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[ALSA] Fixes audiophile usb analog capture with the new device_setup parameter
Modules: Documentation,USB generic driver The patch adds the 'device_setup' module parameter and a specific quirk to correctly initialize the audiophile usb device: this fixes the distorted sound bug on the Analog capture port. Backward compatibility is achieved by simply omitting the new parameter. Signed-off-by: Thibault LE MEUR <Thibault.LeMeur@supelec.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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@ -1411,6 +1411,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
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vid - Vendor ID for the device (optional)
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pid - Product ID for the device (optional)
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device_setup - Device specific magic number (optional)
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- Influence depends on the device
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- Default: 0x0000
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This module supports multiple devices, autoprobe and hotplugging.
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330
Documentation/sound/alsa/Audiophile-Usb.txt
Normal file
330
Documentation/sound/alsa/Audiophile-Usb.txt
Normal file
@ -0,0 +1,330 @@
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Guide to using M-Audio Audiophile USB with ALSA and Jack v1.1
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========================================================
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Thibault Le Meur <Thibault.LeMeur@supelec.fr>
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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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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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The device has 4 audio interfaces, and 2 MIDI ports:
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* Analog Stereo Input (Ai)
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* Analog Stereo Output (Ao)
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* Digital Stereo Input (Di)
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* Digital Stereo Output (Do)
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* Midi In (Mi)
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* Midi Out (Mo)
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The internal DAC/ADC has the following caracteristics:
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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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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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- 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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- 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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- Ai or Ao or Di or Do
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Important facts about the Digital interface:
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--------------------------------------------
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* The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough,
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though I haven't tested it under linux
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- Note that in this setup only the Do interface can be enabled
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* Apart from recording an audio digital stream, enabling the Di port is a way
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to syncrhonize the device to an external sample clock
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- As a consequence, the Di port must be enable only if an active Digital
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source is connected
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- Enabling Di when no digital source is connected can result in a
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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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2.1 - 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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following modules have been loaded:
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* snd-usb-audio
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* snd-seq
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* snd-seq-midi
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No additionnal setting is required.
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2.2 - Audio ports
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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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The default behaviour 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 sample rates and any sample depths supported). 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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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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* 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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Examples:
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* playing a S24_3BE encoded raw file to the Ao port
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% aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
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* recording a S24_3BE encoded raw file from the Ai port
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% arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw
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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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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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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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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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2.2.2.1 - Initializing the working mode of the Audiohile USB
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As far as the Audiohile USB device is concerned, this value let the user
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specify:
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* the sample depth
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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 decribed earlier
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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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* 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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to an active source
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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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* 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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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_ intialize 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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- first trun off the device
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- de-register the snd-usb-audio module
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- change the device_setup parameter (by either manually reprobing the module
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or changing modprobe.conf)
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- turn on the device
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2.2.2.3 - Setting and switching configurations with the device_setup parameter
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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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this is not required to use the parameter and you may skip thi section.
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The device_setup is one byte long and its structure is the following:
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+---+---+---+---+---+---+---+---+
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| b7| b6| b5| b4| b3| b2| b1| b0|
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+---+---+---+---+---+---+---+---+
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| 0 | 0 | 0 | Di|24B|96K|DTS|SET|
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+---+---+---+---+---+---+---+---+
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Where:
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* b0 is the "SET" bit
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- it MUST be set if device_setup is initialized
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* b1 is the "DTS" bit
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- it is set only for Digital output with DTS/AC3
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- this setup is not tested
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* b2 is the Rate selection flag
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- When set to "1" the rate range is 48.1-96kHz
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- Otherwise the sample rate range is 8-48kHz
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* b3 is the bit depth selection flag
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- When set to "1" samples are 24bits long
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- Otherwise they are 16bits long
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- Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits
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samples
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* b4 is the Digital input flag
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- When set to "1" the device assumes that an active digital source is
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connected
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- You shouldn't enable Di if no source is seen on the port (this leads to
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synchronization issues)
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- b4 is implied by b2 (since only one port is enabled at a time no synch
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error can occur)
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* b5 to b7 are reserved for future uses, and must be set to "0"
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- might become Ao, Do, Ai, for b7, b6, b4 respectively
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Caution:
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* there is no check on the value you will give to device_setup
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- for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since
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b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
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* Hardware constraints due to the USB bus limitation aren't checked
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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 - Technical Details for Audiophile Usb
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You may safely skip this section if you're not interrested in driver
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development.
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This section describes some internals aspect of the device and summarize 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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a "USB interface":
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* Usb Interface nb.0
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* Usb Interface nb.1
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- Audio Control function
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* Usb Interface nb.2
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- Analog Output
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* Usb Interface nb.3
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- Digital Output
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* Usb Interface nb.4
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- Analog Input
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* Usb Interface nb.5
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- Digital Input
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* Usb Interface nb.6
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- MIDI interface compliant with the MIDIMAN quirk
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Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
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* Interface 3 (Digital Out) has an extra Alset nb.6
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* Interface 5 (Digital In) does not have Alset nb.3 and 5
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Here is a short description of the AltSettings capabilities:
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* AltSettings 1 corresponds to
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- 24-bit depth, 48.1-96kHz sample mode
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- Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
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* AltSettings 2 corresponds to
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- 24-bit depth, 8-48kHz sample mode
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- Asynch capture and playback (Ao,Ai,Do,Di)
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* AltSettings 3 corresponds to
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- 24-bit depth, 8-48kHz sample mode
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- Synch capture (Ai) and Adaptive playback (Ao,Do)
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* AltSettings 4 corresponds to
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- 16-bit depth, 8-48kHz sample mode
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- Asynch capture and playback (Ao,Ai,Do,Di)
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* AltSettings 5 corresponds to
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- 16-bit depth, 8-48kHz sample mode
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- Synch capture (Ai) and Adaptive playback (Ao,Do)
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* AltSettings 6 corresponds to
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- 16-bit depth, 8-48kHz sample mode
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- Synch playback (Do), audio format type III IEC1937_AC-3
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In order to ensure a correct intialization of the device, the driver
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_must_know_ how the device will be used:
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* if DTS is choosen, only Interface 2 with AltSet nb.6 must be
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registered
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* if 96KHz only AltSets nb.1 of each interface must be selected
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* if samples are using 24bits/48KHz then AltSet 2 must me used if
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Digital input is connected, and only AltSet nb.3 if Digital input
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is not connected
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* if samples are using 16bits/48KHz then AltSet 4 must me used if
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Digital input is connected, and only AltSet nb.5 if Digital input
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is not connected
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When device_setup is given as a parameter to the snd-usb-audio module, the
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parse_audio_enpoint 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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===================================
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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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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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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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# /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
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@ -70,6 +70,7 @@ static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Vendor ID for
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static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Product ID for this card */
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static int nrpacks = 4; /* max. number of packets per urb */
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static int async_unlink = 1;
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static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/
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module_param_array(index, int, NULL, 0444);
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MODULE_PARM_DESC(index, "Index value for the USB audio adapter.");
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@ -85,6 +86,8 @@ module_param(nrpacks, int, 0644);
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MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB.");
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module_param(async_unlink, bool, 0444);
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MODULE_PARM_DESC(async_unlink, "Use async unlink mode.");
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module_param_array(device_setup, int, NULL, 0444);
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MODULE_PARM_DESC(device_setup, "Specific device setup (if needed).");
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/*
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@ -2547,6 +2550,8 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp
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return 0;
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}
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static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
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int iface, int altno);
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static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
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{
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struct usb_device *dev;
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@ -2581,6 +2586,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
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stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ?
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SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
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altno = altsd->bAlternateSetting;
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/* audiophile usb: skip altsets incompatible with device_setup
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*/
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if (chip->usb_id == USB_ID(0x0763, 0x2003) &&
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audiophile_skip_setting_quirk(chip, iface_no, altno))
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continue;
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/* get audio formats */
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fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL);
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@ -2675,7 +2686,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
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continue;
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}
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snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, i, fp->endpoint);
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snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint);
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err = add_audio_endpoint(chip, stream, fp);
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if (err < 0) {
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kfree(fp->rate_table);
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@ -3083,6 +3094,45 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
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return 0;
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}
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/*
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* Setup quirks
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*/
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#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */
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#define AUDIOPHILE_SET_DTS 0x02 /* if set, enable DTS Digital Output */
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#define AUDIOPHILE_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
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#define AUDIOPHILE_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */
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#define AUDIOPHILE_SET_DI 0x10 /* if set, enable Digital Input */
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#define AUDIOPHILE_SET_MASK 0x1F /* bit mask for setup value */
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#define AUDIOPHILE_SET_24B_48K_DI 0x19 /* value for 24bits+48KHz+Digital Input */
|
||||
#define AUDIOPHILE_SET_24B_48K_NOTDI 0x09 /* value for 24bits+48KHz+No Digital Input */
|
||||
#define AUDIOPHILE_SET_16B_48K_DI 0x11 /* value for 16bits+48KHz+Digital Input */
|
||||
#define AUDIOPHILE_SET_16B_48K_NOTDI 0x01 /* value for 16bits+48KHz+No Digital Input */
|
||||
|
||||
static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
|
||||
int iface, int altno)
|
||||
{
|
||||
if (device_setup[chip->index] & AUDIOPHILE_SET) {
|
||||
if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS)
|
||||
&& altno != 6)
|
||||
return 1; /* skip this altsetting */
|
||||
if ((device_setup[chip->index] & AUDIOPHILE_SET_96K)
|
||||
&& altno != 1)
|
||||
return 1; /* skip this altsetting */
|
||||
if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
|
||||
AUDIOPHILE_SET_24B_48K_DI && altno != 2)
|
||||
return 1; /* skip this altsetting */
|
||||
if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
|
||||
AUDIOPHILE_SET_24B_48K_NOTDI && altno != 3)
|
||||
return 1; /* skip this altsetting */
|
||||
if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
|
||||
AUDIOPHILE_SET_16B_48K_DI && altno != 4)
|
||||
return 1; /* skip this altsetting */
|
||||
if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
|
||||
AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5)
|
||||
return 1; /* skip this altsetting */
|
||||
}
|
||||
return 0; /* keep this altsetting */
|
||||
}
|
||||
|
||||
/*
|
||||
* audio-interface quirks
|
||||
|
Loading…
Reference in New Issue
Block a user