io-audio

If 1, don't use software techniques to increase the maximum number of playing channels on cards that have only a single channel in hardware.

You should specify this in the [CTRL] section for each card rather than at this point, to simplify enabling/disabling of the PCM software mixer on a card-by-card basis.

The mixer fragment size, in samples per voice. This is the deprecated method of specifying the mixer fragment size (overridden by sw_mixer_ms if specified). The minimum is 64 bytes, and the maximum 512 KB.

Because you cannot use a samples value to specify the fragment size the PCM input splitter uses, if you use the input splitter, you cannot use this option to specify the fragment size the PCM software mixer uses.

• 16 — 16-bit signed, native endian
• 32 — 32-bit signed, native endian

If you do not use this key, the sample size is determined by the sample size the underlying hardware uses:

• If the hardware is 8- or 16-bit, the mixer uses 16-bit signed, native endian.
• For 24- or 32-bit hardware, the mixer uses 32-bit signed, native endian.

• FAHQ — First active, high quality; the highest quality sample rate that the hardware supports and is a multiple of the rate requested by the first active client (44.1Khz and above).
• FA — First active; the sample rate requested by the first active client.
• Lvalue — Lock to the specified rate, where value is in the range 100–200000.

• Bit 0 DRIVER — Debug information from drivers. (e.g., you can also use INTERRUPT or CODEC for driver-level debug information).
• Bit 1 CONTROL — Debug information from the control path, state machine, etc.
• Bit 2 PCM — Logs control type information.
• Bit 3 MIXER — Mixer related debug debug information, such as volume and mute controls. Debug information from mixer DLL and codec level deva-* DLLs.
• Bit 4 IO_ADO — High-level Audio library debug information.
• Bit 5 RESMGR — Debug information from the resource manager (resmgr_*) callbacks.
• Bit 6 POWER — Debug information related to power management.
• Bit 7 MISC — Miscellaneous debug information.
• Bit 8 CODEC — Codec-level debug information from the PCM software mixer and deva-* DLLs.
• Bit 30 INTERRUPT — Interrupt handler debug information. (dma_interrupt, deva interrupt callbacks, etc.)
• Bit 31 MEMORY — Memory allocation debug information.

If the pool_name refers to typed memory, then you must also provide the pool_size option. If the pool_name refers to a shared memory object, then the pool is automatically sized to the size of the object (pool_size is ignored if provided). The io-audio service first tries to open the name as typed memory; if this fails, it tries to open the name as a shared memory object.

For more information on how io-audio sets the preferred device, see “Preferred device selection”.

For more information on how io-audio sets the preferred device, see “Preferred device selection”.

If this key isn't provided or if the number is a negative, a unique value is generated with the first available number starting at zero. After io-audio starts, the unit (card) number corresponds to the DLL mounted as specified in the audio configuration file. These numbers are reserved for the life of io-audio and can be reused only if the corresponding DLL has previously been unmounted. For more information, see “ Driver-specific configuration file”.

If you specify this parameter without an index suffix, the value applies to every capture-capable PCM device under the card created by this section.

If you specify this parameter with an index suffix, then the applied value (0/1) takes effect only for the PCM device specified by the index, overriding the value specified by an instance of the key without an index suffix.

To ensure that the input splitter can route data to a PCM software mixer device, both devices (input_splitter and PCM software mixer) must operate at the same data fragment size, which you set via the sw_mixer_ms option, and thus the sw_mixer_ms option is required when you're using the input splitter (even if the PCM software mixer device isn't being used).

The input splitter fragment size, in milliseconds. No default.

If this key is not specified, the sw_mixer_ms value from the current [CTRL] section determines the size. If the current [CTRL] section has no sw_mixer_ms key, sw_mixer_ms from the [GLOBAL] section or the command line for this card determines the size.

You must specify either this parameter or sw_mixer_ms (at the command-line, or in the [GLOBAL] or [CTRL] sections) if any AFMs are configured to capture through the PCM input splitter on this card.

To route data between the PCM input splitter and the PCM software mixer via an AFM, both the splitter and mixer must use a data fragment size that can be divided exactly into the AFM’s fragment period. (You can use ms_frag_period to set the AFM's fragment period.)

For example, to configure a Bridge AFM to operate with 8 millisecond fragments, specify ms_frag_period=8. To configure a QNX Acoustics for Voice AFM to use smaller but compatible mixer and splitter fragment sizes, specify sw_mixer_ms=1 and input_splitter_ms=2.

For more information, see ms_frag_period in the [AFM] section.

The method for setting the PCM input splitter sample rate.

If the underlying hardware supports multiple sample rates, the specified method determines which sample rate applies to the splitter:

• FAHQ — First active, high quality; the highest quality sample rate that the hardware supports and is a multiple of the rate requested by the first active client (44.1Khz and above).
• FA — First active; the sample rate requested by the first active client.
• Lvalue — A specified, locked rate, where value is in the range 100–200000.

If the value of this parameter is 0, a PCM Software mixer is instantiated.

If it's 1, io-audio doesn't use software techniques to increase the maximum number of playing channels on cards that have only a single channel in hardware.

For a Bluetooth device, this should be 1 (disable the PCM software mixer).

This parameter overrides the corresponding option from the [GLOBAL] section or the command line for this card.

The mixer fragment size, in milliseconds. No default and this key isn't used if a value isn't specified.

To specify the fragment size for a specific PCM device, add an underscore and the device name to the key. For example, sw_mixer_ms_0.

This parameter overrides the corresponding option from the [GLOBAL] section or the command line for this card.

• You must specify this parameter if any AFMs are configured to play back through the PCM software mixer on this card.
• To route data to the PCM software mixer via an acoustic AFM, the mixer must use a data fragment size that can be divided exactly into the AFM’s fragment period. (You can use ms_frag_period to set the AFM's fragment period.)
• This parameter overrides sw_mixer_samples. See also ms_frag_period in the [AFM] section.

This parameter overrides the corresponding option from the [GLOBAL] section or the command line for this card.

• FAHQ — First active, high quality; the highest quality sample rate that the hardware supports and is a multiple of the rate requested by the first active client (44.1Khz and above).
• FA — First active; the sample rate requested by the first active client.
• Lvalue — Lock to the specified rate, where value is in the range 100–200000.

This parameter overrides the corresponding option from the [GLOBAL] section or the command line for this card.

• 16 — 16-bit signed, native endian
• 32 — 32-bit signed, native endian

If you do not use this key, the sample size is determined by the sample size the underlying hardware uses.

• If the hardware is 8- or 16-bit, the mixer uses signed, native endian
• For 24- or 32-bit hardware, the mixer uses 32-bit signed, native endian

To specify the sample size for a specific PCM software mixer, append an underscore and the device name to the key. For example, sw_mixer_sample_size_0.

If 1, enable a 1 ms soft look ahead limiter on the main media mixing path through the PCM software mixer.

If 0, audio streams are strictly added and hard-clamped.

This value is discarded unless afm_mixer_enable is set to 1 and there are configured AFMs.

Normally, when the AFM mixer is enabled there is an implicit 6 dB attenuation of both the main (audio stream) and the AFM mixing paths in the PCM mixer, so that the combined output of these two paths with never need to be clamped, avoiding the audio artifacts which could arise when performing acoustic echo cancellation.

This implicit attenuation can be disabled for target systems where it is known that the AFM audio and media playback will never run concurrently. This can be true in systems where only the voice AFM is integrated. However if integrating ASD or ICC, which do typically run concurrently with media playback, this key shouldn't be used.

If set to 1, the AFM mixer attenuation is disabled for all devices on a card; otherwise if set to 0, the AFM mixer attenuation is enabled for all devices. An optional _index suffix can be appended to the key to target the option at a specific PCM device.

Normally the AFM mixer gain is 1/afm_cnt to avoid clipping (non-linearity) when mixing, where afm_cnt is the number of AFMs, which have a pcm_local_playback device and an afm (default) sw_mixer_route. If you know that AFMs aren't going to be concurrently used, or if the AFMs are tuned correctly, this restriction can be overridden to a value less than afm_cnt.

If this value is set to 0, the key/value pair is ignored for all devices. An optional _index suffix can be appended to the key to target the option at a specific PCM device.

When set to 0, the card is mounted at startup. When set to 1, mounting of the card is deferred until the mount command is issued. The acoustics option can be used with the mount command to mount all AFM cards that were deferred.

• icc — QNX In-Car Communications (ICC) that provides algorithms to improve the ease of communication between passengers within a vehicle.
• voice — QNX Acoustics for Voice (Voice) that provides algorithms to detect and extract voice from conditions that have lots of noise (wind, road, etc.).
• asd — QNX Active Sound Design (ASD) to use active-noise cancellation to reduce engine tones, as well as, synthesize sound to support a manufacturer's vision of what their vehicle should sound like.
• bridge — bridges audio inputs to audio outputs and assumes the clocking for the inputs and outputs is synchronized

• Library inputs and capture voices are 0-based.
• Only one map to a single capture voice is allowed per library input.
• You must map all library inputs or none.
• Maps for unused library inputs are ignored.

• Library outputs and playback voices are 0-based.
• Maps to multiple voices are allowed per library output.
• You must map all library outputs or none.
• Maps for unused library outputs are ignored.
• Maps for non-existent playback voices are ignored.

• Library inputs and capture voices are 0-based.
• Only one map to a single capture voice is allowed per library input.
• You must map all library inputs or none.
• Maps for unused library inputs are ignored.

These keys are parsed when the AFM is started; if there are errors, the AFM won't start.

• Library inputs and capture voices are 0-based.
• Maps for unused library inputs are ignored.

These keys are parsed when the AFM is started; if there are errors, the AFM won't start.

• voice: libqwa_qvp.so
• icc: libqwa_icc.so
• asd: libqwa_asd.so

The default audio mode is used when a mode is not set using the libasound library snd_afm_set_audio_mode() function.

mode is a string set using the libasound library snd_afm_set_audio_mode() function. It specifies an audio mode that corresponds to a particular audio environment. For example, for Bluetooth hands-free phone calls that operate at 8kHZ (narrowband) and 16 KHz (wideband), you can create the corresponding modes "nb" and "wb". To apply sample rates other than the default to these modes, create keys ap_qcf_wb and ap_qcf_nb.

The default audio mode is used when a mode is not set using the libasound library snd_afm_set_audio_mode() function.

mode is a string set using the libasound library snd_afm_set_audio_mode() function. It specifies an audio mode that corresponds to a particular audio environment.

The default audio mode is used when a mode is not set using the libasound library snd_afm_set_audio_mode() function.

mode is a string set using the libasound library snd_afm_set_audio_mode() function. It specifies an audio mode that corresponds to a particular audio environment.

If 0, the baseline settings don't turn on diagnostics in the acoustic library. If 1, the baseline settings enable the diagnostics module in the acoustic library. Note that the baseline settings can be overridden by the QCF file.

Enable pass through (0 / 1). If 0, the acoustic processing library is used. If 1, the acoustic processing library isn't loaded nor used and the audio data is copied from the input to the output of the AFMs.

• When the value is afm, playback audio is routed through the AFM mixer path in the PCM software mixer.
• When the value is main, playback audio is routed through the main (media) path in the PCM software mixer.

This setting changes the effect that audio ducking can have on the Bridge AFM playback audio. See “ Understanding audio ducking” in the “Audio Management” chapter in the QNX Neutrino Audio Developer's Guide.

When the local playback device is a PCM Software Mixer device, must be a multiple of sw_mixer_ms.

You must always specify this parameter for the Bridge AFM.

The acoustic library fragment period must be a multiple of this value (and implicitly of sw_mixer_ms).

If 0, volume control, that uses the acoustic library gain and mute controls is disabled. If 1, a mixer group that uses the acoustic library gain and mute controls (ICC gain or Voice Receive Gain) to adjust the volume is published.

If 0, mute control using the acoustic library and gain control is disabled. If 1, a mixer interface that uses the acoustic library mute controls to perform muting is published. This key is used if mute control is required, but the library QCF (.qcf) configuration file is used to set a fixed library gain value.

• ICC: 20 dB
• Voice: 10 dB

This parameter must be >= min_vol_db. If this value is set beyond the ICC/QVP library maximum of 40 dB, a warning is logged, but the value is accepted.

• ICC: -10 dB
• Voice: -20 dB

This parameter must be <= max_vol_db. If this value is set beyond the ICC/QVP library minimum of -30 dB, a warning is logged, but the value is accepted.

• ICC: 35
• Voice:100

The initial mix_ctl volume setting at boot (0-100) for speaker output X.This key is ignored if enable_vol_ctl is set to 0. ICC and Voice AFMs only. The value is clamped between 0 and 100 if out of range.

The Bridge is an AFM that connects audio between two devices. For example, it captures audio from a microphone input and plays it out to a speaker output. It times itself on the hardware capture and playback interrupts.

The Bridge can use only capture interrupts, playback interrupts, or both for timing. When capture or playback is specified as a value for this key, input is captured and played out when an interrupt occurs on a specified channel. When both is specified, both capture and playback interrupts are used. Audio is input when a capture interrupt occurs and played out when a playback interrupt occurs.

When set to 1, audio is restarted when overrun or underrun is detected. When set to 0, no correction will be performed if an overrun or underrun is detected.

When set to 1, once capture audio is written to the playback buffer, clear and silence the capture buffer. This prevents stale data from being left behind. When set to 0, the capture buffer isn't overwritten with silence.

This key can be used to differentiate between AFMs if there are multiple AFMs of the same type configured. For example, if there are multiple bridges, their names could be overwritten to bridge1, bridge2 and so forth, and accessed with afm_ctl -a bridge1 or afm_ctl -a bridge2.

Overriding the cardname also impacts the name of the resource file used to store the card mixer settings when deva-util-restore.so is being used. By default, the restore files are named Afm cardname @ cardnumberio-ado-cfg. For example, Afm Icc @ 4.io-ado-cfg. If you override the cardname to my_newname, the restore file would be renamed to my_newname @ 4.io-ado-cfg.

This key/value fair can be used to compensate for variable interrupt timing. For example, if local card interrupts occurs in a cycle of 12ms/18ms/18ms, instead of 16 ms, data may be dropped. Increasing the number of remote fragments to be 2 (or higher) compensates for this inconsistency at a the cost of a higher round trip latency.

When set to 0, the card is mounted at startup. When set to 1, mounting of the card is deferred until the mount command is issued. The acoustics option can be used with the mount command to mount all AFM cards that were deferred.

The default audio mode is used when a mode is not set using the libasound library snd_afm_set_audio_mode() function.

Applies to acoustic AFMs only.

mode is a string set using the libasound library snd_afm_set_audio_mode() function. It specifies an audio mode that corresponds to a particular audio environment. For example, for Bluetooth hands-free phone calls that operate at 8kHZ (narrowband) and 16 KHz (wideband), you can create the corresponding modes "nb" and "wb". To apply sample rates other than the default to these modes, create keys local_rate_wb and local_rate_nb.

The default audio mode is used when a mode is not set using the libasound library snd_afm_set_audio_mode() function.

Applies to the QAV AFM only.

mode is a string set using the libasound library snd_afm_set_audio_mode() function. It specifies an audio mode that corresponds to a particular audio environment. For example, for Bluetooth hands-free phone calls that operate at 8kHZ (narrowband) and 16 KHz (wideband), you can create the corresponding modes "nb" and "wb". To apply sample rates other than the default to these modes, create keys remote_rate_wb and remote_rate_nb.

Applies to the QAV AFM only.

The deva-acs- prefix and the .so suffix are omitted when specifying the module name. For example, link corrsponds to deva-acs-link.so. The link module is a special type of AFM that integrates the Acoustic Control Server (ACS) for tuning purposes; it performs no acoustic processing.

When set to 1, the Link AFM is available to be started. When set to 0, the link AFM isn't available and can't be started.

When set to 0, the card is mounted at startup. When set to 1, mounting of the card is deferred until the mount command is issued. The acoustics option can be used with the mount command to mount all AFM cards that were deferred.