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* ALSA: USB MIDI support for Access Music VirusTISebastien Alaiwan2010-02-16
| | | | | | | | | | | | | | | | | | | | Here's a patch that adds MIDI support through USB for one of the Access Music synths, the VirusTI. The synth uses standard USBMIDI protocol on its USB interface 3, although it does signal "vendor specific" class. A magic string has to be sent on interface 3 to enable the sending of MIDI from the synth (this string was found by sniffing usb communication of the Windows driver). This is all my patch does, and it works on my computer. Please note that the synth can also do standard usb audio I/O on its interfaces 2&3, which already works with the current snd-usb-audio driver, except for the audio input from the synth. I'm going to work on it when I have some time. Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator) Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: usbmixer - add possibility to remap dB valuesJaroslav Kysela2010-02-11
| | | | | | | | | | | USB devices tends to represent dB ranges in different way than ALSA expects. Add possibility to override these values and add guessed values for SoundBlaster MP3+. Also rename 'Capture Input Source' control to 'Capture Source' for SoundBlaster MP3+ and Extigy. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: dummy driver - add model parameterJaroslav Kysela2010-02-02
| | | | | | | This is a cleanup for the dummy driver. The model kernel module parameter is introduced to select the soundcard emulation. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* sound: control: fix minimum TLV lengthClemens Ladisch2010-02-01
| | | | | | | | Allow TLV blocks that do not have any values; the smallest possible TLV is an empty container or one where the information is only in the tag. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* sound: control: actually allow TLV command accessClemens Ladisch2010-02-01
| | | | | | | | | Creating a control with TLV_COMMAND access was not possible because snd_ctl_new1() forgot to include it in the mask of allowable access bits. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: pcm_native - fix runtime->boundary calculationJaroslav Kysela2010-01-27
| | | | | | | | | | | | The code in pcm_lib updating runtime->hw_ptr_interrupt expects that runtime->boundary is divisible with runtime->period_size. Thanks are going to Clemens Ladisch for the notice. Fix the runtime->boundary calculation using buffer_size * period_size as base and find a least common multiple for 32bit platforms when the expression might overflow. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: pcm_lib - return back hw_ptr_interruptJaroslav Kysela2010-01-26
| | | | | | | | | | | | | | | | | | | | Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr update functions" commit: "It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data." Return back the hw_ptr_interrupt variable. The last interrupt pointer is always computed from the latest hw_ptr instead of tracking it separately (in this case all hw_ptr checks and modifications might influence also hw_ptr_interrupt and it is difficult to keep it consistent). Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: hda - AD1988 codec - fix SPDIF-input mixer initialization (unmute)Jaroslav Kysela2010-01-21
| | | | | | | The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate pin to get captured samples instead zeros. Tested on Lenovo Thinkstation. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: pcm_core: Fix wake_up() optimizationJaroslav Kysela2010-01-21
| | | | | | | | This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O" commit. New sleeping queue is introduced to separate user space and kernel space wake_ups. runtime->nowake is renamed to twake (transfer wake). Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* sound: virtuoso: add Xonar DS supportClemens Ladisch2010-01-18
| | | | | | | Add experimental support for the Asus Xonar DS. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* sound: seq_timer: simplify snd_seq_timer_set_tick_resolution() parametersClemens Ladisch2010-01-18
| | | | | | | | | As snd_seq_timer_set_tick_resolution() is always called with the same three fields of struct snd_seq_timer, it suffices to give that as the only parameter. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: pcm_lib - fix wrong delta print for jiffies checkJaroslav Kysela2010-01-13
| | | | | | | The previous jiffies delta was 0 in all cases. Use hw_ptr variable to store and print original value. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: hda - support OLPC XO-1.5 DC inputDaniel Drake2010-01-08
| | | | | | | | | | The XO's audio hardware is wired up to allow DC sensors (e.g. light sensors, thermistors, etc) to be plugged in through the microphone jack. Add sound mixer controls to allow this mode to be enabled and tweaked. Signed-off-by: Daniel Drake <dsd@laptop.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: hda - Configure XO-1.5 microphones at capture timeDaniel Drake2010-01-08
| | | | | | | | | | | | | | The XO-1.5 has a microphone LED designed to indicate to the user when something is being recorded. This light is controlled by the microphone bias voltage and it is currently coming on all the time. This patch defers the microphone port configuration until when recording is actually taking place, fixing the behaviour of the LED. Signed-off-by: Daniel Drake <dsd@laptop.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* Merge branch 'topic/hda' of ↵Jaroslav Kysela2010-01-08
|\ | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into devel
| * ALSA: hda: Refactor powerdown for Realtek HDA codecsDaniel T Chen2009-12-28
| | | | | | | | | | | | | | | | | | | | | | | | | | This patch converts the alc889 Aspire-specific powerdown to a generic one. Like the previous effort, it currently only handles Front and PCM but can be easily extended to cover other nids. The existing hook for alc889 Aspire-specific remains enabled. Upon further testing, I've added its use for ALC861_AUTO as well. Following patches will enable them for other quirks. Tested-by: Dr. David Alan Gilbert <linux@treblig.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda: Add powerdown for Analog Devices HDA codecsDaniel T Chen2009-12-28
| | | | | | | | | | | | | | | | | | This patch ports powerdown fixes to AD198x. Currently we only turn off Front and HP for suspend, but this is easily extended for additional nids. Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Use strict_strtoul()Takashi Iwai2009-12-27
| | | | | | | | | | | | Rewrite the codes to use strict_strtoul() instead of simple_strtoul(). Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Add sanity check for storing the user-defined pin configsTakashi Iwai2009-12-27
| | | | | | | | | | | | | | Check whether the given NID is a pin widget before storing the user-defined pin configs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix click noises at suspend/free with Realtek codecsTakashi Iwai2009-12-27
| | | | | | | | | | | | Call snd_hda_shutup_pins() at suspend and free for avoiding click noises. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Add snd_hda_shutup_pins() helper functionTakashi Iwai2009-12-27
| | | | | | | | | | | | | | Add a common helper function for clearing pin controls before suspend. Use the pincfg array instead of looking through all widget tree. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Add more hints for GPIO setup of IDT/STAC codecsTakashi Iwai2009-12-26
| | | | | | | | | | | | gpio_led, gpio_led_polarity and gpio_mute are added now. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge branch 'fix/hda' into topic/hdaTakashi Iwai2009-12-25
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| | * ALSA: hda - HDMI sticky stream tag supportWu Fengguang2009-12-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When we run the following commands in turn (with CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0), speaker-test -Dhw:0,3 -c2 -twav # HDMI speaker-test -Dhw:0,0 -c2 -twav # Analog The second command will produce sound in the analog lineout _as well as_ HDMI sink. The root cause is, device 0 "reuses" the same stream tag that was used by device 3, and the "intelhdmi - sticky stream id" patch leaves the HDMI codec in a functional state. So the HDMI codec happily accepts the audio samples which reuse its stream tag. The proposed solution is to remember the last device each azx_dev was assigned to, and prefer to 1) reuse the azx_dev (and hence the stream tag) the HDMI codec last used 2) or assign a never-used azx_dev for HDMI With this patch and the above two speaker-test commands, HDMI codec will use stream tag 8 and Analog codec will use 5. The stream tag used by HDMI codec won't be reused by others, as long as we don't run out of the 4 playback azx_dev's. The legacy Analog codec will continue to use stream tag 5 because its device id is 0 (this is a bit tricky). Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * Merge branch 'fix/hda' into for-linusTakashi Iwai2009-12-23
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| | | * ALSA: hda - Add STAC9205 PCI_QUIRK for Dell Vostro 1700Anisse Astier2009-12-23
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Anisse Astier <anisse@astier.eu> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * ALSA: hda - Set mixer name after codec patchTakashi Iwai2009-12-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Postpone the mixer name setup after the codec patch since the codec patch may change the codec name string in itself. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | Merge branch 'fix/asoc' into for-linusTakashi Iwai2009-12-23
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| | | * | ASoC: Do not write to invalid registers on the wm9712.Eric Millbrandt2009-12-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch fixes a bug where "virtual" registers were being written to the ac97 bus. This was causing unrelated registers to become corrupted (headphone 0x04, touchscreen 0x78, etc). This patch duplicates protection that was included in the wm9713 driver. Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | | * | ASoC: add missing parameter to mx27vis_hifi_hw_free()Guennadi Liakhovetski2009-12-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 2ccafed4 added an extra parameter to the DAI .set_pll() method, but it missed this call in sound/soc/imx/mx27vis_wm8974.c. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | ASoC: sh: FSI:: don't check platform_get_irq's return value against zeroUwe Kleine-König2009-12-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | platform_get_irq returns -ENXIO on failure, so !irq was probably always true. Better use (int)irq <= 0. Note that a return value of zero is still handled as error even though this could mean irq0. This is a followup to 305b3228f9ff4d59f49e6d34a7034d44ee8ce2f0 that changed the return value of platform_get_irq from 0 to -ENXIO on error. Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | Merge branch 'fix/misc' into for-linusTakashi Iwai2009-12-23
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| | | * | | ALSA: sound/core/pcm_timer.c: use lib/gcd.cFlorian Fainelli2009-12-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Make sound/core/pcm_timer.c use lib/gcd.c Signed-off-by: Florian Fainelli <florian@openwrt.org> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | | Merge branch 'fix/misc' into for-linusTakashi Iwai2009-12-21
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| | | * | | ALSA: sbawe: fix memory detectionKrzysztof Helt2009-12-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Memory amount is increased before a successful write-read sequence is done. Thus, 512 kB of onboard memory is detected on memoryless cards like SB32. Move the increasing of memory counter after successful read is done. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: fix incorrect rounding direction in snd_interval_ratnum()Krzysztof Helt2009-12-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The direction of rounding is incorrect in the snd_interval_ratnum() It was detected with following parameters (sb8 driver playing 8kHz stereo file): - num is always 1000000 - requested frequency rate is from 7999 to 7999 (single frequency) The first loop calculates div_down(num, freq->min) which is 125. Thus, a frequency range's minimum value is 1000000 / 125 = 8000 Hz. The second loop calculates div_up(num, freq->max) which is 126 The frequency range's maximum value is 1000000 / 126 = 7936 Hz. The range maximum is lower than the range minimum so the function fails due to empty result range. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: Use kzalloc for allocating only one thingJulia Lawall2009-12-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use kzalloc rather than kcalloc(1,...) The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ @@ - kcalloc(1, + kzalloc( ...) // </smpl> Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: AACI: switch to per-pcm lockingRussell King2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We can use finer-grained locking, which makes things easier when we gain DMA support. Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: AACI: add double-rate supportRussell King2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_paramsRussell King2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: AACI: cleanup aaci_pcm_hw_paramsRussell King2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since the recording and playback paths are now the same, eliminate the needless conditionals. Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: AACI: simplify codec rate informationRussell King2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There's no need for a specific rule; ALSA's generic AC'97 support calculates the necessary rate constraint information itself, and we can use this directly. Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: aaci - Fix a typoTakashi Iwai2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fixed a typo of the max buffer size specified for buffer allocation changed in the commit d6797322231af98b9bb4afb175dd614cf511e5f7. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | sound: sgio2audio/pdaudiocf/usb-audio: initialize PCM bufferClemens Ladisch2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When allocating the PCM buffer, use vmalloc_user() instead of vmalloc(). Otherwise, it would be possible for applications to play the previous contents of the kernel memory to the speakers, or to read it directly if the buffer is exported to userspace. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | sound/oss/pss: Fix test of unsigned in pss_reset_dsp() and pss_download_boot()Roel Kluin2009-12-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | limit and jiffies are unsigned so the test did not work. Signed-off-by: Roel Kluin <roel.kluin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2009-12-21
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| | | * | | ASoC: wm8974: fix a wrong bit definitionGuennadi Liakhovetski2009-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The wm8974 datasheet defines BUFIOEN as bit 2. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | | * | | ASoC: Fix disable of SPDIF on STAC9766 codecJon Smirl2009-12-16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Change code so that switching to playing music through the analog output disables SPDIF out instead of disabling it when stream ends. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | ASoC: ak4642: Add default return value in ak4642_modinitKuninori Morimoto2009-12-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If ak4642 driver was compiled without I2C configs, ak4642_modinit return value will become un-stable. This patch modify this bug Reported-by: Magnus Damm <damm@opensource.se> Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | Merge branch 'fix/hda' into for-linusTakashi Iwai2009-12-21
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