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* Merge branch 'linux-omap-audio-3.0' into linux-omap-3.0Simon Wilson2011-07-144-16/+49
|\ | | | | | | Signed-off-by: Simon Wilson <simonwilson@google.com>
| * ASoC: sdp4430: Configure modem's McBSP port for BT callMisael Lopez Cruz2011-07-141-2/+16
| | | | | | | | | | | | | | | | | | | | | | | | Configuration of McBSP port connected to modem requires a special handling because it's not associated to any backend but to the ABE modem frontend. Backends need to explicitly configure modem's McBSP port, it was done for PDM backend but was missing for McBSP backend which is used during BT call. Change-Id: Icb433a09ebe868e148933bbc5df7fd5ed0bda83a Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
| * ASoC: sdp4430: Split BT backend DAI link for UL and DLMisael Lopez Cruz2011-07-142-3/+21
| | | | | | | | | | | | | | | | | | | | | | | | ASoC DSP requires exact stream name match to associate AIF widgets and BEs. Existing BT backend DAI link didn't match any of the BT backend widgets: BT_VX_UL (BT Capture) or BT_VX_DL (BT Playback), so it requires to split the original backend DAI link into two new links, one per each BT AIF widget as their stream name differ. Change-Id: I1902469d0641e61e5302c3ba7559d00df1b4585b Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
| * ASoC: ABE DSP: Fix compilation warningsMisael Lopez Cruz2011-07-141-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | Fix compilation warnings in ABE DSP driver: sound/soc/omap/omap-abe-dsp.c: In function 'abe_irq_pingpong_subroutine': sound/soc/omap/omap-abe-dsp.c:196: warning: unused variable 'abe' sound/soc/omap/omap-abe-dsp.c: In function 'abe_engine_probe': sound/soc/omap/omap-abe-dsp.c:2296: warning: unused variable 'k' Change-Id: I1337e95c2a7f22e4be17d987637f1b4f5b926398 Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
| * ASoC: ABE: Use 16-bits DMA for mono S16_LEMisael Lopez Cruz2011-07-141-9/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ABE handles audio data in 32-bit containers, most of the ABE formats use a 32-bit container for each sample. The exception is STEREO_16_16 format which uses 2 16-bit samples in a 32-bit container. DMA transactions using 32-bits data size is used for above cases. There is a limitation for 16-bits mono streams as every sample should use a separate container. Using 16-bit transactions for this particular scenario allows DMAing 16-bits consecutive samples to/from CPU into 32-bit containers from/to ABE CBPrs. Change-Id: I4e52765442cc2910f2930627f87c352698532e26 Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
* | Merge commit 'v3.0-rc7' into linux-omap-3.0Colin Cross2011-07-129-45/+68
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| * \ 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>
* | | | | Merge commit 'v3.0-rc6' into linux-omap-3.0Colin Cross2011-07-117-10/+14
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| * | | | 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 'linux-omap-audio-3.0' into linux-omap-3.0Simon Wilson2011-07-102-79/+138
|\ \ \ \ \ | | |_|_|/ | |/| | | | | | | | Signed-off-by: Simon Wilson <simonwilson@google.com>
| * | | | ASoC: McPDM: Fix race conditions during stream open/closeMisael Lopez Cruz2011-07-101-10/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | McPDM PM resources and ABE DSP configuration was done in prepare() callback which may not be executed during certain scenarios like stream open and close without audio being written/read. That created race conditions while closing the audio stream if no audio data was sent/received. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: PankajJindal <pankajjindal@ti.com>
| * | | | ASoC: OMAP ABE: Add context save/restoreSebastien Guiriec2011-07-101-69/+110
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add context save/restore to ABE DSP driver. Signed-off-by: Sebastien Guiriec <s-guiriec@ti.com> Signed-off-by: PankajJindal <pankajjindal@ti.com>
* | | | | Merge commit 'v3.0-rc5' into linux-omap-3.0Colin Cross2011-06-299-25/+43
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| * | | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-06-245-13/+5
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| | * | | ASoC: pxa-ssp: Correct check for stream presenceDaniel Mack2011-06-231-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Don't rely on the codec's channels_min information to decide wheter or not allocate a substream's DMA buffer. Rather check if the substream itself was allocated previously. Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | * | | ASoC: imx: add missing module informationsArnaud Patard (Rtp)2011-06-232-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | - add some modules aliases - add module license to avoid tainted kernel when loading the imx-pcm-audio driver Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org> Acked-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entryArnaud Patard (Rtp)2011-06-231-7/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | SND_MXC_SOC_SSI looks to be unused, so kill it. Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org> Acked-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O busMark Brown2011-06-131-3/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This will be removed in -next so let's drop it from mainline as soon as we can in order to minimise surprises. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ALSA: Remove unneeded version.h includes from sound/Jesper Juhl2011-06-242-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the sound/ directory there are two files (flagged by 'make versioncheck'); sound/pci/asihpi/asihpi.c and sound/soc/codecs/wm8991.c that include linux/version.h although they don't need it. This patch removes the unneeded includes. Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: HDA: Pinfix quirk for HP Z200 WorkstationDavid Henningsson2011-06-231-0/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BIOS lists the internal speaker as an internal line-out. Change to internal speaker + model=auto for better auto-mute capabilities. BugLink: http://bugs.launchpad.net/bugs/754964 Reported-by: Marc Legris <marc.legris@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: VIA HDA: Create a master amplifier control for VT1718S.Lydia Wang2011-06-211-0/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Create a master volume and mute control of playback for VT1718S. Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S.Lydia Wang2011-06-211-1/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When switch HP independent mode, mute/unmute connctions of mixer which is connected to headphone for VT1718S. Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: VIA HDA: Modify initial verbs list for VT1718S.Lydia Wang2011-06-211-7/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove some invalid initial verbs and correct some wrong initial verbs for VT1718S codec. Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: hda - Remove ALC268 model override for CPR2000Tony Vroon2011-06-211-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER. This keeps headphone automute and microphone input from operating on at least one laptop from Opti Systems. Without the override, the BIOS parser does a fine job setting the card up and everything works. Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com> Signed-off-by: Tony Vroon <tony@linx.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: HDA: Remove quirk for an HP deviceDavid Henningsson2011-06-201-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The reporter, who is running kernel 2.6.38, reports that he needs to set model=auto for the headphone output to work correctly. BugLink: http://bugs.launchpad.net/bugs/761022 Cc: stable@kernel.org (v2.6.38+) Reported-by: Jo Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'linux-omap-audio-3.0' into linux-omap-3.0Simon Wilson2011-06-244-23/+101
|\ \ \ \ \ | | |_|/ / | |/| | | | | | | | Signed-off-by: Simon Wilson <simonwilson@google.com>
| * | | | ASoC: ABE DAI: support 2 channels for MODEM DAIFrancois Mazard2011-06-241-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | MODEM DAI needs to support 2 channels for dual-mic on capture and to set the McBSP interface as I2S. Signed-off-by: Francois Mazard <f-mazard@ti.com>
| * | | | ASoC: ABE HAL: Fix CMEM address for write_gainMisael Lopez Cruz2011-06-241-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix CMEM offset used for omap_abe_write_gain(), the address must be aligned to 32-bits. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
| * | | | ASoC: ABE DAI: Call modem trigger in bespoke_trigger()Misael Lopez Cruz2011-06-241-1/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Modem DAI is a special case because requires explicit ALSA PCM calls to the DAI associated with the serial port (e.g. McBSP). The trigger() call was missing and the serial port was not initiating any transfer. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
| * | | | ASoC: SDP4430: Update McBSP configuration for modem usecaseSebastien Guiriec2011-06-241-17/+79
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The configuration of McBSP is not updated when McBSP is not a backend, for example when connected to VX_DL/VX_UL frontends for modem usecases. To ensure McBSP port is properly configured in those cases, we need to explicitly set its parameters when the actual backend is configured. Signed-off-by: Francois Mazard <f-mazard@ti.com> Signed-off-by: Sebastien Guiriec <s-guiriec@ti.com>
| * | | | ARM: OMAP4: Enable threshold mode for OMAP4 McBSPMisael Lopez Cruz2011-06-241-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Threshold mode is also available for McBSP in OMAP4, FIFO size is 128x32-bit words for all interfaces. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
generated by cgit v1.1 at 2025-03-22 21:14:13 +0000