aboutsummaryrefslogtreecommitdiffstats
path: root/sound
Commit message (Collapse)AuthorAge
...
| | * | | ALSA: ice1724: increase SPDIF and independent stereo buffer sizesRobert Hancock2009-10-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Increase the default and maximum PCM buffer prellocation size for ice1724's SPDIF and independent stereo pair outputs to 256K, which is the hardware's maximum supported size. This allows a reduction in interrupt rate and potentially power usage when an application is not latency-critical. Signed-off-by: Robert Hancock <hancockrwd@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: opl3: circular locking in the snd_opl3_note_on() and snd_opl3_note_off()Krzysztof Helt2009-10-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix following circular locking in the opl3 driver. ======================================================= [ INFO: possible circular locking dependency detected ] 2.6.32-rc3 #87 ------------------------------------------------------- swapper/0 is trying to acquire lock: (&opl3->voice_lock){..-...}, at: [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] but task is already holding lock: (&opl3->sys_timer_lock){..-...}, at: [<cca75169>] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth] which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&opl3->sys_timer_lock){..-...}: [<c02461d5>] validate_chain+0xa25/0x1040 [<c0246aca>] __lock_acquire+0x2da/0xab0 [<c024731a>] lock_acquire+0x7a/0xa0 [<c044c300>] _spin_lock_irqsave+0x40/0x60 [<cca75046>] snd_opl3_note_on+0x686/0x790 [snd_opl3_synth] [<cca68912>] snd_midi_process_event+0x322/0x590 [snd_seq_midi_emul] [<cca74245>] snd_opl3_synth_event_input+0x15/0x20 [snd_opl3_synth] [<cca4dcc0>] snd_seq_deliver_single_event+0x100/0x200 [snd_seq] [<cca4de07>] snd_seq_deliver_event+0x47/0x1f0 [snd_seq] [<cca4e50b>] snd_seq_dispatch_event+0x3b/0x140 [snd_seq] [<cca5008c>] snd_seq_check_queue+0x10c/0x120 [snd_seq] [<cca5037b>] snd_seq_enqueue_event+0x6b/0xe0 [snd_seq] [<cca4e0fd>] snd_seq_client_enqueue_event+0xdd/0x100 [snd_seq] [<cca4eb7a>] snd_seq_write+0xea/0x190 [snd_seq] [<c02827b6>] vfs_write+0x96/0x160 [<c0282c9d>] sys_write+0x3d/0x70 [<c0202c45>] syscall_call+0x7/0xb -> #0 (&opl3->voice_lock){..-...}: [<c02467e6>] validate_chain+0x1036/0x1040 [<c0246aca>] __lock_acquire+0x2da/0xab0 [<c024731a>] lock_acquire+0x7a/0xa0 [<c044c300>] _spin_lock_irqsave+0x40/0x60 [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [<cca751f0>] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth] [<c022ac46>] run_timer_softirq+0x166/0x1e0 [<c02269e8>] __do_softirq+0x78/0x110 [<c0226ac6>] do_softirq+0x46/0x50 [<c0226e26>] irq_exit+0x36/0x40 [<c0204bd2>] do_IRQ+0x42/0xb0 [<c020328e>] common_interrupt+0x2e/0x40 [<c021092f>] apm_cpu_idle+0x10f/0x290 [<c0201b11>] cpu_idle+0x21/0x40 [<c04443cd>] rest_init+0x4d/0x60 [<c055c835>] start_kernel+0x235/0x280 [<c055c066>] i386_start_kernel+0x66/0x70 other info that might help us debug this: 2 locks held by swapper/0: #0: (&opl3->tlist){+.-...}, at: [<c022abd0>] run_timer_softirq+0xf0/0x1e0 #1: (&opl3->sys_timer_lock){..-...}, at: [<cca75169>] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth] stack backtrace: Pid: 0, comm: swapper Not tainted 2.6.32-rc3 #87 Call Trace: [<c0245188>] print_circular_bug+0xc8/0xd0 [<c02467e6>] validate_chain+0x1036/0x1040 [<c0247f14>] ? check_usage_forwards+0x54/0xd0 [<c0246aca>] __lock_acquire+0x2da/0xab0 [<c024731a>] lock_acquire+0x7a/0xa0 [<cca748fe>] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [<c044c300>] _spin_lock_irqsave+0x40/0x60 [<cca748fe>] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [<c044c307>] ? _spin_lock_irqsave+0x47/0x60 [<cca751f0>] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth] [<c022ac46>] run_timer_softirq+0x166/0x1e0 [<c022abd0>] ? run_timer_softirq+0xf0/0x1e0 [<cca75150>] ? snd_opl3_timer_func+0x0/0xc0 [snd_opl3_synth] [<c02269e8>] __do_softirq+0x78/0x110 [<c044c0fd>] ? _spin_unlock+0x1d/0x20 [<c025915f>] ? handle_level_irq+0xaf/0xe0 [<c0226ac6>] do_softirq+0x46/0x50 [<c0226e26>] irq_exit+0x36/0x40 [<c0204bd2>] do_IRQ+0x42/0xb0 [<c024463c>] ? trace_hardirqs_on_caller+0x12c/0x180 [<c020328e>] common_interrupt+0x2e/0x40 [<c0208d88>] ? default_idle+0x38/0x50 [<c021092f>] apm_cpu_idle+0x10f/0x290 [<c0201b11>] cpu_idle+0x21/0x40 [<c04443cd>] rest_init+0x4d/0x60 [<c055c835>] start_kernel+0x235/0x280 [<c055c210>] ? unknown_bootoption+0x0/0x210 [<c055c066>] i386_start_kernel+0x66/0x70 Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: ICE1712/24 - Change the Multi Track Peak control (level meters) from ↵Pavel Hofman2009-10-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | MIXER to PCM type * PLEASE NOTE - this change requires the corresponding update of envy24control for ice1712 - kind of an ABI change. * The "Multi Track Peak" control is read-only level meters indicator. * The control is VERY confusing to most users since it is currently displayed in regular mixers. E.g. alsamixer ignores its read-only status and allows changing the levels with keys which makes no sense. Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Acked-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | sound: via82xx: move DXS volume controls to PCM interfaceClemens Ladisch2009-10-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The "VIA DXS" controls are actually volume controls that apply to the four PCM substreams, so we better indicate this connection by moving the controls to the PCM interface. Commit b452e08e73c0e3dbb0be82130217be4b7084299e in 2.6.30 broke the restoring of these volumes by "alsactl restore" that most distributions use; the renaming in this patch cures that regression by preventing alsactl from applying the old, wrong volume levels to the new controls. http://bugzilla.kernel.org/show_bug.cgi?id=14151 http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613 Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | Merge branch 'fix/hda' into for-linusTakashi Iwai2009-10-08
| |\ \ \ \ | | | |/ / | | |/| |
| | * | | ALSA: hda - Fix yet another auto-mic bug in ALC268Takashi Iwai2009-10-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since patch_alc268() doesn't call set_capture_mixer() (due to its h/w design different from other siblings), it needs to call fixup_automic_adc() explicitly to set up the auto-mic routing. Otherwise the indices for int/ext mics aren't set properly. Reference: Novell bnc#544899 http://bugzilla.novell.com/show_bug.cgi?id=544899 Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: hda - Don't pick up invalid HP pins in alc_subsystem_id()Takashi Iwai2009-10-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | alc_subsystem_id() tries to pick up a headphone pin if not configured, but this caused side-effects as the problem in commit 15870f05e90a365f8022da416e713be0c5024e2f. This patch fixes the driver behavior to pick up invalid HP pins; at least, the pins that are listed as the primary outputs aren't taken any more. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: hda - Add a workaround for ASUS A7KTakashi Iwai2009-10-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ASUS A7K needs additional GPIO1 bit setup; it has to be cleared. Added a new fixup hook for this laptop so that it works as is. Refernece: Novell bnc#494309 http://bugzilla.novell.com/show_bug.cgi?id=494309 Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | ALSA: hda - Fix invalid initializations for ALC861 auto modeTakashi Iwai2009-10-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent auto-parser doesn't work for machines with a single output with ALC861, such as Toshiba laptops, because alc_subsystem_id() sets the hp_pins[0] while it's listed in line_outs[0]. This ends up with the doubled initialization of the same mixer widget, and it mutes the DAC route because hp_pins has no DAC assigned. To fix this problem, just check spec->autocfg.hp_outs and speaker_outs so that they are really detected pins. Reference: Novell bnc#544161 http://bugzilla.novell.com/show_bug.cgi?id=544161 Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ASoC: Factor out snd_soc_init_card()Mark Brown2009-11-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_soc_init_card() is always called as the last part of the CODEC probe function so we can factor it out into the core card setup rather than have each CODEC replicate the code to do the initialiastation. This will be required to support multiple CODECs per card. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Move sysfs and debugfs functions to head of soc-core.cMark Brown2009-11-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | A fairly hefty change in diff terms but no actual code changes, will be used by the next commit. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Staticise wm8727 driver structureMark Brown2009-11-03
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: Make sure, that the codec is powered on startupPeter Ujfalusi2009-11-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Set the codec->bias_level to SND_SOC_BIAS_OFF before changing the initial bias level to STANDBY. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Add support for the WM8727 DAC.Neil Jones2009-11-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for the Wolfson Microelectronics WM8727 DAC, this is a simple non-configurable DAC. Signed-off-by: Neil Jones <neil.jones@imgtec.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: au1x: convert to platform drivers.Manuel Lauss2009-11-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Convert psc-ac97,i2s to platform drivers similar to the davinci ones. Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: refactor snd_soc_update_bits()Eero Nurkkala2009-10-30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Introduce a wrapper call snd_soc_update_bits_locked() that will take the codec mutex. This call is used when the codec mutex is not already taken. Drivers calling snd_soc_update_bits() may wish to make sure the codec mutex is taken from the driver. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: remove io_mutexEero Nurkkala2009-10-30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove the io_mutex. It has a drawback of serializing all accesses to snd_soc_update_bits() even when multiple codecs are in use. In addition, it fails to actually do its task - during snd_soc_update_bits(), dapm_update_bits() may also be accessing the same register which may result in an outdated register value. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-30
|\ \ \ \ \ | | |_|_|/ | |/| | |
| * | | | ASoC: Amstrad Delta: add info about the line discipline requirement to ↵Janusz Krzysztofik2009-10-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Kconfig help text I thought it could be usefull to add some information on how to get the device fully supported by loading a line discipline on the modem line. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: sh: FSI: Add capture supportKuninori Morimoto2009-10-30
| | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: sh: FSI: Remove DMA supportKuninori Morimoto2009-10-30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | SuperH FSI device have the hardware limitation to use DMA. If DMA is used, LCD output will be broken. Maybe there are some solution. But I don't know how to do it now. This patch remove DMA support and use soft transfer. Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Modifying Kconfig/Makefile for AM3517 EVMAnuj Aggarwal2009-10-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Modifying the Kconfig and Makefile in sound/soc/omap folder to add support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Adding OMAP3517 / AM3517 EVM support in ASOCAnuj Aggarwal2009-10-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Adding support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removalAnuj Aggarwal2009-10-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The pop-removal specific values are configured for TWL4030 codec for OMAP3EVM through this patch. Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: Add APLL supply for the capture pathPeter Ujfalusi2009-10-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Capture path also need the APLL enabled, adding DAPM_SUPPLY for the Virtual ADCs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: Change APLL powering sequencePeter Ujfalusi2009-10-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It seams that certain part of the twl4030 codec needs the APLL enabled before they are enabled. Paths which has any digital processing needs need the APLL enabled before they can function. For example the vibra output will have some random data after it is enabled and before the APLL also enabled. If only analog components are in use (analog bypass), than it seams, that the APLL does not need to be enabled. This lowers the power consumption with around ~0.005A. Adding DAPM_SUPPLY to the Digital playback route and also to the capture route to enable and disable the APLL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: Vibra motor stop fix when it is driven with audioJari Vanhala2009-10-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch fixes vibrator playing incoherently, when driven with audio. There is something wrong in switch 3 at H-bridge and VIBRA_SET still affects PWM generator. Slowest value fixes things. Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: CS4270: export de-emphasis filter as ALSA controlDaniel Mack2009-10-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The CS4270 codec features an de-emphasis filter for compensation of audio material filtered by an 50/15 uS algorithm. Not sure whether we should always enable it for 44100Hz sampling frequency, but it should at least be configurable by the user. Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Minor SMDK64xx WM8580 cleanupsMark Brown2009-10-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix up some comments, remove all enable_pin() calls (edge widgets are all enabled by default) and mark the microphone as disabled by default since it requires a resistor fit to connect it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: Change codec_muted to apll_enabledPeter Ujfalusi2009-10-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | codec_muted is missleading, change it to apll_enabled, which is what it is doing: enabing and disabling the APLL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: Remove bypass trackingPeter Ujfalusi2009-10-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since ASoC core now handling the codec bias differently there is no need to do the tracking of bypass switch states anymore. Handling of the common bit for analog loopbacks is done with DAPM_SUPPLY for the bypass paths. Now this bit is only enabled when there is a complete analog bypass path, compared to the previous implementation, when the global switch was enabled if there were any of the analog bypass switch was on (regardless if there were complete path or not) Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Add regulator support for WM8731Mark Brown2009-10-26
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: Driver registration via twl4030_codec MFDPeter Ujfalusi2009-10-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: use the twl4030-codec.h for register descriptionsPeter Ujfalusi2009-10-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove the register descriptions from the twl4030.h file and use the linux/mfd/twl4030-codec.h instead, which has the codec related register descriptions also. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1Janusz Krzysztofik2009-10-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c, omap_pcm_prepare() unconditionally calls: omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); Current implementation of those two functions found in arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at all, so they both end with BUG() on that machine. That results in ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta. The patch corrects the problem by not calling those two functions when run on OMAP1 class based machines. Created against linux-2.6.32-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: tlv320dac33: typo fix in the headerPeter Ujfalusi2009-10-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix the definition of DAC33_LTM field, the LTM bits in FIFO_IRQ_MODE_B register are starting at bit 6. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Amstrad Delta minor cleanupsJanusz Krzysztofik2009-10-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Hi Mark, Here is a patch that corrects small omissions I have found in my code. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-19
|\| | | |
| * | | | ASoC: Fix possible codec_dai->ops NULL pointer problemsBarry Song2009-10-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc. access the ops field in these DAIs, panic will happen. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: Move dereference after NULL testJulia Lawall2009-10-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If the NULL test on jack is needed, then the derefernce should be after the NULL test. A simplified version of the semantic match that detects this problem is as follows (http://coccinelle.lip6.fr/): // <smpl> @match exists@ expression x, E; identifier fld; @@ * x->fld ... when != \(x = E\|&x\) * x == NULL // </smpl> Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: au1x: psc-ac97: reorganize timeoutsManuel Lauss2009-10-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Codec read/write functions: wait 21us between the pokings of hardware. Add timeouts to unbounded loops waiting for bits to change. Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: au1x: psc-ac97: verify correct codec register was readManuel Lauss2009-10-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Verify that the correct register has been received from the codec. Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | ASoC: TWL4030: Only update the needed bits in *set_dai_sysclkPeter Ujfalusi2009-10-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Do not rewrite the whole register, but only update the needed bits in set_dai_sysclk functions. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-15
|\| | | |
| * | | | ASoC: Serialize access to dapm_power_widgets()Eero Nurkkala2009-10-13
| |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Access to damp_power_widgets() is assumed to be single-threaded. Concurrent accesses to dapm_power_widgets() may result in unpredictable behavior. Calls from: close_delayed_work() soc_codec_close() soc_pcm_prepare() soc_suspend() soc_resume_deferred() to snd_soc_dapm_stream_event() do not have the codec->mutex taken to cover the call to dapm_power_widgets(). Thus, take the mutex in these paths also to assure single-threaded use of dapm_power_widgets(). Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Codec driver for Texas Instruments tlv320dac33 codecPeter Ujfalusi2009-10-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo audio DAC. TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low power audio playback. The digital interface can use I2S, DSP (A or B), Right and Left justified formats. DAC33 has stereo analog input, which can be bypassed to the analog outputs. Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass' mode (default) and nSample mode (FIFO is in use). a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is working synchronously as a normal codec (it needs constant stream of data on the digital interface). b) The nSample mode implementation uses one interrupt line from DAC33 to the host: Alarm threshold is set to 10ms of audio data (limit by the driver implementation). DAC33 will signal an interrupt, when the FIFO level goes under the Alarm threshold. The host will write to nSample register a value (number of stereo samples), to tell DAC33 how many samples it should read in a burst from the host. When the DAC33 received the number of samples, it disables the clocks on the I2S bus. When the FIFO use again goes under the Alarm threshold, DAC33 signals the host with an interrupt, and the process is repeated. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: finally enable support for eXeda and CM-X300Igor Grinberg2009-10-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Igor Grinberg <grinberg@compulab.co.il> Signed-off-by: Mike Rapoport <mike@compulab.co.il> CC: Mark Brown <broonie@opensource.wolfsonmicro.com> CC: alsa-devel@alsa-project.org Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Remove snd_soc_suspend_device()Mark Brown2009-10-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The PM core will grow pm_link infrastructure in 2.6.33 which can be used to implement the intended functionality of the ASoC-specific device suspend and resume callbacks so drop them. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: S3C: Remove <plat/audio.h>Ben Dooks2009-10-13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h as it provides nothing to the current kernel and is not in any future plans for the system. Signed-off-by: Ben Dooks <ben@simtec.co.uk> Signed-off-by: Simtec Linux Team <linux@simtec.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: TPA6130A2: Make tpa6130a2_power as staticPeter Ujfalusi2009-10-12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The power for the amplifier should be handled internally by the tpa6130a2 driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
generated by cgit v1.2.2 (git 2.25.0) at 2024-11-17 05:13:55 -0500