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

[alsa-devel] [PATCH 1/2] ALSA: hda/hdmi - set depop_delay for haswell plus
by libin.yang＠intel.com 19 Aug '14

19 Aug '14

From: Libin Yang <libin.yang(a)intel.com> Both Haswell and Broadwell need set depop_delay to 0. So apply this setting to haswell plus. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 36badba..5e229f7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2330,9 +2330,8 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } - if (is_haswell(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview(codec)) codec->depop_delay = 0; - } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; -- 1.9.1

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[alsa-devel] [PATCH v2] ALSA: hda - restore the gpio led after resume
by Hui Wang 19 Aug '14

19 Aug '14

On some HP laptops, the mute led is controlled by codec gpio. When some machine resume from s3/s4, the codec gpio data will be cleared to 0 by BIOS: Before suspend: IO[3]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 After resume: IO[3]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 To skip the AFG node to enter D3 can't fix this problem. A workaround is to restore the gpio data when the system resume back from s3/s4. It is safe even on the machines without this problem. BugLink: https://bugs.launchpad.net/bugs/1358116 Tested-by: Franz Hsieh <franz.hsieh(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- V2: move the verb write from led_power_filter to alc269_resume sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b32ce08..d71270a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3268,6 +3268,15 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); + + /* on some machine, the BIOS will clear the codec gpio data when enter + * suspend, and won't restore the data after resume, so we restore it + * in the driver. + */ + if (spec->gpio_led) + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); + if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Set TLV_DB_SCALE_MUTE bit for cx5051 vmaster
by Takashi Iwai 19 Aug '14

19 Aug '14

Conexnat HD-audio driver has a workaround for cx5051 (aka CX20561) chip to add fake mute controls to each amp (commit 3868137e). This implies the minimum-as-mute TLV bit in TLV for each corresponding control. Meanwhile we build the virtual master from these, but the TLV bit is missing, even though the slaves have it. This patch simply adds the missing TLV_DB_SCALE_MUTE bit for vmaster, as already done in patch_sigmatel.c. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7627a69ca6d7..6f2fa838b635 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -26,6 +26,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/jack.h> +#include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" @@ -859,6 +860,11 @@ static int patch_conexant_auto(struct hda_codec *codec) if (err < 0) goto error; + if (codec->vendor_id == 0x14f15051) { + /* minimum value is actually mute */ + spec->gen.vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; + } + codec->patch_ops = cx_auto_patch_ops; /* Some laptops with Conexant chips show stalls in S3 resume, -- 2.0.4

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Re: [alsa-devel] [v4 10/12] ASoC: Intel: mrfld: add the DSP DAPM widgets
by Mark Brown 19 Aug '14

19 Aug '14

On Mon, Aug 18, 2014 at 09:58:11AM +0530, Subhransu S. Prusty wrote: > On Thu, Aug 14, 2014 at 02:07:08PM +0100, Mark Brown wrote: > > > for (i = 0; i < w->num_kcontrols; i++) { > > > if (dapm_kcontrol_get_value(w->kcontrols[i])) { > > > val |= 1 << i; > > > } > > > Is this ok? > > No, that doesn't seem safe. Why not read the data from the control - if > > we were calling a function which returned the value that'd be much > > clearer. > So lets step back and see what is required here. > I get a mixer update for one of the inputs of the mixer. This needs to be > communicated to the DSP via the IPC. But the IPC expects that we send all > the values of all the inputs of a mixer to the DSP. > That is why here we are reading the all mixer inputs to find the values and > send the bitmap of all inputs. > So from the driver if I have to find out what is the value of the inputs in > mixer, how do I do that, is there a simpler way? Or should we push > this/<something else> into framework to query The problem is that it looks like you are reading the value from the control, treating it as a boolean and discarding it.

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[alsa-devel] [PATCH 0/5] Convert to use regmap framework's endianness method.
by Xiubo Li 19 Aug '14

19 Aug '14

Changes in V2: - Modified the regmap config to const type. - Added ASRC patch. - Followed Rutland's advice. Xiubo Li (5): ASoC: fsl-asrc: Convert to use regmap framework's endianness method. ASoC: fsl-esai: Convert to use regmap framework's endianness method. ASoC: fsl-spdif: Convert to use regmap framework's endianness method. ASoC: fsl-sai: Convert to use regmap framework's endianness method. ASoC: fsl-sai: rename big_endian_data to is_msb_first. Documentation/devicetree/bindings/sound/fsl-sai.txt | 15 +++++++-------- sound/soc/fsl/fsl_asrc.c | 6 +----- sound/soc/fsl/fsl_esai.c | 5 +---- sound/soc/fsl/fsl_sai.c | 12 ++++-------- sound/soc/fsl/fsl_sai.h | 3 +-- sound/soc/fsl/fsl_spdif.c | 5 +---- 6 files changed, 15 insertions(+), 31 deletions(-) -- 1.8.5

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[alsa-devel] [PATCH v3 2/2] mfd: arizona: Update DT binding to support INn_MODE init_data
by Inha Song 19 Aug '14

19 Aug '14

This patch update DT binding to support INn_MODE init_data. Each input signal path can be configurated either as a Analogue or Digital using the INn_MODE registers. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- Documentation/devicetree/bindings/mfd/arizona.txt | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/Documentation/devicetree/bindings/mfd/arizona.txt b/Documentation/devicetree/bindings/mfd/arizona.txt index 5c7e723..0064b21 100644 --- a/Documentation/devicetree/bindings/mfd/arizona.txt +++ b/Documentation/devicetree/bindings/mfd/arizona.txt @@ -42,6 +42,14 @@ Optional properties: the chip default will be used. If present exactly five values must be specified. + - wlf,inmode : A list of INn_MODE register values, where n is the number + of input signals. Each input signal path can be configurated either as a + Analogue or Digital using the INn_MODE registers. If absent, INn_MODE + registers set to 0 by default. If present, values must be specified less + than or equal to the number of input singals. If values less than the + number of input signals, elements that has not been specifed are set to 0 + by default. + - DCVDD-supply, MICVDD-supply : Power supplies, only need to be specified if they are being externally supplied. As covered in Documentation/devicetree/bindings/regulator/regulator.txt -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH] ALSA: hda/hdmi - apply Valleyview fix-ups to Cherryview display codec
by libin.yang＠intel.com 19 Aug '14

19 Aug '14

From: Libin Yang <libin.yang(a)intel.com> Valleyview and Cherryview have the same behavior on display audio. So this patch defines is_valleyview_plus() to include codecs for both Valleyview and its successor Cherryview, and apply Valleyview fix-ups to Cherryview. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 36badba..99d7d7f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -50,6 +50,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec)) #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) +#define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) +#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1459,7 +1461,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell_plus(codec) || is_valleyview(codec)) + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1598,7 +1600,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * and this can make HW reset converter selection on a pin. */ if (eld->eld_valid && !old_eld_valid && per_pin->setup) { - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || + is_valleyview_plus(codec)) { intel_verify_pin_cvt_connect(codec, per_pin); intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); @@ -1779,7 +1782,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; int pinctl; - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { /* Verify pin:cvt selections to avoid silent audio after S3. * After S3, the audio driver restores pin:cvt selections * but this can happen before gfx is ready and such selection @@ -2330,9 +2333,8 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } - if (is_haswell(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) codec->depop_delay = 0; - } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; -- 1.9.1

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[alsa-devel] Warning Big Yield
by Joël Krähemann 19 Aug '14

19 Aug '14

I'd really like to know if there's a common way to do syncing applications. What frequencies are required to run a GUI? How does Xorg work and how can I prevent interfering with ALSA output? And what base frequency is recommended to threads? Warning you may experience big yield. please take a look at ags_thread_loop() in: http://sourceforge.net/p/ags/code/HEAD/tree/src/ags/thread/ags_thread-posix…

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[alsa-devel] [PATCH alsa-lib] Sync enum snd_hwdep_iface_t with include/asound/asound.h
by Takashi Sakamoto 19 Aug '14

19 Aug '14

Some members in this enumerated type has not updated for 9 years, although kernel-drivers added them during this period. This commit adds them following to a commit 87df9f3 'sync include/asound/asound.h with 3.17-rc1 kernel'. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/hwdep.h | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/include/hwdep.h b/include/hwdep.h index ab12822..6ec421c 100644 --- a/include/hwdep.h +++ b/include/hwdep.h @@ -68,8 +68,13 @@ typedef enum _snd_hwdep_iface { SND_HWDEP_IFACE_USX2Y_PCM, /**< Tascam US122, US224 & US428 raw USB PCM */ SND_HWDEP_IFACE_PCXHR, /**< Digigram PCXHR */ SND_HWDEP_IFACE_SB_RC, /**< SB Extigy/Audigy2NX remote control */ - - SND_HWDEP_IFACE_LAST = SND_HWDEP_IFACE_SB_RC /**< last known hwdep interface */ + SNDRV_HWDEP_IFACE_HDA, /* HD-audio */ + SNDRV_HWDEP_IFACE_USB_STREAM, /* direct access to usb stream */ + SNDRV_HWDEP_IFACE_FW_DICE, /* TC DICE FireWire device */ + SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */ + SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ + + SND_HWDEP_IFACE_LAST = SND_HWDEP_IFACE_FW_BEBOB /**< last known hwdep interface */ } snd_hwdep_iface_t; /** open for reading */ -- 1.9.1

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[alsa-devel] [PATCH v3] ASOC: dapm: add code to configure dai link parameters
by Nikesh Oswal 18 Aug '14

18 Aug '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 | 4 +- sound/soc/soc-dapm.c | 140 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 141 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471..3ee031e 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); @@ -531,6 +532,8 @@ 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 ed9e2d7..51c6c4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -906,6 +906,7 @@ 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 b87d7d8..db1572a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1461,7 +1461,7 @@ 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); @@ -1473,7 +1473,7 @@ 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 cdc837e..41f835b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -729,6 +729,33 @@ 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; + + + /* 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) @@ -2664,6 +2691,9 @@ 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; } @@ -3142,6 +3172,9 @@ 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 += 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; @@ -3222,8 +3255,35 @@ 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); + + 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) { @@ -3231,12 +3291,44 @@ 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; + 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 = kzalloc(sizeof(char *) * num_params, GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; + + for (count = 0 ; count < num_params; count++) { + w_param_text[count] = kzalloc(sizeof(char) * 100, GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto outfree_w_param; + } + + sprintf(w_param_text[count], "SampleRate=%d:Channels=%d:BitsPerSample=%d", + config->rate_min, config->channels_min, + snd_pcm_format_width(ffs(config->formats) - 1)); + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = (const char * const *) w_param_text; 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); memset(&template, 0, sizeof(template)); @@ -3246,6 +3338,25 @@ 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; + private_value = + (unsigned long) kmemdup((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; + template.kcontrol_news = kmemdup(&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); @@ -3253,15 +3364,34 @@ 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++) { + if (w_param_text[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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