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* Merge branch 'upstream/wm8711' into for-2.6.33Mark Brown2009-09-21
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| * ASoC: Add TLV information to WM8711Mark Brown2009-08-18
| | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: WM8711 minor cleanupsMark Brown2009-08-18
| | | | | | | | | | | | Coding style changes only. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: Add SPI support to WM8711Mark Brown2009-08-18
| | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: Factor out WM8711 cache I/OMark Brown2009-08-18
| | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * Merge branch 'topic/asoc' of ↵Mark Brown2009-08-18
| |\ | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into upstream/wm8711
| * | ASoC: Update WM8711 to driver model registration methodMark Brown2009-08-18
| | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: Add WM8711 CODEC driverMike Arthur2009-08-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8711 or WM8711L (WM8711/L) is a low power stereo DAC with an integrated headphone driver. The WM8711/L is designed specifically for portable MP3 audio and speech players. The WM8711/L is also ideal for MD, CD machines and DAT players. Signed-off-by: Mike Arthur <Mike.Arthur@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Convert soc-cache to use C99 style initialisers for the tableMark Brown2009-09-21
| | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Support WM8580 based audio subsystem on SMDK64xx machinesjassi brar2009-09-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | New machine driver for WM8580 I2S i/f on SMDK64XX. By default SoC-Slave is set and WM8580 is configured to use it's PLLA to generate clocks from a 12MHz crystal attached to WM8580. [Added dependency on BROKEN since the IISv4 interface hasn't been merged yet, fixed the PLL API usage and removed the disabling of the PLL in the hw_free function since that'll break simultaneous playback and record -- broonie.] Signed-off-by: Jassi <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-09-18
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| * | | ASoC: DaVinci: Fixes to McASP configurationChaithrika U S2009-09-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | McASP register settings are not correct for DSP mode of operation. There is a channel swap initally. This patch provides fixes to the register values for proper working. Tested on DA830/OMAP-L137 EVM, DM6467 EVM. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Blackfin I2S: fix resuming when device hasn't been usedCliff Cai2009-09-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If the sound system hasn't been utilized yet and we suspend, then we attempt to save/restore using state that doesn't exist. So use a global handle instead to reconfigure properly. Signed-off-by: Cliff Cai <cliff.cai@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Blackfin I2S: add lost platform_device parameter to resume functionBarry Song2009-09-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit dc7d7b830ee1 trimmed the platform_device parameter from all of the suspend functions, but it also accidentally removed it from the resume function in the Blackfin I2S driver. So restore it. Signed-off-by: Barry Song <barry.song@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: fix typos in Blackfin headersBarry Song2009-09-17
| | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Barry Song <barry.song@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned longMike Frysinger2009-09-17
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Blackfin AC97: add a few missing multichannel define handlingCliff Cai2009-09-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was merged, but two places were missed (the probe/resume functions). Restore handling of this option so it gets initialized properly. Signed-off-by: Cliff Cai <cliff.cai@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: remove unused #include <linux/version.h>Huang Weiyi2009-09-16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove unused #include <linux/version.h>('s) in sound/soc/codecs/ad1836.c sound/soc/codecs/ad1938.c sound/soc/codecs/wm8974.c Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: S3C lrsync function made to work with IRQs disabled.Jassi2009-09-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | s3c2412_snd_lrsync() maybe reached with IRQs disabled and if LRCLK is dead due to improper initialization of CPU or CODEC, the system gets stuck in the loop because jiffies may never get updated. Implemented counter based wait mechanism for atleast the same timeout period. Signed-off-by: Jassi <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Fix display of stream name in DAPM debugfsMark Brown2009-09-14
| | | | | | | | | | | | | | | | | | | | | | | | Also display streams all the time while we're here. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Clean up error handling in MPC5200 DMA setupJulia Lawall2009-09-12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Error handling code following a kzalloc should free the allocated data. Error handling code following an ioremap should iounmap the allocated data. The semantic match that finds the first problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @r exists@ local idexpression x; statement S; expression E; identifier f,f1,l; position p1,p2; expression *ptr != NULL; @@ x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...); ... if (x == NULL) S <... when != x when != if (...) { <+...x...+> } ( x->f1 = E | (x->f1 == NULL || ...) | f(...,x->f1,...) ) ...> ( return \(0\|<+...x...+>\|ptr\); | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // </smpl> Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: au1x: PSC-AC97 bugfixesManuel Lauss2009-09-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Fix WM835x Out4 capture enumerationMark Brown2009-09-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It's the 8th enum of a zero indexed array. This is why I don't let new drivers use these arrays of enums... Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
* | | | ASoC: Return correct codec clock in s3c64xx-i2sJassi2009-09-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Instead of always returnig pointer to the 'audio-bus' clock, check which clock is used to generate internal clocks and then return it's pointer. Signed-off-by: Jassi <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Add S3C64xx IIS CDCLK source selectionMark Brown2009-09-16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | CDCLK can either be an output generated by the CPU, intended for use as the CODEC master clock, or an input (probably from the CODEC) providing a master clock for the IIS block. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Davinci: Add audio codec support for DM365 EVMMiguel Aguilar2009-09-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch enables tlv320aic3101 support on DM365 EVM and it was tested on DM365 EVM rev c. Note: this patch was created based on temp/asoc branch. Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836Barry Song2009-09-15
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Barry Song <barry.song@analog.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: S3C I2S LRCLK polarity option.Jassi2009-09-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 1) Explicitly set LRCLK polarity for I2S Vs LSM/MSB modes. 2) Convert from numerical to bit-field values for BCLK selection. 3) Use proper error checking for return value from clk_get Signed-off-by: Jassi <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Provide API for reordering channelsBarry Song2009-09-13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The patch adds an interface to set the relationship between audio channel number and slot number. The interface should be really useful because audio channel n doesn't always use slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: AK4671: add ak4671 codec driverJoonyoung Shim2009-09-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier, Receiver-Amplifier and Headphone-Amplifier. The datasheet for the ak4671 can find at the following url: http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Allow per-route connectedness checks for suppliesMark Brown2009-09-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some chips with complex internal supply (particularly clocking) arragements may have multiple options for some of the supply connections. Since these don't affect user-visible audio routing the expectation would be that they would be managed automatically by one of the drivers. Support these users by allowing routes to have a connected function which is queried before the connectedness of the path is checked as normal. Currently this is only done for supplies, other widgets could be supported but are not currently since the expectation for them is that audio routing will be under the control of userspace. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Factor out I2C 8 bit address 8 bit data I/OJoonyoung Shim2009-09-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch is for the AK4671 codec driver using this format. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Add source argument to PLL configurationMark Brown2009-09-05
|/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | More and more devices feature PLLs and FLLs with the ability to select between multiple input clocks. In order to better support these devices a new argument, source, has been added to the set_pll() configuration API. Using set_clkdiv() is often difficult due to the need to stop the PLL/FLL before any reconfiguration can be done. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Fully specify DC servo bits to update in wm_hubsMark Brown2009-09-02
| | | | | | | | | | | | | | | | | | | | | Avoids potential issues if we read back unexpected values during a read/modify/write cycle. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Debugged improper setting of PLL fields in WM8580 driverjassi brar2009-09-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Bug was caught while trying to use WM8580 as I2S master on SMDK. Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved by referring to WM8580A manual and setting mask value correctly and making the code to not touch 'reserved' bits of PLL4 register. Signed-off-by: Jassi <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: new board driver to connect bfin-5xx with ad1836 codecBarry Song2009-09-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | As discussed, the patch uses the original TDM order without rewriting. For the match between TDM slot number and audio channel number, a new API need be added. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAIJarkko Nikula2009-08-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The McBSP1 port in OMAP3 processors (I believe OMAP2 too but I don't have specifications to check it) have additional CLKR and FSR pins for McBSP1 receiver. Reset default is that receiver is using bit clock and frame sync signal from those pins but it is possible to configure to use also CLKX and FSX pins as well. In fact, other McBSP ports are doing that internally that transmitter and receiver share the CLKX and FSX. Add functionaly that machine drivers can set the CLKR and FSR sources by using the snd_soc_dai_set_sysclk. Thanks to "Aggarwal, Anuj" <anuj.aggarwal@ti.com> for reporting the issue. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: davinci: i2c device creation moved into board filesChaithrika U S2009-08-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | Also, the codec setup data structure has to remain for successful probe. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Don't reconfigure WM8350 FLL if not neededMark Brown2009-08-26
| | | | | | | | | | | | | | | | | | | | | If the requested FLL configuration is the one we're currently running in it's at best pointless to reconfigure the FLL. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Fix s3c-i2s-v2 buildMark Brown2009-08-26
| | | | | | | | | | | | | | | | | | We now need the PCM header to kick the DMA. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: Make platform data optional for TLV320AIC3xMark Brown2009-08-26
| | | | | | | | | | | | | | | | | | | | | | | | Now that we don't need the I2C address for the device the platform data is redundant so allow it to be omitted. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Tested-by: Chaithrika U S <chaithrika@ti.com>
* | | ASoC: Add S3C24xx dependencies for Simtec machinesMark Brown2009-08-26
| | | | | | | | | | | | | | | | | | | | | No point in building them for S3C64xx, certainly no sense in running into build issues there. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: SDP3430: Fix TWL GPIO6 pin mux requestCandelaria Villareal, Jorge2009-08-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: S3C platform: Fix s3c2410_dma_started() called at improper timeShine Liu2009-08-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | s3c24xx dma has the auto reload feature, when the the trnasfer is done, CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So the transmission is repeated. IRQ is issued while auto reload occurs. We change the DISRC and DCON[19:0] in the ISR, but at this time, the auto reload has been performed already. The first block is being re-transmitted by the DMA. So we need rewrite the DISRC and DCON[19:0] for the next block immediatly after the this block has been started to be transported. The function s3c2410_dma_started() is for this perpose, which is called in the form of "s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger(). But it is not correct. DMA transmission won't start until DMA REQ signal arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in s3c24xx_i2s_trigger(). In the current framework, s3c24xx_pcm_trigger() is always called before s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in this function. However, s3c2410_dma_started() is dma related, to call this function we should provide the channel number, which is given by substream->runtime->private_data->params->channel. The private_data points to a struct s3c24xx_runtime_data object, which is define in s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c Fix this by moving the call to signal the DMA started to the DAI drivers. Signed-off-by: Shine Liu <liuxian@redflag-linux.com> Signed-off-by: Shine Liu <shinel@foxmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stopJarkko Nikula2009-08-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Functionality of functions omap_mcbsp_xmit_enable and omap_mcbsp_recv_enable can be merged into omap_mcbsp_start and omap_mcbsp_stop since API of those omap_mcbsp_start and omap_mcbsp_stop was changed recently allowing to start and stop individually the transmitter and receiver. This cleans up the code in arch/arm/plat-omap/mcbsp.c and in sound/soc/omap/omap-mcbsp.c which was the only user for those removed functions. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAIJarkko Nikula2009-08-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit ca6e2ce08679c094878d7f39a0349a7db1d13675 is setting up few XCCR and RCCR bits for I2S and DPS_A formats. Part of the bits are already set for all formats and I believe that XDISABLE and RDISABLE bits are format independent. As XCCR and RCCR are found only from OMAP2430 and OMAP34xx, I move setup of XDISABLE and RDISABLE to where those cpu's are tested and remove format dependent part for simplicity. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | Merge branch 'topic/digital-mixing' into for-2.6.32Mark Brown2009-08-24
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| * | | ASoC: WM8993 digital mixing supportMark Brown2009-08-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8993 provides digital sidetone paths and also allows each channel on the audio interface to be routed separtately to the DACs and ADCs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Add input and output AIF widgetsMark Brown2009-08-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently DAPM interfaces with the audio streams to and from the processor at the DAC and ADC widgets. As the digital capabilities of parts increases this is becoming a less and less able to meet the needs of parts. To meet the needs of these devices create new widgets interfacing with the TDM bus but not integrated into any other functionality. Audio can then be routed to and from these widgets using existing routing widgets. A slot number is provided in the definition but this is currently not used yet. This is intended to support devices which can use more than one TDM slot on a single interface. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Select core DMA when building for S3C64xxMark Brown2009-08-24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Ensure that the core DMA support is available when building for S3C64xx. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
generated by cgit v1.2.2 (git 2.25.0) at 2024-10-04 00:28:29 -0400