord != null) {
mAudioRecord.stop();
mAudioRecord.release();
mAudioRecord = null;
}
}
// Input
/**
include/SDL_audio.h
+81 -14
View File
@@ -278,7 +278,8 @@ extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
* protect data structures that it accesses by calling SDL_LockAudio()
* and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL
* pointer here, and call SDL_QueueAudio() with some frequency, to queue
* more audio samples to be played.
* more audio samples to be played (or for capture devices, call
* SDL_DequeueAudio() with some frequency, to obtain audio samples).
* - \c desired->userdata is passed as the first parameter to your callback
* function. If you passed a NULL callback, this value is ignored.
*
@@ -482,6 +483,10 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
/**
* Queue more audio on non-callback devices.
*
* (If you are looking to retrieve queued audio from a non-callback capture
* device, you want SDL_DequeueAudio() instead. This will return -1 to
* signify an error if you use it with capture devices.)
*
* SDL offers two ways to feed audio to the device: you can either supply a
* callback that SDL triggers with some frequency to obtain more audio
* (pull method), or you can supply no callback, and then SDL will expect
@@ -516,21 +521,76 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
*/
extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
/**
* Dequeue more audio on non-callback devices.
*
* (If you are looking to queue audio for output on a non-callback playback
* device, you want SDL_QueueAudio() instead. This will always return 0
* if you use it with playback devices.)
*
* SDL offers two ways to retrieve audio from a capture device: you can
* either supply a callback that SDL triggers with some frequency as the
* device records more audio data, (push method), or you can supply no
* callback, and then SDL will expect you to retrieve data at regular
* intervals (pull method) with this function.
*
* There are no limits on the amount of data you can queue, short of
* exhaustion of address space. Data from the device will keep queuing as
* necessary without further intervention from you. This means you will
* eventually run out of memory if you aren't routinely dequeueing data.
*
* Capture devices will not queue data when paused; if you are expecting
* to not need captured audio for some length of time, use
* SDL_PauseAudioDevice() to stop the capture device from queueing more
* data. This can be useful during, say, level loading times. When
* unpaused, capture devices will start queueing data from that point,
* having flushed any capturable data available while paused.
*
* This function is thread-safe, but dequeueing from the same device from
* two threads at once does not promise which thread will dequeued data
* first.
*
* You may not dequeue audio from a device that is using an
* application-supplied callback; doing so returns an error. You have to use
* the audio callback, or dequeue audio with this function, but not both.
*
* You should not call SDL_LockAudio() on the device before queueing; SDL
* handles locking internally for this function.
*
* \param dev The device ID from which we will dequeue audio.
* \param data A pointer into where audio data should be copied.
* \param len The number of bytes (not samples!) to which (data) points.
* \return number of bytes dequeued, which could be less than requested.
*
* \sa SDL_GetQueuedAudioSize
* \sa SDL_ClearQueuedAudio
*/
extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
/**
* Get the number of bytes of still-queued audio.
*
* This is the number of bytes that have been queued for playback with
* SDL_QueueAudio(), but have not yet been sent to the hardware.
* For playback device:
*
* Once we've sent it to the hardware, this function can not decide the exact
* byte boundary of what has been played. It's possible that we just gave the
* hardware several kilobytes right before you called this function, but it
* hasn't played any of it yet, or maybe half of it, etc.
* This is the number of bytes that have been queued for playback with
* SDL_QueueAudio(), but have not yet been sent to the hardware. This
* number may shrink at any time, so this only informs of pending data.
*
* Once we've sent it to the hardware, this function can not decide the
* exact byte boundary of what has been played. It's possible that we just
* gave the hardware several kilobytes right before you called this
* function, but it hasn't played any of it yet, or maybe half of it, etc.
*
* For capture devices:
*
* This is the number of bytes that have been captured by the device and
* are waiting for you to dequeue. This number may grow at any time, so
* this only informs of the lower-bound of available data.
*
* You may not queue audio on a device that is using an application-supplied
* callback; calling this function on such a device always returns 0.
* You have to use the audio callback or queue audio with SDL_QueueAudio(),
* but not both.
* You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
* the audio callback, but not both.
*
* You should not call SDL_LockAudio() on the device before querying; SDL
* handles locking internally for this function.
@@ -544,10 +604,17 @@ extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *da
extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
/**
* Drop any queued audio data waiting to be sent to the hardware.
* Drop any queued audio data. For playback devices, this is any queued data
* still waiting to be submitted to the hardware. For capture devices, this
* is any data that was queued by the device that hasn't yet been dequeued by
* the application.
*
* Immediately after this call, SDL_GetQueuedAudioSize() will return 0 and
* the hardware will start playing silence if more audio isn't queued.
* Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
* playback devices, the hardware will start playing silence if more audio
* isn't queued. Unpaused capture devices will start filling the queue again
* as soon as they have more data available (which, depending on the state
* of the hardware and the thread, could be before this function call
* returns!).
*
* This will not prevent playback of queued audio that's already been sent
* to the hardware, as we can not undo that, so expect there to be some
@@ -557,8 +624,8 @@ extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
*
* You may not queue audio on a device that is using an application-supplied
* callback; calling this function on such a device is always a no-op.
* You have to use the audio callback or queue audio with SDL_QueueAudio(),
* but not both.
* You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
* the audio callback, but not both.
*
* You should not call SDL_LockAudio() on the device before clearing the
* queue; SDL handles locking internally for this function.
premake/VisualC/VS2008/SDL2/SDL2.vcproj
-4
View File
@@ -418,10 +418,6 @@
RelativePath="..\..\..\..\src\audio\SDL_audiodev_c.h"
>
</File>
<File
RelativePath="..\..\..\..\src\audio\SDL_audiomem.h"
>
</File>
<File
RelativePath="..\..\..\..\src\audio\SDL_audiotypecvt.c"
>
premake/VisualC/VS2010/SDL2/SDL2.vcxproj
-1
View File
@@ -113,7 +113,6 @@
<ClInclude Include="..\..\..\..\src\SDL_assert_c.h" />
<ClInclude Include="..\..\..\..\src\SDL_error_c.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_sysaudio.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_wave.h" />
premake/VisualC/VS2010/SDL2/SDL2.vcxproj.filters
-3
View File
@@ -123,9 +123,6 @@
<ClInclude Include="..\..\..\..\src\audio\SDL_audiodev_c.h">
<Filter>src\audio</Filter>
</ClInclude>
<ClInclude Include="..\..\..\..\src\audio\SDL_audiomem.h">
<Filter>src\audio</Filter>
</ClInclude>
<ClInclude Include="..\..\..\..\src\audio\SDL_audio_c.h">
<Filter>src\audio</Filter>
</ClInclude>
premake/VisualC/VS2012/SDL2/SDL2.vcxproj
-1
View File
@@ -115,7 +115,6 @@
<ClInclude Include="..\..\..\..\src\SDL_assert_c.h" />
<ClInclude Include="..\..\..\..\src\SDL_error_c.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_audiodev_c.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_audiomem.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_audio_c.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_sysaudio.h" />
<ClInclude Include="..\..\..\..\src\audio\SDL_wave.h" />
premake/VisualC/VS2012/SDL2/SDL2.vcxproj.filters
-3
View File
@@ -123,9 +123,6 @@
<ClInclude Include="..\..\..\..\src\audio\SDL_audiodev_c.h">
<Filter>src\audio</Filter>
</ClInclude>
<ClInclude Include="..\..\..\..\src\audio\SDL_audiomem.h">
<Filter>src\audio</Filter>
</ClInclude>
<ClInclude Include="..\..\..\..\src\audio\SDL_audio_c.h">
<Filter>src\audio</Filter>
</ClInclude>
premake/Xcode-iOS/SDL2/SDL2.xcodeproj/project.pbxproj
-2
View File
@@ -127,7 +127,6 @@
4DBB70D75469728B342373E8 /* SDL_audiocvt.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiocvt.c"; path = "../../../src/audio/SDL_audiocvt.c"; sourceTree = "<group>"; };
48886D482B5239D2429E422D /* SDL_audiodev.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiodev.c"; path = "../../../src/audio/SDL_audiodev.c"; sourceTree = "<group>"; };
227E138737440F101016545F /* SDL_audiodev_c.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_audiodev_c.h"; path = "../../../src/audio/SDL_audiodev_c.h"; sourceTree = "<group>"; };
5C3C744F22823D470BED10D6 /* SDL_audiomem.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_audiomem.h"; path = "../../../src/audio/SDL_audiomem.h"; sourceTree = "<group>"; };
0F175E65628D4137386B7A6D /* SDL_audiotypecvt.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiotypecvt.c"; path = "../../../src/audio/SDL_audiotypecvt.c"; sourceTree = "<group>"; };
77537CFB490A3599736F3830 /* SDL_mixer.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_mixer.c"; path = "../../../src/audio/SDL_mixer.c"; sourceTree = "<group>"; };
591062475F93492D625F7D3B /* SDL_sysaudio.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_sysaudio.h"; path = "../../../src/audio/SDL_sysaudio.h"; sourceTree = "<group>"; };
@@ -369,7 +368,6 @@
4DBB70D75469728B342373E8 /* SDL_audiocvt.c */,
48886D482B5239D2429E422D /* SDL_audiodev.c */,
227E138737440F101016545F /* SDL_audiodev_c.h */,
5C3C744F22823D470BED10D6 /* SDL_audiomem.h */,
0F175E65628D4137386B7A6D /* SDL_audiotypecvt.c */,
77537CFB490A3599736F3830 /* SDL_mixer.c */,
591062475F93492D625F7D3B /* SDL_sysaudio.h */,
premake/Xcode/Xcode3/SDL2/SDL2.xcodeproj/project.pbxproj
-2
View File
@@ -146,7 +146,6 @@
2BA37BD372FE166821D80A1E /* SDL_audiocvt.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiocvt.c"; path = "../../../../src/audio/SDL_audiocvt.c"; sourceTree = "<group>"; };
5D2936CF698D392735D76E9E /* SDL_audiodev.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiodev.c"; path = "../../../../src/audio/SDL_audiodev.c"; sourceTree = "<group>"; };
1F255A29771744AC1DFE48A0 /* SDL_audiodev_c.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_audiodev_c.h"; path = "../../../../src/audio/SDL_audiodev_c.h"; sourceTree = "<group>"; };
14AA3D784A5D4B873D657338 /* SDL_audiomem.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_audiomem.h"; path = "../../../../src/audio/SDL_audiomem.h"; sourceTree = "<group>"; };
76263CFA4F4A3E8E74966406 /* SDL_audiotypecvt.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiotypecvt.c"; path = "../../../../src/audio/SDL_audiotypecvt.c"; sourceTree = "<group>"; };
748562A8151756FF3FE91679 /* SDL_mixer.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_mixer.c"; path = "../../../../src/audio/SDL_mixer.c"; sourceTree = "<group>"; };
7B696A2B3C9847A40FD30FA2 /* SDL_sysaudio.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_sysaudio.h"; path = "../../../../src/audio/SDL_sysaudio.h"; sourceTree = "<group>"; };
@@ -425,7 +424,6 @@
2BA37BD372FE166821D80A1E /* SDL_audiocvt.c */,
5D2936CF698D392735D76E9E /* SDL_audiodev.c */,
1F255A29771744AC1DFE48A0 /* SDL_audiodev_c.h */,
14AA3D784A5D4B873D657338 /* SDL_audiomem.h */,
76263CFA4F4A3E8E74966406 /* SDL_audiotypecvt.c */,
748562A8151756FF3FE91679 /* SDL_mixer.c */,
7B696A2B3C9847A40FD30FA2 /* SDL_sysaudio.h */,
premake/Xcode/Xcode4/SDL2/SDL2.xcodeproj/project.pbxproj
-2
View File
@@ -146,7 +146,6 @@
07B907294E82663A7E91738C /* SDL_audiocvt.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiocvt.c"; path = "../../../../src/audio/SDL_audiocvt.c"; sourceTree = "<group>"; };
5AAD4B726237251050431873 /* SDL_audiodev.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiodev.c"; path = "../../../../src/audio/SDL_audiodev.c"; sourceTree = "<group>"; };
15895798549516351860492E /* SDL_audiodev_c.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_audiodev_c.h"; path = "../../../../src/audio/SDL_audiodev_c.h"; sourceTree = "<group>"; };
0D3062CE47BF5D5934AB598D /* SDL_audiomem.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_audiomem.h"; path = "../../../../src/audio/SDL_audiomem.h"; sourceTree = "<group>"; };
5B0759ED16B35B9A6B027892 /* SDL_audiotypecvt.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_audiotypecvt.c"; path = "../../../../src/audio/SDL_audiotypecvt.c"; sourceTree = "<group>"; };
2B8C7A19218A1FFC6D376B1D /* SDL_mixer.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; name = "SDL_mixer.c"; path = "../../../../src/audio/SDL_mixer.c"; sourceTree = "<group>"; };
09E4653E4CD964410C0E71BA /* SDL_sysaudio.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = "SDL_sysaudio.h"; path = "../../../../src/audio/SDL_sysaudio.h"; sourceTree = "<group>"; };
@@ -425,7 +424,6 @@
07B907294E82663A7E91738C /* SDL_audiocvt.c */,
5AAD4B726237251050431873 /* SDL_audiodev.c */,
15895798549516351860492E /* SDL_audiodev_c.h */,
0D3062CE47BF5D5934AB598D /* SDL_audiomem.h */,
5B0759ED16B35B9A6B027892 /* SDL_audiotypecvt.c */,
2B8C7A19218A1FFC6D376B1D /* SDL_mixer.c */,
09E4653E4CD964410C0E71BA /* SDL_sysaudio.h */,
src/audio/SDL_audio.c
+323 -144
View File
File diff suppressed because it is too large Load Diff
src/audio/SDL_audio_c.h
-3
View File
@@ -29,9 +29,6 @@ extern SDL_AudioFormat SDL_NextAudioFormat(void);
/* Function to calculate the size and silence for a SDL_AudioSpec */
extern void SDL_CalculateAudioSpec(SDL_AudioSpec * spec);
/* The actual mixing thread function */
extern int SDLCALL SDL_RunAudio(void *audiop);
/* this is used internally to access some autogenerated code. */
typedef struct
{
src/audio/SDL_audiomem.h
-25
View File
@@ -1,25 +0,0 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../SDL_internal.h"
#define SDL_AllocAudioMem SDL_malloc
#define SDL_FreeAudioMem SDL_free
/* vi: set ts=4 sw=4 expandtab: */
src/audio/SDL_sysaudio.h
+11 -7
View File
@@ -26,6 +26,10 @@
#include "SDL_mutex.h"
#include "SDL_thread.h"
/* !!! FIXME: These are wordy and unlocalized... */
#define DEFAULT_OUTPUT_DEVNAME "System audio output device"
#define DEFAULT_INPUT_DEVNAME "System audio capture device"
/* The SDL audio driver */
typedef struct SDL_AudioDevice SDL_AudioDevice;
#define _THIS SDL_AudioDevice *_this
@@ -76,6 +80,8 @@ typedef struct SDL_AudioDriverImpl
int (*GetPendingBytes) (_THIS);
Uint8 *(*GetDeviceBuf) (_THIS);
void (*WaitDone) (_THIS);
int (*CaptureFromDevice) (_THIS, void *buffer, int buflen);
void (*FlushCapture) (_THIS);
void (*CloseDevice) (_THIS);
void (*LockDevice) (_THIS);
void (*UnlockDevice) (_THIS);
@@ -90,7 +96,7 @@ typedef struct SDL_AudioDriverImpl
int SkipMixerLock; /* !!! FIXME: do we need this anymore? */
int HasCaptureSupport;
int OnlyHasDefaultOutputDevice;
int OnlyHasDefaultInputDevice;
int OnlyHasDefaultCaptureDevice;
int AllowsArbitraryDeviceNames;
} SDL_AudioDriverImpl;
@@ -157,12 +163,10 @@ struct SDL_AudioDevice
SDL_AudioStreamer streamer;
/* Current state flags */
/* !!! FIXME: should be SDL_bool */
int iscapture;
int enabled; /* true if device is functioning and connected. */
int shutdown; /* true if we are signaling the play thread to end. */
int paused;
int opened;
SDL_atomic_t shutdown; /* true if we are signaling the play thread to end. */
SDL_atomic_t enabled; /* true if device is functioning and connected. */
SDL_atomic_t paused;
SDL_bool iscapture;
/* Fake audio buffer for when the audio hardware is busy */
Uint8 *fake_stream;
src/audio/SDL_wave.c
+1 -1
View File
@@ -504,7 +504,7 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
was_error = 1;
goto done;
}
SDL_memset(spec, 0, (sizeof *spec));
SDL_zerop(spec);
spec->freq = SDL_SwapLE32(format->frequency);
if (IEEE_float_encoded) {
src/audio/alsa/SDL_alsa_audio.c
+101 -59
View File
@@ -32,7 +32,6 @@
#include "SDL_assert.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_alsa_audio.h"
@@ -43,8 +42,10 @@
static int (*ALSA_snd_pcm_open)
(snd_pcm_t **, const char *, snd_pcm_stream_t, int);
static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm);
static snd_pcm_sframes_t(*ALSA_snd_pcm_writei)
static snd_pcm_sframes_t (*ALSA_snd_pcm_writei)
(snd_pcm_t *, const void *, snd_pcm_uframes_t);
static snd_pcm_sframes_t (*ALSA_snd_pcm_readi)
(snd_pcm_t *, void *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_recover) (snd_pcm_t *, int, int);
static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *);
static int (*ALSA_snd_pcm_drain) (snd_pcm_t *);
@@ -86,6 +87,7 @@ static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int);
static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int);
static int (*ALSA_snd_pcm_sw_params_set_avail_min)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_reset)(snd_pcm_t *);
static int (*ALSA_snd_device_name_hint) (int, const char *, void ***);
static char* (*ALSA_snd_device_name_get_hint) (const void *, const char *);
static int (*ALSA_snd_device_name_free_hint) (void **);
@@ -122,6 +124,7 @@ load_alsa_syms(void)
SDL_ALSA_SYM(snd_pcm_open);
SDL_ALSA_SYM(snd_pcm_close);
SDL_ALSA_SYM(snd_pcm_writei);
SDL_ALSA_SYM(snd_pcm_readi);
SDL_ALSA_SYM(snd_pcm_recover);
SDL_ALSA_SYM(snd_pcm_prepare);
SDL_ALSA_SYM(snd_pcm_drain);
@@ -148,6 +151,7 @@ load_alsa_syms(void)
SDL_ALSA_SYM(snd_pcm_nonblock);
SDL_ALSA_SYM(snd_pcm_wait);
SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min);
SDL_ALSA_SYM(snd_pcm_reset);
SDL_ALSA_SYM(snd_device_name_hint);
SDL_ALSA_SYM(snd_device_name_get_hint);
SDL_ALSA_SYM(snd_device_name_free_hint);
@@ -242,37 +246,37 @@ ALSA_WaitDevice(_THIS)
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
#define SWIZ6(T) \
T *ptr = (T *) this->hidden->mixbuf; \
#define SWIZ6(T, buf, numframes) \
T *ptr = (T *) buf; \
Uint32 i; \
for (i = 0; i < this->spec.samples; i++, ptr += 6) { \
for (i = 0; i < numframes; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
}
static SDL_INLINE void
swizzle_alsa_channels_6_64bit(_THIS)
swizzle_alsa_channels_6_64bit(void *buffer, Uint32 bufferlen)
{
SWIZ6(Uint64);
SWIZ6(Uint64, buffer, bufferlen);
}
static SDL_INLINE void
swizzle_alsa_channels_6_32bit(_THIS)
swizzle_alsa_channels_6_32bit(void *buffer, Uint32 bufferlen)
{
SWIZ6(Uint32);
SWIZ6(Uint32, buffer, bufferlen);
}
static SDL_INLINE void
swizzle_alsa_channels_6_16bit(_THIS)
swizzle_alsa_channels_6_16bit(void *buffer, Uint32 bufferlen)
{
SWIZ6(Uint16);
SWIZ6(Uint16, buffer, bufferlen);
}
static SDL_INLINE void
swizzle_alsa_channels_6_8bit(_THIS)
swizzle_alsa_channels_6_8bit(void *buffer, Uint32 bufferlen)
{
SWIZ6(Uint8);
SWIZ6(Uint8, buffer, bufferlen);
}
#undef SWIZ6
@@ -283,18 +287,16 @@ swizzle_alsa_channels_6_8bit(_THIS)
* channels from Windows/Mac order to the format alsalib will want.
*/
static SDL_INLINE void
swizzle_alsa_channels(_THIS)
swizzle_alsa_channels(_THIS, void *buffer, Uint32 bufferlen)
{
if (this->spec.channels == 6) {
const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
if (fmtsize == 16)
swizzle_alsa_channels_6_16bit(this);
else if (fmtsize == 8)
swizzle_alsa_channels_6_8bit(this);
else if (fmtsize == 32)
swizzle_alsa_channels_6_32bit(this);
else if (fmtsize == 64)
swizzle_alsa_channels_6_64bit(this);
switch (SDL_AUDIO_BITSIZE(this->spec.format)) {
case 8: swizzle_alsa_channels_6_8bit(buffer, bufferlen); break;
case 16: swizzle_alsa_channels_6_16bit(buffer, bufferlen); break;
case 32: swizzle_alsa_channels_6_32bit(buffer, bufferlen); break;
case 64: swizzle_alsa_channels_6_64bit(buffer, bufferlen); break;
default: SDL_assert(!"unhandled bitsize"); break;
}
}
/* !!! FIXME: update this for 7.1 if needed, later. */
@@ -304,19 +306,18 @@ swizzle_alsa_channels(_THIS)
static void
ALSA_PlayDevice(_THIS)
{
int status;
const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;
const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) *
const int frame_size = (((int) SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);
swizzle_alsa_channels(this);
swizzle_alsa_channels(this, this->hidden->mixbuf, frames_left);
while ( frames_left > 0 && this->enabled ) {
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
/* !!! FIXME: This works, but needs more testing before going live */
/* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
int status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
if (status == -EAGAIN) {
@@ -346,20 +347,66 @@ ALSA_GetDeviceBuf(_THIS)
return (this->hidden->mixbuf);
}
static int
ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
Uint8 *sample_buf = (Uint8 *) buffer;
const int frame_size = (((int) SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
const int total_frames = buflen / frame_size;
snd_pcm_uframes_t frames_left = total_frames;
SDL_assert((buflen % frame_size) == 0);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
/* !!! FIXME: This works, but needs more testing before going live */
/* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
int status = ALSA_snd_pcm_readi(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
/*printf("ALSA: capture error %d\n", status);*/
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA read failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
return -1;
}
continue;
}
/*printf("ALSA: captured %d bytes\n", status * frame_size);*/
sample_buf += status * frame_size;
frames_left -= status;
}
swizzle_alsa_channels(this, buffer, total_frames - frames_left);
return (total_frames - frames_left) * frame_size;
}
static void
ALSA_FlushCapture(_THIS)
{
ALSA_snd_pcm_reset(this->hidden->pcm_handle);
}
static void
ALSA_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->pcm_handle) {
ALSA_snd_pcm_drain(this->hidden->pcm_handle);
ALSA_snd_pcm_close(this->hidden->pcm_handle);
this->hidden->pcm_handle = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
if (this->hidden->pcm_handle) {
ALSA_snd_pcm_drain(this->hidden->pcm_handle);
ALSA_snd_pcm_close(this->hidden->pcm_handle);
}
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
static int
@@ -492,16 +539,16 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
/* Open the audio device */
/* Name of device should depend on # channels in spec */
status = ALSA_snd_pcm_open(&pcm_handle,
get_audio_device(handle, this->spec.channels),
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
get_audio_device(handle, this->spec.channels),
iscapture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't open audio device: %s",
ALSA_snd_strerror(status));
}
@@ -512,7 +559,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
snd_pcm_hw_params_alloca(&hwparams);
status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't get hardware config: %s",
ALSA_snd_strerror(status));
}
@@ -521,7 +567,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set interleaved access: %s",
ALSA_snd_strerror(status));
}
@@ -575,7 +620,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
}
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
@@ -587,7 +631,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (status < 0) {
status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set audio channels");
}
this->spec.channels = channels;
@@ -598,7 +641,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
&rate, NULL);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set audio frequency: %s",
ALSA_snd_strerror(status));
}
@@ -610,7 +652,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Failed to set desired buffer size, do the best you can... */
status = ALSA_set_period_size(this, hwparams, 1);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
}
}
@@ -618,26 +659,22 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
snd_pcm_sw_params_alloca(&swparams);
status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't get software config: %s",
ALSA_snd_strerror(status));
}
status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, this->spec.samples);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("Couldn't set minimum available samples: %s",
ALSA_snd_strerror(status));
}
status =
ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set start threshold: %s",
ALSA_snd_strerror(status));
}
status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("Couldn't set software audio parameters: %s",
ALSA_snd_strerror(status));
}
@@ -646,13 +683,14 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ALSA_CloseDevice(this);
return SDL_OutOfMemory();
if (!iscapture) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
/* Switch to blocking mode for playback */
ALSA_snd_pcm_nonblock(pcm_handle, 0);
@@ -866,6 +904,10 @@ ALSA_Init(SDL_AudioDriverImpl * impl)
impl->PlayDevice = ALSA_PlayDevice;
impl->CloseDevice = ALSA_CloseDevice;
impl->Deinitialize = ALSA_Deinitialize;
impl->CaptureFromDevice = ALSA_CaptureFromDevice;
impl->FlushCapture = ALSA_FlushCapture;
impl->HasCaptureSupport = SDL_TRUE;
return 1; /* this audio target is available. */
}
src/audio/android/SDL_androidaudio.c
+73 -36
View File
@@ -24,6 +24,7 @@
/* Output audio to Android */
#include "SDL_assert.h"
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_androidaudio.h"
@@ -33,23 +34,22 @@
#include <android/log.h>
static SDL_AudioDevice* audioDevice = NULL;
static SDL_AudioDevice* captureDevice = NULL;
static int
AndroidAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
ANDROIDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format;
SDL_assert((captureDevice == NULL) || !iscapture);
SDL_assert((audioDevice == NULL) || iscapture);
if (iscapture) {
/* TODO: implement capture */
return SDL_SetError("Capture not supported on Android");
captureDevice = this;
} else {
audioDevice = this;
}
if (audioDevice != NULL) {
return SDL_SetError("Only one audio device at a time please!");
}
audioDevice = this;
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
@@ -82,100 +82,137 @@ AndroidAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
this->spec.freq = 48000;
}
/* TODO: pass in/return a (Java) device ID, also whether we're opening for input or output */
this->spec.samples = Android_JNI_OpenAudioDevice(this->spec.freq, this->spec.format == AUDIO_U8 ? 0 : 1, this->spec.channels, this->spec.samples);
SDL_CalculateAudioSpec(&this->spec);
/* TODO: pass in/return a (Java) device ID */
this->spec.samples = Android_JNI_OpenAudioDevice(iscapture, this->spec.freq, this->spec.format == AUDIO_U8 ? 0 : 1, this->spec.channels, this->spec.samples);
if (this->spec.samples == 0) {
/* Init failed? */
return SDL_SetError("Java-side initialization failed!");
}
SDL_CalculateAudioSpec(&this->spec);
return 0;
}
static void
AndroidAUD_PlayDevice(_THIS)
ANDROIDAUDIO_PlayDevice(_THIS)
{
Android_JNI_WriteAudioBuffer();
}
static Uint8 *
AndroidAUD_GetDeviceBuf(_THIS)
ANDROIDAUDIO_GetDeviceBuf(_THIS)
{
return Android_JNI_GetAudioBuffer();
}
static int
ANDROIDAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
return Android_JNI_CaptureAudioBuffer(buffer, buflen);
}
static void
AndroidAUD_CloseDevice(_THIS)
ANDROIDAUDIO_FlushCapture(_THIS)
{
Android_JNI_FlushCapturedAudio();
}
static void
ANDROIDAUDIO_CloseDevice(_THIS)
{
/* At this point SDL_CloseAudioDevice via close_audio_device took care of terminating the audio thread
so it's safe to terminate the Java side buffer and AudioTrack
*/
Android_JNI_CloseAudioDevice();
if (audioDevice == this) {
if (audioDevice->hidden != NULL) {
SDL_free(this->hidden);
this->hidden = NULL;
}
Android_JNI_CloseAudioDevice(this->iscapture);
if (this->iscapture) {
SDL_assert(captureDevice == this);
captureDevice = NULL;
} else {
SDL_assert(audioDevice == this);
audioDevice = NULL;
}
SDL_free(this->hidden);
}
static int
AndroidAUD_Init(SDL_AudioDriverImpl * impl)
ANDROIDAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = AndroidAUD_OpenDevice;
impl->PlayDevice = AndroidAUD_PlayDevice;
impl->GetDeviceBuf = AndroidAUD_GetDeviceBuf;
impl->CloseDevice = AndroidAUD_CloseDevice;
impl->OpenDevice = ANDROIDAUDIO_OpenDevice;
impl->PlayDevice = ANDROIDAUDIO_PlayDevice;
impl->GetDeviceBuf = ANDROIDAUDIO_GetDeviceBuf;
impl->CloseDevice = ANDROIDAUDIO_CloseDevice;
impl->CaptureFromDevice = ANDROIDAUDIO_CaptureFromDevice;
impl->FlushCapture = ANDROIDAUDIO_FlushCapture;
/* and the capabilities */
impl->HasCaptureSupport = 0; /* TODO */
impl->HasCaptureSupport = SDL_TRUE;
impl->OnlyHasDefaultOutputDevice = 1;
impl->OnlyHasDefaultInputDevice = 1;
impl->OnlyHasDefaultCaptureDevice = 1;
return 1; /* this audio target is available. */
}
AudioBootStrap ANDROIDAUD_bootstrap = {
"android", "SDL Android audio driver", AndroidAUD_Init, 0
AudioBootStrap ANDROIDAUDIO_bootstrap = {
"android", "SDL Android audio driver", ANDROIDAUDIO_Init, 0
};
/* Pause (block) all non already paused audio devices by taking their mixer lock */
void AndroidAUD_PauseDevices(void)
void ANDROIDAUDIO_PauseDevices(void)
{
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if(audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
if (audioDevice->paused) {
if (SDL_AtomicGet(&audioDevice->paused)) {
/* The device is already paused, leave it alone */
private->resume = SDL_FALSE;
}
else {
SDL_LockMutex(audioDevice->mixer_lock);
audioDevice->paused = SDL_TRUE;
SDL_AtomicSet(&audioDevice->paused, 1);
private->resume = SDL_TRUE;
}
}
if(captureDevice != NULL && captureDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) captureDevice->hidden;
if (SDL_AtomicGet(&captureDevice->paused)) {
/* The device is already paused, leave it alone */
private->resume = SDL_FALSE;
}
else {
SDL_LockMutex(captureDevice->mixer_lock);
SDL_AtomicSet(&captureDevice->paused, 1);
private->resume = SDL_TRUE;
}
}
}
/* Resume (unblock) all non already paused audio devices by releasing their mixer lock */
void AndroidAUD_ResumeDevices(void)
void ANDROIDAUDIO_ResumeDevices(void)
{
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if(audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
if (private->resume) {
audioDevice->paused = SDL_FALSE;
SDL_AtomicSet(&audioDevice->paused, 0);
private->resume = SDL_FALSE;
SDL_UnlockMutex(audioDevice->mixer_lock);
}
}
if(captureDevice != NULL && captureDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) captureDevice->hidden;
if (private->resume) {
SDL_AtomicSet(&captureDevice->paused, 0);
private->resume = SDL_FALSE;
SDL_UnlockMutex(captureDevice->mixer_lock);
}
}
}
src/audio/android/SDL_androidaudio.h
-2
View File
@@ -34,8 +34,6 @@ struct SDL_PrivateAudioData
int resume;
};
static void AndroidAUD_CloseDevice(_THIS);
#endif /* _SDL_androidaudio_h */
/* vi: set ts=4 sw=4 expandtab: */
src/audio/arts/SDL_artsaudio.c
+7 -18
View File
@@ -32,7 +32,6 @@
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_artsaudio.h"
@@ -203,17 +202,12 @@ ARTS_GetDeviceBuf(_THIS)
static void
ARTS_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->stream) {
SDL_NAME(arts_close_stream) (this->hidden->stream);
this->hidden->stream = 0;
}
SDL_NAME(arts_free) ();
SDL_free(this->hidden);
this->hidden = NULL;
if (this->hidden->stream) {
SDL_NAME(arts_close_stream) (this->hidden->stream);
}
SDL_NAME(arts_free) ();
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
static int
@@ -241,7 +235,7 @@ ARTS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
@@ -267,19 +261,16 @@ ARTS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
}
if (format == 0) {
ARTS_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
if ((rc = SDL_NAME(arts_init) ()) != 0) {
ARTS_CloseDevice(this);
return SDL_SetError("Unable to initialize ARTS: %s",
SDL_NAME(arts_error_text) (rc));
}
if (!ARTS_Suspend()) {
ARTS_CloseDevice(this);
return SDL_SetError("ARTS can not open audio device");
}
@@ -297,7 +288,6 @@ ARTS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Determine the power of two of the fragment size */
for (frag_spec = 0; (0x01 << frag_spec) < this->spec.size; ++frag_spec);
if ((0x01 << frag_spec) != this->spec.size) {
ARTS_CloseDevice(this);
return SDL_SetError("Fragment size must be a power of two");
}
frag_spec |= 0x00020000; /* two fragments, for low latency */
@@ -316,9 +306,8 @@ ARTS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ARTS_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
src/audio/bsd/SDL_bsdaudio.c
+110 -55
View File
@@ -38,7 +38,6 @@
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_bsdaudio.h"
@@ -63,13 +62,17 @@ BSDAUDIO_Status(_THIS)
#ifdef DEBUG_AUDIO
/* *INDENT-OFF* */
audio_info_t info;
const audio_prinfo *prinfo;
if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
fprintf(stderr, "AUDIO_GETINFO failed.\n");
return;
}
prinfo = this->iscapture ? &info.play : &info.record;
fprintf(stderr, "\n"
"[play/record info]\n"
"[%s info]\n"
"buffer size : %d bytes\n"
"sample rate : %i Hz\n"
"channels : %i\n"
@@ -83,18 +86,19 @@ BSDAUDIO_Status(_THIS)
"waiting : %s\n"
"active : %s\n"
"",
info.play.buffer_size,
info.play.sample_rate,
info.play.channels,
info.play.precision,
info.play.encoding,
info.play.seek,
info.play.samples,
info.play.eof,
info.play.pause ? "yes" : "no",
info.play.error ? "yes" : "no",
info.play.waiting ? "yes" : "no",
info.play.active ? "yes" : "no");
this->iscapture ? "record" : "play",
prinfo->buffer_size,
prinfo->sample_rate,
prinfo->channels,
prinfo->precision,
prinfo->encoding,
prinfo->seek,
prinfo->samples,
prinfo->eof,
prinfo->pause ? "yes" : "no",
prinfo->error ? "yes" : "no",
prinfo->waiting ? "yes" : "no",
prinfo->active ? "yes" : "no");
fprintf(stderr, "\n"
"[audio info]\n"
@@ -182,11 +186,15 @@ BSDAUDIO_PlayDevice(_THIS)
break;
}
if (p < written
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
if (p < this->hidden->mixlen
|| ((written < 0) && ((errno == 0) || (errno == EAGAIN)))) {
SDL_Delay(1); /* Let a little CPU time go by */
}
} while (p < written);
} while (p < this->hidden->mixlen);
/* If timer synchronization is enabled, set the next write frame */
if (this->hidden->frame_ticks) {
@@ -197,9 +205,6 @@ BSDAUDIO_PlayDevice(_THIS)
if (written < 0) {
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
}
static Uint8 *
@@ -208,27 +213,74 @@ BSDAUDIO_GetDeviceBuf(_THIS)
return (this->hidden->mixbuf);
}
static int
BSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
{
Uint8 *buffer = (Uint8 *) _buffer;
int br, p = 0;
/* Write the audio data, checking for EAGAIN on broken audio drivers */
do {
br = read(this->hidden->audio_fd, buffer + p, buflen - p);
if (br > 0)
p += br;
if (br == -1 && errno != 0 && errno != EAGAIN && errno != EINTR) {
/* Non recoverable error has occurred. It should be reported!!! */
perror("audio");
return p ? p : -1;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Captured %d bytes of audio data\n", br);
#endif
if (p < buflen
|| ((br < 0) && ((errno == 0) || (errno == EAGAIN)))) {
SDL_Delay(1); /* Let a little CPU time go by */
}
} while (p < buflen);
}
static void
BSDAUDIO_FlushCapture(_THIS)
{
audio_info_t info;
size_t remain;
Uint8 buf[512];
if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
return; /* oh well. */
}
remain = (size_t) (info.record.samples * (SDL_AUDIO_BITSIZE(this->spec.format) / 8));
while (remain > 0) {
const size_t len = SDL_min(sizeof (buf), remain);
const int br = read(this->hidden->audio_fd, buf, len);
if (br <= 0) {
return; /* oh well. */
}
remain -= br;
}
}
static void
BSDAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
this->hidden->audio_fd = -1;
}
SDL_free(this->hidden);
this->hidden = NULL;
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
}
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
static int
BSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
const int flags = iscapture ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT;
SDL_AudioFormat format = 0;
audio_info_t info;
audio_prinfo *prinfo = iscapture ? &info.play : &info.record;
/* We don't care what the devname is...we'll try to open anything. */
/* ...but default to first name in the list... */
@@ -245,7 +297,7 @@ BSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
/* Open the audio device */
this->hidden->audio_fd = open(devname, flags, 0);
@@ -259,9 +311,8 @@ BSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_CalculateAudioSpec(&this->spec);
/* Set to play mode */
info.mode = AUMODE_PLAY;
info.mode = iscapture ? AUMODE_RECORD : AUMODE_PLAY;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) < 0) {
BSDAUDIO_CloseDevice(this);
return SDL_SetError("Couldn't put device into play mode");
}
@@ -270,28 +321,28 @@ BSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
format; format = SDL_NextAudioFormat()) {
switch (format) {
case AUDIO_U8:
info.play.encoding = AUDIO_ENCODING_ULINEAR;
info.play.precision = 8;
prinfo->encoding = AUDIO_ENCODING_ULINEAR;
prinfo->precision = 8;
break;
case AUDIO_S8:
info.play.encoding = AUDIO_ENCODING_SLINEAR;
info.play.precision = 8;
prinfo->encoding = AUDIO_ENCODING_SLINEAR;
prinfo->precision = 8;
break;
case AUDIO_S16LSB:
info.play.encoding = AUDIO_ENCODING_SLINEAR_LE;
info.play.precision = 16;
prinfo->encoding = AUDIO_ENCODING_SLINEAR_LE;
prinfo->precision = 16;
break;
case AUDIO_S16MSB:
info.play.encoding = AUDIO_ENCODING_SLINEAR_BE;
info.play.precision = 16;
prinfo->encoding = AUDIO_ENCODING_SLINEAR_BE;
prinfo->precision = 16;
break;
case AUDIO_U16LSB:
info.play.encoding = AUDIO_ENCODING_ULINEAR_LE;
info.play.precision = 16;
prinfo->encoding = AUDIO_ENCODING_ULINEAR_LE;
prinfo->precision = 16;
break;
case AUDIO_U16MSB:
info.play.encoding = AUDIO_ENCODING_ULINEAR_BE;
info.play.precision = 16;
prinfo->encoding = AUDIO_ENCODING_ULINEAR_BE;
prinfo->precision = 16;
break;
default:
continue;
@@ -303,33 +354,34 @@ BSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
if (!format) {
BSDAUDIO_CloseDevice(this);
return SDL_SetError("No supported encoding for 0x%x", this->spec.format);
}
this->spec.format = format;
AUDIO_INITINFO(&info);
info.play.channels = this->spec.channels;
prinfo->channels = this->spec.channels;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == -1) {
this->spec.channels = 1;
}
AUDIO_INITINFO(&info);
info.play.sample_rate = this->spec.freq;
prinfo->sample_rate = this->spec.freq;
info.blocksize = this->spec.size;
info.hiwat = 5;
info.lowat = 3;
(void) ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info);
(void) ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info);
this->spec.freq = info.play.sample_rate;
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
BSDAUDIO_CloseDevice(this);
return SDL_OutOfMemory();
this->spec.freq = prinfo->sample_rate;
if (!iscapture) {
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
BSDAUDIO_Status(this);
@@ -347,7 +399,10 @@ BSDAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->WaitDevice = BSDAUDIO_WaitDevice;
impl->GetDeviceBuf = BSDAUDIO_GetDeviceBuf;
impl->CloseDevice = BSDAUDIO_CloseDevice;
impl->CaptureFromDevice = BSDAUDIO_CaptureFromDevice;
impl->FlushCapture = BSDAUDIO_FlushCapture;
impl->HasCaptureSupport = SDL_TRUE;
impl->AllowsArbitraryDeviceNames = 1;
return 1; /* this audio target is available. */
src/audio/coreaudio/SDL_coreaudio.c
+171 -54
View File
@@ -22,6 +22,8 @@
#if SDL_AUDIO_DRIVER_COREAUDIO
/* !!! FIXME: clean out some of the macro salsa in here. */
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
@@ -30,11 +32,8 @@
#define DEBUG_COREAUDIO 0
static void COREAUDIO_CloseDevice(_THIS);
#define CHECK_RESULT(msg) \
if (result != noErr) { \
COREAUDIO_CloseDevice(this); \
SDL_SetError("CoreAudio error (%s): %d", msg, (int) result); \
return 0; \
}
@@ -185,7 +184,7 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
#if DEBUG_COREAUDIO
printf("COREAUDIO: Found %s device #%d: '%s' (devid %d)\n",
((iscapture) ? "capture" : "output"),
(int) *devCount, ptr, (int) dev);
(int) i, ptr, (int) dev);
#endif
addfn(ptr, iscapture, dev, addfndata);
}
@@ -268,6 +267,27 @@ device_list_changed(AudioObjectID systemObj, UInt32 num_addr, const AudioObjectP
}
#endif
static int open_playback_devices = 0;
static int open_capture_devices = 0;
static void update_audio_session()
{
#if !MACOSX_COREAUDIO
/* !!! FIXME: move this to AVAudioSession. This is deprecated, and the new version is available as of (ancient!) iOS 3.0 */
UInt32 category;
if (open_playback_devices && open_capture_devices) {
category = kAudioSessionCategory_PlayAndRecord;
} else if (open_capture_devices) {
category = kAudioSessionCategory_RecordAudio;
} else { /* nothing open, or just playing audio. */
category = kAudioSessionCategory_AmbientSound;
}
AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof (UInt32), &category);
#endif
}
/* The CoreAudio callback */
static OSStatus
outputCallback(void *inRefCon,
@@ -283,7 +303,7 @@ outputCallback(void *inRefCon,
UInt32 i;
/* Only do anything if audio is enabled and not paused */
if (!this->enabled || this->paused) {
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
for (i = 0; i < ioData->mNumberBuffers; i++) {
abuf = &ioData->mBuffers[i];
SDL_memset(abuf->mData, this->spec.silence, abuf->mDataByteSize);
@@ -324,18 +344,53 @@ outputCallback(void *inRefCon,
}
}
return 0;
return noErr;
}
static OSStatus
inputCallback(void *inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames,
AudioBufferList * ioData)
AudioBufferList *ioData)
{
/* err = AudioUnitRender(afr->fAudioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, afr->fAudioBuffer); */
/* !!! FIXME: write me! */
SDL_AudioDevice *this = (SDL_AudioDevice *) inRefCon;
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
return noErr; /* just drop this if we're not accepting input. */
}
const OSStatus err = AudioUnitRender(this->hidden->audioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &this->hidden->captureBufferList);
SDL_assert(this->hidden->captureBufferList.mNumberBuffers == 1);
if (err == noErr) {
const AudioBuffer *abuf = &this->hidden->captureBufferList.mBuffers[0];
UInt32 remaining = abuf->mDataByteSize;
const Uint8 *ptr = (const Uint8 *) abuf->mData;
/* No SDL conversion should be needed here, ever, since we accept
any input format in OpenAudio, and leave the conversion to CoreAudio.
*/
while (remaining > 0) {
UInt32 len = this->hidden->bufferSize - this->hidden->bufferOffset;
if (len > remaining)
len = remaining;
/* !!! FIXME: why are we copying here? just pass the buffer to the callback? */
SDL_memcpy((char *)this->hidden->buffer + this->hidden->bufferOffset, ptr, len);
ptr += len;
remaining -= len;
this->hidden->bufferOffset += len;
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
SDL_LockMutex(this->mixer_lock);
(*this->spec.callback)(this->spec.userdata,
this->hidden->buffer, this->hidden->bufferSize);
SDL_UnlockMutex(this->mixer_lock);
this->hidden->bufferOffset = 0;
}
}
}
return noErr;
}
@@ -357,7 +412,7 @@ device_unplugged(AudioObjectID devid, UInt32 num_addr, const AudioObjectProperty
UInt32 size = sizeof (isAlive);
OSStatus error;
if (!this->enabled) {
if (!SDL_AtomicGet(&this->enabled)) {
return 0; /* already known to be dead. */
}
@@ -381,38 +436,39 @@ device_unplugged(AudioObjectID devid, UInt32 num_addr, const AudioObjectProperty
static void
COREAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->audioUnitOpened) {
#if MACOSX_COREAUDIO
/* Unregister our disconnect callback. */
AudioObjectRemovePropertyListener(this->hidden->deviceID, &alive_address, device_unplugged, this);
#endif
const int iscapture = this->iscapture;
if (this->hidden->audioUnitOpened) {
#if MACOSX_COREAUDIO
/* Unregister our disconnect callback. */
AudioObjectRemovePropertyListener(this->hidden->deviceID, &alive_address, device_unplugged, this);
#endif
AURenderCallbackStruct callback;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
const int iscapture = this->iscapture;
const AudioUnitElement bus =
((iscapture) ? input_bus : output_bus);
const AudioUnitScope scope =
((iscapture) ? kAudioUnitScope_Output :
kAudioUnitScope_Input);
AURenderCallbackStruct callback;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
const AudioUnitElement bus = ((iscapture) ? input_bus : output_bus);
/* stop processing the audio unit */
AudioOutputUnitStop(this->hidden->audioUnit);
/* stop processing the audio unit */
AudioOutputUnitStop(this->hidden->audioUnit);
/* Remove the input callback */
SDL_memset(&callback, 0, sizeof(AURenderCallbackStruct));
AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback, sizeof(callback));
AudioComponentInstanceDispose(this->hidden->audioUnit);
this->hidden->audioUnitOpened = 0;
}
SDL_free(this->hidden->buffer);
SDL_free(this->hidden);
this->hidden = NULL;
/* Remove the input callback */
SDL_zero(callback);
AudioUnitSetProperty(this->hidden->audioUnit,
iscapture ? kAudioOutputUnitProperty_SetInputCallback : kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global, bus, &callback, sizeof(callback));
AudioComponentInstanceDispose(this->hidden->audioUnit);
}
SDL_free(this->hidden->captureBufferList.mBuffers[0].mData);
SDL_free(this->hidden->buffer);
SDL_free(this->hidden);
if (iscapture) {
open_capture_devices--;
} else {
open_playback_devices--;
}
update_audio_session();
}
#if MACOSX_COREAUDIO
@@ -480,9 +536,6 @@ prepare_audiounit(_THIS, void *handle, int iscapture,
AudioComponent comp = NULL;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
const AudioUnitElement bus = ((iscapture) ? input_bus : output_bus);
const AudioUnitScope scope = ((iscapture) ? kAudioUnitScope_Output :
kAudioUnitScope_Input);
#if MACOSX_COREAUDIO
if (!prepare_device(this, handle, iscapture)) {
@@ -495,7 +548,7 @@ prepare_audiounit(_THIS, void *handle, int iscapture,
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
#if MACOSX_COREAUDIO
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
desc.componentSubType = iscapture ? kAudioUnitSubType_HALOutput : kAudioUnitSubType_DefaultOutput;
#else
desc.componentSubType = kAudioUnitSubType_RemoteIO;
#endif
@@ -512,10 +565,29 @@ prepare_audiounit(_THIS, void *handle, int iscapture,
this->hidden->audioUnitOpened = 1;
if (iscapture) { /* have to do EnableIO only for capture devices. */
UInt32 enable = 1;
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input, input_bus,
&enable, sizeof (enable));
CHECK_RESULT
("AudioUnitSetProperty (kAudioOutputUnitProperty_EnableIO input bus)");
enable = 0;
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output, output_bus,
&enable, sizeof (enable));
CHECK_RESULT
("AudioUnitSetProperty (kAudioOutputUnitProperty_EnableIO output bus)");
}
#if MACOSX_COREAUDIO
/* this is always on the output_bus, even for capture devices. */
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global, 0,
kAudioUnitScope_Global, output_bus,
&this->hidden->deviceID,
sizeof(AudioDeviceID));
CHECK_RESULT
@@ -525,16 +597,45 @@ prepare_audiounit(_THIS, void *handle, int iscapture,
/* Set the data format of the audio unit. */
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_StreamFormat,
scope, bus, strdesc, sizeof(*strdesc));
iscapture ? kAudioUnitScope_Output : kAudioUnitScope_Input,
iscapture ? input_bus : output_bus,
strdesc, sizeof (*strdesc));
CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_StreamFormat)");
if (iscapture) { /* only need to do this for capture devices. */
void *ptr;
UInt32 framesize = 0;
UInt32 propsize = sizeof (UInt32);
result = AudioUnitGetProperty(this->hidden->audioUnit,
kAudioUnitProperty_MaximumFramesPerSlice,
kAudioUnitScope_Global, output_bus,
&framesize, &propsize);
CHECK_RESULT
("AudioUnitGetProperty (kAudioDevicePropertyBufferFrameSize)");
framesize *= SDL_AUDIO_BITSIZE(this->spec.format) / 8;
ptr = SDL_calloc(1, framesize);
if (ptr == NULL) {
SDL_OutOfMemory();
return 0;
}
this->hidden->captureBufferList.mNumberBuffers = 1;
this->hidden->captureBufferList.mBuffers[0].mNumberChannels = this->spec.channels;
this->hidden->captureBufferList.mBuffers[0].mDataByteSize = framesize;
this->hidden->captureBufferList.mBuffers[0].mData = ptr;
}
/* Set the audio callback */
SDL_memset(&callback, 0, sizeof(AURenderCallbackStruct));
SDL_zero(callback);
callback.inputProc = ((iscapture) ? inputCallback : outputCallback);
callback.inputProcRefCon = this;
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback, sizeof(callback));
iscapture ? kAudioOutputUnitProperty_SetInputCallback : kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
iscapture ? input_bus : output_bus,
&callback, sizeof (callback));
CHECK_RESULT
("AudioUnitSetProperty (kAudioUnitProperty_SetRenderCallback)");
@@ -542,8 +643,14 @@ prepare_audiounit(_THIS, void *handle, int iscapture,
SDL_CalculateAudioSpec(&this->spec);
/* Allocate a sample buffer */
this->hidden->bufferOffset = this->hidden->bufferSize = this->spec.size;
this->hidden->bufferSize = this->spec.size;
this->hidden->bufferOffset = iscapture ? 0 : this->hidden->bufferSize;
this->hidden->buffer = SDL_malloc(this->hidden->bufferSize);
if (this->hidden->buffer == NULL) {
SDL_OutOfMemory();
return 0;
}
result = AudioUnitInitialize(this->hidden->audioUnit);
CHECK_RESULT("AudioUnitInitialize");
@@ -552,6 +659,8 @@ prepare_audiounit(_THIS, void *handle, int iscapture,
result = AudioOutputUnitStart(this->hidden->audioUnit);
CHECK_RESULT("AudioOutputUnitStart");
/* !!! FIXME: what does iOS do when a bluetooth audio device vanishes? Headphones unplugged? */
/* !!! FIXME: (we only do a "default" device on iOS right now...can we do more?) */
#if MACOSX_COREAUDIO
/* Fire a callback if the device stops being "alive" (disconnected, etc). */
AudioObjectAddPropertyListener(this->hidden->deviceID, &alive_address, device_unplugged, this);
@@ -575,10 +684,17 @@ COREAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
if (iscapture) {
open_capture_devices++;
} else {
open_playback_devices++;
}
update_audio_session();
/* Setup a AudioStreamBasicDescription with the requested format */
SDL_memset(&strdesc, '\0', sizeof(AudioStreamBasicDescription));
SDL_zero(strdesc);
strdesc.mFormatID = kAudioFormatLinearPCM;
strdesc.mFormatFlags = kLinearPCMFormatFlagIsPacked;
strdesc.mChannelsPerFrame = this->spec.channels;
@@ -613,7 +729,6 @@ COREAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
if (!valid_datatype) { /* shouldn't happen, but just in case... */
COREAUDIO_CloseDevice(this);
return SDL_SetError("Unsupported audio format");
}
@@ -623,7 +738,6 @@ COREAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
strdesc.mBytesPerFrame * strdesc.mFramesPerPacket;
if (!prepare_audiounit(this, handle, iscapture, &strdesc)) {
COREAUDIO_CloseDevice(this);
return -1; /* prepare_audiounit() will call SDL_SetError()... */
}
@@ -653,17 +767,20 @@ COREAUDIO_Init(SDL_AudioDriverImpl * impl)
AudioObjectAddPropertyListener(kAudioObjectSystemObject, &devlist_address, device_list_changed, NULL);
#else
impl->OnlyHasDefaultOutputDevice = 1;
impl->OnlyHasDefaultCaptureDevice = 1;
/* Set category to ambient sound so that other music continues playing.
You can change this at runtime in your own code if you need different
behavior. If this is common, we can add an SDL hint for this.
*/
/* !!! FIXME: move this to AVAudioSession. This is deprecated, and the new version is available as of (ancient!) iOS 3.0 */
AudioSessionInitialize(NULL, NULL, NULL, nil);
UInt32 category = kAudioSessionCategory_AmbientSound;
AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(UInt32), &category);
#endif
impl->ProvidesOwnCallbackThread = 1;
impl->HasCaptureSupport = 1;
return 1; /* this audio target is available. */
}
src/audio/coreaudio/SDL_coreaudio.h
+1
View File
@@ -48,6 +48,7 @@ struct SDL_PrivateAudioData
void *buffer;
UInt32 bufferOffset;
UInt32 bufferSize;
AudioBufferList captureBufferList;
#if MACOSX_COREAUDIO
AudioDeviceID deviceID;
#endif
src/audio/directsound/SDL_directsound.c
+193 -88
View File
@@ -24,6 +24,7 @@
/* Allow access to a raw mixing buffer */
#include "SDL_assert.h"
#include "SDL_timer.h"
#include "SDL_loadso.h"
#include "SDL_audio.h"
@@ -36,11 +37,13 @@
/* DirectX function pointers for audio */
static void* DSoundDLL = NULL;
typedef HRESULT(WINAPI*fnDirectSoundCreate8)(LPGUID,LPDIRECTSOUND*,LPUNKNOWN);
typedef HRESULT(WINAPI*fnDirectSoundEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
typedef HRESULT(WINAPI*fnDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW,LPVOID);
typedef HRESULT (WINAPI *fnDirectSoundCreate8)(LPGUID,LPDIRECTSOUND*,LPUNKNOWN);
typedef HRESULT (WINAPI *fnDirectSoundEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
typedef HRESULT (WINAPI *fnDirectSoundCaptureCreate8)(LPCGUID,LPDIRECTSOUNDCAPTURE8 *,LPUNKNOWN);
typedef HRESULT (WINAPI *fnDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW,LPVOID);
static fnDirectSoundCreate8 pDirectSoundCreate8 = NULL;
static fnDirectSoundEnumerateW pDirectSoundEnumerateW = NULL;
static fnDirectSoundCaptureCreate8 pDirectSoundCaptureCreate8 = NULL;
static fnDirectSoundCaptureEnumerateW pDirectSoundCaptureEnumerateW = NULL;
static void
@@ -48,6 +51,7 @@ DSOUND_Unload(void)
{
pDirectSoundCreate8 = NULL;
pDirectSoundEnumerateW = NULL;
pDirectSoundCaptureCreate8 = NULL;
pDirectSoundCaptureEnumerateW = NULL;
if (DSoundDLL != NULL) {
@@ -76,6 +80,7 @@ DSOUND_Load(void)
loaded = 1; /* will reset if necessary. */
DSOUNDLOAD(DirectSoundCreate8);
DSOUNDLOAD(DirectSoundEnumerateW);
DSOUNDLOAD(DirectSoundCaptureCreate8);
DSOUNDLOAD(DirectSoundCaptureEnumerateW);
#undef DSOUNDLOAD
@@ -155,7 +160,7 @@ FindAllDevs(LPGUID guid, LPCWSTR desc, LPCWSTR module, LPVOID data)
{
const int iscapture = (int) ((size_t) data);
if (guid != NULL) { /* skip default device */
char *str = WIN_StringToUTF8(desc);
char *str = WIN_LookupAudioDeviceName(desc, guid);
if (str != NULL) {
LPGUID cpyguid = (LPGUID) SDL_malloc(sizeof (GUID));
SDL_memcpy(cpyguid, guid, sizeof (GUID));
@@ -197,7 +202,7 @@ DSOUND_WaitDevice(_THIS)
return;
}
while ((cursor / this->hidden->mixlen) == this->hidden->lastchunk) {
while ((cursor / this->spec.size) == this->hidden->lastchunk) {
/* FIXME: find out how much time is left and sleep that long */
SDL_Delay(1);
@@ -239,9 +244,8 @@ DSOUND_PlayDevice(_THIS)
if (this->hidden->locked_buf) {
IDirectSoundBuffer_Unlock(this->hidden->mixbuf,
this->hidden->locked_buf,
this->hidden->mixlen, NULL, 0);
this->spec.size, NULL, 0);
}
}
static Uint8 *
@@ -265,7 +269,7 @@ DSOUND_GetDeviceBuf(_THIS)
SetDSerror("DirectSound GetCurrentPosition", result);
return (NULL);
}
cursor /= this->hidden->mixlen;
cursor /= this->spec.size;
#ifdef DEBUG_SOUND
/* Detect audio dropouts */
{
@@ -281,17 +285,17 @@ DSOUND_GetDeviceBuf(_THIS)
#endif
this->hidden->lastchunk = cursor;
cursor = (cursor + 1) % this->hidden->num_buffers;
cursor *= this->hidden->mixlen;
cursor *= this->spec.size;
/* Lock the audio buffer */
result = IDirectSoundBuffer_Lock(this->hidden->mixbuf, cursor,
this->hidden->mixlen,
this->spec.size,
(LPVOID *) & this->hidden->locked_buf,
&rawlen, NULL, &junk, 0);
if (result == DSERR_BUFFERLOST) {
IDirectSoundBuffer_Restore(this->hidden->mixbuf);
result = IDirectSoundBuffer_Lock(this->hidden->mixbuf, cursor,
this->hidden->mixlen,
this->spec.size,
(LPVOID *) & this->
hidden->locked_buf, &rawlen, NULL,
&junk, 0);
@@ -310,7 +314,7 @@ DSOUND_WaitDone(_THIS)
/* Wait for the playing chunk to finish */
if (stream != NULL) {
SDL_memset(stream, this->spec.silence, this->hidden->mixlen);
SDL_memset(stream, this->spec.silence, this->spec.size);
DSOUND_PlayDevice(this);
}
DSOUND_WaitDevice(this);
@@ -319,93 +323,106 @@ DSOUND_WaitDone(_THIS)
IDirectSoundBuffer_Stop(this->hidden->mixbuf);
}
static void
DSOUND_CloseDevice(_THIS)
static int
DSOUND_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
if (this->hidden != NULL) {
if (this->hidden->sound != NULL) {
if (this->hidden->mixbuf != NULL) {
/* Clean up the audio buffer */
IDirectSoundBuffer_Release(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
IDirectSound_Release(this->hidden->sound);
this->hidden->sound = NULL;
struct SDL_PrivateAudioData *h = this->hidden;
DWORD junk, cursor, ptr1len, ptr2len;
VOID *ptr1, *ptr2;
SDL_assert(buflen == this->spec.size);
while (SDL_TRUE) {
if (SDL_AtomicGet(&this->shutdown)) { /* in case the buffer froze... */
SDL_memset(buffer, this->spec.silence, buflen);
return buflen;
}
SDL_free(this->hidden);
this->hidden = NULL;
if (IDirectSoundCaptureBuffer_GetCurrentPosition(h->capturebuf, &junk, &cursor) != DS_OK) {
return -1;
}
if ((cursor / this->spec.size) == h->lastchunk) {
SDL_Delay(1); /* FIXME: find out how much time is left and sleep that long */
} else {
break;
}
}
if (IDirectSoundCaptureBuffer_Lock(h->capturebuf, h->lastchunk * this->spec.size, this->spec.size, &ptr1, &ptr1len, &ptr2, &ptr2len, 0) != DS_OK) {
return -1;
}
SDL_assert(ptr1len == this->spec.size);
SDL_assert(ptr2 == NULL);
SDL_assert(ptr2len == 0);
SDL_memcpy(buffer, ptr1, ptr1len);
if (IDirectSoundCaptureBuffer_Unlock(h->capturebuf, ptr1, ptr1len, ptr2, ptr2len) != DS_OK) {
return -1;
}
h->lastchunk = (h->lastchunk + 1) % h->num_buffers;
return ptr1len;
}
static void
DSOUND_FlushCapture(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
DWORD junk, cursor;
if (IDirectSoundCaptureBuffer_GetCurrentPosition(h->capturebuf, &junk, &cursor) == DS_OK) {
h->lastchunk = cursor / this->spec.size;
}
}
static void
DSOUND_CloseDevice(_THIS)
{
if (this->hidden->mixbuf != NULL) {
IDirectSoundBuffer_Stop(this->hidden->mixbuf);
IDirectSoundBuffer_Release(this->hidden->mixbuf);
}
if (this->hidden->sound != NULL) {
IDirectSound_Release(this->hidden->sound);
}
if (this->hidden->capturebuf != NULL) {
IDirectSoundCaptureBuffer_Stop(this->hidden->capturebuf);
IDirectSoundCaptureBuffer_Release(this->hidden->capturebuf);
}
if (this->hidden->capture != NULL) {
IDirectSoundCapture_Release(this->hidden->capture);
}
SDL_free(this->hidden);
}
/* This function tries to create a secondary audio buffer, and returns the
number of audio chunks available in the created buffer.
number of audio chunks available in the created buffer. This is for
playback devices, not capture.
*/
static int
CreateSecondary(_THIS, HWND focus)
CreateSecondary(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
{
LPDIRECTSOUND sndObj = this->hidden->sound;
LPDIRECTSOUNDBUFFER *sndbuf = &this->hidden->mixbuf;
Uint32 chunksize = this->spec.size;
const int numchunks = 8;
HRESULT result = DS_OK;
DSBUFFERDESC format;
LPVOID pvAudioPtr1, pvAudioPtr2;
DWORD dwAudioBytes1, dwAudioBytes2;
WAVEFORMATEX wfmt;
SDL_zero(wfmt);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
wfmt.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
wfmt.wFormatTag = WAVE_FORMAT_PCM;
}
wfmt.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
wfmt.nChannels = this->spec.channels;
wfmt.nSamplesPerSec = this->spec.freq;
wfmt.nBlockAlign = wfmt.nChannels * (wfmt.wBitsPerSample / 8);
wfmt.nAvgBytesPerSec = wfmt.nSamplesPerSec * wfmt.nBlockAlign;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Try to set primary mixing privileges */
if (focus) {
result = IDirectSound_SetCooperativeLevel(sndObj,
focus, DSSCL_PRIORITY);
} else {
result = IDirectSound_SetCooperativeLevel(sndObj,
GetDesktopWindow(),
DSSCL_NORMAL);
}
if (result != DS_OK) {
return SetDSerror("DirectSound SetCooperativeLevel", result);
}
/* Try to create the secondary buffer */
SDL_zero(format);
format.dwSize = sizeof(format);
format.dwFlags = DSBCAPS_GETCURRENTPOSITION2;
if (!focus) {
format.dwFlags |= DSBCAPS_GLOBALFOCUS;
} else {
format.dwFlags |= DSBCAPS_STICKYFOCUS;
}
format.dwBufferBytes = numchunks * chunksize;
if ((format.dwBufferBytes < DSBSIZE_MIN) ||
(format.dwBufferBytes > DSBSIZE_MAX)) {
return SDL_SetError("Sound buffer size must be between %d and %d",
DSBSIZE_MIN / numchunks, DSBSIZE_MAX / numchunks);
}
format.dwReserved = 0;
format.lpwfxFormat = &wfmt;
format.dwFlags |= DSBCAPS_GLOBALFOCUS;
format.dwBufferBytes = bufsize;
format.lpwfxFormat = wfmt;
result = IDirectSound_CreateSoundBuffer(sndObj, &format, sndbuf, NULL);
if (result != DS_OK) {
return SetDSerror("DirectSound CreateSoundBuffer", result);
}
IDirectSoundBuffer_SetFormat(*sndbuf, &wfmt);
IDirectSoundBuffer_SetFormat(*sndbuf, wfmt);
/* Silence the initial audio buffer */
result = IDirectSoundBuffer_Lock(*sndbuf, 0, format.dwBufferBytes,
@@ -420,31 +437,90 @@ CreateSecondary(_THIS, HWND focus)
}
/* We're ready to go */
return (numchunks);
return 0;
}
/* This function tries to create a capture buffer, and returns the
number of audio chunks available in the created buffer. This is for
capture devices, not playback.
*/
static int
CreateCaptureBuffer(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
{
LPDIRECTSOUNDCAPTURE capture = this->hidden->capture;
LPDIRECTSOUNDCAPTUREBUFFER *capturebuf = &this->hidden->capturebuf;
DSCBUFFERDESC format;
// DWORD junk, cursor;
HRESULT result;
SDL_zero(format);
format.dwSize = sizeof (format);
format.dwFlags = DSCBCAPS_WAVEMAPPED;
format.dwBufferBytes = bufsize;
format.lpwfxFormat = wfmt;
result = IDirectSoundCapture_CreateCaptureBuffer(capture, &format, capturebuf, NULL);
if (result != DS_OK) {
return SetDSerror("DirectSound CreateCaptureBuffer", result);
}
result = IDirectSoundCaptureBuffer_Start(*capturebuf, DSCBSTART_LOOPING);
if (result != DS_OK) {
IDirectSoundCaptureBuffer_Release(*capturebuf);
return SetDSerror("DirectSound Start", result);
}
#if 0
/* presumably this starts at zero, but just in case... */
result = IDirectSoundCaptureBuffer_GetCurrentPosition(*capturebuf, &junk, &cursor);
if (result != DS_OK) {
IDirectSoundCaptureBuffer_Stop(*capturebuf);
IDirectSoundCaptureBuffer_Release(*capturebuf);
return SetDSerror("DirectSound GetCurrentPosition", result);
}
this->hidden->lastchunk = cursor / this->spec.size;
#endif
return 0;
}
static int
DSOUND_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const DWORD numchunks = 8;
HRESULT result;
SDL_bool valid_format = SDL_FALSE;
SDL_bool tried_format = SDL_FALSE;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
LPGUID guid = (LPGUID) handle;
DWORD bufsize;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
/* Open the audio device */
result = pDirectSoundCreate8(guid, &this->hidden->sound, NULL);
if (result != DS_OK) {
DSOUND_CloseDevice(this);
return SetDSerror("DirectSoundCreate", result);
if (iscapture) {
result = pDirectSoundCaptureCreate8(guid, &this->hidden->capture, NULL);
if (result != DS_OK) {
return SetDSerror("DirectSoundCaptureCreate8", result);
}
} else {
result = pDirectSoundCreate8(guid, &this->hidden->sound, NULL);
if (result != DS_OK) {
return SetDSerror("DirectSoundCreate8", result);
}
result = IDirectSound_SetCooperativeLevel(this->hidden->sound,
GetDesktopWindow(),
DSSCL_NORMAL);
if (result != DS_OK) {
return SetDSerror("DirectSound SetCooperativeLevel", result);
}
}
while ((!valid_format) && (test_format)) {
@@ -454,10 +530,38 @@ DSOUND_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
case AUDIO_S32:
case AUDIO_F32:
tried_format = SDL_TRUE;
this->spec.format = test_format;
this->hidden->num_buffers = CreateSecondary(this, NULL);
if (this->hidden->num_buffers > 0) {
valid_format = SDL_TRUE;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
bufsize = numchunks * this->spec.size;
if ((bufsize < DSBSIZE_MIN) || (bufsize > DSBSIZE_MAX)) {
SDL_SetError("Sound buffer size must be between %d and %d",
(DSBSIZE_MIN < numchunks) ? 1 : DSBSIZE_MIN / numchunks,
DSBSIZE_MAX / numchunks);
} else {
int rc;
WAVEFORMATEX wfmt;
SDL_zero(wfmt);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
wfmt.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
wfmt.wFormatTag = WAVE_FORMAT_PCM;
}
wfmt.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
wfmt.nChannels = this->spec.channels;
wfmt.nSamplesPerSec = this->spec.freq;
wfmt.nBlockAlign = wfmt.nChannels * (wfmt.wBitsPerSample / 8);
wfmt.nAvgBytesPerSec = wfmt.nSamplesPerSec * wfmt.nBlockAlign;
rc = iscapture ? CreateCaptureBuffer(this, bufsize, &wfmt) : CreateSecondary(this, bufsize, &wfmt);
if (rc == 0) {
this->hidden->num_buffers = numchunks;
valid_format = SDL_TRUE;
}
}
break;
}
@@ -465,15 +569,13 @@ DSOUND_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
if (!valid_format) {
DSOUND_CloseDevice(this);
if (tried_format) {
return -1; /* CreateSecondary() should have called SDL_SetError(). */
}
return SDL_SetError("DirectSound: Unsupported audio format");
}
/* The buffer will auto-start playing in DSOUND_WaitDevice() */
this->hidden->mixlen = this->spec.size;
/* Playback buffers will auto-start playing in DSOUND_WaitDevice() */
return 0; /* good to go. */
}
@@ -500,11 +602,14 @@ DSOUND_Init(SDL_AudioDriverImpl * impl)
impl->WaitDevice = DSOUND_WaitDevice;
impl->WaitDone = DSOUND_WaitDone;
impl->GetDeviceBuf = DSOUND_GetDeviceBuf;
impl->CaptureFromDevice = DSOUND_CaptureFromDevice;
impl->FlushCapture = DSOUND_FlushCapture;
impl->CloseDevice = DSOUND_CloseDevice;
impl->FreeDeviceHandle = DSOUND_FreeDeviceHandle;
impl->Deinitialize = DSOUND_Deinitialize;
impl->HasCaptureSupport = SDL_TRUE;
return 1; /* this audio target is available. */
}
src/audio/directsound/SDL_directsound.h
+2 -1
View File
@@ -35,8 +35,9 @@ struct SDL_PrivateAudioData
{
LPDIRECTSOUND sound;
LPDIRECTSOUNDBUFFER mixbuf;
LPDIRECTSOUNDCAPTURE capture;
LPDIRECTSOUNDCAPTUREBUFFER capturebuf;
int num_buffers;
int mixlen;
DWORD lastchunk;
Uint8 *locked_buf;
};
src/audio/disk/SDL_diskaudio.c
+97 -64
View File
@@ -31,46 +31,33 @@
#include "SDL_rwops.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_diskaudio.h"
/* !!! FIXME: these should be SDL hints, not environment variables. */
/* environment variables and defaults. */
#define DISKENVR_OUTFILE "SDL_DISKAUDIOFILE"
#define DISKDEFAULT_OUTFILE "sdlaudio.raw"
#define DISKENVR_WRITEDELAY "SDL_DISKAUDIODELAY"
#define DISKDEFAULT_WRITEDELAY 150
static const char *
DISKAUD_GetOutputFilename(const char *devname)
{
if (devname == NULL) {
devname = SDL_getenv(DISKENVR_OUTFILE);
if (devname == NULL) {
devname = DISKDEFAULT_OUTFILE;
}
}
return devname;
}
#define DISKENVR_INFILE "SDL_DISKAUDIOFILEIN"
#define DISKDEFAULT_INFILE "sdlaudio-in.raw"
#define DISKENVR_IODELAY "SDL_DISKAUDIODELAY"
/* This function waits until it is possible to write a full sound buffer */
static void
DISKAUD_WaitDevice(_THIS)
DISKAUDIO_WaitDevice(_THIS)
{
SDL_Delay(this->hidden->write_delay);
SDL_Delay(this->hidden->io_delay);
}
static void
DISKAUD_PlayDevice(_THIS)
DISKAUDIO_PlayDevice(_THIS)
{
size_t written;
/* Write the audio data */
written = SDL_RWwrite(this->hidden->output,
this->hidden->mixbuf, 1, this->hidden->mixlen);
const size_t written = SDL_RWwrite(this->hidden->io,
this->hidden->mixbuf,
1, this->spec.size);
/* If we couldn't write, assume fatal error for now */
if (written != this->hidden->mixlen) {
if (written != this->spec.size) {
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
@@ -79,63 +66,105 @@ DISKAUD_PlayDevice(_THIS)
}
static Uint8 *
DISKAUD_GetDeviceBuf(_THIS)
DISKAUDIO_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
DISKAUD_CloseDevice(_THIS)
static int
DISKAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->output != NULL) {
SDL_RWclose(this->hidden->output);
this->hidden->output = NULL;
struct SDL_PrivateAudioData *h = this->hidden;
const int origbuflen = buflen;
SDL_Delay(h->io_delay);
if (h->io) {
const size_t br = SDL_RWread(h->io, buffer, 1, buflen);
buflen -= (int) br;
buffer = ((Uint8 *) buffer) + br;
if (buflen > 0) { /* EOF (or error, but whatever). */
SDL_RWclose(h->io);
h->io = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
/* if we ran out of file, just write silence. */
SDL_memset(buffer, this->spec.silence, buflen);
return origbuflen;
}
static void
DISKAUDIO_FlushCapture(_THIS)
{
/* no op...we don't advance the file pointer or anything. */
}
static void
DISKAUDIO_CloseDevice(_THIS)
{
if (this->hidden->io != NULL) {
SDL_RWclose(this->hidden->io);
}
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
static const char *
get_filename(const int iscapture, const char *devname)
{
if (devname == NULL) {
devname = SDL_getenv(iscapture ? DISKENVR_INFILE : DISKENVR_OUTFILE);
if (devname == NULL) {
devname = iscapture ? DISKDEFAULT_INFILE : DISKDEFAULT_OUTFILE;
}
}
return devname;
}
static int
DISKAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
DISKAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
/* handle != NULL means "user specified the placeholder name on the fake detected device list" */
const char *fname = DISKAUD_GetOutputFilename(handle ? NULL : devname);
const char *envr = SDL_getenv(DISKENVR_WRITEDELAY);
const char *fname = get_filename(iscapture, handle ? NULL : devname);
const char *envr = SDL_getenv(DISKENVR_IODELAY);
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc(sizeof(*this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, sizeof(*this->hidden));
SDL_zerop(this->hidden);
this->hidden->mixlen = this->spec.size;
this->hidden->write_delay =
(envr) ? SDL_atoi(envr) : DISKDEFAULT_WRITEDELAY;
if (envr != NULL) {
this->hidden->io_delay = SDL_atoi(envr);
} else {
this->hidden->io_delay = ((this->spec.samples * 1000) / this->spec.freq);
}
/* Open the audio device */
this->hidden->output = SDL_RWFromFile(fname, "wb");
if (this->hidden->output == NULL) {
DISKAUD_CloseDevice(this);
this->hidden->io = SDL_RWFromFile(fname, iscapture ? "rb" : "wb");
if (this->hidden->io == NULL) {
return -1;
}
/* Allocate mixing buffer */
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
DISKAUD_CloseDevice(this);
return -1;
if (!iscapture) {
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->spec.size);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
#if HAVE_STDIO_H
fprintf(stderr,
"WARNING: You are using the SDL disk writer audio driver!\n"
" Writing to file [%s].\n", fname);
"WARNING: You are using the SDL disk i/o audio driver!\n"
" %s file [%s].\n", iscapture ? "Reading from" : "Writing to",
fname);
#endif
/* We're ready to rock and roll. :-) */
@@ -143,30 +172,34 @@ DISKAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
static void
DISKAUD_DetectDevices(void)
DISKAUDIO_DetectDevices(void)
{
/* !!! FIXME: stole this literal string from DEFAULT_OUTPUT_DEVNAME in SDL_audio.c */
SDL_AddAudioDevice(SDL_FALSE, "System audio output device", (void *) 0x1);
SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, (void *) 0x1);
SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, (void *) 0x2);
}
static int
DISKAUD_Init(SDL_AudioDriverImpl * impl)
DISKAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = DISKAUD_OpenDevice;
impl->WaitDevice = DISKAUD_WaitDevice;
impl->PlayDevice = DISKAUD_PlayDevice;
impl->GetDeviceBuf = DISKAUD_GetDeviceBuf;
impl->CloseDevice = DISKAUD_CloseDevice;
impl->DetectDevices = DISKAUD_DetectDevices;
impl->OpenDevice = DISKAUDIO_OpenDevice;
impl->WaitDevice = DISKAUDIO_WaitDevice;
impl->PlayDevice = DISKAUDIO_PlayDevice;
impl->GetDeviceBuf = DISKAUDIO_GetDeviceBuf;
impl->CaptureFromDevice = DISKAUDIO_CaptureFromDevice;
impl->FlushCapture = DISKAUDIO_FlushCapture;
impl->CloseDevice = DISKAUDIO_CloseDevice;
impl->DetectDevices = DISKAUDIO_DetectDevices;
impl->AllowsArbitraryDeviceNames = 1;
impl->HasCaptureSupport = SDL_TRUE;
return 1; /* this audio target is available. */
}
AudioBootStrap DISKAUD_bootstrap = {
"disk", "direct-to-disk audio", DISKAUD_Init, 1
AudioBootStrap DISKAUDIO_bootstrap = {
"disk", "direct-to-disk audio", DISKAUDIO_Init, 1
};
#endif /* SDL_AUDIO_DRIVER_DISK */
src/audio/disk/SDL_diskaudio.h
+2 -3
View File
@@ -32,10 +32,9 @@
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
SDL_RWops *output;
SDL_RWops *io;
Uint32 io_delay;
Uint8 *mixbuf;
Uint32 mixlen;
Uint32 write_delay;
};
#endif /* _SDL_diskaudio_h */
src/audio/dsp/SDL_dspaudio.c
+43 -31
View File
@@ -44,7 +44,6 @@
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_dspaudio.h"
@@ -60,16 +59,11 @@ DSP_DetectDevices(void)
static void
DSP_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
this->hidden->audio_fd = -1;
}
SDL_free(this->hidden);
this->hidden = NULL;
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
}
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
@@ -106,23 +100,20 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
/* Open the audio device */
this->hidden->audio_fd = open(devname, flags, 0);
if (this->hidden->audio_fd < 0) {
DSP_CloseDevice(this);
return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
}
this->hidden->mixbuf = NULL;
/* Make the file descriptor use blocking writes with fcntl() */
/* Make the file descriptor use blocking i/o with fcntl() */
{
long ctlflags;
ctlflags = fcntl(this->hidden->audio_fd, F_GETFL);
ctlflags &= ~O_NONBLOCK;
if (fcntl(this->hidden->audio_fd, F_SETFL, ctlflags) < 0) {
DSP_CloseDevice(this);
return SDL_SetError("Couldn't set audio blocking mode");
}
}
@@ -130,7 +121,6 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Get a list of supported hardware formats */
if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0) {
perror("SNDCTL_DSP_GETFMTS");
DSP_CloseDevice(this);
return SDL_SetError("Couldn't get audio format list");
}
@@ -187,7 +177,6 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
}
if (format == 0) {
DSP_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
@@ -197,7 +186,6 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if ((ioctl(this->hidden->audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
(value != format)) {
perror("SNDCTL_DSP_SETFMT");
DSP_CloseDevice(this);
return SDL_SetError("Couldn't set audio format");
}
@@ -205,7 +193,6 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
value = this->spec.channels;
if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0) {
perror("SNDCTL_DSP_CHANNELS");
DSP_CloseDevice(this);
return SDL_SetError("Cannot set the number of channels");
}
this->spec.channels = value;
@@ -214,7 +201,6 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
value = this->spec.freq;
if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_SPEED, &value) < 0) {
perror("SNDCTL_DSP_SPEED");
DSP_CloseDevice(this);
return SDL_SetError("Couldn't set audio frequency");
}
this->spec.freq = value;
@@ -225,7 +211,6 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Determine the power of two of the fragment size */
for (frag_spec = 0; (0x01U << frag_spec) < this->spec.size; ++frag_spec);
if ((0x01U << frag_spec) != this->spec.size) {
DSP_CloseDevice(this);
return SDL_SetError("Fragment size must be a power of two");
}
frag_spec |= 0x00020000; /* two fragments, for low latency */
@@ -250,13 +235,14 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
#endif
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
DSP_CloseDevice(this);
return SDL_OutOfMemory();
if (!iscapture) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 0;
@@ -266,14 +252,13 @@ DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
static void
DSP_PlayDevice(_THIS)
{
const Uint8 *mixbuf = this->hidden->mixbuf;
const int mixlen = this->hidden->mixlen;
if (write(this->hidden->audio_fd, mixbuf, mixlen) == -1) {
struct SDL_PrivateAudioData *h = this->hidden;
if (write(h->audio_fd, h->mixbuf, h->mixlen) == -1) {
perror("Audio write");
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", mixlen);
fprintf(stderr, "Wrote %d bytes of audio data\n", h->mixlen);
#endif
}
@@ -283,6 +268,30 @@ DSP_GetDeviceBuf(_THIS)
return (this->hidden->mixbuf);
}
static int
DSP_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
return (int) read(this->hidden->audio_fd, buffer, buflen);
}
static void
DSP_FlushCapture(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
audio_buf_info info;
if (ioctl(h->audio_fd, SNDCTL_DSP_GETISPACE, &info) == 0) {
while (info.bytes > 0) {
char buf[512];
const size_t len = SDL_min(sizeof (buf), info.bytes);
const ssize_t br = read(h->audio_fd, buf, len);
if (br <= 0) {
break;
}
info.bytes -= br;
}
}
}
static int
DSP_Init(SDL_AudioDriverImpl * impl)
{
@@ -292,8 +301,11 @@ DSP_Init(SDL_AudioDriverImpl * impl)
impl->PlayDevice = DSP_PlayDevice;
impl->GetDeviceBuf = DSP_GetDeviceBuf;
impl->CloseDevice = DSP_CloseDevice;
impl->CaptureFromDevice = DSP_CaptureFromDevice;
impl->FlushCapture = DSP_FlushCapture;
impl->AllowsArbitraryDeviceNames = 1;
impl->HasCaptureSupport = SDL_TRUE;
return 1; /* this audio target is available. */
}
src/audio/dummy/SDL_dummyaudio.c
+22 -5
View File
@@ -22,27 +22,44 @@
/* Output audio to nowhere... */
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_dummyaudio.h"
static int
DUMMYAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
DUMMYAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
return 0; /* always succeeds. */
}
static int
DUMMYAUD_Init(SDL_AudioDriverImpl * impl)
DUMMYAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
/* Delay to make this sort of simulate real audio input. */
SDL_Delay((this->spec.samples * 1000) / this->spec.freq);
/* always return a full buffer of silence. */
SDL_memset(buffer, this->spec.silence, buflen);
return buflen;
}
static int
DUMMYAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = DUMMYAUD_OpenDevice;
impl->OpenDevice = DUMMYAUDIO_OpenDevice;
impl->CaptureFromDevice = DUMMYAUDIO_CaptureFromDevice;
impl->OnlyHasDefaultOutputDevice = 1;
impl->OnlyHasDefaultCaptureDevice = 1;
impl->HasCaptureSupport = SDL_TRUE;
return 1; /* this audio target is available. */
}
AudioBootStrap DUMMYAUD_bootstrap = {
"dummy", "SDL dummy audio driver", DUMMYAUD_Init, 1
AudioBootStrap DUMMYAUDIO_bootstrap = {
"dummy", "SDL dummy audio driver", DUMMYAUDIO_Init, 1
};
/* vi: set ts=4 sw=4 expandtab: */
src/audio/emscripten/SDL_emscriptenaudio.c
+214 -61
View File
@@ -61,16 +61,15 @@ HandleAudioProcess(_THIS)
Uint8 *buf = NULL;
int byte_len = 0;
int bytes = SDL_AUDIO_BITSIZE(this->spec.format) / 8;
int bytes_in = SDL_AUDIO_BITSIZE(this->convert.src_format) / 8;
/* Only do soemthing if audio is enabled */
if (!this->enabled)
return;
if (this->paused)
/* Only do something if audio is enabled */
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
return;
}
if (this->convert.needed) {
const int bytes_in = SDL_AUDIO_BITSIZE(this->convert.src_format) / 8;
if (this->hidden->conv_in_len != 0) {
this->convert.len = this->hidden->conv_in_len * bytes_in * this->spec.channels;
}
@@ -136,29 +135,139 @@ HandleAudioProcess(_THIS)
}
static void
Emscripten_CloseDevice(_THIS)
HandleCaptureProcess(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
/* Clean up the audio buffer */
SDL_free(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
Uint8 *buf;
int buflen;
SDL_free(this->hidden);
this->hidden = NULL;
/* Only do something if audio is enabled */
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
return;
}
if (this->convert.needed) {
buf = this->convert.buf;
buflen = this->convert.len_cvt;
} else {
if (!this->hidden->mixbuf) {
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->spec.size);
if (!this->hidden->mixbuf) {
return; /* oh well. */
}
}
buf = this->hidden->mixbuf;
buflen = this->spec.size;
}
EM_ASM_ARGS({
var numChannels = SDL2.capture.currentCaptureBuffer.numberOfChannels;
if (numChannels == 1) { /* fastpath this a little for the common (mono) case. */
var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(0);
if (channelData.length != $1) {
throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
for (var j = 0; j < $1; ++j) {
setValue($0 + (j * 4), channelData[j], 'float');
}
} else {
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(c);
if (channelData.length != $1) {
throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
for (var j = 0; j < $1; ++j) {
setValue($0 + (((j * numChannels) + c) * 4), channelData[j], 'float');
}
}
}
}, buf, (this->spec.size / sizeof (float)) / this->spec.channels);
/* okay, we've got an interleaved float32 array in C now. */
if (this->convert.needed) {
SDL_ConvertAudio(&this->convert);
}
/* Send it to the app. */
(*this->spec.callback) (this->spec.userdata, buf, buflen);
}
static void
EMSCRIPTENAUDIO_CloseDevice(_THIS)
{
EM_ASM_({
if ($0) {
if (SDL2.capture.silenceTimer !== undefined) {
clearTimeout(SDL2.capture.silenceTimer);
}
if (SDL2.capture.scriptProcessorNode !== undefined) {
SDL2.capture.scriptProcessorNode.disconnect();
SDL2.capture.scriptProcessorNode = undefined;
}
if (SDL2.capture.mediaStreamNode !== undefined) {
SDL2.capture.mediaStreamNode.disconnect();
SDL2.capture.mediaStreamNode = undefined;
}
if (SDL2.capture.silenceBuffer !== undefined) {
SDL2.capture.silenceBuffer = undefined
}
SDL2.capture = undefined;
} else {
if (SDL2.audio.scriptProcessorNode != undefined) {
SDL2.audio.scriptProcessorNode.disconnect();
SDL2.audio.scriptProcessorNode = undefined;
}
SDL2.audio = undefined;
}
if ((SDL2.audioContext !== undefined) && (SDL2.audio === undefined) && (SDL2.capture === undefined)) {
SDL2.audioContext.close();
SDL2.audioContext = undefined;
}
}, this->iscapture);
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
static int
Emscripten_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_bool valid_format = SDL_FALSE;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
SDL_AudioFormat test_format;
int i;
float f;
int result;
/* based on parts of library_sdl.js */
/* create context (TODO: this puts stuff in the global namespace...)*/
result = EM_ASM_INT({
if(typeof(SDL2) === 'undefined') {
SDL2 = {};
}
if (!$0) {
SDL2.audio = {};
} else {
SDL2.capture = {};
}
if (!SDL2.audioContext) {
if (typeof(AudioContext) !== 'undefined') {
SDL2.audioContext = new AudioContext();
} else if (typeof(webkitAudioContext) !== 'undefined') {
SDL2.audioContext = new webkitAudioContext();
}
}
return SDL2.audioContext === undefined ? -1 : 0;
}, iscapture);
if (result < 0) {
return SDL_SetError("Web Audio API is not available!");
}
test_format = SDL_FirstAudioFormat(this->spec.format);
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_F32: /* web audio only supports floats */
@@ -181,36 +290,11 @@ Emscripten_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* based on parts of library_sdl.js */
/* create context (TODO: this puts stuff in the global namespace...)*/
result = EM_ASM_INT_V({
if(typeof(SDL2) === 'undefined')
SDL2 = {};
if(typeof(SDL2.audio) === 'undefined')
SDL2.audio = {};
if (!SDL2.audioContext) {
if (typeof(AudioContext) !== 'undefined') {
SDL2.audioContext = new AudioContext();
} else if (typeof(webkitAudioContext) !== 'undefined') {
SDL2.audioContext = new webkitAudioContext();
} else {
return -1;
}
}
return 0;
});
if (result < 0) {
return SDL_SetError("Web Audio API is not available!");
}
SDL_zerop(this->hidden);
/* limit to native freq */
int sampleRate = EM_ASM_INT_V({
return SDL2.audioContext['sampleRate'];
const int sampleRate = EM_ASM_INT_V({
return SDL2.audioContext.sampleRate;
});
if(this->spec.freq != sampleRate) {
@@ -227,26 +311,83 @@ Emscripten_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_CalculateAudioSpec(&this->spec);
/* setup a ScriptProcessorNode */
EM_ASM_ARGS({
SDL2.audio.scriptProcessorNode = SDL2.audioContext['createScriptProcessor']($1, 0, $0);
SDL2.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
SDL2.audio.currentOutputBuffer = e['outputBuffer'];
Runtime.dynCall('vi', $2, [$3]);
};
SDL2.audio.scriptProcessorNode['connect'](SDL2.audioContext['destination']);
}, this->spec.channels, this->spec.samples, HandleAudioProcess, this);
if (iscapture) {
/* The idea is to take the capture media stream, hook it up to an
audio graph where we can pass it through a ScriptProcessorNode
to access the raw PCM samples and push them to the SDL app's
callback. From there, we "process" the audio data into silence
and forget about it. */
/* This should, strictly speaking, use MediaRecorder for capture, but
this API is cleaner to use and better supported, and fires a
callback whenever there's enough data to fire down into the app.
The downside is that we are spending CPU time silencing a buffer
that the audiocontext uselessly mixes into any output. On the
upside, both of those things are not only run in native code in
the browser, they're probably SIMD code, too. MediaRecorder
feels like it's a pretty inefficient tapdance in similar ways,
to be honest. */
EM_ASM_({
var have_microphone = function(stream) {
//console.log('SDL audio capture: we have a microphone! Replacing silence callback.');
if (SDL2.capture.silenceTimer !== undefined) {
clearTimeout(SDL2.capture.silenceTimer);
SDL2.capture.silenceTimer = undefined;
}
SDL2.capture.mediaStreamNode = SDL2.audioContext.createMediaStreamSource(stream);
SDL2.capture.scriptProcessorNode = SDL2.audioContext.createScriptProcessor($1, $0, 1);
SDL2.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {
audioProcessingEvent.outputBuffer.getChannelData(0).fill(0.0);
SDL2.capture.currentCaptureBuffer = audioProcessingEvent.inputBuffer;
Runtime.dynCall('vi', $2, [$3]);
};
SDL2.capture.mediaStreamNode.connect(SDL2.capture.scriptProcessorNode);
SDL2.capture.scriptProcessorNode.connect(SDL2.audioContext.destination);
};
var no_microphone = function(error) {
//console.log('SDL audio capture: we DO NOT have a microphone! (' + error.name + ')...leaving silence callback running.');
};
/* we write silence to the audio callback until the microphone is available (user approves use, etc). */
SDL2.capture.silenceBuffer = SDL2.audioContext.createBuffer($0, $1, SDL2.audioContext.sampleRate);
SDL2.capture.silenceBuffer.getChannelData(0).fill(0.0);
var silence_callback = function() {
SDL2.capture.currentCaptureBuffer = SDL2.capture.silenceBuffer;
Runtime.dynCall('vi', $2, [$3]);
};
SDL2.capture.silenceTimer = setTimeout(silence_callback, ($1 / SDL2.audioContext.sampleRate) * 1000);
if ((navigator.mediaDevices !== undefined) && (navigator.mediaDevices.getUserMedia !== undefined)) {
navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(have_microphone).catch(no_microphone);
} else if (navigator.webkitGetUserMedia !== undefined) {
navigator.webkitGetUserMedia({ audio: true, video: false }, have_microphone, no_microphone);
}
}, this->spec.channels, this->spec.samples, HandleCaptureProcess, this);
} else {
/* setup a ScriptProcessorNode */
EM_ASM_ARGS({
SDL2.audio.scriptProcessorNode = SDL2.audioContext['createScriptProcessor']($1, 0, $0);
SDL2.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
SDL2.audio.currentOutputBuffer = e['outputBuffer'];
Runtime.dynCall('vi', $2, [$3]);
};
SDL2.audio.scriptProcessorNode['connect'](SDL2.audioContext['destination']);
}, this->spec.channels, this->spec.samples, HandleAudioProcess, this);
}
return 0;
}
static int
Emscripten_Init(SDL_AudioDriverImpl * impl)
EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = Emscripten_OpenDevice;
impl->CloseDevice = Emscripten_CloseDevice;
impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice;
impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice;
/* only one output */
impl->OnlyHasDefaultOutputDevice = 1;
/* no threads here */
@@ -254,7 +395,7 @@ Emscripten_Init(SDL_AudioDriverImpl * impl)
impl->ProvidesOwnCallbackThread = 1;
/* check availability */
int available = EM_ASM_INT_V({
const int available = EM_ASM_INT_V({
if (typeof(AudioContext) !== 'undefined') {
return 1;
} else if (typeof(webkitAudioContext) !== 'undefined') {
@@ -267,11 +408,23 @@ Emscripten_Init(SDL_AudioDriverImpl * impl)
SDL_SetError("No audio context available");
}
const int capture_available = available && EM_ASM_INT_V({
if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
return 1;
} else if (typeof(navigator.webkitGetUserMedia) !== 'undefined') {
return 1;
}
return 0;
});
impl->HasCaptureSupport = capture_available ? SDL_TRUE : SDL_FALSE;
impl->OnlyHasDefaultCaptureDevice = capture_available ? SDL_TRUE : SDL_FALSE;
return available;
}
AudioBootStrap EmscriptenAudio_bootstrap = {
"emscripten", "SDL emscripten audio driver", Emscripten_Init, 0
AudioBootStrap EMSCRIPTENAUDIO_bootstrap = {
"emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_EMSCRIPTEN */
src/audio/esd/SDL_esdaudio.c
+6 -16
View File
@@ -32,7 +32,6 @@
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_esdaudio.h"
@@ -174,17 +173,11 @@ ESD_GetDeviceBuf(_THIS)
static void
ESD_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->audio_fd >= 0) {
SDL_NAME(esd_close) (this->hidden->audio_fd);
this->hidden->audio_fd = -1;
}
SDL_free(this->hidden);
this->hidden = NULL;
if (this->hidden->audio_fd >= 0) {
SDL_NAME(esd_close) (this->hidden->audio_fd);
}
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
/* Try to get the name of the program */
@@ -227,7 +220,7 @@ ESD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
this->hidden->audio_fd = -1;
/* Convert audio spec to the ESD audio format */
@@ -252,7 +245,6 @@ ESD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
if (!found) {
ESD_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
@@ -271,7 +263,6 @@ ESD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
get_progname());
if (this->hidden->audio_fd < 0) {
ESD_CloseDevice(this);
return SDL_SetError("Couldn't open ESD connection");
}
@@ -283,9 +274,8 @@ ESD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ESD_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
src/audio/fusionsound/SDL_fsaudio.c
+11 -20
View File
@@ -22,6 +22,8 @@
#if SDL_AUDIO_DRIVER_FUSIONSOUND
/* !!! FIXME: why is this is SDL_FS_* instead of FUSIONSOUND_*? */
/* Allow access to a raw mixing buffer */
#ifdef HAVE_SIGNAL_H
@@ -31,7 +33,6 @@
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_fsaudio.h"
@@ -168,20 +169,14 @@ SDL_FS_GetDeviceBuf(_THIS)
static void
SDL_FS_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->stream) {
this->hidden->stream->Release(this->hidden->stream);
this->hidden->stream = NULL;
}
if (this->hidden->fs) {
this->hidden->fs->Release(this->hidden->fs);
this->hidden->fs = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
if (this->hidden->stream) {
this->hidden->stream->Release(this->hidden->stream);
}
if (this->hidden->fs) {
this->hidden->fs->Release(this->hidden->fs);
}
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
@@ -200,7 +195,7 @@ SDL_FS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
@@ -239,7 +234,6 @@ SDL_FS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
if (format == 0) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
@@ -247,7 +241,6 @@ SDL_FS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Retrieve the main sound interface. */
ret = SDL_NAME(FusionSoundCreate) (&this->hidden->fs);
if (ret) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Unable to initialize FusionSound: %d", ret);
}
@@ -266,7 +259,6 @@ SDL_FS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
this->hidden->fs->CreateStream(this->hidden->fs, &desc,
&this->hidden->stream);
if (ret) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Unable to create FusionSoundStream: %d", ret);
}
@@ -285,9 +277,8 @@ SDL_FS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
SDL_FS_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
src/audio/haiku/SDL_haikuaudio.cc
+9 -15
View File
@@ -49,10 +49,11 @@ FillSound(void *device, void *stream, size_t len,
SDL_AudioDevice *audio = (SDL_AudioDevice *) device;
/* Only do soemthing if audio is enabled */
if (!audio->enabled)
if (!SDL_AtomicGet(&audio->enabled)) {
return;
}
if (!audio->paused) {
if (!SDL_AtomicGet(&audio->paused)) {
if (audio->convert.needed) {
SDL_LockMutex(audio->mixer_lock);
(*audio->spec.callback) (audio->spec.userdata,
@@ -73,16 +74,11 @@ FillSound(void *device, void *stream, size_t len,
static void
HAIKUAUDIO_CloseDevice(_THIS)
{
if (_this->hidden != NULL) {
if (_this->hidden->audio_obj) {
_this->hidden->audio_obj->Stop();
delete _this->hidden->audio_obj;
_this->hidden->audio_obj = NULL;
}
delete _this->hidden;
_this->hidden = NULL;
if (_this->hidden->audio_obj) {
_this->hidden->audio_obj->Stop();
delete _this->hidden->audio_obj;
}
delete _this->hidden;
}
@@ -122,10 +118,10 @@ HAIKUAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (_this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(_this->hidden, 0, (sizeof *_this->hidden));
SDL_zerop(this->hidden);
/* Parse the audio format and fill the Be raw audio format */
SDL_memset(&format, '\0', sizeof(media_raw_audio_format));
SDL_zero(format);
format.byte_order = B_MEDIA_LITTLE_ENDIAN;
format.frame_rate = (float) _this->spec.freq;
format.channel_count = _this->spec.channels; /* !!! FIXME: support > 2? */
@@ -176,7 +172,6 @@ HAIKUAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
if (!valid_datatype) { /* shouldn't happen, but just in case... */
HAIKUAUDIO_CloseDevice(_this);
return SDL_SetError("Unsupported audio format");
}
@@ -195,7 +190,6 @@ HAIKUAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (_this->hidden->audio_obj->Start() == B_NO_ERROR) {
_this->hidden->audio_obj->SetHasData(true);
} else {
HAIKUAUDIO_CloseDevice(_this);
return SDL_SetError("Unable to start Be audio");
}

Some files were not shown because too many files have changed in this diff Show More