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

[alsa-devel] ASoC widgets questions
by Lopez Cruz, Misael 29 Jul '08

29 Jul '08

Hi all, I started to read about ASoC, but I didn't find much information apart from the kernel documentation files. Is there any other documents? Below are some questions mainly related to DAPM for codec driver. How should the interconnections map be created? What is the difference between using "SND_SOC_DAPM_PGA" and "SND_SOC_DAPM_MICBIAS" for an input? When to use each one? Why in some codec drivers there is a distinction between left and right ADC/DACs when creating DAPM widgets and in others not? How should be handled digital gains and mixing (after ADCs/before DACs)? Regards, Misael

2 1

[alsa-devel] app-internal volume control
by Wolfgang Rosenauer 29 Jul '08

29 Jul '08

Hi, might be a dumb question for people knowing the externals of Alsa but I'm not sure :-( What is the best practice to control the volume of a Linux application? I could control Alsa'x mixer for Master or PCM but that looks like a bad thing to do since it changes the volume for every other app as well. Is there another mixer I could use which is application specific? Or do I need to implement an application internal filter to control the volume by modifying the audio stream? Thanks, Wolfgang

1 0

[alsa-devel] Exposing control of my device
by Jon Smirl 29 Jul '08

29 Jul '08

My device has 220 registers, some of these are quite long (32 bytes) and I have split them up into multiple 32b pseudo registers. That results in about 300 registers for a TAS5504 and 450 for a TAS5508. A large number of these registers are for BiQuads, DRC, loudness, etc. They're not general purpose ALSA functions. What is an efficient way to expose several hundred registers of this class? Making snd_kcontrols for all of them seems pretty inefficient. Is there some generic way to do register access from user space that I haven't found yet? Or should I make two special snd_kcontrols, one for read register and write register? -- Jon Smirl jonsmirl(a)gmail.com

2 2

[alsa-devel] [PATCH] asoc: restrict sample rate and size in Freescale MPC8610 sound drivers
by Timur Tabi 29 Jul '08

29 Jul '08

The Freescale MPC8610 SSI device has the option of using one clock for both transmit and receive (synchronous mode), or independent clocks (asynchronous). The SSI driver, however, programs the SSI into synchronous mode and then tries to program the clock registers independently. The result is that the wrong sample size is usually generated during recording. This patch fixes the discrepancy by restricting the sample rate and sample size of the playback and capture streams. The SSI driver remembers which stream is opened first. When a second stream is opened, that stream is constrained to the same sample rate and size as the first stream. A future version of this driver will lift the sample size restriction. Supporting independent sample rates is more difficult, because only certain codecs provide dual independent clocks. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Please apply this patch for 2.6.26.1 and for 2.6.27-rc1. Without this patch, recording on the MPC8610 is just plain broken, so it is a must-fix for 2.6.26. sound/soc/fsl/fsl_dma.c | 7 ++++- sound/soc/fsl/fsl_ssi.c | 66 +++++++++++++++++++++++++++++++++++++++------- 2 files changed, 62 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index da2bc59..7ceea2b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -132,12 +132,17 @@ struct fsl_dma_private { * Since each link descriptor has a 32-bit byte count field, we set * period_bytes_max to the largest 32-bit number. We also have no maximum * number of periods. + * + * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a + * limitation in the SSI driver requires the sample rates for playback and + * capture to be the same. */ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_JOINT_DUPLEX, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 71bff33..da71876 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,6 +67,7 @@ * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI + * @first_stream: stream ID of first stream to program hardware * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened @@ -79,6 +80,7 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; + struct snd_pcm_substream *first_stream; struct device *dev; unsigned int playback; unsigned int capture; @@ -283,6 +285,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) return ret; } + ssi_private->first_stream = substream; + /* * Section 16.5 of the MPC8610 reference manual says that the * SSI needs to be disabled before updating the registers we set @@ -340,6 +344,44 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) * this is bad is because at this point, the PCM driver has not * finished initializing the DMA controller. */ + + } else { + /* This is not the first stream open, so we need to impose + * sample rate and maybe sample size constraints. Note that + * this can cause a race condition if the second stream is + * opened before the first stream is fully initialized. + * + * We provide some protection by checking to make sure the first + * stream is initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample rate or + * size. If the second stream is opened before the first stream + * has received its final parameters, then the second stream may + * be constrained to the wrong sample rate or size. + * + * FIXME: This code does not handle opening and closing streams + * repeatedly. If you open two streams and then close the first + * one, you may not be able to open another stream until you + * close the second one as well. + */ + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + + if (!first_runtime->rate || !first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample rate and size in %s stream first\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, first_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -371,18 +413,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u32 wl; - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + if (substream == ssi_private->first_stream) { + u32 wl; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + /* The SSI should always be disabled at this points (SSIEN=0) */ + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); - - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + } return 0; } @@ -407,9 +447,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -462,6 +506,8 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); free_irq(ssi_private->irq, ssi_private); + + ssi_private->first_stream = NULL; } } -- 1.5.5

1 1

[alsa-devel] [PATCH 1/2] ALSA: Add jack reporting API
by Mark Brown 28 Jul '08

28 Jul '08

Currently very few systems provide information about jack status to user space, even though many have hardware facilities to do detection. Those systems that do use an input device with the existing SW_HEADPHONE_INSERT switch type to do so, often independently of ALSA. This patch introduces a standard method for representing jacks to user space into ALSA. It allows drivers to register jacks for a sound card with the input subsystem, binding the input device to the card to help user space associate the input devices with their sound cards. The created input devices are named in the form "card longname jack" where jack is provided by the driver when allocating a jack. By default the parent for the input device is the sound card but this can be overridden by the card driver. The existing user space API with SW_HEADPHONE_INSERT is preserved. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/core.h | 1 + include/sound/jack.h | 75 +++++++++++++++++++++++ sound/core/jack.c | 163 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 include/sound/jack.h create mode 100644 sound/core/jack.c diff --git a/include/sound/core.h b/include/sound/core.h index 558b962..1a4ff0b 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -63,6 +63,7 @@ typedef int __bitwise snd_device_type_t; #define SNDRV_DEV_INFO ((__force snd_device_type_t) 0x1006) #define SNDRV_DEV_BUS ((__force snd_device_type_t) 0x1007) #define SNDRV_DEV_CODEC ((__force snd_device_type_t) 0x1008) +#define SNDRV_DEV_JACK ((__force snd_device_type_t) 0x1009) #define SNDRV_DEV_LOWLEVEL ((__force snd_device_type_t) 0x2000) typedef int __bitwise snd_device_state_t; diff --git a/include/sound/jack.h b/include/sound/jack.h new file mode 100644 index 0000000..b1b2b8b --- /dev/null +++ b/include/sound/jack.h @@ -0,0 +1,75 @@ +#ifndef __SOUND_JACK_H +#define __SOUND_JACK_H + +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics plc + * + * + * 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; either version 2 of the License, or + * (at your option) any later version. + * + * 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 + * + */ + +#include <sound/core.h> + +struct input_dev; + +/** + * Jack types which can be reported. These values are used as a + * bitmask. + */ +enum snd_jack_types { + SND_JACK_HEADPHONE = 0x0001, + SND_JACK_MICROPHONE = 0x0002, + SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, +}; + +struct snd_jack { + struct input_dev *input_dev; + int registered; + int type; + const char *id; + char name[100]; +}; + +#ifdef CONFIG_SND_JACK + +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack); +void snd_jack_set_parent(struct snd_jack *jack, struct device *parent); + +void snd_jack_report(struct snd_jack *jack, int status); + +#else + +static inline int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack) +{ + return 0; +} + +static inline void snd_jack_set_parent(struct snd_jack *jack, + struct device *parent) +{ +} + +static inline void snd_jack_report(struct snd_jack *jack, int status) +{ +} + +#endif + +#endif diff --git a/sound/core/jack.c b/sound/core/jack.c new file mode 100644 index 0000000..8133a2b --- /dev/null +++ b/sound/core/jack.c @@ -0,0 +1,163 @@ +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics + * + * 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; either version 2 of the License, or + * (at your option) any later version. + * + * 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 + * + */ + +#include <linux/input.h> +#include <sound/jack.h> +#include <sound/core.h> + +static int snd_jack_dev_free(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + + /* If the input device is registered with the input subsystem + * then we need to use a different deallocator. */ + if (jack->registered) + input_unregister_device(jack->input_dev); + else + input_free_device(jack->input_dev); + + kfree(jack); + + return 0; +} + +static int snd_jack_dev_register(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + struct snd_card *card = device->card; + int err; + + snprintf(jack->name, sizeof(jack->name), "%s %s", + card->longname, jack->id); + jack->input_dev->name = jack->name; + + /* Default to the sound card device. */ + if (!jack->input_dev->dev.parent) + jack->input_dev->dev.parent = card->dev; + + err = input_register_device(jack->input_dev); + if (err == 0) + jack->registered = 1; + + return err; +} + +/** + * snd_jack_new - Create a new jack + * @card: the card instance + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jjack: Used to provide the allocated jack object to the caller. + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jjack will be initialised. + */ +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jjack) +{ + struct snd_jack *jack; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_jack_dev_free, + .dev_register = snd_jack_dev_register, + }; + + jack = kzalloc(sizeof(struct snd_jack), GFP_KERNEL); + if (jack == NULL) + return -ENOMEM; + + jack->id = id; + + jack->input_dev = input_allocate_device(); + if (jack->input_dev == NULL) { + err = -ENOMEM; + goto fail_input; + } + + jack->input_dev->phys = "ALSA"; + + jack->type = type; + + if (type & SND_JACK_HEADPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_HEADPHONE_INSERT); + if (type & SND_JACK_MICROPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_MICROPHONE_INSERT); + + err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); + if (err < 0) + goto fail_input; + + *jjack = jack; + + return 0; + +fail_input: + input_free_device(jack->input_dev); + kfree(jack); + return err; +} +EXPORT_SYMBOL(snd_jack_new); + +/** + * snd_jack_set_parent - Set the parent device for a jack + * + * @jack: The jack to configure + * @parent: The device to set as parent for the jack. + * + * Set the parent for the jack input device in the device tree. This + * function is only valid prior to registration of the jack. If no + * parent is configured then the parent device will be the sound card. + */ +void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) +{ + WARN_ON(jack->registered); + + jack->input_dev->dev.parent = parent; +} +EXPORT_SYMBOL(snd_jack_set_parent); + +/** + * snd_jack_report - Report the current status of a jack + * + * @jack: The jack to report status for + * @status: The current status of the jack + */ +void snd_jack_report(struct snd_jack *jack, int status) +{ + if (jack->type & SND_JACK_HEADPHONE) + input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, + status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_MICROPHONE) + input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, + status & SND_JACK_MICROPHONE); + + input_sync(jack->input_dev); +} +EXPORT_SYMBOL(snd_jack_report); + +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("Jack detection support for ALSA"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

1 1

[alsa-devel] [PATCH 1/2] ALSA: Add jack reporting API
by Mark Brown 28 Jul '08

28 Jul '08

Currently very few systems provide information about jack status to user space, even though many have hardware facilities to do detection. Those systems that do use an input device with the existing SW_HEADPHONE_INSERT switch type to do so, often independently of ALSA. This patch introduces a standard method for representing jacks to user space into ALSA. It allows drivers to register jacks for a sound card with the input subsystem, binding the input device to the card to help user space associate the input devices with their sound cards. The created input devices are named in the form "card longname jack" where jack is provided by the driver when allocating a jack. By default the parent for the input device is the sound card but this can be overridden by the card driver. The existing user space API with SW_HEADPHONE_INSERT is preserved. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/core.h | 1 + include/sound/jack.h | 75 +++++++++++++++++++++++ sound/core/jack.c | 163 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 include/sound/jack.h create mode 100644 sound/core/jack.c diff --git a/include/sound/core.h b/include/sound/core.h index 558b962..1a4ff0b 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -63,6 +63,7 @@ typedef int __bitwise snd_device_type_t; #define SNDRV_DEV_INFO ((__force snd_device_type_t) 0x1006) #define SNDRV_DEV_BUS ((__force snd_device_type_t) 0x1007) #define SNDRV_DEV_CODEC ((__force snd_device_type_t) 0x1008) +#define SNDRV_DEV_JACK ((__force snd_device_type_t) 0x1009) #define SNDRV_DEV_LOWLEVEL ((__force snd_device_type_t) 0x2000) typedef int __bitwise snd_device_state_t; diff --git a/include/sound/jack.h b/include/sound/jack.h new file mode 100644 index 0000000..16521a3 --- /dev/null +++ b/include/sound/jack.h @@ -0,0 +1,75 @@ +#ifndef __SOUND_JACK_H +#define __SOUND_JACK_H + +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics plc + * + * + * 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; either version 2 of the License, or + * (at your option) any later version. + * + * 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 + * + */ + +#include <sound/core.h> + +struct input_dev; + +/** + * Jack types which can be reported. These values are used as a + * bitmask. + */ +enum snd_jack_types { + SND_JACK_HEADPHONE = 0x0001, + SND_JACK_MICROPHONE = 0x0002, + SND_JACK_HEADSET = 0x0003, /* Both microphone and headphone */ +}; + +struct snd_jack { + struct input_dev *input_dev; + int registered; + int type; + const char *id; + char name[100]; +}; + +#ifdef CONFIG_SND_JACK + +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack); +void snd_jack_set_parent(struct snd_jack *jack, struct device *parent); + +void snd_jack_report(struct snd_jack *jack, int status); + +#else + +static inline int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack) +{ + return 0; +} + +static inline void snd_jack_set_parent(struct snd_jack *jack, + struct device *parent) +{ +} + +static inline void snd_jack_report(struct snd_jack *jack, int status) +{ +} + +#endif + +#endif diff --git a/sound/core/jack.c b/sound/core/jack.c new file mode 100644 index 0000000..9c05ec8 --- /dev/null +++ b/sound/core/jack.c @@ -0,0 +1,163 @@ +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics + * + * 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; either version 2 of the License, or + * (at your option) any later version. + * + * 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 + * + */ + +#include <linux/input.h> +#include <sound/jack.h> +#include <sound/core.h> + +static int snd_jack_dev_free(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + + /* If the input device is registered with the input subsystem + * then we need to use a different deallocator. */ + if (jack->registered) + input_unregister_device(jack->input_dev); + else + input_free_device(jack->input_dev); + + kfree(jack); + + return 0; +} + +static int snd_jack_dev_register(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + struct snd_card *card = device->card; + int err; + + snprintf(jack->name, sizeof(jack->name), "%s %s", + card->longname, jack->id); + jack->input_dev->name = jack->name; + + /* Default to the sound card device. */ + if (!jack->input_dev->dev.parent) + jack->input_dev->dev.parent = card->dev; + + err = input_register_device(jack->input_dev); + if (err == 0) + jack->registered = 1; + + return err; +} + +/** + * snd_jack_new - Create a new jack + * @card: the card instance + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jjack: Used to provide the allocated jack object to the caller. + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jjack will be initialised. + */ +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jjack) +{ + struct snd_jack *jack; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_jack_dev_free, + .dev_register = snd_jack_dev_register, + }; + + jack = kzalloc(sizeof(struct snd_jack), GFP_KERNEL); + if (jack == NULL) + return -ENOMEM; + + jack->id = id; + + jack->input_dev = input_allocate_device(); + if (jack->input_dev == NULL) { + err = -ENOMEM; + goto fail_input; + } + + jack->input_dev->phys = "ALSA"; + + jack->type = type; + + if (type & SND_JACK_HEADPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_HEADPHONE_INSERT); + if (type & SND_JACK_MICROPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_MICROPHONE_INSERT); + + err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); + if (err < 0) + goto fail_input; + + *jjack = jack; + + return 0; + +fail_input: + input_free_device(jack->input_dev); + kfree(jack); + return err; +} +EXPORT_SYMBOL(snd_jack_new); + +/** + * snd_jack_set_parent - Set the parent device for a jack + * + * @jack: The jack to configure + * @parent: The device to set as parent for the jack. + * + * Set the parent for the jack input device in the device tree. This + * function is only valid prior to registration of the jack. If no + * parent is configured then the parent device will be the sound card. + */ +void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) +{ + BUG_ON(jack->registered); + + jack->input_dev->dev.parent = parent; +} +EXPORT_SYMBOL(snd_jack_set_parent); + +/** + * snd_jack_report - Report the current status of a jack + * + * @jack: The jack to report status for + * @status: The current status of the jack + */ +void snd_jack_report(struct snd_jack *jack, int status) +{ + if (jack->type & SND_JACK_HEADPHONE) + input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, + status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_MICROPHONE) + input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, + status & SND_JACK_MICROPHONE); + + input_sync(jack->input_dev); +} +EXPORT_SYMBOL(snd_jack_report); + +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("Jack detection support for ALSA"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

2 3

[alsa-devel] underrun!!! (at least 1786051083.613 ms long)
by Timur Tabi 28 Jul '08

28 Jul '08

I have a driver that supports simultaneous playback and capture, but only if the same size and sample rate are the same. I use these commands to play and capture audio: aplay -D hw:0,0 -t raw -f S24_BE -r 48000 -c 2 test2.raw arecord -D hw:0,0 -t raw -f S24_BE -r 48000 -c 2 test.raw If I start capture first and playback second, then everything works. If I start playback first and capture second, I get this: underrun!!! (at least 1786051083.587 ms long) underrun!!! (at least 1786051083.586 ms long) underrun!!! (at least 1786051083.585 ms long) underrun!!! (at least 1786051083.585 ms long) ... This repeats indefinitely. Can someone explain how this is even possible? 1786051083.585ms is longer than 20 days! How could ALSA be so wrong about the extend of the underrun? -- Timur Tabi Linux kernel developer at Freescale

1 0

[alsa-devel] sound/soc/pxa/tosa.c: removed duplicated include
by Huang Weiyi 28 Jul '08

28 Jul '08

Removed duplicated include <asm/arch/tosa.h> in sound/soc/pxa/tosa.c. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fe6cca9..22971a0 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -33,7 +33,6 @@ #include <asm/arch/pxa-regs.h> #include <asm/arch/hardware.h> #include <asm/arch/audio.h> -#include <asm/arch/tosa.h> #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h"

3 4

[alsa-devel] [PATCH 0/2] ALSA jack reporting API
by Mark Brown 28 Jul '08

28 Jul '08

This patch series is a repost of the jack reporting API previously discussed on this list. Patch 1 is a cherry pick of the commit from the input subsystem (available from Linus' current tree) adding a switch type for reporting microphone insertion which could be skipped if Linus' current tree were merged in to ALSA.

2 7

Re: [alsa-devel] [PATCH] cs4232: fix crash during chip PNP detection
by krzysztof.h1＠poczta.fm 28 Jul '08

28 Jul '08

Rene Herman napisał(a): > On 23-07-08 07:48, Krzysztof Helt wrote: > > > Looking back, it seems this was broken at least as far back as 2.6.20 > which would nicely show how much testing that code path gets. Do I > understand correctly from the backtrace that you have the PNPBIOS > hardware? If so, good... > Yes, I borrowed old laptop with PNP bios but no ISA PnP devices. I will test my wss_lib on it. > > --- linux-2.6.26/sound/isa/cs423x/cs4236.c~ 2008-07-23 > 07:40:16.129637645 +0200 > > +++ linux-2.6.26/sound/isa/cs423x/cs4236.c 2008-07-23 07:40:32.905641806 > +0200 > > @@ -325,6 +325,7 @@ static int __devinit snd_cs423x_pnp_init > > static int __devinit snd_card_cs4232_pnp(int dev, struct > snd_card_cs4236 *acard, > > struct pnp_dev *pdev) > > { > > + acard->wss = pdev; > > if (snd_cs423x_pnp_init_wss(dev, acard->wss) > 0) > > return -EBUSY; > > cport[dev] = -1; > > > > Acked-by: Rene Herman >rene.herman(a)gmail.com> > > (I believe I noticed earlier that struct snd_cs4236->wss, ctrl and mpu > really needn't/shouldn't be kept around in the first place and you could > just pass pdev directly as well but that's for another cleanup; this has > it look most similar to the card variant) > Feel free to changed it but I prefer to have the wss_lib patches merged first to avoid conflicts and redoing the patches. Krzysztof ---------------------------------------------------------------------- Galeria absurdow. zobacz >>> http://link.interia.pl/f1e5e

4 4