- 21 Jan, 2019 6 commits
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Takashi Sakamoto authored
This commit adds a member for a callback function to switch frame fetching mode to former protocol. Signed-off-by:
Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by:
Takashi Iwai <tiwai@suse.de>
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Takashi Sakamoto authored
This commit adds a member for a callback function to dump status and move existing code to former protocol. Signed-off-by:
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Takashi Sakamoto authored
In a series of Fireface, latter protocol has no way for drivers to retrieve current clock configuration. On the other hand, this driver has proc node for it. This commit removes a proc node to dump both clock configuration and synchronization status in one proc node. Signed-off-by:
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Takashi Sakamoto authored
This commit moves codes for Fireface 400 to a file of former protocol. Signed-off-by:
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Takashi Sakamoto authored
In a series of Fireface, later model supports different protocol from former models. This commit is a preparation to support both of protocols. Signed-off-by:
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Takashi Iwai authored
The unused variable was forgotten to be removed and now we get a compiler warning: sound/pci/hda/hda_codec.c: In function 'hda_codec_runtime_suspend': sound/pci/hda/hda_codec.c:2926:18: warning: unused variable 'pcm' Fixes: 17bc4815 ("ALSA: pci: Remove superfluous snd_pcm_suspend*() calls") Reported-by:
Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by:
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- 18 Jan, 2019 1 commit
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Takashi Iwai authored
Pull the PCM suspend improvement / cleanup. This moves the most of snd_pcm_suspend*() calls into PCM's own device PM ops. There should be no change from the functionality POV. Signed-off-by:
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- 15 Jan, 2019 14 commits
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Takashi Iwai authored
snd_pcm_suspend() is no longer called from outside, so let's make it local static. Also drop a superfluous NULL check there. Reviewed-by:
Jaroslav Kysela <perex@perex.cz> Signed-off-by:
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Takashi Iwai authored
The PCM suspend procedure was changed for drivers, so that they don't have to call snd_pcm_suspend*() in each callback any longer. Update the documentation to adapt the changes. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
The call of snd_pcm_suspend_all() & co became superfluous since we call it in the PCM PM ops. Let's remove them. Reviewed-by:
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Takashi Iwai authored
ATIIXP driver supports the full PCM resume and saves/restores the running PCM pointer. This used to be done in the suspend and resume callbacks together with snd_pcm_suspend() call. But since we moved the snd_pcm_supsend*() call in PCM device PM ops, this should be moved to a more appropriate place, i.e. the trigger callback. Along with the movement of the PCM suspend/resume code, remove the superfluous snd_pcm_suspend_all() call, too. Reviewed-by:
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Takashi Iwai authored
Until now we rely on each driver calling snd_pcm_suspend*() explicitly at its own PM handling. However, this can be done far more easily by setting the PM ops to each actual snd_pcm device object. This patch adds the device_type object for PCM stream and assigns to each PCM stream object. The type contains only the PM ops for system suspend; we don't need to deal with the resume in general. The suspend hook simply calls snd_pcm_suspend_all() for the given PCM streams. This implies that the PM order is correctly put, i.e. PCM is suspended before the main (or codec) driver, which should be true in general. If a special ordering is needed, you'd need to adjust the device PM order manually later. This patch introduces a new flag, snd_pcm.no_device_suspend, too. With this flag set, the PCM device object won't invoke snd_pcm_suspend_all() by itself. This is needed for ASoC who wants to manage the PM call orders in its serialized way, and the flag is set in soc_new_pcm() as default. For the non-ASoC world, we can get rid of the manual snd_pcm_suspend calls. This will be done in the later patches. Reviewed-by:
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- 14 Jan, 2019 5 commits
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Sameer Pujar authored
Program codec stripe through AC_VERB_SET_STRIPE_CONTROL to use multiple sdo lines if supported. Audio needs to be striped across number of sdo lines for simultaneous playbacks of higher resolutions to work. This needs to be implemented only for an Audio Output Converter and only if the stripe bit(AC_WCAP_STRIPE) of Audio Widget Capabilities parameter is 1. Signed-off-by:
Sameer Pujar <spujar@nvidia.com> Reviewed-by:
Mohan Kumar D <mkumard@nvidia.com> Reviewed-by:
Ravindra Lokhande <rlokhande@nvidia.com> Reviewed-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by:
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Sameer Pujar authored
Platforms having multiple SORs and hdmi/dp sinks require higher bandwidth to support simultaneous playbacks of higher resolution. If hda controller supports multiple SDO lines, STRIPE can be used to indicate how many of the SDO lines the stream should be striped across. During stream start stripe control bits are programmed to use given number of sdo lines and the same is cleared during stream stop. Signed-off-by:
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Sameer Pujar authored
bits 16:17 in SD_CTL register refer to stripe control. Added an offset register(AZX_REG_SD_CTL_3B) to have exclusive read/write of corresponding register byte. This helps to avoid unnecessary 32-bit read/write of SD_CTL whenever only stripe or other bits of corresponding byte need to be updated. Also HD audio spec defines SD_CTL as 3 byte register. SD_CTL_STRIPE_MASK(0x3) can be used for stripe control programming and when updating AZX_REG_SD_CTL_3B. Signed-off-by:
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Sameer Pujar authored
Controllers and codecs can support striping of audio out across multiple SDO lines. The number of supported SDO lines can be specific to chip. GCAP register can be read to know the maximum supported SDO lines. snd_hdac_get_stream_stripe_ctl() is exposed to program stripe bits on controller and codec side. stripe value: 0 for 1SDO, 1 for 2SDO, 2 for 4SDO lines, etc., Signed-off-by:
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Sameer Pujar authored
Controllers can support multiple Serial Data Out(SDO) lines, for extended outbound bandwidth, to pump data to all codecs on the link. Codecs can sample data present on SDO. Add verbs AC_VERB_GET_STRIPE_CONTROL and AC_VERB_SET_STRIPE_CONTROL These can be used to program usage of SDO lines for codec. Signed-off-by:
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- 09 Jan, 2019 3 commits
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YueHaibing authored
Fixes gcc '-Wunused-but-set-variable' warning: sound/usb/mixer.c: In function 'parse_audio_feature_unit': sound/usb/mixer.c:1838:28: warning: variable 'first_ch_bits' set but not used [-Wunused-but-set-variable] It never used since 2.6 Signed-off-by:
YueHaibing <yuehaibing@huawei.com> Signed-off-by:
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Keyon Jie authored
To enable SIE(Stream Interrupt Enable) in snd_hdac_stream_start(), we should set both mask and value to be "1 << azx_dev->index" for register update, the mask was 0, here fix it. Signed-off-by:
Keyon Jie <yang.jie@linux.intel.com> Signed-off-by:
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Keyon Jie authored
E.g. for azx_int_enable(), we should set both mask and value to be "AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN"(the mask was 0) to enable controller CIE and GIE. We have similar issues on setting AZX_GCTL_RESET and AZX_GCTL_UNSOL, here try to correct all of them. Signed-off-by:
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- 07 Jan, 2019 11 commits
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Aditya Pakki authored
snd_ctl_add() could fail, so let's check its return value and return its error code upstream upon failure. Signed-off-by:
Aditya Pakki <pakki001@umn.edu> Signed-off-by:
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Aditya Pakki authored
The fix checks if snd_card_register() fails, and if so logs the error via dev_err() consistent with other patches. Signed-off-by:
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Aditya Pakki authored
snd_i2c_sendbytes could fail. The fix checks its return value: if it fails, issues an error message and returns with its error code. Signed-off-by:
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Kangjie Lu authored
snd_ctl_add() could fail, so let's check its status and issue an error message if it indeed fails. Signed-off-by:
Kangjie Lu <kjlu@umn.edu> Signed-off-by:
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Tom Yan authored
There's no reason for us to do that while we initialize dac_mute to 1. Also oxygen_init() has been clearing the OXYGEN_SPDIF_OUT_ENABLE bit anyway. Signed-off-by:
Tom Yan <tom.ty89@gmail.com> Signed-off-by:
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Tom Yan authored
Add control for the de-emphasis filter in the PCM179x DACs Signed-off-by:
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Kailang Yang authored
Dell has new platform for ALC274. This will support to enable headset mode. Signed-off-by:
Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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Hui Peng authored
In `create_composite_quirk`, the terminating condition of for loops is `quirk->ifnum < 0`. So any composite quirks should end with `struct snd_usb_audio_quirk` object with ifnum < 0. for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) { ..... } the data field of Bower's & Wilkins PX headphones usb device device quirks do not end with {.ifnum = -1}, wihch may result in out-of-bound read. This Patch fix the bug by adding an ending quirk object. Fixes: 240a8af9 ("ALSA: usb-audio: Add a quirck for B&W PX headphones") Signed-off-by:Hui Peng <benquike@163.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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Takashi Iwai authored
There are a few places where we access the data without checking the actual object size from the USB audio descriptor. This may result in OOB access, as recently reported. This patch addresses these missing checks. Most of added codes are simple bLength checks in the caller side. For the input and output terminal parsers, we put the length check in the parser functions. For the input terminal, a new argument is added to distinguish between UAC1 and the rest, as they treat different objects. Reported-by:
Mathias Payer <mathias.payer@nebelwelt.net> Reported-by:
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Takashi Iwai authored
We've had some sanity checks of the mixer unit descriptors but they are too loose and some corner cases are overlooked. Add more strict checks in uac_mixer_unit_get_channels() for avoiding possible OOB accesses by malformed descriptors. This also changes the semantics of uac_mixer_unit_get_channels() slightly. Now it returns zero for the cases where the descriptor lacks of bmControls instead of -EINVAL. Then the caller side skips the mixer creation for such unit while it keeps parsing it. This corresponds to the case like Maya44. Cc: <stable@vger.kernel.org> Signed-off-by: -
Takashi Iwai authored
The parser for the processing unit reads bNrInPins field before the bLength sanity check, which may lead to an out-of-bound access when a malformed descriptor is given. Fix it by assignment after the bLength check. Cc: <stable@vger.kernel.org> Signed-off-by:
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