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

[alsa-devel] ALSA lib pcm_hw.c:1466:(_snd_pcm_hw_open) Unknown field chmap
by Alexander Stein 07 Oct '13

07 Oct '13

Hi, I get this error message from several games I got recently. There is either not sound at all or it is choppy. This was with alsa-lib-1.0.27.1. So I downgraded to 1.0.26 and everything was fine. I compiled alsa-lib-1.0.27.1 myself and started them using LD_LIBRARY_PATH: everything worked. So I assumed there is something in the config files which has changed. Comparing them I noticed some changes in /usr/share/alsa/cards/Audigy2.conf. Those lines were added: chmap [ "UNKNOWN" "FL,FR" ] chmap [ "UNKNOWN" "RL,RR" ] chmap [ "UNKNOWN" "FC,LFE" ] chmap [ "UNKNOWN" "SL,SR" ] See alsa-lib commits 5ba11d69dafbd4502469fd957b4580169779664c and ec7acfc408856cfe71a122e3b5387463d7b3d919. Updating to 1.0.27.1 again and commenting those lines out everything is ok now. So I'm wondering what is the reason for this error message? How to fix that? Thanks, Alexander PS: my sound card: 03:07.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value 1102:0008

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[alsa-devel] [PATCH - Ego Sys Waveterminal 192M 1/1] Update driver to support alsa changes happened since last driver commit. Add TLV support for dB scales. Add sampling rate control.
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..286be47 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -4,8 +4,8 @@ * Lowlevel functions for Ego Sys Waveterminal 192M * * Copyright (c) 2006 Guedez Clement <klem.dev(a)gmail.com> - * Some functions are taken from the Prodigy192 driver - * source + * Thanks to the developers of the Prodigy192 driver on which + * I based this driver * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -25,15 +25,25 @@ +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/slab.h> #include <sound/core.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +#include <sound/tlv.h> + +struct wtm_spec { + struct ak4114 *ak4114; + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* @@ -68,6 +78,35 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned char *change) +{ + unsigned char new, old; + int idx; + + /*stac9460 1*/ + for (idx = 0; idx < 7; idx++) { + if (change[idx]) { + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx] = (new != old); + if (change[idx]) + stac9460_put(ice, idx, new); + } + } + + /*stac9460 2*/ + for (idx = 0; idx < 3; idx++) { + if (change[idx + 7]) { + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx + 7] = (new != old); + if (change[idx + 7]) + stac9460_2_put(ice, idx, new); + } + } +} + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, @@ -96,10 +135,14 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char new, old; int id, idx; int change; + mutex_lock(&spec->mute_mutex); + + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; old = stac9460_get(ice, idx); @@ -127,6 +170,7 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, stac9460_2_put(ice, idx - 6, new); } } + mutex_unlock(&spec->mute_mutex); return change; } @@ -175,7 +219,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; - nvol = ucontrol->value.integer.value[0] & 0x7f; + nvol = ucontrol->value.integer.value[0]; tmp = stac9460_get(ice, idx); ovol = 0x7f - (tmp & 0x7f); change = (ovol != nvol); @@ -186,7 +230,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, } else { id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); idx = id + STAC946X_LF_VOLUME; - nvol = ucontrol->value.integer.value[0] & 0x7f; + nvol = ucontrol->value.integer.value[0]; if (id < 6) tmp = stac9460_get(ice, idx); else @@ -339,7 +383,24 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, * MIC / LINE switch fonction */ -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Line In", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items + - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -353,7 +414,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -369,7 +430,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) @@ -380,6 +441,46 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned char changed[10]; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + stac9460_dac_mute_all(ice, 0, changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, changed); + mutex_unlock(&spec->mute_mutex); +} + + + +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ @@ -390,15 +491,19 @@ static struct snd_kcontrol_new stac9640_controls[] = { .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +524,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,18 +543,40 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; /*INIT*/ +static void stac9460_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = entry->private_data; + int reg, val; + /* registers 0x0 - 0x14 */ + for (reg = 0; reg <= 0x15; reg++) { + val = stac9460_get(ice, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + +static void stac9460_proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ice->card, "stac9460_codec", &entry)) + snd_info_set_text_ops(entry, ice, stac9460_proc_regs_read); +} + + static int wtm_add_controls(struct snd_ice1712 *ice) { unsigned int i; @@ -458,46 +588,64 @@ static int wtm_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; } + stac9460_proc_init(ice); return 0; } static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; + /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x67, /* 49MHz crystal, mpu401, + * spdif-in+ 1 stereo ADC, + * 4 stereo DACs + */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /* out-en, out-int, spdif-in */ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, /* GPIO20: 0 = CD drive dig. input + * passthrough, + * 1 = SPDIF-OUT from ice1724 + */ }; -- 1.7.10.4

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[alsa-devel] [PATCH 4/4] ESI W192M: Add text "Line in/Mic".
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add text "Line in/Mic" for selecting input gain state for Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index cf2b6a5..464cca5 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -377,8 +377,23 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Line In", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items + - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -392,7 +407,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -408,7 +423,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) -- 1.8.4.rc3

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[alsa-devel] [PATCH 2/4] ESI W192M: Add TLV Support.
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add TLV support for control value in dB scale for Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index b6b650c..d559916 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <sound/tlv.h> #include "ice1712.h" #include "envy24ht.h" @@ -380,6 +381,10 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ @@ -394,11 +399,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +427,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +446,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; -- 1.8.4.rc3

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[alsa-devel] [PING!] Re: HDMI audio is not working on my new desktop
by Alexander E. Patrakov 05 Oct '13

05 Oct '13

2013/8/14 Alexander E. Patrakov <patrakov(a)gmail.com>: > Hello. > > I built a new desktop PC based on the Gigabyte H87N-WIFI motherboard. > It has one DVI and two HDMI outputs. However, I cannot get any sound > over HDMI. The same OS image works perfectly on my laptop that also > has an HDMI output. > > This is not a pulseaudio problem, as I don't get any sound even if I > test with this command: > > pasuspender -- speaker-test -c2 -D hdmi:0,2 # or, for that matter, 0,0 or 0,1 > > Alsa-info.sh has uploaded its output to > http://www.alsa-project.org/db/?f=1ad7192a15ff4a70b5ddb2579476dbb98adbd929 > > Please note this error in dmesg, it may be relevant: > > [ 1.803595] hda_codec: invalid CONNECT_LIST verb 5[1]:0 > [ 1.803719] hda_codec: invalid CONNECT_LIST verb 7[1]:0 As nobody replied, I have converted this into a formal bug report: https://bugzilla.kernel.org/show_bug.cgi?id=60769 Haswell people, _please_ comment on the bug. -- Alexander E. Patrakov

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[alsa-devel] Microphone does not longer work on Lenovo N581
by Manuel Reimer 05 Oct '13

05 Oct '13

Hello, I'm using Archlinux and so usually run always on the latest kernel. But unfortunately somewhere with Kernel 3.9 my Intel sound got broken. Playback works, but no microphone is found. So I keep downgrading to 3.8.8 for some months, now. Can someone please help me with finding a solution for this? Is it really possible that the alsa driver "snd_had_intel" was broken between 3.8 and 3.9 and noone found this bug, so far? Thank you very much in advance Greetings, Manuel

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[alsa-devel] [PATCH 1/6] ASoC: cs4271: Include linux/of.h header
by Sachin Kamat 04 Oct '13

04 Oct '13

'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/cs4271.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a20f1bb..f6e9534 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -25,6 +25,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> #include <sound/pcm.h> -- 1.7.9.5

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[alsa-devel] us122l
by James Miller 04 Oct '13

04 Oct '13

please ignore my previous email -- James Stewart Miller Bsc(hons) Psych.

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[alsa-devel] tascam us122l
by James Miller 04 Oct '13

04 Oct '13

Hi, I've just checked the kernel source for the tascam us122l (us122l.c us122l.h) and it has changed completely since the 3.4 kernel. I'm having a stack trace problem when trying to use jack with us122l. The card is recognised and I am using the .asoundrc from pub.briata.org as I always have done. When the us122l.c got broken (around about kernel version 3.6 to 3.9) I was able to get it working by recompiling the kernel after amending the us122l.c source code by changing a flag that was pushed in to a vm struct. In the new code I can't find any reference to the vm operation and it seems that the code has changed completely, I am going to try and revert to the 3.4 us122l.c to see if that will work, but I'd be grateful if someone could look into this. Thanks James -- James Stewart Miller Bsc(hons) Psych.

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Re: [alsa-devel] [PATCH] sound: usb: support wireless devices in snd_usb_parse_datainterval
by Greg KH 04 Oct '13

04 Oct '13

On Tue, Oct 01, 2013 at 02:32:57PM -0500, Thomas Pugliese wrote: > Hi, > This patch adds support for dev speed USB_SPEED_WIRELESS in > snd_usb_parse_datainterval which allows the usb sound core to create > ISO urbs with the correct number and size of buffers. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Acked-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org>

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Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Greg KH 04 Oct '13

04 Oct '13

On Tue, Oct 01, 2013 at 02:32:15PM -0500, Thomas Pugliese wrote: > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Acked-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org>

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[alsa-devel] snd_device_name_hint segfaults
by Forest Bond 03 Oct '13

03 Oct '13

Hi, I'm seeing segfaults with VLC similar to those reported here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/059168.ht… This is on Ubuntu 12.04 with the latest alsa-lib package (1.0.25-1ubuntu10.2) and the following patches also applied: * 2cfc8b9b44a8e493c41b3d63d5a00b306a18a5ed snd_pcm_direct_parse_open_conf(): use thread-safe getgrnam_r() * 44c1a623dd1fc9e831616b663bebc54ca98df994 Add snd_lib_error_set_local() to install a thread-local error handler. * 25dbb102810b31c02358904d70d53c960fb0a10e snd_device_name_hint(): do not change the global error handler. * f49b2dc522a2564315c76d075203b15a39941e8a snd_device_name_hint(): do not use global snd_config. * 7f2b2c8c1650a1883b48abfcdb455138943854f9 conf: Fix a memory access violation resulting from improper error propogation So I think there are still issues when calling snd_device_name_hint(-1, ...). Here is the backtrace: #0 0x003f0e78 in __strcpy_chk () from /lib/i386-linux-gnu/libc.so.6 #1 0x0625084b in strcpy (__src=0x0, __dest=0xb6025e80 "pcm.(null)") at /usr/include/i386-li nux-gnu/bits/string3.h:105 #2 try_config (config=0xaf1c4df0, list=0xb52fe804, base=0x323e962 "pcm", name=0x0) at nameh int.c:243 #3 0x06251bc5 in add_software_devices (list=0xb52fe804, config=0xaf1c4df0) at namehint.c:51 2 #4 snd_device_name_hint (card=-1, iface=<optimized out>, hints=0xb52fe90c) at namehint.c:59 1 #5 0x0323c9bb in GetDevices (obj=0xb6271700, item=0x0, prefs_dev=<optimized out>) at alsa.c :721 #6 0x0323d4c0 in Probe (dev=0xb6011df8 "default", obj=0xb6271700) at alsa.c:161 #7 Open (obj=0xb6271700) at alsa.c:539 #8 0x0a6c73d0 in generic_start (func=0x323cd10, ap=0xb52fed98 "\273") at modules/modules.c:422 #9 0x0a6c7a3a in vlc_module_load (p_this=0xb6271700, psz_capability=0xa70b959 "audio output", psz_name=<optimized out>, b_strict=false, probe=0xa6c73c0 <generic_start>) at modules/modules.c:347 #10 0x0a6c8002 in module_need (obj=0xb6271700, cap=0xa70b959 "audio output", name=0xa70d2bd "$aout", strict=false) at modules/modules.c:437 #11 0x0a6b1242 in aout_OutputNew (p_aout=0xb6271700, p_format=0xb52feeac) at audio_output/output.c:59 #12 0x0a6ad77f in aout_DecNew (p_aout=0xb6271700, p_format=0xb52feeac, p_replay_gain=0xaef02a14, p_request_vout=0xb52feed0) at audio_output/dec.c:112 #13 0x0a67066e in aout_new_buffer (p_dec=0xaef02870, i_samples=1024) at input/decoder.c:2286 #14 0x0a672690 in decoder_NewAudioBuffer (p_decoder=0xaef02870, i_size=1024) at input/decoder.c:213 #15 0x04032f8c in DecodeBlock (p_dec=0xaef02870, pp_block=0xb52ff05c) at faad.c:464 #16 0x0a670891 in DecoderDecodeAudio (p_dec=0xaef02870, p_block=0xaef03000) at input/decoder.c:1291 #17 0x0a6716f5 in DecoderProcessAudio (b_flush=false, p_block=0xaef03000, p_dec=0xaef02870) at input/decoder.c:1936 #18 DecoderProcess (p_dec=0xaef02870, p_block=<optimized out>) at input/decoder.c:2059 #19 0x0a671959 in DecoderThread (p_data=0xaef02870) at input/decoder.c:938 #20 0x001fbd4c in start_thread () from /lib/i386-linux-gnu/libpthread.so.0 #21 0x003ddace in clone () from /lib/i386-linux-gnu/libc.so.6 As a work around I will probably patch VLC to not call snd_device_name_hint this way. I see this has been done for other applications: http://code.google.com/p/chromium/issues/detail?id=95797 http://code.mythtv.org/trac/ticket/10809 But I wanted to try diagnosing the issue first. Thoughts? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.rapidrollout.com

4 6

[alsa-devel] [PATCH] ASoC: omap: Fix incorrect ARM dependency
by Jarkko Nikula 03 Oct '13

03 Oct '13

Commit b0e0a4d ("ASoC: omap: Enable COMPILE_TEST build for DT platforms") added two incorrect CONFIG_ARCH_ARM dependencies making impossible to select audio support for Nokia RX-51. Fix this by using correct CONFIG_ARM. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- This is a fix for 3.12-rc. I.e. not for next merge window. --- sound/soc/omap/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index daa78a0..4a07f71 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST) + depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST) select SND_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) + depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: fix start frame mask
by Daniel Mack 03 Oct '13

03 Oct '13

This patch is based on a change proposed by Guido Aulisi in a discussion on LAU which can be followed here: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 Cc: fzu(a)wemgehoertderstaat.de Original-code-by: Guido Aulisi <guido.aulisi(a)gmail.com> Tested-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..cd16fcc 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -328,7 +328,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) + if (likely((urb->start_frame & 0x03FF) == (usX2Y->wait_iso_frame & 0x03FF))) subs->completed_urb = urb; else { usX2Y_error_sequence(usX2Y, subs, urb); -- 1.8.3.1

3 4

[alsa-devel] [PATCH] ASoC: davinci-mcasp: add support for suspend and resume
by Daniel Mack 03 Oct '13

03 Oct '13

When the system returns from suspend, it looses its configuration. Most of it is restored by running a normal audio stream startup, but the DAI format is left unset as that's configured on the audio device creation. Hence, it suffices here to care for the registers which are touched by davinci_mcasp_set_dai_fmt() and restore them when the system is resumed. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/davinci/davinci-mcasp.c | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 12 ++++++++++++ 2 files changed, 51 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 591f853..5023da7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1272,12 +1272,51 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int davinci_mcasp_suspend(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); + a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); + a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); + a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); + a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); + a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); + a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + + return 0; +} + +static int davinci_mcasp_resume(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl); + mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir); + + return 0; +} +#endif + +SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops, + davinci_mcasp_suspend, + davinci_mcasp_resume); + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a9ac0c1..a2e27e1 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -43,6 +43,18 @@ struct davinci_audio_dev { /* McASP FIFO related */ u8 txnumevt; u8 rxnumevt; + +#ifdef CONFIG_PM_SLEEP + struct { + u32 txfmtctl; + u32 rxfmtctl; + u32 txfmt; + u32 rxfmt; + u32 aclkxctl; + u32 aclkrctl; + u32 pdir; + } context; +#endif }; #endif /* DAVINCI_MCASP_H */ -- 1.8.3.1

3 2

[alsa-devel] [PATCH 1/3] ASoC: tas5086: move two variables into private struct
by Daniel Mack 03 Oct '13

03 Oct '13

We need to access the charge_period and start_mid_z values from other places later, so move them to the private struct. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/tas5086.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88..31b5868 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -244,6 +244,8 @@ struct tas5086_private { unsigned int mclk, sclk; unsigned int format; bool deemph; + unsigned int charge_period; + unsigned int pwm_start_mid_z; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -720,13 +722,15 @@ static const int tas5086_charge_period[] = { static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - int charge_period = 1300000; /* hardware default is 1300 ms */ - u8 pwm_start_mid_z = 0; int i, ret; + priv->pwm_start_mid_z = 0; + priv->charge_period = 1300000; /* hardware default is 1300 ms */ + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; - of_property_read_u32(of_node, "ti,charge-period", &charge_period); + of_property_read_u32(of_node, "ti,charge-period", + &priv->charge_period); for (i = 0; i < 6; i++) { char name[25]; @@ -735,7 +739,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) "ti,mid-z-channel-%d", i + 1); if (of_get_property(of_node, name, NULL) != NULL) - pwm_start_mid_z |= 1 << i; + priv->pwm_start_mid_z |= 1 << i; } } @@ -744,25 +748,25 @@ static int tas5086_probe(struct snd_soc_codec *codec) * configure 'part 1' of the PWM starts to use Mid-Z, and tell * all configured mid-z channels to start start under 'part 1'. */ - if (pwm_start_mid_z) + if (priv->pwm_start_mid_z) regmap_write(priv->regmap, TAS5086_PWM_START, TAS5086_PWM_START_MIDZ_FOR_START_1 | - pwm_start_mid_z); + priv->pwm_start_mid_z); /* lookup and set split-capacitor charge period */ - if (charge_period == 0) { + if (priv->charge_period == 0) { regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); } else { i = index_in_array(tas5086_charge_period, ARRAY_SIZE(tas5086_charge_period), - charge_period); + priv->charge_period); if (i >= 0) regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, i + 0x08); else dev_warn(codec->dev, "Invalid split-cap charge period of %d ns.\n", - charge_period); + priv->charge_period); } /* enable factory trim */ -- 1.8.3.1

2 3

[alsa-devel] [PATCH] ARM: omap: edma: add suspend suspend/resume hooks
by Daniel Mack 03 Oct '13

03 Oct '13

This patch adds support for edma driver. I know the intermediate code in arch/arm/common is going to be purged eventually when the real dma-engine driver lands, so I don't know whether it's reasonable to include my patch after all. It might just server as template for the suspend/resume functions in the final driver, or maybe anyone wants to include that intermediate hack for the time being. Successfully tested on an AM335x board with davinci-mcasp enabled audio. A patch to make the latter work after suspend has been posted on alsa-devel. Thanks, Dainel Daniel Mack (1): ARM: omap: edma: add suspend suspend/resume hooks arch/arm/common/edma.c | 133 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 133 insertions(+) -- 1.8.3.1

2 2

[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: remove bogus frame check
by Daniel Mack 02 Oct '13

02 Oct '13

The frame check in i_usX2Y_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This code removes the check code entirely. Cc: fzu(a)wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 22 +++------------------- 1 file changed, 3 insertions(+), 19 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..6234a51 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; -- 1.8.3.1

2 2

[alsa-devel] [PATCH] amixer: fix indentation when printing container TLV contents
by Anssi Hannula 02 Oct '13

02 Oct '13

decode_tlv() adds indentation in the beginning, with the expectation that the TLV will be printed on the line afterwards in a switch-case. However, in the case of a container TLV the switch-case simply adds another level of indentation and calls decode_tlv() for the inner TLVs. This causes the first inner TLV to be printed with too much indentation and double '|'. Fix that by printing "container" and a newline for container TLVs, so that the result is as follows: : values=0,0,0,0,0,0,0,0 | container | chmap-variable=FL,FR | chmap-variable=FL,FR,LFE | chmap-variable=FL,FR,FC Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- amixer/amixer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/amixer/amixer.c b/amixer/amixer.c index b83b0c3..cf82892 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -460,6 +460,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ } switch (type) { case SND_CTL_TLVT_CONTAINER: + printf("container\n"); size += sizeof(unsigned int) -1; size /= sizeof(unsigned int); while (idx < size) { -- 1.8.1.5

2 1

[alsa-devel] alsa-lib 1.0.27.2 and musl
by Waldemar Brodkorb 02 Oct '13

02 Oct '13

Hi Developers, I am trying to cross-compile alsa-lib with a musl¹ libc based toolchain. Unfortunately src/compat/hsearch_r.c is failing to compile, because of some glibc specific macros (like _P) normally defined in sys/cdefs.h. After grepping through the code I can not find any consumer for any functions implemented in hsearch_r.c. When I just comment out the code, everything compiles fine. Seems that that usage of hsearch functions was removed in the code long time ago. commit 273d115de05574251bdd661747ecb68449a5cf1d Author: Jaroslav Kysela <perex(a)perex.cz> Date: Sat Jul 15 10:20:32 2000 +0000 Removed old mixer API functions. Added simple control layer (experimental). Can it just be removed? thanks for any advice or comment. best regards Waldemar ¹ http://www.musl-libc.org/

3 2

Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Daniel Mack 02 Oct '13

02 Oct '13

On 01.10.2013 21:32, Thomas Pugliese wrote: > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. Interesting. What device are you testing this with? Daniel > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> > --- > sound/usb/card.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 64952e2..ef76aa0 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -349,6 +349,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, > case USB_SPEED_LOW: > case USB_SPEED_FULL: > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > break; > default: >

2 1

[alsa-devel] snd-usb-usx2y (Tascam US-122) sequence errors
by Dr Nicholas J Bailey 02 Oct '13

02 Oct '13

I'm on debian jessie, Linux arial 3.10-3-686-pae #1 SMP Debian 3.10.11-1 (2013-09-10) i686 GNU/Linux with a Tascam us-122 that has recently stopped working. When I try to capture with anything, there is a few hundred ms of audio then the capture process stalls and the following dmesg appears: [ 462.596296] Sequence Error!(hcd_frame=0 ep=8in;wait=1024,frame=0). [ 462.596296] Most probably some urb of usb-frame 1024 is still missing. [ 462.596296] Cause could be too long delays in usb-hcd interrupt handling. [ 477.959230] Sequence Error!(hcd_frame=0 ep=8in;wait=1024,frame=0). [ 477.959230] Most probably some urb of usb-frame 1024 is still missing. [ 477.959230] Cause could be too long delays in usb-hcd interrupt handling. Before this kernel, the box worked OK. The only think that leads me to believe it might be my fault is that for a very long time, I've had to plug the box in: [ 445.529271] usb 2-1.6: new full-speed USB device number 5 using ehci-pci [ 445.622076] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8006 [ 445.622086] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 then unplug it and plug it in again: [ 445.972743] usb 2-1.6: USB disconnect, device number 5 [ 447.621519] usb 2-1.6: new full-speed USB device number 6 using ehci-pci [ 447.714462] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8007 [ 447.714471] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 [ 447.834852] usbcore: registered new interface driver snd-usb-usx2y [ 455.194314] usb 2-1.6: USB disconnect, device number 6 [ 457.487433] usb 2-1.6: new full-speed USB device number 7 using ehci-pci [ 457.579960] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8006 [ 457.579969] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 to get the firmware to load [ 457.888884] usb 2-1.6: USB disconnect, device number 7 [ 459.379782] usb 2-1.6: new full-speed USB device number 8 using ehci-pci [ 459.472516] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8007 [ 459.472526] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 (green USB light comes on here). I'm not sure about how to start debugging this; there are many things that could be wrong. If it will help, I'll try and file a bug report; if not please tell me to shut up (prefereably after pointing me at the solution :) ) Nick/.

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Re: [alsa-devel] Realtek ALC286 SONY Vaio Pro 13 Ubuntu 13.10
by David Henningsson 02 Oct '13

02 Oct '13

On 10/02/2013 04:13 AM, Yuval Levy wrote: > On 10/01/2013 02:28 AM, David Henningsson wrote: >> Ok. If things work without extra drivers from Sony or Realtek, then it >> should be possible to get things working in Linux too without extra >> information from these parties... > > I have added this information to the bug report > <https://bugs.launchpad.net/bugs/1228689> > where I also post my comments so that they are more persistent and > available to other users with similar issues. > > >> Ok - given that 0x21 works as expected you should only need to do this >> for pin 0x18. > > All my testing was with doing this for both pins. I only noticed now > that I do not need to do this for pin 0x18. I will try to o this only > for pin 0x21 and reboot tomorrow. Ok, cool. Notice that I wrote "do this for pin 0x18 only, not for pin 0x21", where as you replied "I'll do this for pin 0x21 only, not for pin 0x18". 0x18 is for input switching, 0x21 is for output switching. If you remove jack detection, like you did, you will need to do this switching manually, e g in pavucontrol or gnome sound settings. That is, when you plug your headphone in, you need to manually select "Headphone" too, if you don't, the result will probably be that you hear sound in both headphones and speakers. Anyway, let's guess that the high pitched sound, as well as the lack of headset microphone input, is because your headset is OMTP and that the Vaio only supports CTIA. That's my *guess*, but it could be wrong, and it would be good to have this confirmed with a CTIA headset. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] Problems with i.MX233 driver in 3.12rc2
by Trevor Page 01 Oct '13

01 Oct '13

Hello, First post to this list - I hope this is the appropriate place to ask for this assistance. I'm using the Olinuxino Maxi board with the i.MX233 processor. I'm using the 3.12rc2 kernel at the moment because this has recently introduced the drivers for this processor's audio. I understand that the drivers for this chip's audio are: sound/soc/codecs/mxs-builtin-codec.c sound/soc/mxs/mxs-builtin-audio.c sound/soc/mxs/mxs-builtin-dai.c sound/soc/mxs/mxs-builtin-pcm.c My application for the board is a simple C program that uses the Linphone library to make VOIP calls. The problem I am having is that audio usually works (playback and capture) perfectly fine from boot, but at some point it may suddenly go 'dead'. At this point, there is no longer any audio from the headphone socket. I have not yet pinpointed exactly what it takes to trigger the fault, but I have seen it happen once when the call was put on hold while a Linphone call was in progress. Once the audio output is dead, these are some further observations: - When a Linphone call is received, I still hear a click from the headphone output when the call ends. - I have confirmed that the DAC / HP channels are turned up and unmuted. - If I run speaker-test when the audio is working, speaker-test periodically prints time duration information (presumably on each pass of its internal loop), like "0 - Front Left", "Time per period ...". When the audio is dead, speaker-test just freezes on "0 - Front Left" forever, as if some call is either blocking or returning to say it is busy. This might hopefully therefore be a useful point to start debugging. - I have only ever managed to restore audio using a reboot. I would like to have a go at debugging this. My problem though is that although I'm experienced with C and microprocessor programming, compiling and debugging embedded Linux is new ground for me. I'd be grateful for any pointers on things I could try in order to gain an insight into what's going wrong, and tips on how I could debug the drivers. So far my plan for today is to build the kernel again with the i.MX233 audio drivers as external modules so that I can hopefully play with them a bit and compile them separately. (I assume the development and debug process for audio drivers like this is to compile them as .ko modules and modprobe them in). However without having any prior knowledge of Linux drivers or the ALSA / ASoC framework I'm not sure if I'm going to get very far. Regards, Trevor.

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[alsa-devel] [PATCH] ASoC: kirkwood: fix compilation warning in kirkwood_dma_open
by Vladimir Murzin 01 Oct '13

01 Oct '13

writel() supposes the first argument of type unsigned int. This fix the warning: sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_open': sound/soc/kirkwood/kirkwood-dma.c:164:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] Signed-off-by: Vladimir Murzin <murzin.v(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b238434..1135c30 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -161,7 +161,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) * Enable Error interrupts. We're only ack'ing them but * it's useful for diagnostics */ - writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); + writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } dram = mv_mbus_dram_info(); -- 1.8.1.5

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