aboutsummaryrefslogtreecommitdiffstats
path: root/sound (follow)
AgeCommit message (Collapse)AuthorFilesLines
2011-07-22ALSA: asihpi - Add volume mute controlsEliot Blennerhassett1-1/+50
Mute functionality was recently added to the DSP firmware Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Control name updatesEliot Blennerhassett1-4/+8
Add names corresponding to new HPI node types. Shorten some names so that constructed names don't overflow the maximum name length. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Use size_t for sizeof resultEliot Blennerhassett1-2/+2
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Explicitly include mutex.hEliot Blennerhassett1-0/+1
Because mutex is used in adapter struct defined here. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Add new node and message definesEliot Blennerhassett2-6/+15
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Make local function staticEliot Blennerhassett1-0/+1
Fixes a sparse warning. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Fix minor typos and spellingEliot Blennerhassett3-3/+3
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Remove unused structures, macros and functionsEliot Blennerhassett5-30/+0
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Remove spurious adapter index checkEliot Blennerhassett1-7/+3
Subsystem requests don't have or need a valid adapter index. The adapter index is already checked further on, before it is used to index the adapters array. (Reverts 4a122c10f) Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macroEliot Blennerhassett1-9/+15
Work towards moving the function into alsa common header. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - DSP code loader API now independent of OSEliot Blennerhassett4-122/+97
The loader API has been revised so that OS specific data is kept local to hpidspcd.c, and the public API is unchanged across OSes. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Remove controlex structs and associated special data transfer codeEliot Blennerhassett3-160/+76
Some cobranet control data would not fit in an original HPI message. Now that HPI is able to transfer larger messages, this special handling is no longer required. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Increase request and response buffer sizesEliot Blennerhassett3-19/+34
Allow for up to 256 bytes of extra data on top of standard hpi request and response sizes. Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22ALSA: asihpi - Give more meaningful name to hpi request message typeEliot Blennerhassett6-10/+11
Having a 'request message' makes more sense than a 'message message' Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21ALSA: usb-audio - Add quirk for Roland / BOSS BR-800David G Turner1-0/+30
Add support for Roland/BOSS BR-800 (0582:011e) to snd-usb-audio driver. This allows playback and recording, which has been tested and found to work. The third interface should be MIDI (MTC/SMPTE?) for DAW interface and is set as per ME-25, but this has not been tested. SDHC card access is already supported by usb-storage for Backup/Rhythm Editor/Wave Convertor mode which should not conflict with this. Signed-off-by: David G Turner <dgturner@iee.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21ALSA: hda - Remove a superfluous argument of via_auto_init_output()Takashi Iwai1-14/+10
"force" argument is always true, so let's strip it off. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecsTakashi Iwai1-40/+52
This patch fixes non-working indep-HP control on VT1708* codecs. The problems are that via_independent_hp_put() wasn't fixed to follow the recent change of three HP paths, and hp_indep_path didn't contain the amp nids of mixer elements. Together with the fixes, a few code clean-ups are done. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-20ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.Liam Girdwood1-0/+30
In preparation for ASoC Dynamic PCM (AKA DSP) support. Provide convenience methods to retrieve the soc_card or snd_card from a DAPM context. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20ASoC: SAMSUNG: Add I2S0 internal dma driverSangbeom Kim3-0/+481
I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. Signed-off-by: Sangbeom Kim <sbkim73@samsung.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20ASoC: SAMSUNG: Modify I2S driver to support idmaSangbeom Kim1-0/+7
Previously, I2S driver only can support system dma. In this patch, i2s driver can support internal dma too. IDMA h/w configuration is initialized on idma.c Signed-off-by: Sangbeom Kim <sbkim73@samsung.com> Acked-by: Liam Girdwood <lrg@ti.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20ASoC: davinci: add missing break statementRajashekhara, Sudhakar1-0/+1
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> Cc: stable@kernel.org
2011-07-20ASoC: davinci: fix codec start and stop functionsRajashekhara, Sudhakar1-4/+4
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> Cc: stable@kernel.org
2011-07-20ASoC: Acknowledge WM8962 interrupts before acting on themMark Brown1-3/+3
This closes the small race between a status being read in response to an interrupt and clearing the interrupt, meaning that if the status changes between those periods we might not get a reassertion of the interrupt. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-19ASoC: sgtl5000: guide user when regulator support is neededWolfram Sang1-0/+1
Print a hint when the user has a setup where CONFIG_REGULATOR is really needed to make the driver work. Signed-off-by: Wolfram Sang <w.sang@pengutronix.de> Tested-by: Dong Aisheng <b29396@freescale.com> Tested-by: Shawn Guo <shawn.guo@freescale.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19ASoC: sgtl5000: refactor registering internal ldoWolfram Sang1-38/+31
The code for registering the internal ldo was present twice. Turn it into a function instead. Also, inform the user if LDO is used now. Signed-off-by: Wolfram Sang <w.sang@pengutronix.de> Tested-by: Dong Aisheng <b29396@freescale.com> Tested-by: Shawn Guo <shawn.guo@freescale.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19ASoC: core: make comments fit the codeWolfram Sang1-2/+2
In one comment, cpu_dai was mentioned although codec_dai was used in the code. Also, fix the name for the card dai list which has no seperation into card_dai and codec_dai. Signed-off-by: Wolfram Sang <w.sang@pengutronix.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19ASoC: Mark cache as dirty when suspendingMark Brown1-0/+1
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> Cc: stable@kernel.org
2011-07-18Merge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6Linus Torvalds4-15/+16
* '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
2011-07-18ALSA: hda - Switch HP DAC dynamically with indep-HP switch for VIATakashi Iwai1-34/+125
This patch changes the behavior of independent-HP enum switch. Now instead of returning a busy error, the driver switches dynamically the stream of the HP (and shared) DACs according to the current mode. The logic is similar like the dual-mic ADC switch, but a bit more complicated because of the presence of shared DAC. Together with the change, a mutex is introduced to protect against the possible races for the indep-HP mode setting. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-18ALSA: hda - Implement dynamic loopback control for VIA codecsTakashi Iwai1-140/+260
This patch adds the dynamic control of analog-loopback for VIA codecs. When the loopback is enabled, the inputs from line-ins and mics are mixed with the front DAC, and sent to the front outputs. The very same input is routed to the headhpones and speakers in loopback mode. However, since the loopback mix can't take other than the front DAC, there is no longer individual volume controls for headphones and speakers. Once when the loopback control is off, these volumes take effect. Since the individual volumes are more desired in general use caess, the loopback mode is set to off as default for now. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-18ALSA: virtuoso: fix silent analog output on Xonar Essence ST DeluxeClemens Ladisch1-1/+4
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> Cc: 2.6.38+ <stable@kernel.org>
2011-07-17Merge branches 'omap/prcm' and 'omap/mfd' of git+ssh://master.kernel.org/pub/scm/linux/kernel/git/arm/linux-arm-soc into next/devel-2Arnd Bergmann7-582/+351
2011-07-17Merge branch 'for-3.0' into for-3.1Mark Brown1-3/+4
2011-07-17ASoC: Correct WM8994 MICBIAS supply widget hookupMark Brown1-3/+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
2011-07-17ASoC: Don't use -1 to boostrap subseq so it can be used by driversMark Brown1-1/+1
Makes life a little easier if you want to add subsequences to an existing driver as you can use -1 to put things at the start of sequences. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-17ASoC: Reduce power consumption for idle DAIs in WM8994Mark Brown1-0/+29
If DAIs are idle but their clocks are in use for some reason (eg, as SYSCLK or for accessory detect) then set the clock dividers to the maximum to reduce slightly the power consumption of the unclocked circuits. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-17ASoC: Report an error for unknown adav80x formatsMark Brown1-2/+2
Not only fixes error handling but also some uninitialized variable warnings. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de>
2011-07-17ASoC: Handle failed WM8994 FLL lock waitsMark Brown1-0/+3
Try the completion before we start the FLL so that if an interrupt was delayed long enough for us to miss it we don't wait for the completion it signalled. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-15Merge branch 'fixes-for-arnd' of git://git.pengutronix.de/git/imx/linux-2.6 into imx/fixesArnd Bergmann24-80/+125
2011-07-15ASoC: Handle spurious wm_hubs DC servo done interruptsMark Brown1-14/+16
Don't assume the first fire indicates that we're done. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-15ASoC: WM8983: Initial driverDimitris Papastamos4-0/+2238
The WM8983 is a low power, high quality stereo CODEC designed for portable multimedia applications. Highly flexible analogue mixing functions enable new application features, combining hi-fi quality audio with voice communication. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-15Merge branch 'for-3.0' into for-3.1Mark Brown1-1/+1
2011-07-15ASoC: Fix shift in WM8958 accessory detection default implementationMark Brown1-1/+1
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
2011-07-15ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300Daniel T Chen1-0/+6
BugLink: https://bugs.launchpad.net/bugs/774895 The original reporter states that his volume keys do not change the desired Master and PCM mixer elements together, so apply the hp+mute led quirk for his PCI SSID. Reported-by: Jeffrey Finkelstein Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-14ALSA: hda - Fix krealloc() replacement in hda_codec.cTakashi Iwai1-1/+2
It was obviously wrong, grr.... Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-14ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecsTakashi Iwai1-0/+3
During the rewrite, the check of spec->need_dac_fix and the corresponding num_dacs change was dropped from the channel-mode control. This patch re-adds it, and also enables need_dac_fix for ALC880 as default, as this feature was originally introduced to fix h/w bugs of this chip. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-14ASoC: wm8900: fix a memory leak if wm8900_set_fll failsAxel Lin1-0/+1
Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14ASoC: Log WM8994 FIFO errors from the interruptMark Brown1-0/+14
We should spot them anyway on state changes but logging them gives us better time information about when the misconfiguration happened. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14ASoC: SAMSUNG: 24-bit audio playback on Exynos4210Giridhar Maruthy1-1/+4
Using 256fs or 512fs will result in distortion of 24-bit audio samples. This is because the lrclk generated is not proper. Using 384 fs generates proper output. Signed-off-by: Giridhar Maruthy <giridhar.maruthy@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14ASoC: Don't warn on low WM8994/58 AIFnCLKsMark Brown1-4/+0
We can have valid but very low clocks in accessory detection modes. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Copyright © 1996 – 2023 Jason A. Donenfeld. All Rights Reverse Engineered.