- 07 1月, 2014 2 次提交
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由 Lars-Peter Clausen 提交于
Allow PCMs that do not impose any restrictions on the supported formats to set the formats field to 0, Instead of assuming that this means that the PCM does not support any formats (which doesn't make much sense), assume that it supports all formats. This brings the behavior of DPCM closer to that of non-DPCM. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@linaro.org>
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由 Lars-Peter Clausen 提交于
We have the same code for initializing the runtime pcm on both the playback and the capture path. Factor this out into a common helper function. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 01 1月, 2014 1 次提交
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由 Xiubo Li 提交于
From "ASoC: make snd_soc_dai_link more symmetrical", can we see that the name of CPU DAI maybe omitted. If the DAI name is omitted, try to use the component name instead. Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 30 12月, 2013 1 次提交
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由 Lars-Peter Clausen 提交于
The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 04 12月, 2013 1 次提交
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由 Nicolin Chen 提交于
This patch removed the redundant snd_soc_dai_digital_mute() in close() since it's better to mute in hw_free() which's slightly earlier and symmetrical for the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()-> parepare(unmute)->playing->hw_free(mute)->close() Signed-off-by: NNicolin Chen <b42378@freescale.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 02 12月, 2013 1 次提交
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由 Lars-Peter Clausen 提交于
If there are symmetry constraints between the playback and the capture channel set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 28 11月, 2013 1 次提交
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由 Fabio Estevam 提交于
Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 24 11月, 2013 2 次提交
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由 Nicolin Chen 提交于
We're now applying soc_hw_params_symmetry() to reject unmatched parameters while we clear parameters in soc_pcm_close(). So here's a use case might be broken by this mechanism: aplay -Dhw:0 44100.wav 48000.wav 32000.wav In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free() ->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. As we only clear parameters in soc_pcm_close(). The parameters would be remained in the system even if the playback of 44100.wav is finished. Thus, this patch is trying to move parameters cleaning into hw_free() so that the system can continue to serve this kind of use case. Also, since we set them in hw_params(), it should be better to clear them in hw_free() for symmetry. Signed-off-by: NNicolin Chen <b42378@freescale.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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由 Nicolin Chen 提交于
Some SoCs can only work in mono or stereo mode at one time. So if we let them capture a mono stream while playing a stereo stream, there might be a problem occur to one of these two streams: double paced or slowed down. In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate symmetry. But we don't have one for channels. Likewise, we can treat symmetric_rate as a solution for those SoCs or CODECs which can not handle asymmetrical LRCLK. But it's also impossible for them to handle asymmetrical BCLK. And accodring to BCLK = LRCLK * channel number * slot size(fixed or sample bits), sample bits might also be a problem if they are not using a fixed slot size. Thus, this patch applys symmetry for channels and sample bits. Meanwhile, there might be a race between two substreams if starting simultaneously. Previously, we only added warning to compalin but still using conservative way to let it carry on. However, this patch rejects the second stream with any unmatched parameter to make sure the first existing stream won't be broken. Signed-off-by: NNicolin Chen <b42378@freescale.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 22 11月, 2013 2 次提交
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由 David Henningsson 提交于
current_headset_type should be of the HEADSET_TYPE enum, not the HEADSET_MODE enum. Since ALC_HEADSET_TYPE_UNKNOWN and ALC_HEADSET_MODE_UNKNOWN are both 0, this patch is just janitorial. Signed-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
Some models (or maybe depending on BIOS version) of Sony VAIO with ALC260 give no proper pin configurations as default, resulting in the non-working speaker, etc. Just provide the whole pin configurations via a fixup. Reported-by: NMatthew Markus <mmarkus@hearit.co> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 21 11月, 2013 3 次提交
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由 David Henningsson 提交于
Cc: stable@vger.kernel.org (3.10+) BugLink: https://bugs.launchpad.net/bugs/1253636Signed-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
BIOS sets MISC_NO_PRESENCE bit wrongly to the pin config on NID 0x0f. Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The laptop has a built-in speaker on NID 0x1a. It's an LFE only on the right channel, so we need to provide an explicit chmap, too. There might be other surround speakers, but they can fixed in addition at later point, so let's fix the easier bass speaker at first. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65091Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 11月, 2013 7 次提交
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由 Takashi Iwai 提交于
When a codec is resumed, it keeps the power on while the resuming phase via hda_keep_power_on(), then turns down via snd_hda_power_down(). At that point, snd_hda_power_down() notifies the power down to the controller, and this may confuse the refcount if the codec was already powered up before the resume. In the end result, the controller goes to runtime suspend even before the codec is kicked off to the power save, and the communication stalls happens. The fix is to add the power-up notification together with hda_keep_power_on(), and clears the flag appropriately. Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Richard Fitzgerald 提交于
The FLL must be placed into free-run mode before disabling to allow it to entirely shut down. Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
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由 David Henningsson 提交于
You're looking at a casual headset patch, for a specific hardware it will match, and suddenly, the headset jack will work, so please apply this simple quirk! BugLink: https://bugs.launchpad.net/bugs/1253038Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Kuninori Morimoto 提交于
DMAEngine will stall without this patch Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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由 Dan Carpenter 提交于
If "id == ARRAY_SIZE(routes)" then we read one space beyond the end of the routes[] array. Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com> Acked-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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由 Charles Keepax 提交于
Certain registers require patching after the SYSCLK has been brought up add support for this into the CODEC driver. Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
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由 Lee Jones 提交于
It's been reported that these break audio on Snowball so revert them until a Snowball user has time to investigate. Signed-off-by: NLee Jones <lee.jones@linaro.org> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 19 11月, 2013 3 次提交
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由 David Henningsson 提交于
The docking station is a Thinkpad thing, so it makes sense to check for mute/micmute LEDs for that quirk type too. Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Sakamoto 提交于
'snd_pcm_format_t' is used by amdtp_out_stream_set_pcm_format(). Currently, when just including amdtp.h, compiler cannot find this type because this type is defined in uapi/sound/asound.h and this header is not included by amdtp.h. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Use select FW_LOADER instead of depends on. It's more intuitive and more consistent with other entries. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 11月, 2013 2 次提交
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由 David Henningsson 提交于
We're using the ACPI interface to detect whether we're dealing with a Thinkpad or not. This way we're not loading the thinkpad_acpi module when we're not on a Thinkpad, but at the same time, we give the opportunity to check for, and potentially enable, both present and future Thinkpad with mute/micmute LEDs. At least those running the ALC269 family (269 to 299) of Realtek codecs. Cc: Alex Hung <alex.hung@canonical.com> Tested-by: NHui Wang <hui.wang@canonical.com> Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Kuninori Morimoto 提交于
rsnd_dai_connect() is using mod before NULL checking. This patch fixes it up Reported-by: NDan Carpenter <dan.carpenter@oracle.com> Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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- 15 11月, 2013 6 次提交
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由 Vitaliy Kulikov 提交于
Adds linear EQ filtering for integrated speaker protection Signed-off-by: NVitaliy Kulikov <vitaliy.kulikov@idt.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Brian Austin 提交于
The mask for CS42L52_MIC_CTL_TYPE_MASK was wrong keeping the mic config from being set correctly. Signed-off-by: NBrian Austin <brian.austin@cirrus.com> Signed-off-by: NMark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
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由 Nicolin Chen 提交于
Since gen_pool_dma_alloc() is introduced, we implement it to simplify code. Signed-off-by: NNicolin Chen <b42378@freescale.com> Acked-by: NTakashi Iwai <tiwai@suse.de> Signed-off-by: NAndrew Morton <akpm@linux-foundation.org> Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>
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由 Wolfram Sang 提交于
Use this new function to make code more comprehensible, since we are reinitialzing the completion, not initializing. [akpm@linux-foundation.org: linux-next resyncs] Signed-off-by: NWolfram Sang <wsa@the-dreams.de> Acked-by: Linus Walleij <linus.walleij@linaro.org> (personally at LCE13) Cc: Ingo Molnar <mingo@kernel.org> Signed-off-by: NAndrew Morton <akpm@linux-foundation.org> Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>
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由 Nicolin Chen 提交于
It's safer to turn on regcache_cache_only before disabling regulator since the driver will turn off the regcache_cache_only after enabling regulator. If we remain cache_only false, some command like 'amixer cset' would get failure if being run before wm8962_resume(). Signed-off-by: NNicolin Chen <b42378@freescale.com> Signed-off-by: NMark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
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由 Takashi Iwai 提交于
The recent change in sysfs triggered a kernel WARNING at unloading a sound driver like WARNING: CPU: 3 PID: 2247 at fs/sysfs/group.c:214 sysfs_remove_group+0xe8/0xf0() sysfs group ffffffff81ab7b20 not found for kobject 'event14' for each jack instance. It's because the unregistration of jack input device is done in dev_free callback, which is called after snd_card_disconnect(). Since device_unregister(card->card_dev) is called in snd_card_disconnect(), the whole sysfs entries belonging to card->card_dev have been already removed recursively. Thus this results in a warning as input_unregister_device() yet tries to unregister the already removed sysfs entry. For fixing this mess, we need to unregister the jack input device at dev_disconnect callback so that it's called before unregistering the card->card_dev. Reviwed-by: NMark Brown <broonie@linaro.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 11月, 2013 5 次提交
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由 Takashi Iwai 提交于
The current code may access to the already freed object. The input device must be accessed and unregistered before freeing the top level sound object. Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Oskar Schirmer 提交于
Unbalanced calls to snd_imx_pcm_trigger() may result in endless FIQ activity and thus provoke eternal sound. While on the first glance, the switch statement looks pretty symmetric, the SUSPEND/RESUME pair is not: the suspend case comes along snd_pcm_suspend_all(), which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(), but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME. This way userland may provoke an unbalanced resume, which might cause the fiq_enable counter to increase and never return to zero again, so eventually imx_pcm_fiq is never disabled. Simply removing the fiq_enable will solve the problem, as long as one never goes play and capture game simultaneously, but beware trying both at once, the early TRIGGER_STOP will cut off the other activity prematurely. So now playing and capturing is scrutinized separately, instead of by counting. Signed-off-by: NOskar Schirmer <oskar@scara.com> Signed-off-by: NMark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
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由 Takashi Iwai 提交于
Fixes: 4b2ffc20 ('ASoC: Blackfin I2S: add 8-bit sample support') Reported-by: David Binderman Signed-off-by: NTakashi Iwai <tiwai@suse.de> Signed-off-by: NMark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
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由 Vasily Khoruzhick 提交于
Add support for front jack channel selector which is present on EMU0204. It allows to get 4 channels out of this soundcard. Tested-by: NYury Bushmelev <jay@jay-tech.ru> Signed-off-by: NVasily Khoruzhick <anarsoul@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
snd_hda_codec_reset() is called either in resetting the whole setup at error paths or hwdep clear/reconfig sysfs triggers. But all of these don't assume that the power has to be off, rather they want to keep the power state unchanged (e.g. reconfig_codec() calls the power up/down by itself). Thus, unconditionally clearing the power state in snd_hda_codec_reset() leads to the inconsistency, confuses the further operation. This patch gets rid of the lines doing that bad thing. Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 11月, 2013 3 次提交
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由 Richard Fitzgerald 提交于
Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@linaro.org>
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由 Takashi Iwai 提交于
Similarly as other laptops with AD1981 & co codecs, we can control EAPD on AD1986A more safely depending on the Master switch, in order to save some power. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The only EAPD on AD1986A is on NID 0x1b where usually the speaker. But this doesn't control only the speaker amp but may influence on all outputs, e.g. Lenovo N100 laptop seems to have this issue. Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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