- 21 7月, 2010 1 次提交
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由 Peter Ujfalusi 提交于
When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). 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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- 18 7月, 2010 1 次提交
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由 Jorge Eduardo Candelaria 提交于
In order to reduce pop-noise at powering up/down of the DACs and Drivers, these components have to be handled in a specific sequence. Headset, Handsfree, and Earphone drivers are now registered as PGA components to ensure DACs are enabled first. Also, add a delay to leave time for DACs to settle before continuing power up/down sequence. Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 13 7月, 2010 2 次提交
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由 Peter Ujfalusi 提交于
Restructure the DAPM connections in order to enable only the needed DAC (out of four in twl4030 series). I need to keep the 'AIF Enable' supply connected to the L2/R2 digital path, since the digital loopback needs AIF and APLL running. If no valid route available, than none of the DAC will be powered, but the AIF and APLL is going to be enabled. Furthermore, if only one audio path have valid route, than only the corresponding DAC will be powered. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Peter Ujfalusi 提交于
When the gain is configured using dB value it was not possible to use -24dB since the loopback got muted instead of -24dB. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsomicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 09 7月, 2010 2 次提交
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由 Mark Brown 提交于
The WM8994 can output a clock derived from its internal SYSCLK, called OPCLK. The rate can be selected as a sysclk, with a division from the SYSCLK rate specified (multiplied by 10 since a division of 5.5 is supported) and the clock can be disabled by specifying a divisor of zero. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Mark Brown 提交于
Very handy for debug. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 06 7月, 2010 5 次提交
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由 Raffaele Recalcati 提交于
i2s_accurate_sck switch can be used to have a better approximate sampling frequency. The clock is an externally visible bit clock and it is named i2s continuous serial clock (I2S_SCK). The trade off is between more accurate clock (fast clock) and less accurate clock (slow clock). The waveform will be not symmetric. Probably it is possible to get a better algorithm for calculating the divider, trying to keep a slower clock as possible. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: NRaffaele Recalcati <raffaele.recalcati@bticino.it> Signed-off-by: NDavide Bonfanti <davide.bonfanti@bticino.it> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Acked-by: NSudhakar Rajashekhara <sudhakar.raj@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Raffaele Recalcati 提交于
When McBSP peripheral gets the clock from an external pin, there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR and MCBSP_CLKS. evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different hardware connection and I use MCBSP_CLKS, so I have added this possibility. This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm) Signed-off-by: NRaffaele Recalcati <raffaele.recalcati@bticino.it> Signed-off-by: NDavide Bonfanti <davide.bonfanti@bticino.it> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Acked-by: NSudhakar Rajashekhara <sudhakar.raj@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Raffaele Recalcati 提交于
Added two clocking options for dm365 McBSP peripheral when used with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock from external pin and generates frame sync). A slave clock management can be important when the external codec needs the system clock and the bit clock synchronized (tested with uda1345). This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: NRaffaele Recalcati <raffaele.recalcati@bticino.it> Signed-off-by: NDavide Bonfanti <davide.bonfanti@bticino.it> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Acked-by: NSudhakar Rajashekhara <sudhakar.raj@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Maurus Cuelenaere 提交于
The speaker was enabled when the headphone was plugged in, which isn't the wanted behaviour so correct this. Signed-off-by: NMaurus Cuelenaere <mcuelenaere@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
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 05 7月, 2010 3 次提交
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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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由 Mark Brown 提交于
It will be replaced with automatic deemphasis rate configuration but since we have an enumeration table in this driver this is done in a separate commit to make the renumbering of the enumeration items clear. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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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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- 04 7月, 2010 2 次提交
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由 Maurus Cuelenaere 提交于
This adds sound support for the SmartQ board. The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750 driver is used for driving the WM8987, as they are register compatible. Signed-off-by: NMaurus Cuelenaere <mcuelenaere@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Maurus Cuelenaere 提交于
The WM8987 codec is register compatible with the WM8750, so just add it to the SPI and I²C device table. Signed-off-by: NMaurus Cuelenaere <mcuelenaere@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 02 7月, 2010 1 次提交
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由 Kuninori Morimoto 提交于
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 30 6月, 2010 3 次提交
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由 Vladimir Zapolskiy 提交于
For UDA1341 codec power control is managed in STATUS1 register, and for all other codecs in DATA011 register. Signed-off-by: NVladimir Zapolskiy <vzapolskiy@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Vladimir Zapolskiy 提交于
In UDA1340, UDA1344 and UDA1345 codecs there is one more functional register in part of DATA0 tranfser. For UDA1341 this register coincides with EA register. Signed-off-by: NVladimir Zapolskiy <vzapolskiy@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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- 25 6月, 2010 3 次提交
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由 Eric Bénard 提交于
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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由 Vladimir Zapolskiy 提交于
On initialization ADC/DAC are enabled only for UDA1341, that's why bias_level shall be set to off explicitly, otherwise dapm is misinformed about bias_level on startup. Signed-off-by: NVladimir Zapolskiy <vzapolskiy@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Vladimir Zapolskiy 提交于
This change wipes out a hardcoded macro, which enables codec bias level control. Now is_powered_on_standby value shall be used instead. Signed-off-by: NVladimir Zapolskiy <vzapolskiy@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 24 6月, 2010 1 次提交
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由 Takashi Iwai 提交于
Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
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- 23 6月, 2010 6 次提交
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由 Jarkko Nikula 提交于
This patch adds GPIO jack detection to Nokia N900/RX-51. At the moment only SND_JACK_VIDEOOUT type is reported. More types could be reported after getting more audio features supported and necessary drivers integrated for implementing automated accessory detection. Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Jarkko Nikula 提交于
Nokia RX-51/N900 has multifunction 4-pole audio-video jack that can be used as headphone, headset or audio-video connector. This patch implements the control 'Jack Function' which is used to select the desired function. At the moment only TV-out without audio is supported. Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Eric Bénard 提交于
in tlv320aic23_set_bias_level, for the case SND_SOC_BIAS_ON, the comment says "vref/mid, osc on, dac unmute" but the code doesn't clear the corresponding bits, thus when resuming, several bits are not cleared preventing the codec from working. in tlv320aic23_suspend, clearing the active register is not needed as it will be done by tlv320aic23_set_bias_level, when setting bias to SND_SOC_BIAS_OFF Signed-off-by: NEric Bénard <eric@eukrea.com> Cc: broonie@opensource.wolfsonmicro.com Cc: anuj.aggarwal@ti.com Cc: lrg@slimlogic.co.uk Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Lars-Peter Clausen 提交于
This patch adds ASoC support for the qi_lb60 board. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lars-Peter Clausen 提交于
This patch adds support for the JZ4740 internal codec. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lars-Peter Clausen 提交于
This patch adds ASoC support for JZ4740 SoCs I2S module. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 22 6月, 2010 1 次提交
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由 Grazvydas Ignotas 提交于
After mass production started it was found that several boards exhibit noise problems during sound playback. After some investigation it was determined that CLKX polarity is set incorrectly, and even if most boards can tolerate the wrong setting, there are some that don't. Fix polarity setup in the board file. As the clock settings for input and output now match, merge in and out functions and structures to simplify code. Signed-off-by: NGrazvydas Ignotas <notasas@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 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 4 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 Mark Brown 提交于
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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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