- 22 Mar, 2011 3 commits
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Mark Brown authored
With appropriate firmware the WM8958 can support Virtual Surround Sound or VSS, widening the stereo audio image for improved user experience. Enable support for this mode of operation when the appropriate firmware can be loaded at runtime. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
In preparation for the addition of additional WM8958 algorithms reorganise the code to make it easier to add such support later. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
Allow userspace to supply an update to the ROM firmware. The firmware request is non-blocking so userspace can load the firmware at its leisure without delaying startup, the driver will begin using the firmware the next time MBC is started after it has been supplied. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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- 18 Mar, 2011 6 commits
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
The DSP2 startup requires that the clock be enable so if we've deferred clock startup we need to defer DSP2 startup Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
DSP2 on the WM8958 has a default ROM which provides a multi-band compressor for enhanced performance on mobile devices but can also support runtime download of alternative firmware. In preparation for more exploiting this functionality refactor the code to split the handling of DSP2 into a separate file. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
The first WM8958 revision requires similar treatment. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Since not all registers need to be cached and the cache is entirely optional anyway we shouldn't be checking that a register is in the cached range. If the register is invalid then the actual I/O code can determine that and report an error. Similarly, the step size can and should be enforced by the lower level code if it's important. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.39
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- 16 Mar, 2011 1 commit
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Takashi Iwai authored
Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
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- 11 Mar, 2011 6 commits
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Marek Belisko authored
ASoC audio for mini2440 platform in current kenrel doesn't work. First problem is samsung_asoc_dma device is missing in initialization. Next problem is with codec. Codec is initialized but never probed because no platform_device exist for codec driver. It leads to errors during codec binding to asoc dai. Next problem was platform data which was passed from board to asoc main driver but not passed to codec when called codec_soc_probe(). Following patch should fix issues. But not sure if in correct way. Please review. Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Vasily Khoruzhick authored
MONO was renamed to MONO1. Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
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Vasily Khoruzhick authored
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Christian Glindkamp authored
This patch adds ASoC support for the MAX9850 codec with headphone amplifier. Supported features: - Playback - 16, 20 and 24 bit audio - 8k - 48k sample rates - DAPM Signed-off-by: Christian Glindkamp <christian.glindkamp@taskit.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Takashi Iwai authored
Added a new API function snd_ctl_activate_id() for activate / inactivate the control element dynamically. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- 10 Mar, 2011 1 commit
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Mark Brown authored
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- 09 Mar, 2011 12 commits
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Mark Brown authored
Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Cc: stable@kernel.org
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Mark Brown authored
Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Cc: stable@kernel.org
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Alexander Sverdlin authored
Enable 192kHz sample rate for EP93xx. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Alexander Sverdlin authored
Improve EP93xx I2S clocks management. Some freqs values are set not exact as they requested for MCLK and original code was not able to find divisors for SCLK and LRCLK. This code just picks up nearest value from 3 possible variants. This patch makes 44100 and 192000 rates working and fixes capture function (by selecting SCLK/LRCLK=64 where possible). All other rates should work as before. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Alexander Sverdlin authored
Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx machine driver. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Alexander Sverdlin authored
Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC. This is done to make capture work properly. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Conflicts: sound/soc/codecs/wm8978.c sound/soc/soc-dapm.c
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Mark Brown authored
We're not only prefixing all controls, we're also prefixing the widget names in the runtime data. This causes us to add the prefix twice - once when using the widget name to generate the control name and once when adding the control. Really we shouldn't be prefixing the widget names at all, the matching code should be handing this as we always know which DAPM context a widget came from and always display the widget name in terms of a DAPM context. However, we're quite close to the merge window and that's relatively invasive. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Reported-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
Now we've got multi-component we need to make sure that the DAPM context (and hence register I/O context) we use to apply the pending updates at the end of a DAPM sequence is the one we were processing rather than the one that was used to initate the state change. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
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Mark Brown authored
Now we have a register write minimisation code in DAPM we don't need to worry about the ordering of the enable and disable of the PGA and the output stage. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Jarkko Nikula authored
McBSP sidetone is needed in telephony applications. McBSP sidetone is a configurable FIR filter that forms a loopback from McBSP input to output. This patch enables the McBSP2 sidetone ALSA controls so that it can be used on Nokia RX-51/N900. Sidetone feature can be tested with following commands: (set up codec input and output paths) # Enable and configure sidetone amixer -D hw:0 set 'McBSP2 Sidetone' on amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767 echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps # Do not loop audio via CPU arecord -f dat >/dev/null |aplay /dev/zero Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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- 08 Mar, 2011 9 commits
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Dan Carpenter authored
The "ldo" variable was dereferenced after free on the error path. Signed-off-by: Dan Carpenter <error27@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Mark Brown authored
Currently will ignore prefixes when creating DAPM controls. Since currently all control creation goes through snd_soc_cnew() we can fix this by factoring the prefixing into that function. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
Symmetric rate configuration can fail if the second stream starting tries to apply the symmetric constraint before the first stream has got far enough to pick a rate. Rather than try to enforce a nonsensical rate of 0Hz log a warning and allow the application to carry on. Things might go wrong later on but the user will know about it and there's unlikely to be lasting damage. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
Also respace the CODEC ops a bit for legibility. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
When multi component systems use DAIless amplifiers which require clocking configuration it is at best hard to use the current clocking API as this requires a DAI even though the device may not even have one. Address this by adding set_sysclk() and set_pll() operations and APIs for CODECs. In order to avoid issues with devices which could be used either with or without DAIs make the DAI variants call through to their CODEC counterparts if there is no DAI specific operation. Converting over entirely would create problems for multi-DAI devices which offer per-DAI clocking setup. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
Annoying as the __devinitdata is actually correct. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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Mark Brown authored
Allow a slight simplification of CODEC drivers by allowing DAPM routes and widgets to be provided in a table. They will be instantiated at the end of CODEC probe. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
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- 07 Mar, 2011 2 commits
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Alexander Sverdlin authored
Remove warnings in ep93xx-i2s.c Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Ryan Mallon <ryan@bluewatersys.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Alexander Sverdlin authored
Extend range of supported sample rates for CS4271 CODEC. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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