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  1. 27 4月, 2012 3 次提交
    • L
      ASoC: dpcm: Add runtime dynamic route update · 618dae11
      由 Liam Girdwood 提交于 
      This patch allows DPCM to dynamically alter the FE to BE PCM links
at runtime based on mixer setting updates. DAPM is looked up after
every mixer update and we perform a DPCM runtime update if the
mixer has a change of value.

This patchs adds/changes the following :-

 o Adds DPCM runtime update core.
 o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power()
   to return if a change has occured rather than 0. No other users check
   atm.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      618dae11
    • L
      ASoC: dpcm: Add debugFS support for DPCM · f86dcef8
      由 Liam Girdwood 提交于 
      Add debugFS files for DPCM link management information.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      f86dcef8
    • L
      ASoC: dpcm: Add Dynamic PCM core operations. · 01d7584c
      由 Liam Girdwood 提交于 
      The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.

Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.

e.g. pcm:0,0 routing digital data to 2 external codecs.

FE pcm:0,0  ----> BE (McBSP.0) ----> CODEC 0
             +--> BE (McPDM.0) ----> CODEC 1

e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.

FE pcm:0,0 ---
             +--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---

The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.

DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.

Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.

This patch adds the core DPCM code and contains :-

 o The FE and BE PCM operations.
 o FE and BE DAI link support.
 o FE and BE PCM creation.
 o BE support API.
 o BE and FE link management.
Signed-off-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      01d7584c
  3. 26 4月, 2012 1 次提交
  5. 25 4月, 2012 5 次提交
  7. 24 4月, 2012 4 次提交
  9. 19 4月, 2012 4 次提交
  11. 18 4月, 2012 2 次提交
  13. 17 4月, 2012 8 次提交
    • A
      ASoC: da7210: Add support for PLL and SRM · 570aa7ba
      由 Ashish Chavan 提交于 
      Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.

This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,

(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled

This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.

Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.
Signed-off-by: NAshish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: NDavid Dajun Chen <dchen@diasemi.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      570aa7ba
    • M
      ASoC: Use dai_fmt in Speyside · 26e67811
      由 Mark Brown 提交于 
      Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      26e67811
    • F
      ASoC: soc-dapm: Use '%llx' with 'u64' type. · 516541a0
      由 Fabio Estevam 提交于 
      Fix the following build warning:

sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event':
sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64'

'%llx' should be used with 'u64' type.
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      516541a0
    • M
      ASoC: core: Support transparent CODEC<->CODEC DAI links · c74184ed
      由 Mark Brown 提交于 
      Rather than having the user half start a stream but avoid any DMA to
trigger data flow on links which don't pass through the CPU create a
DAPM route between the two DAI widgets using a hw_params configuration
provided by the machine driver with the new 'params' member of the
dai_link struct.  If no configuration is provided in the dai_link then
use the old style even for CODEC<->CODEC links to avoid breaking
systems.

This greatly simplifies the userspace usage of such links, making them
as simple as analogue connections with the stream configuration being
completely transparent to them.

This is achieved by defining a new dai_link widget type which is created
when CODECs are linked and triggering the configuration of the link via
the normal PCM operations from there.  It is expected that the bias
level callbacks will be used for clock configuration.

Currently only the DAI format, rate and channel count can be configured
and currently the only DAI operations which can be called are hw_params
and digital_mute().  This corresponds well to the majority of CODEC
drivers which only use other callbacks for constraint setting but there
is obviously much room for extension here.  We can't simply call
hw_params() on startup as things like the system clocking configuration
may change at runtime and in future it will be desirable to offer some
configurability of the link parameters.

At present we are also restricted to a single DAPM link for the entire
DAI.  Once we have better support for channel mapping it would also be
desirable to extend this feature so that we can propagate per-channel
power state over the link.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
      c74184ed
    • M
      ASoC: core: Bind DAIs to CODECs at registration time · 054880fe
      由 Mark Brown 提交于 
      We should always have a CODEC already there when registering a CODEC DAI
and for CODEC<->CODEC links a dai_link will have two CODECs so it's much
simpler to do things at registration time.

This results in a slight change in the error handling for failed CODEC
DAI registrations but practically speaking these are never supposed to
fail so there shouldn't be much issue. The change is that we don't fail
the overall CODEC registration if the DAI registration fails; this seems
more robust anyway as we may not need to use a given DAI in a particular
system.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      054880fe
    • M
      ASoC: core: Flip master for CODECs in the CPU slot of a CODEC<->CODEC link · f04209a7
      由 Mark Brown 提交于 
      When two CODEC DAIs are linked directly to each other then if we give the
same master mode settings to both devices things won't work as either
neither will drive or they'll drive against each other. Flip the settings
for the DAI in the CPU slot of the DAI link.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
      f04209a7
    • M
      ASoC: dapm: Allow DAI widgets to be routed through · 1eee1b38
      由 Mark Brown 提交于 
      In order to allow CODEC<->CODEC links to function we will need to allow
DAPM paths to be created that pass through DAIs rather than only ones
that are source or sunk at the DAI.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
      1eee1b38
    • M
      ASoC: core: Return -ENOTSUPP instead of -EINVAL if mute is not supported · 04570c62
      由 Mark Brown 提交于 
      This helps us ignore errors in callers if the operation failed due to not
being available as opposed to an error.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
      04570c62
  15. 15 4月, 2012 1 次提交
  17. 14 4月, 2012 4 次提交
  19. 13 4月, 2012 8 次提交