August 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> 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: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

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[alsa-devel] [PATCH 1/2] PCI: Add a helper to check Power Resource Requirements _PR3 existence
by Kai-Heng Feng 20 Sep '19

20 Sep '19

A driver may want to know the existence of _PR3, to choose different runtime suspend behavior. A user will be add in next patch. This is mostly the same as nouveau_pr3_present(). Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- drivers/pci/pci.c | 20 ++++++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 22 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index 1b27b5af3d55..776af15b92c2 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,26 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct pci_dev *parent_pdev = pci_upstream_bridge(pdev); + struct acpi_device *parent_adev; + + if (acpi_disabled) + return false; + + if (!parent_pdev) + return false; + + parent_adev = ACPI_COMPANION(&parent_pdev->dev); + if (!parent_adev) + return false; + + return parent_adev->power.flags.power_resources && + acpi_has_method(parent_adev->handle, "_PR3"); +} +EXPORT_SYMBOL_GPL(pci_pr3_present); + /** * pci_add_dma_alias - Add a DMA devfn alias for a device * @dev: the PCI device for which alias is added diff --git a/include/linux/pci.h b/include/linux/pci.h index 82e4cd1b7ac3..9b6f7b67fac9 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2348,9 +2348,11 @@ struct irq_domain *pci_host_bridge_acpi_msi_domain(struct pci_bus *bus); void pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); +bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } +static bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH -- 2.17.1

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[alsa-devel] [PATCH v6 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 19 Sep '19

19 Sep '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v5: - updated slave bindings with proper licencing and fixed up examples. Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 62 + .../soundwire/soundwire-controller.yaml | 82 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1345 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

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[alsa-devel] [PATCH v5 0/5] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 18 Sep '19

18 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Changes from v4 to v5: - synopsys/Kconfig: Remove the incorrect dependency change in v4. - rockchip/Kconfig: Add dependency of hdmi-codec when it is really need for jack support. Cheng-Yi Chiang (5): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + include/sound/hdmi-codec.h | 17 +++ sound/soc/codecs/hdmi-codec.c | 46 +++++++ sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 8 files changed, 216 insertions(+), 27 deletions(-) -- 2.22.0.510.g264f2c817a-goog

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[alsa-devel] [PATCH 0/6] soundwire: inits and PM additions for 5.4
by Pierre-Louis Bossart 16 Sep '19

16 Sep '19

This is an update on the RFC, to be applied after the '[PATCH v2 0/3] soundwire: debugfs support for 5.4' and '[PATCH 00/17] soundwire: fixes for 5.4' series. Total that makes 28 patches submitted for review, broken in 3 sets. Changes since RFC (Feedback from GregKH, Vinod, Cezary, Guennadi): Squashed init sequence fixes in one patch, which remains readable. Tested all return values and called update_config() as needed. Fixed hw-reset debugfs (removed -unsafe and noisy dev_info traces) Simplified enable_interrupt() with goto Fixed style, removed typos and FIXMES in pm_runtime code Clarified commit messages Pierre-Louis Bossart (6): soundwire: fix startup sequence for Intel/Cadence soundwire: cadence_master: add hw_reset capability in debugfs soundwire: intel: add helper for initialization soundwire: intel: Add basic power management support soundwire: cadence_master: make clock stop exit configurable on init soundwire: intel: add pm_runtime support drivers/soundwire/cadence_master.c | 135 ++++++++++++++------ drivers/soundwire/cadence_master.h | 5 +- drivers/soundwire/intel.c | 194 +++++++++++++++++++++++++++-- 3 files changed, 289 insertions(+), 45 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: fsl_sai: Implement set_bclk_ratio
by Daniel Baluta 13 Sep '19

13 Sep '19

From: Viorel Suman <viorel.suman(a)nxp.com> 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(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 21 +++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fe126029f4e3..e896b577b1f7 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -137,6 +137,16 @@ static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, return 0; } +static int fsl_sai_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + + sai->bclk_ratio = ratio; + + return 0; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -423,8 +433,14 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, slot_width = sai->slot_width; if (!sai->is_slave_mode) { - ret = fsl_sai_set_bclk(cpu_dai, tx, - slots * slot_width * params_rate(params)); + if (sai->bclk_ratio) + ret = fsl_sai_set_bclk(cpu_dai, tx, + sai->bclk_ratio * + params_rate(params)); + else + ret = fsl_sai_set_bclk(cpu_dai, tx, + slots * slot_width * + params_rate(params)); if (ret) return ret; @@ -640,6 +656,7 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { + .set_bclk_ratio = fsl_sai_set_dai_bclk_ratio, .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .set_tdm_slot = fsl_sai_set_dai_tdm_slot, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3a3f6f8e5595..f96f8d97489d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -177,6 +177,7 @@ struct fsl_sai { unsigned int mclk_streams; unsigned int slots; unsigned int slot_width; + unsigned int bclk_ratio; const struct fsl_sai_soc_data *soc_data; struct snd_dmaengine_dai_dma_data dma_params_rx; -- 2.17.1

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[alsa-devel] [PATCH] ASoC: fsl_sai: Fix noise when using EDMA
by Daniel Baluta 09 Sep '19

09 Sep '19

From: Mihai Serban <mihai.serban(a)nxp.com> EDMA requires the period size to be multiple of maxburst. Otherwise the remaining bytes are not transferred and thus noise is produced. We can handle this issue by adding a constraint on SNDRV_PCM_HW_PARAM_PERIOD_SIZE to be multiple of tx/rx maxburst value. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 15 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 16 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 728307acab90..fe126029f4e3 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -612,6 +612,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE); + /* + * some DMA controllers need period size to be a multiple of + * tx/rx maxburst + */ + if (sai->soc_data->use_constraint_period_size) + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + tx ? sai->dma_params_tx.maxburst : + sai->dma_params_rx.maxburst); + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints); @@ -1011,30 +1021,35 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_imx_pcm = false, .fifo_depth = 32, .reg_offset = 0, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_imx_pcm = true, .fifo_depth = 32, .reg_offset = 0, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_imx_pcm = true, .fifo_depth = 16, .reg_offset = 8, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_imx_pcm = true, .fifo_depth = 128, .reg_offset = 8, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .fifo_depth = 64, .reg_offset = 0, + .use_constraint_period_size = true, }; static const struct of_device_id fsl_sai_ids[] = { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index b89b0ca26053..3a3f6f8e5595 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -157,6 +157,7 @@ struct fsl_sai_soc_data { bool use_imx_pcm; + bool use_constraint_period_size; unsigned int fifo_depth; unsigned int reg_offset; }; -- 2.17.1

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[alsa-devel] about pop chime on pcm api
by Yang, Xiaotao (X.) 09 Sep '19

09 Sep '19

Hi: Why should I call api<snd_pcm_close()> between the interval if I want to play a wav file two times repeatly, Otherwise pop chime occurs? Eg: (fake code as follows): Following codes works good. Pcm.open();//open pcm device Pcm.play(demo.wav);//set parameter and call snd_pcm_writei() to play Pcm.close();//close pcm device Pcm.open(); Pcm.play(demo.wav); Pcm.close(); But following codes works with pop chime Pcm.open(); Pcm.play(demo.wav); Pcm.play(demo.wav); Pcm.close();

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[alsa-devel] [PATCH v2 0/6] ] ASoC: atmel: extend SSC support
by Michał Mirosław 08 Sep '19

08 Sep '19

This series improves support for various configurations using SSC module as implemented in Atmel SAMA5Dx SoCs. Patches are: 1. enable SSC in Kconfig for audio-graph-card support 2. DRY mode setting code 3. implement left-justified data mode 4-6. enable shared FSYNC source for slave mode Patches against tiwai/sound/for-next tree. You can also pull from https://rere.qmqm.pl/git/linux branch: atmel-ssc Michał Mirosław (6): ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig ASoC: atmel_ssc_dai: rework DAI format configuration ASoC: atmel_ssc_dai: implement left-justified data mode dt-bindings: misc: atmel-ssc: LRCLK from TF/RF pin option misc: atmel-ssc: get LRCLK pin selection from DT ASoC: atmel_ssc_dai: Enable shared FSYNC source in frame-slave mode .../devicetree/bindings/misc/atmel-ssc.txt | 5 + drivers/misc/atmel-ssc.c | 9 + include/linux/atmel-ssc.h | 2 + sound/soc/atmel/Kconfig | 30 +- sound/soc/atmel/atmel_ssc_dai.c | 305 ++++++------------ 5 files changed, 137 insertions(+), 214 deletions(-) -- 2.20.1

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[alsa-devel] [RFC PATCH 0/5] ASoC: SOF: Intel: SoundWire initial integration
by Pierre-Louis Bossart 04 Sep '19

04 Sep '19

This RFC is the companion of the other RFC 'soundwire: intel: simplify DAI/PDI handling'. Our purpose at this point is to gather feedback on the interfaces between the Intel SOF parts and the SoundWire code. The suggested solution is a simple init/release inserted at probe/remove and resume/suspend, as well as two callbacks for the SOF driver to generate IPC configurations with the firmware. That level of separation completely hides the details of the SoundWire DAIs and will allow for 'transparent' multi-cpu DAI support, which will be handled in the machine driver and the soundwire DAIs. This solution was tested on IceLake and CometLake, and captures the feedback from SOF contributors on an initial integration that was deemed too complicated (and rightly so). Pierre-Louis Bossart (5): ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add helper to initialize SoundWire IP ASoC: SOF: Intel: hda: add SoundWire IP support ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: add support for SoundWire suspend/resume include/sound/sof/dai-intel.h | 18 ++-- sound/soc/sof/intel/hda-dsp.c | 11 +++ sound/soc/sof/intel/hda.c | 157 ++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++ 4 files changed, 188 insertions(+), 9 deletions(-) base-commit: 3b3aaa017e8072b1bfddda92be296b3463d870be -- 2.20.1

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