<feed xmlns='http://www.w3.org/2005/Atom'>
<title>user/sven/linux.git/include/sound, branch v3.10.36</title>
<subtitle>Linux Kernel
</subtitle>
<id>https://git.stealer.net/cgit.cgi/user/sven/linux.git/atom?h=v3.10.36</id>
<link rel='self' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/atom?h=v3.10.36'/>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/'/>
<updated>2013-12-20T15:45:06Z</updated>
<entry>
<title>ALSA: memalloc.h - fix wrong truncation of dma_addr_t</title>
<updated>2013-12-20T15:45:06Z</updated>
<author>
<name>Stefano Panella</name>
<email>stefano.panella@citrix.com</email>
</author>
<published>2013-12-10T14:20:28Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=44b8b7a7f2e76dd1aea131bcf79417252a1f3f11'/>
<id>urn:sha1:44b8b7a7f2e76dd1aea131bcf79417252a1f3f11</id>
<content type='text'>
commit 932e9dec380c67ec15ac3eb073bb55797d8b4801 upstream.

When running a 32bit kernel the hda_intel driver is still reporting
a 64bit dma_mask if the HW supports it.

From sound/pci/hda/hda_intel.c:

        /* allow 64bit DMA address if supported by H/W */
        if ((gcap &amp; ICH6_GCAP_64OK) &amp;&amp; !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
                pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64));
        else {
                pci_set_dma_mask(pci, DMA_BIT_MASK(32));
                pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32));
        }

which means when there is a call to dma_alloc_coherent from
snd_malloc_dev_pages a machine address bigger than 32bit can be returned.
This can be true in particular if running  the 32bit kernel as a pv dom0
under the Xen Hypervisor or PAE on bare metal.

The problem is that when calling setup_bdle to program the BLE the
dma_addr_t returned from the dma_alloc_coherent is wrongly truncated
from snd_sgbuf_get_addr if running a 32bit kernel:

static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
                                           size_t offset)
{
        struct snd_sg_buf *sgbuf = dmab-&gt;private_data;
        dma_addr_t addr = sgbuf-&gt;table[offset &gt;&gt; PAGE_SHIFT].addr;
        addr &amp;= PAGE_MASK;
        return addr + offset % PAGE_SIZE;
}

where PAGE_MASK in a 32bit kernel is zeroing the upper 32bit af addr.

Without this patch the HW will fetch the 32bit truncated address,
which is not the one obtained from dma_alloc_coherent and will result
to a non working audio but can corrupt host memory at a random location.

The current patch apply to v3.13-rc3-74-g6c843f5

Signed-off-by: Stefano Panella &lt;stefano.panella@citrix.com&gt;
Reviewed-by: Frediano Ziglio &lt;frediano.ziglio@citrix.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ALSA: compress: fix drain calls blocking other compress functions (v6)</title>
<updated>2013-11-29T19:11:45Z</updated>
<author>
<name>Vinod Koul</name>
<email>vinod.koul@intel.com</email>
</author>
<published>2013-11-07T09:08:22Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=86e6de789cfeb2bb6c532281e16a478d797f3598'/>
<id>urn:sha1:86e6de789cfeb2bb6c532281e16a478d797f3598</id>
<content type='text'>
commit f44f2a5417b2968a8724b352cc0b2545a6bcb1f4 upstream.

The drain and drain_notify callback were blocked by low level driver
until the draining was complete. Due to this being invoked with big
fat mutex held, others ops like reading timestamp, calling pause, drop
were blocked.

So to fix this we add a new snd_compr_drain_notify() API. This would
be required to be invoked by low level driver when drain or partial
drain has been completed by the DSP. Thus we make the drain and
partial_drain callback as non blocking and driver returns immediately
after notifying DSP.  The waiting is done while releasing the lock so
that other ops can go ahead.

[ The commit 917f4b5cba78 was wrongly applied from the preliminary
  patch.  This commit corrects to the final version.
  Sorry for inconvenience!  -- tiwai ]

Signed-off-by: Vinod Koul &lt;vinod.koul@intel.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ALSA: compress: fix drain calls blocking other compress functions</title>
<updated>2013-11-29T19:11:45Z</updated>
<author>
<name>Vinod Koul</name>
<email>vinod.koul@intel.com</email>
</author>
<published>2013-10-24T11:07:31Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=16442d4ff3014c84008266feee1e36befd84c8c3'/>
<id>urn:sha1:16442d4ff3014c84008266feee1e36befd84c8c3</id>
<content type='text'>
commit 917f4b5cba78980a527098a910d94139d3e82c8d upstream.

The drain and drain_notify callback were blocked by low level driver untill the
draining was complete. Due to this being invoked with big fat mutex held, others
ops like reading timestamp, calling pause, drop were blocked.

So to fix this we add a new snd_compr_drain_notify() API. This would be required
to be invoked by low level driver when drain or partial drain has been completed
by the DSP. Thus we make the drain and partial_drain callback as non blocking
and driver returns immediately after notifying DSP.
The waiting is done while relasing the lock so that other ops can go ahead.

Signed-off-by: Vinod Koul &lt;vinod.koul@intel.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;

</content>
</entry>
<entry>
<title>ASoC: dapm: Treat DAI widgets like AIF widgets for power</title>
<updated>2013-06-07T14:54:50Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@linaro.org</email>
</author>
<published>2013-06-05T18:36:11Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=4616274d3382fa7698536d61b351e63cf0ce27f0'/>
<id>urn:sha1:4616274d3382fa7698536d61b351e63cf0ce27f0</id>
<content type='text'>
Even though they are virtual widgets DAI widgets still get counted for the
DAPM context power management so we can't just use the active state to
check if they should be powered as they may not be part of a complete path.

Instead split them into input and output widgets and do the same power
checks as we perform on AIFs.

Reported-by: Stephen Warren &lt;swarren@nvidia.com&gt;
Tested-by: Stephen Warren &lt;swarren@nvidia.com&gt;
Signed-off-by: Mark Brown &lt;broonie@linaro.org&gt;
</content>
</entry>
<entry>
<title>Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound</title>
<updated>2013-05-10T14:51:56Z</updated>
<author>
<name>Linus Torvalds</name>
<email>torvalds@linux-foundation.org</email>
</author>
<published>2013-05-10T14:51:56Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=05a88a43604abb816dfbff075bb114224641793b'/>
<id>urn:sha1:05a88a43604abb816dfbff075bb114224641793b</id>
<content type='text'>
Pull sound fixes from Takashi Iwai:
 "This contains small fixes since the previous pull request:

   - A few regression fixes and small updates of HD-audio

   - Yet another fix for Haswell HDMI audio

   - A copule of trivial fixes in ASoC McASP, DPAM and WM8994"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  Revert "ALSA: hda - Don't set up active streams twice"
  ALSA: Add comment for control TLV API
  ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
  ALSA: HDA: Fix Oops caused by dereference NULL pointer
  ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
  ALSA: mips/hal2: Remove redundant platform_set_drvdata()
  ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecs
  sound: Fix make allmodconfig on MIPS
  ALSA: hda - Fix system panic when DMA &gt; 40 bits for Nvidia audio controllers
  ALSA: atmel: Remove redundant platform_set_drvdata()
  ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
  ASoC: wm8994: missing break in wm8994_aif3_hw_params()
  ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format
  ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare
</content>
</entry>
<entry>
<title>ALSA: Add comment for control TLV API</title>
<updated>2013-05-08T13:43:56Z</updated>
<author>
<name>David Henningsson</name>
<email>david.henningsson@canonical.com</email>
</author>
<published>2013-05-08T13:33:11Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=d24f5a9ad9febffa53b1d3fcdf36c9a2283d848e'/>
<id>urn:sha1:d24f5a9ad9febffa53b1d3fcdf36c9a2283d848e</id>
<content type='text'>
Userspace is not meant to have to handle all strange dB ranges,
so add a specification comment.

Signed-off-by: David Henningsson &lt;david.henningsson@canonical.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound</title>
<updated>2013-05-03T16:10:23Z</updated>
<author>
<name>Linus Torvalds</name>
<email>torvalds@linux-foundation.org</email>
</author>
<published>2013-05-03T16:10:23Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=9992ba72327fa0d8bdc9fb624e80f5cce338a711'/>
<id>urn:sha1:9992ba72327fa0d8bdc9fb624e80f5cce338a711</id>
<content type='text'>
Pull sound updates from Takashi Iwai:
 "Mostly many small changes spread as seen in diffstat in sound/*
  directory by this update.  A significant change in the subsystem level
  is the introduction of snd_soc_component, which will help more generic
  handling of SoC and off-SoC components.

  Also, snd_BUG_ON() macro is enabled unconditionally now due to its
  misuses, so people might hit kernel warnings (it's a good thing for
  us).

   - compress-offload: support for capture by Charles Keepax
   - HD-audio: codec delay support by Dylan Reid
   - HD-audio: improvements/fixes in generic parser: better headphone
     mic and headset mic support, jack_modes hint consolidation, proper
     beep attach/detachment, generalized power filter controls by David
     Henningsson, et al
   - HD-audio: Improved management of HDMI codec pins/converters
   - HD-audio: Better pin/DAC assignment for VIA codecs
   - HD-audio: Haswell HDMI workarounds
   - HD-audio: ALC268 codec support, a few new quirks for Chromebooks
   - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
     fix by Clemens Ladisch
   - USB: support for DSD formats by Daniel Mack
   - USB: A few UAC2 device endian/cock fixes by Eldad Zack
   - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
     Yamaha THRxx devices
   - HDSPM: updates for TCO controls by Adrian Knoth
   - ASoC: Add a snd_soc_component object type for generic handling of
     SoC and off-SoC components by Kuninori Morimoto,
   - dmaengine: a large set of cleanups and conversions by Lars-Peter
     Clausen
   - ASoC DAPM: performance optimizations from Ryo Tsutsui
   - ASoC DAPM: support for mixer control sharing by Stephen Warren
   - ASoC: multiplatform ARM cleanups from Arnd Bergmann
   - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
  ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
  ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
  ALSA: pcm_format_to_bits strong-typed conversion
  ALSA: compress: fix the states to check for allowing read
  ALSA: hda - Move Thinkpad X220 to use auto parser
  ALSA: USB: adjust for changed 3.8 USB API
  ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
  sound: oss/dmabuf: use dma_map_single
  ALSA: ali5451: use mdelay instead of large udelay constants
  ALSA: hda - Add the support for ALC286 codec
  ALSA: usb-audio: USB quirk for Yamaha THR10C
  ALSA: usb-audio: USB quirk for Yamaha THR5A
  ALSA: usb-audio: USB quirk for Yamaha THR10
  ALSA: usb-audio: Fix autopm error during probing
  ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
  ALSA: sound kconfig typo
  ALSA: emu10k1: Fix dock firmware loading
  ASoC: ux500: forward declare msp_i2s_platform_data
  ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
  ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
  ...
</content>
</entry>
<entry>
<title>Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial</title>
<updated>2013-04-30T16:36:50Z</updated>
<author>
<name>Linus Torvalds</name>
<email>torvalds@linux-foundation.org</email>
</author>
<published>2013-04-30T16:36:50Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=5d434fcb255dec99189f1c58a06e4f56e12bf77d'/>
<id>urn:sha1:5d434fcb255dec99189f1c58a06e4f56e12bf77d</id>
<content type='text'>
Pull trivial tree updates from Jiri Kosina:
 "Usual stuff, mostly comment fixes, typo fixes, printk fixes and small
  code cleanups"

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (45 commits)
  mm: Convert print_symbol to %pSR
  gfs2: Convert print_symbol to %pSR
  m32r: Convert print_symbol to %pSR
  iostats.txt: add easy-to-find description for field 6
  x86 cmpxchg.h: fix wrong comment
  treewide: Fix typo in printk and comments
  doc: devicetree: Fix various typos
  docbook: fix 8250 naming in device-drivers
  pata_pdc2027x: Fix compiler warning
  treewide: Fix typo in printks
  mei: Fix comments in drivers/misc/mei
  treewide: Fix typos in kernel messages
  pm44xx: Fix comment for "CONFIG_CPU_IDLE"
  doc: Fix typo "CONFIG_CGROUP_CGROUP_MEMCG_SWAP"
  mmzone: correct "pags" to "pages" in comment.
  kernel-parameters: remove outdated 'noresidual' parameter
  Remove spurious _H suffixes from ifdef comments
  sound: Remove stray pluses from Kconfig file
  radio-shark: Fix printk "CONFIG_LED_CLASS"
  doc: put proper reference to CONFIG_MODULE_SIG_ENFORCE
  ...
</content>
</entry>
<entry>
<title>ALSA: pcm_format_to_bits strong-typed conversion</title>
<updated>2013-04-29T11:36:15Z</updated>
<author>
<name>Eldad Zack</name>
<email>eldad@fogrefinery.com</email>
</author>
<published>2013-04-22T23:00:41Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=74c34ca1cc12884703c70d34ed333517d978c2e7'/>
<id>urn:sha1:74c34ca1cc12884703c70d34ed333517d978c2e7</id>
<content type='text'>
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.

Change such conversions to use this function and silence sparse
warnings.

Signed-off-by: Eldad Zack &lt;eldad@fogrefinery.com&gt;
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>Merge tag 'asoc-v3.10-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next</title>
<updated>2013-04-25T11:02:35Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2013-04-25T11:02:35Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=2fc565e4eaf8fc633bfc741b90e1f28dba732ee1'/>
<id>urn:sha1:2fc565e4eaf8fc633bfc741b90e1f28dba732ee1</id>
<content type='text'>
ASoC: More updates for v3.10

A few more fixes, nothing too major though the DMA changes fix modular
builds.
</content>
</entry>
</feed>
