1. 22 10月, 2010 1 次提交
  2. 19 10月, 2010 1 次提交
  3. 18 10月, 2010 1 次提交
  4. 15 10月, 2010 1 次提交
  5. 11 10月, 2010 1 次提交
  6. 01 10月, 2010 1 次提交
  7. 15 9月, 2010 1 次提交
  8. 23 8月, 2010 1 次提交
  9. 19 8月, 2010 1 次提交
  10. 05 8月, 2010 1 次提交
    • M
      ASoC: Add initial WM8962 CODEC driver · 9a76f1ff
      Mark Brown 提交于
      The WM8962 is a low power, high performance stereo CODEC designed for
      portable digital audio applications.
      
      This initial driver release supports the key audio paths of the WM8962.
      Extended functionality, such as microphone detection, digital microphones
      and the advanced DSP signal enhancements provided by the device are not
      yet supported.
      Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      9a76f1ff
  11. 03 8月, 2010 1 次提交
    • I
      ASoC: Initial WM8741 CODEC driver · 992bee40
      Ian Lartey 提交于
      The WM8741 is a very high performance stereo DAC designed for audio
      applications such as professional recording systems, A/V receivers and
      high specification CD, DVD and home theatre systems. The device supports
      PCM data input word lengths from 16 to 32-bits and sampling rates up to
      192kHz.  The WM8741 also supports DSD bit-stream data format, in both
      direct DSD and PCM-converted DSD modes.
      
      TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
      allow for all supported sample rate / Master Clock frequency combinations.
      Fully enable control of supplies.
      Signed-off-by: NIan Lartey <ian@opensource.wolfsonmicro.com>
      Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
      Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      992bee40
  12. 23 6月, 2010 1 次提交
  13. 31 5月, 2010 1 次提交
  14. 30 4月, 2010 1 次提交
    • M
      ASoC: Add WM9090 amplifier driver · 39b8eab7
      Mark Brown 提交于
      The WM9090 is a high performance low power audio subsystem, including
      headphone and class D speaker drivers.
      
      Note that this driver is a standalone CODEC driver and so is only
      immediately suitable for use with the WM9090 as a standalone sound card
      taking line inputs, or with a DAC with no software control.  The pending
      ASoC multi-CODEC support will expand the range of systems that can use
      the driver, or system-specific adaptations can be made.
      Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
      39b8eab7
  15. 19 3月, 2010 2 次提交
  16. 12 3月, 2010 1 次提交
  17. 12 2月, 2010 1 次提交
  18. 02 2月, 2010 1 次提交
    • M
      ASoC: Add WM8994 CODEC driver · 9e6e96a1
      Mark Brown 提交于
      The WM8994 is a highly integrated ultra-low power hi-fi audio subsystem
      designed for smartphones and other portable devices rich in multimedia
      features.  It provides advanced digital mixing facilities enabling low
      power high quality interconnection of CPU, baseband and other audio
      sources through flexible digital and analogue routing, and integrates
      a class W headphone driver and stereo class D speaker drivers.
      Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      9e6e96a1
  19. 28 1月, 2010 1 次提交
  20. 18 12月, 2009 1 次提交
  21. 17 12月, 2009 2 次提交
  22. 05 12月, 2009 1 次提交
  23. 05 11月, 2009 1 次提交
  24. 02 11月, 2009 1 次提交
  25. 15 10月, 2009 1 次提交
    • P
      ASoC: Codec driver for Texas Instruments tlv320dac33 codec · c8bf93f0
      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>
      c8bf93f0
  26. 10 10月, 2009 1 次提交
    • P
      ASoC: TPA6130A2 amplifier driver · 493b67ef
      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>
      493b67ef
  27. 10 9月, 2009 1 次提交
  28. 21 8月, 2009 1 次提交
  29. 19 8月, 2009 1 次提交
  30. 14 8月, 2009 1 次提交
  31. 13 8月, 2009 1 次提交
  32. 06 8月, 2009 1 次提交
    • M
      ASoC: Add WM8776 CODEC driver · 924914ee
      Mark Brown 提交于
      The WM8776 is a high performance, stereo audio CODEC with five channel
      input selector. The WM8776 is ideal for surround sound processing
      applications for home hi-fi, DVD-RW and other audio visual equipment.
      
      This driver implements support for most WM8776 features - currently the
      ADC automatic level control/limiter functionality is omitted.
      Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      924914ee
  33. 23 7月, 2009 1 次提交
    • J
      ASoC: Add support for Conexant CX20442-11 voice modem codec · 459dc352
      Janusz Krzysztofik 提交于
      This patch adds support for Conexant CX20442-11 voice modem codec, suitable
      for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related
      sound card driver will follow.
      
      This codec is an optional part of the Conexant SmartV three chip modem design.
      As such, documentation for its proprietary digital audio interface is not
      available. However, on Amstrad Delta board, thanks to Mark Underwood who
      created an initial, omap-alsa based sound driver a few years ago[1], the codec
      has been discovered to be accessible not only from the modem side, but also
      over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound
      card that can access the codec DAI directly. The DAI configuration parameters
      (sample rate and format, number of channels) has been selected out empirically
      for best user experience.
      
      The codec analogue interface consists of two pairs of analogue I/O pins:
      speakerphone interface or telephone handset/headset interface. Furthermore, it
      seams to provide two operation modes for speakerphone I/O: standard and
      advanced, with automatic gain control and echo cancelation. Even if the codec
      control interface is unknown and not available, all those interfaces and modes
      can be selected over the modem chip using V.253 commands. The driver is able
      to issue necessary commands over a suitable hw_write function if provided by a
      sound card driver. Otherwise, the codec can be controlled over the modem from
      userspace while inactive.
      
      Even if nothig is known about the codec internal power management
      capabilities, DAPM widgets has been used to model the codec audio map.
      Automatically performed powering up/down of those virtual widgets results in
      corresponding V.253 commands being issued.
      
      Some driver features/oddities may be board specific, but I have no way to
      verify that with any board other than Amstrad Delta.
      
      [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html
      
      Created and tested against linux-2.6.31-rc3.
      Applies and works with linux-omap-2.6 commit
      7c5cb7862d32cb344be7831d466535d5255e35ac as well.
      Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
      Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      459dc352
  34. 16 7月, 2009 1 次提交
  35. 15 7月, 2009 1 次提交
  36. 09 7月, 2009 1 次提交
  37. 02 7月, 2009 1 次提交
  38. 13 6月, 2009 1 次提交