- 12 Nov, 2019 7 commits
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Dragos Tarcatu authored
The volume and bytes kcontrols are currently not freeing their memory on initialization failures. When an error occurs, all the widgets loaded so far are unloaded via sof_widget_unload(). But this only happens for the widgets that got successfully loaded. Fix that by kfree()-ing the allocated memory on load error. Fixes: 311ce4fe ("ASoC: SOF: Add support for loading topologies") Reviewed-by:
Paul Olaru <paul.olaru@nxp.com> Reviewed-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by:
Dragos Tarcatu <dragos_tarcatu@mentor.com> Signed-off-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20191111222039.19651-1-pierre-louis.bossart@linux.intel.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Keyon Jie authored
We should suspend audio to D3 by default, for the sake of power saving, change the condition of D0I3 suspending here to that when there is stream with suspend_ignored specified. Signed-off-by:
Keyon Jie <yang.jie@linux.intel.com> Signed-off-by:
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Keyon Jie authored
Add helper to check if the DSP should be put in D0i3. This function returns true if a stream has ignored the SUSPEND trigger to keep the pipelines running in the DSP. Signed-off-by:
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Keyon Jie authored
Add check before seeting d0_substate and return success if Audio DSP is already in the target substate. Signed-off-by:
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Keyon Jie authored
Add Audio DSP state machine with comments. Note that the 'D0<-->runtime D0I3' part is not implemented yet. Signed-off-by:
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Pierre-Louis Bossart authored
We have platforms such as CFL with no known I2S codec being used, and the ACPI tables are currently empty, so fall-back to using the firmware filename used in nocodec mode Signed-off-by:
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Pierre-Louis Bossart authored
Due to firmware manifest/signature differences, we have to use different firmware names, so split CNL machine table in three (CNL, CFL, CML). The CFL table is currently empty since all known platforms use HDaudio, but let's plan ahead. Reviewed-by:
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- 11 Nov, 2019 15 commits
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Michael Walle authored
This adds a new mode WM8904_CLK_AUTO which automatically enables the FLL if a frequency different than the MCLK is set. These additions make the codec work with the simple-card driver in general and especially in systems where the MCLK doesn't match the required clock. Signed-off-by:
Michael Walle <michael@walle.cc> Acked-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20191108203152.19098-1-michael@walle.ccSigned-off-by:
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Yu-Hsuan Hsu authored
On KBL platform, the microphone is attached to external codec(rt5514) instead of PCH. However, TDM slot between PCH and codec is 16 bits only. In order to avoid setting wrong format, we should add a constraint to force to use 16 bits format forever. Signed-off-by:
Yu-Hsuan Hsu <yuhsuan@chromium.org> Acked-by:
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Sebastian Reichel authored
This commit adds the Dialog DA7213 audio codec as a selectable option in the kernel config. Currently the driver can only be selected for Intel Baytrail/Cherrytrail devices or if SND_SOC_ALL_CODECS is enabled. Signed-off-by:
Sebastian Reichel <sebastian.reichel@collabora.com> Link: https://lore.kernel.org/r/20191108174843.11227-3-sebastian.reichel@collabora.comSigned-off-by:
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Curtis Malainey authored
Wait for hardware to startup. If we load before hardware is ready we could end up corrupting the firmware. Signed-off-by:
Curtis Malainey <cujomalainey@chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-12-cujomalainey@chromium.orgSigned-off-by:
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Curtis Malainey authored
Use the PLL to kept the correct 24M clock rate so frequency shift does not occur when using the DSP VAD. Signed-off-by:
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Curtis Malainey authored
The RT5677 DSP needs the I2S MCLK1 to run its DSP. Add a dapm route to SSP0 CODEC IN so the clock is turned on automatically when the DSP is turned on. Signed-off-by:
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Curtis Malainey authored
Due to limitations of the clocking configuration, we have no way of scheduling our hibernation before the bdw dsp hibernates. This causes issues when the system suspends with an open stream. We need userspace to toggle the kcontrol before we are suspended so that any writes on suspend are not lost and we don't corrupt the regmap. Signed-off-by:
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Ben Zhang authored
The irq is disabled at suspend to avoid running the threaded irq handler after the codec has been powered off. At resume, codec irq is re-enabled and the interrupt status register is checked to see if headphone has been pluggnd/unplugged while the device is suspended. There is still a chance that the headphone gets enabled or disabled after the codec is suspended. disable_irq syncs the threaded irq handler, but soc-jack's threaded irq handler schedules a delayed work to poll gpios (for debounce). This is still OK. The codec won't be powered back on again because all audio paths have been suspended, and there are no force enabled supply widgets (MICBIAS1 is disabled). The gpio status read after codec power off could be wrong, so the gpio values are checked again after resume. Signed-off-by:
Ben Zhang <benzh@chromium.org> Signed-off-by:
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Ben Zhang authored
MCLK1 gets disabled at suspend and re-enabled at resume. Before MCLK1 is re-enabled, if the DSP is already on (either the DSP was left on during suspend, or the DSP is turned on early at resume), i2c register read returns garbage and corrupts the regmap cache. This patch stops the DSP before suspend and restarts it after resume with a dalay to ensure MCLK is on while loading firmware. Signed-off-by:
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Ben Zhang authored
The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1 while it's streaming audio over SPI. The DSP firmware turns on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap does not believe that register can change by itself. When BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't read the register first before write, so PLL2 power bit is cleared by accident. Marking MX-64h as volatile in regmap solved the issue. Signed-off-by:
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Ben Zhang authored
Add a DAPM audio path from "DMIC L1" to "DSP Buffer" so that when hotwording is enabled, DAPM does not power off the codec with SND_SOC_BIAS_OFF. Signed-off-by:
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Ben Zhang authored
Before a hotword is detected, GPIO1 pin is configured as IRQ output so that jack detect works. When a hotword is detected, the DSP firmware configures the GPIO1 pin as GPIO1 and drives a 1. rt5677_irq() is called after a rising edge on the GPIO1 pin, due to either jack detect event or hotword event, or both. All possible events are checked and handled in rt5677_irq() where GPIO1 pin is configured back to IRQ output if a hotword is detected. Signed-off-by:
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Ben Zhang authored
This link is needed for the RT5677 DSP to do hotwording Signed-off-by:
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Ben Zhang authored
The firmware rt5677_elf_vad is an ELF binary obtained from request_firmware(). Sections of the ELF are loaded to the DSP via SPI. A model (e.g. en_us.mmap) can optionally be loaded to the DSP at RT5677_MODEL_ADDR to overwrite the baked-in model in rt5677_elf_vad. Then we switch to DSP mode, load firmware, and let DSP run. When a hotword is detected, an interrupt is fired and rt5677_irq() is called. When 'DSP VAD Switch' is turned off, the codec is set back to normal mode. The kcontrol 'DSP VAD Switch' is automatically enabled/disabled when the hotwording PCM stream is opened/closed. Signed-off-by:
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Shengjiu Wang authored
Audmix support two substream, When two substream start to run, the trigger function may be called by two substream in same time, that the priv->tdms may be updated wrongly. The expected priv->tdms is 0x3, but sometimes the result is 0x2, or 0x1. Fixes: be1df61c ("ASoC: fsl: Add Audio Mixer CPU DAI driver") Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by:
Nicolin Chen <nicoleotsuka@gmail.com> Reviewed-by:
Daniel Baluta <daniel.baluta@nxp.com> Link: https://lore.kernel.org/r/1e706afe53fdd1fbbbc79277c48a98f8416ba873.1573458378.git.shengjiu.wang@nxp.comSigned-off-by:
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- 08 Nov, 2019 3 commits
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Keyon Jie authored
Set L1SEN to make sure the system can enter S0ix, and restore it on resume. Signed-off-by:
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Keyon Jie authored
Add check to avoid possible NULL pointer dereference issue. This issue was reported by static analysis tools, we didn't face this issue but we can't rule it out either as a false positive. Reported-by:
Keqiao Zhang <keqiao.zhang@intel.com> Signed-off-by:
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Takashi Iwai authored
Paper over a compile warning: sound/soc/soc-pcm.c:1185:8: warning: unused variable ‘name’ Fixes: 0632fa04 ("ASoC: core: Fix pcm code debugfs error") Signed-off-by:
Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20191107134833.1502-1-tiwai@suse.deSigned-off-by:
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- 07 Nov, 2019 1 commit
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Kuninori Morimoto authored
This patch switches from .txt base to .yaml base Document for FSI. Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by:
Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/878sp4jaqy.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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- 06 Nov, 2019 5 commits
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Dragos Tarcatu authored
When using the example SOF amp widget topology, KASAN dumps this when the AMP bytes kcontrol gets loaded: [ 9.579548] BUG: KASAN: slab-out-of-bounds in sof_control_load+0x8cc/0xac0 [snd_sof] [ 9.588194] Write of size 40 at addr ffff8882314559dc by task systemd-udevd/2411 Fix that by rejecting the topology if the bytes data size > max_size Fixes: 311ce4fe ("ASoC: SOF: Add support for loading topologies") Reviewed-by:
Jaska Uimonen <jaska.uimonen@intel.com> Reviewed-by:
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Mark Brown authored
Merge branch 'for-5.4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-5.5
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Kuninori Morimoto authored
snd_soc_unregister_component() is calling snd_soc_lookup_component() under mutex_lock(). But, snd_soc_lookup_component() itself is using mutex_lock(), thus it will be dead-lock. This patch adds _nolocked version of it, and avoid dead-lock issue. Fixes: ac6a4dd3("ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component()") Reported-by: "kernelci.org bot" <bot@kernelci.org>" Signed-off-by:
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Pierre-Louis Bossart authored
SOF module load/unload tests show nasty recurring warnings: WARNING: CPU: 5 PID: 1339 at sound/core/control.c:466 snd_ctl_remove+0xf0/0x100 [snd] RIP: 0010:snd_ctl_remove+0xf0/0x100 [snd] This regression was introduced by the removal of the call to soc_remove_link_components() before soc_card_free() is invoked. Go back to the initial order but only call soc_remove_link_components() once. Reviewed-by:
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Dragos Tarcatu authored
When using the example SOF amp widget topology, KASAN dumps this when the AMP bytes kcontrol gets loaded: [ 9.579548] BUG: KASAN: slab-out-of-bounds in sof_control_load+0x8cc/0xac0 [snd_sof] [ 9.588194] Write of size 40 at addr ffff8882314559dc by task systemd-udevd/2411 Fix that by rejecting the topology if the bytes data size > max_size Fixes: 311ce4fe ("ASoC: SOF: Add support for loading topologies") Reviewed-by:
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- 05 Nov, 2019 9 commits
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Kuninori Morimoto authored
snd_soc_dobj is used only when SND_SOC_TOPOLOGY was selected. Let's enable it under SND_SOC_TOPOLOGY. Signed-off-by:
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Kuninori Morimoto authored
soc-core has some API which is used from topology, but it is doing topology specific operation at soc-core. soc-core should care about core things, and topology should care about topology things, otherwise, it is very confusable. For example topology type is not related to soc-core, it is topology side issue. This patch removes meaningless check from soc-core. This patch keeps extra initialization/destruction at snd_soc_add_dai_link() / snd_soc_remove_dai_link() which were for topology. From this patch, non-topology card can use it. Signed-off-by:
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Kuninori Morimoto authored
ALSA SoC has 2 functions. snd_soc_register_dai() is used from topology snd_soc_register_dais() is used from snd_soc_add_component() In general, people think like _dai() is called from _dais() with for loop. But in reality, these are very similar but different implementation. We shouldn't have duplicated and confusing implementation. This patch calls snd_soc_register_dai() from snd_soc_register_dais() Signed-off-by:
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Kuninori Morimoto authored
ALSA SoC has 2 functions. snd_soc_register_dai() is used from topology snd_soc_register_dais() is used from snd_soc_add_component() In general, people think like _dai() is called from _dais() with for loop. But in reality, these are very similar but different implementation. We shouldn't have duplicated and confusing implementation. snd_soc_register_dai() is now used from topology. But to reduce duplicated code, it should be used from _dais(), too. Because of topology side specific reason, it is calling snd_soc_dapm_new_dai_widgets(), but it is not needed _dais() side. This patch factorizes snd_soc_register_dai() to topology / _dais() common part, and topology specific part. And do topology specific part at soc-topology. Signed-off-by:
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Kuninori Morimoto authored
ALSA SoC has 2 functions. snd_soc_register_dai() is used from topology snd_soc_register_dais() is used from snd_soc_add_component() In general, people think like _dai() is called from _dais() with for loop. But in reality, these are very similar but different implementation. We shouldn't have duplicated and confusing implementation. snd_soc_register_dai() is now used from topology. But to reduce duplicated code, it should be used from _dais(), too. To prepare it, this patch adds missing parameter legacy_dai_naming to snd_soc_register_dai(). 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 is difficult to debug. This patch adds missing soc_del_dai() and snd_soc_unregister_dai(). Signed-off-by:
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Kuninori Morimoto authored
This patch moves snd_soc_unregister_dais() next to snd_soc_register_dais(). This is prepare for snd_soc_register_dais() cleanup Signed-off-by:
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Kuninori Morimoto authored
This patch moves snd_soc_register_dai() next to snd_soc_register_dais(). This is prepare for snd_soc_register_dais() cleanup. Signed-off-by:
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Kuninori Morimoto authored
snd_soc_unregister_component() is now finding component manually, but we already have snd_soc_lookup_component() to find component; Let's use existing function. Signed-off-by:
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