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2013-04-04ALSA: usb-audio: neaten EXPORT_SYMBOLS placementEldad Zack1-4/+3
Put EXPORT_SYMBOLS directly under the exported function. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placementEldad Zack1-2/+1
Minor style fix, following a general code style in the kernel. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: convert list_for_each to entry variantEldad Zack6-41/+21
Change occurances of list_for_each into list_for_each_entry where applicable. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-03ALSA: usb-audio: Clean up the code in set_sample_rate_v2()Takashi Iwai1-24/+23
Just for cleaning up, introduce a new function get_sample_rate_v2() for replacing two identical calls in set_sample_rate_v2(). No functional change. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-03Merge branch 'for-linus' into for-nextTakashi Iwai13-27/+73
Back-merge for cleaning up usb-audio code the recent commit modified, and further UAC2 autoclock patches.
2013-04-03ALSA: usb: Work around CM6631 sample rate change bugTorstein Hegge1-10/+35
The C-Media CM6631 USB receiver doesn't respond to changes in sample rate while the interface is active. The same behavior is observed in other UAC2 hardware like the VIA VT1731. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all UAC2 devices, as it should not affect a standards compliant device, but it is only necessary for C-Media CM6631, VIA VT1731 and possibly others. Failure to read sample rate from the device is not handled as an error in set_sample_rate_v2(), as (permanent or intermittent) failure to read sample rate isn't essential for a successful sample rate set. Signed-off-by: Torstein Hegge <hegge@resisty.net> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02ALSA: hda - bug fix on HDMI ELD debug messageMengdong Lin1-1/+1
This patch let ELD debug message show 'pin_eld->monitor_present' which reflects the real pin response to verb GET_PIN_SENSE. 'eld->monitor_present' should not be used here because 'eld' is a temp structure now and so its "monitor_present" is not set. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02ALSA: hda - bug fix on return value when getting HDMI ELD infoMengdong Lin1-1/+1
In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0. Otherwise it will be returned uninitialized as non-zero after ELD info is got successfully. Thus hdmi_present_sense() will always assume ELD info is invalid by mistake, and /proc file system cannot show the proper ELD info. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Cc: stable@vger.kernel.org Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02ALSA: hda/ca0132 - Delay HP amp turnon.Chih-Chung Chang1-4/+23
Turing on the headphone amp interferes with the impedance measurement used to detect a TRRS style headset microphone. Delay the HP turn on until 500ms after the jack is detected, allowing the mic detection state machine to run to completion. Signed-off-by: Chih-Chung Chang <chihchung@chromium.org> Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02sound: oss: sb_common: Used kmemdup instead of kmalloc and memcpyAlexandru Gheorghiu1-2/+1
Used kmemdup instead of replicating it's behaviour with kmalloc followed by memcpy. Patch found using coccinelle. Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02sound: oss: uart401: Used kmemdup instead of kmalloc and memcpyAlexandru Gheorghiu1-4/+7
Used kmemdup instead of replicating it's behaviour with kmalloc followed by memcpy. Patch found using coccinelle. Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-26Merge remote-tracking branch 'asoc/fix/spear' into asoc-nextMark Brown1-6/+6
2013-03-26Merge remote-tracking branch 'asoc/fix/si476x' into asoc-nextMark Brown1-0/+1
2013-03-26Merge remote-tracking branch 'asoc/fix/sh' into asoc-nextMark Brown1-2/+2
2013-03-26Merge remote-tracking branch 'asoc/fix/max98090' into asoc-nextMark Brown2-0/+0
2013-03-26Merge remote-tracking branch 'asoc/fix/fsl' into asoc-nextMark Brown2-1/+6
2013-03-26Merge remote-tracking branch 'asoc/fix/dapm' into asoc-nextMark Brown1-0/+14
2013-03-26Merge remote-tracking branch 'asoc/fix/core' into asoc-nextMark Brown1-4/+4
2013-03-26Merge remote-tracking branch 'asoc/fix/adsp' into asoc-nextMark Brown1-2/+3
2013-03-22ALSA: hda - VIA prefers side surrounds over HPTakashi Iwai1-0/+20
The recent fix for the independent HP reduced the availability of the side surround output, because there are only 4 DACs for 7.1 and a HP outputs. Adjust the badness tables for VIA so that 7.1 outputs are activated for the cost of missing independent HP. Once when we implement the dynamic DAC switching to multiple outputs, this conflicts will be eased in future... Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-22ALSA: hda - Lower the badness for independent HP penaltyTakashi Iwai1-1/+1
The lack of independent HP mode shouldn't be too bad, but currently its badness is set a bit too high. Let's lower it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-22ALSA: hda - Allow codec drivers to give own badness tablesTakashi Iwai2-14/+29
The standard badness values don't seem to fit to all preferences. Some configuration prefer the side output over the headphone, some want the speaker over the surround, etc. This patch moves the badness table pointers into hda_gen_spec, so that the codec driver can override them. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-22Merge branch 'for-linus' into for-nextTakashi Iwai5-34/+171
Merge back for-linus branch for the badness table adjustment for VIA codecs * for-linus: ALSA: hda - Fix DAC assignment for independent HP ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader ALSA: hda - Fix typo in checking IEC958 emphasis bit ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls() ALSA: snd-usb: mixer: propagate errors up the call chain ALSA: usb: Parse UAC2 extension unit like for UAC1 ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
2013-03-22ASoC: dma-sh7760: Fix compile errorLars-Peter Clausen1-2/+2
The dma-sh7760 currently fails with the following compile error: sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe': sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *' This is due the misnaming of the snd_soc_platform_driver type name and 'ops' field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC Multi-Component Support"). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-03-21ALSA: hda - Fix DAC assignment for independent HPTakashi Iwai1-0/+46
The generic parser should evaluate the availability of the independent HP when specified. Otherwise a DAC without the direct connection to the corresponding pin may be assigned for the HP, but the driver doesn't check it at all. The problem was actually seen on some machines with VT1708s or equivalent codec, where DAC0 is assigned to HP although it can be connected only via aamix. This patch adds the badness evaluation for the independent HP to make it working properly. Reported-by: Lydia Wang <LydiaWang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-21ALSA: hda - Enable "Headset Mic" name for some Dell Latitude devicesDavid Henningsson1-1/+5
Now that we have a "Headset Mic" name, let's use it for some devices we know for sure has a headset mic jack. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-21ALSA: hda - Introduce "Headset Mic" nameDavid Henningsson2-2/+24
Headset mic jacks, i e TRRS style jacks with Headphone Left, Headphone Right, Mic and GND signals, are becoming increasingly common and are now being shipped by several manufacturers. Unfortunately, the HDA specification does not give us any hint of whether a Mic pin belongs to such a jack or not, but it would still be helpful for the user to know (especially if there is one TRS Mic jack and one TRRS headset jack). This new fixup causes the first (non-dock, non-internal) mic to be a headset mic jack. The algorithm can be later refined if needed. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loaderTakashi Iwai1-23/+109
The current DSP loader code abuses snd_hda_lock_devices() for ensuring the DSP loader not conflicting with the other normal operations. But this trick obviously doesn't work for the PM resume since the streams are kept opened there where snd_hda_lock_devices() returns -EBUSY. That means we need another lock mechanism instead of abuse. This patch provides the new lock state to azx_dev. Theoretically it's possible that the DSP loader conflicts with the stream that has been already assigned for another PCM. If it's running, the DSP loader should simply fail. If not -- it's the case for PM resume --, we should assign this stream temporarily to the DSP loader, and take it back to the PCM after finishing DSP loading. If the PCM is operated during the DSP loading, it should get an error, too. Reported-and-tested-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ALSA: hda - Fix typo in checking IEC958 emphasis bitTakashi Iwai1-1/+1
There is a typo in convert_to_spdif_status() about checking the emphasis IEC958 status bit. It should check the given value instead of the resultant value. Reported-by: Martin Weishart <martin.weishart@telosalliance.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ASoC: core: fix invalid free of devm_ allocated dataSilviu-Mihai Popescu1-2/+0
The objects allocated by devm_* APIs are managed by devres and are freed when the device is detached. Hence there is no need to use kfree() explicitly. Signed-off-by: Silviu-Mihai Popescu <silviupopescu1990@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-20ASoC: spear_pcm: Update to new pcm_new() APILars-Peter Clausen1-6/+6
Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only rtd") updated the pcm_new() callback to take the rtd as the only parameter. The spear PCM driver (which was merged much later) still uses the old API. This patch updates the driver to the new API. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-03-20ASoC:: max98090: Remove executable bitJoe Perches2-0/+0
Source files shouldn't have the executable bit set. Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-20ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()Daniel Mack1-3/+3
Creation of individual mixer controls may fail, but that shouldn't cause the entire mixer creation to fail. Even worse, if the mixer creation fails, that will error out the entire device probing. All the functions called by parse_audio_unit() should return -EINVAL if they find descriptors that are unsupported or believed to be malformed, so we can safely handle this error code as a non-fatal condition in snd_usb_mixer_controls(). That fixes a long standing bug which is commonly worked around by adding quirks which make the driver ignore entire interfaces. Some of them might now be unnecessary. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Rodolfo Thomazelli <pe.soberbo@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ALSA: snd-usb: mixer: propagate errors up the call chainDaniel Mack1-4/+6
In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20ALSA: usb: Parse UAC2 extension unit like for UAC1Torstein Hegge1-1/+4
UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in the same way when parsing the unit. Otherwise parse_audio_unit() fails when it sees an extension unit on a UAC2 device. UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1. Signed-off-by: Torstein Hegge <hegge@resisty.net> Acked-by: Daniel Mack <zonque@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driverTakashi Iwai1-2/+2
I forgot to update spec->gpio_data in the automute hook, so it will be overridden at the init sequence, thus the machine is still silent when no headphone jack is plugged at boot time. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda - Add GPIO-based LED support on HP desktop machinesTakashi Iwai1-0/+63
The new HP desktop machines have Realtek codecs and their LEDs are controlled via GPIO as for many laptop models. Add similar hooks as well as in patch_sigmatel.c for controlling LEDs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda - Make the resume of digital beep setup properTakashi Iwai1-1/+1
The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2 should be executed at resume as well. Use the cached write for it being performed automatically at resume. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda - Fix power-saving during playing beep soundTakashi Iwai2-11/+27
While playing the digital beep tone, the codec shouldn't be turned off. This patch adds proper snd_hda_power_up()/down() calls at each time when the beep is played or off. Also, this fixes automatically an unnecessary codec power-up at detaching the beep device when the beep isn't being played. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda - Move beep attach/detach calls in hda_generic.cTakashi Iwai6-128/+72
Instead of calling snd_hda_attach_beep_device() and snd_hda_detach_beep_device() in each codec driver, move them to the generic parser. The codec driver just needs to set spec->beep_nid for activating the digital beep. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18Merge branch 'for-linus' into for-nextTakashi Iwai19-58/+164
Back-merged for refactoring beep stuff.
2013-03-18ALSA: hda/cirrus - Fix the digital beep registrationTakashi Iwai1-4/+4
The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: hda - Fix missing beep detach in patch_conexant.cTakashi Iwai1-1/+7
This leaks the beep input device after module unload, which leads to Oops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: snd-usb: add delay quirk for "Playback Design" productsDaniel Mack1-0/+7
"Playback Design" products need a 50ms delay after setting the USB interface. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: snd-usb: handle raw data format of UAC2 devicesDaniel Mack1-4/+7
UAC2 compliant audio devices may announce the capability to transport raw audio data on their endpoints. Catch this and handle it as 'special' stream on the ALSA side. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: snd-usb: handle the bmFormats field as unsigned intDaniel Mack3-6/+7
This field may use up to 32 bits, so it should be handled as unsigned int. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: usb-audio: Trust fields given in the quirkMark Hills2-5/+4
The maxpacksize field is given in some quirks, but it gets ignored (in favour of wMaxPacketSize from the first endpoint.) This patch favours the one in the quirk. Digidesign Mbox and Mbox 2 are the only affected quirks and the devices are assumed to be working without this patch. So for safety against the values in the quirk being incorrect, remove them. The datainterval is also ignored but there are not currently any quirks which choose to override this. Cc: Damien Zammit <damien@zamaudio.com> Cc: Chris J Arges <christopherarges@gmail.com> Signed-off-by: Mark Hills <mark@xwax.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18ALSA: usb-audio: Playback and MIDI support for Novation Twitch DJ controllerMark Hills2-0/+55
The hardware also has a PCM capture device which is not implemented in this patch. It may be possible to generalise this to Saffire 6 USB support and some of the other Focusrite interfaces, but as I don't have access to these devices we should wait until capture support is working first. Capture support is not implemented because the code assumes the endpoint to have its own interface (instead, it shares the interface with playback) and some thought will be needed to lift this limitation. Signed-off-by: Mark Hills <mark@xwax.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15ALSA: hda - Fix missing EAPD/GPIO setup for Cirrus codecsTakashi Iwai1-0/+4
During the transition to the generic parser, the hook to the codec specific automute function was forgotten. This resulted in the silent output on some MacBooks. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-15sound: sequencer: cap array index in seq_chn_common_event()Dan Carpenter1-0/+6
"chn" here is a number between 0 and 255, but ->chn_info[] only has 16 elements so there is a potential write beyond the end of the array. If the seq_mode isn't SEQ_2 then we let the individual drivers (either opl3.c or midi_synth.c) handle it. Those functions all do a bounds check on "chn" so I haven't changed anything here. The opl3.c driver has up to 18 channels and not 16. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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