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* | ASoC: OMAP ABE: Update ports in order to support 44.1 KHzSebastien Guiriec2014-10-012-5/+2
| | | | | | | | | | | | | | | | | | | | With new ABE release the driver can support now 44.1 on MM DL and TONES DL port. For MM DL port a true interpolation is done. For TONES DL the ratio 12/11 is used. Change-Id: Ic433e1576e7c86b1477c1854816caf5cefdceeed Signed-off-by: Sebastien Guiriec <s-guiriec@ti.com> Signed-off-by: Chris Kelly <c-kelly@ti.com>
* | HACK: ASoC: OMAP ABE: Save DL1 BE related gainsAxel Castaneda Gonzalez2014-10-011-0/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | MM_EXT_DL gains were being lost when ABE was hitting OFF, because they were not properly saved when DL1 last active BE was muted. Save DL1 BE gains along with ABE context in order to avoid losing their values when OFF is hit. Gain values are saved when gains are muted. These gains were not properly muted if BT_VX_DL or MM_EXT_DL port was enabled and DL2 path was active, since PDM_DL1 port gets enabled(for DL1 or DL2) when ABE McPDM is started. Change-Id: Ib0d658c65c134f7ee4baeb13f7e09f6299b1923a Signed-off-by: Axel Castaneda Gonzalez <x0055901@ti.com>
* | ASoC: DMIC: Implement Errata i653Misael Lopez Cruz2014-10-011-2/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Implement Errata i653: if the DMIC path is reset by SW_DMIC_RST, uplink FIFO is correctly reset but the dma pending signal is kept asserted. This can result in an invalid access by the host DMA to uplink FIFO. The workaround involves: - Enable channels: DMIC_CTRL.SW_DMIC_RST = 1 Enable necessary channels DMIC_CTRL.SW_DMIC_RST = 0 - Disable channels: DMIC_CTRL.SW_DMIC_RST = 1 Disable all channels DMIC_CTRL.SW_DMIC_RST = 0 Change-Id: I03766e32493322094fb48d9ae07a19d15695996a Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
* | [Blaze] ASoC: OMAP4 - DMIC DAI driverLiam Girdwood2014-10-014-0/+651
| | | | | | | | | | | | | | | | | | | | | | | | | | Add both legacy DMA and ABE support for Digital Microphones on the OMAP4 platform. Change-Id: Ib6681c51d78255ba94177821c570e4dd5021a243 Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Chris Kelly <c-kelly@ti.com> Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Conflicts: sound/soc/omap/Kconfig
* | ASoC: ABE HAL: Change ABE port priority table for BT voice callFrancois Mazard2014-10-011-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Up to now the VX ports had a higher priority than BT port. When voice call starts or stops, the ABE reference clock changes between Modem clock and ABE system clock. This change can produce noise on the multimedia tone played in the same time. To avoid that the BT needs to have the priority. Indeed, the ABE ref clock will stay at ABE system clock inside/outside voice call. As Vx ports have ASRC there's no problem of clock drift. Change-Id: Ie04fe6705e6c85fe3c83bf684717ae48a10a43e7 Signed-off-by: Francois Mazard <f-mazard@ti.com>
* | ASoC: ABE: update to ABE firmware 09.51.Sebastien Guiriec2014-10-0113-26871/+557
| | | | | | | | | | | | | | | | | | | | | | | | | | New features: - SRC 44.1 to 48 for MM_DL port (CBPr or Ping/Pong) - Split monitoring of ABE filters saturation - Table for filter saturation 8 or 16 kHz Change-Id: I469a477612d4c927516e2bc96ea07d672bfcf3e0 Signed-off-by: Sebastien Guiriec <s-guiriec@ti.com> Signed-off-by: Chris Kelly <c-kelly@ti.com> Signed-off-by: Barry Woodward <b-woodward@ti.com> Signed-off-by: Gabriel M. Beddingfield <gabrbedd@ti.com>
* | OMAP:ABE: Fix schedule tablet formattingDan Murphy2014-10-011-19/+19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Fixed the formatting of the code when the MultiFrame is set to 0 change = 0/*...*/; to = 0; /*...*/ No operational changes are made to this file Change-Id: I6b49e0f66f2fad16e540968c6ca926e450c60bf5 Signed-off-by: Dan Murphy <dmurphy@ti.com>
* | ASoC: OMAP4 ABE: Allow an alternate ABE firmware to be used.Gabriel M. Beddingfield2014-10-015-5/+25999
| | | | | | | | | | | | | | | | | | | | | | | | | | | | On commit 52c46046 the ABE firmware was updated with a version that included several improvements, but also disabled the digital microphone (DMIC) feature. In order to support DMIC on devices like the Blaze, an alternate firmware (and matching code) must be configured. This commit allows an alternate firmware using CONFIG_SND_OMAP4_ABE_USE_ALT_FW. Change-Id: I2fd4aa8d03be802722dca973a71435b9965e80ea Signed-off-by: Gabriel M. Beddingfield <gabrbedd@ti.com>
* | ASoC: OMAP4 ABE: Add CONFIG_SND_OMAP4_ABE_USE_ALT_FWGabriel M. Beddingfield2014-10-011-0/+11
| | | | | | | | | | | | | | This enables a build to load the alternate ABE Firmware. Change-Id: I5772616c576234845d0e839067d5b85f232f12be Signed-off-by: Gabriel M. Beddingfield <gabrbedd@ti.com>
* | ASoC: OMAP ABE: Restore defaults for DMA paramsGabriel M. Beddingfield2014-10-011-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The static struct omap_abe_dai_dma_params is used to both set defaults and serve as the omap_pcm_dma_data instance for the driver. In some cases, the values in this struct are overridden to suit the end-use. When these overrides happen, there was nothing to set them back to the original default. This fixes an issue where single-channel headset (AMic) recording causes subsequent DMic recordings to be rendered at double-speed. This happened because the the dma data_type was being set to OMAP_DMA_DATA_TYPE_S16 in the static struct omap_abe_dai_dma_params. After being set, the value was never restored to OMAP_DMA_DATA_TYPE_S32 (needed by DMic recording). Change-Id: Iea2b19d451edc10ab864ca6c1a2b6b3449b05bea Signed-off-by: Gabriel M. Beddingfield <gabrbedd@ti.com>
* | omap: omap-mcbsp: fix compilation warningsLeed Aguilar2014-10-011-1/+2
| | | | | | | | | | | | | | | | In function 'omap_mcbsp_dai_set_dai_sysclk': warning: suggest explicit braces to avoid ambiguous 'else' Change-Id: Ic273b811371ae12ae76097cc709327443690d39b Signed-off-by: Leed Aguilar <leed.aguilar@ti.com>
* | ASoC: ABE HAL: Set proper physical ID for MM_EXT portAxel Castaneda Gonzalez2014-10-011-2/+2
| | | | | | | | | | | | | | | | Correct logical to physical port mapping for MM_EXT ports. Change-Id: Ia617ec5cc9fcba85bcdedbb29aa3fe8ce9307985 Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Axel Castaneda Gonzalez <x0055901@ti.com>
* | ASoC: McPDM: Remove extra CTRL register clearMisael Lopez Cruz2014-10-011-5/+0
| | | | | | | | | | | | | | | | | | | | | | Reset value of MCPDM_CTRL is 0, so there is no need to do an extra clear write during McPDM DAIs probe(). One important advantage of removing this extra register access is that it removes the dependency on the McPDM clock which is supplied externally. Change-Id: I8bcad0494cc389523172b0f9ae5834e0f0f96326 Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
* | ASoC: McPDM: Update ABE OPP constraint for McPDMSebastien Guiriec2014-10-011-2/+2
| | | | | | | | | | | | | | | | | | With ABE release 09.50 both UL and DL task for McPDM are at OPP 25 So we just need to set the constrain to OPP 25 instead of 50. Change-Id: Ic250678085d0b763993d3be8f540bc0cf836924e Signed-off-by: Sebastien Guiriec <s-guiriec@ti.com> Signed-off-by: Chris Kelly <c-kelly@ti.com>
* | ASoC: OMAP: lock mutex in aess_hw_paramsFrancois Mazard2014-10-011-1/+5
| | | | | | | | | | | | | | | | | | | | aess_hw_params() calls subroutines that manipulate registers and buffers in the ABE. Therefore it needs exclusive access to the ABE to prevent corruption. Change-Id: I5a8f62a818168bb402dcf48f619e993a9ee26df7 Signed-off-by: Francois Mazard <f-mazard@ti.com> Signed-off-by: Gabriel M. Beddingfield <gabrbedd@ti.com>
* | OMAP4470: HDMI: Audio MCLK support for 4470Leonid Iziumtsev2014-10-011-1/+1
| | | | | | | | | | | | | | | | Add runtime checking during core audio configuration to enable MCLK usage. Change-Id: Ia8a3032c3fe5338980b7af98b4bd42ed814c715e Signed-off-by: Leonid Iziumtsev <x0153368@ti.com>
* | ASoC: twl6040: Convert PLUGINT to no-suspend irqMisael Lopez Cruz2014-09-301-1/+1
| | | | | | | | | | | | | | | | | | | | | | During suspend/resume cycle, PLUGINT could be processed while corresponding irq has not being enabled back, so PLUGINT handler is skipped. Converting PLUGINT to no-suspend type makes this interrupt to be handled properly during resume as it's kept enabled during suspend. Change-Id: I38c344113816afcf05634cb92cdd0d81a908b088 Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
* | MFD: twl6040-codec: Allow passing flags to IRQ requestMisael Lopez Cruz2014-09-301-2/+2
| | | | | | | | | | | | | | Allow passing irqflags to TWL6040 IRQ request API (twl6040_request_irq). Change-Id: I731931c4f8fc9aee57bb3707c94f7b763e140c0a Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
* | Merge branch 'android-omap-3.0' into android-omap-tuna-3.0Todd Poynor2013-03-131-2/+2
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| * \ Merge branch 'android-3.0' into android-omap-3.0Todd Poynor2013-03-131-2/+2
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| | * ASoC: wm2000: Fix sense of speech clarity enableMark Brown2013-01-171-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | commit 267f8fa2e1eef0612b2007e1f1846bcbc35cc1fa upstream. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* | | Merge branch 'android-omap-3.0' into android-omap-tuna-3.0Todd Poynor2013-01-166-8/+25
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| * | Merge branch 'android-3.0' into android-omap-3.0Todd Poynor2013-01-166-8/+25
| |\ \ | | |/ | | | | | | | | | | | | | | | Fixup incompatible code in opp.c: omap_init_opp_table() Change-Id: Iac7d60b814a539285d00e0a3dbb6e3f0060cb683 Signed-off-by: Todd Poynor <toddpoynor@google.com>
| | * ASoC: dapm: Use card_list during DAPM shutdownMisael Lopez Cruz2012-11-261-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 445632ad6dda42f4d3f9df2569a852ca0d4ea608 upstream. DAPM shutdown incorrectly uses "list" field of codec struct while iterating over probed components (codec_dev_list). "list" field refers to codecs registered in the system, "card_list" field is used for probed components. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm8978: pll incorrectly configured when codec is masterEric Millbrandt2012-11-261-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 55c6f4cb6ef49afbb86222c6a3ff85329199c729 upstream. When MCLK is supplied externally and BCLK and LRC are configured as outputs (codec is master), the PLL values are only calculated correctly on the first transmission. On subsequent transmissions, at differenct sample rates, the wrong PLL values are used. Test for f_opclk instead of f_pllout to determine if the PLL values are needed. Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm9712: Fix name of Capture SwitchMark Brown2012-10-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 689185b78ba6fbe0042f662a468b5565909dff7a upstream. Help UIs associate it with the matching gain control. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm9712: Fix microphone source selectionMark Brown2012-09-141-2/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit ccf795847a38235ee4a56a24129ce75147d6ba8f upstream. Currently the microphone input source is not selectable as while there is a DAPM widget it's not connected to anything so it won't be properly instantiated. Add something more correct for the input structure to get things going, even though it's not hooked into the rest of the routing map and so won't actually achieve anything except allowing the relevant register bits to be written. Reported-by: Christop Fritz <chf.fritz@googlemail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm8994: Ensure there are enough BCLKs for four channelsMark Brown2012-08-091-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit b8edf3e5522735c8ce78b81845f7a1a2d4a08626 upstream. Otherwise if someone tries to use all four channels on AIF1 with the device in master mode we won't be able to clock out all the data. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm8962: Allow VMID time to fully rampMark Brown2012-08-091-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 9d40e5582c9c4cfb6977ba2a0ca9c2ed82c56f21 upstream. Required for reliable power up from cold. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: tlv320aic3x: Fix codec pll configure bugHebbar, Gururaja2012-07-162-3/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit c9fe573a6584034670c1a55ee8162d623519cbbf upstream. In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently(like its stretched). Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* | | ASoC: OMAP4: HDMI: add mixer control to get LPCM channelsSimon Wilson2012-06-051-6/+71
| | | | | | | | | | | | | | | | | | | | | | | | Also fix a potential race condition where the state could have changed during the sleep. Change-Id: I189d0cd17ea56f5e0c2315061ab18acfcff2c91c Signed-off-by: Simon Wilson <simonwilson@google.com>
* | | Merge branch 'android-omap-3.0' into android-omap-tuna-3.0Todd Poynor2012-05-081-0/+2
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| * | Merge branch 'android-3.0' into android-omap-3.0Todd Poynor2012-05-081-0/+2
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| | * ASoC: dapm: Ensure power gets managed for line widgetsMark Brown2012-05-071-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 7e1f7c8a6e517900cd84da1b8ae020f08f286c3b upstream. Line widgets had not been included in either the power up or power down sequences so if a widget had an event associated with it that event would never be run. Fix this minimally by adding them to the sequences, we should probably be doing away with the specific widget types as they all have the same priority anyway. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* | | Merge branch 'android-omap-3.0' into android-omap-tuna-3.0Todd Poynor2012-05-011-6/+24
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| * | ASoC: OMAP: HDMI: Defer audio transfer startAxel Castaneda Gonzalez2012-05-011-6/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Defer audio transfer after HDMI AUDIO wrapper is enabled. If audio transmit was started along with audio wrapper enabling, spurious data (zeros) was sent at the beginning of the transfer as part of the of audio sample packets, due an AUDIO FIFO UNDERFLOW, which was shifting audio channels. Change-Id: I48d8c02c0467dd3158ac748eb0720173d3b209ca Signed-off-by: Axel Castaneda Gonzalez <x0055901@ti.com>
* | | Merge branch 'android-omap-3.0' into android-omap-tuna-3.0Todd Poynor2012-04-1918-82/+165
|\ \ \ | |/ / | | | | | | | | | Change-Id: I336ba4d61beba035ecc237ccc68b012e48702a87 Signed-off-by: Todd Poynor <toddpoynor@google.com>
| * | Merge linux-stable 3.0.28 into linux-omap-3.0Todd Poynor2012-04-1918-91/+175
| |\ \ | | |/ | | | | | | | | | Change-Id: I76904a60370e2cb9cc29ccde5d526d9183ff4f8e Signed-off-by: Todd Poynor <toddpoynor@google.com>
| | * ASoC: ak4642: fixup: mute needs +1 stepKuninori Morimoto2012-04-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 1f99e44cf059d2ed43c5a0724fa738b83800f725 upstream. ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute. But current settings didn't care +1 step for mute. This patch adds it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: pxa-ssp: atomically set stream active masksDaniel Mack2012-04-021-25/+36
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 273b72c8ce6b28df6b49423d775c3e59072c73c5 upstream. PXA's SSP engine fails to take its current channel phase into account when enabling a stream while the engine is already running. This results in randomly swapped left/right channels on either the record or the playback side, depending on which one was enabled first. The following patch fixes this by factoring out the bit field modifications in question to a separate function that pauses the engine temporarily, modifies the bits and kicks it off again afterwards. Appearantly, a transition of SSCR0_SSE syncs both directions properly. The patch has been rolled out to quite a number of devices over the last weeks and seems to fix the issue reliably. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Sven Neumann <s.neumann@raumfeld.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: neo1973: fix neo1973 wm8753 initializationDenis 'GNUtoo' Carikli2012-03-191-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit b2ccf065f7b23147ed135a41b01d05a332ca6b7e upstream. The neo1973 driver had wrong codec name which prevented the "sound card" from appearing. Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo@no-log.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: i.MX SSI: Fix DSP_A format.Javier Martin2012-03-121-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 5ed80a75b248bfaf840ea6b38f941edcf6ee7dc7 upstream. According to i.MX27 Reference Manual (p 1593) TXBIT0 bit selects whether the most significant or the less significant part of the data word written to the FIFO is transmitted. As DSP_A is the same as DSP_B with a data offset of 1 bit, it doesn't make any sense to remove TXBIT0 bit here. Signed-off-by: Javier Martin <javier.martin@vista-silicon.com> Acked-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: dapm: Check for bias level when powering downMark Brown2012-03-121-3/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 7679e42ec833ed70aa34790a5f39dcb7e5bda4fe upstream. Recent enhancements in the bias management means that we might not be in standby when the CODEC is idle and can have active widgets without being in full power mode but the shutdown functionality assumes these things. Add checks for the bias level at each stage so that we don't do transitions other than the ON->PREPARE->STANDBY->OFF ones that the drivers are expecting. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm8962: Fix sidetone enumeration textsMark Brown2012-02-291-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 31794bc37bf2db84f085da52b72bfba65739b2d2 upstream. The sidetone enumeration texts have left and right swapped. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm8962: Fix word length configurationSusan Gao2012-02-131-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | commit 2b6712b19531e22455e7fa18371c5ba9eec76699 upstream. Signed-off-by: Susan Gao <sgao@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm_hubs: Correct line input to line output 2 pathsMark Brown2012-02-131-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 43b6cec27e1e50a1de3eff47e66e502f3fe7e66e upstream. The second line output mixer has the controls for the line input bypasses in the opposite order. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm_hubs: Fix routing of input PGAs to line output mixerMark Brown2012-02-131-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit ee76744c51ec342df9822b4a85dbbfc3887b6d60 upstream. IN1L/R is routed to both line output mixers, we don't route IN1 to LINEOUT1 and IN2 to LINEOUT2. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: Ensure we generate a driver nameMark Brown2012-02-131-3/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit f0e8ed858edb327802ee65fd695cc1538286226f upstream. Commit 873bd4c (ASoC: Don't set invalid name string to snd_card->driver field) broke generation of a driver name for all ASoC cards relying on the automatic generation of one. Fix this by using the old default with spaces replaced by underscores. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm_hubs: fix wrong bits for LINEOUT2 N/P mixerUK KIM2012-02-131-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | commit 114395c61ad2eb5a7a5cd163fcadb2414e48245a upstream. Signed-off-by: UK KIM <w0806.kim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
| | * ASoC: wm_hubs: Enable line out VMID buffer for single ended line outputsMark Brown2012-02-131-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 77231abe55433aa17eca712718745275853fa66d upstream. For optimal performance the single ended line outputs require that the line output VMID buffer be enabled. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
generated by cgit v1.1 at 2025-03-21 00:58:03 +0000