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

[alsa-devel] [PATCH] ALSA: virtuoso: add Xonar Xense support
by Harley Griggs 22 Sep '14

22 Sep '14

This patch adds support for the Xonar Xense card. Has been working fine for over a year and with the latest Xonar patches. I have not touched the CS4362A (surround), nor am I sure how to go about it without more code duplication or refactoring. Signed-off-by: Harley Griggs <hgriggs(a)posteo.co.uk> --- sound/pci/oxygen/virtuoso.c | 1 + sound/pci/oxygen/xonar_pcm179x.c | 135 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 136 insertions(+) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 7b317a2..83de6fb 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static const struct pci_device_id xonar_ids[] = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8428) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID(0x1043, 0x85f4) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index e026059..e06acfc 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -212,6 +212,9 @@ #define GPIO_ST_MAGIC 0x0040 #define GPIO_ST_HP 0x0080 +#define GPIO_XENSE_OUTPUT_ENABLE (0x0001 | 0x0010 | 0x0020) +#define GPIO_XENSE_SPEAKERS 0x0080 + #define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ii, /W=0 */ #define I2C_DEVICE_CS2000 0x9c /* 100111, 0, /W=0 */ @@ -499,6 +502,52 @@ static void xonar_stx_init(struct oxygen *chip) xonar_st_init_common(chip); } +static void xonar_xense_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + + data->generic.anti_pop_delay = 100; + data->has_cs2000 = 1; + data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; + data->broken_i2c = true; + + oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, + OXYGEN_RATE_48000 | + OXYGEN_I2S_FORMAT_I2S | + OXYGEN_I2S_MCLK(MCLK_512) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); + + xonar_st_init_i2c(chip); + cs2000_registers_init(chip); + + data->generic.output_enable_bit = GPIO_XENSE_OUTPUT_ENABLE; + data->dacs = chip->model.dac_channels_mixer / 2; + data->hp_gain_offset = 2*-18; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | + GPIO_ST_MAGIC | GPIO_XENSE_SPEAKERS); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | + GPIO_XENSE_SPEAKERS); + + xonar_init_cs53x1(chip); + xonar_enable_output(chip); + + snd_component_add(chip->card, "PCM1796"); + snd_component_add(chip->card, "CS5381"); + snd_component_add(chip->card, "CS2000"); +} + static void xonar_d2_cleanup(struct oxygen *chip) { xonar_disable_output(chip); @@ -859,6 +908,67 @@ static const struct snd_kcontrol_new st_controls[] = { }, }; +static int xense_output_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 gpio; + + gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (gpio & GPIO_XENSE_SPEAKERS) + value->value.enumerated.item[0] = 0; + else if (!(gpio & GPIO_XENSE_SPEAKERS) && (gpio & GPIO_ST_HP_REAR)) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + return 0; +} + +static int xense_output_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + u16 gpio_old, gpio; + + mutex_lock(&chip->mutex); + gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA); + gpio = gpio_old; + switch (value->value.enumerated.item[0]) { + case 0: + gpio |= GPIO_XENSE_SPEAKERS | GPIO_ST_HP_REAR; + break; + case 1: + gpio = (gpio | GPIO_ST_HP_REAR) & ~GPIO_XENSE_SPEAKERS; + break; + case 2: + gpio &= ~(GPIO_XENSE_SPEAKERS | GPIO_ST_HP_REAR); + break; + } + oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio); + data->hp_active = !(gpio & GPIO_XENSE_SPEAKERS); + update_pcm1796_volume(chip); + mutex_unlock(&chip->mutex); + return gpio != gpio_old; +} + +static const struct snd_kcontrol_new xense_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Output", + .info = st_output_switch_info, + .get = xense_output_switch_get, + .put = xense_output_switch_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphones Impedance Playback Enum", + .info = st_hp_volume_offset_info, + .get = st_hp_volume_offset_get, + .put = st_hp_volume_offset_put, + }, +}; + static void xonar_line_mic_ac97_switch(struct oxygen *chip, unsigned int reg, unsigned int mute) { @@ -946,6 +1056,23 @@ static int xonar_st_mixer_init(struct oxygen *chip) return 0; } +static int xonar_xense_mixer_init(struct oxygen *chip) +{ + unsigned int i; + int err; + + for (i = 0; i < ARRAY_SIZE(xense_controls); ++i) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&xense_controls[i], chip)); + if (err < 0) + return err; + } + err = add_pcm1796_controls(chip); + if (err < 0) + return err; + return 0; +} + static void dump_pcm1796_registers(struct oxygen *chip, struct snd_info_buffer *buffer) { @@ -1146,6 +1273,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x8428: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar Xense"; + chip->model.chip = "AV100"; + chip->model.init = xonar_xense_init; + chip->model.mixer_init = xonar_xense_mixer_init; + break; + default: return -EINVAL; } -- 1.8.5.5

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[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Allow to describe master or slave mode
by Fabio Estevam 22 Sep '14

22 Sep '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Provide a mechanism to describe whether the system runs in master or slave mode. In order to keep compatibilty with existing dtb's let slave mode be default one. Tested on a imx6q-sabresd board in both modes. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 5 +++++ sound/soc/fsl/fsl-asoc-card.c | 15 ++++++++++++--- 2 files changed, 17 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..1bbc828 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -59,6 +59,10 @@ Optional properties: - audio-asrc : The phandle of ASRC. It can be absent if there's no need to add ASRC support via DPCM. + - audio-cpu-mode : The operating mode for the audio cpu interface. + "slave" - audio clock comes from the external codec + "master" -audio clock is generated from the audio cpu + Example: sound-cs42888 { compatible = "fsl,imx-audio-cs42888"; @@ -66,6 +70,7 @@ sound-cs42888 { audio-cpu = <&esai>; audio-asrc = <&asrc>; audio-codec = <&cs42888>; + audio-cpu-mode = "master"; audio-routing = "Line Out Jack", "AOUT1L", "Line Out Jack", "AOUT1R", diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 007c772..bd903cc 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -393,6 +393,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct clk *codec_clk; u32 width; int ret; + const char *mode; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -441,6 +442,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Assign a default DAI format, and allow each card to overwrite it */ priv->dai_fmt = DAI_FMT_BASE; + mode = of_get_property(np, "audio-cpu-mode", NULL); + if (mode) { + if (!strcmp(mode, "master")) + priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + else if (!strcmp(mode, "slave")) + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + /* Assign slave mode by default */ + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { priv->card.set_bias_level = NULL; @@ -448,16 +460,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; - priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; - priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; - priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; -- 1.9.1

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[alsa-devel] Wanted you to test music composition application
by Joël Krähemann 20 Sep '14

20 Sep '14

Hi, as mentioned in subject: Production release 0.4.2 planed of Advanced Gtk+ Sequencer. Now, comes your part I would like you to test functionality. I'll do for you a wiki page of aspects to test. Please visit http://ags.sf.net and wiki page https://sourceforge.net/p/ags/wiki/testing-cheatsheet/ Happy to hear of you pretty soon! Joël

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Re: [alsa-devel] [PATCH] ASoC: fsl_ssi: Remove unneeded 'i2s-slave' property
by Fabio.Estevam＠freescale.com 20 Sep '14

20 Sep '14

Sorry, I missed to add alsa-devel(a)alsa-project.org on Cc. ________________________________________ From: Timur Tabi <timur(a)tabi.org> Sent: Thursday, September 18, 2014 3:37 PM To: Nicolin Chen Cc: Estevam Fabio-R49496; broonie(a)kernel.org; Xiubo Li-B47053 Subject: Re: [PATCH] ASoC: fsl_ssi: Remove unneeded 'i2s-slave' property On 09/18/2014 01:32 PM, Nicolin Chen wrote: > On Thu, Sep 18, 2014 at 03:11:17PM -0300, Fabio Estevam wrote: >> There is no need to use 'i2s-slave' property, since master/slave configuration >> are passed via machine layer. >> >> This change does not break existing users because they do check for slave >> mode inside sound/soc/fsl/mpc8610_hpcd.c/p1022_ds.c and p1022_rdk.c > > Besides this change, I'm wondering if we can take a further step to drop > this entire property from Device Tree binding document as newer platforms > may no long need this property in their DT bindings. I think the property is necessary if anyone takes a P1022 or MPC8610 and uses it in a platform that doesn't support i2s-slave. But currently, the code doesn't support anything else. So technically, we could remove it, and if anyone does want to support another configuration, he will have to add that support the "new" way.

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Re: [alsa-devel] alsa-jack issues
by Radoslaw Szkodzinski 20 Sep '14

20 Sep '14

On Thu, Sep 11, 2014 at 1:33 AM, Sergey <sergemp(a)mail.ru> wrote: > Sat, 6 Sep 2014 Radoslaw Szkodzinski wrote: > >>A recent change in alsa-plugins 1.0.28 alsa-jack has changed the poll semantics. >>This breaks audacious and mpv, perhaps more applications. >> >>The regression is caused by: >>Commit: 9217377337cdceb62abeb5969112b738bb5cd551 >>jack: fix polling and recovering > > I'm initial author of this commit and I'd love to fix the issue, > but it never happens to me. I need your help to fix it. > > Can you provide some steps to reproduce this bug? How much time it > takes to reproduce it? Can you reproduce it with alsa 1.0.27 and > alsa-jack plugin 1.0.28? You can download that plugin separately: I'll test this during the weekend. 1.0.27.x was fine with alsa 1.0.27. > By the way, what distribution are you using? > Is there a livecd that I can download to reproduce this problem? This is actually Gentoo, so not really. However, I think it should be reproducible anywhere. >>This might or might not be related to lack of >>snd_pcm_poll_descriptor_revents call or the use of a timer instead of >>repolling. > > It should not. But we can't be sure until we find what causes it. I'm not entirely sure myself. The above mentioned applications cause the problem. The whole setup consists of: Highly patched 3.12.x kernel, no RT patches but in full preempt mode. Jack2 in DBus mode, realtime and mlock enabled. ALSA output to ICE1724 soundcard, sample rate matched everywhere to 44.1kHz. ALSA set to use speexrate_best and 44.1kHz by default if this is important. Applications: Ladish Carla with Caps equalizers and one other personal LADSPA plugin (RT safe). Directly connecting to playback also reproduces the issue. Any period size and nframes, though very long period sizes vastly reduce the likelihood this is reproducible. It is not related to xruns directly, but an xrun can precipitate the issue too. Reproducible in audacious 3.4.1 or older and mpv (git revision, from this month or older), but not with sox or aplay. No problems with direct Jack outputs of the above applications whatsoever. Both of the problematic applications use a sleep timer in addition or instead of polling. (audacious due to a bug, mpv by design I think) > I'm sorry for bothering you. Not a bother at all, I'd like to see this fixed. R.

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[alsa-devel] ASoC: samsung: MACH_SMDKC100
by Paul Bolle 19 Sep '14

19 Sep '14

Kukjin, Your commit 52ad6582ceb2 ("ARM: S5PC100: no more support S5PC100 SoC" landed in next-20140702. It removed the Kconfig symbol MACH_SMDKC100 (and a lot of other stuff). Is the trivial patch to also remove the last two references to MACH_SMDKC100 from sound/soc/samsung/Kconfig/ queued somewhere? I don't think it was part of the series that included the above commit. Paul Bolle

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[alsa-devel] [PATCH 6/11] simple_abst : null check for lib
by Renu Tyagi 19 Sep '14

19 Sep '14

Signed-off-by: renu tyagi <renu.tyagi(a)samsung.com> --- src/mixer/simple_abst.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..5297472 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -70,6 +70,8 @@ static int try_open(snd_mixer_class_t *class, const char *lib) void *h; int err = 0; + if (!lib) + return -ENXIO; path = getenv("ALSA_MIXER_SIMPLE_MODULES"); if (!path) path = SO_PATH; -- 1.7.1

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[alsa-devel] [PATCH 4/11] rawmidi : close handle h
by Renu Tyagi 19 Sep '14

19 Sep '14

Signed-off-by: renu tyagi --- src/rawmidi/rawmidi.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index b835b47..ac699b4 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -256,8 +256,11 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp snd_config_delete(type_conf); if (err >= 0) err = open_func(inputp, outputp, name, rawmidi_root, rawmidi_conf, mode); - if (err < 0) + if (err < 0) { + if (h) + snd_dlclose(h); return err; + } if (inputp) { (*inputp)->dl_handle = h; h = NULL; snd_rawmidi_params_default(*inputp, Â¶ms); -- 1.7.1

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[alsa-devel] [PATCH 3/11] sbase : fixed missing free
by Renu Tyagi 19 Sep '14

19 Sep '14

Signed-off-by: renu tyagi --- modules/mixer/simple/sbase.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/modules/mixer/simple/sbase.c b/modules/mixer/simple/sbase.c index 97feee8..bb2f59d 100644 --- a/modules/mixer/simple/sbase.c +++ b/modules/mixer/simple/sbase.c @@ -377,6 +377,7 @@ static int simple_event_add1(snd_mixer_class_t *class, if (ctype != SND_CTL_ELEM_TYPE_BOOLEAN) { __invalid_type: snd_mixer_selem_id_free(id); + free(hsimple); return -EINVAL; } break; -- 1.7.1

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[alsa-devel] alsa-info.sh was broken recently
by Sergey 19 Sep '14

19 Sep '14

Hello. Something was broken recently. http://www.alsa-project.org/alsa-info.sh leads to Error 404. As well as http://www.alsa-project.org/alsa-info.sh.changelog It breaks many tutorials and documentation references, e.g.: http://www.kernel.org/doc/Documentation/sound/alsa/HD-Audio.txt It also breaks the alsa-info.sh script itself attempting to update from that url. -- Sergey

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