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[ALSA] usb-audio - Fix audiophile-USB quirk for little-endian
Audiophile-usb fix (corrects little-endianness in 16bit modes, resets interfaces at device initialization, and updates the documentation). Signed-off-by: Thibault Le Meur <Thibault.LeMeur@supelec.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This commit is contained in:
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2 changed files with 157 additions and 78 deletions
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@ -1,4 +1,4 @@
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Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3
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Guide to using M-Audio Audiophile USB with ALSA and Jack v1.4
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========================================================
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Thibault Le Meur <Thibault.LeMeur@supelec.fr>
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@ -6,8 +6,17 @@
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This document is a guide to using the M-Audio Audiophile USB (tm) device with
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ALSA and JACK.
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History
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=======
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* v1.4 - Thibault Le Meur (2007-07-11)
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- Added Low Endianness nature of 16bits-modes
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found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
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- Modifying document structure
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1 - Audiophile USB Specs and correct usage
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==========================================
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This part is a reminder of important facts about the functions and limitations
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of the device.
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@ -25,18 +34,18 @@ The device has 4 audio interfaces, and 2 MIDI ports:
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The internal DAC/ADC has the following characteristics:
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* sample depth of 16 or 24 bits
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* sample rate from 8kHz to 96kHz
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* Two ports can't use different sample depths at the same time. Moreover, the
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Audiophile USB documentation gives the following Warning: "Please exit any
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audio application running before switching between bit depths"
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* Two interfaces can't use different sample depths at the same time.
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Moreover, the Audiophile USB documentation gives the following Warning:
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"Please exit any audio application running before switching between bit depths"
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Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be
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activated at the same time depending on the audio mode selected:
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* 16-bit/48kHz ==> 4 channels in/4 channels out
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* 16-bit/48kHz ==> 4 channels in + 4 channels out
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- Ai+Ao+Di+Do
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* 24-bit/48kHz ==> 4 channels in/2 channels out,
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or 2 channels in/4 channels out
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* 24-bit/48kHz ==> 4 channels in + 2 channels out,
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or 2 channels in + 4 channels out
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- Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
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* 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only)
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* 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
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- Ai or Ao or Di or Do
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Important facts about the Digital interface:
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@ -52,44 +61,53 @@ source is connected
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synchronization error (for instance sound played at an odd sample rate)
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2 - Audiophile USB support in ALSA
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==================================
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2 - Audiophile USB MIDI support in ALSA
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=======================================
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2.1 - MIDI ports
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----------------
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The Audiophile USB MIDI ports will be automatically supported once the
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The Audiophile USB MIDI ports will be automatically supported once the
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following modules have been loaded:
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* snd-usb-audio
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* snd-seq-midi
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No additional setting is required.
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2.2 - Audio ports
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-----------------
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3 - Audiophile USB Audio support in ALSA
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========================================
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Audio functions of the Audiophile USB device are handled by the snd-usb-audio
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module. This module can work in a default mode (without any device-specific
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parameter), or in an "advanced" mode with the device-specific parameter called
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"device_setup".
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2.2.1 - Default Alsa driver mode
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3.1 - Default Alsa driver mode
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------------------------------
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The default behavior of the snd-usb-audio driver is to parse the device
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capabilities at startup and enable all functions inside the device (including
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all ports at any supported sample rates and sample depths). This approach
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has the advantage to let the driver easily switch from sample rates/depths
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automatically according to the need of the application claiming the device.
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The default behavior of the snd-usb-audio driver is to list the device
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capabilities at startup and activate the required mode when required
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by the applications: for instance if the user is recording in a
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24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
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the snd-usb-audio module will reconfigure the device on the fly.
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In this case the Audiophile ports are mapped to alsa pcm devices in the
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following way (I suppose the device's index is 1):
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This approach has the advantage to let the driver automatically switch from sample
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rates/depths automatically according to the user's needs. However, those who
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are using the device under windows know that this is not how the device is meant to
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work: under windows applications must be closed before using the m-audio control
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panel to switch the device working mode. Thus as we'll see in next section, this
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Default Alsa driver mode can lead to device misconfigurations.
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Let's get back to the Default Alsa driver mode for now. In this case the
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Audiophile interfaces are mapped to alsa pcm devices in the following
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way (I suppose the device's index is 1):
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* hw:1,0 is Ao in playback and Di in capture
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* hw:1,1 is Do in playback and Ai in capture
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* hw:1,2 is Do in AC3/DTS passthrough mode
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You must note as well that the device uses Big Endian byte encoding so that
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supported audio format are S16_BE for 16-bit depth modes and S24_3BE for
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24-bits depth mode. One exception is the hw:1,2 port which is Little Endian
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compliant and thus uses S16_LE.
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In this mode, the device uses Big Endian byte-encoding so that
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supported audio format are S16_BE for 16-bit depth modes and S24_3BE for
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24-bits depth mode. One exception is the hw:1,2 port which is reported
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to be Little Endian compliant (supposedly supporting S16_LE) but processes
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in fact only S16_BE streams.
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Examples:
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* playing a S24_3BE encoded raw file to the Ao port
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@ -99,21 +117,23 @@ Examples:
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* playing a S16_BE encoded raw file to the Do port
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% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
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If you're happy with the default Alsa driver setup and don't experience any
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If you're happy with the default Alsa driver mode and don't experience any
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issue with this mode, then you can skip the following chapter.
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2.2.2 - Advanced module setup
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3.2 - Advanced module setup
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---------------------------
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Due to the hardware constraints described above, the device initialization made
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by the Alsa driver in default mode may result in a corrupted state of the
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device. For instance, a particularly annoying issue is that the sound captured
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from the Ai port sounds distorted (as if boosted with an excessive high volume
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gain).
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from the Ai interface sounds distorted (as if boosted with an excessive high
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volume gain).
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For people having this problem, the snd-usb-audio module has a new module
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parameter called "device_setup".
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parameter called "device_setup" (this parameter was introduced in kernel
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release 2.6.17)
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2.2.2.1 - Initializing the working mode of the Audiophile USB
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3.2.1 - Initializing the working mode of the Audiophile USB
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As far as the Audiophile USB device is concerned, this value let the user
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specify:
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@ -121,33 +141,57 @@ specify:
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* the sample rate
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* whether the Di port is used or not
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Here is a list of supported device_setup values for this device:
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* device_setup=0x00 (or omitted)
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- Alsa driver default mode
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- maintains backward compatibility with setups that do not use this
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parameter by not introducing any change
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- results sometimes in corrupted sound as described earlier
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When initialized with "device_setup=0x00", the snd-usb-audio module has
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the same behaviour as when the parameter is omitted (see paragraph "Default
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Alsa driver mode" above)
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Others modes are described in the following subsections.
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3.2.1.1 - 16-bit modes
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The two supported modes are:
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* device_setup=0x01
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- 16bits 48kHz mode with Di disabled
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- Ai,Ao,Do can be used at the same time
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- hw:1,0 is not available in capture mode
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- hw:1,2 is not available
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* device_setup=0x11
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- 16bits 48kHz mode with Di enabled
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- Ai,Ao,Di,Do can be used at the same time
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- hw:1,0 is available in capture mode
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- hw:1,2 is not available
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In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
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the devices where reported to be Big-Endian when in fact they were Little-Endian
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so that playing a file was a matter of using:
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% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
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where "test_S16_LE.raw" was in fact a little-endian sample file.
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Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
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these modes) a fix has been committed (expected in kernel 2.6.23) and
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Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
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using:
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% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
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3.2.1.2 - 24-bit modes
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The three supported modes are:
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* device_setup=0x09
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- 24bits 48kHz mode with Di disabled
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- Ai,Ao,Do can be used at the same time
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- hw:1,0 is not available in capture mode
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- hw:1,2 is not available
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* device_setup=0x19
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- 24bits 48kHz mode with Di enabled
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- 3 ports from {Ai,Ao,Di,Do} can be used at the same time
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- hw:1,0 is available in capture mode and an active digital source must be
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connected to Di
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- hw:1,2 is not available
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* device_setup=0x0D or 0x10
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- 24bits 96kHz mode
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- Di is enabled by default for this mode but does not need to be connected
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@ -155,34 +199,61 @@ Here is a list of supported device_setup values for this device:
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- Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
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- hw:1,0 is available in captured mode
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- hw:1,2 is not available
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In these modes the device is only Big-Endian compliant (see "Default Alsa driver
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mode" above for an aplay command example)
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3.2.1.3 - AC3 w/ DTS passthru mode
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This mode is untested, I have no AC3 compliant device to test it. I uses:
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* device_setup=0x03
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- 16bits 48kHz mode with only the Do port enabled
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- AC3 with DTS passthru (not tested)
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- Caution with this setup the Do port is mapped to the pcm device hw:1,0
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2.2.2.2 - Setting and switching configurations with the device_setup parameter
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3.2.2 - How to use the device_setup parameter
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----------------------------------------------
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The parameter can be given:
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* By manually probing the device (as root):
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# modprobe -r snd-usb-audio
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# modprobe snd-usb-audio index=1 device_setup=0x09
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* Or while configuring the modules options in your modules configuration file
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- For Fedora distributions, edit the /etc/modprobe.conf file:
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alias snd-card-1 snd-usb-audio
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options snd-usb-audio index=1 device_setup=0x09
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IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
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-------------------------------------------
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* You may need to _first_ initialize the module with the correct device_setup
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parameter and _only_after_ turn on the Audiophile USB device
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* This is especially true when switching the sample depth:
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CAUTION when initializaing the device
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-------------------------------------
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* Correct initialization on the device requires that device_setup is given to
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the module BEFORE the device is turned on. So, if you use the "manual probing"
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method described above, take care to power-on the device AFTER this initialization.
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* Failing to respect this will lead in a misconfiguration of the device. In this case
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turn off the device, unproble the snd-usb-audio module, then probe it again with
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correct device_setup parameter and then (and only then) turn on the device again.
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* If you've correctly initialized the device in a valid mode and then want to switch
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to another mode (possibly with another sample-depth), please use also the following
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procedure:
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- first turn off the device
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- de-register the snd-usb-audio module (modprobe -r)
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- change the device_setup parameter by changing the device_setup
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option in /etc/modprobe.conf
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- turn on the device
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* A workaround for this last issue has been applied to kernel 2.6.23, but it may not
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be enough to ensure the 'stability' of the device initialization.
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2.2.2.3 - Audiophile USB's device_setup structure
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3.2.3 - Technical details for hackers
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-------------------------------------
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This section is for hackers, wanting to understand details about the device
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internals and how Alsa supports it.
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3.2.3.1 - Audiophile USB's device_setup structure
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If you want to understand the device_setup magic numbers for the Audiophile
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USB, you need some very basic understanding of binary computation. However,
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@ -228,12 +299,12 @@ Caution:
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- choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
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only be able to use one at the same time
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2.2.3 - USB implementation details for this device
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3.2.3.2 - USB implementation details for this device
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You may safely skip this section if you're not interested in driver
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development.
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hacking.
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This section describes some internal aspects of the device and summarize the
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This section describes some internal aspects of the device and summarizes the
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data I got by usb-snooping the windows and Linux drivers.
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The M-Audio Audiophile USB has 7 USB Interfaces:
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@ -293,43 +364,45 @@ parse_audio_endpoints function uses a quirk called
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"audiophile_skip_setting_quirk" in order to prevent AltSettings not
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corresponding to device_setup from being registered in the driver.
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3 - Audiophile USB and Jack support
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4 - Audiophile USB and Jack support
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===================================
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This section deals with support of the Audiophile USB device in Jack.
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The main issue regarding this support is that the device is Big Endian
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compliant.
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3.1 - Using the plug alsa plugin
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--------------------------------
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There are 2 main potential issues when using Jackd with the device:
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* support for Big-Endian devices in 24-bit modes
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* support for 4-in / 4-out channels
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Jack doesn't directly support big endian devices. Thus, one way to have support
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for this device with Alsa is to use the Alsa "plug" converter.
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4.1 - Direct support in Jackd
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-----------------------------
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Jack supports big endian devices only in recent versions (thanks to
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Andreas Steinmetz for his first big-endian patch). I can't remember
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extacly when this support was released into jackd, let's just say that
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with jackd version 0.103.0 it's almost ok (just a small bug is affecting
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16bits Big-Endian devices, but since you've read carefully the above
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paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices
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are now Little Endians ;-) ).
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You can run jackd with the following command for playback with Ao and
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record with Ai:
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% jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
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4.2 - Using Alsa plughw
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-----------------------
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If you don't have a recent Jackd installed, you can downgrade to using
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the Alsa "plug" converter.
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For instance here is one way to run Jack with 2 playback channels on Ao and 2
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capture channels from Ai:
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% jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
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However you may see the following warning message:
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"You appear to be using the ALSA software "plug" layer, probably a result of
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using the "default" ALSA device. This is less efficient than it could be.
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Consider using a hardware device instead rather than using the plug layer."
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3.2 - Patching alsa to use direct pcm device
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--------------------------------------------
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A patch for Jack by Andreas Steinmetz adds support for Big Endian devices.
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However it has not been included in the CVS tree.
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You can find it at the following URL:
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http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
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atid=425939
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After having applied the patch you can run jackd with the following command
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line:
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% jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
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3.2 - Getting 2 input and/or output interfaces in Jack
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4.3 - Getting 2 input and/or output interfaces in Jack
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------------------------------------------------------
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As you can see, starting the Jack server this way will only enable 1 stereo
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|
@ -339,6 +412,7 @@ This is due to the following restrictions:
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* Jack can only open one capture device and one playback device at a time
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* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
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(and optionally hw:1,2)
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If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
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combine the Alsa devices into one logical "complex" device.
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|
@ -348,13 +422,11 @@ It is related to another device (ice1712) but can be adapted to suit
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the Audiophile USB.
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Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
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* patching Jack with the previously mentioned "Big Endian" patch
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* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
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* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
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* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
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* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
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* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
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file
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* start jackd with this device
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||||
I had no success in testing this for now, but this may be due to my OS
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configuration. If you have any success with this kind of setup, please
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drop me an email.
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I had no success in testing this for now, if you have any success with this kind
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of setup, please drop me an email.
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|
|
|
@ -2350,7 +2350,9 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *
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return 1;
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break;
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case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
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return 1;
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if (device_setup[chip->index] == 0x00 ||
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fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3)
|
||||
return 1;
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||||
}
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||||
return 0;
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}
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|
@ -3251,6 +3253,11 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
|
|||
static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
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int iface, int altno)
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{
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/* Reset ALL ifaces to 0 altsetting.
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* Call it for every possible altsetting of every interface.
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*/
|
||||
usb_set_interface(chip->dev, iface, 0);
|
||||
|
||||
if (device_setup[chip->index] & AUDIOPHILE_SET) {
|
||||
if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS)
|
||||
&& altno != 6)
|
||||
|
|
Loading…
Reference in a new issue