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* [ALSA] snd_emu10k1: Added support for 14dB Attenuation PADS on DACs and ADCs.James Courtier-Dutton2007-02-09
| | | | | Signed-off-by: James Courtier-Dutton <James@superbug.co.uk> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] hdsp: support for mixer matrix of RME9632 rev 152Remy Bruno2007-02-09
| | | | | | | | Added the support for mixer matrix of RME9632 rev 152. Signed-off-by: Remy Bruno <remy.bruno@trinnov.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] emu10k1: select FW_LOADERClemens Ladisch2007-02-09
| | | | | | | | Let the emu10k1 driver select FW_LOADER because the new Emu1010 support requires it. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] pci: select FW_LOADER instead of depending on itClemens Ladisch2007-02-09
| | | | | | | | | Let the AudioScience, Echoaudio and Riptide drivers select FW_LOADER instead of depending on it so that they can be configured without having to enable FW_LOADER manually. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] soc-core: fix multi-line string literalClemens Ladisch2007-02-09
| | | | | | | Properly quote a string that had an embedded newline. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC AT91RM92000 buildFrank Mandarino2007-02-09
| | | | | | | | | | This patch adds a Makefile and Kconfig to build the ASoC AT91RM9200 support. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC AT91RM92000 eti_b1 machine supportFrank Mandarino2007-02-09
| | | | | | | | | | This patch adds support for the Endrelia ETI_B1 machine using the WM8731 codec and the AT91RM9200 platform. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC AT91RM92000 I2S supportFrank Mandarino2007-02-09
| | | | | | | | | | | | | | This patch adds I2S support to the Atmel AT91RM9200 CPU. Features:- o Playback/Capture supported. o 16 Bit data size. o 8k - 48k sample rates. o ssc0, ssc1 and ssc2 supported as I2S ports. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC AT91RM92000 audio DMAFrank Mandarino2007-02-09
| | | | | | | | | | | | This patch adds ASoC audio DMA support to the Atmel AT91RM9200 CPU. Features:- o Playback/Capture supported. o 16 Bit data size. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC codecs: build filesRichard Purdie2007-02-09
| | | | | | | | | | This patch adds an ASoC Makefile and Kconfig for the WM8731, WM8750 and WM9712 codecs. Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC codecs: generic AC97 supportRichard Purdie2007-02-09
| | | | | | | | | | | This patch allows the std Alsa AC97 codec driver to use any AsoC AC97 controller driver. Currently, only HiFi playback and Capture are supported atm. Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC codecs: WM9712 supportRichard Purdie2007-02-09
| | | | | | | | | | | | | | This patch adds ASoC support for the WM9712 codec. Supported features:- o Capture/Playback/Sidetone/Bypass. o Aux DAC. o 8k - 48k sample rates. o DAPM. Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC codecs: WM8750 supportRichard Purdie2007-02-09
| | | | | | | | | | | | | | This patch adds ASoC support for the WM8750 codec. Supported features:- o Capture/Playback/Sidetone/Bypass. o 16 & 24 bit audio. o 8k - 96k sample rates. o DAPM. Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC codecs: WM8731 supportRichard Purdie2007-02-09
| | | | | | | | | | | | | | This patch adds ASoC support for the WM8731 codec. Supported features:- o Capture/Playback/Sidetone/Bypass. o 16 & 24 bit audio. o 8k - 96k sample rates. o DAPM. Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC: Build filesLiam Girdwood2007-02-09
| | | | | | | | | This patch adds support for building the ASoC core and the dynamic audio power management support. Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC: dynamic audio power management (DAPM)Richard Purdie2007-02-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ASoC: core codeFrank Mandarino2007-02-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] intel8x0 - Use pci_iomapTakashi Iwai2007-02-09
| | | | | | | | | Use pci_iomap and ioread*/iowrite*() functions for accessing hardwares. pci_iomap is suitable for hardwares like ICH and compatible that have both PIO and MMIO. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] pcm core: add prealloc_max file to substream directory to show ↵Jaroslav Kysela2007-02-09
| | | | | | | | | maximum DMA size Users ask us many times about the maximum DMA size for PCM devices. This file gives them a hint in KB. Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] hda_intel: increase maximum DMA buffer size to 1024MBJaroslav Kysela2007-02-09
| | | | | | See ALSA bug#2481 . Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] emu10k1 - Fix compile warningTakashi Iwai2007-02-09
| | | | | | | Fixed a compile warning regarding print format for size_t. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] alsa core: convert to list_for_each_entry*Johannes Berg2007-02-09
| | | | | | | | | | | This patch converts most uses of list_for_each to list_for_each_entry all across alsa. In some place apparently an item can be on a list with different pointers so of course that isn't compatible with list_for_each, I therefore didn't touch those places. Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] aoa: fix up i2sbus_attach_codecJohannes Berg2007-02-09
| | | | | | | | | | | | This patch changes i2sbus_attach_codec to implement a proper error handling strategy using labels to jump to the right part. Since it has an elaborate set-up sequence it also needs that tear-down, which I had hard-coded inbetween all the checks. This increases readability and should reduce .text size as well. Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] aoa: set device pointer in pcmsJohannes Berg2007-02-09
| | | | | | | | | | This patch makes a few whitespace cleanups and makes i2sbus assign the new struct device pointer in struct snd_pcm so that the proper device symlink shows up in sysfs. Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] alsa core: add struct device pointer to struct snd_pcmJohannes Berg2007-02-09
| | | | | | | | | | | This patch adds a struct device pointer to struct snd_pcm in order to be able to give it a different device than the card. It defaults to the card's device, however, so it should behave identically for drivers not touching the field. Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] allow registering an alsa device with struct device pointerJohannes Berg2007-02-09
| | | | | | | | | | This patch adds snd_register_device_for_dev taking a struct device pointer to link the new device to and makes snd_register_device a simple static inline wrapper around it. Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] Enable the analog loopback of the Revolution 5.1Jochen Voss2007-02-09
| | | | | | | | | | | Enable the analog loopback of the Revolution 5.1 card. This patch adds support for the PT2258 volume controller and modifies the Revolution 5.1 driver to make use of this facility. This allows to control the analog loopback of the card. Signed-off-by: Jochen Voss <voss@seehuhn.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] Enable capture from line-in and CD on Revolution 5.1Jochen Voss2007-02-09
| | | | | | | | | | | | Enable capture from line-in and CD on the Revolution 5.1 card. This patch adds support for switching between the 5 input channels of the AK5365 ADC and modifies the Revolution 5.1 driver to make use of this facility. Previously the capture channel was fixed to channel 0 (microphone on the Revolution 5.1 card). Signed-off-by: Jochen Voss <voss@seehuhn.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] usb-audio: allow pausingClemens Ladisch2007-02-09
| | | | | | | Add pause capabilities for both USB playback and capture streams. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] usb-audio: merge playback/capture hardware information structsClemens Ladisch2007-02-09
| | | | | | | | | The hardware information structures for playback and capture streams, respectively, are the same, so we can use just one structure for both streams. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] snd-emu10k1: Added support for emu1010, including E-Mu 1212m and E-Mu ↵James Courtier-Dutton2007-02-09
| | | | | | | 1820m Signed-off-by: James Courtier-Dutton <James@superbug.co.uk> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [PATCH] ALSA: Fix sysfs breakageTakashi Iwai2007-01-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent change for a new sysfs tree with card* object breaks the /sys/class/sound tree if CONFIG_SYSFS_DEPRECATED is enabled. The device in each entry doesn't point the correct device object: /sys/class/sound ... |-- pcmC0D0c | |-- dev | |-- device -> ../../../class/sound/card0 | |-- pcm_class | |-- power | | `-- wakeup | |-- subsystem -> ../../../class/sound | `-- uevent Also, this change breaks some drivers (like sound/arm/*) referring card->dev directly to obtain the device object for memory handling. This patch reverts the semantics of card->dev to the former version, which points to a real device object. The card* object is stored in a new card->card_dev field, instead. The device parent is chosen either card->dev or card->card_dev according to CONFIG_SYSFS_DEPRECATED to keep the tree compatibility. Also, card* isn't created if CONFIG_SYSFS_DEPRECATED is enabled. The reason of card* object is a root of all beloing devices, and it makes little sense if each sound device points to the real device object directly. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Monty Montgomery <xiphmont@gmail.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* [ALSA] Repair snd-usb-usx2y over OHCIKarsten Wiese2007-01-23
| | | | | | | | | | | | | The previous patch 'Repair snd-usb-usx2y for usb 2.6.18' assumed urb->start_frame roll over beyond MAX_INT for both UHCI & OHCI. This isn't true until now (kernel 2.6.20). Fix this by only looking at the common between OHCI & UHCI Frame number range. This is for mainline and stable kernels >= 2.6.18. Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] usbaudio - Fix kobject_add() error at reconnectionTakashi Iwai2007-01-09
| | | | | | | | | | Fixed the error from kobject_add() at reconnection the usb audio device. This happens when an app keeps opening a device while the device is replugged, due to the confliction of the internal bookkept index and the really empty slot. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] usb: usbmixer error path fixMariusz Kozlowski2007-01-09
| | | | | | | | | Without the patch below namelist[0] will not be freed in case of kmalloc error. Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] _snd_cmipci_uswitch_put doesn't set zero flagsTimofei V. Bondarenko2007-01-09
| | | | | | | | | | | | | | | | Playing with spdif output on cmipci i've noticed the SPDO5V option does not change appropriate bits the register. The _snd_cmipci_uswitch_put checks the change in flags in wrong way. If 'active' state of an option corresponds to a _zero_ bits in a hw register then function fails. The SPDO5V is the sample. In the most cases 'active' state of option is set through an non-zerio bits in a register. This case works fine. The fix attached. Unfortunately i was unable to change spdif output voltage anyway. Although the register changes right at least. From: Timofei V. Bondarenko <tim@ipi.ac.ru> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] hda-codec - Fix NULL dereference in generic hda codeTakashi Iwai2007-01-09
| | | | | | | Fix NULL dereference in hda_generic.c. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] hda_intel: ALSA HD Audio patch for Intel ICH9Jason Gaston2007-01-09
| | | | | | | | This patch adds the Intel ICH9 HD Audio controller DID's for ALSA. Signed-off-by: Jason Gaston <jason.d.gaston@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] usb-audio: work around wrong frequency in CM6501 descriptorsClemens Ladisch2007-01-09
| | | | | | | | The C-Media CM6501 chip's descriptors say that altsetting 5 supports 48 kHz, but it actually plays at 96 kHz. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] Fix potential NULL pointer dereference in echoaudio midiGiuliano Pochini2007-01-09
| | | | | | | | Fix races between the timer handler and the close function. Signed-off-by: Giuliano Pochini <pochini@shiny.it> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] Audio: Add nvidia HD Audio controllers of MCP67 support to hda_intel.cPeer Chen2007-01-09
| | | | | | | | Add the support for HD audio controllers of MCP51,MCP55,MCP61,MCP65 & MCP67. Signed-off-by: Peer Chen <pchen@nvidia.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [SOUND] Sparc CS4231: Use 64 for period_bytes_minDavid S. Miller2007-01-03
| | | | | | | | This matches what the ISA cs4231 driver uses. Tested by Georg Chini. Signed-off-by: David S. Miller <davem@davemloft.net>
* [SOUND] Sparc CS4231: Fix IRQ return value and initialization.Georg Chini2007-01-03
| | | | | | | | | | | SBUS: Change IRQ-handler return value from 0 to IRQ_HANDLED and fix some initialisation problems. Change period_bytes_min from 4096 to 256 to allow driver to work with low latency (VOIP) applications. Hope this does not break EBUS. Signed-off-by: Georg Chini <georg.chini@triaton-webhosting.com> Signed-off-by: David S. Miller <davem@davemloft.net>
* [PATCH] sound: hda: detect ALC883 on MSI K9A Platinum motherboards (MS-7280)Leonard Norrgård2007-01-01
| | | | | | | | Recognize the Realtek ALC883 chip on MSI K9A Platinum motherboards (model no. MS-7280), enabling full sound capabilities. Signed-off-by: Leonard Norrgård <leonard.norrgard@refactor.fi> Signed-off-by: Linus Torvalds <torvalds@osdl.org>
* [ALSA] ac97: Identify CMI9761 chips.James Courtier-Dutton2006-12-20
| | | | | Signed-off-by: James Courtier-Dutton <James@superbug.co.uk> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] ac97_codec - trivial fix for bit update functionsJames C Georgas2006-12-20
| | | | | | | | | | | This patch fixes a couple of bit update functions in alsa-kernel/pci/ac97/ac97_codec.c, which could possibly corrupt bits not in the given mask. Specifically, it'll clobber unset bits in the target that are not in the mask, when the corresponding bit in the given new value is set. Signed-off-by: James C Georgas <jgeorgas@rogers.com> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] snd-ca0106: Fix typos.James Courtier-Dutton2006-12-20
| | | | | Signed-off-by: James Courtier-Dutton <James@superbug.co.uk> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] snd-ca0106: Add new card variant.James Courtier-Dutton2006-12-20
| | | | | | | Fixed ALSA bug#2326 Signed-off-by: James Courtier-Dutton <James@superbug.co.uk> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] sound: fix PCM substream listAkinobu Mita2006-12-20
| | | | | | | | | | | | If snd_pcm_new_stream() fails to initalize a substream (if snd_pcm_substream_proc_init() returns error), snd_pcm_new_stream() immediately return without unlinking that kfree()d substram. It causes oops when snd_pcm_free() iterates the list of substream to free them by invalid reference. Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* [ALSA] sound: initialize rawmidi substream listAkinobu Mita2006-12-20
| | | | | | | | | | | | | | If snd_rawmidi_new() failed to allocate substreams for input (snd_rawmidi_alloc_substreams() failed to populate a &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT]), it will try to free rawmidi instance by snd_rawmidi_free(). But it will cause oops because snd_rawmidi_free() tries to free both of substreams list but list for output (&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]) is not initialized yet. Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
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