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* 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>
| * ALSA: fireworks/bebob: Shorten critical section for stream_stop_duplex()Takashi Sakamoto2014-06-042-4/+4
| | | | | | | | | | | | | | | | All assignment for local variables in these functions are not related to critical section. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: seq: correctly detect input buffer overflowAdam Goode2014-06-041-1/+1
| | | | | | | | | | | | | | | | | | | | snd_seq_event_dup returns -ENOMEM in some buffer-full conditions, but usually returns -EAGAIN. Make -EAGAIN trigger the overflow condition in snd_seq_fifo_event_in so that the fifo is cleared and -ENOSPC is returned to userspace as stated in the alsa-lib docs. Signed-off-by: Adam Goode <agoode@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge tag 'sound-3.16-rc1' of ↵Linus Torvalds2014-06-04300-5237/+35826
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into next Pull sound updates from Takashi Iwai: "At this time, majority of changes come from ASoC world while we got a few new drivers in other places for FireWire and USB. There have been lots of ASoC core cleanups / refactoring, but very little visible to external users. ASoC: - Support for specifying aux CODECs in DT - Removal of the deprecated mux and enum macros - More moves towards full componentisation - Removal of some unused I/O code - Lots of cleanups, fixes and enhancements to the davinci, Freescale, Haswell and Realtek drivers - Several drivers exposed directly in Kconfig for use with simple-card - GPIO descriptor support for jacks - More updates and fixes to the Freescale SSI, Intel and rsnd drivers - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781, and Realtek RT5677 HD-audio: - Clean up Dell headset quirks - Noise fixes for Dell and Sony laptops - Thinkpad T440 dock fix - Realtek codec updates (ALC293,ALC233,ALC3235) - Tegra HD-audio HDMI support FireWire-audio: - FireWire audio stack enhancement (AMDTP, MIDI), support for incoming isochronous stream and duplex streams with timestamp synchronization - BeBoB-based devices support - Fireworks-based device support USB-audio: - Behringer BCD2000 USB device support Misc: - Clean up of a few old drivers, atmel, fm801, etc" * tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (480 commits) ASoC: Fix wrong argument for card remove callbacks ASoC: free jack GPIOs before the sound card is freed ALSA: firewire-lib: Remove a comment about restriction of asynchronous operation ASoC: cache: Fix error code when not using ASoC level cache ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data ASoC: add RT5677 CODEC driver ASoC: intel: The Baytrail/MAX98090 driver depends on I2C ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651 ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error. ASoC: Add helper functions to cast from DAPM context to CODEC/platform ALSA: bebob: sizeof() vs ARRAY_SIZE() typo ASoC: wm9713: correct mono out PGA sources ALSA: synth: emux: soundfont.c: Cleaning up memory leak ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320 ASoC: fsl-ssi: Use regmap ASoC: fsl-ssi: reorder and document fsl_ssi_private ...
| * ASoC: Fix wrong argument for card remove callbacksTakashi Iwai2014-06-037-9/+8
| | | | | | | | | | | | | | | | | | | | | | The commit [e1d4d3c8: ASoC: free jack GPIOs before the sound card is freed] introduced snd_soc_card remove callbacks to a few drivers, but they are implemented with a wrong argument type. The callback should receive snd_soc_card pointer instead of snd_soc_pcm_runtime. Fixes: e1d4d3c854f2 ('ASoC: free jack GPIOs before the sound card is freed') Acked-by: Mark Brown <broonie@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge tag 'asoc-v3.16-2' of ↵Takashi Iwai2014-06-0387-932/+9417
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Final updates for v3.16 A few more updates from the last week of development, nothing too exciting. Highlights include: - GPIO descriptor support for jacks - More updates and fixes to the Freescale SSI, Intel and rsnd drivers. - New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781, and Realtek RT5677.
| | * ASoC: free jack GPIOs before the sound card is freedStephen Warren2014-06-037-34/+69
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is the same change as commit fb6b8e71448a "ASoC: tegra: free jack GPIOs before the sound card is freed", but applied to all other ASoC machine drivers where code inspection indicates the same problem exists. That commit's description is: ========== snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, guard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. ========== Note that I have not even compile-tested this in most cases, since most of the drivers rely on specific mach-* support I don't have enabled, and don't support COMPILE_TEST. Testing by the relevant board maintainers would be useful. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | *-. Merge remote-tracking branches 'asoc/topic/wm8804' and 'asoc/topic/wm9713' ↵Mark Brown2014-06-033-5/+19
| | |\ \ | | | | | | | | | | | | | | | into asoc-next
| | | | * ASoC: wm9713: correct mono out PGA sourcesMatt Reimer2014-06-011-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The mono output PGA input only has four possible sources, so omit the rest. Signed-off-by: Matt Reimer <mreimer@sdgsystems.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | ASoC: wm8804: Allow control of master clock divider in PLL generationDaniel Matuschek2014-05-292-3/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel@matuschek.net> Tested-by: Florian Meier <florian.meier@koalo.de> Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | * | | Merge remote-tracking branch 'asoc/topic/tegra' into asoc-nextMark Brown2014-06-034-14/+45
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| | | * | | ASoC: tegra: free jack GPIOs before the sound card is freedStephen Warren2014-05-264-14/+45
| | | | |/ | | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, gGuard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. This change fixes all Tegra machine drivers. By code inspection, I believe some non-Tegra machine drivers have the same issue. I'll send a patch for that separately, once this is reviewed. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@linaro.org>
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| | *-----. \ \ Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', ↵Mark Brown2014-06-0311-94/+80
| | |\ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next
| | | | | | * | | ASoC: sirf-audio-codec: Simplify the new bitmask value in regmap_update_bitsAxel Lin2014-05-261-5/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Having the binary ones complement operator in the new bitmak value makes the code hard to read. Signed-off-by: Axel Lin <axel.lin@ingics.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | | * | | | ASoC: simple-card: Support setting mclk via a fixed factorAndrew Lunn2014-05-261-0/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some platforms require that the codecs mclk is a fixed multiplication factor of the audio stream rate. Add a optional property to the binding to hold this factor and implement a hw_params() function to make use of it. Signed-off-by: Andrew Lunn <andrew@lunn.ch> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | | * | | | ASoC: simple-card: is_top_level_node parameter to simple_card_dai_link_of()Jyri Sarha2014-04-241-4/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Restore correct parsing of dai-link subnodes with more explicit implementation for applying the "simple-audio-card,"-prefix to dai-link property and subnode names. Signed-off-by: Jyri Sarha <jsarha@ti.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | * | | | | ASoC: sgtl5000: Fix the cache handlingFabio Estevam2014-05-271-60/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. The kernel oops happens inside sgtl5000_restore_regs() as codec->reg_cache is no longer a valid pointer. Add the remaining register entries into sgtl5000_reg_defaults[] and remove sgtl5000_restore_regs() completely, which allows suspend/resume to work fine and make the code simpler. Tested on a im53-qsb board. Reported-by: Shawn Guo <shawn.guo@freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Tested-by: Shawn Guo <shawn.guo@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | | | | | ASoC: samsung: Use params_width()Tushar Behera2014-05-268-24/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 8c5178fca4ce ("ALSA: Add params_width() helpers") introduces a helper to get the sample width. Updating Samsung related sound drivers to use this helper. Signed-off-by: Tushar Behera <tushar.behera@linaro.org> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | | | | | ASoC: samsung: Handle errors when getting the op_clk clockSylwester Nawrocki2014-05-221-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Ensure i2s->op_clk is not used when clk_get() for this clock fails. This prevents working with an incorrectly configured clock in some conditions. Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@linaro.org>
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| | *-. \ \ \ \ \ \ Merge remote-tracking branches 'asoc/topic/rl6231' and 'asoc/topic/rt5677' ↵Mark Brown2014-06-0312-307/+5187
| | |\ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | into asoc-next
| | | | * | | | | | | ASoC: add RT5677 CODEC driverOder Chiou2014-06-014-0/+4955
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds the Realtek ALC5677 codec driver. Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | | | | | | | ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared supportOder Chiou2014-06-015-48/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The patch adds the function "get_clk_info" to RL6231 shared support. Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | | | | | | | ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared ↵Oder Chiou2014-06-018-206/+102
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | support The patch adds the function of the PLL clock calculation to RL6231 shared support. Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | * | | | | | | | ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and ↵Oder Chiou2014-06-017-53/+110
| | | |/ / / / / / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | RT5651 The patch adds the RL6231 class device shared support for RT5640, RT5645 and RT5651. The function of the DMIC clock calculation can be shared by RL6231 shared support. Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@linaro.org>
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| | | *-. \ \ \ \ \ \ Merge branches 'topic/rt5640', 'topic/rt5645' and 'topic/rt5651' of ↵Mark Brown2014-06-016-0/+8653
| | | |\ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rl6231
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| | *-. \ \ \ \ \ \ \ \ \ Merge remote-tracking branches 'asoc/topic/omap' and 'asoc/topic/rcar' into ↵Mark Brown2014-06-0312-66/+264
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| | | | * | | | | | | | | | ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addrKuninori Morimoto2014-05-263-4/+101
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The DMAC src/dst addr needs to be set from driver when DT case. (It was set from SoC/DMAEngine code when non-DT case) This patch adds rsnd_gen_dma_addr() to set DMAC src/dst addr. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | * | | | | | | | | | ASoC: rsnd: care DMA slave channel name for DTKuninori Morimoto2014-05-262-1/+85
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Renesas sound driver is supporting to use DMAEngine. But, DMA slave channel name "tx", "rx" is not enough in DT case. Becuase, it has many ports and path combination. This patch adds rsnd_dma_of_name() to find DMA channel name, for example memory to SSI0 is "mem_ssi0", SSI0 to memory is "ssi0_mem", SSI0 to SRC0 is "ssi0_src0", SRC0 to SSI0 is "src0_ssi0", SRC0 to DVC0 is "src0_dvc0"... Renesas sound want to use PIO transfer mode for some reasons. It will be PIO tranfer mode if device node doesn't have DMA settings. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | * | | | | | | | | | ASoC: rsnd: module name is unifiedKuninori Morimoto2014-05-263-9/+49
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Renesas sound driver uses many modules (= SSI/SRC/DVC), and each module had own name. But, each module name can be used as several purpose, like clock name, DMA name etc... This patch uses common name for each module. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
| | | | * | | | | | | | | | ASoC: rsnd: remove rsnd_src_non_opsKuninori Morimoto2014-05-261-10/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Renesas sound driver is supporting Gen1/Gen2. SRC probe can return error if it was unknown generation. Now, rsnd_src_non_ops is not needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-07 12:33:18 +0000