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* ASoC: tegra: Replace instances of rtd->codec->card with rtd->cardLars-Peter Clausen2014-07-226-16/+10
| | | | | | | | No need to go via the CODEC to get a pointer to the card. This will help to eventually remove the card field from the snd_soc_codec struct. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* ASoC: Remove per card platform listLars-Peter Clausen2014-07-221-2/+0
| | | | | | | | | The platform_dev_list was added in commit f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component Support") and while platforms are added and remove from that list it is otherwise unused. This patch removes it again. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* ASoC: Remove unused 'r' variable from dapm_connect_dai_link_widgets()Lars-Peter Clausen2014-07-171-3/+0
| | | | | | | | It was accidentally added in commit 44ba2641 ("ASoC: dapm: Add support for DAI multicodec"). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* ASoC: pcm: Add soc_dai_hw_params helperBenoit Cousson2014-07-162-39/+32
| | | | | | | | | | | Add a function helper to factorize the hw_params code. Suggested by Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* ASoC: compress: Prevent multicodec for compressed streamBenoit Cousson2014-07-161-0/+5
| | | | | | | | | | | | | Multiple codecs does not make sense in the case of compressed stream. Exit with error if it happens. Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
* ASoC: dapm: Add support for DAI multicodecBenoit Cousson2014-07-161-20/+47
| | | | | | | | | | | Add multicodec support in soc-dapm.c Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Fabien Parent <fparent@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* ASoC: pcm: Add support for DAI multicodecBenoit Cousson2014-07-161-166/+368
| | | | | | | | | | | Add multicodec support in soc-pcm.c Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Fabien Parent <fparent@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* ASoC: core: Add initial support for DAI multicodecBenoit Cousson2014-07-161-90/+203
| | | | | | | | | | | | | | | | | | | | | | | | | | DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). Multi-codec links are not supported in the case of CODEC to CODEC links. Just print a warning if it happens. Based on an original code done by Misael. Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Fabien Parent <fparent@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* Merge remote-tracking branch 'asoc/topic/component' into asoc-multiMark Brown2014-07-166-381/+381
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| * ASoC: core: Move non-shared code paths out of snd_soc_post_component_init()Lars-Peter Clausen2014-07-021-75/+46
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are two call sites for snd_soc_post_component_init(), one passes 0 and the other 1 for the 'dailess' parameter of snd_soc_post_component_init(). Depending on whether 'dailess' is 0 or 1 snd_soc_post_component_init() runs different code at the beginning and the end of the function. The patch moves this conditional code out of snd_soc_post_component_init() and into the call sites. This removes the need for snd_soc_post_component_init() to know whether it is called for a DAI link or a aux dev. Also do the initialization of rtd->card when the rtd struct is allocated. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * ASoC: core: Bind aux devs earlyLars-Peter Clausen2014-07-021-42/+25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently in snd_soc_instantiate_card() we only check if the aux dev exists, but do not yet assign it to its rtd. This means that we need to lookup the aux dev again in soc_probe_aux_dev(). This patch changes the behavior to assign the aux dev to the rtd in soc_check_aux_dev() (and renames it to soc_bind_aux_dev()). This simplifies the implementation a bit and also removes the need for soc_post_component_init() to know about the specific CODEC that was assigned to the rtd. The later is necessary for componentization as the code should work for all types of components not just CODECs. This new behavior is also more in sync with how soc_bind_dai_link()/soc_probe_link_dais() works. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * ASoC: core: Replace soc_find_matching_codec() with soc_find_codec()Lars-Peter Clausen2014-07-021-22/+4
| | | | | | | | | | | | | | | | | | soc_find_matching_codec() works in the same way as soc_find_codec() except that it only works for auxdevs. It can easily be replaced by the generic soc_find_codec(). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * ASoC: core: Remove duplicated rtd->codec initializationLars-Peter Clausen2014-07-021-6/+2
| | | | | | | | | | | | | | | | | | | | rtd->codec is already initialized in soc_bind_dai_link(), so there is no need to do it again in soc_dai_link_init(). Removing the rtd->codec initialization from soc_dai_link_init() also removes the need for soc_dai_link_init() to know about the CODEC at all. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * Merge remote-tracking branch 'asoc/fix/debugfs' into asoc-componentMark Brown2014-06-281-4/+26
| |\ | | | | | | | | | | | | Conflicts: sound/soc/soc-core.c
| | * ASoC: fix debugfs directory creation bugRussell King2014-06-281-4/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Avoid creating duplicate directories by prefixing codecs and platforms with their separate identifiers. This avoids snd-soc-dummy (which can appear both as a dummy platform and a dummy codec on the same card) from clashing. Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk> Tested-by: Andrew Lunn <andrew@lunn.ch> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | ASoC: dapm: Remove platform field from widget and dapm context structLars-Peter Clausen2014-06-212-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The platform field in the snd_soc_dapm_widget and snd_soc_dapm_context structs is now unused can be removed. New code that wants to get the platform for a widget or dapm context should use snd_soc_dapm_to_platform(w->dapm) or snd_soc_dapm_to_platform(dapm). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | ASoC: dapm: Remove DAI DAPM contextLars-Peter Clausen2014-06-211-11/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The DAI DAPM context was added in commit be09ad90 ("ASoC: core: Add platform DAI widget mapping") and the only user was removed again in commit ae10e7e8f ("ASoC: core: Only add platform DAI widgets once."). Now that we have a per component DAPM context it is unlikely that we'll need the DAI DAPM context again. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | ASoC: Add component level stream_event() and seq_notifier() supportLars-Peter Clausen2014-06-211-3/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds stream_event() and seq_notifier() callbacks similar to those found in the snd_soc_codec_driver and snd_soc_platform driver struct to the snd_soc_component_driver struct. This is meant to unify the handling of these callbacks across different types of components and will eventually allow their removal from the CODEC and platfrom driver structs. The new callbacks are slightly different from the old ones in that they take a snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was done since otherwise casting from the DAPM context to the component would typically be the first thing to do in the callback. And the interface becomes slightly cleaner by passing a snd_soc_component to all callbacks in the snd_soc_component_driver struct. The patch also already removes the stream_event() callback from the snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently unused. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | ASoC: Use component DAPM context for platformsLars-Peter Clausen2014-06-211-13/+12
| | | | | | | | | | | | | | | | | | | | | | | | The snd_soc_platform dapm field is not accessed outside of the ASoC core. Switch it over to using the snd_soc_component DAPM context. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | ASoC: Add DAPM support at the component levelLars-Peter Clausen2014-06-212-41/+78
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds full DAPM support at the component level. Previously there was only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor MUXs) for platforms. Having DAPM support at the component level will allow all types of components to use DAPM and also help in consolidating the DAPM support between CODECs and platforms. Since the DAPM context is directly embedded into the snd_soc_codec and snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of drivers moving the field just right now is not possible without causing code churn. The approach this patch takes is to add two new fields to the component struct. One field which is the pointer to the actual DAPM context used by the component and one DAPM context that will be used as the default if no other context was specified. For CODECs and platforms the pointer is initialized to point to the CODEC or platform DAPM context. All generic code when referencing a component's DAPM struct will go via the pointer. This will make it possible to eventually seamlessly move the DAPM context from snd_soc_codec and snd_soc_platform struct over once all direct references have been eliminated. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | ASoC: Add a set_bias_level() callback to the DAPM context structLars-Peter Clausen2014-06-212-8/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently the DAPM code directly looks at the CODEC driver struct to get a handle to the set_bias_level() callback. This patch adds a new set_bias_level() callback to the DAPM context struct. The DAPM code will use this new callback instead of the CODEC callback. For CODECs the new callback is set up to call the CODEC specific set_bias_level callback(). Not looking directly at the CODEC driver struct will allow non CODEC DAPM contexts to implement a set_bias_level() callback. This is also similar to how the seq_notifier() and stream_event() callbacks are currently handled. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | Merge remote-tracking branch 'asoc/fix/core' into asoc-componentMark Brown2014-06-211-13/+16
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| | * | ASoC: dapm: Make sure register value is in sync with DAPM kcontrol stateJarkko Nikula2014-06-091-13/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit c9e065c27fe9 ("ASoC: dapm: Make sure to always update the DAPM graph in _put_volsw()") stopped updating register values in those cases where initial after boot state of kcontrol appears to not change but where register value still needs update because it is not in sync with the kcontrol state. Fix this by doing snd_soc_test_bits() unconditionally as it was before but by using separate flags for kcontrol and register state changes. This allow both DAPM graph to be updated when disabling auto-muted control and update register if it is out-of-sync in respect of kcontrol state. Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | | ASoC: Auto disconnect pins from all DAPM contextsLars-Peter Clausen2014-06-212-18/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently only pins in CODEC DAPM contexts are automatically marked as non-connected if the card has the fully_routed flag set. This makes sense since widgets which qualify for auto-disconnection are only found in CODEC DAPM contexts. But with componentisation this is going to change, so consider all widgets for auto-disconnection. Also it is probably faster to walk the widgets list only once rather than once for each CODEC. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | | ASoC: Split component registration into two stepsLars-Peter Clausen2014-06-211-86/+91
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Split snd_soc_component_register() into snd_soc_component_initialize() and snd_soc_component_add(). Using a 2-stage registration approach has the advantage that it is possible to modify the component after it has been initialized, but before it is made visible to the system. This e.g. allows CODECs or platforms to overwrite some of the default settings made in snd_soc_component_initialize(). Similar snd_soc_component_unregister() is split into two steps as well, snd_soc_component_delete(), which removes the component from the system, and snd_soc_component_cleanup(), which frees all the resources allocated by the component. Furthermore this patch makes sure that if a component is visible on two list (e.g. the component list and the CODEC list) it is added or removed to both lists atomically. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | | ASoC: Move name and id from CODEC/platform to componentLars-Peter Clausen2014-06-216-47/+35
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The component struct already has a name and id field which are initialized to the same values as the same fields in the CODEC and platform structs. So remove them from the CODEC and platform structs and used the ones from the component struct instead. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
| * | | ASoC: Move name_prefix from CODEC to componentLars-Peter Clausen2014-06-212-20/+28
| | |/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | Move the name_prefix from the CODEC struct to the component struct. This will eventually allow to specify prefixes for all types of components. It is also necessary to make the DAPM code component type independent (i.e. a DAPM context does not need to know whether it belongs to a CODEC or a platform or something else). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* | | ASoC: pcm: Refactor soc_pcm_apply_msb for multicodecsBenoit Cousson2014-07-011-11/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Refactor the function to facilitate the migration to multiple codecs. Fix a trailing space in the header as well. No functional change. Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Mark Brown <broonie@linaro.org>
* | | ASoC: core: Change soc_link_dai_widgets signature for multiple codecsBenoit Cousson2014-07-011-4/+4
|/ / | | | | | | | | | | | | | | | | | | | | | | Since multiple codecs DAI will be usable in the future, remove explicit unique codec_dai and cpu_dai parameters. Replace them with snd_soc_pcm_runtime pointer that will contain every instances. No functionale change. Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Mark Brown <broonie@linaro.org>
* | Merge tag 'sound-fix-3.16-rc1' of ↵Linus Torvalds2014-06-1314-54/+145
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Most of changes are small and easy cleanup or fixes: - a few HD-audio Realtek codec fixes and quirks - Intel HDMI audio fixes for Broadwell and Haswell / ValleyView - FireWire sound stack cleanups - a couple of sequencer core fixes - compress ABI fix for 64bit - conversion to modern ktime*() API" * tag 'sound-fix-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits) ALSA: hda/realtek - Add more entry for enable HP mute led ALSA: hda - Add quirk for external mic on Lifebook U904 ALSA: hda - fix a fixup value for codec alc293 in the pin_quirk table ALSA: intel8x0: Use ktime and ktime_get() ALSA: core: Use ktime_get_ts() ALSA: hda - verify pin:converter connection on unsol event for HSW and VLV ALSA: compress: Cancel the optimization of compiler and fix the size of struct for all platform. ALSA: hda - Add quirk for ABit AA8XE Revert "ALSA: hda - mask buggy stream DMA0 for Broadwell display controller" ALSA: hda - using POS_FIX_LPIB on Broadwell HDMI Audio ALSA: hda/realtek - Add support of ALC667 codec ALSA: hda/realtek - Add more codec rename ALSA: hda/realtek - New vendor ID for ALC233 ALSA: hda - add two new pin tables ALSA: hda/realtek - Add support of ALC891 codec ALSA: seq: Continue broadcasting events to ports if one of them fails ALSA: bebob: Remove unused function prototype ALSA: fireworks: Remove meaningless mutex_destroy() ALSA: fireworks: Remove a constant over width to which it's applied ALSA: fireworks: Improve comments about Fireworks transaction ...
| * | ALSA: hda/realtek - Add more entry for enable HP mute ledKailang Yang2014-06-131-0/+14
| | | | | | | | | | | | | | | | | | | | | | | | More HP machine need mute led support. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Add quirk for external mic on Lifebook U904David Henningsson2014-06-131-0/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | According to the bug reporter (Данило Шеган), the external mic starts to work and has proper jack detection if only pin 0x19 is marked properly as an external headset mic. AlsaInfo at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1328587/+attachment/4128991/+files/AlsaInfo.txt Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1328587 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - fix a fixup value for codec alc293 in the pin_quirk tableHui Wang2014-06-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The fixup value for codec alc293 was set to ALC269_FIXUP_DELL1_MIC_NO_PRESENCE by a mistake, if we don't fix it, the Dock mic will be overwriten by the headset mic, this will make the Dock mic can't work. Cc: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: intel8x0: Use ktime and ktime_get()Thomas Gleixner2014-06-121-6/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | do_posix_clock_monotonic_gettime() is a leftover from the initial posix timer implementation which maps to ktime_get_ts() and returns the monotonic time in a timespec. Use ktime based ktime_get() and use the ktime_delta_us() function to calculate the delta instead of open coding the timespec math. Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: core: Use ktime_get_ts()Thomas Gleixner2014-06-121-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | do_posix_clock_monotonic_gettime() is a leftover from the initial posix timer implementation which maps to ktime_get_ts(). Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - verify pin:converter connection on unsol event for HSW and VLVMengdong Lin2014-06-121-1/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch will verify the pin's coverter selection for an active stream when an unsol event reports this pin becomes available again after a display mode change or hot-plug event. For Haswell+ and Valleyview: display mode change or hot-plug can change the transcoder:port connection and make all the involved audio pins share the 1st converter. So the stream using 1st convertor will flow to multiple pins but active streams using other converters will fail. This workaround is to assure the pin selects the right conveter and an assigned converter is not shared by other unused pins. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Add quirk for ABit AA8XEDavid Henningsson2014-06-101-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Bios does not set up the pin config default correctly (everything is set to zero). Reporter claims that 6stack-dig and 6stack-automute solve the problem. Alsa-info at http://www.alsa-project.org/db/?f=376c0804cbdde90bcd2cb94799407cb1cacf5d05 BugLink: https://bugs.launchpad.net/bugs/1319291 Reported-by: Stefano Statuti <stefano.statuti@hotmail.it> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Revert "ALSA: hda - mask buggy stream DMA0 for Broadwell display controller"Libin Yang2014-06-091-6/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit 7189eb9b8f7962474956196c301676470542f253. It will use LPIB to get the DMA position on Broadwell HDMI Audio. Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - using POS_FIX_LPIB on Broadwell HDMI AudioLibin Yang2014-06-091-1/+7
| | | | | | | | | | | | | | | | | | | | | Broadwell HDMI can't use position buffer reliably, force to use LPIB Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Add support of ALC667 codecKailang Yang2014-06-061-0/+1
| | | | | | | | | | | | | | | | | | | | | New codec suooprt of ALC667. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Add more codec renameKailang Yang2014-06-061-0/+15
| | | | | | | | | | | | | | | | | | | | | Some vendor has special bonding options. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - New vendor ID for ALC233Kailang Yang2014-06-061-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | This is compatible with ALC255. It is use for Lenovo. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - add two new pin tablesHui Wang2014-06-061-6/+41
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These two new pin tables can fix headset mic problems for several new Dell machines. And also delete some machines from old quirk table since the existing pin talbes already cover them. Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Add support of ALC891 codecKailang Yang2014-06-051-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | New codec support for ALC891. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: seq: Continue broadcasting events to ports if one of them failsAdam Goode2014-06-041-12/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sometimes PORT_EXIT messages are lost when a process is exiting. This happens if you subscribe to the announce port with client A, then subscribe to the announce port with client B, then kill client A. Client B will not see the PORT_EXIT message because client A's port is closing and is earlier in the announce port subscription list. The for each loop will try to send the announcement to client A and fail, then will stop trying to broadcast to other ports. Killing B works fine since the announcement will already have gone to A. The CLIENT_EXIT message does not get lost. How to reproduce problem: *** termA $ aseqdump -p 0:1 0:1 Port subscribed 0:1 -> 128:0 *** termB $ aseqdump -p 0:1 *** termA 0:1 Client start client 129 0:1 Port start 129:0 0:1 Port subscribed 0:1 -> 129:0 *** termB 0:1 Port subscribed 0:1 -> 129:0 *** termA ^C *** termB 0:1 Client exit client 128 <--- expected Port exit as well (before client exit) Signed-off-by: Adam Goode <agoode@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: bebob: Remove unused function prototypeTakashi Sakamoto2014-06-041-2/+0
| | | | | | | | | | | | | | | | | | | | | snd_bebob_stream_map() is not defined. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: fireworks: Remove meaningless mutex_destroy()Takashi Sakamoto2014-06-041-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently mutex_destroy() is called in module's cleanup function. But after cleaned up, this mutex is automatically released. So this function call is meaningless. [fixed a typo in changelog by tiwai] Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: fireworks: Remove a constant over width to which it's appliedTakashi Sakamoto2014-06-041-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | The constants of enum snd_efw_grp_type is for struct snd_efw_phys_grp.type. But this member is 1 byte. Although the value is between 0x00-0xff, a constant has 0x10000. This constant is meaningless. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: fireworks: Improve comments about Fireworks transactionTakashi Sakamoto2014-06-041-8/+8
| | | | | | | | | | | | | | | | | | | | | It includes descriptions to cause misreading. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: fireworks: Use safer way to arrange ring buffer pointerTakashi Sakamoto2014-06-042-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | To reverse a pointer for the ring buffer, subtraction by buffer size is better than assignment to the beginning of the buffer. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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