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

[alsa-devel] [PATCH v9 0/3] HDA: add extended HDA
by Vinod Koul 11 Jun '15

11 Jun '15

New HDA controllers from Intel have extended HDA capabilities like multilink, pipe processing, SPIB, GTS etc In order to use them we create an extended HDA bus, controller and stream which uses the extended configuration Changes in v9: - removed snd_hdac_ext_bus_map_codec_to_link - updated snd_hdac_ext_bus_get_link per Takahsi comment and drop the array - add index in bus instance Changes in v8: - fixed some dead code crept in - checkpatch cleanup Changes in v7: - Feedback from Takashi - removed the macros for register read/writes - add new macros for reg update - fix bus pointer macro - header cleanup on c/h files - new device name ehdaudioiDn - removed flags for ext caps, use pointer only - moved module info to patch 1 - sparse check on series - remove codec name in link Jeeja KP (3): ALSA: hdac_ext: add extended HDA bus ALSA: hdac_ext: add hdac extended controller ALSA: hdac_ext: add extended stream capabilities include/sound/hdaudio_ext.h | 129 ++++++++++++ sound/hda/Kconfig | 4 + sound/hda/Makefile | 3 + sound/hda/ext/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 117 +++++++++++ sound/hda/ext/hdac_ext_controller.c | 270 ++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 402 ++++++++++++++++++++++++++++++++++++ 7 files changed, 928 insertions(+) create mode 100644 include/sound/hdaudio_ext.h create mode 100644 sound/hda/ext/Makefile create mode 100644 sound/hda/ext/hdac_ext_bus.c create mode 100644 sound/hda/ext/hdac_ext_controller.c create mode 100644 sound/hda/ext/hdac_ext_stream.c -- 1.9.1

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[alsa-devel] [PATCH spi-pxa2xx.c 1/1] : fix spi init of WM510205 codec via ACPI, the chip hangs on chipselect 1
by Christian Hartmann 11 Jun '15

11 Jun '15

Hello, I have one patch that enables the spi initialization of the WM510205 audio codec Without the patch, this message indicates that the codec chip fails with the current chipselect number in spi-pxa2xx: [ 0.296256] pxa2xx-spi 80860F0E:00: cs1 >= max 1 [ 0.296270] spi_master spi32766: failed to add SPI device WM510205:00 from ACPI So raise num_chipselect by one to enable the codec. The WM510202 is on chipselect = 1. With the patch applied I got in dmesg now: [ 6.056829] pxa2xx-spi 80860F0E:00: no DMA channels available, using PIO [ 6.067035] pxa2xx-spi 80860F0E:00: registered master spi32766 (dynamic) [ 6.067214] spi spi-WM510205:00: 8333333 Hz actual, PIO [ 6.067221] spi spi-WM510205:00: setup mode 0, 8 bits/w, 8000000 Hz max --> 0 [ 6.067287] pxa2xx-spi 80860F0E:00: registered child spi-WM510205:00 would be nice to hear some feedback or comments about this patch. Notice: Thats my first attempt to make the codec chip via ACPI working on Linux systems and some other patches are needed to let the codec work together with the intel_sst_acpi driver. Currently I am trying to write the also needed machine driver which binds to this codec chip. But I can need help from others too, cause I am not so familiar with the Intel SST ACPI stuff and the baytrail code itself, nor with the arizona stuff... cheers Chris

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[alsa-devel] [PATCH] ALSA: hda - Fix link power unbalance at device removal
by Takashi Iwai 11 Jun '15

11 Jun '15

snd_hdac_link_power() has to be called after unregistering the codec device. Otherwise the device might be already runtime-suspended, thus the refcount goes under zero, triggering a warning like: WARNING: CPU: 7 PID: 2014 at sound/hda/hdac_i915.c:63 snd_hdac_display_power+0x106/0x120 [snd_hda_core]() CPU: 7 PID: 2014 Comm: modprobe Not tainted 4.1.0-rc7-test+ #1 Call Trace: [<ffffffff81697fe3>] dump_stack+0x4c/0x6e [<ffffffff810696da>] warn_slowpath_common+0x8a/0xc0 [<ffffffff810697ca>] warn_slowpath_null+0x1a/0x20 [<ffffffffa02dd526>] snd_hdac_display_power+0x106/0x120 [snd_hda_core] [<ffffffffa030b422>] azx_intel_link_power+0x12/0x20 [snd_hda_intel] [<ffffffffa037139f>] azx_link_power+0x1f/0x30 [snd_hda_codec] [<ffffffffa02d89fe>] snd_hdac_link_power+0x2e/0x40 [snd_hda_core] [<ffffffffa0368524>] snd_hda_codec_dev_free+0x34/0x50 [snd_hda_codec] [<ffffffffa0252061>] __snd_device_free+0x51/0xa0 [snd] ..... Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d70229841156..497bd0f9dcda 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -886,8 +886,8 @@ static int snd_hda_codec_dev_free(struct snd_device *device) struct hda_codec *codec = device->device_data; codec->in_freeing = 1; - snd_hdac_link_power(&codec->core, false); snd_hdac_device_unregister(&codec->core); + snd_hdac_link_power(&codec->core, false); put_device(hda_codec_dev(codec)); return 0; } -- 2.4.3

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[alsa-devel] [PATCH] ASoC: tas2552: Use devm_gpiod_get_optional for enable-gpio
by Axel Lin 11 Jun '15

11 Jun '15

commit ea178d1456dc ("ASoC: tas2552: Make the enable-gpio really optional") makes enable-gpio optional. devm_gpiod_get_optional() is the better function for optional gpio, so let's switch to use it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas2552.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 169a703..4f25a7d 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -703,13 +703,10 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); - if (IS_ERR(data->enable_gpio)) { - if (PTR_ERR(data->enable_gpio) == -EPROBE_DEFER) - return -EPROBE_DEFER; - - data->enable_gpio = NULL; - } + data->enable_gpio = devm_gpiod_get_optional(dev, "enable", + GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); -- 2.1.0

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[alsa-devel] [PATCH v4 1/3] ASoC: wm_adsp: Add codec_probe and codec_remove stubs
by Richard Fitzgerald 11 Jun '15

11 Jun '15

Currently the only init function in wm_adsp is called by the codec driver early in its probe before the codec has been registered with SOC. This patch adds stubs for the codec_probe and codec_remove stages and calls them from WM5102 and WM5110 codec drivers. This allows us to hang anything that needs setup during the codec probe stage off these functions without further modification of the codec drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 6 ++++++ sound/soc/codecs/wm5110.c | 12 +++++++++++- sound/soc/codecs/wm_adsp.c | 12 ++++++++++++ sound/soc/codecs/wm_adsp.h | 2 ++ 4 files changed, 31 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 36a3921..785b3cf 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1875,6 +1875,10 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = wm_adsp2_codec_probe(&priv->core.adsp[0], codec); + if (ret) + return ret; + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; @@ -1893,6 +1897,8 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) { struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + wm_adsp2_codec_remove(&priv->core.adsp[0], codec); + priv->core.arizona->dapm = NULL; return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c83b0a9..45d109a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1599,7 +1599,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int i, ret; priv->core.arizona->dapm = &codec->dapm; @@ -1607,6 +1607,12 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); + for (i = 0; i < WM5110_NUM_ADSP; ++i) { + ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); + if (ret) + return ret; + } + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) return ret; @@ -1621,6 +1627,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < WM5110_NUM_ADSP; ++i) + wm_adsp2_codec_remove(&priv->core.adsp[i], codec); priv->core.arizona->dapm = NULL; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b62ffd0..9fad2fd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1933,6 +1933,18 @@ err: } EXPORT_SYMBOL_GPL(wm_adsp2_event); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_probe); + +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_remove); + int wm_adsp2_init(struct wm_adsp *dsp) { int ret; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 0e5f07c..5584e34 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -79,6 +79,8 @@ extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, -- 1.7.2.5

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[alsa-devel] [PATCH 1/1] usb-audio: Set correct type for some UAC2 mixer controls.
by Johan Rastén 11 Jun '15

11 Jun '15

Changed ctl type for Input Gain Control and Input Gain Pad Control to USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0 definition. Signed-off-by: Johan Rastén <johan(a)oljud.se> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 8b7e391..6b3acba 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -809,12 +809,12 @@ static struct usb_feature_control_info audio_feature_info[] = { { "Tone Control - Treble", USB_MIXER_S8 }, { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, + { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ { "Bass Boost", USB_MIXER_BOOLEAN }, { "Loudness", USB_MIXER_BOOLEAN }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_U16 }, - { "Input Gain Pad Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16 }, + { "Input Gain Pad Control", USB_MIXER_S16 }, { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; -- 2.1.4

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[alsa-devel] [PATCH] ALSA: hda - Re-add the lost fake mute support
by Takashi Iwai 11 Jun '15

11 Jun '15

Yet another regression by the transition to regmap cache; for better usability, we had the fake mute control using the zero amp value for Conexant codecs, and this was forgotten in the new hda core code. Since the bits 4-7 are unused for the amp registers (as we follow the syntax of AMP_GET verb), the bit 4 is now used to indicate the fake mute. For setting this flag, snd_hda_codec_amp_update() becomes a function from a simple macro. The bonus is that it gained a proper function description. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_regmap.h | 2 ++ sound/hda/hdac_regmap.c | 5 +++++ sound/pci/hda/hda_codec.c | 23 +++++++++++++++++++++++ sound/pci/hda/hda_local.h | 4 ++-- 4 files changed, 32 insertions(+), 2 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index 53a18b3635e2..df705908480a 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -9,6 +9,8 @@ #include <sound/core.h> #include <sound/hdaudio.h> +#define AC_AMP_FAKE_MUTE 0x10 /* fake mute bit set to amp verbs */ + int snd_hdac_regmap_init(struct hdac_device *codec); void snd_hdac_regmap_exit(struct hdac_device *codec); int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec, diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index c4f5e61d4404..1eabcdf69457 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -246,6 +246,9 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) return hda_reg_read_stereo_amp(codec, reg, val); if (verb == AC_VERB_GET_PROC_COEF) return hda_reg_read_coef(codec, reg, val); + if ((verb & 0x700) == AC_VERB_SET_AMP_GAIN_MUTE) + reg &= ~AC_AMP_FAKE_MUTE; + err = snd_hdac_exec_verb(codec, reg, 0, val); if (err < 0) return err; @@ -283,6 +286,8 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) switch (verb & 0xf00) { case AC_VERB_SET_AMP_GAIN_MUTE: + if ((reg & AC_AMP_FAKE_MUTE) && (val & AC_AMP_MUTE)) + val = 0; verb = AC_VERB_SET_AMP_GAIN_MUTE; if (reg & AC_AMP_GET_LEFT) verb |= AC_AMP_SET_LEFT >> 8; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7782212dd64..3a3670168f12 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1376,6 +1376,29 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps); /** + * snd_hda_codec_amp_update - update the AMP mono value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel to update (0 or 1) + * @dir: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Update the AMP values for the given channel, direction and index. + */ +int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, int mask, int val) +{ + unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx); + + if (query_amp_caps(codec, nid, dir) & AC_AMPCAP_MIN_MUTE) + cmd |= AC_AMP_FAKE_MUTE; + return snd_hdac_regmap_update_raw(&codec->core, cmd, mask, val); +} +EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); + +/** * snd_hda_codec_amp_stereo - update the AMP stereo values * @codec: HD-audio codec * @nid: NID to read the AMP value diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 3b567f42296b..bed66c314431 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -129,8 +129,8 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, /* lowlevel accessor with caching; use carefully */ #define snd_hda_codec_amp_read(codec, nid, ch, dir, idx) \ snd_hdac_regmap_get_amp(&(codec)->core, nid, ch, dir, idx) -#define snd_hda_codec_amp_update(codec, nid, ch, dir, idx, mask, val) \ - snd_hdac_regmap_update_amp(&(codec)->core, nid, ch, dir, idx, mask, val) +int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, int mask, int val); int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, int mask, int val); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, -- 2.4.3

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[alsa-devel] [PATCH v8 0/3] HDA: add extended HDA
by Vinod Koul 11 Jun '15

11 Jun '15

New HDA controllers from Intel have extended HDA capabilities like multilink, pipe processing, SPIB, GTS etc In order to use them we create an extended HDA bus, controller and stream which uses the extended configuration Changes in v8: - fixed some dead codec crept in - checkpatch cleanup Changes in v7: - Feedback from Takashi - removed the macros for register read/writes - add new macros for reg update - fix bus pointer macro - header cleanup on c/h files - new device name ehdaudioiDn - removed flags for ext caps, use pointer only - moved module info to patch 1 - sparse check on series - remove codec name in link Jeeja KP (3): ALSA: hdac_ext: add extended HDA bus ALSA: hdac_ext: add hdac extended controller ALSA: hdac_ext: add extended stream capabilities include/sound/hdaudio_ext.h | 129 ++++++++++++ sound/hda/Kconfig | 4 + sound/hda/Makefile | 3 + sound/hda/ext/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 116 +++++++++++ sound/hda/ext/hdac_ext_controller.c | 290 ++++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 402 ++++++++++++++++++++++++++++++++++++ 7 files changed, 947 insertions(+) create mode 100644 include/sound/hdaudio_ext.h create mode 100644 sound/hda/ext/Makefile create mode 100644 sound/hda/ext/hdac_ext_bus.c create mode 100644 sound/hda/ext/hdac_ext_controller.c create mode 100644 sound/hda/ext/hdac_ext_stream.c -- 1.9.1

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[alsa-devel] [PATCH v4 1/3] SoC: rt5645: make RT5650_TDM_CTRL_4 readable
by Oder Chiou 11 Jun '15

11 Jun '15

Register RT5650_TDM_CTRL_4(0x7A) is readable and used for mixer setting. It should be added in rt5645_readable_register function. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d5f0f56..b86474a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -349,6 +349,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: case RT5645_TDM_CTRL_3: + case RT5650_TDM_CTRL_4: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [patch] ASoC: fsl: add an error check
by Dan Carpenter 10 Jun '15

10 Jun '15

My static checker complains that: sound/soc/fsl/imx-wm8962.c:196 imx_wm8962_probe() warn: we tested 'ret' before and it was 'false' The intent was that we use "ret" to check imx_audmux_v2_configure_port(). Fixes: 8de2ae2a7f1f ('ASoC: fsl: add imx-wm8962 machine driver') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- I assume checking for an error here will not change anything but I have not tested this patch. diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index cd146d4..b38b98c 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -190,7 +190,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } - imx_audmux_v2_configure_port(ext_port, + ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) {

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