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* ALSA: usb-audio: disable autopm for MIDI devicesClemens Ladisch2013-04-151-11/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | 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 Cc: <stable@vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix headset mic support for Asus X101CHDavid Henningsson2013-04-151-0/+56
| | | | | | | | | | | | With this patch, a TRRS headset mic cannot be successfully detected on the Asus X101CH, and we can also distinguish between headphone and headset automatically. Buglink: https://bugs.launchpad.net/bugs/1169138 Co-authored-by: Kailang <kailang@realtek.com> Tested-by: Luis Henriques <luis.henriques@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Implement headset jack functionality for some Dell hwDavid Henningsson2013-04-151-0/+386
| | | | | | | | | | | | | | | | On some machines, there is a headset jack that can support both headphone, headsets (of both CTIA and OMTP type) and mic-in. On other machines, the headset jack supports headphone, headsets (both CTIA and OMTP), but not mic-in. This patch implements that functionality as different capture sources. Buglink: https://bugs.launchpad.net/bugs/1169143 Tested-by: David Chen <david.chen@canonical.com> Co-authored-by: Kailang <kailang@realtek.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb: Add quirk for 192KHz recording on E-Mu devicesCalvin Owens2013-04-134-1/+4
| | | | | | | | | | | | | | | | | | | | | 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 <jcalvinowens@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Handle Headphone Mic jack more genericDavid Henningsson2013-04-112-24/+30
| | | | | | | | Now that we have a flag for headphone mics, we can use that flag in the jack creation instead of creating the jack manually. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - add some small convenience functions to auto parserDavid Henningsson2013-04-111-0/+21
| | | | | | | | | | I never liked that we move our speaker and hp pins to line out if there are not any line outs; but now that we do, add some convenience functions to find hp and speaker pins even if they have been moved. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - allow "Headphone Mic" parser flagDavid Henningsson2013-04-113-15/+62
| | | | | | | | | | | | This allows a specific mic to get the "Headphone Mic" name, in addition to the existing "Headset Mic" name. Also, it allows for a special mark: if the sequence number is set to 0xc, that's an indication to prefer it for headset mic, and if it's set to 0xd, that's an indication to prefer it for headphone mic. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: at73c213: Use dev_pm_opsLars-Peter Clausen2013-04-111-9/+13
| | | | | | | | Use dev_pm_ops instead of the deprecated legacy suspend/resume callbacks. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locationsDaniel Mack2013-04-104-6/+15
| | | | | | | | | It turns out the devices from Playback Design need the delay quirk after usb_set_interface from clocks.c as well. Make it a proper quirks function and factor out the code to quirks.c. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Apply codec delay to wallclock.Dylan Reid2013-04-091-0/+18
| | | | | | | | | For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek - Add a quirk for AC700 Chromebook.Dylan Reid2013-04-071-1/+17
| | | | | | | | | | Correct pin configs for the Acer AC700. Most importantly indicate that SPDIF is connected, it routes to HDMI out. Similar to Aspire models, chain in the DMIC fixup and allow it to be applied to this codec (ALC269VB) as well. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/cirrus - Add a quirk for Stumpy ChromeBox.Dylan Reid2013-04-051-1/+18
| | | | | | | The Stumpy ChromeBox needs its pin configs fixed up. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/ca0132 - Update latency based on DSP state.Dylan Reid2013-04-051-2/+53
| | | | | | | | | | | The DSP in the CA0132 codec adds a variable latency to audio depending on what processing is being done. Add a new patch op to return that latency for capture and playback streams. The latency is determined by which blocks are enabled and knowing how much latency is added by each block. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Introduce get_delay codec PCM opsTakashi Iwai2013-04-052-4/+17
| | | | | | | | | | Add a new codec PCM ops, get_delay(), to obtain the codec/stream- specific PCM delay count. When it's NULL, nothing changes. This new feature was requested for CA0132, which has significant delays in the path depending on the running DSP code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: UAC2: support read-only freq controlEldad Zack2013-04-041-11/+26
| | | | | | | | | | | | | | | | | Some clocks might be read-only, e.g., external clocks (see also UAC2 4.7.2.1). In this case, setting the sample frequency will always fail (even if the rate is equal to the current clock rate), therefore do not write, but read the value and compare to the requested rate. If the clock is read only, avoid reading it twice. If it doesn't match, return -ENXIO since the clock is invalid for this configuration. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: show err in set_sample_rate_v2 debugEldad Zack2013-04-041-4/+4
| | | | | | | | Show the error code returned from the USB subsystem in the debug messages. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: UAC2: auto clock selection module paramEldad Zack2013-04-043-1/+6
| | | | | | | | | | Add a module param to disable auto clock selection. This is provided for users that expect the audio stream to fail when the clock source is invalid (e.g., the word clock was unintentionally disconnected). Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: UAC2: try to find and switch to valid clockEldad Zack2013-04-041-3/+66
| | | | | | | | | | If a selector is available on a device, it may be pointing to a clock source which is currently invalid. If there is a valid clock source which can be selected, switch to it. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: UAC2: do clock validity check earlierEldad Zack2013-04-043-18/+21
| | | | | | | | | | | | | | | Move the check that parse_audio_format_rates_v2() do after receiving the clock source entity ID directly into the find function and add a validation flag to the function. This patch does not introduce any logic flow change. It is provided to allow introducing automatic clock switching easier later. By moving this uac_clock_source_is_valid callsite, 2 additional callsites can be avoided. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: use endianness macrosEldad Zack2013-04-041-9/+6
| | | | | | | | Replace the endianness conversions with the kernel-wide swabbing macros in get/set_sample_rate_v2. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: spelling correctionEldad Zack2013-04-043-9/+9
| | | | | | | | Correct spelling of snd_usb_endpoint_implict_feedback_sink in all occurances. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: neaten EXPORT_SYMBOLS placementEldad Zack2013-04-041-4/+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>
* ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placementEldad Zack2013-04-041-2/+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>
* ALSA: usb-audio: convert list_for_each to entry variantEldad Zack2013-04-046-41/+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>
* ALSA: usb-audio: Clean up the code in set_sample_rate_v2()Takashi Iwai2013-04-031-24/+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>
* Merge branch 'for-linus' into for-nextTakashi Iwai2013-04-03887-4477/+9477
|\ | | | | | | | | Back-merge for cleaning up usb-audio code the recent commit modified, and further UAC2 autoclock patches.
| * ALSA: usb: Work around CM6631 sample rate change bugTorstein Hegge2013-04-031-10/+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>
| * ALSA: hda - bug fix on HDMI ELD debug messageMengdong Lin2013-04-021-1/+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>
| * ALSA: hda - bug fix on return value when getting HDMI ELD infoMengdong Lin2013-04-021-1/+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>
| * Merge tag 'asoc-fix-v3.9-rc4' of ↵Takashi Iwai2013-04-02884-4465/+9440
| |\ | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.9 A few more fixes here and there, including quite a few nasty driver specific ones, but nothing that has a major general impact.
| | * Merge remote-tracking branch 'asoc/fix/spear' into asoc-nextMark Brown2013-03-261-6/+6
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| | | * ASoC: spear_pcm: Update to new pcm_new() APILars-Peter Clausen2013-03-201-6/+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
| | * | Merge remote-tracking branch 'asoc/fix/si476x' into asoc-nextMark Brown2013-03-261-0/+1
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| | | * | ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch caseAxel Lin2013-02-201-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Axel Lin <axel.lin@ingics.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | Merge remote-tracking branch 'asoc/fix/sh' into asoc-nextMark Brown2013-03-261-2/+2
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| | | * | | ASoC: dma-sh7760: Fix compile errorLars-Peter Clausen2013-03-221-2/+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
| | * | | Merge remote-tracking branch 'asoc/fix/max98090' into asoc-nextMark Brown2013-03-263-0/+0
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| | | * | | ASoC:: max98090: Remove executable bitJoe Perches2013-03-203-0/+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>
| | * | | Merge remote-tracking branch 'asoc/fix/fsl' into asoc-nextMark Brown2013-03-262-1/+6
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| | | * | | ASoC: pcm030 audio fabric: remove __init from probeMarkus Pargmann2013-03-121-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove probe function from the init section. Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | ASoC: imx-ssi: Fix occasional AC97 reset failureSascha Hauer2013-03-121-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | * | | | Merge remote-tracking branch 'asoc/fix/dapm' into asoc-nextMark Brown2013-03-262-0/+15
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| | | * | | | ASoC: dapm: Fix pointer dereference in is_connected_output_ep()Peter Ujfalusi2013-03-151-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | *path is not yet initialized when we check if the widget is connected. The compiler also warns about this: sound/soc/soc-dapm.c: In function 'is_connected_output_ep': sound/soc/soc-dapm.c:824:18: warning: 'path' may be used uninitialized in this function Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | | ASoC: dapm: Fix handling of loopsMark Brown2013-02-252-0/+16
| | | | |_|/ | | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently if a path loops back on itself we correctly skip over it to avoid going into an infinite loop but this causes us to ignore the need to power up the path as we don't count the loop for the purposes of counting inputs and outputs. This means that internal loopbacks within a device that have powered devices on them won't be powered up. Fix this by treating any path that is currently in the process of being recursed as having a single input or output so that it is counted for the purposes of power decisions. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
| | * | | | Merge remote-tracking branch 'asoc/fix/core' into asoc-nextMark Brown2013-03-261-4/+4
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| | | * | | | ASoC: core: fix invalid free of devm_ allocated dataSilviu-Mihai Popescu2013-03-201-2/+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>
| | | * | | | ASoC: core: fix possible memory leak in snd_soc_bytes_put()Wei Yongjun2013-03-121-2/+4
| | | | |/ / | | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 'data' is malloced in snd_soc_bytes_put() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | Merge remote-tracking branch 'asoc/fix/adsp' into asoc-nextMark Brown2013-03-261-2/+3
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| | | * | | | ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff()Wei Yongjun2013-03-121-2/+3
| | | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 'file' is malloced in wm_adsp_load_coeff() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | Linux 3.9-rc4v3.9-rc4Linus Torvalds2013-03-231-1/+1
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