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

[alsa-devel] SEGFAULT in ALSA lib 1.0.27.1 pcm_direct parser
by khalasa＠piap.pl 14 Jun '13

14 Jun '13

Hi, Fixes a segfault when getgrnam_r() returns 0 and still sets pgrp to NULL when the group simply doesn't exist. --- alsa-lib/src/pcm/pcm_direct.c 2013-05-21 10:48:28.000000000 +0200 +++ alsa-lib/src/pcm/pcm_direct.c 2013-06-14 13:31:28.981013646 +0200 @@ -1636,7 +1636,7 @@ if (buffer == NULL) return -ENOMEM; int st = getgrnam_r(group, &grp, buffer, len, &pgrp); - if (st != 0) { + if (st != 0 || !pgrp) { SNDERR("The field ipc_gid must be a valid group (create group %s)", group); free(buffer); return -EINVAL; -- Krzysztof Halasa Research Institute for Automation and Measurements PIAP Al. Jerozolimskie 202, 02-486 Warsaw, Poland

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[alsa-devel] [PATCH (alsa-utils)] amixer: Print TLV of channel map controls
by David Henningsson 14 Jun '13

14 Jun '13

Previously these were written as "unk-25..." which wasn't very user friendly. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- amixer/amixer.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/amixer/amixer.c b/amixer/amixer.c index fe83b49..7727943 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -443,6 +443,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ unsigned int type = tlv[0]; unsigned int size; unsigned int idx = 0; + const char *chmap_type = NULL; if (tlv_size < 2 * sizeof(unsigned int)) { printf("TLV size error!\n"); @@ -541,6 +542,27 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ } break; #endif +#ifdef SND_CTL_TLVT_CHMAP_FIXED + case SND_CTL_TLVT_CHMAP_FIXED: + chmap_type = "fixed"; + /* Fall through */ + case SND_CTL_TLVT_CHMAP_VAR: + if (!chmap_type) + chmap_type = "variable"; + /* Fall through */ + case SND_CTL_TLVT_CHMAP_PAIRED: + if (!chmap_type) + chmap_type = "paired"; + printf("chmap-%s=", chmap_type); + + while (size > 0) { + printf("%s", snd_pcm_chmap_name(tlv[idx++])); + size -= sizeof(unsigned int); + if (size > 0) + printf(","); + } + break; +#endif default: printf("unk-%i-", type); while (size > 0) { -- 1.8.3

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[alsa-devel] [PATCH v2] ASoC: WM8962: Enable start-up and normal bias after reset in runtime resume
by Nicolin Chen 14 Jun '13

14 Jun '13

This part of bias settings are essential for WM8962 to power up. Without it "wm8962 0-001a: DC servo timed out" might be prompted due to power-up failure that happens to FLL if being used. The driver's also bringing the bias down in the suspend path so it needs to be powered up in the resume path for symmetry. According to dapm_pre_sequence_async(), DAPM would call pm_runtime_get_sync() to let driver finish the bias settings in pm_runtime_resume() before the bias level being set to STANDBY. So no need to worry about disordered settings for VMID of WM8962. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog: v1->v2 * Revised the commit comment to make it clear. sound/soc/codecs/wm8962.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7a7a056..c63b58e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3773,6 +3773,17 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); + + /* Bias enable at 2*5k (fast start-up) */ + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_BIAS_ENA | WM8962_VMID_SEL_MASK, + WM8962_BIAS_ENA | 0x180); + + msleep(5); + return 0; } -- 1.7.1

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[alsa-devel] [PATCH 1/5] ASoC: dapm: Add snd_soc_dapm_switch to the power up/down sequence table
by Lars-Peter Clausen 14 Jun '13

14 Jun '13

The power up/down sequence order for DAPM switch widgets is not explicitly initialized, causing them to be run always as the first widget type for both power up and down. Move it to the same position in the sequence as other mixer widget types. Cc: stable(a)vger.kernel.org Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2201f22..10290c7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -64,6 +64,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_virt_mux] = 5, [snd_soc_dapm_value_mux] = 5, [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_switch] = 7, [snd_soc_dapm_mixer] = 7, [snd_soc_dapm_mixer_named_ctl] = 7, [snd_soc_dapm_pga] = 8, @@ -83,6 +84,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_line] = 2, [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, -- 1.8.0

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[alsa-devel] [PATCH] ASoC: WM8962: Enable start-up and normal bias after reset in runtime resume
by Nicolin Chen 14 Jun '13

14 Jun '13

The analogue circuits in WM8962 require a mid-rail analogue reference voltage, VMID, which generator is enabled by STARTUP_BIAS_ENA bit, while VMID_BUF_ENA bit can enable buffered VMID to avoid audible pops. Besides, these circuits also need a bias current enabled by BIAS_ENA. Then set VMID_SEL to 2x5k divider, the fast start-up mode, to power it up. And let set_bias_level() handle its 2x250k for low power standby and 2x50k for normal operation. This patch might be kinda reverting operation against commit 52c0eee3 but after testing with WM8962 CODEC, these configurations during power up should be essential and soc-core would not do any of them. So the codec drvier need to handle it. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7a7a056..c63b58e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3773,6 +3773,17 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); + + /* Bias enable at 2*5k (fast start-up) */ + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_BIAS_ENA | WM8962_VMID_SEL_MASK, + WM8962_BIAS_ENA | 0x180); + + msleep(5); + return 0; } -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Fix double assignment of .owner in struct snd_soc_card
by Emil Goode 14 Jun '13

14 Jun '13

In struct snd_soc_card zylonite .owner is assigned THIS_MODULE twice, remove one of them. Signed-off-by: Emil Goode <emilgoode(a)gmail.com> --- sound/soc/pxa/zylonite.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ceb6566..db8aadf 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -256,7 +256,6 @@ static struct snd_soc_card zylonite = { .resume_pre = &zylonite_resume_pre, .dai_link = zylonite_dai, .num_links = ARRAY_SIZE(zylonite_dai), - .owner = THIS_MODULE, }; static struct platform_device *zylonite_snd_ac97_device; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: WM8962: Remove remaining direct register cache accesses
by Nicolin Chen 14 Jun '13

14 Jun '13

Also fix return values for headphone switch updates. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 23 ++++++++++++----------- 1 files changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 40f0e71..a24f6dd 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1601,7 +1601,6 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; int ret; /* Apply the update (if any) */ @@ -1610,16 +1609,19 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, - reg_cache[WM8962_HPOUTL_VOLUME]); + ret = snd_soc_read(codec, WM8962_PWR_MGMT_2); + if (ret & WM8962_HPOUTL_PGA_ENA) { + snd_soc_write(codec, WM8962_HPOUTL_VOLUME, + snd_soc_read(codec, WM8962_HPOUTL_VOLUME)); + return 1; + } /* ...otherwise the right. The VU is stereo. */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, - reg_cache[WM8962_HPOUTR_VOLUME]); + if (ret & WM8962_HPOUTR_PGA_ENA) + snd_soc_write(codec, WM8962_HPOUTR_VOLUME, + snd_soc_read(codec, WM8962_HPOUTR_VOLUME)); - return 0; + return 1; } /* The VU bits for the speakers are in a different register to the mute @@ -3376,7 +3378,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct wm8962_pdata *pdata = &wm8962->pdata; - u16 *reg_cache = codec->reg_cache; int i, trigger, irq_pol; bool dmicclk, dmicdat; @@ -3433,8 +3434,8 @@ static int wm8962_probe(struct snd_soc_codec *codec) /* Put the speakers into mono mode? */ if (pdata->spk_mono) - reg_cache[WM8962_CLASS_D_CONTROL_2] - |= WM8962_SPK_MONO; + snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2, + WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); /* Micbias setup, detection enable and detection * threasholds. */ -- 1.7.1

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Re: [alsa-devel] [PATCH] ASoC: dmaengine: add runtime status checking in dmaengine_pcm_dma_complete
by Mark Brown 14 Jun '13

14 Jun '13

On Fri, Jun 07, 2013 at 07:57:29PM +0800, Qiao Zhou wrote: > the dmaengine_pcm_dma_complete callback is usually executed after > the dma interrupt, which uses tasklet_schedule, workqueue, or other > method for quick int handler return. > > in some corner case, where pcm stream is released unexpected, like > media server is killed, the runtime parameter will be freed. if it > happens between the t1 and t2 in below chart, then the callback > will try to access members of paramters which is already freed, > and kernel panics. > > to avoid this issue, add runtime checking before other handling in > dmaengine_pcm_dma_complete. if pcm stream is already released, > just ignore the current handling and return. This doesn't seem like a good or robust way of fixing this, if we're tearing down the resources the DMA is using while the DMA is in progress then in the worst case that might include the memory being DMAed and of course there's races if you just check the pointer - the pointer can be checked at the same time as it's being freed (or between the free and the clear). I think we should be either halting the DMA or waiting for it to finish here.

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[alsa-devel] [PATCH] ASoc: si476x: Do not use binary constants
by James Hogan 14 Jun '13

14 Jun '13

Gcc < 4.3 doesn't understand binary constants (0b*): sound/soc/codecs/si476x.c:172:8: error: invalid suffix "b1111110" on integer constant sound/soc/codecs/si476x.c:219:9: error: invalid suffix "b111" on integer constant sound/soc/codecs/si476x.c:219:56: error: invalid suffix "b111" on integer constant So use hexadecimal constants (0x*) instead. Signed-off-by: James Hogan <james.hogan(a)imgtec.com> Cc: Andrey Smirnov <andrey.smirnov(a)convergeddevices.net> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/si476x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 721587c..73e205c 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -38,9 +38,9 @@ enum si476x_digital_io_output_format { SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT = 8, }; -#define SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK ((0b111 << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | \ - (0b111 << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)) -#define SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK (0b1111110) +#define SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK ((0x7 << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | \ + (0x7 << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)) +#define SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK (0x7e) enum si476x_daudio_formats { SI476X_DAUDIO_MODE_I2S = (0x0 << 1), -- 1.8.1.2

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[alsa-devel] tascam us800
by someone deleted 14 Jun '13

14 Jun '13

my tascam us800 works great in 3.8.0-21 except for one issue... no usb-mixer here is output of lsusb -v http://studio213.us/usbif.txt here is output of alsa-info.sh http://www.alsa-project.org/db/?f=336a5faa6233bc80a856089e657d80da1d32e081 alsamixer -c 1 shows amixer: Mixer hw:1 load error: Connection timed out dmesg when I do that shows [115350.348159] cannot get ctl value: req = 0x83, wValue = 0x101, wIndex = 0x1f00, type = 4 [115351.348158] cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0x1f00, type = 4 [115732.480250] cannot get ctl value: req = 0x83, wValue = 0x101, wIndex = 0x1f00, type = 4 [115733.480241] cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0x1f00, type = 4 plugging in the card shows [ 45.550396] usbcore: registered new interface driver snd-usb-audio in dmesg any info on how I might access this usb-mixer built into this card?

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