[SCM] giada/master: Fixed RtAudio patch
umlaeute at users.alioth.debian.org
umlaeute at users.alioth.debian.org
Mon Oct 30 09:12:42 UTC 2017
The following commit has been merged in the master branch:
commit 6fed4fa8d0697c83a0041548465d8acf3851c0f6
Author: IOhannes m zmölnig <zmoelnig at iem.at>
Date: Mon Oct 30 09:48:11 2017 +0100
Fixed RtAudio patch
Gbp-Dch: ignore
diff --git a/debian/patches/01-rtaudio5.patch b/debian/patches/01-rtaudio5.patch
index 439c1fa..713b217 100644
--- a/debian/patches/01-rtaudio5.patch
+++ b/debian/patches/01-rtaudio5.patch
@@ -8,11 +8,9 @@ Subject: updated bundled and hacked RtAudio to RtAudio5
src/deps/rtaudio-mod/RtAudio.h | 113 +-
3 files changed, 10401 insertions(+), 10294 deletions(-)
-diff --git a/src/core/kernelAudio.cpp b/src/core/kernelAudio.cpp
-index d9fe65c..df56c14 100644
---- a/src/core/kernelAudio.cpp
-+++ b/src/core/kernelAudio.cpp
-@@ -59,7 +59,7 @@ JackState jackState;
+--- giada.orig/src/core/kernelAudio.cpp
++++ giada/src/core/kernelAudio.cpp
+@@ -59,7 +59,7 @@
jack_client_t *jackGetHandle()
{
@@ -21,10 +19,8 @@ index d9fe65c..df56c14 100644
}
#endif
-diff --git a/src/deps/rtaudio-mod/RtAudio.cpp b/src/deps/rtaudio-mod/RtAudio.cpp
-index 1586aaa..62c6858 100755
---- a/src/deps/rtaudio-mod/RtAudio.cpp
-+++ b/src/deps/rtaudio-mod/RtAudio.cpp
+--- giada.orig/src/deps/rtaudio-mod/RtAudio.cpp
++++ giada/src/deps/rtaudio-mod/RtAudio.cpp
@@ -1,10237 +1,10343 @@
-/************************************************************************/
-/*! \class RtAudio
@@ -20586,7 +20582,7 @@ index 1586aaa..62c6858 100755
+
+void *RtAudio :: GIADA_HACK__getJackClient() { /* Monocasual HACK */
+#if defined(__UNIX_JACK__)
-+ RtApiJack*jackapi = dynamic_cast<RtApiJack*>rtapi_;
++ RtApiJack*jackapi = dynamic_cast<RtApiJack*>(rtapi_);
+ if (jackapi && jackapi->stream_.apiHandle) {
+ JackHandle *handle = (JackHandle *) jackapi->stream_.apiHandle;
+ return (void*) handle->client;
@@ -20606,10 +20602,8 @@ index 1586aaa..62c6858 100755
+ //
+ // vim: et sts=2 sw=2
+
-diff --git a/src/deps/rtaudio-mod/RtAudio.h b/src/deps/rtaudio-mod/RtAudio.h
-index ddb42cc..746bcbc 100755
---- a/src/deps/rtaudio-mod/RtAudio.h
-+++ b/src/deps/rtaudio-mod/RtAudio.h
+--- giada.orig/src/deps/rtaudio-mod/RtAudio.h
++++ giada/src/deps/rtaudio-mod/RtAudio.h
@@ -10,7 +10,7 @@
RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
@@ -20633,7 +20627,7 @@ index ddb42cc..746bcbc 100755
#include <iostream>
/*! \typedef typedef unsigned long RtAudioFormat;
-@@ -86,6 +86,7 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi
+@@ -86,6 +86,7 @@
- \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
- \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use.
- \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
@@ -20641,7 +20635,7 @@ index ddb42cc..746bcbc 100755
By default, RtAudio streams pass and receive audio data from the
client in an interleaved format. By passing the
-@@ -111,12 +112,15 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi
+@@ -111,12 +112,15 @@
open the input and/or output stream device(s) for exclusive use.
Note that this is not possible with all supported audio APIs.
@@ -20658,7 +20652,7 @@ index ddb42cc..746bcbc 100755
*/
typedef unsigned int RtAudioStreamFlags;
static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved).
-@@ -124,6 +128,7 @@ static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to s
+@@ -124,6 +128,7 @@
static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others.
static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only).
@@ -20666,7 +20660,7 @@ index ddb42cc..746bcbc 100755
/*! \typedef typedef unsigned long RtAudioStreamStatus;
\brief RtAudio stream status (over- or underflow) flags.
-@@ -195,7 +200,7 @@ typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer,
+@@ -195,7 +200,7 @@
*/
/************************************************************************/
@@ -20675,7 +20669,7 @@ index ddb42cc..746bcbc 100755
{
public:
//! Defined RtAudioError types.
-@@ -214,25 +219,22 @@ class RtAudioError : public std::exception
+@@ -214,25 +219,22 @@
};
//! The constructor.
@@ -20709,7 +20703,7 @@ index ddb42cc..746bcbc 100755
Type type_;
};
-@@ -341,7 +343,7 @@ class RtAudio
+@@ -341,7 +343,7 @@
open the input and/or output stream device(s) for exclusive use.
Note that this is not possible with all supported audio APIs.
@@ -20718,7 +20712,7 @@ index ddb42cc..746bcbc 100755
to select realtime scheduling (round-robin) for the callback thread.
The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
flag is set. It defines the thread's realtime priority.
-@@ -375,7 +377,7 @@ class RtAudio
+@@ -375,7 +377,7 @@
};
//! A static function to determine the current RtAudio version.
@@ -20727,7 +20721,7 @@ index ddb42cc..746bcbc 100755
//! A static function to determine the available compiled audio APIs.
/*!
-@@ -383,7 +385,7 @@ class RtAudio
+@@ -383,7 +385,7 @@
the enumerated list values. Note that there can be more than one
API compiled for certain operating systems.
*/
@@ -20736,7 +20730,7 @@ index ddb42cc..746bcbc 100755
//! The class constructor.
/*!
-@@ -401,18 +403,18 @@ class RtAudio
+@@ -401,18 +403,18 @@
If a stream is running or open, it will be stopped and closed
automatically.
*/
@@ -20759,7 +20753,7 @@ index ddb42cc..746bcbc 100755
//! Return an RtAudio::DeviceInfo structure for a specified device number.
/*!
-@@ -435,7 +437,7 @@ class RtAudio
+@@ -435,7 +437,7 @@
client's responsibility to verify that a device is available
before attempting to open a stream.
*/
@@ -20768,7 +20762,7 @@ index ddb42cc..746bcbc 100755
//! A function that returns the index of the default input device.
/*!
-@@ -445,7 +447,7 @@ class RtAudio
+@@ -445,7 +447,7 @@
client's responsibility to verify that a device is available
before attempting to open a stream.
*/
@@ -20777,7 +20771,7 @@ index ddb42cc..746bcbc 100755
//! A public function for opening a stream with the specified parameters.
/*!
-@@ -477,7 +479,7 @@ class RtAudio
+@@ -477,7 +479,7 @@
from within the callback function.
\param options An optional pointer to a structure containing various
global stream options, including a list of OR'ed RtAudioStreamFlags
@@ -20786,7 +20780,7 @@ index ddb42cc..746bcbc 100755
control stream latency. More buffers typically result in more
robust performance, though at a cost of greater latency. If a
value of zero is specified, a system-specific median value is
-@@ -498,7 +500,7 @@ class RtAudio
+@@ -498,7 +500,7 @@
If a stream is not open, this function issues a warning and
returns (no exception is thrown).
*/
@@ -20795,7 +20789,7 @@ index ddb42cc..746bcbc 100755
//! A function that starts a stream.
/*!
-@@ -528,10 +530,10 @@ class RtAudio
+@@ -528,10 +530,10 @@
void abortStream( void );
//! Returns true if a stream is open and false if not.
@@ -20808,7 +20802,7 @@ index ddb42cc..746bcbc 100755
//! Returns the number of elapsed seconds since the stream was started.
/*!
-@@ -565,14 +567,15 @@ class RtAudio
+@@ -565,14 +567,15 @@
unsigned int getStreamSampleRate( void );
//! Specify whether warning messages should be printed to stderr.
@@ -20828,7 +20822,7 @@ index ddb42cc..746bcbc 100755
};
// Operating system dependent thread functionality.
-@@ -618,7 +621,7 @@ struct CallbackInfo {
+@@ -618,7 +621,7 @@
// Default constructor.
CallbackInfo()
@@ -20837,7 +20831,7 @@ index ddb42cc..746bcbc 100755
};
// **************************************************************** //
-@@ -675,12 +678,6 @@ class RtApi
+@@ -675,12 +678,6 @@
{
public:
@@ -20850,7 +20844,7 @@ index ddb42cc..746bcbc 100755
RtApi();
virtual ~RtApi();
virtual RtAudio::Api getCurrentApi( void ) = 0;
-@@ -790,7 +787,7 @@ protected:
+@@ -790,7 +787,7 @@
"warning" message is reported and FAILURE is returned. A
successful probe is indicated by a return value of SUCCESS.
*/
@@ -20859,7 +20853,7 @@ index ddb42cc..746bcbc 100755
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options );
-@@ -824,6 +821,8 @@ protected:
+@@ -824,6 +821,8 @@
//! Protected common method that sets up the parameters for buffer conversion.
void setConvertInfo( StreamMode mode, unsigned int firstChannel );
@@ -20868,7 +20862,7 @@ index ddb42cc..746bcbc 100755
};
// **************************************************************** //
-@@ -832,22 +831,22 @@ protected:
+@@ -832,22 +831,22 @@
//
// **************************************************************** //
@@ -20899,7 +20893,7 @@ index ddb42cc..746bcbc 100755
// RtApi Subclass prototypes.
-@@ -882,7 +881,7 @@ public:
+@@ -882,7 +881,7 @@
private:
@@ -20908,7 +20902,7 @@ index ddb42cc..746bcbc 100755
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options );
-@@ -916,10 +915,12 @@ public:
+@@ -916,10 +915,12 @@
private:
@@ -20922,7 +20916,7 @@ index ddb42cc..746bcbc 100755
};
#endif
-@@ -952,7 +953,7 @@ public:
+@@ -952,7 +953,7 @@
std::vector<RtAudio::DeviceInfo> devices_;
void saveDeviceInfo( void );
bool coInitialized_;
@@ -20931,7 +20925,7 @@ index ddb42cc..746bcbc 100755
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options );
-@@ -991,7 +992,7 @@ public:
+@@ -991,7 +992,7 @@
bool buffersRolling;
long duplexPrerollBytes;
std::vector<struct DsDevice> dsDevices;
@@ -20940,7 +20934,7 @@ index ddb42cc..746bcbc 100755
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options );
-@@ -1062,7 +1063,7 @@ public:
+@@ -1062,7 +1063,7 @@
std::vector<RtAudio::DeviceInfo> devices_;
void saveDeviceInfo( void );
@@ -20949,7 +20943,7 @@ index ddb42cc..746bcbc 100755
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options );
-@@ -1126,7 +1127,7 @@ public:
+@@ -1126,7 +1127,7 @@
private:
@@ -20958,7 +20952,7 @@ index ddb42cc..746bcbc 100755
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options );
-@@ -1151,7 +1152,7 @@ public:
+@@ -1151,7 +1152,7 @@
private:
--
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