June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: sigmadsp: remove I2C support
by Arnd Bergmann 05 Jun '14

05 Jun '14

The newly added ADAU1781 codec comes with separate drivers for I2C and SPI modes, which gives us a new variation of the dependency problems: If SPI is enabled and the I2C core is a loadable module, selecting SND_SOC_ADAU1781_SPI as built-in leads to a link error because the common base code from sigmadsp.c now also gets built-in: sound/built-in.o: In function `sigma_action_write_i2c': :(.text+0x5d8d4): undefined reference to `i2c_master_send' Rather than adding yet more complex Kconfig logic, this converts the last user of process_sigma_firmware() to process_sigma_firmware_regmap(), which seems trivial after the driver itself now uses regmap as well. With that done, there is no more linker dependency from SND_SOC_ADAU1781_SPI to i2c_master_send. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- I hope this is the last ASoC patch I have for the moment, sorry for spamming everybody ;-) sound/soc/codecs/adau1701.c | 5 +++-- sound/soc/codecs/sigmadsp.c | 27 --------------------------- sound/soc/codecs/sigmadsp.h | 3 --- 3 files changed, 3 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d71c59c..1b5e306 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -239,7 +239,6 @@ static int adau1701_reg_read(void *context, unsigned int reg, static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) { struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *client = to_i2c_client(codec->dev); int ret; if (clkdiv != ADAU1707_CLKDIV_UNSET && @@ -282,7 +281,9 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) * know the correct PLL setup */ if (clkdiv != ADAU1707_CLKDIV_UNSET) { - ret = process_sigma_firmware(client, ADAU1701_FIRMWARE); + ret = process_sigma_firmware_regmap(codec->dev, + adau1701->regmap, + ADAU1701_FIRMWARE); if (ret) { dev_warn(codec->dev, "Failed to load firmware\n"); return ret; diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 4068f24..209f98f 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -10,7 +10,6 @@ #include <linux/delay.h> #include <linux/firmware.h> #include <linux/kernel.h> -#include <linux/i2c.h> #include <linux/regmap.h> #include <linux/module.h> @@ -198,30 +197,6 @@ static int _process_sigma_firmware(struct device *dev, return ret; } -#if IS_ENABLED(CONFIG_I2C) - -static int sigma_action_write_i2c(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return i2c_master_send(control_data, (const unsigned char *)&sa->addr, - len); -} - -int process_sigma_firmware(struct i2c_client *client, const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = client; - ssfw.write = sigma_action_write_i2c; - - return _process_sigma_firmware(&client->dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware); - -#endif - -#if IS_ENABLED(CONFIG_REGMAP) - static int sigma_action_write_regmap(void *control_data, const struct sigma_action *sa, size_t len) { @@ -241,6 +216,4 @@ int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL(process_sigma_firmware_regmap); -#endif - MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index e439cbd..38a455b 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -12,9 +12,6 @@ #include <linux/device.h> #include <linux/regmap.h> -struct i2c_client; - -extern int process_sigma_firmware(struct i2c_client *client, const char *name); extern int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, const char *name);

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[alsa-devel] [PATCH 1/2] ALSA: soundcore: coding style fixes
by Daniel Mack 05 Jun '14

05 Jun '14

sound/sounc_core.c has a larger number of style issues. Fix them to make the code more readable. No functional change. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/sound_core.c | 192 ++++++++++++++++++++++++----------------------------- 1 file changed, 85 insertions(+), 107 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 11e953a..bf9e41e 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -68,21 +68,16 @@ module_exit(cleanup_soundcore); #ifdef CONFIG_SOUND_OSS_CORE /* * OSS sound core handling. Breaks out sound functions to submodules - * - * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> - * - * Fixes: * + * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> * * 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. * - * -------------------- - * * Top level handler for the sound subsystem. Various devices can - * plug into this. The fact they don't all go via OSS doesn't mean + * plug into this. The fact they don't all go via OSS doesn't mean * they don't have to implement the OSS API. There is a lot of logic * to keeping much of the OSS weight out of the code in a compatibility * module, but it's up to the driver to rember to load it... @@ -111,8 +106,7 @@ module_exit(cleanup_soundcore); #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -158,8 +152,7 @@ module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -171,71 +164,66 @@ static const struct file_operations soundcore_fops = * join into it. Called with the lock asserted */ -static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, const struct file_operations *fops, int index, int low, int top) +static int __sound_insert_unit(struct sound_unit *s, struct sound_unit **list, + const struct file_operations *fops, + int index, int low, int top) { - int n=low; + int n = low; if (index < 0) { /* first free */ + while (*list && (*list)->unit_minor < n) + list = &((*list)->next); - while (*list && (*list)->unit_minor<n) - list=&((*list)->next); - - while(n<top) - { + while (n < top) { /* Found a hole ? */ - if(*list==NULL || (*list)->unit_minor>n) + if (*list == NULL || (*list)->unit_minor > n) break; - list=&((*list)->next); - n+=SOUND_STEP; + + list = &((*list)->next); + n += SOUND_STEP; } - if(n>=top) + if (n >= top) return -ENOENT; } else { - n = low+(index*16); + n = low + (index * 16); while (*list) { - if ((*list)->unit_minor==n) + if ((*list)->unit_minor == n) return -EBUSY; - if ((*list)->unit_minor>n) + if ((*list)->unit_minor > n) break; - list=&((*list)->next); + list = &((*list)->next); } - } - - /* - * Fill it in - */ - - s->unit_minor=n; - s->unit_fops=fops; - - /* - * Link it - */ - - s->next=*list; - *list=s; - - + } + + /* Fill it in */ + s->unit_minor = n; + s->unit_fops = fops; + + /* Link it */ + s->next = *list; + *list = s; + return n; } /* * Remove a node from the chain. Called with the lock asserted */ - -static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) +static struct sound_unit *__sound_remove_unit(struct sound_unit **list, + int unit) { - while(*list) - { - struct sound_unit *p=*list; - if(p->unit_minor==unit) - { - *list=p->next; + while (*list) { + struct sound_unit *p = *list; + + if (p->unit_minor == unit) { + *list = p->next; return p; } - list=&(p->next); + + list = &p->next; } + printk(KERN_ERR "Sound device %d went missing!\n", unit); return NULL; } @@ -250,12 +238,15 @@ static DEFINE_SPINLOCK(sound_loader_lock); * Allocate the controlling structure and add it to the sound driver * list. Acquires locks as needed */ - -static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, int index, int low, int top, const char *name, umode_t mode, struct device *dev) +static int sound_insert_unit(struct sound_unit **list, + const struct file_operations *fops, + int index, int low, int top, const char *name, + umode_t mode, struct device *dev) { - struct sound_unit *s = kmalloc(sizeof(*s), GFP_KERNEL); + struct sound_unit *s; int r; + s = kmalloc(sizeof(*s), GFP_KERNEL); if (!s) return -ENOMEM; @@ -263,7 +254,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati retry: r = __sound_insert_unit(s, list, fops, index, low, top); spin_unlock(&sound_loader_lock); - + if (r < 0) goto fail; else if (r < SOUND_STEP) @@ -292,7 +283,8 @@ retry: } device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), - NULL, "%s", s->name+6); + NULL, "%s", s->name + 6); + return s->unit_minor; fail: @@ -305,7 +297,6 @@ fail: * completed the removal before their file operations become * invalid. */ - static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -357,7 +348,6 @@ static struct sound_unit *chains[SOUND_STEP]; * Return: The allocated number is returned on success. On failure, * a negative error code is returned. */ - int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { @@ -367,69 +357,66 @@ int register_sound_special_device(const struct file_operations *fops, int unit, char _name[16]; switch (chain) { - case 0: + case 0: name = "mixer"; break; - case 1: + case 1: name = "sequencer"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 2: + case 2: name = "midi"; break; - case 3: + case 3: name = "dsp"; break; - case 4: + case 4: name = "audio"; break; - case 5: + case 5: name = "dspW"; break; - case 8: + case 8: name = "sequencer2"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 9: + case 9: name = "dmmidi"; break; - case 10: + case 10: name = "dmfm"; break; - case 12: + case 12: name = "adsp"; break; - case 13: + case 13: name = "amidi"; break; - case 14: + case 14: name = "admmidi"; break; - default: - { - __unknown: - sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; - } + default: +__unknown: + sprintf(_name, "unknown%d", chain); + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; break; } + return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, name, S_IRUSR | S_IWUSR, dev); } - EXPORT_SYMBOL(register_sound_special_device); int register_sound_special(const struct file_operations *fops, int unit) { return register_sound_special_device(fops, unit, NULL); } - EXPORT_SYMBOL(register_sound_special); /** @@ -443,13 +430,11 @@ EXPORT_SYMBOL(register_sound_special); * Return: On success, the allocated number is returned. On failure, * a negative error code is returned. */ - int register_sound_mixer(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[0], fops, dev, 0, 128, "mixer", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_mixer); /** @@ -469,14 +454,13 @@ int register_sound_midi(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[2], fops, dev, 2, 130, "midi", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_midi); /* * DSP's are registered as a triple. Register only one and cheat * in open - see below. */ - + /** * register_sound_dsp - register a DSP device * @fops: File operations for the driver @@ -497,7 +481,6 @@ int register_sound_dsp(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[3], fops, dev, 3, 131, "dsp", S_IWUSR | S_IRUSR, NULL); } - EXPORT_SYMBOL(register_sound_dsp); /** @@ -514,7 +497,6 @@ void unregister_sound_special(int unit) { sound_remove_unit(&chains[unit % SOUND_STEP], unit); } - EXPORT_SYMBOL(unregister_sound_special); /** @@ -529,7 +511,6 @@ void unregister_sound_mixer(int unit) { sound_remove_unit(&chains[0], unit); } - EXPORT_SYMBOL(unregister_sound_mixer); /** @@ -544,7 +525,6 @@ void unregister_sound_midi(int unit) { sound_remove_unit(&chains[2], unit); } - EXPORT_SYMBOL(unregister_sound_midi); /** @@ -561,21 +541,19 @@ void unregister_sound_dsp(int unit) { sound_remove_unit(&chains[3], unit); } - - EXPORT_SYMBOL(unregister_sound_dsp); static struct sound_unit *__look_for_unit(int chain, int unit) { struct sound_unit *s; - - s=chains[chain]; - while(s && s->unit_minor <= unit) - { - if(s->unit_minor==unit) + + s = chains[chain]; + while (s && s->unit_minor <= unit) { + if (s->unit_minor == unit) return s; - s=s->next; + s = s->next; } + return NULL; } @@ -586,14 +564,13 @@ static int soundcore_open(struct inode *inode, struct file *file) struct sound_unit *s; const struct file_operations *new_fops = NULL; - chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { - unit&=0xF0; - unit|=3; - chain=3; + chain = unit & 0x0F; + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ + unit &= 0xF0; + unit |= 3; + chain = 3; } - + spin_lock(&sound_loader_lock); s = __look_for_unit(chain, unit); if (s) @@ -608,8 +585,8 @@ static int soundcore_open(struct inode *inode, struct file *file) * ALSA toplevel modules for soundcards, thus we need * load them at first. [Jaroslav Kysela <perex(a)jcu.cz>] */ - request_module("sound-slot-%i", unit>>4); - request_module("sound-service-%i-%i", unit>>4, chain); + request_module("sound-slot-%i", unit >> 4); + request_module("sound-service-%i-%i", unit >> 4, chain); /* * sound-slot/service-* module aliases are scheduled @@ -636,10 +613,11 @@ static int soundcore_open(struct inode *inode, struct file *file) replace_fops(file, new_fops); if (file->f_op->open) - err = file->f_op->open(inode,file); + err = file->f_op->open(inode, file); return err; } + return -ENODEV; } -- 1.9.3

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[alsa-devel] New codec support for ALC891
by Kailang 05 Jun '14

05 Jun '14

Hi Takashi, We have a new codec ALC891. The support patch as attach. Many Thanks. BR, Kailang

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[alsa-devel] [PATCH - ucm 1/1] ucm: set active_verb when set_verb was successful
by nrubinstein＠aldebaran-robotics.com 05 Jun '14

05 Jun '14

From: Noé Rubinstein <nrubinstein(a)aldebaran-robotics.com> This seems to be a bug: execute_sequence returns zero on success, therefore the current code changes the active_verb iff execute_sequence failed. Signed-off-by: Noé Rubinstein <nrubinstein(a)aldebaran-robotics.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 0494728..3c0d7e4 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -623,7 +623,7 @@ static int set_verb(snd_use_case_mgr_t *uc_mgr, &verb->value_list, &uc_mgr->value_list, NULL); - if (enable && err >= 0) + if (enable && err == 0) uc_mgr->active_verb = verb; return err; } -- 1.9.2

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[alsa-devel] Rate or plug plugin issue with snd-pcsp
by Alexander E. Patrakov 05 Jun '14

05 Jun '14

Hello. While testing application support for weird-rate cards (such as snd-pcsp), I found a bug in alsa-lib. Alsa-plugins are installed, so the "libspeex (builtin)" rate converter is used. This bug already exists in alsa-plugins 1.0.27.2, so not a regression and thus should not block 1.0.28. $ file test.wav test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 32000 Hz $ aplay -vvv -D plughw:pcsp test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: NONE PERIOD_TIME: (125006 125007) PERIOD_SIZE: NONE PERIOD_BYTES: (16000 16004) PERIODS: (3 4) BUFFER_TIME: 500000 BUFFER_SIZE: 16000 BUFFER_BYTES: 64000 TICK_TIME: 0 The pcsp device supports the sample rate of 37286 Hz. Files with 44100 Hz sample rate work fine. $ aplay -vv -D plughw:pcsp -f cd /dev/zero Playing raw data '/dev/zero' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0*0,5 + 1*0,5 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 Slave: Rate conversion PCM (37286, sformat=S16_LE) Converter: libspeex (builtin) Protocol version: 10002 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 6206523236469964800 Slave: Hardware PCM card 2 'pcsp' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 37286 exact rate : 37286 (37286/1) msbits : 16 buffer_size : 18644 period_size : 4661 period_time : 125006 tstamp_mode : NONE period_step : 1 avail_min : 4661 period_event : 0 start_threshold : 18644 stop_threshold : 18644 silence_threshold: 0 silence_size : 0 boundary : 5247819465793470464 appl_ptr : 0 hw_ptr : 0 #+ | 00%^C -- Alexander E. Patrakov

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[alsa-devel] [PATCH] ASoC: pxa: add I2C dependencies as needed
by Arnd Bergmann 04 Jun '14

04 Jun '14

We have in the past added 'depends on I2C' for some of the PXA boards after hitting randconfig build bugs. I have seens a couple of new bugs in this area during the linux-next cycle for 3.16, after it became possible to build some more PXA machines with I2C disabled. To shut this up for good, this adds the dependency to every board that uses I2C as the interface to the codec. I have gone through all board files and verified that they all either use AC97 or I2C, and this annotates the latter. Some of these already enable I2C from mach-pxa/Kconfig, but since that can change it's better to be explicit here. The link error that can result otherwise happens when CONFIG_I2C is set to 'm' and the codec driver is built-in as a result of being selected by the platform specific glue. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcf0c08..e195b038 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -40,7 +40,7 @@ config SND_MMP_SOC_SSPA config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" - depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx + depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -49,7 +49,7 @@ config SND_PXA2XX_SOC_CORGI config SND_PXA2XX_SOC_SPITZ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" - depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 + depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -58,7 +58,7 @@ config SND_PXA2XX_SOC_SPITZ config SND_PXA2XX_SOC_Z2 tristate "SoC Audio support for Zipit Z2" - depends on SND_PXA2XX_SOC && MACH_ZIPIT2 + depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -66,7 +66,7 @@ config SND_PXA2XX_SOC_Z2 config SND_PXA2XX_SOC_POODLE tristate "SoC Audio support for Poodle" - depends on SND_PXA2XX_SOC && MACH_POODLE + depends on SND_PXA2XX_SOC && MACH_POODLE && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -181,7 +181,7 @@ config SND_PXA2XX_SOC_HX4700 config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" - depends on SND_PXA2XX_SOC && MACH_MAGICIAN + depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C select SND_PXA2XX_SOC_I2S select SND_PXA_SOC_SSP select SND_SOC_UDA1380

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[alsa-devel] [PATCH] sound: tlv320aic32x4: Fixed Coding style issues
by ShaRab Shaik 04 Jun '14

04 Jun '14

>From 617d7a25f3c0c08e0b95b24de2d1ea0f3f4b98c7 Mon Sep 17 00:00:00 2001 From: Shahina Shaik <sharab.shaik(a)gmail.com> Date: Wed, 4 Jun 2014 16:27:19 +0530 Subject: [PATCH] sound: tlv320aic32x4: Fixed Coding style issues. Fixed brace coding style, lines morethan 80 characters issues. Replaced printk with pr_err as pr_err is prefered than printk. Signed-off-by: Shahina Shaik <sharab.shaik(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 34 +++++++++++++++++++--------------- 1 files changed, 19 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117..9e45490 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -306,7 +306,7 @@ static inline int aic32x4_get_divs(int mclk, int rate) return i; } } - printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); + pr_err(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); return -EINVAL; } @@ -323,7 +323,7 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, aic32x4->sysclk = freq; return 0; } - printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); + pr_err(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); return -EINVAL; } @@ -349,7 +349,7 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_CBS_CFS: break; default: - printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + pr_err(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); return -EINVAL; } @@ -374,7 +374,7 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); break; default: - printk(KERN_ERR "aic32x4: invalid DAI interface format\n"); + pr_err(KERN_ERR "aic32x4: invalid DAI interface format\n"); return -EINVAL; } @@ -395,7 +395,7 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); if (i < 0) { - printk(KERN_ERR "aic32x4: sampling rate not supported\n"); + pr_err(KERN_ERR "aic32x4: sampling rate not supported\n"); return i; } @@ -626,32 +626,36 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } - if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); - } + tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? AIC32X4_LDOCTLEN : 0; snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) tmp_reg |= AIC32X4_LDOIN_18_36; - } - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) tmp_reg |= AIC32X4_LDOIN2HP; - } + snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); /* Mic PGA routing */ if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_IN2R_10K); else - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_CM1L_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_CM1L_10K); if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_IN1L_10K); else - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_CM1R_10K); aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

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[alsa-devel] [PATCH] ASoC: samsung: Add I2C dependency for snow
by Arnd Bergmann 04 Jun '14

04 Jun '14

Both codecs used by snow, max98090 and max98095 require the use of I2C, so we can only select this driver if I2C is there, otherwise we get a build error like: codecs/max98090.c:2494:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(max98090_i2c_driver); ^ codecs/max98095.c:2443:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(max98095_i2c_driver); ^ This adds one more I2C dependency to the hundreds we already have. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Tushar Behera <tushar.behera(a)linaro.org> --- diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a7cb8f6..cb7be4f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -232,7 +232,7 @@ config SND_SOC_LITTLEMILL config SND_SOC_SNOW tristate "Audio support for Google Snow boards" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SOC_MAX98095 select SND_SAMSUNG_I2S

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[alsa-devel] [PATCH v2] ASoC: arizona: Implement TDM support for Arizona devices
by Charles Keepax 04 Jun '14

04 Jun '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Removed check of TDM width to requested stream word length include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/arizona.c | 78 +++++++++++++++++++++++++++++++++++--- 2 files changed, 75 insertions(+), 6 deletions(-) diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 5cf8b91..11783b5 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -110,6 +110,9 @@ struct arizona { int clk32k_ref; struct snd_soc_dapm_context *dapm; + + int tdm_width[ARIZONA_MAX_AIF]; + int tdm_slots[ARIZONA_MAX_AIF]; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f..e77f61c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1185,7 +1185,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int base = dai->driver->base; const int *rates; int i, ret, val; + int channels = params_channels(params); int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int tdm_width = arizona->tdm_width[dai->id - 1]; + int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; if (params_rate(params) % 8000) @@ -1193,18 +1196,27 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - bclk_target = snd_soc_params_to_bclk(params); - if (chan_limit && chan_limit < params_channels(params)) { + if (tdm_slots) { + arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", + tdm_slots, tdm_width); + bclk_target = tdm_slots * tdm_width * params_rate(params); + channels = tdm_slots; + } else { + bclk_target = snd_soc_params_to_bclk(params); + } + + if (chan_limit && chan_limit < channels) { arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); - bclk_target /= params_channels(params); + bclk_target /= channels; bclk_target *= chan_limit; } - /* Force stereo for I2S mode */ + /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); - bclk_target *= 2; + bclk_target /= channels; + bclk_target *= channels + 1; } for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { @@ -1324,9 +1336,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) ARIZONA_AIF1_TRI, reg); } +static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, + unsigned int base, + int channels, unsigned int mask) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int slot, i; + + for (i = 0; i < channels; ++i) { + slot = ffs(mask) - 1; + if (slot < 0) + return; + + regmap_write(arizona->regmap, base + i, slot); + + mask &= ~(1 << slot); + } + + if (mask) + arizona_aif_warn(dai, "Too many channels in TDM mask\n"); +} + +static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int base = dai->driver->base; + int rx_max_chan = dai->driver->playback.channels_max; + int tx_max_chan = dai->driver->capture.channels_max; + + /* Only support TDM for the physical AIFs */ + if (dai->id > ARIZONA_MAX_AIF) + return -ENOTSUPP; + + if (slots == 0) { + tx_mask = (1 << tx_max_chan) - 1; + rx_mask = (1 << rx_max_chan) - 1; + } + + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3, + tx_max_chan, tx_mask); + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11, + rx_max_chan, rx_mask); + + arizona->tdm_width[dai->id - 1] = slot_width; + arizona->tdm_slots[dai->id - 1] = slots; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, + .set_tdm_slot = arizona_set_tdm_slot, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, .set_tristate = arizona_set_tristate, -- 1.7.2.5

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[alsa-devel] [PATCH v2 1/1] ALSA: seq: Continue broadcasting events to ports if one of them fails
by Adam Goode 04 Jun '14

04 Jun '14

Sometimes PORT_EXIT messages are lost when a process is exiting. This happens if you subscribe to the announce port with client A, then subscribe to the announce port with client B, then kill client A. Client B will not see the PORT_EXIT message because client A's port is closing and is earlier in the announce port subscription list. The for each loop will try to send the announcement to client A and fail, then will stop trying to broadcast to other ports. Killing B works fine since the announcement will already have gone to A. The CLIENT_EXIT message does not get lost. How to reproduce problem: *** termA $ aseqdump -p 0:1 0:1 Port subscribed 0:1 -> 128:0 *** termB $ aseqdump -p 0:1 *** termA 0:1 Client start client 129 0:1 Port start 129:0 0:1 Port subscribed 0:1 -> 129:0 *** termB 0:1 Port subscribed 0:1 -> 129:0 *** termA ^C *** termB 0:1 Client exit client 128 <--- expected Port exit as well (before client exit) Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 9ca5e64..225c7315 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -660,7 +660,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, int atomic, int hop) { struct snd_seq_subscribers *subs; - int err = 0, num_ev = 0; + int err, result = 0, num_ev = 0; struct snd_seq_event event_saved; struct snd_seq_client_port *src_port; struct snd_seq_port_subs_info *grp; @@ -685,8 +685,12 @@ static int deliver_to_subscribers(struct snd_seq_client *client, subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL); err = snd_seq_deliver_single_event(client, event, 0, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev++; /* restore original event record */ *event = event_saved; @@ -697,7 +701,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, up_read(&grp->list_mutex); *event = event_saved; /* restore */ snd_seq_port_unlock(src_port); - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } @@ -709,7 +713,7 @@ static int port_broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int num_ev = 0, err = 0; + int num_ev = 0, err, result = 0; struct snd_seq_client *dest_client; struct snd_seq_client_port *port; @@ -724,14 +728,18 @@ static int port_broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev++; } read_unlock(&dest_client->ports_lock); snd_seq_client_unlock(dest_client); event->dest.port = SNDRV_SEQ_ADDRESS_BROADCAST; /* restore */ - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } /* @@ -741,7 +749,7 @@ static int port_broadcast_event(struct snd_seq_client *client, static int broadcast_event(struct snd_seq_client *client, struct snd_seq_event *event, int atomic, int hop) { - int err = 0, num_ev = 0; + int err, result = 0, num_ev = 0; int dest; struct snd_seq_addr addr; @@ -760,12 +768,16 @@ static int broadcast_event(struct snd_seq_client *client, err = snd_seq_deliver_single_event(NULL, event, SNDRV_SEQ_FILTER_BROADCAST, atomic, hop); - if (err < 0) - break; + if (err < 0) { + /* save first error that occurs and continue */ + if (!result) + result = err; + continue; + } num_ev += err; } event->dest = addr; /* restore */ - return (err < 0) ? err : num_ev; + return (result < 0) ? result : num_ev; } -- 2.0.0.526.g5318336

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