- 30 10月, 2009 2 次提交
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由 Anuj Aggarwal 提交于
Modifying the Kconfig and Makefile in sound/soc/omap folder to add support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: NAnuj Aggarwal <anuj.aggarwal@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Anuj Aggarwal 提交于
Adding support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: NAnuj Aggarwal <anuj.aggarwal@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 29 10月, 2009 6 次提交
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由 Anuj Aggarwal 提交于
The pop-removal specific values are configured for TWL4030 codec for OMAP3EVM through this patch. Signed-off-by: NAnuj Aggarwal <anuj.aggarwal@ti.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Capture path also need the APLL enabled, adding DAPM_SUPPLY for the Virtual ADCs. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
It seams that certain part of the twl4030 codec needs the APLL enabled before they are enabled. Paths which has any digital processing needs need the APLL enabled before they can function. For example the vibra output will have some random data after it is enabled and before the APLL also enabled. If only analog components are in use (analog bypass), than it seams, that the APLL does not need to be enabled. This lowers the power consumption with around ~0.005A. Adding DAPM_SUPPLY to the Digital playback route and also to the capture route to enable and disable the APLL. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jari Vanhala 提交于
This patch fixes vibrator playing incoherently, when driven with audio. There is something wrong in switch 3 at H-bridge and VIBRA_SET still affects PWM generator. Slowest value fixes things. Signed-off-by: NJari Vanhala <ext-jari.vanhala@nokia.com> Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Daniel Mack 提交于
The CS4270 codec features an de-emphasis filter for compensation of audio material filtered by an 50/15 uS algorithm. Not sure whether we should always enable it for 44100Hz sampling frequency, but it should at least be configurable by the user. Signed-off-by: NDaniel Mack <daniel@caiaq.de> Acked-by: NTimur Tabi <timur@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Fix up some comments, remove all enable_pin() calls (edge widgets are all enabled by default) and mark the microphone as disabled by default since it requires a resistor fit to connect it. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 28 10月, 2009 2 次提交
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由 Peter Ujfalusi 提交于
codec_muted is missleading, change it to apll_enabled, which is what it is doing: enabing and disabling the APLL. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Since ASoC core now handling the codec bias differently there is no need to do the tracking of bypass switch states anymore. Handling of the common bit for analog loopbacks is done with DAPM_SUPPLY for the bypass paths. Now this bit is only enabled when there is a complete analog bypass path, compared to the previous implementation, when the global switch was enabled if there were any of the analog bypass switch was on (regardless if there were complete path or not) Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 26 10月, 2009 5 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 Peter Ujfalusi 提交于
Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Remove the register descriptions from the twl4030.h file and use the linux/mfd/twl4030-codec.h instead, which has the codec related register descriptions also. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Add needed platform data for the twl4030_codec MFD on boards, where the audio part of the twl4030 codec is used. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NTony Lindgren <tony@atomide.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
New MFD child to twl4030 MFD device. Reason for the twl4030_codec MFD: the vibra control is actually in the codec part of the twl4030. If both the vibra and the audio functionality is needed from the twl4030 at the same time, than they need to control the codec power and APLL at the same time without breaking the other driver. Also these two has to be able to work without the need for the other driver. This MFD device will be used by the drivers, which needs resources from the twl4030 codec like audio and vibra. The platform specific configuration data is passed along to the child drivers (audio, vibra). Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NSamuel Ortiz <sameo@linux.intel.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 22 10月, 2009 3 次提交
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由 Janusz Krzysztofik 提交于
After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c, omap_pcm_prepare() unconditionally calls: omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); Current implementation of those two functions found in arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at all, so they both end with BUG() on that machine. That results in ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta. The patch corrects the problem by not calling those two functions when run on OMAP1 class based machines. Created against linux-2.6.32-rc5. Tested on Amstrad Delta. Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Fix the definition of DAC33_LTM field, the LTM bits in FIFO_IRQ_MODE_B register are starting at bit 6. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Janusz Krzysztofik 提交于
Hi Mark, Here is a patch that corrects small omissions I have found in my code. Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 19 10月, 2009 6 次提交
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由 Mark Brown 提交于
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由 Barry Song 提交于
Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc. access the ops field in these DAIs, panic will happen. Signed-off-by: NBarry Song <21cnbao@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Julia Lawall 提交于
If the NULL test on jack is needed, then the derefernce should be after the NULL test. A simplified version of the semantic match that detects this problem is as follows (http://coccinelle.lip6.fr/): // <smpl> @match exists@ expression x, E; identifier fld; @@ * x->fld ... when != \(x = E\|&x\) * x == NULL // </smpl> Signed-off-by: NJulia Lawall <julia@diku.dk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Manuel Lauss 提交于
Codec read/write functions: wait 21us between the pokings of hardware. Add timeouts to unbounded loops waiting for bits to change. Signed-off-by: NManuel Lauss <manuel.lauss@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Manuel Lauss 提交于
Verify that the correct register has been received from the codec. Signed-off-by: NManuel Lauss <manuel.lauss@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Do not rewrite the whole register, but only update the needed bits in set_dai_sysclk functions. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 15 10月, 2009 4 次提交
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由 Mark Brown 提交于
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由 Peter Ujfalusi 提交于
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo audio DAC. TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low power audio playback. The digital interface can use I2S, DSP (A or B), Right and Left justified formats. DAC33 has stereo analog input, which can be bypassed to the analog outputs. Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass' mode (default) and nSample mode (FIFO is in use). a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is working synchronously as a normal codec (it needs constant stream of data on the digital interface). b) The nSample mode implementation uses one interrupt line from DAC33 to the host: Alarm threshold is set to 10ms of audio data (limit by the driver implementation). DAC33 will signal an interrupt, when the FIFO level goes under the Alarm threshold. The host will write to nSample register a value (number of stereo samples), to tell DAC33 how many samples it should read in a burst from the host. When the DAC33 received the number of samples, it disables the clocks on the I2S bus. When the FIFO use again goes under the Alarm threshold, DAC33 signals the host with an interrupt, and the process is repeated. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Igor Grinberg 提交于
Signed-off-by: NIgor Grinberg <grinberg@compulab.co.il> Signed-off-by: NMike Rapoport <mike@compulab.co.il> CC: Mark Brown <broonie@opensource.wolfsonmicro.com> CC: alsa-devel@alsa-project.org Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
The PM core will grow pm_link infrastructure in 2.6.33 which can be used to implement the intended functionality of the ASoC-specific device suspend and resume callbacks so drop them. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 10月, 2009 2 次提交
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由 Ben Dooks 提交于
Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h as it provides nothing to the current kernel and is not in any future plans for the system. Signed-off-by: NBen Dooks <ben@simtec.co.uk> Signed-off-by: NSimtec Linux Team <linux@simtec.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Eero Nurkkala 提交于
Access to damp_power_widgets() is assumed to be single-threaded. Concurrent accesses to dapm_power_widgets() may result in unpredictable behavior. Calls from: close_delayed_work() soc_codec_close() soc_pcm_prepare() soc_suspend() soc_resume_deferred() to snd_soc_dapm_stream_event() do not have the codec->mutex taken to cover the call to dapm_power_widgets(). Thus, take the mutex in these paths also to assure single-threaded use of dapm_power_widgets(). Signed-off-by: NEero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 12 10月, 2009 1 次提交
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由 Peter Ujfalusi 提交于
The power for the amplifier should be handled internally by the tpa6130a2 driver. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 10 10月, 2009 2 次提交
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由 Mark Brown 提交于
- Staticise ttpa6130a2_client. - Remove unneeded cast from void. - Use explict NULL rather than 0. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Driver for Texas Instruments TPA6130A2 stereo headphone amplifier. The driver provides playback gain control and also pre-defined DAPM_HP widgets and DAPM routings for power management. The DAPM_HP widget names are: "TPA6130A2 Headphone Left" "TPA6130A2 Headphone Right" From soc machine drivers to use with the tpa6130a2 amplifier, the tpa6130a2_add_controls has to be called, which adds the alsa controls and the DAPM routing needed for the tpa6130a2. After that the machine driver can connect the codec's output with 'TPA6130A2 Left' and 'TPA6130A2 Right': {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"}, {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"}, Internally the left and right channels are powered separately. When none of the channels are needed the amplifier is powered down: hard power: valid GPIO number is passed within platform data soft power: Using the software shutdown of the amplifier Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 09 10月, 2009 1 次提交
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由 Nicolas Ferre 提交于
The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file to extend the machine ID checking. Signed-off-by: NNicolas Ferre <nicolas.ferre@atmel.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 08 10月, 2009 1 次提交
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由 Mark Brown 提交于
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- 07 10月, 2009 3 次提交
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由 Mark Brown 提交于
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
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由 Mark Brown 提交于
These should be handled via set_tdm_slot() now and cause build failures as-is. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 06 10月, 2009 2 次提交
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由 Mark Brown 提交于
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由 Mark Brown 提交于
Sometimes it is desirable to have a mux which does not reflect any direct register configuration but which will instead only have an effect implicitly (for example, as a result of changing which parts of the device are powered up). Provide a virtual mux for this purpose. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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