- 18 3月, 2013 9 次提交
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由 Takashi Iwai 提交于
Instead of calling snd_hda_attach_beep_device() and snd_hda_detach_beep_device() in each codec driver, move them to the generic parser. The codec driver just needs to set spec->beep_nid for activating the digital beep. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Back-merged for refactoring beep stuff.
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由 Takashi Iwai 提交于
The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
This leaks the beep input device after module unload, which leads to Oops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
"Playback Design" products need a 50ms delay after setting the USB interface. Signed-off-by: NDaniel Mack <zonque@gmail.com> Reported-by: NAndreas Koch <andreas@akdesigninc.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
UAC2 compliant audio devices may announce the capability to transport raw audio data on their endpoints. Catch this and handle it as 'special' stream on the ALSA side. Signed-off-by: NDaniel Mack <zonque@gmail.com> Reported-by: NAndreas Koch <andreas@akdesigninc.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
This field may use up to 32 bits, so it should be handled as unsigned int. Signed-off-by: NDaniel Mack <zonque@gmail.com> Reported-by: NAndreas Koch <andreas@akdesigninc.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Mark Hills 提交于
The maxpacksize field is given in some quirks, but it gets ignored (in favour of wMaxPacketSize from the first endpoint.) This patch favours the one in the quirk. Digidesign Mbox and Mbox 2 are the only affected quirks and the devices are assumed to be working without this patch. So for safety against the values in the quirk being incorrect, remove them. The datainterval is also ignored but there are not currently any quirks which choose to override this. Cc: Damien Zammit <damien@zamaudio.com> Cc: Chris J Arges <christopherarges@gmail.com> Signed-off-by: NMark Hills <mark@xwax.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Mark Hills 提交于
The hardware also has a PCM capture device which is not implemented in this patch. It may be possible to generalise this to Saffire 6 USB support and some of the other Focusrite interfaces, but as I don't have access to these devices we should wait until capture support is working first. Capture support is not implemented because the code assumes the endpoint to have its own interface (instead, it shares the interface with playback) and some thought will be needed to lift this limitation. Signed-off-by: NMark Hills <mark@xwax.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 17 3月, 2013 1 次提交
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由 Masanari Iida 提交于
Correct spelling typos in Documentation/sound/alsa Signed-off-by: NMasanari Iida <standby24x7@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 15 3月, 2013 5 次提交
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由 Takashi Iwai 提交于
During the transition to the generic parser, the hook to the codec specific automute function was forgotten. This resulted in the silent output on some MacBooks. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Dan Carpenter 提交于
"chn" here is a number between 0 and 255, but ->chn_info[] only has 16 elements so there is a potential write beyond the end of the array. If the seq_mode isn't SEQ_2 then we let the individual drivers (either opl3.c or midi_synth.c) handle it. Those functions all do a bounds check on "chn" so I haven't changed anything here. The opl3.c driver has up to 18 channels and not 16. Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Dylan Reid 提交于
spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset and check it in ca0132_download_dsp(). Signed-off-by: NDylan Reid <dgreid@chromium.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Dylan Reid 提交于
Instead of using the dspload_is_loaded() function, check the dsp_state that is kept in the spec. The dspload_is_loaded() function returns true if the DSP transfer was never started. This false-positive leads to multiple second delays when ca0132_setup_efaults() times out on each write. Signed-off-by: NDylan Reid <dgreid@chromium.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Dylan Reid 提交于
If dspload_image() fails, it was ignored and dspload_wait_loaded() was still called. dsp_loaded should never be set to true in this case, skip it. The check in dspload_wait_loaded() return true if the DSP is loaded or if it never started. Signed-off-by: NDylan Reid <dgreid@chromium.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 3月, 2013 4 次提交
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由 David Henningsson 提交于
If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
So far, the driver doesn't power down the widget at going down to D3 when the widget node has an EAPD capability and EAPD is actually set on all codecs unless codec->power_filter is set explicitly. This caused a problem on some Conexant codecs, leading to click noises, and we set it as NULL there. But it is very unlikely that the problem hits only these codecs. Looking back at the development history, this workaround for EAPD was introduced just for some laptops with STAC9200 codec, then we applied it blindly for all. Now, since it's revealed to have an ill effect, we should disable this workaround per default and apply only for the known requiring systems. The EAPD workaround is implemented now as snd_hda_codec_eapd_power_filter(), and this has to be set explicitly by the codec driver when needed. As of now, only patch_stac9200() sets this one. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Use the dynamic array allocations for pins, converters and PCM arrays instead of the fixed size arrays. The modern HDMI codecs get more and more pins, and we don't know the sensitive limit. Most of the patch are spent for the straight conversions from the fixed array access to snd_array helpers. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 3月, 2013 2 次提交
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Make sure that the allocated buffer for reading the proc file won't expose the uncleared kernel memory. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 3月, 2013 4 次提交
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由 Takashi Iwai 提交于
In the connection list expansion in hda_codec.c and hda_proc.c, the value returned from snd_hda_get_num_raw_conns() is used as the array size to store the connection list. However, the function returns simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the widget list with ranges isn't considered there. Thus it may return a smaller size than the actual list, which results in -ENOSPC in snd_hda_get_raw_conections(). This patch fixes the bug by parsing the connection list correctly also for snd_hda_get_num_raw_conns(). Reported-and-tested-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
The NuForce UDH-100 numbers its interfaces incorrectly, which makes the interface associations come out wrong, which results in the driver erroring out with the message "Audio class v2 interfaces need an interface association". Work around this by searching for the interface association descriptor also in some other place where it might have ended up. Reported-and-tested-by: NDave Helstroom <helstroom@google.com> Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Wei Yongjun 提交于
The dereference should be moved below the NULL test. Signed-off-by: NWei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Yacine Belkadi 提交于
script/kernel-doc reports the following type of warnings (when run in verbose mode): Warning(sound/core/init.c:152): No description found for return value of 'snd_card_create' To fix that: - add missing descriptions of function return values - use "Return:" sections to describe those return values Along the way: - complete some descriptions - fix some typos Signed-off-by: NYacine Belkadi <yacine.belkadi.1@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 11 3月, 2013 10 次提交
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由 Adrian Knoth 提交于
Expose the newly added TCO LTC and sync check functions to userspace. 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
This patch adds new ALSA controls to query the LTC state from userspace. 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
This patch prepares snd_hdspm_get_sync_check() to also check the TCO sync state. The added feature will be exposed to the user in a later commit. 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
Considerably shorten the code by using a macro. Though this won't lower the binary size, it makes the source more readable. 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
For 96kHz, MADI allows to multiplex the samples (SMUX) or to use a dedicated 96K mode. The RME cards default to 96K mode, but since not all external MADI equipment supports this, provide a switch to users that changes the on-wire protocol to SMUX. 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
When using the additional Time Code Option module in slave mode or the SYNC-In wordclock connector, the sample rate needs to be returned by hdspm_external_sample_rate(). Since this sample rate may contain any value with 1Hz granularity, we need to round it to a common rate as done by the OSX driver. [Fixed missing function declarations by tiwai] 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
This commit introduces hdspm_get_pll_freq() to avoid code duplication. Reading the sample rate from the DDS register will be required by upcoming code. Signed-off-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
When a headphone pin is set up as a shared hp/mic pin, we rather want to keep it as a headphone primarily as default, but the driver overrides it always as a mic pin, just because the input controls are created after outputs. Add a check of pin NID and skip the re-initialization of pinctl for such a shared hp/mic pin. Reported-by: NJonathan Woithe <jwoithe@just42.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
snd_seq_timer_open() didn't catch the whole error path but let through if the timer id is a slave. This may lead to Oops by accessing the uninitialized pointer. BUG: unable to handle kernel NULL pointer dereference at 00000000000002ae IP: [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 PGD 785cd067 PUD 76964067 PMD 0 Oops: 0002 [#4] SMP CPU 0 Pid: 4288, comm: trinity-child7 Tainted: G D W 3.9.0-rc1+ #100 Bochs Bochs RIP: 0010:[<ffffffff819b3477>] [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 RSP: 0018:ffff88006ece7d38 EFLAGS: 00010246 RAX: 0000000000000286 RBX: ffff88007851b400 RCX: 0000000000000000 RDX: 000000000000ffff RSI: ffff88006ece7d58 RDI: ffff88006ece7d38 RBP: ffff88006ece7d98 R08: 000000000000000a R09: 000000000000fffe R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: ffff8800792c5400 R14: 0000000000e8f000 R15: 0000000000000007 FS: 00007f7aaa650700(0000) GS:ffff88007f800000(0000) GS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000000002ae CR3: 000000006efec000 CR4: 00000000000006f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process trinity-child7 (pid: 4288, threadinfo ffff88006ece6000, task ffff880076a8a290) Stack: 0000000000000286 ffffffff828f2be0 ffff88006ece7d58 ffffffff810f354d 65636e6575716573 2065756575712072 ffff8800792c0030 0000000000000000 ffff88006ece7d98 ffff8800792c5400 ffff88007851b400 ffff8800792c5520 Call Trace: [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [<ffffffff819b17e9>] snd_seq_queue_timer_open+0x29/0x70 [<ffffffff819ae01a>] snd_seq_ioctl_set_queue_timer+0xda/0x120 [<ffffffff819acb9b>] snd_seq_do_ioctl+0x9b/0xd0 [<ffffffff819acbe0>] snd_seq_ioctl+0x10/0x20 [<ffffffff811b9542>] do_vfs_ioctl+0x522/0x570 [<ffffffff8130a4b3>] ? file_has_perm+0x83/0xa0 [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [<ffffffff811b95ed>] sys_ioctl+0x5d/0xa0 [<ffffffff813663fe>] ? trace_hardirqs_on_thunk+0x3a/0x3f [<ffffffff81faed69>] system_call_fastpath+0x16/0x1b Reported-and-tested-by: NTommi Rantala <tt.rantala@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Christine Spang 提交于
Having snd_BUG_ON() only evaluate its conditional when CONFIG_SND_DEBUG is set leads to frequent bugs, since other similar macros in the kernel have different behavior. Let's make snd_BUG_ON() act like those macros so it will stop being accidentally misused. Signed-off-by: NChristine Spang <christine.spang@oracle.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 08 3月, 2013 5 次提交
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由 Takashi Iwai 提交于
When the headphone mic jack enum control is created (via explicitly specification by user), it doesn't make much sense to change the I/O direction dynamically per capture source change, since the I/O direction is rather controlled over the enum ctl. This also reduces the implicit dependency between the capture source and the hp mic jack enum ctls, which might confuse a program accessing the whole control elements at once like alsactl. In addition, this patch introduces update_hp_automute_hook() function to call the proper hook function. It's just to remove the open codes in multiple places in hda_generic.c. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
There is no big merit to distinguish these two hints. Instead, just have a single flag, add_jack_modes, for creating the jack mode enum ctls for both I/O directions. The hint string parser code is left and translated as add_jack_modes just for keeping compatibility. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The previous commits added the capability to change the pin control of hp/mic shared jack, but it actually didn't work as expected when the value is changed from the output to the input, since I forgot to reset the pin I/O bit in that case. This patch fixes the problem. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
In order to let user test the known workaround more easily, give a few known fixups for ALC260 to the model strings so that it can be passed via the module option. Also, move the unusual setups found in FSC S7020 fixup into a special model, fujitsu-jwse, Jonathan Woithe Special Edition. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
When a headphone jack is configured as a shared hp/mic jack, the jack mode enum needs to handle both input and output directions. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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