July 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] [RFC 9/13] Intel SST sound card vendor0 module
by Vinod Koul 03 Jul '09

03 Jul '09

This adds the support for vendor PMICs for widget control and configurations. The MID platform supports three different vendor implementation of PMIC. The sound card, ie all analog components like DAC, ADC, mixer settings are different for three vendors. This patch implements these settings for one of the vendor. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: R Dharageswari <dharageswari.r(a)intel.com> new file: sound/pci/sst/intelmid_v0_control.c --- sound/pci/sst/intelmid_v0_control.c | 529 +++++++++++++++++++++++++++++++++++ 1 files changed, 529 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intelmid_v0_control.c diff --git a/sound/pci/sst/intelmid_v0_control.c b/sound/pci/sst/intelmid_v0_control.c new file mode 100644 index 0000000..cbab23a --- /dev/null +++ b/sound/pci/sst/intelmid_v0_control.c @@ -0,0 +1,529 @@ +/* + * intelmid_v0_control.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * R Dharageswari <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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 + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file contains the control operations of vendor 1 + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/file.h> +#include <linux/fcntl.h> +#include <linux/uaccess.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include <linux/workqueue.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#include "intelmid_snd_control.h" +#include <asm/ipc_defs.h> + +enum _reg_v1 { + VOICEPORT1 = 0x180, + VOICEPORT2 = 0x181, + AUDIOPORT1 = 0x182, + AUDIOPORT2 = 0x183, + MISCVOICECTRL = 0x184, + MISCAUDCTRL = 0x185, + DMICCTRL1 = 0x186, + DMICCTRL2 = 0x187, + MICCTRL = 0x188, + MICLICTRL1 = 0x189, + MICLICTRL2 = 0x18A, + MICLICTRL3 = 0x18B, + VOICEDACCTRL1 = 0x18C, + STEREOADCCTRL = 0x18D, + AUD15 = 0x18E, + AUD16 = 0x18F, + AUD17 = 0x190, + AUD18 = 0x191, + RMIXOUTSEL = 0x192, + ANALOGLBR = 0x193, + ANALOGLBL = 0x194, + AUD22 = 0x195, + POWERCTRL2 = 0x196, + HEADSETDETECTINT = 0x197, + HEADSETDETECTINTMASK = 0x198, + TRIMENABLE = 0x199, +}; + +int rev_id = 0x20; + +int fs_init_card(void) +{ + + struct sc_reg_access sc_access[17] = {{0,},}; + int retval = 0; + + sst_dbg("FS-B0 initializing card...\n"); + + /*send message to SCU to get the PMIC vendor-*/ + sc_access[0].reg_addr = 0; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + rev_id = sc_access[0].value; + sst_dbg("value of reg 0:0x%02x \n", rev_id); + + + sc_access[0].reg_addr = 0x184; + sc_access[0].value = 0x1c; + sc_access[1].reg_addr = 0x185; + if (rev_id == 0x20) + sc_access[1].value = 0x4A;/*FS-B0*/ + else + sc_access[1].value = 0x48; + + sc_access[2].reg_addr = 0x187 ; + sc_access[2].value = 0x39; + sc_access[3].reg_addr = 0x196 ; + sc_access[3].value = 0x7F; + sc_access[4].reg_addr = 0x180; + sc_access[4].value = 0xF5; + sc_access[5].reg_addr = 0x181; + sc_access[5].value = 0x26; + sc_access[6].reg_addr = 0x182; + if (rev_id == 0x20) + sc_access[6].value = 0xE8;/*FS-B0*/ + else + sc_access[6].value = 0xD8; + sc_access[7].reg_addr = 0x183; + sc_access[7].value = 0x19; + sc_access[8].reg_addr = 0x186; + sc_access[8].value = 0x79; + sc_access[9].reg_addr = 0x189; + sc_access[9].value = 0xB9; + sc_access[10].reg_addr = 0x18a; + sc_access[10].value = 0xB9; + sc_access[11].reg_addr = 0x18e; + sc_access[11].value = 0x39; + sc_access[12].reg_addr = 0x18f; + sc_access[12].value = 0x39; + sc_access[13].reg_addr = 0x190; + sc_access[13].value = 0x39; + sc_access[14].reg_addr = 0x191; + sc_access[14].value = 0x11; + sc_access[15].reg_addr = 0x192; + sc_access[15].value = 0x0E; + sc_access[16].reg_addr = 0x195; + sc_access[16].value = 0x06; + + retval = sst_sc_reg_access(sc_access, PMIC_WRITE, 17); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + + sst_dbg("Initialisation complete!!\n"); + return 0; +} + + +int fs_init_capture_card(void) +{ + return 0; +} + +void fs_set_pcm_params(int sfreq, int word_size) +{ + int config1 = 0, retval = 0; + struct sc_reg_access sc_access[2]; + + switch (sfreq) { + case 8000: + config1 = 0x10; + break; + case 11025: + config1 = 0x11; + break; + case 12000: + config1 = 0x12; + break; + case 16000: + config1 = 0x13; + break; + case 22050: + config1 = 0x14; + break; + case 24000: + config1 = 0x15; + break; + case 26000: + config1 = 0x16; + break; + case 32000: + config1 = 0x17; + break; + case 44100: + config1 = 0x18; + break; + case 48000: + config1 = 0x19; + break; + } + + sst_dbg("sfreq:%d, Register value = %x\n", sfreq, config1); + + sc_access[0].reg_addr = AUDIOPORT2; + sc_access[0].value = config1; + + sst_dbg("Setting word size ...\n"); + switch (word_size) { + case 16: + if (rev_id == 0x20) + config1 = 0xE8; /* FS B0 */ + else + config1 = 0xD8; /* FS A0 */ + break; + case 24: + if (rev_id == 0x20) + config1 = 0xEB; /* FS B0 */ + else + config1 = 0xDB; /* FS A0 */ + break; + } /* switch (word_size) */ + + sst_dbg("rev_id:0x%2x, pcm_wd_sz:%d, Register_value= %x\n", + rev_id, word_size, config1); + + sc_access[1].reg_addr = AUDIOPORT1; + sc_access[1].value = config1; + retval = sst_sc_reg_access(sc_access, PMIC_WRITE, 2); + if (0 != retval) + sst_err("pmic commn failed \n"); + return; +} + +int fs_set_selected_input_dev(int value) +{ + + return 0; +} + +int fs_set_selected_output_dev(int value) +{ + struct sc_reg_access sc_access[2]; + int retval = 0; + + sc_access[0].reg_addr = AUD22; + sc_access[0].mask = MASK6|MASK7|MASK4|MASK3|MASK2|MASK1|MASK0; + + switch (value) { + case STEREO_HEADPHONE: + sc_access[0].value = VALUE2|VALUE1; + sc_access[0].mask = MASK2|MASK1; + sc_access[1].reg_addr = 0x192; + sc_access[1].mask = MASK2; + sc_access[1].value = 1; + break; + + case INTERNAL_SPKR: + break; + + case MONO_HS: + sc_access[0].value = VALUE7; + sc_access[0].mask = MASK7; + sc_access[1].reg_addr = 0x192; + sc_access[1].mask = MASK2; + sc_access[1].value = 0; + break; + } + sst_dbg("dev_id:0x%2x reg:0x%2x value:0x%2x\n", + value, sc_access[0].reg_addr, sc_access[0].value); + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 2); + if (0 != retval) + sst_err("pmic commn failed \n"); + + return retval; +} + +int fs_set_mute(int dev_id, int value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + int retval = 0; + + sst_dbg("dev_id:0x%2x value:0x%2x", dev_id, value); + switch (dev_id) { + case PMIC_SND_INPUT_MUTE_LINE_IN: + break; + + case PMIC_SND_INPUT_MUTE_MIC1: + break; + + case PMIC_SND_INPUT_MUTE_MIC2: + break; + + case PMIC_SND_INPUT_MUTE_DMIC: + break; + + case PMIC_SND_LEFT_HP_MUTE: + sc_access[0].reg_addr = AUD18; + sc_access[0].mask = MASK2; + if (value == MUTE) + sc_access[0].value = VALUE2; + else + sc_access[0].value = 0x00; + break; + + case PMIC_SND_LEFT_SPEAKER_MUTE: + break; + + case PMIC_SND_RIGHT_SPEAKER_MUTE: + break; + + case PMIC_SND_RIGHT_HP_MUTE: + sc_access[0].reg_addr = AUD18; + sc_access[0].mask = MASK1; + if (value == MUTE) + sc_access[0].value = VALUE1; + else + sc_access[0].value = 0x00; + break; + + + case PMIC_SND_MONO_EARPIECE_MUTE: + sc_access[0].reg_addr = AUD18; + sc_access[0].mask = MASK0; + if (value == MUTE) + sc_access[0].value = VALUE0; + else + sc_access[0].value = 0x00; + break; + default: + sst_err("rcvd illegal request"); + /*Illegal case*/ + return -EINVAL; + } + + sst_dbg("dev_id:0x%02x arg_value:%d reg:0x%02x value:0x%02x\n", dev_id, + value, sc_access[0].reg_addr, sc_access[0].value); + + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 1); + if (0 != retval) + sst_err("pmic commn failed \n"); + + return retval; +} + +int fs_set_vol(int dev_id, int value) +{ + struct sc_reg_access sc_access[2] = {{0,},}; + int retval = 0; + + switch (dev_id) { + case PMIC_SND_INPUT_VOL_MIC1: + break; + + case PMIC_SND_INPUT_VOL_MIC2: + break; + + case PMIC_SND_INPUT_VOL_DMIC: + break; + + case PMIC_SND_INPUT_VOL_RIGHT_LINE_IN: + break; + + case PMIC_SND_INPUT_VOL_LEFT_LINE_IN: + break; + + case PMIC_SND_LEFT_SPEAKER_VOL: + break; + + case PMIC_SND_RIGHT_SPEAKER_VOL: + break; + + case PMIC_SND_MONO_EARPIECE_VOL: + sc_access[0].reg_addr = AUD16; + sc_access[0].mask = MASK5|MASK4|MASK3|MASK2|MASK1|MASK0; + sc_access[0].value = (value - MIN_VOL_PMIC_VENDOR0); + sc_access[1].reg_addr = AUD17; + sc_access[1].mask = MASK5|MASK4|MASK3|MASK2|MASK1|MASK0; + sc_access[1].value = (value - MIN_VOL_PMIC_VENDOR0); + + sst_dbg("dev_id:0x%02x arg_value:%d reg:0x%02x value:0x%02x\n", + dev_id, value, sc_access[0].reg_addr, + sc_access[0].value); + sst_dbg("reg2:0x%02x value2:0x%02x\n", + sc_access[1].reg_addr, sc_access[1].value); + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 2); + if (0 != retval) + sst_err("pmic commn failed \n"); + return retval; + + case PMIC_SND_LEFT_HP_VOL: + sc_access[0].reg_addr = AUD16; + sc_access[0].mask = MASK5|MASK4|MASK3|MASK2|MASK1|MASK0; + sc_access[0].value = (value - MIN_VOL_PMIC_VENDOR0); + break; + + case PMIC_SND_RIGHT_HP_VOL: + sc_access[0].reg_addr = AUD17; + sc_access[0].mask = MASK5|MASK4|MASK3|MASK2|MASK1|MASK0; + sc_access[0].value = (value - MIN_VOL_PMIC_VENDOR0); + break; + + default: + sst_err("rcvd illegal request"); + /*Illegal case*/ + return -EINVAL; + } + + sst_dbg("dev_id:0x%2x arg_value:%d reg:0x%2x value:0x%2x\n", dev_id, + value, sc_access[0].reg_addr, sc_access[0].value); + + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 1); + if (0 != retval) + sst_err("pmic commn failed \n"); + return retval; +} + +int fs_get_selected_input_dev(int *value) +{ + return 0; +} + +int fs_get_selected_output_dev(int *value) +{ + return 0; +} + +int fs_get_mute(int dev_id, int *value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + + int retval = 0, temp_value = 0, mask = 0; + + switch (dev_id) { + case PMIC_SND_INPUT_MUTE_LINE_IN: + break; + case PMIC_SND_INPUT_MUTE_MIC1: + break; + case PMIC_SND_INPUT_MUTE_MIC2: + break; + case PMIC_SND_INPUT_MUTE_DMIC: + break; + case PMIC_SND_LEFT_SPEAKER_MUTE: + break; + case PMIC_SND_RIGHT_SPEAKER_MUTE: + break; + case PMIC_SND_LEFT_HP_MUTE: + mask = MASK2; + break; + case PMIC_SND_RIGHT_HP_MUTE: + mask = MASK1; + break; + case PMIC_SND_MONO_EARPIECE_MUTE: + mask = MASK0; + break; + /*line out mute not supported*/ + default: + sst_err("rcvd illegal request"); + /*Illegal case*/ + return -EINVAL; + } + sc_access[0].reg_addr = AUD18; + + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value & mask; + if (0 == temp_value) + *value = UNMUTE; + else + *value = MUTE; + return retval; +} + +int fs_get_vol(int dev_id, int *value) +{ + struct sc_reg_access sc_access[2] = {{0,},}; + int retval = 0, mask = 0, temp_value = 0; + + switch (dev_id) { + case PMIC_SND_INPUT_VOL_MIC1: + break; + + case PMIC_SND_INPUT_VOL_MIC2: + break; + + case PMIC_SND_INPUT_VOL_DMIC: + break; + + case PMIC_SND_INPUT_VOL_RIGHT_LINE_IN: + break; + + case PMIC_SND_INPUT_VOL_LEFT_LINE_IN: + break; + + case PMIC_SND_LEFT_SPEAKER_VOL: + break; + + case PMIC_SND_RIGHT_SPEAKER_VOL: + break; + + case PMIC_SND_LEFT_HP_VOL: + sc_access[0].reg_addr = AUD16; + mask = (MASK5|MASK4|MASK3|MASK2|MASK1|MASK0); + break; + case PMIC_SND_RIGHT_HP_VOL: + sc_access[0].reg_addr = AUD17; + mask = (MASK5|MASK4|MASK3|MASK2|MASK1|MASK0); + break; + case PMIC_SND_MONO_EARPIECE_VOL: + sc_access[0].reg_addr = AUD17; + mask = (MASK5|MASK4|MASK3|MASK2|MASK1|MASK0); + break; + default: + sst_err("rcvd illegal request"); + /*Illegal case*/ + return -EINVAL; + } + + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + sst_err("pmic commn failed \n"); + return retval; + } + + temp_value = sc_access[0].value & mask; + *value = temp_value + MIN_VOL_PMIC_VENDOR0; + + return retval; +} /* fs_get_vol */ -- 1.5.4.5

1 0

[alsa-devel] RFC: Jack-detection control elements on HD-audio
by Takashi Iwai 03 Jul '09

03 Jul '09

Hi, yet another thing I've worked recently on is a bit more generalized framework for jack-sense reporting for HD-audio. This supersedes the existing jack-sense reporting via the input layer. In addition, it gives the corresponding control elements. For example, you'll have "Jack HP-Out" (or a bit more verbose like "Jack HP-Out at Ext Rear") control via control API as well as other mixer controls. This can notify when a jack is plugged/unplugged. But, another big win is that you can read the current pin sense just by reading this element. I think this is a sort of feature Lennart wanted for PA. Lennart, does it make sense? The test codes are already in topic/hda-jack branch on sound-unstable GIT tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git topic/hda-jack Any comments appreciated. This is also intended for 2.6.32, not for 2.6.31 merge, of course :) thanks, Takashi

5 10

[alsa-devel] [PATCH] ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect
by Candelaria Villareal, Jorge 02 Jul '09

02 Jul '09

According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. It is suggested that GPIO6 of TWL4030 be used, but any other gpio can be used instead. This is indicated in machine driver with the following twl4030_setup_data members: -hs_extmute. Set to 1 if board has support for EXTMUTE. -set_hs_extmute. Set to a callback funcion to control an external gpio line. Set to NULL if MUTE[GPIO6] pin is used. Codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/codecs/twl4030.c | 27 +++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 2 ++ 2 files changed, 29 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 06a95f2..4e9e0e1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -620,6 +620,9 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { + struct snd_soc_device *socdev = codec->socdev; + struct twl4030_setup_data *setup = socdev->codec_data; + unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = codec->private_data; /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -629,6 +632,17 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + /* Enable external mute control, this dramatically reduces + * the pop-noise */ + if (setup && setup->hs_extmute) { + if (setup->set_hs_extmute) { + setup->set_hs_extmute(1); + } else { + hs_pop |= TWL4030_EXTMUTE; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } + } + if (ramp) { /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; @@ -636,6 +650,9 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Wait ramp delay time + 1, so the VMID can settle */ + mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / + twl4030->sysclk) + 1); } else { /* Headset ramp-down _not_ according to * the TRM, but in a way that it is working */ @@ -652,6 +669,16 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_pop &= ~TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); } + + /* Disable external mute */ + if (setup && setup->hs_extmute) { + if (setup->set_hs_extmute) { + setup->set_hs_extmute(0); + } else { + hs_pop &= ~TWL4030_EXTMUTE; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } + } } static int headsetlpga_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index fe5f395..2b4bfa2 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -274,6 +274,8 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; unsigned int sysclk; + unsigned int hs_extmute:1; + void (*set_hs_extmute)(int mute); }; #endif /* End of __TWL4030_AUDIO_H__ */ -- 1.6.0.4

5 4

[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: reparent sound device
by Daniel Mack 02 Jul '09

02 Jul '09

The sound device instance needs to be a child of the USB interface, not the USB device. Newer udev versions pay attention to that. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Reported-by: Lennart Poettering <lennart(a)poettering.net> --- sound/usb/caiaq/device.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 0e5db71..de38108 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -349,7 +349,9 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) log("Unable to set up control system (ret=%d)\n", ret); } -static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) +static int create_card(struct usb_device *usb_dev, + struct usb_interface *intf, + struct snd_card **cardp) { int devnum; int err; @@ -374,7 +376,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor), le16_to_cpu(usb_dev->descriptor.idProduct)); spin_lock_init(&dev->spinlock); - snd_card_set_dev(card, &usb_dev->dev); + snd_card_set_dev(card, &intf->dev); *cardp = card; return 0; @@ -461,7 +463,7 @@ static int __devinit snd_probe(struct usb_interface *intf, struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - ret = create_card(device, &card); + ret = create_card(device, intf, &card); if (ret < 0) return ret; -- 1.6.3.1

2 3

[alsa-devel] [PATCH] Enable loudspeakers in ALC889 for Acers and others
by ruben＠ugr.es 02 Jul '09

02 Jul '09

Hi all I managed to get sound working on my acer 6935, which has an ALC889 driven by the intel_hda driver The lack of sound seems to appear in other laptops as well. The bug report is at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4261 I know Takashi Iwai is familiar with the problems, since I used his workaround with hda-verb to get it working initially, so if he is free and wants to get this fixed, I'd be grateful (if anybody else can have a look, that'd be great as well). The patch is very easy (you can download it at the url above). I added snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); at the end of patch_alc883 before the return. Please check whether there is a problem with other models; in that case a new patch_alc889 which calls patch_alc883 plus the line can be added, or more detailed checks on the pci ids can be done. Thanks for your time. Ruben

2 1

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 02 Jul '09

02 Jul '09

The following changes since commit 10121a12e2380fb34b6e646a8e367add06d036c3: Takashi Iwai (1): Merge branch 'for-2.6.31' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Grant Likely (2): ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig ASoC: remove BROKEN from Efika and pcm030 fabric drivers Janusz Krzysztofik (1): ASoC: OMAP: fix OMAP1510 broken PCM pointer callback Mark Brown (1): ASoC: Only disable pxa2xx-i2s clocks if we enabled them sound/soc/fsl/Kconfig | 6 +++--- sound/soc/omap/omap-pcm.c | 11 +++++++---- sound/soc/pxa/pxa2xx-i2s.c | 7 ++++++- 3 files changed, 16 insertions(+), 8 deletions(-)

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[alsa-devel] Problems with snd-hda-intel on hp 6930p (2.6.30)
by Rafael J. Wysocki 02 Jul '09

02 Jul '09

Hi Takashi, I have a problem with snd-hda-intel from 2.6.30 on an hp EliteBook 6930p, which is that the sound is not audible, although everything appears to work just fine. Except maybe that when the box is rebooted and KDE (3.5) is started, the master control is "muted" and the led indicator next to the keyboard shows "no sound". Then, changing the "muted" status of the master control doesn't change the state of the led indicator and vice versa (the two things appear to be totally independent of each other). Can you please tell me what I can do to debug this issue? Best, Rafael

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[alsa-devel] laptop multichannel ethernet OR PCMCIA sound cards
by Matt Flax 02 Jul '09

02 Jul '09

We are after an ultra low latency connection to a multichannel in/out sound card for a laptop. We want a portable system and don't want to network to a different computer to achieve this goal. As we are after ultra low latency, we are interested in both PCMCIA and ethernet (gigabit) solutions. Is anyone aware of any such ultra low latency hardware ? Can anyone suggest any other solution for a laptop which is portable and not based on ethernet or PCMCIA ? I think that USB and firewire are good, but want even lower latency to the hardware if possible. thanks -- Matt Flax, BE, BSc, PhD Research Engineer National Acoustic Laboratories A Division of Australian Hearing 126 Greville Street Chatswood NSW 2067 Australia P +61 2 9412 6985 F +61 2 9411 8273 Please consider the environment before printing this email

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[alsa-devel] [PATCH] Line In for Acer Inspire 6530G model
by Tony Vroon 01 Jul '09

01 Jul '09

The Line In connector is set up as PIN_IN by default, using VREF_HIZ. It is connected to both ADCs, so add it to both input selectors. Also add the ability to use the input mix (on a SoundBlaster one would call this "What You Hear"). Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3345331..67dd883 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1560,18 +1560,22 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { /* Interal mic only available on one ADC */ { - .num_items = 3, + .num_items = 5, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, { "Int Mic", 0xb }, }, }, { - .num_items = 2, + .num_items = 4, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, }, } }; @@ -8189,6 +8193,8 @@ static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),

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[alsa-devel] [PATCH] alsa/soc: add locking to mpc5200-psc-ac97 driver
by Grant Likely 01 Jul '09

01 Jul '09

From: Grant Likely <grant.likely(a)secretlab.ca> AC97 bus register read/write hooks need to provide locking, but the mpc5200-psc-ac97 driver does not. This patch adds a mutex around the register access routines. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc5200_dma.h | 1 + sound/soc/fsl/mpc5200_psc_ac97.c | 17 ++++++++++++++++- 3 files changed, 18 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index efec33a..f0a2d40 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -456,6 +456,7 @@ int mpc5200_audio_dma_create(struct of_device *op) return -ENODEV; spin_lock_init(&psc_dma->lock); + mutex_init(&psc_dma->mutex); psc_dma->id = be32_to_cpu(*prop); psc_dma->irq = irq; psc_dma->psc_regs = regs; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 2000803..8d396bb 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -55,6 +55,7 @@ struct psc_dma { unsigned int irq; struct device *dev; spinlock_t lock; + struct mutex mutex; u32 sicr; uint sysclk; int imr; diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 794a247..7eb5499 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -34,13 +34,20 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) int status; unsigned int val; + mutex_lock(&psc_dma->mutex); + /* Wait for command send status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } + + /* Force clear the data valid bit */ + in_be32(&psc_dma->psc_regs->ac97_data); + /* Send the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); @@ -50,16 +57,19 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } /* Get the data */ val = in_be32(&psc_dma->psc_regs->ac97_data); if (((val >> 24) & 0x7f) != reg) { pr_err("reg echo error on ac97 read\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } val = (val >> 8) & 0xffff; + mutex_unlock(&psc_dma->mutex); return (unsigned short) val; } @@ -68,16 +78,21 @@ static void psc_ac97_write(struct snd_ac97 *ac97, { int status; + mutex_lock(&psc_dma->mutex); + /* Wait for command status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); - return; + goto out; } /* Write data */ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + out: + mutex_unlock(&psc_dma->mutex); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97)

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