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

[alsa-devel] [PATCH v8 0/2] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 02 Jan '15

02 Jan '15

The NXP TDA998x HDMI transmitter may transmit audio to the HDMI link from 2 different sources, I2S and S/PDIF. This patch set first adds an interface between a HDMI transmitter and the HDMI CODEC. The interface is then used by the TDA998x driver to describe its audio capabilities (DAIs), to give the audio constraints of the HDMI device (EDID) to the audio subsystem, and to connect the chosen audio source to the HDMI link. v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (2): ASoC: hdmi: Add a transmitter interface to the HDMI CODEC drm/i2c: tda998x: Use the HDMI audio CODEC interface .../devicetree/bindings/drm/i2c/tda998x.txt | 18 ++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 264 ++++++++++++++++++++- include/sound/hdmi.h | 20 ++ sound/soc/codecs/hdmi.c | 176 +++++++++++++- 5 files changed, 460 insertions(+), 19 deletions(-) create mode 100644 include/sound/hdmi.h -- 2.1.1

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[alsa-devel] [PATCH 0/2 v7] Digidesign Mbox 1 enhancements
by Damien Zammit 30 Dec '14

30 Dec '14

Revision 7. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign Mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 41 +++++++++++---- 3 files changed, 170 insertions(+), 9 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v5 0/4] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 29 Dec '14

29 Dec '14

This patch set adds multi-CODEC support to the simple card. v5: accept only "sound-dai" as the dai link property (Arnd Bergmann) v4: (Mark Brown) - document the new core function - calculation of the number of codecs moved to the core function - split the patch about simple-card.c - less code changes - remove the cast of pointers - less changes in the DT binding example 2 v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (4): ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: simple-card: add multi-CODECs in DT ASoC: simple-card: Remove useless check .../devicetree/bindings/sound/simple-card.txt | 18 +--- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 87 ++++++++-------- sound/soc/soc-core.c | 110 ++++++++++++++++++--- 4 files changed, 149 insertions(+), 69 deletions(-) -- 2.1.3

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[alsa-devel] [PATCH v3 0/2] Disable the ThinkPad HW mute/level control when possible
by Andy Lutomirski 11 Dec '14

11 Dec '14

This series is being resurrected after several years, and it's quite different from before, so I'm restarting the version numbers :) ThinkPad volume and mute controls are a mess. For whatever reason, ThinkPads have mute buttons that send KEY_MUTE *and* control an invisible-to-ALSA mute switch. Some of them have volume controls that interact with this switch as well. This is a perennial source of problems. On most ThinkPads, if you press mute and then unmute using GUI controls, you have no sound, because userspace and ALSA state gets out of sync with the hardware switch. There's a separate "sound card" that exposes the hardware switch, but userspace code generally doesn't understand that. There are already a few _OSI(Linux) overrides to turn all the hardware buttons into regular buttons. Rather than quirking ACPI everywhere, just teach thinkpad-acpi to program the buttons for full software control and to disable hardware controls. That allows us to remove the ACPI quirks and have normal mute controls. This approach should be much simpler than adding even more kludgey ALSA integration for questionable gain. Tested on an X200s (with latching mute by default) and an X220 (with a mute light and toggle mute by default). Everything works as expected. Changes from v2: - Don't try to change the mute mode on IBM ThinkPads. - Further minor cleanups. - Try to restore the mode on module unload. Changes from v1: - Simplified the code a bit. - Improved suspend/hibernate behavior. Andy Lutomirski (2): thinkpad-acpi: Try to use full software mute control acpi: Remove _OSI(Linux) for ThinkPads drivers/acpi/blacklist.c | 54 ---------------- drivers/platform/x86/thinkpad_acpi.c | 116 ++++++++++++++++++++++++++++++++--- 2 files changed, 106 insertions(+), 64 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH v7] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 05 Dec '14

05 Dec '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 154 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 157 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3a4d7da..9b62457 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -533,6 +534,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..db60701 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -942,6 +942,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56..f4956e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1292,7 +1292,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1304,7 +1305,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9c..7e961ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -754,6 +754,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2721,6 +2751,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3206,6 +3239,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config = &config[w->params_select]; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3274,8 +3310,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3283,14 +3350,52 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3298,6 +3403,28 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3305,15 +3432,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

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[alsa-devel] usb-audio: I'd like to change two things in usb audio, but I'm not sure if it's right
by Johan Aires Rastén 05 Dec '14

05 Dec '14

So I'll try again, this time with the proper cc:s and a second question. I've come across two issues in the usb audio module that are causing problems with my sound card, but I'm thinking that changing them might break something for some other card. In sound/usb/mixer.c function check_input_term(...), for the case UAC_INPUT_TERMINAL and UAC_VERSION_2: The recursive call to get clock selectors messes up the naming of my terminals. At this stage the terminals are already properly named Mic and Line but the call overwrites them both with the unnamed clock selector "Unit 22". Are other UAC2 cards different, so that the clock selector contains the correct terminal name, or is there some other reasoning here that I'm not seeing? Could the call be removed, or perhaps only done if no terminal name has been found yet? In sound/usb/stream.c function snd_usb_add_audio_stream(...), near the end of the function: A newly created pcm is added with "list_add(&as->list, &chip->pcm_list);" which causes devices to be listed in reverse order, i.e. device 0 will be at the end of the list, and this leads to that capture streams end up in the wrong device under certain circumstances. The problem arises when the code under "look for an empty stream" earlier in the same function executes. It will attach the capture stream to the first compatible existing stream, but since they are reversed it will find the highest numbered device. In my case analog input will be grouped with digital out (device 1) instead of analog out (device 0). This is assuming that playback streams are defined before capture streams but it could happen in the other direction as long as there are several streams of one type. I also suspect that this is what causes the M-Audio Audiophile USB to have swapped inputs, for which there's a fix in USB-Audio.conf ( http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/USB-Aud… ). Thanks to debianuser on IRC for helping me find it. I would like to change this call to use list_add_tail for adding new items instead, but I'm worried that there are quirks or other fixes that expect the current behaviour. There are very few other references to pcm_list and none of them seem to care about ordering. Sorry for being bad at referencing code, I'm new at this. //Johan Rastén

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[alsa-devel] [PATCH] ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing
by Peter Rosin 04 Dec '14

04 Dec '14

From: Peter Rosin <peda(a)axentia.se> If a snd_soc_card has any DAPM routes when it calls snd_soc_of_parse_audio_routing, those are clobbered without this change. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-core.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56ebc0f..377db486852e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4585,7 +4585,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes; + int num_routes, old_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -4603,7 +4603,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), + old_routes = card->num_dapm_routes; + routes = devm_kzalloc(card->dev, + (old_routes + num_routes) * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -4611,9 +4613,11 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } + memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); + for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[i].sink); + 2 * i, &routes[old_routes + i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4621,7 +4625,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[i].source); + (2 * i) + 1, &routes[old_routes + i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4630,7 +4634,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; + card->num_dapm_routes += num_routes; card->dapm_routes = routes; return 0; -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: rt5677: Add ACPI device probing
by Ben Zhang 04 Dec '14

04 Dec '14

The rt5677 codec driver looks for ACPI device ID "RT5677CE", which is specified in coreboot. This patch allows platform data to be obtained via ACPI Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677.c | 52 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 50 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d317c68..384281d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -20,6 +20,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <linux/firmware.h> #include <linux/gpio.h> #include <sound/core.h> @@ -4525,6 +4526,43 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) return 0; } +#ifdef CONFIG_ACPI + +static unsigned long long rt5677_parse_acpi_entry(struct device *dev, + acpi_string name) +{ + acpi_handle handle = ACPI_HANDLE(dev); + unsigned long long val; + acpi_status status; + + status = acpi_evaluate_integer(handle, name, NULL, &val); + if (ACPI_FAILURE(status)) { + dev_err(dev, "Failed to parse ACPI entry %s, default to 0: %d\n", + name, status); + return 0; + } + return val; +} + +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ + rt5677->pdata.dmic2_clk_pin = (enum rt5677_dmic2_clk) + rt5677_parse_acpi_entry(dev, "DCLK"); + rt5677->pdata.in1_diff = (bool)rt5677_parse_acpi_entry(dev, "IN1"); + rt5677->pdata.in2_diff = (bool)rt5677_parse_acpi_entry(dev, "IN2"); + rt5677->pdata.lout1_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT1"); + rt5677->pdata.lout2_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT2"); + rt5677->pdata.lout3_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT3"); + rt5677->pdata.jd1_gpio = rt5677_parse_acpi_entry(dev, "JD1"); + rt5677->pdata.jd2_gpio = rt5677_parse_acpi_entry(dev, "JD2"); + rt5677->pdata.jd3_gpio = rt5677_parse_acpi_entry(dev, "JD3"); +} +#else +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ +} +#endif + static struct regmap_irq rt5677_irqs[] = { [RT5677_IRQ_JD1] = { .reg_offset = 0, @@ -4604,6 +4642,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (pdata) rt5677->pdata = *pdata; + rt5677->pow_ldo2 = -EINVAL; if (i2c->dev.of_node) { ret = rt5677_parse_dt(rt5677, i2c->dev.of_node); if (ret) { @@ -4611,8 +4650,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, ret); return ret; } - } else { - rt5677->pow_ldo2 = -EINVAL; + } else if (ACPI_HANDLE(&i2c->dev)) { + rt5677_parse_acpi(rt5677, &i2c->dev); } if (gpio_is_valid(rt5677->pow_ldo2)) { @@ -4708,10 +4747,19 @@ static int rt5677_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5677_acpi_id[] = { + { "RT5677CE", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_acpi_id); +#endif + static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5677_acpi_id), }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, -- 2.1.0.rc2.206.gedb03e5

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[alsa-devel] [PATCH v2] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 03 Dec '14

03 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Switch to use TLV_DB_MINMAX as this allows the minimum to be off by only 0.005dB. This is an improvement from the ~500dB is was misreported by before. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..a4c71f9 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -538,9 +538,9 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); -- 2.1.2.330.g565301e

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[alsa-devel] Analog sound (ALC262) lost on linux (any distro and kernel) on a dual boot system
by Ploumistos Alexandros 03 Dec '14

03 Dec '14

Hello, A while back I had installed Ubuntu on a friend's laptop -a Sony Vaio VGN-FW21M- alongside windows Vista. Everything worked fine for a few years, until at some point last year linux went mute, while in windows both the analog and the digital outputs continued working. She couldn't remember doing something "out of the ordinary", but she was sure it was not after an update. Several months later, she upgraded to windows 7 but that didn't change anything. When I got my hands on the laptop, I checked for muted controls and other usual suspects, deprecated quirks (there were none) and all inputs and outputs of the system. HDMI audio works fine and so does the internal mic. Speakers, headphones and the external mic input do not work, even though pulse audio volume meter acts as if there were actually sounds being played. I followed all of Ubuntu's troubleshooting procedures, I removed and reinstalled alsa and pulse, I purged all configuration files, I created a new user and I even installed the latest alsa-packages from the "ALSA daily build snapshots" repository and nothing worked. I also tried booting into windows, setting all volume levels to the max and disabled power saving on all sound and multimedia devices. Next, I tried retasking the jacks one by one to no avail. I remembered that I had performed the installation with an Ubuntu 11.10 live CD, so I booted the computer off of that, but this time I got no sound. I tried with all the live CDs of a multitude of distros I had lying around, with kernels ranging from 2.6.35 to 3.16.6 and none of them worked, so I figured the issue must have something to do with windows. Is it possible that windows somehow started locking the audio chip on shutdown? If that is the case, can it be deduced from the logs and more importantly, can it be fixed? I am attaching the output of alsa-info and lspci as well as kernel logs. Thank you for your time.

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