- 09 Sep, 2019 8 commits
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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 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:
Mark Brown <broonie@kernel.org>
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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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- 05 Sep, 2019 10 commits
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YueHaibing authored
When do compile test, if SND_SOC_SOF_OF is not set, we get: sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_request': imx8.c:(.text+0xb0): undefined reference to `snd_sof_ipc_msgs_rx' sound/soc/sof/imx/imx8.o: In function `imx8_ipc_msg_data': imx8.c:(.text+0xf4): undefined reference to `sof_mailbox_read' sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_reply': imx8.c:(.text+0x160): undefined reference to `sof_mailbox_read' Make SND_SOC_SOF_IMX_TOPLEVEL always depends on SND_SOC_SOF_OF Reported-by:
Hulk Robot <hulkci@huawei.com> Fixes: 202acc56 ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by:
YueHaibing <yuehaibing@huawei.com> Reviewed-by:
Daniel Baluta <daniel.baluta@nxp.com> Reviewed-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190905064400.24800-1-yuehaibing@huawei.comSigned-off-by:
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Jerome Brunet authored
On the sm1, the TDMOUT number of input is extended and the the gain enable bit moved to accommodate this extension Signed-off-by:
Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20190905120120.31752-9-jbrunet@baylibre.comSigned-off-by:
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Jerome Brunet authored
On sm1, the maximum number TODDR inputs is extended to 16. Signed-off-by:
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Jerome Brunet authored
On sm1, the output routing bits have moved to CTRL2 register Signed-off-by:
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Jerome Brunet authored
The TDMIN component, as it, has a maximum of 16 input. Depending on the SoC, these may not all be connected. Instead of decribing only the connected inputs of each SoC, describe them all and let ASoC routing do the rest. Signed-off-by:
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Jerome Brunet authored
The TODDR component, as it, has a maximum of 8 input. Depending on the SoC, these may not all be connected or some input components may not be supported Instead of decribing only the connected inputs, describe them all and let ASoC routing do the rest. Signed-off-by:
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Jerome Brunet authored
The FRDDR component, as it, has a maximum of 8 outputs. Depending on the SoC, these may not all be connected. Instead of decribing only the connected outputs of each SoC, describe them all and let ASoC routing do the rest. Signed-off-by:
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Jerome Brunet authored
The g12a audio subsystem, which is a derivative of the axg subsystem, provides a dedicated reset line for each of the audio components. The axg did not provide that and it is unclear if/when these reset are required. The reset already helped solve a channel mapping issue on the tdm formatter devices. Let's add the reset binding for the other components, so we can describe this in DT. We'll use it later on in the driver when/if needed. Signed-off-by:
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Jerome Brunet authored
Document the compatible strings of the audio devices of the sm1 SoC family Signed-off-by:
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Bjorn Andersson authored
Reading out the link-name earlier and including it in the various error messages makes it much more convenient to figure out what links have unmet dependencies. Signed-off-by:
Bjorn Andersson <bjorn.andersson@linaro.org> Link: https://lore.kernel.org/r/20190905040306.21399-1-bjorn.andersson@linaro.orgSigned-off-by:
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- 04 Sep, 2019 5 commits
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Kuninori Morimoto authored
Basically, driver which setups snd_soc_component_set_jack() need to release it by themselves. But, as framework level robustness, soc_remove_component() also releases it. To avoid code reader confuse, this patch makes it clarify. This patch makes it clarify. Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/875zm8q5n8.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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YueHaibing authored
Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by:
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YueHaibing authored
Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by:
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YueHaibing authored
Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by:
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YueHaibing authored
Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by:
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- 03 Sep, 2019 4 commits
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Katsuhiro Suzuki authored
This patch introduce clock property for MCLK master freq control. Driver will set rate of MCLK master if set_sysclk is called and changing sysclk by board driver. [Modified slightly to apply without an earlier patch in the series due to context diffs -- broonie] Signed-off-by:
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Katsuhiro Suzuki authored
This patch adds missing DT-bindings document for Everest ES8316. Signed-off-by:
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Kuninori Morimoto authored
ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform were unbinded and re-binded again. But, Audio Graph Card might can't rebind again if user do for example unbind CPU or Codec driver bind CPU or Codec driver Because Audio Graph Card is still pointing old/unbinded CPU or Codec driver's DAI name at dlc->dai_name. To avoid this issue, it needs to alloc memory and keep DAI name even though if CPU or Codec driver was unbinded. Or, always do unbind/bind at Sound Card. For now, this patch indicates this issue as FIXME. Signed-off-by:
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Kuninori Morimoto authored
ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform were unbinded and re-binded again. But, Simple Card might can't rebind again if user do for example unbind CPU or Codec driver bind CPU or Codec driver Because Simple Card is still pointing old/unbinded CPU or Codec driver's DAI name at dlc->dai_name. To avoid this issue, it needs to alloc memory and keep DAI name even though if CPU or Codec driver was unbinded. Or, always do unbind/bind at Sound Card. For now, this patch indicates this issue as FIXME. Signed-off-by:
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- 02 Sep, 2019 5 commits
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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_probe_link_components() has paired soc_remove_link_components(), but, these are implemented at different place. So it is difficult to confirm code. This patch moves soc_probe_link_components() next to soc_remove_link_components(). Signed-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-dapm has snd_soc_dapm_free() which cleanups debugfs, widgets, list. But, there is no paired initialize function. This patch adds snd_soc_dapm_init() and initilaizing dapm Signed-off-by:
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Kuninori Morimoto authored
Current ASoC setups some dapm related member at snd_soc_component_initialize() which is called when component was registered, and setups remaining member at soc_probe_component() which is called when component was probed. This kind of setup separation is no meanings, and it is very difficult to read and confusable. This patch setups all dapm settings at one place. Signed-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_probe_comonent() has paired soc_remove_comonent(), but, these are implemented at different place. So it is difficult to confirm code. This patch moves soc_probe_component() next to soc_remove_component(). Signed-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_rtd_init() was soc_post_component_init(), but there was no its paired soc_post_component_free(), but it is done at soc_remove_link_dais(). This means it is difficult to find related code. This patch adds soc_rtd_free() which is paired soc_rtd_init(). soc_rtd_xxx() will be more cleanuped in the future. Signed-off-by:
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- 30 Aug, 2019 8 commits
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Maxime Ripard authored
The Allwinner A64 SoC has an embedded audio codec that uses a separate controller to drive its analog part, which is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by:
Maxime Ripard <maxime.ripard@bootlin.com> Link: https://lore.kernel.org/r/20190828125209.28173-5-mripard@kernel.orgReviewed-by:
Rob Herring <robh@kernel.org> Signed-off-by:
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Maxime Ripard authored
The Allwinner A33 SoC have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by:
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Maxime Ripard authored
Even though the H6 compatible has been properly added, the exeption for the number of DMA channels hasn't been updated, leading in a validation warning. Fix this. Fixes: b2045303 ("dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible") Signed-off-by:
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Tzung-Bi Shih authored
Fix space issues: - No space before ';'. - No trailing blank line. Signed-off-by:
Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20190830074240.195166-5-tzungbi@google.comSigned-off-by:
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Tzung-Bi Shih authored
The forward declaration of mt8183_mt6358_ts3a227_max98357_headset_init is for cyclic dependency between card, headset_dev, and headset_init. It used to be: - card depends on headset_dev - headset_dev depends on headset_init - headset_init depends on card Commit a962a809 ("ASoC: mediatek: mt8183: make headset codec optional") removed the cyclic dependency. Thus, it is safe to remove the forward declaration. Signed-off-by:
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Tzung-Bi Shih authored
Move headset jack descriptor from module global scope to card-specific storage to make its ownership more explicit. Signed-off-by:
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Tzung-Bi Shih authored
Move private structure to the beginning of file to declare earlier so that most functions can see it. Signed-off-by:
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Pierre-Louis Bossart authored
We should only select SND_INTEL_NHLT when ACPI is defined. This was done for the legacy HDAudio driver but not for DSP-enabled cases, leading to compilation errors with randconfig. Fix by aligning on the same solution. For the Skylake driver this is overkill since there is a top-level dependency on ACPI, but it doesn't hurt and it's better to have consistency. Fixes: 68b953ae ('ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number') Reported-by:
Randy Dunlap <rdunlap@infradead.org> Signed-off-by:
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