- 20 6月, 2010 1 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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- 19 6月, 2010 1 次提交
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由 Stuart Longland 提交于
When SX_TLV widgets are read, if the gain is set to a value below 0dB, the mixer control is erroniously read as being at maximum volume. The value read out of the CODEC register is never sign-extended, and when the minimum value is subtracted (read; added, since the minimum is negative) the result is a number greater than the maximum allowed value for the control, and hence it saturates. Solution: Mask the result so that it "wraps around", emulating sign-extension. Signed-off-by: NStuart Longland <redhatter@gentoo.org> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 6月, 2010 1 次提交
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由 Eric Bénard 提交于
* tdm slot has to be configured to get sound working on i.MX25 Signed-off-by: NEric Bénard <eric@eukrea.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 16 6月, 2010 2 次提交
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由 Peter Huewe 提交于
The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn enables the compilation of ak4642.c - however this codec uses I2C to communicate with the HW. Same applies to DA7210. Consequently when I2C is not set, the compilation fails [1] This patch fixes this issues, by adding a depencdency on the related HW- controller. Signed-off-by: NPeter Huewe <peterhuewe@gmx.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 15 6月, 2010 7 次提交
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由 Mark Brown 提交于
The most useful configuration for the WM2000 is to enable the ANC so turn that on by default, and since we're not reflecting chip default state also enable the speaker output by default. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Sudhakar Rajashekhara 提交于
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO support. This FIFO provides additional data buffering. It also provides tolerance to variation in host/DMA controller response times. More details of the FIFO operation can be found at http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf Existing sequence of steps for audio playback/capture are: a. DMA configuration b. McASP configuration (configures and enables FIFO) c. Start DMA d. Start McASP (enables FIFO) During McASP configuration, while FIFO was being configured, FIFO was being enabled in davinci_hw_common_param() function of sound/soc/davinci/davinci-mcasp.c file. This generated a transmit DMA event, which gets serviced when DMA is started. https://patchwork.kernel.org/patch/84611/ patch clears the DMA events before starting DMA, which is the right thing to do. But this resulted in a state where DMA was waiting for an event from McASP (after step c above), but the event which was already there, has got cleared (because of step b above). The fix is not to enable the FIFO during McASP configuration as FIFO was being enabled as part of McASP start. Signed-off-by: NSudhakar Rajashekhara <sudhakar.raj@ti.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> 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@slimlogic.co.uk> 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@slimlogic.co.uk> 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@slimlogic.co.uk> 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@slimlogic.co.uk> 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@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 6月, 2010 1 次提交
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由 Grant Likely 提交于
The header contains an extern that isn't used by anything. Remove. Signed-off-by: NGrant Likely <grant.likely@secretlab.ca> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 10 6月, 2010 1 次提交
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由 Wan ZongShun 提交于
This patch is to change 'auido.h' to 'audio.h' for fixing nuc900 alsa driver build error. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 09 6月, 2010 1 次提交
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由 Ryan Mallon 提交于
Add support for i2s audio on Bluewater Systems Snapper CL15 module Signed-off-by: NRyan Mallon <ryan@bluewatersys.com> Acked-by: NH Hartley Sweeten <hsweeten@visionengravers.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 07 6月, 2010 4 次提交
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由 Wan ZongShun 提交于
This patch to remove the 'break;', when the 'switch' jumps to the 'default' branch, the 'return -EINVAL' will be return with a error number, so the 'break;' code never be run, it is unuseful and should be removed here. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Wan ZongShun 提交于
Remove break after return, it is not needed. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Signed-off-by: NNicolas Ferre <nicolas.ferre@atmel.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Ryan Mallon 提交于
Add ep93xx i2s audio driver Signed-off-by: NRyan Mallon <ryan@bluewatersys.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Upper threshold is used in mode7 of DAC33. Instead of hard wired UTHR, add control to change the upper threshold value. Changing upper threshold is not allowed when the playback is already running, since wrongly timed change in the UTHR can cause problems with the codec. With this control the length of the burst in mode7 can be changed. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 04 6月, 2010 5 次提交
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由 Eric Bénard 提交于
Add the necessary files to support the TLV320AIC23B wired in I2S on our i.MX platforms. Signed-off-by: NEric Bénard <eric@eukrea.com> Acked-by: NSascha Hauer <s.hauer@pengutronix.de> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Eric Bénard 提交于
* introduce 3 new flags to allow a more detailed configuration of the SSI link : IMX_SSI_NET : enable Network Mode IMX_SSI_SYN : enable Synchronous Mode IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode * new platform can use these settings without breaking actual platforms. Signed-off-by: NEric Bénard <eric@eukrea.com> Acked-by: NSascha Hauer <s.hauer@pengutronix.de> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Wan ZongShun 提交于
The variable 'state' of structure 's3c_ac97_info' seems no use here, so this patch is to remove the unnecessary variable. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Wan ZongShun 提交于
The variable 'periods' of structure 'atmel_runtime_data' seems no use in whole atmel alsa driver,so I make a patch to remove the unnecessary variable. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Wan ZongShun 提交于
Use the resource_size function instead of manually calculating the resource size.This patch can reduce the chance of introducing off-by-one errors. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NManuel Lauss <manuel.lauss@googlemail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 03 6月, 2010 2 次提交
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由 Peter Ujfalusi 提交于
OMAP McBSP FIFO is word structured: McBSP2 has 1024 + 256 = 1280 word long buffer, McBSP1,3,4,5 has 128 word long buffer This means, that the size of the FIFO depends on the McBSP word size configuration. For example on McBSP3: 16bit samples: size is 128 * 2 = 256 bytes 32bit samples: size is 128 * 4 = 512 bytes It is simpler to place constraint for buffer and period based on channels. McBSP3 as example again (16 or 32 bit samples): 1 channel (mono): size is 128 frames (128 words) 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) Use the second method to place hw_rule on buffer size, and in threshold mode to period size. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com> Acked-by: NTony Lindgren <tony@atomide.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Peter Ujfalusi 提交于
Save the word length configuration of McBSP, and use that information to calculate, and configure the threshold in McBSP. Previously the calculation was only correct when the stream had 16bit audio. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com> Acked-by: NTony Lindgren <tony@atomide.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 02 6月, 2010 5 次提交
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由 Wan ZongShun 提交于
The size calculation is end - start + 1. But,sometimes, the '1' can be forgotten carelessly, witch will have potential risk, so use resource_size for {request/release}_mem_region and ioremap here should be good habit. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NDaniel Glöckner <dg@emlix.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Wan ZongShun 提交于
This patch is to modify the ac97 delays to minimum, all these 1000 micro seconds delays seem over spec for the AC97 interface. I deleted some unnecessary delays here and changed the AC97 cold and warm reset delays from 1000us to 100us. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Wan ZongShun 提交于
Fix a '#include "nuc900-audio.h' typo, I think it should be 'audio'. At the same time, this patch renames the 'nuc900-auido.h' file to 'nuc900-audio.h'. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Wan ZongShun 提交于
The current implement meant ACTL_ACCON was only accessed once when read or write proceeding, which is not right, if so,we have to wait the 'timeout=0x10000' to end every times. We need to polling the bit AC_R_FINISH and AC_W_FINISH of ACTL_ACCON register to identify whether read or write is finished or not,so I make the patch to fix the issue. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Wan ZongShun 提交于
This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition. There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX, the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather than 'if (PCM_TX == stype)', which makes the codes easy to read. Signed-off-by: NWan ZongShun <mcuos.com@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 01 6月, 2010 2 次提交
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由 Sascha Hauer 提交于
Signed-off-by: NSascha Hauer <s.hauer@pengutronix.de> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Sascha Hauer 提交于
Signed-off-by: NSascha Hauer <s.hauer@pengutronix.de> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 31 5月, 2010 7 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 apatard@mandriva.com 提交于
This patch is adding support for openrd client platforms. It's using the cs42l51 codec and has one mic and one speaker plugs. Signed-off-by: NArnaud Patard <apatard@mandriva.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 apatard@mandriva.com 提交于
This patch enables support for the i2s controller available on kirkwood platforms Signed-off-by: NArnaud Patard <apatard@mandriva.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 apatard@mandriva.com 提交于
This patch is adding a ASoC driver for the cs42l51 from Cirrus Logic. Master mode and spi mode are not supported. Signed-off-by: NArnaud Patard <apatard@mandriva.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.ul> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Seungwhan Youn 提交于
This patch adds spdif dummy codec driver for using spdif-dit as a stand-alone. Until this, spdif-dit can be used only with other codecs like tlv320aci3x in davinci platform. Signed-off-by: NSeungwhan Youn <sw.youn@samsung.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
Machine driver can instruct the codec driver to reset the chip registers to their default values at probe time. If machine driver does not provide setup data, then the registers are going to be reseted to their defaults, to be safe. If the developer on the platform confirms that the register reset is not needed, than it can be skipped, saving ~20ms time in probe. As safety measure do the register reset at remove time also. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Peter Ujfalusi 提交于
Restructure the codec power code in order to be able to hit off when the codec is not in use. Since the audio registers are accessible while the codec is powered down, there is no need for additional safety mechanism. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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