- 17 12月, 2009 6 次提交
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由 Mark Brown 提交于
DAC_VOICE_EN was not defined - looks to have been overly enthusiastically deleted from a previous revision of the patch, pull the value from v1. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Peter Meerwald 提交于
fix precision of PLL computation for TLV320AIC3x SoC driver, test results are at http://pmeerw.net/clkSigned-off-by: NPeter Meerwald <pmeerw@pmeerw.net> Acked-by: NVladimir Barinov <vova.barinov@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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由 Kuninori Morimoto 提交于
This original driver was created by Dialog Semiconductor, and cleanuped by Kuninori Morimoto. Special thanks to David Chen. This became very simple ASoC codec driver, and it is tested by EcoVec24 board. Signed-off-by: NDavid Chen <Dajun.chen@diasemi.com> Signed-off-by: NKuninori Morimoto <morimoto.kuninori@renesas.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Ilkka Koskinen 提交于
Take the regulator framework in use for managing the power sources Signed-off-by: NIlkka Koskinen <ilkka.koskinen@nokia.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NEduardo Valentin <eduardo.valentin@nokia.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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- 07 12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> Cc: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 05 12月, 2009 1 次提交
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由 Mark Brown 提交于
The WM8904 is a high performance ultra-low power stereo CODEC optimised for portable audio applications, with features including a class W amplifier, FLL with free running mode, Mobile ReTune and ground referenced headphone and line outputs. Support for some features, most particularly the digital microphone interface, is not yet present. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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- 04 12月, 2009 2 次提交
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由 Ilkka Koskinen 提交于
Take the regulator framework in use for managing the power sources. Signed-off-by: NIlkka Koskinen <ilkka.koskinen@nokia.com> Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Joonyoung Shim 提交于
This renames from a character / to : of controls. A / occurs below error messages. ASoC: Failed to create IN2RP/VXRP debugfs file ASoC: Failed to create IN2LP/VXRN debugfs file Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 28 11月, 2009 1 次提交
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由 Anuj Aggarwal 提交于
This patch fixes two issues: a) Infinite loop in resume function b) Writes to non-existing registers in resume function Cc: stable@kernel.org Signed-off-by: NAnuj Aggarwal <anuj.aggarwal@ti.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 27 11月, 2009 1 次提交
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由 Takashi Iwai 提交于
The commit fe3e78e0 ASoC: Factor out snd_soc_init_card() removed the error paths that are still valid for wm97* codecs, causing the compile errors like sound/soc/codecs/wm9705.c:399: error: label 'reset_err' used but not defined sound/soc/codecs/wm9712.c:687: error: label 'reset_err' used but not defined sound/soc/codecs/wm9713.c:1237: error: label 'reset_err' used but not defined Revert the removed error path codes. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 11月, 2009 1 次提交
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由 Peter Ujfalusi 提交于
RT workqueue is going away in the near future, replace it with singlethread wq for now, which is still supported. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 11月, 2009 1 次提交
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由 Troy Kisky 提交于
Fix the ordering of sr_valid_mask array. The lower bit of the index represents USB not bosr. Reported-by: NAnuj Aggarwal <anuj.aggarwal@ti.com> Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 13 11月, 2009 1 次提交
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由 Mark Brown 提交于
The DAPM widgets are now insntantiated by the core when creating the card so there is no need for the individual CODEC drivers to do so. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 10 11月, 2009 1 次提交
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由 Peter Ujfalusi 提交于
APLL_CTL register is configured by the twl4030-codec MFD driver. Remove code, which makes changes in the APLL_CTL register, and replace those with checks against the configured audio_mclk configuration done in the MFD driver. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 07 11月, 2009 1 次提交
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由 Takashi Iwai 提交于
Remove the dead code and labels "card_err" in the error paths of some codec drivers. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 05 11月, 2009 2 次提交
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由 Mark Brown 提交于
Probe as a platform driver (ads117x) and remove the call to snd_soc_init_card(). Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Graeme Gregory 提交于
This patch adds support for the TI ADS117x family of multichannel ADCs and was sponsored by Shotspotter Inc. Signed-off-by: NGraeme Gregory <gg@slimlogic.co.uk> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 04 11月, 2009 2 次提交
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由 Mark Brown 提交于
snd_soc_init_card() is always called as the last part of the CODEC probe function so we can factor it out into the core card setup rather than have each CODEC replicate the code to do the initialiastation. This will be required to support multiple CODECs per card. 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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- 03 11月, 2009 1 次提交
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由 Peter Ujfalusi 提交于
Set the codec->bias_level to SND_SOC_BIAS_OFF before changing the initial bias level to STANDBY. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 02 11月, 2009 1 次提交
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由 Neil Jones 提交于
Add support for the Wolfson Microelectronics WM8727 DAC, this is a simple non-configurable DAC. Signed-off-by: NNeil Jones <neil.jones@imgtec.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 29 10月, 2009 4 次提交
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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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- 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 3 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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- 22 10月, 2009 1 次提交
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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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- 19 10月, 2009 1 次提交
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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 2 次提交
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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 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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- 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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- 07 10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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