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

[alsa-devel] [PATCH 3/3 v3] ASoC: TSCS42xx: Add CCF support to get sysclk
by Steven Eckhoff 13 Apr '18

13 Apr '18

The TSCS42xx relies on set_sysclk to get a unique clock id and rate, which prevents it from being used with the simple-card. Remove set_sysclk callback Add CCF support to get clock id and rate Add clocks and clock-names to device tree binding Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- .../devicetree/bindings/sound/tscs42xx.txt | 6 ++ sound/soc/codecs/tscs42xx.c | 104 ++++++++++++++------- sound/soc/codecs/tscs42xx.h | 2 +- 3 files changed, 75 insertions(+), 37 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt index 2ac2f0996697..7eea32e9d078 100644 --- a/Documentation/devicetree/bindings/sound/tscs42xx.txt +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -8,9 +8,15 @@ Required Properties: - reg : <0x71> for analog mic <0x69> for digital mic + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + Example: wookie: codec@69 { compatible = "tempo,tscs42A2"; reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; }; diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d41852a37ac6..0aaa6a826096 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/delay.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -40,6 +41,9 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; + + struct clk *sysclk; + int sysclk_src_id; }; struct coeff_ram_ctl { @@ -1251,13 +1255,46 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return 0; } -static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) +static const struct snd_soc_dai_ops tscs42xx_dai_ops = { + .hw_params = tscs42xx_hw_params, + .mute_stream = tscs42xx_mute_stream, + .set_fmt = tscs42xx_set_dai_fmt, + .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, +}; + +static int part_is_valid(struct tscs42xx *tscs42xx) { - struct snd_soc_component *component = codec_dai->component; + int val; int ret; + unsigned int reg; + + ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); + if (ret < 0) + return ret; - switch (clk_id) { + val = reg << 8; + ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); + if (ret < 0) + return ret; + + val |= reg; + + switch (val) { + case 0x4A74: + case 0x4A73: + return true; + default: + return false; + }; +} + +static int set_sysclk(struct snd_soc_component *component) +{ + struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); + unsigned long freq; + int ret; + + switch (tscs42xx->sysclk_src_id) { case TSCS42XX_PLL_SRC_XTAL: case TSCS42XX_PLL_SRC_MCLK1: ret = snd_soc_component_write(component, R_PLLREFSEL, @@ -1285,6 +1322,7 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } + freq = clk_get_rate(tscs42xx->sysclk); ret = set_pll_ctl_from_input_freq(component, freq); if (ret < 0) { dev_err(component->dev, @@ -1295,41 +1333,13 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static const struct snd_soc_dai_ops tscs42xx_dai_ops = { - .hw_params = tscs42xx_hw_params, - .mute_stream = tscs42xx_mute_stream, - .set_fmt = tscs42xx_set_dai_fmt, - .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, - .set_sysclk = tscs42xx_set_dai_sysclk, -}; - -static int part_is_valid(struct tscs42xx *tscs42xx) +static int tscs42xx_probe(struct snd_soc_component *component) { - int val; - int ret; - unsigned int reg; - - ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); - if (ret < 0) - return ret; - - val = reg << 8; - ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); - if (ret < 0) - return ret; - - val |= reg; - - switch (val) { - case 0x4A74: - case 0x4A73: - return true; - default: - return false; - }; + return set_sysclk(component); } static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { + .probe = tscs42xx_probe, .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1387,11 +1397,15 @@ static const struct reg_sequence tscs42xx_patch[] = { { R_AIC2, RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED }, }; +static char const * const src_names[TSCS42XX_PLL_SRC_CNT] = { + "xtal", "mclk1", "mclk2"}; + static int tscs42xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct tscs42xx *tscs42xx; - int ret = 0; + int src; + int ret; tscs42xx = devm_kzalloc(&i2c->dev, sizeof(*tscs42xx), GFP_KERNEL); if (!tscs42xx) { @@ -1402,6 +1416,24 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, tscs42xx); + for (src = TSCS42XX_PLL_SRC_XTAL; src < TSCS42XX_PLL_SRC_CNT; src++) { + tscs42xx->sysclk = devm_clk_get(&i2c->dev, src_names[src]); + if (!IS_ERR(tscs42xx->sysclk)) { + break; + } else if (PTR_ERR(tscs42xx->sysclk) != -ENOENT) { + ret = PTR_ERR(tscs42xx->sysclk); + dev_err(&i2c->dev, "Failed to get sysclk (%d)\n", ret); + return ret; + } + } + if (src == TSCS42XX_PLL_SRC_CNT) { + ret = -EINVAL; + dev_err(&i2c->dev, "Failed to get a valid clock name (%d)\n", + ret); + return ret; + } + tscs42xx->sysclk_src_id = src; + tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index d4a30bcbf64b..814c8f3c4a68 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -7,10 +7,10 @@ #define __WOOKIE_H__ enum { - TSCS42XX_PLL_SRC_NONE, TSCS42XX_PLL_SRC_XTAL, TSCS42XX_PLL_SRC_MCLK1, TSCS42XX_PLL_SRC_MCLK2, + TSCS42XX_PLL_SRC_CNT, }; #define R_HPVOLL 0x0 -- 2.15.1

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[alsa-devel] [PATCH] ALSA: hda - Silence PM ops build warning
by Lukas Wunner 13 Apr '18

13 Apr '18

The system sleep PM ops azx_suspend() and azx_resume() were previously called by vga_switcheroo, but commit 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") removed their invocation. Unfortunately the commit neglected to update the #ifdef surrounding the two functions, so if CONFIG_PM_SLEEP is *not* enabled but all three of CONFIG_PM, CONFIG_VGA_SWITCHEROO and CONFIG_SND_HDA_CODEC_HDMI *are* enabled, the compiler now emits the following warning: sound/pci/hda/hda_intel.c:1024:12: warning: 'azx_resume' defined but not used [-Wunused-function] static int azx_resume(struct device *dev) ^~~~~~~~~~ sound/pci/hda/hda_intel.c:989:12: warning: 'azx_suspend' defined but not used [-Wunused-function] static int azx_suspend(struct device *dev) ^~~~~~~~~~~ Silence by updating the #ifdef. Because the #ifdef block now uses the same condition as the one immediately succeeding it, the two blocks can be collapsed together, shaving off another two lines. Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Cc: Takashi Iwai <tiwai(a)suse.com> Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Lukas Wunner <lukas(a)wunner.de> --- sound/pci/hda/hda_intel.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec4e6b829ee2..d653c56e9141 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -982,7 +982,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -#if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1063,9 +1063,7 @@ static int azx_resume(struct device *dev) trace_azx_resume(chip); return 0; } -#endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ -#ifdef CONFIG_PM_SLEEP /* put codec down to D3 at hibernation for Intel SKL+; * otherwise BIOS may still access the codec and screw up the driver */ -- 2.16.2

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[alsa-devel] [PATCH] Documentation: sound: Corrected my_chip port element
by Christina Quast 12 Apr '18

12 Apr '18

Hi all! A small correction in the documentation. C.

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[alsa-devel] [PATCH 0/5] for-linux/sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 12 Apr '18

12 Apr '18

Hello, all! This is the syncup version of the sound protocol changes for Linux Kernel with the only difference from the corresponding Xen version being spaces to tabs conversion. Regradless of this only change I have dropped R-b tags received for Xen version. In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - bump protocol version to 2 - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events - add XENSND_OP_HW_PARAM_QUERY request to read/update stream configuration space: request passes desired intervals/formats for the stream parameters and the response returns allowed intervals and formats mask that can be used. - MAJOR: changed req/resp/evt packet sizes from 32 to 64 octets - Reworked XENSND_OP_HW_PARAM_QUERY so it now sends all parameters at once, allowing to check all the configuration space. - Minor documentation cleanup (added missed "reserved" fields) Oleksandr Andrushchenko (5): xen/sndif: Introduce protocol version xen/sndif: Fix missed "reserved" fields in comments xen/sndif: Make requests and responses 64 octets long xen/sndif: Add explicit back and front synchronization xen/sndif: Add explicit back and front parameter negotiation include/xen/interface/io/sndif.h | 322 +++++++++++++++++++++++++++++-- 1 file changed, 306 insertions(+), 16 deletions(-) -- 2.17.0

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Re: [alsa-devel] [PATCH 08/15] ASoC: pxa: remove the dmaengine compat need
by Mark Brown 12 Apr '18

12 Apr '18

On Mon, Apr 02, 2018 at 04:26:49PM +0200, Robert Jarzmik wrote: > As the pxa architecture switched towards the dmaengine slave map, the > old compatibility mechanism to acquire the dma requestor line number and > priority are not needed anymore. Acked-by: Mark Brown <broonie(a)kernel.org> If there's no dependency I'm happy to take this for 4.18.

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[alsa-devel] [PATCH] ASoC: topology: fix some tiny memory leaks
by Dan Carpenter 12 Apr '18

12 Apr '18

These tiny memory leaks don't have a huge real life impact but they cause static checker warnings so let's fix them. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index fa27d0fca6dc..942c6e5eb4b7 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1325,8 +1325,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( ec->hdr.name); kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) + if (kc[i].name == NULL) { + kfree(se); goto err_se; + } kc[i].private_value = (long)se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = ec->hdr.access; @@ -1442,8 +1444,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( be->hdr.name, be->hdr.access); kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) + if (kc[i].name == NULL) { + kfree(sbe); goto err; + } kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = be->hdr.access;

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[alsa-devel] Missing alsa-utils 1.1.6 tag
by Jordi Mallach 12 Apr '18

12 Apr '18

Hello Jaroslav, Can you push a tag for alsa-utils 1.1.6? It's missing from git AFAICT. Thank you, Jordi -- Jordi Mallach Pérez -- Debian developer https://www.debian.org/ jordi(a)sindominio.net jordi(a)debian.org https://www.sindominio.net/ GnuPG public key information available at: https://oskuro.net/

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[alsa-devel] snd_seq_query_port_subscribers not updating subs index
by Charles Flèche 12 Apr '18

12 Apr '18

Hi all, I'm iterating through subscribers with snd_seq_query_port_subscribers to disconnect them. It works most of the time, but when a2j (an alsa -> jack MIDI bridge) runs, snd_seq_query_port_subscribers does not update the subs index, and the next call returns 0, leading to an infinite loop. I have to manually increase the index when the disconnection fails (meaning that I hit an a2j connection). while (snd_seq_query_port_subscribers(seq, subs) >= 0) { addr = *snd_seq_query_subscribe_get_addr(subs); if (snd_seq_disconnect_to(seq, port, addr.client, addr.port) < 0) { snd_seq_query_subscribe_set_index(subs, snd_seq_query_subscribe_get_index(subs) + 1); } } 1. Is the non-updated subs index an expected behavior ? 2. Is my fix a correct one or could it leads to other issues ? Many thanks,

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[alsa-devel] [PATCH] ASoC: fsl_esai: Fix divisor calculation failure at lower ratio
by Nicolin Chen 12 Apr '18

12 Apr '18

When the desired ratio is less than 256, the savesub (tolerance) in the calculation would become 0. This will then fail the loop- search immediately without reporting any errors. But if the ratio is smaller enough, there is no need to calculate the tolerance because PM divisor alone is enough to get the ratio. So a simple fix could be just to set PM directly instead of going into the loop-search. Reported-by: Marek Vasut <marex(a)denx.de> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> --- sound/soc/fsl/fsl_esai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 40a7004..da8fd98 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -144,6 +144,13 @@ static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio, psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + /* Do not loop-search if PM (1 ~ 256) alone can serve the ratio */ + if (ratio <= 256) { + pm = ratio; + fp = 1; + goto out; + } + /* Set the max fluctuation -- 0.1% of the max devisor */ savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix mode setting when changing channel number
by Nicolin Chen 12 Apr '18

12 Apr '18

This is a partial revert (in a cleaner way) of commit ebf08ae3bc90 ("ASoC: fsl_ssi: Keep ssi->i2s_net updated") to fix a regression at test cases when switching between mono and stereo audio. The problem is that ssi->i2s_net is initialized in set_dai_fmt() only, while this set_dai_fmt() is only called during the dai-link probe(). The original patch assumed set_dai_fmt() would be called during every playback instance, so it failed at the overriding use cases. This patch adds the local variable i2s_net back to let regular use cases still follow the mode settings from the set_dai_fmt(). Meanwhile, the original commit of keeping ssi->i2s_net updated was to make set_tdm_slot() clean by checking the ssi->i2s_net directly instead of reading SCR register. However, the change itself is not necessary (or even harmful) because the set_tdm_slot() might fail to check the slot number for Normal-Mode-None-Net settings while mono audio cases still need 2 slots. So this patch can also fix it. And it adds an extra line of comments to declare ssi->i2s_net does not reflect the register value but merely the initial setting from the set_dai_fmt(). Reported-by: Mika Penttilä <mika.penttila(a)nextfour.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Mika Penttilä <mika.penttila(a)nextfour.com> --- sound/soc/fsl/fsl_ssi.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0823b08..89df2d9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -217,6 +217,7 @@ struct fsl_ssi_soc_data { * @dai_fmt: DAI configuration this device is currently used with * @streams: Mask of current active streams: BIT(TX) and BIT(RX) * @i2s_net: I2S and Network mode configurations of SCR register + * (this is the initial settings based on the DAI format) * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for dual FIFO mode @@ -829,16 +830,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } if (!fsl_ssi_is_ac97(ssi)) { + /* + * Keep the ssi->i2s_net intact while having a local variable + * to override settings for special use cases. Otherwise, the + * ssi->i2s_net will lose the settings for regular use cases. + */ + u8 i2s_net = ssi->i2s_net; + /* Normal + Network mode to send 16-bit data in 32-bit frames */ if (fsl_ssi_is_i2s_cbm_cfs(ssi) && sample_size == 16) - ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; + i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; /* Use Normal mode to send mono data at 1st slot of 2 slots */ if (channels == 1) - ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL; + i2s_net = SSI_SCR_I2S_MODE_NORMAL; regmap_update_bits(regs, REG_SSI_SCR, - SSI_SCR_I2S_NET_MASK, ssi->i2s_net); + SSI_SCR_I2S_NET_MASK, i2s_net); } /* In synchronous mode, the SSI uses STCCR for capture */ -- 2.7.4

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