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[alsa-devel] ASoC update for 2.6.31
by Mark Brown 23 Jul '09

23 Jul '09

The following changes since commit b7d4de7ff03085fda8310b2983b907166dd40f43: Daniel Mack (1): ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free 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 Chaithrika U S (1): ASoC: tlv320aic3x: Enable PLL when not bypassed sound/soc/codecs/tlv320aic3x.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-)

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[alsa-devel] ASoC: MAX9877: fix write operation for register
by Joonyoung Shim 23 Jul '09

23 Jul '09

The MAX9877 needs an address of start register when we write values to registers through i2c_master_send(), but the code for this was missed in max9877_write_regs(). If the value of control is 0 in the max9877_set_out_mode(), the value is not increased to 1, but actually the value to write to the register should be 1. And the register bits for out_mode and osc_mode should be cleared before writing. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/max9877.c | 14 +++++++++++--- 1 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index a1f1119..9e7e964 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -25,7 +25,14 @@ static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; static void max9877_write_regs(void) { - if (i2c_master_send(i2c, max9877_regs, 5) != 5) + unsigned int i; + u8 data[6]; + + data[0] = MAX9877_INPUT_MODE; + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + data[i + 1] = max9877_regs[i]; + + if (i2c_master_send(i2c, data, 6) != 6) dev_err(&i2c->dev, "i2c write failed\n"); } @@ -135,12 +142,12 @@ static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, { u8 value = ucontrol->value.integer.value[0]; - if (value) - value += 1; + value += 1; if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) return 0; + max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; max9877_regs[MAX9877_OUTPUT_MODE] |= value; max9877_write_regs(); return 1; @@ -166,6 +173,7 @@ static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) return 0; + max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; max9877_regs[MAX9877_OUTPUT_MODE] |= value; max9877_write_regs(); return 1; -- 1.6.0.4

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[alsa-devel] [RFC] [PATCH 3/3] ASoC: add support for Amstrad E3 (Delta) machine
by Janusz Krzysztofik 23 Jul '09

23 Jul '09

This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC. Created and tested against linux-2.6.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Depends on patch 2 form this series: TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver. Credits to Mark Underwood for his initial, omap-alsa based sound driver for this machine. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- CPU DAI parameters best matching the codec DAI has been selected out empirically for best user experience. Board specific audio function control (with related DAPM widgets) has been modeled after empirically discovered codec capabilities. Unlike other ASoC machine drivers, this one contains a line discipline. This is required for providig the codec driver with support for talking to the modem chip that can control the codec behavoiur. As the line discipline code contains board specific bits, I decided to make it a part of the ASoC machine driver, not the codec, and not a separate source file. Otherwise, some kind of a glue, like a bus over a tty, could help in linking the board specific part (bus device) to a more generic line discipline (bus adapter)[1]. In order to work at all, this driver requires a working McBSP1. On OMAP1510 based machines (not sure if other OMAP1 variants as well), where McBSP1 is a DSP public peripheral, that means the kernel must provide basic DSP support, ie. omap_dsp_init(), in order to power up the DSP. This used to be included in linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c. Without that, the driver would not work, ie. not shift in/out any bits over the CPU DAI[2]. This limitation is not board, but CPU specific, and may apply to other code that makes use of McBSP1/McBSP3 on affected machines. I provide an extra patch [4/3] as a temporary solution. To work correctly in playback mode, this driver requires my prevoiusly submitted patch that corrects pcm pointer calculation for OMAP1510 based machines[3] (already included in linux-2.6.31-rc3). To support codec controls, this driver requires my previously submitted patch that adds support for modem found on Amstrad Delta[4]. [1] http://www.spinics.net/lists/linux-serial/msg01856.html [2] http://www.spinics.net/lists/linux-omap/msg15114.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html [4] http://www.spinics.net/lists/linux-omap/msg15432.html --- linux-2.6.31-rc1/sound/soc/omap/Kconfig.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/Kconfig 2009-07-22 01:06:51.000000000 +0200 @@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_AMS_DELTA + tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + depends on SND_OMAP_SOC && MACH_AMS_DELTA + select SND_OMAP_SOC_MCBSP + select SND_SOC_CX20442 + help + Say Y if you want to add support for SoC audio on Amstrad Delta. + config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C --- linux-2.6.31-rc1/sound/soc/omap/Makefile.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/Makefile 2009-07-22 01:06:51.000000000 +0200 @@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd- # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o @@ -16,6 +17,7 @@ snd-soc-omap3pandora-objs := omap3pandor snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o --- /dev/null 2009-06-24 10:33:03.437003511 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/ams-delta.c 2009-07-22 01:06:51.000000000 +0200 @@ -0,0 +1,618 @@ +/* + * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone + * + * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/gpio.h> +#include <linux/tty.h> + +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include <mach/board-ams-delta.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/cx20442.h" + + +/* + * Controls available after the modem line discipline has been activated. + */ + +/* Virtual switch: audio input/output constellations */ +static const char *ams_delta_audio_mode[] = + {"Mixed", "Handset", "HandsFree", "SpeakerPhone"}; + +/* Selection <-> pin translation */ +#define AMS_DELTA_MOUTHPIECE 0 +#define AMS_DELTA_EARPHONE 1 +#define AMS_DELTA_MICROPHONE 2 +#define AMS_DELTA_SPEAKER 3 +#define AMS_DELTA_AGC 4 + +#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPHONE) | \ + (1 << AMS_DELTA_MICROPHONE)) +#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \ + (1 << AMS_DELTA_EARPHONE)) +#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \ + (1 << AMS_DELTA_SPEAKER)) +#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC)) + +unsigned short ams_delta_audio_mode_pins[] = { + AMS_DELTA_MIXED, + AMS_DELTA_HANDSET, + AMS_DELTA_HANDSFREE, + AMS_DELTA_SPEAKERPHONE, +}; + +static bool ams_delta_audio_agc = 0; + +static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *control = (struct soc_enum *)kcontrol->private_value; + unsigned short function; + bool changed = 0; + + if (ucontrol->value.enumerated.item[0] >= control->max) + return -EINVAL; + + mutex_lock(&codec->mutex); + + /* Translate selection to bitmap */ + function = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]]; + + /* Setup pins after corresponding bits if changed */ + if ((bool)snd_soc_dapm_get_pin_status(codec, "Speaker") != + (bool)(function & (1 << AMS_DELTA_SPEAKER))) { + changed = 1; + if (function & (1 << AMS_DELTA_SPEAKER)) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Microphone") != + (bool)(function & (1 << AMS_DELTA_MICROPHONE))) { + changed = 1; + if (function & (1 << AMS_DELTA_MICROPHONE)) + snd_soc_dapm_enable_pin(codec, "Microphone"); + else + snd_soc_dapm_disable_pin(codec, "Microphone"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Earphone") != + (bool)(function & (1 << AMS_DELTA_EARPHONE))) { + changed = 1; + if (function & (1 << AMS_DELTA_EARPHONE)) + snd_soc_dapm_enable_pin(codec, "Earphone"); + else + snd_soc_dapm_disable_pin(codec, "Earphone"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Mouthpiece") != + (bool)(function & (1 << AMS_DELTA_MOUTHPIECE))) { + changed = 1; + if (function & (1 << AMS_DELTA_MOUTHPIECE)) + snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + else + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + } + if (ams_delta_audio_agc != (bool)(function & (1 << AMS_DELTA_AGC))) { + ams_delta_audio_agc = (bool)(function & (1 << AMS_DELTA_AGC)); + changed = 1; + if (ams_delta_audio_agc) + snd_soc_dapm_enable_pin(codec, "AGCIN"); + else + snd_soc_dapm_disable_pin(codec, "AGCIN"); + } + if (changed) + snd_soc_dapm_sync(codec); + + mutex_unlock(&codec->mutex); + + return changed; +} + +static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned short value, mux; + + value = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") << + AMS_DELTA_MOUTHPIECE) | + (snd_soc_dapm_get_pin_status(codec, "Earphone") << + AMS_DELTA_EARPHONE)); + if (value) + value |= (snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE); + else + value = ((snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE) | + (snd_soc_dapm_get_pin_status(codec, "Speaker") << + AMS_DELTA_SPEAKER) | + ((unsigned short)ams_delta_audio_agc << AMS_DELTA_AGC)); + + for (mux = 0; mux < ARRAY_SIZE(ams_delta_audio_mode); mux++) + if (value == ams_delta_audio_mode_pins[mux]) + break; + + if (mux >= ARRAY_SIZE(ams_delta_audio_mode)) + return -EINVAL; + + ucontrol->value.enumerated.item[0] = mux; + + return 0; +} + +static const struct soc_enum ams_delta_audio_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode), + ams_delta_audio_mode), +}; + +static const struct snd_kcontrol_new ams_delta_audio_controls[] = { + SOC_ENUM_EXT("Audio Function", ams_delta_audio_enum[0], + ams_delta_get_audio_mode, ams_delta_set_audio_mode), +}; + +/* Now that we are able to control the codec over the modem, + * our hook switch can be used for dynamic DAPM reconfiguration */ +static struct snd_soc_jack ams_delta_hook_switch; +static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[]; +static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = { + { + .pin = "Mouthpiece", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Earphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, + { + .pin = "Microphone", + .mask = SND_JACK_MICROPHONE, + .invert = 1, + }, +}; + +/* To actually apply any modem controlled configuration changes to the codec, + * we must connect codec DAI pins to the modem for a moment. Be carefull + * not to interfere with digital mute function that shares the same hardware. */ +static struct timer_list cx81801_timer; +static bool cx81801_cmd_pending = 0; +static bool ams_delta_muted; + +static void cx81801_timeout(unsigned long data) +{ + /* REVISIT - locking? */ + if(!ams_delta_muted) + /* Reconnect the codec DAI back from the modem to the CPU DAI + * only if digital mute still off */ + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0); + cx81801_cmd_pending = 0; +} + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *cx81801_init = "ate0m0q0+fclass=8\r"; + + +/* + * Modem line discipline required for making above controls functional. + * Activated from userspace with ldattach, possibly invoked from udev rule. + */ + +static struct snd_soc_card ams_delta_audio_card; + +/* Line discipline setup */ +static int cx81801_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = ams_delta_audio_card.codec; + int ret, len = strlen(cx81801_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass codec structure address to other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, cx81801_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Can this empty callback be omitted? */ +static void cx81801_wakeup(struct tty_struct *tty) +{ +} + +/* Modem response callback */ +static void cx81801_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + const unsigned char *c; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Initialize timer used for config pulse generation */ + setup_timer(&cx81801_timer, cx81801_timeout, 0); + + /* Set up codec access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + + /* Link hook switch to DAPM pins */ + snd_soc_jack_add_pins(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_pins), + ams_delta_hook_switch_pins); + + /* Add virtual switches */ + snd_soc_add_controls(codec, ams_delta_audio_controls, + ARRAY_SIZE(ams_delta_audio_controls)); + + /* Set DAPM pins after hook switch present state */ +#if 0 + /* Fails for switch state matching initial gpio->state = 0 */ + snd_soc_jack_report(&ams_delta_ams_delta_hook_switch, + gpio_get_value(ams_delta_hook_switch_gpios[0].gpio) ? + 0 : SND_JACK_HEADSET, SND_JACK_HEADSET); +#else + if (gpio_get_value(ams_delta_hook_switch_gpios[0].gpio)) { + snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Earphone"); + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + } else { + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "Microphone"); + snd_soc_dapm_enable_pin(codec, "Earphone"); + snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + } + snd_soc_dapm_sync(codec); +#endif + return; + } + for (c = &cp[count - 1]; c >= cp; c--) { + if (*c != '\r') + continue; + /* Complete modem response received, apply config to codec */ + + /* REVISIT - locking? */ + /* Reconnect the codec to the modem if not already done */ + if (!ams_delta_muted && !cx81801_cmd_pending) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + AMS_DELTA_LATCH2_MODEM_CODEC); + + /* Keep config pulse long enough for proper config activation */ + cx81801_cmd_pending = 1; + mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(200)); + break; + } +} + +static void cx81801_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + del_timer_sync(&cx81801_timer); + + /* Revert back to default audio input/output constellation */ + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_disable_pin(codec, "AGCOUT"); + snd_soc_jack_report(&ams_delta_hook_switch, + SND_JACK_HEADPHONE, SND_JACK_HEADSET); + + /* REVISIT - Don't know how to do that */ + /* + * Remove controls that we expose when over the modem control available: + * - virtual audio mode switch, + * - hook switch to DAPM pins links + */ + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; + + tty->disc_data = NULL; +} + +static struct tty_ldisc_ops cx81801_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx81801", + .owner = THIS_MODULE, + .open = cx81801_open, + .close = cx81801_close, + .receive_buf = cx81801_receive, + .write_wakeup = cx81801_wakeup, +}; + + +/* + * Even if not very usefull, the sound card can still work without any of the + * above functonality activated. You can still control its audio input/output + * constellation and speakerphone gain from userspace by issueing AT commands + * over the modem port. + */ + +static int ams_delta_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set cpu DAI configuration */ + return snd_soc_dai_set_fmt(rtd->dai->cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +} + +static struct snd_soc_ops ams_delta_ops = { + .hw_params = ams_delta_hw_params, +}; + + +/* + * Board specific DAPM widgets + */ + +static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Microphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Earphone", NULL), + SND_SOC_DAPM_MIC("Mouthpiece", NULL), +}; + +static const struct snd_soc_dapm_route ams_delta_audio_map[] = { + {"TELIN", NULL, "Mouthpiece"}, + {"Earphone", NULL, "TELOUT"}, + + {"MIC", NULL, "Microphone"}, + {"Speaker", NULL, "SPKOUT"}, +}; + +/* Hook switch */ +static struct snd_soc_jack ams_delta_hook_switch; +static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = { + { + .gpio = 4, + .name = "hook_switch", + .report = SND_JACK_HEADSET, + .invert = 1, + .debounce_time = 100, + } +}; + +/* Board specific codec bias level control */ +static int ams_delta_set_bias_level(struct snd_soc_card *card, + enum snd_soc_bias_level level) +{ + struct snd_soc_codec *codec = card->codec; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, + AMS_DELTA_LATCH2_MODEM_NRESET); + break; + case SND_SOC_BIAS_OFF: + if (codec->bias_level != SND_SOC_BIAS_OFF) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, + 0); + } + codec->bias_level = level; + + return 0; +} + +/* Digital mute implemented using modem/CPU multiplexer. + * Shares hardware with codec config pulse generation */ +static bool ams_delta_muted = 1; + +static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute) +{ + /* REVISIT - locking? */ + if (ams_delta_muted == mute) + return 0; + ams_delta_muted = mute; + if (cx81801_cmd_pending) + return 0; + + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0); + + return 0; +} + +/* Our codec DAI probably doesn't have its own .ops structure */ +static struct snd_soc_dai_ops ams_delta_dai_ops = { + .digital_mute = ams_delta_digital_mute, +}; + +/* Will be used if the codec ever has its own digital_mute function */ +static int ams_delta_startup(struct snd_pcm_substream *substream) +{ + return ams_delta_digital_mute(NULL, 0); +} + +static void ams_delta_shutdown(struct snd_pcm_substream *substream) +{ + ams_delta_digital_mute(NULL, 1); +} + + +/* + * Card initialization + */ + +static int ams_delta_cx20442_init(struct snd_soc_codec *codec) +{ + struct snd_soc_dai *codec_dai = codec->dai; + /* Codec is ready, now add/activate board specific controls */ + + /* Set up digital mute if not provided by codec */ + if (!codec_dai->ops) { + codec_dai->ops = &ams_delta_dai_ops; + } else if (!codec_dai->ops->digital_mute) { + codec_dai->ops->digital_mute = ams_delta_digital_mute; + } else { + ams_delta_ops.startup = ams_delta_startup; + ams_delta_ops.shutdown = ams_delta_shutdown; + } + + /* Set codec bias level */ + ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + + /* Add board specific DAPM controls */ + if (!snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets, + ARRAY_SIZE(ams_delta_dapm_widgets))) { + if (!snd_soc_dapm_add_routes(codec, ams_delta_audio_map, + ARRAY_SIZE(ams_delta_audio_map))) { + /* Actual pin constellation if no modem control */ + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + snd_soc_dapm_enable_pin(codec, "Earphone"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_disable_pin(codec, "AGCOUT"); + + snd_soc_dapm_sync(codec); + } + } + /* Add hook switch */ + if (!snd_soc_jack_new(&ams_delta_audio_card, "hook_switch", + SND_JACK_HEADSET, &ams_delta_hook_switch)) { + if (!snd_soc_jack_add_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios)) { +#ifdef CONFIG_GPIO_SYSFS + /* Expose hook switch over sysfs if configured */ + gpio_export(ams_delta_hook_switch_gpios[0].gpio, false); +#endif + } + } + /* Register optional line discipline for over the modem control */ + tty_register_ldisc(N_AMSDELTA, &cx81801_ops); + + return 0; +} + +/* DAI glue - connects codec <--> CPU */ +static struct snd_soc_dai_link ams_delta_dai_link = { + .name = "CX20442", + .stream_name = "CX20442", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &cx20442_dai, + .init = ams_delta_cx20442_init, + .ops = &ams_delta_ops, +}; + +/* Audio card driver */ +static struct snd_soc_card ams_delta_audio_card = { + .name = "AMS_DELTA", + .platform = &omap_soc_platform, + .dai_link = &ams_delta_dai_link, + .num_links = 1, + .set_bias_level = ams_delta_set_bias_level, +}; + +/* Audio subsystem */ +static struct snd_soc_device ams_delta_snd_soc_device = { + .card = &ams_delta_audio_card, + .codec_dev = &cx20442_codec_dev, +}; + +/* Module init/exit */ +static struct platform_device *ams_delta_audio_platform_device; +static struct platform_device *cx20442_platform_device; + +static int __init ams_delta_module_init(void) +{ + int ret; + + if (!(machine_is_ams_delta())) + return -ENODEV; + + ams_delta_audio_platform_device = platform_device_alloc("soc-audio", -1); + if (!ams_delta_audio_platform_device) + return -ENOMEM; + + platform_set_drvdata(ams_delta_audio_platform_device, + &ams_delta_snd_soc_device); + ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev; + *(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1; + + ret = platform_device_add(ams_delta_audio_platform_device); + if (ret) + goto err; + + /* + * Codec platform device could be registered from elsewhere (board?), + * but I do it here as it makes sense only if used with the card. + */ + cx20442_platform_device = platform_device_register_simple("cx20442", + -1, NULL, 0); + return 0; +err: + platform_device_put(ams_delta_audio_platform_device); + return ret; +} +module_init(ams_delta_module_init); + +static void __exit ams_delta_module_exit(void) +{ + tty_unregister_ldisc(N_AMSDELTA); + +#ifdef CONFIG_GPIO_SYSFS + gpio_unexport(ams_delta_hook_switch_gpios[0].gpio); +#endif + snd_soc_jack_free_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + + /* Keep modem power on */ + ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + + platform_device_unregister(cx20442_platform_device); + platform_device_unregister(ams_delta_audio_platform_device); +} +module_exit(ams_delta_module_exit); + +MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>"); +MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); +MODULE_LICENSE("GPL");

3 8

[alsa-devel] [PATCH] ASoC: OMAP: Staticise pcm creation function of omap-pcm
by Lopez Cruz, Misael 23 Jul '09

23 Jul '09

Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/omap-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 84a1950..c3c931d 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -330,7 +330,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, +static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; -- 1.5.4.3

2 1

Re: [alsa-devel] distinguish events based on input ports
by Lieven Moors 22 Jul '09

22 Jul '09

Clemens Ladisch wrote: > Lieven Moors wrote: > >> I had a question regarding the use of input(read) ports. >> What I want to do is distinguish events based on the input port >> they came through. Since the destination of the event is mostly >> set to 'broadcast', I cannot extract this information from the event >> itself. >> > > Yes you can; when delivering broadcast or to-all-subscribers events, > ALSA sets the event's destination field to the actual destination. > > > HTH > Clemens > > Thanks for confirming that. it was my own silly mistake, because i had used snd_seq_ev_set_subs before trying to filter... Sorry for that... greets, Lieven

1 0

[alsa-devel] [PATCH] Mixer quirk for QuickCam E 3500
by Alexey Fisher 22 Jul '09

22 Jul '09

1. E3500 report cval->max more than it actually can handel, so if you set 95% capture level it will be muted. 2. betwen cval->min and cval-max(real) is 2940 control units, but real are only 7. 3. we found only 7 real control units, with cval->res = 384, but alsa can't handel less than 10 controls, so make it more and set cval->res = 192. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/usb/usbmixer.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 4bd3a7a..72562ee 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -995,6 +995,13 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, /* is not very clear from datasheets */ /* I hope that the min value is -15360 for newer firmware --jk */ switch (state->chip->usb_id) { + case USB_ID(0x046d, 0x09a4): + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO "using volume control quirk for the QuickCam E3500\n"); + cval->min = 6080; + cval->max = 8768; + cval->res = 192; + } case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): -- 1.6.3.3

2 17

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Enable PLL when not bypassed
by Chaithrika U S 22 Jul '09

22 Jul '09

PLL was not being enabled when it was not bypassed. This patch enables the PLL when it is used. Additionally, it disables the PLL when it is bypassed. Without this patch, the audio on TI DM646x EVM and DM355 EVM does not work properly. The bit clocks and the frame sync signals from the codec are not correct and hence the playback/record are faster than usual for most sample rates. The reason for this was that the PLL was not enabled when it was not bypassed. Tested on DM6467 EVM, playback tested on DM355 EVM. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to master branch of ALSA GIT tree at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=summary sound/soc/codecs/tlv320aic3x.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ab099f4..cb0d1bf 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 pll_d = 1; + u8 reg; /* select data word length */ data = @@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, pll_q &= 0xf; aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); - } else + /* disable PLL if it is bypassed */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + + } else { aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + /* enable PLL when it is used */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + } /* Route Left DAC to left channel input and * right DAC to right channel input */ -- 1.5.6

4 5

[alsa-devel] Blocked in snd_pcm_open
by Nicolas Krieger 22 Jul '09

22 Jul '09

Hello, I don't know if I am in the right place to expose my problem, if not, excuse me. I have a problem with snd_pcm_open function. The program seems to be blocked and gdb output shows this is happening while using snd_pcm_open. I am upgrading a program that reads videos so as to use alsa now. On my computer, there is no problem. But the problem appears on test computer. My questions are : - is it only a configuration problem (.asoundrc) ? - if not, what else could it be ? _rc = snd_pcm_open(&_handle, "defaultTEST", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); <= this is line *42 in ../../src/CAudio.cpp* in gdb output This a part of gdb when the progam is blocked : *0x009877a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 (gdb) bt #0 0x009877a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x00bdbd4b in __open_nocancel () from /lib/tls/libpthread.so.0 #2 0xf6fedb9d in open () from /usr/lib/libesddsp.so.0 #3 0x04e7e5f2 in snd_pcm_hw_open () from /lib/libasound.so.2 #4 0x04e7ea90 in _snd_pcm_hw_open () from /lib/libasound.so.2 #5 0x04e70893 in snd_pcm_close () from /lib/libasound.so.2 #6 0x04e70e78 in snd_pcm_close () from /lib/libasound.so.2 #7 0x04e710c9 in snd_pcm_open_slave () from /lib/libasound.so.2 #8 0x04ea4541 in snd_pcm_dmix_open () from /lib/libasound.so.2 #9 0x04ea4aea in _snd_pcm_dmix_open () from /lib/libasound.so.2 #10 0x04e70893 in snd_pcm_close () from /lib/libasound.so.2 #11 0x04e70e78 in snd_pcm_close () from /lib/libasound.so.2 #12 0x04e710c9 in snd_pcm_open_slave () from /lib/libasound.so.2 #13 0x04ea8186 in _snd_pcm_asym_open () from /lib/libasound.so.2 #14 0x04e70893 in snd_pcm_close () from /lib/libasound.so.2 #15 0x04e70e78 in snd_pcm_close () from /lib/libasound.so.2 #16 0x04e710c9 in snd_pcm_open_slave () from /lib/libasound.so.2 #17 0x04e8e1de in _snd_pcm_plug_open () from /lib/libasound.so.2 #18 0x04e70893 in snd_pcm_close () from /lib/libasound.so.2 #19 0x04e70e78 in snd_pcm_close () from /lib/libasound.so.2 #20 0x08065543 in CAudio::openDevice (this=0x87795e0, sampleRate=@0x8884558<https://messagerie.si.c-s.fr/horde/imp/message.php?index=988#>, nbChannels=@0x888455c<https://messagerie.si.c-s.fr/horde/imp/message.php?index=988#>, devicename=@0xfeee8670<https://messagerie.si.c-s.fr/horde/imp/message.php?index=988#>) at ../../src/CAudio.cpp:42* .asoundrc I am using : *pcm.card0 { type hw card 0 device 0 } pcm.card1 { type hw card 1 device 0 } pcm.dmixerDEF { type dmix ipc_key 1025 slave { pcm card0 period_time 0 period_size 2048 buffer_size 32768 rate 44100 } bindings { 0 0 1 1 } } pcm.dmixerTEST { type dmix ipc_key 1027 slave { pcm card1 format S16_LE period_time 0 period_size 2048 buffer_size 32768 rate 44100 } #bindings { # 0 0 # 1 1 #} } pcm.DEF { type asym playback.pcm "dmixerDEF" capture.pcm "card0" } pcm.TEST { type asym playback.pcm "dmixerTEST" capture.pcm "card1" } pcm.!default { type plug slave.pcm "DEF" } pcm.defaultTEST { type plug slave.pcm "TEST" }* Nicolas Krieger

1 0

[alsa-devel] [PATCH] ASoC: MAX9877: separate callback functions
by Joonyoung Shim 22 Jul '09

22 Jul '09

The callback function to control register was used by whole controls in MAX9877 driver, but this causes using many if statement for double register control or invert. So, the callback function for double register control is separate differently, and the code for invert is added in the callback function. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/max9877.c | 96 +++++++++++++++++++++++++++++--------------- 1 files changed, 63 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 7df9a7c..a1f1119 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -34,16 +34,16 @@ static int max9877_get_reg(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int reg2 = mc->rreg; - int shift = mc->shift; - int mask = mc->max; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - if (reg2) - ucontrol->value.integer.value[1] = - (max9877_regs[reg2] >> shift) & mask; + if (invert) + ucontrol->value.integer.value[0] = + mask - ucontrol->value.integer.value[0]; return 0; } @@ -53,39 +53,69 @@ static int max9877_set_reg(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int reg2 = mc->rreg; - int shift = mc->shift; - int mask = mc->max; - int change = 1; - int change2 = 1; - int ret = 0; - - if (((max9877_regs[reg] >> shift) & mask) == - ucontrol->value.integer.value[0]) + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + + if (invert) + val = mask - val; + + if (((max9877_regs[reg] >> shift) & mask) == val) + return 0; + + max9877_regs[reg] &= ~(mask << shift); + max9877_regs[reg] |= val << shift; + max9877_write_regs(); + + return 1; +} + +static int max9877_get_2reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + + ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; + + return 0; +} + +static int max9877_set_2reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + unsigned int val2 = (ucontrol->value.integer.value[1] & mask); + unsigned int change = 1; + + if (((max9877_regs[reg] >> shift) & mask) == val) change = 0; - if (reg2) - if (((max9877_regs[reg2] >> shift) & mask) == - ucontrol->value.integer.value[1]) - change2 = 0; + if (((max9877_regs[reg2] >> shift) & mask) == val2) + change = 0; if (change) { max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= ucontrol->value.integer.value[0] << shift; - ret = change; - } - - if (reg2 && change2) { + max9877_regs[reg] |= val << shift; max9877_regs[reg2] &= ~(mask << shift); - max9877_regs[reg2] |= ucontrol->value.integer.value[1] << shift; - ret = change2; - } - - if (ret) + max9877_regs[reg2] |= val2 << shift; max9877_write_regs(); + } - return ret; + return change; } static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, @@ -190,7 +220,7 @@ static const struct snd_kcontrol_new max9877_controls[] = { max9877_get_reg, max9877_set_reg, max9877_output_tlv), SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume", MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, - max9877_get_reg, max9877_set_reg, max9877_output_tlv), + max9877_get_2reg, max9877_set_2reg, max9877_output_tlv), SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", MAX9877_INPUT_MODE, 4, 1, 1, max9877_get_reg, max9877_set_reg), -- 1.6.0.4

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[alsa-devel] [RFC] [PATCH 0/3] Add support for Amstrad E3 (Delta) audio to ASoC
by Janusz Krzysztofik 22 Jul '09

22 Jul '09

This series of patches adds support for audio device found on Amstrad E3 (Delta) videophone to ASoC. 1/3: ASoC: Add support for Conexant CX20442-11 voice modem codec 2/3: TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver 3/3: ASoC: add support for Amstrad E3 (Delta) machine 4/3: OMAP1: Readd basic support for DSP initialization Created and tested against linux-2.5.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well Patch 4/3 is a temporary solution that should help interested parties in testing the driver or even making use of it before an official DSP support gets back to the omap or mainline tree. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

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