February 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] ASoC: pcm512x: Add knobs to allow and control overclocking
by Peter Rosin 24 Feb '15

24 Feb '15

From: Peter Rosin <peda(a)axentia.se> Hi! I wasn't sure if I should add Documentation/* for these sysfs knobs or not? A lot of knobs do not seem to have docs (no specific example, just a gut feeling). And I'm not sure how I should name the doc-file since the pcm512x driver handles devices connected with both i2c and spi. And what KernelVersion should I enter? That depends... So, this isn't perfect, suggestions welcome. The first patch is a preparatory patch that makes 2/2 slightly simpler. I can fold them together if that's desired. Or I could split 2/2 up in three logical patches, one for each of the independent PLL/DSP/DAC knobs. Another "feature" that one might want, is that attempts to change the overclocking should fail with EBUSY when the device is already in use. But I haven't invested enough time to find out how I should determine when to fail, so for this version the driver accepts the change but doesn't enforce the new limit until the clock rates are recalculated. Cheers, Peter Peter Rosin (2): ASoC: pcm512x: Rearrange to not repeat dacsrc_rate / dac_div ASoC: pcm512x: Allow independently overclocking PLL, DAC and DSP Documentation/ABI/testing/sysfs-i2c-pcm512x | 35 +++++ sound/soc/codecs/pcm512x.c | 189 ++++++++++++++++++++++++--- 2 files changed, 205 insertions(+), 19 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-i2c-pcm512x -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: hda: controller code - do not export static functions
by Jaroslav Kysela 24 Feb '15

24 Feb '15

It is a bad idea to export static functions. GCC for some platforms shows errors like: error: __ksymtab_azx_get_response causes a section type conflict Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/hda_controller.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ebb7a64..f63d2cb5 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -978,7 +978,6 @@ static int azx_alloc_cmd_io(struct azx *chip) dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n"); return err; } -EXPORT_SYMBOL_GPL(azx_alloc_cmd_io); static void azx_init_cmd_io(struct azx *chip) { @@ -1043,7 +1042,6 @@ static void azx_init_cmd_io(struct azx *chip) azx_writeb(chip, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN); spin_unlock_irq(&chip->reg_lock); } -EXPORT_SYMBOL_GPL(azx_init_cmd_io); static void azx_free_cmd_io(struct azx *chip) { @@ -1053,7 +1051,6 @@ static void azx_free_cmd_io(struct azx *chip) azx_writeb(chip, CORBCTL, 0); spin_unlock_irq(&chip->reg_lock); } -EXPORT_SYMBOL_GPL(azx_free_cmd_io); static unsigned int azx_command_addr(u32 cmd) { @@ -1333,7 +1330,6 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val) else return azx_corb_send_cmd(bus, val); } -EXPORT_SYMBOL_GPL(azx_send_cmd); /* get a response */ static unsigned int azx_get_response(struct hda_bus *bus, @@ -1347,7 +1343,6 @@ static unsigned int azx_get_response(struct hda_bus *bus, else return azx_rirb_get_response(bus, addr); } -EXPORT_SYMBOL_GPL(azx_get_response); #ifdef CONFIG_SND_HDA_DSP_LOADER /* -- 1.9.3

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation
by Nicolin Chen 24 Feb '15

24 Feb '15

According to i.MX6 Series Reference Manual, the formula to calculate the sys clock is sysclk rate = bclk rate * (div2 + 1) * (7 * psr + 1) * (pm + 1) * 2 Commit aafa85e71a75 ("ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series") added the divisor calculation which relies on the clk_round_rate(). However, at that time, clk_round_rate() didn't provide closest clock rates for some cases because it might not use a correct rounding policy. So using the original formula (pm + 1) for PM divisor was not able to give us a desired clock rate. And then we used (pm + 2) to do the trick. However, the clk-divider driver has been refined a lot since commit b11d282dbea2 ("clk: divider: fix rate calculation for fractional rates") Now using (pm + 2) trick would result an incorrect clock rate. So this patch fixes the problem by removing the useless trick. Reported-by: Stephane Cerveau <scerveau(a)voxtok.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- [ I also checked that only 3.18 stable tree has the refinements for clk-divider driver, so I guess it might not be able to apply to previous long-term trees if being treated as bug fix. -- Nicolin ] sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d7365c5..3e7c644 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,7 +603,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + tmprate = freq * factor * (i + 1); if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: sta32x: fix register range in regmap.
by Thomas Niederprüm 24 Feb '15

24 Feb '15

The STA32X_AUTO3 is a writable register that currently does not appear in the regmap ranges(neither read nor write). By adding this register to the register ranges there is no gap anymore and the existing register ranges can be joined. This fixes a regression introduced in commit a1be4cead9b9504aa6fc93b624975601cec8c188 where the driver was moved to direct regmap usage and the STA32X_AUTO3 register was missed. That made it impossible to choose the preset EQ mode set through the STA32X_AUTO3 register. Signed-off-by: Thomas Niederprüm <niederp(a)physik.uni-kl.de> --- changes since v1: - updated to apply to linux-next. - added commit id that introduced the regression in the commit message. sound/soc/codecs/sta32x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index e28c185..fc686ca 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = { }; static const struct regmap_range sta32x_write_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_read_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_volatile_regs_range[] = { -- 2.1.1

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[alsa-devel] [PATCH] ASoC: improve usage of gpiod API
by Uwe Kleine-König 24 Feb '15

24 Feb '15

Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Simplify drivers accordingly. Also there is an *_optional variant that serves well here. The sematics is slightly changed here by using it as error checking is more strict now: If GPIOLIB is not enabled an error is returned instead of just ignoring the gpio. On one hand this is bad for devices that don't "have" the respective gpio as the driver is failing now. On the other hand there is no means to assert that this gpio is really not needed or if only the driver to control it is not available. The latter is a real reason to fail and so it's defensive to fail here, too. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/adau1977.c | 17 +++++------------ sound/soc/codecs/cs35l32.c | 19 ++++++------------- sound/soc/codecs/cs4265.c | 19 ++++++------------- sound/soc/codecs/sta350.c | 30 +++++++++--------------------- sound/soc/codecs/tas2552.c | 13 +++---------- 5 files changed, 29 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 70ab35744aba..7ad8e156e2df 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, adau1977->dvdd_reg = NULL; } - adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); - if (IS_ERR(adau1977->reset_gpio)) { - ret = PTR_ERR(adau1977->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return PTR_ERR(adau1977->reset_gpio); - adau1977->reset_gpio = NULL; - } + adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(adau1977->reset_gpio)) + return PTR_ERR(adau1977->reset_gpio); dev_set_drvdata(dev, adau1977); - if (adau1977->reset_gpio) { - ret = gpiod_direction_output(adau1977->reset_gpio, 0); - if (ret) - return ret; + if (adau1977->reset_gpio) ndelay(100); - } ret = adau1977_power_enable(adau1977); if (ret) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index f2b8aad21274..60598b230341 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, } /* Reset the Device */ - cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs35l32->reset_gpio)) { - ret = PTR_ERR(cs35l32->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs35l32->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs35l32->reset_gpio, 0); - if (ret) - return ret; + cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l32->reset_gpio)) + return PTR_ERR(cs35l32->reset_gpio); + + if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); - } /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, &reg); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ce6086835ebd..cac48ddf3ba6 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, return ret; } - cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs4265->reset_gpio)) { - ret = PTR_ERR(cs4265->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs4265->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs4265->reset_gpio, 0); - if (ret) - return ret; + cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4265->reset_gpio)) + return PTR_ERR(cs4265->reset_gpio); + + if (cs4265->reset_gpio) { mdelay(1); gpiod_set_value_cansleep(cs4265->reset_gpio, 1); - } i2c_set_clientdata(i2c_client, cs4265); diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bda2ee18769e..669e3228241e 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1213,27 +1213,15 @@ static int sta350_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta350->gpiod_nreset)) { - ret = PTR_ERR(sta350->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta350->gpiod_nreset, 0); - } - - sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); - if (IS_ERR(sta350->gpiod_power_down)) { - ret = PTR_ERR(sta350->gpiod_power_down); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_power_down = NULL; - } else { - gpiod_direction_output(sta350->gpiod_power_down, 0); - } + sta350->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_nreset)) + return PTR_ERR(sta350->gpiod_nreset); + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_power_down)) + return PTR_ERR(sta350->gpiod_power_down); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index ae23acdd2708..dfb4ff5cc9ea 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -485,16 +485,9 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable"); - if (IS_ERR(data->enable_gpio)) { - ret = PTR_ERR(data->enable_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - data->enable_gpio = NULL; - } else { - gpiod_direction_output(data->enable_gpio, 0); - } + data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); -- 2.1.4

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[alsa-devel] [PATCH] ASoC: rt5670: Set RT5670_IRQ_CTRL1 non volatile
by Bard Liao 24 Feb '15

24 Feb '15

RT5670_IRQ_CTRL1(0xbd) is a non volatile register. And we need to restore its value after suspend/resume. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 6aa4ab2..04ca247 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg) case RT5670_ADC_EQ_CTRL1: case RT5670_EQ_CTRL1: case RT5670_ALC_CTRL_1: - case RT5670_IRQ_CTRL1: case RT5670_IRQ_CTRL2: case RT5670_INT_IRQ_ST: case RT5670_IL_CMD: -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2 0/4] ASoC: Intel: BSW driver updates
by Vinod Koul 24 Feb '15

24 Feb '15

This series adds rest of the remaining 4 patches posted in v1. This is generated on top of Mark's topic/intel Subhransu S. Prusty (2): ASoC: Intel: Remove ignore suspend support ASoC: Intel: Remove soc pm handling to allow platform driver handle it Vinod Koul (2): ASoC: Intel: add pm support in sst ipc driver ASoC: Intel: Move the fw download to power_control sound/soc/intel/cht_bsw_rt5672.c | 3 - sound/soc/intel/sst/sst.c | 128 +++++++++++++++++++++++++++++-- sound/soc/intel/sst/sst.h | 9 +++ sound/soc/intel/sst/sst_drv_interface.c | 27 ++++++- 4 files changed, 152 insertions(+), 15 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH 00/13] ASoC: Intel: BSW driver updates
by Vinod Koul 24 Feb '15

24 Feb '15

This patch series first adds nonatomic pcm ops support in ASoC (it is already availble in ALSA). Then adds driver support. Later it adds puase/resume and suspend/resume support to the intel driver Subhransu S. Prusty (3): ASoC: Intel: Add memcpy32_fromio as well ASoC: Intel: Remove ignore suspend support ASoC: Intel: Remove soc pm handling to allow platform driver handle it Vinod Koul (10): ASoC: core: allow pcms to be registered as nonatomic ASoC: Intel: mark cht machine driver with nonatomic trigger ASoC: Intel: update MMX ID to 3 ASoC: Intel: add support for pause and resume in sst ASoC: Intel: add support for pcm stream suspend/resume ASoC: Intel: add support for platform suspend ASoC: Intel: save and restore the CSR register ASoC: Intel: reset the DSP while suspending ASoC: Intel: add pm support in sst ipc driver ASoC: Intel: Move the fw download to power_control include/sound/soc.h | 3 + sound/soc/intel/cht_bsw_rt5672.c | 5 +- sound/soc/intel/sst-atom-controls.h | 2 +- sound/soc/intel/sst-mfld-platform-pcm.c | 60 ++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 1 + sound/soc/intel/sst/sst.c | 134 ++++++++++++++++++++++++++++--- sound/soc/intel/sst/sst.h | 12 +++ sound/soc/intel/sst/sst_drv_interface.c | 58 ++++++++++++- sound/soc/intel/sst/sst_loader.c | 10 ++- sound/soc/soc-pcm.c | 1 + 10 files changed, 268 insertions(+), 18 deletions(-) -- 1.7.9.5

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Re: [alsa-devel] Intel HDA / ALC662 analog surround problem
by Raymond Yau 24 Feb '15

24 Feb '15

> I'm using alsa 1.0.28 (lib and utils). Everything works fine so far, i.e. stereo output through speakers and headphone as well as S/PDIF and all possible capturing. > According to my mainboard manual, the codec supports 5.1 (analog) surround by using the rear "line out" jack (for rear surround) and "mic" jack (for center and sub) - which doesn't work. (I don't have even a clue what to do to activate.) > My questions: > 1) Does 5.1 analog surround work for the ALC662? If yes, how? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Document… Channel Mode This is an enum control to change the surround-channel setup, appears only when the surround channels are available. It gives the number of channels to be used, "2ch", "4ch", "6ch", and "8ch". According to the configuration, this also controls the jack-retasking of multi-I/O jacks. >2) If not, is it a limitation of the kernel oder alsa? >HDA-Intel.conf says: > HDA-Intel.pcm.surround51.0 cards.HDA-Intel.pcm.front.0 > Does this mean that 5.1 surround is restricted to stereo in any case? This mean softvol plugin is also used by 5.1 and stereo Do "pcm playback volume" affect the volume after you change channel mode from 2ch to 6 ch so that mic and line in jacks are retasked as output ?

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[alsa-devel] [PATCH 0/6] ASoC: samsung: Spring sound support
by Andreas Färber 23 Feb '15

23 Feb '15

Hello, This series prepares sound support for the HP Chromebook 11. Regards, Andreas Andreas Färber (6): ASoC: max98088: Document DT bindings ASoC: max98088: Add DT bindings ARM: dts: Add max98089 to exynos5250-spring ASoC: samsung: Document binding for max98089 based Snow driver ASoC: samsung: Extend Snow driver to support max98089 ARM: dts: Add sound support to exynos5250-spring .../devicetree/bindings/sound/max98088.txt | 16 ++++++++++++ Documentation/devicetree/bindings/sound/snow.txt | 1 + arch/arm/boot/dts/exynos5250-spring.dts | 30 ++++++++++++++++++++++ sound/soc/codecs/max98088.c | 8 ++++++ sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/snow.c | 1 + 6 files changed, 57 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98088.txt -- 2.1.4

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