Start one or more audio drivers
io-audio [-c config_file] [-v[v]...] [-l level] [-m opt[,opt...]] [-o opt[,opt...]] [-U string] [-d driver[opt[,opt...]]]
QNX Neutrino
If you want to start the driver from a configuration file, all the following driver options can be specified using the options key-value pair in the [CTRL] section, with the exception of the unit option.
All audio drivers support the following card options:
This option allows you specify driver settings in a separate, driver-specific configuration file.
io-audio -d audiopci -dindex=0, restrict_interface=capture
on -u100:100 -A able=io-audio/capturerestricted, allow,nonroot waverec /temp/test.wav
When you use the mount command to dynamically load a driver, you can use the unit number to specify which [CTRL] section in the configuration file to use. For more information on the different scenarios, see the Description section. .
If the smmuman service is running, specify 1 or on to configure io-audio to interface with smmuman to manage ("cage") access for non-CPU initiated reads and writes (i.e., DMA devices).
If you do not use this key, the sample size is determined by the sample size the underlying hardware uses:
sw_mixer_samples = sample_rate (in samples per second) * fragment_length (in seconds)
In QNX Neutrino 7.0 or later, sw_mixer_samples is the number of samples per channel.
In the second form, the primary group is the one specified for user_name in /etc/passwd.
The io-audio service allows you to dynamically load audio drivers when you start io-audio. You use the -d options to specify the drivers to load.
You can run this command as root or a non-root user. When you run this command as a non-root user, it must be granted the proper process abilities. The io-audio manager includes custom permissions to restrict capture and playback. For more information, see the restrict_interface driver option for the driver (-d) option.
Once io-audio has started, you can dynamically load and unload drivers using the mount and umount commands.
Mounting a driver
Here's how you can use the mount command to dynamically load drivers. For example, this command:
io-audio -dusb -daudiopci
gives the same result as this sequence:
io-audio mount -T io-audio usb mount -T io-audio audiopci
Unit (card) numbers are reserved for drivers defined in the [CTRL] sections of the audio configuration file. The numbers are reserved based on the order that the [CTRL] sections appear in the file. Additionally, the unit key in the [CTRL] section can be used to define a specific card number unrelated to the section's position in the file. This ensures that card numbers that are assigned at io-audio startup via the configuration file are deterministic regardless of further runtime loading/unloading of driver DLLs.
If a device is defined in the audio configuration file, its card number is assigned for the lifetime that io-audio runs. If you unload a card (e.g., umount /dev/snd/controlCx) and its number is reserved for a driver in the configuration file, the card number is not released back to the system. However, if a driver is dynamically loaded at runtime (and does not match a number reserved in the configuration file), unloading the card releases its number.
# mount -Tio-audio -ounit=2 deva-ctrl-mcasp.so
#mount -Tio-audio -oconf=mydriver.conf deva-ctrl-mcasp.so #mount -Tio-audio -oconf=mydriver.conf,unit=2 deva-ctrl-mcasp.soThe following driver-specific configuration file (mydriver.conf) has two entries that match the previous mount commands. The first [CTRL] section is used when you don't provide the unit option. If you do provide the option (as in the second example [CTRL] section), then the second [CTRL] section is used:
[CTRL] name=mcasp options=clk_mode=master [CTRL] name=mcasp unit=2 options=clk_mode=master
[CTRL] name=mcasp options=unit=2,clk_mode=masterThe following section with the explicit unit key is correct:
[CTRL] name=mcasp unit=2 options=clk_mode=master
# mount -T io-audio all
# mount -T io-audio acoustics
Unmounting a module
To unload a module, reference its published device name in the unmount command. For example:
umount /dev/snd/controlC0
The audio configuration file is optional, but is easier to use and more flexible than using the corresponding command-line options. In order to load an audio configuration file, specify the -c configuration_file_path option when you start io-audio. Global and memory options provided on the command line override the corresponding audio configuration file settings. If you specify deva-* shared objects both on the command line (via the -d option) and in the audio configuration file (via [CTRL] sections), each is treated as an individual and independent request to load or mount a driver.
Options from the [CTRL] section take precedence over options defined (and supported) in both the [GLOBAL] and [CTRL] sections. Global options provided via the command line (-o) are also overridden by options defined in the [CTRL] section in the audio configuration file. In general, we recommend that you limit the use of [GLOBAL] options and use the options under the [CTRL] section.
The standard location for this file is /etc/system/config/audio/io-audio_platformVariant.conf. If an error occurs while parsing the audio configuration file, io-audio won't start.
When you're using indexed PCM mixer or input splitter keys, the index is that of the parent PCM device that the mixer/splitter is being created for, not the index of the mixer/splitter PCM device that's being created.
The general syntax is as follows:
options=clkmgmt=0 &options=,dindex=1,play_name=defaultp &options=,dindex=2,cap_name=defaultc
options=clkmgmt=0,dindex=1,play_name=defaultp,dindex=2,cap_name=defaultc
Key | Default | Description |
---|---|---|
config_write_delay | -1 | The time in seconds after last change before soundcard settings are written to disk. The default value of -1 prevents the settings from ever being written. |
max_dma_buf_size | 256 | The maximum DMA buffer size, in KB. The minimum value for this parameter is 16 KB. |
disable_sw_mixer | 0 | Disable the PCM Software mixer (0 / 1). If 0, a PCM Software mixer is instantiated. 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. |
sw_mixer_ms | See description |
Set the PCM mixer fragment size, in milliseconds, for any card that doesn't specify the fragment size in the [CTRL] section or at the command line. This key overrides sw_mixer_samples. If you don't specify this key, sw_mixer_samples sets the PCM mixer fragment size. Note:
If any acoustic AFMs play back through a PCM software mixer, the following
configuration is required:
|
sw_mixer_samples | 1024 | 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. |
sw_mixer_sample_size | None | Specifies one of the following sample sizes for the PCM software mixer to
use:
If you do not use this key, the sample size is determined by the sample size the underlying hardware uses:
|
sw_mixer_rate | FAHQ | The method for setting the PCM software mixer sample rate. If the underlying hardware
supports multiple sample rates, the specified method determines which rate the
mixer uses. The selection process may make use of two possible rate "families" — a
set of rates that are multiples of 4 kHz and a set that are multiples of 11.025
kHz:
|
intr_thread_prio | 50 | The interrupt service thread's priority. The value is discarded if it isn't within the valid range of round-robin priorities. |
data_thread_prio | 25 | The minimum thread priority. The value is discarded if it isn't within the valid range of round-robin priorities. |
verbosity | 0 | The logging verbosity level.
using a Bit-masked log level:
|
pool_name | None | The name of a shared memory or typed memory object to map and use as a DMA memory pool.
The object must be physically contiguous memory.
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. |
pool_size | The size of the DMA memory pool to create, in KB. If no pool_name is provided, then an anonymous object is created to the pool_size specified. If the pool_name refers to typed memory, then you must also provide the pool_size. | |
pcmPreferredp | None | The name of the preferred playback device; that is, the device
snd_pcm_open_preferred() opens when
SND_PCM_OPEN_PLAYBACK is specified. For more information on how io-audio sets the preferred device, see Preferred device selection. |
pcmPreferredc | None | The name of preferred capture device; that is, the device opened with
snd_pcm_open_preferred() when
SND_PCM_OPEN_CAPTURE is specified. For more information on how io-audio sets the preferred device, see Preferred device selection. |
smmu | 0 | Use the QNX System Memory Management Unit Manager (SMMUMAN). If the smmuman service is running, specify 1 or on to configure io-audio to interface with smmuman to manage ("cage") access for non-CPU initiated reads and writes (i.e., DMA devices). Otherwise, specify 0 or off. |
[AUDIOMGMT]sections include the following keys:
Key | Default | Description |
---|---|---|
audiomgmt_id | None | (Required) Name of the audio concurrency management context or output. PCM devices can attach to the audio concurrency management context or output using this identifier. |
policy_conf | None | (Required) Path to the audio policy configuration file. The audio policy file contains audio concurrency management policies such as volume ramping and audio ducking for the system. For more information, see the Audio Concurrency Management chapter in the QNX Neutrino Audio Developer's Guide. |
unit | 0 | Card number used in the filesystem mount. The card number (unit) must be
unique. Note: This key is used by the first [AUDIOMGT] section.
Subsequent unit keys in other [AUDIOMGMT] sections are
ignored.
|
Key | Default | Description |
---|---|---|
name | None | The name of the sound card HW interface DLL. Required; omit the deva-ctrl- prefix and the .so suffix when you're specifying the driver name. |
options | None | Driver-specific options as documented for the sound card DLL. As part of
the card options, the following options can be specified for each PCM device:
options=dindex=2,cap_name=devC,play_name=devP |
unit | -1 | Card or unit numbers must be unique, non-negative number. This value is used
as the card number to mount the card on the filesystem. 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. |
input_splitter_enable | 0 | Enable or disable the input splitter on PCM devices that have a capture
device (0 / 1). An optional _index suffix can be appended to
the key to target the option at a specific PCM device. 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). Note: If you use the QNX Acoustic Management Platform 3.0, you must specify this
parameter if multiple AFMs are capturing simultaneously from the same capture
device on this sound card.
|
input_splitter_ms | None | The input splitter fragment size, in milliseconds. No default. To specify the fragment size for a specific PCM device, add an underscore and the device name to the key. For example, input_splitter_ms_0.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 (in the [CTRL] section) or sw_mixer_ms (in the [GLOBAL] or [CTRL] section) 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. |
input_splitter_samples | None |
The input splitter fragment size, in samples per voice. The minimum is 64 bytes, and the maximum 512 KB. Do not use if any AFMs are configured to capture through the PCM input splitter on this card. If you set input_splitter_ms, it has priority over this key. If you are streaming data from the input splitter to the software mixer, the fragment sizes need to be the same. This key is set in the [CTRL] section only. To specify the fragment size for a specific PCM input splitter, append an underscore and the device name to the key. |
input_splitter_hw_sample_size | None | If the underlying capture hardware supports multiple data sizes (formats),
this configuration key allows you to lock the input splitter format to a specific
sample size (e.g., 8 (8-bit), 16 (16-bit), and so on). If you do not use this key, the highest-quality format supported by the underlying hardware is selected by default. To specify the sample size for a specific PCM software mixer, append an underscore and the device name to the key. |
input_splitter_rate | FAHQ | 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. The selection process may make use of two possible rate "families" — a set of rates that are multiples of 4 kHz and a set that are multiples of 11.025 kHz:
To specify the method for a specific PCM device, add an underscore and the device name to the key. For example, input_splitter_rate_0. |
disable_sw_mixer | 0 | Disable the sw mixer (0 / 1). An optional _index suffix
can be appended to the key to target the option at a specific PCM device. 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. Note: If you use the QNX Acoustic Management Platform 3.0, this parameter must be
0 if AFMs are playing simultaneously to the same playback device on this sound
card.
|
sw_mixer_ms | See description |
Set the PCM mixer fragment size, in milliseconds, for this card. This key overrides sw_mixer_samples and any fragment size specified in the [GLOBAL] section or at the command line. If you don't specify this key, the fragment size specified in the [GLOBAL] section or at the command line is used; if no fragment size is specified in [GLOBAL] or at the command line, sw_mixer_samples sets the PCM mixer fragment size. Note:
If any acoustic AFMs play back through a PCM software mixer, the following
configuration is required:
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. |
sw_mixer_samples | 1024 | The mixer fragment size, in samples per voice. An optional
_index suffix can be appended to the key to target the option
at a specific PCM device. This is the deprecated method of specifying the mixer
fragment size (overridden by sw_mixer_ms if specified). The minimum is 64
bytes; the maximum 512 KB. Do not use if any AFMs are configured to capture through the PCM input splitter on this card. This parameter overrides the corresponding option from the [GLOBAL] section or the command line for this card. |
sw_mixer_rate | FAHQ | The method for setting the PCM software mixer sample rate. If the underlying
hardware supports multiple sample rates, the specified method determines which
rate the mixer uses. The selection process may make use of two possible rate
"families" — a set of rates that are multiples of 4 kHz and a set that are
multiples of 11.025 kHz:
This parameter overrides the corresponding option from the [GLOBAL] section or the command line for this card. An optional _index suffix can be appended to the key to target the option at a specific PCM device. |
sw_mixer_sample_size | None | Specifies one of the following sample sizes for the PCM software mixer to
use:
If you do not use this key, the sample size is determined by the sample size the underlying hardware uses.
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. This parameter overrides the corresponding key from the [GLOBAL] section or the command line for this card. |
sw_mixer_cap_name | pcmCxDyp_ref | The symbolic name for PCM software mixer's reference devices, where
x is the numeric index of the card and y is the numeric index of the device. An optional
_index suffix can be appended to the key to target the option
at a specific PCM device. If you create multiple PCM mixers, and you provide this
key without an index suffix, then a name collision occurs, and the PCM mixer
creation fails. Note: If you have QNX Acoustics Management Platform 3.0, you can
use the symbolic name instead of the PCM device name in the audio configuration
file to simplify AFM device assignments.
|
sw_mixer_limiter | 0 | Enable the PCM software mixer limiter. An optional _index
suffix can be appended to the key to target the option at a specific PCM
device. 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. |
sw_mixer_max_references | 0 | The maximum number of media references available for capture from the PCM
software mixer (maximum is 10). This value should be equal to or greater than the total number of media references required by all configured AFMs. A reference is required for each pcm_media_capture device (used with ICC and QAV AFMs) that points to the PCM mixer reference capture device (typically specified by sw_mixer_cap_name). An optional _index suffix can be appended to the key to target the option at a specific PCM device. |
audiomgmt_id | None | Name of the audio concurrency management context/output to be used. An optional _index suffix can be appended to the key to target the option at a specific PCM device. |
defer_mount | 0 | Defer mounting of the DLL until a later time. 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. |
Note: You use the following keys only if you have QNX Acoustic Management Platform
3.0.
|
||
input_splitter_tap | 0 | Enable an Acoustic Control Server (ACS) tap point in the input splitter. Specify 1 to create a tap point that LiveAMP can stream or inject audio to using the ACS protocol, or 0 for no tap point. Optionally, append an _index suffix to the key to target a specific PCM device. |
sw_mixer_tap | 0 | Enable an Acoustic Control Server (ACS) audio tap point in the PCM software mixer. Specify 1 to create a tap point that LiveAMP can stream or inject audio to using the ACS protocol, or 0 for no tap point. Optionally, append an _index suffix to the key to target a specific PCM device.Note: This option is intended for systems where SFO or SPM are not being
used and you still want access to the PCM mixer audio in LiveAMP. SPM and SFO
have internal audio tap points that provide access to the PCM mixer audio; if
they are enabled, you don't need to specify sw_mixer_tap.
|
sw_mixer_spm | 0 | Enable the Speaker Management (SPM) module in the PCM software
mixer. Specify 1 to create a module, or 0 for no module. Optionally, append an _index suffix to the key to target a specific PCM device. |
sw_mixer_sfo | 0 | Enable the Sound Field Optimization (SFO) module in the PCM software
mixer. Specify 1 to create a module, or 0 for no module. Optionally, append an _index suffix to the key to target a specific PCM device. |
spm_qcf | None | Specify the path to the SPM tuning file. The recommended directory is
/etc/acoustic. For example:spm_qcf=/etc/acoustic/spm.qcf Optionally, append an _index suffix to the key to target a specific PCM device. |
sfo_qcf | None | Specify the path to the SFO tuning file. The recommended directory is /etc/acoustic. For example: sfo_qcf=/etc/acoustic/sfo.qcf Optionally, append an _index suffix to the key to target a specific PCM device. |
spm_dataset_qcf_name | None | Specify the path to the SPM dataset qcf file to apply when the dataset name is set by snd_pcm_load_apx_dataset(). |
sfo_dataset_qcf_name | None | Specify the path to the SFO dataset qcf file to apply when the dataset name is set by snd_pcm_load_apx_dataset(). |
sw_mixer_spm_chmap | The hardware channel mapping | Specify channel mapping names within the mixer before SPM is
applied. Separate the channel name strings with colons (similar to how drivers overwrite their channel map; for example, fl:fr). This mapping allows you to associate the correct channel names with channels within the mixer but does not change the mapping set in the SPM tuning file. It is only used when SPM is enabled. The following channel name strings are valid: fl (front left), fr (front right), rl (rear left), rr (rear right), fc (front center), lfe (low-frequency effects (LFE)), sl (side left), sr (side right), rc (rear center), flc (front left center), frc (front right center), rlc (rear left center), rrc (rear right center), flw (front left wide), frw (front right wide), flh (front left high), fch (front center high), frh (front right high), tc (top center), tfl (top front left), tfr (top front right), tfc (top front center), trl (top rear left), trr (top rear right), trc (top rear center), tflc (top front left center), tfrc (top front right center), tsl (top side left), tsr (top side right), llfe (left LFE), rlfe (right LFE), bc (bottom center), blc (bottom left center), brc (bottom right center). Optionally, append an _index suffix to the key to target a specific PCM device. |
sfo_audio_type | None | Specify the name of the audio type used to calculate an internal user
volume. SFO calculates the internal user volume using the software mixer volume, the audio type volume, the codec volume, and the external volume. You can use snd_pcm_set_apx_user_volume() to override the calculated value. Optionally, append an _index suffix to the key to target a specific PCM device. |
csa_audio_type | None | Specify the name of the audio type used to create the CSA reference. If the PCM mixer reference capture device (typically specified by sw_mixer_cap_name) is used as the pcm_ref_capture device for the CSA AFM, the CSA reference is a mix of all the subchannels whose audio type matches this key. If this key is not specified (and the pcm_ref_capture device for the CSA AFM points at the PCM mixer reference capture device), the CSA reference is not created and the CSA AFM will not start. Optionally, append an _index suffix to the key to target a specific PCM device. |
csa_limiter | 1 | Enable the PCM software mixer limiter in the CSA mix audio path. Specify 1 to enable a 1 ms soft look-ahead limiter on the CSA mixing path through the PCM software mixer. Specify 0 to add the CSA mix strictly and make it hard-clamped. Optionally, append an _index suffix to the key to target a specific PCM device. |
afm_mixer_enable | 0 | Enable mixing of the AFM outputs separately from the media streams in the PCM
software mixer. Specify 1 to enable the AFM mixer or 0 to disable it. This value is discarded if there are no configured AFMs. An optional _index suffix can be appended to the key to target the option at a specific PCM device. |
afm_mixer_max_references | 0 | The maximum number of AFM references available for capture from the AFM mixer
(maximum is 10). An optional _index suffix can be appended to the key to target the option at a specific PCM device. This value is discarded unless afm_mixer_enable is 1 and there are configured AFMs.This value should be equal to or greater than the total number of AFM references (non-media) required by all configured AFMs. A reference is required for each pcm_ref_capture device that points to the PCM mixer reference capture device (typically specified by sw_mixer_cap_name). A minimum of one AFM reference is required for the ICC AFM if not using loopback as the reference. Using a reference with the QAV AFM is optional and depends on whether pcm_ref_capture is specified. |
afm_mixer_atten_disable | 0 | Disable AFM mixer attenuation. 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 QAV 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. |
max_concurrent_afms | 0 | The maximum number of AFMs that are expected to be concurrently active. This
value is discarded unless afm_mixer_enable is set to 1 and there are AFMs
that are configured. 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. |
clock_domain | 0 | Identify a PCM device that is in a separate clock domain. When an AFM is used to bridge hardware devices in separate clock domains, an asynchronous sample rate converter (ASRC) is created to translate between the domains. Optionally, append an _index suffix to the key to target a specific PCM device. |
Key | Default | Description |
---|---|---|
name | None | The name of the DLL, which is usually an AMP (Acoustic Management Platform) functional
module (AFM); this parameter is required and must be one of the following modular
libraries:
|
unit | -1 | Card number used in the filesystem mount. Card numbers must be unique. If this key isn't provided or if the value is a negative number, the first available, unique number after zero is generated for the AFM. |
audio_type | None | Audio type that's used by the system to handle audio ducking. Not used with the CSA AFM. |
pcm_local_capture | None | The name of the local capture device. Required for all AFMS. |
pcm_local_playback | None | The name of the local playback device. Required for all AFMs except CSA. |
pcm_remote_capture | None | The name of the remote capture device. Required for the QAV AFM. |
pcm_remote_playback | None | The name of the remote playback device. Required for the QAV AFM. |
pcm_ref_capture | None | The name of the acoustic reference capture device. For ICC, if the value is loopback rather than a PCM device name, the ICC output is routed internally to the ICC reference input. Required for the ICC and CSA AFMs; optional for the QAV AFM. |
pcm_media_capture | None | The name of the media reference capture device. Optional key for ICC and QAV AFMs. |
map_mic_X | A 1:1 mapping of the capture voice to library input (e.g., map_mic_0=0, map_mic_1=1, etc.) | Map the capture voice to the acoustic processing microphone input X.
The Bridge AFM is a special case, since this is a mapping from capture voice to playback voice (rather than
library input), so there are exceptions to the above:
|
map_spk_X | A 1:1 mapping of library output to playback voice (e.g., map_spk_0=0, map_spk_1=1, etc.). | Map acoustic processing speaker output X to playback voice(s); you can
specify multiple voices, delimited by commas.
Note: When acoustic library volume and mute control is enabled
(enable_vol_ctl) for an AFM, these keys are used to instantiate the
corresponding mixer controls, so they are parsed when the AFM is loaded when
io-audio starts. If there are errors,
io-audio won't start. These keys are also parsed each time
the AFM is started. If there are errors, the AFM won't start.
|
map_ref_X | A 1:1 mapping of acoustic reference voice to library reference input (e.g., map_ref_0=0, map_ref_1=1, etc.) | Map the acoustic reference voice to acoustic processing reference input
X. CSA, ICC, and QAV AFMs only.
These keys are parsed when the AFM is started; if there are errors, the AFM won't start. |
map_media_ref_X | A 1:1 mapping of media reference voice to library external reference input (e.g., map_media_ref_0=0, map_media_ref_1=1, etc.) | Map a media reference voice to acoustic processing external reference input
X. ICC and QAV AFMs only.
These keys are parsed when the AFM is started; if there are errors, the AFM won't start. |
ap_dlname | The acoustic processing DLL name corresponding to the AFM type:
|
Acoustic processing library .so name |
ap_qcf_default | None | The path to the acoustic tuning file when in the default audio mode. QNX recommends that
you store tuning files in the directory /etc/acoustic (for
example, ap_qcf_default=/etc/acoustic/handsfree.qcf). The default audio mode is used when a mode is not set using the libasound library snd_afm_set_audio_mode() function. |
ap_qcf_mode | None | The path to the acoustic tuning file to use when in the specified
mode. 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 8 kHZ (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. |
ap_samp_play_dir_default | None | The path to the acoustic sample player directory. ASD AFM only. |
ap_samp_play_dir_mode | None |
The path to the acoustic sample player directory when the mode string is set by snd_afm_set_audio_mode(). ASD AFM only. |
ap_gran_syn_dir_default | None | The path to the acoustic granular synthesis files to use when in the default mode. ASD AFM
only. The default audio mode is used when a mode is not set using the libasound library snd_afm_set_audio_mode() function. |
ap_gran_syn_dir_mode | None | The path to the acoustic granular synthesis files to use when in the specified mode. ASD
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. |
ap_calib_default | None | The path to the acoustic calibration file (ASD and CSA AFMs only) or beamformer calibration
directory (QAV AFM only) to use when in the default mode. The default audio mode is used when a mode is not set using the libasound library snd_afm_set_audio_mode() function. |
ap_calib_mode | None | The path to the acoustic calibration file (ASD and CSA AFMs only) or beamformer calibration
directory (QAV AFM only) to use when in the specified
mode. 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. |
ap_dataset_qcf_name | None | The path to the dataset QCF to apply when the dataset name string set with snd_afm_load_ap_dataset(). |
ap_enable_diagnostics | 0 | Enable acoustic library diagnostics; ICC and QAV AFMs only. 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 | 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. Note: This option can result in extreme feedback in the case of the ICC and Bridge
AFMs.
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pcm_play_name | voicep | The symbolic name for the PCM interface published for playback. QAV AFM only. |
pcm_cap_name | voicec | The symbolic name for the PCM interface published for capture. QAV AFM only. |
pcm_disable | 0 | Disable PCM interface (0 / 1). QAV AFM only. If 0, the voice PCM interface is enabled; if 1, it's disabled. |
sw_mixer_route | afm | The route to use when mixing playback audio (main or afm) 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 Concurrency Management chapter in the QNX Neutrino Audio Developer's Guide. |
ms_frag_period | sw_mixer_ms (when the local playback device is a PCM Software Mixer device) | The fragment size, in milliseconds, to use for audio hardware interfacing. When the local
playback device is a PCM software mixer, the value of this key:
This key must be explicitly specified:
For acoustic AFMs, the value of ms_frag_period should be a factor of the acoustic library fragment period. If the hardware requires a sw_mixer_ms value (and, implicity, ms_frag_period) that is larger than the acoustic library fragment size, you can set ms_frag_period to a multiple of the acoustic library fragment size. However, using a fragment size that is larger than the acoustic library fragment size will have an impact on the AFMs performance. |
slip_guard_frags | 1 | Specify how many additional fragments to add in the playback path to ensure consistent
startup timing. If set to 0 (zero), audio is buffered on the playback path by either the maximum of the acoustic library fragment period or ms_frag_period, whichever is larger. Each additional slip guard fragment you specify adds an additional period of latency as specified by ms_frag_period. |
enable_vol_ctl | 0 | Enable acoustic library volume and mute control; ICC and QAV AFMs only. 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 QAV Gain) to adjust the volume is published. Note: If both enable_vol_ctrl and enable_mute_ctrl_only are set
to 1, enable_vol_ctl takes precedence.
|
enable_mute_ctl_only | 0 | Only enable acoustic library mute control; ICC and QAV AFMs only. 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. Note: If both enable_vol_ctrl and enable_mute_ctrl_only are set
to 1, enable_vol_ctl takes precedence.
|
max_vol_db_X |
|
The maximum output fixed gain setting for speaker output X corresponding
to 100% volume set with mix_ctl. This key is ignored if
enable_vol_ctl is set to 0. ICC and QAV AFMs only. This parameter must be >= min_vol_db. If this value is set beyond the ICC/QAV library maximum of 40 dB, a warning is logged, but the value is accepted. |
min_vol_db_X |
|
The minimum output fixed gain setting for speaker output X corresponding
to 0% volume set with mix_ctl. This key is ignored if
enable_vol_ctl is set to 0. ICC and QAV AFMs only; This parameter must be <= max_vol_db. If this value is set beyond the ICC/QAV library minimum of -30 dB, a warning is logged, but the value is accepted. |
init_vol_X |
|
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 QAV AFMs only. The value is clamped between 0 and 100 if out of range. |
start_on_boot | 0 | If 1, start the AFM when io-audio starts, after all cards and AFMs have been successfully loaded; if 0, the AFM is idle after io-audio starts. |
master | capture | Master channel for bridge audio transfer; Bridge AFM only. 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 the 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. |
resync_on_xrun | 0 | Force audio resynchronization when an overrun or an underrun occurs; Bridge AFM only.
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. |
artifact_protection | 0 | Clear the capture buffer before advancing internal references (pointers) to it.
Bridge AFM only.
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. |
cardname_override | None | Not used. |
sym_name | Same as the AFM name specified by name | Symbolic name for the AFM that you can use with snd_afm_open_name()
and utilities such as afm_ctl to access the AFM. This key can be used to differentiate between AFMs if your configuration has multiple AFMs of the same type. For example, if there are multiple bridges, set the symbolic names to bridge1, bridge2, and so on, and access them with afm_ctl -a bridge1 or afm_ctl -a bridge2. |
pcm_mode_remote_frags | 1 | Number of remote fragments used for QAV PCM mode, which are used to compensate for
variable-interrupt timing. 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. |
defer_mount | 0 | Defer mounting of the DLL until a later time. 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. |
local_rate_default | 0 | Sample rate to use when you open pcm_local_capture and
pcm_local_playback when in the default audio mode. If you do not
specify this key, the highest sample rate that the underlying hardware supports is
used. 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.
|
local_rate_mode | 0 | Sample rate to use when you open pcm_local_capture and
pcm_local_playback devices when in the specified audio mode. If you do
not specify this key, the highest sample rate that the underlying hardware
supports is used. 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 8 kHZ (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. Applies to acoustic AFMs only.
|
remote_rate_default | 0 | Sample rate to use when you open pcm_remote_capture and
pcm_remote_playback when in the default audio mode. If you do not
specify this key, the highest sample rate that the underlying hardware supports is
used. 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.
|
remote_rate_mode | 0 | Sample rate to use when you open pcm_remote_capture and
pcm_remote_playback devices when in the specified audio mode. For information on mode, see local_rate_mode. Applies to the QAV AFM only.
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ref_rate_default | 0 | Sample rate to use when you open any reference device (e.g.,
pcm_ref_capture) when in the default audio mode. 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.
|
ref_rate_mode | 0 | Sample rate to use when you open any reference device (e.g.,
pcm_ref_capture) when in the specified audio mode. For information on mode, see local_rate_mode. Applies to acoustic AFMs only.
|
ep_notify_default | 0 | Enable endpoint notifications when the audio is considered "voiced" or
"non-voiced" in the default audio mode. Clients can read notifications using
snd_ctrl_read(). 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. For more information, see the QNX Acoustics for Voice Developer’s Guide. |
ep_notify_mode | 0 | Enable endpoint notifications when the audio is considered "voiced" or "non-voiced" in the specified audio mode.
For information on mode, see local_rate_mode. Applies to the QAV AFM only. For more information, see the QNX Acoustics for Voice Developer’s Guide. |
ep_min_voiced | 300 | Specify the minimum duration of a voiced event that produces an endpoint notification, in milliseconds. A voiced event that is shorter than this value does not produce a notification. Applies to the QAV AFM only. For more information, see the QNX Acoustics for Voice Developer’s Guide. |
ep_min_non_voiced | 200 | Specify the minimum duration of a non-voiced event that produces an end-of-speech notification, in milliseconds. If the time between two voiced events is shorter than this value, they are handled as one event. Applies to the QAV AFM only. For more information, see the QNX Acoustics for Voice Developer’s Guide. |
buffer_ms | 5000 | Specify the maximum time to buffer audio, in milliseconds. Applies to the TAADIS AFM only. |
Key | Default | Description |
---|---|---|
name | None | The name of the link. This is a required key.
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. |
unit | -1 | Card number used in the filesystem mount. Card numbers must be unique. If this key isn't provided or if the value is a negative number, the first available, unique number after zero is generated. |
enable_acs | None | (Required) Enable the ACS link.
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. |
server_port | 4000 | The ACS TCP/IP server port. |
ms_frag_period | None | Not used. |
tap_split_cap | None | Not used. |
tap_media_ref | None | Not used. |
defer_mount | 0 | Defer mounting of the DLL until a later time. 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. After the DLL is mounted, it cannot be unmounted. |
[global] verbosity=3 sw_mixer_ms=2 ######################################## # Audio Concurrency Management Output Context ######################################## [audiomgmt] audiomgmt_id=Zone1 policy_conf=/etc/system/config/audio/audio_policy_zone1.conf ######################################## # On-board audio ######################################## # This card's PCM software mixer and input splitter will be the preferred devices [ctrl] name= your_driver_name options= your_driver_specific_options input_splitter_enable=1 sw_mixer_ms=16 audiomgmt_id=Zone1
When you use snd_pcm_open_preferred() to open an audio interface, the function attempts to open the device configured as the preferred device.
The io-audio service searches for the preferred device to use in the following system resources and order:
pcmPreferredp card_name device_number pcmPreferredc card_name device_number
The io-audio service compares available card names to the preferred card name in the following order and selects the first card name that matches:
(The card name is set by the driver code and can be queried via card_get_longname(), card_get_name(), and so on.)
Whenever a new device is mounted or unmounted, io-audio repeats the search and updates the preferred device if it finds a better match.
mount -T io-audio -o conf=mydriver.conf deva-ctrl-mcasp.soHere's the example mydriver.conf file that corresponds to the above command:
[CTRL] name=mcasp options=clk_mode=master
Start io-audio and load the audio configuration from a file:
io-audio -c /etc/system/config/audio/io-audio.conf
Load the AudioPCI driver, specifying the size of the DMA memory pool as 512 KB:
io-audio -vv -m pool_size=512 -d audiopci
Lock the sample rate to 8 kHz (if the hardware supports 8 kHz natively):
io-audio -o sw_mixer_rate=L8000 -d my_audio_driver
Start an audio driver, specifying the card options:
io-audio -d my_audio_driver cap_name=capture,play_name=playback
The /dev/snd directory will look something like this:
# ls -l /dev/snd total 0 lrw-rw-rw- 1 root root 0 May 31 11:11 capture -> pcmC0D0c -rw-rw-rw- 1 root root 0 May 31 11:11 controlC0 -rw-rw-rw- 1 root root 0 May 31 11:11 mixerC0D0 -rw-rw-rw- 1 root root 0 May 31 11:11 pcmC0D0c -rw-rw-rw- 1 root root 0 May 31 11:11 pcmC0D0p lrw-rw-rw- 1 root root 0 May 31 11:11 pcmPreferredc -> pcmC0D0c lrw-rw-rw- 1 root root 0 May 31 11:11 pcmPreferredp -> pcmC0D0p lrw-rw-rw- 1 root root 0 May 31 11:11 playback -> pcmC0D0p
In addition to the deva-ctrl-*, deva-afm-*, and dev-acs-link drivers, the io-audio command can load the following shared objects: