- 14 4月, 2011 17 次提交
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由 Lars-Peter Clausen 提交于
Use the newly introduced dapm_widgets, dpam_routes and controls fields of the snd_soc_dai_driver struct to setup controls and DAPM. 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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由 Lars-Peter Clausen 提交于
Use SND_SOC_DAPM_EVENT_OFF for determining whether the speaker should be turned on or off instead of open coding 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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由 Lars-Peter Clausen 提交于
This patch changes the qi_lb60 setup code to use gpio_request_array instead of manually calling gpio_request and gpio_direction_output for each gpio. Doing so makes the code a bit more compact. 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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由 Lars-Peter Clausen 提交于
Use the newly introduced dapm_widgets, dpam_routes and fields of the snd_soc_card struct to setup DAPM. 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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由 Lars-Peter Clausen 提交于
Those should not be modified (and are not) by the core code, so make them const. This also makes them consistent with the same members of snd_soc_codec. 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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由 Mark Brown 提交于
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由 Lars-Peter Clausen 提交于
Commit ce6120cc(ASoC: Decouple DAPM from CODECs) changed the signature of snd_soc_dapm_widgets_new to take an pointer to a snd_soc_dapm_context instead of a snd_soc_codec. The call to snd_soc_dapm_widgets_new in jz4740_codec_dev_probe was not updated to reflect this change, which results in a compiletime warning and a runtime OOPS. Since the core code calls snd_soc_dapm_widgets_new after the codec has been registered it can be dropped here. 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> Cc: stable@kernel.org
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由 Mark Brown 提交于
Allow audio paths through the Speyside system to be kept active while the system is suspended (for example, when on a voice call) by marking all the external widgets and the DAI link to the WM1250-EV1 baseband module as ignoring suspend. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
Demonstrate the connection of a baseband to the system. We add a DAI for the link to the baseband. This will become visible to the application layer - audio should be started from the application layer using an application such as this: http://opensource.wolfsonmicro.com/~gg/bluetooth-pcm/bluetooth_pcm.c which starts up audio as for CPU based playback and record up to the point where data is streamed. Due to non-availability of baseband simulation hardware we reuse the configuration for the CPU link with the CODEC acting as clock master, allowing signals to be observed with a scope. A more standard system would have separate configuration for the baseband with its own ops structure and operations. Normally the baseband would be clock master as the baseband audio will be synchronised to the external telephony network. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
Pin switches enable direct control of the DAPM state from userspace, enabling simple enabling and disabling of the path. This is especially useful for outputs such as the speaker which are composed of several physical devices as it allows them to be controlled as a group. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
Speyside makes use of support the WM8915 has for detecting the polarity of the microphone and ground connections on headsets, using a GPIO to control the polarity of the ground connection and switching between the two microphone bias supplies available on the device in order to do so. As a result of this the detection support is more involved than for most other CODECs, using a callback to configure the current polarity of the jack and translate this into the board-specific connections required for the current scenario. On Android some additional work is required to hook this up to the application layer as the Android HeadsetObserver monitors a custom drivers/switch API rather than the standard Linux APIs. This can be done by either updating HeadsetObserver or modifying the ALSA core to report via drivers/switch as well. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
Speyside includes a WM9081 configured as an external speaker driver taking an analogue input from HPOUT2 on the WM8915 on the system. Add support for this to the driver, using a prefix of "Sub" for the WM9081 controls to ensure we avoid collisions with controls on the WM8915. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
Dynamically enable and disable the FLL on the WM8915, configuring the system clock to 256fs for 48kHz when the device is active but reverting to using the input 32.768kHz clock directly at other times to support features such as jack detection with minimal power consumption. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
Provide widgets for the basic widgets connected directly to the WM8915 on Speyside - the headphones, speaker, digital and analogue microphones. For the outputs this is just documentation, for the inputs this ensures that the relevant microphone biases are enabled when they are in use. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
This is minimal code required to get audio out of the Speyside audio subsystem on the Wolfson Cragganmore 6410 reference platform. It sets up the link between the CPU and AIF1 of the WM8915 on the system, enabling audio playback via the headphone and speaker outputs of the device (which require no further configuration except runtime). It allows verification of basic functionality of the system. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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由 Mark Brown 提交于
This helps with things like setting up the initial state. 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
Since the WM8915 FLL is not tied to a particular audio interface move it to a CODEC wide operation. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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- 12 4月, 2011 13 次提交
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
The WM1250-EV1 Springbank audio I/O module for the Wolfson Glenfarclas reference platform provides a simple audio I/O with an independant clock domain, intended to simulate cellular modem and bluetooth subsystems within the platform. The card supports some limited GPIO based control but this is currently not implemented. 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
The WM8915 is an ultra low power mobile CODEC designed for smartphones, featuring a mixture of digital and analogue I/O with flexible mixing options and advanced low power accessory detection functionality in a compact package. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Kuninori Morimoto 提交于
card->num_rtd should be 0 after soc_romve_dai_link Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Sangbeom Kim 提交于
This patch add WM8580 PCM machine driver to support PCM audio on SMDKC110, SMDKV210, SMDK6450, SMDK6440 boards. Playback and Capture supports 8kHz sampling rates. and It is tested on SMDKC110, SMDKV210, SMDK6450 Signed-off-by: NSangbeom Kim <sbkim73@samsung.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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由 Mark Brown 提交于
The output PGA was not being powered up in headphone and speaker paths, removing the ability to offer volume control and mute with the output PGA. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
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由 Lu Guanqun 提交于
According to the comments in include/linux/init.h: "Pointers to __devexit functions must use __devexit_p(function_name), the wrapper will insert either the function_name or NULL, depending on the config options." Fix this issue in codecs sn95031. Signed-off-by: NLu Guanqun <guanqun.lu@intel.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Sangbeom Kim 提交于
Fix the inverted clocks handling for pcm cpu driver. By using SND_SOC_DAIFMT_NB_NF, Audio noise can be generated on SMDK. Signed-off-by: NSangbeom Kim <sbkim73@samsung.com> Acked-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lu Guanqun 提交于
Fix the possible dead lock shown below: spin_lock sst_get_stream_status sst_period_elapsed intel_sst_interrupt handle_IRQ_event handle_fasteoi_irq do_IRQ common_interrupt spin_lock sst_set_stream_status sst_platform_pcm_trigger Signed-off-by: NLu Guanqun <guanqun.lu@intel.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Kuninori Morimoto 提交于
free_irq and pm_runtime_disable should be called before snd_soc_unregister_xxx Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Kuninori Morimoto 提交于
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Kuninori Morimoto 提交于
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 09 4月, 2011 4 次提交
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由 Mark Brown 提交于
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由 Jassi Brar 提交于
Signed-off-by: NJassi Brar <jassisinghbrar@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
The DAPM pin operations currently require that the specific DAPM context that the pin being operated in is contained in be specified. With multi component and especially with the addition of a per-card DAPM context this isn't ideal as it means that things like disabling unused pins on CODECs require looking up the CODEC DAPM context. Fix this by falling back to matching a widget in any context if there isn't a match in the current context. The code isn't ideal currently but will do the job. 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 we allow all DAPM contexts to determine their own bias level. While this should in general work in most situations and will deliver the lowest possible power it causes problems for our integration with the card bias level as we're calling the card bias level functions for each DAPM context even though they're card wide but don't say which CODEC we're calling them for. Mitigate against this by forcing everything to be in the same state. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@ti.com>
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- 08 4月, 2011 6 次提交
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由 Mark Brown 提交于
Since we recently explicitly set the register for registerless widgets to no register there is no longer any need to special case power updates for them, we can allow them to be handled with the register compression code as other widgets are. As this is the only remaining user of dapm_generic_apply_power() and dapm_update_bits() also remove those functions. Noticed-by: NLu Guanqun <guanqun.lu@intel.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lu Guanqun 提交于
Add several include files to fix the below compile error. drivers/staging/intel_sst/intelmid.c: In function ‘snd_intelmad_sst_register’: drivers/staging/intel_sst/intelmid.c:805:2: error: ‘sst_drv_ctx’ undeclared (first use in this function) drivers/staging/intel_sst/intelmid.c:805:2: note: each undeclared identifier is reported only once for each function it appears in Signed-off-by: NLu Guanqun <guanqun.lu@intel.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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由 Mike Frysinger 提交于
The ssm2602 codec has a SPI interface as well as I2C, so add the simple bit of glue to make it usable. Signed-off-by: NMike Frysinger <vapier@gentoo.org> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Allow CODEC and card drivers to point to an array of controls from their driver structure rather than explicitly calling snd_soc_add_controls(). Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Dilan Lee 提交于
Update the headphone and line out mixers and PGAs use the same logical set of register bits and sequencing as the speaker mixer/PGA. This allows ALSA controls for mute and volume on headphone and line out to operate correctly. Per conversation on alsa-devel, earlier datasheets indicated that the POWER_MANAGEMENT_* register bits 0 and 1 were aliases to ANALOG_* register bits 0 and 4, and hence only one copy of those bits was programmed. However, later datasheets corrected this. From: Dilan Lee <dilee@nvidia.com> [swarren: Applied same change to headphone widgets] Signed-off-by: NStephen Warren <swarren@nvidia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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