Skip to main content. Select Product Version. All Products. Additionally, if the client detects a round-trip delay of more than 25ms, it automatically flips to a low-bandwidth codec for example, Siren instead of G. More Information. To enable this feature, install the following cumulative update: Description of the cumulative update for Lync November General information about Call Admission Control.

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Stack Overflow for Teams is a private, secure spot for you and your coworkers to find and share information. Can anyone with experience using both libraries recommend one over the other? Here's a relevant thread that came up on the Linux audio devel list. Getting jack sample rate and buffersize with portaudio.

The PortAudio Java bindings are oversimplified. For example, you cannot set exclusive mode or shared mode for device access, which is a shame because it effectively steals 7ms of less latency from you -- especially if you capture microphone input where usually no shared access is necessary under Windows But are there Java bindings? Wasn't asked in the question, I know, but still Another, rather new project is XtAudio.

It exposes exclusive and shared access to devices as different sets of devices to the frontend and in general seems to have a better API support. Don't think it supports MIDI tho. Learn more. RtAudio or PortAudio, which one to use?

Ask Question. Asked 9 years, 4 months ago. Active 25 days ago. Viewed 11k times. Rafael Vega. Rafael Vega Rafael Vega 4, 4 4 gold badges 25 25 silver badges 45 45 bronze badges. Did you eventually find a good library? I have exactly the same problem I just did little experiments with PortAudio and it seems quite complex very low levelit lacks of an official forum I hate mail list.

RTAudio lacks of documentation. STK has annoying include policy. Active Oldest Votes. I have been locking for the perfect match with that need for a quite long time. Daniel Russell 49 5 5 bronze badges. Thanks for the answer!


It is an old thread anyway : I use visual on windows, or qtcreator for qt. However for port audio I used just makefiles and gedit on linux.

If you work on linux you can do a makefile or use a tool like scons, or cmake anyway if you work on linux, just try to find a tool that is portable on mac and windows with mingw and everything should be right.The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.

Any person wishing to distribute modifications to the Software is asked to send the modifications to the original developer so that they can be incorporated into the canonical version.

This is, however, not a binding provision of this license. The constructor performs minor initialization tasks. An exception can be thrown if no API support is compiled. The values returned in the std::vector can be compared against the enumerated list values. Note that there can be more than one API compiled for certain operating systems. This obtains a short lower-case name used for identification purposes. This value is guaranteed to remain identical across library versions. If the API is unknown, this function will return the empty string.

This obtains a long name used for display purposes. A case insensitive comparison will check the specified name against the list of compiled APIs, and return the one which matches. This function performs a system query of available devices each time it is called, thus supporting devices connected after instantiation. If a system error occurs during processing, a warning will be issued.

Return an RtAudio::DeviceInfo structure for a specified device number. Any device integer between 0 and getDeviceCount - 1 is valid. If a device is busy or otherwise unavailable, the structure member "probed" will have a value of "false" and all other members are undefined. If the specified device is the current default input or output device, the corresponding "isDefault" member will have a value of "true".

If the underlying audio API does not provide a "default device", or if no devices are available, the return value will be 0. Note that this is a valid device identifier and it is the client's responsibility to verify that a device is available before attempting to open a stream. A warning is issued if the stream is already running.

A warning is issued if the stream is already stopped. For duplex streams, the returned value will represent the sum of the input and output latencies. If the API does not report latency, the return value will be zero. On some systems, the sample rate used may be slightly different than that specified in the stream parameters. Audio API specifier arguments. The class constructor.

The destructor. If a stream is running or open, it will be stopped and closed automatically. A static function to determine the available compiled audio APIs.It provides a well-documented API that operates consistently regardless of the sound driver it connects to.

It performs no buffering or processing on your behalf; instead exposing the raw power of the underlying backend. It even handles out of memory conditions correctly. The primary developer of libsoundio is Andrew Kelleywho can be found on GitHub. Raw devices give you the best performance but prevent other applications from using them. Shared devices are default and usually provide sample rate conversion and format conversion.

Exposes both device id and friendly name. Supports optimal usage of each supported backend. C library. Depends only on the respective backend API libraries and libc. Errors are communicated via return codes, not logging to stdio.

Supports channel layouts also known as channel mapsimportant for surround sound applications. Ability to monitor devices and get an event when available devices change. Ability to get an event when the backend is disconnected, for example when the JACK server or PulseAudio server shuts down.

Detects which input device is default and which output device is default. Ability to connect to multiple backends at once.

Meticulously checks all return codes and memory allocations and uses meaningful error codes. MIT licensed.RTAudio is a Microsoft produced adaptive wide-band speech codec.


Some of the target applications include games, audio conferencing, and wireless applications over IP. The RTAudio encoder is capable of encoding single-channel mono16 bit per sample audio signals.

The RTAudio decoder has a built-in jitter control module as well as an error concealment module. RTAudio is a proprietary codec. Like RTVideothis protocol can also be licensed from Microsoft. From Wikipedia, the free encyclopedia.

Licensing [ edit ] RTAudio is a proprietary codec.

CppCon 2016: Timur Doumler “Want fast C++? Know your hardware!\

Multimedia compression and container formats. RTP Ogg. Comparison of audio coding formats Comparison of video codecs List of codecs. See Compression methods for techniques and Compression software for codecs. Categories : Audio codecs Speech codecs Microsoft proprietary codecs Wideband codecs Microsoft software stubs. Hidden categories: All stub articles.

Namespaces Article Talk. Views Read Edit View history. Help Community portal Recent changes Upload file. Download as PDF Printable version. This article about software created or produced by Microsoft is a stub. You can help Wikipedia by expanding it.Please contact mpub-help umich.

For more information, read Michigan Publishing's access and usage policy. Scavone gary ccrma. RtAudio provides a flexible, easy to use application programming interface API which allows complete audio system control, including device capability querying, multiple concurrent streams, blocking and callback functionality. As a result, an attempt to provide multi-platform support for an audio application can prove difficult at best.


To further complicate matters, multiple audio driver interfaces often exist for a single operating system. RtAudio was designed to provide a common interface across a variety of these APIs in as flexible, yet simple, manner as possible.

However, the latest release of RtAudio version 2. RtAudio significantly simplifies the process of interfacing with computer audio hardware.

RTP payload formats

Available audio devices and their capabilities can be enumerated and then specified when opening a stream. Multiple streams can run at the same time and, when allowed by the underlying audio API, a single device can serve multiple streams. All public RtAudio functions are thread-safe. This allows users to safely RtAudio offers uniform support for 8-bit, 16 -bit, bit, and bit signed integer data formats, as well as bit and bit floating point formats.

When an audio device does not natively support a requested user format, RtAudio provides automatic format conversion. In addition, internal routines will automatically perform any byte-swapping, channel number compensation, and channel de-interleaving required by the underlying audio driver or hardware. The ALSA driver model was recently incorporated into the Linux development kernel and will likely gain wide acceptance in the near future.

All source code for RtAudio is made freely available, allowing full user extensibility and customization. Likewise, all uses of RtAudio must end with class destruction. An RtError can be caught by type, providing a means for error correction or at a minimum, more detailed error reporting. Almost all RtAudio methods can "throw" an RtError, most typically if an invalid stream identifier is supplied to a method or a driver error occurs.

There are a number of cases within RtAudio where warning messages may be displayed but an exception is not thrown. Alternately, a stream can be opened after instantiation in much the same way. When successful, a stream identifier is returned which must be used for subsequent function calls on the stream. Audio devices are identified by integer values of one and higher, as enumerated by the getDeviceInfo function.


When a device identifier of zero is If that fails, an attempt is made to find a device or set of devices which will meet the given parameters. If all attempts are unsuccessful, an RtError is thrown. When a positive, non-zero device value is specified, no additional devices are probed. Example program code is provided in the appendix of this paper.The details of media encoding, such as signal sampling rate, frame size and timing, are specified in an RTP payload format.

The standard also describes the process of registering new payload types with IANA; additional payload formats and payload types are defined in the following specifications:. Payload identifiers 96— are used for payloads defined dynamically during a session. It is recommended to dynamically assigned port numbers, although port numbers and have been registered for use of the profile when a dynamically assigned port is not required.

From Wikipedia, the free encyclopedia. Redirected from RTP audio video profile. For instance, video codecs typically use a clock rate of so their frames can be more precisely aligned with the RTCP NTP timestamp, even though video sampling rates are typically in the range of 1—60 samples per second.

So at sampling rates of, ora packetization rate of "4" is rounded down to 3. Schulzrinne, S. Casner, The Internet Society July May Retrieved March August Retrieved on Categories : VoIP protocols. Hidden categories: Pages using RFC magic links.

Namespaces Article Talk. Views Read Edit View history. Help Community portal Recent changes Upload file. Download as PDF Printable version. ITU-T G. RFC Experimental Linear Predictive Coding audio 5. RFC - Page Linear PCM bit Stereo audio RFCPage Linear PCM bit audio Qualcomm Code Excited Linear Prediction.

Comfort noise. Payload type used with audio codecs that do not support comfort noise as part of the codec itself such as G. Sun CellB video [5]. Internet low Bitrate Codec Adaptive Multi-Rate audio. Linear PCM 8-bit audio with offset. RFC Section 4. RFC Section 3.

cross-platform audio input and output

EVRC audio. EVRC-B audio. ATRAC 3 audio.

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