- 18 Nov, 2019 7 commits
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Peter Ujfalusi authored
On boards where the RST line is not pulled up, but it is connected to a GPIO line this property must present in order to be able to enable the codec. Signed-off-by:
Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20191113124734.27984-2-peter.ujfalusi@ti.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Chuhong Yuan authored
The driver forgets to call pm_runtime_disable in remove and probe failure. Add the calls to fix it. Signed-off-by:
Chuhong Yuan <hslester96@gmail.com> Acked-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20191118073707.28298-1-hslester96@gmail.comSigned-off-by:
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Chuhong Yuan authored
This driver misses calls to pm_runtime_disable and regulator_bulk_disable in remove and a call to free_irq in probe failure. Add the calls to fix it. Signed-off-by:
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Vinod Koul authored
Qualcomm DSPs also support the flac decoder, so add support for FLAC decoder and convert the snd_dec_flac params to qdsp format. Co-developed-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by:
Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20191115102705.649976-4-vkoul@kernel.orgSigned-off-by:
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Srinivas Kandagatla authored
Qualcomm DSPs expect flac config to be set for flac decoders, so add the API to program the flac config to the DSP Signed-off-by:
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Vinod Koul authored
The current design of sending codec parameters assumes that decoders will have parsers so they can parse the encoded stream for parameters and configure the decoder. But this assumption may not be universally true and we know some DSPs which do not contain the parsers so additional parameters are required to be passed. So add these parameters starting with FLAC decoder. The size of snd_codec_options is still 120 bytes after this change (due to this being a union) Co-developed-by:
Takashi Iwai <tiwai@suse.de> Signed-off-by:
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kbuild test robot authored
Fixes: 21c00e5d ("ASoC: rt5677: Enable jack detect while DSP is running") Signed-off-by:
kbuild test robot <lkp@intel.com> Link: https://lore.kernel.org/r/20191114153304.n4pyix7qadu76tx4@4978f4969bb8Signed-off-by:
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- 15 Nov, 2019 6 commits
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Peter Ujfalusi authored
dma_request_slave_channel_reason() is: #define dma_request_slave_channel_reason(dev, name) \ dma_request_chan(dev, name) Signed-off-by:
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Peter Ujfalusi authored
dma_request_slave_channel_reason() is: #define dma_request_slave_channel_reason(dev, name) \ dma_request_chan(dev, name) Signed-off-by:
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Kuninori Morimoto authored
soc_cleanup_card_resources() will call card->remove(), but it should be called if card->probe() or card->late_probe() are called. snd_soc_bind_card() might be error before calling card->probe() / card->late_probe(). In that time, card->remove() will be called. This patch adds card_probed parameter to judge it. Fixes: bfce78a5 ("ASoC: soc-core: tidyup soc_init_dai_link()") Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by:
Jon Hunter <jonathanh@nvidia.com> Link: https://lore.kernel.org/r/87o8xg4ltr.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Kuninori Morimoto authored
To makes code readable, this patch moves snd_soc_unbind_card() next to snd_soc_bind_card(). 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. snd_soc_bind_card() is calling snd_soc_dapm_init() for both card and component. Let's call paired snd_soc_dapm_shutdown() at paired soc_cleanup_card_resources(). Signed-off-by:
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Colin Ian King authored
There is a block that is indented too deeply, remove the extraneous tabs. Signed-off-by:
Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20191112190218.282337-1-colin.king@canonical.comSigned-off-by:
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- 12 Nov, 2019 16 commits
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Rob Herring authored
'fsia' and 'fsib' property prefixes were added as schema in commit 2f52475b ("ASoC: fsi: switch to yaml base Documentation"). Unfortunately to do checks on actual vendor prefixes, we have to track the handful of prefixes which are not vendors like 'fsia' and 'fsib'. Fixes: 2f52475b ASoC: fsi: switch to yaml base Documentation Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Mark Brown <broonie@kernel.org> Signed-off-by:
Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/20191108153538.11970-1-robh@kernel.orgSigned-off-by:
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Kuninori Morimoto authored
snd_soc_bind_card() is calling many initialize functions for each card / link / dai / aux etc, etc, etc... When error happen, the message is indicated at snd_soc_bind_card(), not at each functions. But, only soc_probe_aux_devices() case is indicating error at functions, not at snd_soc_bind_card(). It is not an issue, but unbalanced. This patch moves error message to snd_soc_bind_card(). Also avoids deep-nested code. Signed-off-by:
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Kuninori Morimoto authored
having both soc_bind_card() and snd_soc_instantiate_card() is very confusable. Let's merge these. Signed-off-by:
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Kuninori Morimoto authored
having both soc_remove_component() and soc_cleanup_component() is very confusable. Let's merge these. Signed-off-by:
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Kuninori Morimoto authored
We don't need to separete snd_soc_remove_dai_link() and soc_unbind_dai_link() anymore. Let's merge these. Signed-off-by:
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Kuninori Morimoto authored
We don't need to separete snd_soc_add_dai_link() and soc_bind_dai_link() anymore. Let's merge these. One note is that before this patch, it adds list (A) eventhough if it had dai_link->ignore (1), or already bounded dai_link (2). But I guess it is wrong. This patch also solve this issue. /* BEFORE */ int soc_bind_dai_link(...) { ... (1) if (dai_link->ignore) return 0; (2) if (soc_is_dai_link_bound(...)) return 0; ... } int snd_soc_add_dai_link(...) { ... => ret = soc_bind_dai_link(...); => if (ret < 0) => return ret; (A) list_add_tail(&dai_link->list, &card->dai_link_list); ... } /* AFTER */ int snd_soc_add_dai_link(...) { ... (1) if (dai_link->ignore) return 0; (2) if (soc_is_dai_link_bound(...)) return 0; ... (A) list_add_tail(&dai_link->list, &card->dai_link_list); return 0; } Signed-off-by: -
Kuninori Morimoto authored
We don't need to separete snd_soc_unregister_dai() and soc_del_dai() anymore. Let's merge these Signed-off-by:
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Kuninori Morimoto authored
We don't need to separete snd_soc_register_dai() and soc_add_dai() anymore. Let's merge these Signed-off-by:
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Kuninori Morimoto authored
Because complex separeted "card pre-listed component" and "topology added component" duplicated operation is now becoming simple, we don't need to check already bound dai_link which is not exist anymore. This patch removes soc_is_dai_link_bound(). Signed-off-by:
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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:
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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 11 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:
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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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