September 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] ASoC: ab8500-codec: info leak in anc_status_control_put()
by Dan Carpenter 13 Sep '13

13 Sep '13

If the user passes an invalid value it leads to an info leak when we print the error message or it could oops. This is called with user supplied data from snd_ctl_elem_write(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b8ba0ad..80555d7 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; - int req, status; + unsigned int req; + int status; dev_dbg(dev, "%s: Enter.\n", __func__); mutex_lock(&drvdata->anc_lock); req = ucontrol->value.integer.value[0]; + if (req >= ARRAY_SIZE(enum_anc_state)) { + status = -EINVAL; + goto cleanup; + } if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",

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[alsa-devel] [patch] ASoC: 88pm860x: array overflow in snd_soc_put_volsw_2r_st()
by Dan Carpenter 13 Sep '13

13 Sep '13

This is called from snd_ctl_elem_write() with user supplied data so we need to add some bounds checking. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 8af0434..259d1ac 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -349,6 +349,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; val2 = ucontrol->value.integer.value[1]; + if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table)) + return -EINVAL; + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); if (err < 0) return err;

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[alsa-devel] [patch] ASoC: max98095: a couple array underflows
by Dan Carpenter 13 Sep '13

13 Sep '13

The ->put() function are called from snd_ctl_elem_write() with user supplied data. The limit checks here could underflow leading to a crash. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd16..8dbcacd 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_eq_channel(kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_eq_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; @@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave;

2 1

[alsa-devel] [PATCH] ASoC: rt5640: Provide more useful hw_params error reasons.
by Liam Girdwood 13 Sep '13

13 Sep '13

Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/codecs/rt5640.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 2f6bb16..de40bd9 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1609,7 +1609,8 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, rt5640->lrck[dai->id] = params_rate(params); pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting\n"); + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5640->lrck[dai->id], dai->id); return -EINVAL; } frame_size = snd_soc_params_to_frame_size(params); -- 1.8.1.2

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[alsa-devel] virtual alsa mixer device
by André Roth 13 Sep '13

13 Sep '13

hello, I'd like to implement a mixer control device for an external amplifier, which can be controlled via rs232. would be nice to adjust volume with any alsa mixer program.. is there a way to do this in user space ? any hints for the right api / examples ? -- andré

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[alsa-devel] [PATCH] ASoC: codecs: da9055: Update driver name to fix breakage due to pmic driver with same name
by Ashish Chavan 13 Sep '13

13 Sep '13

This patch updates i2c driver name and device id of da9055 codec driver. DA9055 is a PMIC + CODEC and currently, the corresponding PMIC driver also registers itself with the same name as codec, i.e. "da9055". Because of this the codec driver was broken. Now codec driver uses "da9055-codec" as driver name instead of "da9055". Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 2e1bf5d..e146f68 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1544,7 +1544,7 @@ static int da9055_remove(struct i2c_client *client) } static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, + { "da9055-codec", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); @@ -1552,7 +1552,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, .probe = da9055_i2c_probe, -- 1.7.1

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[alsa-devel] No sound from internal speaker on 2013 Macbook Air 6, 1
by Ian Munsie 13 Sep '13

13 Sep '13

Hey all, I'm trying to get sound working on my new mid 2013 Macbook Air 6,1. Sound works via the headphone jack, but I haven't managed to get the internal speaker to work. I've checked all the controls in the mixer, tried a few different mac related model= options to snd-hda-intel and even tried tweaking things with hda_analyzer to no avail. I've uninstalled pulseaudio (for now) to get rid of it's ALSA hook, but had the same result both with and without it. I'm running on a 3.11-rc6+ built today (had a few other fixes to make for this machine) - I didn't bother including the hda i915 driver as I don't need sound via the display port, but the hda driver does print a warning about that missing module. The speaker wasn't working on the 3.10 kernel from Debian either which included that module. Any help would be appreciated - I've attached my alsa-info output taken after a reboot. Cheers, -Ian -- http://sites.google.com/site/DarkStarJunkSpace -- http://darkstarshout.blogspot.com/ -- Please avoid sending me Word or PowerPoint attachments. See http://www.gnu.org/philosophy/no-word-attachments.html

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[alsa-devel] [PATCH 1/1] ALSA: rme9652: Remove redundant break
by Sachin Kamat 13 Sep '13

13 Sep '13

'break' after return statement is not necessary. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/rme9652/hdspm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 3cde55b..2907e68 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3996,7 +3996,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) return 1; } return 0; - break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); if (status & HDSPM_tcoLockAes) { @@ -4006,9 +4005,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) return 1; } return 0; - - break; - case RayDAT: case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); @@ -4018,7 +4014,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) if (status & 0x4000000) return 1; /* Lock */ return 0; /* No signal */ - break; default: break; -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ALSA: au88x0: Remove redundant break
by Sachin Kamat 13 Sep '13

13 Sep '13

'break' after a return statement is redundant. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/au88x0/au88x0_synth.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 8bef473..922a84b 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -219,7 +219,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_RUN(wt), val); return 0xc; - break; case 1: /* param 0 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -227,7 +226,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 0), val); return 0xc; - break; case 2: /* param 1 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -235,7 +233,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 1), val); return 0xc; - break; case 3: /* param 2 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -243,7 +240,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 2), val); return 0xc; - break; case 4: /* param 3 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -251,7 +247,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 3), val); return 0xc; - break; case 6: /* mute */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -259,20 +254,17 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_MUTE(wt), val); return 0xc; - break; case 0xb: - { /* delay */ - /* - printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", - WT_DELAY(wt,0), (int)val); - */ - hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 0), val); - return 0xc; - } - break; + /* delay */ + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_DELAY(wt,0), (int)val); + */ + hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 0), val); + return 0xc; /* Global WT block parameters */ case 5: /* sramp */ ecx = WT_SRAMP(wt); @@ -291,7 +283,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, break; default: return 0; - break; } /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: ctxfi: Staticize local symbols
by Sachin Kamat 13 Sep '13

13 Sep '13

Local symbols used only in this file are made static. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/ctxfi/ctdaio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 0c00eb4..84f86bf 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -33,7 +33,7 @@ struct daio_rsc_idx { unsigned short right; }; -struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { +static struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x00, .right = 0x01}, [LINEO2] = {.left = 0x18, .right = 0x19}, [LINEO3] = {.left = 0x08, .right = 0x09}, @@ -44,7 +44,7 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [SPDIFI1] = {.left = 0x95, .right = 0x9d}, }; -struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { +static struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x40, .right = 0x41}, [LINEO2] = {.left = 0x60, .right = 0x61}, [LINEO3] = {.left = 0x50, .right = 0x51}, -- 1.7.9.5

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