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Version 4.4.3

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This commit is contained in:
Gary Scavone
2013-09-29 23:21:29 +02:00
committed by Stephen Sinclair
parent baca57040b
commit 0aec39260a
223 changed files with 26190 additions and 11130 deletions
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23
include/RtAudio.h
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@@ -10,7 +10,7 @@
RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
RtAudio: realtime audio i/o C++ classes
Copyright (c) 2001-2010 Gary P. Scavone
Copyright (c) 2001-2011 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
@@ -42,7 +42,7 @@
\file RtAudio.h
*/
// RtAudio: Version 4.0.7
// RtAudio: Version 4.0.10
#ifndef __RTAUDIO_H
#define __RTAUDIO_H
@@ -59,10 +59,12 @@
internal routines will automatically take care of any necessary
byte-swapping between the host format and the soundcard. Thus,
endian-ness is not a concern in the following format definitions.
Note that 24-bit data is expected to be encapsulated in a 32-bit
format.
- \e RTAUDIO_SINT8: 8-bit signed integer.
- \e RTAUDIO_SINT16: 16-bit signed integer.
- \e RTAUDIO_SINT24: Upper 3 bytes of 32-bit signed integer.
- \e RTAUDIO_SINT24: Lower 3 bytes of 32-bit signed integer.
- \e RTAUDIO_SINT32: 32-bit signed integer.
- \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
- \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
@@ -84,6 +86,7 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi
- \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved).
- \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).
By default, RtAudio streams pass and receive audio data from the
client in an interleaved format. By passing the
@@ -111,12 +114,17 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi
If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
to select realtime scheduling (round-robin) for the callback thread.
If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
open the "default" PCM device when using the ALSA API. Note that this
will override any specified input or output device id.
*/
typedef unsigned int RtAudioStreamFlags;
static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved).
static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency.
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).
/*! \typedef typedef unsigned long RtAudioStreamStatus;
\brief RtAudio stream status (over- or underflow) flags.
@@ -248,6 +256,7 @@ class RtAudio
- \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_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
- \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only).
By default, RtAudio streams pass and receive audio data from the
client in an interleaved format. By passing the
@@ -276,7 +285,11 @@ class RtAudio
If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt
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.
flag is set. It defines the thread's realtime priority.
If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to
open the "default" PCM device when using the ALSA API. Note that this
will override any specified input or output device id.
The \c numberOfBuffers parameter can be used to control stream
latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
@@ -292,7 +305,7 @@ class 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). */
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). */