December 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: dapm: Replace siggen widget type with mic widgets
by Charles Keepax 02 Dec '15

02 Dec '15

The siggen widget type doesn't really require its own special handling. It can be handled identically to a mic widget, ie. it is a source of a signal and causes things to power up when an output is connected. Additionally, several places (cs42l52, cs42l56, wm8962, arizona_haptics) tie siggens in to the input framework and the pattern for this is usually to use snd_soc_dapm_enable_pin/disable_pin on the siggen widget. However, as siggen widgets present as always on (the power check always returns 1), this leads to some odd interactions with DAPM. Enabling/disabling the widget then connecting it to a path works as expected, however once in a path enabling the widget has no effect. dapm_widget_set_power will return immediately as the current power state of the widget will return 1, meaning we never check peer power states. Treating the widget as a mic widget causes the expected behaviour in all cases. This does cause a minor cosmetic side effect that siggen widgets no longer show as always on through debugfs, although in the case described above that is more representive of their behaviour. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-dapm.c | 6 +----- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..67f176f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -47,7 +47,7 @@ struct device; /* platform domain */ #define SND_SOC_DAPM_SIGGEN(wname) \ -{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \ +{ .id = snd_soc_dapm_mic, .name = wname, .kcontrol_news = NULL, \ .num_kcontrols = 0, .reg = SND_SOC_NOPM } #define SND_SOC_DAPM_SINK(wname) \ { .id = snd_soc_dapm_sink, .name = wname, .kcontrol_news = NULL, \ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a8156b5..ea6f477 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1817,12 +1817,9 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Supplies and micbiases only bring the * context up to STANDBY as unless something * else is active and passing audio they - * generally don't require full power. Signal - * generators are virtual pins and have no - * power impact themselves. + * generally don't require full power. */ switch (w->id) { - case snd_soc_dapm_siggen: case snd_soc_dapm_vmid: break; case snd_soc_dapm_supply: @@ -3354,7 +3351,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_vmid: - case snd_soc_dapm_siggen: w->is_ep = SND_SOC_DAPM_EP_SOURCE; w->power_check = dapm_always_on_check_power; break; -- 2.1.4

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[alsa-devel] snd-fireworks requires larger period sizes than FFADO
by Dominic Sacré 02 Dec '15

02 Dec '15

Hi, I'm trying to use an Echo AudioFire 4 with the snd-fireworks driver. With FFADO the device works reasonably well at a period size of 64 samples (on a PREEMPT_RT kernel), whereas with snd-fireworks the lowest possible period size seems to be 256 samples. This is what JACK says when I try to start it with a period size of 128 samples: configuring for 48000Hz, period = 128 frames (2.7 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: cannot set period size to 128 frames for capture The lowest achievable analog roundtrip latency seems to be around 19ms (at 2x256 frames), as opposed to 8.3ms with FFADO (at 2x64). Are there any known limitations of the snd-fireworks driver in this regard? Thanks, Dominic

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[alsa-devel] [PATCH 0/14] ASoC: rsnd: TDM Extend Mode support
by Kuninori Morimoto 02 Dec '15

02 Dec '15

Hi Mark These patch-set adds TDM Extend Mode to Renesas R-Car sound driver. 1) - 9) tidyup / cleanup patches 10) - 14) TDM Extend Mode support Unfortunately, our board can't use this TDM Extend Mode, because our SoC and codec are using different style TDM. Thus I have tested this patches on special board which doesn't have codec chip, and I checked signal via oscilloscope. oKuninori Morimoto (14): 1) ASoC: rsnd: don't use normal *mod in adg.c 2) ASoC: rsnd: add missing ADINR::CHNUM on DVC/SRC/SSIU 3) ASoC: rsnd: add missing SRC_O_BUSIF_MODE register 4) ASoC: rsnd: src: rename rsnd_src_soft_reset() to rsnd_src_activation() 5) ASoC: rsnd: mix: rename rsnd_mix_soft_reset() to rsnd_mix_activation() 6) ASoC: rsnd: dvc: rename rsnd_dvc_soft_reset() to rsnd_dvc_activation() 7) ASoC: rsnd: src: add rsnd_src_halt() 8) ASoC: rsnd: mix: add rsnd_mix_halt() 9) ASoC: rsnd: dvc: add rsnd_dvc_halt() 10) ASoC: rsnd: add rsnd_ssi_config_init() 11) ASoC: rsnd: set SSIWSR setting on rsnd_ssi_config_init() 12) ASoC: rsnd: ssi enables non-stereo sound 13) ASoC: rsnd: dvc enables non-stereo sound 14) ASoC: rsnd: add TDM Extend Mode support sound/soc/sh/rcar/adg.c | 30 +++++++++++------------------- sound/soc/sh/rcar/core.c | 65 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/dvc.c | 32 +++++++++++++++++++++++++++----- sound/soc/sh/rcar/gen.c | 11 ++++++++++- sound/soc/sh/rcar/mix.c | 12 ++++++++++-- sound/soc/sh/rcar/rsnd.h | 19 +++++++++++++++++-- sound/soc/sh/rcar/src.c | 18 ++++++++++++++---- sound/soc/sh/rcar/ssi.c | 126 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++--------------------------------------------------- sound/soc/sh/rcar/ssiu.c | 12 +++++++++++- 9 files changed, 235 insertions(+), 90 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: rsnd: fixup wrong snd_soc_dai_driver pointer access
by Kuninori Morimoto 01 Dec '15

01 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> drv pointer should be "base + offset" instead of "current + offset". This patch fixup this issue, otherwise third and subsequent pointer will be broken Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index b187a89..f1d7af1 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -603,7 +603,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) struct device_node *playback, *capture; struct rsnd_dai_stream *io_playback; struct rsnd_dai_stream *io_capture; - struct snd_soc_dai_driver *drv; + struct snd_soc_dai_driver *rdrv, *drv; struct rsnd_dai *rdai; struct device *dev = rsnd_priv_to_dev(priv); int nr, dai_i, io_i, np_i; @@ -616,15 +616,15 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) goto rsnd_dai_probe_done; } - drv = devm_kzalloc(dev, sizeof(*drv) * nr, GFP_KERNEL); + rdrv = devm_kzalloc(dev, sizeof(*rdrv) * nr, GFP_KERNEL); rdai = devm_kzalloc(dev, sizeof(*rdai) * nr, GFP_KERNEL); - if (!drv || !rdai) { + if (!rdrv || !rdai) { ret = -ENOMEM; goto rsnd_dai_probe_done; } priv->rdai_nr = nr; - priv->daidrv = drv; + priv->daidrv = rdrv; priv->rdai = rdai; /* @@ -633,7 +633,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) dai_i = 0; for_each_child_of_node(dai_node, dai_np) { rdai = rsnd_rdai_get(priv, dai_i); - drv = drv + dai_i; + drv = rdrv + dai_i; io_playback = &rdai->playback; io_capture = &rdai->capture; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: rsrc-card: check return value of snd_soc_of_get_dai_name()
by Kuninori Morimoto 01 Dec '15

01 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch adds missing check of snd_soc_of_get_dai_name(). It might not be able to use sound card, because it might returns -EPROBE_DEFER. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index d61db9c..a3ec13f 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -208,7 +208,9 @@ static int rsrc_card_parse_links(struct device_node *np, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; dai_link->cpu_of_node = args.np; - snd_soc_of_get_dai_name(np, &dai_link->cpu_dai_name); + ret = snd_soc_of_get_dai_name(np, &dai_link->cpu_dai_name); + if (ret < 0) + return ret; /* set dai_name */ snprintf(dai_props->dai_name, DAI_NAME_NUM, "fe.%s", @@ -240,7 +242,9 @@ static int rsrc_card_parse_links(struct device_node *np, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = rsrc_card_be_hw_params_fixup; dai_link->codec_of_node = args.np; - snd_soc_of_get_dai_name(np, &dai_link->codec_dai_name); + ret = snd_soc_of_get_dai_name(np, &dai_link->codec_dai_name); + if (ret < 0) + return ret; /* additional name prefix */ if (of_data) { -- 1.9.1

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[alsa-devel] [PATCH] ASoC: arizona: Add 32uS delay after putting FLL into freerun
by Charles Keepax 01 Dec '15

01 Dec '15

When switching between two clock sources using the FLL freerun to smooth the transition we should wait 32uS after putting the FLL into freerun before we proceed. In practice we appear to be getting enough delay from the surrounding code, but better to make it explicit. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e76ecc7..a23f7d1 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2212,9 +2212,9 @@ static int arizona_enable_fll(struct arizona_fll *fll) /* Facilitate smooth refclk across the transition */ regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, ARIZONA_FLL1_GAIN_MASK, 0); - regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, - ARIZONA_FLL1_FREERUN); + regmap_update_bits(fll->arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); + udelay(32); } /* -- 2.1.4

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[alsa-devel] [PATCH 1/2] ASoC: sunxi: Remove useless comments and variable
by Maxime Ripard 01 Dec '15

01 Dec '15

The comment is misleading on how we should support external power amps, and the variable is not used and generates a warning. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/soc/sunxi/sun4i-codec.c | 22 ---------------------- 1 file changed, 22 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 30c9e9260491..516c7c2b479a 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -109,11 +109,6 @@ struct sun4i_codec { static void sun4i_codec_start_playback(struct sun4i_codec *scodec) { - /* - * FIXME: according to the BSP, we might need to drive a PA - * GPIO high here on some boards - */ - /* Flush TX FIFO */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), @@ -127,11 +122,6 @@ static void sun4i_codec_start_playback(struct sun4i_codec *scodec) static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) { - /* - * FIXME: according to the BSP, we might need to drive a PA - * GPIO low here on some boards - */ - /* Disable DAC DRQ */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), @@ -140,11 +130,6 @@ static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) static void sun4i_codec_start_capture(struct sun4i_codec *scodec) { - /* - * FIXME: according to the BSP, we might need to drive a PA - * GPIO high here on some boards - */ - /* Enable ADC DRQ */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), @@ -153,11 +138,6 @@ static void sun4i_codec_start_capture(struct sun4i_codec *scodec) static void sun4i_codec_stop_capture(struct sun4i_codec *scodec) { - /* - * FIXME: according to the BSP, we might need to drive a PA - * GPIO low here on some boards - */ - /* Disable ADC DRQ */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), 0); @@ -358,8 +338,6 @@ static int sun4i_codec_hw_params_capture(struct sun4i_codec *scodec, struct snd_pcm_hw_params *params, unsigned int hwrate) { - u32 val; - /* Set ADC sample rate */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, 7 << SUN4I_CODEC_ADC_FIFOC_ADC_FS, -- 2.6.3

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[alsa-devel] [PATCH] ASoC: tegra_alc5632: check return value
by Sudip Mukherjee 01 Dec '15

01 Dec '15

We have been returning success even if snd_soc_card_jack_new() fails. Lets check the return value and return error if it fails. Fixes: 12cc6d1dca4d ("ASoC: tegra_alc5632: Register jacks at the card level") Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/tegra/tegra_alc5632.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index ba272e2..deb597f 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -101,12 +101,16 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { + int ret; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, - &tegra_alc5632_hs_jack, - tegra_alc5632_hs_jack_pins, - ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, + &tegra_alc5632_hs_jack, + tegra_alc5632_hs_jack_pins, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); + if (ret) + return ret; if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: hdac_hdmi: check error return
by Sudip Mukherjee 01 Dec '15

01 Dec '15

As hdac->num_nodes is unsigned we can not check if snd_hdac_get_sub_nodes() has returned error or success. Lets have a temporary int to check the error value. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v2: made num_nodes as int, but Takashi said that num_nodes should really be an unsigned. v1: removed the error check as that condition is always false. sound/soc/codecs/hdac_hdmi.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 205f2c2..1a2f33b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -409,17 +409,18 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev, static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev) { hda_nid_t nid; - int i; + int i, num_nodes; struct hdac_device *hdac = &edev->hdac; struct hdac_hdmi_priv *hdmi = edev->private_data; int cvt_nid = 0, pin_nid = 0; - hdac->num_nodes = snd_hdac_get_sub_nodes(hdac, hdac->afg, &nid); - if (!nid || hdac->num_nodes < 0) { + num_nodes = snd_hdac_get_sub_nodes(hdac, hdac->afg, &nid); + if (!nid || num_nodes < 0) { dev_warn(&hdac->dev, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } + hdac->num_nodes = num_nodes; hdac->start_nid = nid; for (i = 0; i < hdac->num_nodes; i++, nid++) { -- 1.9.1

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[alsa-devel] [asoc:topic/da7218 2/2] sound/soc/codecs/da7218.c:3214:9-10: WARNING: return of 0/1 in function 'da7218_volatile_register' with return type bool
by kbuild test robot 01 Dec '15

01 Dec '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/da7218 head: 4d50934abd2261fd467320d52c470efff309fd74 commit: 4d50934abd2261fd467320d52c470efff309fd74 [2/2] ASoC: da7218: Add da7218 codec driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/da7218.c:3214:9-10: WARNING: return of 0/1 in function 'da7218_volatile_register' with return type bool Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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