September 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] fm801: Fix double free in case of error in tuner detection
by Ben Hutchings 22 Sep '11

22 Sep '11

Commit 9676001559fce06e37c7dc230ab275f605556176 ("ALSA: fm801: add error handling if auto-detect fails") added incorrect error handling. Once we have successfully called snd_device_new(), the cleanup function fm801_free() will automatically be called by snd_card_free() and we must *not* also call fm801_free() directly. Reported-by: Hor Jiun Shyong <jiunshyong(a)gmail.com> References: http://bugs.debian.org/641946 Signed-off-by: Ben Hutchings <ben(a)decadent.org.uk> Cc: stable(a)kernel.org [v3.0+] --- sound/pci/fm801.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index a7ec703..0dd8963 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1236,7 +1236,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, (tea575x_tuner & TUNER_TYPE_MASK) < 4) { if (snd_tea575x_init(&chip->tea)) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - snd_fm801_free(chip); return -ENODEV; } } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) { @@ -1251,7 +1250,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, } if (tea575x_tuner == 4) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - snd_fm801_free(chip); return -ENODEV; } } -- 1.7.5.4

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[alsa-devel] [PATCH 2/2] fm801: Gracefully handle failure of tuner auto-detect
by Ben Hutchings 22 Sep '11

22 Sep '11

Commit 9676001559fce06e37c7dc230ab275f605556176 ("ALSA: fm801: add error handling if auto-detect fails") seems to break systems that were previously working without a tuner. As a bonus, this should fix init and cleanup for the case where the tuner is explicitly disabled. Reported-and-tested-by: Hor Jiun Shyong <jiunshyong(a)gmail.com> References: http://bugs.debian.org/641946 Signed-off-by: Ben Hutchings <ben(a)decadent.org.uk> Cc: stable(a)kernel.org [v3.0+] --- sound/pci/fm801.c | 13 ++++++++++--- 1 files changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 0dd8963..ecce948 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only)."); +#define TUNER_DISABLED (1<<3) #define TUNER_ONLY (1<<4) #define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) @@ -1150,7 +1151,8 @@ static int snd_fm801_free(struct fm801 *chip) __end_hw: #ifdef CONFIG_SND_FM801_TEA575X_BOOL - snd_tea575x_exit(&chip->tea); + if (!(chip->tea575x_tuner & TUNER_DISABLED)) + snd_tea575x_exit(&chip->tea); #endif if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1250,10 +1252,15 @@ static int __devinit snd_fm801_create(struct snd_card *card, } if (tea575x_tuner == 4) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - return -ENODEV; + chip->tea575x_tuner = TUNER_DISABLED; } } - strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card)); + if (!(chip->tea575x_tuner & TUNER_DISABLED)) { + strlcpy(chip->tea.card, + snd_fm801_tea575x_gpios[(tea575x_tuner & + TUNER_TYPE_MASK) - 1].name, + sizeof(chip->tea.card)); + } #endif *rchip = chip; -- 1.7.5.4

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[alsa-devel] [PATCH 2/4] src/pcm/pcm_mmap.c: add missing free
by Julia Lawall 22 Sep '11

22 Sep '11

From: Julia Lawall <julia(a)diku.dk> Something that is allocated using calloc is not freed on an error path. Signed-off-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Suman Saha <sumsaha(a)gmail.com> --- Compile-tested only. src/pcm/pcm_mmap.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_mmap.c b/src/pcm/pcm_mmap.c index 4621fe6..2a009c8 100644 --- a/src/pcm/pcm_mmap.c +++ b/src/pcm/pcm_mmap.c @@ -320,8 +320,11 @@ int snd_pcm_mmap(snd_pcm_t *pcm) snd_pcm_channel_info_t *i = &pcm->mmap_channels[c]; i->channel = c; err = snd_pcm_channel_info(pcm, i); - if (err < 0) + if (err < 0) { + free(pcm->mmap_channels); + pcm->mmap_channels = NULL; return err; + } } for (c = 0; c < pcm->channels; ++c) { snd_pcm_channel_info_t *i = &pcm->mmap_channels[c];

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Re: [alsa-devel] Using the timer
by Clemens Ladisch 22 Sep '11

22 Sep '11

Ilya Dmitrichenko wrote: > On 21 September 2011 13:54, Clemens Ladisch <clemens(a)ladisch.de> wrote: > > What hardware timer? > > I meant the clock of the soundcard. The HDSP does have a sample clock, but it cannot be used independently from a running PCM stream. Using this PCM slave timer in the sequencer is possible, but the period length is likely to be much larger than the desired sequencer timestamp granularity. What exactly do you want to do? Regards, Clemens

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[alsa-devel] [PATCH 1/4] src/pcm/pcm_rate.c: add missing free
by Julia Lawall 22 Sep '11

22 Sep '11

From: Julia Lawall <julia(a)diku.dk> Something that is allocated using calloc is not freed on one or more error paths. Signed-off-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Suman Saha <sumsaha(a)gmail.com> --- Compile-tested only. src/pcm/pcm_rate.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 70e30e5..eb35e4a 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1392,11 +1392,13 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, } else { SNDERR("Invalid type for rate converter"); snd_pcm_close(pcm); + free(rate); return -EINVAL; } if (err < 0) { SNDERR("Cannot find rate converter"); snd_pcm_close(pcm); + free(rate); return -ENOENT; } #else @@ -1405,6 +1407,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, err = open_func(SND_PCM_RATE_PLUGIN_VERSION, &rate->obj, &rate->ops); if (err < 0) { snd_pcm_close(pcm); + free(rate); return err; } #endif @@ -1413,6 +1416,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, ! rate->ops.input_frames || ! rate->ops.output_frames) { SNDERR("Inproper rate plugin %s initialization", type); snd_pcm_close(pcm); + free(rate); return err; }

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[alsa-devel] [GIT PULL] OMAP McPDM updates for 3.2
by Péter Ujfalusi 22 Sep '11

22 Sep '11

Hello Mark, as we discussed you can pull the OMAP McPDM updates. As for reference: ASoC: OMAP4: McPDM: Fix legacy support [4 patch] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/042401.html ASoC: omap-mcpdm: Replace legacy driver: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/043317.html All received acks Regards, Péter The following changes since commit 0722d055ac2236da4e319d22a99c9f7e82dbdd5d: ASoC: tpa6130a2: Remove model_id from platform data (2011-09-21 16:07:45 +0100) are available in the git repository at: git://gitorious.org/omap-audio/linux-audio.git peter/topic/for-mark/mcpdm_for-3.2 Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (4): OMAP: McPDM: Convert McPDM device to omap_device OMAP4: hwmod: enable mcpdm hwmod device. ASoC: omap-mcpdm: Fix threshold and dma configuration ASoC: OMAP4: McPDM: Convert to hwmod/omap_device arch/arm/mach-omap2/devices.c | 33 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/plat-omap/devices.c | 36 --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 470 ---------------------------- sound/soc/omap/mcpdm.h | 153 --------- sound/soc/omap/omap-mcpdm.c | 442 ++++++++++++++++++++------ sound/soc/omap/omap-mcpdm.h | 95 ++++++ sound/soc/omap/sdp4430.c | 4 +- 9 files changed, 472 insertions(+), 765 deletions(-) delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.h

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[alsa-devel] [PATCH] HDA - add "Independent HP" switch for ad1988
by Raymond Yau 22 Sep '11

22 Sep '11

Add "Independent HP" switch for ad1988 - add playback device 2 "AD1988 HP" for 7.1+2 multi streaming playback - add "Independent HP" switch to enable/disable the feature switch is inactive and write protect when device 0 or device 2 is opened - remove 6stack-dig-fp model

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[alsa-devel] [PATCH] ASoC: Include delay.h in 88pm860x
by Mark Brown 22 Sep '11

22 Sep '11

Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/88pm860x-codec.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index df7b4a0..5ca122e 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -15,6 +15,7 @@ #include <linux/platform_device.h> #include <linux/mfd/88pm860x.h> #include <linux/slab.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 1.7.6.3

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[alsa-devel] [PATCH 1/2] ASoC: Add platform data for WM1250 EV1 GPIOs
by Mark Brown 22 Sep '11

22 Sep '11

The WM1250 EV1 has some GPIOs which can be used to control the behaviour at runtime. Request them all if supplied and add a set_bias_level() function to start and stop the clocks. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm1250-ev1.h | 27 +++++++++ sound/soc/codecs/wm1250-ev1.c | 121 ++++++++++++++++++++++++++++++++++++++++- 2 files changed, 145 insertions(+), 3 deletions(-) create mode 100644 include/sound/wm1250-ev1.h diff --git a/include/sound/wm1250-ev1.h b/include/sound/wm1250-ev1.h new file mode 100644 index 0000000..7dff828 --- /dev/null +++ b/include/sound/wm1250-ev1.h @@ -0,0 +1,27 @@ +/* + * linux/sound/wm1250-ev1.h - Platform data for WM1250-EV1 + * + * Copyright 2011 Wolfson Microelectronics. PLC. + * + * 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. + */ + +#ifndef __LINUX_SND_WM1250_EV1_H +#define __LINUX_SND_WM1250_EV1_H + +#define WM1250_EV1_NUM_GPIOS 5 + +#define WM1250_EV1_GPIO_CLK_ENA 0 +#define WM1250_EV1_GPIO_CLK_SEL0 1 +#define WM1250_EV1_GPIO_CLK_SEL1 2 +#define WM1250_EV1_GPIO_OSR 3 +#define WM1250_EV1_GPIO_MASTER 4 + + +struct wm1250_ev1_pdata { + int gpios[WM1250_EV1_NUM_GPIOS]; +}; + +#endif diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 4523c4c..0554b88 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -12,10 +12,59 @@ #include <linux/init.h> #include <linux/module.h> +#include <linux/slab.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/wm1250-ev1.h> + +static const char *wm1250_gpio_names[WM1250_EV1_NUM_GPIOS] = { + "WM1250 CLK_ENA", + "WM1250 CLK_SEL0", + "WM1250 CLK_SEL1", + "WM1250 OSR", + "WM1250 MASTER", +}; + +struct wm1250_priv { + struct gpio gpios[WM1250_EV1_NUM_GPIOS]; +}; + +static int wm1250_ev1_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm1250_priv *wm1250 = dev_get_drvdata(codec->dev); + int ena; + + if (wm1250) + ena = wm1250->gpios[WM1250_EV1_GPIO_CLK_ENA].gpio; + else + ena = -1; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (ena >= 0) + gpio_set_value_cansleep(ena, 1); + break; + + case SND_SOC_BIAS_OFF: + if (ena >= 0) + gpio_set_value_cansleep(ena, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} static const struct snd_soc_dapm_widget wm1250_ev1_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0), @@ -53,12 +102,66 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { .num_dapm_widgets = ARRAY_SIZE(wm1250_ev1_dapm_widgets), .dapm_routes = wm1250_ev1_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm1250_ev1_dapm_routes), + + .set_bias_level = wm1250_ev1_set_bias_level, }; +static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) +{ + struct wm1250_ev1_pdata *pdata = dev_get_platdata(&i2c->dev); + struct wm1250_priv *wm1250; + int i, ret; + + if (!pdata) + return 0; + + wm1250 = kzalloc(sizeof(*wm1250), GFP_KERNEL); + if (!wm1250) { + dev_err(&i2c->dev, "Unable to allocate private data\n"); + ret = -ENOMEM; + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm1250->gpios); i++) { + wm1250->gpios[i].gpio = pdata->gpios[i]; + wm1250->gpios[i].label = wm1250_gpio_names[i]; + wm1250->gpios[i].flags = GPIOF_OUT_INIT_LOW; + } + wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].flags = GPIOF_OUT_INIT_HIGH; + wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].flags = GPIOF_OUT_INIT_HIGH; + + ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret); + goto err_alloc; + } + + dev_set_drvdata(&i2c->dev, wm1250); + + return ret; + +err_alloc: + kfree(wm1250); +err: + return ret; +} + +static void wm1250_ev1_free(struct i2c_client *i2c) +{ + struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev); + + if (wm1250) { + gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); + kfree(wm1250); + } +} + static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { - int id, board, rev; + int id, board, rev, ret; + + dev_set_drvdata(&i2c->dev, NULL); board = i2c_smbus_read_byte_data(i2c, 0); if (board < 0) { @@ -76,13 +179,25 @@ static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %d\n", rev + 1); - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, - &wm1250_ev1_dai, 1); + ret = wm1250_ev1_pdata(i2c); + if (ret != 0) + return ret; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, + &wm1250_ev1_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + wm1250_ev1_free(i2c); + return ret; + } + + return 0; } static int __devexit wm1250_ev1_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); + wm1250_ev1_free(i2c); return 0; } -- 1.7.6.3

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[alsa-devel] sound card matrix defaced
by Patrick Shirkey 22 Sep '11

22 Sep '11

Hi, Someone needs to restore the sound card matrix as it has been defaced by spammers. Cheers -- Patrick Shirkey Boost Hardware Ltd

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