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| * | | ASoC: TPA6130A2 amplifier driverPeter Ujfalusi2009-10-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-06
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| * | | ASoC: Add virtual enumeration support for DAPM muxesMark Brown2009-10-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sometimes it is desirable to have a mux which does not reflect any direct register configuration but which will instead only have an effect implicitly (for example, as a result of changing which parts of the device are powered up). Provide a virtual mux for this purpose. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: add support for multiple cards/codecs in debugfsPeter Ujfalusi2009-10-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In order to support multiple codecs on the same system in the debugfs the directory hierarchy need to be changed by adding directory per codec under the asoc direcorty: debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg /dapm_pop_time /dapm/{widgets} With the original implementation only the debugfs files are only created for the first codec, other codecs loaded later would fail to create the debugfs files (since they are already exist). Furthermore in this situation any of the codecs has been removed, would cause the debugfs entries to disappear, regardless if the codec, which created them are still loaded (the one which loaded first). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Add PDM DAI format definitionLopez Cruz, Misael2009-09-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Add DAI format definition for PDM interfaces. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-09-18
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| * | | | ASoC: Provide API for reordering channelsBarry Song2009-09-13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The patch adds an interface to set the relationship between audio channel number and slot number. The interface should be really useful because audio channel n doesn't always use slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: Allow per-route connectedness checks for suppliesMark Brown2009-09-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some chips with complex internal supply (particularly clocking) arragements may have multiple options for some of the supply connections. Since these don't affect user-visible audio routing the expectation would be that they would be managed automatically by one of the drivers. Support these users by allowing routes to have a connected function which is queried before the connectedness of the path is checked as normal. Currently this is only done for supplies, other widgets could be supported but are not currently since the expectation for them is that audio routing will be under the control of userspace. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: Add source argument to PLL configurationMark Brown2009-09-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | More and more devices feature PLLs and FLLs with the ability to select between multiple input clocks. In order to better support these devices a new argument, source, has been added to the set_pll() configuration API. Using set_clkdiv() is often difficult due to the need to stop the PLL/FLL before any reconfiguration can be done. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | Merge branch 'topic/misc' into for-linusTakashi Iwai2009-12-04
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| * | | | | ALSA: opti-miro: expose ACI mixer to outside driversKrzysztof Helt2009-11-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The ACI mixer is used to control the radio FM module installed on the Miro PCM20 sound card. Expose ACI mixer outside the sound card driver. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: opti-miro: make miro.h header available outside the alsa directoryKrzysztof Helt2009-11-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Move the miro.h header to the include/sound directory. It can be used in the Miro PCM20 radio driver (v4l). Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: cs4236: update control namesKrzysztof Helt2009-11-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Update control names to be more closer to their meaning. Change the "Mono" name to the "Beep" as this line is usually used to forward the PC beeper signal to sound card's output. Update names for both cs423x and wss. Clean up cs4235 controls according to the cs4235 doc. Rename some of the cs4235 controls to be consistent with the cs4236's ones. Also, delete one misnamed cs4231 register define. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: cs4236: detect chip in one passKrzysztof Helt2009-11-05
| | |_|_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The cs4236 was two step detection with call to the snd_wss_free() between two steps. The snd_wss_free() did not free a sound device created in the snd_wss_create(). This caused an OOPS during module removal as the same sound device was released twice. The same OOPS happened if the cs4236 module loading failed. Fix this by adapting the snd_cs4236_create() to correctly work with chips less capable then cs4236. The snd_cs4236_create() behaves the same as the snd_wss_create() if the chip is less capable than the cs4236. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'topic/beep-rename' into topic/core-changeTakashi Iwai2009-12-01
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| * | | | ALSA: sh: add SuperH DAC audio driver for ALSA V4Rafael Ignacio Zurita2009-11-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is a port of the sound/oss/sh_dac_audio.c driver. The driver uses an on-chip 8-bit D/A converter, which has a speaker connected to one of its channels, found in several ancient HP machines. For interrupts it uses a high-resolution timer (hrtimer). Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx). Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver would be obsolete soon, and it could be removed. Signed-off-by: Rafael Ignacio Zurita <rizurita@yahoo.com> Acked-by: Paul Mundt <lethal@linux-sh.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: sscape - Remove sscap_ioctl.h from include/sound/KbuildTakashi Iwai2009-10-02
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: sscape: convert to firmware loader frameworkKrzysztof Helt2009-10-01
| | |_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional firmware initialization code has been moved from the OSS driver. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | sound: rawmidi: record a substream's owner processClemens Ladisch2009-11-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Record the pid of the task that opened a RawMIDI substream. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | sound: pcm: record a substream's owner processClemens Ladisch2009-11-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Record the pid of the task that opened a PCM substream. For sound cards with hardware mixing, this allows determining which process is associated with a specific substream's volume control. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | control: use reference-counted pidClemens Ladisch2009-11-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Instead of storing the PID number, take a reference to the task's pid structure. This protects against duplicates due to PID overflows, and using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is correct as seen from the current namespace. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | control: remove snd_konctrol_volatile::owner_pid fieldClemens Ladisch2009-11-06
|/ / / | | | | | | | | | | | | | | | | | | | | | We do not need to save the ID of the process that locked a control because that information is already available in the owner's file data. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge branch 'topic/ymfpci' into for-linusTakashi Iwai2009-09-10
|\ \ \ | | | | | | | | | | | | | | | | * topic/ymfpci: sound: ymfpci: increase timer resolution to 96 kHz
| * | | sound: ymfpci: increase timer resolution to 96 kHzClemens Ladisch2009-08-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Allow the interval timer to be programmed with its full 96 kHz precision. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | Merge branch 'topic/tlv-minmax' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/tlv-minmax: ALSA: usb-audio - Correct bogus volume dB information ALSA: usb-audio - Use the new TLV_DB_MINMAX type ALSA: Add new TLV types for dBwith min/max
| * | | | ALSA: Add new TLV types for dBwith min/maxTakashi Iwai2009-06-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add new types for TLV dB scale specified with min/max values instead of min/step since the resolution can't match always with the one a device provides. For example, usb audio devices give 1/256 dB resolution while ALSA TLV is based on 1/100 dB resolution. The new min/max types have less problems because the possible rounding error happens only at min/max. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'topic/snd-printk' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/snd-printk: ALSA: Fixed a typo of printk() ALSA: Add debug module option ALSA: core - strip too long file names in snd_print*()
| * | | | | ALSA: Fixed a typo of printk()Takashi Iwai2009-08-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fixed a silly typo of printk() included in the previous patch... Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: Add debug module optionTakashi Iwai2009-08-27
| | |/ / / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add debug module option to snd core. This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE is set, you can suppress the debug messages by giving or changing this parameter to a lower value. debug=0 means no debug messsages. As default, it's set to the verbose level 2. Since this option can be changed dynamically via sysfs file, you can suppress the verbose debug messages on the fly, which wasn't possible before. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'topic/pcm-drain-nonblock' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/pcm-drain-nonblock: ALSA: pcm - Increase protocol version ALSA: pcm - Fix drain behavior in non-blocking mode
| * | | | | ALSA: pcm - Increase protocol versionTakashi Iwai2009-08-27
| |/ / / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Increase the PCM protocol version to indicate the drain ioctl behavior change. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'topic/misc' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/misc: ALSA: Remove unneeded ifdef from sound/core.h ALSA: Remove struct snd_monitor_file from public sound/core.h ALSA: Release v1.0.21
| * | | | | ALSA: Remove unneeded ifdef from sound/core.hTakashi Iwai2009-09-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove the old hack that was needed for building alsa-driver modules externally for old kernels. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: Remove struct snd_monitor_file from public sound/core.hTakashi Iwai2009-09-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The struct snd_monitor_file is used locally only in sound/core/init.c, thus it should be moved there from the public sound/core.h. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: Release v1.0.21Jaroslav Kysela2009-09-03
| |/ / / / | | | | | | | | | | | | | | | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'topic/dummy' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/dummy: ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128 ALSA: dummy - Add debug proc file ALSA: Add const prefix to proc helper functions ALSA: Re-export snd_pcm_format_name() function ALSA: dummy - Fake buffer allocations ALSA: dummy - Fix the timer calculation in systimer mode ALSA: dummy - Add more description ALSA: dummy - Better jiffies handling ALSA: dummy - Support high-res timer mode
| * | | | | ALSA: Add const prefix to proc helper functionsTakashi Iwai2009-09-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add appropriate const prefix to char * arguments in proc helper functions. Also fixed the caller side to be proper const pointers. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: Re-export snd_pcm_format_name() functionTakashi Iwai2009-09-08
| |/ / / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Re-export snd_pcm_format_name() function to be used outside the PCM core. As a first example, usbaudio is changed to use it now again. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'topic/dma-sgbuf' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ | |_|_|_|/ |/| | | | | | | | | | | | | | * topic/dma-sgbuf: ALSA: Fix SG-buffer DMA with non-coherent architectures
| * | | | ALSA: Fix SG-buffer DMA with non-coherent architecturesTakashi Iwai2009-07-08
| |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Using SG-buffers with dma_alloc_coherent() is often very inefficient on non-coherent architectures because a tracking record could be allocated in addition for each dma_alloc_coherent() call. Instead, simply disable SG-buffers but just allocate normal continuous buffers on non-supported (currently all but x86) architectures. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | / ASoC: Remove unuused hw_read_tMark Brown2009-09-07
| |_|/ |/| | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | Merge branch 'topic/digital-mixing' into for-2.6.32Mark Brown2009-08-24
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| * | | ASoC: Add input and output AIF widgetsMark Brown2009-08-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently DAPM interfaces with the audio streams to and from the processor at the DAC and ADC widgets. As the digital capabilities of parts increases this is becoming a less and less able to meet the needs of parts. To meet the needs of these devices create new widgets interfacing with the TDM bus but not integrated into any other functionality. Audio can then be routed to and from these widgets using existing routing widgets. A slot number is provided in the definition but this is currently not used yet. This is intended to support devices which can use more than one TDM slot on a single interface. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Add DAPM widget power decision debugfs filesMark Brown2009-08-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently when built with DEBUG DAPM will dump information about the power state decisions it is taking for each widget to dmesg. This isn't an ideal way of getting the information - it requires a kernel build to turn it on and off and for large hub CODECs the volume of information is so large as to be illegible. When the output goes to the console it can also cause a noticable impact on performance simply to print it out. Improve the situation by adding a dapm directory to our debugfs tree containing a file per widget with the same information in it. This still requires a decision to build with debugfs support but is easier to navigate and much less intrusive. In addition to the previously displayed information active streams are also shown in these files. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Add SuperH FSI driver support for ALSAKuninori Morimoto2009-08-20
|/ / / | | | | | | | | | | | | | | | | | | | | | | | | This driver is very simple. It support playback only now. This patch is tested by ms7724se board. Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ALSA: Allow passing platform_data for pxa2xx-ac97Marek Vasut2009-08-13
| | | | | | | | | | | | | | | | | | | | | | | | This patch adds support for passing platform data to ac97 bus devices from PXA2xx-AC97 driver.. Signed-off-by: Marek Vasut <marek.vasut@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | Merge commit 'a5479e389e989acfeca9c32eeb0083d086202280' into for-2.6.32Mark Brown2009-08-11
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| * | | ASoC: change set_tdm_slot api to allow slot_width override.Daniel Ribeiro2009-08-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Define more formats for the AC97 CODECsMark Brown2009-08-09
| | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | Merge branch 'reg-cache' into for-2.6.32Mark Brown2009-08-07
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