- 23 9月, 2011 2 次提交
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由 Dan Carpenter 提交于
Smatch has a new check for Rosenberg type information leaks where structs are copied to the user with uninitialized stack data in them. The status struct has a hole in it, and on some paths not all the members were initialized. struct hdspm_status { unsigned char card_type; /* 0 1 */ /* XXX 3 bytes hole, try to pack */ enum hdspm_syncsource autosync_source; /* 4 4 */ long long unsigned int card_clock; /* 8 8 */ The hdspm_version struct had holes in it as well. struct hdspm_version { unsigned char card_type; /* 0 1 */ char cardname[20]; /* 1 20 */ /* XXX 3 bytes hole, try to pack */ unsigned int serial; /* 24 4 */ short unsigned int firmware_rev; /* 28 2 */ /* XXX 2 bytes hole, try to pack */ int addons; /* 32 4 */ Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Use macro to improve readability. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 22 9月, 2011 2 次提交
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由 Ben Hutchings 提交于
Commit 96760015 ("ALSA: fm801: add error handling if auto-detect fails") seems to break systems that were previously working without a tuner. As a bonus, this should fix init and cleanup for the case where the tuner is explicitly disabled. Reported-and-tested-by: NHor Jiun Shyong <jiunshyong@gmail.com> References: http://bugs.debian.org/641946Signed-off-by: NBen Hutchings <ben@decadent.org.uk> Cc: stable@kernel.org [v3.0+] Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Ben Hutchings 提交于
Commit 96760015 ("ALSA: fm801: add error handling if auto-detect fails") added incorrect error handling. Once we have successfully called snd_device_new(), the cleanup function fm801_free() will automatically be called by snd_card_free() and we must *not* also call fm801_free() directly. Reported-by: NHor Jiun Shyong <jiunshyong@gmail.com> References: http://bugs.debian.org/641946Signed-off-by: NBen Hutchings <ben@decadent.org.uk> Cc: stable@kernel.org [v3.0+] Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 9月, 2011 5 次提交
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由 David Henningsson 提交于
Cc: stable@kernel.org BugLink: http://bugs.launchpad.net/bugs/854468Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Add the PCM rule to allow disabling the PCM playback SRC. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Add the PCM rule to allow disabling the PCM playback SRC. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Add the PCM rules to allow disabling the PCM playback and capture SRCs. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
The installation of the minimum period size constraint in the PCM open callbacks was not checked for errors. Add this check, and move the call to the beginning of the function to avoid having to do any cleanups in the error case. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 19 9月, 2011 1 次提交
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由 Takashi Iwai 提交于
When the system has only the headphone and the line-out jacks without speakers, the current auto-mute code doesn't work. It's because the spec->automute_lines flag is wrongly referred in update_speakers(). This flag must be meaningless when spec->automute_hp_lo isn't set, thus they should be always coupled. The patch fixes the problem and add a comment to indicate the relationship briefly. BugLink: http://bugs.launchpad.net/bugs/851697Reported-by: NDavid Henningsson <david.henningsson@canonical.com> Tested-By: NJayne Han <jayne.han@canonical.com> Cc: stable@kernel.org (3.0) Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 9月, 2011 2 次提交
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由 David Henningsson 提交于
This patch fixes "Surround Speaker Playback Volume" being cut off. (Commit b4dabfc4 was probably meant to fix this, but it fixed only the "Switch" name, not the "Volume" name.) Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
The semantics of snd_mpu401_uart_new()'s interrupt parameters are somewhat counterintuitive: To prevent the function from allocating its own interrupt, either the irq number must be invalid, or the irq_flags parameter must be zero. At the same time, the irq parameter being invalid specifies that the mpu401 code has to work without an interrupt allocated by the caller. This implies that, if there is an interrupt and it is allocated by the caller, the irq parameter must be set to a valid-looking number which then isn't actually used. With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value, which forces us to handle the parameters differently. This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the device interrupt is handled by the caller, and makes the allocation of the interrupt to depend only on the irq parameter. As suggested by Takashi, the irq_flags parameter was dropped because, when used, it had the constant value IRQF_DISABLED. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 9月, 2011 1 次提交
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由 Takashi Iwai 提交于
The recursive search of widget connections in snd_hda_get_conn_index() must be terminated at the pin and the audio-out widgets. Otherwise you'll get "too deep connection" warnings unnecessarily. Reported-by: NFrancis Moreau <francis.moro@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 9月, 2011 1 次提交
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由 Raymond Yau 提交于
YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth. Signed-off-by: NRaymond Yau <superquad.vortex2@gmail.com> Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 25 8月, 2011 1 次提交
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由 David Henningsson 提交于
Headphones has stopped working for the original reported (a regression compared to 2.6.38). This is because Speaker and Headphones share the same DAC, in which case no Headphones volume control was created. This patch fixes so that both Speaker and Headphones volume controls are created in such scenario. BugLink: http://bugs.launchpad.net/bugs/817943Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 24 8月, 2011 2 次提交
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由 Takashi Iwai 提交于
The ADC-switching can work also in the auto-mic mode, too. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Maarten Lankhorst 提交于
There are references in the code to 256 sources, so I tested it with 256 aplays, of which the first and last with real data and the rest playing /dev/zero . Also increase amount of page tables, so the default aplay size works. Signed-off-by: NMaarten Lankhorst <m.b.lankhorst@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 23 8月, 2011 3 次提交
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由 Takashi Iwai 提交于
When multiple headphones or speakers are assigned but no individual DACs are available, the driver should take the first HP/SPK DAC instead of another primary output. The patch adds a bit-flag to dac field of struct pin_dac_pair indicating that it's a slave DAC. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The internal states, jack_present and line_jack_present should be updated upon unsolicited events even if no automute is set. Otherwise the wrong state is referred when the automute behavior is changed by the mixer control. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
When the headphone or speaker output has no own DAC, initialize the path using the primary DAC. Otherwise the path won't be set properly and can result in the silence. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 8月, 2011 2 次提交
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由 Takashi Iwai 提交于
snd_hda_get_conn_index() returns a negative value while the current code stores it in an unsigned int. It must be stored in a signed integer. Reported-by: NJesper Juhl <jj@chaosbits.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Currently HD-audio driver shows the all error ELD byte as an error in the kernel message. This is annoying when the video driver doesn't set the correct ELD from the beginning. e.g. radeon sends a zero-byte data, but we still check ELD with the fixed 128 byte as a workaround for some broken devices, it spews 128-times errors. For avoiding this, the driver aborts reading when the first byte is invalid. In such a case, the whole data is certainly invalid. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 8月, 2011 1 次提交
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由 Takashi Iwai 提交于
The new parser may use "PCM" volume, but it was missing the vmaster slave list, thus "Master" volume didn't control it. Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 16 8月, 2011 1 次提交
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由 Takashi Iwai 提交于
Fix the duplicated creation of capture-mixer elements for some static ALC268 configurations. The capture mixers must be put to cap_mixer field instead of mixers array. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 15 8月, 2011 10 次提交
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由 Takashi Iwai 提交于
Refactoring the code using snd_pcm_hw_constraint_pow2() helper function. Tested-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
AES32 supports the non-standard 128kHZ, and this is enabled only when SNDRV_PCM_RATE_KNOT is set in hw.rates field. Tested-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Some modesl can support up to 8192 frames per period. Tested-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
On the Xonar Essence ST/STX, the connector J14 has been confirmed to be a digital input, so enable it in the driver. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel T Chen 提交于
BugLink: https://bugs.launchpad.net/bugs/826081 The original reporter needs 'Headphone Jack Sense' enabled to have audible audio, so add his PCI SSID to the whitelist. Reported-and-tested-by: Muhammad Khurram Khan Cc: <stable@kernel.org> Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Adrian Knoth 提交于
Newer RME cards like RayDAT and AIO support 32 samples per period. This value is encoded as {1,1,1} in the HDSP_LatencyMask bits in the control register. Since {1,1,1} is also the representation for 8192 samples/period on older RME cards, we have to special case 32 samples and 32768 bytes according to the actual card. Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Adrian Knoth 提交于
Currently, hdspm_decode_latency is called several times, violating the DRY principle. Given that we need to distinguish between old and new cards when decoding the latency bits in the control register, introduce hdspm_get_latency() to provide the required functionality. Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Adrian Knoth 提交于
On newer RME cards like RayDAT and AIO, the 8192 samples per period size are no longer supported. Instead, setting all three bits of HDSP_LatencyMask to one ({1,1,1}) now corresponds to 32 samples per period. To make this more obvious to future developers, let's reorder the array according to their bit representation, starting at 64 ({0,0,0}) up to 4096 ({1,1,0}) and finally 32 ({1,1,1}). Note that this patch doesn't change semantics. Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Adrian Knoth 提交于
On newer RME cards like RayDAT and AIO, the lower bound is 32 samples per period in contrast to 64 samples as seen on older cards. We hence lower period_bytes_min to 32 * 4. Four bytes per sample. Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Adrian Knoth 提交于
Older RME cards like MADI and AES support period sizes of 8192 samples. The original hdspm driver already featured this value, apparently, it was lost during the rewrite. Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 8月, 2011 2 次提交
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由 Julia Lawall 提交于
snd_azf3328_dbgcallenter is called at the very beginning of the function, so it could be useful to call snd_azf3328_dbgcallleave at all exit points. Signed-off-by: NJulia Lawall <julia@diku.dk> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Wang Shaoyan 提交于
In commit 45eebda7, it add new function stac_vrefout_set, but it is only used in code between CONFIG_SND_HDA_POWER_SAVE macro, so add the macro to avoid such warning: sound/pci/hda/patch_sigmatel.c:676:12: warning: 'stac_vrefout_set' defined but not used Signed-off-by: NWang Shaoyan <wangshaoyan.pt@taobao.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 07 8月, 2011 2 次提交
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由 Takashi Iwai 提交于
CONFIG_SND_TEA575X is enabled by RADIO_SF16FMR2, but the latter one is no PCI device. Since tea575x-tuner itself is independent from the board bus type, the config should be moved out of SND_PCI dependency. Reported-by: NRandy Dunlap <rdunlap@xenotime.net> Acked-by: NRandy Dunlap <rdunlap@xenotime.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Thomas Meyer 提交于
Use kzalloc rather than kmalloc followed by memset with 0 This considers some simple cases that are common and easy to validate Note in particular that there are no ...s in the rule, so all of the matched code has to be contiguous The semantic patch that makes this output is available in scripts/coccinelle/api/alloc/kzalloc-simple.cocci. More information about semantic patching is available at http://coccinelle.lip6.fr/Signed-off-by: NThomas Meyer <thomas@m3y3r.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 05 8月, 2011 2 次提交
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由 Wang Shaoyan 提交于
sound/pci/hda/patch_via.c:2087: warning: 'dac' may be used uninitialized in this function Signed-off-by: NWang Shaoyan <wangshaoyan.pt@taobao.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Put the exception checks for io_type switch() for possible mistakes in future. Also this shuts up annoying compile warnings. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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