- 20 Sep, 2019 1 commit
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Takashi Iwai authored
It's reported that the garbled sound on HP Envy x360 13z-ag000 (Ryzen Laptop) is fixed by the same workaround applied to other AMD chips. Update the driver_data entry for Raven (1022:15e3) to use the newly introduced preset, AZX_DCAPS_PRESET_AMD_SB. Since it already contains AZX_DCAPS_PM_RUNTIME, we can drop that bit, too. Reported-and-tested-by:
Dennis Padiernos <depadiernos@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20190920073040.31764-1-tiwai@suse.deSigned-off-by:
Takashi Iwai <tiwai@suse.de>
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- 17 Sep, 2019 1 commit
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Ilya Pshonkin authored
This patch adds quirk VID ID for Hiby portable players family with native DSD playback support. Signed-off-by:
Ilya Pshonkin <sudokamikaze@protonmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20190917074937.157802-1-ilya.pshonkin@netforce.uaSigned-off-by:
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- 16 Sep, 2019 3 commits
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James McDonnell authored
Headset microphone quirk for alienware 15r3. Without this using a headset with mic attached will not work. Signed-off-by:
James McDonnell <james_mcdonnell@hotmail.com> Link: https://lore.kernel.org/r/QB1PR01MB2337D0367C2E3ADB0010134F808C0@QB1PR01MB2337.CANPRD01.PROD.OUTLOOK.COMSigned-off-by:
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Takashi Sakamoto authored
At higher sampling rate (e.g. 192.0 kHz), Alesis iO26 transfers 4 data channels per data block in CIP. Both iO14 and iO26 have the same contents in their configuration ROM. For this reason, ALSA Dice driver attempts to distinguish them according to the value of TX0_AUDIO register at probe callback. Although the way is valid at lower and middle sampling rate, it's lastly invalid at higher sampling rate because because the two models returns the same value for read transaction to the register. In the most cases, users just plug-in the device and ALSA dice driver detects it. In the case, the device runs at lower sampling rate and the driver detects expectedly. For this reason, this commit leaves the way to detect as is. Fixes: 28b208f6 ("ALSA: dice: add parameters of stream formats for models produced by Alesis") Cc: <stable@vger.kernel.org> # v4.18+ Signed-off-by:
Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20190916101851.30409-1-o-takashi@sakamocchi.jpSigned-off-by:
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Takashi Iwai authored
Merge tag 'asoc-v5.4-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Final merge window fixes for v5.4 A few small fixes and one feature that came in since I sent you the earlier pull request.
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- 15 Sep, 2019 3 commits
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Mark Brown authored
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Mark Brown authored
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Saiyam Doshi authored
Remove excess semicolon after closing parenthesis. Signed-off-by:
Saiyam Doshi <saiyamdoshi.in@gmail.com> Link: https://lore.kernel.org/r/20190914031133.GA28447@SDSigned-off-by:
Mark Brown <broonie@kernel.org>
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- 14 Sep, 2019 1 commit
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Takashi Iwai authored
Signed-off-by:
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- 13 Sep, 2019 1 commit
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Viorel Suman authored
This is to allow machine drivers to set a certain bitclk rate which might not be exactly rate * frame size. Cc: NXP Linux Team <linux-imx@nxp.com> Signed-off-by:
Viorel Suman <viorel.suman@nxp.com> Signed-off-by:
Daniel Baluta <daniel.baluta@nxp.com> Acked-by:
Nicolin Chen <nicoleotsuka@gmail.com> Link: https://lore.kernel.org/r/20190830215910.31590-1-daniel.baluta@nxp.comSigned-off-by:
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- 11 Sep, 2019 2 commits
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Peter Ujfalusi authored
While it is safe to use strncpy in this case, the advice is to move to strscpy or strscpy_pad. Suggested-by:
Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20190911083331.16801-1-peter.ujfalusi@ti.comSigned-off-by:
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Saiyam Doshi authored
All these functions declares and initializes variable ret with '0' and without modifying 'ret' variable, it is returned. This patch removes this redundancy and returns '0' directly. Signed-off-by:
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- 10 Sep, 2019 3 commits
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Takashi Sakamoto authored
2 bytes in MSB of register for clock status is zero during intermediate state after changing status of sampling clock in models of TASCAM FireWire series. The duration of this state differs depending on cases. During the state, it's better to retry reading the register for current status of the clock. In current implementation, the intermediate state is checked only when getting current sampling transmission frequency, then retry reading. This care is required for the other operations to read the register. This commit moves the codes of check and retry into helper function commonly used for operations to read the register. Fixes: e453df44 ("ALSA: firewire-tascam: add PCM functionality") Cc: <stable@vger.kernel.org> # v4.4+ Signed-off-by:
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Takashi Sakamoto authored
The return value of snd_tscm_stream_get_clock() is ignored. This commit checks the value and handle error. Fixes: e453df44 ("ALSA: firewire-tascam: add PCM functionality") Cc: <stable@vger.kernel.org> # v4.4+ Signed-off-by:
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/soundTakashi Iwai authored
ASoC: Updates for v5.4 Quite a big update this time around, particularly in the core where we've had a lot of cleanups from Morimoto-san - there's not much functional change but quite a bit of modernization going on. We've also seen a lot of driver work, a lot of it cleanups but also some particular drivers. - Lots and lots of cleanups from Morimoto-san and Yue Haibing. - Lots of cleanups and enhancements to the Freescale, sunxi dnd Intel rivers. - Initial Sound Open Firmware suppot for i.MX8. - Removal of w90x900 and nuc900 drivers as the platforms are being removed. - New support for Cirrus Logic CS47L15 and CS47L92, Freescale i.MX 7ULP and 8MQ, Meson G12A and NXP UDA1334
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- 09 Sep, 2019 25 commits
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Mark Brown authored
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Mark Brown authored
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Cheng-Yi Chiang authored
Add an op in hdmi_codec_ops so codec driver can register callback function to handle plug event. Driver in DRM can use this callback function to report connector status. Signed-off-by:
Cheng-Yi Chiang <cychiang@chromium.org> Link: https://lore.kernel.org/r/20190717083327.47646-2-cychiang@chromium.orgReviewed-by:
Tzung-Bi Shih <tzungbi@google.com> Signed-off-by:
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Ben Zhang authored
Instead of clearing RT5677_PWR_ANLG2 (MX-64h) to 0 at SND_SOC_BIAS_OFF, we only clear the RT5677_PWR_CORE bit which is set at SND_SOC_BIAS_PREPARE. MICBIAS control bits are left unchanged. This fixed the bug where if MICBIAS1 widget is forced on, MICBIAS control bits will be cleared at suspend and never turned back on again, since DAPM thinks the widget is always on. Signed-off-by:
Ben Zhang <benzh@chromium.org> Signed-off-by:
Curtis Malainey <cujomalainey@chromium.org> Link: https://lore.kernel.org/r/20190906194636.217881-3-cujomalainey@chromium.orgSigned-off-by:
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Curtis Malainey authored
In order to simplify understanding what register values are being written to the codec for debugging more advanced features (such as hotwording) it is best to remove magic numbers Signed-off-by:
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Kuninori Morimoto authored
Current soc_unbind_aux_dev() implementation is very half, thus it is very unreadable. for_each_comp_order(order) { for_each_card_auxs_safe(card, comp, _comp) { (1) if (comp->driver->remove_order == order) { ... => soc_unbind_aux_dev(comp); } } soc_unbind_aux_dev() itself is not related to remove_order (1). And, it is called from soc_remove_aux_devices(), even though its paired function soc_bind_aux_dev() is called from snd_soc_instantiate_card(). It is very unbalance, and very difficult to understand. This patch do 1) update soc_bind_aux_dev() to self contained 2) call it from soc_cleanup_card_resources() to make up balance Signed-off-by:Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87r24wor0z.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Kuninori Morimoto authored
It is easy to read code if it is cleanly using paired function/naming, like start <-> stop, register <-> unregister, etc, etc. But, current ALSA SoC code is very random, unbalance, not paired, etc. It is easy to create bug at the such code, and it will be difficult to debug. soc-core.c has soc_bind_aux_dev(), but, there is no its paired soc_unbind_aux_dev(). This patch adds soc_unbind_aux_dev(). Signed-off-by:
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Kuninori Morimoto authored
Current soc_bind_aux_dev() implementation is very half, thus it is very unreadable. for_each_card_pre_auxs(xxx) { => ret = soc_bind_aux_dev(xxx); ... } This patch does all for_each_xxx() under soc_bind_aux_dev(), and makes it to self contained. Signed-off-by: -
Kuninori Morimoto authored
It is easy to read code if it is cleanly using paired function/naming, like start <-> stop, register <-> unregister, etc, etc. But, current ALSA SoC code is very random, unbalance, not paired, etc. It is easy to create bug at the such code, and it will be difficult to debug. This patch moves soc_probe_link_dais() next to soc_remove_link_dais() which is paired function. Signed-off-by:
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Kuninori Morimoto authored
Current soc_probe_link_dais() implementation is very half, thus it is very difficult to read. for_each_comp_order(xxx) { for_each_card_rtds(xxx) => soc_probe_link_dais(xxx); } This patch does all for_each_xxx() under soc_probe_link_dais(), and makes it to self contained. Signed-off-by: -
Kuninori Morimoto authored
Current soc_probe_link_dais() (1) is called under probe_order (2), and it will initialize dai_link related settings at *Last* turn (3)(B). It is very complex code. static int soc_probe_link_dais(..., order) { (A) /* probe DAIs here */ ... (3) if (order != SND_SOC_COMP_ORDER_LAST) return 0; (B) /* initialize dai_link related settings */ ... } static int snd_soc_instantiate_card(...) { ... (2) for_each_comp_order(order) { for_each_card_rtds(...) { (1) ret = soc_probe_link_dais(..., order); } } } This patch separes soc_probe_link_dais() into "DAI probe" portion (A), and dai_link settings portion (B). The later is named as soc_link_init() by this patch. Signed-off-by: -
Kuninori Morimoto authored
It is easy to read code if it is cleanly using paired function/naming, like start <-> stop, register <-> unregister, etc, etc. But, current ALSA SoC code is very random, unbalance, not paired, etc. It is easy to create bug at the such code, and it will be difficult to debug. This patch moves soc_probe_dai() next to soc_remove_dai() which is paired function. Signed-off-by:
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Kuninori Morimoto authored
Current soc_remove_link_dais() implementation is very half, thus it is very difficult to read. for_each_comp_order(xxx) { for_each_card_rtds(xxx) => soc_remove_link_dais(xxx); } This patch does all for_each_xxx() under soc_remove_link_dais(), and makes it to self contained. Signed-off-by: -
Kuninori Morimoto authored
Current soc_remove_link_components() implementation is very half, thus it is very difficult to read. for_each_comp_order(xxx) { for_each_card_rtds(xxx) => soc_remove_link_components(xxx); } This patch does all for_each_xxx() under soc_remove_link_components(), and makes it to self contained. Signed-off-by: -
Kuninori Morimoto authored
Current soc_probe_link_components() implementation is very half, thus it is very difficult to read. for_each_comp_order(xxx) { for_each_card_rtds(xxx) { => ret = soc_probe_link_components(xxx); ... } } This patch does all for_each_xxx() under soc_probe_link_components(), and makes it to self contained. Signed-off-by: -
Colin Ian King authored
Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 82 bytes. Before: text data bss dec hex filename 26548 7288 64 33900 846c sound/soc/codecs/rt1308.o After: text data bss dec hex filename 26370 7384 64 33818 841a sound/soc/codecs/rt1308.o (gcc version 9.2.1, amd64) Signed-off-by:
Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20190907074634.22144-1-colin.king@canonical.comSigned-off-by:
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Colin Ian King authored
Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 93 bytes. Before: text data bss dec hex filename 38961 9784 64 48809 bea9 sound/soc/codecs/rt1305.o After: text data bss dec hex filename 38804 9848 64 48716 be4c sound/soc/codecs/rt1305.o (gcc version 9.2.1, amd64) Signed-off-by:
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Colin Ian King authored
Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 100 bytes. Before: text data bss dec hex filename 51463 13016 128 64607 fc5f sound/soc/codecs/rt1011.o After: text data bss dec hex filename 51299 13080 128 64507 fbfb sound/soc/codecs/rt1011.o (gcc version 9.2.1, amd64) Signed-off-by:
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Katsuhiro Suzuki authored
This patch set 0Hz to sysclk when shutdown the card. Some codecs set rate constraints that derives from sysclk. This mechanism works correctly if machine drivers give fixed frequency. But simple-audio and audio-graph card set variable clock rate if 'mclk-fs' property exists. In this case, rate constraints will go bad scenario. For example a codec accepts three limited rates (mclk / 256, mclk / 384, mclk / 512). Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - Codec set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 24kHz (1/512) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk to 8.192MHz (= fs * mclk-fs = 32k * 256) - Codec set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 16kHz (1/512) - Play 48kHz again, but it's NOT acceptable because constraints do not allow 48kHz So codecs treat 0Hz sysclk as signal of applying no constraints to avoid this problem. Signed-off-by:Katsuhiro Suzuki <katsuhiro@katsuster.net> Link: https://lore.kernel.org/r/20190907174501.19833-1-katsuhiro@katsuster.netSigned-off-by:
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Katsuhiro Suzuki authored
This patch ignores sysclk setting if it is 0Hz. Some codecs treat 0Hz sysclk as signal of applying no constraints. This driver does not have such feature but current implementation outputs 'Failed to set mclk' error message if machine driver sets 0Hz sysclk to this driver. Signed-off-by:
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Colin Ian King authored
Don't populate the arrays on the stack but instead make them static const. Makes the object code smaller by 37 bytes. Before: text data bss dec hex filename 16253 7200 0 23453 5b9d sound/soc/codecs/ad193x.o After: text data bss dec hex filename 16056 7360 0 23416 5b78 sound/soc/codecs/ad193x.o (gcc version 9.2.1, amd64) Signed-off-by:
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Katsuhiro Suzuki authored
This patch supports some type of machine drivers that set 0 to mclk when sound device goes to idle state. After applied this patch, sysclk == 0 means there is no constraint of sound rate and other values will set constraints which is derived by sysclk setting. Original code refuses sysclk == 0 setting. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Signed-off-by: -
Katsuhiro Suzuki authored
This patch removes redundant null checks for optional MCLK clock. And fix DT binding document for changing clock property to optional from required. Signed-off-by:
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Janusz Krzysztofik authored
Since commit 1137ceee ("ARM: OMAP1: ams-delta: Don't request unused GPIOs"), on-board audio has appeared muted. It has been discovered that believed to be unused GPIO pins "hookflash1" and "hookflash2" need to be set low for audible sound in handsfree and handset mode respectively. According to Amstrad E3 wiki, the purpose of both pins hasn't been clearly identified. Original Amstrad software used to produce a high pulse on them when the phone was taken off hook or recall was pressed. With the current findings, we can assume the pins provide a kind of audio mute function, separately for handset and handsfree operation modes. Commit 2afdb4c4 ("ARM: OMAP1: ams-delta: Fix audio permanently muted") attempted to fix the issue temporarily by hogging the GPIO pin "hookflash1" renamed to "audio_mute", however the fix occurred incomplete as it restored audible sound only for handsfree mode. Stop hogging that pin, rename the pins to "handsfree_mute" and "handset_mute" respectively and implement appropriate DAPM event callbacks for "Speaker" and "Earpiece" DAPM widgets. Fixes: 1137ceee ("ARM: OMAP1: ams-delta: Don't request unused GPIOs") Signed-off-by:
Janusz Krzysztofik <jmkrzyszt@gmail.com> Reviewed-by:
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Peter Ujfalusi authored
Some tools use the snd_pcm_info_get_name() to try to identify PCMs or for other purposes. Currently it is left empty with the dmaengine-pcm, in this case copy the pcm->id string as pcm->name. For example IGT is using this to find the HDMI PCM for testing audio on it. Signed-off-by:
Arthur She <arthur.she@linaro.org> Link: https://lore.kernel.org/r/20190906055524.7393-1-peter.ujfalusi@ti.comSigned-off-by:
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