diff --git a/.gitattributes b/.gitattributes
new file mode 100755
index 0000000000000000000000000000000000000000..77510b228d66e174cad661a19350ad00b2ac41b1
--- /dev/null
+++ b/.gitattributes
@@ -0,0 +1 @@
+*.pd binary
diff --git a/.gitignore b/.gitignore
index ae3c406d9349c527a5403db831d1bcf6d5269be8..71d246e17d8fc1cd9c7ae74e060e5309071fdbe9 100644
--- a/.gitignore
+++ b/.gitignore
@@ -2,3 +2,4 @@
*.o
*.so
*.os
+*.import
diff --git a/GdpdExample/Main.tscn b/GdpdExample/Main.tscn
index 9c9c91d01212770da591b055664da4a17323640b..8899168122e119736af3b4183b813d2987b3cad2 100644
--- a/GdpdExample/Main.tscn
+++ b/GdpdExample/Main.tscn
@@ -3,19 +3,15 @@
[sub_resource type="GDScript" id=1]
script/source = "extends Control
-var _gdpd
-
-var _patch = 0
+onready var _gdpd = load(\"res://addons/gdpd/bin/gdpd.gdns\").new()
func _ready():
- _gdpd = load(\"res://addons/gdpd/bin/gdpd.gdns\").new()
pass
func _process(delta):
while _gdpd.has_message() :
- print(\"got msg\")
var msg = _gdpd.get_next()
- print(msg)
+ print(\"got message from pd \", msg)
func _load_patch(pd_patch) :
#separate file name from directory
@@ -26,27 +22,44 @@ func _load_patch(pd_patch) :
_gdpd.openfile(patch_name, patch_dir)
func _on_Start_pressed() :
+ #retrieve the list of available input and outputs
var inps = _gdpd.get_available_input_devices()
var outs = _gdpd.get_available_output_devices()
+
+ #initialise the first ones
_gdpd.init_devices(inps[0], outs[0])
#the patch path should be the absolute one
_load_patch(ProjectSettings.globalize_path(\"res://patch1.pd\"))
_load_patch(ProjectSettings.globalize_path(\"res://patch2.pd\"))
+ #send message to [receive from_godot] with one symbol
_gdpd.start_message(1)
- _gdpd.add_symbol(\"from godot\")
- _gdpd.finish_list(\"blup\")
-
- _gdpd.subscribe(\"toBfWeb\")
+ _gdpd.add_symbol(\"hello\")
+ _gdpd.finish_list(\"from_godot\")
-func _on_Stop_pressed():
+ #listen to messages sent with [send to_godot]
+ _gdpd.subscribe(\"to_godot\")
+func _on_Stop_pressed():
_gdpd.closefile(\"patch1.pd\")
_gdpd.closefile(\"patch2.pd\")
_gdpd.stop()
+func _on_HSlider_value_changed(value):
+ #send message to [receive from_godot] with three elements
+ _gdpd.start_message(3)
+ _gdpd.add_symbol(\"patch1\")
+ _gdpd.add_symbol(\"pitch\")
+ _gdpd.add_float(value)
+ _gdpd.finish_list(\"from_godot\")
+func _on_HSlider2_value_changed(value):
+ _gdpd.start_message(3)
+ _gdpd.add_symbol(\"patch2\")
+ _gdpd.add_symbol(\"pitch\")
+ _gdpd.add_float(value)
+ _gdpd.finish_list(\"from_godot\")
"
[node name="Control" type="Control"]
@@ -58,28 +71,60 @@ __meta__ = {
}
[node name="Stop" type="Button" parent="."]
-margin_left = 73.0
-margin_top = 198.0
-margin_right = 176.0
-margin_bottom = 244.0
+margin_left = 213.0
+margin_top = 83.0
+margin_right = 316.0
+margin_bottom = 129.0
text = "Stop"
__meta__ = {
"_edit_use_anchors_": false
}
[node name="Start" type="Button" parent="."]
-margin_left = 83.0
-margin_top = 93.0
-margin_right = 187.0
-margin_bottom = 144.0
+margin_left = 73.0
+margin_top = 83.0
+margin_right = 177.0
+margin_bottom = 134.0
text = "Start"
-[node name="Load" type="Button" parent="."]
-margin_left = 103.0
-margin_top = 159.0
-margin_right = 145.0
-margin_bottom = 179.0
-text = "Load"
+[node name="HSlider" type="HSlider" parent="."]
+margin_left = 73.0
+margin_top = 193.0
+margin_right = 406.0
+margin_bottom = 209.0
+max_value = 12.0
+__meta__ = {
+"_edit_use_anchors_": false
+}
+
+[node name="Label" type="Label" parent="."]
+margin_left = 73.0
+margin_top = 170.0
+margin_right = 113.0
+margin_bottom = 184.0
+text = "Send pitch to patch1"
+
+[node name="HSlider2" type="HSlider" parent="."]
+margin_left = 73.0
+margin_top = 275.0
+margin_right = 406.0
+margin_bottom = 291.0
+max_value = 12.0
+__meta__ = {
+"_edit_use_anchors_": false
+}
+
+[node name="Label2" type="Label" parent="."]
+margin_left = 73.0
+margin_top = 246.0
+margin_right = 204.0
+margin_bottom = 260.0
+text = "Send pitch to patch2"
+__meta__ = {
+"_edit_use_anchors_": false
+}
+
[connection signal="pressed" from="Stop" to="." method="_on_Stop_pressed"]
[connection signal="pressed" from="Start" to="." method="_on_Start_pressed"]
-[connection signal="pressed" from="Load" to="." method="_on_Load_pressed"]
+[connection signal="value_changed" from="HSlider" to="." method="_on_HSlider_value_changed"]
+[connection signal="value_changed" from="HSlider2" to="." method="_on_HSlider2_value_changed"]
diff --git a/GdpdExample/icon.png.import b/GdpdExample/icon.png.import
index 96cbf4629a557eb7339eef87ad61b7e7d8c8f0e7..a4c02e6e27173699ab89f60c32661726fadfdbcd 100644
--- a/GdpdExample/icon.png.import
+++ b/GdpdExample/icon.png.import
@@ -28,6 +28,7 @@ process/fix_alpha_border=true
process/premult_alpha=false
process/HDR_as_SRGB=false
process/invert_color=false
+process/normal_map_invert_y=false
stream=false
size_limit=0
detect_3d=true
diff --git a/GdpdExample/patch1.pd b/GdpdExample/patch1.pd
index 3c3f092e53e168b62056dfd4cffeee438ef2843c..295f29d7515db9dca6c5e2786ea944188d612441 100644
Binary files a/GdpdExample/patch1.pd and b/GdpdExample/patch1.pd differ
diff --git a/GdpdExample/patch2.pd b/GdpdExample/patch2.pd
index 099314ff1570d8d522778c1d39ceece6d758e04f..e4495276a278f03fd8fe1962354f243a9d8667e1 100644
Binary files a/GdpdExample/patch2.pd and b/GdpdExample/patch2.pd differ
diff --git a/GdpdExample/project.godot b/GdpdExample/project.godot
index efa53133e88dc4f346b0297d5503b25119e90be8..1daf32513ea13ea5cd4b3482417820a0bd9c6849 100644
--- a/GdpdExample/project.godot
+++ b/GdpdExample/project.godot
@@ -10,7 +10,6 @@ config_version=4
_global_script_classes=[ ]
_global_script_class_icons={
-
}
[application]
diff --git a/README.md b/README.md
index 09b66642817c0585eb074640c23f5c9e32ce936f..a9ec8098befc7b8ac9a2ecb57cd6052c0d49819e 100644
--- a/README.md
+++ b/README.md
@@ -7,8 +7,9 @@ Godot add-on for loading and processing Pure Data patches
## Compiling on GNU/Linux
-- Open a terminal
-- Type the following commandes :
+- Install the following packages :
+ - scons
+- Open a terminal and type the following commands :
- git clone https://gitlab.univ-lille.fr/ivmi/gdpd.git
- cd gdpd
- ./update.sh linux
@@ -16,7 +17,7 @@ Godot add-on for loading and processing Pure Data patches
## Compiling on Windows
- Open the Windows Powershell
-- Type the following commandes :
+- Type the following commands :
- git clone https://gitlab.univ-lille.fr/ivmi/gdpd.git
- cd gdpd
- ./update.sh windows
diff --git a/SConstruct b/SConstruct
index a7fdfe20c9a3884d3bc45a009ad3c39bfcbb2379..a7076dd7583cbe8156ace4527eb728bf64c89b14 100644
--- a/SConstruct
+++ b/SConstruct
@@ -4,7 +4,7 @@ import os, subprocess
opts = Variables([], ARGUMENTS)
# Gets the standard flags CC, CCX, etc.
-env = DefaultEnvironment()
+env = Environment(ENV = os.environ)
# Define our options
opts.Add(EnumVariable('target', "Compilation target", 'release', ['d', 'debug', 'r', 'release']))
@@ -61,12 +61,13 @@ if env['platform'] == "osx":
env.Append(CCFLAGS=['-g', '-O3', '-arch', 'x86_64'])
elif env['platform'] in ('x11', 'linux'):
- env['CC'] = 'gcc-7'
- env['CXX'] = 'g++-7'
+ env['CC'] = 'gcc'
+ env['CXX'] = 'g++'
env['target_path'] += 'x11/'
cpp_library += '.linux'
env.Append(CPPDEFINES=['__UNIX_JACK__', 'LIBPD_EXTRA'])
- env.Append(LINKFLAGS=['-ljack','-pthread'])
+ env.Append(LIBS=['jack', 'pthread'])
+ env.Append(LDPATH=['/usr/lib/x86_64-linux-gnu'])
if env['target'] in ('debug', 'd'):
env.Append(CCFLAGS=['-fPIC', '-g3', '-Og'])
env.Append(CFLAGS=['-std=c11'])
@@ -80,6 +81,7 @@ elif env['platform'] == "windows":
env['target_path'] += 'win64/'
cpp_library += '.windows'
env.Append(ENV=os.environ)
+ env.Append(CPPDEFINES=['NOMINMAX'])
if not env['use_mingw']:
# MSVC
diff --git a/src/gdpd.hpp b/src/gdpd.hpp
index 3bf3ea1c3725b8ce6c0c8b0726d3639fdd0d21d2..9069ffed72b710ea5b2fcc8016f4f4a00a60a899 100644
--- a/src/gdpd.hpp
+++ b/src/gdpd.hpp
@@ -5,6 +5,7 @@
#include <stdlib.h>
#include <string.h>
#include <map>
+#include <algorithm>
#include <Godot.hpp>
#include <AudioStreamPlayer.hpp>
diff --git a/src/rtaudio/RtAudio.cpp b/src/rtaudio/RtAudio.cpp
index 0837d98ef1e8d6eafecd51505ba6d6f7f569552d..40fb6a9091b7a21aee3c9481bf120c802650881c 100644
--- a/src/rtaudio/RtAudio.cpp
+++ b/src/rtaudio/RtAudio.cpp
@@ -11,7 +11,7 @@
RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
RtAudio: realtime audio i/o C++ classes
- Copyright (c) 2001-2019 Gary P. Scavone
+ Copyright (c) 2001-2021 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
@@ -39,7 +39,7 @@
*/
/************************************************************************/
-// RtAudio: Version 5.1.0
+// RtAudio: Version 5.2.0
#include "RtAudio.h"
#include <iostream>
@@ -56,7 +56,7 @@ const unsigned int RtApi::SAMPLE_RATES[] = {
32000, 44100, 48000, 88200, 96000, 176400, 192000
};
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
+#if defined(_WIN32) || defined(__CYGWIN__)
#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
#define MUTEX_DESTROY(A) DeleteCriticalSection(A)
#define MUTEX_LOCK(A) EnterCriticalSection(A)
@@ -64,12 +64,17 @@ const unsigned int RtApi::SAMPLE_RATES[] = {
#include "tchar.h"
- static std::string convertCharPointerToStdString(const char *text)
+ template<typename T> inline
+ std::string convertCharPointerToStdString(const T *text);
+
+ template<> inline
+ std::string convertCharPointerToStdString(const char *text)
{
return std::string(text);
}
- static std::string convertCharPointerToStdString(const wchar_t *text)
+ template<> inline
+ std::string convertCharPointerToStdString(const wchar_t *text)
{
int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
std::string s( length-1, '\0' );
@@ -77,15 +82,12 @@ const unsigned int RtApi::SAMPLE_RATES[] = {
return s;
}
-#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+#elif defined(__unix__) || defined(__APPLE__)
// pthread API
#define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
#define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
#define MUTEX_LOCK(A) pthread_mutex_lock(A)
#define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
-#else
- #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
- #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
#endif
// *************************************************** //
@@ -262,7 +264,7 @@ RtAudio :: RtAudio( RtAudio::Api api )
// It should not be possible to get here because the preprocessor
// definition __RTAUDIO_DUMMY__ is automatically defined if no
// API-specific definitions are passed to the compiler. But just in
- // case something weird happens, we'll thow an error.
+ // case something weird happens, we'll throw an error.
std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
}
@@ -405,13 +407,27 @@ void RtApi :: openStream( RtAudio::StreamParameters *oParams,
unsigned int RtApi :: getDefaultInputDevice( void )
{
- // Should be implemented in subclasses if possible.
+ // Should be reimplemented in subclasses if necessary.
+ unsigned int nDevices = getDeviceCount();
+ for ( unsigned int i = 0; i < nDevices; i++ ) {
+ if ( getDeviceInfo( i ).isDefaultInput ) {
+ return i;
+ }
+ }
+
return 0;
}
unsigned int RtApi :: getDefaultOutputDevice( void )
{
- // Should be implemented in subclasses if possible.
+ // Should be reimplemented in subclasses if necessary.
+ unsigned int nDevices = getDeviceCount();
+ for ( unsigned int i = 0; i < nDevices; i++ ) {
+ if ( getDeviceInfo( i ).isDefaultOutput ) {
+ return i;
+ }
+ }
+
return 0;
}
@@ -506,6 +522,8 @@ unsigned int RtApi :: getStreamSampleRate( void )
#if defined(__MACOSX_CORE__)
+#include <unistd.h>
+
// The OS X CoreAudio API is designed to use a separate callback
// procedure for each of its audio devices. A single RtAudio duplex
// stream using two different devices is supported here, though it
@@ -1478,15 +1496,17 @@ void RtApiCore :: closeStream( void )
errorText_ = "RtApiCore::closeStream(): error removing property listener!";
error( RtAudioError::WARNING );
}
- }
- if ( stream_.state == STREAM_RUNNING )
- AudioDeviceStop( handle->id[0], callbackHandler );
+
#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
-#else
- // deprecated in favor of AudioDeviceDestroyIOProcID()
- AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[0], handle->procId[0] );
+ AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[0], callbackHandler );
+ AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
#endif
+ }
}
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
@@ -1501,15 +1521,17 @@ void RtApiCore :: closeStream( void )
errorText_ = "RtApiCore::closeStream(): error removing property listener!";
error( RtAudioError::WARNING );
}
- }
- if ( stream_.state == STREAM_RUNNING )
- AudioDeviceStop( handle->id[1], callbackHandler );
+
#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
- AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
-#else
- // deprecated in favor of AudioDeviceDestroyIOProcID()
- AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[1], handle->procId[1] );
+ AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[1], callbackHandler );
+ AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
#endif
+ }
}
for ( int i=0; i<2; i++ ) {
@@ -1542,15 +1564,19 @@ void RtApiCore :: startStream( void )
return;
}
- #if defined( HAVE_GETTIMEOFDAY )
+#if defined( HAVE_GETTIMEOFDAY )
gettimeofday( &stream_.lastTickTimestamp, NULL );
- #endif
+#endif
OSStatus result = noErr;
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceStart( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
result = AudioDeviceStart( handle->id[0], callbackHandler );
+#endif
if ( result != noErr ) {
errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
@@ -1561,7 +1587,11 @@ void RtApiCore :: startStream( void )
if ( stream_.mode == INPUT ||
( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceStart( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
result = AudioDeviceStart( handle->id[1], callbackHandler );
+#endif
if ( result != noErr ) {
errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
errorText_ = errorStream_.str();
@@ -1596,7 +1626,11 @@ void RtApiCore :: stopStream( void )
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceStop( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
result = AudioDeviceStop( handle->id[0], callbackHandler );
+#endif
if ( result != noErr ) {
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
@@ -1606,7 +1640,11 @@ void RtApiCore :: stopStream( void )
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceStop( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
result = AudioDeviceStop( handle->id[1], callbackHandler );
+#endif
if ( result != noErr ) {
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
errorText_ = errorStream_.str();
@@ -2790,13 +2828,13 @@ static long asioMessages( long selector, long value, void* message, double* opt
RtApiAsio :: RtApiAsio()
{
- // ASIO cannot run on a multi-threaded appartment. You can call
- // CoInitialize beforehand, but it must be for appartment threading
+ // ASIO cannot run on a multi-threaded apartment. You can call
+ // CoInitialize beforehand, but it must be for apartment threading
// (in which case, CoInitilialize will return S_FALSE here).
coInitialized_ = false;
HRESULT hr = CoInitialize( NULL );
if ( FAILED(hr) ) {
- errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+ errorText_ = "RtApiAsio::ASIO requires a single-threaded apartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
error( RtAudioError::WARNING );
}
coInitialized_ = true;
@@ -2819,6 +2857,18 @@ unsigned int RtApiAsio :: getDeviceCount( void )
return (unsigned int) drivers.asioGetNumDev();
}
+// We can only load one ASIO driver, so the default output is always the first device.
+unsigned int RtApiAsio :: getDefaultOutputDevice( void )
+{
+ return 0;
+}
+
+// We can only load one ASIO driver, so the default input is always the first device.
+unsigned int RtApiAsio :: getDefaultInputDevice( void )
+{
+ return 0;
+}
+
RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
{
RtAudio::DeviceInfo info;
@@ -3799,6 +3849,17 @@ if ( objectPtr )\
typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
+#ifndef __IAudioClient3_INTERFACE_DEFINED__
+MIDL_INTERFACE( "00000000-0000-0000-0000-000000000000" ) IAudioClient3
+{
+ virtual HRESULT GetSharedModeEnginePeriod( WAVEFORMATEX*, UINT32*, UINT32*, UINT32*, UINT32* ) = 0;
+ virtual HRESULT InitializeSharedAudioStream( DWORD, UINT32, WAVEFORMATEX*, LPCGUID ) = 0;
+};
+#ifdef __CRT_UUID_DECL
+__CRT_UUID_DECL( IAudioClient3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 )
+#endif
+#endif
+
//-----------------------------------------------------------------------------
// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
@@ -4053,7 +4114,7 @@ public:
#endif
}
- void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount )
+ void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount, int maxOutSampleCount = -1 )
{
unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
if ( _sampleRatio == 1 )
@@ -4064,7 +4125,15 @@ public:
return;
}
- unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
+ unsigned int outputBufferSize = 0;
+ if ( maxOutSampleCount != -1 )
+ {
+ outputBufferSize = _bytesPerSample * _channelCount * maxOutSampleCount;
+ }
+ else
+ {
+ outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
+ }
IMFMediaBuffer* rInBuffer;
IMFSample* rInSample;
@@ -4486,34 +4555,6 @@ Exit:
return info;
}
-//-----------------------------------------------------------------------------
-
-unsigned int RtApiWasapi::getDefaultOutputDevice( void )
-{
- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
- if ( getDeviceInfo( i ).isDefaultOutput ) {
- return i;
- }
- }
-
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
-unsigned int RtApiWasapi::getDefaultInputDevice( void )
-{
- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
- if ( getDeviceInfo( i ).isDefaultInput ) {
- return i;
- }
- }
-
- return 0;
-}
-
-//-----------------------------------------------------------------------------
-
void RtApiWasapi::closeStream( void )
{
if ( stream_.state == STREAM_CLOSED ) {
@@ -4600,17 +4641,16 @@ void RtApiWasapi::stopStream( void )
error( RtAudioError::WARNING );
return;
}
+ if ( stream_.state == STREAM_STOPPING ) {
+ errorText_ = "RtApiWasapi::stopStream: The stream is already stopping.";
+ error( RtAudioError::WARNING );
+ return;
+ }
// inform stream thread by setting stream state to STREAM_STOPPING
stream_.state = STREAM_STOPPING;
- // wait until stream thread is stopped
- while( stream_.state != STREAM_STOPPED ) {
- Sleep( 1 );
- }
-
- // Wait for the last buffer to play before stopping.
- Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
+ WaitForSingleObject( ( void* ) stream_.callbackInfo.thread, INFINITE );
// close thread handle
if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
@@ -4633,14 +4673,16 @@ void RtApiWasapi::abortStream( void )
error( RtAudioError::WARNING );
return;
}
+ if ( stream_.state == STREAM_STOPPING ) {
+ errorText_ = "RtApiWasapi::abortStream: The stream is already stopping.";
+ error( RtAudioError::WARNING );
+ return;
+ }
// inform stream thread by setting stream state to STREAM_STOPPING
stream_.state = STREAM_STOPPING;
- // wait until stream thread is stopped
- while ( stream_.state != STREAM_STOPPED ) {
- Sleep( 1 );
- }
+ WaitForSingleObject( ( void* ) stream_.callbackInfo.thread, INFINITE );
// close thread handle
if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
@@ -4849,7 +4891,7 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
stream_.doConvertBuffer[mode] = true;
if ( stream_.doConvertBuffer[mode] )
- setConvertInfo( mode, 0 );
+ setConvertInfo( mode, firstChannel );
// Allocate necessary internal buffers
bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
@@ -4941,7 +4983,7 @@ void RtApiWasapi::wasapiThread()
// declare local stream variables
RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
BYTE* streamBuffer = NULL;
- unsigned long captureFlags = 0;
+ DWORD captureFlags = 0;
unsigned int bufferFrameCount = 0;
unsigned int numFramesPadding = 0;
unsigned int convBufferSize = 0;
@@ -4960,7 +5002,7 @@ void RtApiWasapi::wasapiThread()
RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
// Attempt to assign "Pro Audio" characteristic to thread
- HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
+ HMODULE AvrtDll = LoadLibraryW( L"AVRT.dll" );
if ( AvrtDll ) {
DWORD taskIndex = 0;
TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr =
@@ -4985,12 +5027,37 @@ void RtApiWasapi::wasapiThread()
captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
if ( !captureClient ) {
- hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
- loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
- 0,
- 0,
- captureFormat,
- NULL );
+ IAudioClient3* captureAudioClient3 = nullptr;
+ captureAudioClient->QueryInterface( __uuidof( IAudioClient3 ), ( void** ) &captureAudioClient3 );
+ if ( captureAudioClient3 && !loopbackEnabled )
+ {
+ UINT32 Ignore;
+ UINT32 MinPeriodInFrames;
+ hr = captureAudioClient3->GetSharedModeEnginePeriod( captureFormat,
+ &Ignore,
+ &Ignore,
+ &MinPeriodInFrames,
+ &Ignore );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient3->InitializeSharedAudioStream( AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ MinPeriodInFrames,
+ captureFormat,
+ NULL );
+ }
+ else
+ {
+ hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ 0,
+ 0,
+ captureFormat,
+ NULL );
+ }
+
if ( FAILED( hr ) ) {
errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
goto Exit;
@@ -5071,12 +5138,37 @@ void RtApiWasapi::wasapiThread()
renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
if ( !renderClient ) {
- hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
- 0,
- 0,
- renderFormat,
- NULL );
+ IAudioClient3* renderAudioClient3 = nullptr;
+ renderAudioClient->QueryInterface( __uuidof( IAudioClient3 ), ( void** ) &renderAudioClient3 );
+ if ( renderAudioClient3 )
+ {
+ UINT32 Ignore;
+ UINT32 MinPeriodInFrames;
+ hr = renderAudioClient3->GetSharedModeEnginePeriod( renderFormat,
+ &Ignore,
+ &Ignore,
+ &MinPeriodInFrames,
+ &Ignore );
+ if ( FAILED( hr ) ) {
+ errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient3->InitializeSharedAudioStream( AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ MinPeriodInFrames,
+ renderFormat,
+ NULL );
+ }
+ else
+ {
+ hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ 0,
+ 0,
+ renderFormat,
+ NULL );
+ }
+
if ( FAILED( hr ) ) {
errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
goto Exit;
@@ -5140,18 +5232,18 @@ void RtApiWasapi::wasapiThread()
if ( stream_.mode == INPUT )
{
using namespace std; // for ceilf
- convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ convBuffSize = ( unsigned int ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
}
else if ( stream_.mode == OUTPUT )
{
- convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ convBuffSize = ( unsigned int ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
}
else if ( stream_.mode == DUPLEX )
{
- convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
- ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+ convBuffSize = std::max( ( unsigned int ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ ( unsigned int ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
}
@@ -5177,11 +5269,6 @@ void RtApiWasapi::wasapiThread()
if ( captureAudioClient )
{
int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
- if ( captureSrRatio != 1 )
- {
- // account for remainders
- samplesToPull--;
- }
convBufferSize = 0;
while ( convBufferSize < stream_.bufferSize )
@@ -5203,7 +5290,8 @@ void RtApiWasapi::wasapiThread()
captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
convBuffer,
samplesToPull,
- convSamples );
+ convSamples,
+ convBufferSize == 0 ? -1 : stream_.bufferSize - convBufferSize );
convBufferSize += convSamples;
samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
@@ -5471,14 +5559,14 @@ Exit:
CoUninitialize();
- // update stream state
- stream_.state = STREAM_STOPPED;
-
if ( !errorText.empty() )
{
errorText_ = errorText;
error( errorType );
}
+
+ // update stream state
+ stream_.state = STREAM_STOPPED;
}
//******************** End of __WINDOWS_WASAPI__ *********************//
@@ -6863,7 +6951,7 @@ void RtApiDs :: callbackEvent()
if ( stream_.mode == DUPLEX ) {
if ( safeReadPointer < endRead ) {
if ( duplexPrerollBytes <= 0 ) {
- // Pre-roll time over. Be more agressive.
+ // Pre-roll time over. Be more aggressive.
int adjustment = endRead-safeReadPointer;
handle->xrun[1] = true;
@@ -7018,19 +7106,18 @@ static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
// If good device, then save its name and guid.
std::string name = convertCharPointerToStdString( description );
- //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
- if ( lpguid == NULL )
- name = "Default Device";
+
if ( validDevice ) {
for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
if ( dsDevices[i].name == name ) {
- dsDevices[i].found = true;
- if ( probeInfo.isInput ) {
- dsDevices[i].id[1] = lpguid;
+ if ( probeInfo.isInput && dsDevices[i].id[1] == lpguid)
+ {
+ dsDevices[i].found = true;
dsDevices[i].validId[1] = true;
}
- else {
- dsDevices[i].id[0] = lpguid;
+ else if (dsDevices[i].id[0] == lpguid)
+ {
+ dsDevices[i].found = true;
dsDevices[i].validId[0] = true;
}
return TRUE;
@@ -7126,7 +7213,11 @@ struct AlsaHandle {
bool runnable;
AlsaHandle()
- :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
+#if _cplusplus >= 201103L
+ :handles{nullptr, nullptr}, synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
+#else
+ : synchronized(false), runnable(false) { handles[0] = NULL; handles[1] = NULL; xrun[0] = false; xrun[1] = false; }
+#endif
};
static void *alsaCallbackHandler( void * ptr );
@@ -7148,6 +7239,13 @@ unsigned int RtApiAlsa :: getDeviceCount( void )
char name[64];
snd_ctl_t *handle = 0;
+ strcpy(name, "default");
+ result = snd_ctl_open( &handle, "default", 0 );
+ if (result == 0) {
+ nDevices++;
+ snd_ctl_close( handle );
+ }
+
// Count cards and devices
card = -1;
snd_card_next( &card );
@@ -7180,12 +7278,6 @@ unsigned int RtApiAlsa :: getDeviceCount( void )
snd_card_next( &card );
}
- result = snd_ctl_open( &handle, "default", 0 );
- if (result == 0) {
- nDevices++;
- snd_ctl_close( handle );
- }
-
return nDevices;
}
@@ -7195,13 +7287,21 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
info.probed = false;
unsigned nDevices = 0;
- int result, subdevice, card;
+ int result=-1, subdevice=-1, card=-1;
char name[64];
snd_ctl_t *chandle = 0;
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices++ == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
+ }
+ }
+ if ( chandle )
+ snd_ctl_close( chandle );
+
// Count cards and devices
- card = -1;
- subdevice = -1;
snd_card_next( &card );
while ( card >= 0 ) {
sprintf( name, "hw:%d", card );
@@ -7235,15 +7335,6 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
snd_card_next( &card );
}
- result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
- if ( result == 0 ) {
- if ( nDevices == device ) {
- strcpy( name, "default" );
- goto foundDevice;
- }
- nDevices++;
- }
-
if ( nDevices == 0 ) {
errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
error( RtAudioError::INVALID_USE );
@@ -7464,11 +7555,13 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
}
// Get the device name
- char *cardname;
- result = snd_card_get_name( card, &cardname );
- if ( result >= 0 ) {
- sprintf( name, "hw:%s,%d", cardname, subdevice );
- free( cardname );
+ if (strncmp(name, "default", 7)!=0) {
+ char *cardname;
+ result = snd_card_get_name( card, &cardname );
+ if ( result >= 0 ) {
+ sprintf( name, "hw:%s,%d", cardname, subdevice );
+ free( cardname );
+ }
}
info.name = name;
@@ -7495,8 +7588,12 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
{
#if defined(__RTAUDIO_DEBUG__)
- snd_output_t *out;
- snd_output_stdio_attach(&out, stderr, 0);
+ struct SndOutputTdealloc {
+ SndOutputTdealloc() : _out(NULL) { snd_output_stdio_attach(&_out, stderr, 0); }
+ ~SndOutputTdealloc() { snd_output_close(_out); }
+ operator snd_output_t*() { return _out; }
+ snd_output_t *_out;
+ } out;
#endif
// I'm not using the "plug" interface ... too much inconsistent behavior.
@@ -7506,9 +7603,23 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
char name[64];
snd_ctl_t *chandle;
- if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
- snprintf(name, sizeof(name), "%s", "default");
+ if ( device == 0
+ || (options && options->flags & RTAUDIO_ALSA_USE_DEFAULT) )
+ {
+ strcpy(name, "default");
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ snd_ctl_close( chandle );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ }
+
else {
+ nDevices++;
// Count cards and devices
card = -1;
snd_card_next( &card );
@@ -7536,17 +7647,6 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
snd_card_next( &card );
}
- result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
- if ( result == 0 ) {
- if ( nDevices == device ) {
- strcpy( name, "default" );
- snd_ctl_close( chandle );
- goto foundDevice;
- }
- nDevices++;
- }
- snd_ctl_close( chandle );
-
if ( nDevices == 0 ) {
// This should not happen because a check is made before this function is called.
errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
@@ -8284,7 +8384,7 @@ void RtApiAlsa :: callbackEvent()
}
if ( result < (int) stream_.bufferSize ) {
- // Either an error or overrun occured.
+ // Either an error or overrun occurred.
if ( result == -EPIPE ) {
snd_pcm_state_t state = snd_pcm_state( handle[1] );
if ( state == SND_PCM_STATE_XRUN ) {
@@ -8354,7 +8454,7 @@ void RtApiAlsa :: callbackEvent()
}
if ( result < (int) stream_.bufferSize ) {
- // Either an error or underrun occured.
+ // Either an error or underrun occurred.
if ( result == -EPIPE ) {
snd_pcm_state_t state = snd_pcm_state( handle[0] );
if ( state == SND_PCM_STATE_XRUN ) {
@@ -8424,10 +8524,27 @@ static void *alsaCallbackHandler( void *ptr )
#include <pulse/error.h>
#include <pulse/simple.h>
+#include <pulse/pulseaudio.h>
#include <cstdio>
+static pa_mainloop_api *rt_pa_mainloop_api = NULL;
+struct PaDeviceInfo {
+ PaDeviceInfo() : sink_index(-1), source_index(-1) {}
+ int sink_index;
+ int source_index;
+ std::string sink_name;
+ std::string source_name;
+ RtAudio::DeviceInfo info;
+};
+static struct {
+ std::vector<PaDeviceInfo> dev;
+ std::string default_sink_name;
+ std::string default_source_name;
+ int default_rate;
+} rt_pa_info;
+
static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
- 44100, 48000, 96000, 0};
+ 44100, 48000, 96000, 192000, 0};
struct rtaudio_pa_format_mapping_t {
RtAudioFormat rtaudio_format;
@@ -8436,6 +8553,7 @@ struct rtaudio_pa_format_mapping_t {
static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
{RTAUDIO_SINT16, PA_SAMPLE_S16LE},
+ {RTAUDIO_SINT24, PA_SAMPLE_S24LE},
{RTAUDIO_SINT32, PA_SAMPLE_S32LE},
{RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
{0, PA_SAMPLE_INVALID}};
@@ -8449,35 +8567,219 @@ struct PulseAudioHandle {
PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
};
+static void rt_pa_mainloop_api_quit(int ret) {
+ rt_pa_mainloop_api->quit(rt_pa_mainloop_api, ret);
+}
+
+static void rt_pa_set_server_info(pa_context *context, const pa_server_info *info, void *data){
+ (void)context;
+ (void)data;
+ pa_sample_spec ss;
+
+ if (!info) {
+ rt_pa_mainloop_api_quit(1);
+ return;
+ }
+
+ ss = info->sample_spec;
+
+ rt_pa_info.default_rate = ss.rate;
+ rt_pa_info.default_sink_name = info->default_sink_name;
+ rt_pa_info.default_source_name = info->default_source_name;
+}
+
+static void rt_pa_set_sink_info(pa_context * /*c*/, const pa_sink_info *i,
+ int eol, void * /*userdata*/)
+{
+ if (eol) return;
+ PaDeviceInfo inf;
+ inf.info.name = pa_proplist_gets(i->proplist, "device.description");
+ inf.info.probed = true;
+ inf.info.outputChannels = i->sample_spec.channels;
+ inf.info.preferredSampleRate = i->sample_spec.rate;
+ inf.info.isDefaultOutput = (rt_pa_info.default_sink_name == i->name);
+ inf.sink_index = i->index;
+ inf.sink_name = i->name;
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+ inf.info.sampleRates.push_back( *sr );
+ for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats;
+ fm->rtaudio_format; ++fm )
+ inf.info.nativeFormats |= fm->rtaudio_format;
+ for (size_t i=0; i < rt_pa_info.dev.size(); i++)
+ {
+ /* Attempt to match up sink and source records by device description. */
+ if (rt_pa_info.dev[i].info.name == inf.info.name) {
+ rt_pa_info.dev[i].sink_index = inf.sink_index;
+ rt_pa_info.dev[i].sink_name = inf.sink_name;
+ rt_pa_info.dev[i].info.outputChannels = inf.info.outputChannels;
+ rt_pa_info.dev[i].info.isDefaultOutput = inf.info.isDefaultOutput;
+ /* Assume duplex channels are minimum of input and output channels. */
+ /* Uncomment if we add support for DUPLEX
+ if (rt_pa_info.dev[i].source_index > -1)
+ (inf.info.outputChannels < rt_pa_info.dev[i].info.inputChannels)
+ ? inf.info.outputChannels : rt_pa_info.dev[i].info.inputChannels;
+ */
+ return;
+ }
+ }
+ /* try to ensure device #0 is the default */
+ if (inf.info.isDefaultOutput)
+ rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf);
+ else
+ rt_pa_info.dev.push_back(inf);
+}
+
+static void rt_pa_set_source_info_and_quit(pa_context * /*c*/, const pa_source_info *i,
+ int eol, void * /*userdata*/)
+{
+ if (eol) {
+ rt_pa_mainloop_api_quit(0);
+ return;
+ }
+ PaDeviceInfo inf;
+ inf.info.name = pa_proplist_gets(i->proplist, "device.description");
+ inf.info.probed = true;
+ inf.info.inputChannels = i->sample_spec.channels;
+ inf.info.preferredSampleRate = i->sample_spec.rate;
+ inf.info.isDefaultInput = (rt_pa_info.default_source_name == i->name);
+ inf.source_index = i->index;
+ inf.source_name = i->name;
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+ inf.info.sampleRates.push_back( *sr );
+ for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats;
+ fm->rtaudio_format; ++fm )
+ inf.info.nativeFormats |= fm->rtaudio_format;
+
+ for (size_t i=0; i < rt_pa_info.dev.size(); i++)
+ {
+ /* Attempt to match up sink and source records by device description. */
+ if (rt_pa_info.dev[i].info.name == inf.info.name) {
+ rt_pa_info.dev[i].source_index = inf.source_index;
+ rt_pa_info.dev[i].source_name = inf.source_name;
+ rt_pa_info.dev[i].info.inputChannels = inf.info.inputChannels;
+ rt_pa_info.dev[i].info.isDefaultInput = inf.info.isDefaultInput;
+ /* Assume duplex channels are minimum of input and output channels. */
+ /* Uncomment if we add support for DUPLEX
+ if (rt_pa_info.dev[i].sink_index > -1) {
+ rt_pa_info.dev[i].info.duplexChannels =
+ (inf.info.inputChannels < rt_pa_info.dev[i].info.outputChannels)
+ ? inf.info.inputChannels : rt_pa_info.dev[i].info.outputChannels;
+ }
+ */
+ return;
+ }
+ }
+ /* try to ensure device #0 is the default */
+ if (inf.info.isDefaultInput)
+ rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf);
+ else
+ rt_pa_info.dev.push_back(inf);
+}
+
+static void rt_pa_context_state_callback(pa_context *context, void *userdata) {
+ (void)userdata;
+
+ auto state = pa_context_get_state(context);
+ switch (state) {
+ case PA_CONTEXT_CONNECTING:
+ case PA_CONTEXT_AUTHORIZING:
+ case PA_CONTEXT_SETTING_NAME:
+ break;
+
+ case PA_CONTEXT_READY:
+ rt_pa_info.dev.clear();
+ pa_context_get_server_info(context, rt_pa_set_server_info, NULL);
+ pa_context_get_sink_info_list(context, rt_pa_set_sink_info, NULL);
+ pa_context_get_source_info_list(context, rt_pa_set_source_info_and_quit, NULL);
+ break;
+
+ case PA_CONTEXT_TERMINATED:
+ rt_pa_mainloop_api_quit(0);
+ break;
+
+ case PA_CONTEXT_FAILED:
+ default:
+ rt_pa_mainloop_api_quit(1);
+ }
+}
+
RtApiPulse::~RtApiPulse()
{
if ( stream_.state != STREAM_CLOSED )
closeStream();
}
-unsigned int RtApiPulse::getDeviceCount( void )
+void RtApiPulse::collectDeviceInfo( void )
{
- return 1;
-}
+ pa_context *context = NULL;
+ pa_mainloop *m = NULL;
+ char *server = NULL;
+ int ret = 1;
-RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
-{
- RtAudio::DeviceInfo info;
- info.probed = true;
- info.name = "PulseAudio";
- info.outputChannels = 2;
- info.inputChannels = 2;
- info.duplexChannels = 2;
- info.isDefaultOutput = true;
- info.isDefaultInput = true;
+ if (!(m = pa_mainloop_new())) {
+ errorStream_ << "RtApiPulse::DeviceInfo pa_mainloop_new() failed.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto quit;
+ }
- for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
- info.sampleRates.push_back( *sr );
+ rt_pa_mainloop_api = pa_mainloop_get_api(m);
- info.preferredSampleRate = 48000;
- info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
+ if (!(context = pa_context_new_with_proplist(rt_pa_mainloop_api, NULL, NULL))) {
+ errorStream_ << "pa_context_new() failed.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto quit;
+ }
- return info;
+ pa_context_set_state_callback(context, rt_pa_context_state_callback, NULL);
+
+ if (pa_context_connect(context, server, PA_CONTEXT_NOFLAGS, NULL) < 0) {
+ errorStream_ << "RtApiPulse::DeviceInfo pa_context_connect() failed: "
+ << pa_strerror(pa_context_errno(context));
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto quit;
+ }
+
+ if (pa_mainloop_run(m, &ret) < 0) {
+ errorStream_ << "pa_mainloop_run() failed.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto quit;
+ }
+
+ if (ret != 0) {
+ errorStream_ << "could not get server info.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto quit;
+ }
+
+quit:
+ if (context)
+ pa_context_unref(context);
+
+ if (m) {
+ pa_mainloop_free(m);
+ }
+
+ pa_xfree(server);
+}
+
+unsigned int RtApiPulse::getDeviceCount( void )
+{
+ collectDeviceInfo();
+ return rt_pa_info.dev.size();
+}
+
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device )
+{
+ if (rt_pa_info.dev.size()==0)
+ collectDeviceInfo();
+ if (device < rt_pa_info.dev.size())
+ return rt_pa_info.dev[device].info;
+ return RtAudio::DeviceInfo();
}
static void *pulseaudio_callback( void * user )
@@ -8679,15 +8981,18 @@ void RtApiPulse::stopStream( void )
stream_.state = STREAM_STOPPED;
MUTEX_LOCK( &stream_.mutex );
- if ( pah && pah->s_play ) {
- int pa_error;
- if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
- errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
- pa_strerror( pa_error ) << ".";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
+ if ( pah ) {
+ pah->runnable = false;
+ if ( pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
}
}
@@ -8713,15 +9018,18 @@ void RtApiPulse::abortStream( void )
stream_.state = STREAM_STOPPED;
MUTEX_LOCK( &stream_.mutex );
- if ( pah && pah->s_play ) {
- int pa_error;
- if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
- errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
- pa_strerror( pa_error ) << ".";
- errorText_ = errorStream_.str();
- MUTEX_UNLOCK( &stream_.mutex );
- error( RtAudioError::SYSTEM_ERROR );
- return;
+ if ( pah ) {
+ pah->runnable = false;
+ if ( pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
}
}
@@ -8738,15 +9046,51 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
unsigned long bufferBytes = 0;
pa_sample_spec ss;
- if ( device != 0 ) return false;
- if ( mode != INPUT && mode != OUTPUT ) return false;
- if ( channels != 1 && channels != 2 ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
+ if ( device >= rt_pa_info.dev.size() ) return false;
+ if ( firstChannel != 0 ) {
+ errorText_ = "PulseAudio does not support channel offset mapping.";
+ return false;
+ }
+
+ /* these may be NULL for default, but we've already got the names */
+ const char *dev_input = NULL;
+ const char *dev_output = NULL;
+ if (!rt_pa_info.dev[device].source_name.empty())
+ dev_input = rt_pa_info.dev[device].source_name.c_str();
+ if (!rt_pa_info.dev[device].sink_name.empty())
+ dev_output = rt_pa_info.dev[device].sink_name.c_str();
+
+ if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels == 0) {
+ errorText_ = "PulseAudio device does not support input.";
+ return false;
+ }
+ if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels == 0) {
+ errorText_ = "PulseAudio device does not support output.";
+ return false;
+ }
+ if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels == 0) {
+ /* Note: will always error, DUPLEX not yet supported */
+ errorText_ = "PulseAudio device does not support duplex.";
+ return false;
+ }
+
+ if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels < channels) {
+ errorText_ = "PulseAudio: unsupported number of input channels.";
return false;
}
- ss.channels = channels;
- if ( firstChannel != 0 ) return false;
+ if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels < channels) {
+ errorText_ = "PulseAudio: unsupported number of output channels.";
+ return false;
+ }
+
+ if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels < channels) {
+ /* Note: will always error, DUPLEX not yet supported */
+ errorText_ = "PulseAudio: unsupported number of duplex channels.";
+ return false;
+ }
+
+ ss.channels = channels;
bool sr_found = false;
for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
@@ -8758,8 +9102,8 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
}
}
if ( !sr_found ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
- return false;
+ stream_.sampleRate = sampleRate;
+ ss.rate = sampleRate;
}
bool sf_found = 0;
@@ -8783,7 +9127,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
else stream_.userInterleaved = true;
stream_.deviceInterleaved[mode] = true;
- stream_.nBuffers = 1;
+ stream_.nBuffers = options ? options->numberOfBuffers : 1;
stream_.doByteSwap[mode] = false;
stream_.nUserChannels[mode] = channels;
stream_.nDeviceChannels[mode] = channels + firstChannel;
@@ -8796,6 +9140,8 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
stream_.doConvertBuffer[mode] = true;
if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] )
+ stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers.
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
@@ -8851,24 +9197,47 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
int error;
if ( options && !options->streamName.empty() ) streamName = options->streamName;
switch ( mode ) {
- case INPUT:
pa_buffer_attr buffer_attr;
+ case INPUT:
buffer_attr.fragsize = bufferBytes;
buffer_attr.maxlength = -1;
- pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
+ pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD,
+ dev_input, "Record", &ss, NULL, &buffer_attr, &error );
if ( !pah->s_rec ) {
errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
goto error;
}
break;
- case OUTPUT:
- pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
+ case OUTPUT: {
+ pa_buffer_attr * attr_ptr;
+
+ if ( options && options->numberOfBuffers > 0 ) {
+ // pa_buffer_attr::fragsize is recording-only.
+ // Hopefully PortAudio won't access uninitialized fields.
+ buffer_attr.maxlength = bufferBytes * options->numberOfBuffers;
+ buffer_attr.minreq = -1;
+ buffer_attr.prebuf = -1;
+ buffer_attr.tlength = -1;
+ attr_ptr = &buffer_attr;
+ } else {
+ attr_ptr = nullptr;
+ }
+
+ pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK,
+ dev_output, "Playback", &ss, NULL, attr_ptr, &error );
if ( !pah->s_play ) {
errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
goto error;
}
break;
+ }
+ case DUPLEX:
+ /* Note: We could add DUPLEX by synchronizing multiple streams,
+ but it would mean moving from Simple API to Asynchronous API:
+ https://freedesktop.org/software/pulseaudio/doxygen/streams.html#sync_streams */
+ errorText_ = "RtApiPulse::probeDeviceOpen: duplex not supported for PulseAudio.";
+ goto error;
default:
goto error;
}
@@ -10117,24 +10486,19 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
// data interleaving/deinterleaving. 24-bit integers are assumed to occupy
// the lower three bytes of a 32-bit integer.
- // Clear our device buffer when in/out duplex device channels are different
- if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
- ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
+ // Clear our duplex device output buffer if there are more device outputs than user outputs
+ if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX && info.outJump > info.inJump )
memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
int j;
if (info.outFormat == RTAUDIO_FLOAT64) {
- Float64 scale;
Float64 *out = (Float64 *)outBuffer;
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
- scale = 1.0 / 127.5;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]] / 128.0;
}
in += info.inJump;
out += info.outJump;
@@ -10142,12 +10506,9 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
}
else if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
- scale = 1.0 / 32767.5;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]] / 32768.0;
}
in += info.inJump;
out += info.outJump;
@@ -10155,12 +10516,9 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
}
else if (info.inFormat == RTAUDIO_SINT24) {
Int24 *in = (Int24 *)inBuffer;
- scale = 1.0 / 8388607.5;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]].asInt() / 8388608.0;
}
in += info.inJump;
out += info.outJump;
@@ -10168,12 +10526,9 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
- scale = 1.0 / 2147483647.5;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]] / 2147483648.0;
}
in += info.inJump;
out += info.outJump;
@@ -10202,17 +10557,13 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
}
}
else if (info.outFormat == RTAUDIO_FLOAT32) {
- Float32 scale;
Float32 *out = (Float32 *)outBuffer;
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
- scale = (Float32) ( 1.0 / 127.5 );
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]] / 128.f;
}
in += info.inJump;
out += info.outJump;
@@ -10220,12 +10571,9 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
}
else if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
- scale = (Float32) ( 1.0 / 32767.5 );
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]] / 32768.f;
}
in += info.inJump;
out += info.outJump;
@@ -10233,12 +10581,9 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
}
else if (info.inFormat == RTAUDIO_SINT24) {
Int24 *in = (Int24 *)inBuffer;
- scale = (Float32) ( 1.0 / 8388607.5 );
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]].asInt() / 8388608.f;
}
in += info.inJump;
out += info.outJump;
@@ -10246,12 +10591,9 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
- scale = (Float32) ( 1.0 / 2147483647.5 );
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
- out[info.outOffset[j]] += 0.5;
- out[info.outOffset[j]] *= scale;
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]] / 2147483648.f;
}
in += info.inJump;
out += info.outJump;
@@ -10329,7 +10671,8 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+ // Use llround() which returns `long long` which is guaranteed to be at least 64 bits.
+ out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 2147483648.f), 2147483647LL);
}
in += info.inJump;
out += info.outJump;
@@ -10339,7 +10682,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+ out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 2147483648.0), 2147483647LL);
}
in += info.inJump;
out += info.outJump;
@@ -10396,7 +10739,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+ out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 8388608.f), 8388607LL);
}
in += info.inJump;
out += info.outJump;
@@ -10406,7 +10749,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+ out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 8388608.0), 8388607LL);
}
in += info.inJump;
out += info.outJump;
@@ -10461,7 +10804,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+ out[info.outOffset[j]] = (Int16) std::min(std::llround(in[info.inOffset[j]] * 32768.f), 32767LL);
}
in += info.inJump;
out += info.outJump;
@@ -10471,7 +10814,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+ out[info.outOffset[j]] = (Int16) std::min(std::llround(in[info.inOffset[j]] * 32768.0), 32767LL);
}
in += info.inJump;
out += info.outJump;
@@ -10525,7 +10868,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+ out[info.outOffset[j]] = (signed char) std::min(std::llround(in[info.inOffset[j]] * 128.f), 127LL);
}
in += info.inJump;
out += info.outJump;
@@ -10535,7 +10878,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+ out[info.outOffset[j]] = (signed char) std::min(std::llround(in[info.inOffset[j]] * 128.0), 127LL);
}
in += info.inJump;
out += info.outJump;
diff --git a/src/rtaudio/RtAudio.h b/src/rtaudio/RtAudio.h
index 17f442b46247f694fbd0349b37a6507845dba5f4..1aa42e0d04b962ac8473db2c28053c14589b55be 100644
--- a/src/rtaudio/RtAudio.h
+++ b/src/rtaudio/RtAudio.h
@@ -11,7 +11,7 @@
RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
RtAudio: realtime audio i/o C++ classes
- Copyright (c) 2001-2019 Gary P. Scavone
+ Copyright (c) 2001-2021 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
@@ -46,7 +46,7 @@
#ifndef __RTAUDIO_H
#define __RTAUDIO_H
-#define RTAUDIO_VERSION "5.1.0"
+#define RTAUDIO_VERSION "5.2.0"
#if defined _WIN32 || defined __CYGWIN__
#if defined(RTAUDIO_EXPORT)
@@ -226,12 +226,12 @@ class RTAUDIO_DLL_PUBLIC RtAudioError : public std::runtime_error
UNSPECIFIED, /*!< The default, unspecified error type. */
NO_DEVICES_FOUND, /*!< No devices found on system. */
INVALID_DEVICE, /*!< An invalid device ID was specified. */
- MEMORY_ERROR, /*!< An error occured during memory allocation. */
+ MEMORY_ERROR, /*!< An error occurred during memory allocation. */
INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
INVALID_USE, /*!< The function was called incorrectly. */
- DRIVER_ERROR, /*!< A system driver error occured. */
- SYSTEM_ERROR, /*!< A system error occured. */
- THREAD_ERROR /*!< A thread error occured. */
+ DRIVER_ERROR, /*!< A system driver error occurred. */
+ SYSTEM_ERROR, /*!< A system error occurred. */
+ THREAD_ERROR /*!< A thread error occurred. */
};
//! The constructor.
@@ -299,30 +299,21 @@ class RTAUDIO_DLL_PUBLIC RtAudio
struct DeviceInfo {
bool probed; /*!< true if the device capabilities were successfully probed. */
std::string name; /*!< Character string device identifier. */
- unsigned int outputChannels; /*!< Maximum output channels supported by device. */
- unsigned int inputChannels; /*!< Maximum input channels supported by device. */
- unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */
- bool isDefaultOutput; /*!< true if this is the default output device. */
- bool isDefaultInput; /*!< true if this is the default input device. */
+ unsigned int outputChannels{}; /*!< Maximum output channels supported by device. */
+ unsigned int inputChannels{}; /*!< Maximum input channels supported by device. */
+ unsigned int duplexChannels{}; /*!< Maximum simultaneous input/output channels supported by device. */
+ bool isDefaultOutput{false}; /*!< true if this is the default output device. */
+ bool isDefaultInput{false}; /*!< true if this is the default input device. */
std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
- unsigned int preferredSampleRate; /*!< Preferred sample rate, e.g. for WASAPI the system sample rate. */
- RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */
-
- // Default constructor.
- DeviceInfo()
- :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
- isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), nativeFormats(0) {}
+ unsigned int preferredSampleRate{}; /*!< Preferred sample rate, e.g. for WASAPI the system sample rate. */
+ RtAudioFormat nativeFormats{}; /*!< Bit mask of supported data formats. */
};
- //! The structure for specifying input or ouput stream parameters.
+ //! The structure for specifying input or output stream parameters.
struct StreamParameters {
- unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */
- unsigned int nChannels; /*!< Number of channels. */
- unsigned int firstChannel; /*!< First channel index on device (default = 0). */
-
- // Default constructor.
- StreamParameters()
- : deviceId(0), nChannels(0), firstChannel(0) {}
+ unsigned int deviceId{}; /*!< Device index (0 to getDeviceCount() - 1). */
+ unsigned int nChannels{}; /*!< Number of channels. */
+ unsigned int firstChannel{}; /*!< First channel index on device (default = 0). */
};
//! The structure for specifying stream options.
@@ -383,14 +374,10 @@ class RTAUDIO_DLL_PUBLIC RtAudio
RtAudio with Jack, each instance must have a unique client name.
*/
struct StreamOptions {
- RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
- unsigned int numberOfBuffers; /*!< Number of stream buffers. */
+ RtAudioStreamFlags flags{}; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */
+ unsigned int numberOfBuffers{}; /*!< Number of stream buffers. */
std::string streamName; /*!< A stream name (currently used only in Jack). */
- int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
-
- // Default constructor.
- StreamOptions()
- : flags(0), numberOfBuffers(0), priority(0) {}
+ int priority{}; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
};
//! A static function to determine the current RtAudio version.
@@ -527,7 +514,7 @@ class RTAUDIO_DLL_PUBLIC RtAudio
lowest allowable value is used. The actual value used is
returned via the structure argument. The parameter is API dependent.
\param errorCallback A client-defined function that will be invoked
- when an error has occured.
+ when an error has occurred.
*/
void openStream( RtAudio::StreamParameters *outputParameters,
RtAudio::StreamParameters *inputParameters,
@@ -616,7 +603,7 @@ class RTAUDIO_DLL_PUBLIC RtAudio
};
// Operating system dependent thread functionality.
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
+#if defined(_WIN32) || defined(__CYGWIN__)
#ifndef NOMINMAX
#define NOMINMAX
@@ -628,18 +615,22 @@ class RTAUDIO_DLL_PUBLIC RtAudio
typedef uintptr_t ThreadHandle;
typedef CRITICAL_SECTION StreamMutex;
-#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+#else
+
// Using pthread library for various flavors of unix.
#include <pthread.h>
typedef pthread_t ThreadHandle;
typedef pthread_mutex_t StreamMutex;
-#else // Setup for "dummy" behavior
+#endif
+
+// Setup for "dummy" behavior if no apis specified.
+#if !(defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) \
+ || defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) \
+ || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__))
#define __RTAUDIO_DUMMY__
- typedef int ThreadHandle;
- typedef int StreamMutex;
#endif
@@ -647,19 +638,15 @@ class RTAUDIO_DLL_PUBLIC RtAudio
// between the private RtAudio stream structure and global callback
// handling functions.
struct CallbackInfo {
- void *object; // Used as a "this" pointer.
- ThreadHandle thread;
- void *callback;
- void *userData;
- void *errorCallback;
- void *apiInfo; // void pointer for API specific callback information
- bool isRunning;
- bool doRealtime;
- int priority;
-
- // Default constructor.
- CallbackInfo()
- :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false), priority(0) {}
+ void *object{}; // Used as a "this" pointer.
+ ThreadHandle thread{};
+ void *callback{};
+ void *userData{};
+ void *errorCallback{};
+ void *apiInfo{}; // void pointer for API specific callback information
+ bool isRunning{false};
+ bool doRealtime{false};
+ int priority{};
};
// **************************************************************** //
@@ -686,9 +673,9 @@ class S24 {
S24() {}
S24& operator = ( const int& i ) {
- c3[0] = (i & 0x000000ff);
- c3[1] = (i & 0x0000ff00) >> 8;
- c3[2] = (i & 0x00ff0000) >> 16;
+ c3[0] = (unsigned char)(i & 0x000000ff);
+ c3[1] = (unsigned char)((i & 0x0000ff00) >> 8);
+ c3[2] = (unsigned char)((i & 0x00ff0000) >> 16);
return *this;
}
@@ -895,20 +882,20 @@ public:
RtApiCore();
~RtApiCore();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- unsigned int getDefaultOutputDevice( void );
- unsigned int getDefaultInputDevice( void );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::MACOSX_CORE; }
+ unsigned int getDeviceCount( void ) override;
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override;
+ unsigned int getDefaultOutputDevice( void ) override;
+ unsigned int getDefaultInputDevice( void ) override;
+ void closeStream( void ) override;
+ void startStream( void ) override;
+ void stopStream( void ) override;
+ void abortStream( void ) override;
// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesireable results!
+ // will most likely produce highly undesirable results!
bool callbackEvent( AudioDeviceID deviceId,
const AudioBufferList *inBufferList,
const AudioBufferList *outBufferList );
@@ -918,7 +905,7 @@ public:
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
+ RtAudio::StreamOptions *options ) override;
static const char* getErrorCode( OSStatus code );
};
@@ -932,18 +919,18 @@ public:
RtApiJack();
~RtApiJack();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::UNIX_JACK; }
+ unsigned int getDeviceCount( void ) override;
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override;
+ void closeStream( void ) override;
+ void startStream( void ) override;
+ void stopStream( void ) override;
+ void abortStream( void ) override;
// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesireable results!
+ // will most likely produce highly undesirable results!
bool callbackEvent( unsigned long nframes );
private:
@@ -951,7 +938,7 @@ public:
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
+ RtAudio::StreamOptions *options ) override;
bool shouldAutoconnect_;
};
@@ -966,18 +953,20 @@ public:
RtApiAsio();
~RtApiAsio();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_ASIO; }
+ unsigned int getDeviceCount( void ) override;
+ unsigned int getDefaultOutputDevice( void ) override;
+ unsigned int getDefaultInputDevice( void ) override;
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override;
+ void closeStream( void ) override;
+ void startStream( void ) override;
+ void stopStream( void ) override;
+ void abortStream( void ) override;
// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesireable results!
+ // will most likely produce highly undesirable results!
bool callbackEvent( long bufferIndex );
private:
@@ -988,7 +977,7 @@ public:
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
+ RtAudio::StreamOptions *options ) override;
};
#endif
@@ -1001,20 +990,20 @@ public:
RtApiDs();
~RtApiDs();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; }
- unsigned int getDeviceCount( void );
- unsigned int getDefaultOutputDevice( void );
- unsigned int getDefaultInputDevice( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_DS; }
+ unsigned int getDeviceCount( void ) override;
+ unsigned int getDefaultOutputDevice( void ) override;
+ unsigned int getDefaultInputDevice( void ) override;
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override;
+ void closeStream( void ) override;
+ void startStream( void ) override;
+ void stopStream( void ) override;
+ void abortStream( void ) override;
// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesireable results!
+ // will most likely produce highly undesirable results!
void callbackEvent( void );
private:
@@ -1026,7 +1015,7 @@ public:
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
+ RtAudio::StreamOptions *options ) override;
};
#endif
@@ -1041,15 +1030,13 @@ public:
RtApiWasapi();
virtual ~RtApiWasapi();
- RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- unsigned int getDefaultOutputDevice( void );
- unsigned int getDefaultInputDevice( void );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_WASAPI; }
+ unsigned int getDeviceCount( void ) override;
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override;
+ void closeStream( void ) override;
+ void startStream( void ) override;
+ void stopStream( void ) override;
+ void abortStream( void ) override;
private:
bool coInitialized_;
@@ -1058,7 +1045,7 @@ private:
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int* bufferSize,
- RtAudio::StreamOptions* options );
+ RtAudio::StreamOptions* options ) override;
static DWORD WINAPI runWasapiThread( void* wasapiPtr );
static DWORD WINAPI stopWasapiThread( void* wasapiPtr );
@@ -1076,18 +1063,18 @@ public:
RtApiAlsa();
~RtApiAlsa();
- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
+ RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_ALSA; }
+ unsigned int getDeviceCount( void ) override;
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override;
+ void closeStream( void ) override;
+ void startStream( void ) override;
+ void stopStream( void ) override;
+ void abortStream( void ) override;
// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesireable results!
+ // will most likely produce highly undesirable results!
void callbackEvent( void );
private:
@@ -1097,7 +1084,7 @@ public:
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
+ RtAudio::StreamOptions *options ) override;
};
#endif
@@ -1108,28 +1095,27 @@ class RtApiPulse: public RtApi
{
public:
~RtApiPulse();
- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
+ RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_PULSE; }
+ unsigned int getDeviceCount( void ) override;
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override;
+ void closeStream( void ) override;
+ void startStream( void ) override;
+ void stopStream( void ) override;
+ void abortStream( void ) override;
// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesireable results!
+ // will most likely produce highly undesirable results!
void callbackEvent( void );
private:
- std::vector<RtAudio::DeviceInfo> devices_;
- void saveDeviceInfo( void );
+ void collectDeviceInfo( void );
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
+ RtAudio::StreamOptions *options ) override;
};
#endif
@@ -1142,18 +1128,18 @@ public:
RtApiOss();
~RtApiOss();
- RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; }
- unsigned int getDeviceCount( void );
- RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
- void closeStream( void );
- void startStream( void );
- void stopStream( void );
- void abortStream( void );
+ RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_OSS; }
+ unsigned int getDeviceCount( void ) override;
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override;
+ void closeStream( void ) override;
+ void startStream( void ) override;
+ void stopStream( void ) override;
+ void abortStream( void ) override;
// This function is intended for internal use only. It must be
// public because it is called by the internal callback handler,
// which is not a member of RtAudio. External use of this function
- // will most likely produce highly undesireable results!
+ // will most likely produce highly undesirable results!
void callbackEvent( void );
private:
@@ -1161,7 +1147,7 @@ public:
bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
- RtAudio::StreamOptions *options );
+ RtAudio::StreamOptions *options ) override;
};
#endif
@@ -1173,20 +1159,20 @@ class RtApiDummy: public RtApi
public:
RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); }
- RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; }
- unsigned int getDeviceCount( void ) { return 0; }
- RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; }
- void closeStream( void ) {}
- void startStream( void ) {}
- void stopStream( void ) {}
- void abortStream( void ) {}
+ RtAudio::Api getCurrentApi( void ) override { return RtAudio::RTAUDIO_DUMMY; }
+ unsigned int getDeviceCount( void ) override { return 0; }
+ RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) override { RtAudio::DeviceInfo info; return info; }
+ void closeStream( void ) override {}
+ void startStream( void ) override {}
+ void stopStream( void ) override {}
+ void abortStream( void ) override {}
private:
bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
- RtAudio::StreamOptions * /*options*/ ) { return false; }
+ RtAudio::StreamOptions * /*options*/ ) override { return false; }
};
#endif
diff --git a/src/rtaudio/RtAudio.os b/src/rtaudio/RtAudio.os
index f2dfd26c1a6b2994d015badbc29eeb137ae99ef1..7da86472b3334fa616dd67ddc74050896c89bd0d 100644
Binary files a/src/rtaudio/RtAudio.os and b/src/rtaudio/RtAudio.os differ