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* ASoC: ad193x: fix dac word len settingScott Jiang2011-10-032-1/+3
| | | | | | | | | | | | commit 95c93d8525ebce1024bda7316f602ae45c36cd6f upstream. dac word len value should left shift before setting Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com> Acked-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: ad193x: fix registers definitionScott Jiang2011-10-031-2/+2
| | | | | | | | | | | commit bf545ed72f2eeac664695a8ea2199d9ddaef6020 upstream. fix dac word len mask and adc tdm fmt shift value Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: soc-jack: Fix checking return value of request_any_context_irqAxel Lin2011-10-031-1/+1
| | | | | | | | | | | | commit d2b4c7bd7eabfaa2e3e5b8107d5eeb56ac879813 upstream. request_any_context_irq() returns a negative value on failure. On success, it returns either IRQC_IS_HARDIRQ or IRQC_IS_NESTED. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: ac97: Add HP Compaq dc5100 SFF(PT003AW) to Headphone Jack Sense whitelistDaniel T Chen2011-08-291-0/+1
| | | | | | | | | | | | | | | commit eade7b281c9fc18401b989c77d5e5e660b25a3b7 upstream. BugLink: https://bugs.launchpad.net/bugs/826081 The original reporter needs 'Headphone Jack Sense' enabled to have audible audio, so add his PCI SSID to the whitelist. Reported-and-tested-by: Muhammad Khurram Khan Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: snd_usb_caiaq: track submitted output urbsDaniel Mack2011-08-292-4/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | commit da6094ea7d3c2295473d8f5134279307255d6ebf upstream. The snd_usb_caiaq driver currently assumes that output urbs are serviced in time and doesn't track when and whether they are given back by the USB core. That usually works fine, but due to temporary limitations of the XHCI stack, we faced that urbs were submitted more than once with this approach. As it's no good practice to fire and forget urbs anyway, this patch introduces a proper bit mask to track which requests have been submitted and given back. That alone however doesn't make the driver work in case the host controller is broken and doesn't give back urbs at all, and the output stream will stop once all pre-allocated output urbs are consumed. But it does prevent crashes of the controller stack in such cases. See http://bugzilla.kernel.org/show_bug.cgi?id=40702 for more details. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Matej Laitl <matej@laitl.cz> Cc: Sarah Sharp <sarah.a.sharp@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: usb-audio - Fix missing mixer dB informationTakashi Iwai2011-08-291-1/+2
| | | | | | | | | | | | | | | | | commit 38b65190c6ab0be8ce7cff69e734ca5b5e7fa309 upstream. The recent fix for testing dB range at the mixer creation time seems to cause regressions in some devices. In such devices, reading the dB info at probing time gives an error, thus both dBmin and dBmax are still zero, and TLV flag isn't set although the later read of dB info succeeds. This patch adds a workaround for such a case by assuming that the later read will succeed. In future, a similar test should be performed in a case where a wrong dB range is seen even in the later read. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: Tegra: wm8903 machine driver: Allow re-insertion of moduleStephen Warren2011-08-171-4/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | commit 29591ed4ac6fe00e3ff23b5be0cdc7016ef9c47e upstream. Two issues were preventing module snd-soc-tegra-wm8903.ko from being removed and re-inserted: a) The speaker-enable GPIO is hosted by the WM8903 chip. This GPIO must be freed before snd_soc_unregister_card() is called, because that triggers wm8903.c:wm8903_remove(), which calls gpiochip_remove(), which then fails if any of the GPIOs are in use. To solve this, free all GPIOs first, so the code doesn't care where they come from. b) We need to call snd_soc_jack_free_gpios() to match the call to snd_soc_jack_add_gpios() during initialization. Without this, the call to snd_soc_jack_add_gpios() fails during any subsequent modprobe and initialization, since the GPIO and IRQ are already registered. In turn, this causes the headphone state not to be monitored, so the headphone is assumed not to be plugged in, and the audio path to it is never enabled. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: Tegra: tegra_pcm_deallocate_dma_buffer: Don't OOPSStephen Warren2011-08-171-2/+7
| | | | | | | | | | | | | | | commit a96edd59b2bc88b3d1ea47e0ba48076d65db9302 upstream. Not all PCM devices have all sub-streams. Specifically, the SPDIF driver only supports playback and hence has no capture substream. Check whether a substream exists before dereferencing it, when de-allocating DMA buffers in tegra_pcm_deallocate_dma_buffer. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: Fix binding of WM8750 on JiveMark Brown2011-08-171-1/+1
| | | | | | | | | | | commit 6678050442e90a4e9511a9ed14b9bdfc5e393323 upstream. The I2C address is misformatted and would never match. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: snd-usb-caiaq: Correct offset fields of outbound iso_frame_descDaniel Mack2011-08-171-2/+4
| | | | | | | | | | | | | | | | commit 15439bde3af7ff88459ea2b5520b77312e958df2 upstream. This fixes faulty outbount packets in case the inbound packets received from the hardware are fragmented and contain bogus input iso frames. The bug has been there for ages, but for some strange reasons, it was only triggered by newer machines in 64bit mode. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: William Light <wrl@illest.net> Reported-by: Pedro Ribeiro <pedrib@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: snd-usb: Accept UAC2 FORMAT_TYPE descriptors with bLength > 6Clemens Ladisch2011-08-151-1/+1
| | | | | | | | | | | | | | commit 824818b148db42173446707df4cbd61cd7133272 upstream. The Focusrite Scarlett 18i6 USB has them that way, which is probably a bug. Anyway, the driver should simply ignore this fact. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: snd-usb: operate on given mixer interface onlyDaniel Mack2011-08-152-12/+9
| | | | | | | | | | | | | | | | | commit 1faa5d07a93fc5b0a4a5254fc940a79e20b55540 upstream. When creating the mixers for an USB audio device, the current code looks at the host interface stored in mixer->chip->ctrl_if. Change this and rather keep a local pointer to the interface that was given when snd_usb_create_mixer() was called. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com> Reported-by: Lean-Yves LENHOF <jean-yves@lenhof.eu.org> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: snd-usb: avoid dividing by zero on invalid inputNicolai Krakowiak2011-08-151-0/+5
| | | | | | | | | | | commit 60c961a9e1ed879a4d151df6076bf1203f595f73 upstream. Signed-off-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com> Acked-by: Daniel Mack <zonque@gmail.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: snd-usb-caiaq: Fix keymap for RigKontrol3Daniel Mack2011-08-151-1/+1
| | | | | | | | | | commit f4389489b5cbe60b3441869c68bb4afe760969c4 upstream. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Renato <naretobh@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: timer - Fix Oops at closing slave timerTakashi Iwai2011-08-151-0/+2
| | | | | | | | | | | | | | commit 0584ffa548b6e59aceb027112f23a55f0133400e upstream. A slave-timer instance has no timer reference, and this results in NULL-dereference at stopping the timer, typically called at closing the device. Reference: https://bugzilla.kernel.org/show_bug.cgi?id=40682 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: sgtl5000: fix cache handlingWolfram Sang2011-08-151-93/+35
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 151798f872d6b386d82cd1707ad703e981fef8f2 upstream. Cache handling in this driver is broken. The chip has 16-bit registers, yet the register numbers also increase by 2 per register, i.e. there are only even-numbered registers. The cache in this driver, though, simply increments register numbers, so it does need some mapping as seen in sgtl5000_restore_regs(), note the '>> 1': snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); That, of course, won't work with snd_soc_update_bits(). (Thus, we won't even notice the missing register 0x1c in the default regs which shifted all follwing registers to wrong values.) Noticed on the MX28EVK where enabling the regulators simply locked up the chip. Refactor the routines and use a properly sized default_regs array which matches the register layout of the underlying chip, i.e. create a truly flat cache. This also saves some code which should make up for the bigger array a little. When soc-core will somewhen have another cache type which handles a step size, this conversion will also ease the transition. Signed-off-by: Wolfram Sang <w.sang@pengutronix.de> Tested-by: Dong Aisheng <b29396@freescale.com> Tested-by: Shawn Guo <shawn.guo@linaro.org> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: sound/core/pcm_compat.c: adjust array indexJulia Lawall2011-08-041-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | commit ca9380fd68514c7bc952282c1b4fc70607e9fe43 upstream. Convert array index from the loop bound to the loop index. A simplified version of the semantic patch that fixes this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e1,e2,ar; @@ for(e1 = 0; e1 < e2; e1++) { <... ar[ - e2 + e1 ] ...> } // </smpl> Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: hda - Fix duplicated DAC assignments for RealtekTakashi Iwai2011-08-041-2/+5
| | | | | | | | | | | | commit c48a8fb0d31d6147d8d76b8e2ad7f51a2fbb5c4d upstream. Copying hp_pins and speaker_pins from line_out_pins may confuse the parser, and it can lead to duplicated initializations for the same pin with a wrong DAC assignment. The problem appears in 3.0 kernel code. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ALSA: virtuoso: fix silent analog output on Xonar Essence ST DeluxeClemens Ladisch2011-08-041-1/+4
| | | | | | | | | | | | | | | commit c81c6b356b52d3fcb4d531d149573fc100aad643 upstream. Commit dd203fa97bd5 (ALSA: virtuoso: remove non-working controls on Essence ST Deluxe) made it impossible to adjust the volume after the driver initialized it to muted. Ensure that those DACs that can be accessed with I2C are initialized to the same volume that is the reset default of the DAC without I2C. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: Mark cache as dirty when suspendingMark Brown2011-08-041-0/+1
| | | | | | | | | | | | | | | | | | commit 7be4ba24a3ea53bc8ade841635e4d4a59e98ceb5 upstream. Since quite a few drivers are not managing to flag the cache as needing to be resynced after suspend and it's a reasonable thing to do flag the cache as needing sync automatically when suspending. The expectation is that systems will mainly only keep the CODEC powered when doing audio through the CODEC so we won't actually suspend the device anyway; drivers which want to can override this behaviour when they resume. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: davinci: fix codec start and stop functionsRajashekhara, Sudhakar2011-08-041-4/+4
| | | | | | | | | | | | | | | | | | commit 3012f43eaf7592d8121426918e43e3b5db013aff upstream. According to DM365 voice codec data sheet at [1], before starting recording or playback, ADC/DAC modules should follow a reset and enable cycle. Writing a 1 to the ADC/DAC bit in the register resets the module and clearing the bit to 0 will enable the module. But the driver seems to be doing the reverse of it. [1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: davinci: add missing break statementRajashekhara, Sudhakar2011-08-041-0/+1
| | | | | | | | | | | | | | commit 82d1d521036eb3f5aae48b847f939d99a44c18bb upstream. In davinci_vcif_trigger() function, a break() statement was missing causing the davinci_vcif_stop() function to be called as a fallback after calling davinci_vcif_start(). Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* Merge branch 'fix/asoc' of ↵Linus Torvalds2011-07-184-15/+16
|\ | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ASoC: Correct WM8994 MICBIAS supply widget hookup ASoC: Fix shift in WM8958 accessory detection default implementation ASoC: sh: fsi-hdmi: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-ak4642: fixup snd_soc_card name
| * ASoC: Correct WM8994 MICBIAS supply widget hookupMark Brown2011-07-171-3/+4
| | | | | | | | | | | | | | | | | | | | | | | | The WM8994 and WM8958 series of devices have two MICBIAS supplies rather than one, the current widget actually manages the microphone detection control register bit (which is managed separately by the relevant API). Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2 widgets. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * ASoC: Fix shift in WM8958 accessory detection default implementationMark Brown2011-07-151-1/+1
| | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * ASoC: sh: fsi-hdmi: fixup snd_soc_card nameKuninori Morimoto2011-07-091-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | it shouldn't contain space letters and special letters like parentheses. aplay will be "Segmentation fault" without this patch special thanks to Takashi. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: sh: fsi-da7210: fixup snd_soc_card nameKuninori Morimoto2011-07-091-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | it shouldn't contain space letters and special letters like parentheses. aplay will be "Segmentation fault" without this patch. special thanks to Takashi. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: sh: fsi-ak4642: fixup snd_soc_card nameKuninori Morimoto2011-07-091-8/+8
| | | | | | | | | | | | | | | | | | | | | | | | it shouldn't contain space letters and special letters like parentheses. aplay will be "Segmentation fault" without this patch. special thanks to Takashi. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-07-098-35/+45
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| * Merge branch 'for-3.0' of ↵Takashi Iwai2011-07-092-27/+4
| |\ | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc
| | * ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2Kuninori Morimoto2011-07-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | mask didn't cover update-data Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | * ASoC: Manage WM8731 ACTIVE bit as a supply widgetMark Brown2011-07-051-26/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | Now we have supply widgets there's no need to open code the handling of the ACTIVE bit. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com> Acked-by: Liam Girdwood <lrg@ti.com>
| * | ASoC: Don't set invalid name string to snd_card->driver fieldTakashi Iwai2011-07-051-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The snd_card->driver field contains a driver name string, and in general it shouldn't contain space or special letters. The commit 2b39535b9e54888649923beaab443af212b6c0fd changed the string copy from card->name, but the long name string may contain such letters, thus it may still lead to a segfault. A temporary fix is not to copy the long name string but just keep it empty as the earlier version did. Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'for-3.0' of ↵Takashi Iwai2011-07-053-2/+19
| |\ \ | | |/ | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc
| | * ASoC: Ensure we delay long enough for WM8994 FLL to lock when startingMark Brown2011-07-041-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | This delay is very conservative. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Cc: stable@kernel.org
| | * ASoC: Tegra: I2S: Ensure clock is enabled when writing regsStephen Warren2011-07-041-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The I2S controller needs a clock to respond to register writes. Without this, register writes will at worst hang the CPU. In practice, I've only observed writes being dropped. Luckily, the dropped register writes historically had no effect: TEGRA_I2S_TIMING: The value we wrote was the reset default. TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data when one slot was empty. The requested value was for the FIFOs to request when four slots were empty. The DMA controller in the mainline kernel is configured to burst a single entry at a time into the FIFO, hence there was no issue. The only negative effect was on bus efficiency losses due to an increased number of arbitration attempts. However, in various non-upstream changes, the DMA controller now bursts four entries at a time into the FIFO. If there is only space for one entry, the data is simply dropped. In practice, this resulted in 3/4 of samples being dropped, and playback at 4x the expected rate and pitch. By fixing the clocking issue, this is solved. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Fix Blackfin I2S _pointer() implementation return in bounds valuesMark Brown2011-06-291-2/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Blackfin DMA controller can report one frame beyond the end of the buffer in the wraparound case but ALSA requires that the pointer always be in the buffer. Do the wraparound to handle this. A similar bug is likely to apply to the other Blackfin PCM drivers but the code is less obvious to inspection and I don't have a user to test. Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * | Merge branch 'for-3.0' of ↵Takashi Iwai2011-07-052-4/+19
| |\ \ | | |/ | |/| | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc
| | * ASoC: tlv320aic3x: Do soft reset to codec when going to bias off stateJarkko Nikula2011-05-231-0/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | TLV320AIC33, TLV320AIC34 and I believe others too in this family have some hw bugs that cause that analogue and digital VDD supplies remain leaking up to a few mA of current after certain use cases even the hw blocks inside codec are driven to off. Highest leakages occur after using the bypass paths inside codec but it is possible to get smaller leakages just by toggling mute switches in unused audio paths (i.e. no DAPM changes) while codec is on due another active audio path. While some cases are able to workaroud by making sure that e.g. output mixer switches are muted before powering down the output stage this doesn't help all the cases. Therefore use the software reset command to clear possible leakage currents since that works in every cases and affects only this codec instance. Only drawback is that now cache sync is required everytime when codec bias comes out from bias off state, not only when supply regulators were off. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
| | * ASoC: tlv320aic3x: Don't sync first two registers from register cacheJarkko Nikula2011-05-231-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There is no need to sync first two registers from cache to hw after a reset. First one is used to select page for register access and this driver is normally accessing page 0 only. Second one does a software reset which is obviously unneeded after hardware or previous software reset command. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
| | * audio: tlv320aic26: fix PLL register configurationMichael Williamson2011-05-211-3/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current PLL configuration code for the tlc320aic26 codec appears to assume a hardcoded system clock of 12 MHz. Use the clock value provided by the DAI_OPS API for the calculation. Tested using a MityDSP-L138 platform providing a 24.576 MHz clock. Signed-off-by: Michael Williamson <michael.williamson@criticallink.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
* | | ALSA: hda - Fix a copmile warningTakashi Iwai2011-07-091-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | It's harmless but annyoing. sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’: sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Change all ADCs for dual-adc switching mode for RealtekTakashi Iwai2011-07-071-10/+23
| | | | | | | | | | | | | | | | | | | | | | | | | | | When the dual-adc switching mode is active in Realtek auto-parser, we need to couple all ADCs as a single capture-volume. Currently, the volume control changes only the first ADC, thus others may remain silent. This patch fixes the problem. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Fix compile warnings with PPCTakashi Iwai2011-06-301-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The char can be unsigned on some architectures. Since the code checks the negative values, they should be declared as signed char explicitly. sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: cs5535 - Fix invalid big-endian conversionsTakashi Iwai2011-06-301-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | Fix the wrongly converted short values: sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDMI - fix ELD monitor name lengthWu Fengguang2011-06-291-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | I noticed that the last character of the ELD monitor name is lost, this fixes the issue. This fix should be confirming to the HDA spec, and works together with the DRM part of the ELD patch. The HDA spec does not mention that Monitor_Name_String is an '\0' ending string, and it allows NML to be 1, which is only valid when MNL does not count the possible ending '\0'. Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: atmel - update author email for ABDAC, AC97C and AT73C213Hans-Christian Egtvedt2011-06-283-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | This patch updates the email address of the sound drivers supported by me to an email account I will use on a more regular basis in the future. Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDA: Add model=auto quirk for Acer Aspire 3830TGDavid Henningsson2011-06-281-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | Since we're not using the new auto parser as a fallback yet, add it manually as a quirk. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDA: Add a new Conexant codec ID (506c)David Henningsson2011-06-281-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Conexant ID 506c was found on Acer Aspire 3830TG. As users report no playback, sending to stable should be safe. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/783582 Reported-by: andROOM Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-06-245-13/+5
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