May 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Regression post-ALSA 1.0.22 for Lenovo G555 Internal mic
by Roo - Andrew Baines 24 May '11

24 May '11

Dear All, I have been told I should send an email to this list from this bug<https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5367> (0005367). There is a regression in ALSA 1.0.23 and above for Lenovo sound. The internal mic no longer works. A helpful guy 'Raymond' on the bug identified that pin-ctls is correct in ALSA 1.0.21 and 1.0.22, but incorrect in higher versions due to SND_PCI_QUIRK_VENDOR. This is based on alsa-info.sh run on systems using different versions of ALSA but on the same system. The jacksense has never actually worked but I have always managed to use hda-verb to do this for me - the main issue is the regression with the internal mic. Regards Andrew

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[alsa-devel] [PATCH] ASoC: core - fix module reference counting for CPU DAIs
by Liam Girdwood 24 May '11

24 May '11

Currently CODEC and platform drivers have their module reference count incremented soc_probe_dai_link() whilst CPU DAI drivers have their reference count incremented in soc_bind_dai_link(). CPU DAIs should have their reference count incremented in soc_probe_dai_link() just like the CODEC and platform drivers. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5968745..f051e85 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1306,10 +1306,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { if (!strcmp(cpu_dai->name, dai_link->cpu_dai_name)) { - - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; - rtd->cpu_dai = cpu_dai; goto find_codec; } @@ -1621,12 +1617,17 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ + if (!cpu_dai->probed) { + if (!try_module_get(cpu_dai->dev->driver->owner)) + return -ENODEV; + if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n", cpu_dai->name); + module_put(cpu_dai->dev->driver->owner); return ret; } } -- 1.7.4.1

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[alsa-devel] [PATCH] ALSA: ASoC: fix raumfeld platform
by Daniel Mack 24 May '11

24 May '11

Commit f0fba2ad (ASoC: multi-component - ASoC Multi-Component Support) broke support for Raumfeld platforms as it didn't take into account the different hardware features on individual devices. In particular, Raumfeld speakers have no S/PDIF output, so the members of the snd_soc_card struct must be set dynamically. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/pxa/raumfeld.c | 92 +++++++++++++++++++++++++++++----------------- 1 files changed, 58 insertions(+), 34 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 2afabaf..1a591f1 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = { .hw_params = raumfeld_cs4270_hw_params, }; -static int raumfeld_line_suspend(struct snd_soc_card *card) +static int raumfeld_analog_suspend(struct snd_soc_card *card) { raumfeld_enable_audio(false); return 0; } -static int raumfeld_line_resume(struct snd_soc_card *card) +static int raumfeld_analog_resume(struct snd_soc_card *card) { raumfeld_enable_audio(true); return 0; @@ -225,32 +225,53 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .hw_params = raumfeld_ak4104_hw_params, }; -static struct snd_soc_dai_link raumfeld_dai[] = { +#define DAI_LINK_CS4270 \ +{ \ + .name = "CS4270", \ + .stream_name = "CS4270", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "cs4270-hifi", \ + .codec_name = "cs4270-codec.0-0048", \ + .ops = &raumfeld_cs4270_ops, \ +} + +#define DAI_LINK_AK4104 \ +{ \ + .name = "ak4104", \ + .stream_name = "Playback", \ + .cpu_dai_name = "pxa-ssp-dai.1", \ + .codec_dai_name = "ak4104-hifi", \ + .platform_name = "pxa-pcm-audio", \ + .ops = &raumfeld_ak4104_ops, \ + .codec_name = "spi0.0", \ +} + +static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { - .name = "ak4104", - .stream_name = "Playback", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "ak4104-hifi", - .platform_name = "pxa-pcm-audio", - .ops = &raumfeld_ak4104_ops, - .codec_name = "ak4104-codec.0", -}, + DAI_LINK_CS4270, + DAI_LINK_AK4104, +}; + +static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { - .name = "CS4270", - .stream_name = "CS4270", - .cpu_dai_name = "pxa-ssp-dai.0", - .platform_name = "pxa-pcm-audio", - .codec_dai_name = "cs4270-hifi", - .codec_name = "cs4270-codec.0-0048", - .ops = &raumfeld_cs4270_ops, -},}; - -static struct snd_soc_card snd_soc_raumfeld = { - .name = "Raumfeld", - .dai_link = raumfeld_dai, - .suspend_post = raumfeld_line_suspend, - .resume_pre = raumfeld_line_resume, - .num_links = ARRAY_SIZE(raumfeld_dai), + DAI_LINK_CS4270, +}; + +static struct snd_soc_card snd_soc_raumfeld_connector = { + .name = "Raumfeld Connector", + .dai_link = snd_soc_raumfeld_connector_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, +}; + +static struct snd_soc_card snd_soc_raumfeld_speaker = { + .name = "Raumfeld Speaker", + .dai_link = snd_soc_raumfeld_speaker_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, }; static struct platform_device *raumfeld_audio_device; @@ -271,22 +292,25 @@ static int __init raumfeld_audio_init(void) set_max9485_clk(MAX9485_MCLK_FREQ_122880); - /* Register LINE and SPDIF */ + /* Register analog device */ raumfeld_audio_device = platform_device_alloc("soc-audio", 0); if (!raumfeld_audio_device) return -ENOMEM; - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld); - ret = platform_device_add(raumfeld_audio_device); - - /* no S/PDIF on Speakers */ if (machine_is_raumfeld_speaker()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_speaker); + + if (machine_is_raumfeld_connector()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_connector); + + ret = platform_device_add(raumfeld_audio_device); + if (ret < 0) return ret; raumfeld_enable_audio(true); - - return ret; + return 0; } static void __exit raumfeld_audio_exit(void) -- 1.7.5.1

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[alsa-devel] Clarrification about the alsa volume init framework.
by Luke Yelavich 24 May '11

24 May '11

Hi, So I am looking at cleaning up Ubuntu's collection of volume init db data originally set up in Ubuntu's/Debian old big init script. The switch has already been made to the new init system, and any existing settings were migrated from our old init script to the new system, however this is somewhat unclean. Is my understanding correct in that the new volume init system was designed to have quirks for hardware that needs particular volume settings? If so, how would upstream like these to be submitted, and would you want to have several dozen or more quirk files in the init db for different hardware? Thanks in advance. Luke

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[alsa-devel] Problems with snd_hda_intel in Linux kernel 2.6.38
by Svante Signell 24 May '11

24 May '11

Now I'm subscribed to the list. It does not seem possible to get a message in without subscription. Would be nice if alsa had and used bug reporting pages outside the mailing list. > Trying again to get this message to the alsa-devel list, as recommended > by the Debian maintainer of the kernel, without being subscribed. Does > alsa not have a bug report page? > > Booting kernel 2.6.38 hangs at: > During boot of kernel 2.6.38 (and 2.6.37) udev bugs out: > Waiting for /dev to be fully populated > BUG: Unable to handle kernel paging request at ffffc90013cd8000 > axz_probe+ ... [snd_hda_intel] > ...lots of output lost... > udevadm timeout 180 sec ... > udevd[390]: worker [439] failed while handling > '/devices/pci0000:80/0000:80:01.0' > > No sound card is recognized after the timeout of the boot as shown above. > According to the /proc/asound/cards there are no sound cards > > See bug Debian bug #619034 for more information. > http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=619034 > > This problem was also reported as Debian bug #613979, as kernel ticket > > https://bugzilla.kernel.org/show_bug.cgi?id=30552 > > and sent to this mailing list in February 2011. > > snd_hda_intel works properly with 2.6.32-5-amd64 (and earlier kernels). >

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[alsa-devel] [PATCH v3 0/3] ASoC: add support for audio on iPaq hx4700
by Dmitry Artamonow 24 May '11

24 May '11

Ok, here's another version of patches adding audio support for good ole iPAQ hx4700. It took a bit faster to make v3 than I expected, so here is it. Previous versions can be found here: https://lkml.org/lkml/2010/11/20/50 (v1) https://lkml.org/lkml/2011/5/16/462 (v2) This patchset is based on 'for-next' branch of Mark's ASoC git tree (fetched today), also applies cleanly on linux-next-20110518. Lightly tested on real hardware - playback/recording/mixer seem to work without any glitches. Changes v2 -> v3 * data based init is now used in codec and machine audio driver for DAPM and controls * machine audio driver now is a proper platform driver, instantiating sound card with snd_soc_register_card() * platform code registers corresponding platform_device for machine audio driver * ak4641_sync() is dropped in favour of snd_soc_cache_sync() * snd_soc_update_bits() is used instead of snd_soc_read()/update/snd_soc_write() sequences. Also return value of snd_soc_update_bits is propagated further for catching errors (previously result of snd_soc_write wasn't checked) * gpio_request_array is used now in machine audio driver * some other minor cleanup in the couple of places Dmitry Artamonow (3): ASoC: Asahi Kasei AK4641 codec driver ASoC: add iPAQ hx4700 machine driver pxa/hx4700: add platform device and I2C info for AK4641 codec arch/arm/mach-pxa/hx4700.c | 25 ++ arch/arm/mach-pxa/include/mach/hx4700.h | 4 +- include/sound/ak4641.h | 26 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4641.c | 664 +++++++++++++++++++++++++++++++ sound/soc/codecs/ak4641.h | 47 +++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/hx4700.c | 255 ++++++++++++ 10 files changed, 1036 insertions(+), 2 deletions(-) create mode 100644 include/sound/ak4641.h create mode 100644 sound/soc/codecs/ak4641.c create mode 100644 sound/soc/codecs/ak4641.h create mode 100644 sound/soc/pxa/hx4700.c -- 1.7.4.rc3

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[alsa-devel] [PATCH] ASoC: Convert 16x16 write to use cpu_to_be16()
by Mark Brown 24 May '11

24 May '11

Make it clear what we're doing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 10 ++++------ 1 files changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 06b7b81..abdf8d1 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -277,14 +277,12 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[4]; + u16 data[2]; - data[0] = (reg >> 8) & 0xff; - data[1] = reg & 0xff; - data[2] = (value >> 8) & 0xff; - data[3] = value & 0xff; + data[0] = cpu_to_be16(reg); + data[1] = cpu_to_be16(value); - return do_hw_write(codec, reg, value, data, 4); + return do_hw_write(codec, reg, value, data, sizeof(data)); } /* Primitive bulk write support for soc-cache. The data pointed to by -- 1.7.5.1

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[alsa-devel] ALSA and MediaController
by pl bossart 23 May '11

23 May '11

During the ALSA/Embedded audio workshop two weeks ago we talked about using the MediaController API (merged in 2.6.39) to provide userspace with a graph-like representation of the transforms implemented in firmware or hardware. Two main applications were mentioned: - volume control, at the moment in PulseAudio we get a flat list of volume controls and we use a convoluted mixer logic to figure out if 'pcm playback volume' is done before or after 'master playback volume'. - tuning applications. With literally dozens of controls exposed in embedded solutions, a flat list is going to be very hard to use. It would make sense during the device tuning steps to figure out which controls is located where to better understand its impacts. This would help create UCM configuration files as well. At the moment the Media Controller does not provide means to set any values/parameters in a media_entity. We would need to cross-reference each media_entity with ALSA controls and use the existing control get/set interfaces. This raises 2 main questions: - Is the list of ALSA controls is completely defined after the card initialization step? Or do we have cases where ALSA controls are created dynamically after the init? I tend to believe the former is true but want to verify this with others on the mailing list. It's my understanding that only connections may be changed in such a graph. - Laurent suggested a new ioctl to export media_entity information to userspace, as a binary buffer containing a list of TLV (Type-Length-Value) field. However it looks like there's already an ELEM_BYTES control type. It might make more sense to expose this information using native ALSA controls, but again feedback from smart people on this list would be welcome. Thanks, -Pierre

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[alsa-devel] [PATCH v5] [resend] tea575x: convert to control framework
by Ondrej Zary 23 May '11

23 May '11

Convert tea575x-tuner to use the new V4L2 control framework. Also add ext_init() callback that can be used by a card driver for additional initialization right before registering the video device (for SF16-FMR2). Also embed struct video_device to struct snd_tea575x to simplify the code. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-05-13 19:39:23.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-05-19 17:27:45.000000000 +0200 @@ -23,8 +23,8 @@ */ #include <linux/videodev2.h> +#include <media/v4l2-ctrls.h> #include <media/v4l2-dev.h> -#include <media/v4l2-ioctl.h> #define TEA575X_FMIF 10700 @@ -42,7 +42,7 @@ struct snd_tea575x_ops { }; struct snd_tea575x { - struct video_device *vd; /* video device */ + struct video_device vd; /* video device */ bool tea5759; /* 5759 chip is present */ bool mute; /* Device is muted? */ bool stereo; /* receiving stereo */ @@ -54,6 +54,8 @@ struct snd_tea575x { void *private_data; u8 card[32]; u8 bus_info[32]; + struct v4l2_ctrl_handler ctrl_handler; + int (*ext_init)(struct snd_tea575x *tea); }; int snd_tea575x_init(struct snd_tea575x *tea); --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-05-13 19:39:23.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-05-19 17:31:45.000000000 +0200 @@ -22,11 +22,11 @@ #include <asm/io.h> #include <linux/delay.h> -#include <linux/interrupt.h> #include <linux/init.h> #include <linux/slab.h> #include <linux/version.h> -#include <sound/core.h> +#include <media/v4l2-dev.h> +#include <media/v4l2-ioctl.h> #include <sound/tea575x-tuner.h> MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); @@ -62,17 +62,6 @@ module_param(radio_nr, int, 0); #define TEA575X_BIT_DUMMY (1<<15) /* buffer */ #define TEA575X_BIT_FREQ_MASK 0x7fff -static struct v4l2_queryctrl radio_qctrl[] = { - { - .id = V4L2_CID_AUDIO_MUTE, - .name = "Mute", - .minimum = 0, - .maximum = 1, - .default_value = 1, - .type = V4L2_CTRL_TYPE_BOOLEAN, - } -}; - /* * lowlevel part */ @@ -266,47 +255,17 @@ static int vidioc_s_audio(struct file *f return 0; } -static int vidioc_queryctrl(struct file *file, void *priv, - struct v4l2_queryctrl *qc) +static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) { - int i; - - for (i = 0; i < ARRAY_SIZE(radio_qctrl); i++) { - if (qc->id && qc->id == radio_qctrl[i].id) { - memcpy(qc, &(radio_qctrl[i]), - sizeof(*qc)); - return 0; - } - } - return -EINVAL; -} - -static int vidioc_g_ctrl(struct file *file, void *priv, - struct v4l2_control *ctrl) -{ - struct snd_tea575x *tea = video_drvdata(file); + struct snd_tea575x *tea = container_of(ctrl->handler, struct snd_tea575x, ctrl_handler); switch (ctrl->id) { case V4L2_CID_AUDIO_MUTE: - ctrl->value = tea->mute; + tea->mute = ctrl->val; + snd_tea575x_set_freq(tea); return 0; } - return -EINVAL; -} -static int vidioc_s_ctrl(struct file *file, void *priv, - struct v4l2_control *ctrl) -{ - struct snd_tea575x *tea = video_drvdata(file); - - switch (ctrl->id) { - case V4L2_CID_AUDIO_MUTE: - if (tea->mute != ctrl->value) { - tea->mute = ctrl->value; - snd_tea575x_set_freq(tea); - } - return 0; - } return -EINVAL; } @@ -355,16 +314,17 @@ static const struct v4l2_ioctl_ops tea57 .vidioc_s_input = vidioc_s_input, .vidioc_g_frequency = vidioc_g_frequency, .vidioc_s_frequency = vidioc_s_frequency, - .vidioc_queryctrl = vidioc_queryctrl, - .vidioc_g_ctrl = vidioc_g_ctrl, - .vidioc_s_ctrl = vidioc_s_ctrl, }; static struct video_device tea575x_radio = { .name = "tea575x-tuner", .fops = &tea575x_fops, .ioctl_ops = &tea575x_ioctl_ops, - .release = video_device_release, + .release = video_device_release_empty, +}; + +static const struct v4l2_ctrl_ops tea575x_ctrl_ops = { + .s_ctrl = tea575x_s_ctrl, }; /* @@ -373,7 +333,6 @@ static struct video_device tea575x_radio int snd_tea575x_init(struct snd_tea575x *tea) { int retval; - struct video_device *tea575x_radio_inst; tea->mute = 1; @@ -384,40 +343,45 @@ int snd_tea575x_init(struct snd_tea575x tea->in_use = 0; tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; tea->freq = 90500 * 16; /* 90.5Mhz default */ + snd_tea575x_set_freq(tea); - tea575x_radio_inst = video_device_alloc(); - if (tea575x_radio_inst == NULL) { - printk(KERN_ERR "tea575x-tuner: not enough memory\n"); - return -ENOMEM; - } + tea->vd = tea575x_radio; + video_set_drvdata(&tea->vd, tea); - memcpy(tea575x_radio_inst, &tea575x_radio, sizeof(tea575x_radio)); + v4l2_ctrl_handler_init(&tea->ctrl_handler, 1); + tea->vd.ctrl_handler = &tea->ctrl_handler; + v4l2_ctrl_new_std(&tea->ctrl_handler, &tea575x_ctrl_ops, V4L2_CID_AUDIO_MUTE, 0, 1, 1, 1); + retval = tea->ctrl_handler.error; + if (retval) { + printk(KERN_ERR "tea575x-tuner: can't initialize controls\n"); + v4l2_ctrl_handler_free(&tea->ctrl_handler); + return retval; + } - strcpy(tea575x_radio.name, tea->tea5759 ? - "TEA5759 radio" : "TEA5757 radio"); + if (tea->ext_init) { + retval = tea->ext_init(tea); + if (retval) { + v4l2_ctrl_handler_free(&tea->ctrl_handler); + return retval; + } + } - video_set_drvdata(tea575x_radio_inst, tea); + v4l2_ctrl_handler_setup(&tea->ctrl_handler); - retval = video_register_device(tea575x_radio_inst, - VFL_TYPE_RADIO, radio_nr); + retval = video_register_device(&tea->vd, VFL_TYPE_RADIO, radio_nr); if (retval) { printk(KERN_ERR "tea575x-tuner: can't register video device!\n"); - kfree(tea575x_radio_inst); + v4l2_ctrl_handler_free(&tea->ctrl_handler); return retval; } - snd_tea575x_set_freq(tea); - tea->vd = tea575x_radio_inst; - return 0; } void snd_tea575x_exit(struct snd_tea575x *tea) { - if (tea->vd) { - video_unregister_device(tea->vd); - tea->vd = NULL; - } + video_unregister_device(&tea->vd); + v4l2_ctrl_handler_free(&tea->ctrl_handler); } static int __init alsa_tea575x_module_init(void) -- Ondrej Zary

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[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Don't sync first two registers from register cache
by Jarkko Nikula 23 May '11

23 May '11

There is no need to sync first two registers from cache to hw after a reset. First one is used to select page for register access and this driver is normally accessing page 0 only. Second one does a software reset which is obviously unneeded after hardware or previous software reset command. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c3d96fc..9047bb1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1114,7 +1114,7 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) /* Sync reg_cache with the hardware */ codec->cache_only = 0; - for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++) + for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++) snd_soc_write(codec, i, cache[i]); if (aic3x->model == AIC3X_MODEL_3007) aic3x_init_3007(codec); -- 1.7.4.4

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