Updates for release 4.0.8, including new python binding, new teststops.cpp program...

[rtaudio.git] / RtAudio.h

diff --git a/RtAudio.h b/RtAudio.h
index d1624cfe0239b16ab125230b66ed7928c017c357..d8591c066c51957fb191b1881c11aa892c6bfa23 100644 (file)
--- a/RtAudio.h
+++ b/RtAudio.h
@@ -86,6 +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
@@ -113,12 +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.
@@ -250,6 +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
@@ -278,7 +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
@@ -294,7 +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). */