January 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] imac27 12,2 (2011) model for patch_cirrus.c
by Jérémy Lal 09 Jan '12

09 Jan '12

Hi, it'd be glad to have patch_cirrus.c recognize model=imac27_122, (i'm adding a model because it's harder to auto-detect it), please find attached patch, tested on linux 3.2-rc7. I don't know anything about pin configs, just that it works with the ones in the patch. Note that i tried with model=auto (no sound) and model=imac27 (front speakers not muted when headphones are plugged in, and surround speakers not properly detected). Regards, Jérémy.

2 9

[alsa-devel] TWL4030 low-latency
by Edgar Berdahl 09 Jan '12

09 Jan '12

Dear Alsa Devel List, I would like to send some feedback on the TWL4030 driver when used on the Beagle Board. We have tested it under Ubuntu and Angstrom, and we find it does not work well at low latencies (e.g. input-to-output in 5ms, 10ms, or 20ms). Perhaps it was never tested extensively at these latencies. I thought I would send you this information in case you would look into it, or at least pass it on to the relevant contact. I can also run further tests if someone informs me what I should try. 1) To get the drivers to run at particularly low latencies (e.g. 20ms), it is necessary to operate the drivers in NON-realtime mode. It would interest me to know whether this means that the realtime functionality of the Beagle Board xM hardware or OS is hampered, or whether the TWL4030 driver is not operating up to specification. (If instead the TWL4030 driver is operating in realtime mode, then Jack stops running when an unrelated low-latency USB-serial device is opened.) 2) Some audio software (e.g. Pure Data, ChucK, etc.) will not connect to the TWL4030 ALSA drivers directly, instead they will only start when jack intermediates between the audio software and the TWL4030 ALSA drivers. 3) In some situations, the TWL4030 drivers appear not to start using the requested settings. For example, when we start jack requesting two periods per buffer, it is apparently getting initialized at eight periods per buffer, see the final lines below: Calling $ jackd -r -dalsa -dhw:0 -p128 -n2 -i2 -o2 -s -S -r on the Beagle under Ubuntu 10.10 and Angstrom demo image results in the following jackd 0.118.0 Copyright 2001-2009 Paul Davis, Stephane Letz, Jack O'Quinn, Torben Hohn and ot. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. loading driver .. apparent rate = 44100 creating alsa driver ... hw:0|hw:0|128|2|44100|2|2|nomon|swmeter|soft-mode|16bit control device hw:0 configuring for 44100Hz, period = 128 frames (2.9 ms), buffer = 2 periods ALSA: final selected sample format for capture: 16bit little-endian ALSA: use **8** periods for capture ALSA: final selected sample format for playback: 16bit little-endian ALSA: use **8** periods for playback Thanks very much for listening! Best, Edgar Berdahl Lecturer Center for Computer Research in Music and Acoustics (CCRMA) Stanford University http://ccrma.stanford.edu/courses/250a

6 8

[alsa-devel] [PATCH v2 4/4] ASoC: OMAP: ams-delta: drop .set_bias_level callback
by Janusz Krzysztofik 09 Jan '12

09 Jan '12

This functionality has just been implemented in the cx20442 codec driver, no need to keep it here duplicated. Once done, remove the no longer used AMS_DELTA_LATCH2_MODEM_NRESET symbol from the board header file and a call to the regulator_toggle() helper function from the old API wrapper found in the board file. While being at it, simplify the way the modem .pm callback handles the regulator, and drop that no longer used helper function and its related consumer setup elements completely. Depends on patch 3/4 "ARM: OMAP1: ams-delta: update the modem to use regulator API" Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes against initial version: * don't move consumer setup elements, now named to indicated their modem related purpose, down the file, * don't track the regulator enavble/disable state, compare new target power state with the old one instead; thanks to Mark Brown for suggesting this simplification, * actually drop all references to AMS_DELTA_LATCH2_MODEM_NRESET. arch/arm/mach-omap1/board-ams-delta.c | 42 +++----------------- arch/arm/plat-omap/include/plat/board-ams-delta.h | 1 - sound/soc/omap/ams-delta.c | 32 ---------------- 3 files changed, 7 insertions(+), 68 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 3389237..704fee6 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -18,7 +18,6 @@ #include <linux/input.h> #include <linux/interrupt.h> #include <linux/leds.h> -#include <linux/mutex.h> #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/regulator/fixed.h> @@ -291,35 +290,8 @@ static struct platform_device modem_nreset_device = { struct modem_private_data { struct regulator *regulator; - struct { - struct mutex lock; - bool enabled; - } consumer; }; -static int regulator_toggle(struct modem_private_data *priv, bool enable) -{ - int err = 0; - - mutex_lock(&priv->consumer.lock); - if (IS_ERR(priv->regulator)) { - err = PTR_ERR(priv->regulator); - } else if (enable) { - if (!priv->consumer.enabled) { - err = regulator_enable(priv->regulator); - priv->consumer.enabled = true; - } - } else { - if (priv->consumer.enabled) { - err = regulator_disable(priv->regulator); - priv->consumer.enabled = false; - } - } - mutex_unlock(&priv->consumer.lock); - - return err; -} - static struct modem_private_data modem_priv; void ams_delta_latch_write(int base, int ngpio, u16 mask, u16 value) @@ -330,8 +302,6 @@ void ams_delta_latch_write(int base, int ngpio, u16 mask, u16 value) for (; bit < ngpio; bit++, bitpos = bitpos << 1) { if (!(mask & bitpos)) continue; - else if (base + bit == AMS_DELTA_GPIO_PIN_MODEM_NRESET) - regulator_toggle(&modem_priv, (value & bitpos) != 0); else gpio_set_value(base + bit, (value & bitpos) != 0); } @@ -530,10 +500,16 @@ static void modem_pm(struct uart_port *port, unsigned int state, unsigned old) { struct modem_private_data *priv = port->private_data; + if (IS_ERR(priv->regulator)) + return; + if (state == old) return; - regulator_toggle(priv, state == 0); + if (state == 0) + regulator_enable(priv->regulator); + else if (old == 0) + regulator_disable(priv->regulator); } static struct plat_serial8250_port ams_delta_modem_ports[] = { @@ -593,7 +569,6 @@ static int __init late_init(void) gpio_direction_input(AMS_DELTA_GPIO_PIN_MODEM_IRQ); /* Initialize the modem_nreset regulator consumer before use */ - mutex_init(&modem_priv.consumer.lock); modem_priv.regulator = ERR_PTR(-ENODEV); ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, @@ -606,9 +581,6 @@ static int __init late_init(void) /* * Once the modem device is registered, the modem_nreset * regulator can be requested on behalf of that device. - * In addition to the modem .pm callback, that regulator - * is still used via the ams_delta_latch_write() wrapper - * by the ASoC driver until updated. */ modem_priv.regulator = regulator_get(&ams_delta_modem_device.dev, "RESET#"); diff --git a/arch/arm/plat-omap/include/plat/board-ams-delta.h b/arch/arm/plat-omap/include/plat/board-ams-delta.h index 027e79e..ad6f865 100644 --- a/arch/arm/plat-omap/include/plat/board-ams-delta.h +++ b/arch/arm/plat-omap/include/plat/board-ams-delta.h @@ -30,7 +30,6 @@ #define AMD_DELTA_LATCH2_SCARD_RSTIN 0x0400 #define AMD_DELTA_LATCH2_SCARD_CMDVCC 0x0800 -#define AMS_DELTA_LATCH2_MODEM_NRESET 0x1000 #define AMS_DELTA_LATCH2_MODEM_CODEC 0x2000 #define AMS_DELTA_GPIO_PIN_KEYBRD_DATA 0 diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 0462c7e..d213d04 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -426,29 +426,6 @@ static struct snd_soc_ops ams_delta_ops = { }; -/* Board specific codec bias level control */ -static int ams_delta_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - switch (level) { - case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: - case SND_SOC_BIAS_STANDBY: - if (card->dapm.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 (card->dapm.bias_level != SND_SOC_BIAS_OFF) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, - 0); - } - card->dapm.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; @@ -512,9 +489,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) ams_delta_ops.shutdown = ams_delta_shutdown; } - /* Set codec bias level */ - ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); - /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ ret = snd_soc_jack_new(rtd->codec, "hook_switch", @@ -597,7 +571,6 @@ static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", .dai_link = &ams_delta_dai_link, .num_links = 1, - .set_bias_level = ams_delta_set_bias_level, }; /* Module init/exit */ @@ -646,11 +619,6 @@ static void __exit ams_delta_module_exit(void) 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, - &ams_delta_audio_card.rtd[0].codec->dapm, - SND_SOC_BIAS_STANDBY); - platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); } -- 1.7.3.4

3 2

[alsa-devel] [PATCH - ice1724 1/4] ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
by pavel.hofman＠ivitera.com 09 Jan '12

09 Jan '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Add the capture pcm only if there is at least one ADC configured in the SYSCONF register. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/pci/ice1712/ice1724.c b/pci/ice1712/ice1724.c index 4353e76..2a0be2c 100644 --- a/pci/ice1712/ice1724.c +++ b/pci/ice1712/ice1724.c @@ -1117,14 +1117,18 @@ static struct snd_pcm_ops snd_vt1724_capture_pro_ops = { static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) { struct snd_pcm *pcm; - int err; + int capt, err; - err = snd_pcm_new(ice->card, "ICE1724", device, 1, 1, &pcm); + capt = (~ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_ADC_MASK) ? + 1 : 0; + err = snd_pcm_new(ice->card, "ICE1724", device, 1, capt, &pcm); if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_vt1724_playback_pro_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_vt1724_capture_pro_ops); + if (capt) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_vt1724_capture_pro_ops); pcm->private_data = ice; pcm->info_flags = 0; -- 1.7.0.4

4 15

[alsa-devel] [PATCH] Revert "ASoC: Fix build dependency for SND_ATMEL_SOC_SSC"
by Axel Lin 09 Jan '12

09 Jan '12

This reverts commit 739be96ab83755e10fd0c2b6a34c8a73254527f7 because it introduces below build warnings: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/Kconfig | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a4d6742..d1fcc81 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -9,7 +9,6 @@ config SND_ATMEL_SOC config SND_ATMEL_SOC_SSC tristate depends on SND_ATMEL_SOC - select ATMEL_SSC help Say Y or M if you want to add support for codecs the ATMEL SSC interface. You will also needs to select the individual -- 1.7.5.4

2 4

[alsa-devel] [PATCH] ALSA au88x0: Fix IRQ fifo error and channels swap of 4 channels playback
by Raymond Yau 09 Jan '12

09 Jan '12

[PATCH ] ALSA au88x0 - Fix IRQ fifo error and channels swap of 4 channels playback - Fix IRQ fifo error when playing stereo by set stereo flag of fifo control This also fix the swap of front and rear channels on au8830 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1840 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1726

3 5

[alsa-devel] [PATCH] ASoC: introduce SOC_SINGLE_S8_TLV macro
by Patrick Lai 09 Jan '12

09 Jan '12

ASoC currently has macro for double 8-bit signed TLV. Enhance info,get,put callback functions of SOC_DOUBLE_S8_TLV for single single 8-bit signed TLV. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- include/sound/soc.h | 10 ++++++++++ sound/soc/soc-core.c | 28 ++++++++++++++++++++-------- 2 files changed, 30 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 11cfb59..558e9ff 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -55,6 +55,16 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .min = xmin, .max = xmax, \ + .platform_max = xmax} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5d3685..3d1ba5f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2399,13 +2399,15 @@ int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; int platform_max; int min = mc->min; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; if (!mc->platform_max) mc->platform_max = mc->max; platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; + uinfo->count = shift == rshift ? 1 : 2; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; return 0; @@ -2428,13 +2430,16 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; int min = mc->min; int val = snd_soc_read(codec, reg); ucontrol->value.integer.value[0] = - ((signed char)(val & 0xff))-min; - ucontrol->value.integer.value[1] = - ((signed char)((val >> 8) & 0xff))-min; + ((signed char)((val >> shift) & 0xff))-min; + if (shift != rshift) + ucontrol->value.integer.value[1] = + ((signed char)((val >> rshift) & 0xff))-min; return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); @@ -2455,13 +2460,20 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; int min = mc->min; - unsigned int val; + unsigned int val, val2, val_mask; - val = (ucontrol->value.integer.value[0]+min) & 0xff; - val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + val = ((ucontrol->value.integer.value[0]+min) & 0xff) << shift; + val_mask = 0xff << shift; + if (shift != rshift) { + val2 = (ucontrol->value.integer.value[1]+min) & 0xff; + val |= val2 << rshift; + val_mask |= 0xff << rshift; + } - return snd_soc_update_bits_locked(codec, reg, 0xffff, val); + return snd_soc_update_bits_locked(codec, reg, val_mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); -- 1.7.3.3 -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum. --

3 5

[alsa-devel] [PATCH] ASoC: Fix idma build after update for channel count check
by Mark Brown 09 Jan '12

09 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/idma.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 3ba6aba..c227c31 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -398,6 +398,7 @@ static int idma_new(struct snd_soc_pcm_runtime *rtd) if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = preallocate_idma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); + } return ret; } -- 1.7.7.3

3 2

Re: [alsa-devel] Not able to restart audio after hibernation
by Mark Brown 09 Jan '12

09 Jan '12

On Mon, Jan 09, 2012 at 08:46:45AM +0530, Shiraz Hashim wrote: > On Sat, Jan 07, 2012 at 02:32:05AM +0800, Mark Brown wrote: > > This suggests that you have a bug in your suspend and resume > > implementation, possibly outside of this driver. > The driver works absolutely fine in case of suspend to ram where the > power domains (which includes the I2S controller) is switched off. That doesn't mean everything is OK, for example the clock tree or DMA controller might not be preserving state properly. > As the soc-core.c misses hibernation call backs of dev_pm_ops, we were > wondering whether it has been tested before on some other platform. Not to my knowledge, but I don't see any reason why anything more than the trivial implementation should really be needed except for the general pointlessness of resuming after hibernating.

1 0

[alsa-devel] cannot submit syncpipe, usb2 audio
by alexander 08 Jan '12

08 Jan '12

|I asked in alsa-users about the state of usb2 audio class compliance some time ago but I never got a definitive answer.. I've done some digging on my own and found someone had a similar issue with the same error about a year ago. This is what dmesg says when I connect the device in usb2 mode, the last row is the error when I try to use it. [ 2384.749399] usb 1-2: USB disconnect, device number 2 [ 2390.729825] usb 1-2: new high speed USB device number 3 using ehci_hcd [ 2390.863654] generic-usb 0003:0926:0202.0003: hiddev0,hidraw0: USB HID v1.01 Device [Sound Devices USBPre2] on usb-0000:00:02.1-2/input3 [ 2418.388135] cannot submit syncpipe for urb 0, error -90: internal error Here are the lsusb -v as usb1.1 http://pastebin.com/Jj6mXHxJ and here as usb2 http://pastebin.com/8uCBAkNb I've looked at them and the only thing that differs is the|bInterval, wich is 1 in usb1.1 and 4 in usb2

1 1