- 16 Mar, 2011 4 commits
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Takashi Iwai authored
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Takashi Iwai authored
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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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Stephen Rothwell authored
fixes this error: sound/firewire/fcp.c: In function 'fcp_avc_transaction': sound/firewire/fcp.c:103: error: implicit declaration of function 'msleep' Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 15 Mar, 2011 4 commits
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Clemens Ladisch authored
Add an AMDTP stream error state that occurs when we fail to queue another packet. In this case, the stream is stopped, and the error can be reported when the application tries to restart the PCM stream. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Clemens Ladisch authored
For correct cache coherency on some architectures, DMA buffers must be allocated in a different cache line than data that is concurrently used by the CPU. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Clemens Ladisch authored
In non-blocking mode, the SYT_INTERVAL is larger than the number of audio frames in each packet, so there are packets that do not contain any frame to which the SYT could be applied. For these packets, the SYT must not be the timestamp of the next valid SYT frame, but the special no-info SYT value. This fixes broken playback on the FireWave at 44.1 kHz. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Clemens Ladisch authored
Add a driver for two playback-only FireWire devices based on the OXFW970 chip. v2: better AMDTP API abstraction; fix fw_unit leak; small fixes v3: cache the iPCR value v4: FireWave constraints; fix fw_device reference counting; fix PCR caching; small changes and fixes v5: volume/mute support; fix crashing due to pcm stop races v6: fix build; one-channel volume for LaCie v7: use signed values to make volume (range checks) work; fix function block IDs for volume/mute; always use channel 0 for LaCie volume Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Acked-by: Stefan Richter <stefanr@s5r6.in-berlin.de> Tested-by: Jay Fenlason <fenlason@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 14 Mar, 2011 5 commits
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Takashi Iwai authored
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Vitaliy Kulikov authored
This patch replaces use of the harcoded arrays of pins, muxes, digital mics and adcs with the auto-generated ones using codec parsing and auto-discovers all actually connected digital mic pins on 92HD8X-like codecs This patch also adds the support for d-mic on pin 0x20. Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Vitaliy Kulikov authored
When the mux for digital mic is different from the mux for other mics, the current auto-parser doesn't handle them in a right way but provides only one mic. This patch fixes the issue. Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Move the default input-src selection code for alc268/269 to the init part instead of the parser. The input-src selection might be overwritten by init verbs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Currently some special handling for the unusual case like dual-ADCs or a single-input-src is done in the tree-parse time in set_capture_mixer(). But this setup could be overwritten by static init verbs. This patch moves the initialization into the init phase so that such input-src setup won't be lost. Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 11 Mar, 2011 5 commits
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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 6 commits
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Adam Lackorzynski authored
azx_init_pci() always writes PCI config register ICH6_PCIREG_TCSEL although this looks to be only defined on Intel systems and has a different meaning on AMD systems. On AMD systems the PCI interrupt pin control register is modified instead. Since the meaning of offset 0x44 in device specific configuration space is unknown for devices by other vendors, we only exclude AMD systems to retain the current behaviour. Signed-off-by: Adam Lackorzynski <adam@os.inf.tu-dresden.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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David Henningsson authored
Do not initialize again the what has already been initialized as multi outs, as this breaks surround speakers. Tested-by: Bartłomiej Żogała <nusch88@gmail.com> Cc: <stable@kernel.org> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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David Henningsson authored
Without this change, a volume control named "Surround" or "Side" would get an unnecessary index, causing it to be ignored by the vmaster and PulseAudio. Tested-by: Bartłomiej Żogała <nusch88@gmail.com> Cc: <stable@kernel.org> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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David Henningsson authored
When more than one pair of internal speakers is present, allow names according to their channels. Tested-by: Bartłomiej Żogała <nusch88@gmail.com> Cc: <stable@kernel.org> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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David Henningsson authored
The pin config values would change the association instead of the sequence, this commit fixes that up. Tested-by: Bartłomiej Żogała <nusch88@gmail.com> Cc: <stable@kernel.org> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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 4 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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