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

[alsa-devel] [PATCH v3 0/2] reducing memory footprint of ALSA
by Jie Yang 27 May '15

27 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... We plan to enable those reducing items only for EXPERT, and add knobs in sound/core/Kconfig. These first two patches is a start, adding SND_PROC_FS to config disable/enable ALSA proc FS, this disabling can save about 9KB code size on x86_64 platform. Changes in v3: 1. drop SND_REDUCED_MEMORY_FOOTPRINT and make configuration available only for EXPERT; 2. move the configure into sound/core/Kconfig; 3. Add missing Signed-off-by for PATCH 2/2. Changes in v2: 1. Prompt "Sound Proc FS Support" appears only when SND_REDUCED_MEMORY_FOOTPRINT is true. Jie Yang (2): ALSA: Kconfig: add config item SND_PROC_FS for expert ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS include/sound/info.h | 8 ++++---- sound/core/Kconfig | 9 +++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 61 insertions(+), 52 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH 0/2] reducing memory footprint of ALSA
by Jie Yang 27 May '15

27 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... These first two patches is a start, add config menu in Kconfig, and disable ALSA proc FS once reducing memory footprint is selected, which can save about 9KB code size on x86_64 platform. Changes in v2: Prompt "Sound Proc FS Support" appears only when SND_REDUCED_MEMORY_FOOTPRINT is true. Jie Yang (2): ALSA: Kconfig: add config menu for reduced memory footprint ALSA: set no sound proc fs for reduced memory footprint include/sound/info.h | 8 ++++---- sound/Kconfig | 16 ++++++++++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 68 insertions(+), 52 deletions(-) -- 1.9.1

4 10

[alsa-devel] [PATCH] ALSA: hda - Set stream_pm ops automatically by generic parser
by Takashi Iwai 27 May '15

27 May '15

This allows user to test power_save_node feature via sysfs or patch firmware even on the codecs that don't specify it. It'll also save a few lines. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 5 ++++- sound/pci/hda/patch_realtek.c | 1 - sound/pci/hda/patch_sigmatel.c | 1 - sound/pci/hda/patch_via.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1c8678775f40..ac0db1679f09 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4926,9 +4926,12 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused || codec->power_save_node) + if (spec->power_down_unused || codec->power_save_node) { if (!codec->power_filter) codec->power_filter = snd_hda_gen_path_power_filter; + if (!codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; + } if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fc0ccc78bdc7..e3bf72bb278c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5724,7 +5724,6 @@ static int patch_alc269(struct hda_codec *codec) set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; - codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; #ifdef CONFIG_PM codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 43c99ce4a520..0de0fd95144a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4403,7 +4403,6 @@ static const struct hda_codec_ops stac_patch_ops = { #ifdef CONFIG_PM .suspend = stac_suspend, #endif - .stream_pm = snd_hda_gen_stream_pm, .reboot_notify = stac_shutup, }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 31a95cca015d..742087ef378f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -472,7 +472,6 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = snd_hda_jack_unsol_event, - .stream_pm = snd_hda_gen_stream_pm, #ifdef CONFIG_PM .suspend = via_suspend, .check_power_status = via_check_power_status, -- 2.4.1

1 0

[alsa-devel] [PATCH] ASoC: fsl_spdif: Don't try to round-up for clock divisor calculation
by Nicolin Chen 27 May '15

27 May '15

As commit 6c8ca30eec7b ("ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation") mentioned that there's no more need to use a round up work around to get a better divisor since the clk-divider driver has been refined a lot. So this patch applies the same modification to fsl_spdif driver. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Zidan Wang <zidan.wang(a)freescale.com> --- Fabio and Zidan, Theoretically, it should have the same problem as fsl_ssi driver had. But I don't have an S/PDIF test environment. So I need your helps to confirm it. Thank you. sound/soc/fsl/fsl_spdif.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 91eb3ae..8e93221 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -417,11 +417,9 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, if (clk != STC_TXCLK_SPDIF_ROOT) goto clk_set_bypass; - /* - * The S/PDIF block needs a clock of 64 * fs * txclk_df. - * So request 64 * fs * (txclk_df + 1) to get rounded. - */ - ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (txclk_df + 1)); + /* The S/PDIF block needs a clock of 64 * fs * txclk_df */ + ret = clk_set_rate(spdif_priv->txclk[rate], + 64 * sample_rate * txclk_df); if (ret) { dev_err(&pdev->dev, "failed to set tx clock rate\n"); return ret; @@ -1060,7 +1058,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * (txclk_df + 1) * 64; + rate_ideal = rate[index] * txclk_df * 64; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else -- 1.9.1

4 6

[alsa-devel] [PATCH v2] ASoC: Add GPIO based jack device
by Dylan Reid 27 May '15

27 May '15

Add a jack device that allows for separate headphone and mic detect GPIOs. This device will be used as an aux device and will registered the jack with the card at init time. The gpios are each optional allowing for a headphone, microphone, or combination devices to be created depending on the board configuration. A board will be able to have several of these jacks, one for each physical connection. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Tested on Acer Chromebook 13 Tegra device. I'm travelling this week, I'll try to get the rockchip team to integrate this with simple card for the 3288. Changes since v1: Static component_driver Coding style Improve error handling Get GPIOs in device probe to better handle EPROBE_DEFER --- .../devicetree/bindings/sound/gpio-audio-jack.txt | 39 ++++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/gpio-audio-jack.c | 201 +++++++++++++++++++++ 4 files changed, 247 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/gpio-audio-jack.txt create mode 100644 sound/soc/codecs/gpio-audio-jack.c diff --git a/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt b/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt new file mode 100644 index 0000000..5505a49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt @@ -0,0 +1,39 @@ +Gpio-Audio-Jack: + +GPIO based audio jacks. This can represent a headphone, mic, or combo jack. + +Required properties: + +- compatible : "linux,gpio-audio-jack" + +Optional properties: + +- gpio-audio-jack,jack-name : Name of the jack. Normally + 'Headphone Jack', 'Mic Jack', or + 'Headset Jack' +- gpio-audio-jack,sw-det-gpios : Reference to GPIOs that signals when + a jack is plugged. +- gpio-audio-jack,gpio-names : Names of the GPIOs. Must provide one + per sw-det-gpio entry. +- gpio-audio-jack,report-masks : Jack report mask from + dt-bindings/sound/audio-jack-events.h. + Must have one per sw-det-gpio entry. +- gpio-audio-jack,debounce-times : Debounce time in milliseconds for each + sw-det-gpio entry. + +Example of Headphone/Mic combo jack: + +audio_jack: gpio-audio-jack { + compatible = "linux,gpio-audio-jack"; + gpio-audio-jack,jack-name = "Headset Jack"; + gpio-audio-jack,sw-det-gpios = + <&gpio TEGRA_GPIO(I, 7) GPIO_ACTIVE_HIGH>, + <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_LOW>; + gpio-audio-jack,gpio-names = "Headphones", "Mic Jack"; + gpio-audio-jack,report-masks = <JACK_HEADPHONE>, <JACK_MICROPHONE>; + gpio-audio-jack,debounce-times = <150>, <150>; +}; + +sound { + nvidia,headset-dev = <&audio_jack>; + ... diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..4147593 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_GPIO_AUDIO_JACK select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,10 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_GPIO_AUDIO_JACK + tristate "GPIO based audio jack detection" + depends on GPIOLIB || COMPILE_TEST + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..ae570f5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-gpio-audio-jack-objs := gpio-audio-jack.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_GPIO_AUDIO_JACK) += snd-soc-gpio-audio-jack.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/gpio-audio-jack.c b/sound/soc/codecs/gpio-audio-jack.c new file mode 100644 index 0000000..29fdd8a --- /dev/null +++ b/sound/soc/codecs/gpio-audio-jack.c @@ -0,0 +1,201 @@ +/* + * GPIO based audio jack detection + * + * Copyright (C) 2015 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gpio.h> +#include <linux/init.h> +#include <linux/module.h> +#include <dt-bindings/sound/audio-jack-events.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +#define JACK_NAME_CON_ID "gpio-audio-jack,jack-name" +#define GPIO_CON_ID "gpio-audio-jack,sw-det" +#define NAME_CON_ID "gpio-audio-jack,gpio-names" +#define REPORT_MASK_CON_ID "gpio-audio-jack,report-masks" +#define DEBOUNCE_CON_ID "gpio-audio-jack,debounce-times" + +struct gpio_audio_jack { + struct snd_soc_jack jack; + struct snd_soc_jack_gpio *gpios; + const char *jack_name; + int report_mask; + int gpio_count; +}; + +static int dt_jack_to_alsa_report(int dt_jack) +{ + switch (dt_jack) { + case JACK_HEADPHONE: + return SND_JACK_HEADPHONE; + case JACK_MICROPHONE: + return SND_JACK_MICROPHONE; + case JACK_LINEOUT: + return SND_JACK_LINEOUT; + case JACK_LINEIN: + return SND_JACK_LINEIN; + default: + return 0; + } +} + +static int gpio_audio_component_probe(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct gpio_audio_jack *priv = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_card_jack_new(component->card, priv->jack_name, + priv->report_mask, &priv->jack, NULL, 0); + if (ret) + return ret; + + return snd_soc_jack_add_gpiods(dev, &priv->jack, + priv->gpio_count, priv->gpios); +} + +static const struct snd_soc_component_driver gpio_jack_drv = { + .name = "gpio-audio-jack-dev", + .probe = gpio_audio_component_probe, +}; + +static int gpio_audio_jack_probe(struct platform_device *pdev) +{ + struct gpio_audio_jack *priv; + struct device *dev = &pdev->dev; + const char **gpio_names; + u32 *debounce; + u32 *gpio_report; + int ret; + int i; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + + ret = device_property_read_string(dev, JACK_NAME_CON_ID, + &priv->jack_name); + if (ret < 0) { + dev_err(dev, "Failed to read jack name %d\n", ret); + return ret; + } + + priv->gpio_count = gpiod_count(dev, GPIO_CON_ID); + if (priv->gpio_count <= 0) + return priv->gpio_count; + + gpio_names = devm_kcalloc(dev, priv->gpio_count, sizeof(*gpio_names), + GFP_KERNEL); + if (!gpio_names) + return -ENOMEM; + + gpio_report = devm_kcalloc(dev, priv->gpio_count, sizeof(*gpio_report), + GFP_KERNEL); + if (!gpio_report) + return -ENOMEM; + + debounce = devm_kcalloc(dev, priv->gpio_count, sizeof(*debounce), + GFP_KERNEL); + if (!debounce) + return -ENOMEM; + + priv->gpios = devm_kcalloc(dev, priv->gpio_count, sizeof(*priv->gpios), + GFP_KERNEL); + if (!priv->gpios) + return -ENOMEM; + + ret = device_property_read_string_array(dev, NAME_CON_ID, gpio_names, + priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio names %d\n", ret); + return ret; + } + + ret = device_property_read_u32_array(dev, REPORT_MASK_CON_ID, + gpio_report, priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio masks %d\n", ret); + return ret; + } + + ret = device_property_read_u32_array(dev, DEBOUNCE_CON_ID, + debounce, priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio debounce times %d\n", ret); + return ret; + } + + for (i = 0; i < priv->gpio_count; i++) { + priv->gpios[i].desc = gpiod_get_index(dev, GPIO_CON_ID, + i, GPIOD_IN); + if (IS_ERR(priv->gpios[i].desc)) { + ret = PTR_ERR(priv->gpios[i].desc); + dev_err(priv->gpios[i].gpiod_dev, + "Cannot get jack gpio at index %d: %d", + i, ret); + goto gpio_err; + } + + priv->gpios[i].name = gpio_names[i]; + priv->gpios[i].report = dt_jack_to_alsa_report(gpio_report[i]); + priv->gpios[i].debounce_time = debounce[i]; + priv->report_mask |= gpio_report[i]; + } + + devm_kfree(dev, gpio_names); + devm_kfree(dev, gpio_report); + devm_kfree(dev, debounce); + + return devm_snd_soc_register_component(&pdev->dev, &gpio_jack_drv, + NULL, 0); + +gpio_err: + for (i = 0; i < priv->gpio_count; i++) { + if (!IS_ERR(priv->gpios[i].desc)) + gpiod_put(priv->gpios[i].desc); + } + return ret; +} + +static int gpio_audio_jack_remove(struct platform_device *pdev) +{ + struct gpio_audio_jack *priv = dev_get_drvdata(&pdev->dev); + int i; + + for (i = 0; i < priv->gpio_count; i++) { + if (!IS_ERR(priv->gpios[i].desc)) + gpiod_put(priv->gpios[i].desc); + } + + return 0; +} + +static const struct of_device_id gpio_audio_of_match[] = { + { .compatible = "linux,gpio-audio-jack", }, + { } +}; +MODULE_DEVICE_TABLE(of, gpio_audio_of_match); + +static struct platform_driver gpio_audio_jack = { + .driver = { + .name = "gpio-audio-jack", + .of_match_table = gpio_audio_of_match, + }, + .probe = gpio_audio_jack_probe, + .remove = gpio_audio_jack_remove, +}; +module_platform_driver(gpio_audio_jack); + +MODULE_DESCRIPTION("ASoC GPIO audio jack driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPL v2"); -- 2.2.1.62.g3f15098

2 1

[alsa-devel] [PATCH] ASoC: max98925: Fix mask for setting DAI invert mode
by Axel Lin 26 May '15

26 May '15

The M98925_DAI_WCI_MASK bit is not updated with current code. To properly set the DAI invert mode, the mask should be M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98925.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 9b5a17d..aad6642 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -346,7 +346,7 @@ static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, } regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_BCI_MASK, invert); + M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK, invert); return 0; } -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: Simplify format_register_str() without stack usages
by Takashi Iwai 26 May '15

26 May '15

Instead of allocating two string buffers on stack and copying them back, manipulate directly the target string buffer. This simplifies the code well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-core.c | 27 +++++++++------------------ 1 file changed, 9 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 23732523f87c..ee86f84b7ccc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -92,30 +92,21 @@ static int format_register_str(struct snd_soc_codec *codec, int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; int regsize = codec->driver->reg_word_size * 2; int ret; - char tmpbuf[len + 1]; - char regbuf[regsize + 1]; - - /* since tmpbuf is allocated on the stack, warn the callers if they - * try to abuse this function */ - WARN_ON(len > 63); /* +2 for ': ' and + 1 for '\n' */ if (wordsize + regsize + 2 + 1 != len) return -EINVAL; - ret = snd_soc_read(codec, reg); - if (ret < 0) { - memset(regbuf, 'X', regsize); - regbuf[regsize] = '\0'; - } else { - snprintf(regbuf, regsize + 1, "%.*x", regsize, ret); - } - - /* prepare the buffer */ - snprintf(tmpbuf, len + 1, "%.*x: %s\n", wordsize, reg, regbuf); - /* copy it back to the caller without the '\0' */ - memcpy(buf, tmpbuf, len); + sprintf(buf, "%.*x: ", wordsize, reg); + buf += wordsize + 2; + ret = snd_soc_read(codec, reg); + if (ret < 0) + memset(buf, 'X', regsize); + else + sprintf(buf, "%.*x", regsize, ret); + buf[regsize] = '\n'; + /* no NUL-termination needed */ return 0; } -- 2.4.1

2 1

Re: [alsa-devel] ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets
by Dan Carpenter 26 May '15

26 May '15

Hello Takashi Sakamoto, The patch 6fc6b9ce41c6: "ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets" from May 22, 2015, leads to the following static checker warning: sound/firewire/amdtp.c:857 in_stream_callback() warn: unsigned 'data_blocks' is never less than zero. sound/firewire/amdtp.c 855 856 data_blocks = handle_in_packet(s, payload_quadlets, buffer); 857 if (data_blocks < 0) { ^^^^^^^^^^^^^^^ Never true. But handle_in_packet() does return negative error codes. 858 s->packet_index = -1; 859 break; 860 } 861 regards, dan carpenter

2 1

[alsa-devel] [PATCH 0/2] reducing memory footprint of ALSA
by Jie Yang 26 May '15

26 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... These first two patches is a start, add config menu in Kconfig, and disable ALSA proc FS once reducing memory footprint is selected, which can save about 9KB code size on x86_64 platform. Comments are welcomed. ~Keyon Jie Yang (2): ALSA: Kconfig: add config menu for reduced memory footprint ALSA: set no sound proc fs for reduced memory footprint include/sound/info.h | 8 ++++---- sound/Kconfig | 16 ++++++++++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 68 insertions(+), 52 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH 0/5] Trivial cleanups
by Takashi Iwai 26 May '15

26 May '15

Hi, here is a series of patches to clean up / fix the issues spotted by sparse. Takashi === Takashi Iwai (5): ALSA: dummy: make local data static ALSA: lx6464es: Fix duplicated SSID entries ALSA: lx6464es: Use NULL for pointers ALSA: bcd2000: Make local data static sound: sound_firmware: Fix invalid use of vfs_read() sound/drivers/dummy.c | 14 +++++++------- sound/pci/lx6464es/lx6464es.c | 16 ++++++++-------- sound/sound_firmware.c | 4 +--- sound/usb/bcd2000/bcd2000.c | 2 +- 4 files changed, 17 insertions(+), 19 deletions(-) -- 2.4.1

1 5