May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] ASoC: stm32: sai: add support of iec958 controls
by Olivier Moysan 07 Jun '18

07 Jun '18

This patchset adds support of iec958 controls to STM32 SAI driver. The patch makes use of iec958 control status helpers previously proposed and discussed through the following threads: https://patchwork.kernel.org/patch/8062601/ https://patchwork.kernel.org/patch/8091551/ (v2) https://patchwork.kernel.org/patch/8462961/ (v3) https://patchwork.kernel.org/patch/8533731/ (v4) Olivier Moysan (3): ALSA: pcm: add IEC958 channel status control helper ASoC: stm32: sai: add iec958 controls support ASoC: dmaengine_pcm: document process callback include/sound/dmaengine_pcm.h | 2 + include/sound/pcm_iec958.h | 19 +++++++ sound/core/pcm_iec958.c | 114 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 101 ++++++++++++++++++++++++++++++++----- 5 files changed, 225 insertions(+), 12 deletions(-) -- 1.9.1

4 13

[alsa-devel] [PATCH v2 1/3] ASoC: topology: Improve backwards compatibility with v4 topology files
by Guenter Roeck 06 Jun '18

06 Jun '18

From: Guenter Roeck <groeck(a)chromium.org> Commit dc31e741db49 ("ASoC: topology: ABI - Add the types for BE DAI") introduced sound topology files version 5. Initially, this change made the topology code incompatible with v4 topology files. Backwards compatibility with v4 configuration files was subsequently added with commit 288b8da7e992 ("ASoC: topology: Support topology file of ABI v4"). Unfortunately, backwards compatibility was never fully implemented. First, the manifest size in (Skylake) v4 configuration files is set to 0, which causes manifest_new_ver() to bail out with error messages similar to the following. snd_soc_skl 0000:00:1f.3: ASoC: invalid manifest size snd_soc_skl 0000:00:1f.3: tplg component load failed-22 snd_soc_skl 0000:00:1f.3: Failed to init topology! snd_soc_skl 0000:00:1f.3: ASoC: failed to probe component -22 skl_n88l25_m98357a skl_n88l25_m98357a: ASoC: failed to instantiate card -22 skl_n88l25_m98357a: probe of skl_n88l25_m98357a failed with error -22 After this problem is fixed, the following error message is seen instead. snd_soc_skl 0000:00:1f.3: ASoC: old version of manifest snd_soc_skl 0000:00:1f.3: Invalid descriptor token 1093938482 snd_soc_skl 0000:00:1f.3: ASoC: failed to load widget media0_in cpr 0 snd_soc_skl 0000:00:1f.3: tPlg component load failed-22 This message is seen because backwards compatibility for loading widgets was never implemented. The lack of audio support when running the upstream kernel on recent Chromebooks has been reported in various forums, and can be traced back to this problem. Attempts to fix the problem, usually by providing v5 configuration files, were only partially successful. Let's implement backward compatibility properly to solve the problem for good. Signed-off-by: Guenter Roeck <groeck(a)chromium.org> --- Tested on Caroline (Samsung Chromebook Pro) and Chell (HP Chromebook 13 G1) running v4.17-rc6 plus this patch, with original (v4) configuration files. Also tested on several other Chromebooks with this patch on top of chromeos-4.14. v2: - Move on from RFC/RFT to real patch - Move v4 structure definitions into header file - Add support for copying private capabilities - Declare skl_dfw_v4_module_caps as __packed - Drop duplicate assignment of mconfig->pipe->state sound/soc/intel/skylake/skl-topology.c | 169 +++++++++++++++++++ sound/soc/intel/skylake/skl-tplg-interface.h | 74 ++++++++ sound/soc/soc-topology.c | 7 +- 3 files changed, 248 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 3b1dca419883..9e4c2cb88dea 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -19,6 +19,7 @@ #include <linux/slab.h> #include <linux/types.h> #include <linux/firmware.h> +#include <linux/uuid.h> #include <sound/soc.h> #include <sound/soc-topology.h> #include <uapi/sound/snd_sst_tokens.h> @@ -2724,6 +2725,167 @@ static int skl_tplg_get_desc_blocks(struct device *dev, return -EINVAL; } +/* Functions to parse private data from configuration file format v4 */ + +/* + * Add pipeline from topology binary into driver pipeline list + * + * If already added we return that instance + * Otherwise we create a new instance and add into driver list + */ +static int skl_tplg_add_pipe_v4(struct device *dev, + struct skl_module_cfg *mconfig, struct skl *skl, + struct skl_dfw_v4_pipe *dfw_pipe) +{ + struct skl_pipeline *ppl; + struct skl_pipe *pipe; + struct skl_pipe_params *params; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == dfw_pipe->pipe_id) { + mconfig->pipe = ppl->pipe; + return 0; + } + } + + ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL); + if (!ppl) + return -ENOMEM; + + pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL); + if (!pipe) + return -ENOMEM; + + params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + + pipe->ppl_id = dfw_pipe->pipe_id; + pipe->memory_pages = dfw_pipe->memory_pages; + pipe->pipe_priority = dfw_pipe->pipe_priority; + pipe->conn_type = dfw_pipe->conn_type; + pipe->state = SKL_PIPE_INVALID; + pipe->p_params = params; + INIT_LIST_HEAD(&pipe->w_list); + + ppl->pipe = pipe; + list_add(&ppl->node, &skl->ppl_list); + + mconfig->pipe = pipe; + + return 0; +} + +static void skl_fill_module_pin_info_v4(struct skl_dfw_v4_module_pin *dfw_pin, + struct skl_module_pin *m_pin, + bool is_dynamic, int max_pin) +{ + int i; + + for (i = 0; i < max_pin; i++) { + m_pin[i].id.module_id = dfw_pin[i].module_id; + m_pin[i].id.instance_id = dfw_pin[i].instance_id; + m_pin[i].in_use = false; + m_pin[i].is_dynamic = is_dynamic; + m_pin[i].pin_state = SKL_PIN_UNBIND; + } +} + +static void skl_tplg_fill_fmt_v4(struct skl_module_pin_fmt *dst_fmt, + struct skl_dfw_v4_module_fmt *src_fmt, + int pins) +{ + int i; + + for (i = 0; i < pins; i++) { + dst_fmt[i].fmt.channels = src_fmt[i].channels; + dst_fmt[i].fmt.s_freq = src_fmt[i].freq; + dst_fmt[i].fmt.bit_depth = src_fmt[i].bit_depth; + dst_fmt[i].fmt.valid_bit_depth = src_fmt[i].valid_bit_depth; + dst_fmt[i].fmt.ch_cfg = src_fmt[i].ch_cfg; + dst_fmt[i].fmt.ch_map = src_fmt[i].ch_map; + dst_fmt[i].fmt.interleaving_style = + src_fmt[i].interleaving_style; + dst_fmt[i].fmt.sample_type = src_fmt[i].sample_type; + } +} + +static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w, + struct skl *skl, struct device *dev, + struct skl_module_cfg *mconfig) +{ + struct skl_dfw_v4_module *dfw = + (struct skl_dfw_v4_module *)tplg_w->priv.data; + int ret; + + dev_dbg(dev, "Parsing Skylake v4 widget topology data\n"); + + ret = guid_parse(dfw->uuid, (guid_t *)mconfig->guid); + if (ret) + return ret; + mconfig->id.module_id = -1; + mconfig->id.instance_id = dfw->instance_id; + mconfig->module->resources[0].cps = dfw->max_mcps; + mconfig->module->resources[0].ibs = dfw->ibs; + mconfig->module->resources[0].obs = dfw->obs; + mconfig->core_id = dfw->core_id; + mconfig->module->max_input_pins = dfw->max_in_queue; + mconfig->module->max_output_pins = dfw->max_out_queue; + mconfig->module->loadable = dfw->is_loadable; + skl_tplg_fill_fmt_v4(mconfig->module->formats[0].inputs, dfw->in_fmt, + MAX_IN_QUEUE); + skl_tplg_fill_fmt_v4(mconfig->module->formats[0].outputs, dfw->out_fmt, + MAX_OUT_QUEUE); + + mconfig->params_fixup = dfw->params_fixup; + mconfig->converter = dfw->converter; + mconfig->m_type = dfw->module_type; + mconfig->vbus_id = dfw->vbus_id; + mconfig->module->resources[0].is_pages = dfw->mem_pages; + + ret = skl_tplg_add_pipe_v4(dev, mconfig, skl, &dfw->pipe); + if (ret) + return ret; + + mconfig->dev_type = dfw->dev_type; + mconfig->hw_conn_type = dfw->hw_conn_type; + mconfig->time_slot = dfw->time_slot; + mconfig->formats_config.caps_size = dfw->caps.caps_size; + + mconfig->m_in_pin = devm_kzalloc(dev, + MAX_IN_QUEUE * sizeof(*mconfig->m_in_pin), + GFP_KERNEL); + if (!mconfig->m_in_pin) + return -ENOMEM; + + mconfig->m_out_pin = devm_kzalloc(dev, + MAX_OUT_QUEUE * sizeof(*mconfig->m_out_pin), + GFP_KERNEL); + if (!mconfig->m_out_pin) + return -ENOMEM; + + skl_fill_module_pin_info_v4(dfw->in_pin, mconfig->m_in_pin, + dfw->is_dynamic_in_pin, + mconfig->module->max_input_pins); + skl_fill_module_pin_info_v4(dfw->out_pin, mconfig->m_out_pin, + dfw->is_dynamic_out_pin, + mconfig->module->max_output_pins); + + if (mconfig->formats_config.caps_size) { + mconfig->formats_config.set_params = dfw->caps.set_params; + mconfig->formats_config.param_id = dfw->caps.param_id; + mconfig->formats_config.caps = + devm_kzalloc(dev, mconfig->formats_config.caps_size, + GFP_KERNEL); + if (!mconfig->formats_config.caps) + return -ENOMEM; + memcpy(mconfig->formats_config.caps, dfw->caps.caps, + dfw->caps.caps_size); + } + + return 0; +} + /* * Parse the private data for the token and corresponding value. * The private data can have multiple data blocks. So, a data block @@ -2739,6 +2901,13 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, char *data; int ret; + /* + * v4 configuration files have a valid UUID at the start of + * the widget's private data. + */ + if (uuid_is_valid((char *)tplg_w->priv.data)) + return skl_tplg_get_pvt_data_v4(tplg_w, skl, dev, mconfig); + /* Read the NUM_DATA_BLOCKS descriptor */ array = (struct snd_soc_tplg_vendor_array *)tplg_w->priv.data; ret = skl_tplg_get_desc_blocks(dev, array); diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h index f8d1749a2e0c..b0e3d376594c 100644 --- a/sound/soc/intel/skylake/skl-tplg-interface.h +++ b/sound/soc/intel/skylake/skl-tplg-interface.h @@ -169,4 +169,78 @@ enum skl_tuple_type { SKL_TYPE_DATA }; +/* v4 configuration data */ + +struct skl_dfw_v4_module_pin { + u16 module_id; + u16 instance_id; +} __packed; + +struct skl_dfw_v4_module_fmt { + u32 channels; + u32 freq; + u32 bit_depth; + u32 valid_bit_depth; + u32 ch_cfg; + u32 interleaving_style; + u32 sample_type; + u32 ch_map; +} __packed; + +struct skl_dfw_v4_module_caps { + u32 set_params:2; + u32 rsvd:30; + u32 param_id; + u32 caps_size; + u32 caps[HDA_SST_CFG_MAX]; +} __packed; + +struct skl_dfw_v4_pipe { + u8 pipe_id; + u8 pipe_priority; + u16 conn_type:4; + u16 rsvd:4; + u16 memory_pages:8; +} __packed; + +struct skl_dfw_v4_module { + char uuid[SKL_UUID_STR_SZ]; + + u16 module_id; + u16 instance_id; + u32 max_mcps; + u32 mem_pages; + u32 obs; + u32 ibs; + u32 vbus_id; + + u32 max_in_queue:8; + u32 max_out_queue:8; + u32 time_slot:8; + u32 core_id:4; + u32 rsvd1:4; + + u32 module_type:8; + u32 conn_type:4; + u32 dev_type:4; + u32 hw_conn_type:4; + u32 rsvd2:12; + + u32 params_fixup:8; + u32 converter:8; + u32 input_pin_type:1; + u32 output_pin_type:1; + u32 is_dynamic_in_pin:1; + u32 is_dynamic_out_pin:1; + u32 is_loadable:1; + u32 rsvd3:11; + + struct skl_dfw_v4_pipe pipe; + struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE]; + struct skl_dfw_v4_module_fmt out_fmt[MAX_OUT_QUEUE]; + struct skl_dfw_v4_module_pin in_pin[MAX_IN_QUEUE]; + struct skl_dfw_v4_module_pin out_pin[MAX_OUT_QUEUE]; + struct skl_dfw_v4_module_caps caps; +} __packed; + #endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 986b8b2f90fb..d66b2e5ccd67 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2293,8 +2293,11 @@ static int manifest_new_ver(struct soc_tplg *tplg, *manifest = NULL; if (src->size != sizeof(*src_v4)) { - dev_err(tplg->dev, "ASoC: invalid manifest size\n"); - return -EINVAL; + dev_warn(tplg->dev, "ASoC: invalid manifest size %d\n", + src->size); + if (src->size) + return -EINVAL; + src->size = sizeof(*src_v4); } dev_warn(tplg->dev, "ASoC: old version of manifest\n"); -- 2.17.0.441.gb46fe60e1d-goog

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[alsa-devel] [PATCH] ASoC: ssm2602: Fix ADC powerup sequencing
by Marco Felsch 05 Jun '18

05 Jun '18

From: Philipp Zabel <p.zabel(a)pengutronix.de> According to the ssm2603 data sheet (control register sequencing), the digital core should be activated only after all necessary bits in the power register are enabled, and a delay determined by the decoupling capacitor on the VMID pin has passed. If the digital core is activated too early, or even before the ADC is powered up, audible artifacts appear at the beginning of the recorded signal. The digital core is also needed for playback, so when recording starts it may already be enabled. This means we cannot get the power sequence correct when we want to be able to start recording after playback. As a workaround put the MIC mute switch into the DAPM routes. This way we can keep the recording disabled until the MIC Bias has settled and thus get rid of audible artifacts. The delay we have to wait depends on a board specific capacitor connected to the VMID pins, so make the delay configurable in the device tree. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- .../devicetree/bindings/sound/adi,ssm2602.txt | 7 +++++ sound/soc/codecs/ssm2602.c | 30 +++++++++++++++++-- 2 files changed, 35 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2602.txt b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt index 3b3302fe399b..9334d48c0462 100644 --- a/Documentation/devicetree/bindings/sound/adi,ssm2602.txt +++ b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt @@ -11,9 +11,16 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + + - startup-delay-us : delay between powering on and activating the digital + core, determined by the decoupling capacitor C on the + VMID pin: delay [µs] = C [µF] * 25000 / 3.5 + Example: ssm2602: ssm2602@1a { compatible = "adi,ssm2602"; reg = <0x1a>; + startup-delay-us = <34000>; /* 4.7 µF * 25000 / 3.5 -> ~34 ms */ }; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 501a4e73b185..1b99f5d44f8d 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -26,9 +26,11 @@ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA */ +#include <linux/delay.h> #include <linux/module.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/of.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -46,6 +48,7 @@ struct ssm2602_priv { enum ssm2602_type type; unsigned int clk_out_pwr; + u32 startup_delay_us; }; /* @@ -111,7 +114,6 @@ SOC_SINGLE_TLV("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1, SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0), -SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), }; /* Output Mixer */ @@ -121,10 +123,26 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0), SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), }; +static const struct snd_kcontrol_new mic_ctl = + SOC_DAPM_SINGLE("Switch", SSM2602_APANA, 1, 1, 1); + /* Input mux */ static const struct snd_kcontrol_new ssm2602_input_mux_controls = SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]); +static int ssm2602_mic_switch_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); + + if (ssm2602->startup_delay_us) + usleep_range(ssm2602->startup_delay_us, + ssm2602->startup_delay_us * 2); + + return 0; +} + static const struct snd_soc_dapm_widget ssm260x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), @@ -146,6 +164,9 @@ SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls), SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1), +SND_SOC_DAPM_SWITCH_E("Mic Switch", SSM2602_APANA, 1, 1, &mic_ctl, + ssm2602_mic_switch_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_OUTPUT("LHPOUT"), SND_SOC_DAPM_OUTPUT("RHPOUT"), SND_SOC_DAPM_INPUT("MICIN"), @@ -178,9 +199,11 @@ static const struct snd_soc_dapm_route ssm2602_routes[] = { {"LHPOUT", NULL, "Output Mixer"}, {"Input Mux", "Line", "Line Input"}, - {"Input Mux", "Mic", "Mic Bias"}, + {"Input Mux", "Mic", "Mic Switch"}, {"ADC", NULL, "Input Mux"}, + {"Mic Switch", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "MICIN"}, }; @@ -645,6 +668,9 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type, if (ssm2602 == NULL) return -ENOMEM; + of_property_read_u32(dev->of_node, "startup-delay-us", + &ssm2602->startup_delay_us); + dev_set_drvdata(dev, ssm2602); ssm2602->type = type; ssm2602->regmap = regmap; -- 2.17.0

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[alsa-devel] [PATCH] [PATCH v3] ASoC: Intel: kbl: Move codec sysclk config to codec_init function
by mac.chiang＠intel.com 04 Jun '18

04 Jun '18

From: Mac Chiang <mac.chiang(a)intel.com> On APL, commit fd0f237572ad ("ASoC: Intel: bxt: Move codec sysclk config to codec_init function") fixed an issue related to jack detection. The MCLK for DA7219 does not change in this platform, but is currently being configured everytime as part of the platform_clock event handler for DAPM. The upshot of this is that we have unnecessary calls to this function, and it also means that if a stream hasn't yet been started, DA7219 driver does not have the correct MCLK rates programmed and so the HP detection feature does not operate as expected. The same fix is needed on KBL. This patch rectifies this issue by moving the sysclk call to codec_init function so it's only called once at initialisation. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Notes(changelog): v2: - correct the MCLK to 24.576M v3: - described the comments clearly --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index c332a32..ad3029d 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -66,14 +66,6 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, - SND_SOC_CLOCK_IN); - if (ret) { - dev_err(card->dev, "can't set codec sysclk configuration\n"); - return ret; - } - if (SND_SOC_DAPM_EVENT_OFF(event)) { ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); @@ -170,9 +162,18 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_jack *jack; int ret; + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + /* * Headset buttons map to the google Reference headset. * These can be configured by userspace. -- 2.7.4

3 4

[alsa-devel] Remove clkdiv and pll setters from pxa dais
by Daniel Mack 02 Jun '18

02 Jun '18

Hi, As discussed in the simple-card thread the other day, the .set_clkdiv and .set_pll callbacks for DAIs are considered legacy. In order to support PXA platforms with DT, we need get rid of those for the DAIs of PXA hardware. On the good side, pxa-ssp is the only problematic one of the PXA dais, the ac97, i2s don't have any special clocking knbos, and the mmp-sspa should be straight forward to fix. I looked into pxa-ssp and it seems the conversion to calculate the audio clock dividers automatically seems doable. There are currently 4 platforms that use this cpu dai in mainline that also use snd_soc_dai_set_pll() or snd_soc_dai_set_clkdiv(): * Zylonite * Brownstone * Magician * Raumfeld I can test modifications on the latter one, but for the others, I need to understand what the machine drivers are doing, and there are some bits I don't grok. For instance, code for Zylonite does this: /* Add 1 to the width for the leading clock cycle */ pll_out = rate * (width + 1) * 8; The commit which introduced these lines dates back to 2009 and was done by you, Mark. Can you remember what the reason for this was? I've never seen sample frames with 17 bits :) This is a setup that we can't generically describe through .hw_params() or .dai_fmt() in the cpu dai, correct? In the magician code, you'll find this: case 22050: acps = 5622000; switch (width) { case 16: /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } break; case 44100: acps = 5622000; switch (width) { case 16: /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } break; So for both 22050 and 44100, the base frequency and all dividers are the same, which can't be right. I assume these rates have never been used. I'll ignore this and implement the table in the datasheet which should do the right thing. Philipp? What we need, however, is a way to describe whether the dai is mclk master or slave. Would we add a DT propery for that? Thanks, Daniel

3 6

[alsa-devel] [PATCH v2] ASoC: simple-card: set cpu dai clk in hw_params
by Daniel Mack 01 Jun '18

01 Jun '18

The simple-card driver currently accepts a clock node in the cpu dai sub-node and only uses it as an alternative to the 'system-clock-frequency' property to get the current frequency. This patch adds another use of the passed clock node. If mclk-fs is specified, the clocks in cpu and codec dai sub-nodes will be set to the calculated rate (stream rate * mclk_fs) in hw_params. This allows platforms to pass tuneable clocks as phandle that will automatically be set to the right rates. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- This is the only patch that's left from the series of three that I posted earlier. v1 → v2: * Apply the setting to both the codec and the cpu dai * Check for errors from clk_set_rate() and avoid unnecessary rate changes. .../devicetree/bindings/sound/simple-card.txt | 5 +++++ sound/soc/generic/simple-card.c | 21 +++++++++++++++++++++ 2 files changed, 26 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 17c13e74667d..a4c72d09cd45 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -86,6 +86,11 @@ Optional CPU/CODEC subnodes properties: in dai startup() and disabled with clk_disable_unprepare() in dai shutdown(). + If a clock is specified and a + multiplication factor is given with + mclk-fs, the clock will be set to the + calculated mclk frequency when the + stream starts. - system-clock-direction-out : specifies clock direction as 'out' on initialization. It is useful for some aCPUs with fixed clocks. diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6959a74a6f49..4a516c428b3d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -135,6 +135,18 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) asoc_simple_card_clk_disable(&dai_props->codec_dai); } +static int asoc_simple_set_clk_rate(struct asoc_simple_dai *simple_dai, + unsigned long rate) +{ + if (!simple_dai->clk) + return 0; + + if (clk_get_rate(simple_dai->clk) == rate) + return 0; + + return clk_set_rate(simple_dai->clk, rate); +} + static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -154,6 +166,15 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, if (mclk_fs) { mclk = params_rate(params) * mclk_fs; + + ret = asoc_simple_set_clk_rate(&dai_props->codec_dai, mclk); + if (ret < 0) + return ret; + + ret = asoc_simple_set_clk_rate(&dai_props->cpu_dai, mclk); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) -- 2.14.3

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[alsa-devel] [PATCH] ASoC: Intel: kbl: Move codec sysclk config to codec_init function
by mac.chiang＠intel.com 01 Jun '18

01 Jun '18

From: Mac Chiang <mac.chiang(a)intel.com> unexpected error: PLL input clock 0 below valid range backport from broonie's tree commit fd0f237572ad ("ASoC: Intel: bxt: Move codec sysclk config to codec_init function") The MCLK for DA7219 does not change in this platform, but is currently being configured everytime as part of the platform_clock event handler for DAPM. The upshot of this is that we have unnecessary calls to this function, and it also means that if a stream hasn't yet been started, DA7219 driver does not have the correct MCLK rates programmed and so the HP detection feature does not operate as expected. This patch rectifies this issue by moving the sysclk call to codec_init function so it's only called once at initialisation. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Acked-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Singed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 60e739f..88fbc6a 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -65,14 +65,6 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, - SND_SOC_CLOCK_IN); - if (ret) { - dev_err(card->dev, "can't set codec sysclk configuration\n"); - return ret; - } - if (SND_SOC_DAPM_EVENT_OFF(event)) { ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); @@ -169,9 +161,18 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_jack *jack; int ret; + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 19200000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + /* * Headset buttons map to the google Reference headset. * These can be configured by userspace. -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream()
by Bo Chen 01 Jun '18

01 Jun '18

When 'kzalloc()' fails in 'snd_hda_attach_pcm_stream()', a new pcm instance is created without setting its operators via 'snd_pcm_set_ops()'. Following operations on the new pcm instance can trigger kernel null pointer dereferences and cause kernel oops. This bug was found with my work on building a gray-box fault-injection tool for linux-kernel-module binaries. A kernel null pointer dereference was confirmed from line 'substream->ops->open()' in function 'snd_pcm_open_substream()' in file 'sound/core/pcm_native.c'. This patch fixes the bug by calling 'snd_device_free()' in the error handling path of 'kzalloc()', which removes the new pcm instance from the snd card before returns with an error code. Signed-off-by: Bo Chen <chenbo(a)pdx.edu> --- sound/pci/hda/hda_controller.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index d1eb14842340..a12e594d4e3b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -748,8 +748,10 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, return err; strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); - if (apcm == NULL) + if (apcm == NULL) { + snd_device_free(chip->card, pcm); return -ENOMEM; + } apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; -- 2.17.0

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[alsa-devel] [PATCH v2 0/2] slimbus: Add QCOM SLIMBus NGD driver
by Srinivas Kandagatla 01 Jun '18

01 Jun '18

This patchset adds support to basic version of Qualcomm NGD SLIMBus controller driver found SoCs from B family. This controller is light-weight SLIMBus controller driver responsible for communicating with slave HW directly over the bus using messaging interface, and communicating with master component residing on ADSP for bandwidth and data-channel management. Tested this patchset on DB820c with WCD9335 codec. I have pushed my working branch to [1] incase someone want to try. This patch has dependency on https://lkml.org/lkml/2018/5/17/251 Changes sinc v1 (https://lwn.net/Articles/754679): - Fixed DT binding making NGD slave of the slim controller. - Fixed few typos spotted by Vinod. - Removed unnessary dmaengine_slave_config suggested by Vinod - Removed few unnessary initializations. - fixed commit log to remove reference to B family suggested by Trilok Thanks, srini [1] https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimbus-… Srinivas Kandagatla (2): slimbus: ngd: dt-bindings: Add slim ngd dt bindings slimbus: ngd: Add qcom SLIMBus NGD driver .../bindings/slimbus/slim-ngd-qcom-ctrl.txt | 84 ++ drivers/slimbus/Kconfig | 11 + drivers/slimbus/Makefile | 3 + drivers/slimbus/qcom-ngd-ctrl.c | 1342 ++++++++++++++++++++ drivers/slimbus/slimbus.h | 8 + 5 files changed, 1448 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt create mode 100644 drivers/slimbus/qcom-ngd-ctrl.c -- 2.16.2

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[alsa-devel] [PATCH v2 16/21] ASoC: max98088: use match_string() helper
by Yisheng Xie 01 Jun '18

01 Jun '18

match_string() returns the index of an array for a matching string, which can be used instead of open coded variant. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- v2: - split ret=xxx and move assignment to the line before 'if' - fix error return value - both per Andy sound/soc/codecs/max98088.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 865f64c..fb515aa 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1382,15 +1382,12 @@ static int max98088_set_bias_level(struct snd_soc_component *component, static int max98088_get_channel(struct snd_soc_component *component, const char *name) { - int i; + int ret; - for (i = 0; i < ARRAY_SIZE(eq_mode_name); i++) - if (strcmp(name, eq_mode_name[i]) == 0) - return i; - - /* Shouldn't happen */ - dev_err(component->dev, "Bad EQ channel name '%s'\n", name); - return -EINVAL; + ret = match_string(eq_mode_name, ARRAY_SIZE(eq_mode_name), name); + if (ret < 0) + dev_err(component->dev, "Bad EQ channel name '%s'\n", name); + return ret; } static void max98088_setup_eq1(struct snd_soc_component *component) -- 1.7.12.4

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