September 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] sound/pci/hda/patch_ca0132.c: potential null dereference 'dma_engine'
by Fengguang Wu 27 Sep '12

27 Sep '12

Hi Ian, FYI, there are new smatch warnings show up in commit 0b37d4c7982442730061a737c16939e8cc363861 Author: Ian Minett <ian_minett(a)creativelabs.com> Commit: Takashi Iwai <tiwai(a)suse.de> CommitDate: Fri Sep 21 09:54:29 2012 +0200 + sound/pci/hda/patch_ca0132.c:2056 dspxfr_image() error: potential null dereference 'dma_engine'. (kzalloc returns null) + sound/pci/hda/patch_ca0132.c:2056 dspxfr_image() error: we previously assumed 'dma_engine' could be null (see line 1963) --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [PATCH 1/2] ASoC: arizona: Add more clock rates
by Mark Brown 26 Sep '12

26 Sep '12

Some devices support additional clock rates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5764960..c03b65a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -270,6 +270,9 @@ static unsigned int arizona_sysclk_48k_rates[] = { 12288000, 22579200, 49152000, + 73728000, + 98304000, + 147456000, }; static unsigned int arizona_sysclk_44k1_rates[] = { @@ -277,6 +280,9 @@ static unsigned int arizona_sysclk_44k1_rates[] = { 11289600, 24576000, 45158400, + 67737600, + 90316800, + 135475200, }; static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: pcm: Report error code when we fail to init platform
by Mark Brown 26 Sep '12

26 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a67b239..40c4edb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2097,7 +2097,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - dev_err(platform->dev, "pcm constructor failed\n"); + dev_err(platform->dev, "pcm constructor failed: %d\n", + ret); return ret; } } -- 1.7.10.4

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[alsa-devel] [PATCH 1/3] ASoC: wm0010: Allow slow GPIO for reset
by Mark Brown 26 Sep '12

26 Sep '12

We never set the GPIO from atomic context so there's no reason why we can't support a GPIO that needs to sleep when configuring. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f8d6c31..4722acf 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -168,7 +168,8 @@ static void wm0010_halt(struct snd_soc_codec *codec) case WM0010_STAGE2: case WM0010_FIRMWARE: /* Remember to put chip back into reset */ - gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + gpio_set_value_cansleep(wm0010->gpio_reset, + wm0010->gpio_reset_value); /* Disable the regulators */ regulator_disable(wm0010->dbvdd); regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), @@ -387,7 +388,7 @@ static int wm0010_boot(struct snd_soc_codec *codec) } /* Release reset */ - gpio_set_value(wm0010->gpio_reset, !wm0010->gpio_reset_value); + gpio_set_value_cansleep(wm0010->gpio_reset, !wm0010->gpio_reset_value); spin_lock_irqsave(&wm0010->irq_lock, flags); wm0010->state = WM0010_OUT_OF_RESET; spin_unlock_irqrestore(&wm0010->irq_lock, flags); @@ -918,7 +919,8 @@ static int __devexit wm0010_spi_remove(struct spi_device *spi) if (wm0010->gpio_reset) { /* Remember to put chip back into reset */ - gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + gpio_set_value_cansleep(wm0010->gpio_reset, + wm0010->gpio_reset_value); } if (wm0010->irq) -- 1.7.10.4

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[alsa-devel] ASoC fix for 3.6
by Mark Brown 26 Sep '12

26 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to d0e12f3ff3472cbd8f52d3c0e6ee07a841787c40: ASoC: wm2000: Correct register size (2012-09-26 12:06:20 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.6 One small and obvious driver-specific fix. ---------------------------------------------------------------- Mark Brown (1): ASoC: wm2000: Correct register size sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

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[alsa-devel] [PATCH] ASoC: wm2000: Correct register size
by Mark Brown 26 Sep '12

26 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index b723e91..683dc43 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -726,7 +726,7 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg) } static const struct regmap_config wm2000_regmap = { - .reg_bits = 8, + .reg_bits = 16, .val_bits = 8, .max_register = WM2000_REG_IF_CTL, -- 1.7.10.4

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[alsa-devel] Merging alsa-driver and alsa-kmirror to one GIT repo
by Jaroslav Kysela 26 Sep '12

26 Sep '12

Hi all, I believe, it's time to manage only one GIT repository for the alsa-driver package now. The merge has benefits mainly for the end users - bisecting may work, the correlation between kernel sources and our out-of-kernel build framework becomes more strong. I merged the alsa-driver and alsa-kmirror repos with full history. I also added some code changes to enable the "build-in" mirror tree in Makefile and gitcompile from v1.0.18 (but I don't think that it will be used). The structure of new repository is similar, but the kernel mirrored code is located in the mirror/ subdirectory. The paths are exactly same as in the Linux kernel tree, so possible patches can be applied without any path modifications. Of course, we may create a bash script to do proper and full cherry-picks between kernel and mirrored trees (comments, author, dates). New temporary repository has name alsa-driver.new: http://git.alsa-project.org/?p=alsa-driver.new.git;a=summary The another difference is that the releases are now branches not tags. It allows me to do quick fixes out of the standard development or separate the main development in the release time window (to not include some very new code). The previous branches won't contain probably any additional commits, but it's probably better to play with only one reference scheme. I think that the change from alsa-driver.git to alsa-driver.new.git should be quick. Please, report any objections or acks now. Schedule: When accepted, I will rename the original alsa-driver.git repository to alsa-driver.old.git and move alsa-driver.new.git to alsa-driver.git quickly. 1.0.26 release: This was my last big change (with the ALSA server upgrade) which blocked me to do the 1.0.26 release. I'm going to create new release after the repos switch ASAP. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH 1/2] ASoC: fsl_ssi: Let it understand mono
by Fabio Estevam 25 Sep '12

25 Sep '12

Currently there are two variants for the FSL SSI driver: - fsl_ssi.c is the device tree driver version for the FSL SSI driver (used by PowerPC and i.MX) - imx-ssi.c is the non device tree version that runs on some i.MX processors that are not yet converted to dt. Commit 0865a7 (ASoC: imx-ssi: Remove mono support) attempted to fix mono audio playback issue inside imx-ssi.c, which according to 104c22 (ASoC: Revert 'ASoC: imx-ssi: Remove mono support' ) caused problem on another i.MX platform. The original mono playback issue was later fixed inside the mc13783 codec driver. As the long term goal is to move other platforms to dt, then fsl_ssi needs to have the same fix for channels_min as in imx-ssi. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Timur, Can you please try to play a mono track on your PowerPC platform with these series applied? sound/soc/fsl/fsl_ssi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4ed2afd..7768b93 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -559,14 +559,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { /* Template for the CPU dai driver structure */ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: core: Configure pin muxing via pinctrl when registering a DAI
by Peter Ujfalusi 25 Sep '12

25 Sep '12

pinctrl framework now becoming widely available for SoCs to configure pin multiplexing. Instead of adding the same code to all dai drivers the core can take care of this transparently. Do not make too much noise if pinctrl is not provided via DT for the dai but just use dev_info(). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam, I was about to add the pinctrl support for OMAP dais (McBSP/McPDM/DMIC/HDMI) when I realized that I just kept duplicating the same code all over. It is better to leave this to the core. Regards, Peter sound/soc/soc-core.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9a6daf9..37464c5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -33,6 +33,7 @@ #include <linux/ctype.h> #include <linux/slab.h> #include <linux/of.h> +#include <linux/pinctrl/consumer.h> #include <sound/ac97_codec.h> #include <sound/core.h> #include <sound/jack.h> @@ -3686,6 +3687,19 @@ static inline char *fmt_multiple_name(struct device *dev, return kstrdup(dai_drv->name, GFP_KERNEL); } +/* + * Simple function to be used in snd_soc_dai_register_dai/s to set the default + * pinmuxing for the dai. + */ +static inline void snd_soc_configure_dai_pins(struct device *dev) +{ + struct pinctrl *pinctrl; + + pinctrl = devm_pinctrl_get_select_default(dev); + if (IS_ERR(pinctrl)) + dev_info(dev, "pins are not configured for the DAI\n"); +} + /** * snd_soc_register_dai - Register a DAI with the ASoC core * @@ -3734,6 +3748,8 @@ int snd_soc_register_dai(struct device *dev, mutex_unlock(&client_mutex); + snd_soc_configure_dai_pins(dev); + pr_debug("Registered DAI '%s'\n", dai->name); return 0; @@ -3828,6 +3844,8 @@ int snd_soc_register_dais(struct device *dev, pr_debug("Registered DAI '%s'\n", dai->name); } + snd_soc_configure_dai_pins(dev); + return 0; err: -- 1.7.12

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[alsa-devel] [PATCH] ALSA: hda - Remove ignore_misc_bit
by David Henningsson 25 Sep '12

25 Sep '12

The purpose of this flag is unclear. If the problem is that some machines have broken misc/NO_PRESENCE bits, they should be fixed by pin fixups. In addition, this causes jack detection functionality to be flawed on the M31EI, where there are two jacks without jack detection (which is properly marked as NO_PRESENCE), but due to ignore_misc_bit, these jacks are instead being reported as being present but always unplugged. BugLink: https://bugs.launchpad.net/bugs/939161 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 4 ---- sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_jack.c | 5 ++--- 3 files changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4f7d2df..9acd5a9 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -141,7 +141,6 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, memset(sequences_hp, 0, sizeof(sequences_hp)); assoc_line_out = 0; - codec->ignore_misc_bit = true; end_nid = codec->start_nid + codec->num_nodes; for (nid = codec->start_nid; nid < end_nid; nid++) { unsigned int wid_caps = get_wcaps(codec, nid); @@ -157,9 +156,6 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); - if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) - codec->ignore_misc_bit = false; conn = get_defcfg_connect(def_conf); if (conn == AC_JACK_PORT_NONE) continue; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8fd9f63..ee4ae8e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,7 +861,6 @@ struct hda_codec { unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ - unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index aaccc02..c9333c9 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -26,9 +26,8 @@ bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) return false; if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) return false; - if (!codec->ignore_misc_bit && - (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) + if (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE) return false; if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) return false; -- 1.7.9.5

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