1. 12 8月, 2010 1 次提交
    • L
      ASoC: multi-component - ASoC Multi-Component Support · f0fba2ad
      Liam Girdwood 提交于
      This patch extends the ASoC API to allow sound cards to have more than one
      CODEC and more than one platform DMA controller. This is achieved by dividing
      some current ASoC structures that contain both driver data and device data into
      structures that only either contain device data or driver data. i.e.
      
       struct snd_soc_codec    --->  struct snd_soc_codec (device data)
                                +->  struct snd_soc_codec_driver (driver data)
      
       struct snd_soc_platform --->  struct snd_soc_platform (device data)
                                +->  struct snd_soc_platform_driver (driver data)
      
       struct snd_soc_dai      --->  struct snd_soc_dai (device data)
                                +->  struct snd_soc_dai_driver (driver data)
      
       struct snd_soc_device   --->  deleted
      
      This now allows ASoC to be more tightly aligned with the Linux driver model and
      also means that every ASoC codec, platform and (platform) DAI is a kernel
      device. ASoC component private data is now stored as device private data.
      
      The ASoC sound card struct snd_soc_card has also been updated to store lists
      of it's components rather than a pointer to a codec and platform. The PCM
      runtime struct soc_pcm_runtime now has pointers to all its components.
      
      This patch adds DAPM support for ASoC multi-component and removes struct
      snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
      or runtime PCM level basis rather than using snd_soc_socdev.
      
      Other notable multi-component changes:-
      
       * Stream operations now de-reference less structures.
       * close_delayed work() now runs on a DAI basis rather than looping all DAIs
         in a card.
       * PM suspend()/resume() operations can now handle N CODECs and Platforms
         per sound card.
       * Added soc_bind_dai_link() to bind the component devices to the sound card.
       * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
         DAI link components.
       * sysfs entries can now be registered per component per card.
       * snd_soc_new_pcms() functionailty rolled into dai_link_probe().
       * snd_soc_register_codec() now does all the codec list and mutex init.
      
      This patch changes the probe() and remove() of the CODEC drivers as follows:-
      
       o Make CODEC driver a platform driver
       o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
       o Removed all static codec pointers (drivers now support > 1 codec dev)
       o snd_soc_register_pcms() now done by core.
       o snd_soc_register_dai() folded into snd_soc_register_codec().
      
      CS4270 portions:
      Acked-by: NTimur Tabi <timur@freescale.com>
      
      Some TLV320aic23 and Cirrus platform fixes.
      Signed-off-by: NRyan Mallon <ryan@bluewatersys.com>
      
      TI CODEC and OMAP fixes
      Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
      Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
      Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
      
      Samsung platform and misc fixes :-
      Signed-off-by: NChanwoo Choi <cw00.choi@samsung.com>
      Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com>
      Signed-off-by: NKyungmin Park <kyungmin.park@samsung.com>
      Reviewed-by: NJassi Brar <jassi.brar@samsung.com>
      Signed-off-by: NSeungwhan Youn <sw.youn@samsung.com>
      
      MPC8610 and PPC fixes.
      Signed-off-by: NTimur Tabi <timur@freescale.com>
      
      i.MX fixes and some core fixes.
      Signed-off-by: NSascha Hauer <s.hauer@pengutronix.de>
      
      J4740 platform fixes:-
      Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
      
      CC: Tony Lindgren <tony@atomide.com>
      CC: Nicolas Ferre <nicolas.ferre@atmel.com>
      CC: Kevin Hilman <khilman@deeprootsystems.com>
      CC: Sascha Hauer <s.hauer@pengutronix.de>
      CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
      CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
      CC: Daniel Gloeckner <dg@emlix.com>
      CC: Manuel Lauss <mano@roarinelk.homelinux.net>
      CC: Mike Frysinger <vapier.adi@gmail.com>
      CC: Arnaud Patard <apatard@mandriva.com>
      CC: Wan ZongShun <mcuos.com@gmail.com>
      Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
      f0fba2ad
  2. 04 8月, 2010 1 次提交
    • P
      ASoC: TWL4030: Capture route runtime DAPM ordering fix · bda7d2a8
      Peter Ujfalusi 提交于
      Fix the ordering problem in DAPM domain, when the user
      changes between digital and analog sources during active
      capture (or loopback) scenario.
      Before this patch, when the user changed from analog source
      to digital there were a short time, when the codec enabled
      analog mic bias (2.2 volts) instead of the correct digital
      mic bias (1.8 volts) to the digital microphones.
      This behaviour caused by the former implementation of
      selecting the correct type of bias. This was done at the
      POST_REG event of the DAPM_MUX_E("TXx Capture Route")
      widget.
      By moving the bias type selection as DAPM_SUPPLY and
      connecting it to the corresponding digimic widget the
      problematic situation can be avoided.
      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>
      bda7d2a8
  3. 03 8月, 2010 13 次提交
  4. 02 8月, 2010 4 次提交
  5. 30 7月, 2010 3 次提交
  6. 29 7月, 2010 2 次提交
  7. 27 7月, 2010 1 次提交
  8. 23 7月, 2010 1 次提交
  9. 21 7月, 2010 1 次提交
    • P
      ASoC: TWL4030: Add configurable delay after digimic enable · 01ea6ba2
      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>
      01ea6ba2
  10. 20 7月, 2010 7 次提交
  11. 18 7月, 2010 6 次提交