- 19 5月, 2009 2 次提交
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由 Lopez Cruz, Misael 提交于
Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com> Acked-by: NPeter Ujflausi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Ujfalusi 提交于
AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 5月, 2009 1 次提交
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由 Misael Lopez Cruz 提交于
Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 15 5月, 2009 2 次提交
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Marek Vasut 提交于
The WM9712 can be configured by resistor strapping GPIO4 to behave like the WM9713 and default to leaving the AC97 link disabled after cold reset until a warm reset occurs. In this configuration we need to issue a warm reset after cold to bring the link up so do so. The warm reset will be harmless on systems that don't need it. [Changelog rewritten to document the reasoning. -- broonie] Signed-off-by: NMarek Vasut <marek.vasut@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 12 5月, 2009 3 次提交
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由 Joonyoung Shim 提交于
The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Greg Kroah-Hartman 提交于
In the near future, the driver core is going to not allow direct access to the driver_data pointer in struct device. Instead, the functions dev_get_drvdata() and dev_set_drvdata() should be used. These functions have been around since the beginning, so are backwards compatible with all older kernel versions. Signed-off-by: NGreg Kroah-Hartman <gregkh@suse.de> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Roel Kluin 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 08 5月, 2009 1 次提交
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由 Peter Ujfalusi 提交于
Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAIN It has not caused problems, since TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x30 Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 07 5月, 2009 1 次提交
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由 Daniel Mack 提交于
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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- 06 5月, 2009 1 次提交
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由 Daniel Mack 提交于
Replace the magic 0x80 value with a suitable macro definition. 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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- 05 5月, 2009 4 次提交
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由 Peter Ujfalusi 提交于
This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lopez Cruz, Misael 提交于
This patch add voice digital loopback (sidetone) to the twl4030 driver. It mixes voice uplink attenuated (by sidetone gain) with voice downlink when the codec is working in option2 (voice/audio mode). Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lopez Cruz, Misael 提交于
This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: NMisael Lopez Cruz <x0052729@ti.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jinyoung Park 提交于
The mis-typing exist in dapm controller definitions and dapm route definitions, so happen mis-matched error when snd_soc_dapm_add_routes(). Cc: stable@kernel.org Signed-off-by: NJinyoung Park <parkjy@mtekvision.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
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- 02 5月, 2009 1 次提交
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由 Mark Brown 提交于
The AC97 wire format is completely fixed so CODECs don't have any choice about the formats they accept but controllers accept a variety of data formats and render them down onto the bus. Have a shared define so all the CODEC drivers will interoperate with any of our controller drivers. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 28 4月, 2009 7 次提交
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由 Joonyoung Shim 提交于
The gain control for earpiece amplifier uses 0dB ~ 12dB according to the TRM, but the present code is implemented to -6dB ~ 6dB. Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> 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>
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由 Mark Brown 提交于
We need to check only if the WM8350 is master and only when starting the stream so if either is not true then we can skip the check. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Daniel Mack 提交于
This adds a new control named 'Master Playback Switch' for cs4270 codecs. It is implemented using the new SOC_DOUBLE_EXT macro to catch the put function and store the information about manually set mute controls from userspace. When a manual mute is set, we don't want the soc core to un-mute the outputs. Renamed cs4270_mute() to cs4270_dai_mute() to avoid confusion. 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 Daniel Mack 提交于
The control modifies the MUTE register, hence the polarity must be inverted. 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 Jonathan Cameron 提交于
Signed-off-by: NJonathan Cameron <jic23@cam.ac.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
It's expected behaviour for the CODEC header to provide them but the WM8350 doesn't due to having all the registers together under drivers/mfd. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 23 4月, 2009 9 次提交
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由 Peter Ujfalusi 提交于
Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Joonyoung Shim 提交于
Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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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
This is now handled by symmetric_rates. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Save a little extra power by enabling the DC servo offset correction for the output channels only when the relevant channels are enabled. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Modify the default startup sequence in the chip to set the DC servo dither level for optimal performance. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
CLK_DSP provides a master clock for the DAC and ADC related functionality on the device. 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>
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- 22 4月, 2009 1 次提交
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由 Mark Brown 提交于
The driver is out of sync with the core functions it is using. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 21 4月, 2009 1 次提交
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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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- 20 4月, 2009 1 次提交
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由 Joonyoung Shim 提交于
Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 4月, 2009 1 次提交
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由 Russell King - ARM Linux 提交于
I notice that the fixes were merged, minus one: sound/soc/codecs/wm9705.c: At top level: sound/soc/codecs/wm9705.c:445: warning: initialization from incompatible pointer type so you might find this trivial patch useful. Signed-off-by: NRussell King <rmk+kernel@arm.linux.org.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 17 4月, 2009 1 次提交
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由 Peter Ujfalusi 提交于
The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 16 4月, 2009 1 次提交
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由 Mark Brown 提交于
It has a shared LRCLK. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 4月, 2009 2 次提交
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由 Mark Brown 提交于
The WM8960 is a low power, high quality stereo codec designed for portable digital audio applications. Stereo class D speaker drivers provide 1W per channel into 8W loads. Guaranteed low leakage, excellent PSRR and pop/click suppression mechanisms enable direct battery connection for the speaker supply. The device also integrates a complete microphone interface and a stereo headphone driver. External component requirements are drastically reduced as no separate microphone, speaker or headphone amplifiers are required. Advanced on-chip digital signal processing performs automatic level control for the microphone or line input. Stereo 24-bit sigma-delta ADCs and DACs are used with low power over-sampling digital interpolation and decimation filters and a flexible digital audio interface. The master clock can be input directly or generated internally by an onboard PLL, supporting most commonly-used clocking schemes. This driver was originally written by Liam Girdwood, with substantial subsequent additions and updates for feature completeness and changes in the ASoC framework from me. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
This ensures that we sync with the DAPM powerdown sequencing properly and don't need to bounce the power on the voice DAC so often. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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