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

[alsa-devel] Keyboard volume buttons
by Harsha, Priya 05 Mar '09

05 Mar '09

Hi, Can anyone tell me how keyboard's volume buttons are converted into real volume change? Will there be an application that monitors the keyboard volume button events and call ALSA API to set the volume? Thanks Harsha

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[alsa-devel] [PATCH 1/3] ASoC: Add GPIO support for jack reporting interface
by Lopez Cruz, Misael 03 Mar '09

03 Mar '09

Add GPIO support for jack reporting framework in ASoC, by using gpiolib calls. It's only required to append GPIO information (gpio number, irq handler and flags) to standard jack_pin declaration. GPIO request, data direction configuration and irq request are handled by the utility. The minimal GPIO information that can be provided is the gpio number, in that case default handler/flags will be used. Default handler queues a work that reads the current value in the gpio pin and informs to the jack framework. If the GPIO support is not required, the "gpio" field ot jack_pin structure must be set to NO_JACK_PIN_GPIO. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc.h | 15 ++++++++++ sound/soc/soc-jack.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 83 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 68d8149..846e2c1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -16,6 +16,8 @@ #include <linux/platform_device.h> #include <linux/types.h> #include <linux/workqueue.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/control.h> @@ -254,14 +256,27 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported + * @gpio: gpio number associated to the pin (gpiolib calls will be used) + * @irqflags IRQ flags + * @handler: handler for servicing interrupt events on gpio pin */ struct snd_soc_jack_pin { + struct snd_soc_jack *jack; + struct snd_soc_jack_gpio *gpio_pin; struct list_head list; const char *pin; int mask; bool invert; + /* GPIO */ + unsigned int gpio; + unsigned int irq; + unsigned long irqflags; + irq_handler_t handler; + struct work_struct work; }; +#define NO_JACK_PIN_GPIO UINT_MAX + struct snd_soc_jack { struct snd_jack *jack; struct snd_soc_card *card; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 8cc00c3..0d048b2 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,9 @@ #include <sound/jack.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/workqueue.h> /** * snd_soc_jack_new - Create a new jack @@ -96,6 +99,32 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_jack_report); +/* Default IRQ handler for a GPIO jack pin, it will queue a + * work that reads current value in GPIO pin and reports it + * to the jack framework. + */ +static irqreturn_t gpio_interrupt(int irq, void *data) +{ + struct snd_soc_jack_pin *pin = data; + + return IRQ_RETVAL(schedule_work(&pin->work)); +} + +static void gpio_work(struct work_struct *work) +{ + struct snd_soc_jack_pin *pin; + int report; + + pin = container_of(work, struct snd_soc_jack_pin, work); + report = pin->jack->status & pin->mask; + if (gpio_get_value(pin->gpio)) + report |= pin->mask; + else + report &= ~pin->mask; + + snd_soc_jack_report(pin->jack, report, pin->jack->jack->type); +} + /** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * @@ -106,11 +135,18 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * After this function has been called the DAPM pins specified in the * pins array will have their status updated to reflect the current * state of the jack whenever the jack status is updated. + * + * A GPIO pin (using gpiolib) can be used to detect events. It requieres + * an IRQ handler and flags to be set in jack_pin structure; if they are + * not provided, default handler/flags will be used instead. If this + * feature is not desired, "gpio" field of jack_pin structure must be + * set to NO_JACK_PIN_GPIO. */ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins) { - int i; + unsigned int gpio = 0; + int i, ret = 0; for (i = 0; i < count; i++) { if (!pins[i].pin) { @@ -123,6 +159,32 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, return -EINVAL; } + if (pins[i].gpio != NO_JACK_PIN_GPIO) { + pins[i].jack = jack; + gpio = pins[i].gpio; + ret = gpio_request(gpio, pins[i].pin); + if (ret) + return ret; + + ret = gpio_direction_input(gpio); + if (ret) + goto out; + pins[i].irq = gpio_to_irq(gpio); + /* If none set, use the default handler */ + if (!pins[i].handler) { + pins[i].handler = gpio_interrupt; + pins[i].irqflags = IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING; + INIT_WORK(&pins[i].work, gpio_work); + } + ret = request_irq(pins[i].irq, + pins[i].handler, + pins[i].irqflags, + jack->card->dev->driver->name, + &pins[i]); + if (ret) + goto out; + } INIT_LIST_HEAD(&pins[i].list); list_add(&(pins[i].list), &jack->pins); } @@ -133,6 +195,10 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, */ snd_soc_jack_report(jack, 0, 0); - return 0; +out: + if (ret) + gpio_free(gpio); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); -- 1.5.4.3

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[alsa-devel] Plugging in headphones doesn't mute speakers, internal microphone does not work
by Emilio López 03 Mar '09

03 Mar '09

Hello, This is my first mail to the list, I hope you can help me. I have been talking with people on #alsa, but they couldn't help me, so that's why I'm writing to you. Well, as lspci says, my Acer Aspire 6930 laptop has an "Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)". With this card, and without adding any mode=... to /etc/modprobe.d/alsa-base, playback works perfectly, but when I plug in headphones, the integrated speakers do not mute. I can mute the speakers muting "front" and "side" and use the headphones, but it should be automatic. Setting mode=auto makes the headphones, when plugged, mute the speakers. But the problem with this is, that when using the speakers, I only get sound from the speaker that was "front" in the other mode. The microphone does not work in this mode either. I also tried other modes, including (but not limited to) acer, acer-aspire. I ran alsa-info.sh, here is the URL: http://www.alsa-project.org/db/?f=7910da439e8e190a8c2a6f18167d4879c74881c9 And before I forget, regarding the microphone, if I plug an external one on the microphone plug, I can record. And I tried the internal one on windows, and it works, so I believe it's an alsa problem. Looking forward to hearing from you! Emilio

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[alsa-devel] [PATCH] cs4270: add support for slave mode configurations
by Daniel Mack 02 Mar '09

02 Mar '09

Added support for scenarios where the Cirrus CS4270 audio codec is slave to the bitclk and lrclk. Mixed setups are unsupported. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/codecs/cs4270.c | 24 +++++++++++++++++++++++- 1 files changed, 23 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index cd4a9ee..339e0f6 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -109,6 +109,7 @@ struct cs4270_private { u8 reg_cache[CS4270_NUMREGS]; unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ + unsigned int slave_mode; }; /** @@ -247,6 +248,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, struct cs4270_private *cs4270 = codec->private_data; int ret = 0; + /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: @@ -257,6 +259,21 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, ret = -EINVAL; } + /* set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4270->slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4270->slave_mode = 0; + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* unsupported - cs4270 can eigther be slave or master to + * both the bitclk and the lrclk. */ + default: + ret = -EINVAL; + } + return ret; } @@ -399,7 +416,12 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg = snd_soc_read(codec, CS4270_MODE); reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); - reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk; + reg |= cs4270_mode_ratios[i].mclk; + + if (cs4270->slave_mode) + reg |= CS4270_MODE_SLAVE; + else + reg |= cs4270_mode_ratios[i].speed_mode; ret = snd_soc_write(codec, CS4270_MODE, reg); if (ret < 0) { -- 1.6.1.3

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[alsa-devel] arecord channel limitation
by Tim Blechmann 02 Mar '09

02 Mar '09

hi all, is there any reason, why arecord limits the channel count to 32? best, tim -- tim(a)klingt.org http://tim.klingt.org Desperation is the raw material of drastic change. Only those who can leave behind everything they have ever believed in can hope to escape. William S. Burroughs

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[alsa-devel] [PATCH] Arcam AV Control Plugin Update
by Peter Stokes 02 Mar '09

02 Mar '09

arcam_av: Refactor event callback to facilitate addition of "numid" optimisation Signed-off-by: Peter Stokes <linux(a)dadeos.co.uk>

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[alsa-devel] [PATCH] opl3sa2 fix irq releasing and short name of card
by Krzysztof Helt 02 Mar '09

02 Mar '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Two simple fixes: 1. Use the same pointer for the free_irq() and the request_irq() calls. 2. A short name of card is appended with '2' or '3' character depending on a detected chip. Remove the '2' character from the short name. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The short name without the patch is "OPL3-SA23" for opl3-sa3. sound/isa/opl3sa2.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 58c972b..9a53ab4 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -613,7 +613,7 @@ static void snd_opl3sa2_free(struct snd_card *card) { struct snd_opl3sa2 *chip = card->private_data; if (chip->irq >= 0) - free_irq(chip->irq, (void *)chip); + free_irq(chip->irq, card); release_and_free_resource(chip->res_port); } @@ -626,7 +626,7 @@ static struct snd_card *snd_opl3sa2_card_new(int dev) if (card == NULL) return NULL; strcpy(card->driver, "OPL3SA2"); - strcpy(card->shortname, "Yamaha OPL3-SA2"); + strcpy(card->shortname, "Yamaha OPL3-SA"); chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; -- 1.5.2.2 ---------------------------------------------------------------------- 169 stacji TV! Naziemne i satelitarne! Sprawdz program TV w kilka sekund >> http://link.interia.pl/f2050

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[alsa-devel] pointer callback question
by Tim Blechmann 28 Feb '09

28 Feb '09

hi all, one question about the pointer callback. the docs say: > This callback is called when the PCM middle layer inquires the current > hardware position on the buffer. The position must be returned in frames, > ranging from 0 to buffer_size - 1. i am using a ringbuffer to communicate with my audio device. but when the position is reset to 0, alsa requests to stop the stream (SNDRV_PCM_TRIGGER_STOP). the brief log is (4 periods of 64 frames): stream_pointer at 64 stream_pointer at 128 stream_pointer at 192 stream_pointer at 256 stream_pointer at 0 after returning 0, a stop is requested. am i missing something? thanks, tim -- tim(a)klingt.org http://tim.klingt.org Contrary to general belief, an artist is never ahead of his time but most people are far behind theirs. Edgar Varèse

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[alsa-devel] Xonar U1 volume controls
by Elie Morisse 28 Feb '09

28 Feb '09

Hi! I'm answering to cladisch call for the "lsusb -v" ouput for this device on Asus forums: http://vip.asus.com/forum/view.aspx?id=20090105132008643&board_id=21&model=… us The analog ouput and microphone input work flawlessly with alsa-driver-1.0.19 but the volume controls don't, although they are listed in evtest pushing or rotating the knob(X not running) does nothing : $ sudo evtest /dev/input/event2 Input driver version is 1.0.0 Input device ID: bus 0x3 vendor 0xb05 product 0x1743 version 0x100 Input device name: "C-Media Electronics Inc. USB Advanced Audio Device" Supported events: Event type 0 (Reset) Event code 0 (Reset) Event code 1 (Key) Event code 4 (?) Event type 1 (Key) Event code 113 (Mute) Event code 114 (VolumeDown) Event code 115 (VolumeUp) Event type 4 (?) Event code 4 (?) Testing ... (interrupt to exit) Besides it also has a LED that can "pulse" and can be blue, purple or red, and there doesn't seem to be any way to control this either. On Vista the Asus driver defaults it to pulsing blue when ouputting sound while on Linux it blinks alternatively purple and blue, how ugly and less relaxing!.. Hope the lsusb output will be useful and thanks a lot for all your hard work, Elie

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[alsa-devel] E-MU 0404 Support
by Kristian Lumme 27 Feb '09

27 Feb '09

Hi! I reported a bug (or what I think is a bug) with the E-MU 0404 USB driver (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4410) (this probably concerns other E-MU cards as well). Now, the supported soundcards list states that E-MU (or Creative) have supplied ALSA developers with datasheets to help the development of the driver. Well... in my bug report I was referred to this discussion on the ALSA devel list: http://thread.gmane.org/gmane.linux.alsa.devel/56288/focus=56355 If I understand this correctly, people that are offering to help develop the driver can't get in touch with the person who has the datasheets (and there hasn't been any update from him in a long time... or so it seems--sorry if I'm mistaken). Surely, there's some way to make these specifications available to interested developers so the support for the E-MU cards can be improved?

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