- 01 Oct, 2019 39 commits
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Liam Girdwood authored
Helps with FW debug as it provides DSP IPC processing context. Signed-off-by:
Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190927200538.660-13-pierre-louis.bossart@linux.intel.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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Marcin Rajwa authored
This patch prevents the reset of host period bytes and uses no_stream_position to record requests for stream position. Signed-off-by:
Marcin Rajwa <marcin.rajwa@linux.intel.com> Signed-off-by:
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Marcin Rajwa authored
The host period bytes value needs to be passed to firmware. However current implementation uses this field for different purpose - to indicate whether FW should send stream position to the host. Therefore this patch introduces another field "no_stream_position", a boolean value aimed to store information about position tracking. This way host_period_bytes preserves its original value. Signed-off-by:
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Ranjani Sridharan authored
fw_ready should be a mandatory op. Make sure fw_ready ops is set during probe. Signed-off-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by:
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Liam Girdwood authored
Add config option to prevent DSP entering D3 after any FW exception. This can then be used to dump FW context for debug. Signed-off-by:
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Pierre-Louis Bossart authored
In a number of debug cases, the DMA-based trace can add problems (e.g. with HDaudio channel allocation). It also generates additional traffic on the bus and if the DMA handling is unreliable will prevent audio use-cases from working normally. Using the trace also requires tools to be installed on the target. The trace can be instead handled as dynamic debug. We can use a Kconfig to force the trace to be enabled in all cases, or use a module parameter to enable it on a need-basis, e.g. by setting "options snd_sof sof_debug=0x1" in a /etc/modprobe.d file. Signed-off-by:
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Charles Keepax authored
Some Madera CODECs have a third MCLK pin allow this to be sourced for the FLLs and system clocks. Signed-off-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20191001135700.31017-1-ckeepax@opensource.cirrus.comSigned-off-by:
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Frank Shi authored
add tas2770 smart PA kernel driver Signed-off-by:
Frank Shi <shifu0704@thundersoft.com> Link: https://lore.kernel.org/r/1568962709-19185-2-git-send-email-shifu0704@thundersoft.comSigned-off-by:
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Peter Ujfalusi authored
On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by:
Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by:
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Shengjiu Wang authored
There is error "aplay: pcm_write:2023: write error: Input/output error" on i.MX8QM/i.MX8QXP platform for S24_3LE format. In i.MX8QM/i.MX8QXP, the DMA is EDMA, which don't support 24bit sample, but we didn't add any constraint, that cause issues. So we need to query the caps of dma, then update the hw parameters according to the caps. Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by:
Nicolin Chen <nicoleotsuka@gmail.com> Link: https://lore.kernel.org/r/b6a4de2bbf960ef291ee902afe4388bd0fc1d347.1569493933.git.shengjiu.wang@nxp.comSigned-off-by:
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Shengjiu Wang authored
When set the runtime hardware parameters, we may need to query the capability of DMA to complete the parameters. This patch is to Extract this operation from dmaengine_pcm_set_runtime_hwparams function to a separate function snd_dmaengine_pcm_refine_runtime_hwparams, that other components which need this feature can call this function. Signed-off-by:
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Shengjiu Wang authored
The ASRC support 24bit/16bit/8bit input width, which is data width, not slot width. For the S20_3LE format, the data with is 20bit, slot width is 24bit, if we set ASRMCR1n.IWD to be 24bits, the result is the volume is lower than expected, it likes 24bit data right shift 4 bits So replace S20_3LE with S24_3LE in supported list and add S8 format in TX supported list Signed-off-by:
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Shengjiu Wang authored
snd_pcm_format_t is more formal than enum asrc_word_width, which has two property, width and physical width, which is more accurate than enum asrc_word_width. So it is better to use in(out)put_format instead of in(out)put_word_width. Signed-off-by:
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Lars-Peter Clausen authored
The adau1761 has a automatic level control block that can adjust the gain for the differential input PGA. This patch adds ALSA controls for enabling and changing the parameter settings for the ALC. Signed-off-by:
Lars-Peter Clausen <lars@metafoo.de> Signed-off-by:
Alexandru Ardelean <alexandru.ardelean@analog.com> Link: https://lore.kernel.org/r/20190926115012.24049-2-alexandru.ardelean@analog.comSigned-off-by:
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Lars-Peter Clausen authored
The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by:
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Daniel Baluta authored
For a sound card ignore_machine means that existing FEs links should be ignored and existing BEs links should be overridden with some information from the matching component driver. Current code make some confusions about this so fix it! Signed-off-by:
Daniel Baluta <daniel.baluta@nxp.com> Link: https://lore.kernel.org/r/20190925183358.11955-1-daniel.baluta@nxp.comSigned-off-by:
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Colin Ian King authored
There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by:
Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20190925112621.9312-1-colin.king@canonical.comSigned-off-by:
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Colin Ian King authored
There is an if statement that is indented one level too deeply, remove the extraneous tabs. Signed-off-by:
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Colin Ian King authored
There are two break statements that are indented one level too deeply, remove the extraneous tabs. Signed-off-by:
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Colin Ian King authored
There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by:
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Frank Shi authored
Add tas2770 smart PA dt bindings Signed-off-by:
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Kuninori Morimoto authored
If we can use devm_kzalloc(rtd->dev, xxx) for rtdcom, we don't need to call snd_soc_rtdcom_del_all() for kfree(). This patch uses devm_kzalloc(rtd->dev, xxx) for rtdcom, and remove snd_soc_rtdcom_del_all(). Signed-off-by:
Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/875zlyf7ln.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Kuninori Morimoto authored
This patch allocs dev first at soc_new_pcm_runtime(). This is prepare for rtd->dev, rtd->codec_dais alloc cleanup Signed-off-by:
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Kuninori Morimoto authored
"rtd" is handled by soc_xxx_pcm_runtime(), and "rtd->dev" is handled by soc_rtd_xxx(). There is no reason to separate these, and it makes code complex. We can free these in the same time. Here soc_rtd_free() (A) which frees rtd->dev is called from soc_remove_link_dais() many times (1). Then, it is using dev_registered flags to avoid multi kfree() (2). This is no longer needed if we can merge these functions. static void soc_remove_link_dais(...) { ... (1) for_each_comp_order(order) { (1) for_each_card_rtds(card, rtd) { (A) soc_rtd_free(rtd); ... } } } (A) static void soc_rtd_free(...) { (2) if (rtd->dev_registered) { /* we don't need to call kfree() for rtd->dev */ device_unregister(rtd->dev); (2) rtd->dev_registered = 0; } } This patch merges soc_rtd_free() into soc_free_pcm_runtime(). Signed-off-by: -
Kuninori Morimoto authored
"rtd" is handled by soc_xxx_pcm_runtime(), and "rtd->dev" is handled by soc_rtd_xxx(). There is no reason to separate these, and it makes code complex. We can create these in the same time. This patch merges soc_rtd_init() into soc_new_pcm_runtime(). Signed-off-by:
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Kuninori Morimoto authored
soc_new_pcm_runtime() allocs rtd and rtd->codec_dais. This patch allocs both first, and setup these after that. This is prepare for soc_new_pcm_runtime() cleanup. Signed-off-by:
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Kuninori Morimoto authored
Current ALSA SoC is calling list_del(&rtd->list) at (1) static void soc_remove_pcm_runtimes(...) { ... for_each_card_rtds_safe(card, rtd, _rtd) { (1) list_del(&rtd->list); (2) soc_free_pcm_runtime(rtd); } ... } But, we will call soc_free_pcm_runtime() after that (2). &rtd->list is connected at soc_new_pcm_runtime(), Thus, it should be disconnected at soc_free_pcm_runtime(). This patch calls list_del(&rtd->list) at soc_free_pcm_runtime(). Signed-off-by: -
Kuninori Morimoto authored
We have soc_new_pcm_runtime() which allocs rtd and its related memory, and soc_add_pcm_runtime() which connects rtd to card. But we don't need to separate these, we can alloc and connect rtd in the same time. Current implementation is just makes code complex. This patch merges these into one. Signed-off-by:
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Kuninori Morimoto authored
This patch moves soc_free_pcm_runtime() next to soc_new_pcm_runtime(). This is prepare for soc_xxx_pcm_runtime() cleanup. Signed-off-by:
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Sylwester Nawrocki authored
Add compatible string for boards with WM1811 CODEC to the list. Acked-by:
Krzysztof Kozlowski <krzk@kernel.org> Signed-off-by:
Sylwester Nawrocki <s.nawrocki@samsung.com> Link: https://lore.kernel.org/r/20190920130218.32690-7-s.nawrocki@samsung.comSigned-off-by:
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Sylwester Nawrocki authored
As an intermediate step before covering the clocking subsystem of the CODEC entirely by the clk API add handling of external CODEC's master clocks in DAPM events when the AIFn clocks are sourced directly from MCLKn; when FLLn are used we enable/disable respective MCLKn before/after FLLn is enabled/disabled. Signed-off-by:
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Sylwester Nawrocki authored
Extend the set_sysclk() handler so we also set frequency of the MCLK1, MCLK2 clocks through clk API when those clocks are specified in DT. Reviewed-by:
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Tzung-Bi Shih authored
Headset codec is optional. Add missing update to DT binding document. Fixes: a962a809 ("ASoC: mediatek: mt8183: make headset codec optional") Signed-off-by:
Tzung-Bi Shih <tzungbi@google.com> Link: https://lore.kernel.org/r/20190920112320.166052-1-tzungbi@google.comSigned-off-by:
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Kuninori Morimoto authored
soc-component.h already has SPDX License, thus, GPL explanation is not needed. This patch removes it. Signed-off-by:
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Markus Elfring authored
Simplify this function implementation by using a known wrapper function. This issue was detected by using the Coccinelle software. Signed-off-by:
Markus Elfring <elfring@users.sourceforge.net> Link: https://lore.kernel.org/r/8f7cf483-6ab3-d00f-5606-863e9f5b31fc@web.deSigned-off-by:
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Maxime Ripard authored
The Allwinner A23 SoC and later have 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> Reviewed-by:
Rob Herring <robh@kernel.org> Link: https://lore.kernel.org/r/20190906151221.3148-2-mripard@kernel.orgSigned-off-by:
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Sam McNally authored
As of commit 648e9218 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL"), the cht_bsw_rt5645 driver needs to enable the clock it's using for the codec's mclk. It does this from commit 7735bce0 ("ASoC: Intel: boards: use devm_clk_get() unconditionally"), enabling pmc_plt_clk_3. However, Strago family Chromebooks use pmc_plt_clk_0 for the codec mclk, resulting in white noise with some digital microphones. Add a DMI-based quirk for Strago family Chromebooks to use pmc_plt_clk_0 instead - mirroring the changes made to cht_bsw_max98090_ti in commit a182ecd3 ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0") and making use of the existing dmi_check_system() call and related infrastructure added in commit 22af2911 ("ASoC: Intel: cht-bsw-rt5645: add quirks for SSP0/AIF1/AIF2 routing"). Signed-off-by:
Sam McNally <sammc@chromium.org> Acked-by:
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Shengjiu Wang authored
MQS (medium quality sound), is used to generate medium quality audio via a standard digital output pin. It can be used to connect stereo speakers or headphones simply via power amplifier stages without an additional DAC chip. It only accepts 2-channel, LSB-valid 16bit, MSB shift-out first, frame sync asserting with the first bit of the frame, data shifted with the posedge of bit clock, 44.1 kHz or 48 kHz signals from SAI1 in left justified format; and it provides the SNR target as no more than 20dB for the signals below 10 kHz. The signals above 10 kHz will have worse THD+N values. MQS provides only simple audio reproduction. No internal pop, click or distortion artifact reduction methods are provided. The MQS receives the audio data from the SAI1 Tx section. Signed-off-by:
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Shengjiu Wang authored
Add the DT binding documentation for NXP MQS driver Signed-off-by:
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- 30 Sep, 2019 1 commit
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Linus Torvalds authored
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