<feed xmlns='http://www.w3.org/2005/Atom'>
<title>user/sven/linux.git/sound, branch v3.2.45</title>
<subtitle>Linux Kernel
</subtitle>
<id>https://git.stealer.net/cgit.cgi/user/sven/linux.git/atom?h=v3.2.45</id>
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<updated>2013-05-13T14:02:33Z</updated>
<entry>
<title>vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helper</title>
<updated>2013-05-13T14:02:33Z</updated>
<author>
<name>Linus Torvalds</name>
<email>torvalds@linux-foundation.org</email>
</author>
<published>2013-04-19T17:01:04Z</published>
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<id>urn:sha1:dfdc98ac7e4a90c07bc988ab1870fd1528bd2100</id>
<content type='text'>
commit 0fe09a45c4848b5b5607b968d959fdc1821c161d upstream.

This is my example conversion of a few existing mmap users.  The pcm
mmap case is one of the more straightforward ones.

Acked-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Linus Torvalds &lt;torvalds@linux-foundation.org&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ALSA: usb-audio: Fix autopm error during probing</title>
<updated>2013-05-13T14:02:26Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2013-04-25T05:38:15Z</published>
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<id>urn:sha1:42c72c8c4efa25812f711c96b0f04f7a1091a2b9</id>
<content type='text'>
commit 60af3d037eb8c670dcce31401501d1271e7c5d95 upstream.

We've got strange errors in get_ctl_value() in mixer.c during
probing, e.g. on Hercules RMX2 DJ Controller:

  ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4
  ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4
  ....

It turned out that the culprit is autopm: snd_usb_autoresume() returns
-ENODEV when called during card-&gt;probing = 1.

Since the call itself during card-&gt;probing = 1 is valid, let's fix the
return value of snd_usb_autoresume() as success.

Reported-and-tested-by: Daniel Schürmann &lt;daschuer@mixxx.org&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT</title>
<updated>2013-05-13T14:02:25Z</updated>
<author>
<name>Daniel Mack</name>
<email>zonque@gmail.com</email>
</author>
<published>2013-04-24T17:38:42Z</published>
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<id>urn:sha1:7b475d0e41f0979b8f8f475ae5fd3e026216e318</id>
<content type='text'>
commit ebfc594c02148b6a85c2f178cf167a44a3c3ce10 upstream.

The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually
stuffed directly after the standard USB endpoint descriptor, and this is
where the driver currently expects it to be.

There are, however, devices in the wild that have it the other way
around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes
*before* the standard enpoint. Devices known to implement it that way
are "Sennheiser BTD-500" and Plantronics USB headsets.

When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to
change sample rates, as the bitmask for the validity of this command is
storen in bmAttributes of that descriptor.

Fix this by searching the entire interface instead of just the extra
bytes of the first endpoint, in case the latter fails.

Signed-off-by: Daniel Mack &lt;zonque@gmail.com&gt;
Reported-and-tested-by: Torstein Hegge &lt;hegge@resisty.net&gt;
Reported-and-tested-by: Yves G &lt;alsa-user@vivigatt.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ASoC: max98088: Fix logging of hardware revision.</title>
<updated>2013-05-13T14:02:20Z</updated>
<author>
<name>Dylan Reid</name>
<email>dgreid@chromium.org</email>
</author>
<published>2013-04-17T03:02:34Z</published>
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<id>urn:sha1:929628fcc44a4be40bb886f8558838324e63e833</id>
<content type='text'>
commit 98682063549bedd6e2d2b6b7222f150c6fbce68c upstream.

The hardware revision of the codec is based at 0x40.  Subtract that
before convering to ASCII.  The same as it is done for 98095.

Signed-off-by: Dylan Reid &lt;dgreid@chromium.org&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ALSA: usb-audio: disable autopm for MIDI devices</title>
<updated>2013-05-13T14:02:17Z</updated>
<author>
<name>Clemens Ladisch</name>
<email>clemens@ladisch.de</email>
</author>
<published>2013-04-15T13:59:51Z</published>
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<id>urn:sha1:7766cdc32e03471a00dec6dd4cb487ceca811501</id>
<content type='text'>
commit cbc200bca4b51a8e2406d4b654d978f8503d430b upstream.

Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend)
introduced autopm for all USB audio/MIDI devices.  However, many MIDI
devices, such as synthesizers, do not merely transmit MIDI messages but
use their MIDI inputs to control other functions.  With autopm, these
devices would get powered down as soon as the last MIDI port device is
closed on the host.

Even some plain MIDI interfaces could get broken: they automatically
send Active Sensing messages while powered up, but as soon as these
messages cease, the receiving device would interpret this as an
accidental disconnection.

Commit f5f165418cab (ALSA: usb-audio: Fix missing autopm for MIDI input)
introduced another regression: some devices (e.g. the Roland GAIA SH-01)
are self-powered but do a reset whenever the USB interface's power state
changes.

To work around all this, just disable autopm for all USB MIDI devices.

Reported-by: Laurens Holst
Signed-off-by: Clemens Ladisch &lt;clemens@ladisch.de&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ALSA: usb: Add quirk for 192KHz recording on E-Mu devices</title>
<updated>2013-05-13T14:02:17Z</updated>
<author>
<name>Calvin Owens</name>
<email>jcalvinowens@gmail.com</email>
</author>
<published>2013-04-13T03:33:59Z</published>
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<id>urn:sha1:40de8047654debace75db0f04aebbbaace5bdd26</id>
<content type='text'>
commit 1539d4f82ad534431cc67935e8e442ccf107d17d upstream.

When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.

Userspace expected:  L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1

Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.

Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.

Signed-off-by: Calvin Owens &lt;jcalvinowens@gmail.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
[bwh: Backported to 3.2: adjust filenames, context]
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ALSA: hda - fix typo in proc output</title>
<updated>2013-04-25T19:25:47Z</updated>
<author>
<name>David Henningsson</name>
<email>david.henningsson@canonical.com</email>
</author>
<published>2013-04-04T09:47:13Z</published>
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<id>urn:sha1:b11b9b2c5e21f294be556c8ab9fe6cee48fca567</id>
<content type='text'>
commit aeb3a97222832e5457c4b72d72235098ce4bfe8d upstream.

Rename "Digitial In" to "Digital In". This function is only used for
proc output, so should not cause any problems to change.

Signed-off-by: David Henningsson &lt;david.henningsson@canonical.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ALSA: hda - Enabling Realtek ALC 671 codec</title>
<updated>2013-04-25T19:25:47Z</updated>
<author>
<name>Rainer Koenig</name>
<email>Rainer.Koenig@ts.fujitsu.com</email>
</author>
<published>2013-04-04T06:40:38Z</published>
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<id>urn:sha1:4e41c2fc993103bb300b1608aa905cfeb9fc0e2e</id>
<content type='text'>
commit 1d87caa69c04008e09f5ff47b5e6acb6116febc7 upstream.

* Added the device ID to the modalias list and assinged ALC662 patches
for it
* Added 4 port support for the device ID 0671 in alc662_parse_auto_config

Signed-off-by: Rainer Koenig &lt;Rainer.Koenig@ts.fujitsu.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running</title>
<updated>2013-04-25T19:25:35Z</updated>
<author>
<name>Alban Bedel</name>
<email>alban.bedel@avionic-design.de</email>
</author>
<published>2013-04-09T15:13:59Z</published>
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<id>urn:sha1:6248d3d8c0611025c48e86481b9c509ac107e5b1</id>
<content type='text'>
commit f1ca493b0b5e8f42d3b2dc8877860db2983f47b6 upstream.

The Charge Pump needs the DSP clock to work properly, without it the
bypass to HP/LINEOUT is not working properly. This requirement is not
mentioned in the datasheet but has been confirmed by Mark Brown from
Wolfson.

Signed-off-by: Alban Bedel &lt;alban.bedel@avionic-design.de&gt;
Signed-off-by: Mark Brown &lt;broonie@opensource.wolfsonmicro.com&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
<entry>
<title>ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*</title>
<updated>2013-04-25T19:25:33Z</updated>
<author>
<name>Eldad Zack</name>
<email>eldad@fogrefinery.com</email>
</author>
<published>2013-04-05T18:49:46Z</published>
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<id>urn:sha1:74e038af9a2696bad0b69bb04c43e48aba484eea</id>
<content type='text'>
commit 889d66848b12d891248b03abcb2a42047f8e172a upstream.

The usb_control_msg() function expects __u16 types and performs
the endianness conversions by itself.
However, in three places, a conversion is performed before it is
handed over to usb_control_msg(), which leads to a double conversion
(= no conversion):
* snd_usb_nativeinstruments_boot_quirk()
* snd_nativeinstruments_control_get()
* snd_nativeinstruments_control_put()

Caught by sparse:

sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:512:38:    expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:512:38:    got restricted __le16 [usertype] &lt;noident&gt;
sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types)
sound/usb/mixer_quirks.c:543:35:    expected unsigned short [unsigned] [usertype] value
sound/usb/mixer_quirks.c:543:35:    got restricted __le16 [usertype] &lt;noident&gt;
sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:543:56:    expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:543:56:    got restricted __le16 [usertype] &lt;noident&gt;
sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types)
sound/usb/quirks.c:502:35:    expected unsigned short [unsigned] [usertype] value
sound/usb/quirks.c:502:35:    got restricted __le16 [usertype] &lt;noident&gt;

Signed-off-by: Eldad Zack &lt;eldad@fogrefinery.com&gt;
Acked-by: Daniel Mack &lt;zonque@gmail.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Ben Hutchings &lt;ben@decadent.org.uk&gt;
</content>
</entry>
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