Contributions:AudioExtension - Revision history

Nluczak at 15:27, 28 October 2025

2025-10-28T15:27:03Z

← Older revision		Revision as of 15:27, 28 October 2025
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	===Notes===		===Notes===

	Use '''~~ASIO~~''' for devices supporting ≥4 inputs.		Use '''WASAPI''' for devices supporting ≥4 inputs.

	'''Debug builds''' may drop audio—use '''Release''' for deployment.		'''Debug builds''' may drop audio—use, use '''Release''' for deployment.

	Increasing <code>AudioBufferSize</code> can reduce dropouts.		Increasing <code>AudioBufferSize</code> can reduce dropouts.

	~~To record more than 4 input envelopes, modify and recompile:~~

	~~<pre> #define NUM_INPUT_ENVELOPES 4 #define NUM_OUTPUT_ENVELOPES 4 </pre>~~

	~~in <code>AudioExtension.cpp</code>.~~

	====Troubleshooting====		====Troubleshooting====

	If only two input channels are available, ensure the device driver and host API are ~~ASIO~~.		If only two input channels are available, ensure the device driver and host API are WASAPI.

	For stability issues, increase buffer size or disable envelope computation.		For stability issues, increase buffer size or disable envelope computation.

	~~Restart trials if DirectSound (non-ASIO) audio desynchronizes.~~

	==See also==		==See also==

Nluczak: /* Configuring AudioExtension for 4-Channel Recording */

2025-10-13T19:35:08Z

Configuring AudioExtension for 4-Channel Recording

← Older revision		Revision as of 19:35, 13 October 2025
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	===Building with ASIO Support===		===Building with ASIO Support===

	Download the '''ASIO SDK''' from [https://www.steinberg.net/en/company/developers.html		Download the '''ASIO SDK''' from [https://www.steinberg.net/en/company/developers.html Steinberg’s developer site].
	Steinberg’s developer site].
	Extract the downloaded archive into:		Extract the downloaded archive into:

Nluczak: Added tutorial for 4 channel recording

2025-10-13T17:28:19Z

Added tutorial for 4 channel recording

Show changes

Mellinger: /* AudioRecordingFormat */

2025-06-03T16:35:06Z

AudioRecordingFormat

← Older revision		Revision as of 16:35, 3 June 2025
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	*Lossless - Records to FLAC formatted files. These files are slightly smaller than RAW files, but have no quality loss.		*Lossless - Records to FLAC formatted files. These files are slightly smaller than RAW files, but have no quality loss.
	*Lossy - Records to Ogg Vorbis files. These files are similar to MP3 but do not have the associated licensing issues. They are compressed using a lossy algorithm, so the resulting files are very small but sound slightly worse than lossless encoding. This format is good for long recordings where perfect quality is not necessary.		*Lossy - Records to Ogg Vorbis files. These files are similar to MP3 but do not have the associated licensing issues. They are compressed using a lossy algorithm, so the resulting files are very small but sound slightly worse than lossless encoding. This format is good for long recordings where perfect quality is not necessary.

			Output files are located in the current BCI2000 output directory, and bear the <tt>.dat</tt> file's name, with its extension replaced with <tt>_in.wav</tt> or <tt>_out.wav</tt>, respectively (<tt>.flac</tt> and <tt>.ogg</tt> for the remaining two file format options).

	===AudioInputFilterbank, AudioOutputFilterbank===		===AudioInputFilterbank, AudioOutputFilterbank===
	A filterbank which filters audio input and output before rectification/smoothing for envelope extraction. These butterworth filters will not be applied to the audible signal. The format of the filter bank is as follows:		A filterbank which filters audio input and output before rectification/smoothing for envelope extraction. These butterworth filters will not be applied to the audible signal. The format of the filter bank is as follows:

Mellinger: /* Integration into BCI2000 */

2024-12-17T17:19:51Z

Integration into BCI2000

← Older revision		Revision as of 17:19, 17 December 2024
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	==Integration into BCI2000==		==Integration into BCI2000==
	Compile the extension into your source module by enabling contributed extensions in your CMake configuration. You can do this by going into your root build folder and deleting <code>CMakeCache.txt</code> and re-running the project batch file, or by running <code>cmake -i</code> and enabling '''BUILD_AUDIOEXTENSION'''. Once the extension is built into the source module, enable it by starting the source module with the <code>--EnableAudioExtension=1</code> command line argument ~~(NB, as explained below, the numeric value here matters, and denotes the audio API to be used: =1 means DirectSound)~~.		Compile the extension into your source module by enabling contributed extensions in your CMake configuration. You can do this by going into your root build folder and deleting <code>CMakeCache.txt</code> and re-running the project batch file, or by running <code>cmake -i</code> and enabling '''BUILD_AUDIOEXTENSION'''. Once the extension is built into the source module, enable it by starting the source module with the <code>--EnableAudioExtension=1</code> command line argument.

	===Building with ASIO support===		===Building with ASIO support===

Mellinger: /* Known Issues */

2024-12-17T17:19:23Z

Known Issues

← Older revision		Revision as of 17:19, 17 December 2024
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	===Known Issues===		===Known Issues===
	* Using DirectSound when suspending and resuming states can cause an issue where the file recorded drops samples, this can be fixed by suspending and resuming until the audio clears up. ''Luckily, ~~AudioExtensions~~ plays back what has been recorded so its easy to detect when this issue happens'', just restart the trial to fix or use ASIO where no known issues exist.		* Using DirectSound when suspending and resuming states can cause an issue where the file recorded drops samples, this can be fixed by suspending and resuming until the audio clears up. ''Luckily, AudioExtension plays back what has been recorded so its easy to detect when this issue happens'', just restart the trial to fix or use ASIO where no known issues exist.

	*When compiling in Debug mode the audio clips and some data may be lost, this '''DOES NOT''' occur in release mode.		*When compiling in Debug mode the audio clips and some data may be lost, this '''DOES NOT''' occur in release mode.

Mellinger: /* Source Code Revisions */

2024-12-17T17:18:36Z

Source Code Revisions

← Older revision		Revision as of 17:18, 17 December 2024
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	===Source Code Revisions===		===Source Code Revisions===
	*Initial development: 4095		*Initial development: 4095
	*Tested under: ~~5896~~		*Tested under: 8516
	*Known to compile under: ~~5896~~		*Known to compile under: 8516
	*Broken since: --		*Broken since: --

Mellinger: /* AudioEnvelopeSmoothing */

2024-12-17T17:17:26Z

AudioEnvelopeSmoothing

← Older revision		Revision as of 17:17, 17 December 2024
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	===AudioEnvelopeSmoothing===		===AudioEnvelopeSmoothing===
	The cutoff frequency for the low pass filter which is applied to the filtered and full-wave rectified audio data. This should be set to the highest frequency you want to see in the resulting audio envelope.		The cutoff frequency for the low pass filter which is applied to the filtered and full-wave rectified audio data. This should be set to the highest frequency you want to see in the resulting audio envelope.

			===ComputeEnvelopes===
			When set to 0 (false), the relatively CPU intensive envelope computation for input and output signal is omitted. This may be useful if you experience audio sample loss.

	==State Variables==		==State Variables==

Mellinger: /* AudioMixer */

2024-12-17T17:14:33Z

AudioMixer

← Older revision		Revision as of 17:14, 17 December 2024
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	When set to 0 (auto), the audio extension will choose a preferred host API. On Windows, DirectSound is chosen over WASAPI although WASAPI allows for lower latency because DirectSound is more compatible across multiple devices for input and output.		When set to 0 (auto), the audio extension will choose a preferred host API. On Windows, DirectSound is chosen over WASAPI although WASAPI allows for lower latency because DirectSound is more compatible across multiple devices for input and output.

			===AudioBufferSize===
			The size of the audio buffer, in audio frames. When set to "auto", defaults to 2048 (50ms).
			Enlarge in steps of 1024 if you experience audio frame loss.

	===AudioMixer===		===AudioMixer===

Mellinger: /* AudioExtensionHostAPI */

2024-12-17T17:10:41Z

AudioExtensionHostAPI

← Older revision		Revision as of 17:10, 17 December 2024
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	===AudioExtensionHostAPI===		===AudioExtensionHostAPI===
	This parameter is an audio host API selector. The following values of this parameter are valid. NOTE: Not all audio APIs are available on all platforms.		This parameter is an audio host API selector. The following values of this parameter are valid. NOTE: Not all audio APIs are available on all platforms.
	**[0] - ~~Disabled~~		**[0] - auto
	**[1] - DirectSound		**[1] - DirectSound
	**[2] - MME		**[2] - MME
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	**[16] - PulseAudio		**[16] - PulseAudio
	**[17] - Sndio		**[17] - Sndio

			When set to 0 (auto), the audio extension will choose a preferred host API. On Windows, DirectSound is chosen over WASAPI although WASAPI allows for lower latency because DirectSound is more compatible across multiple devices for input and output.

	===AudioMixer===		===AudioMixer===