- 27 4月, 2012 4 次提交
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由 Liam Girdwood 提交于
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's necessary to allow some flexability wrt to PCM operations here so that we can define a bespoke DPCM trigger() PCM operation for such HW. A bespoke DPCM trigger() allows exact ordering and timing of component triggering by allowing a component driver to manage the final enable and disable configurations without adding extra complexity to other component drivers. e.g. The McPDM DAI and ABE are tightly coupled on OMAP4 so we have a bespoke trigger to manage the trigger to improve performance and reduce complexity when triggering new McPDM BEs. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
Some component drivers will need to be able to look up their DAI link substream and RTD data. Provide a mechanism for this. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
Add debugFS files for DPCM link management information. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
The Dynamic PCM core allows digital audio data to be dynamically routed between different ALSA PCMs and DAI links on SoC CPUs with on chip DSP devices. e.g. audio data could be played on pcm:0,0 and routed to any (or all) SoC DAI links. Dynamic PCM introduces the concept of Front End (FE) PCMs and Back End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that they can dynamically route digital audio data to any supported BE PCM. A BE PCM has no ALSA device, but represents a DAI link and it's substream and audio HW parameters. e.g. pcm:0,0 routing digital data to 2 external codecs. FE pcm:0,0 ----> BE (McBSP.0) ----> CODEC 0 +--> BE (McPDM.0) ----> CODEC 1 e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec. FE pcm:0,0 --- +--> BE (McBSP.0) ----> CODEC FE pcm:0,1 --- The digital audio routing is controlled by the usual ALSA method of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the routing based upon the mixer settings and configures the BE PCMs based on routing and the FE HW params. DPCM is designed so that most ASoC component drivers will need no modification at all. It's intended that existing CODEC, DAI and platform drivers can be used in DPCM based audio devices without any changes. However, there will be some cases where minor changes are required (e.g. for very tightly coupled HW) and there are helpers to support this too. Somethimes the HW params of a FE and BE do not match or are incompatible, so in these cases the machine driver can reconfigure any hw_params and make any DSP perform sample rate / format conversion. This patch adds the core DPCM code and contains :- o The FE and BE PCM operations. o FE and BE DAI link support. o FE and BE PCM creation. o BE support API. o BE and FE link management. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 24 4月, 2012 2 次提交
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由 Kristoffer KARLSSON 提交于
Added support for a control that strobes a bit in a register to high then back to low (or the inverse). This is typically useful for hardware that requires strobing a singe bit to trigger some functionality and where exposing the bit in a normal single control would require the user to first manually set then again unset the bit again for the strobe to trigger. Added convenience macro. SOC_SINGLE_STROBE Added accessor implementations. snd_soc_get_strobe snd_soc_put_strobe Signed-off-by: NKristoffer KARLSSON <kristoffer.karlsson@stericsson.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Kristoffer KARLSSON 提交于
Added control type that can span multiple consecutive codec registers forming a single signed value in a MSB/LSB manner. The control dynamically adjusts to the register word size configured in driver. Added convenience macro. SOC_SINGLE_XR_SX Added accessor implementations. snd_soc_info_xr_sx snd_soc_get_xr_sx snd_soc_put_xr_sx Signed-off-by: NKristoffer KARLSSON <kristoffer.karlsson@stericsson.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 17 4月, 2012 1 次提交
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由 Mark Brown 提交于
Rather than having the user half start a stream but avoid any DMA to trigger data flow on links which don't pass through the CPU create a DAPM route between the two DAI widgets using a hw_params configuration provided by the machine driver with the new 'params' member of the dai_link struct. If no configuration is provided in the dai_link then use the old style even for CODEC<->CODEC links to avoid breaking systems. This greatly simplifies the userspace usage of such links, making them as simple as analogue connections with the stream configuration being completely transparent to them. This is achieved by defining a new dai_link widget type which is created when CODECs are linked and triggering the configuration of the link via the normal PCM operations from there. It is expected that the bias level callbacks will be used for clock configuration. Currently only the DAI format, rate and channel count can be configured and currently the only DAI operations which can be called are hw_params and digital_mute(). This corresponds well to the majority of CODEC drivers which only use other callbacks for constraint setting but there is obviously much room for extension here. We can't simply call hw_params() on startup as things like the system clocking configuration may change at runtime and in future it will be desirable to offer some configurability of the link parameters. At present we are also restricted to a single DAPM link for the entire DAI. Once we have better support for channel mapping it would also be desirable to extend this feature so that we can propagate per-channel power state over the link. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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- 03 4月, 2012 1 次提交
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由 Brian Austin 提交于
Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: NBrian Austin <brian.austin@cirrus.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 01 4月, 2012 5 次提交
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由 Mark Brown 提交于
In version 3.4 the driver core acquired probe deferral which is a core way of doing essentially the same thing as ASoC has been doing since forever to make sure that all the devices needed to make up the card are present without needing open coding in the subsystem. Make basic use of this probe deferral mechanism for the cards, removing the need to handle partially instantiated cards. We should be able to remove even more code than this, though some of the checks we're currently doing should stay since they're about things like suppressing unneeded DAPM runs rather than deferring probes. In order to avoid robustness issues with our teardown paths (which do need quite a bit of TLC) add a check for aux_devs prior to attempting to set things up, this means that we've got a reasonable idea that everything will be there before we start. As with the removal of partial instantiation support more work will be needed to make this work neatly. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
Currently operations on jack reporting take the CODEC mutex both to protect the current jack status and also to protect the DAPM run which is triggered on status updates. Since the addition of a DAPM-specific lock we no longer need to worry about locking DAPM as it has its own finer grained lock so create a per jack lock to take care of the jack status. This is both cleaner where the jack isn't specifically associated with a CODEC and clearer as it's much more obvious what the lock is protecting. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
Change SND_SOC_CARD_CLASS_PCM to SND_SOC_CARD_CLASS_RUNTIME to better describe all uses for this mutex subclass and align with DAPM too. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
It has now become necessary to use a DAPM mutex instead of the codec mutex to lock the DAPM operations. This is due to the recent multi component support and forth coming Dynamic PCM updates. Currently we lock DAPM operations with the codec mutex of the calling RTD context. However, DAPM operations can span the whole card context and all components. This patch updates the DAPM operations that use the codec mutex to now use the DAPM mutex PCM subclass for all DAPM ops. We also add a mutex subclass for DAPM init and PCM operations. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
This is the first part of a change that is intended to improve ASoC locking protection for DAPM and PCM operations. This part of the series adds a mutex class for the soc_card mutex. The SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic PCM operations. The new mutex classes are required otherwise we will see a false positive mutex deadlock warning between the card initialisation and the PCM operations (something that would never deadlock in real life). Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 07 3月, 2012 1 次提交
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由 Liam Girdwood 提交于
Add mutex support for platform IO operations. e.g. can be used for platform DAPM widget IO ops. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 22 2月, 2012 2 次提交
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由 Mark Brown 提交于
Chip designers frequently include things like the enable and disable controls for algorithms in the register blocks which also hold the coefficients. Since it's desirable to split out the enable/disable control from userspace the plain SND_SOC_BYTES() isn't optimal for these devices. Add a SND_SOC_BYTES_MASK() which allows a bitmask from the first word of the block to be excluded from the control. This supports the needs of devices I've looked at and lets us have a reasonably simple API. Further controls can be added in future if that's needed. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
Allow devices to export blocks of registers to the application layer, intended for use for reading and writing coefficient data which can't usefully be worked with by the kernel at runtime (for example, due to requiring complex and expensive calculations or being the results of callibration procedures). Currently drivers are using platform data to provide configurations for coefficient blocks which isn't at all convenient for runtime management or configuration development. Currently only devices using regmap are supported, an error will be generated for any attempt to work with a byte control on a non-regmap device. There's no fundamental block to other devices so support could be added if required. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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- 18 2月, 2012 1 次提交
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由 Mark Brown 提交于
Neater and avoids warnings when used in other places where const strings are desired. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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- 16 2月, 2012 1 次提交
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由 Sebastien Guiriec 提交于
Allow platform widgets to be visible in debugfs like codec widgets. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 09 2月, 2012 1 次提交
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由 Mark Brown 提交于
This is usually not a use case dependant flag anyway. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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- 04 2月, 2012 1 次提交
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由 Liam Girdwood 提交于
Currently ASoC can only add kcontrols using codec and platform component device handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for SoC card machine drivers too. This allows the kcontrol to have a direct handle to the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily get it's private data. This change makes snd_soc_add_controls() static and wraps it in the folowing calls (card and dai are new) :- snd_soc_add_card_controls() snd_soc_add_codec_controls() snd_soc_add_dai_controls() snd_soc_add_platform_controls() This patch also does a lot of small mechanical changes in individual codec drivers to replace snd_soc_add_controls() with snd_soc_add_codec_controls(). It also updates the McBSP DAI driver to use snd_soc_add_dai_controls(). Finally, it updates the existing machine drivers that register controls to either :- 1) Use snd_soc_add_card_controls() where no direct codec control is required. 2) Use snd_soc_add_codec_controls() where there is direct codec control. In the case of 1) above we also update the machine drivers to get the correct component data pointers from the kcontrol (rather than getting the machine pointer via the codec pointer). Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 22 1月, 2012 1 次提交
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由 Mark Brown 提交于
If a driver is using regmap directly ensure that we're coherent with non-ASoC register updates by using the regmap API directly to do our read/modify/write cycles. This will bypass the ASoC cache but drivers using regmap directly should not be using the ASoC cache. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 20 1月, 2012 1 次提交
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由 Mark Brown 提交于
Most devices accept data in formats that don't correspond directly to their internal format. ALSA allows us to set a msbits constraint which tells userspace about this in case it finds it useful (for example, in order to avoid wasting effort dithering bits that will be ignored when raising the sample size of data) so provide a mechanism for drivers to specify the number of bits that are actually significant on a DAI and add the appropriate constraints along with all the others. This is done slightly awkwardly as the constraint is specified per sample size - we loop over every possible sample size, including ones that the device doesn't support and including ones that have fewer bits than are actually used, but this is harmless as the upper layers do the right thing in these cases. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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- 11 1月, 2012 1 次提交
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由 Mark Brown 提交于
The device model needs a release() function so it can free devices when they become dereferenced. Do that for rtds. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 23 12月, 2011 1 次提交
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由 Mark Brown 提交于
Ensure that everything is seeing the same declaration by moving it to a header file rather than putting the declaration in soc-core.c Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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- 22 12月, 2011 1 次提交
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由 Stephen Warren 提交于
DAI link endpoints and platform (DMA) devices are currently specified by name. When instantiating sound cards from device tree, it may be more convenient to refer to these devices by phandle in the device tree, and for code to describe DAI links using the "struct device_node *" ("of_node") those phandles map to. This change adds new fields to snd_soc_dai_link which can "name" devices using of_node, enhances soc_bind_dai_link() to allow binding based on of_node, and enhances snd_soc_register_card() to ensure that illegal combinations of name and of_node are not used. Signed-off-by: NStephen Warren <swarren@nvidia.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 20 12月, 2011 2 次提交
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由 Stephen Warren 提交于
Implement snd_soc_of_parse_audio_routing(), a utility function that can parses a simple DAPM route table from device tree.The machine driver specifies the DT property to use, since this is binding-specific. Signed-off-by: NStephen Warren <swarren@nvidia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Stephen Warren 提交于
Implement snd_soc_of_parse_card_name(), a utility function that sets a card's name from device tree. The machine driver specifies the DT property to use, since this is binding-specific. Signed-off-by: NStephen Warren <swarren@nvidia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 12月, 2011 1 次提交
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由 Mark Brown 提交于
All users now use regmap directly so delete the ASoC version of the code. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 02 12月, 2011 1 次提交
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由 Lars-Peter Clausen 提交于
The existence of this parameter is purely historical. None of the CODEC drivers uses it and we always pass in the same value anyway, so it should be safe to remove it. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 24 11月, 2011 1 次提交
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由 Stephen Warren 提交于
A card is fully routed if the DAPM route table describes all connections on the board. When a card is fully routed, some operations can be automated by the ASoC core. The first, and currently only, such operation is described below, and implemented by this patch. Codecs often have a large number of external pins, and not all of these pins will be connected on all board designs. Some machine drivers therefore call snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core never to activate them. However, when a card is fully routed, the information needed to derive the set of unused pins is present in card->dapm_routes. In this case, have the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused codec pin. This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c. Signed-off-by: NStephen Warren <swarren@nvidia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 10 11月, 2011 2 次提交
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由 Mark Brown 提交于
There are no current users and new drivers ought to be using the regmap API and its cache implementation directly so just delete the ASoC copy. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
My usual technique for finding definitions is to search for "name {" which breaks with the extra space. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 27 10月, 2011 1 次提交
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With this flag, each dai_link in machine driver can choose to ignore pmdown_time during DAPM shut down sequence. If the ignore_pmdown_time is set, the DAPM for corresponding DAI will be executed immediately. Signed-off-by: NRamesh Babu K V <ramesh.babu@linux.intel.com> Signed-off-by: NVinod Koul <vinod.koul@linux.intel.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 15 10月, 2011 1 次提交
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由 Peter Ujfalusi 提交于
With this flag codec drivers can indicate that it is desired to ignore the pmdown_time for DAPM shutdown sequence when playback stream is stopped. The DAPM sequence will be executed without delay in this case. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 07 10月, 2011 1 次提交
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由 Peter Ujfalusi 提交于
By accident few places still uses the _2r calls from the core. This is a quick fix, the drivers using the old callbacks going to be changed. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 06 10月, 2011 5 次提交
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由 Peter Ujfalusi 提交于
We do not have users for snd_soc_put_volsw_2r anymore. It can be removed. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Handle the put_volsw/put_volsw_2r in one function. To avoid build breakage in twl6040 keep the snd_soc_put_volsw_2r as define, and map it snd_soc_put_volsw. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Handle the get_volsw/get_volsw_2r in one function. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Handle the info_volsw/info_volsw_2r in one function. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
SOC_SINGLE/DOUBLE_VALUE is used for mixer controls, where the bits are within one register. Assign .rreg to be the same as .reg for these types. With this change we can tell if the mixer in question: is mono: mc->reg == mc->rreg && mc->shift == mc->rshift is stereo, within single register: mc->reg == mc->rreg && mc->shift != mc->rshift is stereo, in two registers: mc->reg != mc->rreg The patch provide a small inline function to query, if the mixer is stereo, or mono. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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