The RtError class declaration and definition have been extracted to a separate file (RtError.h). This was done in preparation for a new release of the RtMidi class (planned for Summer 2004).
The first thing that must be done when using RtAudio is to create an instance of the class. The default constructor scans the underlying audio system to verify that at least one device is available. RtAudio often uses C++ exceptions to report errors, necessitating try/catch blocks around most member functions. The following code example demonstrates default object construction and destruction:
@@ -66,9 +66,9 @@ The first thing that must be done when using delete audio;
}
-Obviously, this example doesn't demonstrate any of the real functionality of RtAudio. However, all uses of RtAudio must begin with a constructor (either default or overloaded varieties) and must end with class destruction. Further, it is necessary that all class methods which can throw a C++ exception be called within a try/catch block.
-The nativeFormats member of the RtAudioDeviceInfo structure is a bit mask of the above formats which are natively supported by the device. However, RtAudio will automatically provide format conversion if a particular format is not natively supported. When the probed member of the RtAudioDeviceInfo structure is false, the remaining structure members are undefined and the device is probably unuseable.
+The nativeFormats member of the RtAudioDeviceInfo structure is a bit mask of the above formats that are natively supported by the device. However, RtAudio will automatically provide format conversion if a particular format is not natively supported. When the probed member of the RtAudioDeviceInfo structure is false, the remaining structure members are undefined and the device is probably unuseable.
While some audio devices may require a minimum channel value greater than one, RtAudio will provide automatic channel number compensation when the number of channels set by the user is less than that required by the device. Channel compensation is NOT possible when the number of channels set by the user is greater than that supported by the device.
It should be noted that the capabilities reported by a device driver or underlying audio API are not always accurate and/or may be dependent on a combination of device settings. For this reason, RtAudio does not typically rely on the queried values when attempting to open a stream.
+Obviously, this example doesn't demonstrate any of the real functionality of RtAudio. However, all uses of RtAudio must begin with a constructor (either default or overloaded varieties) and must end with class destruction. Further, it is necessary that all class methods that can throw a C++ exception be called within a try/catch block.
Error Handling
-RtAudio uses a C++ exception handler called RtError, which is declared and defined in RtError.h. The RtError class is quite simple but it does allow errors to be "caught" by RtError::Type. Almost all RtAudio methods can "throw" an RtError, most typically if a driver error occurs or a stream function is called when no stream is open. There are a number of cases within RtAudio where warning messages may be displayed but an exception is not thrown. There is a protected RtAudio method, error(), which can be modified to globally control how these messages are handled and reported. By default, error messages are not automatically displayed in RtAudio unless the preprocessor definition __RTAUDIO_DEBUG__ is defined. Messages associated with caught exceptions can be displayed with, for example, the RtError::printMessage() function.
+RtAudio uses a C++ exception handler called RtError, which is declared and defined in RtError.h. The RtError class is quite simple but it does allow errors to be "caught" by RtError::Type. Almost all RtAudio methods can "throw" an RtError, most typically if a driver error occurs or a stream function is called when no stream is open. There are a number of cases within RtAudio where warning messages may be displayed but an exception is not thrown. There is a protected RtAudio method, error(), that can be modified to globally control how these messages are handled and reported. By default, error messages are not automatically displayed in RtAudio unless the preprocessor definition __RTAUDIO_DEBUG__ is defined. Messages associated with caught exceptions can be displayed with, for example, the RtError::printMessage() function.
Probing Device Capabilities
A programmer may wish to query the available audio device capabilities before deciding which to use. The following example outlines how this can be done.// probe.cpp
@@ -136,7 +136,7 @@ The following data formats are defined and fully supported by static const RtAudioFormat RTAUDIO_FLOAT32; // 32-bit float normalized between +/- 1.0
static const RtAudioFormat RTAUDIO_FLOAT64; // 64-bit double normalized between +/- 1.0
Device Settings
@@ -176,9 +176,9 @@ The next step in using RtAudio is to
return 0;
}
-The RtAudio::openStream() method attempts to open a stream with a specified set of parameter values. In this case, we attempt to open a two channel playback stream with the default output device, 32-bit floating point data, a sample rate of 44100 Hz, a frame rate of 256 sample frames per read/write, and 4 internal device buffers. When device = 0, RtAudio first attempts to open the default audio device with the given parameters. If that attempt fails, RtAudio searches through the remaining available devices in an effort to find a device which will meet the given parameters. If all attempts are unsuccessful, an RtError is thrown. When a non-zero device value is specified, an attempt is made to open that device ONLY (device = 1 specifies the first identified device, as reported by RtAudio::getDeviceInfo()).
-RtAudio provides four signed integer and two floating point data formats which can be specified using the RtAudioFormat parameter values mentioned earlier. If the opened device does not natively support the given format, RtAudio will automatically perform the necessary data format conversion.
-The bufferSize parameter specifies the desired number of sample frames which will be written to and/or read from a device per write/read operation. The nBuffers parameter is used in setting the underlying device buffer parameters. Both the bufferSize and nBuffers parameters can be used to control stream latency though there is no guarantee that the passed values will be those used by a device (the nBuffers parameter is ignored when using the OS X CoreAudio, Linux Jack, and the Windows ASIO APIs). In general, lower values for both parameters will produce less latency but perhaps less robust performance. Both parameters can be specified with values of zero, in which case the smallest allowable values will be used. The bufferSize parameter is passed as a pointer and the actual value used by the stream is set during the device setup procedure. bufferSize values should be a power of two. Optimal and allowable buffer values tend to vary between systems and devices. Check the API Notes section for general guidelines.
+The RtAudio::openStream() method attempts to open a stream with a specified set of parameter values. In this case, we attempt to open a two channel playback stream with the default output device, 32-bit floating point data, a sample rate of 44100 Hz, a frame rate of 256 sample frames per read/write, and 4 internal device buffers. When device = 0, RtAudio first attempts to open the default audio device with the given parameters. If that attempt fails, RtAudio searches through the remaining available devices in an effort to find a device that will meet the given parameters. If all attempts are unsuccessful, an RtError is thrown. When a non-zero device value is specified, an attempt is made to open that device ONLY (device = 1 specifies the first identified device, as reported by RtAudio::getDeviceInfo()).
+RtAudio provides four signed integer and two floating point data formats that can be specified using the RtAudioFormat parameter values mentioned earlier. If the opened device does not natively support the given format, RtAudio will automatically perform the necessary data format conversion.
+The bufferSize parameter specifies the desired number of sample frames that will be written to and/or read from a device per write/read operation. The nBuffers parameter is used in setting the underlying device buffer parameters. Both the bufferSize and nBuffers parameters can be used to control stream latency though there is no guarantee that the passed values will be those used by a device (the nBuffers parameter is ignored when using the OS X CoreAudio, Linux Jack, and the Windows ASIO APIs). In general, lower values for both parameters will produce less latency but perhaps less robust performance. Both parameters can be specified with values of zero, in which case the smallest allowable values will be used. The bufferSize parameter is passed as a pointer and the actual value used by the stream is set during the device setup procedure. bufferSize values should be a power of two. Optimal and allowable buffer values tend to vary between systems and devices. Check the API Notes section for general guidelines.
As noted earlier, the device capabilities reported by a driver or underlying audio API are not always accurate and/or may be dependent on a combination of device settings. Because of this, RtAudio does not attempt to query a device's capabilities or use previously reported values when opening a device. Instead, RtAudio simply attempts to set the given parameters on a specified device and then checks whether the setup is successful or not.
@@ -219,7 +219,7 @@ Once the device is open for playback, there are only a few final steps necessary goto cleanup; } - // An example loop which runs for 40000 sample frames + // An example loop that runs for 40000 sample frames count = 0; while (count < 40000) { // Generate your samples and fill the buffer with bufferSize sample frames of data @@ -256,7 +256,7 @@ The first thing to notice in this example is that we attempt to open a stream du Assuming the constructor is successful, it is necessary to get a pointer to the buffer, provided by RtAudio, for use in feeding data to/from the opened stream. Note that the user should NOT attempt to deallocate the stream buffer memory ... memory management for the stream buffer will be automatically controlled by RtAudio. After starting the stream with RtAudio::startStream(), one simply fills that buffer, which is of length equal to the returned bufferSize value, with interleaved audio data (in the specified format) for playback. Finally, a call to the RtAudio::tickStream() routine triggers a blocking write call for the stream.
In general, one should call the RtAudio::stopStream() and RtAudio::closeStream() methods after finishing with a stream. However, both methods will implicitly be called during object destruction if necessary.
+The primary difference in using RtAudio with callback functionality involves the creation of a user-defined callback function. Here is an example that produces a sawtooth waveform for playback.
#include <iostream> #include "RtAudio.h" @@ -331,7 +331,7 @@ The primary difference in using RtAudioreturn 0; }
-After opening the device in exactly the same way as the previous example (except with a data format change), we must set our callback function for the stream using RtAudio::setStreamCallback(). When the underlying audio API uses blocking calls (OSS, ALSA, SGI, and Windows DirectSound), this method will spawn a new process (or thread) which automatically calls the callback function when more data is needed. Callback-based audio APIs (OS X CoreAudio Linux Jack, and ASIO) implement their own event notification schemes. Note that the callback function is called only when the stream is "running" (between calls to the RtAudio::startStream() and RtAudio::stopStream() methods). The last argument to RtAudio::setStreamCallback() is a pointer to arbitrary data that you wish to access from within your callback function.
+After opening the device in exactly the same way as the previous example (except with a data format change), we must set our callback function for the stream using RtAudio::setStreamCallback(). When the underlying audio API uses blocking calls (OSS, ALSA, SGI, and Windows DirectSound), this method will spawn a new process (or thread) that automatically calls the callback function when more data is needed. Callback-based audio APIs (OS X CoreAudio Linux Jack, and ASIO) implement their own event notification schemes. Note that the callback function is called only when the stream is "running" (between calls to the RtAudio::startStream() and RtAudio::stopStream() methods). The last argument to RtAudio::setStreamCallback() is a pointer to arbitrary data that you wish to access from within your callback function.
In this example, we stop the stream with an explicit call to RtAudio::stopStream(). When using callback functionality, it is also possible to stop a stream by returning a non-zero value from the callback function.
Once set with RtAudio::setStreamCallback, the callback process exists for the life of the stream (until the stream is closed with RtAudio::closeStream() or the RtAudio instance is deleted). It is possible to disassociate a callback function and cancel its process for an open stream using the RtAudio::cancelStreamCallback() method. The stream can then be used with blocking functionality or a new callback can be associated with it.
The example compiler statements above could be used to compile the probe.cpp example file, assuming that probe.cpp, RtAudio.h, RtError.h, and RtAudio.cpp all exist in the same directory.
__RTAUDIO_DEBUG__ to the compiler (or uncomment the definition at the bottom of RtAudio.h). A variety of warning messages will be displayed which may help in determining the problem. Also try using the programs included in the test directory. The program info displays the queried capabilities of all hardware devices found.__RTAUDIO_DEBUG__ to the compiler (or uncomment the definition at the bottom of RtAudio.h). A variety of warning messages will be displayed that may help in determining the problem. Also try using the programs included in the test directory. The program info displays the queried capabilities of all hardware devices found.-The ALSA library includes OSS emulation support. That means that you can run programs compiled for the OSS API even when using the ALSA drivers and library. It should be noted however that OSS emulation under ALSA is not perfect. Specifically, channel number queries seem to consistently produce invalid results. While OSS emulation is successful for the majority of RtAudio tests, it is recommended that the native ALSA implementation of RtAudio be used on systems which have ALSA drivers installed.
+The ALSA library includes OSS emulation support. That means that you can run programs compiled for the OSS API even when using the ALSA drivers and library. It should be noted however that OSS emulation under ALSA is not perfect. Specifically, channel number queries seem to consistently produce invalid results. While OSS emulation is successful for the majority of RtAudio tests, it is recommended that the native ALSA implementation of RtAudio be used on systems that have ALSA drivers installed.
The ALSA implementation of RtAudio makes no use of the ALSA "plug" interface. All necessary data format conversions, channel compensation, de-interleaving, and byte-swapping is handled by internal RtAudio routines.
The Jack API is based on a callback scheme. RtAudio provides blocking functionality, in addition to callback functionality, within the context of that behavior. It should be noted, however, that the best performance is achieved when using RtAudio's callback functionality with the Jack API. At the moment, only one RtAudio instance can be connected to the Jack server. Because RtAudio does not provide a mechanism for allowing the user to specify particular channels (or ports) of a device, it simply opens the first N enumerated Jack ports for input/output.
+The DirectSound version of RtAudio can be compiled with or without the UNICODE preprocessor definition.
-A number of ASIO source and header files are required for use with RtAudio. Specifically, an RtAudio project must include the following files: asio.h,cpp; asiodrivers.h,cpp; asiolist.h,cpp; asiodrvr.h; asiosys.h; ginclude.h; iasiodrv.h. The Visual C++ projects found in /tests/Windows/ compile both ASIO and DirectSound support.
asio.h,cpp; asiodrivers.h,cpp; asiolist.h,cpp; asiodrvr.h; asiosys.h; ginclude.h; iasiodrv.h; iasiothiscallresolver.h,cpp. The Visual C++ projects found in /tests/Windows/ compile both ASIO and DirectSound support.
+The Steinberg provided asiolist class does not compile when the preprocessor definition UNICODE is defined. Note that this could be an issue when using RtAudio with Qt, though Qt programs appear to compile without the UNICODE definition (try DEFINES -= UNICODE in your .pro file). RtAudio with ASIO support has been tested using the MinGW compiler under Windows XP, as well as in the Visual Studio environment.
+Many thanks to the following people for providing bug fixes and improvements:
The RtAudio API incorporates many of the concepts developed in the PortAudio project by Phil Burk and Ross Bencina. Early development also incorporated ideas from Bill Schottstaedt's sndlib. The CCRMA SoundWire group provided valuable feedback during the API proposal stages.
The early 2.0 version of RtAudio was slowly developed over the course of many months while in residence at the Institut Universitari de L'Audiovisual (IUA) in Barcelona, Spain and the Laboratory of Acoustics and Audio Signal Processing at the Helsinki University of Technology, Finland. Much subsequent development happened while working at the Center for Computer Research in Music and Acoustics (CCRMA) at Stanford University. The most recent version of RtAudio was finished while working as an assistant professor of Music Technology at McGill University. This work was supported in part by the United States Air Force Office of Scientific Research (grant #F49620-99-1-0293).