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[alsa-devel] [PATCH RESEND 1/2] ASoC: rt298: enable IRQ for jack detection
by Bard Liao 24 Feb '16

24 Feb '16

There are some registers needed for enabling rt298 IRQ missed on current driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 6 ++++++ sound/soc/codecs/rt298.h | 8 ++++++++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 30c6de6..e67ef5f 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1225,6 +1225,12 @@ static int rt298_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt298->regmap, RT298_WIND_FILTER_CTRL, 0x0082, 0x0082); regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); + + regmap_write(rt298->regmap, RT298_UNSOLICITED_INLINE_CMD, 0x81); + regmap_write(rt298->regmap, RT298_UNSOLICITED_HP_OUT, 0x82); + regmap_write(rt298->regmap, RT298_UNSOLICITED_MIC1, 0x84); + regmap_update_bits(rt298->regmap, RT298_IRQ_FLAG_CTRL, 0x2, 0x2); + rt298->is_hp_in = -1; if (rt298->i2c->irq) { diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index 31da162..d66f884 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -34,6 +34,7 @@ #define RT298_HP_OUT 0x21 #define RT298_MIXER_IN1 0x22 #define RT298_MIXER_IN2 0x23 +#define RT298_INLINE_CMD 0x55 #define RT298_SET_PIN_SFT 6 #define RT298_SET_PIN_ENABLE 0x40 @@ -124,6 +125,12 @@ VERB_CMD(AC_VERB_SET_COEF_INDEX, RT298_VENDOR_REGISTERS, 0) #define RT298_PROC_COEF\ VERB_CMD(AC_VERB_SET_PROC_COEF, RT298_VENDOR_REGISTERS, 0) +#define RT298_UNSOLICITED_INLINE_CMD\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_INLINE_CMD, 0) +#define RT298_UNSOLICITED_HP_OUT\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_HP_OUT, 0) +#define RT298_UNSOLICITED_MIC1\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_MIC1, 0) /* Index registers */ #define RT298_A_BIAS_CTRL1 0x01 @@ -148,6 +155,7 @@ #define RT298_DEPOP_CTRL2 0x67 #define RT298_DEPOP_CTRL3 0x68 #define RT298_DEPOP_CTRL4 0x69 +#define RT298_IRQ_FLAG_CTRL 0x7c /* SPDIF (0x06) */ #define RT298_SPDIF_SEL_SFT 0 -- 1.8.1.1.439.g50a6b54

1 1

[alsa-devel] [PATCH 2/2] ASoC: max9867: fix type of variable containing error codes
by Andrzej Hajda 24 Feb '16

24 Feb '16

value variable can contain error values and is compared with zero. Its type must be signed. The problem has been detected using coccinelle script scripts/coccinelle/tests/unsigned_lesser_than_zero.cocci Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/soc/codecs/max9867.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index ab7a94d..4fb6fc7 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -180,7 +180,8 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); - unsigned int ni_h, ni_l, value; + unsigned int ni_h, ni_l; + int value; value = get_ni_value(max9867->sysclk, params_rate(params)); if (value < 0) -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: max9867: Use params_width instead of open-coded
by Axel Lin 24 Feb '16

24 Feb '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max9867.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index ab7a94d..e3ee391 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -206,10 +206,9 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, unsigned long int bclk_rate, pclk_bclk_ratio; int bclk_value; - bclk_rate = params_rate(params) * 2 * - snd_pcm_format_width(params_format(params)); + bclk_rate = params_rate(params) * 2 * params_width(params); pclk_bclk_ratio = max9867->pclk/bclk_rate; - switch (snd_pcm_format_width(params_format(params))) { + switch (params_width(params)) { case 8: case 16: switch (pclk_bclk_ratio) { -- 2.1.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: adau17x1: Minor fixes and DT support
by Andreas Irestål 24 Feb '16

24 Feb '16

This set of patches fixes some minor typos and adds DT support for the adau17x1 codec drivers. I've only been able to test the 1761 part on real hardware since I don't have any adau1781 hardware at the moment. Currently, quite a lot of options are provided through the platform_data struct, so it made sense parsing the DT and create a platform_data struct from these parameters not to break anything. I've peeked at a few other drivers and the TI tlv320aic31xx driver works this way as well. Andreas Irestål (4): ASoC: adau17x1: Fix incorrect BCLK ratio definitions ASoc: adau17x1: Correct typos in file headers ASoC: adau17x1: Add basic DT support for adau17x1 ASoC: adau17x1: Add option to provide platform data via DT .../devicetree/bindings/sound/adi,adau17x1.txt | 54 +++++++++ include/dt-bindings/sound/adau17x1.h | 14 +++ include/linux/platform_data/adau17x1.h | 2 +- sound/soc/codecs/adau1761-i2c.c | 14 ++- sound/soc/codecs/adau1761-spi.c | 14 ++- sound/soc/codecs/adau1761.c | 133 ++++++++++++++++++++- sound/soc/codecs/adau1781-i2c.c | 10 ++ sound/soc/codecs/adau1781-spi.c | 10 ++ sound/soc/codecs/adau1781.c | 59 ++++++++- sound/soc/codecs/adau17x1.h | 6 +- 10 files changed, 308 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau17x1.txt create mode 100644 include/dt-bindings/sound/adau17x1.h -- 2.1.4

3 8

[alsa-devel] [PATCH v2 0/5] Minor fixes and DT support
by Andreas Irestål 24 Feb '16

24 Feb '16

This set of patches fixes some minor typos and adds DT support for the adau17x1 codec drivers. I've only been able to test the 1761 part on real hardware since I don't have any adau1781 hardware at the moment. Currently, quite a lot of options are provided through the platform_data struct, so it made sense parsing the DT and create a platform_data struct from these parameters not to break anything. Changes since v1: - Added vendor prefix "adi," to DT properties - DT parsing moved to main probe function Andreas Irestål (5): ASoC: adau17x1: Fix incorrect BCLK ratio definitions ASoC: adau17x1: Correct typos in file headers ASoC: adau17x1: Add adau17x1 bindings document ASoC: adau17x1: Add basic DT support for adau17x1 ASoC: adau17x1: Support platform data via DT .../devicetree/bindings/sound/adi,adau17x1.txt | 55 +++++++++ include/dt-bindings/sound/adau17x1.h | 14 +++ include/linux/platform_data/adau17x1.h | 2 +- sound/soc/codecs/adau1761-i2c.c | 14 ++- sound/soc/codecs/adau1761-spi.c | 14 ++- sound/soc/codecs/adau1761.c | 129 ++++++++++++++++++++- sound/soc/codecs/adau1781-i2c.c | 10 ++ sound/soc/codecs/adau1781-spi.c | 10 ++ sound/soc/codecs/adau1781.c | 50 +++++++- sound/soc/codecs/adau17x1.h | 6 +- 10 files changed, 296 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau17x1.txt create mode 100644 include/dt-bindings/sound/adau17x1.h -- 2.1.4

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[alsa-devel] [PATCH v8 1/2] ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec
by Subhransu S. Prusty 24 Feb '16

24 Feb '16

Powering off codec immediately after sending D3 verb may not set the node to D3 state. So wait for a confirmation response before shutting down codec. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 2445b43..aa953a5 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1486,6 +1486,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + unsigned long timeout; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1495,10 +1496,19 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) return 0; /* Power down afg */ - if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) + if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) { snd_hdac_codec_write(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + /* Wait till power state is set to D3 */ + timeout = jiffies + msecs_to_jiffies(1000); + while (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3) + && time_before(jiffies, timeout)) { + + msleep(50); + } + } + err = snd_hdac_display_power(bus, false); if (err < 0) { dev_err(bus->dev, "Cannot turn on display power on i915\n"); -- 1.9.1

3 4

[alsa-devel] [PATCH v9 1/2] ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec
by Subhransu S. Prusty 24 Feb '16

24 Feb '16

Powering off codec immediately after sending D3 verb may not set the node to D3 state. So wait for a confirmation response before shutting down codec. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 2445b43..aa953a5 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1486,6 +1486,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + unsigned long timeout; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1495,10 +1496,19 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) return 0; /* Power down afg */ - if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) + if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) { snd_hdac_codec_write(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + /* Wait till power state is set to D3 */ + timeout = jiffies + msecs_to_jiffies(1000); + while (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3) + && time_before(jiffies, timeout)) { + + msleep(50); + } + } + err = snd_hdac_display_power(bus, false); if (err < 0) { dev_err(bus->dev, "Cannot turn on display power on i915\n"); -- 1.9.1

2 2

[alsa-devel] [PATCH v2] ASoC: wm9713: fix regmap free path
by Robert Jarzmik 24 Feb '16

24 Feb '16

In the conversion to regmap, I assumed that the devm_() variant could be used in the soc probe function. As a mater of fact with the current code the regmap is freed twice because of the devm_() call: (mutex_lock) from [<c01f6624>] (debugfs_remove_recursive+0x50/0x1d0) (debugfs_remove_recursive) from [<c02bf800>] (regmap_debugfs_exit+0x1c/0xd4) (regmap_debugfs_exit) from [<c02ba1f8>] (regmap_exit+0x28/0xc8) (regmap_exit) from [<c02aa258>] (release_nodes+0x18c/0x204) (release_nodes) from [<c02a278c>] (device_release+0x18/0x90) (device_release) from [<c0239030>] (kobject_release+0x90/0x1bc) (kobject_release) from [<c0395c94>] (wm9713_soc_remove+0x1c/0x24) (wm9713_soc_remove) from [<c0384884>] (soc_remove_component+0x50/0x7c) (soc_remove_component) from [<c0386c28>] (soc_remove_dai_links+0x118/0x228) (soc_remove_dai_links) from [<c038721c>] (snd_soc_register_card+0x4e4/0xdd4) (snd_soc_register_card) from [<c0393c54>] (devm_snd_soc_register_card+0x34/0x70) Fix this by replacing the devm_regmap initialization code with the non devm_() variant. Fixes: 700dadfefc3d ASoC: wm9713: convert to regmap Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: changed the fix into removing the devm_() variant --- sound/soc/codecs/wm9713.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4e522302bb8a..edd270f6d8e5 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1212,7 +1212,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); - regmap = devm_regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); + regmap = regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); if (IS_ERR(regmap)) { snd_soc_free_ac97_codec(wm9713->ac97); return PTR_ERR(regmap); -- 2.1.4

4 3

[alsa-devel] [PATCH] ASoC: rockchip: i2s: add default values for registers
by Sugar Zhang 24 Feb '16

24 Feb '16

this patch add default values for registers according description from TRM. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 6561c4c..a0b0942 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -440,11 +440,21 @@ static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) } } +static const struct reg_default rockchip_i2s_reg_defaults[] = { + {0x00, 0x0000000f}, + {0x04, 0x0000000f}, + {0x08, 0x00071f1f}, + {0x10, 0x001f0000}, + {0x14, 0x01f00000}, +}; + static const struct regmap_config rockchip_i2s_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, .max_register = I2S_RXDR, + .reg_defaults = rockchip_i2s_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rockchip_i2s_reg_defaults), .writeable_reg = rockchip_i2s_wr_reg, .readable_reg = rockchip_i2s_rd_reg, .volatile_reg = rockchip_i2s_volatile_reg, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: compress: Add DSP private metadata interface
by Tim Sheridan 23 Feb '16

23 Feb '16

Provides an area to communicate with a DSP to implement support for weird and wonderful features of DSP hardware that aren't supported in the compress offload API. Signed-off-by: Tim Sheridan <tim.sheridan(a)imgtec.com> --- include/uapi/sound/compress_offload.h | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index e00d8cb..65ac445 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 3) /** * struct snd_compressed_buffer - compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -150,6 +150,8 @@ struct snd_compr_metadata { * SNDRV_COMPRESS_SET_PARAMS: Set codec and stream parameters * Note: only codec params can be changed runtime and stream params cant be * SNDRV_COMPRESS_GET_PARAMS: Query codec params + * SNDRV_COMPRESS_SET_DSP_PRIV: Set private DSP metadata + * SNDRV_COMPRESS_GET_DSP_PRIV: Get private DSP metadata * SNDRV_COMPRESS_TSTAMP: get the current timestamp value * SNDRV_COMPRESS_AVAIL: get the current buffer avail value. * This also queries the tstamp properties @@ -171,6 +173,8 @@ struct snd_compr_metadata { struct snd_compr_metadata) #define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ struct snd_compr_metadata) +#define SNDRV_COMPRESS_SET_DSP_PRIV _IOW('C', 0x16, struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_DSP_PRIV _IOWR('C', 0x17, struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) -- 1.7.1

4 6

[alsa-devel] [RFC PATCH] ALSA: hda - hdmi eld control created based on pcm
by libin.yang＠linux.intel.com 23 Feb '16

23 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> eld control is created based on pcm now. When monitor is connected, eld control will be bound to pin automatically. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 74 ++++++++++++++++++++++++++++++---------------- 1 file changed, 48 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 541986f..d77509a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -86,7 +86,6 @@ struct hdmi_spec_per_pin { struct hdmi_eld sink_eld; struct mutex lock; struct delayed_work work; - struct snd_kcontrol *eld_ctl; struct hdmi_pcm *pcm; /* pointer to spec->pcm_rec[n] dynamically*/ int pcm_idx; /* which pcm is attached. -1 means no pcm is attached */ int repoll_count; @@ -136,6 +135,7 @@ struct hdmi_ops { struct hdmi_pcm { struct hda_pcm *pcm; struct snd_jack *jack; + struct snd_kcontrol *eld_ctl; }; struct hdmi_spec { @@ -471,17 +471,22 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; - int pin_idx; + int pcm_idx; uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; - pin_idx = kcontrol->private_value; - per_pin = get_pin(spec, pin_idx); + pcm_idx = kcontrol->private_value; + mutex_lock(&spec->pcm_lock); + per_pin = pcm_idx_to_pin(spec, pcm_idx); + if (!per_pin) { + /* no pin is bound to the pcm */ + uinfo->count = 0; + mutex_unlock(&spec->pcm_lock); + return 0; + } eld = &per_pin->sink_eld; - - mutex_lock(&per_pin->lock); uinfo->count = eld->eld_valid ? eld->eld_size : 0; - mutex_unlock(&per_pin->lock); + mutex_unlock(&spec->pcm_lock); return 0; } @@ -493,16 +498,23 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; - int pin_idx; + int pcm_idx; - pin_idx = kcontrol->private_value; - per_pin = get_pin(spec, pin_idx); + pcm_idx = kcontrol->private_value; + mutex_lock(&spec->pcm_lock); + per_pin = pcm_idx_to_pin(spec, pcm_idx); + if (!per_pin) { + /* no pin is bound to the pcm */ + memset(ucontrol->value.bytes.data, 0, + ARRAY_SIZE(ucontrol->value.bytes.data)); + mutex_unlock(&spec->pcm_lock); + return 0; + } eld = &per_pin->sink_eld; - mutex_lock(&per_pin->lock); if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || eld->eld_size > ELD_MAX_SIZE) { - mutex_unlock(&per_pin->lock); + mutex_unlock(&spec->pcm_lock); snd_BUG(); return -EINVAL; } @@ -512,7 +524,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, if (eld->eld_valid) memcpy(ucontrol->value.bytes.data, eld->eld_buffer, eld->eld_size); - mutex_unlock(&per_pin->lock); + mutex_unlock(&spec->pcm_lock); return 0; } @@ -525,7 +537,7 @@ static struct snd_kcontrol_new eld_bytes_ctl = { .get = hdmi_eld_ctl_get, }; -static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pcm_idx, int device) { struct snd_kcontrol *kctl; @@ -535,14 +547,17 @@ static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, kctl = snd_ctl_new1(&eld_bytes_ctl, codec); if (!kctl) return -ENOMEM; - kctl->private_value = pin_idx; + kctl->private_value = pcm_idx; kctl->id.device = device; - err = snd_hda_ctl_add(codec, get_pin(spec, pin_idx)->pin_nid, kctl); + /* no pin nid is associated with the kctl now + * tbd: associate pin nid to eld ctl later + */ + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - get_pin(spec, pin_idx)->eld_ctl = kctl; + get_hdmi_pcm(spec, pcm_idx)->eld_ctl = kctl; return 0; } @@ -1841,7 +1856,10 @@ static void update_eld(struct hda_codec *codec, struct hdmi_spec *spec = codec->spec; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; + int pcm_idx = -1; + /* for monitor disconnection, save pcm_idx firstly */ + pcm_idx = per_pin->pcm_idx; if (spec->dyn_pcm_assign) { if (eld->eld_valid) { hdmi_attach_hda_pcm(spec, per_pin); @@ -1851,6 +1869,11 @@ static void update_eld(struct hda_codec *codec, hdmi_detach_hda_pcm(spec, per_pin); } } + /* if pcm_idx == -1, it means this is in monitor connection event + * we can get the correct pcm_idx now. + */ + if (pcm_idx == -1) + pcm_idx = per_pin->pcm_idx; if (eld->eld_valid) snd_hdmi_show_eld(codec, &eld->info); @@ -1884,11 +1907,11 @@ static void update_eld(struct hda_codec *codec, hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } - if (eld_changed) + if (eld_changed && pcm_idx >= 0) snd_ctl_notify(codec->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, - &per_pin->eld_ctl->id); + &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id); } /* update ELD and jack state via HD-audio verbs */ @@ -2629,17 +2652,16 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pcm_idx); - } - - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); /* add control for ELD Bytes */ - err = hdmi_create_eld_ctl(codec, pin_idx, - get_pcm_rec(spec, pin_idx)->device); - + err = hdmi_create_eld_ctl(codec, pcm_idx, + get_pcm_rec(spec, pcm_idx)->device); if (err < 0) return err; + } + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hdmi_present_sense(per_pin, 0); } -- 1.9.1

3 3

[alsa-devel] [alsa-lib][PATCH 0/6] control: add APIs for control element set
by Takashi Sakamoto 23 Feb '16

23 Feb '16

This patchset adds some APIs for control element set, according to design of ALSA control core. Additionally, for the APIs, some API documentation is improved. In ALSA control core, each sound card can have control elements. The elements are managed according to below design. * element set * A set of elements with the same attribute (i.e. name, get/put operations). Some element sets can be added to a sound card. [0] * element * An element can be identified by userspace applications. * channel * An element has channels which can be changed by userspace applications or hardware device itself. * value * Each channel has its own value. The place to kept the value depends on design of each device driver (i.e. in hardware register or kernel space). There're two ways to identify the element; by identification number(numid), or by a combination of name of the element set and index from the first element in the element set [1]. Both of device drivers in kernel land and ALSA control applications in user land can add the element set. When userspace applications add them, ALSA control core works as Inter Process Communication mechanism between event-listening applications and usual control applications. Such element is referred as 'user-defined control elements' in alsa-lib. Typically, device drivers in kernel land adds one element set for a single element. So users and userspace developers are hard to realize that an existence of the element set. There are more items to describe the design. But here I focus on what relates to this patchset. Well, current alsa-lib implementation includes small bugs, some misleadings, a lack of feature related to user-defined control elements. This patchset is my attempt to improves them. [0] In kernel, the element set is represented by 'struct snd_kcontrol'. Some developers call it as 'kernel control'. The others call it as 'control group'. Here, I suggest 'element set', because from userspace the 'struct snd_kcontrol' is not visible and any elements are just visible. Furthermore, I'd like to use unique name against Linux control groups. [1] Actually, when identifying an element according to name/index pair, the other members of 'struct snd_ctl_elem_id' such as iface, device and subdevice are required. They're required to seek element set, then index is used to find an element. Regards Takashi Sakamoto (6): control: rename variables so that it represents the number of channels which an element has control: improve API documentation for functions to add elements control: improve API documentation for functions to set/get channel values for an element control: rename variables so that it represents a channel in element control: add a function to add an element of bytes type control: add functions to add element set include/control.h | 55 +++-- src/control/control.c | 589 ++++++++++++++++++++++++++++++++++++++------------ 2 files changed, 487 insertions(+), 157 deletions(-) -- 2.5.0

3 10

[alsa-devel] Compute Stick baytrail audio problem
by Giacomo Comes 23 Feb '16

23 Feb '16

Hi, I have recently bought a Intel Compute Stick and I am having trouble with the audio. Intel provides a custom Ubuntu release that support audio, but I want to run the linux distribution of my choice on it (openSUSE 13.2 for the moment). In order to have HDMI audio support it is necessary to compile a kernel with this patch (https://raw.githubusercontent.com/01org/baytrailaudio/master/hdmi_audio_201…) provided by Intel for kernel 3.16.7 (https://01.org/ubuntu-hdmi) I compiled the openSUSE kernel using the default openSUSE .config plus the following: CONFIG_SUPPORT_HDMI=y CONFIG_SND_SOC=m CONFIG_SND_SOC_INTEL_SST=m CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=m I booted the new kernel and aplay -l gives this output: **** List of PLAYBACK Hardware Devices **** card 0: IntelHDMI [IntelHDMI], device 0: IntelHDMI [IntelHDMI] Subdevices: 0/1 Subdevice #0: subdevice #0 but when I run: aplay /usr/share/sounds/alsa/test.wav the audio plays for .2 seconds and then nothing more. Sometime it will play later a little bit more and stop again. At some point these messages appears: underrun!!! (at least 29325.946 ms long) underrun!!! (at least 1535.889 ms long) underrun!!! (at least 3196.213 ms long) I have looked around on google for a hint on how to fix this issue but I found nothing. Any idea where to look for a solution? Attached there are the dmesg and lsmod output for ubuntu with the working audio and openSUSE 13.2 plus patch without the working audio. I can provide any other information you need. Thanks for any help. Giacomo

2 2

[alsa-devel] [PATCH v1 0/2] patches to add support for rockchip rt5640 card
by Sugar Zhang 23 Feb '16

23 Feb '16

add default values for i2s registers from TRM. rt5640: add master clock handing. Sugar Zhang (2): ASoC: rockchip: i2s: add default values for registers ASoC: rt5640: add master clock handling for rt5640 Documentation/devicetree/bindings/sound/rt5640.txt | 3 +++ sound/soc/codecs/rt5640.c | 31 ++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2 ++ sound/soc/rockchip/rockchip_i2s.c | 10 +++++++ 4 files changed, 46 insertions(+) -- 1.9.1

3 6

[alsa-devel] Compress offload: DSP buffer timestamp passing
by Tim Sheridan 22 Feb '16

22 Feb '16

Hi, I've been working on the Imagination Technologies AXD (audio DSP) compress offload driver. One of the features of the DSP is scheduled playback of audio using timestamps (that are obtained by an application from userspace). A current problem with the compress offload API is that there's no "blessed" way to get these timestamps from tinycompress through to our compress offload driver. Currently, I've added a SNDRV_COMPRESS_ENCODER_PTS value to the sndrv_compress_encoder enum, and exposed this with an API in tinycompress, handling this in my driver to pass it to the DSP. Does this sound like a reasonable approach to go about adding compress offload API support for this? Or is there some other part of the ALSA API which would be more appropriate to use instead of this? Cheers, Tim -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

2 2

[alsa-devel] [PATCH] ALSA: hda - Apply clock gate workaround to Skylake, too
by Takashi Iwai 22 Feb '16

22 Feb '16

Some Skylake machines show the codec probe errors in certain situations, e.g. HP Z240 desktop fails to probe the onboard Realtek codec at reloading the snd-hda-intel module like: snd_hda_intel 0000:00:1f.3: spurious response 0x200:0x2, last cmd=0x000000 snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to polling mode: lastcmd=0x000f0000 snd_hda_intel 0000:00:1f.3: No response from codec, disabling MSI: last cmd=0x000f0000 snd_hda_intel 0000:00:1f.3: Codec #0 probe error; disabling it... hdaudio hdaudioC0D2: no AFG or MFG node found snd_hda_intel 0000:00:1f.3: no codecs initialized Also, HP G470 G3 suffers from the similar problem, as reported in bugzilla below. On this machine, the codec probe error appears even at a fresh boot. As Libin suggested, the same workaround used for Broxton in the commit [6639484ddaf6: ALSA: hda - disable dynamic clock gating on Broxton before reset] can be applied for Skylake in order to fix this problem. The Intel HW team also confirmed that this is needed for SKL. This patch makes the workaround applied to both SKL and BXT platforms. The referred macros are moved and one superfluous macro (IS_BROXTON()) is another one (IS_BXT()) as well. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=112731 Suggested-by: Libin Yang <libin.yang(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ce6b97f31390..e5240cb3749f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -363,7 +363,10 @@ enum { ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c)) -#define IS_BROXTON(pci) ((pci)->device == 0x5a98) +#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) +#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) +#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) +#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -540,13 +543,13 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, true); - if (IS_BROXTON(pci)) { + if (IS_SKL_PLUS(pci)) { pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); val = val & ~INTEL_HDA_CGCTL_MISCBDCGE; pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); } azx_init_chip(chip, full_reset); - if (IS_BROXTON(pci)) { + if (IS_SKL_PLUS(pci)) { pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); val = val | INTEL_HDA_CGCTL_MISCBDCGE; pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); @@ -555,7 +558,7 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) snd_hdac_set_codec_wakeup(bus, false); /* reduce dma latency to avoid noise */ - if (IS_BROXTON(pci)) + if (IS_BXT(pci)) bxt_reduce_dma_latency(chip); } @@ -977,11 +980,6 @@ static int azx_resume(struct device *dev) /* put codec down to D3 at hibernation for Intel SKL+; * otherwise BIOS may still access the codec and screw up the driver */ -#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) -#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) -#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) -#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) - static int azx_freeze_noirq(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); -- 2.7.1

1 0

[alsa-devel] [PATCH] ALSA: mts64: use new parport device model
by Sudip Mukherjee 22 Feb '16

22 Feb '16

Modify mts64 driver to use the new parallel port device model. The advantage of using the device model is that the driver gets binded to the hardware, we get the feature of hotplug, we can bind/unbind the driver at runtime. The changes are in the way the driver gets registered with the parallel port subsystem and the temporary device to probe mts64 card is removed and mts64_probe() is used in the probe callback. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- exactly same changes as done in portman2x4. BTW, there is one portman2x4 available on ebay. I was bidding for it but someone is more desperate than I am and outbidding me everytime :( sound/drivers/mts64.c | 95 +++++++++++++++++++++++---------------------------- 1 file changed, 43 insertions(+), 52 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 2a008a9..c76bd87 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -65,8 +65,6 @@ struct mts64 { struct snd_card *card; struct snd_rawmidi *rmidi; struct pardevice *pardev; - int pardev_claimed; - int open_count; int current_midi_output_port; int current_midi_input_port; @@ -850,30 +848,6 @@ __out: spin_unlock(&mts->lock); } -static int snd_mts64_probe_port(struct parport *p) -{ - struct pardevice *pardev; - int res; - - pardev = parport_register_device(p, DRIVER_NAME, - NULL, NULL, NULL, - 0, NULL); - if (!pardev) - return -EIO; - - if (parport_claim(pardev)) { - parport_unregister_device(pardev); - return -EIO; - } - - res = mts64_probe(p); - - parport_release(pardev); - parport_unregister_device(pardev); - - return res; -} - static void snd_mts64_attach(struct parport *p) { struct platform_device *device; @@ -907,10 +881,20 @@ static void snd_mts64_detach(struct parport *p) /* nothing to do here */ } +static int snd_mts64_dev_probe(struct pardevice *pardev) +{ + if (strcmp(pardev->name, DRIVER_NAME)) + return -ENODEV; + + return 0; +} + static struct parport_driver mts64_parport_driver = { - .name = "mts64", - .attach = snd_mts64_attach, - .detach = snd_mts64_detach + .name = "mts64", + .probe = snd_mts64_dev_probe, + .match_port = snd_mts64_attach, + .detach = snd_mts64_detach, + .devmodel = true, }; /********************************************************************* @@ -922,8 +906,7 @@ static void snd_mts64_card_private_free(struct snd_card *card) struct pardevice *pardev = mts->pardev; if (pardev) { - if (mts->pardev_claimed) - parport_release(pardev); + parport_release(pardev); parport_unregister_device(pardev); } @@ -938,6 +921,12 @@ static int snd_mts64_probe(struct platform_device *pdev) struct snd_card *card = NULL; struct mts64 *mts = NULL; int err; + struct pardev_cb mts64_cb = { + .preempt = NULL, + .wakeup = NULL, + .irq_func = snd_mts64_interrupt, /* ISR */ + .flags = PARPORT_DEV_EXCL, /* flags */ + }; p = platform_get_drvdata(pdev); platform_set_drvdata(pdev, NULL); @@ -946,8 +935,6 @@ static int snd_mts64_probe(struct platform_device *pdev) return -ENODEV; if (!enable[dev]) return -ENOENT; - if ((err = snd_mts64_probe_port(p)) < 0) - return err; err = snd_card_new(&pdev->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -960,23 +947,32 @@ static int snd_mts64_probe(struct platform_device *pdev) sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); - pardev = parport_register_device(p, /* port */ - DRIVER_NAME, /* name */ - NULL, /* preempt */ - NULL, /* wakeup */ - snd_mts64_interrupt, /* ISR */ - PARPORT_DEV_EXCL, /* flags */ - (void *)card); /* private */ - if (pardev == NULL) { + mts64_cb.private = card; /* private */ + pardev = parport_register_dev_model(p, /* port */ + DRIVER_NAME, /* name */ + &mts64_cb, /* callbacks */ + pdev->id); /* device number */ + if (!pardev) { snd_printd("Cannot register pardevice\n"); err = -EIO; goto __err; } + /* claim parport */ + if (parport_claim(pardev)) { + snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); + err = -EIO; + goto free_pardev; + } + err = mts64_probe(p); + if (err) { + err = -EIO; + goto release_pardev; + } + if ((err = snd_mts64_create(card, pardev, &mts)) < 0) { snd_printd("Cannot create main component\n"); - parport_unregister_device(pardev); - goto __err; + goto release_pardev; } card->private_data = mts; card->private_free = snd_mts64_card_private_free; @@ -986,14 +982,6 @@ static int snd_mts64_probe(struct platform_device *pdev) goto __err; } - /* claim parport */ - if (parport_claim(pardev)) { - snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); - err = -EIO; - goto __err; - } - mts->pardev_claimed = 1; - /* init device */ if ((err = mts64_device_init(p)) < 0) goto __err; @@ -1009,6 +997,10 @@ static int snd_mts64_probe(struct platform_device *pdev) snd_printk(KERN_INFO "ESI Miditerminal 4140 on 0x%lx\n", p->base); return 0; +release_pardev: + parport_release(pardev); +free_pardev: + parport_unregister_device(pardev); __err: snd_card_free(card); return err; @@ -1024,7 +1016,6 @@ static int snd_mts64_remove(struct platform_device *pdev) return 0; } - static struct platform_driver snd_mts64_driver = { .probe = snd_mts64_probe, .remove = snd_mts64_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH] control: keep consistent usage of assertion
by Takashi Sakamoto 22 Feb '16

22 Feb '16

In most codes for control APIs, assertions are used to check given arguments from applications. If the arguments are not expected, the application is forced to abort. When NDEBUG is defined in the beginning of the codes, no checks are performed. Although, in snd_ctl_elem_set_bytes(), assertion is used with condition statement. There's an intention to check the arguments regardless of NDEBUG. However, this is not the same fashion in the codes. This commit applies the same fashion to the function, to keep consistent usage of assertion. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 328920d..70d168d 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2627,10 +2627,7 @@ void snd_ctl_elem_value_set_byte(snd_ctl_elem_value_t *obj, unsigned int idx, un void snd_ctl_elem_set_bytes(snd_ctl_elem_value_t *obj, void *data, size_t size) { assert(obj); - if (size >= ARRAY_SIZE(obj->value.bytes.data)) { - assert(0); - return; - } + assert(size < ARRAY_SIZE(obj->value.bytes.data)); memcpy(obj->value.bytes.data, data, size); } -- 2.5.0

2 1

[alsa-devel] [alsa-lib][PATCH] control: correct assertion in snd_ctl_elem_set_bytes()
by Takashi Sakamoto 22 Feb '16

22 Feb '16

The assert(3) aborts running process when expression arguments is false (zero). Although, the snd_ctl_elem_set_bytes() has return statement after assert(3). It has meaningless. This commit corrects the statements. Fixes: 7893ea238d5a('Added mode argument to open functions where it was missing. First part of CTL documentation') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 328920d..70d168d 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2627,10 +2627,7 @@ void snd_ctl_elem_value_set_byte(snd_ctl_elem_value_t *obj, unsigned int idx, un void snd_ctl_elem_set_bytes(snd_ctl_elem_value_t *obj, void *data, size_t size) { assert(obj); - if (size >= ARRAY_SIZE(obj->value.bytes.data)) { - assert(0); - return; - } + assert(size < ARRAY_SIZE(obj->value.bytes.data)); memcpy(obj->value.bytes.data, data, size); } -- 2.5.0

2 5

[alsa-devel] [PATCH] ASoC: mediatek: Add machine driver for ALC5650 codec
by PC Liao 22 Feb '16

22 Feb '16

From: Koro Chen <koro.chen(a)mediatek.com> This is the DPCM based machine driver with rt5650 Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 15 ++ sound/soc/mediatek/Kconfig | 10 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8173-rt5650.c | 236 ++++++++++++++++++++ 4 files changed, 262 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650.txt create mode 100644 sound/soc/mediatek/mt8173-rt5650.c diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt new file mode 100644 index 0000000..fe5a5ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -0,0 +1,15 @@ +MT8173 with RT5650 CODECS + +Required properties: +- compatible : "mediatek,mt8173-rt5650" +- mediatek,audio-codec: the phandles of rt5650 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650"; + mediatek,audio-codec = <&rt5650>; + mediatek,platform = <&afe>; + }; + diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 9769676..6c88087 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -17,6 +17,16 @@ config SND_SOC_MT8173_MAX98090 Select Y if you have such device. If unsure select "N". +config SND_SOC_MT8173_RT5650 + tristate "ASoC Audio driver for MT8173 with RT5650 codec" + depends on SND_SOC_MEDIATEK && I2C + select SND_SOC_RT5645 + help + This adds ASoC driver for Mediatek MT8173 boards + with the RT5650 audio codec. + Select Y if you have such device. + If unsure select "N". + config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" depends on SND_SOC_MEDIATEK && I2C diff --git a/sound/soc/mediatek/Makefile b/sound/soc/mediatek/Makefile index 75effbe..bcdcb06 100644 --- a/sound/soc/mediatek/Makefile +++ b/sound/soc/mediatek/Makefile @@ -3,3 +3,4 @@ obj-$(CONFIG_SND_SOC_MEDIATEK) += mtk-afe-pcm.o # Machine support obj-$(CONFIG_SND_SOC_MT8173_MAX98090) += mt8173-max98090.o obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o +obj-$(CONFIG_SND_SOC_MT8173_RT5650) += mt8173-rt5650.o diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c new file mode 100644 index 0000000..bb09bb1 --- /dev/null +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -0,0 +1,236 @@ +/* + * mt8173-rt5650.c -- MT8173 machine driver with RT5650 codecs + * + * Copyright (c) 2016 MediaTek Inc. + * Author: Koro Chen <koro.chen(a)mediatek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" + +#define MCLK_FOR_CODECS 12288000 + +static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Headset Mic", NULL, "micbias1"}, + {"Headset Mic", NULL, "micbias2"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, +}; + +static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + + /* pll from mclk 12.288M */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, + params_rate(params) * 512); + if (ret) + return ret; + + /* sysclk from pll */ + ret = snd_soc_dai_set_sysclk(codec_dai, 1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + return 0; +} + +static struct snd_soc_ops mt8173_rt5650_ops = { + .hw_params = mt8173_rt5650_hw_params, +}; + +static struct snd_soc_jack mt8173_rt5650_jack; + +static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + int ret; + + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + /* enable jack detection */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &mt8173_rt5650_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "Can't new Headset Jack %d\n", ret); + return ret; + } + + return rt5645_set_jack_detect(codec, + &mt8173_rt5650_jack, + &mt8173_rt5650_jack, + &mt8173_rt5650_jack); +} + +static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { + { + .dai_name = "rt5645-aif1", + }, +}; + +enum { + DAI_LINK_PLAYBACK, + DAI_LINK_CAPTURE, + DAI_LINK_CODEC_I2S, +}; + +/* Digital audio interface glue - connects codec <---> CPU */ +static struct snd_soc_dai_link mt8173_rt5650_dais[] = { + /* Front End DAI links */ + [DAI_LINK_PLAYBACK] = { + .name = "rt5650 Playback", + .stream_name = "rt5650 Playback", + .cpu_dai_name = "DL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, + [DAI_LINK_CAPTURE] = { + .name = "rt5650 Capture", + .stream_name = "rt5650 Capture", + .cpu_dai_name = "VUL", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_capture = 1, + }, + /* Back End DAI links */ + [DAI_LINK_CODEC_I2S] = { + .name = "Codec", + .cpu_dai_name = "I2S", + .no_pcm = 1, + .codecs = mt8173_rt5650_codecs, + .num_codecs = 1, + .init = mt8173_rt5650_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &mt8173_rt5650_ops, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +static struct snd_soc_card mt8173_rt5650_card = { + .name = "mtk-rt5650", + .owner = THIS_MODULE, + .dai_link = mt8173_rt5650_dais, + .num_links = ARRAY_SIZE(mt8173_rt5650_dais), + .controls = mt8173_rt5650_controls, + .num_controls = ARRAY_SIZE(mt8173_rt5650_controls), + .dapm_widgets = mt8173_rt5650_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8173_rt5650_widgets), + .dapm_routes = mt8173_rt5650_routes, + .num_dapm_routes = ARRAY_SIZE(mt8173_rt5650_routes), +}; + +static int mt8173_rt5650_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mt8173_rt5650_card; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_dais[i].platform_name) + continue; + mt8173_rt5650_dais[i].platform_of_node = platform_node; + } + + mt8173_rt5650_codecs[0].of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); + if (!mt8173_rt5650_codecs[0].of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + return ret; +} + +static const struct of_device_id mt8173_rt5650_dt_match[] = { + { .compatible = "mediatek,mt8173-rt5650", }, + { } +}; +MODULE_DEVICE_TABLE(of, mt8173_rt5650_dt_match); + +static struct platform_driver mt8173_rt5650_driver = { + .driver = { + .name = "mtk-rt5650", + .of_match_table = mt8173_rt5650_dt_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, + .probe = mt8173_rt5650_dev_probe, +}; + +module_platform_driver(mt8173_rt5650_driver); + +/* Module information */ +MODULE_DESCRIPTION("MT8173 RT5650 SoC machine driver"); +MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mtk-rt5650"); + -- 1.7.9.5

2 3

[alsa-devel] [PATCH 0/3] ALSA: hda: Move chmap helpers to core
by Subhransu S. Prusty 22 Feb '16

22 Feb '16

The HDMI channel map helper APIs and control registration can be reused by ASoC skylake HDMI drivers. A common chmap object is created, channel map control callback handlers, helper APIs and data structures are moved to the core. Driver object store an instance of this chmap object to register chmap controls and program channel allocation for HDMI infoframe. Driver registers ops for specific operations. Subhransu S. Prusty (3): ALSA: hda - Create common chmap object ALSA: hda - Move chmap support helpers/structures to core ALSA: hda - Use the helpers from HDA core include/sound/hdaudio.h | 1 + include/sound/hdmi_chmap.h | 84 +++++ sound/hda/Makefile | 2 +- sound/hda/hdmi_chmap.c | 763 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_eld.c | 31 +- sound/pci/hda/patch_hdmi.c | 794 ++++----------------------------------------- 6 files changed, 923 insertions(+), 752 deletions(-) create mode 100644 include/sound/hdmi_chmap.h create mode 100644 sound/hda/hdmi_chmap.c -- 1.9.1

2 7

[alsa-devel] [PATCH] ASoC: rockchip: add bindings for rk3399 i2s
by jianqun.xu 22 Feb '16

22 Feb '16

From: Jianqun Xu <jay.xu(a)rock-chips.com> Add devicetree bindings for i2s controller found on rk3399 processors from rockchip. It's helpful to add full set of compatible strings for serials of Rockchip SoCs (rk3066, rk3188, rk3288, rk3399). Acked-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Jianqun Xu <jay.xu(a)rock-chips.com> --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 1 + sound/soc/rockchip/rockchip_i2s.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index b7f3a93..6e86d8a 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -9,6 +9,7 @@ Required properties: - "rockchip,rk3066-i2s": for rk3066 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 + - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the I2S interrupt. diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 6561c4c..531a6d2 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -575,6 +575,9 @@ static int rockchip_i2s_remove(struct platform_device *pdev) static const struct of_device_id rockchip_i2s_match[] = { { .compatible = "rockchip,rk3066-i2s", }, + { .compatible = "rockchip,rk3188-i2s", }, + { .compatible = "rockchip,rk3288-i2s", }, + { .compatible = "rockchip,rk3399-i2s", }, {}, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: topology: fix build warning
by Sudip Mukherjee 22 Feb '16

22 Feb '16

We are getting build warning about: sound/soc/soc-topology.c:249:31: warning: 'get_dobj_type' defined but not used commit 64527e8a3529 has removed the only caller of get_dobj_type() but missed to remove the function which is now unused. And after removing get_dobj_type() we do not have any use of get_dobj_mixer_type(). Fixes: 64527e8a3529 ("ASoC: topology: Add FE DAIs dynamically") CC: Mengdong Lin <mengdong.lin(a)linux.intel.com> Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/soc-topology.c | 45 --------------------------------------------- 1 file changed, 45 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0eb01e8..07f61a6 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -223,51 +223,6 @@ static int get_widget_id(int tplg_type) return -EINVAL; } -static enum snd_soc_dobj_type get_dobj_mixer_type( - struct snd_soc_tplg_ctl_hdr *control_hdr) -{ - if (control_hdr == NULL) - return SND_SOC_DOBJ_NONE; - - switch (control_hdr->ops.info) { - case SND_SOC_TPLG_CTL_VOLSW: - case SND_SOC_TPLG_CTL_VOLSW_SX: - case SND_SOC_TPLG_CTL_VOLSW_XR_SX: - case SND_SOC_TPLG_CTL_RANGE: - case SND_SOC_TPLG_CTL_STROBE: - return SND_SOC_DOBJ_MIXER; - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_CTL_ENUM_VALUE: - return SND_SOC_DOBJ_ENUM; - case SND_SOC_TPLG_CTL_BYTES: - return SND_SOC_DOBJ_BYTES; - default: - return SND_SOC_DOBJ_NONE; - } -} - -static enum snd_soc_dobj_type get_dobj_type(struct snd_soc_tplg_hdr *hdr, - struct snd_soc_tplg_ctl_hdr *control_hdr) -{ - switch (hdr->type) { - case SND_SOC_TPLG_TYPE_MIXER: - return get_dobj_mixer_type(control_hdr); - case SND_SOC_TPLG_TYPE_DAPM_GRAPH: - case SND_SOC_TPLG_TYPE_MANIFEST: - return SND_SOC_DOBJ_NONE; - case SND_SOC_TPLG_TYPE_DAPM_WIDGET: - return SND_SOC_DOBJ_WIDGET; - case SND_SOC_TPLG_TYPE_DAI_LINK: - return SND_SOC_DOBJ_DAI_LINK; - case SND_SOC_TPLG_TYPE_PCM: - return SND_SOC_DOBJ_PCM; - case SND_SOC_TPLG_TYPE_CODEC_LINK: - return SND_SOC_DOBJ_CODEC_LINK; - default: - return SND_SOC_DOBJ_NONE; - } -} - static inline void soc_bind_err(struct soc_tplg *tplg, struct snd_soc_tplg_ctl_hdr *hdr, int index) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fix memory leak
by Sudip Mukherjee 22 Feb '16

22 Feb '16

If dai_link is already bound then we just returned and leaked rtd and rtd->codec_dais which were allocated by soc_new_pcm_runtime(). We do not need this newly allocated rtd to check if dai_link is already binded. Lets check first if it is already binded before allocating this memory. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 790ee2b..d2e62b15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -986,16 +986,16 @@ static int soc_bind_dai_link(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); - rtd = soc_new_pcm_runtime(card, dai_link); - if (!rtd) - return -ENOMEM; - if (soc_is_dai_link_bound(card, dai_link)) { dev_dbg(card->dev, "ASoC: dai link %s already bound\n", dai_link->name); return 0; } + rtd = soc_new_pcm_runtime(card, dai_link); + if (!rtd) + return -ENOMEM; + cpu_dai_component.name = dai_link->cpu_name; cpu_dai_component.of_node = dai_link->cpu_of_node; cpu_dai_component.dai_name = dai_link->cpu_dai_name; -- 1.9.1

2 1

[alsa-devel] [PATCH v2]ASoC: topology: Fix setting of stream rates, rate_min and rate_max
by mengdong.lin＠linux.intel.com 22 Feb '16

22 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Directly set a stream's rates, rate_min and rate_max from the topology info. Also define set_stream_info to wrap setting of the stream info. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0eb01e8..726cea7 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1575,6 +1575,18 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } +static void set_stream_info(struct snd_soc_pcm_stream *stream, + struct snd_soc_tplg_stream_caps *caps) +{ + stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + stream->channels_min = caps->channels_min; + stream->channels_max = caps->channels_max; + stream->rates = caps->rates; + stream->rate_min = caps->rate_min; + stream->rate_max = caps->rate_max; + stream->formats = caps->formats; +} + static int soc_tplg_dai_create(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *pcm) { @@ -1593,25 +1605,13 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - - stream->stream_name = kstrdup(caps->name, GFP_KERNEL); - stream->channels_min = caps->channels_min; - stream->channels_max = caps->channels_max; - stream->rates = snd_pcm_rate_range_to_bits(caps->rate_min, - caps->rate_max); - stream->formats = caps->formats; + set_stream_info(stream, caps); } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - - stream->stream_name = kstrdup(caps->name, GFP_KERNEL); - stream->channels_min = caps->channels_min; - stream->channels_max = caps->channels_max; - stream->rates = snd_pcm_rate_range_to_bits(caps->rate_min, - caps->rate_max); - stream->formats = caps->formats; + set_stream_info(stream, caps); } /* pass control to component driver for optional further init */ -- 2.5.0

2 1

[alsa-devel] [RFC 0/4] Add support for DAI link addition dynamically
by Vaibhav Agarwal 22 Feb '16

22 Feb '16

Following patches are based on for-next branch from broonie tree. First 2 patches include changes in existing soc, core framework to prepare for adding support for dynamic DAI link addition/ removal Patch 3 & 4 contains actual changes to enable dynamic DAI link support NOTE: Currently, the code is tested on Pandaboad ES revB3 for playback usecase. Vaibhav Agarwal (4): ASoc: Use ref_count for soc DAI & component alsa: add locked variant for snd_ctl_remove_id ASoC: Enable dynamic DAIlink insertion & removal ASoC: Change soc-card codec_conf array to a list include/sound/control.h | 1 + include/sound/soc-dai.h | 1 + include/sound/soc-dapm.h | 7 +- include/sound/soc-dpcm.h | 1 + include/sound/soc.h | 18 ++- sound/core/control.c | 23 +++ sound/soc/Kconfig | 4 + sound/soc/soc-core.c | 359 ++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/soc-dapm.c | 105 +++++++++++--- sound/soc/soc-pcm.c | 25 ++++ 10 files changed, 501 insertions(+), 43 deletions(-) -- 2.1.4

5 22

Re: [alsa-devel] [PATCH] ASoC: topology: Fix setting of stream rates, rate_min and rate_max
by Mark Brown 22 Feb '16

22 Feb '16

On Tue, Feb 16, 2016 at 02:27:32PM +0800, kbuild test robot wrote: > sound/soc/soc-topology.c: In function 'set_stream_info': > >> sound/soc/soc-topology.c:1588:1: warning: no return statement in function returning non-void [-Wreturn-type] > } > ^ This is a real bug that 0day has identified.

2 1

[alsa-devel] [PATCH v7 0/2] ASoC: Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 22 Feb '16

22 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK --- Changes since v6: - removed legacy dapm components that are no longer needed. - changed dt-bindings patch subject styling to match that of the sound subsystem Changes since v5: - fixed warning that had sneaked in. Changes since v4: - corrected indentation issues - removed old macros - removed unnecessary enabling of apb clock Changes since v3: - removed code required for spdif capture - added disable/enable of apb clk to suspend/resume. - removed unnecessary dt checks in probe - fixed error conditions in probe/resume - modified pm_runtime functionality and confirmed that driver worked when PM was disabled. - removed .owner from platform driver - renamed bindings file Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): ASoC: sunxi :add sun4i SPDIF dt bindings ASoC: sunxi: Add support for the SPDIF block .../bindings/sound/sunxi,sun4i-spdif.txt | 39 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 542 +++++++++++++++++++++ 4 files changed, 590 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.1

3 4

[alsa-devel] [PATCH 1/3] ASoC: Intel: Skylake: fix pointer scaling
by Vinod Koul 22 Feb '16

22 Feb '16

From: Alan <gnomes(a)lxorguk.ukuu.org.uk> skl_tplg_tlv_control_set does pointer maths on data but forgets that data is not uint8_t so the maths is already scaled in the pointer type. Signed-off-by: Alan Cox <alan(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ba0d02d1613e..22a9e6fe1c05 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1111,7 +1111,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, return -EFAULT; } else { if (copy_from_user(ac->params, - data + 2 * sizeof(u32), size)) + data + 2, size)) return -EFAULT; } -- 1.9.1

2 9

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: remove bogus comparison of an array with NULL
by Vinod Koul 22 Feb '16

22 Feb '16

From: Alan Cox <alan(a)linux.intel.com> dfw_ac->params is an array not a pointer. It will never be NULL. The check on ac->max appears sufficient. Signed-off-by: Alan Cox <alan(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22a9e6fe1c05..99f50a034e1d 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1657,8 +1657,7 @@ static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be, if (!ac->params) return -ENOMEM; - if (dfw_ac->params) - memcpy(ac->params, dfw_ac->params, ac->max); + memcpy(ac->params, dfw_ac->params, ac->max); } be->dobj.private = ac; -- 1.9.1

1 0

[alsa-devel] Applied "regmap: Add explict native endian flag to DT bindings" to the asoc tree
by Mark Brown 22 Feb '16

22 Feb '16

The patch regmap: Add explict native endian flag to DT bindings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a06c488da0b0c1eebf710017675f00b9d5f9cf42 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Tue, 26 Jan 2016 17:59:30 +0000 Subject: [PATCH] regmap: Add explict native endian flag to DT bindings Currently the binding document says that if no endianness is configured we use native endian but this is not in fact true for all binding types and we do have some devices that really want native endianness such as Broadcom MIPS SoCs where switching the endianness of the CPU also switches the endianness of external IPs. Provide an explicit option for this. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/regmap/regmap.txt | 11 +++++++---- drivers/base/regmap/regmap.c | 2 ++ 2 files changed, 9 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/regmap/regmap.txt b/Documentation/devicetree/bindings/regmap/regmap.txt index b494f8b..e98a9652 100644 --- a/Documentation/devicetree/bindings/regmap/regmap.txt +++ b/Documentation/devicetree/bindings/regmap/regmap.txt @@ -5,15 +5,18 @@ Index Device Endianness properties --------------------------------------------------- 1 BE 'big-endian' 2 LE 'little-endian' +3 Native 'native-endian' For one device driver, which will run in different scenarios above on different SoCs using the devicetree, we need one way to simplify this. -Required properties: -- {big,little}-endian: these are boolean properties, if absent - meaning that the CPU and the Device are in the same endianness mode, - these properties are for register values and all the buffers only. +Optional properties: +- {big,little,native}-endian: these are boolean properties, if absent + then the implementation will choose a default based on the device + being controlled. These properties are for register values and all + the buffers only. Native endian means that the CPU and device have + the same endianness. Examples: Scenario 1 : CPU in LE mode & device in LE mode. diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index ee54e84..3432634 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -557,6 +557,8 @@ enum regmap_endian regmap_get_val_endian(struct device *dev, endian = REGMAP_ENDIAN_BIG; else if (of_property_read_bool(np, "little-endian")) endian = REGMAP_ENDIAN_LITTLE; + else if (of_property_read_bool(np, "native-endian")) + endian = REGMAP_ENDIAN_NATIVE; /* If the endianness was specified in DT, use that */ if (endian != REGMAP_ENDIAN_DEFAULT) -- 2.7.0

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[alsa-devel] Applied "MIPS: dt: Explicitly specify native endian behaviour for syscon" to the asoc tree
by Mark Brown 22 Feb '16

22 Feb '16

The patch MIPS: dt: Explicitly specify native endian behaviour for syscon has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 25d6463e48ea41149c88dc40648a888699e77158 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Tue, 26 Jan 2016 18:08:06 +0000 Subject: [PATCH] MIPS: dt: Explicitly specify native endian behaviour for syscon On many MIPS systems the endianness of IP blocks is kept the same as that of the CPU by the hardware. This includes the system controllers on these systems which are controlled via syscon which uses the regmap API which used readl() and writel() to interact with the hardware, meaning that all writes are converted to little endian when writing to the hardware. This caused a bad interaction with the regmap core in big endian mode since it was not aware of the byte swapping and so ended up performing little endian writes. Unfortunately when this issue was noticed it was addressed by updating the DT for the affected devices to specify them as little endian. This happened to work since it resulted in two endianness swaps which cancelled each other out and gave little endian behaviour but meant that the DT was clearly not accurately describing the hardware. The intention of commit 29bb45f25ff305 (regmap-mmio: Use native endianness for read/write) was to fix this by making regmap default to native endianness but this breaks most other MMIO users where the hardware has a fixed endianness and the implementation uses the __raw accessors which are not intended to be used outside of architecture code. Instead use the newly added native-endian DT property to say exactly what we want for these systems. Fixes: 29bb45f25ff305 (regmap-mmio: Use native endianness for read/write) Reported-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> Acked-by: Ralf Baechle <ralf(a)linux-mips.org> --- arch/mips/boot/dts/brcm/bcm6328.dtsi | 1 + arch/mips/boot/dts/brcm/bcm6368.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7125.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7346.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7358.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7360.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7362.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7420.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7425.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7435.dtsi | 1 + 10 files changed, 10 insertions(+), 1 deletion(-) diff --git a/arch/mips/boot/dts/brcm/bcm6328.dtsi b/arch/mips/boot/dts/brcm/bcm6328.dtsi index 459b9b2..9d19236 100644 --- a/arch/mips/boot/dts/brcm/bcm6328.dtsi +++ b/arch/mips/boot/dts/brcm/bcm6328.dtsi @@ -74,6 +74,7 @@ timer: timer@10000040 { compatible = "syscon"; reg = <0x10000040 0x2c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm6368.dtsi b/arch/mips/boot/dts/brcm/bcm6368.dtsi index 9c8d3fe2..1f6b9b5 100644 --- a/arch/mips/boot/dts/brcm/bcm6368.dtsi +++ b/arch/mips/boot/dts/brcm/bcm6368.dtsi @@ -54,7 +54,7 @@ periph_cntl: syscon@10000000 { compatible = "syscon"; reg = <0x10000000 0x14>; - little-endian; + native-endian; }; reboot: syscon-reboot@10000008 { diff --git a/arch/mips/boot/dts/brcm/bcm7125.dtsi b/arch/mips/boot/dts/brcm/bcm7125.dtsi index 4fc7ece..3ae1605 100644 --- a/arch/mips/boot/dts/brcm/bcm7125.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7125.dtsi @@ -98,6 +98,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7125-sun-top-ctrl", "syscon"; reg = <0x404000 0x60c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7346.dtsi b/arch/mips/boot/dts/brcm/bcm7346.dtsi index a3039bb..be79919 100644 --- a/arch/mips/boot/dts/brcm/bcm7346.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7346.dtsi @@ -118,6 +118,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7346-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7358.dtsi b/arch/mips/boot/dts/brcm/bcm7358.dtsi index 4274ff4..060805b 100644 --- a/arch/mips/boot/dts/brcm/bcm7358.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7358.dtsi @@ -112,6 +112,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7358-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7360.dtsi b/arch/mips/boot/dts/brcm/bcm7360.dtsi index 0dcc9163..bcdb09b 100644 --- a/arch/mips/boot/dts/brcm/bcm7360.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7360.dtsi @@ -112,6 +112,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7360-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7362.dtsi b/arch/mips/boot/dts/brcm/bcm7362.dtsi index 2f3f9fc..d3b1b76 100644 --- a/arch/mips/boot/dts/brcm/bcm7362.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7362.dtsi @@ -118,6 +118,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7362-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7420.dtsi b/arch/mips/boot/dts/brcm/bcm7420.dtsi index bee221b..3302a1b 100644 --- a/arch/mips/boot/dts/brcm/bcm7420.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7420.dtsi @@ -99,6 +99,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7420-sun-top-ctrl", "syscon"; reg = <0x404000 0x60c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7425.dtsi b/arch/mips/boot/dts/brcm/bcm7425.dtsi index 571f30f..15b27aa 100644 --- a/arch/mips/boot/dts/brcm/bcm7425.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7425.dtsi @@ -100,6 +100,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7425-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7435.dtsi b/arch/mips/boot/dts/brcm/bcm7435.dtsi index 614ee21..adb33e3 100644 --- a/arch/mips/boot/dts/brcm/bcm7435.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7435.dtsi @@ -114,6 +114,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7425-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { -- 2.7.0

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[alsa-devel] Applied "regmap: mmio: Convert to regmap_bus and fix accessor usage" to the asoc tree
by Mark Brown 22 Feb '16

22 Feb '16

The patch regmap: mmio: Convert to regmap_bus and fix accessor usage has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 922a9f936e40001f9b921379aab90047d5990923 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 27 Jan 2016 04:50:07 +0000 Subject: [PATCH] regmap: mmio: Convert to regmap_bus and fix accessor usage Currently regmap-mmio uses the __raw accessors to read and write from memory. This is not safe as these interact poorly with spinlocks and are not guaranteed to generate emulated instructions on at least ARM where regmap is commonly used. The APIs that are provided all provide some byte swapping so this is difficult to do with the current regmap-mmio implementation which attempts to use the regmap core byte swapping. We can fix this by modernising the MMIO implementation to use reg_read() and reg_write() operations which were added after the API was implemented and pass simple unsigned integers through to the bus, making use of the formatting provided by the I/O accessors using a similar pattern to that used by the core. This will be less efficient for block I/O operations since we now enable and disable any required clocks per register but it is not clear that any users of regmap-mmio actually use block I/O and there is room to optimise later. This removes support for big endian I/O on 64 bit registers since no I/O accessors are provided, no current users were found and support can be added easily once they are available. In addition make the default endianness little endian. This was the behaviour prior to 29bb45f25ff305 (regmap-mmio: Use native endianness for read/write) and is the behaviour desired by most existing users, the users have been audited and those that need native endianness converted to request it explicitly. Previously native was documented as the default but due to the byte swapping in the accessors this was not correctly implemented. Fixes: 29bb45f25ff305 (regmap-mmio: Use native endianness for read/write) Reported-by: Johannes Berg <johannes(a)sipsolutions.net> Tested-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regmap-mmio.c | 259 ++++++++++++++++++++------------------ 1 file changed, 139 insertions(+), 120 deletions(-) diff --git a/drivers/base/regmap/regmap-mmio.c b/drivers/base/regmap/regmap-mmio.c index 8812bfb..7526906 100644 --- a/drivers/base/regmap/regmap-mmio.c +++ b/drivers/base/regmap/regmap-mmio.c @@ -25,26 +25,14 @@ struct regmap_mmio_context { void __iomem *regs; - unsigned reg_bytes; unsigned val_bytes; - unsigned pad_bytes; struct clk *clk; -}; -static inline void regmap_mmio_regsize_check(size_t reg_size) -{ - switch (reg_size) { - case 1: - case 2: - case 4: -#ifdef CONFIG_64BIT - case 8: -#endif - break; - default: - BUG(); - } -} + void (*reg_write)(struct regmap_mmio_context *ctx, + unsigned int reg, unsigned int val); + unsigned int (*reg_read)(struct regmap_mmio_context *ctx, + unsigned int reg); +}; static int regmap_mmio_regbits_check(size_t reg_bits) { @@ -88,72 +76,62 @@ static int regmap_mmio_get_min_stride(size_t val_bits) return min_stride; } -static inline void regmap_mmio_count_check(size_t count, u32 offset) +static void regmap_mmio_write8(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) +{ + writeb(val, ctx->regs + reg); +} + +static void regmap_mmio_write16le(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) { - BUG_ON(count <= offset); + writew(val, ctx->regs + reg); } -static inline unsigned int -regmap_mmio_get_offset(const void *reg, size_t reg_size) +static void regmap_mmio_write16be(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) { - switch (reg_size) { - case 1: - return *(u8 *)reg; - case 2: - return *(u16 *)reg; - case 4: - return *(u32 *)reg; + iowrite16be(val, ctx->regs + reg); +} + +static void regmap_mmio_write32le(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) +{ + writel(val, ctx->regs + reg); +} + +static void regmap_mmio_write32be(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) +{ + iowrite32be(val, ctx->regs + reg); +} + #ifdef CONFIG_64BIT - case 8: - return *(u64 *)reg; -#endif - default: - BUG(); - } +static void regmap_mmio_write64le(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) +{ + writeq(val, ctx->regs + reg); } +#endif -static int regmap_mmio_gather_write(void *context, - const void *reg, size_t reg_size, - const void *val, size_t val_size) +static int regmap_mmio_write(void *context, unsigned int reg, unsigned int val) { struct regmap_mmio_context *ctx = context; - unsigned int offset; int ret; - regmap_mmio_regsize_check(reg_size); - if (!IS_ERR(ctx->clk)) { ret = clk_enable(ctx->clk); if (ret < 0) return ret; } - offset = regmap_mmio_get_offset(reg, reg_size); - - while (val_size) { - switch (ctx->val_bytes) { - case 1: - __raw_writeb(*(u8 *)val, ctx->regs + offset); - break; - case 2: - __raw_writew(*(u16 *)val, ctx->regs + offset); - break; - case 4: - __raw_writel(*(u32 *)val, ctx->regs + offset); - break; -#ifdef CONFIG_64BIT - case 8: - __raw_writeq(*(u64 *)val, ctx->regs + offset); - break; -#endif - default: - /* Should be caught by regmap_mmio_check_config */ - BUG(); - } - val_size -= ctx->val_bytes; - val += ctx->val_bytes; - offset += ctx->val_bytes; - } + ctx->reg_write(ctx, reg, val); if (!IS_ERR(ctx->clk)) clk_disable(ctx->clk); @@ -161,59 +139,56 @@ static int regmap_mmio_gather_write(void *context, return 0; } -static int regmap_mmio_write(void *context, const void *data, size_t count) +static unsigned int regmap_mmio_read8(struct regmap_mmio_context *ctx, + unsigned int reg) { - struct regmap_mmio_context *ctx = context; - unsigned int offset = ctx->reg_bytes + ctx->pad_bytes; + return readb(ctx->regs + reg); +} + +static unsigned int regmap_mmio_read16le(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return readw(ctx->regs + reg); +} + +static unsigned int regmap_mmio_read16be(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return ioread16be(ctx->regs + reg); +} + +static unsigned int regmap_mmio_read32le(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return readl(ctx->regs + reg); +} - regmap_mmio_count_check(count, offset); +static unsigned int regmap_mmio_read32be(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return ioread32be(ctx->regs + reg); +} - return regmap_mmio_gather_write(context, data, ctx->reg_bytes, - data + offset, count - offset); +#ifdef CONFIG_64BIT +static unsigned int regmap_mmio_read64le(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return readq(ctx->regs + reg); } +#endif -static int regmap_mmio_read(void *context, - const void *reg, size_t reg_size, - void *val, size_t val_size) +static int regmap_mmio_read(void *context, unsigned int reg, unsigned int *val) { struct regmap_mmio_context *ctx = context; - unsigned int offset; int ret; - regmap_mmio_regsize_check(reg_size); - if (!IS_ERR(ctx->clk)) { ret = clk_enable(ctx->clk); if (ret < 0) return ret; } - offset = regmap_mmio_get_offset(reg, reg_size); - - while (val_size) { - switch (ctx->val_bytes) { - case 1: - *(u8 *)val = __raw_readb(ctx->regs + offset); - break; - case 2: - *(u16 *)val = __raw_readw(ctx->regs + offset); - break; - case 4: - *(u32 *)val = __raw_readl(ctx->regs + offset); - break; -#ifdef CONFIG_64BIT - case 8: - *(u64 *)val = __raw_readq(ctx->regs + offset); - break; -#endif - default: - /* Should be caught by regmap_mmio_check_config */ - BUG(); - } - val_size -= ctx->val_bytes; - val += ctx->val_bytes; - offset += ctx->val_bytes; - } + *val = ctx->reg_read(ctx, reg); if (!IS_ERR(ctx->clk)) clk_disable(ctx->clk); @@ -232,14 +207,11 @@ static void regmap_mmio_free_context(void *context) kfree(context); } -static struct regmap_bus regmap_mmio = { +static const struct regmap_bus regmap_mmio = { .fast_io = true, - .write = regmap_mmio_write, - .gather_write = regmap_mmio_gather_write, - .read = regmap_mmio_read, + .reg_write = regmap_mmio_write, + .reg_read = regmap_mmio_read, .free_context = regmap_mmio_free_context, - .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, - .val_format_endian_default = REGMAP_ENDIAN_NATIVE, }; static struct regmap_mmio_context *regmap_mmio_gen_context(struct device *dev, @@ -265,24 +237,71 @@ static struct regmap_mmio_context *regmap_mmio_gen_context(struct device *dev, if (config->reg_stride < min_stride) return ERR_PTR(-EINVAL); - switch (config->reg_format_endian) { - case REGMAP_ENDIAN_DEFAULT: - case REGMAP_ENDIAN_NATIVE: - break; - default: - return ERR_PTR(-EINVAL); - } - ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return ERR_PTR(-ENOMEM); ctx->regs = regs; ctx->val_bytes = config->val_bits / 8; - ctx->reg_bytes = config->reg_bits / 8; - ctx->pad_bytes = config->pad_bits / 8; ctx->clk = ERR_PTR(-ENODEV); + switch (config->reg_format_endian) { + case REGMAP_ENDIAN_DEFAULT: + case REGMAP_ENDIAN_LITTLE: +#ifdef __LITTLE_ENDIAN + case REGMAP_ENDIAN_NATIVE: +#endif + switch (config->val_bits) { + case 8: + ctx->reg_read = regmap_mmio_read8; + ctx->reg_write = regmap_mmio_write8; + break; + case 16: + ctx->reg_read = regmap_mmio_read16le; + ctx->reg_write = regmap_mmio_write16le; + break; + case 32: + ctx->reg_read = regmap_mmio_read32le; + ctx->reg_write = regmap_mmio_write32le; + break; +#ifdef CONFIG_64BIT + case 64: + ctx->reg_read = regmap_mmio_read64le; + ctx->reg_write = regmap_mmio_write64le; + break; +#endif + default: + ret = -EINVAL; + goto err_free; + } + break; + case REGMAP_ENDIAN_BIG: +#ifdef __BIG_ENDIAN + case REGMAP_ENDIAN_NATIVE: +#endif + switch (config->val_bits) { + case 8: + ctx->reg_read = regmap_mmio_read8; + ctx->reg_write = regmap_mmio_write8; + break; + case 16: + ctx->reg_read = regmap_mmio_read16be; + ctx->reg_write = regmap_mmio_write16be; + break; + case 32: + ctx->reg_read = regmap_mmio_read32be; + ctx->reg_write = regmap_mmio_write32be; + break; + default: + ret = -EINVAL; + goto err_free; + } + break; + default: + ret = -EINVAL; + goto err_free; + } + if (clk_id == NULL) return ctx; -- 2.7.0

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[alsa-devel] [PATCH 1/2] regmap: Return an error if a caller attempts to do an unsupported raw read
by Mark Brown 22 Feb '16

22 Feb '16

regmaps without raw I/O access can't implement raw I/O operations, return an error if someone tries to do that rather than crashing. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regmap.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 343263449aff..e2f68807d970 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -2255,6 +2255,9 @@ static int _regmap_raw_read(struct regmap *map, unsigned int reg, void *val, WARN_ON(!map->bus); + if (!map->bus || !map->bus->read) + return -EINVAL; + range = _regmap_range_lookup(map, reg); if (range) { ret = _regmap_select_page(map, &reg, range, -- 2.7.0.rc3

3 7

[alsa-devel] Applied "regmap: cache: Fall back to register by register read for cache defaults" to the asoc tree
by Mark Brown 22 Feb '16

22 Feb '16

The patch regmap: cache: Fall back to register by register read for cache defaults has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3245d460a1eb55b5c3ca31dde7b5c5ac71546edf Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Tue, 2 Feb 2016 10:16:51 -0200 Subject: [PATCH] regmap: cache: Fall back to register by register read for cache defaults If we are unable to read the cache defaults for a regmap then fall back on attempting to read them word by word. This is going to be painfully slow for large regmaps but might be adequate for smaller ones. Signed-off-by: Mark Brown <broonie(a)kernel.org> [maciej: Use cache_bypass around read and skipping of unreadable regs] Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> Tested-by: Fabio Estevam <fabio.estevam(a)nxp.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regcache.c | 41 ++++++++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 11 deletions(-) diff --git a/drivers/base/regmap/regcache.c b/drivers/base/regmap/regcache.c index 348be3a..cccceb5 100644 --- a/drivers/base/regmap/regcache.c +++ b/drivers/base/regmap/regcache.c @@ -30,7 +30,7 @@ static int regcache_hw_init(struct regmap *map) int i, j; int ret; int count; - unsigned int val; + unsigned int reg, val; void *tmp_buf; if (!map->num_reg_defaults_raw) @@ -67,27 +67,46 @@ static int regcache_hw_init(struct regmap *map) ret = regmap_raw_read(map, 0, tmp_buf, map->num_reg_defaults_raw); map->cache_bypass = cache_bypass; - if (ret < 0) - goto err_cache_free; - - map->reg_defaults_raw = tmp_buf; - map->cache_free = 1; + if (ret == 0) { + map->reg_defaults_raw = tmp_buf; + map->cache_free = 1; + } else { + kfree(tmp_buf); + } } /* fill the reg_defaults */ for (i = 0, j = 0; i < map->num_reg_defaults_raw; i++) { - if (regmap_volatile(map, i * map->reg_stride)) + reg = i * map->reg_stride; + + if (!regmap_readable(map, reg)) continue; - val = regcache_get_val(map, map->reg_defaults_raw, i); - map->reg_defaults[j].reg = i * map->reg_stride; + + if (regmap_volatile(map, reg)) + continue; + + if (map->reg_defaults_raw) { + val = regcache_get_val(map, map->reg_defaults_raw, i); + } else { + bool cache_bypass = map->cache_bypass; + + map->cache_bypass = true; + ret = regmap_read(map, reg, &val); + map->cache_bypass = cache_bypass; + if (ret != 0) { + dev_err(map->dev, "Failed to read %d: %d\n", + reg, ret); + goto err_free; + } + } + + map->reg_defaults[j].reg = reg; map->reg_defaults[j].def = val; j++; } return 0; -err_cache_free: - kfree(tmp_buf); err_free: kfree(map->reg_defaults); -- 2.7.0

1 0

[alsa-devel] [PATCH 3/3] ASoC: fsl_ssi: remove register defaults
by Maciej S. Szmigiero 22 Feb '16

22 Feb '16

There is no guarantee that on fsl_ssi module load SSI registers will have their power-on-reset values. In fact, if the driver is reloaded the values in registers will be whatever they were set to previously. This fixes hard lockup on fsl_ssi module reload, at least in AC'97 mode. Fixes: 05cf237972fe ("ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast") Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 40dfd8a36484..f6ef7d64acb3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,20 +112,6 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; -static const struct reg_default fsl_ssi_reg_defaults[] = { - {CCSR_SSI_SCR, 0x00000000}, - {CCSR_SSI_SIER, 0x00003003}, - {CCSR_SSI_STCR, 0x00000200}, - {CCSR_SSI_SRCR, 0x00000200}, - {CCSR_SSI_STCCR, 0x00040000}, - {CCSR_SSI_SRCCR, 0x00040000}, - {CCSR_SSI_SACNT, 0x00000000}, - {CCSR_SSI_STMSK, 0x00000000}, - {CCSR_SSI_SRMSK, 0x00000000}, - {CCSR_SSI_SACCEN, 0x00000000}, - {CCSR_SSI_SACCDIS, 0x00000000}, -}; - static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -190,8 +176,6 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .reg_defaults = fsl_ssi_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg,

4 27

[alsa-devel] [PATCH v2] ASoC: wm9713: fix regmap free path
by Robert Jarzmik 22 Feb '16

22 Feb '16

In the conversion to regmap, I assumed that the 2 following functions was working symetrically: - snd_soc_codec_init_regmap() - snd_soc_codec_exit_regmap(codec) As a mater of fact with the current code the regmap is freed twice because of the devm_() call: (mutex_lock) from (debugfs_remove_recursive+0x50/0x1d0) (debugfs_remove_recursive) from (regmap_debugfs_exit+0x1c/0xd4) (regmap_debugfs_exit) from (regmap_exit+0x28/0xc8) (regmap_exit) from (release_nodes+0x18c/0x204) (release_nodes) from (device_release+0x18/0x90) (device_release) from (kobject_release+0x90/0x1bc) (kobject_release) from (wm9713_soc_remove+0x1c/0x24) (wm9713_soc_remove) from (soc_remove_component+0x50/0x7c) (soc_remove_component) from (soc_remove_dai_links+0x118/0x228) (soc_remove_dai_links) from (snd_soc_register_card+0x4e4/0xdd4) (snd_soc_register_card) from (devm_snd_soc_register_card+0x34/0x70) Fix this by removing the doubled regmap free. Fixes: 700dadfefc3d ASoC: wm9713: convert to regmap Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: fix the diff second hunk, god knows where it came from. --- sound/soc/codecs/wm9713.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 79e143625ac3..dd31a5207527 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1230,7 +1230,6 @@ static int wm9713_soc_remove(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9713->ac97); return 0; } -- 2.1.4

3 3

[alsa-devel] [alsa-lib][PATCH] control: initialize element channel values according to identification number at adding a new element
by Takashi Sakamoto 21 Feb '16

21 Feb '16

In ALSA control core, there're two ways to identify elements. One is by numid. Another is by name/index. The former way is lighter than the latter way. In control API, after adding elements by snd_ctl_elem_add_integer() and snd_ctl_elem_add_integer64(), these functions initialize value of each channel. When executing this operation, used ID information is still given by applications. Although, in this time, the element can be identified according to numid. This commit uses the numid instead of name/index pair. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index a44ae96..8a5d530 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -292,7 +292,7 @@ int snd_ctl_elem_add_integer(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, if (err < 0) return err; snd_ctl_elem_value_alloca(&val); - val->id = *id; + val->id = info->id; for (i = 0; i < count; i++) val->value.integer.value[i] = min; err = ctl->ops->element_write(ctl, val); @@ -330,7 +330,7 @@ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, if (err < 0) return err; snd_ctl_elem_value_alloca(&val); - val->id = *id; + val->id = info->id; for (i = 0; i < count; i++) val->value.integer64.value[i] = min; err = ctl->ops->element_write(ctl, val); -- 2.5.0

2 1

[alsa-devel] [alsa-lib][PATCH] control: fulfill a description about element type with integer64
by Takashi Sakamoto 21 Feb '16

21 Feb '16

In UAPI of asound.h, there're 6 types of control element; boolean, integer, enumerated, bytes, iec958 and integer64 (The none type seems to be added to start the value at non-zero). Although, current description in alsa-lib has a lack of integer64 type. This commit fulfil the type. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control.c b/src/control/control.c index 70d168d..a44ae96 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -35,7 +35,7 @@ also interface notifying about control and structure changes. \section control_general_overview General overview -The primitive controls can be integer, boolean, enumerators, bytes +The primitive controls can be integer, inter64, boolean, enumerators, bytes and IEC958 structure. */ -- 2.5.0

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm_adsp: Avoid narrow race condition on compr pointer
by Charles Keepax 20 Feb '16

20 Feb '16

There is a very small window between then wm_adsp_compr_free gets call and when the DSP is actually powered down. If we get an IRQ from the DSP in this window then the wm_adsp_compr pointer will be NULL. This patch adds a check for this into the IRQ handler to avoid any issues when this happens. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ed90e12..00a7d68 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2857,7 +2857,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) goto out; } - if (compr->stream) + if (compr && compr->stream) snd_compr_fragment_elapsed(compr->stream); out: -- 2.1.4

2 7

[alsa-devel] [PATCH] ASoC: max98926: Constify max98926_reg and max98926_regmap
by Axel Lin 20 Feb '16

20 Feb '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98926.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 5245e10..8d14ada 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -40,7 +40,7 @@ static const char *const max98926_hpf_cutoff_txt[] = { "200Hz", "400Hz", "800Hz", }; -static struct reg_default max98926_reg[] = { +static const struct reg_default max98926_reg[] = { { 0x0B, 0x00 }, /* IRQ Enable0 */ { 0x0C, 0x00 }, /* IRQ Enable1 */ { 0x0D, 0x00 }, /* IRQ Enable2 */ @@ -506,7 +506,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98926 = { .num_dapm_widgets = ARRAY_SIZE(max98926_dapm_widgets), }; -static struct regmap_config max98926_regmap = { +static const struct regmap_config max98926_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = MAX98926_VERSION, -- 2.1.4

3 2

[alsa-devel] [PATCH 1/3] ASoC: max9867: Add CONFIG_SND_SOC_MAX9867 to build
by Axel Lin 20 Feb '16

20 Feb '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Makefile | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 621c0a3..185a712 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -283,6 +283,7 @@ obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o +obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o -- 2.1.4

3 6

[alsa-devel] [PATCH] ASoC: wm9713: fix regmap free path
by Robert Jarzmik 20 Feb '16

20 Feb '16

In the conversion to regmap, I assumed that the 2 following functions was working symetrically: - snd_soc_codec_init_regmap() - snd_soc_codec_exit_regmap(codec) As a mater of fact with the current code the regmap is freed twice because of the devm_() call: (mutex_lock) from (debugfs_remove_recursive+0x50/0x1d0) (debugfs_remove_recursive) from (regmap_debugfs_exit+0x1c/0xd4) (regmap_debugfs_exit) from (regmap_exit+0x28/0xc8) (regmap_exit) from (release_nodes+0x18c/0x204) (release_nodes) from (device_release+0x18/0x90) (device_release) from (kobject_release+0x90/0x1bc) (kobject_release) from (wm9713_soc_remove+0x1c/0x24) (wm9713_soc_remove) from (soc_remove_component+0x50/0x7c) (soc_remove_component) from (soc_remove_dai_links+0x118/0x228) (soc_remove_dai_links) from (snd_soc_register_card+0x4e4/0xdd4) (snd_soc_register_card) from (devm_snd_soc_register_card+0x34/0x70) Fix this by removing the doubled regmap free. Fixes: 700dadfefc3d ASoC: wm9713: convert to regmap Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/codecs/wm9713.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 79e143625ac3..0242b7dcc1bb 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1230,7 +1230,6 @@ static int wm9713_soc_remove(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9713->ac97); return 0; } @@ -1272,3 +1271,2 @@ static int wm9713_remove(struct platform_device *pdev) return 0; } -- 2.1.4

2 1

[alsa-devel] [PATCH 0/11] ASoC: rsnd: cleanup SSI multi mode and sequence
by Kuninori Morimoto 20 Feb '16

20 Feb '16

Hi Mark Current rsnd driver is supporting many features, and now we are confirming these register settings sequence. These patches fixup not good sequence (and bug) Kuninori Morimoto (11): ASoC: rsnd: rename RSND_DVC_CHANNELS to RSND_MAX_CHANNELS ASoC: rsnd: setup SRC_ROUTE_MODE0 before SRC_SRCIR ASoC: rsnd: indicates status failed SSI ASoC: rsnd: SSI function parameter uses "mod" instead of "ssi" ASoC: rsnd: fixup forever loop bug on SSI ASoC: rsnd: add missing .irq callback for DMA ASoC: rsnd: tidyup SSI init/start sequence ASoC: rsnd: Parent SSI attach is not needed if not clock master ASoC: rsnd: move rsnd_ssi_irq() position ASoC: rsnd: judge multi SSI in runtime ASoC: rsnd: judge work SSI in runtime sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/sh/rcar/dvc.c | 6 ++--- sound/soc/sh/rcar/rsnd.h | 6 ++--- sound/soc/sh/rcar/src.c | 3 ++- sound/soc/sh/rcar/ssi.c | 166 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------------------------------- sound/soc/sh/rcar/ssiu.c | 6 ++--- 6 files changed, 126 insertions(+), 65 deletions(-)

2 22

[alsa-devel] [PATCH v5 00/14] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 20 Feb '16

20 Feb '16

This patch series adds DP audio and hotplug notification support. In skylake platform the DP is on a different port which is not enabled by default. A vendor widget is programmed to enable playback on all ports. This also enables all the converters. Codec Dais, widgets, graph are now created dynamically based on the nodes enumerated, thus removing the hardcoding. Mux controls are used to establish routing stream to a specific port. For DP audio support, infoframe needs to be different. Based on ELD queried we either pack HDMI or DP infoframe. Also with this series, Jack notification support is added and Jack event is reported to userspace as is done in case of legacy driver. Few fixes are added to check for possible memory leak, reconfiguring register during resume from D3 etc. The 8th patch is adding a small macro for getting connection type in drm header. It is reviewed by drm folks, CCed them. Please merge it through sound trees. Changes in v5: - Added more comments and reworded some commit messages - Jack reporting support to user space added - Fix to check for possible memory leak in hw_params - Fix not to fail in dai startup to make userland APIs happy Cc: Jani Nikula <jani.nikula(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Daniel Vetter <daniel.vetter(a)intel.com> Jeeja KP (2): ASoC: hdac_hdmi: Add jack reporting ASoC: hdac_hdmi: Add PM support Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Subhransu S. Prusty (11): ASoC: hdac_hdmi: Add hotplug notification and read eld ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi: create dais based on number of cvts ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Enable playback on all enumerated ports drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume include/drm/drm_edid.h | 12 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1197 +++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/hdac_hdmi.h | 6 + 4 files changed, 1094 insertions(+), 122 deletions(-) create mode 100644 sound/soc/codecs/hdac_hdmi.h -- 1.9.1

4 25

[alsa-devel] [PATCH v1 00/14] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. Most of these patches are acked by Kenneth. These patches are tested on DB410C with Headset Mic, secondary mic. Thanks, srini Changes since RFC: - made patches bisectable, reported by 0day testing - merged two patches as suggested by Kenneth - used corect device for snd_dma alloc/free apis, spotted by Kenneth - renamed LPAIF_I2SCTL_MICMODE_QUAD02 to LPAIF_I2SCTL_MICMODE_QUAD23, spotted by Kenneth. - Fixed minor issues suggested by Kenneth. Srinivas Kandagatla (14): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma register definitions ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: add mic support ASoC: qcom: apq8016-sbc: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 +++++++++++++++++------ sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 116 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 243 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 390 insertions(+), 177 deletions(-) -- 1.9.1

3 26

[alsa-devel] [PATCH v3 00/14] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. All these patches are acked by Kenneth. These patches are tested on DB410C with Headset Mic, secondary mic. Thanks, srini Changes since v1/v2: - Fixed patch version number. - renamed rdma_port to dma_port in patch 3, as suggested by Kenneth. - fixed correct device for snd_dma_alloc/free as suggested by Kenneth in patch 13. Changes since RFC: - made patches bisectable, reported by 0day testing - merged two patches as suggested by Kenneth - used corect device for snd_dma alloc/free apis, spotted by Kenneth - renamed LPAIF_I2SCTL_MICMODE_QUAD02 to LPAIF_I2SCTL_MICMODE_QUAD23, spotted by Kenneth. - Fixed minor issues suggested by Kenneth. Srinivas Kandagatla (14): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma register definitions ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: add mic support ASoC: qcom: apq8016-sbc: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 ++++++++++++++++++----- sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 116 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 244 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 391 insertions(+), 177 deletions(-) -- 1.9.1

2 17

[alsa-devel] [PATCH RFC 00/15] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. These patches are tested on DB410C with Headset Mic. Thanks, srini Srinivas Kandagatla (15): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma dma channel start ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add wrdma register definations ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: apq8016-sbc: add mic support ASoC: qcom: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 +++++++++++++++++------ sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 118 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 243 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 392 insertions(+), 177 deletions(-) -- 1.9.1

3 45

[alsa-devel] [PATCH] ASoC: cht_bsw_rt5645: Enable jack detection
by Carlo Caione 20 Feb '16

20 Feb '16

From: Carlo Caione <carlo(a)endlessm.com> Add missing DAPM pins and enable jack detection on those pins for Cherrytrail and Braswell. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index e6cf800..c7af271 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -147,6 +147,17 @@ static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Ext Spk"), }; +static struct snd_soc_jack_pin cht_bsw_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -202,9 +213,9 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) else jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; - ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", + ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, &ctx->jack, - NULL, 0); + cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); if (ret) { dev_err(runtime->dev, "Headset jack creation failed %d\n", ret); return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH v4 0/2] ASoC: Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 20 Feb '16

20 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver. For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK --- Changes since v3: - removed code required for spdif capture - added disable/enable of apb clk to suspend/resume. - removed unnecessary dt checks in probe - fixed error conditions in probe/resume - modified pm_runtime functionality(confirmed driver worked with PM disabled). - removed .owner from platform driver - renamed bindings file Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): dt-bindings: sunxi :add sun4i SPDIF transceiver ASoC: sunxi: Add support for the SPDIF block .../bindings/sound/sunxi,sun4i-spdif.txt | 39 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 561 +++++++++++++++++++++ 4 files changed, 609 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.0

4 5

[alsa-devel] [PATCH] SoC: cht_bsw_rt5645: Fix writing to string literal
by Carlo Caione 20 Feb '16

20 Feb '16

From: Carlo Caione <carlo(a)endlessm.com> We cannot use strcpy() to write to a const char * location. This is causing a 'BUG: unable to handle kernel paging request' error at boot when using the cht-bsw-rt5645 driver. With this patch we also fix a wrong indexing in the driver where the codec_name of the wrong dai_link is being overwritten. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/atom/sst-atom-controls.h | 1 + sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index e011311..6f88d1c 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -30,6 +30,7 @@ enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_DEEP_BUFFER, MERR_DPCM_COMPR, + MERR_DPCM_BE, }; /* define a bit for each mixer input */ diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index e6cf800..89b4c032 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -282,7 +282,7 @@ static struct snd_soc_dai_link cht_dailink[] = { }, /* CODEC<->CODEC link */ /* back ends */ - { + [MERR_DPCM_BE] = { .name = "SSP2-Codec", .be_id = 1, .cpu_dai_name = "ssp2-port", @@ -357,7 +357,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + card->dai_link[MERR_DPCM_BE].codec_name = kstrdup(codec_name, GFP_KERNEL); snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 1.9.1

3 3

[alsa-devel] [PATCH] fireworks: serialize transactions to update connections at bus reset
by Takashi Sakamoto 20 Feb '16

20 Feb '16

In IEC 61883-1, at bus-reset, applications can continue isochronous streaming by updating connections. In ALSA fireworks driver, the operation is executed in 'update' handler for bus driver. The connection resources are also changed in process contexts of PCM/MIDI applications. Therefore, bus-reset handling has race condition against connection. Current ALSA fireworks driver has a bug for the condition. This commit fixes the bug, by expand critical section with mutex. As a result, connection updating operation in bus-reset handler and connection changing operation in process context are serialized. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 3 +++ sound/firewire/fireworks/fireworks_stream.c | 6 ++---- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index d5b19bc..8f27b67 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -301,7 +301,10 @@ static void efw_update(struct fw_unit *unit) struct snd_efw *efw = dev_get_drvdata(&unit->device); snd_efw_transaction_bus_reset(efw->unit); + + mutex_lock(&efw->mutex); snd_efw_stream_update_duplex(efw); + mutex_unlock(&efw->mutex); } static void efw_remove(struct fw_unit *unit) diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 968a40a..425db8d 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -313,12 +313,10 @@ void snd_efw_stream_stop_duplex(struct snd_efw *efw) void snd_efw_stream_update_duplex(struct snd_efw *efw) { - if ((cmp_connection_update(&efw->out_conn) < 0) || - (cmp_connection_update(&efw->in_conn) < 0)) { - mutex_lock(&efw->mutex); + if (cmp_connection_update(&efw->out_conn) < 0 || + cmp_connection_update(&efw->in_conn) < 0) { stop_stream(efw, &efw->rx_stream); stop_stream(efw, &efw->tx_stream); - mutex_unlock(&efw->mutex); } else { amdtp_stream_update(&efw->rx_stream); amdtp_stream_update(&efw->tx_stream); -- 2.5.0

2 2

[alsa-devel] [PATCH 0/4] ALSA: bebob: give up updating streams at bus reset
by Takashi Sakamoto 20 Feb '16

20 Feb '16

Hi, This patchset is to simplify codes of ALSA bebob driver by giving up updating streams at bus reset. DM1000/DM1100/DM1500 chipsets have a quirk to transfer packets with discontinuous value in 'dbc' field of CIP header. This is against packet streaming protocol (IEC 61883-1/6). In the specification, units should continue packet streaming during 1 seconds after bus reset. During this period, connections are updated. Else, the streaming stops. The discontinuity causes firewire-lib module (packet streaming layer) to stop streaming immediately. This is done far earlier than executing bus-reset handler. Therefore, updating streams at the bus-reset handler has no meaning. This patchset will have little impacts for userspace applications because most users won't update IEEE 1394 bus when playbacking/recording. And PCM substream recovery is still available. Regards Takashi Sakamoto (4): ALSA: bebob: simplify bus-reset handling ALSA: bebob: move mutex from function callee to callers ALSA: bebob: change type of substream counter from atomic_t to unsigned int ALSA: bebob: give up updating streams at bus reset handler sound/firewire/bebob/bebob.c | 21 ++++++++++---- sound/firewire/bebob/bebob.h | 5 +--- sound/firewire/bebob/bebob_midi.c | 16 +++++++--- sound/firewire/bebob/bebob_pcm.c | 28 +++++++++++++----- sound/firewire/bebob/bebob_stream.c | 58 +++---------------------------------- 5 files changed, 53 insertions(+), 75 deletions(-) -- 2.5.0

2 5

Re: [alsa-devel] Issues with ALSA:ice1724 sound cards on powerpc machines
by Julian Margetson 20 Feb '16

20 Feb '16

On 2/17/2016 2:29 PM, Julian Margetson wrote: > Hi > > Having an issue issue with ice1724 sound cards on powerpc motherboards. > Motherboards in question Acube Sam460ex . > The card appears as a valid device but there is no sound output on the > majority of boots of the computer. > Logs attached. > On the first set boot audio playback produced sound ok. > On the second set boot no sound was produce with audio playback > > > Regards > Julian Margetson > > > Issue also with A-Eon Cyrus powerpc motherboard .

1 0

[alsa-devel] [PATCH v6 0/2] ASoC: Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 19 Feb '16

19 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver. For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK --- Changes since v5: - fixed warning that had sneaked in.' Changes since v4: - corrected indentation issues - removed old macros - removed unnecessary enabling of apb clock Changes since v3: - removed code required for spdif capture - added disable/enable of apb clk to suspend/resume. - removed unnecessary dt checks in probe - fixed error conditions in probe/resume - modified pm_runtime functionality and confirmed that driver worked when PM was disabled. - removed .owner from platform driver - renamed bindings file Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): dt-bindings: sunxi :add sun4i SPDIF transceiver ASoC: sunxi: Add support for the SPDIF block .../bindings/sound/sunxi,sun4i-spdif.txt | 39 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 550 +++++++++++++++++++++ 4 files changed, 598 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.1

3 6

[alsa-devel] [PATCH 0/2] ALSA: oxfw: implement retries for scs1x at transaction failure
by Takashi Sakamoto 19 Feb '16

19 Feb '16

Hi, Current ALSA oxfw driver has a TODO to retry IEEE 1394 asynchronous transactions when transaction failure occurs. This patchset is for it. Additionally, this patchset classifies reason of the failure and stop the retries to prevent from endless loop of scheduling work. Regards Takashi Sakamoto (2): ALSA: oxfw: retry MIDI transferring for scs1x at transaction failure ALSA: oxfw: discontinue MIDI substream for scs1x at transaction failure sound/firewire/oxfw/oxfw-scs1x.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) -- 2.5.0

2 11

[alsa-devel] [PATCH v7 0/9] ASoC: hdac_hdmi: Update to HDMI driver
by Subhransu S. Prusty 19 Feb '16

19 Feb '16

This is remaining patches of v6 and appended platform changes for enumerating HDMI codec Jeeja KP (1): ASoC: hdac_hdmi: Add PM support Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Subhransu S. Prusty (6): ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec ASoC: Intel: Skylake: Fix possible memory corruption in codec dai dma params ASoC: Intel: Skylake: Add HDMI FE and BE cpu dais Vinod Koul (1): ASoC: Intel: Skylake: Add i915 enabling in skl probe sound/soc/codecs/hdac_hdmi.c | 178 +++++++++++++++++++++++++++++++------- sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/skl-pcm.c | 84 ++++++++++++++++-- sound/soc/intel/skylake/skl.c | 51 +++++++++++ 4 files changed, 275 insertions(+), 39 deletions(-) -- 1.9.1

2 16

[alsa-devel] [PATCH 0/4] Add I2S/ADV7511 audio support for ARC AXS10x boards.
by Carlos Palminha 19 Feb '16

19 Feb '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Jose Abreu (4): drm/i2c/adv7511: Add audio support ARC: axs10x: Update defconfigs so that audio is enabled ASoC: dwc: Add I2S HDMI audio support using custom platform ARC: axs10x: Update defconfigs so that I2S is enabled arch/arc/boot/dts/axs10x_mb.dtsi | 20 +- arch/arc/configs/axs101_defconfig | 4 + arch/arc/configs/axs103_defconfig | 6 + arch/arc/configs/axs103_smp_defconfig | 6 + drivers/gpu/drm/i2c/Kconfig | 8 + drivers/gpu/drm/i2c/Makefile | 2 + drivers/gpu/drm/i2c/adv7511.c | 1024 --------------------------------- drivers/gpu/drm/i2c/adv7511.h | 41 ++ drivers/gpu/drm/i2c/adv7511_audio.c | 310 ++++++++++ drivers/gpu/drm/i2c/adv7511_core.c | 1005 ++++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/Makefile | 4 +- sound/soc/dwc/designware_i2s.c | 139 ++++- sound/soc/dwc/designware_pcm.c | 264 +++++++++ sound/soc/dwc/designware_pcm.h | 21 + 16 files changed, 1813 insertions(+), 1043 deletions(-) delete mode 100644 drivers/gpu/drm/i2c/adv7511.c create mode 100644 drivers/gpu/drm/i2c/adv7511_audio.c create mode 100644 drivers/gpu/drm/i2c/adv7511_core.c create mode 100644 sound/soc/dwc/designware_pcm.c create mode 100644 sound/soc/dwc/designware_pcm.h -- 2.5.0

1 3

[alsa-devel] [RFC v2 0/6] sti: add audio interface to the hdmi driver
by Arnaud Pouliquen 19 Feb '16

19 Feb '16

V2: - patch: ALSA: pcm: add IEC958 channel status control helper - Return 1 instead of 0 in snd_pcm_iec958_put - Add .access field in control structure - I have kept condition on mutex for flexibility (but could be cleaned to force user to use a mutex) - patch: video: hdmi: add helper function for N and CTS - Duplicate function to have a separte treatment for coherent and non-coherent clcocks - Add ratio fiel for alternate CTS value - Clock frequency in Hz for TMDS and audio clocks - Add information concerning clocks and CTS calculation. V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Arnaud Pouliquen (6): video: hdmi: add helper function for N and CTS ALSA: pcm: add IEC958 channel status control helper ASoC: core: add code to complete dai init after pcm creation drm: sti: Add ASoC generic hdmi codec support. ASoc: hdmi-codec: add IEC control. ARM: DT: b2120: add audio HDMI dai link in audio card arch/arm/boot/dts/stihxxx-b2120.dtsi | 15 ++- drivers/gpu/drm/sti/Kconfig | 1 + drivers/gpu/drm/sti/sti_hdmi.c | 248 ++++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 10 ++ drivers/video/hdmi.c | 222 +++++++++++++++++++++++++++++++ include/linux/hdmi.h | 22 ++++ include/sound/hdmi-codec.h | 1 + include/sound/pcm_iec958.h | 16 +++ include/sound/soc-dai.h | 7 + sound/core/pcm_iec958.c | 99 ++++++++++++++ sound/soc/codecs/hdmi-codec.c | 59 +++++++-- sound/soc/soc-core.c | 14 ++ 12 files changed, 682 insertions(+), 32 deletions(-) -- 1.9.1

4 17

[alsa-devel] [PATCH] ALSA: hda - hdmi get jack from hda_jack_tbl when not dyn_pcm_assign
by libin.yang＠linux.intel.com 19 Feb '16

19 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> On Intel platform, if !dyn_pcm_assign, spec->pcm_rec[].jack is not NULL even after snd_hda_jack_tbl_clear() is called to free snd_jack. This may cause access invalid memory when calling snd_jack_report. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 30 ++++++++++++++++++++++++++---- 1 file changed, 26 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f4443b5..541986f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1956,6 +1956,29 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, return ret; } +static struct snd_jack *pin_idx_to_jack(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_jack *jack = NULL; + struct hda_jack_tbl *jack_tbl; + + /* if !dyn_pcm_assign, get jack from hda_jack_tbl + * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not + * NULL even after snd_hda_jack_tbl_clear() is called to + * free snd_jack. This may cause access invalid memory + * when calling snd_jack_report + */ + if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) + jack = spec->pcm_rec[per_pin->pcm_idx].jack; + else if (!spec->dyn_pcm_assign) { + jack_tbl = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + if (jack_tbl) + jack = jack_tbl->jack; + } + return jack; +} + /* update ELD and jack state via audio component */ static void sync_eld_via_acomp(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) @@ -1989,11 +2012,10 @@ static void sync_eld_via_acomp(struct hda_codec *codec, /* pcm_idx >=0 before update_eld() means it is in monitor * disconnected event. Jack must be fetched before update_eld() */ - if (per_pin->pcm_idx >= 0) - jack = spec->pcm_rec[per_pin->pcm_idx].jack; + jack = pin_idx_to_jack(codec, per_pin); update_eld(codec, per_pin, eld); - if (jack == NULL && per_pin->pcm_idx >= 0) - jack = spec->pcm_rec[per_pin->pcm_idx].jack; + if (jack == NULL) + jack = pin_idx_to_jack(codec, per_pin); if (jack == NULL) goto unlock; snd_jack_report(jack, -- 1.9.1

2 1

Re: [alsa-devel] [pulseaudio-discuss] [PATCH 0/1] alsa: Make profiles' availability dynamic
by Raymond Yau 19 Feb '16

19 Feb '16

2016-2-18 下午5:38於 "Bob Ham" <bob.ham(a)collabora.com>寫道： > > Make ALSA profiles stick around and update their availability if the > HDMI ELD changes. This alters the presence of card profiles quite > dramatically. Specifically, the existing autogeneration of profiles > generates a large set of input/output mapping combinations and all of > these combinations are reported by the API, marked as unavailable. To > give an idea of the consequences, here is what pavucontrol looks like > with this patch: > > https://people.collabora.com/~rah/pavucontrol-unavailable-profiles.png > > Similarly, here is the output of "pactl list cards": > > https://people.collabora.com/~rah/pactl-list-cards-output.txt > AFAIK, 1) mono is not supported by hda-intel 2) those desktop with three jacks at rear panel need retasking input jacks to output jacks by changing control channel mode from 2ch to 6ch, you may need to add hook for alsa devices to change the channel mode in alsa HDA-Intel.conf 3) even HDMI codec support 8 channels, it still require your hdmi monitor support multi channels

1 0

[alsa-devel] [PATCH] ASoC: samsung: Use IRQ safe spin lock calls
by Charles Keepax 19 Feb '16

19 Feb '16

Lockdep warns of a potential lock inversion, i2s->lock is held numerous times whilst we are under the substream lock (snd_pcm_stream_lock). If we use the IRQ unsafe spin lock calls, you can also end up locking snd_pcm_stream_lock whilst under i2s->lock (if an IRQ happens whilst we are holding i2s->lock). This could result in deadlock. [ 18.147001] CPU0 CPU1 [ 18.151509] ---- ---- [ 18.156022] lock(&(&pri_dai->spinlock)->rlock); [ 18.160701] local_irq_disable(); [ 18.166622] lock(&(&substream->self_group.lock)->rlock); [ 18.174595] lock(&(&pri_dai->spinlock)->rlock); [ 18.181806] <Interrupt> [ 18.184408] lock(&(&substream->self_group.lock)->rlock); [ 18.190045] [ 18.190045] *** DEADLOCK *** This patch changes to using the irq safe spinlock calls, to avoid this issue. Fixes: ce8bcdbb61d9 ("ASoC: samsung: i2s: Protect more registers with a spinlock") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/i2s.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 84d9e77..70a2559 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -481,10 +481,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, unsigned int cdcon_mask = 1 << i2s_regs->cdclkcon_off; unsigned int rsrc_mask = 1 << i2s_regs->rclksrc_off; u32 mod, mask, val = 0; + unsigned long flags; - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); switch (clk_id) { case SAMSUNG_I2S_OPCLK: @@ -575,11 +576,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); mod = (mod & ~mask) | val; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); return 0; } @@ -590,6 +591,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, struct i2s_dai *i2s = to_info(dai); int lrp_shift, sdf_shift, sdf_mask, lrp_rlow, mod_slave; u32 mod, tmp = 0; + unsigned long flags; lrp_shift = i2s->variant_regs->lrp_off; sdf_shift = i2s->variant_regs->sdf_off; @@ -649,7 +651,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); /* * Don't change the I2S mode if any controller is active on this @@ -657,7 +659,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, */ if (any_active(i2s) && ((mod & (sdf_mask | lrp_rlow | mod_slave)) != tmp)) { - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; @@ -666,7 +668,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); return 0; } @@ -676,6 +678,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, { struct i2s_dai *i2s = to_info(dai); u32 mod, mask = 0, val = 0; + unsigned long flags; if (!is_secondary(i2s)) mask |= (MOD_DC2_EN | MOD_DC1_EN); @@ -744,11 +747,11 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); mod = (mod & ~mask) | val; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); -- 2.1.4

4 3

[alsa-devel] Closing the pcm streams
by Andy Ng 19 Feb '16

19 Feb '16

Hello, I am working on a system that a DSP provides the audio algorithms and in case where the DSP crashes, I would like to shutdown nicely any opened PCM streams. The DSP has an RPC protocol to the external world. I use a small watchdog on ARM to monitor the "health" of the DSP. When the DSP goes down, I reset it. However, I would like to close down smoothly the PCM streams and inform the user that there was an error while playing. Is there any way to inform the ALSA kernel framework in the kernel that a specific PCM stream gone bad? Where is the best point to reset ALSA kernel framework? Thank you Andy

2 1

[alsa-devel] [PATCH] Changed the if-else case to switch case
by anish kumar 18 Feb '16

18 Feb '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/codecs/max9867.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index b9d989f..5a8481b 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -212,16 +212,22 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, switch (snd_pcm_format_width(params_format(params))) { case 8: case 16: - if (pclk_bclk_ratio == 2) + switch (pclk_bclk_ratio) { + case 2: bclk_value = MAX9867_IFC1B_PCLK_2; - else if (pclk_bclk_ratio == 4) + break; + case 4: bclk_value = MAX9867_IFC1B_PCLK_4; - else if (pclk_bclk_ratio == 8) + break; + case 8: bclk_value = MAX9867_IFC1B_PCLK_8; - else if (pclk_bclk_ratio == 16) + break; + case 16: bclk_value = MAX9867_IFC1B_PCLK_16; - else { - dev_err(codec->dev, "unsupported sampling rate\n"); + break; + default: + dev_err(codec->dev, + "unsupported sampling rate\n"); return -EINVAL; } break; -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: Add max9867 codec driver
by anish kumar 18 Feb '16

18 Feb '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max9867.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max9867.c | 537 +++++++++++++++++++++ sound/soc/codecs/max9867.h | 83 ++++ 5 files changed, 642 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max9867.txt create mode 100755 sound/soc/codecs/max9867.c create mode 100755 sound/soc/codecs/max9867.h diff --git a/Documentation/devicetree/bindings/sound/max9867.txt b/Documentation/devicetree/bindings/sound/max9867.txt new file mode 100644 index 0000000..394cd4e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9867.txt @@ -0,0 +1,17 @@ +max9867 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max9867" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max9867: max9867@0x18 { + compatible = "maxim,max9867"; + reg = <0x18>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c3178e3..913c83c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C @@ -490,6 +491,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX9867 + tristate + config SND_SOC_MAX98925 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 84843f3..f39dee7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -70,6 +70,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c new file mode 100755 index 0000000..b9d989f --- /dev/null +++ b/sound/soc/codecs/max9867.c @@ -0,0 +1,537 @@ +/* + * max9867.c -- max9867 ALSA SoC Audio driver + * + * Copyright 2013-15 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max9867.h" + +static const char *const max9867_spmode[] = { + "Stereo Diff", "Mono Diff", + "Stereo Cap", "Mono Cap", + "Stereo Single", "Mono Single", + "Stereo Single Fast", "Mono Single Fast" +}; +static const char *const max9867_sidetone_text[] = { + "None", "Left", "Right", "LeftRight", "LeftRightDiv2", +}; +static const char *const max9867_filter_text[] = {"IIR", "FIR"}; + +static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, + max9867_filter_text); +static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, + max9867_spmode); +static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, + max9867_sidetone_text); +static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); +static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); +static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); +static const unsigned int max98088_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const struct snd_kcontrol_new max9867_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 63, 1), + SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, + 0, 15, 1, max9860_capture_tlv), + SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), + SOC_ENUM("Digital Sidetone Src", max9867_sidetone), + SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), + SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), + SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), + SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, + 4, 15, 1, max9860_adc_left_tlv), + SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, + 0, 15, 1, max9860_adc_right_tlv), + SOC_ENUM("Speaker Mode", max9867_spkmode), + SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), + SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_ENUM("DSP Filter", max9867_filter), +}; + +static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; + +static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, + MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, + max9867_mux); + +static const struct snd_kcontrol_new max9867_dapm_mux_controls = + SOC_DAPM_ENUM("Route", max9867_mux_enum); + +static const struct snd_kcontrol_new max9867_left_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); +static const struct snd_kcontrol_new max9867_right_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); +static const struct snd_kcontrol_new max9867_line_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); + +static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HPOUT"), + + SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), + SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &max9867_dapm_mux_controls), + + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, + &max9867_left_dapm_control), + SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, + &max9867_right_dapm_control), + SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, + &max9867_line_dapm_control), + SND_SOC_DAPM_INPUT("LINE_IN"), +}; + +static const struct snd_soc_dapm_route max9867_audio_map[] = { + {"Left DAC", NULL, "DAI_OUT"}, + {"Right DAC", NULL, "DAI_OUT"}, + {"Output Mixer", NULL, "Left DAC"}, + {"Output Mixer", NULL, "Right DAC"}, + {"HPOUT", NULL, "Output Mixer"}, + + {"Left ADC", NULL, "DAI_IN"}, + {"Right ADC", NULL, "DAI_IN"}, + {"Input Mixer", NULL, "Left ADC"}, + {"Input Mixer", NULL, "Right ADC"}, + {"Input Mux", "Line", "Input Mixer"}, + {"Input Mux", "Mic", "Input Mixer"}, + {"Input Mux", "Mic_Line", "Input Mixer"}, + {"Right Line", "Switch", "Input Mux"}, + {"Left Line", "Switch", "Input Mux"}, + {"LINE_IN", NULL, "Left Line"}, + {"LINE_IN", NULL, "Right Line"}, +}; + +enum rates { + pcm_rate_8, pcm_rate_16, pcm_rate_24, + pcm_rate_32, pcm_rate_44, + pcm_rate_48, max_pcm_rate, +}; + +struct ni_div_rates { + u32 mclk; + u16 ni[max_pcm_rate]; +} ni_div[] = { + {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, + {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, + {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, + {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, + {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, + {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, + {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, + {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +}; + +static inline int get_ni_value(int mclk, int rate) +{ + int i, ret = 0; + + /* find the closest rate index*/ + for (i = 0; i < ARRAY_SIZE(ni_div); i++) { + if (ni_div[i].mclk >= mclk) + break; + } + + switch (rate) { + case 8000: + return ni_div[i].ni[pcm_rate_8]; + case 16000: + return ni_div[i].ni[pcm_rate_16]; + case 32000: + return ni_div[i].ni[pcm_rate_32]; + case 44100: + return ni_div[i].ni[pcm_rate_44]; + case 48000: + return ni_div[i].ni[pcm_rate_48]; + default: + pr_err("%s wrong rate %d\n", __func__, rate); + ret = -EINVAL; + } + return ret; +} + +static int max9867_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + unsigned int ni_h, ni_l, value; + + value = get_ni_value(max9867->sysclk, params_rate(params)); + if (value < 0) + return value; + + ni_h = (0xFF00 & value) >> 8; + ni_l = 0x00FF & value; + /* set up the ni value */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_NI_HIGH_MASK, ni_h); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_NI_LOW_MASK, ni_l); + if (!max9867->master) { + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); + } else { + unsigned long int bclk_rate, pclk_bclk_ratio; + int bclk_value; + + bclk_rate = params_rate(params) * 2 * + snd_pcm_format_width(params_format(params)); + pclk_bclk_ratio = max9867->pclk/bclk_rate; + switch (snd_pcm_format_width(params_format(params))) { + case 8: + case 16: + if (pclk_bclk_ratio == 2) + bclk_value = MAX9867_IFC1B_PCLK_2; + else if (pclk_bclk_ratio == 4) + bclk_value = MAX9867_IFC1B_PCLK_4; + else if (pclk_bclk_ratio == 8) + bclk_value = MAX9867_IFC1B_PCLK_8; + else if (pclk_bclk_ratio == 16) + bclk_value = MAX9867_IFC1B_PCLK_16; + else { + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + break; + case 24: + bclk_value = MAX9867_IFC1B_24BIT; + break; + case 32: + bclk_value = MAX9867_IFC1B_32BIT; + break; + default: + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + regmap_update_bits(max9867->regmap, MAX9867_IFC1B, + MAX9867_IFC1B_BCLK_MASK, bclk_value); + } + return 0; +} + +static int max9867_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, MAX9867_SHTDOWN_MASK); + return 0; +} + +static int max9867_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + if (mute) + regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); + else + regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + MAX9867_DAC_MUTE_MASK, 0); + return 0; +} + +static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + int value = 0; + + /* Set the prescaler based on the master clock frequency*/ + if (freq >= 10000000 && freq <= 20000000) { + value |= MAX9867_PSCLK_10_20; + max9867->pclk = freq; + } else if (freq >= 20000000 && freq <= 40000000) { + value |= MAX9867_PSCLK_20_40; + max9867->pclk = freq/2; + } else if (freq >= 40000000 && freq <= 60000000) { + value |= MAX9867_PSCLK_40_60; + max9867->pclk = freq/4; + } else { + pr_err("bad clock frequency %d", freq); + return -EINVAL; + } + value = value << MAX9867_PSCLK_SHIFT; + max9867->sysclk = freq; + /* exact integer mode is not supported */ + value &= ~MAX9867_FREQ_MASK; + regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, + MAX9867_PSCLK_MASK, value); + return 0; +} + +static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + u8 iface1A = 0, iface1B = 0; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + max9867->master = 1; + iface1A |= MAX9867_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + max9867->master = 0; + iface1A &= ~MAX9867_MASTER; + break; + default: + return -EINVAL; + } + + /* for i2s compatible mode */ + iface1A |= MAX9867_I2S_DLY; + /* SDOUT goes to hiz state after all data is transferred */ + iface1A |= MAX9867_SDOUT_HIZ; + + /* Clock inversion bits, BCI and WCI */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface1A |= MAX9867_WCI_MODE | MAX9867_BCI_MODE; + break; + case SND_SOC_DAIFMT_IB_NF: + iface1A |= MAX9867_BCI_MODE; + break; + case SND_SOC_DAIFMT_NB_IF: + iface1A |= MAX9867_WCI_MODE; + break; + default: + return -EINVAL; + } + + ret = regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); + ret = regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; +} + +static struct snd_soc_dai_ops max9867_dai_ops = { + .set_fmt = max9867_dai_set_fmt, + .set_sysclk = max9867_set_dai_sysclk, + .prepare = max9867_prepare, + .digital_mute = max9867_mute, + .hw_params = max9867_dai_hw_params, +}; + +#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_driver max9867_dai[] = { + { + .name = "max9867-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9867_RATES, + .formats = MAX9867_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9867_RATES, + .formats = MAX9867_FORMATS, + }, + .ops = &max9867_dai_ops, + } +}; + +#ifdef CONFIG_PM_SLEEP +static int max9867_suspend(struct device *dev) +{ + struct max9867_priv *max9867 = dev_get_drvdata(dev); + + /* Drop down to power saving mode when system is suspended */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, ~MAX9867_SHTDOWN_MASK); + return 0; +} + +static int max9867_resume(struct device *dev) +{ + struct max9867_priv *max9867 = dev_get_drvdata(dev); + + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, MAX9867_SHTDOWN_MASK); + return 0; +} +#endif + +static int max9867_probe(struct snd_soc_codec *codec) +{ + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "max98090_probe\n"); + max9867->codec = codec; + return 0; +} + +static struct snd_soc_codec_driver max9867_codec = { + .probe = max9867_probe, + .controls = max9867_snd_controls, + .num_controls = ARRAY_SIZE(max9867_snd_controls), + .dapm_routes = max9867_audio_map, + .num_dapm_routes = ARRAY_SIZE(max9867_audio_map), + .dapm_widgets = max9867_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9867_dapm_widgets), +}; + +static bool max9867_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9867_STATUS: + case MAX9867_JACKSTATUS: + case MAX9867_AUXHIGH: + case MAX9867_AUXLOW: + return true; + default: + return false; + } +} + +static struct reg_default max9867_reg[] = { + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x0B, 0x00 }, + { 0x0C, 0x00 }, + { 0x0D, 0x00 }, + { 0x0E, 0x00 }, + { 0x0F, 0x00 }, + { 0x10, 0x00 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static struct regmap_config max9867_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX9867_REVISION, + .reg_defaults = max9867_reg, + .num_reg_defaults = ARRAY_SIZE(max9867_reg), + .volatile_reg = max9867_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max9867_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max9867_priv *max9867; + int ret = 0, reg; + + max9867 = devm_kzalloc(&i2c->dev, + sizeof(*max9867), GFP_KERNEL); + if (!max9867) + return -ENOMEM; + + i2c_set_clientdata(i2c, max9867); + max9867->regmap = devm_regmap_init_i2c(i2c, &max9867_regmap); + if (IS_ERR(max9867->regmap)) { + ret = PTR_ERR(max9867->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + ret = regmap_read(max9867->regmap, + MAX9867_REVISION, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read: %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device revision: %x\n", reg); + ret = snd_soc_register_codec(&i2c->dev, &max9867_codec, + max9867_dai, ARRAY_SIZE(max9867_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max9867_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max9867_i2c_id[] = { + { "max9867", 0 }, +}; + +static const struct of_device_id max9867_of_match[] = { + { .compatible = "maxim,max9867", }, + { } +}; + +MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); + +static const struct dev_pm_ops max9867_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(max9867_suspend, max9867_resume) +}; + +static struct i2c_driver max9867_i2c_driver = { + .driver = { + .name = "max9867", + .of_match_table = of_match_ptr(max9867_of_match), + .pm = &max9867_pm_ops, + }, + .probe = max9867_i2c_probe, + .remove = max9867_i2c_remove, + .id_table = max9867_i2c_id, +}; + +module_i2c_driver(max9867_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX9867 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h new file mode 100755 index 0000000..65590b4 --- /dev/null +++ b/sound/soc/codecs/max9867.h @@ -0,0 +1,83 @@ +/* + * max9867.h -- MAX9867 ALSA SoC Audio driver + * + * Copyright 2013-2015 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX9867_H +#define _MAX9867_H + +/* MAX9867 register space */ + +#define MAX9867_STATUS 0x00 +#define MAX9867_JACKSTATUS 0x01 +#define MAX9867_AUXHIGH 0x02 +#define MAX9867_AUXLOW 0x03 +#define MAX9867_INTEN 0x04 +#define MAX9867_SYSCLK 0x05 +#define MAX9867_FREQ_MASK 0xF +#define MAX9867_PSCLK_SHIFT 0x4 +#define MAX9867_PSCLK_WIDTH 0x2 +#define MAX9867_PSCLK_MASK (0x03<<MAX9867_PSCLK_SHIFT) +#define MAX9867_PSCLK_10_20 0x1 +#define MAX9867_PSCLK_20_40 0x2 +#define MAX9867_PSCLK_40_60 0x3 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_WIDTH 0x7 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0x7F +#define MAX9867_NI_LOW_SHIFT 0x1 +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_RAPID_LOCK 0x01 +#define MAX9867_IFC1A 0x08 +#define MAX9867_MASTER (1<<7) +#define MAX9867_I2S_DLY (1<<4) +#define MAX9867_SDOUT_HIZ (1<<3) +#define MAX9867_TDM_MODE (1<<2) +#define MAX9867_WCI_MODE (1<<6) +#define MAX9867_BCI_MODE (1<<5) +#define MAX9867_IFC1B 0x09 +#define MAX9867_IFC1B_BCLK_MASK 7 +#define MAX9867_IFC1B_32BIT 0x01 +#define MAX9867_IFC1B_24BIT 0x02 +#define MAX9867_IFC1B_PCLK_2 4 +#define MAX9867_IFC1B_PCLK_4 5 +#define MAX9867_IFC1B_PCLK_8 6 +#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_CODECFLTR 0x0a +#define MAX9867_DACGAIN 0x0b +#define MAX9867_DACLEVEL 0x0c +#define MAX9867_DAC_MUTE_SHIFT 0x6 +#define MAX9867_DAC_MUTE_WIDTH 0x1 +#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) +#define MAX9867_ADCLEVEL 0x0d +#define MAX9867_LEFTLINELVL 0x0e +#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_LEFTVOL 0x10 +#define MAX9867_RIGHTVOL 0x11 +#define MAX9867_LEFTMICGAIN 0x12 +#define MAX9867_RIGHTMICGAIN 0x13 +#define MAX9867_INPUTCONFIG 0x14 +#define MAX9867_INPUT_SHIFT 0x6 +#define MAX9867_MICCONFIG 0x15 +#define MAX9867_MODECONFIG 0x16 +#define MAX9867_PWRMAN 0x17 +#define MAX9867_SHTDOWN_MASK (1<<7) +#define MAX9867_REVISION 0xff + +#define MAX9867_CACHEREGNUM 10 + +/* codec private data */ +struct max9867_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + unsigned int sysclk; + unsigned int pclk; + unsigned int master; +}; +#endif -- 1.9.3

2 2

[alsa-devel] [RFC v4 0/8] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 18 Feb '16

18 Feb '16

Hi, these patches add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. The accesses to the hcp->eld field are now locked by a mutex, and I have added a patch to use the hdmi helper function [2] for N/CTS calculation. Changes since v3: - Add mutex to lock hcp->eld - Use helper function for N and CTS calculation The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/8337071/ ("drm/mediatek: Add HDMI support"), and need this patch: [2] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") The asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the machine driver Philipp Zabel (7): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver ASoC: mediatek: address dai link array entries by enum video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: hdmi-codec: Add ELD control drm/mediatek: hdmi: use helper function for N and CTS calculation .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 148 +++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 89 +++++----- drivers/gpu/drm/mediatek/mtk_hdmi.h | 10 +- drivers/gpu/drm/mediatek/mtk_hdmi_hw.c | 179 ++++----------------- drivers/gpu/drm/mediatek/mtk_hdmi_hw.h | 4 +- drivers/video/Makefile | 2 +- drivers/video/hdmi-not.c | 61 +++++++ include/linux/hdmi-not.h | 39 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/hdmi-codec.c | 123 +++++++++++++- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 66 +++++++- 14 files changed, 528 insertions(+), 216 deletions(-) create mode 100644 drivers/video/hdmi-not.c create mode 100644 include/linux/hdmi-not.h -- 2.7.0

2 11

[alsa-devel] [PATCH] ALSA: oxfw: use workqueue instead of tasklet for scs1x
by Takashi Sakamoto 18 Feb '16

18 Feb '16

This commit replaces tasklet with workqueue for scs1x functionality of ALSA oxfw driver. This driver transfers MIDI message specific for SCS.1m and SCS.1d. This task is currently done in software IRQ context of tasklet. In a view of system, this context is limited resources and some important drivers (at least, more important than ALSA oxfw driver) use the context as its bottom-harf. If the work to transfer MIDI messages is done within a time, it's better to use the other context for the work. Actually, with recent CPUs, the work will be scheduled within a time. This is a reason of this commit. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-scs1x.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c index bb53eb3..f7ac124 100644 --- a/sound/firewire/oxfw/oxfw-scs1x.c +++ b/sound/firewire/oxfw/oxfw-scs1x.c @@ -26,7 +26,7 @@ struct fw_scs1x { u8 output_bytes; bool output_escaped; bool output_escape_high_nibble; - struct tasklet_struct tasklet; + struct work_struct work; wait_queue_head_t idle_wait; u8 buffer[HSS1394_MAX_PACKET_SIZE]; bool transaction_running; @@ -129,7 +129,7 @@ static void scs_write_callback(struct fw_card *card, int rcode, ; /* TODO: retry this packet */ scs->transaction_running = false; - tasklet_schedule(&scs->tasklet); + schedule_work(&scs->work); } static bool is_valid_running_status(u8 status) @@ -165,9 +165,9 @@ static bool is_invalid_cmd(u8 status) status == 0xfd; } -static void scs_output_tasklet(unsigned long data) +static void scs_output_work(struct work_struct *work) { - struct fw_scs1x *scs = (struct fw_scs1x *)data; + struct fw_scs1x *scs = container_of(work, struct fw_scs1x, work); struct snd_rawmidi_substream *stream; unsigned int i; u8 byte; @@ -311,7 +311,7 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *stream, int up) scs->output_idle = false; ACCESS_ONCE(scs->output) = stream; - tasklet_schedule(&scs->tasklet); + schedule_work(&scs->work); } else { ACCESS_ONCE(scs->output) = NULL; } @@ -395,7 +395,7 @@ int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw) snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &midi_playback_ops); - tasklet_init(&scs->tasklet, scs_output_tasklet, (unsigned long)scs); + INIT_WORK(&scs->work, scs_output_work); init_waitqueue_head(&scs->idle_wait); scs->output_idle = true; -- 2.5.0

2 1

[alsa-devel] [PATCH v5] portman2x4 - use new parport device model
by Sudip Mukherjee 18 Feb '16

18 Feb '16

Modify portman driver to use the new parallel port device model. The advantage of using the device model is that the device gets binded to the hardware, we get the feature of hotplug, we can bind/unbind the driver at runtime. The changes are in the way the driver gets registered with the parallel port subsystem and the temporary device to probe portman card is removed and portman_probe() is used in the probe callback. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v5: claimed flag and one call to parport_claim() removed. v4: temporary device to probe is removed. v3: changed commit message v2: 1. pardev_cb is initialized while declaring, thus removing the use of memset. 2. used pdev->id. 3. v1 did not have the parport probe callback, but we will need the probe callback for binding as the name of the driver and the name of the device is different. 4. in v1 I missed modifying snd_portman_probe_port(). sound/drivers/portman2x4.c | 93 +++++++++++++++++++++------------------------- 1 file changed, 42 insertions(+), 51 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 464385a..362533f 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -83,8 +83,6 @@ struct portman { struct snd_card *card; struct snd_rawmidi *rmidi; struct pardevice *pardev; - int pardev_claimed; - int open_count; int mode[PORTMAN_NUM_INPUT_PORTS]; struct snd_rawmidi_substream *midi_input[PORTMAN_NUM_INPUT_PORTS]; @@ -648,30 +646,6 @@ static void snd_portman_interrupt(void *userdata) spin_unlock(&pm->reg_lock); } -static int snd_portman_probe_port(struct parport *p) -{ - struct pardevice *pardev; - int res; - - pardev = parport_register_device(p, DRIVER_NAME, - NULL, NULL, NULL, - 0, NULL); - if (!pardev) - return -EIO; - - if (parport_claim(pardev)) { - parport_unregister_device(pardev); - return -EIO; - } - - res = portman_probe(p); - - parport_release(pardev); - parport_unregister_device(pardev); - - return res ? -EIO : 0; -} - static void snd_portman_attach(struct parport *p) { struct platform_device *device; @@ -705,10 +679,20 @@ static void snd_portman_detach(struct parport *p) /* nothing to do here */ } +static int snd_portman_dev_probe(struct pardevice *pardev) +{ + if (strcmp(pardev->name, DRIVER_NAME)) + return -ENODEV; + + return 0; +} + static struct parport_driver portman_parport_driver = { - .name = "portman2x4", - .attach = snd_portman_attach, - .detach = snd_portman_detach + .name = "portman2x4", + .probe = snd_portman_dev_probe, + .match_port = snd_portman_attach, + .detach = snd_portman_detach, + .devmodel = true, }; /********************************************************************* @@ -720,8 +704,7 @@ static void snd_portman_card_private_free(struct snd_card *card) struct pardevice *pardev = pm->pardev; if (pardev) { - if (pm->pardev_claimed) - parport_release(pardev); + parport_release(pardev); parport_unregister_device(pardev); } @@ -736,6 +719,12 @@ static int snd_portman_probe(struct platform_device *pdev) struct snd_card *card = NULL; struct portman *pm = NULL; int err; + struct pardev_cb portman_cb = { + .preempt = NULL, + .wakeup = NULL, + .irq_func = snd_portman_interrupt, /* ISR */ + .flags = PARPORT_DEV_EXCL, /* flags */ + }; p = platform_get_drvdata(pdev); platform_set_drvdata(pdev, NULL); @@ -745,9 +734,6 @@ static int snd_portman_probe(struct platform_device *pdev) if (!enable[dev]) return -ENOENT; - if ((err = snd_portman_probe_port(p)) < 0) - return err; - err = snd_card_new(&pdev->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -759,23 +745,32 @@ static int snd_portman_probe(struct platform_device *pdev) sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); - pardev = parport_register_device(p, /* port */ - DRIVER_NAME, /* name */ - NULL, /* preempt */ - NULL, /* wakeup */ - snd_portman_interrupt, /* ISR */ - PARPORT_DEV_EXCL, /* flags */ - (void *)card); /* private */ + portman_cb.private = card; /* private */ + pardev = parport_register_dev_model(p, /* port */ + DRIVER_NAME, /* name */ + &portman_cb, /* callbacks */ + pdev->id); /* device number */ if (pardev == NULL) { snd_printd("Cannot register pardevice\n"); err = -EIO; goto __err; } + /* claim parport */ + if (parport_claim(pardev)) { + snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); + err = -EIO; + goto free_pardev; + } + err = portman_probe(p); + if (err) { + err = -EIO; + goto release_pardev; + } + if ((err = portman_create(card, pardev, &pm)) < 0) { snd_printd("Cannot create main component\n"); - parport_unregister_device(pardev); - goto __err; + goto release_pardev; } card->private_data = pm; card->private_free = snd_portman_card_private_free; @@ -785,14 +780,6 @@ static int snd_portman_probe(struct platform_device *pdev) goto __err; } - /* claim parport */ - if (parport_claim(pardev)) { - snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); - err = -EIO; - goto __err; - } - pm->pardev_claimed = 1; - /* init device */ if ((err = portman_device_init(pm)) < 0) goto __err; @@ -808,6 +795,10 @@ static int snd_portman_probe(struct platform_device *pdev) snd_printk(KERN_INFO "Portman 2x4 on 0x%lx\n", p->base); return 0; +release_pardev: + parport_release(pardev); +free_pardev: + parport_unregister_device(pardev); __err: snd_card_free(card); return err; -- 1.9.1

2 2

[alsa-devel] another locking issue with nonatomic ops?
by Vinod Koul 18 Feb '16

18 Feb '16

Hi Takashi, We found another issue with non atomic ops. This occurs when we have snd_pcm_do_prepare being invoked which acquires the lock snd_pcm_link_rwsem twice, first in snd_pcm_common_ioctl1, and then again in dpcm_dapm_stream_event. Normally this is fine, but if we have another stream get closed in between two read accesses we get stuck. First thread: [ 419.657259] AudioOut_2 D ffff8800704a3a60 0 3691 1 0x20020004 [ 419.665946] ffff8800704a3a60 00000000704a3a60 ffff88006d1fddd0 ffffffff8252ffd8 [ 419.674678] ffff8800704a4000 ffffffff8252ffc0 ffffffffffffffff ffffffff8252ffd8 [ 419.683513] 0000000000000000 ffff8800704a3a80 ffffffff81ca2987 ffffffffffffffff [ 419.683574] Call Trace: [ 419.692290] [<ffffffff81ca2987>] schedule+0x37/0x90 [ 419.692306] [<ffffffff81ca6f6d>] rwsem_down_read_failed+0xdd/0x130 [ 419.701050] [<ffffffff814df454>] call_rwsem_down_read_failed+0x14/0x30 [ 419.709758] [<ffffffff810e793b>] ? down_read_nested+0x5b/0x70 [ 419.709805] [<ffffffff81adfda9>] ? snd_pcm_stream_lock+0xa9/0x120 [ 419.723012] [<ffffffff81adfda9>] snd_pcm_stream_lock+0xa9/0x120 < ============ 2nd Read lock is acquired here, without releasing previous one. [ 419.723026] [<ffffffff81adfe4c>] snd_pcm_stream_lock_irq+0x2c/0x30 [ 419.731801] [<ffffffff81b01f5e>] ? dpcm_dapm_stream_event+0xbe/0xd0 [ 419.740565] [<ffffffff81b0079c>] dpcm_set_fe_update_state+0x3c/0xb0 [ 419.749252] [<ffffffff81b04514>] dpcm_fe_dai_prepare+0xc4/0x150 [ 419.749301] [<ffffffff81adf7eb>] snd_pcm_do_prepare+0x1b/0x30 [ 419.758083] [<ffffffff81adf3bf>] snd_pcm_action_single+0x2f/0x70 [ 419.766897] [<ffffffff81ca6827>] ? down_read+0x47/0x60 [ 419.766962] [<ffffffff81adf577>] ? snd_pcm_action_nonatomic+0x27/0x80 [ 419.775565] [<ffffffff81adf5c6>] snd_pcm_action_nonatomic+0x76/0x80 < ============ 1st Read lock is acquired here [ 419.784419] [<ffffffff81ae2c62>] snd_pcm_common_ioctl1+0x802/0xd30 [ 419.784495] [<ffffffff81ae332b>] snd_pcm_playback_ioctl1+0x19b/0x280 [ 419.793106] [<ffffffff81230915>] ? __fget+0x5/0x210 [ 419.801943] [<ffffffff81ae3dca>] snd_pcm_ioctl_compat+0x41a/0x770 [ 419.801959] [<ffffffff81230915>] ? __fget+0x5/0x210 [ 419.810793] [<ffffffff8126d670>] compat_SyS_ioctl+0xd0/0x13b0 [ 419.819491] [<ffffffff81ca7bc4>] ? do_nanosleep+0x94/0x190 [ 419.819535] [<ffffffff810eb696>] ? trace_hardirqs_on_caller+0x16/0x210 [ 419.828306] [<ffffffff814df4f8>] ? trace_hardirqs_on_thunk+0x17/0x19 [ 419.837104] [<ffffffff81cabf0e>] sysenter_dispatch+0x7/0x1f Second thread: [ 419.543062] mediaserver D ffff880170957cd8 0 3690 1 0x20020004 [ 419.543191] ffff880170957cd8 0000000070957cd8 ffff88006de49ad0 ffffffff8252ffd8 [ 419.551920] ffff880170958000 ffff88006de49ad0 ffffffff8252ffc0 ffffffff00000001 [ 419.560555] ffffffff8252ffd8 ffff880170957cf8 ffffffff81ca2987 ffffffff8252ffc0 [ 419.569368] Call Trace: [ 419.569392] [<ffffffff81ca2987>] schedule+0x37/0x90 [ 419.578026] [<ffffffff81ca71f5>] rwsem_down_write_failed+0x235/0x450 [ 419.586884] [<ffffffff81ca7005>] ? rwsem_down_write_failed+0x45/0x450 [ 419.586941] [<ffffffff814df483>] call_rwsem_down_write_failed+0x13/0x20 [ 419.595620] [<ffffffff81ca689f>] ? down_write+0x5f/0x80 < ============ Write lock acquired here. If this occurs between two read lock's then it's hangs [ 419.604428] [<ffffffff81aded58>] ? snd_pcm_unlink+0x28/0x120 [ 419.604465] [<ffffffff81aded58>] snd_pcm_unlink+0x28/0x120 [ 419.613215] [<ffffffff81adee5e>] pcm_release_private+0xe/0x10 [ 419.622058] [<ffffffff81ae0d57>] snd_pcm_release_substream.part.34+0x67/0x90 [ 419.622143] [<ffffffff81ae0e68>] snd_pcm_release+0xb8/0xc0 [ 419.630889] [<ffffffff8120f1aa>] __fput+0xea/0x200 [ 419.639680] [<ffffffff8120f30e>] ____fput+0xe/0x10 [ 419.639744] [<ffffffff810ba43c>] task_work_run+0xcc/0x100 [ 419.648435] [<ffffffff81002e41>] do_notify_resume+0x51/0x70 [ 419.657205] [<ffffffff81ca986c>] int_signal+0x12/0x17 Thanks -- ~Vinod

4 13

[alsa-devel] [PATCH] ALSA: hda - hdmi get jack from hda_jack_tbl when not dyn_pcm_assign
by libin.yang＠linux.intel.com 18 Feb '16

18 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> On Intel platform, if !dyn_pcm_assign, spec->pcm_rec[].jack is not NULL even after snd_hda_jack_tbl_clear() is called to free snd_jack. This may cause access invalid memory when calling snd_jack_report. Please see more detail from: https://bugs.freedesktop.org/show_bug.cgi?id=94079 Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f4443b5..3b47101 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1962,6 +1962,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; + struct hda_jack_tbl *jack_tbl; struct snd_jack *jack = NULL; int size; @@ -1989,10 +1990,21 @@ static void sync_eld_via_acomp(struct hda_codec *codec, /* pcm_idx >=0 before update_eld() means it is in monitor * disconnected event. Jack must be fetched before update_eld() */ - if (per_pin->pcm_idx >= 0) + /* if !dyn_pcm_assign, get jack from hda_jack_tbl + * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not + * NULL even after snd_hda_jack_tbl_clear() is called to + * free snd_jack. This may cause access invalid memory + * when calling snd_jack_report + */ + if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) jack = spec->pcm_rec[per_pin->pcm_idx].jack; + else { + jack_tbl = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + if (jack_tbl) + jack = jack_tbl->jack; + } update_eld(codec, per_pin, eld); - if (jack == NULL && per_pin->pcm_idx >= 0) + if (jack == NULL && per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) jack = spec->pcm_rec[per_pin->pcm_idx].jack; if (jack == NULL) goto unlock; -- 1.9.1

3 10

Re: [alsa-devel] Auto mute mode
by Takashi Iwai 18 Feb '16

18 Feb '16

On Wed, 17 Feb 2016 09:31:55 +0100, Kailang wrote: > > Hi Takashi, > > I add the ALC225 for Dell headset mode supported. > But it has the issue for speaker don't mute when headphone or headset was pluged. Is alc_update_headset_mode() called when you plug/unplug the headphone or the headset? If yes, which value is set to new_headset_mode? > The issue is occur to Mic in pluged. > When Mic in is pluged, auto mute mode will change to disable. > But when plug headphone or headset into jack, auto mute mode still keep on disable. > So, speaker will have output at the same time. The same question applies. This is a Dell machine with ALC255_FIXUP_DELL1_MIC_NO_PRESENCE fix up. The mic pin has no jack detection but the detection is triggered only by the headset (HP) pin. All these imply likely that the detection of the headset mode doesn't work properly. I haven't touched with these Dell quirks, but Canonical guys did it. They should have a better clue. Now Cc'ed. Takashi

4 4

[alsa-devel] [PATCH] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 17 Feb '16

17 Feb '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This changes adds support for exposing the card number for kernel clients as well as the PID for user client. It supports kernels with and without the required support. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 1 + include/sound/asequencer.h | 5 +++-- src/seq/seq.c | 11 +++++++++++ src/seq/seq_hw.c | 8 +++++++- 4 files changed, 22 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..77996e5 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,7 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_owner(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..7ebf7fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,8 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int owner; /* RO: card number[kernel] / PID[user] */ + char reserved[64 - sizeof(int)]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..2505293 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,17 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number (kernel) or owning PID + * \param info client_info container + * \return card number/PID/-1 if value is not available. + */ +int snd_seq_client_info_get_owner(const snd_seq_client_info_t *info) +{ + assert(info); + return info->owner; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..a4c083a 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int micro_version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,8 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) + info->owner = -1; return 0; } @@ -368,6 +371,8 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) + info->owner = -1; return 0; } @@ -480,6 +485,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->micro_version = SNDRV_PROTOCOL_MICRO(ver); if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.6.2

3 16

[alsa-devel] [PATCH] ASoC: hdac_hdmi: Dereference 'edev' after checking for NULL
by Fabio Estevam 17 Feb '16

17 Feb '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 4a3478debf36 ("ASoC: hdac_hdmi: Add jack reporting") leads to the following Smatch complaint: sound/soc/codecs/hdac_hdmi.c:971 hdac_hdmi_present_sense() warn: variable dereferenced before check 'edev' (see line 967) ,so deference 'edev' after checking for NULL. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/hdac_hdmi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 36f1200..920af2f 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -964,13 +964,15 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) { struct hdac_ext_device *edev = pin->edev; - struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_priv *hdmi; struct hdac_hdmi_pcm *pcm; int val; if (!edev) return; + hdmi = edev->private_data; + pin->repoll_count = repoll; pm_runtime_get_sync(&edev->hdac.dev); -- 1.9.1

3 3

[alsa-devel] [PATCH v2] ASoC: hdac_hdmi: Remove 'edev' NULL check
by Fabio Estevam 17 Feb '16

17 Feb '16

Commit 4a3478debf36 ("ASoC: hdac_hdmi: Add jack reporting") leads to the following Smatch complaint: sound/soc/codecs/hdac_hdmi.c:971 hdac_hdmi_present_sense() warn: variable dereferenced before check 'edev' (see line 967) As explained by Subhransu S. Prusty: "This NULL check is not required, can be removed. edev object is expected to be present when the present_sense is called. This is called from places where this edev object is already available." So remove the evdev NULL check. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Remove NULL check (Subhransu) sound/soc/codecs/hdac_hdmi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 36f1200..2884dbf 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -968,9 +968,6 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) struct hdac_hdmi_pcm *pcm; int val; - if (!edev) - return; - pin->repoll_count = repoll; pm_runtime_get_sync(&edev->hdac.dev); -- 1.9.1

3 2

[alsa-devel] [PATCH RFT] ASoC: max98926: Fix blr_clk_ratio calculation
by Axel Lin 17 Feb '16

17 Feb '16

Current code calculates blr_clk_ratio before setting max98926->ch_size, fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Anish, I don't have this h/w, please review and test this patch. thanks, Axel sound/soc/codecs/max98926.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 66884ebd..5245e10 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -383,8 +383,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream, int rate = params_rate(params), i; struct snd_soc_codec *codec = dai->codec; struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); - /* BCLK/LRCLK ratio calculation */ - int blr_clk_ratio = params_channels(params) * max98926->ch_size; + int blr_clk_ratio; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -414,6 +413,9 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = params_channels(params) * max98926->ch_size; + switch (blr_clk_ratio) { case 32: regmap_update_bits(max98926->regmap, -- 2.1.4

3 2

[alsa-devel] [RFC v1 0/9] ASoC: Add support to Qualcomm msm8x16-wcd codec
by Srinivas Kandagatla 17 Feb '16

17 Feb '16

This patchset aims at adding msm8x16-wcd codec support. msm8x16-wcd codec is found in Qualcomm msm8916 and apq8016 processors. This codec IP is split in to two parts(Digital & Analog), Analog part is integrated in to PMIC PM8916 and the digital part is integrated into Application processor. Register access to the analog part is done via SPMI interface to PMIC, and registers on the Application processor are memory mapped. Data transfer between Analog and Digital Die is done via a internal bus called PDM. I did split the driver into sensible patches to easy the review process. Most of this driver is ported from downstream Andriod v3.10 kernel. Jack support and Headset button detection code is not added in to this series. This codec support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary Mic. - 2 Digital Microphones. - 2 Mic Bias Circuits. - Earpiece - Headset - Loud Speaker. - Jack Detect. - Headset Button detection circuit. Tested this code on v4.5-rc4 with lpass driver capture support patches for Headset (Mic and speakers), Secondary Mic and Loudspeaker on DB410C. DMIC should work too. Thanks, srini Srinivas Kandagatla (9): ASoC: msm8x176: Add Device Tree bindings. ASoC: msm8x16: add driver structure ASoC: msm8x16: add codec registers definitions ASoC: msm8x16: add ranges for default, readonly ASoC: msm8x16: add codec intialization setup ASoC: msm8x16: add codec dais ASoC: msm8x16: Add sound mixer controls. ASoC: msm8x16: add dapm widgets ASoC: msm8x16: add dapm routes .../devicetree/bindings/sound/qcom,msm8x16-wcd.txt | 91 + include/dt-bindings/sound/msm8x16-wcd.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 +- sound/soc/codecs/msm8x16-wcd-registers.h | 518 +++++ sound/soc/codecs/msm8x16-wcd.c | 2132 ++++++++++++++++++++ sound/soc/codecs/msm8x16-wcd.h | 764 +++++++ 7 files changed, 3518 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8x16-wcd.txt create mode 100644 include/dt-bindings/sound/msm8x16-wcd.h create mode 100644 sound/soc/codecs/msm8x16-wcd-registers.h create mode 100644 sound/soc/codecs/msm8x16-wcd.c create mode 100644 sound/soc/codecs/msm8x16-wcd.h -- 1.9.1

3 18

[alsa-devel] [PATCH v4] portman2x4 - use new parport device model
by Sudip Mukherjee 17 Feb '16

17 Feb '16

Modify portman driver to use the new parallel port device model. The advantage of using the device model is that the device gets binded to the hardware, we get the feature of hotplug, we can bind/unbind the driver at runtime. The changes are in the way the driver gets registered with the parallel port subsystem and the temporary device to probe portman card is removed and portman_probe() is used in the probe callback. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v4: temporary device to probe is removed. v3: changed commit message v2: 1. pardev_cb is initialized while declaring, thus removing the use of memset. 2. used pdev->id. 3. v1 did not have the parport probe callback, but we will need the probe callback for binding as the name of the driver and the name of the device is different. 4. in v1 I missed modifying snd_portman_probe_port(). sound/drivers/portman2x4.c | 79 +++++++++++++++++++++++----------------------- 1 file changed, 40 insertions(+), 39 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 464385a..0741a3b 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -648,30 +648,6 @@ static void snd_portman_interrupt(void *userdata) spin_unlock(&pm->reg_lock); } -static int snd_portman_probe_port(struct parport *p) -{ - struct pardevice *pardev; - int res; - - pardev = parport_register_device(p, DRIVER_NAME, - NULL, NULL, NULL, - 0, NULL); - if (!pardev) - return -EIO; - - if (parport_claim(pardev)) { - parport_unregister_device(pardev); - return -EIO; - } - - res = portman_probe(p); - - parport_release(pardev); - parport_unregister_device(pardev); - - return res ? -EIO : 0; -} - static void snd_portman_attach(struct parport *p) { struct platform_device *device; @@ -705,10 +681,20 @@ static void snd_portman_detach(struct parport *p) /* nothing to do here */ } +static int snd_portman_dev_probe(struct pardevice *pardev) +{ + if (strcmp(pardev->name, DRIVER_NAME)) + return -ENODEV; + + return 0; +} + static struct parport_driver portman_parport_driver = { - .name = "portman2x4", - .attach = snd_portman_attach, - .detach = snd_portman_detach + .name = "portman2x4", + .probe = snd_portman_dev_probe, + .match_port = snd_portman_attach, + .detach = snd_portman_detach, + .devmodel = true, }; /********************************************************************* @@ -736,6 +722,12 @@ static int snd_portman_probe(struct platform_device *pdev) struct snd_card *card = NULL; struct portman *pm = NULL; int err; + struct pardev_cb portman_cb = { + .preempt = NULL, + .wakeup = NULL, + .irq_func = snd_portman_interrupt, /* ISR */ + .flags = PARPORT_DEV_EXCL, /* flags */ + }; p = platform_get_drvdata(pdev); platform_set_drvdata(pdev, NULL); @@ -745,9 +737,6 @@ static int snd_portman_probe(struct platform_device *pdev) if (!enable[dev]) return -ENOENT; - if ((err = snd_portman_probe_port(p)) < 0) - return err; - err = snd_card_new(&pdev->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -759,23 +748,33 @@ static int snd_portman_probe(struct platform_device *pdev) sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); - pardev = parport_register_device(p, /* port */ - DRIVER_NAME, /* name */ - NULL, /* preempt */ - NULL, /* wakeup */ - snd_portman_interrupt, /* ISR */ - PARPORT_DEV_EXCL, /* flags */ - (void *)card); /* private */ + portman_cb.private = card; /* private */ + pardev = parport_register_dev_model(p, /* port */ + DRIVER_NAME, /* name */ + &portman_cb, /* callbacks */ + pdev->id); /* device number */ if (pardev == NULL) { snd_printd("Cannot register pardevice\n"); err = -EIO; goto __err; } + if (parport_claim(pardev)) { + err = -EIO; + goto free_pardev; + } + + err = portman_probe(p); + parport_release(pardev); + + if (err) { + err = -EIO; + goto free_pardev; + } + if ((err = portman_create(card, pardev, &pm)) < 0) { snd_printd("Cannot create main component\n"); - parport_unregister_device(pardev); - goto __err; + goto free_pardev; } card->private_data = pm; card->private_free = snd_portman_card_private_free; @@ -808,6 +807,8 @@ static int snd_portman_probe(struct platform_device *pdev) snd_printk(KERN_INFO "Portman 2x4 on 0x%lx\n", p->base); return 0; +free_pardev: + parport_unregister_device(pardev); __err: snd_card_free(card); return err; -- 1.9.1

2 1

Re: [alsa-devel] ASoC: hdac_hdmi: Add jack reporting
by Dan Carpenter 17 Feb '16

17 Feb '16

Hello Jeeja KP, This is a semi-automatic email about new static checker warnings. The patch 4a3478debf36: "ASoC: hdac_hdmi: Add jack reporting" from Feb 12, 2016, leads to the following Smatch complaint: sound/soc/codecs/hdac_hdmi.c:971 hdac_hdmi_present_sense() warn: variable dereferenced before check 'edev' (see line 967) sound/soc/codecs/hdac_hdmi.c 966 struct hdac_ext_device *edev = pin->edev; 967 struct hdac_hdmi_priv *hdmi = edev->private_data; ^^^^^^^^^^^^^^^^^^ Patch introduces a new dereference. 968 struct hdac_hdmi_pcm *pcm; 969 int val; 970 971 if (!edev) ^^^^^ Existing code assumed that edev could be NULL. 972 return; 973 regards, dan carpenter

1 0

[alsa-devel] [PATCH v2] sound: hdmi: avoid dereferencing uninitialized 'jack' pointer
by Arnd Bergmann 16 Feb '16

16 Feb '16

When CONFIG_SND_JACK is disabled, the hdmi driver tries to create a new output jack structure and dereferences it even though the pointer is never assigned: sound/pci/hda/patch_hdmi.c: In function 'generic_hdmi_build_jack': sound/pci/hda/patch_hdmi.c:2534:30: error: 'jack' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/pci/hda/patch_hdmi.c:2526:19: note: 'jack' was declared here This makes snd_jack_new() return an error when SND_JACK is disabled, forcing all users to handle errors gracefully and avoiding the problem in all callers that correctly check the return code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 788d441a164c ("ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling") --- include/sound/jack.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 23bede121c78..1e84a2e413f8 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -99,7 +99,7 @@ void snd_jack_report(struct snd_jack *jack, int status); static inline int snd_jack_new(struct snd_card *card, const char *id, int type, struct snd_jack **jack, bool initial_kctl, bool phantom_jack) { - return 0; + return -ENXIO; } static inline int snd_jack_add_new_kctl(struct snd_jack *jack, const char * name, int mask) -- 2.7.0

1 0

[alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Jeeja KP 16 Feb '16

16 Feb '16

We have no mechanism to configure hw_params for DAIs which are hostless. Like a loop or DAIs connected to internal sinks/sources. Problem Statement: During a BE-BE loop of DSP or a SRC->SINK path, we cannot program hw_params. There are no hw_params from user as these are hostless. This scenario can happens in below 2 cases: Case 1: BE-BE loop aka hostless stream through SoC DSP Hostless stream can be defined by BE to BE connection and in this case the path is enabled or disabled by the state of the DAPM graph. This usually means there is a mixer control that can be used to connect or disconnect the path between both DAIs. Example: |--------------/----------------| | | | | codec out modem_in | | | | ssp0_TX ssp1_RX | | | | AIFI_Playback dummy_capture In the above case (SSP0,codec playback,SSp1 dai needs to be configured and turned on with the selected hw_params). Case 2: SoC Internal Sink or Source Example: WOV SINK (DAPM Sink) | | | Switch | | w1 | dmic rx (BE dai widget) | Capture(codec dai widget) | DMIC AIF | DMic | Soc DMic (MIC Widget) (SRC) In the above case BE (dmic RX) dai hw_params needs to applied. Approach: 1. Allow params to be defined for other BE dai link also. this currently applicable only for codec_codec link. 2. when params is defined for BE dai link, create params kcontrols and associate it with the dai widgets. this will allow params to be selected via Kcontrol for each dai defined in the dai link. 3. Before powering on the dai widget and if the dai is not powered with the stream event, then dai_ops(hw_params) for the dai widget will be called.This will apply the selected hw_params. 4. with the above, fix_up can be removed and can use .params to select the parameter. Core Changes: 1. params definition needs to changed, currently this is defined to used only for codec codec link. define new flag to identifying codec codec link. 2. if params is defined for any BE dai link, then create kcontrol for each dai defined in the dai_link and associated the kcontrol to dai widget. 3. In the powering sequence check if the dai widget and if the event is not a stream event, SND_SOC_DAPM_STREAM_NOP , then call dai ops with the selected params. This will apply the hw_params for dai. 1. params definition: - Add new flag in struct snd_soc_dai_link. - Based on this flag create, the codec codec link instead of params struct snd_soc_dai_link { const struct snd_soc_pcm_stream *params; unsigned int num_params; + /* flag to create codec codec based on the flag*/ + unsigned int codec_link:1; + unsigned int dai_fmt; /* format to set on init */ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ - creation of dai link based on the codec_link flag static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } } else { - if (!dai_link->params) { + if (!dai_link->codec_link) { /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { 2. when params is defined for BE dai link, create params kcontrols and associate it with the dai widgets. this will allow params to be selected via Kcontrol for each dai defined in the dai link. a. Move the param control creation from snd_soc_dapm_new_pcm and add new API to created kcontrols with the params. +snd_soc_dapm_create_param_control(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params) { /* create kcontrol based on the params */ } - modify the snd_soc_dapm_new_pcm to create the param control 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) { template.kcontrol_news = snd_soc_dapm_create_param_control(card, params, num_params); 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; + goto outfree_link_name; } b. add new API snd_soc_dapm_new_dai_params() to create kcontrol for dai and associated it to dai. +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *dai_w) +{ + struct snd_kcontrol_new *kcontrol_news; + + kcontrol_news = snd_soc_dapm_create_param_control(card, params, + num_params); + if (!kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + dai_w->name); + return -EINVAL; + } + + dai_w->num_kcontrols = 1; + dai_w->kcontrol_news = kcontrol_news; + + dai_w->params = params; + dai_w->num_params = num_params; + + return 0; + +} + +++ b/include/sound/soc-dapm.h +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *dai_w); c. if the be dai link has .params defined, call new API snd_soc_dapm_new_dai_params +static int soc_link_be_params_widget(struct snd_soc_card *card, + struct snd_soc_dai *dai, + const struct snd_soc_pcm_stream *params, + unsigned int num_params) +{ + int ret; + struct snd_soc_dapm_widget *w; + + if (!strcmp(dai->name, "snd-soc-dummy")) + return 0; + + if (dai->playback_widget) { + w = dai->playback_widget; + ret = snd_soc_dapm_new_dai_params(card, params, + num_params, w); + + if (ret != 0) { + dev_err(card->dev, + "ASoC: Can't create params widget=%s %d\n", + w->name, ret); + return ret; + } + } + + if (dai->capture_widget) { + w = dai->capture_widget; + ret = snd_soc_dapm_new_dai_params(card, params, + num_params, w); + + if (ret != 0) { + dev_err(card->dev, + "ASoC: Can't create params widget=%s %d\n", + w->name, ret); + return ret; + } + } + + return 0; +} + +static int soc_link_be_params_dai_widgets(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret, i; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec = rtd->codec_dais[i]; + + ret = soc_link_be_params_widget(card, codec, + dai_link->params, + dai_link->num_params); + + if (!ret) + return ret; + } + + return soc_link_be_params_widget(card, cpu_dai, dai_link->params, + dai_link->num_params); +} + @@ -1651,6 +1714,13 @@ static int soc_probe_link_dais(struct snd_soc_card *card, dai_link->stream_name, ret); return ret; } + if (dai_link->params && dai_link->no_pcm) { + /* create dai link controls */ + ret = soc_link_be_params_dai_widgets(card, + dai_link, rtd); + if (ret) + return ret; + } } else { INIT_DELAYED_WORK(&rtd->delayed_work, codec2codec_close_delayed_work); 3. Applying the hw_params for dai widget --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1398,9 +1398,64 @@ static void dapm_seq_check_event(struct snd_soc_card * card, } } +static int snd_soc_dai_params(struct snd_soc_dapm_widget *w, int event) +{ + struct snd_soc_dai *dai; + const struct snd_soc_pcm_stream *config = w->params + w->params_select; + struct snd_pcm_substream substream; + struct snd_pcm_hw_params *params = NULL; + u64 fmt; + int ret = 0; + + if (WARN_ON(!config)) + return -EINVAL; + + dai = w->priv; + + if (config->formats) { + fmt = ffs(config->formats) - 1; + } else { + dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n", + config->formats); + fmt = 0; + } + + /* Currently very limited parameter selection */ + params = kzalloc(sizeof(*params), GFP_KERNEL); + if (!params) { + ret = -ENOMEM; + goto out; + } + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); + + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = + config->rate_min; + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = + config->rate_max; + + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min + = config->channels_min; + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max + = config->channels_max; + + memset(&substream, 0, sizeof(substream)); + if (w->id == snd_soc_dapm_dai_in) + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + else + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + + ret = soc_dai_hw_params(&substream, params, dai); + if (ret < 0) + goto out; +out: + kfree(params); + return ret; +} + /* Apply the coalesced changes from a DAPM sequence */ static void dapm_seq_run_coalesced(struct snd_soc_card *card, - struct list_head *pending) + struct list_head *pending, + int event) { struct snd_soc_dapm_context *dapm; struct snd_soc_dapm_widget *w; @@ -1413,6 +1468,14 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, dapm = w->dapm; list_for_each_entry(w, pending, power_list) { + if ((event == SND_SOC_DAPM_STREAM_NOP) && + ((w->id == snd_soc_dapm_dai_out) || + (w->id == snd_soc_dapm_dai_in))) { + snd_soc_dai_params(w, SND_SOC_DAPM_PRE_PMU); + } + } + + list_for_each_entry(w, pending, power_list) { WARN_ON(reg != w->reg || dapm != w->dapm); w->power = w->new_power; @@ -1447,6 +1510,13 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMU); dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMD); } + list_for_each_entry(w, pending, power_list) { + if ((event == SND_SOC_DAPM_STREAM_NOP) && + ((w->id == snd_soc_dapm_dai_out) || + (w->id == snd_soc_dapm_dai_in))) { + snd_soc_dai_params(w, SND_SOC_DAPM_POST_PMD); + } + } } /* Apply a DAPM power sequence. @@ -1482,7 +1552,7 @@ static void dapm_seq_run(struct snd_soc_card *card, if (sort[w->id] != cur_sort || w->reg != cur_reg || w->dapm != cur_dapm || w->subseq != cur_subseq) { if (!list_empty(&pending)) - dapm_seq_run_coalesced(card, &pending); + dapm_seq_run_coalesced(card, &pending, event); if (cur_dapm && cur_dapm->seq_notifier) { for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) @@ -1545,7 +1615,7 @@ static void dapm_seq_run(struct snd_soc_card *card, } if (!list_empty(&pending)) - dapm_seq_run_coalesced(card, &pending); + dapm_seq_run_coalesced(card, &pending, event); if (cur_dapm && cur_dapm->seq_notifier) { for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) -- 1.7.9.5 Comments welcome, based on discussion we will send patches. --Jeeja --

1 0

[alsa-devel] sound: list corruption in delete_and_unsubscribe_port
by Dmitry Vyukov 16 Feb '16

16 Feb '16

Hello, Here is a new one on 18558cae0272f8fd9647e69d3fec1565a7949865 (4.5-rc4). But need to note that sound become much more stable, I've seen only 2 of these over night. The following program causes list corruption: ------------[ cut here ]------------ WARNING: CPU: 2 PID: 12546 at lib/list_debug.c:62 __list_del_entry+0x10b/0x1e0() list_del corruption, ffff880063512388->next is LIST_POISON1 (dead000000000100) Modules linked in: CPU: 2 PID: 12546 Comm: a.out Not tainted 4.5.0-rc4+ #328 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87b05080 ffff8800608b7a48 ffffffff82be46cf ffffffff81477fb8 fffffbfff0f60a10 ffff8800608b7ab8 ffff8800637d97c0 ffffffff86ad3780 0000000000000009 000000000000003e ffff8800608b7a88 ffffffff81355139 Call Trace: [<ffffffff81355249>] warn_slowpath_fmt+0xa9/0xd0 kernel/panic.c:494 [<ffffffff82c4c36b>] __list_del_entry+0x10b/0x1e0 lib/list_debug.c:60 [<ffffffff82c4c44d>] list_del+0xd/0x70 lib/list_debug.c:86 [<ffffffff852c38e3>] delete_and_unsubscribe_port+0x1e3/0x2f0 sound/core/seq/seq_ports.c:545 [<ffffffff852c43fa>] clear_subscriber_list+0x15a/0x260 sound/core/seq/seq_ports.c:250 [<ffffffff852c456a>] port_delete+0x6a/0x1c0 sound/core/seq/seq_ports.c:266 [<ffffffff852c5242>] snd_seq_delete_all_ports+0x242/0x350 sound/core/seq/seq_ports.c:330 [<ffffffff852ae1cf>] seq_free_client1+0x2f/0x290 sound/core/seq/seq_clientmgr.c:272 [<ffffffff852ae495>] seq_free_client+0x65/0x160 sound/core/seq/seq_clientmgr.c:299 [<ffffffff852b118d>] snd_seq_release+0x4d/0xb0 sound/core/seq/seq_clientmgr.c:380 [<ffffffff817c3256>] __fput+0x236/0x780 fs/file_table.c:208 [<ffffffff817c3825>] ____fput+0x15/0x20 fs/file_table.c:244 [<ffffffff813b3100>] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] tracehook_notify_resume include/linux/tracehook.h:191 [<ffffffff810066b1>] exit_to_usermode_loop+0x1d1/0x210 arch/x86/entry/common.c:251 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:282 [<ffffffff810084ea>] syscall_return_slowpath+0x2ba/0x340 arch/x86/entry/common.c:344 [<ffffffff866626e2>] int_ret_from_sys_call+0x25/0x9f arch/x86/entry/entry_64.S:281 ---[ end trace 4cad985f706f8ace ]--- ------------[ cut here ]------------ // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> long r[143]; void* thr(void* arg) { switch ((long)arg) { case 0: syscall(SYS_mmap, 0x20000000ul, 0x40000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 1: r[2] = syscall(SYS_open, "/dev/snd/seq", 0x400ul, 0, 0, 0); break; case 2: *(uint8_t*)0x2000df50 = (uint8_t)0xffffffffffffff80; *(uint8_t*)0x2000df51 = (uint8_t)0x1; *(uint8_t*)0x2000df52 = (uint8_t)0xfff; *(uint8_t*)0x2000df53 = (uint8_t)0xb0; *(uint8_t*)0x2000df54 = (uint8_t)0x401; *(uint8_t*)0x2000df55 = (uint8_t)0x7; *(uint8_t*)0x2000df56 = (uint8_t)0x7; *(uint8_t*)0x2000df57 = (uint8_t)0x0; *(uint8_t*)0x2000df58 = (uint8_t)0x1; *(uint8_t*)0x2000df59 = (uint8_t)0x401; *(uint8_t*)0x2000df5a = (uint8_t)0xffffffffffff0001; *(uint8_t*)0x2000df5b = (uint8_t)0x5; *(uint8_t*)0x2000df5c = (uint8_t)0x6; *(uint8_t*)0x2000df5d = (uint8_t)0x0; *(uint8_t*)0x2000df5e = (uint8_t)0xffff; *(uint8_t*)0x2000df5f = (uint8_t)0x3; *(uint8_t*)0x2000df60 = (uint8_t)0x0; *(uint8_t*)0x2000df61 = (uint8_t)0xffffffffffffffc0; *(uint8_t*)0x2000df62 = (uint8_t)0x6; *(uint8_t*)0x2000df63 = (uint8_t)0xfff; *(uint8_t*)0x2000df64 = (uint8_t)0x624c; *(uint8_t*)0x2000df65 = (uint8_t)0x53; *(uint8_t*)0x2000df66 = (uint8_t)0x0; *(uint8_t*)0x2000df67 = (uint8_t)0xfffffffffffffffc; *(uint8_t*)0x2000df68 = (uint8_t)0x3f; *(uint8_t*)0x2000df69 = (uint8_t)0x2; *(uint8_t*)0x2000df6a = (uint8_t)0x4; *(uint8_t*)0x2000df6b = (uint8_t)0x401; *(uint8_t*)0x2000df6c = (uint8_t)0x100000000; *(uint8_t*)0x2000df6d = (uint8_t)0x5; *(uint8_t*)0x2000df6e = (uint8_t)0x1; *(uint8_t*)0x2000df6f = (uint8_t)0x9; *(uint8_t*)0x2000df70 = (uint8_t)0x40; *(uint8_t*)0x2000df71 = (uint8_t)0xfff; *(uint8_t*)0x2000df72 = (uint8_t)0x6; *(uint8_t*)0x2000df73 = (uint8_t)0xffffffffffffff2b; *(uint8_t*)0x2000df74 = (uint8_t)0x1f; *(uint8_t*)0x2000df75 = (uint8_t)0x2; *(uint8_t*)0x2000df76 = (uint8_t)0x4; *(uint8_t*)0x2000df77 = (uint8_t)0x68; *(uint8_t*)0x2000df78 = (uint8_t)0x9c33; *(uint8_t*)0x2000df79 = (uint8_t)0x80; *(uint8_t*)0x2000df7a = (uint8_t)0x3; *(uint8_t*)0x2000df7b = (uint8_t)0x100; *(uint8_t*)0x2000df7c = (uint8_t)0xc1b1; *(uint8_t*)0x2000df7d = (uint8_t)0x3; *(uint8_t*)0x2000df7e = (uint8_t)0x0; *(uint8_t*)0x2000df7f = (uint8_t)0x8; *(uint8_t*)0x2000df80 = (uint8_t)0x3; *(uint8_t*)0x2000df81 = (uint8_t)0x8; *(uint8_t*)0x2000df82 = (uint8_t)0x8; *(uint8_t*)0x2000df83 = (uint8_t)0x5d; *(uint8_t*)0x2000df84 = (uint8_t)0x1; *(uint8_t*)0x2000df85 = (uint8_t)0x9; *(uint8_t*)0x2000df86 = (uint8_t)0x1; *(uint8_t*)0x2000df87 = (uint8_t)0x41; *(uint8_t*)0x2000df88 = (uint8_t)0x3; *(uint8_t*)0x2000df89 = (uint8_t)0x6; *(uint8_t*)0x2000df8a = (uint8_t)0x3f; *(uint8_t*)0x2000df8b = (uint8_t)0x3; *(uint8_t*)0x2000df8c = (uint8_t)0x9; *(uint8_t*)0x2000df8d = (uint8_t)0xffffffffffffff01; *(uint8_t*)0x2000df8e = (uint8_t)0x0; *(uint8_t*)0x2000df8f = (uint8_t)0x6; *(uint8_t*)0x2000df90 = (uint8_t)0xe9c; *(uint8_t*)0x2000df91 = (uint8_t)0x0; *(uint32_t*)0x2000df94 = (uint32_t)0x7b; *(uint32_t*)0x2000df98 = (uint32_t)0x1002; *(uint32_t*)0x2000df9c = (uint32_t)0x9; *(uint32_t*)0x2000dfa0 = (uint32_t)0x1000; *(uint32_t*)0x2000dfa4 = (uint32_t)0x80; *(uint32_t*)0x2000dfa8 = (uint32_t)0x10000; *(uint32_t*)0x2000dfac = (uint32_t)0x8; *(uint64_t*)0x2000dfb0 = (uint64_t)0x0; *(uint32_t*)0x2000dfb8 = (uint32_t)0x3; *(uint32_t*)0x2000dfbc = (uint32_t)0x39; *(uint8_t*)0x2000dfc0 = (uint8_t)0x0; *(uint8_t*)0x2000dfc1 = (uint8_t)0x0; *(uint8_t*)0x2000dfc2 = (uint8_t)0x0; *(uint8_t*)0x2000dfc3 = (uint8_t)0x0; *(uint8_t*)0x2000dfc4 = (uint8_t)0x0; *(uint8_t*)0x2000dfc5 = (uint8_t)0x0; *(uint8_t*)0x2000dfc6 = (uint8_t)0x0; *(uint8_t*)0x2000dfc7 = (uint8_t)0x0; *(uint8_t*)0x2000dfc8 = (uint8_t)0x0; *(uint8_t*)0x2000dfc9 = (uint8_t)0x0; *(uint8_t*)0x2000dfca = (uint8_t)0x0; *(uint8_t*)0x2000dfcb = (uint8_t)0x0; *(uint8_t*)0x2000dfcc = (uint8_t)0x0; *(uint8_t*)0x2000dfcd = (uint8_t)0x0; *(uint8_t*)0x2000dfce = (uint8_t)0x0; *(uint8_t*)0x2000dfcf = (uint8_t)0x0; *(uint8_t*)0x2000dfd0 = (uint8_t)0x0; *(uint8_t*)0x2000dfd1 = (uint8_t)0x0; *(uint8_t*)0x2000dfd2 = (uint8_t)0x0; *(uint8_t*)0x2000dfd3 = (uint8_t)0x0; *(uint8_t*)0x2000dfd4 = (uint8_t)0x0; *(uint8_t*)0x2000dfd5 = (uint8_t)0x0; *(uint8_t*)0x2000dfd6 = (uint8_t)0x0; *(uint8_t*)0x2000dfd7 = (uint8_t)0x0; *(uint8_t*)0x2000dfd8 = (uint8_t)0x0; *(uint8_t*)0x2000dfd9 = (uint8_t)0x0; *(uint8_t*)0x2000dfda = (uint8_t)0x0; *(uint8_t*)0x2000dfdb = (uint8_t)0x0; *(uint8_t*)0x2000dfdc = (uint8_t)0x0; *(uint8_t*)0x2000dfdd = (uint8_t)0x0; *(uint8_t*)0x2000dfde = (uint8_t)0x0; *(uint8_t*)0x2000dfdf = (uint8_t)0x0; *(uint8_t*)0x2000dfe0 = (uint8_t)0x0; *(uint8_t*)0x2000dfe1 = (uint8_t)0x0; *(uint8_t*)0x2000dfe2 = (uint8_t)0x0; *(uint8_t*)0x2000dfe3 = (uint8_t)0x0; *(uint8_t*)0x2000dfe4 = (uint8_t)0x0; *(uint8_t*)0x2000dfe5 = (uint8_t)0x0; *(uint8_t*)0x2000dfe6 = (uint8_t)0x0; *(uint8_t*)0x2000dfe7 = (uint8_t)0x0; *(uint8_t*)0x2000dfe8 = (uint8_t)0x0; *(uint8_t*)0x2000dfe9 = (uint8_t)0x0; *(uint8_t*)0x2000dfea = (uint8_t)0x0; *(uint8_t*)0x2000dfeb = (uint8_t)0x0; *(uint8_t*)0x2000dfec = (uint8_t)0x0; *(uint8_t*)0x2000dfed = (uint8_t)0x0; *(uint8_t*)0x2000dfee = (uint8_t)0x0; *(uint8_t*)0x2000dfef = (uint8_t)0x0; *(uint8_t*)0x2000dff0 = (uint8_t)0x0; *(uint8_t*)0x2000dff1 = (uint8_t)0x0; *(uint8_t*)0x2000dff2 = (uint8_t)0x0; *(uint8_t*)0x2000dff3 = (uint8_t)0x0; *(uint8_t*)0x2000dff4 = (uint8_t)0x0; *(uint8_t*)0x2000dff5 = (uint8_t)0x0; *(uint8_t*)0x2000dff6 = (uint8_t)0x0; *(uint8_t*)0x2000dff7 = (uint8_t)0x0; *(uint8_t*)0x2000dff8 = (uint8_t)0x0; *(uint8_t*)0x2000dff9 = (uint8_t)0x0; *(uint8_t*)0x2000dffa = (uint8_t)0x0; r[138] = syscall(SYS_ioctl, r[2], 0xc0a85320ul, 0x2000df50ul, 0, 0, 0); break; case 3: r[139] = syscall(SYS_read, r[2], 0x20025000ul, 0x75ul, 0, 0, 0); break; case 4: r[140] = syscall(SYS_close, r[2], 0, 0, 0, 0, 0); break; case 5: memcpy((void*)0x20022000, "\x2f\x64\x65\x76\x2f\x73\x65\x71\x75\x65\x6e\x63\x65\x72", 14); syscall(SYS_open, "/dev/sequencer", 0x4000ul, 0, 0, 0); break; } return 0; } int main() { long i; pthread_t th[6]; srand(getpid()); memset(r, -1, sizeof(r)); for (i = 0; i < 6; i++) { pthread_create(&th[i], 0, thr, (void*)i); usleep(10000); } for (i = 0; i < 6; i++) { pthread_create(&th[i], 0, thr, (void*)i); if (rand() % 2 == 0) usleep(rand() % 10000); } usleep(100000); return 0; }

2 3

[alsa-devel] [PATCH v6 00/15] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 16 Feb '16

16 Feb '16

This patch series adds DP audio and hotplug notification support. In skylake platform the DP is on a different port which is not enabled by default. A vendor widget is programmed to enable playback on all ports. This also enables all the converters. Codec Dais, widgets, graph are now created dynamically based on the nodes enumerated, thus removing the hardcoding. Mux controls are used to establish routing stream to a specific port. For DP audio support, infoframe needs to be different. Based on ELD queried we either pack HDMI or DP infoframe. Also with this series, Jack notification support is added and Jack event is reported to userspace as is done in case of legacy driver. Few fixes are added to check for possible memory leak, reconfiguring register during resume from D3 etc. The 8th patch is adding a small macro for getting connection type in drm header. It is reviewed by drm folks, CCed them. Please merge it through sound trees. changes in v6: - Fixed some review comments - Fix afg node D3 state Cc: Jani Nikula <jani.nikula(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Daniel Vetter <daniel.vetter(a)intel.com> Jeeja KP (2): ASoC: hdac_hdmi: Add jack reporting ASoC: hdac_hdmi: Add PM support Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Subhransu S. Prusty (12): ASoC: hdac_hdmi: Add hotplug notification and read ELD ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi: create dais based on number of cvts ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Enable playback on all enumerated ports drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec include/drm/drm_edid.h | 12 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1208 +++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/hdac_hdmi.h | 6 + 4 files changed, 1104 insertions(+), 123 deletions(-) create mode 100644 sound/soc/codecs/hdac_hdmi.h -- 1.9.1

4 28

[alsa-devel] [PATCH v2 0/3] ASoC: topology: Add support for FE DAI & DAI links
by mengdong.lin＠linux.intel.com 16 Feb '16

16 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows topology to create FE DAI and DAI links from the PCM topology objects defined by the user space. History: v2: Remove first 2 patches in v1 that were accepted. Add Ack by Takashi to patch 'ALSA: pcm: Add snd_pcm_rate_range_to_bits()' and add his comments on the function's assumption to c file. Mengdong Lin (3): ALSA: pcm: Add snd_pcm_rate_range_to_bits() ASoC: topology: Add FE DAIs dynamically ASoC: topology: Add FE DAI links dynamically include/sound/pcm.h | 2 + include/sound/soc-topology.h | 21 +++-- include/sound/soc.h | 2 +- sound/core/pcm_misc.c | 30 +++++++ sound/soc/soc-topology.c | 195 ++++++++++++++++++++++++++++++++++--------- 5 files changed, 197 insertions(+), 53 deletions(-) -- 2.5.0

6 14

[alsa-devel] [PATCH v3 0/2] ASoC: topology: Add support for FE DAI & DAI links
by mengdong.lin＠linux.intel.com 16 Feb '16

16 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows topology to create FE DAI and DAI links from the PCM topology objects defined by the user space. History: v2: Remove first 2 patches in v1 that were accepted. Add Ack by Takashi to patch 'ALSA: pcm: Add snd_pcm_rate_range_to_bits()' and add his comments on the function's assumption to c file. v3: remove 1st patch in v2 that was accepted. Revise setting of a stream's rates/rate_min/rate_max as Lars suggested. Mengdong Lin (2): ASoC: topology: Add FE DAIs dynamically ASoC: topology: Add FE DAI links dynamically include/sound/soc-topology.h | 21 +++-- include/sound/soc.h | 2 +- sound/soc/soc-topology.c | 195 ++++++++++++++++++++++++++++++++++--------- 3 files changed, 165 insertions(+), 53 deletions(-) -- 2.5.0

2 3

[alsa-devel] [PATCH] ASoC: topology: Fix setting of stream rates, rate_min and rate_max
by mengdong.lin＠linux.intel.com 16 Feb '16

16 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Directly set a stream's rates, rate_min and rate_max from the topology info. Also define set_stream_info to wrap setting of the stream info. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0eb01e8..2f7da32 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1575,6 +1575,18 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } +static int set_stream_info(struct snd_soc_pcm_stream *stream, + struct snd_soc_tplg_stream_caps *caps) +{ + stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + stream->channels_min = caps->channels_min; + stream->channels_max = caps->channels_max; + stream->rates = caps->rates; + stream->rate_min = caps->rate_min; + stream->rate_max = caps->rate_max; + stream->formats = caps->formats; +} + static int soc_tplg_dai_create(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *pcm) { @@ -1593,25 +1605,13 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - - stream->stream_name = kstrdup(caps->name, GFP_KERNEL); - stream->channels_min = caps->channels_min; - stream->channels_max = caps->channels_max; - stream->rates = snd_pcm_rate_range_to_bits(caps->rate_min, - caps->rate_max); - stream->formats = caps->formats; + set_stream_info(stream, caps); } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - - stream->stream_name = kstrdup(caps->name, GFP_KERNEL); - stream->channels_min = caps->channels_min; - stream->channels_max = caps->channels_max; - stream->rates = snd_pcm_rate_range_to_bits(caps->rate_min, - caps->rate_max); - stream->formats = caps->formats; + set_stream_info(stream, caps); } /* pass control to component driver for optional further init */ -- 2.5.0

1 0

[alsa-devel] [asoc:topic/topology 1/2] sound/soc/soc-topology.c:1572:19: error: implicit declaration of function 'snd_pcm_rate_range_to_bits'
by kbuild test robot 15 Feb '16

15 Feb '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/topology head: acfc7d46cddcf71cf18772bbe8717b84eac5f672 commit: 64527e8a352968bda529f01df1c9dd5fe581ff04 [1/2] ASoC: topology: Add FE DAIs dynamically config: i386-allmodconfig (attached as .config) reproduce: git checkout 64527e8a352968bda529f01df1c9dd5fe581ff04 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/soc-topology.c: In function 'soc_tplg_dai_create': >> sound/soc/soc-topology.c:1572:19: error: implicit declaration of function 'snd_pcm_rate_range_to_bits' [-Werror=implicit-function-declaration] stream->rates = snd_pcm_rate_range_to_bits(caps->rate_min, ^ sound/soc/soc-topology.c: At top level: sound/soc/soc-topology.c:249:31: warning: 'get_dobj_type' defined but not used [-Wunused-function] static enum snd_soc_dobj_type get_dobj_type(struct snd_soc_tplg_hdr *hdr, ^ cc1: some warnings being treated as errors vim +/snd_pcm_rate_range_to_bits +1572 sound/soc/soc-topology.c 1566 stream = &dai_drv->playback; 1567 caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; 1568 1569 stream->stream_name = kstrdup(caps->name, GFP_KERNEL); 1570 stream->channels_min = caps->channels_min; 1571 stream->channels_max = caps->channels_max; > 1572 stream->rates = snd_pcm_rate_range_to_bits(caps->rate_min, 1573 caps->rate_max); 1574 stream->formats = caps->formats; 1575 } --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2 00/14] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 15 Feb '16

15 Feb '16

This patch series adds DP audio and hotplug notification support. On Skylake two DP ports are available and to enable DP on both ports all pins need to be enabled. There is a special vendor widget which need to be programmed to enable all pins and converters. This series adds hotplug notification, read/set constraint based on ELD, enable all pin/cvts, DP1.2, programs the audio infoframe for DP. There is a one to one mapping between converter and stream, so the dais are created based on the no of streams supported on hdmi codec. Even though cvts can be mapped dynamically to the streams, currently it is statically mapped as simultaneous playback on both DP and HDMI is not supported as of now. Pin muxes and controls are created dynamically to map converter to pin widget. So at run time specific pin is mapped to the dai based on the control selected (based on the display type DP/HDMI connected). Finally the DP audio infoframe programming is added to support the DP feature. Also with hotplug notification support, ELD is read and capabilities are set for rate, formats and channels. drm_eld sound/core framework is updated to limit the formats based on ELD. There are few fixes one fixing the static checker warning and other one not to fail if no connection list is found for a pin widget. Pls note, the 11th patch is adding a small macro for getting connection type in drm header, we have CCed drm folks on that and this one. Pls ack so that we can have this series merged thru sound trees changes in v2: - Address review comments - Dropped devm_xxx for cvt and pin memory allocation. - Dropped the ELD helper APIs from core and copied to driver. - Updated notification patch to add PM fixes from legacy hda. - commit message and change log updated for display power fix. - Updated DRM helper APIs and the caller Jeeja KP (1): ASoC: hdac_hdmi: Add codec suspend/resume handler Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep display active while enumerating codec Subhransu S. Prusty (12): ASoC: hdac_hdmi: Fix to check num nodes correctly ASoC: hdac_hdmi: Fix to warn instead of err for no connected nids ASoC: hdac_hdmi - Use list to add pins and converters ASoC: hdac_hdmi: Add hotplug notification and read eld ALSA: pcm: Add DRM helper to set constraint for format ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi - create dais based on number of streams ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Assign pin for stream based on dapm connection drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio include/drm/drm_edid.h | 12 + sound/core/pcm_drm_eld.c | 42 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1019 ++++++++++++++++++++++++++++++++++++------ 4 files changed, 925 insertions(+), 149 deletions(-) -- 1.9.1

4 32

[alsa-devel] [PATCH 0/5] ASoC: topology: Add support for FE DAI & DAI links
by mengdong.lin＠linux.intel.com 15 Feb '16

15 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows topology to create FE DAI and DAI links from the PCM topology objects defined by the user space. Mengdong Lin (5): ASoC: Define soc_add_dai() to add a DAI to a component ASoC: Support registering a DAI dynamically ALSA: pcm: Add snd_pcm_rate_range_to_bits() ASoC: topology: Add FE DAIs dynamically ASoC: topology: Add FE DAI links dynamically include/sound/pcm.h | 2 + include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 21 +++-- include/sound/soc.h | 5 +- sound/core/pcm_misc.c | 27 ++++++ sound/soc/soc-core.c | 128 ++++++++++++++++++++-------- sound/soc/soc-topology.c | 195 ++++++++++++++++++++++++++++++++++--------- 7 files changed, 291 insertions(+), 88 deletions(-) -- 2.5.0

3 12

[alsa-devel] [PATCH] ASoC: Add max9867 codec driver
by anish kumar 15 Feb '16

15 Feb '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max9867.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max9867.c | 528 +++++++++++++++++++++ sound/soc/codecs/max9867.h | 83 ++++ 5 files changed, 633 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max9867.txt create mode 100755 sound/soc/codecs/max9867.c create mode 100755 sound/soc/codecs/max9867.h diff --git a/Documentation/devicetree/bindings/sound/max9867.txt b/Documentation/devicetree/bindings/sound/max9867.txt new file mode 100644 index 0000000..394cd4e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9867.txt @@ -0,0 +1,17 @@ +max9867 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max9867" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max9867: max9867@0x18 { + compatible = "maxim,max9867"; + reg = <0x18>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c3178e3..913c83c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C @@ -490,6 +491,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX9867 + tristate + config SND_SOC_MAX98925 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 84843f3..f39dee7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -70,6 +70,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c new file mode 100755 index 0000000..57e4c93 --- /dev/null +++ b/sound/soc/codecs/max9867.c @@ -0,0 +1,528 @@ +/* + * max9867.c -- max9867 ALSA SoC Audio driver + * + * Copyright 2013-15 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max9867.h" + +static const char *const max9867_spmode[] = { + "Stereo Diff", "Mono Diff", + "Stereo Cap", "Mono Cap", + "Stereo Single", "Mono Single", + "Stereo Single Fast", "Mono Single Fast" +}; +static const char *const max9867_sidetone_text[] = { + "None", "Left", "Right", "LeftRight", "LeftRightDiv2", +}; +static const char *const max9867_filter_text[] = {"IIR", "FIR"}; + +static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, + max9867_filter_text); +static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, + max9867_spmode); +static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, + max9867_sidetone_text); + +static const struct snd_kcontrol_new max9867_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 63, 1), + SOC_DOUBLE_R("Capture Volume", MAX9867_LEFTLINELVL, + MAX9867_RIGTHLINELVL, 0, 15, 1), + SOC_DOUBLE_R("Mic Gain", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 31, 1), + SOC_DOUBLE_R("Mic Preamp Gain", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 3, 0), + SOC_ENUM("Digital Sidetone Src", max9867_sidetone), + SOC_SINGLE("Sidetone Gain", MAX9867_DACGAIN, 0, 31, 1), + SOC_SINGLE("DAC Gain", MAX9867_DACLEVEL, 4, 3, 0), + SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), + SOC_DOUBLE("ADC Left Gain", MAX9867_ADCLEVEL, 4, 0, 15, 1), + SOC_DOUBLE("ADC Right Gain", MAX9867_ADCLEVEL, 0, 0, 15, 1), + SOC_ENUM("Speaker Mode", max9867_spkmode), + SOC_SINGLE("Volume Smoothing", MAX9867_MODECONFIG, 6, 1, 0), + SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_ENUM("DSP Filter", max9867_filter), +}; + +static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; + +static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, + MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, + max9867_mux); + +static const struct snd_kcontrol_new max9867_dapm_mux_controls = + SOC_DAPM_ENUM("Route", max9867_mux_enum); + +static const struct snd_kcontrol_new max9867_left_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); +static const struct snd_kcontrol_new max9867_right_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); +static const struct snd_kcontrol_new max9867_line_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); + +static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HPOUT"), + + SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), + SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &max9867_dapm_mux_controls), + + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, + &max9867_left_dapm_control), + SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, + &max9867_right_dapm_control), + SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, + &max9867_line_dapm_control), + SND_SOC_DAPM_INPUT("LINE_IN"), +}; + +static const struct snd_soc_dapm_route max9867_audio_map[] = { + {"Left DAC", NULL, "DAI_OUT"}, + {"Right DAC", NULL, "DAI_OUT"}, + {"Output Mixer", NULL, "Left DAC"}, + {"Output Mixer", NULL, "Right DAC"}, + {"HPOUT", NULL, "Output Mixer"}, + + {"Left ADC", NULL, "DAI_IN"}, + {"Right ADC", NULL, "DAI_IN"}, + {"Input Mixer", NULL, "Left ADC"}, + {"Input Mixer", NULL, "Right ADC"}, + {"Input Mux", "Line", "Input Mixer"}, + {"Input Mux", "Mic", "Input Mixer"}, + {"Input Mux", "Mic_Line", "Input Mixer"}, + {"Right Line", "Switch", "Input Mux"}, + {"Left Line", "Switch", "Input Mux"}, + {"LINE_IN", NULL, "Left Line"}, + {"LINE_IN", NULL, "Right Line"}, +}; + +enum rates { + pcm_rate_8, pcm_rate_16, pcm_rate_24, + pcm_rate_32, pcm_rate_44, + pcm_rate_48, max_pcm_rate, +}; + +struct ni_div_rates { + u32 mclk; + u16 ni[max_pcm_rate]; +} ni_div[] = { + {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, + {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, + {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, + {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, + {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, + {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, + {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, + {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +}; + +static inline int get_ni_value(int mclk, int rate) +{ + int i, ret = 0; + + /* find the closest rate index*/ + for (i = 0; i < ARRAY_SIZE(ni_div); i++) { + if (ni_div[i].mclk >= mclk) + break; + } + + switch (rate) { + case 8000: + return ni_div[i].ni[pcm_rate_8]; + case 16000: + return ni_div[i].ni[pcm_rate_16]; + case 32000: + return ni_div[i].ni[pcm_rate_32]; + case 44100: + return ni_div[i].ni[pcm_rate_44]; + case 48000: + return ni_div[i].ni[pcm_rate_48]; + default: + pr_err("%s wrong rate %d\n", __func__, rate); + ret = -EINVAL; + } + return ret; +} + +static int max9867_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + if (!max9867->master) { + unsigned int ni_h, ni_l, value; + + value = get_ni_value(max9867->sysclk, params_rate(params)); + if (value < 0) + return value; + + ni_h = (0xFF00 & value) >> 8; + ni_l = 0x00FF & value; + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + /* set up the ni value */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_NI_HIGH_MASK, ni_h); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_NI_LOW_MASK, ni_l); + /*rapid lock mode*/ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); + } else { + unsigned long int bclk_rate, pclk_bclk_ratio; + int bclk_value; + + bclk_rate = params_rate(params) * 2 * + snd_pcm_format_width(params_format(params)); + pclk_bclk_ratio = max9867->pclk/bclk_rate; + switch (snd_pcm_format_width(params_format(params))) { + case 8: + case 16: + if (pclk_bclk_ratio == 2) + bclk_value = MAX9867_IFC1B_PCLK_2; + else if (pclk_bclk_ratio == 4) + bclk_value = MAX9867_IFC1B_PCLK_4; + else if (pclk_bclk_ratio == 8) + bclk_value = MAX9867_IFC1B_PCLK_8; + else if (pclk_bclk_ratio == 16) + bclk_value = MAX9867_IFC1B_PCLK_16; + else { + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + break; + case 24: + bclk_value = MAX9867_IFC1B_24BIT; + break; + case 32: + bclk_value = MAX9867_IFC1B_32BIT; + break; + default: + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + regmap_update_bits(max9867->regmap, MAX9867_IFC1B, + MAX9867_IFC1B_BCLK_MASK, bclk_value); + } + return 0; +} + +static int max9867_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, MAX9867_SHTDOWN_MASK); + return 0; +} + +static int max9867_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + if (mute) + regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); + else + regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + MAX9867_DAC_MUTE_MASK, 0); + return 0; +} + +static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + int value = 0; + + /* Set the prescaler based on the master clock frequency*/ + if (freq >= 10000000 && freq <= 20000000) { + value |= MAX9867_PSCLK_10_20; + max9867->pclk = freq; + } else if (freq >= 20000000 && freq <= 40000000) { + value |= MAX9867_PSCLK_20_40; + max9867->pclk = freq/2; + } else if (freq >= 40000000 && freq <= 60000000) { + value |= MAX9867_PSCLK_40_60; + max9867->pclk = freq/4; + } else { + pr_err("bad clock frequency %d", freq); + return -EINVAL; + } + value = value << MAX9867_PSCLK_SHIFT; + max9867->sysclk = freq; + /* exact integer mode is not supported */ + value &= ~MAX9867_FREQ_MASK; + regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, + MAX9867_PSCLK_MASK, value); + return 0; +} + +static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + u8 iface1A = 0, iface1B = 0; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + max9867->master = 1; + iface1A |= MAX9867_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + max9867->master = 0; + iface1A &= ~MAX9867_MASTER; + break; + default: + return -EINVAL; + } + + /* for i2s compatible mode */ + iface1A |= MAX9867_I2S_DLY; + /* SDOUT goes to hiz state after all data is transferred */ + iface1A |= MAX9867_SDOUT_HIZ; + + /* Clock inversion bits, BCI and WCI */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface1A |= MAX9867_WCI_MODE | MAX9867_BCI_MODE; + break; + case SND_SOC_DAIFMT_IB_NF: + iface1A |= MAX9867_BCI_MODE; + break; + case SND_SOC_DAIFMT_NB_IF: + iface1A |= MAX9867_WCI_MODE; + break; + default: + return -EINVAL; + } + + ret = regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); + ret = regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; +} + +static struct snd_soc_dai_ops max9867_dai_ops = { + .set_fmt = max9867_dai_set_fmt, + .set_sysclk = max9867_set_dai_sysclk, + .prepare = max9867_prepare, + .digital_mute = max9867_mute, + .hw_params = max9867_dai_hw_params, +}; + +#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_driver max9867_dai[] = { + { + .name = "max9867-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9867_RATES, + .formats = MAX9867_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9867_RATES, + .formats = MAX9867_FORMATS, + }, + .ops = &max9867_dai_ops, + } +}; + +#ifdef CONFIG_PM_SLEEP +static int max9867_suspend(struct device *dev) +{ + struct max9867_priv *max9867 = dev_get_drvdata(dev); + + /* Drop down to power saving mode when system is suspended */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, ~MAX9867_SHTDOWN_MASK); + return 0; +} + +static int max9867_resume(struct device *dev) +{ + struct max9867_priv *max9867 = dev_get_drvdata(dev); + + msleep(20); + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, MAX9867_SHTDOWN_MASK); + return 0; +} +#endif + +static int max9867_probe(struct snd_soc_codec *codec) +{ + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + max9867->codec = codec; + codec->control_data = max9867->regmap; + return 0; +} + +static struct snd_soc_codec_driver max9867_codec = { + .probe = max9867_probe, + .controls = max9867_snd_controls, + .num_controls = ARRAY_SIZE(max9867_snd_controls), + .dapm_routes = max9867_audio_map, + .num_dapm_routes = ARRAY_SIZE(max9867_audio_map), + .dapm_widgets = max9867_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9867_dapm_widgets), +}; + +static bool max9867_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9867_STATUS: + case MAX9867_JACKSTATUS: + case MAX9867_AUXHIGH: + case MAX9867_AUXLOW: + return true; + default: + return false; + } +} + +static struct reg_default max9867_reg[] = { + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x0B, 0x00 }, + { 0x0C, 0x00 }, + { 0x0D, 0x00 }, + { 0x0E, 0x00 }, + { 0x0F, 0x00 }, + { 0x10, 0x00 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static struct regmap_config max9867_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX9867_REVISION, + .reg_defaults = max9867_reg, + .num_reg_defaults = ARRAY_SIZE(max9867_reg), + .volatile_reg = max9867_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max9867_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max9867_priv *max9867; + int ret = 0, reg; + + max9867 = devm_kzalloc(&i2c->dev, + sizeof(*max9867), GFP_KERNEL); + if (!max9867) + return -ENOMEM; + + i2c_set_clientdata(i2c, max9867); + max9867->regmap = devm_regmap_init_i2c(i2c, &max9867_regmap); + if (IS_ERR(max9867->regmap)) { + ret = PTR_ERR(max9867->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + ret = regmap_read(max9867->regmap, + MAX9867_REVISION, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read: %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device revision: %x\n", reg); + ret = snd_soc_register_codec(&i2c->dev, &max9867_codec, + max9867_dai, ARRAY_SIZE(max9867_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max9867_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max9867_i2c_id[] = { + { "max9867", 0 }, +}; + +static const struct of_device_id max9867_of_match[] = { + { .compatible = "maxim,max9867", }, + { } +}; + +MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); + +static const struct dev_pm_ops max9867_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(max9867_suspend, max9867_resume) +}; + +static struct i2c_driver max9867_i2c_driver = { + .driver = { + .name = "max9867", + .of_match_table = of_match_ptr(max9867_of_match), + .pm = &max9867_pm_ops, + }, + .probe = max9867_i2c_probe, + .remove = max9867_i2c_remove, + .id_table = max9867_i2c_id, +}; + +module_i2c_driver(max9867_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX9867 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h new file mode 100755 index 0000000..65590b4 --- /dev/null +++ b/sound/soc/codecs/max9867.h @@ -0,0 +1,83 @@ +/* + * max9867.h -- MAX9867 ALSA SoC Audio driver + * + * Copyright 2013-2015 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX9867_H +#define _MAX9867_H + +/* MAX9867 register space */ + +#define MAX9867_STATUS 0x00 +#define MAX9867_JACKSTATUS 0x01 +#define MAX9867_AUXHIGH 0x02 +#define MAX9867_AUXLOW 0x03 +#define MAX9867_INTEN 0x04 +#define MAX9867_SYSCLK 0x05 +#define MAX9867_FREQ_MASK 0xF +#define MAX9867_PSCLK_SHIFT 0x4 +#define MAX9867_PSCLK_WIDTH 0x2 +#define MAX9867_PSCLK_MASK (0x03<<MAX9867_PSCLK_SHIFT) +#define MAX9867_PSCLK_10_20 0x1 +#define MAX9867_PSCLK_20_40 0x2 +#define MAX9867_PSCLK_40_60 0x3 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_WIDTH 0x7 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0x7F +#define MAX9867_NI_LOW_SHIFT 0x1 +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_RAPID_LOCK 0x01 +#define MAX9867_IFC1A 0x08 +#define MAX9867_MASTER (1<<7) +#define MAX9867_I2S_DLY (1<<4) +#define MAX9867_SDOUT_HIZ (1<<3) +#define MAX9867_TDM_MODE (1<<2) +#define MAX9867_WCI_MODE (1<<6) +#define MAX9867_BCI_MODE (1<<5) +#define MAX9867_IFC1B 0x09 +#define MAX9867_IFC1B_BCLK_MASK 7 +#define MAX9867_IFC1B_32BIT 0x01 +#define MAX9867_IFC1B_24BIT 0x02 +#define MAX9867_IFC1B_PCLK_2 4 +#define MAX9867_IFC1B_PCLK_4 5 +#define MAX9867_IFC1B_PCLK_8 6 +#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_CODECFLTR 0x0a +#define MAX9867_DACGAIN 0x0b +#define MAX9867_DACLEVEL 0x0c +#define MAX9867_DAC_MUTE_SHIFT 0x6 +#define MAX9867_DAC_MUTE_WIDTH 0x1 +#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) +#define MAX9867_ADCLEVEL 0x0d +#define MAX9867_LEFTLINELVL 0x0e +#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_LEFTVOL 0x10 +#define MAX9867_RIGHTVOL 0x11 +#define MAX9867_LEFTMICGAIN 0x12 +#define MAX9867_RIGHTMICGAIN 0x13 +#define MAX9867_INPUTCONFIG 0x14 +#define MAX9867_INPUT_SHIFT 0x6 +#define MAX9867_MICCONFIG 0x15 +#define MAX9867_MODECONFIG 0x16 +#define MAX9867_PWRMAN 0x17 +#define MAX9867_SHTDOWN_MASK (1<<7) +#define MAX9867_REVISION 0xff + +#define MAX9867_CACHEREGNUM 10 + +/* codec private data */ +struct max9867_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + unsigned int sysclk; + unsigned int pclk; + unsigned int master; +}; +#endif -- 1.9.3

2 1

[alsa-devel] Error while doing ./alsa-utils-alsa-compile.sh --driver --kmodules'
by Dieter Sauerwein 15 Feb '16

15 Feb '16

File environment.base is cached. > cd /tmp/alsa-compile-script File environment.compile is cached. alsa-driver tree alsa-driver-1.0.25.3584.gb044d is present. Reusing it. Use './alsa-utils-alsa-compile.sh --clean=alsa-driver' command to refetch and rebuild. Sources unpacked to alsa-driver-1.0.25.3584.gb044d > cd alsa-driver-1.0.25.3584.gb044d Running make: make dep make[1]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include' make -C sound prepare make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/sound' make prepare2 make[4]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/sound' make[4]: Nothing to be done for 'prepare2'. make[4]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/sound' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/sound' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/ioctl32' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/ioctl32' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/oss' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/oss' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/seq' make[4]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/seq/oss' make[4]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/seq/oss' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/seq' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/i2c' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/i2c/l3' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/i2c/l3' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/i2c/other' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/i2c/other' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/i2c' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/mpu401' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/mpu401' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/opl3' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/opl3' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/opl4' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/opl4' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/pcsp' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/pcsp' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/vx' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers/vx' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/drivers' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/ad1816a' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/ad1816a' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/ad1848' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/ad1848' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/cs423x' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/cs423x' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/es1688' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/es1688' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/galaxy' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/galaxy' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/gus' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/gus' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/msnd' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/msnd' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/opti9xx' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/opti9xx' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/sb' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/sb' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/wavefront' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/wavefront' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/wss' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa/wss' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/isa' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/synth' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/synth/emux' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/synth/emux' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/synth' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ac97' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ac97' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ali5451' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ali5451' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/asihpi' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/asihpi' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/au88x0' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/au88x0' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/aw2' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/aw2' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ca0106' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ca0106' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/cs46xx' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/cs46xx' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/cs5535audio' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/cs5535audio' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ctxfi' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ctxfi' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/echoaudio' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/echoaudio' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/emu10k1' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/emu10k1' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ice1712' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ice1712' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/korg1212' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/korg1212' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/lola' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/lola' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/lx6464es' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/lx6464es' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/mixart' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/mixart' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/nm256' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/nm256' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/oxygen' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/oxygen' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/pcxhr' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/pcxhr' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/pdplus' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/pdplus' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/riptide' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/riptide' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/rme9652' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/rme9652' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/trident' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/trident' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/vx222' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/vx222' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ymfpci' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/ymfpci' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/codecs' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/codecs' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/core' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/core' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/fabrics' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/fabrics' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/soundbus' make[4]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/soundbus/i2sbus' make[4]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/soundbus/i2sbus' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa/soundbus' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/aoa' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/atmel' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/atmel' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/au1x' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/au1x' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/blackfin' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/blackfin' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/cirrus' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/cirrus' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/codecs' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/codecs' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/davinci' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/davinci' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/dwc' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/dwc' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/fsl' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/fsl' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/generic' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/generic' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/jz4740' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/jz4740' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/kirkwood' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/kirkwood' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/mid-x86' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/mid-x86' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/mxs' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/mxs' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/nuc900' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/nuc900' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/omap' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/omap' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/pxa' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/pxa' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/s6000' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/s6000' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/samsung' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/samsung' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/sh' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/sh' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/spear' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/spear' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/tegra' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/tegra' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/txx9' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/txx9' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/ux500' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc/ux500' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/soc' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/6fire' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/6fire' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/caiaq' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/caiaq' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/hiface' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/hiface' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/misc' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/misc' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/usx2y' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb/usx2y' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/usb' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pcmcia' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pcmcia/pdaudiocf' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pcmcia/pdaudiocf' make[3]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pcmcia/vx' make[3]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pcmcia/vx' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pcmcia' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/misc' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/misc' make[2]: Entering directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/firewire' make[2]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/firewire' make[1]: Leaving directory '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d' make -C /lib/modules/3.16.0-4-amd64/build SUBDIRS=/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d CPP="gcc -E" CC="gcc" modules make[1]: Entering directory '/usr/src/linux-headers-3.16.0-4-amd64' Makefile:10: *** mixed implicit and normal rules: deprecated syntax make[1]: Entering directory `/usr/src/linux-headers-3.16.0-4-amd64' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.c: In function ‘snd_info_version_read’: /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.c:1065:22: error: macro "__DATE__" might prevent reproducible builds [-Werror=date-time] "Compiled on " __DATE__ " for kernel %s" ^ cc1: some warnings being treated as errors /usr/src/linux-headers-3.16.0-4-common/scripts/Makefile.build:262: recipe for target '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o' failed make[5]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o] Error 1 /usr/src/linux-headers-3.16.0-4-common/scripts/Makefile.build:409: recipe for target '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore' failed make[4]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore] Error 2 /usr/src/linux-headers-3.16.0-4-common/Makefile:1352: recipe for target '_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d' failed make[3]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d] Error 2 Makefile:181: recipe for target 'sub-make' failed make[2]: *** [sub-make] Error 2 Makefile:8: recipe for target 'all' failed make[1]: *** [all] Error 2 make[1]: Leaving directory '/usr/src/linux-headers-3.16.0-4-amd64' Makefile:167: recipe for target 'compile' failed make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list.

1 0

[alsa-devel] [PATCH] ALSA: hda - Cancel probe work instead of flush at remove
by Takashi Iwai 15 Feb '16

15 Feb '16

The commit [991f86d7ae4e: ALSA: hda - Flush the pending probe work at remove] introduced the sync of async probe work at remove for fixing the race. However, this may lead to another hangup when the module removal is performed quickly before starting the probe work, because it issues flush_work() and it's blocked forever. The workaround is to use cancel_work_sync() instead of flush_work() there. Fixes: 991f86d7ae4e ('ALSA: hda - Flush the pending probe work at remove') Cc: <stable(a)vger.kernel.org> # v3.17+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4045dca3d699..ce6b97f31390 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2168,10 +2168,10 @@ static void azx_remove(struct pci_dev *pci) struct hda_intel *hda; if (card) { - /* flush the pending probing work */ + /* cancel the pending probing work */ chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - flush_work(&hda->probe_work); + cancel_work_sync(&hda->probe_work); snd_card_free(card); } -- 2.7.1

1 0

[alsa-devel] sound: memory leak in snd_seq_pool_init
by Dmitry Vyukov 15 Feb '16

15 Feb '16

Hello, The following program causes memory leak of a 24K object allocated in snd_seq_pool_init: // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> long r[148]; void* thr(void* arg) { switch ((long)arg) { case 0: r[0] = syscall(SYS_mmap, 0x20000000ul, 0xe000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 1: r[2] = syscall(SYS_open, "/dev/snd/seq", 0x1ul, 0, 0, 0); break; case 2: *(uint32_t*)0x20008a90 = (uint32_t)0x6; *(uint32_t*)0x20008a94 = (uint32_t)0x4; *(uint32_t*)0x20008a98 = (uint32_t)0x200; *(uint8_t*)0x20008a9c = (uint8_t)0xfa5; *(uint8_t*)0x20008a9d = (uint8_t)0x100000000; *(uint8_t*)0x20008a9e = (uint8_t)0x0; *(uint8_t*)0x20008a9f = (uint8_t)0x3; *(uint8_t*)0x20008aa0 = (uint8_t)0x1; *(uint8_t*)0x20008aa1 = (uint8_t)0x1; *(uint8_t*)0x20008aa2 = (uint8_t)0x0; *(uint8_t*)0x20008aa3 = (uint8_t)0x40; *(uint8_t*)0x20008aa4 = (uint8_t)0x9; *(uint8_t*)0x20008aa5 = (uint8_t)0x9; *(uint8_t*)0x20008aa6 = (uint8_t)0x6; *(uint8_t*)0x20008aa7 = (uint8_t)0x80; *(uint8_t*)0x20008aa8 = (uint8_t)0x0; *(uint8_t*)0x20008aa9 = (uint8_t)0x5d34b2c3; *(uint8_t*)0x20008aaa = (uint8_t)0x3a7; *(uint8_t*)0x20008aab = (uint8_t)0x9; *(uint8_t*)0x20008aac = (uint8_t)0x8d62; *(uint8_t*)0x20008aad = (uint8_t)0x5; *(uint8_t*)0x20008aae = (uint8_t)0x7; *(uint8_t*)0x20008aaf = (uint8_t)0x20; *(uint8_t*)0x20008ab0 = (uint8_t)0x8; *(uint8_t*)0x20008ab1 = (uint8_t)0x7; *(uint8_t*)0x20008ab2 = (uint8_t)0xfffffffffffffffc; *(uint8_t*)0x20008ab3 = (uint8_t)0x7; *(uint8_t*)0x20008ab4 = (uint8_t)0x7ff; *(uint8_t*)0x20008ab5 = (uint8_t)0x1; *(uint8_t*)0x20008ab6 = (uint8_t)0x0; *(uint8_t*)0x20008ab7 = (uint8_t)0x9; *(uint8_t*)0x20008ab8 = (uint8_t)0x6; *(uint8_t*)0x20008ab9 = (uint8_t)0xfffffffffffffe00; *(uint8_t*)0x20008aba = (uint8_t)0x1000; *(uint8_t*)0x20008abb = (uint8_t)0xfffffffffffffffb; *(uint8_t*)0x20008abc = (uint8_t)0xcfd3; *(uint8_t*)0x20008abd = (uint8_t)0x101; *(uint8_t*)0x20008abe = (uint8_t)0x9; *(uint8_t*)0x20008abf = (uint8_t)0xfffffffffffffffc; *(uint8_t*)0x20008ac0 = (uint8_t)0xf2; *(uint8_t*)0x20008ac1 = (uint8_t)0x6; *(uint8_t*)0x20008ac2 = (uint8_t)0x0; *(uint8_t*)0x20008ac3 = (uint8_t)0x7fffffff; *(uint8_t*)0x20008ac4 = (uint8_t)0x4; *(uint8_t*)0x20008ac5 = (uint8_t)0x5; *(uint8_t*)0x20008ac6 = (uint8_t)0x4; *(uint8_t*)0x20008ac7 = (uint8_t)0x4; *(uint8_t*)0x20008ac8 = (uint8_t)0x20; *(uint8_t*)0x20008ac9 = (uint8_t)0x401; *(uint8_t*)0x20008aca = (uint8_t)0x6; *(uint8_t*)0x20008acb = (uint8_t)0x6; *(uint8_t*)0x20008acc = (uint8_t)0x5; *(uint8_t*)0x20008acd = (uint8_t)0x3; *(uint8_t*)0x20008ace = (uint8_t)0xfffffffffffffffc; *(uint8_t*)0x20008acf = (uint8_t)0xb0c; *(uint8_t*)0x20008ad0 = (uint8_t)0x2; *(uint8_t*)0x20008ad1 = (uint8_t)0xfffffffffffffff9; *(uint8_t*)0x20008ad2 = (uint8_t)0x8000; *(uint8_t*)0x20008ad3 = (uint8_t)0x5; *(uint8_t*)0x20008ad4 = (uint8_t)0xb0c; *(uint8_t*)0x20008ad5 = (uint8_t)0xfffffffffffffeff; *(uint8_t*)0x20008ad6 = (uint8_t)0x5; *(uint8_t*)0x20008ad7 = (uint8_t)0x101; *(uint8_t*)0x20008ad8 = (uint8_t)0xbfd; *(uint8_t*)0x20008ad9 = (uint8_t)0x7ff; *(uint8_t*)0x20008ada = (uint8_t)0x7f; *(uint8_t*)0x20008adb = (uint8_t)0x5; *(uint32_t*)0x20008adc = (uint32_t)0x6; *(uint8_t*)0x20008ae0 = (uint8_t)0x0; *(uint8_t*)0x20008ae1 = (uint8_t)0x0; *(uint8_t*)0x20008ae2 = (uint8_t)0x0; *(uint8_t*)0x20008ae3 = (uint8_t)0x0; *(uint8_t*)0x20008ae4 = (uint8_t)0x0; *(uint8_t*)0x20008ae5 = (uint8_t)0x0; *(uint8_t*)0x20008ae6 = (uint8_t)0x0; *(uint8_t*)0x20008ae7 = (uint8_t)0x0; *(uint8_t*)0x20008ae8 = (uint8_t)0x0; *(uint8_t*)0x20008ae9 = (uint8_t)0x0; *(uint8_t*)0x20008aea = (uint8_t)0x0; *(uint8_t*)0x20008aeb = (uint8_t)0x0; *(uint8_t*)0x20008aec = (uint8_t)0x0; *(uint8_t*)0x20008aed = (uint8_t)0x0; *(uint8_t*)0x20008aee = (uint8_t)0x0; *(uint8_t*)0x20008aef = (uint8_t)0x0; *(uint8_t*)0x20008af0 = (uint8_t)0x0; *(uint8_t*)0x20008af1 = (uint8_t)0x0; *(uint8_t*)0x20008af2 = (uint8_t)0x0; *(uint8_t*)0x20008af3 = (uint8_t)0x0; *(uint8_t*)0x20008af4 = (uint8_t)0x0; *(uint8_t*)0x20008af5 = (uint8_t)0x0; *(uint8_t*)0x20008af6 = (uint8_t)0x0; *(uint8_t*)0x20008af7 = (uint8_t)0x0; *(uint8_t*)0x20008af8 = (uint8_t)0x0; *(uint8_t*)0x20008af9 = (uint8_t)0x0; *(uint8_t*)0x20008afa = (uint8_t)0x0; *(uint8_t*)0x20008afb = (uint8_t)0x0; *(uint8_t*)0x20008afc = (uint8_t)0x0; *(uint8_t*)0x20008afd = (uint8_t)0x0; *(uint8_t*)0x20008afe = (uint8_t)0x0; *(uint8_t*)0x20008aff = (uint8_t)0x0; *(uint8_t*)0x20008b00 = (uint8_t)0x0; *(uint8_t*)0x20008b01 = (uint8_t)0x0; *(uint8_t*)0x20008b02 = (uint8_t)0x0; *(uint8_t*)0x20008b03 = (uint8_t)0x0; *(uint8_t*)0x20008b04 = (uint8_t)0x0; *(uint8_t*)0x20008b05 = (uint8_t)0x0; *(uint8_t*)0x20008b06 = (uint8_t)0x0; *(uint8_t*)0x20008b07 = (uint8_t)0x0; *(uint8_t*)0x20008b08 = (uint8_t)0x0; *(uint8_t*)0x20008b09 = (uint8_t)0x0; *(uint8_t*)0x20008b0a = (uint8_t)0x0; *(uint8_t*)0x20008b0b = (uint8_t)0x0; *(uint8_t*)0x20008b0c = (uint8_t)0x0; *(uint8_t*)0x20008b0d = (uint8_t)0x0; *(uint8_t*)0x20008b0e = (uint8_t)0x0; *(uint8_t*)0x20008b0f = (uint8_t)0x0; *(uint8_t*)0x20008b10 = (uint8_t)0x0; *(uint8_t*)0x20008b11 = (uint8_t)0x0; *(uint8_t*)0x20008b12 = (uint8_t)0x0; *(uint8_t*)0x20008b13 = (uint8_t)0x0; *(uint8_t*)0x20008b14 = (uint8_t)0x0; *(uint8_t*)0x20008b15 = (uint8_t)0x0; *(uint8_t*)0x20008b16 = (uint8_t)0x0; *(uint8_t*)0x20008b17 = (uint8_t)0x0; *(uint8_t*)0x20008b18 = (uint8_t)0x0; *(uint8_t*)0x20008b19 = (uint8_t)0x0; *(uint8_t*)0x20008b1a = (uint8_t)0x0; *(uint8_t*)0x20008b1b = (uint8_t)0x0; r[131] = syscall(SYS_ioctl, r[2], 0xc08c5332ul, 0x20008a90ul, 0, 0, 0); break; case 3: *(uint32_t*)0x20009fd0 = (uint32_t)0x30; *(uint32_t*)0x20009fd4 = (uint32_t)0xfff; *(uint64_t*)0x20009fd8 = (uint64_t)0x0; *(uint64_t*)0x20009fe0 = (uint64_t)0x1ce; *(uint64_t*)0x20009fe8 = (uint64_t)0x2; *(uint64_t*)0x20009ff0 = (uint64_t)0xff; *(uint32_t*)0x20009ff8 = (uint32_t)0x10001; r[139] = syscall(SYS_write, r[2], 0x20009fd0ul, 0x30ul, 0, 0, 0); break; case 4: *(uint32_t*)0x2000bb13 = (uint32_t)0x30; *(uint32_t*)0x2000bb17 = (uint32_t)0x2; *(uint64_t*)0x2000bb1b = (uint64_t)0x0; *(uint64_t*)0x2000bb23 = (uint64_t)0x53; *(uint64_t*)0x2000bb2b = (uint64_t)0xfffffffffffffbff; *(uint64_t*)0x2000bb33 = (uint64_t)0x5; *(uint32_t*)0x2000bb3b = (uint32_t)0x5; r[147] = syscall(SYS_write, r[2], 0x2000bb13ul, 0x30ul, 0, 0, 0); break; } return 0; } int main() { long i; pthread_t th[5]; memset(r, -1, sizeof(r)); for (i = 0; i < 5; i++) { pthread_create(&th[i], 0, thr, (void*)i); usleep(10000); } for (i = 0; i < 5; i++) { pthread_create(&th[i], 0, thr, (void*)i); if (i % 2 == 0) usleep(10000); } usleep(100000); return 0; } unreferenced object 0xffffc90003663000 (size 24000): comm "a.out", pid 1770, jiffies 4295717383 (age 7.217s) hex dump (first 32 bytes): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ backtrace: [<ffffffff86644303>] kmemleak_alloc+0x63/0xa0 mm/kmemleak.c:916 [<ffffffff81720da9>] __vmalloc_node_range+0x469/0x640 mm/vmalloc.c:1682 [< inline >] __vmalloc_node mm/vmalloc.c:1710 [< inline >] __vmalloc_node_flags mm/vmalloc.c:1724 [<ffffffff8172104b>] vmalloc+0x5b/0x70 mm/vmalloc.c:1739 [<ffffffff852b92cb>] snd_seq_pool_init+0xbb/0x300 sound/core/seq/seq_memory.c:389 [<ffffffff852b5ff4>] snd_seq_write+0x4d4/0x570 sound/core/seq/seq_clientmgr.c:1029 [<ffffffff817bbfc3>] __vfs_write+0x113/0x4b0 fs/read_write.c:528 [<ffffffff817bda87>] vfs_write+0x167/0x4a0 fs/read_write.c:577 [< inline >] SYSC_write fs/read_write.c:624 [<ffffffff817c0d71>] SyS_write+0x111/0x220 fs/read_write.c:616 [<ffffffff866613f6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 [<ffffffffffffffff>] 0xffffffffffffffff On commit 388f7b1d6e8ca06762e2454d28d6c3c55ad0fe95 (4.5-rc3)

2 3

[alsa-devel] sound: spinlock lockup in snd_seq_oss_write
by Dmitry Vyukov 15 Feb '16

15 Feb '16

Hello, The following program hangs machine: https://gist.githubusercontent.com/dvyukov/1c9eeb67414a9e7c5e52/raw/127b660… BUG: spinlock lockup suspected on CPU#2, a.out/7470 lock: 0xffff880036a9ef50, .magic: dead4ead, .owner: a.out/7466, .owner_cpu: 3 CPU: 2 PID: 7470 Comm: a.out Not tainted 4.5.0-rc2+ #319 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff88005f5ff660 ffffffff82be30ad ffff880036a9ef50 ffff88005d3d5f00 ffff88005d0b4740 ffff88005f5ff698 ffffffff8146768d ffffffff85241dce ffff880000000000 ffff880036a9ef50 ffff880036a9ef60 Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82be30ad>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 [<ffffffff8146768d>] spin_dump+0x14d/0x280 kernel/locking/spinlock_debug.c:67 [< inline >] __spin_lock_debug kernel/locking/spinlock_debug.c:117 [<ffffffff814679ad>] do_raw_spin_lock+0x15d/0x2b0 kernel/locking/spinlock_debug.c:137 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:119 [<ffffffff8665ea57>] _raw_spin_lock_irqsave+0xa7/0xd0 kernel/locking/spinlock.c:159 [<ffffffff85241dce>] snd_timer_start+0x10e/0x2b0 sound/core/timer.c:491 [< inline >] seq_timer_start sound/core/seq/seq_timer.c:393 [<ffffffff852bf860>] snd_seq_timer_start+0x1a0/0x2b0 sound/core/seq/seq_timer.c:405 [< inline >] snd_seq_queue_process_event sound/core/seq/seq_queue.c:687 [<ffffffff852ba514>] snd_seq_control_queue+0x304/0x8b0 sound/core/seq/seq_queue.c:748 [<ffffffff852c01b5>] event_input_timer+0x25/0x30 sound/core/seq/seq_system.c:118 [<ffffffff852b2de4>] snd_seq_deliver_single_event.constprop.11+0x3f4/0x740 sound/core/seq/seq_clientmgr.c:634 [<ffffffff852b3248>] snd_seq_deliver_event+0x118/0x800 sound/core/seq/seq_clientmgr.c:831 [<ffffffff852b4636>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2404 [<ffffffff852c92eb>] send_timer_event.isra.0+0x10b/0x150 sound/core/seq/oss/seq_oss_timer.c:153 [<ffffffff852c981a>] snd_seq_oss_timer_start+0x1ca/0x310 sound/core/seq/oss/seq_oss_timer.c:174 [<ffffffff852c9c04>] snd_seq_oss_process_timer_event+0x2a4/0x340 sound/core/seq/oss/seq_oss_timer.c:106 [< inline >] insert_queue sound/core/seq/oss/seq_oss_rw.c:170 [<ffffffff852cf7ab>] snd_seq_oss_write+0x20b/0x810 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff852c74f9>] odev_write+0x59/0xa0 sound/core/seq/oss/seq_oss.c:179 [<ffffffff817bacd3>] __vfs_write+0x113/0x4b0 fs/read_write.c:528 [<ffffffff817bc797>] vfs_write+0x167/0x4a0 fs/read_write.c:577 [< inline >] SYSC_write fs/read_write.c:624 [<ffffffff817bfa81>] SyS_write+0x111/0x220 fs/read_write.c:616 [<ffffffff8665f0b6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 Sending NMI to all CPUs: NMI backtrace for cpu 0 CPU: 0 PID: 7473 Comm: a.out Not tainted 4.5.0-rc2+ #319 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88003376df00 ti: ffff880032440000 task.ti: ffff880032440000 RIP: 0010:[<ffffffff82c103a7>] [<ffffffff82c103a7>] delay_tsc+0x27/0x70 RSP: 0018:ffff8800324479f8 EFLAGS: 00000006 RAX: 00000000a0066ab3 RBX: ffff880036a9ef50 RCX: 000000000000001e RDX: 0000000000000100 RSI: 00000100a0066a95 RDI: 0000000000000001 RBP: ffff8800324479f8 R08: 0000000000000000 R09: 0000000000000001 R10: ffff88003376df00 R11: ffff880036a9ef68 R12: ffff880036a9ef60 R13: 000000009a9d2d40 R14: ffff880036a9ef58 R15: 0000000099db9d6d FS: 00007f94a370e700(0000) GS:ffff88003ec00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 000000c82010e120 CR3: 000000003334f000 CR4: 00000000000006f0 Stack: ffff880032447a08 ffffffff82c102ea ffff880032447a40 ffffffff81467999 ffff880036a9ef50 ffff880036a9ee20 0000000000000282 ffff880035d8ab70 0000000000000000 ffff880032447a60 ffffffff8665dfbb ffffffff8523dd64 Call Trace: [<ffffffff82c102ea>] __delay+0xa/0x10 arch/x86/lib/delay.c:153 [< inline >] __spin_lock_debug kernel/locking/spinlock_debug.c:114 [<ffffffff81467999>] do_raw_spin_lock+0x149/0x2b0 kernel/locking/spinlock_debug.c:137 [< inline >] __raw_spin_lock include/linux/spinlock_api_smp.h:145 [<ffffffff8665dfbb>] _raw_spin_lock+0x3b/0x50 kernel/locking/spinlock.c:151 [< inline >] spin_lock include/linux/spinlock.h:302 [<ffffffff8523dd64>] snd_timer_start_slave+0xd4/0x200 sound/core/timer.c:460 [<ffffffff85241880>] snd_timer_continue+0x200/0x290 sound/core/timer.c:598 [< inline >] snd_timer_user_continue sound/core/timer.c:1822 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1889 [<ffffffff85246a9d>] snd_timer_user_ioctl+0xe1d/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [<ffffffff817faf7c>] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [<ffffffff817fbe2f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [<ffffffff8665f0b6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 Code: 00 00 00 00 55 65 ff 05 c8 4a 40 7d 48 89 e5 65 44 8b 05 8d 2d 40 7d 0f ae e8 0f 31 48 c1 e2 20 48 89 d6 48 09 c6 0f ae e8 0f 31 <48> c1 e2 20 48 09 d0 48 89 c1 48 29 f1 48 39 f9 73 27 65 ff 0d NMI backtrace for cpu 1 CPU: 1 PID: 7421 Comm: stress Not tainted 4.5.0-rc2+ #319 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88003330df00 ti: ffff880032230000 task.ti: ffff880032230000 RIP: 0010:[<ffffffff82c103a7>] [<ffffffff82c103a7>] delay_tsc+0x27/0x70 RSP: 0018:ffff88003ed07c80 EFLAGS: 00000006 RAX: 00000000a0066d38 RBX: ffff880036a9ef50 RCX: 000000000000001e RDX: 0000000000000100 RSI: 00000100a0066d0f RDI: 0000000000000001 RBP: ffff88003ed07c80 R08: 0000000000000001 R09: 0000000000000001 R10: ffff88003330df00 R11: ffff880036a9ef68 R12: ffff880036a9ef60 R13: 000000009a9d2d40 R14: ffff880036a9ef58 R15: 000000009a82fe90 FS: 000000c820127700(0000) GS:ffff88003ed00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 000000c8201004e0 CR3: 0000000032076000 CR4: 00000000000006e0 Stack: ffff88003ed07c90 ffffffff82c102ea ffff88003ed07cc8 ffffffff81467999 ffff880036a9ef50 0000000000000082 ffff880036a9ee20 00000000000f4240 ffff880032795068 ffff88003ed07cf0 ffffffff8665ea57 ffffffff85242019 Call Trace: d [<ffffffff82c102ea>] __delay+0xa/0x10 arch/x86/lib/delay.c:153 [< inline >] __spin_lock_debug kernel/locking/spinlock_debug.c:114 [<ffffffff81467999>] do_raw_spin_lock+0x149/0x2b0 kernel/locking/spinlock_debug.c:137 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:119 [<ffffffff8665ea57>] _raw_spin_lock_irqsave+0xa7/0xd0 kernel/locking/spinlock.c:159 [<ffffffff85242019>] snd_timer_interrupt+0xa9/0xc80 sound/core/timer.c:718 [<ffffffff85248cb9>] snd_hrtimer_callback+0x169/0x230 sound/core/hrtimer.c:54 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1248 [<ffffffff814b7dc3>] __hrtimer_run_queues+0x363/0xc20 kernel/time/hrtimer.c:1312 [<ffffffff814b9de2>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1346 [<ffffffff8124e1d2>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff812516f9>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff8665feac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:520 [<ffffffff817ab804>] create_object+0x34/0x2d0 mm/kmemleak.c:546 [<ffffffff86641fb3>] kmemleak_alloc+0x63/0xa0 mm/kmemleak.c:916 [< inline >] kmemleak_alloc_recursive include/linux/kmemleak.h:47 [< inline >] slab_post_alloc_hook mm/slub.c:1337 [< inline >] slab_alloc_node mm/slub.c:2596 [< inline >] slab_alloc mm/slub.c:2604 [<ffffffff81760843>] kmem_cache_alloc+0x153/0x2e0 mm/slub.c:2609 [<ffffffff8181b1b5>] dup_fd+0x85/0xa90 fs/file.c:298 [< inline >] copy_files kernel/fork.c:1033 [<ffffffff8134efed>] copy_process.part.35+0x188d/0x57a0 kernel/fork.c:1436 [< inline >] copy_process kernel/fork.c:1274 [<ffffffff8135322c>] _do_fork+0x1bc/0xcb0 kernel/fork.c:1723 [< inline >] SYSC_clone kernel/fork.c:1832 [<ffffffff81353df7>] SyS_clone+0x37/0x50 kernel/fork.c:1826 [<ffffffff8665f0b6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 Code: 00 00 00 00 55 65 ff 05 c8 4a 40 7d 48 89 e5 65 44 8b 05 8d 2d 40 7d 0f ae e8 0f 31 48 c1 e2 20 48 89 d6 48 09 c6 0f ae e8 0f 31 <48> c1 e2 20 48 09 d0 48 89 c1 48 29 f1 48 39 f9 73 27 65 ff 0d NMI backtrace for cpu 2 CPU: 2 PID: 7470 Comm: a.out Not tainted 4.5.0-rc2+ #319 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88005d0b4740 ti: ffff88005f5f8000 task.ti: ffff88005f5f8000 RIP: 0010:[<ffffffff81262066>] [<ffffffff81262066>] flat_send_IPI_mask+0x156/0x290 RSP: 0018:ffff88005f5ff5d0 EFLAGS: 00000046 RAX: 0000000000000000 RBX: 0000000000000c00 RCX: 0000000000000000 RDX: 0000000000000c00 RSI: 0000000000000002 RDI: ffffffffff5fc300 RBP: ffff88005f5ff5f8 R08: 0000000000000001 R09: 0000000000000000 R10: fffffbfff11c14aa R11: 1ffffffff13552ed R12: 0000000000000082 BUG: spinlock lockup suspected on CPU#1, stress/7421 lock: 0xffff880036a9ef50, .magic: dead4ead, .owner: a.out/7466, .owner_cpu: 3 CPU: 1 PID: 7421 Comm: stress Not tainted 4.5.0-rc2+ #319 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff88003ed07c58 ffffffff82be30ad ffff880036a9ef50 ffff88005d3d5f00 ffff88003330df00 ffff88003ed07c90 ffffffff8146768d ffffffff85242019 0000000000000000 ffff880036a9ef50 ffff880036a9ef60 Call Trace: <IRQ> [< inline >] __dump_stack lib/dump_stack.c:15 <IRQ> [<ffffffff82be30ad>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 [<ffffffff8146768d>] spin_dump+0x14d/0x280 kernel/locking/spinlock_debug.c:67 [< inline >] __spin_lock_debug kernel/locking/spinlock_debug.c:117 [<ffffffff814679ad>] do_raw_spin_lock+0x15d/0x2b0 kernel/locking/spinlock_debug.c:137 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:119 [<ffffffff8665ea57>] _raw_spin_lock_irqsave+0xa7/0xd0 kernel/locking/spinlock.c:159 [<ffffffff85242019>] snd_timer_interrupt+0xa9/0xc80 sound/core/timer.c:718 [<ffffffff85248cb9>] snd_hrtimer_callback+0x169/0x230 sound/core/hrtimer.c:54 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1248 [<ffffffff814b7dc3>] __hrtimer_run_queues+0x363/0xc20 kernel/time/hrtimer.c:1312 [<ffffffff814b9de2>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1346 [<ffffffff8124e1d2>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff812516f9>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff8665feac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:520 [<ffffffff817ab804>] create_object+0x34/0x2d0 mm/kmemleak.c:546 [<ffffffff86641fb3>] kmemleak_alloc+0x63/0xa0 mm/kmemleak.c:916 [< inline >] kmemleak_alloc_recursive include/linux/kmemleak.h:47 [< inline >] slab_post_alloc_hook mm/slub.c:1337 [< inline >] slab_alloc_node mm/slub.c:2596 [< inline >] slab_alloc mm/slub.c:2604 [<ffffffff81760843>] kmem_cache_alloc+0x153/0x2e0 mm/slub.c:2609 [<ffffffff8181b1b5>] dup_fd+0x85/0xa90 fs/file.c:298 [< inline >] copy_files kernel/fork.c:1033 [<ffffffff8134efed>] copy_process.part.35+0x188d/0x57a0 kernel/fork.c:1436 [< inline >] copy_process kernel/fork.c:1274 [<ffffffff8135322c>] _do_fork+0x1bc/0xcb0 kernel/fork.c:1723 [< inline >] SYSC_clone kernel/fork.c:1832 [<ffffffff81353df7>] SyS_clone+0x37/0x50 kernel/fork.c:1826 [<ffffffff8665f0b6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 R13: 000000000f000000 R14: ffffffff87b00c80 R15: 0000000000000002 FS: 00007fcabc427700(0000) GS:ffff88006d600000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 0000000020028073 CR3: 000000005c829000 CR4: 00000000000006e0 Stack: ffffffff87b00c80 ffffffff888be0a0 0000000000000007 fffffbfff111761c ffff88006d71a540 ffff88005f5ff618 ffffffff81257bfe ffffffff86ac7020 dffffc0000000000 ffff88005f5ff688 ffffffff82bee6ac ffffffff811aa610 Call Trace: [<ffffffff81257bfe>] nmi_raise_cpu_backtrace+0x5e/0x80 arch/x86/kernel/apic/hw_nmi.c:33 [<ffffffff82bee6ac>] nmi_trigger_all_cpu_backtrace+0x54c/0x5f0 lib/nmi_backtrace.c:85 [<ffffffff81257c54>] arch_trigger_all_cpu_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 [< inline >] trigger_all_cpu_backtrace include/linux/nmi.h:41 [< inline >] __spin_lock_debug kernel/locking/spinlock_debug.c:119 [<ffffffff814679b7>] do_raw_spin_lock+0x167/0x2b0 kernel/locking/spinlock_debug.c:137 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:119 [<ffffffff8665ea57>] _raw_spin_lock_irqsave+0xa7/0xd0 kernel/locking/spinlock.c:159 [<ffffffff85241dce>] snd_timer_start+0x10e/0x2b0 sound/core/timer.c:491 [< inline >] seq_timer_start sound/core/seq/seq_timer.c:393 [<ffffffff852bf860>] snd_seq_timer_start+0x1a0/0x2b0 sound/core/seq/seq_timer.c:405 [< inline >] snd_seq_queue_process_event sound/core/seq/seq_queue.c:687 [<ffffffff852ba514>] snd_seq_control_queue+0x304/0x8b0 sound/core/seq/seq_queue.c:748 [<ffffffff852c01b5>] event_input_timer+0x25/0x30 sound/core/seq/seq_system.c:118 [<ffffffff852b2de4>] snd_seq_deliver_single_event.constprop.11+0x3f4/0x740 sound/core/seq/seq_clientmgr.c:634 [<ffffffff852b3248>] snd_seq_deliver_event+0x118/0x800 sound/core/seq/seq_clientmgr.c:831 On commit b37a05c083c85c2657dca9bbe1f5d79dccf756d5 + latest Takashi fixes.

2 2

[alsa-devel] SGTL5000 startup error on reboot
by Erik Friesen 15 Feb '16

15 Feb '16

I find that the codec is getting a write failure at sgtl5000_fill_defaults at { SGTL5000_CHIP_ANA_POWER, 0x7060 }, about half the time following a software reboot, and with external vddd powered with 1.8v. Reading this register shows that it is already at 0x7060, so I am not sure what is up with that, but I would suspect a hardware bug in the codec. Using this code has fixed this so far. static int sgtl5000_fill_defaults(struct sgtl5000_priv *sgtl5000) { int i, ret, val, index; int reg; for (i = 0; i < ARRAY_SIZE(sgtl5000_reg_defaults); i++) { val = sgtl5000_reg_defaults[i].def; index = sgtl5000_reg_defaults[i].reg; ret = regmap_read(sgtl5000->regmap, index, &reg); if (ret) { printk(KERN_WARNING "SGTL5000 probe read failed at %i\n", i); return ret; } if (reg != val) { ret = regmap_write(sgtl5000->regmap, index, val); } if (ret) { printk(KERN_WARNING "SGTL5000 probe write failed %x at %i index %i @ %x\n", val, i, index, reg); return ret; } } return 0; }

1 1

[alsa-devel] kernel crash in snd_hda_intel
by Gabriel Feceoru 15 Feb '16

15 Feb '16

!!! This caused a regression in the i-g-t drv_module_reload_basic test. Reproducible easily on HSW (i5-4460) with: #rmmod snd_hda_intel The bisect shows this as the offending commit: commit 25e4abb33df3aafa7d1efba8f82f9178268efab1 Author: Libin Yang <libin.yang(a)linux.intel.com> Date: Tue Jan 12 11:13:27 2016 +0800 ALSA: hda - hdmi jack created based on pcm Jack is created based on pcm. Apply the acomp jack rule to dyn_pcm_assign. For dyn_pcm_assign: Driver does not use hda_jack. It operates snd_jack directly. snd_jack pointer will be stored in spec->pcm.jack instead of the current spec->acomp_jack. When pcm is assigned to pin, jack will be assigned to pin automatically. For !dyn_pcm_assign: Driver continues using hda_jack for less impact on the old cases. Pcm is statically assigned to pin. So is jack. spec->pcm.jack saves the snd_jack pointer created in hda_jack. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> [ 79.020523] BUG: unable to handle kernel paging request at 0000000000015d80 [ 79.021314] IP: [<ffffffff810a486b>] queued_spin_lock_slowpath+0xeb/0x180 [ 79.022125] PGD 0 [ 79.022881] Oops: 0002 [#1] SMP [ 79.023644] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel(-) snd_hda_codec x86_pkg_temp_thermal snd_hwdep snd_hda_core i915 i2c_algo_bit drm_kms_helper syscopyarea sysfillrect sysimgblt ehci_pci fb_sys_fops r8169 ehci_hcd mii drm xhci_pci xhci_hcd video [ 79.026790] CPU: 0 PID: 1294 Comm: rmmod Tainted: G U 4.5.0-rc1+ #73 [ 79.028322] Hardware name: Dell Inc. Inspiron 3847/088DT1 , BIOS A06 01/15/2015 [ 79.029854] task: ffff8800dac90000 ti: ffff880212700000 task.ti: ffff880212700000 [ 79.031384] RIP: 0010:[<ffffffff810a486b>] [<ffffffff810a486b>] queued_spin_lock_slowpath+0xeb/0x180 [ 79.032924] RSP: 0018:ffff8802127039b0 EFLAGS: 00010002 [ 79.033698] RAX: 000000000000350c RBX: ffff8800d434ef90 RCX: ffff88021fa15d80 [ 79.035224] RDX: 0000000000015d80 RSI: 00000000d434ef88 RDI: ffff8800d434ef90 [ 79.036751] RBP: ffff8802127039b0 R08: 0000000000040000 R09: 0000000000000000 [ 79.038277] R10: ffff8800daef8998 R11: 0000000000000004 R12: 0000000000000282 [ 79.039802] R13: 0000000000000000 R14: 0000000000000006 R15: ffff8800d434ef90 [ 79.041331] FS: 00007f75be1ad700(0000) GS:ffff88021fa00000(0000) knlGS:0000000000000000 [ 79.042863] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 79.043638] CR2: 0000000000015d80 CR3: 00000002101b1000 CR4: 00000000001406f0 [ 79.045166] Stack: [ 79.045926] ffff8802127039d0 ffffffff81748e59 ffff8800d434ed80 0000000000000005 [ 79.047462] ffff880212703a10 ffffffff81597e73 ffffffff817472d2 0000000000000003 [ 79.048995] ffff8800d434ecc0 0000000000000000 0000000000000001 ffff8800daef8800 [ 79.050528] Call Trace: [ 79.051293] [<ffffffff81748e59>] _raw_spin_lock_irqsave+0x39/0x50 [ 79.052071] [<ffffffff81597e73>] input_event+0x43/0x80 [ 79.052844] [<ffffffff817472d2>] ? mutex_lock+0x12/0x30 [ 79.053619] [<ffffffff8160d9fe>] snd_jack_report+0xee/0x110 [ 79.054396] [<ffffffffc02da4ea>] hdmi_present_sense+0x13a/0x390 [snd_hda_codec_hdmi] [ 79.055921] [<ffffffff8150ffee>] ? regmap_unlock_mutex+0xe/0x10 [ 79.056699] [<ffffffffc02da78d>] generic_hdmi_resume+0x4d/0x60 [snd_hda_codec_hdmi] [ 79.058235] [<ffffffffc02a8bb0>] ? snd_hda_add_imux_item+0x120/0x120 [snd_hda_codec] [ 79.059771] [<ffffffffc02a85ae>] hda_call_codec_resume+0xce/0x120 [snd_hda_codec] [ 79.061303] [<ffffffffc02a8bb0>] ? snd_hda_add_imux_item+0x120/0x120 [snd_hda_codec] [ 79.062835] [<ffffffffc02a8be5>] hda_codec_runtime_resume+0x35/0x50 [snd_hda_codec] [ 79.064368] [<ffffffff81503d68>] __rpm_callback+0x28/0x70 [ 79.065143] [<ffffffffc02a8bb0>] ? snd_hda_add_imux_item+0x120/0x120 [snd_hda_codec] [ 79.066676] [<ffffffff81503dd4>] rpm_callback+0x24/0x80 [ 79.067449] [<ffffffffc02a8bb0>] ? snd_hda_add_imux_item+0x120/0x120 [snd_hda_codec] [ 79.068983] [<ffffffff81504646>] rpm_resume+0x426/0x620 [ 79.069758] [<ffffffff815055ae>] __pm_runtime_resume+0x4e/0x70 [ 79.070536] [<ffffffff814f9453>] __device_release_driver+0x43/0x160 [ 79.071314] [<ffffffff814f9593>] device_release_driver+0x23/0x30 [ 79.072085] [<ffffffff814f8b71>] bus_remove_device+0x101/0x170 [ 79.072862] [<ffffffff814f50f9>] device_del+0x139/0x270 [ 79.073638] [<ffffffffc00a51cb>] ? widget_tree_free.isra.1+0x8b/0xa0 [snd_hda_core] [ 79.075170] [<ffffffffc00a3831>] snd_hdac_device_unregister+0x21/0x30 [snd_hda_core] [ 79.076703] [<ffffffffc02a708d>] snd_hda_codec_dev_free+0x1d/0x40 [snd_hda_codec] [ 79.078235] [<ffffffff8160ac5c>] __snd_device_free+0x4c/0x80 [ 79.079010] [<ffffffff8160afb0>] snd_device_free_all+0x30/0x60 [ 79.079787] [<ffffffff816058a4>] release_card_device+0x34/0x90 [ 79.080563] [<ffffffff814f4892>] device_release+0x32/0x90 [ 79.081338] [<ffffffff813f74da>] kobject_release+0x7a/0x190 [ 79.082113] [<ffffffff813f7397>] kobject_put+0x27/0x50 [ 79.082886] [<ffffffff814f4b67>] put_device+0x17/0x20 [ 79.083659] [<ffffffff816068e8>] snd_card_free+0x58/0x70 [ 79.084434] [<ffffffffc00b3fd1>] azx_remove+0x31/0x40 [snd_hda_intel] [ 79.085214] [<ffffffff81433da9>] pci_device_remove+0x39/0xc0 [ 79.085990] [<ffffffff814f94b1>] __device_release_driver+0xa1/0x160 [ 79.086767] [<ffffffff814f9fe6>] driver_detach+0xa6/0xb0 [ 79.087540] [<ffffffff814f8eb5>] bus_remove_driver+0x55/0xd0 [ 79.088314] [<ffffffff814fa5dc>] driver_unregister+0x2c/0x50 [ 79.089084] [<ffffffff814324d1>] pci_unregister_driver+0x21/0x90 [ 79.089863] [<ffffffffc00b5d88>] azx_driver_exit+0x10/0x288 [snd_hda_intel] [ 79.090644] [<ffffffff810dde01>] SyS_delete_module+0x1b1/0x250 [ 79.091422] [<ffffffff81002017>] ? trace_hardirqs_on_thunk+0x17/0x19 [ 79.092200] [<ffffffff8174925b>] entry_SYSCALL_64_fastpath+0x16/0x6e [ 79.092977] Code: c1 e0 10 45 31 c9 85 c0 74 44 48 89 c2 c1 e8 12 48 c1 ea 0c 83 e8 01 83 e2 30 48 98 48 81 c2 80 5d 01 00 48 03 14 c5 e0 b4 d4 81 <48> 89 0a 8b 41 08 85 c0 75 09 f3 90 8b 41 08 85 c0 74 f7 4c 8b [ 79.095373] RIP [<ffffffff810a486b>] queued_spin_lock_slowpath+0xeb/0x180 [ 79.096155] RSP <ffff8802127039b0> [ 79.096920] CR2: 0000000000015d80 [ 79.097687] ---[ end trace 5594cbffe0158d7f ]--- $lspci -nn 00:00.0 Host bridge [0600]: Intel Corporation 4th Gen Core Processor DRAM Controller [8086:0c00] (rev 06) 00:01.0 PCI bridge [0604]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor PCI Express x16 Controller [8086:0c01] (rev 06) 00:02.0 VGA compatible controller [0300]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor Integrated Graphics Controller [8086:0412] (rev 06) 00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06) 00:14.0 USB controller [0c03]: Intel Corporation 8 Series/C220 Series Chipset Family USB xHCI [8086:8c31] (rev 05) 00:16.0 Communication controller [0780]: Intel Corporation 8 Series/C220 Series Chipset Family MEI Controller #1 [8086:8c3a] (rev 04) 00:1a.0 USB controller [0c03]: Intel Corporation 8 Series/C220 Series Chipset Family USB EHCI #2 [8086:8c2d] (rev 05) 00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 05) 00:1c.0 PCI bridge [0604]: Intel Corporation 8 Series/C220 Series Chipset Family PCI Express Root Port #1 [8086:8c10] (rev d5) 00:1c.3 PCI bridge [0604]: Intel Corporation 8 Series/C220 Series Chipset Family PCI Express Root Port #4 [8086:8c16] (rev d5) 00:1c.4 PCI bridge [0604]: Intel Corporation 8 Series/C220 Series Chipset Family PCI Express Root Port #5 [8086:8c18] (rev d5) 00:1d.0 USB controller [0c03]: Intel Corporation 8 Series/C220 Series Chipset Family USB EHCI #1 [8086:8c26] (rev 05) 00:1f.0 ISA bridge [0601]: Intel Corporation C220 Series Chipset Family H81 Express LPC Controller [8086:8c5c] (rev 05) 00:1f.2 SATA controller [0106]: Intel Corporation 8 Series/C220 Series Chipset Family 6-port SATA Controller 1 [AHCI mode] [8086:8c02] (rev 05) 00:1f.3 SMBus [0c05]: Intel Corporation 8 Series/C220 Series Chipset Family SMBus Controller [8086:8c22] (rev 05) 03:00.0 Ethernet controller [0200]: Realtek Semiconductor Co., Ltd. RTL8111/8168/8411 PCI Express Gigabit Ethernet Controller [10ec:8168] (rev 0c) 04:00.0 Network controller [0280]: Qualcomm Atheros QCA9565 / AR9565 Wireless Network Adapter [168c:0036] (rev 01) Regards, Gabriel

2 5

[alsa-devel] [RFC][PATCH 0/5] alsa-lib: add APIs for control element set
by Takashi Sakamoto 15 Feb '16

15 Feb '16

Hi, This patchset adds some APIs for control element set, according to design of ALSA Ctl core. In ALSA Ctl core, each sound card can have control elements. The elements are managed according to below design. * element set * A set of elements with the same attribute (i.e. name, get/put operations). * element * An element can be identified by userspace applications. * channel * An element has channels which can be changed by userspace applications or hardware device itself. * value * Each channel has its own value. The place to kept the value depends on design of each device driver (i.e. in hardware register or kernel space). There're two ways to identify the element; by identification number (numid), or by a combination of name of the element set and index from the first element in the element set. Both of device driver in kernel land and ALSA Ctl application in user land can add the element set. When userspace applications add them, ALSA Ctl core works as Inter Process Communication mechanism between event-listening applications and usual control applications. Such element is refered as 'user-defined control elements' in alsa-lib. Typically, device driver in kernel land adds one element set for a single element. So users and userspace developers are hard to realize that an existence of the element set. There're more items to describe the design. But here I focus on what relates to this patchset. Well, current alsa-lib implementation includes small bugs, some misleadings, a lack of feature related to user-defined control elements. This patchset is my attempt to improves such issues. Regards Takashi Sakamoto (5): control: initialize element channel values according to identification number control: rename variables so that it represents the number of channels which an element has control: improve API documentation control: add a function to add an element of bytes type control: add functions to add element set include/control.h | 35 +++- src/control/control.c | 484 +++++++++++++++++++++++++++++++++++++++----------- 2 files changed, 413 insertions(+), 106 deletions(-) -- 2.5.0

1 5

[alsa-devel] Plantronics DA45 cannot get freq at ep 0x4
by denk 15 Feb '16

15 Feb '16

Hi, using the Plantronics DA45 Headset Adapter leads to many lines of cannot get freq at ep 0x4 and cannot get freq at ep 0x84. I found a similar error in the following discussion: http://www.spinics.net/lists/linux-usb/msg125029.html I added the USB ID of the DA45, recompiled the kernel and the messages are gone. I would be happy if you could accept this simple addition made by the following patch: --- a/sound/usb/quirks.c 2015-12-15 06:41:43.000000000 +0100 +++ b/sound/usb/quirks.c 2016-01-28 08:15:35.207785555 +0100 @@ -1122,6 +1122,7 @@ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ return true; } return false; If there are any questions or something missing, please ask. Thanks in advance. Best denk

2 1

[alsa-devel] sound compress_driver.h: missing kernel-doc notation for a few fields
by Randy Dunlap 15 Feb '16

15 Feb '16

Hi Richard, Please add kernel-doc notation for these fields: ..//include/sound/compress_driver.h:160: warning: No description found for parameter 'id[64]' ..//include/sound/compress_driver.h:160: warning: No description found for parameter 'proc_root' ..//include/sound/compress_driver.h:160: warning: No description found for parameter 'proc_info_entry' thanks, -- ~Randy

1 0

[alsa-devel] [PATCH] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 13 Feb '16

13 Feb '16

From: Martin Koegler <martin.koegler(a)chello.at> Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 1 + include/sound/asequencer.h | 5 +++-- src/seq/seq.c | 11 +++++++++++ src/seq/seq_hw.c | 8 +++++++- 4 files changed, 22 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..77996e5 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,7 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_owner(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..7ebf7fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,8 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int owner; /* RO: card number[kernel] / PID[user] */ + char reserved[64 - sizeof(int)]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..2505293 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,17 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number (kernel) or owning PID + * \param info client_info container + * \return card number/PID/-1 if value is not available. + */ +int snd_seq_client_info_get_owner(const snd_seq_client_info_t *info) +{ + assert(info); + return info->owner; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..a4c083a 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int micro_version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,8 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) + info->owner = -1; return 0; } @@ -368,6 +371,8 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) + info->owner = -1; return 0; } @@ -480,6 +485,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->micro_version = SNDRV_PROTOCOL_MICRO(ver); if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.1.4

2 2

[alsa-devel] [PATCH] Show sequencer sound card numer/PID via aconnect
by Martin Koegler 13 Feb '16

13 Feb '16

From: Martin Koegler <martin.koegler(a)chello.at> Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- seq/aconnect/aconnect.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 8d6cebb..3508e1b 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -166,11 +166,19 @@ static void print_port(snd_seq_t *seq, snd_seq_client_info_t *cinfo, snd_seq_port_info_t *pinfo, int count) { if (! count) { - printf(_("client %d: '%s' [type=%s]\n"), + printf(_("client %d: '%s' [type=%s"), snd_seq_client_info_get_client(cinfo), snd_seq_client_info_get_name(cinfo), (snd_seq_client_info_get_type(cinfo) == SND_SEQ_USER_CLIENT ? _("user") : _("kernel"))); + int owner = snd_seq_client_info_get_owner(cinfo); + if (owner != -1) { + if (snd_seq_client_info_get_type(cinfo) == SND_SEQ_USER_CLIENT) + printf(",pid=%d", owner); + else + printf(",card=%d", owner); + } + printf("]\n"); } printf(" %3d '%-16s'\n", snd_seq_port_info_get_port(pinfo), -- 2.1.4

2 1

[alsa-devel] [PATCH V2 0/3] ASOC: bcm2835: move bcm2835-i2s to use clock framework
by kernel＠martin.sperl.org 13 Feb '16

13 Feb '16

From: Martin Sperl <kernel(a)martin.sperl.org> This patchset enables the bcm2835-i2s driver to use the clock framework which was introduced with commit 94cb7f76caa0 ("ARM: bcm2835: Switch to using the new clock driver support."). This commit resulted in the fact that the bcm2835-i2s driver was no longer working due to some register addresses used by 2 drivers (clk-bcm2835 and bcm2835-i2s). This patchset requires that the patchset for PCM-clock support as well as fractional/mash support is applied to the clk-bcm2835 driver and the corresponding device tree, but as the current version of the driver is not working, that should not be a problem. Note that there is one change: right now the current driver tries to calculate an optimal bclk_ratio based on its knowledge that it is using the 19.2Mhz oscillator. This computation would recommend the use of 40 or 80 bits instead of 32/64 bits that are required. Some of the DACs can not handle this, so most downstream drivers would set snd_soc_dai_set_bclk_ratio explicitly to disable this "non-power-of-2" automatic selection. So it seems wise to leave it out of the current patchset. If it is deemed necessary, then I can provide a separate patch that implements this again. Changelog: V1->V2: * moving clock patches into a separate patchset, which fixes also other issues in the clock framework * remove unnecessary bclk_size assignments Martin Sperl (3): ASoC: bcm2835: move to use the clock framework ARM: bcm2835: I2S: use new register-range and clock framework dt-bindings: bsm2835: fix bindings documentation to use new clock framework .../devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 7 +- arch/arm/boot/dts/bcm2835.dtsi | 5 +- sound/soc/bcm/bcm2835-i2s.c | 284 +++++--------------- 3 files changed, 69 insertions(+), 227 deletions(-) -- 1.7.10.4

6 14

[alsa-devel] [PATCH] midi: avoid freeing umidi object twice
by Andrey Konovalov 13 Feb '16

13 Feb '16

The 'umidi' object will be free'd on the error path by snd_usbmidi_free() when tearing down the rawmidi interface. So we shouldn't try to free it in snd_usbmidi_create() after having registered the rawmidi interface. Found by KASAN. Acked-by: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Andrey Konovalov <andreyknvl(a)gmail.com> --- sound/usb/midi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index cc39f63..007cf58 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2455,7 +2455,6 @@ int snd_usbmidi_create(struct snd_card *card, else err = snd_usbmidi_create_endpoints(umidi, endpoints); if (err < 0) { - snd_usbmidi_free(umidi); return err; } -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: qcom: Don't specify LE device endianness" to the asoc tree
by Mark Brown 12 Feb '16

12 Feb '16

The patch ASoC: qcom: Don't specify LE device endianness has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8faa268ab780a7379cfe54b882c6f6e0083233f2 Mon Sep 17 00:00:00 2001 From: Stephen Boyd <sboyd(a)codeaurora.org> Date: Fri, 12 Feb 2016 09:30:17 -0800 Subject: [PATCH] ASoC: qcom: Don't specify LE device endianness This reverts commit 18560a4e3 (ASoC: qcom: Specify LE device endianness). The commit that caused us to specify LE device endianness here, 29bb45f25ff3 (regmap-mmio: Use native endianness for read/write, 2015-10-29), has been reverted in mainline so now when we specify LE it actively breaks big endian kernels because the byte swapping in regmap-mmio is incorrect. Let's revert this change because it will 1) fix the big endian kernels and 2) be redundant to specify LE because that will become the default soon. Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/lpass-cpu.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 00b6c9d039cf..e5101e0d2d37 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -355,7 +355,6 @@ static struct regmap_config lpass_cpu_regmap_config = { .readable_reg = lpass_cpu_regmap_readable, .volatile_reg = lpass_cpu_regmap_volatile, .cache_type = REGCACHE_FLAT, - .val_format_endian = REGMAP_ENDIAN_LITTLE, }; int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) -- 2.7.0

1 1

[alsa-devel] [PATCH] ALSA: at73c213: add error handling of clk_enable()
by Alexey Khoroshilov 12 Feb '16

12 Feb '16

Recent addition of SSC clock management does not check if clk_enable() succeed. The patch adds such checks where it is appropriate. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> --- sound/spi/at73c213.c | 25 ++++++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index fac7e6eb9529..36cfaa316eab 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -221,7 +221,11 @@ static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) runtime->hw = snd_at73c213_playback_hw; chip->substream = substream; - clk_enable(chip->ssc->clk); + err = clk_enable(chip->ssc->clk); + if (err < 0) { + chip->substream = NULL; + return err; + } return 0; } @@ -787,7 +791,9 @@ static int snd_at73c213_chip_init(struct snd_at73c213 *chip) goto out; /* Enable DAC master clock. */ - clk_enable(chip->board->dac_clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + goto out; /* Initialize at73c213 on SPI bus. */ retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); @@ -900,7 +906,9 @@ static int snd_at73c213_dev_init(struct snd_card *card, chip->card = card; chip->irq = -1; - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->ssc->clk); + if (retval) + return retval; retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); if (retval) { @@ -1099,9 +1107,16 @@ static int snd_at73c213_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; + int retval; - clk_enable(chip->board->dac_clk); - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + return retval; + retval = clk_enable(chip->ssc->clk); + if (retval) { + clk_disable(chip->board->dac_clk); + return retval; + } ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); return 0; -- 1.9.1

1 0

[alsa-devel] [PATCH 1/2] ALSA: timer: Protect the whole snd_timer_close() with open race
by Takashi Iwai 12 Feb '16

12 Feb '16

In order to make the open/close more robust, widen the register_mutex protection over the whole snd_timer_close() function. Also, the close procedure is slightly shuffled to be in the safer order, as well as a few code refactoring. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is no urgent fix, so queued for 4.6. sound/core/timer.c | 48 +++++++++++++++++++++--------------------------- 1 file changed, 21 insertions(+), 27 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index dca817fc7894..b572a9bc31ad 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -318,25 +318,14 @@ int snd_timer_close(struct snd_timer_instance *timeri) if (snd_BUG_ON(!timeri)) return -ENXIO; + mutex_lock(&register_mutex); + list_del(&timeri->open_list); + /* force to stop the timer */ snd_timer_stop(timeri); - if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) { - /* wait, until the active callback is finished */ - spin_lock_irq(&slave_active_lock); - while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) { - spin_unlock_irq(&slave_active_lock); - udelay(10); - spin_lock_irq(&slave_active_lock); - } - spin_unlock_irq(&slave_active_lock); - mutex_lock(&register_mutex); - list_del(&timeri->open_list); - mutex_unlock(&register_mutex); - } else { - timer = timeri->timer; - if (snd_BUG_ON(!timer)) - goto out; + timer = timeri->timer; + if (timer) { /* wait, until the active callback is finished */ spin_lock_irq(&timer->lock); while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) { @@ -345,11 +334,7 @@ int snd_timer_close(struct snd_timer_instance *timeri) spin_lock_irq(&timer->lock); } spin_unlock_irq(&timer->lock); - mutex_lock(&register_mutex); - list_del(&timeri->open_list); - if (list_empty(&timer->open_list_head) && - timer->hw.close) - timer->hw.close(timer); + /* remove slave links */ spin_lock_irq(&slave_active_lock); spin_lock(&timer->lock); @@ -363,18 +348,27 @@ int snd_timer_close(struct snd_timer_instance *timeri) } spin_unlock(&timer->lock); spin_unlock_irq(&slave_active_lock); - /* release a card refcount for safe disconnection */ - if (timer->card) - put_device(&timer->card->card_dev); - mutex_unlock(&register_mutex); + + /* slave doesn't need to release timer resources below */ + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + timer = NULL; } - out: + if (timeri->private_free) timeri->private_free(timeri); kfree(timeri->owner); kfree(timeri); - if (timer) + + if (timer) { + if (list_empty(&timer->open_list_head) && timer->hw.close) + timer->hw.close(timer); + /* release a card refcount for safe disconnection */ + if (timer->card) + put_device(&timer->card->card_dev); module_put(timer->module); + } + + mutex_unlock(&register_mutex); return 0; } -- 2.7.1

1 1

[alsa-devel] Usb audio - unsuported sampling rates
by Gordon Dilan 12 Feb '16

12 Feb '16

Hello, I'm using an usb mic which(according to it's producer should support up to 48kHz sampling rate). The problem is ALSA allows me to use only 16 kHz for this device. I do not want to resample the data, I want to sample it at 48kHz in order to be able to record some high pitch sounds 18-20kHz. I would want to try to do any modifications necessary in order to try support 48kHz, but I'm not sure where to start. Some information: cat /proc/asound/modules ... 1 snd_usb_audio arecord --list-devices **** List of CAPTURE Hardware Devices **** .... card 1: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 when I try to use: arecord -v -D hw:1,0 -f S16_LE -c 4 -r 48000 gigi.wav I get this: Recording WAVE 'gigi.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Warning: rate is not accurate (requested = 48000Hz, got = 16000Hz) please, try the plug plugin Hardware PCM card 1 'USB Camera-B4.09.24.1' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 1 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 Any sampling rate I would try higher or lower than 16kHz, I will always get 16000 kHz. Some more information: lsusb Bus 001 Device 007: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 7, If 0, Class=vend., Driver=ov534, 480M |__ Port 2: Dev 7, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 7, If 2, Class=audio, Driver=snd-usb-audio, 480M At the moment I'm looking inside ALSA's source code to try to see from where this limitation of 16000kHz for my device appears, but I have to admit that I'm kind of lost. Any pointers on what direction should I go to try make my desired rate supported would be greatly appreciated! Thank you and best regards, Gordon

3 6

[alsa-devel] [PATCH 0/2] ALSA: dice: improve Dice notification handling
by Takashi Sakamoto 12 Feb '16

12 Feb '16

Hi, This patchset improves Dice notification handling mainly for old firmwares. Dice-based models with earlier firmwares brings 'fail to ensure phase lock' message to ALSA dice driver. This is due to no Dice transaction at confirming clock status. Although this patchet enables ALSA dice driver to handle Dice-based models with old firmwares, I reccomend users to work with updated firmwares because they may include some bug fixes and improve device's behaviours. Regards Takashi Sakamoto (2): ALSA: dice: change notification mask to detect lock status change ALSA: dice: old firmware optimization for Dice notification sound/firewire/dice/dice-stream.c | 17 ++++++++++++++--- sound/firewire/dice/dice-transaction.c | 2 +- 2 files changed, 15 insertions(+), 4 deletions(-) -- 2.5.0

2 3

[alsa-devel] ASoC fixes for v4.5
by Mark Brown 12 Feb '16

12 Feb '16

The following changes since commit 388f7b1d6e8ca06762e2454d28d6c3c55ad0fe95: Linux 4.5-rc3 (2016-02-07 15:38:30 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.5-rc4 for you to fetch changes up to f11aec0d7c835c3b83e74e3fd60d2e48a850c857: Merge remote-tracking branches 'asoc/fix/rt5659', 'asoc/fix/sigmadsp', 'asoc/fix/simple', 'asoc/fix/wm5110' and 'asoc/fix/wm8960' into asoc-linus (2016-02-10 19:23:09 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.5 A rather large batch of fixes here, almost all in the Intel driver. The changes that got merged in this merge window for Skylake were rather large and as well as issues that you'd expect in a large block of new code there were some problems created for older processors which needed fixing up. Things are largely settling down now hopefully. ---------------------------------------------------------------- Aaro Koskinen (1): ASoC: simple-card: don't fail if sysclk setting is not supported Arnd Bergmann (2): ASoC: mediatek: add i2c dependency ASoC: rt5659: avoid unused variable warning for rt5659_acpi_match Axel Lin (1): ASoC: rt5659: Fix irq leak Bard Liao (2): ASoC: rt286: fix capture doesn't work at some cases ASoC: rt5645: fix the shift bit of IN1 boost Dan Carpenter (1): ASoC: AMD: free memory on error Dharageswari.R (1): ASoC: Intel: Skylake: Fix pipe memory allocation leak Guneshwor Singh (2): ASoC: Intel: Skylake: Fix delay wrap condition ASoC: Intel: Skylake: Fix mcps freeup after module unbind failure Jeeja KP (6): ASoC: Intel: Skylake: Fix return of skl_get_queue_index ASoC: Intel: Skylake: Fix the module state check condition ASoC: Intel: Skylake: Fix not to stop sink pipe in pga pmd event ASoC: Intel: Skylake: Fix bind of source with multiple sinks ASoC: Intel: Skylake: Add missing PRE/POST_PMU handlers for vmixer ASoC: Intel: Skylake: Fix stereo DMIC record Jon Medhurst (Tixy) (1): ASoC: dwc: Ensure i2s_reg_comp{1,2} is always initialised Lars-Peter Clausen (2): ASoC: imx-spdif: Fix crash on suspend ASoC: dapm: Don't prefix autodisable widgets twice Maciej S. Szmigiero (1): ASoC: fsl_ssi: remove explicit register defaults Mans Rullgard (1): ASoC: mxs-saif: fix clk_prepare() without matching clk_unprepare() Mark Brown (7): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel-kconfig' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/amd', 'asoc/fix/arizona', 'asoc/fix/dpcm', 'asoc/fix/dwc', 'asoc/fix/fsl' and 'asoc/fix/fsl-ssi' into asoc-linus Merge remote-tracking branches 'asoc/fix/imx-spdif', 'asoc/fix/mtk', 'asoc/fix/mxs-saif', 'asoc/fix/qcom' and 'asoc/fix/rt286' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5659', 'asoc/fix/sigmadsp', 'asoc/fix/simple', 'asoc/fix/wm5110' and 'asoc/fix/wm8960' into asoc-linus Pascal Huerst (1): ASoC: sigmadsp: Fix missleading return value Pierre-Louis Bossart (2): ASoC: Intel: Atom: fix regression on compress DAI ASoC: Intel: Load the atom DPCM driver only Richard Fitzgerald (2): ASoC: wm5110: Unregister compressed platform when driver is removed ASoC: arizona: fref must be limited in pseudo-fractional mode Srinivas Kandagatla (1): ASoC: qcom: use correct device pointer in dma allocation Stuart Henderson (2): ASoC: wm8960: Fix input boost mixer left/right naming ASoC: wm8960: Fix WM8960_SYSCLK_PLL mode Sudip Mukherjee (1): ASoC: Intel: Skylake: Fix memory leak Vinod Koul (6): ASoC: dpcm: fix the BE state on hw_free ASoC: Intel: Skylake: Fix the memory overwrite of tlv buffer ASoC: Intel: Skylake: Remove autosuspend delay ASoC: Intel: Revert "ASoC: Intel: fix ACPI probe regression with Atom DPCM driver" ASoC: Intel: Create independent acpi match module ASoC: Intel: Add module tags for common match module Xiubo Li (1): ASoC: fsl: document DT compatible string "fsl,imx-audio-wm8960" .../devicetree/bindings/sound/fsl-asoc-card.txt | 2 + sound/soc/amd/acp-pcm-dma.c | 1 + sound/soc/codecs/arizona.c | 43 ++++++++++++- sound/soc/codecs/rt286.c | 26 +------- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5659.c | 31 ++++----- sound/soc/codecs/rt5659.h | 1 - sound/soc/codecs/sigmadsp-i2c.c | 5 +- sound/soc/codecs/wm5110.c | 1 + sound/soc/codecs/wm8960.c | 40 ++++++------ sound/soc/dwc/designware_i2s.c | 5 +- sound/soc/fsl/fsl_ssi.c | 42 ++++++------ sound/soc/fsl/imx-spdif.c | 2 - sound/soc/generic/simple-card.c | 2 +- sound/soc/intel/Kconfig | 13 +++- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 + sound/soc/intel/boards/skl_rt286.c | 5 +- sound/soc/intel/common/Makefile | 9 +-- sound/soc/intel/common/sst-acpi.c | 4 ++ sound/soc/intel/common/sst-match-acpi.c | 3 + sound/soc/intel/skylake/skl-messages.c | 6 +- sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/intel/skylake/skl-topology.c | 75 +++++++++++++++------- sound/soc/intel/skylake/skl.c | 2 - sound/soc/mediatek/Kconfig | 4 +- sound/soc/mxs/mxs-saif.c | 13 +++- sound/soc/qcom/lpass-platform.c | 15 ++--- sound/soc/soc-dapm.c | 8 +-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 221 insertions(+), 144 deletions(-)

2 1

[alsa-devel] Linking ALSA sequencer kernel clients to their device
by Martin Koegler 12 Feb '16

12 Feb '16

In my domain, people use multiple identical USB-MIDI keyboards - each is assigned a different functions. They expect to kept this mapping across reboots. The sound card number are dynamic, therefore they can change after reboots. The only stable device ID is the sysfs path of the used USB port. This means, that I need to associate a kernel sequencer client number with a sysfs device. rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. So I thought about exposing the missing information via sysfs: >From 776dd02d8d7eae89c2699be27eedf81eaf690a0e Mon Sep 17 00:00:00 2001 From: Martin Koegler <martin.koegler(a)chello.at> Date: Mon, 8 Feb 2016 22:53:42 +0100 Subject: [PATCH] Provide sequencer client number via sysfs Provides the missing link between real hardware devices and kernel sequencer clients. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- sound/core/seq/seq_midi.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 5dd0ee2..0d807e2 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -271,6 +271,31 @@ static void snd_seq_midisynth_delete(struct seq_midisynth *msynth) snd_midi_event_free(msynth->parser); } +static ssize_t +client_number_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_seq_device *device = container_of(dev, struct snd_seq_device, dev); + struct snd_card *card = device->card; + struct seq_midisynth_client *client; + + client = card ? synths[card->number] : NULL; + return snprintf(buf, PAGE_SIZE, "%i\n", client ? client->seq_client : -1); +} + +static DEVICE_ATTR(client, S_IRUGO, client_number_show_attr, NULL); + +static ssize_t +device_number_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_seq_device *device = container_of(dev, struct snd_seq_device, dev); + + return snprintf(buf, PAGE_SIZE, "%i\n", device->device); +} + +static DEVICE_ATTR(device, S_IRUGO, device_number_show_attr, NULL); + /* register new midi synth port */ static int snd_seq_midisynth_probe(struct device *_dev) @@ -287,6 +312,7 @@ snd_seq_midisynth_probe(struct device *_dev) struct snd_card *card = dev->card; int device = dev->device; unsigned int input_count = 0, output_count = 0; + int error; if (snd_BUG_ON(!card || device < 0 || device >= SNDRV_RAWMIDI_DEVICES)) return -EINVAL; @@ -332,6 +358,16 @@ snd_seq_midisynth_probe(struct device *_dev) kfree(info); return -ENOMEM; } + error = device_create_file(&dev->dev, &dev_attr_client); + if (error >= 0) + error = device_create_file(&dev->dev, &dev_attr_device); + if (error < 0) { + snd_seq_delete_kernel_client(client->seq_client); + kfree(client); + mutex_unlock(&register_mutex); + kfree(info); + return error; + } } msynth = kcalloc(ports, sizeof(struct seq_midisynth), GFP_KERNEL); @@ -451,6 +487,8 @@ snd_seq_midisynth_remove(struct device *_dev) kfree(msynth); client->num_ports--; if (client->num_ports <= 0) { + device_remove_file(&dev->dev, &dev_attr_client); + device_remove_file(&dev->dev, &dev_attr_device); snd_seq_delete_kernel_client(client->seq_client); synths[card->number] = NULL; kfree(client); -- 2.1.4

2 2

[alsa-devel] Resend [GIT PULL]: Update sst audio firmware for Skylake
by Vinod Koul 11 Feb '16

11 Feb '16

Resending again as previous email bounced for linux-firmware(a)kernel.org... Hello Linux firmware maintainers, Please PULL to receive the updated version of Skylake audio firmware. This PULL request has two patches, first one creates a sym link for binary name which helps us to keep different versions of firmware in git Second one updates to latest FW v869 The following changes since commit 52442afee9907bc32a058f22bb3295d040677c26: brcm: add firmware for bcm4366 device (2016-01-25 15:15:34 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to f7996657e8f25386541593242cb8e906d9808ec7: linux-firmware: intel: Update Skylake audio firmware (2016-02-04 10:36:52 +0530) ---------------------------------------------------------------- Vinod Koul (2): linux-firmware: intel: add symbolic link and file versions linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | Bin 233472 -> 23 bytes intel/dsp_fw_release_v827.bin | Bin 0 -> 233472 bytes intel/dsp_fw_release_v869.bin | Bin 0 -> 233472 bytes 4 files changed, 2 insertions(+), 2 deletions(-) mode change 100644 => 120000 intel/dsp_fw_release.bin create mode 100644 intel/dsp_fw_release_v827.bin create mode 100644 intel/dsp_fw_release_v869.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH] conf: topology: Add topolgy for skylake i2s configuration
by Subhransu S. Prusty 11 Feb '16

11 Feb '16

This patch adds basic playback/capture support for skylake i2s platform. DSP topology module data are passed through the binary files. The framework parses these files and puts the data in the widget private section for the corresponding widget. This is parsed by kernel driver and stored as module config for the DSP. Based on usecase these data are sent to the DSP through IPCs for further processing. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- configure.ac | 2 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/skl/Makefile.am | 5 + src/conf/topology/skl/i2slinux/Makefile.am | 5 + src/conf/topology/skl/i2slinux/codec0_in-cpr-1.bin | Bin 0 -> 4268 bytes src/conf/topology/skl/i2slinux/codec0_in-mi.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/codec0_out-cpr-4.bin | Bin 0 -> 4268 bytes src/conf/topology/skl/i2slinux/codec0_out-mo.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/codec1_out-cpr-5.bin | Bin 0 -> 4268 bytes src/conf/topology/skl/i2slinux/codec1_out-mo.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/dmic01_hifi_in-cpr-3.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/dmic01_hifi_in-mi.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/hdmi1_pt_out-cpr-7.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/hdmi1_pt_out-cpr-8.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/hdmi2_pt_out-cpr-10.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/hdmi2_pt_out-cpr-9.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/hdmi3_pt_out-cpr-11.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/hdmi3_pt_out-cpr-12.bin | Bin 0 -> 4268 bytes src/conf/topology/skl/i2slinux/media0_in-cpr-0.bin | Bin 0 -> 4268 bytes src/conf/topology/skl/i2slinux/media0_in-mi.bin | Bin 0 -> 4268 bytes .../topology/skl/i2slinux/media0_out-cpr-6.bin | Bin 0 -> 4268 bytes src/conf/topology/skl/i2slinux/media0_out-mo.bin | Bin 0 -> 4268 bytes src/conf/topology/skl/skl_i2s_linux.conf | 337 +++++++++++++++++++++ 23 files changed, 350 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/skl/Makefile.am create mode 100644 src/conf/topology/skl/i2slinux/Makefile.am create mode 100644 src/conf/topology/skl/i2slinux/codec0_in-cpr-1.bin create mode 100644 src/conf/topology/skl/i2slinux/codec0_in-mi.bin create mode 100644 src/conf/topology/skl/i2slinux/codec0_out-cpr-4.bin create mode 100644 src/conf/topology/skl/i2slinux/codec0_out-mo.bin create mode 100644 src/conf/topology/skl/i2slinux/codec1_out-cpr-5.bin create mode 100644 src/conf/topology/skl/i2slinux/codec1_out-mo.bin create mode 100644 src/conf/topology/skl/i2slinux/dmic01_hifi_in-cpr-3.bin create mode 100644 src/conf/topology/skl/i2slinux/dmic01_hifi_in-mi.bin create mode 100644 src/conf/topology/skl/i2slinux/hdmi1_pt_out-cpr-7.bin create mode 100644 src/conf/topology/skl/i2slinux/hdmi1_pt_out-cpr-8.bin create mode 100644 src/conf/topology/skl/i2slinux/hdmi2_pt_out-cpr-10.bin create mode 100644 src/conf/topology/skl/i2slinux/hdmi2_pt_out-cpr-9.bin create mode 100644 src/conf/topology/skl/i2slinux/hdmi3_pt_out-cpr-11.bin create mode 100644 src/conf/topology/skl/i2slinux/hdmi3_pt_out-cpr-12.bin create mode 100644 src/conf/topology/skl/i2slinux/media0_in-cpr-0.bin create mode 100644 src/conf/topology/skl/i2slinux/media0_in-mi.bin create mode 100644 src/conf/topology/skl/i2slinux/media0_out-cpr-6.bin create mode 100644 src/conf/topology/skl/i2slinux/media0_out-mo.bin create mode 100644 src/conf/topology/skl/skl_i2s_linux.conf diff --git a/configure.ac b/configure.ac index 9490d39..af6ea6e 100644 --- a/configure.ac +++ b/configure.ac @@ -660,6 +660,8 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ + src/conf/topology/skl/Makefile \ + src/conf/topology/skl/i2slinux/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ alsalisp/Makefile aserver/Makefile \ test/Makefile test/lsb/Makefile \ diff --git a/src/conf/topology/Makefile.am b/src/conf/topology/Makefile.am index f56a96c..ce011f3 100644 --- a/src/conf/topology/Makefile.am +++ b/src/conf/topology/Makefile.am @@ -1 +1 @@ -SUBDIRS=broadwell +SUBDIRS=broadwell skl diff --git a/src/conf/topology/skl/Makefile.am b/src/conf/topology/skl/Makefile.am new file mode 100644 index 0000000..623f1d0 --- /dev/null +++ b/src/conf/topology/skl/Makefile.am @@ -0,0 +1,5 @@ +SUBDIRS=i2slinux +alsaconfigdir = @ALSA_CONFIG_DIR@ +topologydir = $(alsaconfigdir)/topology/skl +topology_DATA = skl_i2s_linux.conf +EXTRA_DIST = $(topology_DATA) diff --git a/src/conf/topology/skl/i2slinux/Makefile.am b/src/conf/topology/skl/i2slinux/Makefile.am new file mode 100644 index 0000000..c57da9f --- /dev/null +++ b/src/conf/topology/skl/i2slinux/Makefile.am @@ -0,0 +1,5 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +i2slinuxdir = $(alsaconfigdir)/topology/skl/i2slinux +i2slinux_DATA = codec0_in-cpr-1.bin codec0_out-mo.bin dmic01_hifi_in-cpr-3.bin hdmi1_pt_out-cpr-8.bin hdmi3_pt_out-cpr-11.bin media0_in-cpr-0.bin media0_out-mo.bin codec0_in-mi.bin codec1_out-cpr-5.bin dmic01_hifi_in-mi.bin hdmi2_pt_out-cpr-10.bin hdmi3_pt_out-cpr-12.bin media0_in-mi.bin codec0_out-cpr-4.bin codec1_out-mo.bin hdmi1_pt_out-cpr-7.bin hdmi2_pt_out-cpr-9.bin media0_out-cpr-6.bin + +EXTRA_DIST = $(i2slinux_DATA) diff 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+SectionData."codec0_out cpr 4" { + file "skl/i2slinux/codec0_out-cpr-4.bin" +} +SectionData."codec1_out mo" { + file "skl/i2slinux/codec1_out-mo.bin" +} +SectionData."codec1_out cpr 5" { + file "skl/i2slinux/codec1_out-cpr-5.bin" +} +SectionData."codec0_in cpr 1" { + file "skl/i2slinux/codec0_in-cpr-1.bin" +} +SectionData."codec0_in mi" { + file "skl/i2slinux/codec0_in-mi.bin" +} +SectionData."dmic01_hifi_in cpr 3" { + file "skl/i2slinux/dmic01_hifi_in-cpr-3.bin" +} +SectionData."dmic01_hifi_in mi" { + file "skl/i2slinux/dmic01_hifi_in-mi.bin" +} +SectionData."hdmi1_pt_out cpr 7" { + file "skl/i2slinux/hdmi1_pt_out-cpr-7.bin" +} +SectionData."hdmi1_pt_out cpr 8" { + file "skl/i2slinux/hdmi1_pt_out-cpr-8.bin" +} +SectionData."hdmi2_pt_out cpr 9" { + file "skl/i2slinux/hdmi2_pt_out-cpr-9.bin" +} +SectionData."hdmi2_pt_out cpr 10" { + file "skl/i2slinux/hdmi2_pt_out-cpr-10.bin" +} +SectionData."hdmi3_pt_out cpr 11" { + file "skl/i2slinux/hdmi3_pt_out-cpr-11.bin" +} +SectionData."hdmi3_pt_out cpr 12" { + file "skl/i2slinux/hdmi3_pt_out-cpr-12.bin" +} + +SectionControlMixer."media0_in mi Switch" { + index"1" + invert "false" + max "1" + no_pm "true" + channel."fl" { + reg "-1" + shift "0" + } + channel."fr" { + reg "-1" + shift "0" + } + ops."ctl" { + get "64" + put "64" + info "64" + } +} +SectionControlMixer."codec0_in mi Switch" { + index"1" + invert "false" + max "1" + no_pm "true" + channel."fl" { + reg "-1" + shift "0" + } + channel."fr" { + reg "-1" + shift "0" + } + ops."ctl" { + get "64" + put "64" + info "64" + } +} +SectionControlMixer."dmic01_hifi_in mi Switch" { + index"1" + invert "false" + max "1" + no_pm "true" + channel."fl" { + reg "-1" + shift "0" + } + channel."fr" { + reg "-1" + shift "0" + } + ops."ctl" { + get "64" + put "64" + info "64" + } +} + +SectionWidget."media0_in cpr 0" { + index"1" + type"mixer" + no_pm "true" + event_type "3" + event_flags "9" + data "media0_in cpr 0" +} +SectionWidget."media0_in mi" { + index"1" + type"pga" + no_pm "true" + event_type "4" + event_flags "9" + subseq "10" + data "media0_in mi" +} +SectionWidget."media0_out mo" { + index"1" + type"mixer" + no_pm "true" + event_type "1" + event_flags "15" + subseq "10" + data "media0_out mo" + mixer [ + "media0_in mi Switch" + "codec0_in mi Switch" + "dmic01_hifi_in mi Switch" + ] +} +SectionWidget."media0_out cpr 6" { + index"1" + type"pga" + no_pm "true" + event_type "4" + data "media0_out cpr 6" +} +SectionWidget."codec0_out mo" { + index"1" + type"mixer" + no_pm "true" + event_type "1" + event_flags "15" + subseq "10" + data "codec0_out mo" + mixer [ + "media0_in mi Switch" + "codec0_in mi Switch" + "dmic01_hifi_in mi Switch" + ] +} +SectionWidget."codec0_out cpr 4" { + index"1" + type"pga" + no_pm "true" + event_type "4" + data "codec0_out cpr 4" +} +SectionWidget."codec0_out" { + index"1" + type"aif_out" + no_pm "true" +} +SectionWidget."codec1_out mo" { + index"1" + type"mixer" + no_pm "true" + event_type "1" + event_flags "15" + subseq "10" + data "codec1_out mo" + mixer [ + "media0_in mi Switch" + "codec0_in mi Switch" + "dmic01_hifi_in mi Switch" + ] +} +SectionWidget."codec1_out cpr 5" { + index"1" + type"pga" + no_pm "true" + event_type "4" + data "codec1_out cpr 5" +} +SectionWidget."codec1_out" { + index"1" + type"aif_out" + no_pm "true" +} +SectionWidget."codec0_in cpr 1" { + index"1" + type"mixer" + no_pm "true" + event_type "3" + event_flags "9" + data "codec0_in cpr 1" +} +SectionWidget."codec0_in mi" { + index"1" + type"pga" + no_pm "true" + event_type "4" + event_flags "9" + subseq "10" + data "codec0_in mi" +} +SectionWidget."codec0_in" { + index"1" + type"aif_in" + no_pm "true" +} +SectionWidget."dmic01_hifi_in cpr 3" { + index"1" + type"mixer" + no_pm "true" + event_type "3" + event_flags "9" + data "dmic01_hifi_in cpr 3" +} +SectionWidget."dmic01_hifi_in mi" { + index"1" + type"pga" + no_pm "true" + event_type "4" + event_flags "9" + subseq "10" + data "dmic01_hifi_in mi" +} +SectionWidget."dmic01_hifi" { + index"1" + type"aif_in" + no_pm "true" +} +SectionWidget."hdmi1_pt_out cpr 7" { + index"1" + type"mixer" + no_pm "true" + event_type "3" + event_flags "9" + data "hdmi1_pt_out cpr 7" +} +SectionWidget."hdmi1_pt_out cpr 8" { + index"1" + type"pga" + no_pm "true" + event_type "4" + data "hdmi1_pt_out cpr 8" +} +SectionWidget."iDisp1_out" { + index"1" + type"aif_out" + no_pm "true" +} +SectionWidget."hdmi2_pt_out cpr 9" { + index"1" + type"mixer" + no_pm "true" + event_type "3" + event_flags "9" + data "hdmi2_pt_out cpr 9" +} +SectionWidget."hdmi2_pt_out cpr 10" { + index"1" + type"pga" + no_pm "true" + event_type "4" + data "hdmi2_pt_out cpr 10" +} +SectionWidget."iDisp2_out" { + index"1" + type"aif_out" + no_pm "true" +} +SectionWidget."hdmi3_pt_out cpr 11" { + index"1" + type"mixer" + no_pm "true" + event_type "3" + event_flags "9" + data "hdmi3_pt_out cpr 11" +} +SectionWidget."hdmi3_pt_out cpr 12" { + index"1" + type"pga" + no_pm "true" + event_type "4" + data "hdmi3_pt_out cpr 12" +} +SectionGraph."Pipeline 1 Graph" { + index"1" + lines [ + "media0_in mi, , media0_in cpr 0" + "media0_in cpr 0, , System Playback" + "media0_out mo, media0_in mi Switch, media0_in mi" + "media0_out mo, codec0_in mi Switch, codec0_in mi" + "media0_out mo, dmic01_hifi_in mi Switch, dmic01_hifi_in mi" + "media0_out cpr 6, , media0_out mo" + "System Capture, , media0_out cpr 6" + "codec0_out mo, media0_in mi Switch, media0_in mi" + "codec0_out mo, codec0_in mi Switch, codec0_in mi" + "codec0_out mo, dmic01_hifi_in mi Switch, dmic01_hifi_in mi" + "codec0_out cpr 4, , codec0_out mo" + "codec0_out, , codec0_out cpr 4" + "codec1_out mo, media0_in mi Switch, media0_in mi" + "codec1_out mo, codec0_in mi Switch, codec0_in mi" + "codec1_out mo, dmic01_hifi_in mi Switch, dmic01_hifi_in mi" + "codec1_out cpr 5, , codec1_out mo" + "codec1_out, , codec1_out cpr 5" + "codec0_in mi, , codec0_in cpr 1" + "codec0_in cpr 1, , codec0_in" + "dmic01_hifi_in mi, , dmic01_hifi_in cpr 3" + "dmic01_hifi_in cpr 3, , dmic01_hifi" + "hdmi1_pt_out cpr 8, , hdmi1_pt_out cpr 7" + "hdmi1_pt_out cpr 7, , HDMI1 Playback" + "iDisp1_out, , hdmi1_pt_out cpr 8" + "hdmi2_pt_out cpr 10, , hdmi2_pt_out cpr 9" + "hdmi2_pt_out cpr 9, , HDMI2 Playback" + "iDisp2_out, , hdmi2_pt_out cpr 10" + "hdmi3_pt_out cpr 12, , hdmi3_pt_out cpr 11" + "hdmi3_pt_out cpr 11, , HDMI3 Playback" + "iDisp1_out, , hdmi3_pt_out cpr 12" + ] +} -- 1.9.1

4 11

Re: [alsa-devel] [PATCH 18/19] ASoC: Intel: Atom: add support for RT5642
by Luka Karinja 11 Feb '16

11 Feb '16

> The machine driver is not loaded when the BIOS uses the 10EC5642 > _HID. Add it to the white list of known _HIDs, codec_name is > already taken care of by previous commit > > Tested on Asus T100TAF. > I'm trying to get sound working on my T100TAF i did apply all the 19 patches on top of broonie/sound and added UCM files to my system. Does the TAF need different UCM file than the T100TA? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: bytcrrt5640 [bytcr-rt5640], device 0: Baytrail Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: bytcrrt5640 [bytcr-rt5640], device 1: Deep-Buffer Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 dmesg: [ 5.209239] Adding 1049596k swap on /dev/mmcblk0p4. Priority:-1 extents:1 across:1049596k SSFS [ 5.219099] input: PC Speaker as /devices/platform/pcspkr/input/input6 [ 5.273802] i2c i2c-10: Added multiplexed i2c bus 13 [ 5.274629] intel_sst_acpi 80860F28:00: LPE base: 0x50a00000 size:0x200000 [ 5.274638] intel_sst_acpi 80860F28:00: IRAM base: 0x50ac0000 [ 5.274671] intel_sst_acpi 80860F28:00: DRAM base: 0x50b00000 [ 5.274685] intel_sst_acpi 80860F28:00: SHIM base: 0x50b40000 [ 5.274700] intel_sst_acpi 80860F28:00: Mailbox base: 0x50b44000 [ 5.274712] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [ 5.274842] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [ 5.290156] Error: Driver 'pcspkr' is already registered, aborting... [ 5.311924] random: nonblocking pool is initialized [ 5.397912] EXT4-fs (mmcblk0p2): mounted filesystem with ordered data mode. Opts: data=ordered [ 5.506882] genirq: Flags mismatch irq 8. 00000083 (volume_down) vs. 00000000 (rtc0) [ 5.507025] gpio-keys gpio-keys.0.auto: Unable to claim irq 8; error -16 [ 5.507283] gpio-keys: probe of gpio-keys.0.auto failed with error -16 [ 5.507518] input: gpio-keys as /devices/platform/gpio-keys.1.auto/input/input8 [ 5.519513] iTCO_vendor_support: vendor-support=0 [ 5.591021] iTCO_wdt: Intel TCO WatchDog Timer Driver v1.11 [ 5.591153] iTCO_wdt: Found a Bay Trail SoC TCO device (Version=3, TCOBASE=0x0460) [ 5.604447] iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) [ 5.613586] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 5.613794] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [ 5.613922] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [ 5.615555] bytcr_rt5640 bytcr_rt5640: rt5640-aif1 <-> ssp2-port mapping ok [ 5.642183] asus_wmi: ASUS WMI generic driver loaded [ 5.663786] usbcore: registered new interface driver usbmouse [ 5.665362] usbcore: registered new interface driver usbkbd [ 5.681990] asus_wmi: Initialization: 0x1 [ 5.682121] asus_wmi: BIOS WMI version: 7.9 [ 5.682219] asus_wmi: SFUN value: 0x37 [ 5.685337] input: Asus WMI hotkeys as /devices/platform/asus-nb-wmi/input/input9 [ 5.708007] asus_wmi: Number of fans: 1 [ 5.801413] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.801529] input: SIS0817:00 0457:1071 as /devices/platform/80860F41:03/i2c-11/i2c-SIS0817:00/0018:0457:1071.0004/input/input10 [ 5.802167] hid-multitouch 0018:0457:1071.0004: input,hidraw3: I2C HID v1.00 Device [SIS0817:00 0457:1071] on i2c-SIS0817:00 [ 5.816775] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.816996] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.817204] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.817405] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.817616] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.818884] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.819109] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! [ 5.911351] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.911578] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.911787] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.911997] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.912201] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.912417] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.912634] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.915142] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! [ 6.042680] intel_rapl: Found RAPL domain package [ 6.042692] intel_rapl: Found RAPL domain core [ 6.247467] Bluetooth: BNEP (Ethernet Emulation) ver 1.3 [ 6.247489] Bluetooth: BNEP filters: protocol multicast [ 6.247515] Bluetooth: BNEP socket layer initialized [ 7.524852] brcmf_sdio_drivestrengthinit: No SDIO Drive strength init done for chip 43340 rev 2 pmurev 20 [ 7.525341] usbcore: registered new interface driver brcmfmac [ 7.722004] brcmf_c_preinit_dcmds: Firmware version = wl0: Jun 17 2014 11:48:43 version 6.10.190.49 (r485840) FWID 01-8744bcd2 [ 7.739335] brcmf_cfg80211_reg_notifier: not a ISO3166 code [ 7.899345] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 7.921532] audit: type=1006 audit(1450805770.516:2): pid=1018 uid=0 old-auid=4294967295 auid=1000 old-ses=4294967295 ses=1 res=1 [ 7.984230] audit: type=1006 audit(1450805770.580:3): pid=1091 uid=0 old-auid=4294967295 auid=1000 old-ses=4294967295 ses=2 res=1 [ 7.996410] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 8.151283] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 9.038571] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 11.245916] Bluetooth: hci1: BCM: chip id 82 [ 11.245928] Bluetooth: hci1: BCM43341B0 (002.001.014) build 0000 [ 11.755123] Bluetooth: hci1: BCM (002.001.014) build 0164 [ 14.513027] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x0 fw_state 0x1 [ 14.513045] intel_sst_acpi 80860F28:00: fw download failed -16 [ 14.513067] intel_sst_acpi 80860F28:00: FW download fail -16 [ 14.513864] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.514676] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.515574] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.518058] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.518941] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.520293] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.525383] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.527248] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.528335] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.534111] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port

2 4

[alsa-devel] [PATCH 0/6 v2] ALSA: dice: constrain PCM substreams to current sampling transfer frequency
by Takashi Sakamoto 11 Feb '16

11 Feb '16

Hi, This patchset is to update my former RFCv2 and for merge request to upstream. [alsa-devel][RFC][PATCH 00/10 v2] ALSA: dice: stabiliza packet streaming http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101897.h… Current ALSA Dice driver is written with somewhat ignoring actual hardware design to perform 'dynamic sample rate selection'. As a result, the driver includes some behaviors against users' expectation or functional misleading. This patchset purges such over-engineering. As a result, userspace applications can start PCM substreams just at current sampling transfer frequency. If users want to start at different sampling rate, they must set favorite rate in advance by tools. Currently, ffado-mixer with ffado-dbus-server[1] and hinawa-dice-common-cui[2] are available as such tool. For technical details, please refer to the post or our discussion in RFCv1: [alsa-devel] [RFC][PATCH 0/8] ALSA: dice: constrain PCM substreams to current sampling transfer frequency http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100525.h… [1] as a part of FFADO utility http://subversion.ffado.org/ [2] in hinawa-utils repository https://github.com/takaswie/hinawa-utils Regards Takashi Sakamoto (6): ALSA: dice: limit to current sampling transfer frequency ALSA: dice: limit stream to current sampling transfer frequency. ALSA: dice: add MIDI ports according to current number of MIDI substreams ALSA: dice: get the number of MBLA data channel at opening PCM substream ALSA: dice: purge generating channel cache ALSA: dice: ensure phase lock before starting streaming sound/firewire/dice/dice-midi.c | 23 +++- sound/firewire/dice/dice-pcm.c | 199 +++++++++------------------------ sound/firewire/dice/dice-stream.c | 69 +++++++----- sound/firewire/dice/dice-transaction.c | 54 --------- sound/firewire/dice/dice.c | 67 +---------- sound/firewire/dice/dice.h | 8 -- 6 files changed, 116 insertions(+), 304 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: arizona: fref must be limited in pseudo-fractional mode
by Richard Fitzgerald 10 Feb '16

10 Feb '16

When the FLL is in pseudo-fractional mode there is an additional limit on fref based on the fratio, to prevent aliasing around the Nyquist frequency. If fref exceeds this limit the refclk divider must be increased and the calculation tried again until a suitable combination of fref and fratio is found or we have to fall back to integer mode. This patch also adds some debug log prints around this code. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a32cfb8..913cfa8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1899,6 +1899,25 @@ static struct { { 1000000, 13500000, 0, 1 }, }; +static const unsigned int pseudo_fref_max[ARIZONA_FLL_MAX_FRATIO] = { + 13500000, + 6144000, + 6144000, + 3072000, + 3072000, + 2822400, + 2822400, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 768000, +}; + static struct { unsigned int min; unsigned int max; @@ -2012,16 +2031,32 @@ static int arizona_calc_fratio(struct arizona_fll *fll, /* Adjust FRATIO/refdiv to avoid integer mode if possible */ refdiv = cfg->refdiv; + arizona_fll_dbg(fll, "pseudo: initial ratio=%u fref=%u refdiv=%u\n", + init_ratio, Fref, refdiv); + while (div <= ARIZONA_FLL_MAX_REFDIV) { for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / - (fll->vco_mult * ratio) < Fref) + (fll->vco_mult * ratio) < Fref) { + arizona_fll_dbg(fll, "pseudo: hit VCO corner\n"); break; + } + + if (Fref > pseudo_fref_max[ratio - 1]) { + arizona_fll_dbg(fll, + "pseudo: exceeded max fref(%u) for ratio=%u\n", + pseudo_fref_max[ratio - 1], + ratio); + break; + } if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); return ratio; } } @@ -2030,6 +2065,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll, if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); return ratio; } } @@ -2038,6 +2076,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll, Fref /= 2; refdiv++; init_ratio = arizona_find_fratio(Fref, NULL); + arizona_fll_dbg(fll, + "pseudo: change fref=%u refdiv=%d(%d) ratio=%u\n", + Fref, refdiv, div, init_ratio); } arizona_fll_warn(fll, "Falling back to integer mode operation\n"); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: sigmadsp: Fix missleading return value
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> Forwarding the return value of i2c_master_send, leads to errors later on, since i2c_master_send returns the number of bytes transmittet. Check for ret < 0 instead and return 0 otherwise. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/sigmadsp-i2c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 21ca3a5..d374c18 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data, kfree(buf); - return ret; + if (ret < 0) + return ret; + + return 0; } static int sigmadsp_read_i2c(void *control_data, -- 2.4.3

3 2

[alsa-devel] [PATCH] ASoC: sigmadsp: Fix missleading return value
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> Forwarding the return value of i2c_master_send, leads to errors later on, since i2c_master_send returns the number of bytes transmittet. Check for ret < 0 instead and return 0 otherwise. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/sigmadsp-i2c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 21ca3a5..d374c18 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data, kfree(buf); - return ret; + if (ret < 0) + return ret; + + return 0; } static int sigmadsp_read_i2c(void *control_data, -- 2.4.3

2 1

[alsa-devel] topic/hda-mst branch merged to for-next
by Takashi Iwai 10 Feb '16

10 Feb '16

JFYI, I merged topic/hda-mst branch to for-next branch for wider testing. It means that this will be included in 4.6. I'll keep topic/hda-mst branch for a while, but for future patches, it might be better to based on for-next branch. Takashi

2 1

[alsa-devel] [PATCH] ASoC: cs4271: add regulator consumer support
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> The cs4271 has three power domains: vd, vl and va. Enable them all, as long as the codec is in use. While at it, factored out the reset code into its own function. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 7 +++ sound/soc/codecs/cs4271.c | 69 +++++++++++++++++++--- 2 files changed, 68 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index e2cd1d7..6e699ce 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -33,12 +33,19 @@ Optional properties: Note that this is not needed in case the clocks are stable throughout the entire runtime of the codec. + - vd-supply: Digital power + - vl-supply: Logic power + - va-supply: Analog Power + Examples: codec_i2c: cs4271@10 { compatible = "cirrus,cs4271"; reg = <0x10>; reset-gpio = <&gpio 23 0>; + vd-supply = <&vdd_3v3_reg>; + vl-supply = <&vdd_3v3_reg>; + va-supply = <&vdd_3v3_reg>; }; codec_spi: cs4271@0 { diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e770ee6..0c0010b 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/tlv.h> @@ -157,6 +158,10 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) return reg == CS4271_CHIPID; } +static const char * const supply_names[] = { + "vd", "vl", "va" +}; + struct cs4271_private { unsigned int mclk; bool master; @@ -170,6 +175,7 @@ struct cs4271_private { int gpio_disable; /* enable soft reset workaround */ bool enable_soft_reset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = { @@ -487,6 +493,20 @@ static struct snd_soc_dai_driver cs4271_dai = { .symmetric_rates = 1, }; +static int cs4271_reset(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(cs4271->gpio_nreset)) { + gpio_set_value(cs4271->gpio_nreset, 0); + mdelay(1); + gpio_set_value(cs4271->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec) { @@ -499,6 +519,9 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; } @@ -507,6 +530,16 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) int ret; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + + /* Do a proper reset after power up */ + cs4271_reset(codec); + /* Restore codec state */ ret = regcache_sync(cs4271->regmap); if (ret < 0) @@ -553,19 +586,24 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) } #endif + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + if (cs4271plat) { amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_is_valid(cs4271->gpio_nreset)) { - /* Reset codec */ - gpio_direction_output(cs4271->gpio_nreset, 0); - mdelay(1); - gpio_set_value(cs4271->gpio_nreset, 1); - /* Give the codec time to wake up */ - mdelay(1); - } + /* Reset codec */ + cs4271_reset(codec); + + ret = regcache_sync(cs4271->regmap); + if (ret < 0) + return ret; ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, @@ -595,6 +633,9 @@ static int cs4271_codec_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(cs4271->gpio_nreset, 0); + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; }; @@ -617,6 +658,7 @@ static int cs4271_common_probe(struct device *dev, { struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; + int i, ret; cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) @@ -638,6 +680,17 @@ static int cs4271_common_probe(struct device *dev, return ret; } + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + cs4271->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + *c = cs4271; return 0; } -- 2.4.3

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[alsa-devel] [PATCH] MAINTAINERS: add msm mailing list to sound/soc/qcom
by Srinivas Kandagatla 10 Feb '16

10 Feb '16

This patch adds msm mailing list to sound/soc/qcom so that get_maintainers would return correct list to CC to. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 7f1fa4f..213be2a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5820,6 +5820,7 @@ F: include/uapi/linux/ipmi* QCOM AUDIO (ASoC) DRIVERS M: Patrick Lai <plai(a)codeaurora.org> M: Banajit Goswami <bgoswami(a)codeaurora.org> +L: linux-arm-msm(a)vger.kernel.org L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/qcom/ -- 1.9.1

1 0

[alsa-devel] Should snd_card_free() check for null pointer?
by Jerome Marchand 10 Feb '16

10 Feb '16

Hi, Before commit f24640648186b (ALSA: Use standard device refcount for card accounting), snd_card_free() would return -EINVAL on a null pointer. Now it ends up in a null pointer dereference. There is at least one driver that can call snd_card_free() with null argument: saa7134_alsa. It can easily be triggered by just inserting and removing the module (no need to have the hardware). I don't think that is a rule, but it seems that the standard behavior of *_free() functions is to check for null pointer. What do you think? Thanks, Jerome

2 3

[alsa-devel] [PATCH] ASoC: rt5645: add "coreboot" to dmi_system_id list
by Bard Liao 09 Feb '16

09 Feb '16

There are some Google projects with manufacturer name different from "GOOGLE". So we need to add DMI list for those projects. However some OEMs / vendors want to keep secret before product launch. So, we can't use DMI_SYS_VENDOR for them. Fortunately, the BIOS vendor field is the same for all chrome os projects. It allows us to match chrome projects by DMI_BIOS_VENDOR = "coreboot". Signed-off-by: John Lin <john.lin(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 93e8c90..8521b22 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3552,6 +3552,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, + { + .ident = "Google Chrome", + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "coreboot"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v2 00/22] Sharing media resources across ALSA and au0828 drivers
by Shuah Khan 09 Feb '16

09 Feb '16

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API and Media Controller API to share media resource (tuner). This Patch v2 series is based on linux_media master. This work addresses Mauro and Takashi's comments. Composite or S-Video connector is the input case is still under discussion and isn't addressed in this series. Patches 3 and 4 need documentation updates and will be sent later this week. Shuah Khan (22): uapi/media.h: Declare interface types for ALSA media: Add ALSA Media Controller function entities media: Media Controller register/unregister entity_notify API media: Media Controller enable/disable source handler API media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add enable/disable source common interfaces media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: Change v4l-core to check if source is free media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_source() media: au0828 change to use Managed Media Controller API media: au0828 handle media_init and media_register window media: au0828 create tuner to decoder link in disabled state media: au0828 disable tuner to demod link media: au0828 Use au8522_media_pads enum for pad defines media: au0828-core register entity_notify hook media: au0828 add enable, disable source handlers sound/usb: Use Media Controller API to share media resources media: Ensure media device unregister is done only once drivers/media/dvb-core/dvb_frontend.c | 139 ++---------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 73 +++++- drivers/media/media-devnode.c | 15 +- drivers/media/media-entity.c | 51 ++++- drivers/media/usb/au0828/au0828-core.c | 272 +++++++++++++++++++++-- drivers/media/usb/au0828/au0828-video.c | 75 +------ drivers/media/usb/au0828/au0828.h | 4 + drivers/media/v4l2-core/Makefile | 2 +- drivers/media/v4l2-core/v4l2-fh.c | 2 + drivers/media/v4l2-core/v4l2-ioctl.c | 30 +++ drivers/media/v4l2-core/v4l2-mc.c | 60 +++++ drivers/media/v4l2-core/videobuf2-core.c | 4 + include/media/media-device.h | 44 ++++ include/media/media-devnode.h | 17 ++ include/media/media-entity.h | 12 + include/media/v4l2-dev.h | 1 + include/media/v4l2-mc.h | 52 +++++ include/uapi/linux/media.h | 33 +++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 1 + sound/usb/media.c | 319 +++++++++++++++++++++++++++ sound/usb/media.h | 72 ++++++ sound/usb/mixer.h | 1 + sound/usb/pcm.c | 28 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 3 + 33 files changed, 1117 insertions(+), 236 deletions(-) create mode 100644 drivers/media/v4l2-core/v4l2-mc.c create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.5.0

3 44

[alsa-devel] [PATCH v3 1/3] portman2x4 - whitespace fixes
by Sudip Mukherjee 09 Feb '16

09 Feb '16

To improve the readability and grep-ability of the code trailing whitespace, multiple blank lines and space before tabs have been removed. This patch takes care of these issues related to whitespace changes and there was no other change. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v3: changed commit message v2: combined 3 patches of v1 series into this patch. sound/drivers/portman2x4.c | 77 ++++++++++++++++++++++------------------------ 1 file changed, 37 insertions(+), 40 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 464385a..65b0650 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -57,7 +57,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -static struct platform_device *platform_devices[SNDRV_CARDS]; +static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; module_param_array(index, int, NULL, S_IRUGO); @@ -105,7 +105,7 @@ static int portman_create(struct snd_card *card, *rchip = NULL; pm = kzalloc(sizeof(struct portman), GFP_KERNEL); - if (pm == NULL) + if (pm == NULL) return -ENOMEM; /* Init chip specific data */ @@ -123,24 +123,24 @@ static int portman_create(struct snd_card *card, *********************************************************************/ /* Standard PC parallel port status register equates. */ -#define PP_STAT_BSY 0x80 /* Busy status. Inverted. */ -#define PP_STAT_ACK 0x40 /* Acknowledge. Non-Inverted. */ -#define PP_STAT_POUT 0x20 /* Paper Out. Non-Inverted. */ -#define PP_STAT_SEL 0x10 /* Select. Non-Inverted. */ -#define PP_STAT_ERR 0x08 /* Error. Non-Inverted. */ +#define PP_STAT_BSY 0x80 /* Busy status. Inverted. */ +#define PP_STAT_ACK 0x40 /* Acknowledge. Non-Inverted. */ +#define PP_STAT_POUT 0x20 /* Paper Out. Non-Inverted. */ +#define PP_STAT_SEL 0x10 /* Select. Non-Inverted. */ +#define PP_STAT_ERR 0x08 /* Error. Non-Inverted. */ /* Standard PC parallel port command register equates. */ -#define PP_CMD_IEN 0x10 /* IRQ Enable. Non-Inverted. */ -#define PP_CMD_SELI 0x08 /* Select Input. Inverted. */ -#define PP_CMD_INIT 0x04 /* Init Printer. Non-Inverted. */ -#define PP_CMD_FEED 0x02 /* Auto Feed. Inverted. */ +#define PP_CMD_IEN 0x10 /* IRQ Enable. Non-Inverted. */ +#define PP_CMD_SELI 0x08 /* Select Input. Inverted. */ +#define PP_CMD_INIT 0x04 /* Init Printer. Non-Inverted. */ +#define PP_CMD_FEED 0x02 /* Auto Feed. Inverted. */ #define PP_CMD_STB 0x01 /* Strobe. Inverted. */ /* Parallel Port Command Register as implemented by PCP2x4. */ -#define INT_EN PP_CMD_IEN /* Interrupt enable. */ +#define INT_EN PP_CMD_IEN /* Interrupt enable. */ #define STROBE PP_CMD_STB /* Command strobe. */ -/* The parallel port command register field (b1..b3) selects the +/* The parallel port command register field (b1..b3) selects the * various "registers" within the PC/P 2x4. These are the internal * address of these "registers" that must be written to the parallel * port command register. @@ -148,7 +148,7 @@ static int portman_create(struct snd_card *card, #define RXDATA0 (0 << 1) /* PCP RxData channel 0. */ #define RXDATA1 (1 << 1) /* PCP RxData channel 1. */ #define GEN_CTL (2 << 1) /* PCP General Control Register. */ -#define SYNC_CTL (3 << 1) /* PCP Sync Control Register. */ +#define SYNC_CTL (3 << 1) /* PCP Sync Control Register. */ #define TXDATA0 (4 << 1) /* PCP TxData channel 0. */ #define TXDATA1 (5 << 1) /* PCP TxData channel 1. */ #define TXDATA2 (6 << 1) /* PCP TxData channel 2. */ @@ -218,7 +218,7 @@ static inline void portman_write_data(struct portman *pm, u8 value) parport_write_data(pm->pardev->port, value); } -static void portman_write_midi(struct portman *pm, +static void portman_write_midi(struct portman *pm, int port, u8 mididata) { int command = ((port + 4) << 1); @@ -234,20 +234,20 @@ static void portman_write_midi(struct portman *pm, */ command |= INT_EN; - /* Disable interrupts so that the process is not interrupted, then - * write the address associated with the current Tx channel to the + /* Disable interrupts so that the process is not interrupted, then + * write the address associated with the current Tx channel to the * PP Command Reg. Do not set the Strobe signal yet. */ do { portman_write_command(pm, command); - /* While the address lines settle, write parallel output data to + /* While the address lines settle, write parallel output data to * PP Data Reg. This has no effect until Strobe signal is asserted. */ portman_write_data(pm, mididata); - + /* If PCP channel's TxEmpty is set (TxEmpty is read through the PP * Status Register), then go write data. Else go back and wait. */ @@ -284,7 +284,6 @@ static void portman_write_midi(struct portman *pm, /* Data sent. */ } - /* * Read MIDI byte from port * Attempt to read input byte from specified hardware input port (0..). @@ -357,7 +356,6 @@ static int portman_read_midi(struct portman *pm, int port) portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */ portman_write_data(pm, 0); /* Return data clock low. */ - /* De-assert Strobe and return data. */ portman_write_command(pm, cmdout); /* Output saved address+IE. */ @@ -370,11 +368,12 @@ static int portman_read_midi(struct portman *pm, int port) /* * Checks if any input data on the given channel is available - * Checks RxAvail + * Checks RxAvail */ static int portman_data_avail(struct portman *pm, int channel) { int command = INT_EN; + switch (channel) { case 0: command |= RXDATA0; @@ -393,7 +392,6 @@ static int portman_data_avail(struct portman *pm, int channel) return 0; } - /* * Flushes any input */ @@ -456,7 +454,7 @@ static int portman_probe(struct parport *p) * hardware handshake lines to midi box: * * Strobe = 0 - * Interrupt Enable = 0 + * Interrupt Enable = 0 */ /* 2 */ parport_write_control(p, 0); @@ -481,7 +479,7 @@ static int portman_probe(struct parport *p) /* 7 */ parport_write_control(p, 0); /* Reset Strobe=0. */ - /* Check if Tx circuitry is functioning properly. If initialized + /* Check if Tx circuitry is functioning properly. If initialized * unit TxEmpty is false, send out char and see if if goes true. */ /* 8 */ @@ -567,12 +565,12 @@ static int snd_portman_rawmidi_create(struct snd_card *card) struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; int err; - - err = snd_rawmidi_new(card, CARD_NAME, 0, - PORTMAN_NUM_OUTPUT_PORTS, - PORTMAN_NUM_INPUT_PORTS, + + err = snd_rawmidi_new(card, CARD_NAME, 0, + PORTMAN_NUM_OUTPUT_PORTS, + PORTMAN_NUM_INPUT_PORTS, &rmidi); - if (err < 0) + if (err < 0) return err; rmidi->private_data = pm; @@ -584,9 +582,9 @@ static int snd_portman_rawmidi_create(struct snd_card *card) pm->rmidi = rmidi; /* register rawmidi ops */ - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_portman_midi_output); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_portman_midi_input); /* name substreams */ @@ -621,7 +619,7 @@ static void snd_portman_interrupt(void *userdata) /* While any input data is waiting */ while ((portman_read_status(pm) & INT_REQ) == INT_REQ) { - /* If data available on channel 0, + /* If data available on channel 0, read it and stuff it into the queue. */ if (portman_data_avail(pm, 0)) { /* Read Midi */ @@ -632,7 +630,7 @@ static void snd_portman_interrupt(void *userdata) &midivalue, 1); } - /* If data available on channel 1, + /* If data available on channel 1, read it and stuff it into the queue. */ if (portman_data_avail(pm, 1)) { /* Read Midi */ @@ -658,7 +656,7 @@ static int snd_portman_probe_port(struct parport *p) 0, NULL); if (!pardev) return -EIO; - + if (parport_claim(pardev)) { parport_unregister_device(pardev); return -EIO; @@ -742,7 +740,7 @@ static int snd_portman_probe(struct platform_device *pdev) if (dev >= SNDRV_CARDS) return -ENODEV; - if (!enable[dev]) + if (!enable[dev]) return -ENOENT; if ((err = snd_portman_probe_port(p)) < 0) @@ -756,7 +754,7 @@ static int snd_portman_probe(struct platform_device *pdev) } strcpy(card->driver, DRIVER_NAME); strcpy(card->shortname, CARD_NAME); - sprintf(card->longname, "%s at 0x%lx, irq %i", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); pardev = parport_register_device(p, /* port */ @@ -779,7 +777,7 @@ static int snd_portman_probe(struct platform_device *pdev) } card->private_data = pm; card->private_free = snd_portman_card_private_free; - + if ((err = snd_portman_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); goto __err; @@ -823,7 +821,6 @@ static int snd_portman_remove(struct platform_device *pdev) return 0; } - static struct platform_driver snd_portman_driver = { .probe = snd_portman_probe, .remove = snd_portman_remove, @@ -844,7 +841,7 @@ static void snd_portman_unregister_all(void) platform_device_unregister(platform_devices[i]); platform_devices[i] = NULL; } - } + } platform_driver_unregister(&snd_portman_driver); parport_unregister_driver(&portman_parport_driver); } -- 1.9.1

2 14

[alsa-devel] [PATCH] ALSA: timer: Fix race at concurrent reads
by Takashi Iwai 09 Feb '16

09 Feb '16

snd_timer_user_read() has a potential race among parallel reads, as qhead and qused are updated outside the critical section due to copy_to_user() calls. Move them into the critical section, and also sanitize the relevant code a bit. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 34 +++++++++++++++------------------- 1 file changed, 15 insertions(+), 19 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index a0405b0078c6..dca817fc7894 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1933,6 +1933,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, { struct snd_timer_user *tu; long result = 0, unit; + int qhead; int err = 0; tu = file->private_data; @@ -1944,7 +1945,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { err = -EAGAIN; - break; + goto _error; } set_current_state(TASK_INTERRUPTIBLE); @@ -1959,42 +1960,37 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, if (tu->disconnected) { err = -ENODEV; - break; + goto _error; } if (signal_pending(current)) { err = -ERESTARTSYS; - break; + goto _error; } } + qhead = tu->qhead++; + tu->qhead %= tu->queue_size; spin_unlock_irq(&tu->qlock); - if (err < 0) - goto _error; if (tu->tread) { - if (copy_to_user(buffer, &tu->tqueue[tu->qhead++], - sizeof(struct snd_timer_tread))) { + if (copy_to_user(buffer, &tu->tqueue[qhead], + sizeof(struct snd_timer_tread))) err = -EFAULT; - goto _error; - } } else { - if (copy_to_user(buffer, &tu->queue[tu->qhead++], - sizeof(struct snd_timer_read))) { + if (copy_to_user(buffer, &tu->queue[qhead], + sizeof(struct snd_timer_read))) err = -EFAULT; - goto _error; - } } - tu->qhead %= tu->queue_size; - - result += unit; - buffer += unit; - spin_lock_irq(&tu->qlock); tu->qused--; + if (err < 0) + goto _error; + result += unit; + buffer += unit; } - spin_unlock_irq(&tu->qlock); _error: + spin_unlock_irq(&tu->qlock); return result > 0 ? result : err; } -- 2.7.1

1 0

[alsa-devel] [PATCH] ALSA: firewire-digi00x: Drop bogus const type qualifier on dot_scrt()
by Geert Uytterhoeven 09 Feb '16

09 Feb '16

sound/firewire/digi00x/amdtp-dot.c:67: warning: type qualifiers ignored on function return type Drop the bogus "const" type qualifier on the return type of dot_scrt() to fix this. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/firewire/digi00x/amdtp-dot.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index b02a5e8cad448c40..0ac92aba5bc1c9a4 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -63,7 +63,7 @@ struct amdtp_dot { #define BYTE_PER_SAMPLE (4) #define MAGIC_DOT_BYTE (2) #define MAGIC_BYTE_OFF(x) (((x) * BYTE_PER_SAMPLE) + MAGIC_DOT_BYTE) -static const u8 dot_scrt(const u8 idx, const unsigned int off) +static u8 dot_scrt(const u8 idx, const unsigned int off) { /* * the length of the added pattern only depends on the lower nibble -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: rsnd_ssi_is_multi_slave() macro uses rsnd_ssi_multi_slaves()
by Kuninori Morimoto 09 Feb '16

09 Feb '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> b4c83b171 ("ASoC: rsnd: add Multi channel support") added Multi channel support, and current rsnd_ssi_is_multi_slave()'s check method is !SSI equals SSIM1/2/3. But, SSI parent also hit to this macro. Because of this reason, some stream which needs SSI parent clock can't work correctly. This patch uses rsnd_ssi_multi_slaves() to solve this issue. This issue was reported by Dung. Reported-by: Nguyen Viet Dung <nv-dung(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Dung-san Can you please check/test this patch ? sound/soc/sh/rcar/ssi.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 7ee89da..72e8420 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -96,7 +96,8 @@ struct rsnd_ssi { #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) #define rsnd_ssi_mode_flags(p) ((p)->flags) #define rsnd_ssi_is_parent(ssi, io) ((ssi) == rsnd_io_to_mod_ssip(io)) -#define rsnd_ssi_is_multi_slave(ssi, io) ((mod) != rsnd_io_to_mod_ssi(io)) +#define rsnd_ssi_is_multi_slave(mod, io) \ + (rsnd_ssi_multi_slaves(io) & (1 << rsnd_mod_id(mod))) int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) { @@ -175,9 +176,6 @@ static int rsnd_ssi_irq_disable(struct rsnd_mod *ssi_mod) u32 rsnd_ssi_multi_slaves(struct rsnd_dai_stream *io) { struct rsnd_mod *mod; - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct device *dev = rsnd_priv_to_dev(priv); enum rsnd_mod_type types[] = { RSND_MOD_SSIM1, RSND_MOD_SSIM2, @@ -185,16 +183,6 @@ u32 rsnd_ssi_multi_slaves(struct rsnd_dai_stream *io) }; int i, mask; - switch (runtime->channels) { - case 2: /* Multi channel is not needed for Stereo */ - return 0; - case 6: - break; - default: - dev_err(dev, "unsupported channel\n"); - return 0; - } - mask = 0; for (i = 0; i < ARRAY_SIZE(types); i++) { mod = rsnd_io_to_mod(io, types[i]); -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: hda - Fix bad dereference of jack object
by Takashi Iwai 09 Feb '16

09 Feb '16

The hda_jack_tbl entries are managed by snd_array for allowing multiple jacks. It's good per se, but the problem is that struct hda_jack_callback keeps the hda_jack_tbl pointer. Since snd_array doesn't preserve each pointer at resizing the array, we can't keep the original pointer but have to deduce the pointer at each time via snd_array_entry() instead. Actually, this resulted in the deference to the wrong pointer on codecs that have many pins such as CS4208. This patch replaces the pointer to the NID value as the search key. As an unexpected good side effect, this even simplifies the code, as only NID is needed in most cases. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_jack.h | 2 +- sound/pci/hda/patch_ca0132.c | 5 ++++- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 6 +++--- 7 files changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 30c8efe0f80a..7ca5b89f088a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4028,9 +4028,9 @@ static void pin_power_callback(struct hda_codec *codec, struct hda_jack_callback *jack, bool on) { - if (jack && jack->tbl->nid) + if (jack && jack->nid) sync_power_state_change(codec, - set_pin_power_jack(codec, jack->tbl->nid, on)); + set_pin_power_jack(codec, jack->nid, on)); } /* callback only doing power up -- called at first */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index c945e257d368..a33234e04d4f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -259,7 +259,7 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, if (!callback) return ERR_PTR(-ENOMEM); callback->func = func; - callback->tbl = jack; + callback->nid = jack->nid; callback->next = jack->callback; jack->callback = callback; } diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 858708a044f5..e9814c0168ea 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -21,7 +21,7 @@ struct hda_jack_callback; typedef void (*hda_jack_callback_fn) (struct hda_codec *, struct hda_jack_callback *); struct hda_jack_callback { - struct hda_jack_tbl *tbl; + hda_nid_t nid; hda_jack_callback_fn func; unsigned int private_data; /* arbitrary data */ struct hda_jack_callback *next; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 4ef2259f88ca..9ceb2bc36e68 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4427,13 +4427,16 @@ static void ca0132_process_dsp_response(struct hda_codec *codec, static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) { struct ca0132_spec *spec = codec->spec; + struct hda_jack_tbl *tbl; /* Delay enabling the HP amp, to let the mic-detection * state machine run. */ cancel_delayed_work_sync(&spec->unsol_hp_work); schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); - cb->tbl->block_report = 1; + tbl = snd_hda_jack_tbl_get(codec, cb->nid); + if (tbl) + tbl->block_report = 1; } static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2191e2359315..8ee78dbd4c60 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1194,7 +1194,7 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid) static void jack_callback(struct hda_codec *codec, struct hda_jack_callback *jack) { - check_presence_and_report(codec, jack->tbl->nid); + check_presence_and_report(codec, jack->nid); } static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b43c0f7994eb..efd4980cffb8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -282,7 +282,7 @@ static void alc_update_knob_master(struct hda_codec *codec, uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (!uctl) return; - val = snd_hda_codec_read(codec, jack->tbl->nid, 0, + val = snd_hda_codec_read(codec, jack->nid, 0, AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); val &= HDA_AMP_VOLMASK; uctl->value.integer.value[0] = val; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2c7c5eb8b1e9..37b70f8e878f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -493,9 +493,9 @@ static void jack_update_power(struct hda_codec *codec, if (!spec->num_pwrs) return; - if (jack && jack->tbl->nid) { - stac_toggle_power_map(codec, jack->tbl->nid, - snd_hda_jack_detect(codec, jack->tbl->nid), + if (jack && jack->nid) { + stac_toggle_power_map(codec, jack->nid, + snd_hda_jack_detect(codec, jack->nid), true); return; } -- 2.7.1

1 0

[alsa-devel] [PATCH 0/7] ASoC: Intel: Skylake: Topology sequence changes
by Vinod Koul 09 Feb '16

09 Feb '16

This series does some minor tweaks to the topology sequences and supports a little more variety of topologies like a pipe that does not have mixer at the end and is connected by a switch. We add some helper functions and then support default params for internal sinks and sources Shutdown callback helps to get hardware to a cleaner state MCLK configuration allows codec to configure and control the mclk for these systems. While at it, also fix the delay code to be simple if case Dharageswari.R (3): ASoC: Intel: Skylake: Get node id using helper ASoC: Intel: Skylake: Add skl_tplg_be_get_cpr_module() helper ASoC: Intel: Skylake: Add MCLK configuration Jeeja KP (3): ASoC: Intel: Skylake: Allow module parameter set after bind ASoC: Intel: Skylake: Add shutdown callback ASoC: Intel: Skylake: Fill BE blob with default params Vinod Koul (1): ASoC: Intel: Skylake: Update the delay check sound/soc/intel/skylake/skl-messages.c | 62 ++++++++- sound/soc/intel/skylake/skl-pcm.c | 22 ++- sound/soc/intel/skylake/skl-topology.c | 193 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 27 ++++ sound/soc/intel/skylake/skl-tplg-interface.h | 3 +- sound/soc/intel/skylake/skl.c | 26 ++++ 6 files changed, 325 insertions(+), 8 deletions(-) -- 1.9.1

2 16

[alsa-devel] [PATCH 0/4] ASoC: Intel: Really fix the BYT/HSW regressions
by Vinod Koul 08 Feb '16

08 Feb '16

So we had some issues due to common acpi matching code which caused regression on BYT first and fixes on that caused regression on HSW. So this series revert the old fix and properly fixes by making matching module as a separate ones and then adding BYT PCI ids conditionally. This was tested on BYT/CHT platforms, HSW/BDW and SKL so test coverge should ensure that this is really fixed this time. Thanks to Jie Yang and Pierre for testing and suggesting fixes Pierre-Louis Bossart (1): ASoC: Intel: Load the atom DPCM driver only Vinod Koul (3): ASoC: Intel: Revert "ASoC: Intel: fix ACPI probe regression with Atom DPCM driver" ASoC: Intel: Create independent acpi match module ASoC: Intel: Add module tags for common match module sound/soc/intel/Kconfig | 13 +++++++++++-- sound/soc/intel/common/Makefile | 9 +++------ sound/soc/intel/common/sst-acpi.c | 4 ++++ sound/soc/intel/common/sst-match-acpi.c | 3 +++ 4 files changed, 21 insertions(+), 8 deletions(-) -- 1.9.1

2 8

[alsa-devel] [PATCH] ASoC: rt5514: add rt5514 SPI driver
by Oder Chiou 08 Feb '16

08 Feb '16

The patch adds the rt5514 SPI driver for loading the firmware of DSP and retrieving the voice data after the system is waked up by voice. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5514-spi.c | 494 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514-spi.h | 36 +++ sound/soc/codecs/rt5514.c | 243 +++++++++++++++++++++ sound/soc/codecs/rt5514.h | 1 + 6 files changed, 782 insertions(+) create mode 100644 sound/soc/codecs/rt5514-spi.c create mode 100644 sound/soc/codecs/rt5514-spi.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4383966..6a36c3a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -621,6 +621,12 @@ config SND_SOC_RT298 config SND_SOC_RT5514 tristate + default y if SND_SOC_RT5514_SPI=y + default m if SND_SOC_RT5514_SPI=m + +config SND_SOC_RT5514_SPI + tristate + depends on SPI_MASTER config SND_SOC_RT5616 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cd05e70..c04d5b6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -96,6 +96,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o snd-soc-rt5514-objs := rt5514.o +snd-soc-rt5514-spi-objs := rt5514-spi.o snd-soc-rt5616-objs := rt5616.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -304,6 +305,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o +obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c new file mode 100644 index 0000000..5889533 --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.c @@ -0,0 +1,494 @@ +/* + * rt5514-spi.c -- RT5514 SPI driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/input.h> +#include <linux/spi/spi.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/sched.h> +#include <linux/kthread.h> +#include <linux/uaccess.h> +#include <linux/miscdevice.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_qos.h> +#include <linux/sysfs.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5514-spi.h" + +static struct spi_device *rt5514_spi; + +struct rt5514_dsp { + struct device *dev; + struct delayed_work copy_work; + struct mutex dma_lock; + struct snd_pcm_substream *substream; + unsigned int buf_base, buf_limit, buf_rp; + size_t buf_size; + size_t dma_offset; + size_t dsp_offset; +}; + +static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = 0x20000 / 8, + .periods_min = 8, + .periods_max = 8, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 0x20000, +}; + +static struct snd_soc_dai_driver rt5514_spi_dai = { + .name = "rt5514-dsp-cpu-dai", + .id = 0, + .capture = { + .stream_name = "DSP Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static void rt5514_spi_copy_work(struct work_struct *work) +{ + struct rt5514_dsp *rt5514_dsp = + container_of(work, struct rt5514_dsp, copy_work.work); + struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; + size_t period_bytes, truncated_bytes = 0; + + mutex_lock(&rt5514_dsp->dma_lock); + if (!rt5514_dsp->substream) { + dev_err(rt5514_dsp->dev, "No pcm substream\n"); + goto done; + } + + period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + + if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) + period_bytes = rt5514_dsp->buf_size - rt5514_dsp->dsp_offset; + + if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) { + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + period_bytes); + + if (rt5514_dsp->buf_rp + period_bytes == rt5514_dsp->buf_limit) + rt5514_dsp->buf_rp = rt5514_dsp->buf_base; + else + rt5514_dsp->buf_rp += period_bytes; + } else { + truncated_bytes = rt5514_dsp->buf_limit - rt5514_dsp->buf_rp; + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + truncated_bytes); + + rt5514_spi_burst_read(rt5514_dsp->buf_base, + runtime->dma_area + rt5514_dsp->dma_offset + + truncated_bytes, period_bytes - truncated_bytes); + + rt5514_dsp->buf_rp = rt5514_dsp->buf_base + + period_bytes - truncated_bytes; + } + + rt5514_dsp->dma_offset += period_bytes; + if (rt5514_dsp->dma_offset >= runtime->dma_bytes) + rt5514_dsp->dma_offset = 0; + + rt5514_dsp->dsp_offset += period_bytes; + + snd_pcm_period_elapsed(rt5514_dsp->substream); + + if (rt5514_dsp->dsp_offset < rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 5); +done: + mutex_unlock(&rt5514_dsp->dma_lock); +} + +/* PCM for streaming audio from the DSP buffer */ +static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &rt5514_spi_pcm_hardware); + + return 0; +} + +static int rt5514_spi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + int ret; + + mutex_lock(&rt5514_dsp->dma_lock); + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + rt5514_dsp->substream = substream; + mutex_unlock(&rt5514_dsp->dma_lock); + + return ret; +} + +static int rt5514_spi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + mutex_lock(&rt5514_dsp->dma_lock); + cancel_delayed_work_sync(&rt5514_dsp->copy_work); + rt5514_dsp->substream = NULL; + mutex_unlock(&rt5514_dsp->dma_lock); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int rt5514_spi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + rt5514_dsp->dma_offset = 0; + rt5514_dsp->dsp_offset = 0; + rt5514_spi_read_addr(RT5514_BUFFER_VOICE_BASE, &rt5514_dsp->buf_base); + rt5514_spi_read_addr(RT5514_BUFFER_VOICE_LIMIT, &rt5514_dsp->buf_limit); + rt5514_spi_read_addr(RT5514_BUFFER_VOICE_RP, &rt5514_dsp->buf_rp); + rt5514_spi_read_addr(RT5514_BUFFER_VOICE_SIZE, &rt5514_dsp->buf_size); + + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && + rt5514_dsp->buf_rp && rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 0); + + return 0; +} + +static snd_pcm_uframes_t rt5514_spi_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + return bytes_to_frames(runtime, rt5514_dsp->dma_offset); +} + +static struct snd_pcm_ops rt5514_spi_pcm_ops = { + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .prepare = rt5514_spi_prepare, + .pointer = rt5514_spi_pcm_pointer, + .mmap = snd_pcm_lib_mmap_vmalloc, + .page = snd_pcm_lib_get_vmalloc_page, +}; + +static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) +{ + struct rt5514_dsp *rt5514_dsp; + + rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), + GFP_KERNEL); + + rt5514_dsp->dev = &rt5514_spi->dev; + mutex_init(&rt5514_dsp->dma_lock); + INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work); + snd_soc_platform_set_drvdata(platform, rt5514_dsp); + + return 0; +} + +static struct snd_soc_platform_driver rt5514_spi_platform = { + .probe = rt5514_spi_pcm_probe, + .ops = &rt5514_spi_pcm_ops, +}; + +static const struct snd_soc_component_driver rt5514_spi_dai_component = { + .name = "rt5514-spi-dai", +}; + +int rt5514_spi_read_addr(unsigned int addr, unsigned int *val) +{ + struct spi_device *spi = rt5514_spi; + struct spi_message message; + struct spi_transfer x[3]; + u8 spi_cmd = RT5514_SPI_CMD_32_READ; + int status; + u8 write_buf[5]; + u8 read_buf[4]; + + write_buf[0] = spi_cmd; + write_buf[1] = (addr & 0xff000000) >> 24; + write_buf[2] = (addr & 0x00ff0000) >> 16; + write_buf[3] = (addr & 0x0000ff00) >> 8; + write_buf[4] = (addr & 0x000000ff) >> 0; + + spi_message_init(&message); + memset(x, 0, sizeof(x)); + + x[0].len = 5; + x[0].tx_buf = write_buf; + spi_message_add_tail(&x[0], &message); + + x[1].len = 4; + x[1].tx_buf = write_buf; + spi_message_add_tail(&x[1], &message); + + x[2].len = 4; + x[2].rx_buf = read_buf; + spi_message_add_tail(&x[2], &message); + + status = spi_sync(spi, &message); + + *val = read_buf[3] | read_buf[2] << 8 | read_buf[1] << 16 | + read_buf[0] << 24; + + return status; +} +EXPORT_SYMBOL_GPL(rt5514_spi_read_addr); + +int rt5514_spi_write_addr(unsigned int addr, unsigned int val) +{ + struct spi_device *spi = rt5514_spi; + u8 spi_cmd = RT5514_SPI_CMD_32_WRITE; + int status; + u8 write_buf[10]; + + write_buf[0] = spi_cmd; + write_buf[1] = (addr & 0xff000000) >> 24; + write_buf[2] = (addr & 0x00ff0000) >> 16; + write_buf[3] = (addr & 0x0000ff00) >> 8; + write_buf[4] = (addr & 0x000000ff) >> 0; + write_buf[5] = (val & 0xff000000) >> 24; + write_buf[6] = (val & 0x00ff0000) >> 16; + write_buf[7] = (val & 0x0000ff00) >> 8; + write_buf[8] = (val & 0x000000ff) >> 0; + write_buf[9] = spi_cmd; + + status = spi_write(spi, write_buf, sizeof(write_buf)); + + if (status) + dev_err(&spi->dev, "%s error %d\n", __func__, status); + + return status; +} +EXPORT_SYMBOL_GPL(rt5514_spi_write_addr); + +/** + * rt5514_spi_burst_read - Read data from SPI by rt5514 address. + * @addr: Start address. + * @rxbuf: Data Buffer for reading. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_READ; + int status; + u8 write_buf[8]; + unsigned int i, end, offset = 0; + + struct spi_message message; + struct spi_transfer x[3]; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + spi_message_init(&message); + memset(x, 0, sizeof(x)); + + x[0].len = 5; + x[0].tx_buf = write_buf; + spi_message_add_tail(&x[0], &message); + + x[1].len = 4; + x[1].tx_buf = write_buf; + spi_message_add_tail(&x[1], &message); + + x[2].len = end; + x[2].rx_buf = rxbuf + offset; + spi_message_add_tail(&x[2], &message); + + status = spi_sync(rt5514_spi, &message); + + if (status) + return false; + + offset += RT5514_SPI_BUF_LEN; + } + + for (i = 0; i < len; i += 8) { + write_buf[0] = rxbuf[i + 0]; + write_buf[1] = rxbuf[i + 1]; + write_buf[2] = rxbuf[i + 2]; + write_buf[3] = rxbuf[i + 3]; + write_buf[4] = rxbuf[i + 4]; + write_buf[5] = rxbuf[i + 5]; + write_buf[6] = rxbuf[i + 6]; + write_buf[7] = rxbuf[i + 7]; + + rxbuf[i + 0] = write_buf[7]; + rxbuf[i + 1] = write_buf[6]; + rxbuf[i + 2] = write_buf[5]; + rxbuf[i + 3] = write_buf[4]; + rxbuf[i + 4] = write_buf[3]; + rxbuf[i + 5] = write_buf[2]; + rxbuf[i + 6] = write_buf[1]; + rxbuf[i + 7] = write_buf[0]; + } + + return true; +} +EXPORT_SYMBOL_GPL(rt5514_spi_burst_read); + +/** + * rt5514_spi_burst_write - Write data to SPI by rt5514 address. + * @addr: Start address. + * @txbuf: Data Buffer for writng. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_WRITE; + u8 *write_buf; + unsigned int i, end, offset = 0; + + write_buf = kmalloc(RT5514_SPI_BUF_LEN + 6, GFP_KERNEL); + + if (write_buf == NULL) + return -ENOMEM; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + for (i = 0; i < end; i += 8) { + write_buf[i + 12] = txbuf[offset + i + 0]; + write_buf[i + 11] = txbuf[offset + i + 1]; + write_buf[i + 10] = txbuf[offset + i + 2]; + write_buf[i + 9] = txbuf[offset + i + 3]; + write_buf[i + 8] = txbuf[offset + i + 4]; + write_buf[i + 7] = txbuf[offset + i + 5]; + write_buf[i + 6] = txbuf[offset + i + 6]; + write_buf[i + 5] = txbuf[offset + i + 7]; + } + + write_buf[end + 5] = spi_cmd; + + spi_write(rt5514_spi, write_buf, end + 6); + + offset += RT5514_SPI_BUF_LEN; + } + + kfree(write_buf); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5514_spi_burst_write); + +static int rt5514_spi_probe(struct spi_device *spi) +{ + int ret; + + rt5514_spi = spi; + + ret = snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register platform.\n"); + goto err_plat; + } + + ret = snd_soc_register_component(&spi->dev, &rt5514_spi_dai_component, + &rt5514_spi_dai, 1); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register component.\n"); + goto err_comp; + } + + return 0; +err_comp: + snd_soc_unregister_platform(&spi->dev); +err_plat: + + return 0; +} + +static int rt5514_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_component(&spi->dev); + snd_soc_unregister_platform(&spi->dev); + + return 0; +} + +#if defined(CONFIG_OF) +static const struct of_device_id rt5514_of_match[] = { + { .compatible = "realtek,rt5514", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5514_of_match); +#endif + +static struct spi_driver rt5514_spi_driver = { + .driver = { + .name = "rt5514", + .of_match_table = of_match_ptr(rt5514_of_match), + }, + .probe = rt5514_spi_probe, + .remove = rt5514_spi_remove, +}; +module_spi_driver(rt5514_spi_driver); + +MODULE_DESCRIPTION("RT5514 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h new file mode 100644 index 0000000..68b73f4 --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.h @@ -0,0 +1,36 @@ +/* + * rt5514-spi.h -- RT5514 driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5514_SPI_H__ +#define __RT5514_SPI_H__ + +#define RT5514_SPI_BUF_LEN 240 +#define RT5514_BUFFER_VOICE_BASE 0x18001034 +#define RT5514_BUFFER_VOICE_LIMIT 0x18001038 +#define RT5514_BUFFER_VOICE_RP 0x1800103c +#define RT5514_BUFFER_VOICE_SIZE 0x18001040 + +/* SPI Command */ +enum { + RT5514_SPI_CMD_16_READ = 0, + RT5514_SPI_CMD_16_WRITE, + RT5514_SPI_CMD_32_READ, + RT5514_SPI_CMD_32_WRITE, + RT5514_SPI_CMD_BURST_READ, + RT5514_SPI_CMD_BURST_WRITE, +}; + +int rt5514_spi_read_addr(unsigned int addr, unsigned int *val); +int rt5514_spi_write_addr(unsigned int addr, unsigned int val); +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len); +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5514_SPI_H__ */ diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 879bf60..a3838c0 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -30,6 +30,9 @@ #include "rl6231.h" #include "rt5514.h" +#if defined(CONFIG_SND_SOC_RT5514_SPI) +#include "rt5514-spi.h" +#endif static const struct reg_sequence rt5514_i2c_patch[] = { {0x1800101c, 0x00000000}, @@ -110,6 +113,159 @@ static const struct reg_default rt5514_reg[] = { {RT5514_VENDOR_ID2, 0x10ec5514}, }; +static void rt5514_enable_dsp_clock(struct rt5514_priv *rt5514) +{ + /* Reset */ + regmap_write(rt5514->i2c_regmap, 0x18002000, 0x000010ec); + /* LDO_I_limit */ + regmap_write(rt5514->i2c_regmap, 0x18002200, 0x00028604); + /* (for reset DSP) mini-core reset */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x0005514b); + /* (for reset DSP) mini-core reset */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055149); + /* DMIC_OUT1/2=HiZ */ + regmap_write(rt5514->i2c_regmap, 0x180020a0, 0x00000000); + /* PIN config */ + regmap_write(rt5514->i2c_regmap, 0x18002070, 0x00000040); + /* pll3(QN)=RCOSC*(10+2) */ + regmap_write(rt5514->i2c_regmap, 0x18002240, 0x0000000a); + /* pll3 source=RCOSC, fsi=rt_clk */ + regmap_write(rt5514->i2c_regmap, 0x18002100, 0x0000000b); + /* PU RCOSC, pll3 */ + regmap_write(rt5514->i2c_regmap, 0x18002004, 0x00808b81); + /* PD ADC1/2 */ + regmap_write(rt5514->i2c_regmap, 0x18002008, 0x00220000); + /* DSP clk source=pll3, ENABLE DSP clk */ + regmap_write(rt5514->i2c_regmap, 0x18002f08, 0x00000005); + /* 256fs=/4,DMIC_CLK_OUT=/16(disable ad2) */ + regmap_write(rt5514->i2c_regmap, 0x18002104, 0x10023541); + /* clk_sys source=mux_out */ + regmap_write(rt5514->i2c_regmap, 0x18002108, 0x00000000); + /* DSP clk= 15M*(20+1)/32 =10M */ + regmap_write(rt5514->i2c_regmap, 0x18001100, 0x00000214); + /* DB, pointer */ + regmap_write(rt5514->i2c_regmap, 0x18002148, 0x80000000); + /* DB, pop=4x */ + regmap_write(rt5514->i2c_regmap, 0x18002140, 0x3fff00fa); + /* DB, pointer */ + regmap_write(rt5514->i2c_regmap, 0x18002148, 0x00000000); + /* DFLL reset */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0x00220012); + /* DFLL, set m/n */ + regmap_write(rt5514->i2c_regmap, 0x18002110, 0x000101f4); + /* DFLL,reset DFLL */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0x80220012); + /* DFLL */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0xc0220012); + /* (I2S) i2s format, TDM 4ch */ + regmap_write(rt5514->i2c_regmap, 0x18002010, 0x10000772); + /* (I2S) source sel; tdm_0=ad0, tdm_1=ad1 */ + regmap_write(rt5514->i2c_regmap, 0x180020ac, 0x44000eee); + /* (ad0) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x18002190, 0x0002042f); + /* (ad0) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x18002194, 0x0002042f); + /* (ad0) DMIC-IN1 L/R select */ + regmap_write(rt5514->i2c_regmap, 0x18002198, 0x10000362); + /* (ad1) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x180021a0, 0x0002042f); + /* (ad1) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x180021a4, 0x0002042f); + /* (ad1) DMIC-IN2 L/R select */ + regmap_write(rt5514->i2c_regmap, 0x180021a8, 0x10000362); + /* (ad2) gain=24dB for WOV */ + regmap_write(rt5514->i2c_regmap, 0x180020d0, 0x00008a2f); + /* dsp clk auto switch */ + regmap_write(rt5514->i2c_regmap, 0x18001114, 0x00000001); + /* reduce DSP power */ + regmap_write(rt5514->i2c_regmap, 0x18001118, 0x00000001); + /* UART clk=off */ + regmap_write(rt5514->i2c_regmap, 0x18001104, 0x00000003); + /* (pitch VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x1800201c, 0x69f32067); + /* (pitch VAD)fix2 */ + regmap_write(rt5514->i2c_regmap, 0x18002020, 0x50d500a5); + /* (pitch VAD)fix3 */ + regmap_write(rt5514->i2c_regmap, 0x18002024, 0x000a0206); + /* (pitch VAD)fix4 */ + regmap_write(rt5514->i2c_regmap, 0x18002028, 0x01800114); + /* (hello VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x18002038, 0x00100010); + /* (ok VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x1800204c, 0x000503c8); + /* (ok VAD)fix2 */ + regmap_write(rt5514->i2c_regmap, 0x18002050, 0x001a0308); + /* (ok VAD)fix3 */ + regmap_write(rt5514->i2c_regmap, 0x18002054, 0x50020502); + /* (ok VAD)fix4 */ + regmap_write(rt5514->i2c_regmap, 0x18002058, 0x50000d18); + /* (ok VAD)fix5 */ + regmap_write(rt5514->i2c_regmap, 0x1800205c, 0x640c0b14); + /* (ok VAD)fix6 */ + regmap_write(rt5514->i2c_regmap, 0x18002060, 0x00100001); + /* (FW) SENSORY_SVSCORE(for UDT+SID) */ + regmap_write(rt5514->i2c_regmap, 0x18002fa4, 0x00000000); + /* (FW) SENSORY_THRS(for UDT+SID) */ + regmap_write(rt5514->i2c_regmap, 0x18002fa8, 0x00000000); + /* (FW) DLY_BUF_LTC_OFFSET (for ok/hello VAD) */ + regmap_write(rt5514->i2c_regmap, 0x18002fbc, 0x00000000); +} + +static void rt5514_enable_dsp(struct rt5514_priv *rt5514) +{ + /* (FW) DRIVER_CTRL0: (1)VAD timeout[7:0]: unit = 0.1s + * (2)buffer mode[15:14]: + * 0 = key phrase+voice command + * 1 = voice command + * 2 = key phrase + */ + regmap_write(rt5514->i2c_regmap, 0x18001028, 0x0000000a); + /* PU RCOSC, pll3 */ + regmap_write(rt5514->i2c_regmap, 0x18002004, 0x00808b81); + /* PD ADC1/2 */ + regmap_write(rt5514->i2c_regmap, 0x18002008, 0x00220000); + /* dsp stop */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055149); + /* CLK, 256fs=/4,DMIC_CLK_OUT=/16(enable ad2) */ + regmap_write(rt5514->i2c_regmap, 0x18002104, 0x14023541); + /* CLK, clk_sys=mux_out/1 */ + regmap_write(rt5514->i2c_regmap, 0x18002108, 0x00000000); + /* (PATH) DMIC_IN1(ri)->ad2, ad2(db_PCM)->IB2 */ + regmap_write(rt5514->i2c_regmap, 0x180020a4, 0x00808002); + /* DSP clk source=pll3, ENABLE DSP clk */ + regmap_write(rt5514->i2c_regmap, 0x18002f08, 0x00000005); + /* (opt)VAD, clr and enable pitch/hello VAD + * (0x800007d2 for ok VAD, 0x800007d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x800007d3); + /* (opt)VAD, clr and enable ok VAD + * (0x80000003 for ok VAD, 0x80000002 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x80000002); + /* (opt)VAD type sel(..90h:hello, ..80h:pitch, ..A0h:ok) */ + regmap_write(rt5514->i2c_regmap, 0x1800206c, 0x80000080); + /* dsp run */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); + /* clear IRQ */ + regmap_write(rt5514->i2c_regmap, 0x18002e04, 0x00000000); + /* (opt)VAD, release and enable pitch/hello VAD + * (0x7d2 for ok VAD, 0x7d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x000007d3); + /* (opt)VAD, release and enable ok VAD + * (0x3 for ok VAD, 0x2 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x00000002); + /* (opt)VAD, release and enable pitch/hello VAD + * (0x7d2 for ok VAD, 0x7d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x000007d3); + /* (opt)VAD, release and enable ok VAD + * (0x3 for ok VAD, 0x2 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x00000002); +} + static bool rt5514_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { @@ -248,6 +404,64 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static int rt5514_dsp_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt5514->dsp_mode; + + return 0; +} + +static int rt5514_dsp_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + const struct firmware *fw = NULL; + + if (ucontrol->value.integer.value[0] == rt5514->dsp_mode) + return 0; + + rt5514->dsp_mode = ucontrol->value.integer.value[0]; + + if (rt5514->dsp_mode) { + rt5514_enable_dsp_clock(rt5514); + + request_firmware(&fw, "0x4ff60000.dat", codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ff60000, fw->data, + ((fw->size/8)+1)*8); +#endif + release_firmware(fw); + fw = NULL; + } + + request_firmware(&fw, "0x4ffc0000.dat", codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ffc0000, fw->data, + ((fw->size/8)+1)*8); +#endif + release_firmware(fw); + fw = NULL; + } + + rt5514_enable_dsp(rt5514); + } else { + regcache_mark_dirty(rt5514->i2c_regmap); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->i2c_regmap); + regcache_sync(rt5514->regmap); + } + + return 0; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -257,6 +471,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_R_TLV("ADC2 Capture Volume", RT5514_DOWNFILTER1_CTRL1, RT5514_DOWNFILTER1_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, adc_vol_tlv), + SOC_SINGLE_EXT("DSP Control", SND_SOC_NOPM, 0, 1, 0, + rt5514_dsp_mode_get, rt5514_dsp_mode_put), }; /* ADC Mixer*/ @@ -799,6 +1015,32 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5514_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (snd_soc_codec_get_bias_level(codec) == + SND_SOC_BIAS_STANDBY) { + if (rt5514->dsp_mode) { + rt5514->dsp_mode = 0; + regcache_mark_dirty(rt5514->i2c_regmap); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->i2c_regmap); + regcache_sync(rt5514->regmap); + } + } + break; + + default: + break; + } + + return 0; +} + static int rt5514_probe(struct snd_soc_codec *codec) { struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); @@ -857,6 +1099,7 @@ struct snd_soc_dai_driver rt5514_dai[] = { static struct snd_soc_codec_driver soc_codec_dev_rt5514 = { .probe = rt5514_probe, + .set_bias_level = rt5514_set_bias_level, .idle_bias_off = true, .controls = rt5514_snd_controls, .num_controls = ARRAY_SIZE(rt5514_snd_controls), diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 6ad8a61..80a5492 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -247,6 +247,7 @@ struct rt5514_priv { int pll_src; int pll_in; int pll_out; + int dsp_mode; }; #endif /* __RT5514_H__ */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v5 0/2] ASoC: Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 08 Feb '16

08 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver. For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK --- Changes since v4: - corrected indentation issues - removed old macros - removed unnecessary enabling of apb clock Changes since v3: - removed code required for spdif capture - added disable/enable of apb clk to suspend/resume. - removed unnecessary dt checks in probe - fixed error conditions in probe/resume - modified pm_runtime functionality and confirmed that driver worked when PM was disabled. - removed .owner from platform driver - renamed bindings file Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): dt-bindings: sunxi :add sun4i SPDIF transceiver ASoC: sunxi: Add support for the SPDIF block .../bindings/sound/sunxi,sun4i-spdif.txt | 39 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 549 +++++++++++++++++++++ 4 files changed, 597 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.1

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[alsa-devel] [PATCH] ASoC: rt5514: Fix up calculation for bclk_ms
by Axel Lin 08 Feb '16

08 Feb '16

The equation to calculate bclk_ms is wrong, fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 879bf60..5bfe720 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -574,7 +574,7 @@ static int rt5514_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_ms = frame_size > 32; + bclk_ms = frame_size >> 32; rt5514->bclk = rt5514->lrck * (32 << bclk_ms); dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", -- 2.1.4

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[alsa-devel] [PATCH] [PATCH] ALSA: Remove deprecated AU1X00 AC97 driver
by Lars-Peter Clausen 08 Feb '16

08 Feb '16

The AU1X00 AC97 ALSA driver was deprecated in commit 7137c6bcb7ff ("ALSA: deprecate MIPS AU1X00 AC97 driver") in favor of the newer and better ASoC driver for the same hardware. This was almost 5 years ago and this driver has not been in use in the mainline kernel since, it should be safe to remove it at this point. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 2153d31..4a47050 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -23,17 +23,5 @@ config SND_SGI_HAL2 help Sound support for the SGI Indy and Indigo2 Workstation. - -config SND_AU1X00 - tristate "Au1x00 AC97 Port Driver (DEPRECATED)" - depends on MIPS_ALCHEMY - select SND_PCM - select SND_AC97_CODEC - help - ALSA Sound driver for the Au1x00's AC97 port. - - Newer drivers for ASoC are available, please do not use - this driver as it will be removed in the future. - endif # SND_MIPS diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 861ec0a..b977c44 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -2,11 +2,9 @@ # Makefile for ALSA # -snd-au1x00-objs := au1x00.o snd-sgi-o2-objs := sgio2audio.o ad1843.o snd-sgi-hal2-objs := hal2.o # Toplevel Module Dependency -obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c deleted file mode 100644 index 1e30e84..0000000 --- a/sound/mips/au1x00.c +++ /dev/null @@ -1,734 +0,0 @@ -/* - * BRIEF MODULE DESCRIPTION - * Driver for AMD Au1000 MIPS Processor, AC'97 Sound Port - * - * Copyright 2004 Cooper Street Innovations Inc. - * Author: Charles Eidsness <charles(a)cooper-street.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN - * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, - * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT - * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF - * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON - * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT - * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF - * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - * History: - * - * 2004-09-09 Charles Eidsness -- Original verion -- based on - * sa11xx-uda1341.c ALSA driver and the - * au1000.c OSS driver. - * 2004-09-09 Matt Porter -- Added support for ALSA 1.0.6 - * - */ - -#include <linux/ioport.h> -#include <linux/interrupt.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/initval.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/ac97_codec.h> -#include <asm/mach-au1x00/au1000.h> -#include <asm/mach-au1x00/au1000_dma.h> - -MODULE_AUTHOR("Charles Eidsness <charles(a)cooper-street.com>"); -MODULE_DESCRIPTION("Au1000 AC'97 ALSA Driver"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{AMD,Au1000 AC'97}}"); - -#define PLAYBACK 0 -#define CAPTURE 1 -#define AC97_SLOT_3 0x01 -#define AC97_SLOT_4 0x02 -#define AC97_SLOT_6 0x08 -#define AC97_CMD_IRQ 31 -#define READ 0 -#define WRITE 1 -#define READ_WAIT 2 -#define RW_DONE 3 - -struct au1000_period -{ - u32 start; - u32 relative_end; /*realtive to start of buffer*/ - struct au1000_period * next; -}; - -/*Au1000 AC97 Port Control Reisters*/ -struct au1000_ac97_reg { - u32 volatile config; - u32 volatile status; - u32 volatile data; - u32 volatile cmd; - u32 volatile cntrl; -}; - -struct audio_stream { - struct snd_pcm_substream *substream; - int dma; - spinlock_t dma_lock; - struct au1000_period * buffer; - unsigned int period_size; - unsigned int periods; -}; - -struct snd_au1000 { - struct snd_card *card; - struct au1000_ac97_reg volatile *ac97_ioport; - - struct resource *ac97_res_port; - spinlock_t ac97_lock; - struct snd_ac97 *ac97; - - struct snd_pcm *pcm; - struct audio_stream *stream[2]; /* playback & capture */ - int dmaid[2]; /* tx(0)/rx(1) DMA ids */ -}; - -/*--------------------------- Local Functions --------------------------------*/ -static void -au1000_set_ac97_xmit_slots(struct snd_au1000 *au1000, long xmit_slots) -{ - u32 volatile ac97_config; - - spin_lock(&au1000->ac97_lock); - ac97_config = au1000->ac97_ioport->config; - ac97_config = ac97_config & ~AC97C_XMIT_SLOTS_MASK; - ac97_config |= (xmit_slots << AC97C_XMIT_SLOTS_BIT); - au1000->ac97_ioport->config = ac97_config; - spin_unlock(&au1000->ac97_lock); -} - -static void -au1000_set_ac97_recv_slots(struct snd_au1000 *au1000, long recv_slots) -{ - u32 volatile ac97_config; - - spin_lock(&au1000->ac97_lock); - ac97_config = au1000->ac97_ioport->config; - ac97_config = ac97_config & ~AC97C_RECV_SLOTS_MASK; - ac97_config |= (recv_slots << AC97C_RECV_SLOTS_BIT); - au1000->ac97_ioport->config = ac97_config; - spin_unlock(&au1000->ac97_lock); -} - - -static void -au1000_release_dma_link(struct audio_stream *stream) -{ - struct au1000_period * pointer; - struct au1000_period * pointer_next; - - stream->period_size = 0; - stream->periods = 0; - pointer = stream->buffer; - if (! pointer) - return; - do { - pointer_next = pointer->next; - kfree(pointer); - pointer = pointer_next; - } while (pointer != stream->buffer); - stream->buffer = NULL; -} - -static int -au1000_setup_dma_link(struct audio_stream *stream, unsigned int period_bytes, - unsigned int periods) -{ - struct snd_pcm_substream *substream = stream->substream; - struct snd_pcm_runtime *runtime = substream->runtime; - struct au1000_period *pointer; - unsigned long dma_start; - int i; - - dma_start = virt_to_phys(runtime->dma_area); - - if (stream->period_size == period_bytes && - stream->periods == periods) - return 0; /* not changed */ - - au1000_release_dma_link(stream); - - stream->period_size = period_bytes; - stream->periods = periods; - - stream->buffer = kmalloc(sizeof(struct au1000_period), GFP_KERNEL); - if (! stream->buffer) - return -ENOMEM; - pointer = stream->buffer; - for (i = 0; i < periods; i++) { - pointer->start = (u32)(dma_start + (i * period_bytes)); - pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1); - if (i < periods - 1) { - pointer->next = kmalloc(sizeof(struct au1000_period), GFP_KERNEL); - if (! pointer->next) { - au1000_release_dma_link(stream); - return -ENOMEM; - } - pointer = pointer->next; - } - } - pointer->next = stream->buffer; - return 0; -} - -static void -au1000_dma_stop(struct audio_stream *stream) -{ - if (snd_BUG_ON(!stream->buffer)) - return; - disable_dma(stream->dma); -} - -static void -au1000_dma_start(struct audio_stream *stream) -{ - if (snd_BUG_ON(!stream->buffer)) - return; - - init_dma(stream->dma); - if (get_dma_active_buffer(stream->dma) == 0) { - clear_dma_done0(stream->dma); - set_dma_addr0(stream->dma, stream->buffer->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - set_dma_addr1(stream->dma, stream->buffer->next->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - } else { - clear_dma_done1(stream->dma); - set_dma_addr1(stream->dma, stream->buffer->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - set_dma_addr0(stream->dma, stream->buffer->next->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - } - enable_dma_buffers(stream->dma); - start_dma(stream->dma); -} - -static irqreturn_t -au1000_dma_interrupt(int irq, void *dev_id) -{ - struct audio_stream *stream = (struct audio_stream *) dev_id; - struct snd_pcm_substream *substream = stream->substream; - - spin_lock(&stream->dma_lock); - switch (get_dma_buffer_done(stream->dma)) { - case DMA_D0: - stream->buffer = stream->buffer->next; - clear_dma_done0(stream->dma); - set_dma_addr0(stream->dma, stream->buffer->next->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - enable_dma_buffer0(stream->dma); - break; - case DMA_D1: - stream->buffer = stream->buffer->next; - clear_dma_done1(stream->dma); - set_dma_addr1(stream->dma, stream->buffer->next->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - enable_dma_buffer1(stream->dma); - break; - case (DMA_D0 | DMA_D1): - printk(KERN_ERR "DMA %d missed interrupt.\n",stream->dma); - au1000_dma_stop(stream); - au1000_dma_start(stream); - break; - case (~DMA_D0 & ~DMA_D1): - printk(KERN_ERR "DMA %d empty irq.\n",stream->dma); - } - spin_unlock(&stream->dma_lock); - snd_pcm_period_elapsed(substream); - return IRQ_HANDLED; -} - -/*-------------------------- PCM Audio Streams -------------------------------*/ - -static unsigned int rates[] = {8000, 11025, 16000, 22050}; -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct snd_pcm_hardware snd_au1000_hw = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | \ - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050), - .rate_min = 8000, - .rate_max = 22050, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = 32, - .period_bytes_max = 16*1024, - .periods_min = 8, - .periods_max = 255, - .fifo_size = 16, -}; - -static int -snd_au1000_playback_open(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[PLAYBACK]->substream = substream; - au1000->stream[PLAYBACK]->buffer = NULL; - substream->private_data = au1000->stream[PLAYBACK]; - substream->runtime->hw = snd_au1000_hw; - return (snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0); -} - -static int -snd_au1000_capture_open(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[CAPTURE]->substream = substream; - au1000->stream[CAPTURE]->buffer = NULL; - substream->private_data = au1000->stream[CAPTURE]; - substream->runtime->hw = snd_au1000_hw; - return (snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0); -} - -static int -snd_au1000_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[PLAYBACK]->substream = NULL; - return 0; -} - -static int -snd_au1000_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[CAPTURE]->substream = NULL; - return 0; -} - -static int -snd_au1000_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct audio_stream *stream = substream->private_data; - int err; - - err = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - return err; - return au1000_setup_dma_link(stream, - params_period_bytes(hw_params), - params_periods(hw_params)); -} - -static int -snd_au1000_hw_free(struct snd_pcm_substream *substream) -{ - struct audio_stream *stream = substream->private_data; - au1000_release_dma_link(stream); - return snd_pcm_lib_free_pages(substream); -} - -static int -snd_au1000_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - if (runtime->channels == 1) - au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_4); - else - au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4); - snd_ac97_set_rate(au1000->ac97, AC97_PCM_FRONT_DAC_RATE, runtime->rate); - return 0; -} - -static int -snd_au1000_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - if (runtime->channels == 1) - au1000_set_ac97_recv_slots(au1000, AC97_SLOT_4); - else - au1000_set_ac97_recv_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4); - snd_ac97_set_rate(au1000->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate); - return 0; -} - -static int -snd_au1000_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct audio_stream *stream = substream->private_data; - int err = 0; - - spin_lock(&stream->dma_lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - au1000_dma_start(stream); - break; - case SNDRV_PCM_TRIGGER_STOP: - au1000_dma_stop(stream); - break; - default: - err = -EINVAL; - break; - } - spin_unlock(&stream->dma_lock); - return err; -} - -static snd_pcm_uframes_t -snd_au1000_pointer(struct snd_pcm_substream *substream) -{ - struct audio_stream *stream = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - long location; - - spin_lock(&stream->dma_lock); - location = get_dma_residue(stream->dma); - spin_unlock(&stream->dma_lock); - location = stream->buffer->relative_end - location; - if (location == -1) - location = 0; - return bytes_to_frames(runtime,location); -} - -static struct snd_pcm_ops snd_card_au1000_playback_ops = { - .open = snd_au1000_playback_open, - .close = snd_au1000_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_au1000_hw_params, - .hw_free = snd_au1000_hw_free, - .prepare = snd_au1000_playback_prepare, - .trigger = snd_au1000_trigger, - .pointer = snd_au1000_pointer, -}; - -static struct snd_pcm_ops snd_card_au1000_capture_ops = { - .open = snd_au1000_capture_open, - .close = snd_au1000_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_au1000_hw_params, - .hw_free = snd_au1000_hw_free, - .prepare = snd_au1000_capture_prepare, - .trigger = snd_au1000_trigger, - .pointer = snd_au1000_pointer, -}; - -static int -snd_au1000_pcm_new(struct snd_au1000 *au1000) -{ - struct snd_pcm *pcm; - int err; - unsigned long flags; - - if ((err = snd_pcm_new(au1000->card, "AU1000 AC97 PCM", 0, 1, 1, &pcm)) < 0) - return err; - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), 128*1024, 128*1024); - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_card_au1000_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_card_au1000_capture_ops); - - pcm->private_data = au1000; - pcm->info_flags = 0; - strcpy(pcm->name, "Au1000 AC97 PCM"); - - spin_lock_init(&au1000->stream[PLAYBACK]->dma_lock); - spin_lock_init(&au1000->stream[CAPTURE]->dma_lock); - - flags = claim_dma_lock(); - au1000->stream[PLAYBACK]->dma = request_au1000_dma(au1000->dmaid[0], - "AC97 TX", au1000_dma_interrupt, 0, - au1000->stream[PLAYBACK]); - if (au1000->stream[PLAYBACK]->dma < 0) { - release_dma_lock(flags); - return -EBUSY; - } - au1000->stream[CAPTURE]->dma = request_au1000_dma(au1000->dmaid[1], - "AC97 RX", au1000_dma_interrupt, 0, - au1000->stream[CAPTURE]); - if (au1000->stream[CAPTURE]->dma < 0){ - release_dma_lock(flags); - return -EBUSY; - } - /* enable DMA coherency in read/write DMA channels */ - set_dma_mode(au1000->stream[PLAYBACK]->dma, - get_dma_mode(au1000->stream[PLAYBACK]->dma) & ~DMA_NC); - set_dma_mode(au1000->stream[CAPTURE]->dma, - get_dma_mode(au1000->stream[CAPTURE]->dma) & ~DMA_NC); - release_dma_lock(flags); - au1000->pcm = pcm; - return 0; -} - - -/*-------------------------- AC97 CODEC Control ------------------------------*/ - -static unsigned short -snd_au1000_ac97_read(struct snd_ac97 *ac97, unsigned short reg) -{ - struct snd_au1000 *au1000 = ac97->private_data; - u32 volatile cmd; - u16 volatile data; - int i; - - spin_lock(&au1000->ac97_lock); -/* would rather use the interrupt than this polling but it works and I can't -get the interrupt driven case to work efficiently */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) - printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); - - cmd = (u32) reg & AC97C_INDEX_MASK; - cmd |= AC97C_READ; - au1000->ac97_ioport->cmd = cmd; - - /* now wait for the data */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) { - printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); - spin_unlock(&au1000->ac97_lock); - return 0; - } - - data = au1000->ac97_ioport->cmd & 0xffff; - spin_unlock(&au1000->ac97_lock); - - return data; - -} - - -static void -snd_au1000_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) -{ - struct snd_au1000 *au1000 = ac97->private_data; - u32 cmd; - int i; - - spin_lock(&au1000->ac97_lock); -/* would rather use the interrupt than this polling but it works and I can't -get the interrupt driven case to work efficiently */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) - printk(KERN_ERR "au1000 AC97: AC97 command write timeout\n"); - - cmd = (u32) reg & AC97C_INDEX_MASK; - cmd &= ~AC97C_READ; - cmd |= ((u32) val << AC97C_WD_BIT); - au1000->ac97_ioport->cmd = cmd; - spin_unlock(&au1000->ac97_lock); -} - -/*------------------------------ Setup / Destroy ----------------------------*/ - -static void snd_au1000_free(struct snd_card *card) -{ - struct snd_au1000 *au1000 = card->private_data; - - if (au1000->stream[PLAYBACK]) { - if (au1000->stream[PLAYBACK]->dma >= 0) - free_au1000_dma(au1000->stream[PLAYBACK]->dma); - kfree(au1000->stream[PLAYBACK]); - } - - if (au1000->stream[CAPTURE]) { - if (au1000->stream[CAPTURE]->dma >= 0) - free_au1000_dma(au1000->stream[CAPTURE]->dma); - kfree(au1000->stream[CAPTURE]); - } - - if (au1000->ac97_res_port) { - /* put internal AC97 block into reset */ - if (au1000->ac97_ioport) { - au1000->ac97_ioport->cntrl = AC97C_RS; - iounmap(au1000->ac97_ioport); - au1000->ac97_ioport = NULL; - } - release_and_free_resource(au1000->ac97_res_port); - au1000->ac97_res_port = NULL; - } -} - -static struct snd_ac97_bus_ops ops = { - .write = snd_au1000_ac97_write, - .read = snd_au1000_ac97_read, -}; - -static int au1000_ac97_probe(struct platform_device *pdev) -{ - int err; - void __iomem *io; - struct resource *r; - struct snd_card *card; - struct snd_au1000 *au1000; - struct snd_ac97_bus *pbus; - struct snd_ac97_template ac97; - - err = snd_card_new(&pdev->dev, -1, "AC97", THIS_MODULE, - sizeof(struct snd_au1000), &card); - if (err < 0) - return err; - - au1000 = card->private_data; - au1000->card = card; - spin_lock_init(&au1000->ac97_lock); - - /* from here on let ALSA call the special freeing function */ - card->private_free = snd_au1000_free; - - /* TX DMA ID */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) { - err = -ENODEV; - snd_printk(KERN_INFO "no TX DMA platform resource!\n"); - goto out; - } - au1000->dmaid[0] = r->start; - - /* RX DMA ID */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) { - err = -ENODEV; - snd_printk(KERN_INFO "no RX DMA platform resource!\n"); - goto out; - } - au1000->dmaid[1] = r->start; - - au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), - GFP_KERNEL); - if (!au1000->stream[PLAYBACK]) { - err = -ENOMEM; - goto out; - } - au1000->stream[PLAYBACK]->dma = -1; - - au1000->stream[CAPTURE] = kmalloc(sizeof(struct audio_stream), - GFP_KERNEL); - if (!au1000->stream[CAPTURE]) { - err = -ENOMEM; - goto out; - } - au1000->stream[CAPTURE]->dma = -1; - - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { - err = -ENODEV; - goto out; - } - - err = -EBUSY; - au1000->ac97_res_port = request_mem_region(r->start, resource_size(r), - pdev->name); - if (!au1000->ac97_res_port) { - snd_printk(KERN_ERR "ALSA AC97: can't grab AC97 port\n"); - goto out; - } - - io = ioremap(r->start, resource_size(r)); - if (!io) - goto out; - - au1000->ac97_ioport = (struct au1000_ac97_reg *)io; - - /* configure pins for AC'97 - TODO: move to board_setup.c */ - au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC); - - /* Initialise Au1000's AC'97 Control Block */ - au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE; - udelay(10); - au1000->ac97_ioport->cntrl = AC97C_CE; - udelay(10); - - /* Initialise External CODEC -- cold reset */ - au1000->ac97_ioport->config = AC97C_RESET; - udelay(10); - au1000->ac97_ioport->config = 0x0; - mdelay(5); - - /* Initialise AC97 middle-layer */ - err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus); - if (err < 0) - goto out; - - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = au1000; - err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97); - if (err < 0) - goto out; - - err = snd_au1000_pcm_new(au1000); - if (err < 0) - goto out; - - strcpy(card->driver, "Au1000-AC97"); - strcpy(card->shortname, "AMD Au1000-AC97"); - sprintf(card->longname, "AMD Au1000--AC97 ALSA Driver"); - - err = snd_card_register(card); - if (err < 0) - goto out; - - printk(KERN_INFO "ALSA AC97: Driver Initialized\n"); - - platform_set_drvdata(pdev, card); - - return 0; - - out: - snd_card_free(card); - return err; -} - -static int au1000_ac97_remove(struct platform_device *pdev) -{ - return snd_card_free(platform_get_drvdata(pdev)); -} - -struct platform_driver au1000_ac97c_driver = { - .driver = { - .name = "au1000-ac97c", - .owner = THIS_MODULE, - }, - .probe = au1000_ac97_probe, - .remove = au1000_ac97_remove, -}; - -module_platform_driver(au1000_ac97c_driver); -- 2.1.4

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[alsa-devel] SGTL5000
by Erik Friesen 07 Feb '16

07 Feb '16

I was sort of directed to this list from linux-arm. I have been working with an embeddedarm ts4900 and built a baseboard that uses the sgtl5000. I also have worked with the RiotBoard, which uses the same chip, both using the i.mx6 arm. I find that occasionally the sgtl5000 is resetting, and there is no way to bring it back online. As near as I can tell, the i2c bus comes back alive, for example alsamixer communicates fine, coms happen on the i2c bus per scope. However, its obvious the chip never gets re initialized by how it acts. I know the thing shouldn't reset, that is the primary problem, but I still want some backup method. Its probably due to ground bounce from a non earth grounded setup. It could be that using the external regulator will help, we will see. Is there a suggestion on a way the sgtl5000.c could be extended to support some sort of re initialize routine? I am using the embeddedarm repo as a base kernel. sgtl5000.c has been updated to include ret = sgtl5000_fill_defaults(sgtl5000); but the base kernel is 3.10.53. However, the riotboard runs 4.3 with similar results.

1 2

[alsa-devel] Sony PCM-D100 source code
by Martin Langer 07 Feb '16

07 Feb '16

Hello, I just figured out that Sony released (in 2014?) the source code of their audio recorder PCM-D100: http://oss.sony.net/Products/Linux/Audio/PCM-D100.html A patched ALSA version and a audio driver (GPL!) for the ICX platform (ARM) is available :-) Enjoy it! Martin

1 0

[alsa-devel] About planning of libhinawa ITP to debian project
by Takashi Sakamoto 07 Feb '16

07 Feb '16

Hi, In this message, I explain about a plan of libhinawa ITP[0] to debian project. Fortunately, I got a help of a debian package maintainer, Kentaro Hayashi. He maintains groonga package[1] and experienced related works[2]. Currently, we plan to work for it in next month, February 2016. I have no idea about when FTP master acked to our ITP and when the package is released from debian project. I with the package will be available as early as possible, and this work will be a part of his great work as a debian package maintainer. FYI, [0] ITP = Intent To Packaging [1] Debian -- Details of package groonga in sid: https://packages.debian.org/sid/groonga [2] Debian Bug report logs - #732055 ITP: groonga -- Fulltext search engine https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=732055 Takashi Sakamoto

1 4

[alsa-devel] [PATCH] ALSA: hda - Fix speaker output from VAIO AiO machines
by Takashi Iwai 07 Feb '16

07 Feb '16

Some Sony VAIO AiO models (VGC-JS4EF and VGC-JS25G, both with PCI SSID 104d:9044) need the same quirk to make the speaker working properly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=112031 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a733e5dc701d..b43c0f7994eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2223,6 +2223,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), + SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), /* All Apple entries are in codec SSIDs */ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), -- 2.7.0

1 0

[alsa-devel] sound: heap out-of-bounds write in dummy_systimer_prepare
by Dmitry Vyukov 07 Feb '16

07 Feb '16

Hello, I am still seeing these eap out-of-bounds writes in dummy_systimer_prepare. Even when we've disabled the hrtimer sysctl. The fuzzer does not change sysctls at the moment and I think we've overlooked a very simple possibility that can happen when sysctls are not changed (i.e. triggered by an unprivileged user). dummy_pcm_open does: static int dummy_pcm_open(struct snd_pcm_substream *substream) { ... dummy->timer_ops = &dummy_systimer_ops; #ifdef CONFIG_HIGH_RES_TIMERS if (hrtimer) dummy->timer_ops = &dummy_hrtimer_ops; #endif There is a small window of time when we switch timer_ops to dummy_systimer_ops and then restore it back to dummy_hrtimer_ops. If another thread executes a timer op during that window, it happily smashes heap thinking that there is a systimer there. KTSAN, a data race detector (https://github.com/google/ktsan), would catch this in a blink of an eye and give a detailed explanation of what happens... ================================================================== [ 134.926046] BUG: KASAN: slab-out-of-bounds in dummy_systimer_prepare+0x268/0x2a0 at addr ffff880030f00b10 [ 134.926046] Write of size 4 by task syz-executor/32211 [ 134.926046] ============================================================================= [ 134.926046] BUG kmalloc-192 (Not tainted): kasan: bad access detected [ 134.926046] ----------------------------------------------------------------------------- [ 134.926046] [ 134.926046] INFO: Allocated in dummy_hrtimer_create+0x49/0x1a0 age=5 cpu=0 pid=32211 [ 134.926046] ___slab_alloc+0x564/0x5b0 [ 134.926046] __slab_alloc+0x66/0xc0 [ 134.926046] kmem_cache_alloc_trace+0x25c/0x300 [ 134.931676] dummy_hrtimer_create+0x49/0x1a0 [ 134.931676] dummy_pcm_open+0xef/0x570 [ 134.932135] snd_pcm_open_substream+0x188/0x430 [ 134.932135] snd_pcm_oss_open.part.17+0x5a4/0x1110 [ 134.932135] snd_pcm_oss_open+0x35/0x50 [ 134.932135] soundcore_open+0x30f/0x640 [ 134.932135] chrdev_open+0x22a/0x4c0 [ 134.932135] do_dentry_open+0x6a2/0xcb0 [ 134.932135] vfs_open+0x17b/0x1f0 [ 134.932135] path_openat+0xde9/0x5e30 [ 134.932135] do_filp_open+0x18e/0x250 [ 134.932135] do_sys_open+0x1fc/0x420 [ 134.932135] SyS_open+0x2d/0x40 [ 134.932135] INFO: Freed in snd_pcm_plugin_free+0xb1/0xe0 age=20 cpu=0 pid=32142 [ 134.932135] __slab_free+0x1fc/0x320 [ 134.932135] kfree+0x2ac/0x2c0 [ 134.932135] snd_pcm_plugin_free+0xb1/0xe0 [ 134.932135] snd_pcm_oss_release_substream+0x113/0x1f0 [ 134.932135] snd_pcm_release_substream.part.38+0x1e8/0x2f0 [ 134.937048] snd_pcm_release_substream+0x59/0x70 [ 134.937048] snd_pcm_oss_release_file+0x7b/0xb0 [ 134.937048] snd_pcm_oss_release+0xfa/0x280 [ 134.937048] __fput+0x236/0x780 [ 134.937048] ____fput+0x15/0x20 [ 134.937048] task_work_run+0x170/0x210 [ 134.937048] do_exit+0x8b5/0x2cb0 [ 134.937048] do_group_exit+0x108/0x330 [ 134.937048] get_signal+0x5e4/0x14f0 [ 134.937048] do_signal+0x83/0x1c90 [ 134.937048] exit_to_usermode_loop+0x1a5/0x210 [ 134.937048] Call Trace: [ 134.937048] [<ffffffff82be30ad>] dump_stack+0x6f/0xa2 [ 134.937048] [<ffffffff8175c984>] print_trailer+0xf4/0x150 [ 134.937048] [<ffffffff8176370f>] object_err+0x2f/0x40 [ 134.937048] [<ffffffff81765fb6>] kasan_report_error+0x256/0x550 [ 134.937048] [<ffffffff81455748>] ? mark_held_locks+0xc8/0x120 [ 134.937048] [<ffffffff85223e23>] ? snd_power_wait+0x2e3/0x380 [ 134.937048] [<ffffffff817664ae>] __asan_report_store4_noabort+0x3e/0x40 [ 134.937048] [<ffffffff852e0f68>] ? dummy_systimer_prepare+0x268/0x2a0 [ 134.937048] [<ffffffff852e0f68>] dummy_systimer_prepare+0x268/0x2a0 [ 134.937048] [<ffffffff852e141b>] dummy_pcm_prepare+0x7b/0xa0 [ 134.937048] [<ffffffff852563ca>] snd_pcm_do_prepare+0x5a/0x90 [ 134.937048] [<ffffffff852556b6>] snd_pcm_action_single+0x76/0x120 [ 134.937048] [<ffffffff85255ab5>] snd_pcm_action_nonatomic+0x95/0xa0 [ 134.937048] [<ffffffff85260a05>] snd_pcm_common_ioctl1+0x1045/0x21a0 [ 134.937048] [<ffffffff86659c34>] ? __mutex_unlock_slowpath+0x1f4/0x430 [ 134.937048] [<ffffffff8525f9c0>] ? snd_pcm_status_user+0x140/0x140 [ 134.937048] [<ffffffff86659e79>] ? mutex_unlock+0x9/0x10 [ 134.937048] [<ffffffff8529649c>] ? snd_pcm_oss_change_params+0xb2c/0x3580 [ 134.937048] [<ffffffff85261e02>] snd_pcm_playback_ioctl1+0x2a2/0x5e0 [ 134.937048] [<ffffffff85295970>] ? snd_pcm_hw_param_near.constprop.26+0x8f0/0x8f0 [ 134.937048] [<ffffffff85261b60>] ? snd_pcm_common_ioctl1+0x21a0/0x21a0 [ 134.937048] [<ffffffff85262906>] snd_pcm_kernel_ioctl+0x136/0x160 [ 134.937048] [<ffffffff852908bb>] snd_pcm_oss_prepare+0x4b/0x200 [ 134.937048] [<ffffffff85298f9e>] snd_pcm_oss_make_ready+0xae/0x120 [ 134.937048] [<ffffffff8529f58f>] snd_pcm_oss_write+0x15f/0x700 [ 134.937048] [<ffffffff81456670>] ? debug_check_no_locks_freed+0x3c0/0x3c0 [ 134.937048] [<ffffffff817bacd3>] __vfs_write+0x113/0x4b0 [ 134.937048] [<ffffffff8529f430>] ? snd_pcm_oss_ioctl_compat+0x30/0x30 [ 134.937048] [<ffffffff817babc0>] ? vfs_iter_write+0x360/0x360 [ 134.937048] [<ffffffff81493f27>] ? debug_lockdep_rcu_enabled+0x77/0x90 [ 134.937048] [<ffffffff829e63c5>] ? common_file_perm+0x155/0x3a0 [ 134.937048] [<ffffffff829e6822>] ? apparmor_file_permission+0x22/0x30 [ 134.937048] [<ffffffff8291d07c>] ? security_file_permission+0x8c/0x1f0 [ 134.937048] [<ffffffff817bc162>] ? rw_verify_area+0x102/0x2c0 [ 134.937048] [<ffffffff817bc797>] vfs_write+0x167/0x4a0 [ 134.937048] [<ffffffff817bfa81>] SyS_write+0x111/0x220 [ 134.937048] [<ffffffff817bf970>] ? SyS_read+0x220/0x220 [ 134.937048] [<ffffffff81005017>] ? trace_hardirqs_on_thunk+0x17/0x19 [ 134.937048] [<ffffffff8665f176>] entry_SYSCALL_64_fastpath+0x16/0x7a [ 134.937048] ================================================================== On commit df48ab3c2f5ffca88b7803ffbadd074bd5a0a2ef.

2 1

[alsa-devel] [PATCH] Revert "ALSA: hda - Fix noise on Gigabyte Z170X mobo"
by Takashi Iwai 05 Feb '16

05 Feb '16

This reverts commit 0c25ad80408e95e0a4fbaf0056950206e95f726f. The original commit disabled the aamixer path due to the noise problem, but it turned out that some mobo with the same PCI SSID doesn't suffer from the issue, and the disabled function (analog loopback) is still demanded by users. Since the recent commit [e7fdd52779a6: ALSA: hda - Implement loopback control switch for Realtek and other codecs], we have the dynamic mixer switch to enable/disable the aamix path, and we don't have to disable the path statically any longer. So, let's revert the disablement, so that only the user suffering from the noise problem can turn off the aamix on the fly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=108301 Reported-by: <mutedbytes(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 21992fb7035d..a733e5dc701d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1787,7 +1787,6 @@ enum { ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, ALC887_FIXUP_BASS_CHMAP, - ALC882_FIXUP_DISABLE_AAMIX, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1949,8 +1948,6 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, static void alc_fixup_bass_chmap(struct hda_codec *codec, const struct hda_fixup *fix, int action); -static void alc_fixup_disable_aamix(struct hda_codec *codec, - const struct hda_fixup *fix, int action); static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { @@ -2188,10 +2185,6 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, - [ALC882_FIXUP_DISABLE_AAMIX] = { - .type = HDA_FIXUP_FUNC, - .v.func = alc_fixup_disable_aamix, - }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2259,7 +2252,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), - SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 2.7.0

1 0

[alsa-devel] [PATCH v3] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by tim.howe＠cirrus.com 05 Feb '16

05 Feb '16

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- .../devicetree/bindings/sound/cs53l30.txt | 21 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 933 +++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 272 ++++++ 5 files changed, 1233 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..a2dff95 +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,21 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3317da..f59aa4b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -444,6 +445,10 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cdadb9b..4f1ab1d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -51,6 +51,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o +snd-soc-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o +obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..3d5cac9 +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,933 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs53l30.h" + +struct cs53l30_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 asp_config_ctl; + u32 mclk; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, THMS_PDN }, + { CS53L30_MCLKCTL, MCLK_DIV_DFLT }, + { CS53L30_INT_SR_CTL, INTERNAL_FS_DFLT }, + { CS53L30_MICBIAS_CTL, MIC_BIAS_DFLT }, + { CS53L30_ASPCFG_CTL, ASP_RATE_48K }, + { CS53L30_ASP1_CTL, ASP1_TDM_PDN }, + { CS53L30_ASP1_TDMTX_CTL1, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL2, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL3, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL4, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_EN1, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN2, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN3, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN4, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN5, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN6, ASP_TX_DISABLED }, + { CS53L30_ASP2_CTL, ASP2_CTRL_DFLT }, + { CS53L30_SFT_RAMP, SFT_RMP_DFLT }, + { CS53L30_LRCLK_CTL1, LRCK_CTLX_DFLT }, + { CS53L30_LRCLK_CTL2, LRCK_CTLX_DFLT }, + { CS53L30_MUTEP_CTL1, MUTEP_CTRL1_DFLT }, + { CS53L30_MUTEP_CTL2, MUTE_PDN_ULP }, + { CS53L30_INBIAS_CTL1, INBIAS_CTLX_DFLT }, + { CS53L30_INBIAS_CTL2, INBIAS_CTLX_DFLT }, + { CS53L30_DMIC1_STR_CTL, DMIC1_STEREO_DFLT }, + { CS53L30_DMIC2_STR_CTL, DMIC2_STEREO_DFLT }, + { CS53L30_ADCDMIC1_CTL1, ADC1_ON_CH_AB_IN }, + { CS53L30_ADCDMIC1_CTL2, ADC1_DMIC1_CTL2_DFLT }, + { CS53L30_ADC1_CTL3, ADC1_HPF_EN }, + { CS53L30_ADC1_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADCDMIC2_CTL1, ADC2_ON_CH_AB_IN }, + { CS53L30_ADCDMIC2_CTL2, ADCX_ZERO_DFLT }, + { CS53L30_ADC2_CTL3, ADC2_HPF_EN }, + { CS53L30_ADC2_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_INT_MASK, DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP1_CTL: + case CS53L30_ASP1_TDMTX_CTL1: + case CS53L30_ASP1_TDMTX_CTL2: + case CS53L30_ASP1_TDMTX_CTL3: + case CS53L30_ASP1_TDMTX_CTL4: + case CS53L30_ASP1_TDMTX_EN1: + case CS53L30_ASP1_TDMTX_EN2: + case CS53L30_ASP1_TDMTX_EN3: + case CS53L30_ASP1_TDMTX_EN4: + case CS53L30_ASP1_TDMTX_EN5: + case CS53L30_ASP1_TDMTX_EN6: + case CS53L30_ASP2_CTL: + case CS53L30_SFT_RAMP: + case CS53L30_LRCLK_CTL1: + case CS53L30_LRCLK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); + +static const char * const input1_sel_text[] = { "DMIC1 On AB In", + "DMIC1 On A In", "DMIC1 On B In", "ADC1 On AB In", "ADC1 On A In", + "ADC1 On B In", "DMIC1 Off ADC1 Off", }; + +unsigned int const input1_sel_values[] = { DMIC1_ON_CH_AB_IN, DMIC1_ON_CH_A_IN, + DMIC1_ON_CH_B_IN, ADC1_ON_CH_AB_IN, ADC1_ON_CH_A_IN, ADC1_ON_CH_B_IN, + DMIC1_OFF_ADC1_OFF, }; + +static const char * const input2_sel_text[] = { "DMIC2 On AB In", + "DMIC2 On A In", "DMIC2 On B In", "ADC2 On AB In", "ADC2 On A In", + "ADC2 On B In", "DMIC2 Off ADC2 Off", }; + +unsigned int const input2_sel_values[] = { DMIC2_ON_CH_AB_IN, DMIC2_ON_CH_A_IN, + DMIC2_ON_CH_B_IN, ADC2_ON_CH_AB_IN, ADC2_ON_CH_A_IN, ADC2_ON_CH_B_IN, + DMIC2_OFF_ADC2_OFF, }; + +static const char * const input1_route_sel_text[] = { "ADC1_SEL", + "DMIC1_SEL" }; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input1_route_sel_text), input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input1_sel_text, input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { "ADC2_SEL", + "DMIC2_SEL" }; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input2_sel_text, input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms" }; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB"}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +/* ADC Preamp gain select */ +static const char * const cs53l30_preamp_gain_sel_text[] = { + "0dB", "10dB", "20dB"}; + +static const struct soc_enum adc1a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc1b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +/* Set MIC Bias Voltage Control */ +static const char * const cs53l30_micbias_text[] = { + "HiZ", "1.8V", "2.75V"}; + +static const struct soc_enum micbias_enum = + SOC_ENUM_SINGLE(CS53L30_MICBIAS_CTL, 0, + ARRAY_SIZE(cs53l30_micbias_text), + cs53l30_micbias_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz"}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, 5, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", + CS53L30_ADC1_CTL3, 0, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", + CS53L30_ADC2_CTL3, 0, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", + CS53L30_ADCDMIC1_CTL2, 7, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", + CS53L30_ADCDMIC2_CTL2, 7, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", + CS53L30_ADCDMIC1_CTL2, 4, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", + CS53L30_ADCDMIC1_CTL2, 5, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", + CS53L30_ADCDMIC2_CTL2, 4, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", + CS53L30_ADCDMIC2_CTL2, 5, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", + CS53L30_ADC1_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", + CS53L30_ADC2_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + SOC_ENUM("ADC1A Pre Amp Gain", adc1a_preamp_gain_enum), + SOC_ENUM("ADC1B Pre Amp Gain", adc1b_preamp_gain_enum), + SOC_ENUM("ADC2A Pre Amp Gain", adc2a_preamp_gain_enum), + SOC_ENUM("ADC2B Pre Amp Gain", adc2b_preamp_gain_enum), + SOC_ENUM("Mic Bias Voltage Select", micbias_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static int cs53l30_asp_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 1); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT1", NULL, 0, + CS53L30_ASP1_CTL, ASP1_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT2", NULL, 0, + CS53L30_ASP2_CTL, ASP2_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, 7, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, 2, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, 2, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_audio_map[] = { + + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, + + /* Output Paths */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"ASP1 Capture", NULL, "ASP_SDOUT1"}, + {"ASP2 Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, 1, 1}, + {5644800, 22050, 0x8, 1, 1}, + {5644800, 44100, 0xC, 1, 1}, + + {6000000, 8000, 0x1, 0, 1}, + {6000000, 11025, 0x2, 0, 1}, + {6000000, 12000, 0x4, 0, 1}, + {6000000, 16000, 0x5, 0, 1}, + {6000000, 22050, 0x6, 0, 1}, + {6000000, 24000, 0x8, 0, 1}, + {6000000, 32000, 0x9, 0, 1}, + {6000000, 44100, 0xA, 0, 1}, + {6000000, 48000, 0xC, 0, 1}, + + {6144000, 8000, 0x1, 1, 1}, + {6144000, 11025, 0x2, 1, 1}, + {6144000, 12000, 0x4, 1, 1}, + {6144000, 16000, 0x5, 1, 1}, + {6144000, 22050, 0x6, 1, 1}, + {6144000, 24000, 0x8, 1, 1}, + {6144000, 32000, 0x9, 1, 1}, + {6144000, 44100, 0xA, 1, 1}, + {6144000, 48000, 0xC, 1, 1}, + + {6400000, 8000, 0x1, 1, 1}, + {6400000, 11025, 0x2, 1, 1}, + {6400000, 12000, 0x4, 1, 1}, + {6400000, 16000, 0x5, 1, 1}, + {6400000, 22050, 0x6, 1, 1}, + {6400000, 24000, 0x8, 1, 1}, + {6400000, 32000, 0x9, 1, 1}, + {6400000, 44100, 0xA, 1, 1}, + {6400000, 48000, 0xC, 1, 1}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, 0}, + {6000000, 1, 0}, + {6144000, 1, 0}, + {11289600, 2, 1}, + {12288000, 2, 1}, + {12000000, 2, 1}, + {19200000, 3, 2}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk == mclk && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk; + unsigned int mclk_ctl; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + mclk_ctl &= ~MCLK_DIV; + mclk_ctl |= cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->mclk = mclk; + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int asp_config_ctl; + + regmap_read(priv->regmap, CS53L30_ASPCFG_CTL, &asp_config_ctl); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + asp_config_ctl |= ASP_MS; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + asp_config_ctl &= ~ASP_MS; + break; + + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + asp_config_ctl |= ASP_SCLK_INV; + else + asp_config_ctl &= ~ASP_SCLK_INV; + + priv->asp_config_ctl = asp_config_ctl; + return 0; +} + +static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + int mclk_coeff; + int srate = params_rate(params); + unsigned int int_sr_ctl, mclk_ctl; + + /* MCLK -> srate */ + mclk_coeff = + cs53l30_get_mclk_coeff(priv->mclk, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_write(priv->regmap, CS53L30_ASPCFG_CTL, priv->asp_config_ctl); + regmap_read(priv->regmap, CS53L30_INT_SR_CTL, &int_sr_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio) + int_sr_ctl |= INTERNAL_FS_RATIO; + else + int_sr_ctl &= ~INTERNAL_FS_RATIO; + regmap_write(priv->regmap, CS53L30_INT_SR_CTL, int_sr_ctl); + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale) + mclk_ctl |= MCLK_INT_SCALE; + else + mclk_ctl &= ~MCLK_INT_SCALE; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->asp_config_ctl &= ASP_CNFG_MASK; + priv->asp_config_ctl |= (cs53l30_mclk_coeffs[mclk_coeff].asp_rate + & ASP_RATE); + regmap_write(priv->regmap, CS53L30_MCLKCTL, priv->asp_config_ctl); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, PDN_LP); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, 0); + /* If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & DIGSFT) + inter_max_check = PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, PDN_ULP); + msleep(20); /* PDN_DONE will take a min of 20ms to be set.*/ + regmap_read(priv->regmap, CS53L30_IS, &reg); /* Clr status */ + for (i = 0; i < inter_max_check; i++) { + usleep_range(1000, 1000); + msleep(1); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & PDN_DONE) + break; + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, MCLK_DIS); + break; + } + dapm->bias_level = level; + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, ASP1_3ST, + (ASP1_3ST_VAL(tristate) & ASP1_3ST)); +} + +unsigned int const cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &src_constraints); + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, +}; + +static struct snd_soc_dai_driver cs53l30_dai[] = { + { + .name = "cs53l30-asp1", + .id = CS53L30_ASP1, + .capture = { + .stream_name = "ASP1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + }, + { + .name = "cs53l30-asp2", + .id = CS53L30_ASP2, + .capture = { + .stream_name = "ASP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs53l30 = { + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs53l30_audio_map), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs53l30_private *cs53l30; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs53l30 = devm_kzalloc(&client->dev, + sizeof(struct cs53l30_private), GFP_KERNEL); + if (!cs53l30) { + dev_err(&client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) + return PTR_ERR(cs53l30->reset_gpio); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + /* initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(&client->dev, + "CS53L30 Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + return ret; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(&client->dev, + "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs53l30, cs53l30_dai, + ARRAY_SIZE(cs53l30_dai)); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + {"cs53l30", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + .owner = THIS_MODULE, + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, + +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..f45e91f +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,272 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP1_CTL 0x0D /* ASP1 Control. */ +#define CS53L30_ASP1_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP1_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP1_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP1_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP1_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP1_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP1_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP1_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP1_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP1_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP2_CTL 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCLK_CTL1 0x1B /* LRCLK Control 1. */ +#define CS53L30_LRCLK_CTL2 0x1C /* LRCLK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + +/* Bitfield Definitions */ + +/* CS53L30_PWRCTL */ +#define PDN_ULP (1 << 7) +#define PDN_LP (1 << 6) +#define DISCHARGE_FILT (1 << 5) +#define THMS_PDN (1 << 4) + +/* CS53L30_MCLKCTL */ +#define MCLK_DIS (1 << 7) +#define MCLK_INT_SCALE (1 << 6) +#define DMIC_DRIVE (1 << 5) +#define MCLK_DIV (3 << 2) +#define MCLK_DIV_DFLT (1 << 2) +#define SYNC_EN (1 << 1) + +/* CS53L30_INT_SR_CTL */ +#define INTERNAL_FS_RATIO (1 << 4) +#define INTERNAL_FS_DFLT (7 << 2) +#define MCLK_19MHZ_EN (1 << 0) + +/* CS53L30_MICBIAS_CTL */ +#define MIC4_BIAS_PDN (1 << 7) +#define MIC3_BIAS_PDN (1 << 6) +#define MIC2_BIAS_PDN (1 << 5) +#define MIC1_BIAS_PDN (1 << 4) +#define VP_MIN (1 << 2) +#define MIC_BIAS_CTRL (3 << 0) +#define MIC_BIAS_ALL_PDN 0xF0 +#define MIC_BIAS_DFLT (MIC_BIAS_ALL_PDN | VP_MIN) + +/* CS53L30_ASPCFG_CTL */ +#define ASP_MS (1 << 7) +#define ASP_SCLK_INV (1 << 4) +#define ASP_RATE_48K (3 << 2) +#define ASP_RATE 0x0F +#define ASP_CNFG_MASK 0xF0 + +/* CS53L30_ASP1_CTL */ +#define ASP1_TDM_PDN (1 << 7) +#define ASP1_SDOUT_PDN (1 << 6) +#define ASP1_3ST (1 << 5) +#define SHIFT_LEFT (1 << 4) +#define ASP1_DRIVE (1 << 0) +#define ASP1_3ST_VAL(x) ((x) << 5) + +/* CS53L30_ASP1_TDMTX_CTL */ +#define ASP1_CHX_TX_STATE (1 << 7) +#define ASP1_CHX_TX_LOC 0x3F +#define ASP1_CHX_TX_DFLT_SLT47 0x2F +#define ASP_TX_DISABLED 0x00 + +/* CS53L30_ASP2_CTL */ +#define ASP2_SDOUT_PDN (1 << 6) +#define ASP2_DRIVE (1 << 0) +#define ASP2_CTRL_DFLT 0x00 + +/* CS53L30_SFT_RAMP */ +#define DIGSFT (1 << 5) +#define SFT_RMP_DFLT 0x00 + +/* CS53L30_LRCLK_CTL2 */ +#define LRCK_50_NPW (1 << 3) +#define LRCK_TPWH (7 << 0) +#define LRCK_CTLX_DFLT 0x00 + +/* CS53L30_MUTEP_CTL */ +#define MUTE_PDN_ULP (1 << 7) +#define MUTE_PDN_LP (1 << 6) +#define MUTE_M4B_PDN (1 << 4) +#define MUTE_M3B_PDN (1 << 3) +#define MUTE_M2B_PDN (1 << 2) +#define MUTE_M1B_PDN (1 << 1) +#define MUTE_MB_ALL_PDN (1 << 0) +#define MUTEP_CTRL1_DFLT 0x00 + +/* CS53L30_MUTEP_CTL2 */ +#define MUTE_PIN_POLARITY (1 << 7) +#define MUTE_ASP_TDM_PDN (1 << 6) +#define MUTE_ASP_SDOUT2_PDN (1 << 5) +#define MUTE_ASP_SDOUT1_PDN (1 << 4) +#define MUTE_ADC2B_PDN (1 << 3) +#define MUTE_ADC2A_PDN (1 << 2) +#define MUTE_ADC1B_PDN (1 << 1) +#define MUTE_ADC1A_PDN (1 << 0) + +/* CS53L30_INBIAS_CTL1 */ +#define IN4M_BIAS (3 << 6) +#define IN4P_BIAS (3 << 4) +#define IN3M_BIAS (3 << 2) +#define IN3P_BIAS (3 << 0) + +/* CS53L30_INBIAS_CTL2 */ +#define IN2M_BIAS (3 << 6) +#define IN2P_BIAS (3 << 4) +#define IN1M_BIAS (3 << 2) +#define IN1P_BIAS (3 << 0) +#define INBIAS_CTLX_DFLT 0xAA + +/* CS53L30_DMIC1_STR_CTL */ +#define DMIC1_STEREO_ENB (1 << 5) +#define DMIC1_STEREO_DFLT 0xA8 + +/* CS53L30_DMIC2_STR_CTL */ +#define DMIC2_STEREO_EN (1 << 5) +#define DMIC2_STEREO_DFLT 0xEC + +/* CS53L30_ADCDMIC1_CTL1 */ +#define ADC1B_PDN (1 << 7) +#define ADC1A_PDN (1 << 6) +#define DMIC1_PDN (1 << 2) +#define DMIC1_SCLK_DIV (1 << 1) +#define CH_TYPE (1 << 0) +#define DMIC1_ON_CH_AB_IN (CH_TYPE) +#define DMIC1_ON_CH_A_IN (ADC1B_PDN | CH_TYPE) +#define DMIC1_ON_CH_B_IN (ADC1A_PDN | CH_TYPE) +#define ADC1_ON_CH_AB_IN (DMIC1_PDN) +#define ADC1_ON_CH_A_IN (ADC1B_PDN | DMIC1_PDN) +#define ADC1_ON_CH_B_IN (ADC1A_PDN | DMIC1_PDN) +#define DMIC1_OFF_ADC1_OFF (ADC1A_PDN | ADC1B_PDN | DMIC1_PDN) +#define ADC1_DMIC1_PDN_MASK 0xFF + +/* CS53L30_ADCDMIC1_CTL2 */ +#define ADC1_NOTCH_DIS (1 << 7) +#define ADC1B_INV (1 << 5) +#define ADC1A_INV (1 << 4) +#define ADC1B_DIG_BOOST (1 << 1) +#define ADC1A_DIG_BOOST (1 << 0) +#define ADC1_DMIC1_CTL2_DFLT 0x00 + +/* CS53L30_ADC1_CTL3 */ +#define ADC1_HPF_EN (1 << 3) +#define ADC1_HPF_CF (3 << 1) +#define ADC1_NG_ALL (1 << 0) + +/* CS53L30_ADC1_NG_CTL */ +#define ADC1B_NG (1 << 7) +#define ADC1A_NG (1 << 6) +#define ADC1_NG_BOOST (1 << 5) +#define ADC1_NG_THRESH (7 << 2) +#define ADC1_NG_DELAY (3 << 0) +#define ADCX_ZERO_DFLT 0x00 + +/* CS53L30_ADC1A_AFE_CTL */ +#define ADC1A_PREAMP (3 << 6) +#define ADC1A_PGA_VOL 0x3F + +/* CS53L30_ADC1B_AFE_CTL */ +#define ADC1B_PREAMP (3 << 6) +#define ADC1B_PGA_VOL 0x3F + +/* CS53L30_ADCXX_DIG_VOL */ +#define MUTE_DIG_OUT (1 << 7) + +/* CS53L30_ADCDMIC2_CTL1 */ +#define ADC2B_PDN (1 << 7) +#define ADC2A_PDN (1 << 6) +#define DMIC2_PDN (1 << 2) +#define DMIC2_CLKDIV (1 << 1) +#define DMIC2_ON_CH_AB_IN 0x00 /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_A_IN (ADC2B_PDN) /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_B_IN (ADC2A_PDN) /* CH_TYPE must = 1 */ +#define ADC2_ON_CH_AB_IN (DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_A_IN (ADC2B_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_B_IN (ADC2A_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define DMIC2_OFF_ADC2_OFF (ADC2A_PDN | ADC2B_PDN | DMIC2_PDN) + +/* CS53L30_ADCDMIC2_CTL2 */ +#define ADC2_NOTCH_DIS (1 << 7) +#define ADC2B_INV (1 << 5) +#define ADC2A_INV (1 << 4) +#define ADC2B_DIG_BOOST (1 << 1) +#define ADC2A_DIG_BOOST (1 << 0) + +/* CS53L30_ADC2_CTL3 */ +#define ADC2_HPF_EN (1 << 3) +#define ADC2_HPF_CF (3 << 1) +#define ADC2_NG_ALL (1 << 0) + +/* CS53L30_INT */ +#define PDN_DONE (1 << 7) +#define THMS_TRIP (1 << 6) +#define SYNC_DONE (1 << 5) +#define ADC2B_OVFL (1 << 4) +#define ADC2A_OVFL (1 << 3) +#define ADC1B_OVFL (1 << 2) +#define ADC1A_OVFL (1 << 1) +#define MUTE_PIN (1 << 0) +#define DEVICE_INT_MASK 0xFF + +/* Serial Ports */ +#define CS53L30_ASP1 0 +#define CS53L30_ASP2 1 + +#endif /* __CS53L30_H__ */ -- 2.4.5

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[alsa-devel] [PATCH] ASoC: mxs-saif: check BUSY bit in hw_params() only if not mclk_in_use
by Mans Rullgard 05 Feb '16

05 Feb '16

If something else, typically a codec, has enabled mclk, the BUSY bit may be set when hw_params() is called without this being an error. This check thus causes intermittent failures to configure the sound device when used in such a manner. Fix this by making the test conditional on !saif->mclk_in_use. Signed-off-by: Mans Rullgard <mans(a)mansr.com> --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index a6c7b8d87cd2..13631003cb7c 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -418,7 +418,7 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, } stat = __raw_readl(saif->base + SAIF_STAT); - if (stat & BM_SAIF_STAT_BUSY) { + if (!saif->mclk_in_use && (stat & BM_SAIF_STAT_BUSY)) { dev_err(cpu_dai->dev, "error: busy\n"); return -EBUSY; } -- 2.7.0

2 6

[alsa-devel] [PATCH v2] ASoC: arizona: Add support for SNDRV_PCM_RATE_KNOT
by Charles Keepax 05 Feb '16

05 Feb '16

The Arizona CODECs support several rates that do not have simple defines in ALSA. This patch adds support for SNDRV_PCM_RATE_KNOT so that users can open stream at these rates. As part of this we should always set constraints in arizona_startup, currently we only set the constraints if we already have a clock to limit rates to that family of sample rates. This patch updates this to set a constraint of all rates supported by the chip if we do not already know which family of rates to limit to. Finally we also reduce the list of rates supported in the constraints to only include those that are supported on current parts. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Thanks, for the pointer there Lars, sorry for missing what you were talking about the first time. It does indeed look much nicer this way. Changes since v1: - Use the mask field in the constaint list as suggested by Lars. Thanks, Charles sound/soc/codecs/arizona.c | 62 ++++++++++++---------------------------------- sound/soc/codecs/arizona.h | 2 ++ sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- 7 files changed, 23 insertions(+), 51 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 33143fe..a32cfb8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1398,29 +1398,6 @@ static const int arizona_48k_bclk_rates[] = { 24576000, }; -static const unsigned int arizona_48k_rates[] = { - 12000, - 24000, - 48000, - 96000, - 192000, - 384000, - 768000, - 4000, - 8000, - 16000, - 32000, - 64000, - 128000, - 256000, - 512000, -}; - -static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { - .count = ARRAY_SIZE(arizona_48k_rates), - .list = arizona_48k_rates, -}; - static const int arizona_44k1_bclk_rates[] = { -1, 44100, @@ -1443,22 +1420,7 @@ static const int arizona_44k1_bclk_rates[] = { 22579200, }; -static const unsigned int arizona_44k1_rates[] = { - 11025, - 22050, - 44100, - 88200, - 176400, - 352800, - 705600, -}; - -static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { - .count = ARRAY_SIZE(arizona_44k1_rates), - .list = arizona_44k1_rates, -}; - -static int arizona_sr_vals[] = { +static const unsigned int arizona_sr_vals[] = { 0, 12000, 24000, @@ -1485,13 +1447,21 @@ static int arizona_sr_vals[] = { 512000, }; +#define ARIZONA_48K_RATE_MASK 0x0F003E +#define ARIZONA_44K1_RATE_MASK 0x003E00 +#define ARIZONA_RATE_MASK (ARIZONA_48K_RATE_MASK | ARIZONA_44K1_RATE_MASK) + +static const struct snd_pcm_hw_constraint_list arizona_constraint = { + .count = ARRAY_SIZE(arizona_sr_vals), + .list = arizona_sr_vals, +}; + static int arizona_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; - const struct snd_pcm_hw_constraint_list *constraint; unsigned int base_rate; if (!substream->runtime) @@ -1509,16 +1479,15 @@ static int arizona_startup(struct snd_pcm_substream *substream, } if (base_rate == 0) - return 0; - - if (base_rate % 8000) - constraint = &arizona_44k1_constraint; + dai_priv->constraint.mask = ARIZONA_RATE_MASK; + else if (base_rate % 8000) + dai_priv->constraint.mask = ARIZONA_44K1_RATE_MASK; else - constraint = &arizona_48k_constraint; + dai_priv->constraint.mask = ARIZONA_48K_RATE_MASK; return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - constraint); + &dai_priv->constraint); } static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, @@ -1911,6 +1880,7 @@ int arizona_init_dai(struct arizona_priv *priv, int id) struct arizona_dai_priv *dai_priv = &priv->dai[id]; dai_priv->clk = ARIZONA_CLK_SYSCLK; + dai_priv->constraint = arizona_constraint; return 0; } diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 0c64a57..1ea8e4e 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -68,6 +68,8 @@ struct wm_adsp; struct arizona_dai_priv { int clk; + + struct snd_pcm_hw_constraint_list constraint; }; struct arizona_priv { diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index b8d3cd0..576087b 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -923,7 +923,7 @@ static int cs47l24_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define CS47L24_RATES SNDRV_PCM_RATE_8000_192000 +#define CS47L24_RATES SNDRV_PCM_RATE_KNOT #define CS47L24_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index aa1e96a..a8b3e3f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1738,7 +1738,7 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM5102_RATES SNDRV_PCM_RATE_8000_192000 +#define WM5102_RATES SNDRV_PCM_RATE_KNOT #define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 302a8d3..83ba70f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2024,7 +2024,7 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM5110_RATES SNDRV_PCM_RATE_8000_192000 +#define WM5110_RATES SNDRV_PCM_RATE_KNOT #define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index b4dba3a..52d766e 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -943,7 +943,7 @@ static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 +#define WM8997_RATES SNDRV_PCM_RATE_KNOT #define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 7719bc5..0123960 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1170,7 +1170,7 @@ static const struct snd_soc_dapm_route wm8998_dapm_routes[] = { { "DRC1 Signal Activity", NULL, "DRC1R" }, }; -#define WM8998_RATES SNDRV_PCM_RATE_8000_192000 +#define WM8998_RATES SNDRV_PCM_RATE_KNOT #define WM8998_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: adau17x1: Cache writes when core clock is disabled
by Andreas Irestål 05 Feb '16

05 Feb '16

In some configurations, the dai registers get written before the bias level is changed in the codec driver. This leads to a situation where an initial write to the serial port register gets ignored, and future writes may as well, since regmap thinks that the codec already holds the value. More specifically, configuring the codec as i2s master would in fact result in the codec running as slave, a situation where no i2s clocks are generated and hence no data is transferred. This change makes sure that regmap only caches writes when the core clock is disabled, and syncs regmap whenever enabling the core clock again. Signed-off-by: Andreas Irestål <andire(a)axis.com> --- sound/soc/codecs/adau1761.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 2f12477..e7136b1 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -456,13 +456,17 @@ static int adau1761_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + regcache_cache_only(adau->regmap, false); regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) + regcache_sync(adau->regmap); break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, 0); + regcache_cache_only(adau->regmap, true); break; } @@ -783,6 +787,10 @@ int adau1761_probe(struct device *dev, struct regmap *regmap, if (ret) return ret; + /* Enable cache only mode as we could miss writes before bias level + * reaches standby and the core clock is enabled */ + regcache_cache_only(regmap, true); + return snd_soc_register_codec(dev, &adau1761_codec_driver, dai_drv, 1); } EXPORT_SYMBOL_GPL(adau1761_probe); -- 2.1.4

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: fsl-asoc-card: add cs4271 and cs4272 support
by Felipe F. Tonello 05 Feb '16

05 Feb '16

add cs4271 and cs42727 support for fsl-asoc-card Signed-off-by: Felipe F. Tonello <eu(a)felipetonello.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 9 +++++++++ sound/soc/fsl/Kconfig | 4 ++-- sound/soc/fsl/fsl-asoc-card.c | 7 +++++++ 3 files changed, 18 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index 4da41bf1888e..ceaef5126989 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -24,6 +24,9 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-cs42888" + "fsl,imx-audio-cs427x" + (compatible with CS4271 and CS4272) + "fsl,imx-audio-wm8962" (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt) @@ -63,6 +66,12 @@ Optional properties: - audio-asrc : The phandle of ASRC. It can be absent if there's no need to add ASRC support via DPCM. +Optional unless SSI is selected as a CPU DAI: + + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) + + - mux-ext-port : The external port of the i.MX audio muxer + Example: sound-cs42888 { compatible = "fsl,imx-audio-cs42888"; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 14dfdee05fd5..35aabf9dc503 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -292,8 +292,8 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_SSI help ALSA SoC Audio support with ASRC feature for Freescale SoCs that have - ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 - and SGTL5000. + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888, + CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card. endif # SND_IMX_SOC diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 562b3bd22d9a..56b6b07d1922 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -28,6 +28,8 @@ #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" +#define CS427x_SYSCLK_MCLK 0 + #define RX 0 #define TX 1 @@ -528,6 +530,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { + codec_dai_name = "cs4271-hifi"; + priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; @@ -681,6 +687,7 @@ fail: static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-cs427x", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, -- 2.7.0

5 6

[alsa-devel] [PATCH 0/4] ALSA: firewire-tascam: add support for FW-1804
by Takashi Sakamoto 05 Feb '16

05 Feb '16

Hi, This patchset adds support for FW-1804. Additionally it fix some bugs. Nothing else ;) This commit also adds no functionality to control internal multiplexer of supported models, following to basic design of ALSA firewire stack. It should be implemented in userspace. Currently, there's no udev rules for fw character devices corresponding to models supported by this module. Thus, permission of the devices may be 0600. When userspace applications touches the devices, they needs effective UID or GID as 0. This is not better environment for users. Further works are required for related softwares. Regards Takashi Sakamoto (4): ALSA: firewire-tascam: fix NULL pointer dereference when model identification fails ALSA: firewire-tascam: add support for FW-1804 ALSA: firewire-tascam: remove a flag for controller ALSA: firewire-tascam: remove needless member for control and status message sound/firewire/tascam/tascam-transaction.c | 6 ++++++ sound/firewire/tascam/tascam.c | 12 +++++++++--- sound/firewire/tascam/tascam.h | 4 ---- 3 files changed, 15 insertions(+), 7 deletions(-) -- 2.5.0

2 5

[alsa-devel] [PATCH] ALSA: hda - Fix static checker warning in patch_hdmi.c
by David Henningsson 05 Feb '16

05 Feb '16

The static checker warning is: sound/pci/hda/patch_hdmi.c:460 hdmi_eld_ctl_get() error: __memcpy() 'eld->eld_buffer' too small (256 vs 512) I have a hard time figuring out if this can ever cause an information leak (I don't think so), but nonetheless it does not hurt to increase the robustness of the code. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1f52b55..2191e235 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -448,7 +448,8 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, eld = &per_pin->sink_eld; mutex_lock(&per_pin->lock); - if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data)) { + if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || + eld->eld_size > ELD_MAX_SIZE) { mutex_unlock(&per_pin->lock); snd_BUG(); return -EINVAL; -- 1.9.1

2 1

Re: [alsa-devel] ALSA: hda - hdmi: Do not expose eld data when eld is invalid
by Dan Carpenter 05 Feb '16

05 Feb '16

Hello David Henningsson, The patch 68e03de98507: "ALSA: hda - hdmi: Do not expose eld data when eld is invalid" from Feb 19, 2013, leads to the following static checker warning: sound/pci/hda/patch_hdmi.c:460 hdmi_eld_ctl_get() error: __memcpy() 'eld->eld_buffer' too small (256 vs 512) sound/pci/hda/patch_hdmi.c 437 static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, 438 struct snd_ctl_elem_value *ucontrol) 439 { 440 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 441 struct hdmi_spec *spec = codec->spec; 442 struct hdmi_spec_per_pin *per_pin; 443 struct hdmi_eld *eld; 444 int pin_idx; 445 446 pin_idx = kcontrol->private_value; 447 per_pin = get_pin(spec, pin_idx); 448 eld = &per_pin->sink_eld; 449 450 mutex_lock(&per_pin->lock); 451 if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data)) { In the original code we always copied ELD_MAX_SIZE (256) bytes. Now it looks like we can copy up to 512 bytes which seems like an information leak. I don't know where eld->eld_size is set so I can't say if this is a real issue or not. Perhaps it's always a valid value. 452 mutex_unlock(&per_pin->lock); 453 snd_BUG(); 454 return -EINVAL; 455 } 456 457 memset(ucontrol->value.bytes.data, 0, 458 ARRAY_SIZE(ucontrol->value.bytes.data)); 459 if (eld->eld_valid) 460 memcpy(ucontrol->value.bytes.data, eld->eld_buffer, 461 eld->eld_size); 462 mutex_unlock(&per_pin->lock); 463 464 return 0; 465 } regards, dan carpenter

1 0

[alsa-devel] linux-api scope (Re: [PATCH v2 11/22] media: dvb-frontend invoke enable/disable_source handlers)
by Andy Lutomirski 04 Feb '16

04 Feb '16

[cc list heavily trimmed] On Wed, Feb 3, 2016 at 8:03 PM, Shuah Khan <shuahkh(a)osg.samsung.com> wrote: > Change dvb frontend to check if tuner is free when > device opened in RW mode. Call to enable_source > handler either returns with an active pipeline to > tuner or error if tuner is busy. Tuner is released > when frontend is released calling the disable_source > handler. As an actual subscriber to linux-api, I prefer for the linux-api list to be lowish-volume and mostly limited to API-related things. Is this API related? Do people think that these series should be sent to linux-api? Thanks, Andy

3 3

[alsa-devel] [PATCH v3 1/3] ALSA: mts64: whitespace fixes
by Sudip Mukherjee 04 Feb '16

04 Feb '16

To improve the readability and grep-ability of the code trailing whitespace, multiple blank lines, space before start of a line and a required space before opening brace have been removed. This patch takes care of these issues related to whitespace changes and there was no other change. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v3: changed commit message v2: combined 3 patches of v1 series into this patch. sound/drivers/mts64.c | 112 ++++++++++++++++++++++++-------------------------- 1 file changed, 54 insertions(+), 58 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 2a008a9..f7d938d 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1,14 +1,14 @@ -/* +/* * ALSA Driver for Ego Systems Inc. (ESI) Miditerminal 4140 * Copyright (c) 2006 by Matthias König <mk(a)phasorlab.de> * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * @@ -38,7 +38,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -static struct platform_device *platform_devices[SNDRV_CARDS]; +static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; module_param_array(index, int, NULL, S_IRUGO); @@ -92,7 +92,7 @@ static int snd_mts64_create(struct snd_card *card, *rchip = NULL; mts = kzalloc(sizeof(struct mts64), GFP_KERNEL); - if (mts == NULL) + if (mts == NULL) return -ENOMEM; /* Init chip specific data */ @@ -118,9 +118,9 @@ static int snd_mts64_create(struct snd_card *card, /* Control Bits */ #define MTS64_CTL_READOUT 0x08 /* enable readout */ -#define MTS64_CTL_WRITE_CMD 0x06 -#define MTS64_CTL_WRITE_DATA 0x02 -#define MTS64_CTL_STROBE 0x01 +#define MTS64_CTL_WRITE_CMD 0x06 +#define MTS64_CTL_WRITE_DATA 0x02 +#define MTS64_CTL_STROBE 0x01 /* Command */ #define MTS64_CMD_RESET 0xfe @@ -130,7 +130,7 @@ static int snd_mts64_create(struct snd_card *card, #define MTS64_CMD_SMPTE_STOP 0xef #define MTS64_CMD_SMPTE_FPS_24 0xe3 #define MTS64_CMD_SMPTE_FPS_25 0xe2 -#define MTS64_CMD_SMPTE_FPS_2997 0xe4 +#define MTS64_CMD_SMPTE_FPS_2997 0xe4 #define MTS64_CMD_SMPTE_FPS_30D 0xe1 #define MTS64_CMD_SMPTE_FPS_30 0xe0 #define MTS64_CMD_COM_OPEN 0xf8 /* setting the communication mode */ @@ -159,7 +159,6 @@ static void mts64_write_command(struct parport *p, u8 c); static void mts64_write_data(struct parport *p, u8 c); static void mts64_write_midi(struct mts64 *mts, u8 c, int midiport); - /* Enables the readout procedure * * Before we can read a midi byte from the device, we have to set @@ -171,10 +170,10 @@ static void mts64_enable_readout(struct parport *p) c = parport_read_control(p); c |= MTS64_CTL_READOUT; - parport_write_control(p, c); + parport_write_control(p, c); } -/* Disables readout +/* Disables readout * * Readout is disabled by clearing bit 3 of control */ @@ -201,9 +200,9 @@ static int mts64_device_ready(struct parport *p) for (i = 0; i < 0xffff; ++i) { c = parport_read_status(p); c &= MTS64_STAT_BSY; - if (c != 0) + if (c != 0) return 1; - } + } return 0; } @@ -234,7 +233,7 @@ static int mts64_device_init(struct parport *p) return -EIO; } -/* +/* * Opens the device (set communication mode) */ static int mts64_device_open(struct mts64 *mts) @@ -248,7 +247,7 @@ static int mts64_device_open(struct mts64 *mts) return 0; } -/* +/* * Close device (clear communication mode) */ static int mts64_device_close(struct mts64 *mts) @@ -265,14 +264,14 @@ static int mts64_device_close(struct mts64 *mts) } /* map hardware port to substream number - * + * * When reading a byte from the device, the device tells us * on what port the byte is. This HW port has to be mapped to * the midiport (substream number). * substream 0-3 are Midiports 1-4 * substream 4 is SMPTE Timecode * The mapping is done by the table: - * HW | 0 | 1 | 2 | 3 | 4 + * HW | 0 | 1 | 2 | 3 | 4 * SW | 0 | 1 | 4 | 2 | 3 */ static u8 mts64_map_midi_input(u8 c) @@ -282,10 +281,9 @@ static u8 mts64_map_midi_input(u8 c) return map[c]; } - /* Probe parport for device * - * Do we have a Miditerminal 4140 on parport? + * Do we have a Miditerminal 4140 on parport? * Returns: * 0 device found * -ENODEV no device @@ -298,13 +296,13 @@ static int mts64_probe(struct parport *p) mts64_write_command(p, MTS64_CMD_PROBE); msleep(50); - + c = mts64_read(p); c &= 0x00ff; - if (c != MTS64_CMD_PROBE) + if (c != MTS64_CMD_PROBE) return -ENODEV; - else + else return 0; } @@ -330,7 +328,7 @@ static u16 mts64_read(struct parport *p) /* Read a byte from device * * Note, that readout mode has to be enabled. - * readout procedure is as follows: + * readout procedure is as follows: * - Write number of the Bit to read to DATA * - Read STATUS * - Bit 5 of STATUS indicates if Bit is set @@ -348,10 +346,10 @@ static u8 mts64_read_char(struct parport *p) parport_write_data(p, i); c >>= 1; status = parport_read_status(p); - if (status & MTS64_STAT_BIT_SET) + if (status & MTS64_STAT_BIT_SET) c |= 0x80; } - + return c; } @@ -369,9 +367,9 @@ static void mts64_smpte_start(struct parport *p, u8 seconds, u8 frames, u8 idx) { - static u8 fps[5] = { MTS64_CMD_SMPTE_FPS_24, + static u8 fps[5] = { MTS64_CMD_SMPTE_FPS_24, MTS64_CMD_SMPTE_FPS_25, - MTS64_CMD_SMPTE_FPS_2997, + MTS64_CMD_SMPTE_FPS_2997, MTS64_CMD_SMPTE_FPS_30D, MTS64_CMD_SMPTE_FPS_30 }; @@ -405,7 +403,7 @@ static void mts64_write_command(struct parport *p, u8 c) parport_write_control(p, MTS64_CTL_WRITE_CMD); } -/* Write a data byte to device +/* Write a data byte to device */ static void mts64_write_data(struct parport *p, u8 c) { @@ -652,7 +650,6 @@ static struct snd_kcontrol_new mts64_ctl_smpte_fps = { .put = snd_mts64_ctl_smpte_fps_put }; - static int snd_mts64_ctl_create(struct snd_card *card, struct mts64 *mts) { @@ -669,7 +666,7 @@ static int snd_mts64_ctl_create(struct snd_card *card, for (i = 0; control[i]; ++i) { err = snd_ctl_add(card, snd_ctl_new1(control[i], mts)); if (err < 0) { - snd_printd("Cannot create control: %s\n", + snd_printd("Cannot create control: %s\n", control[i]->name); return err; } @@ -688,8 +685,8 @@ static int snd_mts64_rawmidi_open(struct snd_rawmidi_substream *substream) struct mts64 *mts = substream->rmidi->private_data; if (mts->open_count == 0) { - /* We don't need a spinlock here, because this is just called - if the device has not been opened before. + /* We don't need a spinlock here, because this is just called + if the device has not been opened before. So there aren't any IRQs from the device */ mts64_device_open(mts); @@ -746,8 +743,8 @@ static void snd_mts64_rawmidi_input_trigger(struct snd_rawmidi_substream *substr if (up) mts->mode[substream->number] |= MTS64_MODE_INPUT_TRIGGERED; else - mts->mode[substream->number] &= ~MTS64_MODE_INPUT_TRIGGERED; - + mts->mode[substream->number] &= ~MTS64_MODE_INPUT_TRIGGERED; + spin_unlock_irqrestore(&mts->lock, flags); } @@ -771,42 +768,42 @@ static int snd_mts64_rawmidi_create(struct snd_card *card) struct snd_rawmidi_substream *substream; struct list_head *list; int err; - - err = snd_rawmidi_new(card, CARD_NAME, 0, - MTS64_NUM_OUTPUT_PORTS, - MTS64_NUM_INPUT_PORTS, + + err = snd_rawmidi_new(card, CARD_NAME, 0, + MTS64_NUM_OUTPUT_PORTS, + MTS64_NUM_INPUT_PORTS, &rmidi); - if (err < 0) + if (err < 0) return err; rmidi->private_data = mts; strcpy(rmidi->name, CARD_NAME); rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | - SNDRV_RAWMIDI_INFO_INPUT | - SNDRV_RAWMIDI_INFO_DUPLEX; + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; mts->rmidi = rmidi; /* register rawmidi ops */ - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mts64_rawmidi_output_ops); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mts64_rawmidi_input_ops); /* name substreams */ /* output */ - list_for_each(list, + list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { substream = list_entry(list, struct snd_rawmidi_substream, list); sprintf(substream->name, "Miditerminal %d", substream->number+1); } /* input */ - list_for_each(list, + list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) { substream = list_entry(list, struct snd_rawmidi_substream, list); mts->midi_input_substream[substream->number] = substream; - switch(substream->number) { + switch (substream->number) { case MTS64_SMPTE_SUBSTREAM: strcpy(substream->name, "Miditerminal SMPTE"); break; @@ -840,7 +837,7 @@ static void snd_mts64_interrupt(void *private) if (status & MTS64_STAT_PORT) { mts->current_midi_input_port = mts64_map_midi_input(data); } else { - if (mts->current_midi_input_port == -1) + if (mts->current_midi_input_port == -1) goto __out; substream = mts->midi_input_substream[mts->current_midi_input_port]; if (mts->mode[substream->number] & MTS64_MODE_INPUT_TRIGGERED) @@ -860,7 +857,7 @@ static int snd_mts64_probe_port(struct parport *p) 0, NULL); if (!pardev) return -EIO; - + if (parport_claim(pardev)) { parport_unregister_device(pardev); return -EIO; @@ -944,7 +941,7 @@ static int snd_mts64_probe(struct platform_device *pdev) if (dev >= SNDRV_CARDS) return -ENODEV; - if (!enable[dev]) + if (!enable[dev]) return -ENOENT; if ((err = snd_mts64_probe_port(p)) < 0) return err; @@ -957,7 +954,7 @@ static int snd_mts64_probe(struct platform_device *pdev) } strcpy(card->driver, DRIVER_NAME); strcpy(card->shortname, "ESI " CARD_NAME); - sprintf(card->longname, "%s at 0x%lx, irq %i", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); pardev = parport_register_device(p, /* port */ @@ -980,7 +977,7 @@ static int snd_mts64_probe(struct platform_device *pdev) } card->private_data = mts; card->private_free = snd_mts64_card_private_free; - + if ((err = snd_mts64_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); goto __err; @@ -1024,7 +1021,6 @@ static int snd_mts64_remove(struct platform_device *pdev) return 0; } - static struct platform_driver snd_mts64_driver = { .probe = snd_mts64_probe, .remove = snd_mts64_remove, @@ -1045,7 +1041,7 @@ static void snd_mts64_unregister_all(void) platform_device_unregister(platform_devices[i]); platform_devices[i] = NULL; } - } + } platform_driver_unregister(&snd_mts64_driver); parport_unregister_driver(&mts64_parport_driver); } -- 1.9.1

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[alsa-devel] sound: use-after-free in snd_timer_start1
by Dmitry Vyukov 04 Feb '16

04 Feb '16

Hello, The following program triggers use-after-free in snd_timer_start1: // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> int main() { syscall(SYS_mmap, 0x20000000ul, 0x664000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); int fd = syscall(SYS_open, "/dev/snd/timer", 0x40000ul, 0, 0, 0); *(uint32_t*)0x20653000 = (uint32_t)0x1; *(uint32_t*)0x20653004 = (uint32_t)0xfffffffffffffffe; *(uint32_t*)0x20653008 = (uint32_t)0x8; *(uint32_t*)0x2065300c = (uint32_t)0x0; *(uint32_t*)0x20653010 = (uint32_t)0x0; *(uint8_t*)0x20653014 = (uint8_t)0x0; *(uint8_t*)0x20653015 = (uint8_t)0x0; *(uint8_t*)0x20653016 = (uint8_t)0x0; *(uint8_t*)0x20653017 = (uint8_t)0x0; *(uint8_t*)0x20653018 = (uint8_t)0x0; *(uint8_t*)0x20653019 = (uint8_t)0x0; *(uint8_t*)0x2065301a = (uint8_t)0x0; *(uint8_t*)0x2065301b = (uint8_t)0x0; *(uint8_t*)0x2065301c = (uint8_t)0x0; *(uint8_t*)0x2065301d = (uint8_t)0x0; *(uint8_t*)0x2065301e = (uint8_t)0x0; *(uint8_t*)0x2065301f = (uint8_t)0x0; *(uint8_t*)0x20653020 = (uint8_t)0x0; *(uint8_t*)0x20653021 = (uint8_t)0x0; *(uint8_t*)0x20653022 = (uint8_t)0x0; *(uint8_t*)0x20653023 = (uint8_t)0x0; *(uint8_t*)0x20653024 = (uint8_t)0x0; *(uint8_t*)0x20653025 = (uint8_t)0x0; *(uint8_t*)0x20653026 = (uint8_t)0x0; *(uint8_t*)0x20653027 = (uint8_t)0x0; *(uint8_t*)0x20653028 = (uint8_t)0x0; *(uint8_t*)0x20653029 = (uint8_t)0x0; *(uint8_t*)0x2065302a = (uint8_t)0x0; *(uint8_t*)0x2065302b = (uint8_t)0x0; *(uint8_t*)0x2065302c = (uint8_t)0x0; *(uint8_t*)0x2065302d = (uint8_t)0x0; *(uint8_t*)0x2065302e = (uint8_t)0x0; *(uint8_t*)0x2065302f = (uint8_t)0x0; *(uint8_t*)0x20653030 = (uint8_t)0x0; *(uint8_t*)0x20653031 = (uint8_t)0x0; *(uint8_t*)0x20653032 = (uint8_t)0x0; *(uint8_t*)0x20653033 = (uint8_t)0x0; syscall(SYS_ioctl, fd, 0x40345410ul, 0x20653000ul, 0, 0, 0); syscall(SYS_ioctl, fd, 0x54a0ul, 0, 0, 0, 0); return 0; } ================================================================== BUG: KASAN: use-after-free in __list_add+0x233/0x22016/02/03 13:44:32 executing program 0: Read of size 8 by task syz-executor/27661 INFO: Allocated in snd_timer_instance_new+0x52/0x3a0 age=7 cpu=0 pid=27660 [< inline >] slab_alloc_node mm/slub.c:2562 [< inline >] slab_alloc mm/slub.c:2604 [< none >] kmem_cache_alloc_trace+0x25c/0x300 mm/slub.c:2621 [< inline >] kmalloc include/linux/slab.h:463 [< inline >] kzalloc include/linux/slab.h:607 [< none >] snd_timer_instance_new+0x52/0x3a0 sound/core/timer.c:106 [< none >] snd_timer_open+0x522/0xd20 sound/core/timer.c:289 [< inline >] snd_timer_user_tselect sound/core/timer.c:1598 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1874 [< none >] snd_timer_user_ioctl+0x8db/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 INFO: Freed in snd_timer_close+0x3ee/0x750 age=9 cpu=3 pid=27661 [< inline >] slab_free mm/slub.c:2835 [< none >] kfree+0x2ac/0x2c0 mm/slub.c:3664 [< none >] snd_timer_close+0x3ee/0x750 sound/core/timer.c:375 [< inline >] snd_timer_user_tselect sound/core/timer.c:1588 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1874 [< none >] snd_timer_user_ioctl+0x7b6/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 Call Trace: [< inline >] kasan_report mm/kasan/report.c:274 [<ffffffff8176640e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:295 [<ffffffff82c4a7e3>] __list_add+0x233/0x250 lib/list_debug.c:39 [< inline >] list_move_tail include/linux/list.h:77 [<ffffffff85240fdc>] snd_timer_start1+0x6c/0x2c0 sound/core/timer.c:432 [<ffffffff85241a7b>] snd_timer_start+0x20b/0x2b0 sound/core/timer.c:499 [< inline >] snd_timer_user_start sound/core/timer.c:1799 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1883 [<ffffffff85245fb7>] snd_timer_user_ioctl+0x767/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [<ffffffff817fa9ec>] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [<ffffffff817fb89f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [<ffffffff8665ebb6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 ================================================================== Sometimes it manifests as: ------------[ cut here ]------------ WARNING: CPU: 1 PID: 6548 at lib/list_debug.c:42 __list_add+0x1db/0x250() list_add corruption. prev->next should be next (ffff8800697c10e8), but was ffff880030441fd0. (prev=ffff880030441fd0). Modules linked in: CPU: 1 PID: 6548 Comm: a.out Not tainted 4.5.0-rc2+ #309 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff88002fa17918 ffffffff82be2c8d ffff88002fa17988 ffff88002fd62f80 ffffffff86acf380 ffff88002fa17958 ffffffff81355139 ffffffff82c4a78b ffffed0005f42f2d ffffffff86acf380 000000000000002a Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82be2c8d>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 [<ffffffff81355139>] warn_slowpath_common+0xd9/0x140 kernel/panic.c:482 [<ffffffff81355249>] warn_slowpath_fmt+0xa9/0xd0 kernel/panic.c:494 [<ffffffff82c4a78b>] __list_add+0x1db/0x250 lib/list_debug.c:39 [< inline >] list_move_tail include/linux/list.h:77 [<ffffffff85240fdc>] snd_timer_start1+0x6c/0x2c0 sound/core/timer.c:432 [<ffffffff85241a7b>] snd_timer_start+0x20b/0x2b0 sound/core/timer.c:499 [< inline >] snd_timer_user_start sound/core/timer.c:1799 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1883 [<ffffffff85245fb7>] snd_timer_user_ioctl+0x767/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [<ffffffff817fa9ec>] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [<ffffffff817fb89f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [<ffffffff8665ebb6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 ---[ end trace 77da812dd399e78b ]--- On commit 34229b277480f46c1e9a19f027f30b074512e68b + recent fixes from Takashi.

2 3

[alsa-devel] [PATCH 00/11] ASoC: Intel: Skylake: Driver fixes
by Vinod Koul 04 Feb '16

04 Feb '16

Now that 4.5-rc1 has audio functional with driver, time to fix some corner cases in driver. This series tries to fix: o resource leakage on error cases while allocation and freeup o wrap of delay reporting o fix of various corner cases in topology and support of multiple sinks/source ona pipe and without a mixer termination o tlv buffer overwrite as suggested by Takashi o removal of autosuspend delay o 2 ch support on rt286 machine Although patches are 11, the diffstat is only +59, -29 Dharageswari.R (1): ASoC: Intel: Skylake: Fix pipe memory allocation leak Guneshwor Singh (2): ASoC: Intel: Skylake: Fix delay wrap condition ASoC: Intel: Skylake: Fix mcps freeup after module unbind failure Jeeja KP (6): ASoC: Intel: Skylake: Fix return of skl_get_queue_index ASoC: Intel: Skylake: Fix the module state check condition ASoC: Intel: Skylake: Fix not to stop sink pipe in pga pmd event ASoC: Intel: Skylake: Fix bind of source with multiple sinks ASoC: Intel: Skylake: Add missing PRE/POST_PMU handlers for vmixer ASoC: Intel: Skylake: Fix stereo DMIC record Vinod Koul (2): ASoC: Intel: Skylake: Fix the memory overwrite of tlv buffer ASoC: Intel: Skylake: Remove autosuspend delay sound/soc/intel/boards/skl_rt286.c | 5 ++- sound/soc/intel/skylake/skl-messages.c | 6 +-- sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/intel/skylake/skl-topology.c | 74 +++++++++++++++++++++++----------- sound/soc/intel/skylake/skl.c | 2 - 5 files changed, 59 insertions(+), 29 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: arizona: Add support for SNDRV_PCM_RATE_KNOT
by Charles Keepax 04 Feb '16

04 Feb '16

The Arizona CODECs support several rates that do not have simple defines in ALSA. This patch adds support for SNDRV_PCM_RATE_KNOT so that users can open stream at these rates. As part of this we should always set constraints in arizona_startup, currently we only set the constraints if we already have a clock to limit rates to that family of sample rates. This patch updates this to set a constraint of all rates supported by the chip if we do not already know which family of rates to limit to. Finally we also reduce the list of rates supported in the constraints to only include those that are supported on current parts. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 47 ++++++++++++++++++---------------------------- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- 6 files changed, 23 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 33143fe..5b16834 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1398,23 +1398,10 @@ static const int arizona_48k_bclk_rates[] = { 24576000, }; -static const unsigned int arizona_48k_rates[] = { - 12000, - 24000, - 48000, - 96000, - 192000, - 384000, - 768000, - 4000, - 8000, - 16000, - 32000, - 64000, - 128000, - 256000, - 512000, -}; +#define ARIZONA_48K_RATES 12000, 24000, 48000, 96000, 192000, 4000, 8000, \ + 16000, 32000 + +static const unsigned int arizona_48k_rates[] = { ARIZONA_48K_RATES }; static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { .count = ARRAY_SIZE(arizona_48k_rates), @@ -1443,15 +1430,9 @@ static const int arizona_44k1_bclk_rates[] = { 22579200, }; -static const unsigned int arizona_44k1_rates[] = { - 11025, - 22050, - 44100, - 88200, - 176400, - 352800, - 705600, -}; +#define ARIZONA_44K1_RATES 11025, 22050, 44100, 88200, 176400 + +static const unsigned int arizona_44k1_rates[] = { ARIZONA_44K1_RATES }; static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { .count = ARRAY_SIZE(arizona_44k1_rates), @@ -1485,6 +1466,15 @@ static int arizona_sr_vals[] = { 512000, }; +static const unsigned int arizona_rates[] = { + ARIZONA_48K_RATES, ARIZONA_44K1_RATES +}; + +static const struct snd_pcm_hw_constraint_list arizona_constraint = { + .count = ARRAY_SIZE(arizona_rates), + .list = arizona_rates, +}; + static int arizona_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1509,9 +1499,8 @@ static int arizona_startup(struct snd_pcm_substream *substream, } if (base_rate == 0) - return 0; - - if (base_rate % 8000) + constraint = &arizona_constraint; + else if (base_rate % 8000) constraint = &arizona_44k1_constraint; else constraint = &arizona_48k_constraint; diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index b8d3cd0..576087b 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -923,7 +923,7 @@ static int cs47l24_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define CS47L24_RATES SNDRV_PCM_RATE_8000_192000 +#define CS47L24_RATES SNDRV_PCM_RATE_KNOT #define CS47L24_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index aa1e96a..a8b3e3f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1738,7 +1738,7 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM5102_RATES SNDRV_PCM_RATE_8000_192000 +#define WM5102_RATES SNDRV_PCM_RATE_KNOT #define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 302a8d3..83ba70f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2024,7 +2024,7 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM5110_RATES SNDRV_PCM_RATE_8000_192000 +#define WM5110_RATES SNDRV_PCM_RATE_KNOT #define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index b4dba3a..52d766e 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -943,7 +943,7 @@ static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 +#define WM8997_RATES SNDRV_PCM_RATE_KNOT #define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 7719bc5..0123960 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1170,7 +1170,7 @@ static const struct snd_soc_dapm_route wm8998_dapm_routes[] = { { "DRC1 Signal Activity", NULL, "DRC1R" }, }; -#define WM8998_RATES SNDRV_PCM_RATE_8000_192000 +#define WM8998_RATES SNDRV_PCM_RATE_KNOT #define WM8998_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.1.4

3 7

[alsa-devel] [PATCH] ASoC: dapm: Don't prefix autodisable widgets twice
by Lars-Peter Clausen 04 Feb '16

04 Feb '16

When a DAPM context has a prefix the autodisable widgets get prefixed twice, once for the control and once for the widget. To avoid this use the un-prefixed control name to construct the autodisable widget name. This change is purely cosmetic. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5a2812f..0d37079 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -310,7 +310,7 @@ struct dapm_kcontrol_data { }; static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol) + struct snd_kcontrol *kcontrol, const char *ctrl_name) { struct dapm_kcontrol_data *data; struct soc_mixer_control *mc; @@ -333,7 +333,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, if (mc->autodisable) { struct snd_soc_dapm_widget template; - name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name, + name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name, "Autodisable"); if (!name) { ret = -ENOMEM; @@ -371,7 +371,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, if (e->autodisable) { struct snd_soc_dapm_widget template; - name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name, + name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name, "Autodisable"); if (!name) { ret = -ENOMEM; @@ -871,7 +871,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w, kcontrol->private_free = dapm_kcontrol_free; - ret = dapm_kcontrol_data_alloc(w, kcontrol); + ret = dapm_kcontrol_data_alloc(w, kcontrol, name); if (ret) { snd_ctl_free_one(kcontrol); goto exit_free; -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Don't add DAPM routes for ASRC if it doesn't exist
by Nicolin Chen 04 Feb '16

04 Feb '16

From: Nicolin Chen <nicolinc(a)nvidia.com> There are a pair of warnings when ASRC is absent in the DTB: fsl-asoc-card sound: ASoC: no source widget found for ASRC-Playback fsl-asoc-card sound: ASoC: Failed to add route ASRC-Playback -> direct -> CPU-Playback fsl-asoc-card sound: ASoC: no sink widget found for ASRC-Capture fsl-asoc-card sound: ASoC: Failed to add route CPU-Capture -> direct -> ASRC-Capture This is because the driver is still trying to add DAPM routes for ASRC even if it doesn't exist on that platform. The warnings are harmless but it might be annoying. So this patch drops the DAPM routes of ASRC when it's absent in the DAI link. Signed-off-by: Nicolin Chen <nicolinc(a)nvidia.com> --- sound/soc/fsl/fsl-asoc-card.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 562b3bd..3d40797 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -99,19 +99,26 @@ struct fsl_asoc_card_priv { /** * This dapm route map exsits for DPCM link only. * The other routes shall go through Device Tree. + * + * Note: keep all ASRC routes in the second half + * to drop them easily for non-ASRC cases. */ static const struct snd_soc_dapm_route audio_map[] = { - {"CPU-Playback", NULL, "ASRC-Playback"}, + /* 1st half -- Normal DAPM routes */ {"Playback", NULL, "CPU-Playback"}, - {"ASRC-Capture", NULL, "CPU-Capture"}, {"CPU-Capture", NULL, "Capture"}, + /* 2nd half -- ASRC DAPM routes */ + {"CPU-Playback", NULL, "ASRC-Playback"}, + {"ASRC-Capture", NULL, "CPU-Capture"}, }; static const struct snd_soc_dapm_route audio_map_ac97[] = { - {"AC97 Playback", NULL, "ASRC-Playback"}, + /* 1st half -- Normal DAPM routes */ {"Playback", NULL, "AC97 Playback"}, - {"ASRC-Capture", NULL, "AC97 Capture"}, {"AC97 Capture", NULL, "Capture"}, + /* 2nd half -- ASRC DAPM routes */ + {"AC97 Playback", NULL, "ASRC-Playback"}, + {"ASRC-Capture", NULL, "AC97 Capture"}, }; /* Add all possible widgets into here without being redundant */ @@ -593,6 +600,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); + /* Drop the second half of DAPM routes -- ASRC */ + if (!asrc_pdev) + priv->card.num_dapm_routes /= 2; + memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: cs42xx8: fix the noise in the right dac channel when playback mono wav with record in background
by Zidan Wang 04 Feb '16

04 Feb '16

When playback mono wav with record in background, there will be some nosie in the right dac channel. It seems that the ADC data has been routed to the dac channel. The cs42888 have 8 dac channels, it's appropriate to mute the unused dac channels, and the noise will disappear. Steps to reproduce this issue: arecord -D hw:0,0 -f S16_LE -r 48000 -c 1 a.wav & aplay -Dhw:0,0 audio48k16M.wav Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/cs42xx8.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index d562e1b..1179101 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -44,6 +44,7 @@ struct cs42xx8_priv { bool slave_mode; unsigned long sysclk; + u32 tx_channels; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -257,6 +258,9 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, u32 ratio = cs42xx8->sysclk / params_rate(params); u32 i, fm, val, mask; + if (tx) + cs42xx8->tx_channels = params_channels(params); + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { if (cs42xx8_ratios[i].ratio == ratio) break; @@ -283,9 +287,11 @@ static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + u8 dac_unmute = cs42xx8->tx_channels ? + ~((0x1 << cs42xx8->tx_channels) - 1) : 0; - regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE, - CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0); + regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, + mute ? CS42XX8_DACMUTE_ALL : dac_unmute); return 0; } -- 1.9.1

2 1

[alsa-devel] [GIT PULL]: Update sst audio firmware for Skylake
by Vinod Koul 04 Feb '16

04 Feb '16

Hello Linux firmware maintainers, Please PULL to receive the updated version of Skylake audio firmware. This PULL request has two patches, first one creates a sym link for binary name which helps us to keep different versions of firmware in git Second one updates to latest FW v869 The following changes since commit 52442afee9907bc32a058f22bb3295d040677c26: brcm: add firmware for bcm4366 device (2016-01-25 15:15:34 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to f7996657e8f25386541593242cb8e906d9808ec7: linux-firmware: intel: Update Skylake audio firmware (2016-02-04 10:36:52 +0530) ---------------------------------------------------------------- Vinod Koul (2): linux-firmware: intel: add symbolic link and file versions linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | Bin 233472 -> 23 bytes intel/dsp_fw_release_v827.bin | Bin 0 -> 233472 bytes intel/dsp_fw_release_v869.bin | Bin 0 -> 233472 bytes 4 files changed, 2 insertions(+), 2 deletions(-) mode change 100644 => 120000 intel/dsp_fw_release.bin create mode 100644 intel/dsp_fw_release_v827.bin create mode 100644 intel/dsp_fw_release_v869.bin Thanks -- ~Vinod

1 0

[alsa-devel] [PATCH 00/31] Sharing media resources across ALSA and au0828 drivers
by Shuah Khan 03 Feb '16

03 Feb '16

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API and Media Controller API to share tuner. Media Controller Next Generation API has been enhanced to add two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Mauro and Takashi: I am hoping you can both coordinate and decide on ALSA patches once the reviews are done and the patches look good. Design Highlights: 1. ALSA to check for resources and hold in snd_usb_hw_params(), and release from snd_usb_hw_free(). This change fixed the lockdep warnings seen when resources were held in TRIGGER_START and released from TRIGGER_STOP which could run in IRQ context. I acknowledge Clemens Ladisch for suggesting the correct places to hold/free resources to avoid IRQ path complications. 2. The Bridge driver (au0828) owns and drives the graph creation as well as enabling and disabling tuner. It also keeps state information to avoid graph walks in enable_source and disable_source handler. I acknowledge Hans Verkuil for his suggestions and ideas for this change. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. Please find the graphs generated using mc_nextgen_test tool at various points during testing at: https://drive.google.com/folderview?id=0B0NIL0BQg-AlRndaaXViSXdPeTA&usp=sha… This patch series is a rebase and update to the latest Media Controller Next Gen API of the Media Controller Next Generation API port. Rebase highlights: 1. Changes to videobuf2 framework required additional v4l2 helper function to enable the tuner from vb2_core_streamon(). Patch 9 has this work. 2. Media device initialization and registration steps are now split in the latest Media Controller Next Gen API. This required changes to au0828 and ALSA to coordinate initialization and registration of the Managed Media Device they share. Patch 19 shows au0828 changes. ALSA changes for this are folded into the main ALSA patch 26. 3. Media device unregistration does more cleanup now. As a result, au0828 and ALSA need to coordinate so media device is done only once. These changes are in patches: 29, 30, and 31. 4. In addition, media device resources cleanup gets done before snd_card_disconnect(). This work is in patch 30. Media Controller Next Generation API port from Media Controller API: The patch series below was a port of the Patch v3 - Media Controller API to Next Gen API and Mixer patch update to address comments and bug fixes: https://www.mail-archive.com/linux-media@vger.kernel.org/msg93086.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg93417.html Patch v3 - Media Controller API: https://www.mail-archive.com/linux-media@vger.kernel.org/msg92572.html Mauro Carvalho Chehab (1): uapi/media.h: Declare interface types for ALSA Shuah Khan (30): media: Add ALSA Media Controller function entities media: Media Controller register/unregister entity_notify API media: Media Controller enable/disable source handler API media: Media Controller fix to not let stream_count go negative media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add v4l_enable/disable_media_tuner() helper functions media: v4l2-core add v4l_vb2q_enable_media_tuner() helper media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: au0828 fix au0828_create_media_graph() entity checks media: Change v4l-core to check for tuner availability media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_tuner() media: au0828 change to use Managed Media Controller API media: au0828 handle media_init and media_register window media: au0828 change to register/unregister entity_notify hook media: au0828 create tuner to decoder link in deactivated state media: dvb-core create tuner to demod pad link in disabled state media: au0828 implement enable_source and disable_source handlers media: au0828 fix null pointer reference in au0828_create_media_graph() media: au0828 fix to not call media_device_unregister_entity_notify() sound/usb: Update ALSA driver to use Managed Media Controller API sound/usb: Create media mixer function and control interface entities media: au0828 create link between ALSA Mixer and decoder media: track media device unregister in progress sound/usb: Check media device unregister progress state media: au0828 change to check media device unregister progress state drivers/media/dvb-core/dvb_frontend.c | 139 ++--------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-core/dvbdev.c | 3 +- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 66 ++++- drivers/media/media-entity.c | 65 +++-- drivers/media/usb/au0828/au0828-core.c | 355 ++++++++++++++++++++++----- drivers/media/usb/au0828/au0828-video.c | 75 +----- drivers/media/usb/au0828/au0828.h | 9 + drivers/media/v4l2-core/v4l2-dev.c | 48 ++++ drivers/media/v4l2-core/v4l2-fh.c | 1 + drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 4 + include/media/media-device.h | 61 +++++ include/media/media-entity.h | 12 + include/media/v4l2-dev.h | 5 + include/uapi/linux/media.h | 17 +- sound/usb/Makefile | 15 +- sound/usb/card.c | 16 ++ sound/usb/card.h | 1 + sound/usb/media.c | 325 ++++++++++++++++++++++++ sound/usb/media.h | 75 ++++++ sound/usb/mixer.h | 1 + sound/usb/pcm.c | 26 +- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 1 + sound/usb/usbaudio.h | 2 + 30 files changed, 1111 insertions(+), 270 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.5.0

5 78

Re: [alsa-devel] bytcr_rt5651 on wintron 7.0
by Tobias Schramm 03 Feb '16

03 Feb '16

Hi, I'm tinkering on the wintron 7.0 together with Tobias Mädel and have tested your patch with your UCM config on Linux 4.5.0-rc2. It looks like we are getting somewhere. The soundcard is now being recognized by ALSA. But there are still some errors in that seem to be related to the audio driver. As soon as I try to output any sound I get an error: [ 88.906739] intel_sst_acpi 80860F28:00: fw download failed -16 [ 88.909262] intel_sst_acpi 80860F28:00: FW download fail -16 [ 88.948434] Audio Port: ASoC: no backend DAIs enabled for Audio Port Does anybody have an idea why the sound chip is not responding and the fw download timeouts? I've uploaded dmesg, lsmod and tree /proc/asound output here: https://github.com/TobleMiner/wintron7.0/tree/master/logs/4.5.0-rc2-ARCH-ts… Regards, Tobias Mädel and Tobias Schramm

2 1

[alsa-devel] [PATCH v2] ASoC: rt5514: add rt5514 codec driver
by Oder Chiou 03 Feb '16

03 Feb '16

This is the initial codec driver for rt5514. The codec includes a low power DSP for voice wake up. The register address is incremental by 4 in the DSP memory map. In order to recover the codec settings in the codec mode and manipulate the DSP mode for voice wake up, we use the multi-level register map. One is for ALSA API in codec mode that can be recovered by cache before recording. Another is for DSP related settings that can be accessed with 32bit address of the DSP in the application of voice wake up. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5514.txt | 25 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5514.c | 982 +++++++++++++++++++++ sound/soc/codecs/rt5514.h | 252 ++++++ 5 files changed, 1267 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5514.txt create mode 100644 sound/soc/codecs/rt5514.c create mode 100644 sound/soc/codecs/rt5514.h diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt new file mode 100644 index 0000000..e24436f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -0,0 +1,25 @@ +RT5514 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5514". + +- reg : The I2C address of the device. + +Pins on the device (for linking into audio routes) for RT5514: + + * DMIC1L + * DMIC1R + * DMIC2L + * DMIC2R + * AMICL + * AMICR + +Example: + +codec: rt5514@57 { + compatible = "realtek,rt5514"; + reg = <0x57>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 50693c8..dabd479 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -95,6 +95,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C + select SND_SOC_RT5514 if I2C select SND_SOC_RT5616 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -565,6 +566,7 @@ config SND_SOC_PCM512x_SPI config SND_SOC_RL6231 tristate + default y if SND_SOC_RT5514=y default y if SND_SOC_RT5616=y default y if SND_SOC_RT5640=y default y if SND_SOC_RT5645=y @@ -572,6 +574,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5659=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y + default m if SND_SOC_RT5514=m default m if SND_SOC_RT5616=m default m if SND_SOC_RT5640=m default m if SND_SOC_RT5645=m @@ -595,6 +598,9 @@ config SND_SOC_RT298 tristate depends on I2C +config SND_SOC_RT5514 + tristate + config SND_SOC_RT5616 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d44f7d3..79f95dd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -92,6 +92,7 @@ snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o +snd-soc-rt5514-objs := rt5514.o snd-soc-rt5616-objs := rt5616.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -296,6 +297,7 @@ obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o +obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c new file mode 100644 index 0000000..879bf60 --- /dev/null +++ b/sound/soc/codecs/rt5514.c @@ -0,0 +1,982 @@ +/* + * rt5514.c -- RT5514 ALSA SoC audio codec driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/fs.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" +#include "rt5514.h" + +static const struct reg_sequence rt5514_i2c_patch[] = { + {0x1800101c, 0x00000000}, + {0x18001100, 0x0000031f}, + {0x18001104, 0x00000007}, + {0x18001108, 0x00000000}, + {0x1800110c, 0x00000000}, + {0x18001110, 0x00000000}, + {0x18001114, 0x00000001}, + {0x18001118, 0x00000000}, + {0x18002f08, 0x00000006}, + {0x18002f00, 0x00055149}, + {0x18002f00, 0x0005514b}, + {0x18002f00, 0x00055149}, + {0xfafafafa, 0x00000001}, + {0x18002f10, 0x00000001}, + {0x18002f10, 0x00000000}, + {0x18002f10, 0x00000001}, + {0xfafafafa, 0x00000001}, + {0x18002000, 0x000010ec}, + {0xfafafafa, 0x00000000}, +}; + +static const struct reg_sequence rt5514_patch[] = { + {RT5514_DIG_IO_CTRL, 0x00000040}, + {RT5514_CLK_CTRL1, 0x38020041}, + {RT5514_SRC_CTRL, 0x44000eee}, + {RT5514_ANA_CTRL_LDO10, 0x00028604}, + {RT5514_ANA_CTRL_ADCFED, 0x00000800}, +}; + +static const struct reg_default rt5514_reg[] = { + {RT5514_RESET, 0x00000000}, + {RT5514_PWR_ANA1, 0x00808880}, + {RT5514_PWR_ANA2, 0x00220000}, + {RT5514_I2S_CTRL1, 0x00000330}, + {RT5514_I2S_CTRL2, 0x20000000}, + {RT5514_VAD_CTRL6, 0xc00007d2}, + {RT5514_EXT_VAD_CTRL, 0x80000080}, + {RT5514_DIG_IO_CTRL, 0x00000040}, + {RT5514_PAD_CTRL1, 0x00804000}, + {RT5514_DMIC_DATA_CTRL, 0x00000005}, + {RT5514_DIG_SOURCE_CTRL, 0x00000002}, + {RT5514_SRC_CTRL, 0x44000eee}, + {RT5514_DOWNFILTER2_CTRL1, 0x0000882f}, + {RT5514_PLL_SOURCE_CTRL, 0x00000004}, + {RT5514_CLK_CTRL1, 0x38020041}, + {RT5514_CLK_CTRL2, 0x00000000}, + {RT5514_PLL3_CALIB_CTRL1, 0x00400200}, + {RT5514_PLL3_CALIB_CTRL5, 0x40220012}, + {RT5514_DELAY_BUF_CTRL1, 0x7fff006a}, + {RT5514_DELAY_BUF_CTRL3, 0x00000000}, + {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, + {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, + {RT5514_DOWNFILTER0_CTRL3, 0x00000362}, + {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, + {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, + {RT5514_DOWNFILTER1_CTRL3, 0x00000362}, + {RT5514_ANA_CTRL_LDO10, 0x00028604}, + {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, + {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, + {RT5514_ANA_CTRL_ADC21, 0x00001180}, + {RT5514_ANA_CTRL_ADC22, 0x0000aaa8}, + {RT5514_ANA_CTRL_ADC23, 0x00151427}, + {RT5514_ANA_CTRL_MICBST, 0x00002000}, + {RT5514_ANA_CTRL_ADCFED, 0x00000800}, + {RT5514_ANA_CTRL_INBUF, 0x00000143}, + {RT5514_ANA_CTRL_VREF, 0x00008d50}, + {RT5514_ANA_CTRL_PLL3, 0x0000000e}, + {RT5514_ANA_CTRL_PLL1_1, 0x00000000}, + {RT5514_ANA_CTRL_PLL1_2, 0x00030220}, + {RT5514_DMIC_LP_CTRL, 0x00000000}, + {RT5514_MISC_CTRL_DSP, 0x00000000}, + {RT5514_DSP_CTRL1, 0x00055149}, + {RT5514_DSP_CTRL3, 0x00000006}, + {RT5514_DSP_CTRL4, 0x00000001}, + {RT5514_VENDOR_ID1, 0x00000001}, + {RT5514_VENDOR_ID2, 0x10ec5514}, +}; + +static bool rt5514_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5514_VENDOR_ID1: + case RT5514_VENDOR_ID2: + return true; + + default: + return false; + } +} + +static bool rt5514_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5514_RESET: + case RT5514_PWR_ANA1: + case RT5514_PWR_ANA2: + case RT5514_I2S_CTRL1: + case RT5514_I2S_CTRL2: + case RT5514_VAD_CTRL6: + case RT5514_EXT_VAD_CTRL: + case RT5514_DIG_IO_CTRL: + case RT5514_PAD_CTRL1: + case RT5514_DMIC_DATA_CTRL: + case RT5514_DIG_SOURCE_CTRL: + case RT5514_SRC_CTRL: + case RT5514_DOWNFILTER2_CTRL1: + case RT5514_PLL_SOURCE_CTRL: + case RT5514_CLK_CTRL1: + case RT5514_CLK_CTRL2: + case RT5514_PLL3_CALIB_CTRL1: + case RT5514_PLL3_CALIB_CTRL5: + case RT5514_DELAY_BUF_CTRL1: + case RT5514_DELAY_BUF_CTRL3: + case RT5514_DOWNFILTER0_CTRL1: + case RT5514_DOWNFILTER0_CTRL2: + case RT5514_DOWNFILTER0_CTRL3: + case RT5514_DOWNFILTER1_CTRL1: + case RT5514_DOWNFILTER1_CTRL2: + case RT5514_DOWNFILTER1_CTRL3: + case RT5514_ANA_CTRL_LDO10: + case RT5514_ANA_CTRL_LDO18_16: + case RT5514_ANA_CTRL_ADC12: + case RT5514_ANA_CTRL_ADC21: + case RT5514_ANA_CTRL_ADC22: + case RT5514_ANA_CTRL_ADC23: + case RT5514_ANA_CTRL_MICBST: + case RT5514_ANA_CTRL_ADCFED: + case RT5514_ANA_CTRL_INBUF: + case RT5514_ANA_CTRL_VREF: + case RT5514_ANA_CTRL_PLL3: + case RT5514_ANA_CTRL_PLL1_1: + case RT5514_ANA_CTRL_PLL1_2: + case RT5514_DMIC_LP_CTRL: + case RT5514_MISC_CTRL_DSP: + case RT5514_DSP_CTRL1: + case RT5514_DSP_CTRL3: + case RT5514_DSP_CTRL4: + case RT5514_VENDOR_ID1: + case RT5514_VENDOR_ID2: + return true; + + default: + return false; + } +} + +static bool rt5514_i2c_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case RT5514_DSP_MAPPING | RT5514_RESET: + case RT5514_DSP_MAPPING | RT5514_PWR_ANA1: + case RT5514_DSP_MAPPING | RT5514_PWR_ANA2: + case RT5514_DSP_MAPPING | RT5514_I2S_CTRL1: + case RT5514_DSP_MAPPING | RT5514_I2S_CTRL2: + case RT5514_DSP_MAPPING | RT5514_VAD_CTRL6: + case RT5514_DSP_MAPPING | RT5514_EXT_VAD_CTRL: + case RT5514_DSP_MAPPING | RT5514_DIG_IO_CTRL: + case RT5514_DSP_MAPPING | RT5514_PAD_CTRL1: + case RT5514_DSP_MAPPING | RT5514_DMIC_DATA_CTRL: + case RT5514_DSP_MAPPING | RT5514_DIG_SOURCE_CTRL: + case RT5514_DSP_MAPPING | RT5514_SRC_CTRL: + case RT5514_DSP_MAPPING | RT5514_DOWNFILTER2_CTRL1: + case RT5514_DSP_MAPPING | RT5514_PLL_SOURCE_CTRL: + case RT5514_DSP_MAPPING | RT5514_CLK_CTRL1: + case RT5514_DSP_MAPPING | RT5514_CLK_CTRL2: + case RT5514_DSP_MAPPING | RT5514_PLL3_CALIB_CTRL1: + case RT5514_DSP_MAPPING | RT5514_PLL3_CALIB_CTRL5: + case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL1: + case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL3: + case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL1: + case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL2: + case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL3: + case RT5514_DSP_MAPPING | RT5514_DOWNFILTER1_CTRL1: + case RT5514_DSP_MAPPING | RT5514_DOWNFILTER1_CTRL2: + case RT5514_DSP_MAPPING | RT5514_DOWNFILTER1_CTRL3: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_LDO10: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_LDO18_16: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_ADC12: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_ADC21: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_ADC22: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_ADC23: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_MICBST: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_ADCFED: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_INBUF: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_VREF: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_PLL3: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_PLL1_1: + case RT5514_DSP_MAPPING | RT5514_ANA_CTRL_PLL1_2: + case RT5514_DSP_MAPPING | RT5514_DMIC_LP_CTRL: + case RT5514_DSP_MAPPING | RT5514_MISC_CTRL_DSP: + case RT5514_DSP_MAPPING | RT5514_DSP_CTRL1: + case RT5514_DSP_MAPPING | RT5514_DSP_CTRL3: + case RT5514_DSP_MAPPING | RT5514_DSP_CTRL4: + case RT5514_DSP_MAPPING | RT5514_VENDOR_ID1: + case RT5514_DSP_MAPPING | RT5514_VENDOR_ID2: + return true; + + default: + return false; + } +} + +/* {-3, 0, +3, +4.5, +7.5, +9.5, +12, +14, +17} dB */ +static const DECLARE_TLV_DB_RANGE(bst_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-300, 300, 0), + 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(750, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(950, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(1200, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(1400, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(1700, 0, 0) +); + +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); + +static const struct snd_kcontrol_new rt5514_snd_controls[] = { + SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, + RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), + SOC_DOUBLE_R_TLV("ADC1 Capture Volume", RT5514_DOWNFILTER0_CTRL1, + RT5514_DOWNFILTER0_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, + adc_vol_tlv), + SOC_DOUBLE_R_TLV("ADC2 Capture Volume", RT5514_DOWNFILTER1_CTRL1, + RT5514_DOWNFILTER1_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, + adc_vol_tlv), +}; + +/* ADC Mixer*/ +static const struct snd_kcontrol_new rt5514_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("DMIC Switch", RT5514_DOWNFILTER0_CTRL1, + RT5514_AD_DMIC_MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("ADC Switch", RT5514_DOWNFILTER0_CTRL1, + RT5514_AD_AD_MIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5514_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("DMIC Switch", RT5514_DOWNFILTER0_CTRL2, + RT5514_AD_DMIC_MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("ADC Switch", RT5514_DOWNFILTER0_CTRL2, + RT5514_AD_AD_MIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5514_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("DMIC Switch", RT5514_DOWNFILTER1_CTRL1, + RT5514_AD_DMIC_MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("ADC Switch", RT5514_DOWNFILTER1_CTRL1, + RT5514_AD_AD_MIX_BIT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5514_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("DMIC Switch", RT5514_DOWNFILTER1_CTRL2, + RT5514_AD_DMIC_MIX_BIT, 1, 1), + SOC_DAPM_SINGLE("ADC Switch", RT5514_DOWNFILTER1_CTRL2, + RT5514_AD_AD_MIX_BIT, 1, 1), +}; + +/* DMIC Source */ +static const char * const rt5514_dmic_src[] = { + "DMIC1", "DMIC2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5514_stereo1_dmic_enum, RT5514_DIG_SOURCE_CTRL, + RT5514_AD0_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); + +static const struct snd_kcontrol_new rt5514_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5514_stereo1_dmic_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5514_stereo2_dmic_enum, RT5514_DIG_SOURCE_CTRL, + RT5514_AD1_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); + +static const struct snd_kcontrol_new rt5514_sto2_dmic_mux = + SOC_DAPM_ENUM("Stereo2 DMIC Source", rt5514_stereo2_dmic_enum); + +/** + * rt5514_calc_dmic_clk - Calculate the frequency divider parameter of dmic. + * + * @rate: base clock rate. + * + * Choose divider parameter that gives the highest possible DMIC frequency in + * 1MHz - 3MHz range. + */ +static int rt5514_calc_dmic_clk(struct snd_soc_codec *codec, int rate) +{ + int div[] = {2, 3, 4, 8, 12, 16, 24, 32}; + int i; + + if (rate < 1000000 * div[0]) { + pr_warn("Base clock rate %d is too low\n", rate); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(div); i++) { + /* find divider that gives DMIC frequency below 3.072MHz */ + if (3072000 * div[i] >= rate) + return i; + } + + dev_warn(codec->dev, "Base clock rate %d is too high\n", rate); + return -EINVAL; +} + +static int rt5514_set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + int idx; + + idx = rt5514_calc_dmic_clk(codec, rt5514->sysclk); + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL1, + RT5514_CLK_DMIC_OUT_SEL_MASK, + idx << RT5514_CLK_DMIC_OUT_SEL_SFT); + + return idx; +} + +static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + if (rt5514->sysclk_src == RT5514_SCLK_S_PLL1) + return 1; + else + return 0; +} + +static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1L"), + SND_SOC_DAPM_INPUT("DMIC1R"), + SND_SOC_DAPM_INPUT("DMIC2L"), + SND_SOC_DAPM_INPUT("DMIC2R"), + + SND_SOC_DAPM_INPUT("AMICL"), + SND_SOC_DAPM_INPUT("AMICR"), + + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + rt5514_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY("ADC CLK", RT5514_CLK_CTRL1, + RT5514_CLK_AD_ANA1_EN_BIT, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("LDO18 IN", RT5514_PWR_ANA1, + RT5514_POW_LDO18_IN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO18 ADC", RT5514_PWR_ANA1, + RT5514_POW_LDO18_ADC_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO21", RT5514_PWR_ANA1, RT5514_POW_LDO21_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("BG LDO18 IN", RT5514_PWR_ANA1, + RT5514_POW_BG_LDO18_IN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BG LDO21", RT5514_PWR_ANA1, + RT5514_POW_BG_LDO21_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BG MBIAS", RT5514_PWR_ANA2, + RT5514_POW_BG_MBIAS_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MBIAS", RT5514_PWR_ANA2, RT5514_POW_MBIAS_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF2", RT5514_PWR_ANA2, RT5514_POW_VREF2_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF1", RT5514_PWR_ANA2, RT5514_POW_VREF1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Power", SND_SOC_NOPM, 0, 0, NULL, 0), + + + SND_SOC_DAPM_SUPPLY("LDO16L", RT5514_PWR_ANA2, RT5514_POWL_LDO16_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1L", RT5514_PWR_ANA2, RT5514_POW_ADC1_L_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("BSTL2", RT5514_PWR_ANA2, RT5514_POW2_BSTL_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("BSTL", RT5514_PWR_ANA2, RT5514_POW_BSTL_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCFEDL", RT5514_PWR_ANA2, RT5514_POW_ADCFEDL_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCL Power", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("LDO16R", RT5514_PWR_ANA2, RT5514_POWR_LDO16_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1R", RT5514_PWR_ANA2, RT5514_POW_ADC1_R_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("BSTR2", RT5514_PWR_ANA2, RT5514_POW2_BSTR_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("BSTR", RT5514_PWR_ANA2, RT5514_POW_BSTR_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCFEDR", RT5514_PWR_ANA2, RT5514_POW_ADCFEDR_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCR Power", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("PLL1 LDO ENABLE", RT5514_ANA_CTRL_PLL1_2, + RT5514_EN_LDO_PLL1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1 LDO", RT5514_PWR_ANA2, + RT5514_POW_PLL1_LDO_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5514_PWR_ANA2, RT5514_POW_PLL1_BIT, 0, + NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5514_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5514_sto2_dmic_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5514_CLK_CTRL1, + RT5514_CLK_AD0_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("adc stereo2 filter", RT5514_CLK_CTRL1, + RT5514_CLK_AD1_EN_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5514_sto1_adc_l_mix, ARRAY_SIZE(rt5514_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5514_sto1_adc_r_mix, ARRAY_SIZE(rt5514_sto1_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5514_sto2_adc_l_mix, ARRAY_SIZE(rt5514_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5514_sto2_adc_r_mix, ARRAY_SIZE(rt5514_sto2_adc_r_mix)), + + SND_SOC_DAPM_ADC("Stereo1 ADC MIXL", NULL, RT5514_DOWNFILTER0_CTRL1, + RT5514_AD_AD_MUTE_BIT, 1), + SND_SOC_DAPM_ADC("Stereo1 ADC MIXR", NULL, RT5514_DOWNFILTER0_CTRL2, + RT5514_AD_AD_MUTE_BIT, 1), + SND_SOC_DAPM_ADC("Stereo2 ADC MIXL", NULL, RT5514_DOWNFILTER1_CTRL1, + RT5514_AD_AD_MUTE_BIT, 1), + SND_SOC_DAPM_ADC("Stereo2 ADC MIXR", NULL, RT5514_DOWNFILTER1_CTRL2, + RT5514_AD_AD_MUTE_BIT, 1), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { + { "DMIC1", NULL, "DMIC1L" }, + { "DMIC1", NULL, "DMIC1R" }, + { "DMIC2", NULL, "DMIC2L" }, + { "DMIC2", NULL, "DMIC2R" }, + + { "DMIC1L", NULL, "DMIC CLK" }, + { "DMIC1R", NULL, "DMIC CLK" }, + { "DMIC2L", NULL, "DMIC CLK" }, + { "DMIC2R", NULL, "DMIC CLK" }, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + + { "Sto1 ADC MIXL", "DMIC Switch", "Stereo1 DMIC Mux" }, + { "Sto1 ADC MIXL", "ADC Switch", "AMICL" }, + { "Sto1 ADC MIXR", "DMIC Switch", "Stereo1 DMIC Mux" }, + { "Sto1 ADC MIXR", "ADC Switch", "AMICR" }, + + { "ADC Power", NULL, "LDO18 IN" }, + { "ADC Power", NULL, "LDO18 ADC" }, + { "ADC Power", NULL, "LDO21" }, + { "ADC Power", NULL, "BG LDO18 IN" }, + { "ADC Power", NULL, "BG LDO21" }, + { "ADC Power", NULL, "BG MBIAS" }, + { "ADC Power", NULL, "MBIAS" }, + { "ADC Power", NULL, "VREF2" }, + { "ADC Power", NULL, "VREF1" }, + + { "ADCL Power", NULL, "LDO16L" }, + { "ADCL Power", NULL, "ADC1L" }, + { "ADCL Power", NULL, "BSTL2" }, + { "ADCL Power", NULL, "BSTL" }, + { "ADCL Power", NULL, "ADCFEDL" }, + + { "ADCR Power", NULL, "LDO16R" }, + { "ADCR Power", NULL, "ADC1R" }, + { "ADCR Power", NULL, "BSTR2" }, + { "ADCR Power", NULL, "BSTR" }, + { "ADCR Power", NULL, "ADCFEDR" }, + + { "AMICL", NULL, "ADC CLK" }, + { "AMICL", NULL, "ADC Power" }, + { "AMICL", NULL, "ADCL Power" }, + { "AMICR", NULL, "ADC CLK" }, + { "AMICR", NULL, "ADC Power" }, + { "AMICR", NULL, "ADCR Power" }, + + { "PLL1 LDO", NULL, "PLL1 LDO ENABLE" }, + { "PLL1", NULL, "PLL1 LDO" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" }, + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, + { "Stereo1 ADC MIX", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5514_is_sys_clk_from_pll }, + + { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, + + { "Sto2 ADC MIXL", "DMIC Switch", "Stereo2 DMIC Mux" }, + { "Sto2 ADC MIXL", "ADC Switch", "AMICL" }, + { "Sto2 ADC MIXR", "DMIC Switch", "Stereo2 DMIC Mux" }, + { "Sto2 ADC MIXR", "ADC Switch", "AMICR" }, + + { "Stereo2 ADC MIXL", NULL, "Sto2 ADC MIXL" }, + { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, + + { "Stereo2 ADC MIX", NULL, "Stereo2 ADC MIXL" }, + { "Stereo2 ADC MIX", NULL, "Stereo2 ADC MIXR" }, + { "Stereo2 ADC MIX", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", rt5514_is_sys_clk_from_pll }, + + { "AIF1TX", NULL, "Stereo1 ADC MIX"}, + { "AIF1TX", NULL, "Stereo2 ADC MIX"}, +}; + +static int rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + int pre_div, bclk_ms, frame_size; + unsigned int val_len = 0; + + rt5514->lrck = params_rate(params); + pre_div = rl6231_get_clk_info(rt5514->sysclk, rt5514->lrck); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + + bclk_ms = frame_size > 32; + rt5514->bclk = rt5514->lrck * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5514->bclk, rt5514->lrck); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len = RT5514_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len = RT5514_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len = RT5514_I2S_DL_8; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_I2S_DL_MASK, + val_len); + regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL2, + RT5514_CLK_SYS_DIV_OUT_MASK | RT5514_SEL_ADC_OSR_MASK, + pre_div << RT5514_CLK_SYS_DIV_OUT_SFT | + pre_div << RT5514_SEL_ADC_OSR_SFT); + + return 0; +} + +static int rt5514_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_NB_IF: + reg_val |= RT5514_I2S_LR_INV; + break; + + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5514_I2S_BP_INV; + break; + + case SND_SOC_DAIFMT_IB_IF: + reg_val |= RT5514_I2S_BP_INV | RT5514_I2S_LR_INV; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5514_I2S_DF_LEFT; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5514_I2S_DF_PCM_A; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5514_I2S_DF_PCM_B; + break; + + default: + return -EINVAL; + } + + regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, + RT5514_I2S_DF_MASK | RT5514_I2S_BP_MASK | RT5514_I2S_LR_MASK, + reg_val); + + return 0; +} + +static int rt5514_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5514->sysclk && clk_id == rt5514->sysclk_src) + return 0; + + switch (clk_id) { + case RT5514_SCLK_S_MCLK: + reg_val |= RT5514_CLK_SYS_PRE_SEL_MCLK; + break; + + case RT5514_SCLK_S_PLL1: + reg_val |= RT5514_CLK_SYS_PRE_SEL_PLL; + break; + + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL2, + RT5514_CLK_SYS_PRE_SEL_MASK, reg_val); + + rt5514->sysclk = freq; + rt5514->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5514_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5514->pll_in = 0; + rt5514->pll_out = 0; + regmap_update_bits(rt5514->regmap, RT5514_CLK_CTRL2, + RT5514_CLK_SYS_PRE_SEL_MASK, + RT5514_CLK_SYS_PRE_SEL_MCLK); + + return 0; + } + + if (source == rt5514->pll_src && freq_in == rt5514->pll_in && + freq_out == rt5514->pll_out) + return 0; + + switch (source) { + case RT5514_PLL1_S_MCLK: + regmap_update_bits(rt5514->regmap, RT5514_PLL_SOURCE_CTRL, + RT5514_PLL_1_SEL_MASK, RT5514_PLL_1_SEL_MCLK); + break; + + case RT5514_PLL1_S_BCLK: + regmap_update_bits(rt5514->regmap, RT5514_PLL_SOURCE_CTRL, + RT5514_PLL_1_SEL_MASK, RT5514_PLL_1_SEL_SCLK); + break; + + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + regmap_write(rt5514->regmap, RT5514_ANA_CTRL_PLL1_1, + pll_code.k_code << RT5514_PLL_K_SFT | + pll_code.n_code << RT5514_PLL_N_SFT | + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5514_PLL_M_SFT); + regmap_update_bits(rt5514->regmap, RT5514_ANA_CTRL_PLL1_2, + RT5514_PLL_M_BP, pll_code.m_bp << RT5514_PLL_M_BP_SFT); + + rt5514->pll_in = freq_in; + rt5514->pll_out = freq_out; + rt5514->pll_src = source; + + return 0; +} + +static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= RT5514_TDM_MODE; + + if (slots == 4) + val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; + + + switch (slot_width) { + case 20: + val |= RT5514_CH_LEN_RX_20 | RT5514_CH_LEN_TX_20; + break; + + case 24: + val |= RT5514_CH_LEN_RX_24 | RT5514_CH_LEN_TX_24; + break; + + case 32: + val |= RT5514_CH_LEN_RX_32 | RT5514_CH_LEN_TX_32; + break; + + case 16: + default: + break; + } + + regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_TDM_MODE | + RT5514_TDMSLOT_SEL_RX_MASK | RT5514_TDMSLOT_SEL_TX_MASK | + RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK, val); + + return 0; +} + +static int rt5514_probe(struct snd_soc_codec *codec) +{ + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + rt5514->codec = codec; + + return 0; +} + +static int rt5514_i2c_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5514_priv *rt5514 = i2c_get_clientdata(client); + + regmap_read(rt5514->i2c_regmap, reg | RT5514_DSP_MAPPING, val); + + return 0; +} + +static int rt5514_i2c_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5514_priv *rt5514 = i2c_get_clientdata(client); + + regmap_write(rt5514->i2c_regmap, reg | RT5514_DSP_MAPPING, val); + + return 0; +} + +#define RT5514_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5514_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5514_aif_dai_ops = { + .hw_params = rt5514_hw_params, + .set_fmt = rt5514_set_dai_fmt, + .set_sysclk = rt5514_set_dai_sysclk, + .set_pll = rt5514_set_dai_pll, + .set_tdm_slot = rt5514_set_tdm_slot, +}; + +struct snd_soc_dai_driver rt5514_dai[] = { + { + .name = "rt5514-aif1", + .id = 0, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = RT5514_STEREO_RATES, + .formats = RT5514_FORMATS, + }, + .ops = &rt5514_aif_dai_ops, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5514 = { + .probe = rt5514_probe, + .idle_bias_off = true, + .controls = rt5514_snd_controls, + .num_controls = ARRAY_SIZE(rt5514_snd_controls), + .dapm_widgets = rt5514_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5514_dapm_widgets), + .dapm_routes = rt5514_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5514_dapm_routes), +}; + +static const struct regmap_config rt5514_i2c_regmap = { + .name = "i2c", + .reg_bits = 32, + .val_bits = 32, + + .max_register = RT5514_DSP_MAPPING | RT5514_VENDOR_ID2, + .readable_reg = rt5514_i2c_readable_register, + + .cache_type = REGCACHE_NONE, +}; + +static const struct regmap_config rt5514_regmap = { + .reg_bits = 16, + .val_bits = 32, + + .max_register = RT5514_VENDOR_ID2, + .volatile_reg = rt5514_volatile_register, + .readable_reg = rt5514_readable_register, + .reg_read = rt5514_i2c_read, + .reg_write = rt5514_i2c_write, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5514_reg, + .num_reg_defaults = ARRAY_SIZE(rt5514_reg), + .use_single_rw = true, +}; + +static const struct i2c_device_id rt5514_i2c_id[] = { + { "rt5514", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5514_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt5514_of_match[] = { + { .compatible = "realtek,rt5514", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5514_of_match); +#endif + +static int rt5514_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5514_priv *rt5514; + int ret; + unsigned int val; + + rt5514 = devm_kzalloc(&i2c->dev, sizeof(struct rt5514_priv), + GFP_KERNEL); + if (rt5514 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5514); + + rt5514->i2c_regmap = devm_regmap_init_i2c(i2c, &rt5514_i2c_regmap); + if (IS_ERR(rt5514->i2c_regmap)) { + ret = PTR_ERR(rt5514->i2c_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt5514->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5514_regmap); + if (IS_ERR(rt5514->regmap)) { + ret = PTR_ERR(rt5514->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5514->regmap, RT5514_VENDOR_ID2, &val); + if (val != RT5514_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5514\n", val); + return -ENODEV; + } + + ret = regmap_register_patch(rt5514->i2c_regmap, rt5514_i2c_patch, + ARRAY_SIZE(rt5514_i2c_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply i2c_regmap patch: %d\n", + ret); + + ret = regmap_register_patch(rt5514->regmap, rt5514_patch, + ARRAY_SIZE(rt5514_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5514, + rt5514_dai, ARRAY_SIZE(rt5514_dai)); +} + +static int rt5514_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt5514_i2c_driver = { + .driver = { + .name = "rt5514", + .of_match_table = of_match_ptr(rt5514_of_match), + }, + .probe = rt5514_i2c_probe, + .remove = rt5514_i2c_remove, + .id_table = rt5514_i2c_id, +}; +module_i2c_driver(rt5514_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5514 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h new file mode 100644 index 0000000..6ad8a61 --- /dev/null +++ b/sound/soc/codecs/rt5514.h @@ -0,0 +1,252 @@ +/* + * rt5514.h -- RT5514 ALSA SoC audio driver + * + * Copyright 2015 Realtek Microelectronics + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5514_H__ +#define __RT5514_H__ + +#define RT5514_DEVICE_ID 0x10ec5514 + +#define RT5514_RESET 0x2000 +#define RT5514_PWR_ANA1 0x2004 +#define RT5514_PWR_ANA2 0x2008 +#define RT5514_I2S_CTRL1 0x2010 +#define RT5514_I2S_CTRL2 0x2014 +#define RT5514_VAD_CTRL6 0x2030 +#define RT5514_EXT_VAD_CTRL 0x206c +#define RT5514_DIG_IO_CTRL 0x2070 +#define RT5514_PAD_CTRL1 0x2080 +#define RT5514_DMIC_DATA_CTRL 0x20a0 +#define RT5514_DIG_SOURCE_CTRL 0x20a4 +#define RT5514_SRC_CTRL 0x20ac +#define RT5514_DOWNFILTER2_CTRL1 0x20d0 +#define RT5514_PLL_SOURCE_CTRL 0x2100 +#define RT5514_CLK_CTRL1 0x2104 +#define RT5514_CLK_CTRL2 0x2108 +#define RT5514_PLL3_CALIB_CTRL1 0x2110 +#define RT5514_PLL3_CALIB_CTRL5 0x2124 +#define RT5514_DELAY_BUF_CTRL1 0x2140 +#define RT5514_DELAY_BUF_CTRL3 0x2148 +#define RT5514_DOWNFILTER0_CTRL1 0x2190 +#define RT5514_DOWNFILTER0_CTRL2 0x2194 +#define RT5514_DOWNFILTER0_CTRL3 0x2198 +#define RT5514_DOWNFILTER1_CTRL1 0x21a0 +#define RT5514_DOWNFILTER1_CTRL2 0x21a4 +#define RT5514_DOWNFILTER1_CTRL3 0x21a8 +#define RT5514_ANA_CTRL_LDO10 0x2200 +#define RT5514_ANA_CTRL_LDO18_16 0x2204 +#define RT5514_ANA_CTRL_ADC12 0x2210 +#define RT5514_ANA_CTRL_ADC21 0x2214 +#define RT5514_ANA_CTRL_ADC22 0x2218 +#define RT5514_ANA_CTRL_ADC23 0x221c +#define RT5514_ANA_CTRL_MICBST 0x2220 +#define RT5514_ANA_CTRL_ADCFED 0x2224 +#define RT5514_ANA_CTRL_INBUF 0x2228 +#define RT5514_ANA_CTRL_VREF 0x222c +#define RT5514_ANA_CTRL_PLL3 0x2240 +#define RT5514_ANA_CTRL_PLL1_1 0x2260 +#define RT5514_ANA_CTRL_PLL1_2 0x2264 +#define RT5514_DMIC_LP_CTRL 0x2e00 +#define RT5514_MISC_CTRL_DSP 0x2e04 +#define RT5514_DSP_CTRL1 0x2f00 +#define RT5514_DSP_CTRL3 0x2f08 +#define RT5514_DSP_CTRL4 0x2f10 +#define RT5514_VENDOR_ID1 0x2ff0 +#define RT5514_VENDOR_ID2 0x2ff4 + +#define RT5514_DSP_MAPPING 0x18000000 + +/* RT5514_PWR_ANA1 (0x2004) */ +#define RT5514_POW_LDO18_IN (0x1 << 5) +#define RT5514_POW_LDO18_IN_BIT 5 +#define RT5514_POW_LDO18_ADC (0x1 << 4) +#define RT5514_POW_LDO18_ADC_BIT 4 +#define RT5514_POW_LDO21 (0x1 << 3) +#define RT5514_POW_LDO21_BIT 3 +#define RT5514_POW_BG_LDO18_IN (0x1 << 2) +#define RT5514_POW_BG_LDO18_IN_BIT 2 +#define RT5514_POW_BG_LDO21 (0x1 << 1) +#define RT5514_POW_BG_LDO21_BIT 1 + +/* RT5514_PWR_ANA2 (0x2008) */ +#define RT5514_POW_PLL1 (0x1 << 18) +#define RT5514_POW_PLL1_BIT 18 +#define RT5514_POW_PLL1_LDO (0x1 << 16) +#define RT5514_POW_PLL1_LDO_BIT 16 +#define RT5514_POW_BG_MBIAS (0x1 << 15) +#define RT5514_POW_BG_MBIAS_BIT 15 +#define RT5514_POW_MBIAS (0x1 << 14) +#define RT5514_POW_MBIAS_BIT 14 +#define RT5514_POW_VREF2 (0x1 << 13) +#define RT5514_POW_VREF2_BIT 13 +#define RT5514_POW_VREF1 (0x1 << 12) +#define RT5514_POW_VREF1_BIT 12 +#define RT5514_POWR_LDO16 (0x1 << 11) +#define RT5514_POWR_LDO16_BIT 11 +#define RT5514_POWL_LDO16 (0x1 << 10) +#define RT5514_POWL_LDO16_BIT 10 +#define RT5514_POW_ADC2 (0x1 << 9) +#define RT5514_POW_ADC2_BIT 9 +#define RT5514_POW_INPUT_BUF (0x1 << 8) +#define RT5514_POW_INPUT_BUF_BIT 8 +#define RT5514_POW_ADC1_R (0x1 << 7) +#define RT5514_POW_ADC1_R_BIT 7 +#define RT5514_POW_ADC1_L (0x1 << 6) +#define RT5514_POW_ADC1_L_BIT 6 +#define RT5514_POW2_BSTR (0x1 << 5) +#define RT5514_POW2_BSTR_BIT 5 +#define RT5514_POW2_BSTL (0x1 << 4) +#define RT5514_POW2_BSTL_BIT 4 +#define RT5514_POW_BSTR (0x1 << 3) +#define RT5514_POW_BSTR_BIT 3 +#define RT5514_POW_BSTL (0x1 << 2) +#define RT5514_POW_BSTL_BIT 2 +#define RT5514_POW_ADCFEDR (0x1 << 1) +#define RT5514_POW_ADCFEDR_BIT 1 +#define RT5514_POW_ADCFEDL (0x1 << 0) +#define RT5514_POW_ADCFEDL_BIT 0 + +/* RT5514_I2S_CTRL1 (0x2010) */ +#define RT5514_TDM_MODE (0x1 << 28) +#define RT5514_TDM_MODE_SFT 28 +#define RT5514_I2S_LR_MASK (0x1 << 26) +#define RT5514_I2S_LR_SFT 26 +#define RT5514_I2S_LR_NOR (0x0 << 26) +#define RT5514_I2S_LR_INV (0x1 << 26) +#define RT5514_I2S_BP_MASK (0x1 << 25) +#define RT5514_I2S_BP_SFT 25 +#define RT5514_I2S_BP_NOR (0x0 << 25) +#define RT5514_I2S_BP_INV (0x1 << 25) +#define RT5514_I2S_DF_MASK (0x7 << 16) +#define RT5514_I2S_DF_SFT 16 +#define RT5514_I2S_DF_I2S (0x0 << 16) +#define RT5514_I2S_DF_LEFT (0x1 << 16) +#define RT5514_I2S_DF_PCM_A (0x2 << 16) +#define RT5514_I2S_DF_PCM_B (0x3 << 16) +#define RT5514_TDMSLOT_SEL_RX_MASK (0x3 << 10) +#define RT5514_TDMSLOT_SEL_RX_SFT 10 +#define RT5514_TDMSLOT_SEL_RX_4CH (0x1 << 10) +#define RT5514_CH_LEN_RX_MASK (0x3 << 8) +#define RT5514_CH_LEN_RX_SFT 8 +#define RT5514_CH_LEN_RX_16 (0x0 << 8) +#define RT5514_CH_LEN_RX_20 (0x1 << 8) +#define RT5514_CH_LEN_RX_24 (0x2 << 8) +#define RT5514_CH_LEN_RX_32 (0x3 << 8) +#define RT5514_TDMSLOT_SEL_TX_MASK (0x3 << 6) +#define RT5514_TDMSLOT_SEL_TX_SFT 6 +#define RT5514_TDMSLOT_SEL_TX_4CH (0x1 << 6) +#define RT5514_CH_LEN_TX_MASK (0x3 << 4) +#define RT5514_CH_LEN_TX_SFT 4 +#define RT5514_CH_LEN_TX_16 (0x0 << 4) +#define RT5514_CH_LEN_TX_20 (0x1 << 4) +#define RT5514_CH_LEN_TX_24 (0x2 << 4) +#define RT5514_CH_LEN_TX_32 (0x3 << 4) +#define RT5514_I2S_DL_MASK (0x3 << 0) +#define RT5514_I2S_DL_SFT 0 +#define RT5514_I2S_DL_16 (0x0 << 0) +#define RT5514_I2S_DL_20 (0x1 << 0) +#define RT5514_I2S_DL_24 (0x2 << 0) +#define RT5514_I2S_DL_8 (0x3 << 0) + +/* RT5514_DIG_SOURCE_CTRL (0x20a4) */ +#define RT5514_AD1_DMIC_INPUT_SEL (0x1 << 1) +#define RT5514_AD1_DMIC_INPUT_SEL_SFT 1 +#define RT5514_AD0_DMIC_INPUT_SEL (0x1 << 0) +#define RT5514_AD0_DMIC_INPUT_SEL_SFT 0 + +/* RT5514_PLL_SOURCE_CTRL (0x2100) */ +#define RT5514_PLL_1_SEL_MASK (0x7 << 12) +#define RT5514_PLL_1_SEL_SFT 12 +#define RT5514_PLL_1_SEL_SCLK (0x3 << 12) +#define RT5514_PLL_1_SEL_MCLK (0x4 << 12) + +/* RT5514_CLK_CTRL1 (0x2104) */ +#define RT5514_CLK_AD_ANA1_EN (0x1 << 31) +#define RT5514_CLK_AD_ANA1_EN_BIT 31 +#define RT5514_CLK_AD1_EN (0x1 << 24) +#define RT5514_CLK_AD1_EN_BIT 24 +#define RT5514_CLK_AD0_EN (0x1 << 23) +#define RT5514_CLK_AD0_EN_BIT 23 +#define RT5514_CLK_DMIC_OUT_SEL_MASK (0x7 << 8) +#define RT5514_CLK_DMIC_OUT_SEL_SFT 8 + +/* RT5514_CLK_CTRL2 (0x2108) */ +#define RT5514_CLK_SYS_DIV_OUT_MASK (0x7 << 8) +#define RT5514_CLK_SYS_DIV_OUT_SFT 8 +#define RT5514_SEL_ADC_OSR_MASK (0x7 << 4) +#define RT5514_SEL_ADC_OSR_SFT 4 +#define RT5514_CLK_SYS_PRE_SEL_MASK (0x3 << 0) +#define RT5514_CLK_SYS_PRE_SEL_SFT 0 +#define RT5514_CLK_SYS_PRE_SEL_MCLK (0x2 << 0) +#define RT5514_CLK_SYS_PRE_SEL_PLL (0x3 << 0) + +/* RT5514_DOWNFILTER_CTRL (0x2190 0x2194 0x21a0 0x21a4) */ +#define RT5514_AD_DMIC_MIX (0x1 << 11) +#define RT5514_AD_DMIC_MIX_BIT 11 +#define RT5514_AD_AD_MIX (0x1 << 10) +#define RT5514_AD_AD_MIX_BIT 10 +#define RT5514_AD_AD_MUTE (0x1 << 7) +#define RT5514_AD_AD_MUTE_BIT 7 +#define RT5514_AD_GAIN_MASK (0x7f << 0) +#define RT5514_AD_GAIN_SFT 0 + +/* RT5514_ANA_CTRL_MICBST (0x2220) */ +#define RT5514_SEL_BSTL_MASK (0xf << 4) +#define RT5514_SEL_BSTL_SFT 4 +#define RT5514_SEL_BSTR_MASK (0xf << 0) +#define RT5514_SEL_BSTR_SFT 0 + +/* RT5514_ANA_CTRL_PLL1_1 (0x2260) */ +#define RT5514_PLL_K_MAX 0x1f +#define RT5514_PLL_K_MASK (RT5514_PLL_K_MAX << 16) +#define RT5514_PLL_K_SFT 16 +#define RT5514_PLL_N_MAX 0x1ff +#define RT5514_PLL_N_MASK (RT5514_PLL_N_MAX << 7) +#define RT5514_PLL_N_SFT 4 +#define RT5514_PLL_M_MAX 0xf +#define RT5514_PLL_M_MASK (RT5514_PLL_M_MAX << 0) +#define RT5514_PLL_M_SFT 0 + +/* RT5514_ANA_CTRL_PLL1_2 (0x2264) */ +#define RT5514_PLL_M_BP (0x1 << 2) +#define RT5514_PLL_M_BP_SFT 2 +#define RT5514_PLL_K_BP (0x1 << 1) +#define RT5514_PLL_K_BP_SFT 1 +#define RT5514_EN_LDO_PLL1 (0x1 << 0) +#define RT5514_EN_LDO_PLL1_BIT 0 + +#define RT5514_PLL_INP_MAX 40000000 +#define RT5514_PLL_INP_MIN 256000 + +/* System Clock Source */ +enum { + RT5514_SCLK_S_MCLK, + RT5514_SCLK_S_PLL1, +}; + +/* PLL1 Source */ +enum { + RT5514_PLL1_S_MCLK, + RT5514_PLL1_S_BCLK, +}; + +struct rt5514_priv { + struct snd_soc_codec *codec; + struct regmap *i2c_regmap, *regmap; + int sysclk; + int sysclk_src; + int lrck; + int bclk; + int pll_src; + int pll_in; + int pll_out; +}; + +#endif /* __RT5514_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] Can't use adau1361 as i2s master due to bad write ordering
by Andreas Irestål 03 Feb '16

03 Feb '16

I've run into problems using the adau1361 codec (adau1761) as i2s master. I'm using i2c to set up the codec, and according to the datasheet there is an enable bit, which must be set in order to be able to access any other register than the first two registers. This enable bit is set when the codec enters bias level standby, however, before entering this bias level, some other registers are written which will have no effect due to the inner workings of this codec. Oddly enough, it does ack all writes when measuring the i2c bus. One of these registers is the serial port 0 register, which configures the i2s bus. This results in the situation that the CODEC dai thinks it is running as i2s slave and the CPU dai thinks too as well, and the bus is dead. Since the other register fields does not change with our dai configuration, the register never gets physically updated by an i2c write. If I reset/reboot the system, which in our case does not power cycle the codec, the i2s bus is correctly set up and everything works as one would expect. So, either the driver behaves incorrectly relying on bias level being set before any other registers are written, or the underlying subsystem are behaving incorrectly. We're using the simple-card board driver as it is sufficient for our needs. /Andreas

2 1

[alsa-devel] sound: another WARNING in rawmidi_transmit_ack
by Dmitry Vyukov 03 Feb '16

03 Feb '16

Hello, The following program triggers a splash of WARNINGs in rawmidi_transmit_ack. Takashi, I am on commit 36f90b0a2ddd60823fe193a85e60ff1906c2a9b3 + a bunch of your recent fixes: https://gist.githubusercontent.com/dvyukov/40640128a433ad16a56a/raw/ab3a086… ------------[ cut here ]------------ WARNING: CPU: 2 PID: 6954 at sound/core/rawmidi.c:1133 rawmidi_transmit_ack+0x24a/0x3b0() Modules linked in: CPU: 2 PID: 6954 Comm: syz-executor Not tainted 4.5.0-rc2+ #306 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff8800309d78b8 ffffffff82be2c0d 0000000000000000 ffff880031238000 ffffffff8719a820 ffff8800309d78f8 ffffffff81355139 ffffffff8527e69a ffffffff8719a820 000000000000046d 0000000000000005 Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82be2c0d>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 [<ffffffff81355139>] warn_slowpath_common+0xd9/0x140 kernel/panic.c:482 [<ffffffff81355369>] warn_slowpath_null+0x29/0x30 kernel/panic.c:515 [<ffffffff8527e69a>] rawmidi_transmit_ack+0x24a/0x3b0 sound/core/rawmidi.c:1133 [<ffffffff8527e851>] snd_rawmidi_transmit_ack+0x51/0x80 sound/core/rawmidi.c:1163 [<ffffffff852d9046>] snd_virmidi_output_trigger+0x2b6/0x570 sound/core/seq/seq_virmidi.c:185 [< inline >] snd_rawmidi_output_trigger sound/core/rawmidi.c:150 [<ffffffff85285a0b>] snd_rawmidi_kernel_write1+0x4bb/0x760 sound/core/rawmidi.c:1252 [<ffffffff85287b73>] snd_rawmidi_write+0x543/0xb30 sound/core/rawmidi.c:1302 [<ffffffff817ba5f3>] __vfs_write+0x113/0x480 fs/read_write.c:528 [<ffffffff817bc087>] vfs_write+0x167/0x4a0 fs/read_write.c:577 [< inline >] SYSC_write fs/read_write.c:624 [<ffffffff817bf371>] SyS_write+0x111/0x220 fs/read_write.c:616 [<ffffffff86660276>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 ---[ end trace 6f686d9dad133d99 ]--- // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> long r[28]; void* thr(void* arg) { switch ((long)arg) { case 0: r[0] = syscall(SYS_mmap, 0x20000000ul, 0xd90000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 1: memcpy( (void*)0x20d8f000, "\x2f\x64\x65\x76\x2f\x73\x65\x71\x75\x65\x6e\x63\x65\x72\x32", 15); r[2] = syscall(SYS_open, "/dev/sequencer2", 0x181800ul, 0, 0, 0); break; case 2: memcpy((void*)0x20d8df5b, "\x2f\x64\x65\x76\x2f\x73\x6e\x64\x2f\x6d" "\x69\x64\x69\x43\x23\x44\x23", 17); r[4] = syscall(SYS_open, "/dev/snd/midiC2D0", 0x802ul, 0, 0, 0); break; case 3: r[5] = syscall(SYS_mmap, 0x20d90000ul, 0x1000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 4: r[6] = syscall(SYS_mmap, 0x20d90000ul, 0x1000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 5: r[7] = syscall(SYS_mmap, 0x20000000ul, 0x0ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 6: r[8] = syscall(SYS_mmap, 0x20d91000ul, 0x1000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 7: *(uint32_t*)0x20d91000 = (uint32_t)0xf834; r[10] = syscall(SYS_ioctl, r[2], 0x40045201ul, 0x20d91000ul, 0, 0, 0); break; case 8: r[11] = syscall(SYS_mmap, 0x20d91000ul, 0x1000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 9: r[12] = syscall(SYS_ioctl, r[2], 0x80404509ul, 0x20d91000ul, 0, 0, 0); break; case 10: r[13] = syscall(SYS_mmap, 0x20d90000ul, 0x1000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 11: *(uint64_t*)0x20d90ffd = (uint64_t)0x20a962b0; *(uint64_t*)0x20d91005 = (uint64_t)0xd3; *(uint64_t*)0x20d9100d = (uint64_t)0x20d902e4; *(uint64_t*)0x20d91015 = (uint64_t)0xf6; *(uint64_t*)0x20d9101d = (uint64_t)0x20d907cd; *(uint64_t*)0x20d91025 = (uint64_t)0x67; *(uint64_t*)0x20d9102d = (uint64_t)0x20000000; *(uint64_t*)0x20d91035 = (uint64_t)0x0; memcpy((void*)0x20a962b0, "\xea\x7e\xf1\x21\xe0\xc7\xaa\xa3\x6d\x6c\xb3\x94\x27\x70" "\xed\x74\xab\xee\x27\x7c\x1b\x9e\x66\x76\x79\xa6\x4f\x69" "\xf6\x63\x0a\x5c\x31\xe3\xef\x43\x43\x93\x6a\x40\x53\x03" "\x0d\x61\xd9\xb5\xac\xc0\xe9\x10\x1c\x5d\x50\x6e\x7f\xe1" "\x0a\x65\x7e\x09\xa5\x57\x89\x33\x52\x49\x01\xaf\x5f\xdd" "\x55\xb2\x59\xca\xf6\x0d\x39\x26\xa6\xad\x9f\x93\x7e\xda" "\x06\x6f\xae\x4e\xce\x3d\xd9\xbd\x4f\x6a\xec\x7b\xb7\xc8" "\xe0\xd8\x25\x85\xca\xda\x7f\x04\x7a\x0e\x23\x66\x63\xc0" "\xfb\x9a\xe7\x87\xb2\x85\x83\x6e\x07\xd5\x8c\x8e\xb1\x8a" "\x9a\x43\xb8\x5c\x2d\xe4\xe9\x9b\x8d\xfb\x23\x52\xfd\xf9" "\x7e\xa4\xc1\x8b\x90\xf9\x14\x98\xba\x75\xa5\xf2\x88\xbf" "\x8f\x28\x8f\xee\xf3\xc7\x20\xfc\xa3\x53\xd2\x1c\x1b\x02" "\xc5\x2b\x1b\x9a\x17\xb6\xef\xd5\x6b\x46\x4c\x66\x75\x45" "\xcc\xb4\x1d\x41\x13\xb1\x1f\xc7\x76\x7f\x28\x8b\x3d\x85" "\x16\x38\x2f\x27\xdb\x17\x71\x05\xd6\x0e\x5e\x53\x3b\x19" "\xbb", 211); memcpy((void*)0x20d902e4, "\xe8\x61\x5b\xd6\x03\xc7\x69\x3a\xa4\x17\x85\xce\xf4\x49" "\x07\x42\xee\x24\x4a\x9e\xe5\xff\x3e\xa1\xbc\x97\xc7\x66" "\x01\x9f\x34\x57\x1b\xf6\x99\x2b\xd0\x45\xa0\xc9\x2b\x2d" "\x45\x45\x7a\xb0\xf9\x9d\xc8\x3b\x53\x78\x3a\x93\x42\xcc" "\x88\xa9\xf4\x79\xab\x1c\xfe\x48\x8d\x61\x6a\x84\x26\x2d" "\x74\xb6\x2d\xa7\xbb\x8c\x33\x6e\xca\x38\x57\xf8\x39\x91" "\xc7\x57\x6f\x28\xbc\x2e\x9e\xca\xc8\xcb\x08\x0c\x1e\xe6" "\x49\x5d\x32\xe6\x33\xd0\x92\x20\xb5\x7f\x96\x2a\x47\x3f" "\xa5\x9d\x9b\xeb\x67\xd9\x36\x48\xeb\x0d\x16\xd4\xc9\x48" "\x7d\xa4\xfd\x42\xde\x2d\xf6\xae\x48\xe6\x50\x1c\x24\x86" "\xe5\x0d\x23\x04\xa6\xcc\xfb\x98\xbe\x61\xb2\xed\x59\x28" "\x82\xd1\x80\x06\xf1\x90\xae\xd4\x99\x28\x92\x92\x30\xdc" "\x12\x29\x5e\x47\x69\x74\x7b\x1b\x82\x14\xce\xaa\x35\x06" "\xa3\x24\x00\x34\xf8\xeb\x3f\x24\xf7\xdf\x30\x59\xb5\x54" "\x1a\x60\xfa\x23\x1b\x67\xf8\x3c\x59\x56\x0b\xef\xaf\x09" "\x7d\xaa\xe5\x4b\xce\x0b\x6a\x3b\xde\x94\xa1\x80\x37\xae" "\x5c\x8c\xac\x0f\x1a\xfd\x36\x9a\xa8\x8a\xd2\x2d\xe5\xf1" "\xb2\x8e\x29\x30\xe1\x39\x72\xef", 246); memcpy((void*)0x20d907cd, "\x3e\x14\x32\x60\x81\x87\xfb\x49\xb5\x66\xa3\x99\xe5\x9b" "\xfd\xae\xec\xa8\x66\x6b\x6b\x52\xa4\x13\xbd\x8a\x2a\xe7" "\x26\x6a\x48\x8d\x09\x1f\x6e\x60\x25\x75\xd0\x62\xfa\x32" "\xc0\x40\x16\x21\xc0\x41\xa9\xd3\x30\x77\x64\x5d\xf4\x50" "\xfd\x63\x06\xbf\xc2\xad\xfa\xf6\xa5\x2a\x51\x60\x5b\x47" "\x60\x4d\xbe\x25\x16\x2e\xd7\xdc\x43\xe3\xc9\x57\x4a\x00" "\x9f\x4f\x23\x40\xb0\x9b\xc7\x25\x5f\xef\xb1\x97\x9e\xb3" "\x09\xc8\x99\x34\xf2", 103); r[25] = syscall(SYS_writev, r[4], 0x20d90ffdul, 0x4ul, 0, 0, 0); break; case 12: memcpy( (void*)0x20d8a029, "\x41\x30\x85\x16\x79\x50\x97\xdd\x70\xc7\x66\xcd\x0b\xa5\x9e" "\x92\xe8\xba\x6c\x3f\x30\x00\x24\x10\x48\xa9\x47\xe4\x96\x91" "\x7d\x4b\x68\x92\xe9\x81\x54\xd5\xe8\x72\xdf\x42\x12\xa3\xfd" "\xe7\x30\xa0\xd7\x9f\xd0\x88\xeb\x6b\x42\x43\x78\x80\xfb\x51" "\x39\xcd\xab\x77\x9a\xb4\x9a\x79\xa4\x0b\xa0\x3a\x9e\x09\x0d" "\x1d\xda\x32\x68\x64\xf9\x03\x3f\xed\x07\xd2\xd3\xea\x82\xb8" "\x8e\x47\xf9\x7a\x78\x73\x6f\x46\x9a\x6e\xb3\xa8\xb7\x9f\xac" "\x1c\x47\x66\x0a\x67\x59\xb4\x6b\x52\xab\x3b\xe6\x6d\x74\xe2" "\xf0\x84\xfe\x1b\x3a\xe8\x82\x2b\x2a\xad\x3e\x61\x69\xdb\xe0" "\xff\x65\xa9\xab\xf9\xa1\x69\x30\xcd\xc2\xbb\x2e\xac\x11\xf6" "\x3c\x31\x25\x2e\x77\x12\x86\xdf\x24\x37\xf8\xdc\x9a\xc3\x11" "\xd4\x25\xce\x12\xbe\x2f\xb0\x4a\x50\x64\x8a\xf1\xb3\xf0\x75" "\xae\x5a\x0b\x5d\xf3\x8d\xe8\x75\x07\xaa\x0a\x93\x6c\x85\x2c" "\x48\x3e\xae\xbc\xe1\x49\xab\xa9\x7b\xe3\x57\x7e\xa8\x90\x1a" "\x04\x5f\x10\x9a\x96\x5b\xf4\xd0\xa1\xec\xed\x0f\xd8\x0a\x40" "\xd8\x75\x05\x0d\x14\xc6\x28\xbb\x7a\xc5\x42\x69\xb0\xea\x26" "\x0d\x2f\xb6\xba\xcc\x23\x28\x28\x61\x6b\x55\xf8\x10\xb8\xb8" "\x2d\xba\x9f\xfd\x2e\x1d\xa7\xf1\x3f\x8e\x54\x64\xf5\x6f\xfe" "\x33\x93\x1d\xcb\x82\x60\x8f\x90\x85\xf7\x58\xc2\x3b\x9d\x9a" "\x7f\xe0\x55\x2e\xa2\x1a\xd5\x39\xe6\xfd\xcf\x2c\x2c\x5b\x7e" "\xfb\xfd\x81\x17\x92\x95\x97\x40\x74\x9d\xfb\x81\xc2\x02\x85" "\x0c\x55\xed\xbf\x1f\xce\xad\x58\x88\xd2\x10\xf0\x4e\x18\x28" "\x86\xe0\x1c\xa7\xce\x7c\xdb\xdd\x15\x3c\xd2\xd8\x1d\x9e\x43" "\x18\x0e\x1d\xc2\xb8\x8e\xf0\x0f\xd6\x6b\x89\x40\x00\x28\x1d" "\x5a\xa2\x5a\xeb\x2f\x60\x7b\x02\xc9\x86\x88\xfb\x94\x89\x81" "\xec\x4c\x9f\xe8\xaf\xcb\x7b\x68\x21\xe1\xa3\x60\x32\x7d\x23" "\xcf\x71\x20\x22\x1b\x8e\xec\x51\x17\x05\x70\x2c\x92\x70\x59" "\x35\x6a\x25\x71\x39\x2c\x66\x42\x3f\x46\x78\x88\x7f\xdb\x71" "\xb8\xd6\x7c\xce\x8b\x4a\xb1\x8d\xa2\xc5\xb9\xac\xe7\x89\x4b" "\x7a\x6c\x91\x11\x6d\xe9\x2b\x27\x80\x49\x32\x66\x16\xed\xf1" "\x74\x51\xcd\x44\x4a\x94\x5c\xfc\xe5\x11\x54\xf5\x61\xbe\xed" "\xbc\x00\xbd\xde\x02\xd4\x40\xa5\x13\x4f\x48\x6e\x43\x1a\xc2" "\x56\xe8\x95\x16\xcc\x5f\x4d\xf8\x80\x40\x2c\x59\x07\x32\x49" "\x84\xff\x31\x53\xd9\x26\x08\xc1\x06\xf2\x69\x4b\x35\xa4\x41" "\xe5\x00\xdd\x64\x23\x1a\xb7\x29\xde\xef\x90\x03\xe3\xe1\x7d" "\x61\x74\xfd\xe5\xd1\xa6\xa8\x9b\xc5\x07\x90\xfb\x75\x5e\x64" "\xa3\x22\xcc\xd2\xc9\x29\xdb\x71\x84\x5c\x67\x51\x75\xc6\xdc" "\x22\x78\xb8\x6d\x2e\x8f\xf5\x51\x87\x53\xd5\xc5\x10\x53\xc3" "\x0c\x96\xf0\x6c\xca\x61\x37\xfc\x63\xff\xe0\xd4\x77\x3d\xea" "\x64\xe0\x43\xfb\xca\xf6\xd0\x2d\xd2\xc4\x67\x35\x58\x70\xa5" "\x6f\x0c\x19\x93\x45\xdd\xf4\x0f\xad\x60\xf4\x8e\x8d\x6a\xb6" "\x8d\x66\xaa\xe6\x19\xc3\x60\x4a\x77\xaa\x7b\x15\x9b\x89\x65" "\x11\x70\xc6\xa1\x86\xf1\x7c\xac\xac\xe4\x94\xb3\x08\xdd\xd0" "\x7c\xfb\x60\x2f\x5f\x42\xe6\x97\xfc\x1c\x40\xc1\xb1\x2c\x4b" "\x9d\xc0\x66\x48\xf1\x1b\x74\x54\x5b\x97\xc8\xb1\xba\x0b\xa5" "\x36\x7b\xf9\x70\x09\x2e\x7e\x57\x07\x98\xaa\xa4\x40\x7b\xc7" "\x6c\xe3\x43\x8b\xc0\xff\x87\x1a\x91\xaa\x07\x1a\x39\x29\xf2" "\x81\xf8\xfe\x22\x77\x31\xbe\x89\x30\x91\x31\xe8\xa2\xe6\xd1" "\x68\x99\x2d\x5a\xbf\x42\x26\x3b\x19\x5a\x18\x20\xf9\xd6\x21" "\x8b\x6c\x2b\x1d\x6a\x99\xbe\x02\x39\xe9\x50\x0c\xf1\xb2\x9f" "\x10\xdd\x7d\x85\x95\xa7\x0d\x08\x90\x73\x26\xf8\x52\xb3\x39" "\x2c\x6a\x42\x60\x99\xf5\x65\xae\x4b\xd6\xd3\xc5\x21\x19\x96" "\x85\x08\xcb\xd6\xb3\xce\xda\xd1\x75\xc3\x86\xd2\x7c\x79\x78" "\xea\x8b\x88\xf4\xe1\x82\x12\x1a\xe4\x57\x57\xa6\x51\x3c\xa4" "\xf0\x42\xbb\x21\xca\x5d\xec\xcd\x89\xec\x0c\x1b\xb8\x5b\x21" "\x17\xb3\xca\x32\x50\xc3\xad\x06\x05\xfa\x5e\x9c\x19\xfe\x81" "\x8a\x18\x06\x76\x46\xda\x93\xd7\x90\xee\xb7\x30\x23\x09\xfd" "\xd9\xa8\x40\x5c\xf0\xfc\xb3\xf7\xb1\x6f\x2b\x01\xe1\x0f\x27" "\x98\xff\xf9\xd4\x01\x89\x91\x79\x71\x11\xdb\x30\xc0\x83\x1e" "\x12\x62\x0e\x50\xa1\x47\x80\x99\xa4\xb6\x78\x19\xc5\x50\x4f" "\x82\x87\x35\x3e\x9c\x22\x20\x81\xef\xfe\xc3\x3e\x79\x2c\xf4" "\x7c\x7a\xcc\xfd\xbd\x63\xb4\x6b\x98\xd3\x56\x1c\xeb\xed\x1a" "\x3a\xff\x5c\x8e\x0e\x7e\x70\x78\xfc\x91\xfc\x29\x14\xa3\x8e" "\xcf\x47\x78\x0f\x98\x97\xb4\x21\x3d\x9c\x19\xbf\x27\x9b\xb4" "\x16\xe6\x50\xad\xe4\x6b\xa9\x1b\xa1\xe2\x05\xc1\x0a\x83\x1d" "\x03\xaf\x7b\x7d\x8c\x88\xa3\x11\x7e\x5d\x72\xe0\x5d\xad\x50" "\xc4\xc8\x75\xc8\xcc\xaa\x7d\x8f\xb4\x47\x5e\x34\xe8\x04\x71" "\x10\xdb\x92\xb7\xfe\x9b\x77\x9f\x04\xe3\x74\x72\x16\xb7\x4e" "\x31\xf8\x00\x98\x5c\x59\xf7\x4d\x7d\x89\xd7\x1c\xbd\xd7\x85" 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"\x02\xfb\x93\x10\xe6\xbe\xcc\xc2\xcf\x5d\x71\x77\x86\x03\x6c" "\xa3\xbd\x6a\x06\xd4\x4c\xb0\xf5\xc8\xb8\xc7\x11\x2a\xda\x99" "\xb6\xf7\x68\x39\xf3\x2c\x2f\x45\xed\xaa\x24\x56\x46\xfe\x05" "\x8b\x9f\x93\x6a\xc7\x8e\x0b\x9f\xb3\xe9\x45\xee\x59\xcd\x69" "\xf1\x5a\xe9\x14\x22\x66\xa0\x3f\x00\x27\x87\x6d\x67\x68\x79" "\x54\xb8\x7f\xd2\x63\xb4\x3f\x48\xf5\xf6\x3e\xa1\x9b\x60\x66" "\x2d\x85\xe7\x08\xd3\x37\x37\xca\x3b\xc1\xb0\xcd\x07\x3d\xec" "\x39\x0b\x79\xd3\xd2\x7d\xfe\x1b\xaa\x35\xf6\xab\x01\xf7\x64" "\x52\x42\xdd\x9c\xce\xea\x12\xd5\xb6\xfe\xda\x76\xf0\x28\x24" "\x64\x61\xb2\x8b\x52", 4100); r[27] = syscall(SYS_write, r[4], 0x20d8a029ul, 0x1004ul, 0, 0, 0); break; } return 0; } int main() { long i; pthread_t th[13]; memset(r, -1, sizeof(r)); for (i = 0; i < 13; i++) { pthread_create(&th[i], 0, thr, (void*)i); usleep(10000); } for (i = 0; i < 13; i++) { pthread_create(&th[i], 0, thr, (void*)i); if (i % 2 == 0) usleep(10000); } usleep(100000); return 0; }

2 5

[alsa-devel] sound: deadlock between snd_rawmidi_kernel_open/snd_seq_port_connect
by Dmitry Vyukov 03 Feb '16

03 Feb '16

Hello, While running syzkaller fuzzer I've got the following lockdep report: ====================================================== [ INFO: possible circular locking dependency detected ] 4.4.0+ #276 Not tainted ------------------------------------------------------- syz-executor/21025 is trying to acquire lock: (register_mutex#5){+.+.+.}, at: [<ffffffff84f889cb>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 but task is already holding lock: (&grp->list_mutex/1){+.+...}, at: [<ffffffff84fc4afa>] snd_seq_port_connect+0x1ba/0x840 sound/core/seq/seq_ports.c:506 which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #2 (&grp->list_mutex/1){+.+...}: [<ffffffff8145ad8c>] lock_acquire+0x1dc/0x430 kernel/locking/lockdep.c:3585 [<ffffffff8144851a>] down_write_nested+0x4a/0xa0 kernel/locking/rwsem.c:149 [<ffffffff84fc4afa>] snd_seq_port_connect+0x1ba/0x840 sound/core/seq/seq_ports.c:506 [<ffffffff84fb6f54>] snd_seq_ioctl_subscribe_port+0x1c4/0x290 sound/core/seq/seq_clientmgr.c:1464 [<ffffffff84fb126d>] snd_seq_do_ioctl+0x19d/0x1c0 sound/core/seq/seq_clientmgr.c:2209 [<ffffffff84fb2c2b>] snd_seq_kernel_client_ctl+0xdb/0x170 sound/core/seq/seq_clientmgr.c:2423 [<ffffffff88413c20>] snd_seq_oss_create_client+0x253/0x2d5 sound/core/seq/oss/seq_oss_init.c:119 [<ffffffff8841393e>] alsa_seq_oss_init+0x1af/0x23e sound/core/seq/oss/seq_oss.c:89 [<ffffffff81002259>] do_one_initcall+0x159/0x380 init/main.c:794 [< inline >] do_initcall_level init/main.c:859 [< inline >] do_initcalls init/main.c:867 [< inline >] do_basic_setup init/main.c:885 [<ffffffff88315c1a>] kernel_init_freeable+0x474/0x52d init/main.c:1010 [<ffffffff86312683>] kernel_init+0x13/0x150 init/main.c:936 [<ffffffff86336fef>] ret_from_fork+0x3f/0x70 arch/x86/entry/entry_64.S:468 -> #1 (&grp->list_mutex){++++.+}: [<ffffffff8145ad8c>] lock_acquire+0x1dc/0x430 kernel/locking/lockdep.c:3585 [<ffffffff863323a7>] down_read+0x47/0x60 kernel/locking/rwsem.c:22 [< inline >] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:679 [<ffffffff84fb5509>] snd_seq_deliver_event+0x5a9/0x800 sound/core/seq/seq_clientmgr.c:817 [<ffffffff84fb6466>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2401 [<ffffffff84fc1e62>] snd_seq_system_broadcast+0xb2/0xf0 sound/core/seq/seq_system.c:101 [<ffffffff84fb248e>] snd_seq_create_kernel_client+0x21e/0x300 sound/core/seq/seq_clientmgr.c:2280 [< inline >] snd_virmidi_dev_attach_seq sound/core/seq/seq_virmidi.c:372 [<ffffffff84fdc55f>] snd_virmidi_dev_register+0x29f/0x750 sound/core/seq/seq_virmidi.c:439 [<ffffffff84f824ec>] snd_rawmidi_dev_register+0x30c/0xd40 sound/core/rawmidi.c:1589 [<ffffffff84f33583>] __snd_device_register.part.0+0x63/0xc0 sound/core/device.c:164 [< inline >] __snd_device_register sound/core/device.c:162 [<ffffffff84f341ed>] snd_device_register_all+0xad/0x110 sound/core/device.c:212 [<ffffffff84f277cf>] snd_card_register+0xef/0x6a0 sound/core/init.c:749 [<ffffffff84fefc0f>] snd_virmidi_probe+0x3ef/0x590 sound/drivers/virmidi.c:123 [<ffffffff832feb6c>] platform_drv_probe+0x8c/0x160 drivers/base/platform.c:562 [< inline >] really_probe drivers/base/dd.c:377 [<ffffffff832f862e>] driver_probe_device+0x37e/0xc90 drivers/base/dd.c:499 [<ffffffff832f926e>] __device_attach_driver+0x19e/0x250 drivers/base/dd.c:584 [<ffffffff832f23df>] bus_for_each_drv+0x13f/0x1d0 drivers/base/bus.c:464 [<ffffffff832f809f>] __device_attach+0x1ef/0x2e0 drivers/base/dd.c:641 [<ffffffff832f93ba>] device_initial_probe+0x1a/0x20 drivers/base/dd.c:688 [<ffffffff832f5939>] bus_probe_device+0x1e9/0x290 drivers/base/bus.c:558 [<ffffffff832ef11b>] device_add+0x84b/0x1490 drivers/base/core.c:1120 [<ffffffff832fe2c9>] platform_device_add+0x389/0x790 drivers/base/platform.c:403 [<ffffffff832fff26>] platform_device_register_full+0x396/0x4c0 drivers/base/platform.c:535 [< inline >] platform_device_register_resndata include/linux/platform_device.h:111 [< inline >] platform_device_register_simple include/linux/platform_device.h:140 [<ffffffff884146d1>] alsa_card_virmidi_init+0x104/0x1da sound/drivers/virmidi.c:172 [<ffffffff81002259>] do_one_initcall+0x159/0x380 init/main.c:794 [< inline >] do_initcall_level init/main.c:859 [< inline >] do_initcalls init/main.c:867 [< inline >] do_basic_setup init/main.c:885 [<ffffffff88315c1a>] kernel_init_freeable+0x474/0x52d init/main.c:1010 [<ffffffff86312683>] kernel_init+0x13/0x150 init/main.c:936 [<ffffffff86336fef>] ret_from_fork+0x3f/0x70 arch/x86/entry/entry_64.S:468 -> #0 (register_mutex#5){+.+.+.}: [< inline >] check_prev_add kernel/locking/lockdep.c:1853 [< inline >] check_prevs_add kernel/locking/lockdep.c:1958 [< inline >] validate_chain kernel/locking/lockdep.c:2144 [<ffffffff8145742b>] __lock_acquire+0x31eb/0x4700 kernel/locking/lockdep.c:3206 [<ffffffff8145ad8c>] lock_acquire+0x1dc/0x430 kernel/locking/lockdep.c:3585 [< inline >] __mutex_lock_common kernel/locking/mutex.c:518 [<ffffffff8632cf61>] mutex_lock_nested+0xb1/0xa50 kernel/locking/mutex.c:618 [<ffffffff84f889cb>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 [<ffffffff84fdda77>] midisynth_subscribe+0xf7/0x340 sound/core/seq/seq_midi.c:188 [<ffffffff84fc2a4e>] subscribe_port.isra.2+0x14e/0x2b0 sound/core/seq/seq_ports.c:426 [<ffffffff84fc4dd0>] snd_seq_port_connect+0x490/0x840 sound/core/seq/seq_ports.c:527 [<ffffffff84fb6f54>] snd_seq_ioctl_subscribe_port+0x1c4/0x290 sound/core/seq/seq_clientmgr.c:1464 [<ffffffff84fb126d>] snd_seq_do_ioctl+0x19d/0x1c0 sound/core/seq/seq_clientmgr.c:2209 [<ffffffff84fb2c2b>] snd_seq_kernel_client_ctl+0xdb/0x170 sound/core/seq/seq_clientmgr.c:2423 [<ffffffff84fd69e4>] snd_seq_oss_midi_open+0x3b4/0x610 sound/core/seq/oss/seq_oss_midi.c:375 [<ffffffff84fd6ccb>] snd_seq_oss_midi_open_all+0x8b/0xd0 sound/core/seq/oss/seq_oss_midi.c:306 [<ffffffff84fca1d5>] snd_seq_oss_open+0x5c5/0x8d0 sound/core/seq/oss/seq_oss_init.c:276 [<ffffffff84fc903a>] odev_open+0x6a/0x90 sound/core/seq/oss/seq_oss.c:138 [<ffffffff84f2279f>] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [<ffffffff817be1fa>] chrdev_open+0x22a/0x4c0 fs/char_dev.c:388 [<ffffffff817a9c02>] do_dentry_open+0x6a2/0xcb0 fs/open.c:736 [<ffffffff817ad2db>] vfs_open+0x17b/0x1f0 fs/open.c:853 [< inline >] do_last fs/namei.c:3254 [<ffffffff817e00d9>] path_openat+0xde9/0x5e30 fs/namei.c:3386 [<ffffffff817e895e>] do_filp_open+0x18e/0x250 fs/namei.c:3421 [<ffffffff817ada5c>] do_sys_open+0x1fc/0x420 fs/open.c:1022 [< inline >] SYSC_open fs/open.c:1040 [<ffffffff817adcad>] SyS_open+0x2d/0x40 fs/open.c:1035 [<ffffffff86336c36>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 other info that might help us debug this: Chain exists of: register_mutex#5 --> &grp->list_mutex --> &grp->list_mutex/1 Possible unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&grp->list_mutex/1); lock(&grp->list_mutex); lock(&grp->list_mutex/1); lock(register_mutex#5); *** DEADLOCK *** 3 locks held by syz-executor/21025: #0: (register_mutex#4){+.+.+.}, at: [<ffffffff84fc902f>] odev_open+0x5f/0x90 sound/core/seq/oss/seq_oss.c:137 #1: (&grp->list_mutex){++++.+}, at: [<ffffffff84fc4ae2>] snd_seq_port_connect+0x1a2/0x840 sound/core/seq/seq_ports.c:505 #2: (&grp->list_mutex/1){+.+...}, at: [<ffffffff84fc4afa>] snd_seq_port_connect+0x1ba/0x840 sound/core/seq/seq_ports.c:506 stack backtrace: CPU: 2 PID: 21025 Comm: syz-executor Not tainted 4.4.0+ #276 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff88006d0af0c0 ffffffff82999e2d ffffffff88fa1eb0 ffffffff88fa2f90 ffffffff88fa2060 ffff88006d0af110 ffffffff81450658 ffff880065c8df00 ffff880065c8e79a 0000000000000000 ffff880065c8e778 Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82999e2d>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 [<ffffffff81450658>] print_circular_bug+0x288/0x340 kernel/locking/lockdep.c:1226 [< inline >] check_prev_add kernel/locking/lockdep.c:1853 [< inline >] check_prevs_add kernel/locking/lockdep.c:1958 [< inline >] validate_chain kernel/locking/lockdep.c:2144 [<ffffffff8145742b>] __lock_acquire+0x31eb/0x4700 kernel/locking/lockdep.c:3206 [<ffffffff8145ad8c>] lock_acquire+0x1dc/0x430 kernel/locking/lockdep.c:3585 [< inline >] __mutex_lock_common kernel/locking/mutex.c:518 [<ffffffff8632cf61>] mutex_lock_nested+0xb1/0xa50 kernel/locking/mutex.c:618 [<ffffffff84f889cb>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 [<ffffffff84fdda77>] midisynth_subscribe+0xf7/0x340 sound/core/seq/seq_midi.c:188 [<ffffffff84fc2a4e>] subscribe_port.isra.2+0x14e/0x2b0 sound/core/seq/seq_ports.c:426 [<ffffffff84fc4dd0>] snd_seq_port_connect+0x490/0x840 sound/core/seq/seq_ports.c:527 [<ffffffff84fb6f54>] snd_seq_ioctl_subscribe_port+0x1c4/0x290 sound/core/seq/seq_clientmgr.c:1464 [<ffffffff84fb126d>] snd_seq_do_ioctl+0x19d/0x1c0 sound/core/seq/seq_clientmgr.c:2209 [<ffffffff84fb2c2b>] snd_seq_kernel_client_ctl+0xdb/0x170 sound/core/seq/seq_clientmgr.c:2423 [<ffffffff84fd69e4>] snd_seq_oss_midi_open+0x3b4/0x610 sound/core/seq/oss/seq_oss_midi.c:375 [<ffffffff84fd6ccb>] snd_seq_oss_midi_open_all+0x8b/0xd0 sound/core/seq/oss/seq_oss_midi.c:306 [<ffffffff84fca1d5>] snd_seq_oss_open+0x5c5/0x8d0 sound/core/seq/oss/seq_oss_init.c:276 [<ffffffff84fc903a>] odev_open+0x6a/0x90 sound/core/seq/oss/seq_oss.c:138 [<ffffffff84f2279f>] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [<ffffffff817be1fa>] chrdev_open+0x22a/0x4c0 fs/char_dev.c:388 [<ffffffff817a9c02>] do_dentry_open+0x6a2/0xcb0 fs/open.c:736 [<ffffffff817ad2db>] vfs_open+0x17b/0x1f0 fs/open.c:853 [< inline >] do_last fs/namei.c:3254 [<ffffffff817e00d9>] path_openat+0xde9/0x5e30 fs/namei.c:3386 [<ffffffff817e895e>] do_filp_open+0x18e/0x250 fs/namei.c:3421 [<ffffffff817ada5c>] do_sys_open+0x1fc/0x420 fs/open.c:1022 [< inline >] SYSC_open fs/open.c:1040 [<ffffffff817adcad>] SyS_open+0x2d/0x40 fs/open.c:1035 [<ffffffff86336c36>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 On commit 30f05309bde49295e02e45c7e615f73aa4e0ccc2.

2 5

[alsa-devel] sound: out-of-bounds write in snd_rawmidi_kernel_write1
by Dmitry Vyukov 03 Feb '16

03 Feb '16

Hello, The following program triggers an out-of-bounds write in snd_rawmidi_kernel_write1 (run in parallel loop). It seems to try to copy -1 bytes (aka 4GB) from user space into kernel smashing all on its way. ================================================================== BUG: KASAN: use-after-free in memset+0x1a/0x30 at addr ffff8800326e3cef Write of size 4294950912 by task a.out/7292 ============================================================================= BUG kmalloc-4096 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- INFO: Allocated in open_substream+0x2ff/0x780 age=288 cpu=1 pid=7287 [< none >] ___slab_alloc+0x564/0x5b0 mm/slub.c:2470 [< none >] __slab_alloc+0x66/0xc0 mm/slub.c:2499 [< inline >] slab_alloc_node mm/slub.c:2562 [< inline >] slab_alloc mm/slub.c:2604 [< none >] kmem_cache_alloc_trace+0x25c/0x300 mm/slub.c:2621 [< inline >] kmalloc include/linux/slab.h:463 [< inline >] snd_rawmidi_runtime_create sound/core/rawmidi.c:127 [< none >] open_substream+0x2ff/0x780 sound/core/rawmidi.c:266 [< none >] rawmidi_open_priv+0x144/0x300 sound/core/rawmidi.c:312 [< none >] snd_rawmidi_open+0x3fb/0xa90 sound/core/rawmidi.c:416 [< none >] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [< none >] chrdev_open+0x22a/0x4c0 fs/char_dev.c:388 [< none >] do_dentry_open+0x6a2/0xcb0 fs/open.c:736 [< none >] vfs_open+0x17b/0x1f0 fs/open.c:853 [< inline >] do_last fs/namei.c:3254 [< none >] path_openat+0xde9/0x5e30 fs/namei.c:3386 [< none >] do_filp_open+0x18e/0x250 fs/namei.c:3421 [< none >] do_sys_open+0x1fc/0x420 fs/open.c:1022 [< inline >] SYSC_open fs/open.c:1040 [< none >] SyS_open+0x2d/0x40 fs/open.c:1035 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 Call Trace: [<ffffffff817654b7>] __asan_storeN+0x127/0x1a0 mm/kasan/kasan.c:522 [<ffffffff8176582a>] memset+0x1a/0x30 mm/kasan/kasan.c:280 [<ffffffff82c15294>] copy_user_handle_tail+0xb4/0xd0 arch/x86/lib/usercopy_64.c:86 [< inline >] copy_from_user ./arch/x86/include/asm/uaccess.h:735 [<ffffffff852855da>] snd_rawmidi_kernel_write1+0x34a/0x760 sound/core/rawmidi.c:1250 [<ffffffff852878b3>] snd_rawmidi_write+0x543/0xb30 sound/core/rawmidi.c:1318 [<ffffffff817b8be1>] do_loop_readv_writev+0x141/0x1e0 fs/read_write.c:719 [<ffffffff817bcb38>] do_readv_writev+0x5f8/0x6e0 fs/read_write.c:849 [<ffffffff817bcd56>] vfs_writev+0x86/0xc0 fs/read_write.c:886 [< inline >] SYSC_writev fs/read_write.c:919 [<ffffffff817bfec1>] SyS_writev+0x111/0x2b0 fs/read_write.c:911 [<ffffffff86661376>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 ================================================================== // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> long r[19]; int main() { memset(r, -1, sizeof(r)); r[0] = syscall(SYS_mmap, 0x20000000ul, 0x22000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); r[2] = syscall(SYS_open, "/dev/midi3", 0x2ul, 0, 0, 0); *(uint64_t*)0x20013ffe = (uint64_t)0x20013f69; *(uint64_t*)0x20014006 = (uint64_t)0x97; *(uint64_t*)0x2001400e = (uint64_t)0x20003ffe; *(uint64_t*)0x20014016 = (uint64_t)0x3; *(uint64_t*)0x2001401e = (uint64_t)0x2001ef70; *(uint64_t*)0x20014026 = (uint64_t)0x92; *(uint64_t*)0x2001402e = (uint64_t)0x20014730; *(uint64_t*)0x20014036 = (uint64_t)0x1; *(uint64_t*)0x2001403e = (uint64_t)0x2001e000; *(uint64_t*)0x20014046 = (uint64_t)0x1000; memcpy((void*)0x20013f69, "\x0e\x6f\xfe\xa4\xc8\xbf\xaa\x28\x69\x82\xe0\x8c\x45\xab\x90" "\xb9\x6a\xeb\x50\x3d\x1a\x59\x59\x33\x0a\xa5\xb8\xfe\x83\x75" "\x1f\xb5\x25\x76\x65\x53\xb6\xe7\xe2\xe7\xe3\xd7\x55\x36\xb6" "\x59\x2f\xf8\x61\xda\xdb\x73\x32\x94\xb1\x0e\x17\x87\xad\x8f" "\x73\x6c\xc7\xeb\x47\xa2\x65\x77\xfb\xfc\xec\x70\x5c\x9e\x01" "\x78\xef\x1b\x69\xab\xef\x58\x13\x35\x99\xf0\x2b\x25\x99\xdb" "\xba\x16\xa7\x43\xc5\x19\x32\x91\x13\x24\x15\xd1\x5e\xb0\x1a" "\x5f\x63\xe2\xc3\x61\x9a\x79\x82\x57\x60\x21\x79\x6e\xf5\x21" "\x74\x18\xcf\x8f\x73\x93\x70\x3a\xfe\xf8\xe3\xde\x6a\x85\x3b" "\x5a\xfc\x0b\xfe\x02\x8f\x32\xe6\xc5\x14\x1c\x07\x24\xce\xcb" "\x95", 151); memcpy((void*)0x20003ffe, "\x3d\x07\x31", 3); memcpy((void*)0x2001ef70, "\x5c\x39\xe0\x3f\xe7\xdd\x08\xeb\x7c\xb0\xe7\x31\x09\xdf\x84" "\x1a\xbb\x98\xe1\xc8\x08\x00\x3b\xe9\xef\x62\x4a\x18\x10\xff" "\xa1\x7f\xa0\xd6\x81\x86\xed\xb5\x43\x33\x7c\xa6\x01\xb8\xfc" "\xe1\xee\x43\x95\x05\xf1\x5d\xa6\xb4\xb1\x62\x74\x82\xc6\x8b" "\x9f\xfb\x9b\xe8\x41\xee\x2e\x44\xc0\x20\x37\xca\xe5\x33\xf3" "\xeb\x9e\x2e\x15\x56\xce\x26\x5e\x6c\xee\xca\x49\x00\xe6\xf2" "\x21\xd4\x07\xf1\x00\xee\x21\x5a\xd5\x7a\x84\x4c\x3e\x7c\x8d" "\xe5\xb5\x35\x57\x69\xe4\x18\x09\x5b\xa7\xdb\xe6\x2f\x56\x78" "\x83\xa1\x0c\xdd\x07\x74\x32\xb2\x85\x75\xeb\x51\x5b\x22\xd1" "\x57\x95\xcd\xc2\xc4\xf9\x2e\xd8\xce\x9a\x52", 146); memcpy((void*)0x20014730, "\xc3", 1); memcpy( (void*)0x2001e000, "\xff\x20\xcc\xa3\x99\x07\x76\x17\xc4\x29\xef\xa5\x6a\xef\x9c\x9b" "\x58\x94\x16\xfc\x87\x60\xb1\x04\x16\x80\x1d\xcb\xfd\x8b\xd7\x9a" "\x41\x87\xb2\xdf\xe8\xc3\x17\x50\xd6\x5a\x01\x06\xc7\xc5\xb1\xf3" "\x56\x57\x81\xa8\x2c\x0b\x81\x67\x93\x59\xc1\xbb\xb5\x2d\x45\xc5" "\x27\x05\x66\x0b\xfd\x28\x1e\xdb\x45\x89\x34\x0b\x03\xfe\x5f\xa7" "\x47\xeb\xa1\xe0\xff\xe1\xdb\x12\xa7\x11\x37\xe0\x96\x53\xa2\x29" "\x1d\x3c\x3c\xda\x8e\x97\xeb\x58\x64\x8d\xff\x13\x70\x08\x96\x28" "\x32\x4e\xce\x4c\xb3\x17\x1c\x74\xa8\x8d\x7b\x78\x32\x4b\xcc\xf8" "\xf7\xcb\x33\x5c\xd5\x6b\xa3\x2a\x9c\x7c\x41\x3e\x87\x92\xec\x22" "\x44\x36\x28\x2d\x76\xde\x44\xa8\xef\x9d\x2c\xa2\xf2\xc5\x94\x2b" "\x3a\x17\xa4\xd7\x82\x4e\xef\xee\x1a\x4a\x6b\x98\xd5\xe4\x40\xbc" "\x5a\xe1\x76\x47\x17\x2b\x45\xb6\xb6\xa4\xeb\x5a\x54\x82\x6a\x98" "\x14\xf0\x4e\x5a\x8c\x30\x6a\x0f\xcb\x0f\x1b\x62\x8c\x48\x54\x16" "\xb8\x94\x28\xd0\xcc\x47\x4f\xb4\x01\x40\x3c\xb7\xed\xf0\x96\x2a" "\x55\xb2\xac\xf6\xa2\xab\x93\x19\x3f\xf6\xf8\xad\x8f\xa3\x5a\x6f" "\x15\x26\xa3\x25\xbb\x58\x6e\x02\x58\xfa\x3a\xcf\xd9\xf8\x17\xb9" "\xa1\x4c\x45\x16\x45\xcb\x7a\x84\x37\x3a\xb4\x3a\xc6\x98\xa9\x5c" "\xf4\x9e\xdb\xcf\xa9\xbb\xcc\xcb\x15\x36\x52\xb0\xcc\x3d\xd0\xfa" "\xd6\xc0\xff\xf6\xb7\x0e\x80\x31\x8e\x76\xfc\xe9\xd8\xf6\xb4\xbc" "\xfa\xec\x94\x13\x64\x9d\x13\x7e\x91\x9c\x10\x27\xda\xc8\xea\x9b" "\xc2\x09\x58\x81\x9e\x44\xf2\xc3\xef\x94\x6d\x2f\xa9\xa5\x19\xd7" "\x78\x61\x2e\xd3\x01\x8a\xc3\xac\x1b\xb9\xaa\x00\x12\xc5\x7e\x7f" "\x7b\x34\x27\x4c\x38\x00\x9c\x57\xa6\xe2\x5a\x5a\xe6\x00\x01\x56" "\xa2\xed\xa6\xcd\x95\x0b\xde\x5e\x09\x62\x61\x7c\x2b\x16\x13\xb4" "\x04\xf8\x1a\x5b\x15\x9d\x62\xd5\xac\x4b\x5b\x35\x70\xd0\xe2\x26" "\xf9\xc0\x7c\x73\xad\x9c\xd5\xef\x7b\x9f\x01\x8b\xb6\xb3\x85\xbf" "\x34\xa2\x8c\xce\x8a\xaf\xe1\x5b\xdd\x7f\xa6\x23\x55\x8b\xa4\xa3" "\x06\x2f\xf1\xa0\xd8\x2a\xcc\x9a\x94\x19\x22\xfe\x65\x69\x9b\xcb" "\x6d\x6c\x1c\x38\x25\x82\x9b\xec\xe1\x75\xb4\xbe\xac\x21\x4b\xec" "\xfd\x22\x76\xcb\x35\x4b\xaa\x5c\x3a\x7b\xaa\x6f\x3c\x68\xa7\xa5" "\x3d\xa2\xd5\x25\x9a\xe4\x6f\x80\xc5\xb5\xcf\x76\x37\xd8\xd9\x25" "\xe6\xc7\x26\x40\xdf\x1d\xd3\x41\xb2\x58\x69\xa7\x18\x57\xf8\x35" "\x26\x70\x30\x08\x76\xed\x2b\x37\x2a\x54\x7d\xa5\x7a\x66\x8d\xbf" "\x4a\x20\x3d\x1b\xd1\x28\x1e\x7e\xfe\xd3\xb1\x6c\x64\x87\x70\x27" "\x9e\xcf\x07\xd9\x5a\x9e\xbb\xdb\x3b\xb2\xcb\x50\x78\x3c\x45\x76" "\xec\x59\xa9\xca\xb2\x80\x8c\xc5\x62\x76\x8c\x76\x61\xb5\xf2\xcf" "\x0f\x37\x7d\x44\x7f\xc3\x2c\xef\x7a\x07\x3b\xe2\xb9\x9b\x15\xc6" "\xa6\xb5\xdf\x02\x12\x75\x7d\x8c\x09\x1a\x50\x91\x70\x46\xbb\x18" "\xf7\x44\x05\xc8\xdb\x78\xa2\x87\xd6\x0a\x5f\x14\xf0\xaf\x61\xd7" "\x14\x91\x31\xae\xad\xea\x05\xb3\xd7\xc3\xae\x57\xe5\xbe\x9a\x43" "\xde\x55\xcc\x8e\x91\xac\x5c\xb5\xb9\x0a\x68\x28\x41\xdc\x09\x69" "\x0b\x86\xa1\x46\x67\x96\x04\xf6\x2e\x6e\x07\x11\x62\xb5\x96\x09" "\x2b\x5f\xcb\x7a\x9b\xcb\x77\x1b\x79\xb0\xab\x05\x89\x15\x5a\xcd" "\xab\xd6\x82\x8c\xb0\x65\xc1\x88\x6c\x14\x60\x3d\x77\xf7\xb4\xc1" "\xdf\x43\x0a\x80\x37\xc4\x82\x31\x27\x93\x2d\x93\x06\xfb\x92\xce" "\x19\x3b\xb8\xcf\x4f\x42\xf6\x44\x7a\x5d\xed\xe7\x09\x79\x19\x42" "\xb8\x30\x7e\x4b\x36\xd9\x75\x46\xd8\x7b\xba\x32\x01\x29\x8b\xec" "\xdb\x66\xcf\x4b\x04\xde\x8d\x5e\x1d\xf1\x58\xc1\x3b\xcb\x04\x13" "\x3d\x8a\x9e\xa8\x8f\xce\x0c\xed\x8c\x1e\xf0\x3e\x8c\x59\x14\x53" "\x17\x9d\xb8\x47\x33\xbc\xa3\xe2\xdc\x16\xaf\xd2\x28\xe4\xfe\xa1" "\xaf\x98\x7a\xc9\x4b\x3b\x38\xea\x8e\x1a\x36\x3d\xb4\xb8\x9d\x28" "\xbc\xc6\x9f\xd4\x21\xa9\x52\xbd\x1c\x77\x69\xb8\x41\x0e\x66\x9a" "\x29\x99\x50\x4c\x76\x46\x99\xcc\xbc\x5a\x23\x1c\x19\xbb\x25\x07" "\xf5\xb3\x5c\x38\x9d\xed\xc5\x85\xea\xb4\xd6\x14\xdb\xd1\x54\xb7" "\x13\xec\xcb\x24\xce\x8c\xf9\xb5\xc7\xbe\x54\x17\x29\x19\xa2\xaf" "\xb6\xd3\x14\xae\x83\xa7\x43\xb7\xbe\x28\xba\x2c\x52\xc0\xaa\x37" "\x98\x14\x87\xe9\xbd\x2e\x1c\x93\x29\xd4\xad\x87\x4e\x9a\x7b\x94" "\x30\x72\x68\x31\x2c\xa5\x2b\xed\x52\xc9\x31\x43\xf1\x2c\x77\xcb" "\x73\x64\x07\x3f\x8a\x59\x59\xf3\x8a\x9c\x9e\xb0\xb6\x7b\x8d\x0e" "\x6b\x5a\x33\xcd\x04\x5e\x77\x93\xd4\x23\xb0\xbe\xca\x08\x95\xd6" "\x02\xd2\x22\xbb\x8d\x4c\xbc\x68\x6b\xc7\x6f\x47\x3d\x78\x4e\x56" "\xaf\x86\x11\x9f\x8b\x16\x22\x8b\x93\x89\x10\x58\xaa\x1a\xaf\x97" "\xc7\x33\x97\xcd\x6d\xb2\x17\xa3\x90\xd5\x92\x5e\xb7\xeb\x9f\x13" "\x02\x34\xa3\x2c\xe8\x0c\xc9\xf0\xe6\xda\x06\x70\xe5\x87\xe9\xbf" "\x8e\x68\x34\x5a\xfb\xe5\x39\x6a\xde\xc4\x15\xae\x4c\xe0\x6e\x0b" "\x42\x2f\x2a\x77\xd3\x5b\x69\xa0\x07\xaf\x1c\x55\xa8\x28\x24\x81" "\xc0\x34\xd5\xde\xca\xcf\xa9\x85\x58\x90\xd4\x5b\xf3\x73\xd1\xfb" "\xc4\x25\x64\xb0\x15\xf6\x98\x73\x65\x38\x89\x72\xa6\x5b\x5e\xdb" "\xca\x92\xeb\x39\xb9\x90\x29\x18\x72\xeb\xee\xa0\x90\x1a\x34\x4c" "\x62\x4f\xd9\x2f\x69\x49\x88\x4c\x24\x26\x02\x3b\x74\x12\xf1\x4c" "\x78\x25\xbb\xd3\x26\xc9\x25\xf0\xda\x68\x5a\x50\x7b\x69\x90\x9e" 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"\xbf\x26\x23\x46\xc6\xcf\xef\x2a\xed\xa7\xc0\x52\xda\x40\x2e\xd6" "\x2e\xc4\xf1\xb2\x2f\x69\xac\x73\xe4\x2d\x75\xea\x19\x4c\xb5\x50" "\xbc\x03\x51\x9c\x05\xda\x8a\x3b\xc9\x98\xd2\x43\x40\xc6\xd9\x0f" "\x53\x2e\x24\x56\x55\x4e\x41\xc9\xcd\xa9\x06\xe8\xbc\x69\x10\xaa" "\x99\x3d\x56\x9e\xd2\xbc\x48\xcd\x58\x2d\xe3\x16\x6a\x23\x3d\x85" "\xc5\x14\x80\x46\x07\x96\xcf\xd9\xdc\xff\x14\x39\xfd\x5c\xc2\x2d" "\xb9\xf9\x51\x09\x5d\xd3\xe4\xf6\x3f\x51\x62\x10\xf6\xd4\x76\x78" "\x11\xf0\xd5\x63\xb8\x53\x2b\xb2\xa0\x58\x4e\x71\x90\xea\xb6\xc6" "\x92\x59\x06\x8d\x53\x63\x0a\x0c\xd4\x62\x2d\x57\x12\xf9\xf8\xd8" "\xdb\x10\x40\x2e\x09\x83\x1b\xbb\x8c\xdd\xbc\x0c\xa0\xc8\x14\x23" "\xa5\x93\xed\xd9\x2d\x0b\xc0\xfd\x0e\x1d\xd1\xbb\x95\x96\xb6\xbc" "\x81\x31\x6f\x20\x6f\x72\x1c\x4a\x87\xfb\x66\x2d\x79\x3c\xa4\x6d" "\x05\xc1\x71\x50\x83\xad\xea\x03\x26\x2b\xe4\x10\xc8\x33\x0b\x61" "\xdf\x30\x4b\x27\x8f\xfe\xbd\xcc\x8d\xda\x4f\xaf\x8f\xad\xaa\x25" "\xa1\x50\x95\x95\x7d\x8e\x35\xec\xe8\x7e\xd4\x98\x68\xe4\x95\xac" "\xa6\x98\x95\x38\xf8\xd2\xab\xdc\xb9\x77\xfc\xa9\xbf\xb5\x18\xed" "\xca\x15\xb2\xe7\xd0\x23\xcb\xc4\x73\xc0\x85\x0b\x57\xeb\xf4\xe6" "\x11\x97\xb1\x03\xa6\x66\x0a\x24\x3b\xb6\xe8\x2f\xfe\x9c\x78\x0a" "\x42\x01\xe9\x35\x58\xf7\x55\x13\x80\x24\x23\x30\x62\x6c\x7b\x86" "\xff\x7e\x12\x34\xf3\x33\x4c\x1c\xed\x5a\x96\xd6\xcd\x04\xfb\xf9" "\x7c\xde\xc0\xd5\x70\x52\x4e\x0f\xa2\x8e\x6b\xa6\x8f\x3a\x08\xf9" "\x08\x0f\x2f\xd7\xa2\xb1\x91\x53\x65\x63\x63\xb8\x31\xba\xf4\x21" "\xb5\x17\xd7\x59\x94\xd0\x0e\x8c\x91\x5c\x37\xf8\x3a\xd8\x8c\x7f" "\x76\xd0\x63\x7a\xd9\xb4\xa6\x3d\xf8\xfb\x72\xd5\x65\xf0\x10\x4a" "\x42\x15\x70\xb1\x9b\x5f\xf8\xf9\x9f\x5b\x29\x0a\x28\x2a\x68\x6b" "\xaf\xec\x84\x94\xbc\x22\xdf\xe8\x3e\xcb\xee\xc3\xc8\x03\x00\xf3" "\x02\xff\x07\x4d\x95\x62\x12\x9e\x51\x1a\x20\x4a\x5a\xa2\x66\x81" "\x3d\x29\x98\x23\x17\x11\x95\x84\x51\x9f\x40\x7a\x3d\x6f\x61\xbe" "\x2b\x1b\x03\x82\xad\xb6\x8f\x69\x3d\x31\x39\xc7\x8b\x0a\x77\x6e" "\x02\x88\xdd\x13\x6a\x9c\x72\x46\x5d\xf0\x92\xb4\x76\xac\xc0\xbd" "\x8d\x2e\x3c\xcb\xa3\x89\x31\x07\x36\x68\x35\x03\x4c\x95\x6d" "\xbd", 4096); r[18] = syscall(SYS_writev, r[2], 0x20013ffeul, 0x5ul, 0, 0, 0); return 0; } On commit 34229b277480f46c1e9a19f027f30b074512e68b.

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[alsa-devel] sound: WARNING in snd_rawmidi_kernel_write1
by Dmitry Vyukov 03 Feb '16

03 Feb '16

Hello, The following program triggers WARNING in snd_rawmidi_kernel_write1: // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <unistd.h> #include <sys/types.h> #include <sys/stat.h> #include <sys/uio.h> #include <fcntl.h> int main() { int fd; struct iovec iov; fd = open("/dev/snd/midiC3D0", O_RDWR); iov.iov_base = ""; iov.iov_len = 0; writev(fd, &iov, 1); return 0; } ------------[ cut here ]------------ WARNING: CPU: 1 PID: 27775 at sound/core/rawmidi.c:1181 snd_rawmidi_kernel_write1+0x4e1/0x780() Modules linked in: CPU: 1 PID: 27775 Comm: syz-executor Not tainted 4.5.0-rc1+ #305 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff88003158f958[ 845.661205] ------------[ cut here ]------------ WARNING: CPU: 3 PID: 27781 at sound/core/rawmidi.c:1181 snd_rawmidi_kernel_write1+0x4e1/0x780() Modules linked in: ffffffff82be11ad 0000000000000000 ffff880032138000 ffffffff87188ec0 ffff88003158f998 ffffffff813536d9 ffffffff8527ab31 ffffffff87188ec0 000000000000049d ffff8800321e4b88 Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82be11ad>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 [<ffffffff813536d9>] warn_slowpath_common+0xd9/0x140 kernel/panic.c:482 [<ffffffff81353909>] warn_slowpath_null+0x29/0x30 kernel/panic.c:515 [<ffffffff8527ab31>] snd_rawmidi_kernel_write1+0x4e1/0x780 sound/core/rawmidi.c:1181 [<ffffffff8527cc93>] snd_rawmidi_write+0x543/0xb30 sound/core/rawmidi.c:1273 [<ffffffff817b7611>] do_loop_readv_writev+0x141/0x1e0 fs/read_write.c:719 [<ffffffff817bb478>] do_readv_writev+0x5f8/0x6e0 fs/read_write.c:849 [<ffffffff817bb696>] vfs_writev+0x86/0xc0 fs/read_write.c:886 [< inline >] SYSC_writev fs/read_write.c:919 [<ffffffff817be801>] SyS_writev+0x111/0x2b0 fs/read_write.c:911 [<ffffffff866531b6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 ---[ end trace 35fa82e449e91fae ]--- On commit 26cd83670f2f5a3d5b5514a1f7d96567cdb9558b

2 4

[alsa-devel] [PATCH] ALSA: hda - Add fixup for Mac Mini 7,1 model
by Takashi Iwai 03 Feb '16

03 Feb '16

Mac Mini 7,1 model with CS4208 codec reports the headphone jack detection wrongly in an inverted way. Moreover, the advertised pins for the audio input and SPDIF output have actually no jack detection. This patch addresses these issues. The inv_jack_detect flag is set for fixing the headphone jack detection, and the pin configs for audio input and SPDIF output are marked as non-detectable. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=105161 Report-and-tested-by: moosotc(a)gmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a12ae8ac0914..c1c855a6c0af 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -614,6 +614,7 @@ enum { CS4208_MAC_AUTO, CS4208_MBA6, CS4208_MBP11, + CS4208_MACMINI, CS4208_GPIO0, }; @@ -621,6 +622,7 @@ static const struct hda_model_fixup cs4208_models[] = { { .id = CS4208_GPIO0, .name = "gpio0" }, { .id = CS4208_MBA6, .name = "mba6" }, { .id = CS4208_MBP11, .name = "mbp11" }, + { .id = CS4208_MACMINI, .name = "macmini" }, {} }; @@ -632,6 +634,7 @@ static const struct snd_pci_quirk cs4208_fixup_tbl[] = { /* codec SSID matching */ static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x5e00, "MacBookPro 11,2", CS4208_MBP11), + SND_PCI_QUIRK(0x106b, 0x6c00, "MacMini 7,1", CS4208_MACMINI), SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6), SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6), SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11), @@ -666,6 +669,24 @@ static void cs4208_fixup_mac(struct hda_codec *codec, snd_hda_apply_fixup(codec, action); } +/* MacMini 7,1 has the inverted jack detection */ +static void cs4208_fixup_macmini(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x18, 0x00ab9150 }, /* mic (audio-in) jack: disable detect */ + { 0x21, 0x004be140 }, /* SPDIF: disable detect */ + { } + }; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + /* HP pin (0x10) has an inverted detection */ + codec->inv_jack_detect = 1; + /* disable the bogus Mic and SPDIF jack detections */ + snd_hda_apply_pincfgs(codec, pincfgs); + } +} + static int cs4208_spdif_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -709,6 +730,12 @@ static const struct hda_fixup cs4208_fixups[] = { .chained = true, .chain_id = CS4208_GPIO0, }, + [CS4208_MACMINI] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs4208_fixup_macmini, + .chained = true, + .chain_id = CS4208_GPIO0, + }, [CS4208_GPIO0] = { .type = HDA_FIXUP_FUNC, .v.func = cs4208_fixup_gpio0, -- 2.7.0

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[alsa-devel] [PATCH v3 0/2] Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 03 Feb '16

03 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver. For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery This is version 3 of the patch series. It's been quite a while since v2 and I'm hoping now that all the previous comments have been acted on. Thanks in advance, CK --- Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): dt-bindings: add sun4i SPDIF transceiver bindings ASOC: sunxi: Add support for the SPDIF block .../devicetree/bindings/sound/sun4i,spdif.txt | 46 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 586 +++++++++++++++++++++ 4 files changed, 641 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sun4i,spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.0

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[alsa-devel] New codec support for ALC225
by Kailang 03 Feb '16

03 Feb '16

Hi Takashi, Attach has 3 patches for support ALC225. The sequence is below. 1. 0000-new-codec-of-alc225.patch 2. 0000-support-headset-mode-for-alc225.patch 3. 0000-dell-headset-mode-for-alc225.patch BR, Kailang

2 1

[alsa-devel] [PATCH 1/1] alsabat: rename to avoid naming conflict
by han.lu＠intel.com 03 Feb '16

03 Feb '16

From: "Lu, Han" <han.lu(a)intel.com> alsa-utils as well as bareos-bat (as well a some Bacula packages) all contain a program called /usr/bin/bat, which causes conflicts on various distributions ("basic audio tester" vs "bareos administration tool"("bacula administration tool")). Rename to avoid conflict. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/Makefile.am b/bat/Makefile.am index f0dc5ab..8dfafa9 100644 --- a/bat/Makefile.am +++ b/bat/Makefile.am @@ -1,9 +1,9 @@ -bin_PROGRAMS = bat -man_MANS = bat.1 +bin_PROGRAMS = alsabat +man_MANS = alsabat.1 -EXTRA_DIST = bat.1 +EXTRA_DIST = alsabat.1 -bat_SOURCES = \ +alsabat_SOURCES = \ bat.c \ common.c \ analyze.c \ @@ -21,4 +21,4 @@ noinst_HEADERS = \ AM_CPPFLAGS = \ -Wall -I$(top_srcdir)/include -bat_LDADD = @FFTW_LIB@ +alsabat_LDADD = @FFTW_LIB@ diff --git a/bat/alsabat.1 b/bat/alsabat.1 new file mode 100644 index 0000000..8d0b9c0 --- /dev/null +++ b/bat/alsabat.1 @@ -0,0 +1,159 @@ +.TH ALSABAT 1 "20th October 2015" +.SH NAME +alsabat \- command\-line sound tester for ALSA sound card driver + +.SH SYNOPSIS +\fBalsabat\fP [\fIflags\fP] + +.SH DESCRIPTION +\fBALSABAT(ALSA Basic Audio Tester)\fP is a simple command\-line utility +intended to help automate audio driver and sound server testing with little +human interaction. ALSABAT can be used to test audio quality, stress test +features and test audio before and after PM state changes. + +ALSABAT's design is relatively simple. ALSABAT plays an audio stream and +captures the same stream in either a digital or analog loop back. It then +compares the captured stream using a FFT to the original to determine if +the test case passes or fails. + +ALSABAT can either run wholly on the target machine being tested (standalone +mode) or can run as a client/server mode where by alsabat client runs on the +target and runs as a server on a separate tester machine. The client/server +mode still requires some manual interaction for synchronization, but this +is actively being developed for future releases. + +The hardware testing configuration may require the use of an analog cable +connecting target to tester machines or a cable to create an analog +loopback if no loopback mode is not available on the sound hardware that +is being tested. +An analog loopback cable can be used to connect the "line in" to "line out" +jacks to create a loopback. If only headphone and mic jacks (or combo jack) +are available then the following simple circuit can be used to create an +analog loopback :- + +https://source.android.com/devices/audio/loopback.html + +.SH OPTIONS +.TP +\fI\-h, \-\-help\fP +Help: show syntax. +.TP +\fI\-D\fP +Select sound card to be tested by name. +.TP +\fI\-P\fP +Select the playback PCM device. +.TP +\fI\-C\fP +Select the capture PCM device. +.TP +\fI\-f\fP +Sample format +.br +Recognized sample formats are: U8 S16_LE S24_3LE S32_LE +.br +Some of these may not be available on selected hardware +.br +The available format shortcuts are: +.nf +\-f cd (16 bit little endian, 44100, stereo) [\-f S16_LE \-c2 \-r44100] +\-f dat (16 bit little endian, 48000, stereo) [\-f S16_LE \-c2 \-r48000] +.fi +If no format is given S16_LE is used. +.TP +\fI\-c\fP +The number of channels. The default is one channel. +Valid values at the moment are 1 or 2. +.TP +\fI\-r\fP +Sampling rate in Hertz. The default rate is 44100 Hertz. +Valid values depends on hardware support. +.TP +\fI\-n\fP +Duration of generated signal. +The value could be either of the two forms: +.br +1. Decimal integer, means number of frames; +.br +2. Floating point with suffix 's', means number of seconds. +.br +The default is 2 seconds. +.TP +\fI\-k\fP +Sigma k value for analysis. +.br +The analysis function reads data from WAV file, run FFT against the data +to get magnitude of frequency vectors, and then calculates the average +value and standard deviation of frequency vectors. After that, we define +a threshold: +.br +threshold = k * standard_deviation + mean_value +.br +Frequencies with amplitude larger than threshold will be recognized as a +peak, and the frequency with largest peak value will be recognized as a +detected frequency. +.br +ALSABAT then compares the detected frequency to target frequency, to +decide if the detecting passes or fails. +.br +The default value is 3.0. +.TP +\fI\-F\fP +Target frequency for signal generation and analysis, in Hertz. +The default is 997.0 Hertz. +Valid range is (DC_THRESHOLD, 40% * Sampling rate). +.TP +\fI\-p\fP +Total number of periods to play or capture. +.TP +\fI\-\-log=#\fP +Write stderr and stdout output to this log file. +.TP +\fI\-\-file=#\fP +Input WAV file for playback. +.TP +\fI\-\-saveplay=#\fP +Target WAV file to save capture test content. +.TP +\fI\-\-local\fP +Internal loopback mode. +Playback, capture and analysis internal to ALSABAT only. This is intended +for developers to test new ALSABAT features as no audio is routed outside +of ALSABAT. + +.SH EXAMPLES + +.TP +\fBalsabat \-P plughw:0,0 \-C plughw:0,0 \-c 2 \-f S32_LE \-F 250\fR +Generate and play a sine wave of 250 Hertz with 2 channel and S32_LE format, +and then capture and analyze. + +.TP +\fBalsabat \-P plughw:0,0 \-C plughw:0,0 \-\-file 500Hz.wav\fR +Play the RIFF WAV file "500Hz.wav" which contains 500 Hertz waveform LPCM +data, and then capture and analyze. + +.SH RETURN VALUE +.br +On success, returns 0. +.br +If no peak be detected, returns -1001; +.br +If only DC be detected, returns -1002; +.br +If peak frequency does not match with the target frequency, returns -1003. + +.SH SEE ALSO +\fB +aplay(1) +\fP + +.SH BUGS +Currently only support RIFF WAV format with PCM data. Please report any bugs to +the alsa-devel mailing list. + +.SH AUTHOR +\fBalsabat\fP is by Liam Girdwood <liam.r.girdwood(a)linux.intel.com>, Bernard +Gautier <bernard.gautier(a)intel.com> and Han Lu <han.lu(a)intel.com>. +This document is by Liam Girdwood <liam.r.girdwood(a)linux.intel.com> and Han Lu +<han.lu(a)intel.com>. diff --git a/bat/bat.1 b/bat/bat.1 deleted file mode 100644 index e00fc27..0000000 --- a/bat/bat.1 +++ /dev/null @@ -1,158 +0,0 @@ -.TH BAT 1 "20th October 2015" -.SH NAME -bat \- command\-line sound tester for ALSA sound card driver - -.SH SYNOPSIS -\fBbat\fP [\fIflags\fP] - -.SH DESCRIPTION -\fBBAT(Basic Audio Tester)\fP is a simple command\-line utility intended -to help automate audio driver and sound server testing with little human -interaction. BAT can be used to test audio quality, stress test features -and test audio before and after PM state changes. - -BAT's design is relatively simple. BAT plays an audio stream and captures -the same stream in either a digital or analog loop back. It then compares -the captured stream using a FFT to the original to determine if the test -case passes or fails. - -BAT can either run wholly on the target machine being tested (standalone -mode) or can run as a client/server mode where by bat client runs on the -target and runs as a server on a separate tester machine. The client/server -mode still requires some manual interaction for synchronization, but this -is actively being developed for future releases. - -The hardware testing configuration may require the use of an analog cable -connecting target to tester machines or a cable to create an analog -loopback if no loopback mode is not available on the sound hardware that -is being tested. -An analog loopback cable can be used to connect the "line in" to "line out" -jacks to create a loopback. If only headphone and mic jacks (or combo jack) -are available then the following simple circuit can be used to create an -analog loopback :- - -https://source.android.com/devices/audio/loopback.html - -.SH OPTIONS -.TP -\fI\-h, \-\-help\fP -Help: show syntax. -.TP -\fI\-D\fP -Select sound card to be tested by name. -.TP -\fI\-P\fP -Select the playback PCM device. -.TP -\fI\-C\fP -Select the capture PCM device. -.TP -\fI\-f\fP -Sample format -.br -Recognized sample formats are: U8 S16_LE S24_3LE S32_LE -.br -Some of these may not be available on selected hardware -.br -The available format shortcuts are: -.nf -\-f cd (16 bit little endian, 44100, stereo) [\-f S16_LE \-c2 \-r44100] -\-f dat (16 bit little endian, 48000, stereo) [\-f S16_LE \-c2 \-r48000] -.fi -If no format is given S16_LE is used. -.TP -\fI\-c\fP -The number of channels. The default is one channel. -Valid values at the moment are 1 or 2. -.TP -\fI\-r\fP -Sampling rate in Hertz. The default rate is 44100 Hertz. -Valid values depends on hardware support. -.TP -\fI\-n\fP -Duration of generated signal. -The value could be either of the two forms: -.br -1. Decimal integer, means number of frames; -.br -2. Floating point with suffix 's', means number of seconds. -.br -The default is 2 seconds. -.TP -\fI\-k\fP -Sigma k value for analysis. -.br -The analysis function reads data from WAV file, run FFT against the data -to get magnitude of frequency vectors, and then calculates the average -value and standard deviation of frequency vectors. After that, we define -a threshold: -.br -threshold = k * standard_deviation + mean_value -.br -Frequencies with amplitude larger than threshold will be recognized as a -peak, and the frequency with largest peak value will be recognized as a -detected frequency. -.br -BAT then compares the detected frequency to target frequency, to decide -if the detecting passes or fails. -.br -The default value is 3.0. -.TP -\fI\-F\fP -Target frequency for signal generation and analysis, in Hertz. -The default is 997.0 Hertz. -Valid range is (DC_THRESHOLD, 40% * Sampling rate). -.TP -\fI\-p\fP -Total number of periods to play or capture. -.TP -\fI\-\-log=#\fP -Write stderr and stdout output to this log file. -.TP -\fI\-\-file=#\fP -Input WAV file for playback. -.TP -\fI\-\-saveplay=#\fP -Target WAV file to save capture test content. -.TP -\fI\-\-local\fP -Internal loopback mode. -Playback, capture and analysis internal to BAT only. This is intended for -developers to test new BAT features as no audio is routed outside of BAT. - -.SH EXAMPLES - -.TP -\fBbat \-P plughw:0,0 \-C plughw:0,0 \-c 2 \-f S32_LE \-F 250\fR -Generate and play a sine wave of 250 Hertz with 2 channel and S32_LE format, -and then capture and analyze. - -.TP -\fBbat \-P plughw:0,0 \-C plughw:0,0 \-\-file 500Hz.wav\fR -Play the RIFF WAV file "500Hz.wav" which contains 500 Hertz waveform LPCM -data, and then capture and analyze. - -.SH RETURN VALUE -.br -On success, returns 0. -.br -If no peak be detected, returns -1001; -.br -If only DC be detected, returns -1002; -.br -If peak frequency does not match with the target frequency, returns -1003. - -.SH SEE ALSO -\fB -aplay(1) -\fP - -.SH BUGS -Currently only support RIFF WAV format with PCM data. Please report any bugs to -the alsa-devel mailing list. - -.SH AUTHOR -\fBbat\fP is by Liam Girdwood <liam.r.girdwood(a)linux.intel.com>, Bernard Gautier -<bernard.gautier(a)intel.com> and Han Lu <han.lu(a)intel.com>. -This document is by Liam Girdwood <liam.r.girdwood(a)linux.intel.com> and Han Lu -<han.lu(a)intel.com>. diff --git a/bat/bat.c b/bat/bat.c index 086b9fa..ddb60b7 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -272,7 +272,7 @@ static void test_capture(struct bat *bat) static void usage(struct bat *bat) { fprintf(bat->log, -_("Usage: bat [-options]...\n" +_("Usage: alsabat [-options]...\n" "\n" " -h, --help this help\n" " -D pcm device for both playback and capture\n" -- 2.5.0

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[alsa-devel] [PATCH] ALSA: hda - build chmap kctl based on pcm in hdmi audio
by libin.yang＠linux.intel.com 03 Feb '16

03 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> Build chmap kctl based on pcm. The first chmap kctl will be mapped to the first pcm, and so on. When a monitor is connected to a pin, the chmap kctl can find the pin and the monitor through the pcm index. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 40 +++++++++++++++++++++++++++++++++------- 1 file changed, 33 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 275b68a..1a46f45 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -437,6 +437,20 @@ static int hinfo_to_pin_index(struct hda_codec *codec, return -EINVAL; } +static struct hdmi_spec_per_pin *pcm_idx_to_pin(struct hdmi_spec *spec, + int pcm_idx) +{ + int i; + struct hdmi_spec_per_pin *per_pin; + + for (i = 0; i < spec->num_pins; i++) { + per_pin = get_pin(spec, i); + if (per_pin->pcm_idx == pcm_idx) + return per_pin; + } + return NULL; +} + static int cvt_nid_to_cvt_index(struct hda_codec *codec, hda_nid_t cvt_nid) { struct hdmi_spec *spec = codec->spec; @@ -2398,10 +2412,16 @@ static int hdmi_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; - int pin_idx = kcontrol->private_value; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + int pcm_idx = kcontrol->private_value; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); int i; + if (!per_pin) { + for (i = 0; i < spec->channels_max; i++) + ucontrol->value.integer.value[i] = 0; + return 0; + } + for (i = 0; i < ARRAY_SIZE(per_pin->chmap); i++) ucontrol->value.integer.value[i] = per_pin->chmap[i]; return 0; @@ -2413,13 +2433,19 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; - int pin_idx = kcontrol->private_value; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + int pcm_idx = kcontrol->private_value; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); unsigned int ctl_idx; struct snd_pcm_substream *substream; unsigned char chmap[8]; int i, err, ca, prepared = 0; + /* No monitor is connected in dyn_pcm_assign. + * It's invalid to setup the chmap + */ + if (!per_pin) + return 0; + ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); substream = snd_pcm_chmap_substream(info, ctl_idx); if (!substream || !substream->runtime) @@ -2596,18 +2622,18 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) } /* add channel maps */ - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + for (pcm_idx = 0; pcm_idx < spec->pcm_used; pcm_idx++) { struct hda_pcm *pcm; struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; - pcm = get_pcm_rec(spec, pin_idx); + pcm = get_pcm_rec(spec, pcm_idx); if (!pcm || !pcm->pcm) break; err = snd_pcm_add_chmap_ctls(pcm->pcm, SNDRV_PCM_STREAM_PLAYBACK, - NULL, 0, pin_idx, &chmap); + NULL, 0, pcm_idx, &chmap); if (err < 0) return err; /* override handlers */ -- 1.9.1

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Re: [alsa-devel] [sound:topic/core-fixes 21/21] sound/core/seq/seq_virmidi.c:192:6-12: preceding lock on line 176
by Julia Lawall 03 Feb '16

03 Feb '16

Looks suspicious, please check. julia On Wed, 3 Feb 2016, kbuild test robot wrote: > CC: kbuild-all(a)01.org > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/core-fixes > head: be33492635ee8fc19a185f37293b533069e5cc8f > commit: be33492635ee8fc19a185f37293b533069e5cc8f [21/21] ALSA: rawmidi: Make snd_rawmidi_transmit() race-free > :::::: branch date: 2 hours ago > :::::: commit date: 2 hours ago > > >> sound/core/seq/seq_virmidi.c:192:6-12: preceding lock on line 176 > > git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git > git remote update sound > git checkout be33492635ee8fc19a185f37293b533069e5cc8f > vim +192 sound/core/seq/seq_virmidi.c > > ^1da177e Linus Torvalds 2005-04-16 170 if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { > 62c5549e Takashi Iwai 2006-02-22 171 if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) > ^1da177e Linus Torvalds 2005-04-16 172 return; > ^1da177e Linus Torvalds 2005-04-16 173 vmidi->event.type = SNDRV_SEQ_EVENT_NONE; > ^1da177e Linus Torvalds 2005-04-16 174 } > ^1da177e Linus Torvalds 2005-04-16 175 while (1) { > be334926 Takashi Iwai 2016-01-31 @176 spin_lock_irqsave(&substream->runtime->lock, flags); > be334926 Takashi Iwai 2016-01-31 177 count = __snd_rawmidi_transmit_peek(substream, buf, sizeof(buf)); > ^1da177e Linus Torvalds 2005-04-16 178 if (count <= 0) > ^1da177e Linus Torvalds 2005-04-16 179 break; > ^1da177e Linus Torvalds 2005-04-16 180 pbuf = buf; > ^1da177e Linus Torvalds 2005-04-16 181 while (count > 0) { > ^1da177e Linus Torvalds 2005-04-16 182 res = snd_midi_event_encode(vmidi->parser, pbuf, count, &vmidi->event); > ^1da177e Linus Torvalds 2005-04-16 183 if (res < 0) { > ^1da177e Linus Torvalds 2005-04-16 184 snd_midi_event_reset_encode(vmidi->parser); > ^1da177e Linus Torvalds 2005-04-16 185 continue; > ^1da177e Linus Torvalds 2005-04-16 186 } > be334926 Takashi Iwai 2016-01-31 187 __snd_rawmidi_transmit_ack(substream, res); > ^1da177e Linus Torvalds 2005-04-16 188 pbuf += res; > ^1da177e Linus Torvalds 2005-04-16 189 count -= res; > ^1da177e Linus Torvalds 2005-04-16 190 if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { > 62c5549e Takashi Iwai 2006-02-22 191 if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) > ^1da177e Linus Torvalds 2005-04-16 @192 return; > ^1da177e Linus Torvalds 2005-04-16 193 vmidi->event.type = SNDRV_SEQ_EVENT_NONE; > ^1da177e Linus Torvalds 2005-04-16 194 } > ^1da177e Linus Torvalds 2005-04-16 195 } > > :::::: The code at line 192 was first introduced by commit > :::::: 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Linux-2.6.12-rc2 > > :::::: TO: Linus Torvalds <torvalds(a)ppc970.osdl.org> > :::::: CC: Linus Torvalds <torvalds(a)ppc970.osdl.org> > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation >

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: Atom: fix regression on compress DAI
by Pierre-Louis Bossart 03 Feb '16

03 Feb '16

Commit a106804 ("ASoC: compress: Fix compress device direction check") added a dependency on the compress-cpu-dai channel_min field which was removed earlier by commit 77095796 ("ASoC: Intel: Atom: clean-up compressed DAI definition") as part of the baytrail cleanups. The net result was a regression at probe on all Atom platforms with no sound card created. Fix by adding explicit initialization for channel_min to 1 for the compress-cpu-dai. Reported-by: Tobias Mädel <alsa-devel(a)tbspace.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- This should be applied before 4.5 is released sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 55c33dc..52ed434 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -528,6 +528,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .ops = &sst_compr_dai_ops, .playback = { .stream_name = "Compress Playback", + .channels_min = 1, }, }, /* BE CPU Dais */ -- 1.9.1

2 1

[alsa-devel] [PATCH v5 0/8] Add the family patches to support for kylin board
by Caesar Wang 03 Feb '16

03 Feb '16

Hi all, This series patches are based on kernel 4.5-rc1 version, also is based on Heiko's branch. (kernel/git/mmind/linux-rockchip.git -b for-next & v4.5-clk/fixes) e.g: boot log.. Linux version 4.5.0-rc1+ (wxt@ubuntu) ....#60 SMP Thu Jan 28 12:54:50 CST 2016 1) The kylin history series patches: V0: https://lkml.org/lkml/2015/12/22/168 V1: https://lkml.org/lkml/2015/12/17/319 V2: https://lkml.org/lkml/2016/1/7/66 V3: http://www.spinics.net/lists/arm-kernel/msg474796.html V4: https://lkml.org/lkml/2016/1/28/132 This series patches are verified on the following github: https://github.com/Caesar-github/rockchip/tree/kylin/next This series has 8 patches: (8--->1) c455bf5 ARM: dts: rockchip: support the spi for rk3036 41bbd5d ASoC: rt5616: add the mclk for the codec driver 6ee2dc0 ASoC: rt5616: trivial: fix the typo 2775e41 ASoC: rt5616: add mclk property for rt5616 document b4ec10b ARM: dts: rockchip: add mclk for rt5616 on kylin board 8d272a1 ARM: dts: rockchip: add support emac for RK3036 02e501f clk: rockchip: rk3036: fix and add node id for emac clock b948f42 ARM: dts: rockchip: add hdmi/vop device node for rk3036 .... This series patches can apply into Heiko for-next branch. (https://git.kernel.org/cgit/linux/kernel/git/mmind/linux-rockchip.git/log/?…) Also, ths ASoC can apply into Mark branch. --- 2) This series patches have the following decriptions: PATCH[1/8]: ====> add hdmi/vop display node for rk3036 ARM: dts: rockchip: add hdmi/vop device node for rk3036 This Patch is based on Ykk's Inno HDMI V7 patches. That's verified on my github. bd5ed3e ARM: dts: rockchip: add hdmi/vop device node for rk3036 438e7bd FROMLIST: dt-bindings: add document for Innosilicon HDMI on Rockchip platform ((am from https://patchwork.kernel.org/patch/8117611/)) ad0623b FROMLIST: drm/rockchip: hdmi: add Innosilicon HDMI support ((am https://patchwork.kernel.org/patch/8117571/)) PATCH[2/8-3/8]: Add the emac patch The emac clock is depend APLL, That shoud be incorrect. We should fix it on the DPLL. df91ab ARM: dts: rockchip: add support emac for RK3036 d5c2781 clk: rockchip: rk3036: fix and add node id for emac clock That's verified on kylin board with ubuntu os. How to test and verify? You can refer to the following wiki document. http://rockchip.wikidot.com/linux-develop-guide PATCH[4/8-7/8]: Make the codec work on kylin board 41bbd5d ASoC: rt5616: add the mclk for the codec driver 6ee2dc0 ASoC: rt5616: trivial: fix the typo 2775e41 ASoC: rt5616: add mclk property for rt5616 document b4ec10b ARM: dts: rockchip: add mclk for rt5616 on kylin board As the previous discussed on https://patchwork.kernel.org/patch/8041001/ Move the mclk into codec driver to enable and disable. How to test and verify? You can refer to the following wiki document. http://rockchip.wikidot.com/linux-develop-guide PATCH[8/8]: Make the codec work on kylin board Add the spi node for rk3036 dts, that's seem not related to the kylin. Maybe the future will support the spi on kylin board or connect the spi devices. f0fff43 ARM: dts: rockchip: support the spi for rk3036 --- Thanks your reviewing! :) - Caesar Changes in v5: - remove unused commit content. - Fix the wrong word, as the Rob comments on https://patchwork.kernel.org/patch/8147731/. - Cc the related alsa experts. -Cc releated alsa experts. - Cc related alsa experts. - Remove the unused, as Heiko comments on https://patchwork.kernel.org/patch/8147761/ - Modify the commit. Changes in v4: - solve the lastest conflict, picked up from https://patchwork.kernel.org/patch/8040961/. - fix the commit, pick up from the https://patchwork.kernel.org/patch/7976631/. - The emac parent shouldn't depend on the APLL. instead of DPLL. - included in the kylin series patches. - This patch picked up from https://patchwork.kernel.org/patch/7924971/ - Change to solve the conflict based on the Heiko's branch. - Make the emac parent as the DPLL. - AS the previous discussed by them, add the mclk for codec. (https://patchwork.kernel.org/patch/8041001/) - Add this patch included in kylin series patches. - Add this patch included in kylin series patches. - AS the previous discussed by them, add the mclk for codec. (https://patchwork.kernel.org/patch/8041001/) - Add this patch included in kylin series patches. Caesar Wang (6): ARM: dts: rockchip: add hdmi/vop device node for rk3036 ARM: dts: rockchip: add mclk for rt5616 on kylin board ASoC: rt5616: add mclk property for rt5616 document ASoC: rt5616: trivial: fix the typo ASoC: rt5616: add the mclk for the codec driver ARM: dts: rockchip: support the spi for rk3036 zhengxing (2): clk: rockchip: rk3036: fix and add node id for emac clock ARM: dts: rockchip: add support emac for RK3036 Documentation/devicetree/bindings/sound/rt5616.txt | 6 + arch/arm/boot/dts/rk3036-evb.dts | 23 ++ arch/arm/boot/dts/rk3036-kylin.dts | 35 ++ arch/arm/boot/dts/rk3036.dtsi | 145 ++++++++ drivers/clk/rockchip/clk-rk3036.c | 9 +- include/dt-bindings/clock/rk3036-cru.h | 2 + sound/soc/codecs/rt5616.c | 410 +++++++++++---------- 7 files changed, 442 insertions(+), 188 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Add missing const from region definitions
by Charles Keepax 02 Feb '16

02 Feb '16

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 653dbff..28d3851 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -325,7 +325,7 @@ struct wm_adsp_buffer_region_def { unsigned int size_offset; }; -static struct wm_adsp_buffer_region_def default_regions[] = { +static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, .base_offset = HOST_BUFFER_FIELD(X_buf_base), @@ -347,7 +347,7 @@ struct wm_adsp_fw_caps { u32 id; struct snd_codec_desc desc; int num_regions; - struct wm_adsp_buffer_region_def *region_defs; + const struct wm_adsp_buffer_region_def *region_defs; }; static const struct wm_adsp_fw_caps ctrl_caps[] = { -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: dwc: Ensure i2s_reg_comp{1, 2} is always initialised
by Jon Medhurst (Tixy) 02 Feb '16

02 Feb '16

In the case that the driver is configured from device-tree i2s_reg_comp1 and i2s_reg_comp2 aren't initialised, breaking the driver. Fix this by unconditionally setting these values before checking for quirks. Fixes: a242cac1d3aa ("ASoC: dwc: add quirk to override COMP_PARAM_1 register") Signed-off-by: Jon Medhurst <tixy(a)linaro.org> --- sound/soc/dwc/designware_i2s.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index d8afd8e..5f4e6aa 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -636,6 +636,8 @@ static int dw_i2s_probe(struct platform_device *pdev) dev->dev = &pdev->dev; + dev->i2s_reg_comp1 = I2S_COMP_PARAM_1; + dev->i2s_reg_comp2 = I2S_COMP_PARAM_2; if (pdata) { dev->capability = pdata->cap; clk_id = NULL; @@ -643,9 +645,6 @@ static int dw_i2s_probe(struct platform_device *pdev) if (dev->quirks & DW_I2S_QUIRK_COMP_REG_OFFSET) { dev->i2s_reg_comp1 = pdata->i2s_reg_comp1; dev->i2s_reg_comp2 = pdata->i2s_reg_comp2; - } else { - dev->i2s_reg_comp1 = I2S_COMP_PARAM_1; - dev->i2s_reg_comp2 = I2S_COMP_PARAM_2; } ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata); } else { -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: dpcm: fix the BE state on hw_free
by Vinod Koul 02 Feb '16

02 Feb '16

While performing hw_free, DPCM checks the BE state but leaves out the suspend state. The suspend state needs to be checked as well, as we might be suspended and then usermode closes rather than resuming the audio stream. This was found by a stress testing of system with playback in loop and killed after few seconds running in background and second script running suspend-resume test in loop Cc: stable(a)vger.kernel.org Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index bd5d3757acf6..659a7e9b2b81 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1753,7 +1753,8 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) continue; dev_dbg(be->dev, "ASoC: hw_free BE %s\n", -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Return sample rate in wm_adsp_compr_pointer
by Charles Keepax 02 Feb '16

02 Feb '16

We should return a valid sample rate from the pointer callback, this patch adds this into the driver. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 28d3851..ed90e12 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -292,6 +292,8 @@ struct wm_adsp_compr { u32 *raw_buf; unsigned int copied_total; + + unsigned int sample_rate; }; #define WM_ADSP_DATA_WORD_SIZE 3 @@ -2484,6 +2486,8 @@ int wm_adsp_compr_set_params(struct snd_compr_stream *stream, if (!compr->raw_buf) return -ENOMEM; + compr->sample_rate = params->codec.sample_rate; + return 0; } EXPORT_SYMBOL_GPL(wm_adsp_compr_set_params); @@ -2923,6 +2927,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, tstamp->copied_total = compr->copied_total; tstamp->copied_total += buf->avail * WM_ADSP_DATA_WORD_SIZE; + tstamp->sampling_rate = compr->sample_rate; out: mutex_unlock(&dsp->pwr_lock); -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Atom: fix regression on compress DAI
by Pierre-Louis Bossart 02 Feb '16

02 Feb '16

Commit a1068045883ed4a18363a4ebad0c3d55e473b716 added a dependency on the compress-cpu-dai channel_min field which was removed earlier by commit 77095796ae9cbaf315f80611edb5aa569796e339 as part of the baytrail cleanups. The net result was a regression at probe on all Atom platforms with no sound card created. Fix by adding explicit initialization for channel_min to 1 for the compress-cpu-dai. Reported-by: Tobias Mädel <alsa-devel(a)tbspace.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 55c33dc..52ed434 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -528,6 +528,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .ops = &sst_compr_dai_ops, .playback = { .stream_name = "Compress Playback", + .channels_min = 1, }, }, /* BE CPU Dais */ -- 1.9.1

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Re: [alsa-devel] sound: use-after-free in snd_seq_deliver_single_event
by Mark Brown 02 Feb '16

02 Feb '16

On Mon, Feb 01, 2016 at 10:34:23PM +0900, Takashi Sakamoto wrote: > Well, I don't mind to receive them because it's a good information for > me to get current state of ALSA core functionality. If I had some > spare time, I was also going to fix it. But currently I have little > time for it due to my development for my minor hardwares... Yeah, my concern is that I'm getting to the point where I mentally tag them as irrelevant and delete them unread (the subject lines don't help here). I've had quite a few and none of them have been for code I even recall working on.

1 0

[alsa-devel] [PATCH] ALSA: timer: Sync timer deletion at closing the system timer
by Takashi Iwai 02 Feb '16

02 Feb '16

ALSA timer core framework has no sync point at stopping because it's called inside the spinlock. Thus we need a sync point at close for avoiding the stray timer task. This is simply done by implementing the close callback just calling del_timer_sync(). (It's harmless to call it unconditionally, as the core timer itself cares of the already deleted timer instance.) Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/core/timer.c b/sound/core/timer.c index 12db60dd147b..b419e612f987 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1058,11 +1058,21 @@ static int snd_timer_s_stop(struct snd_timer * timer) return 0; } +static int snd_timer_s_close(struct snd_timer *timer) +{ + struct snd_timer_system_private *priv; + + priv = (struct snd_timer_system_private *)timer->private_data; + del_timer_sync(&priv->tlist); + return 0; +} + static struct snd_timer_hardware snd_timer_system = { .flags = SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_TASKLET, .resolution = 1000000000L / HZ, .ticks = 10000000L, + .close = snd_timer_s_close, .start = snd_timer_s_start, .stop = snd_timer_s_stop }; -- 2.7.0

1 0

[alsa-devel] [PATCH for-4.6] ALSA: dummy: Implement timer backend switching more safely
by Takashi Iwai 02 Feb '16

02 Feb '16

Currently the selected timer backend is referred at any moment from the running PCM callbacks. When the backend is switched, it's possible to lead to inconsistency from the running backend. This was pointed by syzkaller fuzzer, and the commit [7ee96216c31a: ALSA: dummy: Disable switching timer backend via sysfs] disabled the dynamic switching for avoiding the crash. This patch improves the handling of timer backend switching. It keeps the reference to the selected backend during the whole operation of an opened stream so that it won't be changed by other streams. Together with this change, the hrtimer parameter is reenabled as writable now. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a patch for 4.6, not meant for 4.5 as a fix for now. sound/drivers/dummy.c | 37 +++++++++++++++++++------------------ 1 file changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index bde33308f0d6..c0f8f613f1f1 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -87,7 +87,7 @@ MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); module_param(fake_buffer, bool, 0444); MODULE_PARM_DESC(fake_buffer, "Fake buffer allocations."); #ifdef CONFIG_HIGH_RES_TIMERS -module_param(hrtimer, bool, 0444); +module_param(hrtimer, bool, 0644); MODULE_PARM_DESC(hrtimer, "Use hrtimer as the timer source."); #endif @@ -109,6 +109,9 @@ struct dummy_timer_ops { snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *); }; +#define get_dummy_ops(substream) \ + (*(const struct dummy_timer_ops **)(substream)->runtime->private_data) + struct dummy_model { const char *name; int (*playback_constraints)(struct snd_pcm_runtime *runtime); @@ -137,7 +140,6 @@ struct snd_dummy { int iobox; struct snd_kcontrol *cd_volume_ctl; struct snd_kcontrol *cd_switch_ctl; - const struct dummy_timer_ops *timer_ops; }; /* @@ -231,6 +233,8 @@ static struct dummy_model *dummy_models[] = { */ struct dummy_systimer_pcm { + /* ops must be the first item */ + const struct dummy_timer_ops *timer_ops; spinlock_t lock; struct timer_list timer; unsigned long base_time; @@ -366,6 +370,8 @@ static const struct dummy_timer_ops dummy_systimer_ops = { */ struct dummy_hrtimer_pcm { + /* ops must be the first item */ + const struct dummy_timer_ops *timer_ops; ktime_t base_time; ktime_t period_time; atomic_t running; @@ -492,31 +498,25 @@ static const struct dummy_timer_ops dummy_hrtimer_ops = { static int dummy_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_dummy *dummy = snd_pcm_substream_chip(substream); - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - return dummy->timer_ops->start(substream); + return get_dummy_ops(substream)->start(substream); case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - return dummy->timer_ops->stop(substream); + return get_dummy_ops(substream)->stop(substream); } return -EINVAL; } static int dummy_pcm_prepare(struct snd_pcm_substream *substream) { - struct snd_dummy *dummy = snd_pcm_substream_chip(substream); - - return dummy->timer_ops->prepare(substream); + return get_dummy_ops(substream)->prepare(substream); } static snd_pcm_uframes_t dummy_pcm_pointer(struct snd_pcm_substream *substream) { - struct snd_dummy *dummy = snd_pcm_substream_chip(substream); - - return dummy->timer_ops->pointer(substream); + return get_dummy_ops(substream)->pointer(substream); } static struct snd_pcm_hardware dummy_pcm_hardware = { @@ -562,17 +562,19 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) struct snd_dummy *dummy = snd_pcm_substream_chip(substream); struct dummy_model *model = dummy->model; struct snd_pcm_runtime *runtime = substream->runtime; + const struct dummy_timer_ops *ops; int err; - dummy->timer_ops = &dummy_systimer_ops; + ops = &dummy_systimer_ops; #ifdef CONFIG_HIGH_RES_TIMERS if (hrtimer) - dummy->timer_ops = &dummy_hrtimer_ops; + ops = &dummy_hrtimer_ops; #endif - err = dummy->timer_ops->create(substream); + err = ops->create(substream); if (err < 0) return err; + get_dummy_ops(substream) = ops; runtime->hw = dummy->pcm_hw; if (substream->pcm->device & 1) { @@ -594,7 +596,7 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) err = model->capture_constraints(substream->runtime); } if (err < 0) { - dummy->timer_ops->free(substream); + get_dummy_ops(substream)->free(substream); return err; } return 0; @@ -602,8 +604,7 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) static int dummy_pcm_close(struct snd_pcm_substream *substream) { - struct snd_dummy *dummy = snd_pcm_substream_chip(substream); - dummy->timer_ops->free(substream); + get_dummy_ops(substream)->free(substream); return 0; } -- 2.7.0

1 0

[alsa-devel] [PATCH] ALSA: hda - chmap supports dynamic pcm assignment
by libin.yang＠linux.intel.com 02 Feb '16

02 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> Bind chmap to pcm statically. chmap is now mapped to pin dynamically based on mapping between pcm and pin. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 40 +++++++++++++++++++++++++++++++++------- 1 file changed, 33 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 275b68a..1a46f45 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -437,6 +437,20 @@ static int hinfo_to_pin_index(struct hda_codec *codec, return -EINVAL; } +static struct hdmi_spec_per_pin *pcm_idx_to_pin(struct hdmi_spec *spec, + int pcm_idx) +{ + int i; + struct hdmi_spec_per_pin *per_pin; + + for (i = 0; i < spec->num_pins; i++) { + per_pin = get_pin(spec, i); + if (per_pin->pcm_idx == pcm_idx) + return per_pin; + } + return NULL; +} + static int cvt_nid_to_cvt_index(struct hda_codec *codec, hda_nid_t cvt_nid) { struct hdmi_spec *spec = codec->spec; @@ -2398,10 +2412,16 @@ static int hdmi_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; - int pin_idx = kcontrol->private_value; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + int pcm_idx = kcontrol->private_value; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); int i; + if (!per_pin) { + for (i = 0; i < spec->channels_max; i++) + ucontrol->value.integer.value[i] = 0; + return 0; + } + for (i = 0; i < ARRAY_SIZE(per_pin->chmap); i++) ucontrol->value.integer.value[i] = per_pin->chmap[i]; return 0; @@ -2413,13 +2433,19 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; - int pin_idx = kcontrol->private_value; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + int pcm_idx = kcontrol->private_value; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); unsigned int ctl_idx; struct snd_pcm_substream *substream; unsigned char chmap[8]; int i, err, ca, prepared = 0; + /* No monitor is connected in dyn_pcm_assign. + * It's invalid to setup the chmap + */ + if (!per_pin) + return 0; + ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); substream = snd_pcm_chmap_substream(info, ctl_idx); if (!substream || !substream->runtime) @@ -2596,18 +2622,18 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) } /* add channel maps */ - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + for (pcm_idx = 0; pcm_idx < spec->pcm_used; pcm_idx++) { struct hda_pcm *pcm; struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; - pcm = get_pcm_rec(spec, pin_idx); + pcm = get_pcm_rec(spec, pcm_idx); if (!pcm || !pcm->pcm) break; err = snd_pcm_add_chmap_ctls(pcm->pcm, SNDRV_PCM_STREAM_PLAYBACK, - NULL, 0, pin_idx, &chmap); + NULL, 0, pcm_idx, &chmap); if (err < 0) return err; /* override handlers */ -- 1.9.1

3 2

[alsa-devel] bytcr_rt5651 on wintron 7.0
by Tobias Mädel 02 Feb '16

02 Feb '16

Hello, we're currently trying to get the new bytcr_rt5651 driver to run on a TrekStor SurfTab wintron 7.0 (x86 tablet). Unlike the previous driver for the rt5640 chipset, the new bytcr_rt5651 driver actually detects the card out-of-the-box. This is the dmesg output on the device: bytcr_rt5651: snd-soc-dummy-dai <-> media-cpu-dai mapping ok bytcr_rt5651: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok bytcr_rt5651: Invalid direction for compress P 0, C 0 bytcr_rt5651: ASoC: can't create compress Compress bytcr_rt5651: ASoC: failed to instantiate card -22 bytcr_rt5651: devm_snd_soc_register_card failed -22 bytcr_rt5651: probe of bytcr_rt5651 failed with error -22 So -- apparently it is successfully initializing 2 of the DAIs, but it can't find the Compress-DAI. My first guess would just be to try and disable the Compress DAI in the code just for debugging, but I'm sure that there's a better way. I'm running a bleeding edge kernel build from the main linux git tree, build from yesterday's github.com/torvalds/linux.git repo. Does anyone have any ideas on how to get alsa to detect the card properly? Here's the complete dmesg output: https://gist.github.com/Manawyrm/70d90e95e9c578a7fb26 Attached is a (binary) dump of the ACPI tables of the device. Best regards, Tobias Schramm and Tobias Mädel -- This email was encrypted with 2ROT-13. Tobias Mädel t.maedel(a)alfeld.de PGP: 3A69 E9AF 3FA9 3BCF FE50 9694 494E 2F56 C304 60E1 http://tbspace.de

3 3

[alsa-devel] [PATCH] ASoC: fsl_ssi: remove explicit register defaults
by Maciej S. Szmigiero 01 Feb '16

01 Feb '16

There is no guarantee that on fsl_ssi module load SSI registers will have their power-on-reset values. In fact, if the driver is reloaded the values in registers will be whatever they were set to previously. However, the cache needs to be fully populated at probe time to avoid non-atomic allocations during register access. Special case here is imx21-class SSI, since according to datasheet it don't have SACC{ST,EN,DIS} regs. This fixes hard lockup on fsl_ssi module reload, at least in AC'97 mode. Fixes: 05cf237972fe ("ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast") Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 20 deletions(-) This replaces "ASoC: fsl_ssi: remove register defaults" submission. It may be good to delay merging this until commit d51fe1f393a6 ("regmap: pass buffer size to regmap_raw_read() in regcache_hw_init()") reaches ASoC tree since without that regmap fix fsl_ssi will report error at probe time. diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 40dfd8a36484..ed8de1035cda 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,20 +112,6 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; -static const struct reg_default fsl_ssi_reg_defaults[] = { - {CCSR_SSI_SCR, 0x00000000}, - {CCSR_SSI_SIER, 0x00003003}, - {CCSR_SSI_STCR, 0x00000200}, - {CCSR_SSI_SRCR, 0x00000200}, - {CCSR_SSI_STCCR, 0x00040000}, - {CCSR_SSI_SRCCR, 0x00040000}, - {CCSR_SSI_SACNT, 0x00000000}, - {CCSR_SSI_STMSK, 0x00000000}, - {CCSR_SSI_SRMSK, 0x00000000}, - {CCSR_SSI_SACCEN, 0x00000000}, - {CCSR_SSI_SACCDIS, 0x00000000}, -}; - static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -190,8 +176,7 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .reg_defaults = fsl_ssi_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), + .num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1, .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg, @@ -201,6 +186,7 @@ static const struct regmap_config fsl_ssi_regconfig = { struct fsl_ssi_soc_data { bool imx; + bool imx21regs; /* imx21-class SSI - no SACC{ST,EN,DIS} regs */ bool offline_config; u32 sisr_write_mask; }; @@ -303,6 +289,7 @@ static struct fsl_ssi_soc_data fsl_ssi_mpc8610 = { static struct fsl_ssi_soc_data fsl_ssi_imx21 = { .imx = true, + .imx21regs = true, .offline_config = true, .sisr_write_mask = 0, }; @@ -586,8 +573,12 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) */ regmap_write(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV); - regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); - regmap_write(regs, CCSR_SSI_SACCEN, 0x300); + + /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ + if (!ssi_private->soc->imx21regs) { + regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); + regmap_write(regs, CCSR_SSI_SACCEN, 0x300); + } /* * Enable SSI, Transmit and Receive. AC97 has to communicate with the @@ -1397,6 +1388,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource *res; void __iomem *iomem; char name[64]; + struct regmap_config regconfig = fsl_ssi_regconfig; of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) @@ -1444,15 +1436,25 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(iomem); ssi_private->ssi_phys = res->start; + if (ssi_private->soc->imx21regs) { + /* + * According to datasheet imx21-class SSI + * don't have SACC{ST,EN,DIS} regs. + */ + regconfig.max_register = CCSR_SSI_SRMSK; + regconfig.num_reg_defaults_raw = + CCSR_SSI_SRMSK / sizeof(uint32_t) + 1; + } + ret = of_property_match_string(np, "clock-names", "ipg"); if (ret < 0) { ssi_private->has_ipg_clk_name = false; ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, - &fsl_ssi_regconfig); + &regconfig); } else { ssi_private->has_ipg_clk_name = true; ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, - "ipg", iomem, &fsl_ssi_regconfig); + "ipg", iomem, &regconfig); } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n");

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[alsa-devel] [PATCH REBASE 4.5 00/31] Sharing media resources across ALSA and au0828 drivers
by Shuah Khan 01 Feb '16

01 Feb '16

Please note that I am sending just the 4 patches that changed during series rebase to Linux 4.5. The rebase was pain free. media_device_init() type change required changes to patches 18 and 26. I re-tested the series and everything looks good. [PATCH 16/31] media: au0828 video remove au0828_enable_analog_tuner() [PATCH 18/31] media: au0828 change to use Managed Media Controller API [PATCH 22/31] media: dvb-core create tuner to demod pad link in disabled state [PATCH v3 26/31] sound/usb: Update ALSA driver to use Managed Media Controller API Links: PATCH Series that contains all 31 patches: https://lkml.org/lkml/2016/1/6/668 ALSA patches went through reviews and links to PATCH V3 (ALSA): https://lkml.org/lkml/2016/1/18/556 https://lkml.org/lkml/2016/1/18/557 https://lkml.org/lkml/2016/1/18/558 Compile warns patches: https://lkml.org/lkml/2016/1/19/520 https://lkml.org/lkml/2016/1/19/518 https://lkml.org/lkml/2016/1/19/519 Mauro and Takashi: Please let me know if you would rather see the entire patch series generated for 4.5-rc1 and sent out in a complete series instead of just the rebased patches. Mauro Carvalho Chehab (1): uapi/media.h: Declare interface types for ALSA Shuah Khan (30): media: Add ALSA Media Controller function entities media: Media Controller register/unregister entity_notify API media: Media Controller enable/disable source handler API media: Media Controller fix to not let stream_count go negative media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add v4l_enable/disable_media_tuner() helper functions media: v4l2-core add v4l_vb2q_enable_media_tuner() helper media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: au0828 fix au0828_create_media_graph() entity checks media: Change v4l-core to check for tuner availability media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_tuner() media: au0828 change to use Managed Media Controller API media: au0828 handle media_init and media_register window media: au0828 change to register/unregister entity_notify hook media: au0828 create tuner to decoder link in deactivated state media: dvb-core create tuner to demod pad link in disabled state media: au0828 implement enable_source and disable_source handlers media: au0828 fix null pointer reference in au0828_create_media_graph() media: au0828 fix to not call media_device_unregister_entity_notify() sound/usb: Update ALSA driver to use Managed Media Controller API sound/usb: Create media mixer function and control interface entities media: au0828 create link between ALSA Mixer and decoder media: track media device unregister in progress sound/usb: Check media device unregister progress state media: au0828 change to check media device unregister progress state drivers/media/dvb-core/dvb_frontend.c | 139 ++--------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-core/dvbdev.c | 4 +- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 66 +++++- drivers/media/media-entity.c | 65 +++-- drivers/media/usb/au0828/au0828-core.c | 340 +++++++++++++++++++++++---- drivers/media/usb/au0828/au0828-video.c | 75 +----- drivers/media/usb/au0828/au0828.h | 9 + drivers/media/v4l2-core/v4l2-dev.c | 48 ++++ drivers/media/v4l2-core/v4l2-fh.c | 1 + drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 4 + include/media/media-device.h | 61 +++++ include/media/media-entity.h | 12 + include/media/v4l2-dev.h | 5 + include/uapi/linux/media.h | 17 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 16 ++ sound/usb/card.h | 1 + sound/usb/media.c | 299 +++++++++++++++++++++++ sound/usb/media.h | 76 ++++++ sound/usb/mixer.h | 1 + sound/usb/pcm.c | 28 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 5 + sound/usb/stream.c | 1 + sound/usb/usbaudio.h | 2 + 31 files changed, 1070 insertions(+), 261 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.5.0

3 10

[alsa-devel] [PATCH] topology: fix debug output to print correct "max" value.
by Liam Girdwood 01 Feb '16

01 Feb '16

Debug log is printing num_regs instead of max in the max section. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/ctl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/topology/ctl.c b/src/topology/ctl.c index 1c073f7..c250227 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -350,7 +350,7 @@ int tplg_parse_control_bytes(snd_tplg_t *tplg, return -EINVAL; be->max = atoi(val); - tplg_dbg("\t%s: %d\n", id, be->num_regs); + tplg_dbg("\t%s: %d\n", id, be->max); continue; } -- 2.5.0

2 1

[alsa-devel] [PATCH v2 1/3] [ALSA] portman2x4 - whitespace fixes
by Sudip Mukherjee 01 Feb '16

01 Feb '16

checkpatch was complaining about trailing whitespace, multiple blank lines and space before tabs. This patch takes care of these issues related to whitespace changes and there was no other change. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v2: combined 3 patches of v1 series into this patch. sound/drivers/portman2x4.c | 77 ++++++++++++++++++++++------------------------ 1 file changed, 37 insertions(+), 40 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 464385a..65b0650 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -57,7 +57,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -static struct platform_device *platform_devices[SNDRV_CARDS]; +static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; module_param_array(index, int, NULL, S_IRUGO); @@ -105,7 +105,7 @@ static int portman_create(struct snd_card *card, *rchip = NULL; pm = kzalloc(sizeof(struct portman), GFP_KERNEL); - if (pm == NULL) + if (pm == NULL) return -ENOMEM; /* Init chip specific data */ @@ -123,24 +123,24 @@ static int portman_create(struct snd_card *card, *********************************************************************/ /* Standard PC parallel port status register equates. */ -#define PP_STAT_BSY 0x80 /* Busy status. Inverted. */ -#define PP_STAT_ACK 0x40 /* Acknowledge. Non-Inverted. */ -#define PP_STAT_POUT 0x20 /* Paper Out. Non-Inverted. */ -#define PP_STAT_SEL 0x10 /* Select. Non-Inverted. */ -#define PP_STAT_ERR 0x08 /* Error. Non-Inverted. */ +#define PP_STAT_BSY 0x80 /* Busy status. Inverted. */ +#define PP_STAT_ACK 0x40 /* Acknowledge. Non-Inverted. */ +#define PP_STAT_POUT 0x20 /* Paper Out. Non-Inverted. */ +#define PP_STAT_SEL 0x10 /* Select. Non-Inverted. */ +#define PP_STAT_ERR 0x08 /* Error. Non-Inverted. */ /* Standard PC parallel port command register equates. */ -#define PP_CMD_IEN 0x10 /* IRQ Enable. Non-Inverted. */ -#define PP_CMD_SELI 0x08 /* Select Input. Inverted. */ -#define PP_CMD_INIT 0x04 /* Init Printer. Non-Inverted. */ -#define PP_CMD_FEED 0x02 /* Auto Feed. Inverted. */ +#define PP_CMD_IEN 0x10 /* IRQ Enable. Non-Inverted. */ +#define PP_CMD_SELI 0x08 /* Select Input. Inverted. */ +#define PP_CMD_INIT 0x04 /* Init Printer. Non-Inverted. */ +#define PP_CMD_FEED 0x02 /* Auto Feed. Inverted. */ #define PP_CMD_STB 0x01 /* Strobe. Inverted. */ /* Parallel Port Command Register as implemented by PCP2x4. */ -#define INT_EN PP_CMD_IEN /* Interrupt enable. */ +#define INT_EN PP_CMD_IEN /* Interrupt enable. */ #define STROBE PP_CMD_STB /* Command strobe. */ -/* The parallel port command register field (b1..b3) selects the +/* The parallel port command register field (b1..b3) selects the * various "registers" within the PC/P 2x4. These are the internal * address of these "registers" that must be written to the parallel * port command register. @@ -148,7 +148,7 @@ static int portman_create(struct snd_card *card, #define RXDATA0 (0 << 1) /* PCP RxData channel 0. */ #define RXDATA1 (1 << 1) /* PCP RxData channel 1. */ #define GEN_CTL (2 << 1) /* PCP General Control Register. */ -#define SYNC_CTL (3 << 1) /* PCP Sync Control Register. */ +#define SYNC_CTL (3 << 1) /* PCP Sync Control Register. */ #define TXDATA0 (4 << 1) /* PCP TxData channel 0. */ #define TXDATA1 (5 << 1) /* PCP TxData channel 1. */ #define TXDATA2 (6 << 1) /* PCP TxData channel 2. */ @@ -218,7 +218,7 @@ static inline void portman_write_data(struct portman *pm, u8 value) parport_write_data(pm->pardev->port, value); } -static void portman_write_midi(struct portman *pm, +static void portman_write_midi(struct portman *pm, int port, u8 mididata) { int command = ((port + 4) << 1); @@ -234,20 +234,20 @@ static void portman_write_midi(struct portman *pm, */ command |= INT_EN; - /* Disable interrupts so that the process is not interrupted, then - * write the address associated with the current Tx channel to the + /* Disable interrupts so that the process is not interrupted, then + * write the address associated with the current Tx channel to the * PP Command Reg. Do not set the Strobe signal yet. */ do { portman_write_command(pm, command); - /* While the address lines settle, write parallel output data to + /* While the address lines settle, write parallel output data to * PP Data Reg. This has no effect until Strobe signal is asserted. */ portman_write_data(pm, mididata); - + /* If PCP channel's TxEmpty is set (TxEmpty is read through the PP * Status Register), then go write data. Else go back and wait. */ @@ -284,7 +284,6 @@ static void portman_write_midi(struct portman *pm, /* Data sent. */ } - /* * Read MIDI byte from port * Attempt to read input byte from specified hardware input port (0..). @@ -357,7 +356,6 @@ static int portman_read_midi(struct portman *pm, int port) portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */ portman_write_data(pm, 0); /* Return data clock low. */ - /* De-assert Strobe and return data. */ portman_write_command(pm, cmdout); /* Output saved address+IE. */ @@ -370,11 +368,12 @@ static int portman_read_midi(struct portman *pm, int port) /* * Checks if any input data on the given channel is available - * Checks RxAvail + * Checks RxAvail */ static int portman_data_avail(struct portman *pm, int channel) { int command = INT_EN; + switch (channel) { case 0: command |= RXDATA0; @@ -393,7 +392,6 @@ static int portman_data_avail(struct portman *pm, int channel) return 0; } - /* * Flushes any input */ @@ -456,7 +454,7 @@ static int portman_probe(struct parport *p) * hardware handshake lines to midi box: * * Strobe = 0 - * Interrupt Enable = 0 + * Interrupt Enable = 0 */ /* 2 */ parport_write_control(p, 0); @@ -481,7 +479,7 @@ static int portman_probe(struct parport *p) /* 7 */ parport_write_control(p, 0); /* Reset Strobe=0. */ - /* Check if Tx circuitry is functioning properly. If initialized + /* Check if Tx circuitry is functioning properly. If initialized * unit TxEmpty is false, send out char and see if if goes true. */ /* 8 */ @@ -567,12 +565,12 @@ static int snd_portman_rawmidi_create(struct snd_card *card) struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; int err; - - err = snd_rawmidi_new(card, CARD_NAME, 0, - PORTMAN_NUM_OUTPUT_PORTS, - PORTMAN_NUM_INPUT_PORTS, + + err = snd_rawmidi_new(card, CARD_NAME, 0, + PORTMAN_NUM_OUTPUT_PORTS, + PORTMAN_NUM_INPUT_PORTS, &rmidi); - if (err < 0) + if (err < 0) return err; rmidi->private_data = pm; @@ -584,9 +582,9 @@ static int snd_portman_rawmidi_create(struct snd_card *card) pm->rmidi = rmidi; /* register rawmidi ops */ - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_portman_midi_output); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_portman_midi_input); /* name substreams */ @@ -621,7 +619,7 @@ static void snd_portman_interrupt(void *userdata) /* While any input data is waiting */ while ((portman_read_status(pm) & INT_REQ) == INT_REQ) { - /* If data available on channel 0, + /* If data available on channel 0, read it and stuff it into the queue. */ if (portman_data_avail(pm, 0)) { /* Read Midi */ @@ -632,7 +630,7 @@ static void snd_portman_interrupt(void *userdata) &midivalue, 1); } - /* If data available on channel 1, + /* If data available on channel 1, read it and stuff it into the queue. */ if (portman_data_avail(pm, 1)) { /* Read Midi */ @@ -658,7 +656,7 @@ static int snd_portman_probe_port(struct parport *p) 0, NULL); if (!pardev) return -EIO; - + if (parport_claim(pardev)) { parport_unregister_device(pardev); return -EIO; @@ -742,7 +740,7 @@ static int snd_portman_probe(struct platform_device *pdev) if (dev >= SNDRV_CARDS) return -ENODEV; - if (!enable[dev]) + if (!enable[dev]) return -ENOENT; if ((err = snd_portman_probe_port(p)) < 0) @@ -756,7 +754,7 @@ static int snd_portman_probe(struct platform_device *pdev) } strcpy(card->driver, DRIVER_NAME); strcpy(card->shortname, CARD_NAME); - sprintf(card->longname, "%s at 0x%lx, irq %i", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); pardev = parport_register_device(p, /* port */ @@ -779,7 +777,7 @@ static int snd_portman_probe(struct platform_device *pdev) } card->private_data = pm; card->private_free = snd_portman_card_private_free; - + if ((err = snd_portman_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); goto __err; @@ -823,7 +821,6 @@ static int snd_portman_remove(struct platform_device *pdev) return 0; } - static struct platform_driver snd_portman_driver = { .probe = snd_portman_probe, .remove = snd_portman_remove, @@ -844,7 +841,7 @@ static void snd_portman_unregister_all(void) platform_device_unregister(platform_devices[i]); platform_devices[i] = NULL; } - } + } platform_driver_unregister(&snd_portman_driver); parport_unregister_driver(&portman_parport_driver); } -- 1.9.1

2 4

[alsa-devel] [sound:topic/core-fixes 20/20] ERROR: "__snd_rawmidi_transmit_ack" [sound/core/seq/snd-seq-virmidi.ko] undefined!
by kbuild test robot 01 Feb '16

01 Feb '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/core-fixes head: 9e8e5e25fabf400957a5e1f51ae8107ac949403f commit: 9e8e5e25fabf400957a5e1f51ae8107ac949403f [20/20] ALSA: rawmidi: Make snd_rawmidi_transmit() race-free config: i386-randconfig-s1-201605 (attached as .config) reproduce: git checkout 9e8e5e25fabf400957a5e1f51ae8107ac949403f # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): >> ERROR: "__snd_rawmidi_transmit_ack" [sound/core/seq/snd-seq-virmidi.ko] undefined! --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] sound: use-after-free in snd_seq_deliver_single_event
by Dmitry Vyukov 01 Feb '16

01 Feb '16

Hello, The following program triggers use-after-free in snd_seq_deliver_single_event (run in a tight parallel loop): // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> #ifndef SYS_mmap #define SYS_mmap 9 #endif #ifndef SYS_syz_open_dev #define SYS_syz_open_dev 1000001 #endif #ifndef SYS_ioctl #define SYS_ioctl 16 #endif #ifndef SYS_getpgrp #define SYS_getpgrp 111 #endif #ifndef SYS_migrate_pages #define SYS_migrate_pages 256 #endif #ifndef SYS_read #define SYS_read 0 #endif #ifndef SYS_write #define SYS_write 1 #endif #ifndef SYS_dup3 #define SYS_dup3 292 #endif long r[59]; void* thr(void* arg) { switch ((long)arg) { case 0: r[0] = syscall(SYS_mmap, 0x20000000ul, 0x47000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 1: memcpy((void*)0x20000000, "\x2f\x64\x65\x76\x2f\x73\x65\x71\x75\x65\x6e\x63\x65\x72", 14); r[2] = syscall(SYS_open, "/dev/sequencer", 0x2ul, 0, 0, 0); break; case 2: r[3] = syscall(SYS_mmap, 0x20047000ul, 0x1000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 3: r[4] = syscall(SYS_ioctl, r[2], 0x540ful, 0x20047ffcul, 0, 0, 0); if (r[4] != -1) r[5] = *(uint32_t*)0x20047ffc; break; case 4: r[6] = syscall(SYS_getpgrp, r[5], 0, 0, 0, 0, 0); break; case 5: r[7] = syscall(SYS_mmap, 0x20047000ul, 0x1000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 6: r[8] = syscall(SYS_mmap, 0x20047000ul, 0x1000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); break; case 7: *(uint64_t*)0x20047000 = (uint64_t)0x5; *(uint64_t*)0x20047000 = (uint64_t)0xfffffffffffffff8; r[11] = syscall(SYS_migrate_pages, r[6], 0xfful, 0x20047000ul, 0x20047000ul, 0, 0); break; case 8: memcpy((void*)0x20043684, "\x2f\x64\x65\x76\x2f\x64\x6d\x6d\x69\x64\x69\x23", 12); r[13] = syscall(SYS_open, "/dev/dmmidi2", 0x8a000ul, 0, 0, 0); break; case 9: r[14] = syscall(SYS_read, r[13], 0x20045000ul, 0x36ul, 0, 0, 0); break; case 10: *(uint8_t*)0x2000042c = (uint8_t)0x0; *(uint8_t*)0x2000042d = (uint8_t)0x9cf; *(uint8_t*)0x2000042e = (uint8_t)0x9; *(uint8_t*)0x2000042f = (uint8_t)0x0; *(uint64_t*)0x20000444 = (uint64_t)0x0; *(uint64_t*)0x2000044c = (uint64_t)0x989680; *(uint8_t*)0x20000454 = (uint8_t)0x2; *(uint8_t*)0x20000455 = (uint8_t)0x100000000; *(uint8_t*)0x20000456 = (uint8_t)0x8d; *(uint8_t*)0x20000457 = (uint8_t)0x7; *(uint8_t*)0x20000464 = (uint8_t)0x51b2; *(uint8_t*)0x20000465 = (uint8_t)0xfffffffffffff15a; *(uint8_t*)0x20000466 = (uint8_t)0x4; *(uint8_t*)0x20000467 = (uint8_t)0x4; *(uint32_t*)0x20000468 = (uint32_t)0xffffffff; *(uint8_t*)0x2000046c = (uint8_t)0x5; *(uint8_t*)0x2000046d = (uint8_t)0xfffffffffffffff8; *(uint8_t*)0x2000046e = (uint8_t)0x401; *(uint8_t*)0x2000046f = (uint8_t)0x5; *(uint64_t*)0x20000484 = (uint64_t)0x4; *(uint64_t*)0x2000048c = (uint64_t)0x0; *(uint8_t*)0x20000494 = (uint8_t)0xc2c1; *(uint8_t*)0x20000495 = (uint8_t)0x1; *(uint8_t*)0x20000496 = (uint8_t)0x6b; *(uint8_t*)0x20000497 = (uint8_t)0x6; *(uint8_t*)0x200004a0 = (uint8_t)0xfffffffffffffffa; *(uint8_t*)0x200004a1 = (uint8_t)0x81; *(uint8_t*)0x200004a2 = (uint8_t)0x0; *(uint8_t*)0x200004a3 = (uint8_t)0xa6; *(uint8_t*)0x200004a4 = (uint8_t)0x400; *(uint8_t*)0x200004a5 = (uint8_t)0x3ff; *(uint8_t*)0x200004a6 = (uint8_t)0x8; *(uint8_t*)0x200004a7 = (uint8_t)0x3; *(uint64_t*)0x200004bc = (uint64_t)0x0; *(uint64_t*)0x200004c4 = (uint64_t)0x0; *(uint8_t*)0x200004cc = (uint8_t)0x9e; *(uint8_t*)0x200004cd = (uint8_t)0x7ff; *(uint8_t*)0x200004ce = (uint8_t)0x7fff; *(uint8_t*)0x200004cf = (uint8_t)0x9; *(uint8_t*)0x200004e0 = (uint8_t)0xa; *(uint32_t*)0x200004e4 = (uint32_t)0x6; *(uint32_t*)0x200004e8 = (uint32_t)0xffffffff; r[57] = syscall(SYS_write, r[2], 0x2000042cul, 0x78ul, 0, 0, 0); break; case 11: r[58] = syscall(SYS_dup3, r[13], r[2], 0x80000ul, 0, 0, 0); break; } return 0; } int main() { long i; pthread_t th[12]; memset(r, -1, sizeof(r)); for (i = 0; i < 12; i++) { pthread_create(&th[i], 0, thr, (void*)i); usleep(10000); } for (i = 0; i < 12; i++) { pthread_create(&th[i], 0, thr, (void*)i); if (i % 2 == 0) usleep(10000); } usleep(100000); return 0; } ================================================================== BUG: KASAN: use-after-free in do_raw_spin_lock+0x281/0x2b0 at addr ffff88002abf1654 Read of size 4 by task syz-executor/13987 ============================================================================= BUG kmalloc-96 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- INFO: Allocated in snd_midi_event_new+0x73/0x200 age=10 cpu=0 pid=13996 [< none >] ___slab_alloc+0x564/0x5b0 mm/slub.c:2470 [< none >] __slab_alloc+0x66/0xc0 mm/slub.c:2499 [< inline >] slab_alloc_node mm/slub.c:2562 [< inline >] slab_alloc mm/slub.c:2604 [< none >] kmem_cache_alloc_trace+0x25c/0x300 mm/slub.c:2621 [< inline >] kmalloc include/linux/slab.h:463 [< inline >] kzalloc include/linux/slab.h:607 [< none >] snd_midi_event_new+0x73/0x200 sound/core/seq/seq_midi_event.c:120 [< none >] snd_virmidi_input_open+0xfd/0x380 sound/core/seq/seq_virmidi.c:214 [< none >] open_substream+0x3eb/0x780 sound/core/rawmidi.c:269 [< none >] rawmidi_open_priv+0x144/0x300 sound/core/rawmidi.c:312 [< none >] snd_rawmidi_open+0x3fb/0xa90 sound/core/rawmidi.c:416 [< none >] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [< none >] chrdev_open+0x22a/0x4c0 fs/char_dev.c:388 [< none >] do_dentry_open+0x6a2/0xcb0 fs/open.c:736 [< none >] vfs_open+0x17b/0x1f0 fs/open.c:853 [< inline >] do_last fs/namei.c:3254 [< none >] path_openat+0xde9/0x5e30 fs/namei.c:3386 [< none >] do_filp_open+0x18e/0x250 fs/namei.c:3421 [< none >] do_sys_open+0x1fc/0x420 fs/open.c:1022 [< inline >] SYSC_open fs/open.c:1040 [< none >] SyS_open+0x2d/0x40 fs/open.c:1035 INFO: Freed in snd_midi_event_free+0x43/0x60 age=17 cpu=0 pid=13996 [< none >] __slab_free+0x1fc/0x320 mm/slub.c:2680 [< inline >] slab_free mm/slub.c:2835 [< none >] kfree+0x2ac/0x2c0 mm/slub.c:3664 [< none >] snd_midi_event_free+0x43/0x60 sound/core/seq/seq_midi_event.c:142 [< none >] snd_virmidi_input_close+0x87/0x120 sound/core/seq/seq_virmidi.c:261 [< none >] close_substream.part.14+0xda/0x540 sound/core/rawmidi.c:486 [< inline >] close_substream sound/core/rawmidi.c:514 [< none >] rawmidi_release_priv+0x1e4/0x250 sound/core/rawmidi.c:504 [< none >] snd_rawmidi_release+0x62/0xf0 sound/core/rawmidi.c:539 [< none >] __fput+0x236/0x780 fs/file_table.c:208 [< none >] ____fput+0x15/0x20 fs/file_table.c:244 [< none >] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [< none >] do_exit+0x8b5/0x2cb0 kernel/exit.c:748 [< none >] do_group_exit+0x108/0x330 kernel/exit.c:878 [< none >] get_signal+0x5e4/0x14f0 kernel/signal.c:2307 [< none >] do_signal+0x83/0x1c90 arch/x86/kernel/signal.c:712 [< none >] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:247 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:282 [< none >] syscall_return_slowpath+0x2ba/0x340 arch/x86/entry/common.c:344 INFO: Slab 0xffffea0000aafc00 objects=28 used=11 fp=0xffff88002abf0470 flags=0x1fffc0000004080 INFO: Object 0xffff88002abf1630 @offset=5680 fp=0xffff88002abf06a8 CPU: 1 PID: 13987 Comm: syz-executor Tainted: G B 4.5.0-rc1+ #305 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff880061aaf7d8 ffffffff82be11ad ffff88003e804900 ffff88002abf1630 ffff88002abf0000 ffff880061aaf808 ffffffff8175b454 ffff88003e804900 ffffea0000aafc00 ffff88002abf1630 000000000000000a Call Trace: [<ffffffff81764e3e>] __asan_report_load4_noabort+0x3e/0x40 mm/kasan/report.c:294 [< inline >] debug_spin_lock_before kernel/locking/spinlock_debug.c:83 [<ffffffff814663d1>] do_raw_spin_lock+0x281/0x2b0 kernel/locking/spinlock_debug.c:135 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:119 [<ffffffff86652b67>] _raw_spin_lock_irqsave+0xa7/0xd0 kernel/locking/spinlock.c:159 [<ffffffff852bb6a7>] snd_midi_event_decode+0x1f7/0x5c0 sound/core/seq/seq_midi_event.c:391 [< inline >] snd_virmidi_dev_receive_event sound/core/seq/seq_virmidi.c:95 [<ffffffff852ce754>] snd_virmidi_event_input+0x214/0x310 sound/core/seq/seq_virmidi.c:133 [<ffffffff852a7934>] snd_seq_deliver_single_event.constprop.11+0x3f4/0x740 sound/core/seq/seq_clientmgr.c:634 [<ffffffff852a7da2>] snd_seq_deliver_event+0x122/0x800 sound/core/seq/seq_clientmgr.c:828 [<ffffffff852a9186>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2401 [< inline >] snd_seq_oss_dispatch sound/core/seq/oss/seq_oss_device.h:151 [<ffffffff852ca423>] snd_seq_oss_midi_reset+0x303/0x4a0 sound/core/seq/oss/seq_oss_midi.c:481 [<ffffffff852bd490>] snd_seq_oss_reset+0x130/0x260 sound/core/seq/oss/seq_oss_init.c:469 [<ffffffff852bd631>] snd_seq_oss_release+0x71/0x130 sound/core/seq/oss/seq_oss_init.c:425 [<ffffffff852bbcca>] odev_release+0x5a/0x80 sound/core/seq/oss/seq_oss.c:155 [<ffffffff817c0156>] __fput+0x236/0x780 fs/file_table.c:208 [<ffffffff817c0725>] ____fput+0x15/0x20 fs/file_table.c:244 [<ffffffff813b14e0>] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] tracehook_notify_resume include/linux/tracehook.h:191 [<ffffffff810066b1>] exit_to_usermode_loop+0x1d1/0x210 arch/x86/entry/common.c:251 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:282 [<ffffffff810084ea>] syscall_return_slowpath+0x2ba/0x340 arch/x86/entry/common.c:344 [<ffffffff86653322>] int_ret_from_sys_call+0x25/0x9f arch/x86/entry/entry_64.S:281 ================================================================== general protection fault: 0000 [#1] SMP DEBUG_PAGEALLOC KASAN Modules linked in: CPU: 1 PID: 14982 Comm: syz-executor Tainted: G W 4.5.0-rc1+ #305 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88005e362f80 ti: ffff88005fba8000 task.ti: ffff88005fba8000 RIP: 0010:[<ffffffff852ce6d5>] [<ffffffff852ce6d5>] snd_virmidi_event_input+0x195/0x310 RSP: 0018:ffff88005fbaf608 EFLAGS: 00010202 RAX: 1bd5a00000000023 RBX: dead000000000100 RCX: 0000000000000002 RDX: 0000000000000004 RSI: 0000000000000100 RDI: dead00000000011c RBP: ffff88005fbaf6b8 R08: 0000000000000001 R09: 0000000000000000 R10: ffff88005e362f80 R11: ffff8800009fe110 R12: dffffc0000000000 R13: ffff88005fbaf900 R14: ffff8800009fe130 R15: 1ffff1000bf75ec6 FS: 00007fb93c201700(0000) GS:ffff88003ed00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 0000000020045000 CR3: 000000005f9bd000 CR4: 00000000000006e0 Stack: ffff8800009fe0f8 ffffed000bf75f20 ffff88005fbaf901 0000000000000000 ffff88003639fa50 0000000041b58ab3 ffffffff879486ce ffffffff852ce540 ffff88003639fa28 0000000000000000 ffff88003639fa38 ffff88005fbaf678 Call Trace: [<ffffffff852a7934>] snd_seq_deliver_single_event.constprop.11+0x3f4/0x740 sound/core/seq/seq_clientmgr.c:634 [<ffffffff852a7da2>] snd_seq_deliver_event+0x122/0x800 sound/core/seq/seq_clientmgr.c:828 [<ffffffff852a9186>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2401 [< inline >] snd_seq_oss_dispatch sound/core/seq/oss/seq_oss_device.h:151 [<ffffffff852ca527>] snd_seq_oss_midi_reset+0x407/0x4a0 sound/core/seq/oss/seq_oss_midi.c:475 [<ffffffff852bd490>] snd_seq_oss_reset+0x130/0x260 sound/core/seq/oss/seq_oss_init.c:469 [<ffffffff852bd631>] snd_seq_oss_release+0x71/0x130 sound/core/seq/oss/seq_oss_init.c:425 [<ffffffff852bbcca>] odev_release+0x5a/0x80 sound/core/seq/oss/seq_oss.c:155 [<ffffffff817c0156>] __fput+0x236/0x780 fs/file_table.c:208 [<ffffffff817c0725>] ____fput+0x15/0x20 fs/file_table.c:244 [<ffffffff813b14e0>] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [<ffffffff8135ca55>] do_exit+0x8b5/0x2cb0 kernel/exit.c:748 [<ffffffff8135efc8>] do_group_exit+0x108/0x330 kernel/exit.c:878 [<ffffffff81382144>] get_signal+0x5e4/0x14f0 kernel/signal.c:2307 [<ffffffff811a0db3>] do_signal+0x83/0x1c90 arch/x86/kernel/signal.c:712 [<ffffffff81006685>] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:247 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:282 [<ffffffff810084ea>] syscall_return_slowpath+0x2ba/0x340 arch/x86/entry/common.c:344 [<ffffffff86653322>] int_ret_from_sys_call+0x25/0x9f arch/x86/entry/entry_64.S:281 Code: 3c 20 00 0f 85 31 01 00 00 48 8b 1b 49 39 de 0f 84 d9 00 00 00 e8 ec bb 29 fc 48 8d 7b 1c 48 89 f8 48 89 fa 48 c1 e8 03 83 e2 07 <42> 0f b6 04 20 38 d0 7f 08 84 c0 0f 85 3d 01 00 00 f6 43 1c 01 RIP [< inline >] snd_virmidi_dev_receive_event sound/core/seq/seq_virmidi.c:88 RIP [<ffffffff852ce6d5>] snd_virmidi_event_input+0x195/0x310 sound/core/seq/seq_virmidi.c:133 RSP <ffff88005fbaf608> ---[ end trace 35fa82e449e91fba ]--- Fixing recursive fault but reboot is needed! On commit 26cd83670f2f5a3d5b5514a1f7d96567cdb9558b.

3 7

Re: [alsa-devel] sound: WARNING in snd_rawmidi_transmit_ack
by Takashi Iwai 01 Feb '16

01 Feb '16

On Wed, 27 Jan 2016 13:31:05 +0100, Dmitry Vyukov wrote: > > On Tue, Jan 26, 2016 at 3:51 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > On Sun, 24 Jan 2016 17:29:12 +0100, > > Dmitry Vyukov wrote: > >> > >> Hello, > >> > >> The following program triggers WARNING in snd_rawmidi_transmit_ack: > >> > >> ------------[ cut here ]------------ > >> WARNING: CPU: 1 PID: 20739 at sound/core/rawmidi.c:1136 > >> snd_rawmidi_transmit_ack+0x275/0x400() > >> Modules linked in: > >> CPU: 1 PID: 20739 Comm: syz-executor Tainted: G W 4.4.0+ #276 > >> Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 > >> 00000000ffffffff ffff8800346af8d8 ffffffff82999e2d 0000000000000000 > >> ffff88002fea97c0 ffffffff86d3f920 ffff8800346af918 ffffffff81352089 > >> ffffffff84f80bd5 ffffffff86d3f920 0000000000000470 ffff8800600b9b68 > >> Call Trace: > >> [< inline >] __dump_stack lib/dump_stack.c:15 > >> [<ffffffff82999e2d>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 > >> [<ffffffff81352089>] warn_slowpath_common+0xd9/0x140 kernel/panic.c:482 > >> [<ffffffff813522b9>] warn_slowpath_null+0x29/0x30 kernel/panic.c:515 > >> [<ffffffff84f80bd5>] snd_rawmidi_transmit_ack+0x275/0x400 > >> sound/core/rawmidi.c:1136 > >> [<ffffffff84fdb3c1>] snd_virmidi_output_trigger+0x4b1/0x5a0 > >> sound/core/seq/seq_virmidi.c:163 > >> [< inline >] snd_rawmidi_output_trigger sound/core/rawmidi.c:150 > >> [<ffffffff84f87ed9>] snd_rawmidi_kernel_write1+0x549/0x780 > >> sound/core/rawmidi.c:1223 > >> [<ffffffff84f89fd3>] snd_rawmidi_write+0x543/0xb30 sound/core/rawmidi.c:1273 > >> [<ffffffff817b0323>] __vfs_write+0x113/0x480 fs/read_write.c:528 > >> [<ffffffff817b1db7>] vfs_write+0x167/0x4a0 fs/read_write.c:577 > >> [< inline >] SYSC_write fs/read_write.c:624 > >> [<ffffffff817b50a1>] SyS_write+0x111/0x220 fs/read_write.c:616 > >> [<ffffffff86336c36>] entry_SYSCALL_64_fastpath+0x16/0x7a > >> arch/x86/entry/entry_64.S:185 > >> ---[ end trace 93509ba74cc3e259 ]--- > >> > >> > >> // autogenerated by syzkaller (http://github.com/google/syzkaller) > >> #include <pthread.h> > >> #include <stdint.h> > >> #include <string.h> > >> #include <sys/syscall.h> > >> #include <unistd.h> > >> #include <fcntl.h> > >> > >> long r[11]; > >> > >> void* thr(void* arg) > >> { > >> switch ((long)arg) { > >> case 0: > >> r[0] = syscall(SYS_mmap, 0x20000000ul, 0x10000ul, 0x3ul, 0x32ul, > >> 0xfffffffffffffffful, 0x0ul); > >> break; > >> case 1: > >> r[1] = syscall(SYS_open, "/dev/dmmidi2", O_RDWR, 0); > >> break; > >> case 2: > >> *(uint32_t*)0x20006fda = (uint32_t)0x28; > >> *(uint32_t*)0x20006fde = (uint32_t)0x7; > >> *(uint64_t*)0x20006fe2 = (uint64_t)0x0; > >> *(uint64_t*)0x20006fea = (uint64_t)0x7; > >> *(uint64_t*)0x20006ff2 = (uint64_t)0x8; > >> *(uint16_t*)0x20006ffa = (uint16_t)0xfffffffffffffff8; > >> r[8] = syscall(SYS_write, r[1], 0x20006fdaul, 0x28ul, 0, 0, 0); > >> break; > >> case 3: > >> memcpy((void*)0x20007000, > >> "\x3c\x3f\xbc\xd7\x07\xee\x01\x21\x84\x26\x11\x6d\xc6\xc3" > >> "\xac\x7f\x14\xa3\x15\xda\xa1\x34\x67\xe9\xb6\xf8\xa8\xdd" > >> "\xb5\xe0\x2a\x64\x6e\x04\x7c\x40\xdb\x64\x4f\x53\x41\x90" > >> "\x02\x3b\x48\xba\xf6\xd7\x32\xe5\x11\x3f\xe5\xc5\xf2\x5f" > >> "\x4e\x6f\x5b\xe9\x0e\x76\x1a\x72\x77\x68\xad\xd3\x1b\x8d" > >> "\x8f\x71\x2f\x40\xde\xf4\xbf\x78\x48\xc2\xc3\xe5\xe9\x11" > >> "\x11\xd2\x9a\x91\x94\x81\x57\x22\xef\x2a\x5b\xe2\x9e\xce" > >> "\x3f\xa6\x07\x54\xd4\xb6\x1f\x30\xfd\x09\x35\x54\xcc\x78" > >> "\x12\x37\x59\xf7\x5c\x51\xd2\x47\xb8\x40\x98\x2b\xb4\x8d" > >> "\xbf\xd5\x34\x26\xcd\xb5\xe4\x98\x2d\x0e\xb1\x71\x0f\x12" > >> "\x06\xb0\x2c\x78\x74\xba\x3d\x32\x9c\x72\x96\x0d\x91\x50" > >> "\x9a\xe3\xbc\x4f\x1e\x26\xb2\xe3\xb2\xa2\x68\x6f\x10\xb5" > >> "\xc1\x7f\xfc\xfd\xf0\xe6\xd9\x0c\xa0\xa7\x54\x2c\x62\x0d" > >> "\xee\x04\x5b\x9d\x73\xf4\xcc\x29\x37\x01\xc2\x35\x8d\x3c" > >> "\x8a\x2f\xe3\xf5\x55\x42\x77\x55\x2d\x49\x6f\x46\xf7\x34" > >> "\xb1\x5f\xe7\x98\x55\xb5\x7c\x4e\xd7\x47\x18\xd1\xf9\xf2" > >> "\xd2\xfe\x47\xff\x22\x04\xa9\x32\xe9\xd6\x28\xda\xe1\x85" > >> "\xcb\xc5\xe6\xc1\xca\x9e\x3f\x1f\x80\xe6", > >> 248); > >> r[10] = syscall(SYS_write, r[1], 0x20007000ul, 0xf8ul, 0, 0, 0); > >> break; > >> } > >> return 0; > >> } > >> > >> int main() > >> { > >> long i; > >> pthread_t th[4]; > >> > >> memset(r, -1, sizeof(r)); > >> for (i = 0; i < 4; i++) { > >> pthread_create(&th[i], 0, thr, (void*)i); > >> usleep(10000); > >> } > >> for (i = 0; i < 4; i++) { > >> pthread_create(&th[i], 0, thr, (void*)i); > >> if (i % 2 == 0) > >> usleep(10000); > >> } > >> usleep(100000); > >> return 0; > >> } > >> > >> On commit 30f05309bde49295e02e45c7e615f73aa4e0ccc2. > > > > Could you check the patch below? It's untested yet in my side due to > > other bugs... > > No, I still see the same warning. > Please try to reproduce it locally and test. This wasn't triggered on my machine. How did you start the test program? Meanwhile below is another patch to be applied in addition to the previous fix. There is a possible race in calling snd_rawmidi_transmit(), and this may result in such a warning. Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: rawmidi: Make snd_rawmidi_transmit() race-free Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/rawmidi.c | 95 ++++++++++++++++++++++++++++++++++------------------ 1 file changed, 62 insertions(+), 33 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index a7759846fbaa..3a27944c821c 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1054,24 +1054,9 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) } EXPORT_SYMBOL(snd_rawmidi_transmit_empty); -/** - * snd_rawmidi_transmit_peek - copy data from the internal buffer - * @substream: the rawmidi substream - * @buffer: the buffer pointer - * @count: data size to transfer - * - * Copies data from the internal output buffer to the given buffer. - * - * Call this in the interrupt handler when the midi output is ready, - * and call snd_rawmidi_transmit_ack() after the transmission is - * finished. - * - * Return: The size of copied data, or a negative error code on failure. - */ -int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, +static int rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { - unsigned long flags; int result, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -1081,7 +1066,6 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, return -EINVAL; } result = 0; - spin_lock_irqsave(&runtime->lock, flags); if (runtime->avail >= runtime->buffer_size) { /* warning: lowlevel layer MUST trigger down the hardware */ goto __skip; @@ -1106,25 +1090,39 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, } } __skip: - spin_unlock_irqrestore(&runtime->lock, flags); return result; } -EXPORT_SYMBOL(snd_rawmidi_transmit_peek); /** - * snd_rawmidi_transmit_ack - acknowledge the transmission + * snd_rawmidi_transmit_peek - copy data from the internal buffer * @substream: the rawmidi substream - * @count: the transferred count + * @buffer: the buffer pointer + * @count: data size to transfer * - * Advances the hardware pointer for the internal output buffer with - * the given size and updates the condition. - * Call after the transmission is finished. + * Copies data from the internal output buffer to the given buffer. * - * Return: The advanced size if successful, or a negative error code on failure. + * Call this in the interrupt handler when the midi output is ready, + * and call snd_rawmidi_transmit_ack() after the transmission is + * finished. + * + * Return: The size of copied data, or a negative error code on failure. */ -int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) +int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, + unsigned char *buffer, int count) { + struct snd_rawmidi_runtime *runtime = substream->runtime; + int result; unsigned long flags; + + spin_lock_irqsave(&runtime->lock, flags); + result = rawmidi_transmit_peek(substream, buffer, count); + spin_unlock_irqrestore(&runtime->lock, flags); + return result; +} +EXPORT_SYMBOL(snd_rawmidi_transmit_peek); + +static int rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) +{ struct snd_rawmidi_runtime *runtime = substream->runtime; if (runtime->buffer == NULL) { @@ -1132,7 +1130,6 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) "snd_rawmidi_transmit_ack: output is not active!!!\n"); return -EINVAL; } - spin_lock_irqsave(&runtime->lock, flags); snd_BUG_ON(runtime->avail + count > runtime->buffer_size); runtime->hw_ptr += count; runtime->hw_ptr %= runtime->buffer_size; @@ -1142,9 +1139,31 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) if (runtime->drain || snd_rawmidi_ready(substream)) wake_up(&runtime->sleep); } - spin_unlock_irqrestore(&runtime->lock, flags); return count; } + +/** + * snd_rawmidi_transmit_ack - acknowledge the transmission + * @substream: the rawmidi substream + * @count: the transferred count + * + * Advances the hardware pointer for the internal output buffer with + * the given size and updates the condition. + * Call after the transmission is finished. + * + * Return: The advanced size if successful, or a negative error code on failure. + */ +int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + int result; + unsigned long flags; + + spin_lock_irqsave(&runtime->lock, flags); + result = rawmidi_transmit_ack(substream, count); + spin_unlock_irqrestore(&runtime->lock, flags); + return result; +} EXPORT_SYMBOL(snd_rawmidi_transmit_ack); /** @@ -1160,12 +1179,22 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_ack); int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { + struct snd_rawmidi_runtime *runtime = substream->runtime; + int result; + unsigned long flags; + + spin_lock_irqsave(&runtime->lock, flags); if (!substream->opened) - return -EBADFD; - count = snd_rawmidi_transmit_peek(substream, buffer, count); - if (count < 0) - return count; - return snd_rawmidi_transmit_ack(substream, count); + result = -EBADFD; + else { + count = rawmidi_transmit_peek(substream, buffer, count); + if (count <= 0) + result = count; + else + result = rawmidi_transmit_ack(substream, count); + } + spin_unlock_irqrestore(&runtime->lock, flags); + return result; } EXPORT_SYMBOL(snd_rawmidi_transmit); -- 2.7.0

2 1

Re: [alsa-devel] [PATCH] ASoC: nau8825: reduce stanby power consumption
by John Hsu 01 Feb '16

01 Feb '16

Hi Mark, This power patch modifies some widgets to fix power saving issue when standby. But we want to submit a depop patch to fine tune widgets and fix the pop noise now. It has overlap with the power patch. Can we make the depop patch based on the power patch? Could you give us advice how to make submit patch in this case? Thank you very much. BR. John On 1/15/2016 11:53 PM, AC30 KCHsu0 wrote: > Fix unnecessary widget power up when standby; And decrease internal clock frequency to save power. > > Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> > --- > sound/soc/codecs/nau8825.c | 49 +++++++++++++++++++++++++++++++++------------- > 1 file changed, 35 insertions(+), 14 deletions(-) > > diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 52882e4..f9752cc 100644 > --- a/sound/soc/codecs/nau8825.c > +++ b/sound/soc/codecs/nau8825.c > @@ -262,6 +262,29 @@ static int nau8825_playback_event(struct snd_soc_dapm_widget *w, > return 0; > } > > +static int nau8825_dacr_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) { > + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); > + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); > + > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, > + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); > + break; > + case SND_SOC_DAPM_POST_PMD: > + if (!nau8825->irq) > + regmap_update_bits(nau8825->regmap, > + NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, 0); > + break; > + default: > + return -EINVAL; > + } > + > + return 0; > +} > + > > static const char * const nau8825_adc_decimation[] = { > "32", "64", "128", "256" > @@ -337,16 +360,17 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { > 0), > > /* ADC for button press detection */ > - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, > - NAU8825_SAR_ADC_EN_SFT, 0), > + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, > + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), > > SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), > SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), > SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), > SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), > > - SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, > - NAU8825_ENABLE_DACR_SFT, 0), > + SND_SOC_DAPM_DAC_E("DDACR", NULL, SND_SOC_NOPM, 0, 0, > + nau8825_dacr_event, SND_SOC_DAPM_POST_PMU | > + SND_SOC_DAPM_POST_PMD), > SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, > NAU8825_ENABLE_DACL_SFT, 0), > SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), @@ -878,13 +902,6 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) > > nau8825->dapm = dapm; > > - /* The interrupt clock is gated by x1[10:8], > - * one of them needs to be enabled all the time for > - * interrupts to happen. > - */ > - snd_soc_dapm_force_enable_pin(dapm, "DDACR"); > - snd_soc_dapm_sync(dapm); > - > /* Unmask interruptions. Handler uses dapm object so we can enable > * interruptions only after dapm is fully initialized. > */ > @@ -1075,6 +1092,12 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, > NAU8825_DCO_EN); > regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, > NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); > + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, > + NAU8825_CLK_MCLK_SRC_MASK, 0xf); > + regmap_update_bits(regmap, NAU8825_REG_FLL1, > + NAU8825_FLL_RATIO_MASK, 0x10); > + regmap_update_bits(regmap, NAU8825_REG_FLL6, > + NAU8825_SDM_EN, NAU8825_SDM_EN); > if (nau8825->mclk_freq) { > clk_disable_unprepare(nau8825->mclk); > nau8825->mclk_freq = 0; > @@ -1309,9 +1332,7 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) > /* Enable internal VCO needed for interruptions */ > nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); > > - /* Enable DDACR needed for interrupts > - * It is the same as force_enable_pin("DDACR") we do later > - */ > + /* Enable DDACR needed for interrupts */ > regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, > NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); > > -- > 2.6.4 > > . > >

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[alsa-devel] [PATCH] ASoC: fsl: add imx-cs427x machine driver
by Felipe F. Tonello 31 Jan '16

31 Jan '16

This is the initial imx-cs427x device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Felipe F. Tonello <eu(a)felipetonello.com> --- .../devicetree/bindings/sound/imx-audio-cs427x.txt | 47 +++++ sound/soc/fsl/Kconfig | 12 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-cs427x.c | 218 +++++++++++++++++++++ 4 files changed, 279 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-cs427x.txt create mode 100644 sound/soc/fsl/imx-cs427x.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-cs427x.txt b/Documentation/devicetree/bindings/sound/imx-audio-cs427x.txt new file mode 100644 index 000000000000..295f60b19418 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-cs427x.txt @@ -0,0 +1,47 @@ +Freescale i.MX audio complex with CS4271 or CS4272 codec + +Required properties: + + - compatible : "fsl,imx-audio-cs427x" + + - model : The user-visible name of this sound complex + + - ssi-controller : The phandle of the i.MX SSI controller + + - audio-codec : The phandle of the CS4271 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, CS427x + pins, and the jacks on the board: + + Board connectors: + * Mic Jack + * Headphone Jack + + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) + + - mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx6-rex-cs427x", + "fsl,imx-audio-cs427x"; + model = "audio-cs427x"; + ssi-controller = <&ssi1>; + audio-codec = <&cs4271>; + audio-routing = + "Mic Jack", "AINA", + "Mic Jack", "AINB", + "Headphone Jack", "AOUTA+", + "Headphone Jack", "AOUTA-", + "Headphone Jack", "AOUTB+", + "Headphone Jack", "AOUTB-"; + mux-int-port = <1>; + mux-ext-port = <3>; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 14dfdee05fd5..8696b5c42dd8 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -263,6 +263,18 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. +config SND_SOC_IMX_CS427x + tristate "SoC Audio support for i.MX boards with cs4271 or cs4272" + depends on OF && (I2C || SPI) + select SND_SOC_CS4271_I2C if I2C + select SND_SOC_CS4271_SPI if SPI_MASTER + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + help + Say Y if you want to add support for SoC audio on an i.MX board with + a CS4271 or CS4272 codec over either SPI or I2C. + config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25c9375..334d183e3a17 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-cs427x-objs := imx-cs427x.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_CS427x) += snd-soc-imx-cs427x.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-cs427x.c b/sound/soc/fsl/imx-cs427x.c new file mode 100644 index 000000000000..45c56013822b --- /dev/null +++ b/sound/soc/fsl/imx-cs427x.c @@ -0,0 +1,218 @@ +/* + * Copyright 2015 ROLI Ltd. + * Author: Felipe F. Tonello <felipe.tonello(a)roli.com> + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <sound/soc.h> + +#include "imx-audmux.h" + +#define CS427x_SYSCLK_MCLK 0 + +struct imx_cs427x_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +static int imx_cs427x_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct imx_cs427x_data *data = snd_soc_card_get_drvdata(rtd->card); + struct device *dev = rtd->card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, CS427x_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret) + dev_err(dev, "failed to set sysclk\n"); + + return ret; +} + +static const struct snd_soc_dapm_widget imx_cs427x_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static int imx_cs427x_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct i2c_client *codec_dev; + struct imx_cs427x_data *data = NULL; + int int_port, ext_port; + int ret; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + ret = imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + ret = clk_prepare_enable(data->codec_clk); + if (ret) { + dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); + goto fail; + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "cs4271-hifi"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_of_node = ssi_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.init = &imx_cs427x_dai_init; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + data->card.num_links = 1; + data->card.owner = THIS_MODULE; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_cs427x_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_cs427x_dapm_widgets); + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +fail: + if (data && !IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); + of_node_put(ssi_np); + of_node_put(codec_np); + + return ret; +} + +static int imx_cs427x_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_cs427x_data *data = snd_soc_card_get_drvdata(card); + + clk_put(data->codec_clk); + + return 0; +} + +static const struct of_device_id imx_cs427x_dt_ids[] = { + { .compatible = "fsl,imx-audio-cs427x", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_cs427x_dt_ids); + +static struct platform_driver imx_cs427x_driver = { + .driver = { + .name = "imx-cs427x", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_cs427x_dt_ids, + }, + .probe = imx_cs427x_probe, + .remove = imx_cs427x_remove, +}; +module_platform_driver(imx_cs427x_driver); + +MODULE_AUTHOR("Felipe F. Tonello <felipe.tonello(a)roli.com>"); +MODULE_DESCRIPTION("Freescale i.MX CS427x ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-cs427x"); -- 2.7.0

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[alsa-devel] Meaning of the name 'hinawa' and the intension
by Takashi Sakamoto 31 Jan '16

31 Jan '16

Hi, After published libhinwa, I sometimes got a question about the name of 'hinawa'. Here, I describe it to the persions who are curious. At first, I split: FireWire = Fire + Wire (Japanese/Chinese translation in the meaning) -> U+706B[0] + U+7E04[1] (in ISO/IEC 10646) (Japanese translation in the pronounciation) -> Hi + Nawa Actually, the meaning of 'Nawa' is near rope. In Japanese history, Hinawa was used to fire musket guns introduced in Sengoku period, so the musket gun is called as Hinawa-ju (Ju means general guns). I think there'is a word in European culture corresponding to the hi-nawa, but I don't know it. I select this name because in general the name of library (soname) must be unique in a system. In my case, I judged it better to pick up words from non-European language group. [0] U+706B http://www.unicode.org/cgi-bin/GetUnihanData.pl?codepoint=706b [1] U+7E04 http://www.unicode.org/cgi-bin/GetUnihanData.pl?codepoint=7e04 Regards Takashi Sakamoto

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[alsa-devel] sound: deadlock between snd_pcm_oss_write/snd_pcm_oss_mmap
by Dmitry Vyukov 31 Jan '16

31 Jan '16

Hello, I've got the following deadlock report while running syzkaller fuzzer: [ INFO: possible circular locking dependency detected ] 4.5.0-rc1+ #305 Not tainted ------------------------------------------------------- syz-executor/14254 is trying to acquire lock: (&runtime->oss.params_lock){+.+.+.}, at: [<ffffffff8528a504>] snd_pcm_oss_change_params+0xd4/0x3540 sound/core/oss/pcm_oss.c:852 but task is already holding lock: (&mm->mmap_sem){++++++}, at: [<ffffffff816b267c>] vm_mmap_pgoff which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&mm->mmap_sem){++++++}: [<ffffffff8145b9dc>] lock_acquire+0x1dc/0x430 kernel/locking/lockdep.c:3587 [<ffffffff816def51>] __might_fault+0x141/0x1d0 mm/memory.c:3802 [< inline >] copy_from_user ./arch/x86/include/asm/uaccess.h:714 [< inline >] snd_pcm_oss_write1 sound/core/oss/pcm_oss.c:1376 [<ffffffff852940f0>] snd_pcm_oss_write+0x250/0x700 sound/core/oss/pcm_oss.c:2694 [<ffffffff817b90b3>] __vfs_write+0x113/0x480 fs/read_write.c:528 [<ffffffff817bab47>] vfs_write+0x167/0x4a0 fs/read_write.c:577 [< inline >] SYSC_write fs/read_write.c:624 [<ffffffff817bde31>] SyS_write+0x111/0x220 fs/read_write.c:616 [<ffffffff866531b6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 -> #0 (&runtime->oss.params_lock){+.+.+.}: [< inline >] check_prev_add kernel/locking/lockdep.c:1855 [< inline >] check_prevs_add kernel/locking/lockdep.c:1960 [< inline >] validate_chain kernel/locking/lockdep.c:2146 [<ffffffff8145807b>] __lock_acquire+0x31eb/0x4700 kernel/locking/lockdep.c:3208 [<ffffffff8145b9dc>] lock_acquire+0x1dc/0x430 kernel/locking/lockdep.c:3587 [< inline >] __mutex_lock_common kernel/locking/mutex.c:518 [<ffffffff8664891c>] mutex_lock_interruptible_nested+0xbc/0xbe0 kernel/locking/mutex.c:647 [<ffffffff8528a504>] snd_pcm_oss_change_params+0xd4/0x3540 sound/core/oss/pcm_oss.c:852 [<ffffffff8528f01d>] snd_pcm_oss_mmap+0x3dd/0x4c0 sound/core/oss/pcm_oss.c:2807 [<ffffffff81705747>] mmap_region+0x897/0x1010 mm/mmap.c:1624 [<ffffffff81706614>] do_mmap+0x754/0x990 mm/mmap.c:1403 [< inline >] do_mmap_pgoff include/linux/mm.h:1982 [<ffffffff816b26af>] vm_mmap_pgoff+0x15f/0x1b0 mm/util.c:328 [< inline >] SYSC_mmap_pgoff mm/mmap.c:1453 [<ffffffff816ff85a>] SyS_mmap_pgoff+0x34a/0x580 mm/mmap.c:1411 [< inline >] SYSC_mmap arch/x86/kernel/sys_x86_64.c:95 [<ffffffff811aeeb6>] SyS_mmap+0x16/0x20 arch/x86/kernel/sys_x86_64.c:86 [<ffffffff866531b6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 other info that might help us debug this: Possible unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&mm->mmap_sem); lock(&runtime->oss.params_lock); lock(&mm->mmap_sem); lock(&runtime->oss.params_lock); *** DEADLOCK *** 1 lock held by syz-executor/14254: #0: (&mm->mmap_sem){++++++}, at: [<ffffffff816b267c>] vm_mmap_pgoff+0x12c/0x1b0 mm/util.c:327 stack backtrace: CPU: 2 PID: 14254 Comm: syz-executor Not tainted 4.5.0-rc1+ #305 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff88003214f780 ffffffff82be11ad ffffffff8959ac60 ffffffff8959ac60 ffffffff89573f60 ffff88003214f7d0 ffffffff814512a8 ffff8800333cdf00 ffff8800333ce742 0000000000000000 ffff8800333ce720 Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82be11ad>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 [<ffffffff814512a8>] print_circular_bug+0x288/0x340 kernel/locking/lockdep.c:1228 [< inline >] check_prev_add kernel/locking/lockdep.c:1855 [< inline >] check_prevs_add kernel/locking/lockdep.c:1960 [< inline >] validate_chain kernel/locking/lockdep.c:2146 [<ffffffff8145807b>] __lock_acquire+0x31eb/0x4700 kernel/locking/lockdep.c:3208 [<ffffffff8145b9dc>] lock_acquire+0x1dc/0x430 kernel/locking/lockdep.c:3587 [< inline >] __mutex_lock_common kernel/locking/mutex.c:518 [<ffffffff8664891c>] mutex_lock_interruptible_nested+0xbc/0xbe0 kernel/locking/mutex.c:647 [<ffffffff8528a504>] snd_pcm_oss_change_params+0xd4/0x3540 sound/core/oss/pcm_oss.c:852 [<ffffffff8528f01d>] snd_pcm_oss_mmap+0x3dd/0x4c0 sound/core/oss/pcm_oss.c:2807 [<ffffffff81705747>] mmap_region+0x897/0x1010 mm/mmap.c:1624 [<ffffffff81706614>] do_mmap+0x754/0x990 mm/mmap.c:1403 [< inline >] do_mmap_pgoff include/linux/mm.h:1982 [<ffffffff816b26af>] vm_mmap_pgoff+0x15f/0x1b0 mm/util.c:328 [< inline >] SYSC_mmap_pgoff mm/mmap.c:1453 [<ffffffff816ff85a>] SyS_mmap_pgoff+0x34a/0x580 mm/mmap.c:1411 [< inline >] SYSC_mmap arch/x86/kernel/sys_x86_64.c:95 [<ffffffff811aeeb6>] SyS_mmap+0x16/0x20 arch/x86/kernel/sys_x86_64.c:86 On commit 26cd83670f2f5a3d5b5514a1f7d96567cdb9558b.

2 2

[alsa-devel] sound: use-after-free in snd_timer_notify1
by Dmitry Vyukov 31 Jan '16

31 Jan '16

Hello, The following program causes use-after-free in snd_timer_notify1: ================================================================== BUG: KASAN: use-after-free in snd_timer_notify1+0x411/0x460 at addr ffff880035a433e0 Read of size 8 by task syz-executor/11116 ============================================================================= BUG kmalloc-256 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- INFO: Allocated in snd_timer_instance_new+0x52/0x3a0 age=1 cpu=1 pid=11106 [< inline >] kzalloc include/linux/slab.h:607 [< none >] snd_timer_instance_new+0x52/0x3a0 sound/core/timer.c:105 [< none >] snd_timer_open+0x522/0xce0 sound/core/timer.c:288 [< none >] snd_seq_timer_open+0x223/0x560 sound/core/seq/seq_timer.c:279 [< none >] snd_seq_queue_use+0x147/0x230 sound/core/seq/seq_queue.c:528 [< none >] snd_seq_queue_alloc+0x36a/0x4d0 sound/core/seq/seq_queue.c:199 [< none >] snd_seq_ioctl_create_queue+0xdb/0x2b0 sound/core/seq/seq_clientmgr.c:1536 [< none >] snd_seq_do_ioctl+0x19d/0x1c0 sound/core/seq/seq_clientmgr.c:2209 [< none >] snd_seq_ioctl+0x54/0xa0 sound/core/seq/seq_clientmgr.c:2224 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 INFO: Freed in snd_timer_close+0x3a8/0x700 age=19 cpu=3 pid=11114 [< none >] kfree+0x2b7/0x2e0 mm/slub.c:3664 [< none >] snd_timer_close+0x3a8/0x700 sound/core/timer.c:368 [< none >] snd_seq_timer_close+0x97/0x130 sound/core/seq/seq_timer.c:312 [< none >] snd_seq_queue_timer_close+0x28/0x50 sound/core/seq/seq_queue.c:475 [< none >] snd_seq_ioctl_set_queue_timer+0x159/0x300 sound/core/seq/seq_clientmgr.c:1809 [< none >] snd_seq_do_ioctl+0x19d/0x1c0 sound/core/seq/seq_clientmgr.c:2209 [< none >] snd_seq_ioctl+0x54/0xa0 sound/core/seq/seq_clientmgr.c:2224 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 INFO: Slab 0xffffea0000d69000 objects=22 used=16 fp=0xffff880035a42d80 flags=0x1fffc0000004080 INFO: Object 0xffff880035a43330 @offset=13104 fp=0xffff880064ccee20 CPU: 0 PID: 11116 Comm: syz-executor Tainted: G B 4.4.0+ #276 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff880064bcf560 ffffffff82999e2d ffff88003e807000 ffff880035a43330 ffff880035a40000 ffff880064bcf590 ffffffff81757354 ffff88003e807000 ffffea0000d69000 ffff880035a43330 ffff880064bcf718 Call Trace: [<ffffffff817609ee>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:295 [<ffffffff84f40da1>] snd_timer_notify1+0x411/0x460 sound/core/timer.c:416 [<ffffffff84f41025>] _snd_timer_stop+0x235/0x5c0 sound/core/timer.c:524 [<ffffffff84f41d3a>] snd_timer_pause+0x1a/0x20 sound/core/timer.c:583 [< inline >] snd_seq_timer_stop sound/core/seq/seq_timer.c:325 [<ffffffff84fc1658>] snd_seq_timer_start+0x148/0x1a0 sound/core/seq/seq_timer.c:366 [< inline >] snd_seq_queue_process_event sound/core/seq/seq_queue.c:687 [<ffffffff84fbc344>] snd_seq_control_queue+0x304/0x8b0 sound/core/seq/seq_queue.c:748 [<ffffffff84fc1da5>] event_input_timer+0x25/0x30 sound/core/seq/seq_system.c:118 [<ffffffff84fb4c14>] snd_seq_deliver_single_event.constprop.11+0x3f4/0x740 sound/core/seq/seq_clientmgr.c:634 [<ffffffff84fb5082>] snd_seq_deliver_event+0x122/0x800 sound/core/seq/seq_clientmgr.c:828 [<ffffffff84fb65a9>] snd_seq_dispatch_event+0xf9/0x510 sound/core/seq/seq_clientmgr.c:902 [<ffffffff84fba85b>] snd_seq_check_queue+0x3fb/0x560 sound/core/seq/seq_queue.c:293 [<ffffffff84fbac0d>] snd_seq_enqueue_event+0x24d/0x400 sound/core/seq/seq_queue.c:357 [<ffffffff84fb5974>] snd_seq_client_enqueue_event+0x214/0x430 sound/core/seq/seq_clientmgr.c:961 [<ffffffff84fb5e7f>] snd_seq_write+0x2ef/0x570 sound/core/seq/seq_clientmgr.c:1075 [<ffffffff817b0323>] __vfs_write+0x113/0x480 fs/read_write.c:528 [<ffffffff817b1db7>] vfs_write+0x167/0x4a0 fs/read_write.c:577 [< inline >] SYSC_write fs/read_write.c:624 [<ffffffff817b50a1>] SyS_write+0x111/0x220 fs/read_write.c:616 [<ffffffff86336c36>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 ================================================================== // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> long r[254]; int main() { memset(r, -1, sizeof(r)); r[0] = syscall(SYS_mmap, 0x20000000ul, 0x1c000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); r[2] = open("/dev/snd/seq", 0x1ul, 0, 0, 0); *(uint32_t*)0x20006000 = (uint32_t)0xffffffffffffffff; *(uint32_t*)0x20006004 = (uint32_t)0x10000; *(uint32_t*)0x20006008 = (uint32_t)0x2; *(uint8_t*)0x2000600c = (uint8_t)0x7; *(uint8_t*)0x2000600d = (uint8_t)0x2; *(uint8_t*)0x2000600e = (uint8_t)0xec0; *(uint8_t*)0x2000600f = (uint8_t)0x4; *(uint8_t*)0x20006010 = (uint8_t)0x7; *(uint8_t*)0x20006011 = (uint8_t)0x9; *(uint8_t*)0x20006012 = (uint8_t)0x80000000; *(uint8_t*)0x20006013 = (uint8_t)0x2; *(uint8_t*)0x20006014 = (uint8_t)0x2; *(uint8_t*)0x20006015 = (uint8_t)0x3; *(uint8_t*)0x20006016 = (uint8_t)0x8; *(uint8_t*)0x20006017 = (uint8_t)0x9; *(uint8_t*)0x20006018 = (uint8_t)0x4; *(uint8_t*)0x20006019 = (uint8_t)0xffffffffffffff5c; *(uint8_t*)0x2000601a = (uint8_t)0x5ab; *(uint8_t*)0x2000601b = (uint8_t)0x4; *(uint8_t*)0x2000601c = (uint8_t)0x0; *(uint8_t*)0x2000601d = (uint8_t)0x38; *(uint8_t*)0x2000601e = (uint8_t)0x9d; *(uint8_t*)0x2000601f = (uint8_t)0x8; *(uint8_t*)0x20006020 = (uint8_t)0x3; *(uint8_t*)0x20006021 = (uint8_t)0x47221059; *(uint8_t*)0x20006022 = (uint8_t)0x400; *(uint8_t*)0x20006023 = (uint8_t)0x1000; *(uint8_t*)0x20006024 = (uint8_t)0x2; *(uint8_t*)0x20006025 = (uint8_t)0x3; *(uint8_t*)0x20006026 = (uint8_t)0x3; *(uint8_t*)0x20006027 = (uint8_t)0x80000000; *(uint8_t*)0x20006028 = (uint8_t)0x5; *(uint8_t*)0x20006029 = (uint8_t)0x200; *(uint8_t*)0x2000602a = (uint8_t)0x1; *(uint8_t*)0x2000602b = (uint8_t)0x30; *(uint8_t*)0x2000602c = (uint8_t)0x4; *(uint8_t*)0x2000602d = (uint8_t)0x0; *(uint8_t*)0x2000602e = (uint8_t)0x3; *(uint8_t*)0x2000602f = (uint8_t)0xffffffff8a5c645b; *(uint8_t*)0x20006030 = (uint8_t)0x4; *(uint8_t*)0x20006031 = (uint8_t)0x1; *(uint8_t*)0x20006032 = (uint8_t)0x3ff; *(uint8_t*)0x20006033 = (uint8_t)0x200; *(uint8_t*)0x20006034 = (uint8_t)0x3; *(uint8_t*)0x20006035 = (uint8_t)0xea3e; *(uint8_t*)0x20006036 = (uint8_t)0x9; *(uint8_t*)0x20006037 = (uint8_t)0x200; *(uint8_t*)0x20006038 = (uint8_t)0x0; *(uint8_t*)0x20006039 = (uint8_t)0x5; *(uint8_t*)0x2000603a = (uint8_t)0xfdc; *(uint8_t*)0x2000603b = (uint8_t)0x1000; *(uint8_t*)0x2000603c = (uint8_t)0x467; *(uint8_t*)0x2000603d = (uint8_t)0xea; *(uint8_t*)0x2000603e = (uint8_t)0x40; *(uint8_t*)0x2000603f = (uint8_t)0x9e98; *(uint8_t*)0x20006040 = (uint8_t)0x7; *(uint8_t*)0x20006041 = (uint8_t)0x7; *(uint8_t*)0x20006042 = (uint8_t)0x0; *(uint8_t*)0x20006043 = (uint8_t)0x20; *(uint8_t*)0x20006044 = (uint8_t)0x1; *(uint8_t*)0x20006045 = (uint8_t)0x4; *(uint8_t*)0x20006046 = (uint8_t)0x2; *(uint8_t*)0x20006047 = (uint8_t)0x9; *(uint8_t*)0x20006048 = (uint8_t)0x5; *(uint8_t*)0x20006049 = (uint8_t)0x6; *(uint8_t*)0x2000604a = (uint8_t)0x8f2; *(uint8_t*)0x2000604b = (uint8_t)0x0; *(uint32_t*)0x2000604c = (uint32_t)0x6; *(uint8_t*)0x20006050 = (uint8_t)0x0; *(uint8_t*)0x20006051 = (uint8_t)0x0; *(uint8_t*)0x20006052 = (uint8_t)0x0; *(uint8_t*)0x20006053 = (uint8_t)0x0; *(uint8_t*)0x20006054 = (uint8_t)0x0; *(uint8_t*)0x20006055 = (uint8_t)0x0; *(uint8_t*)0x20006056 = (uint8_t)0x0; *(uint8_t*)0x20006057 = (uint8_t)0x0; *(uint8_t*)0x20006058 = (uint8_t)0x0; *(uint8_t*)0x20006059 = (uint8_t)0x0; *(uint8_t*)0x2000605a = (uint8_t)0x0; *(uint8_t*)0x2000605b = (uint8_t)0x0; *(uint8_t*)0x2000605c = (uint8_t)0x0; *(uint8_t*)0x2000605d = (uint8_t)0x0; *(uint8_t*)0x2000605e = (uint8_t)0x0; *(uint8_t*)0x2000605f = (uint8_t)0x0; *(uint8_t*)0x20006060 = (uint8_t)0x0; *(uint8_t*)0x20006061 = (uint8_t)0x0; *(uint8_t*)0x20006062 = (uint8_t)0x0; *(uint8_t*)0x20006063 = (uint8_t)0x0; *(uint8_t*)0x20006064 = (uint8_t)0x0; *(uint8_t*)0x20006065 = (uint8_t)0x0; *(uint8_t*)0x20006066 = (uint8_t)0x0; *(uint8_t*)0x20006067 = (uint8_t)0x0; *(uint8_t*)0x20006068 = (uint8_t)0x0; *(uint8_t*)0x20006069 = (uint8_t)0x0; *(uint8_t*)0x2000606a = (uint8_t)0x0; *(uint8_t*)0x2000606b = (uint8_t)0x0; *(uint8_t*)0x2000606c = (uint8_t)0x0; *(uint8_t*)0x2000606d = (uint8_t)0x0; *(uint8_t*)0x2000606e = (uint8_t)0x0; *(uint8_t*)0x2000606f = (uint8_t)0x0; *(uint8_t*)0x20006070 = (uint8_t)0x0; *(uint8_t*)0x20006071 = (uint8_t)0x0; *(uint8_t*)0x20006072 = (uint8_t)0x0; *(uint8_t*)0x20006073 = (uint8_t)0x0; *(uint8_t*)0x20006074 = (uint8_t)0x0; *(uint8_t*)0x20006075 = (uint8_t)0x0; *(uint8_t*)0x20006076 = (uint8_t)0x0; *(uint8_t*)0x20006077 = (uint8_t)0x0; *(uint8_t*)0x20006078 = (uint8_t)0x0; *(uint8_t*)0x20006079 = (uint8_t)0x0; *(uint8_t*)0x2000607a = (uint8_t)0x0; *(uint8_t*)0x2000607b = (uint8_t)0x0; *(uint8_t*)0x2000607c = (uint8_t)0x0; *(uint8_t*)0x2000607d = (uint8_t)0x0; *(uint8_t*)0x2000607e = (uint8_t)0x0; *(uint8_t*)0x2000607f = (uint8_t)0x0; *(uint8_t*)0x20006080 = (uint8_t)0x0; *(uint8_t*)0x20006081 = (uint8_t)0x0; *(uint8_t*)0x20006082 = (uint8_t)0x0; *(uint8_t*)0x20006083 = (uint8_t)0x0; *(uint8_t*)0x20006084 = (uint8_t)0x0; *(uint8_t*)0x20006085 = (uint8_t)0x0; *(uint8_t*)0x20006086 = (uint8_t)0x0; *(uint8_t*)0x20006087 = (uint8_t)0x0; *(uint8_t*)0x20006088 = (uint8_t)0x0; *(uint8_t*)0x20006089 = (uint8_t)0x0; *(uint8_t*)0x2000608a = (uint8_t)0x0; *(uint8_t*)0x2000608b = (uint8_t)0x0; r[131] = syscall(SYS_ioctl, r[2], 0xc08c5332ul, 0x20006000ul, 0, 0, 0); *(uint32_t*)0x20006fd7 = (uint32_t)0x0; *(uint32_t*)0x20006fdb = (uint32_t)0x0; *(uint32_t*)0x20006fdf = (uint32_t)0x6; *(uint32_t*)0x20006fe3 = (uint32_t)0x81; *(uint32_t*)0x20006fe7 = (uint32_t)0x7; *(uint32_t*)0x20006feb = (uint32_t)0x9; *(uint32_t*)0x20006fef = (uint32_t)0x401; *(uint8_t*)0x20006ff3 = (uint8_t)0x0; *(uint8_t*)0x20006ff4 = (uint8_t)0x0; *(uint8_t*)0x20006ff5 = (uint8_t)0x0; *(uint8_t*)0x20006ff6 = (uint8_t)0x0; *(uint8_t*)0x20006ff7 = (uint8_t)0x0; *(uint8_t*)0x20006ff8 = (uint8_t)0x0; *(uint8_t*)0x20006ff9 = (uint8_t)0x0; *(uint8_t*)0x20006ffa = (uint8_t)0x0; *(uint8_t*)0x20006ffb = (uint8_t)0x0; *(uint8_t*)0x20006ffc = (uint8_t)0x0; *(uint8_t*)0x20006ffd = (uint8_t)0x0; *(uint8_t*)0x20006ffe = (uint8_t)0x0; *(uint8_t*)0x20006fff = (uint8_t)0x0; *(uint8_t*)0x20007000 = (uint8_t)0x0; *(uint8_t*)0x20007001 = (uint8_t)0x0; *(uint8_t*)0x20007002 = (uint8_t)0x0; *(uint8_t*)0x20007003 = (uint8_t)0x0; *(uint8_t*)0x20007004 = (uint8_t)0x0; *(uint8_t*)0x20007005 = (uint8_t)0x0; *(uint8_t*)0x20007006 = (uint8_t)0x0; *(uint8_t*)0x20007007 = (uint8_t)0x0; *(uint8_t*)0x20007008 = (uint8_t)0x0; *(uint8_t*)0x20007009 = (uint8_t)0x0; *(uint8_t*)0x2000700a = (uint8_t)0x0; *(uint8_t*)0x2000700b = (uint8_t)0x0; *(uint8_t*)0x2000700c = (uint8_t)0x0; *(uint8_t*)0x2000700d = (uint8_t)0x0; *(uint8_t*)0x2000700e = (uint8_t)0x0; *(uint8_t*)0x2000700f = (uint8_t)0x0; *(uint8_t*)0x20007010 = (uint8_t)0x0; *(uint8_t*)0x20007011 = (uint8_t)0x0; *(uint8_t*)0x20007012 = (uint8_t)0x0; *(uint8_t*)0x20007013 = (uint8_t)0x0; *(uint8_t*)0x20007014 = (uint8_t)0x0; *(uint8_t*)0x20007015 = (uint8_t)0x0; *(uint8_t*)0x20007016 = (uint8_t)0x0; *(uint8_t*)0x20007017 = (uint8_t)0x0; *(uint8_t*)0x20007018 = (uint8_t)0x0; *(uint8_t*)0x20007019 = (uint8_t)0x0; *(uint8_t*)0x2000701a = (uint8_t)0x0; *(uint8_t*)0x2000701b = (uint8_t)0x0; *(uint8_t*)0x2000701c = (uint8_t)0x0; *(uint8_t*)0x2000701d = (uint8_t)0x0; *(uint8_t*)0x2000701e = (uint8_t)0x0; *(uint8_t*)0x2000701f = (uint8_t)0x0; *(uint8_t*)0x20007020 = (uint8_t)0x0; *(uint8_t*)0x20007021 = (uint8_t)0x0; *(uint8_t*)0x20007022 = (uint8_t)0x0; *(uint8_t*)0x20007023 = (uint8_t)0x0; *(uint8_t*)0x20007024 = (uint8_t)0x0; *(uint8_t*)0x20007025 = (uint8_t)0x0; *(uint8_t*)0x20007026 = (uint8_t)0x0; *(uint8_t*)0x20007027 = (uint8_t)0x0; *(uint8_t*)0x20007028 = (uint8_t)0x0; *(uint8_t*)0x20007029 = (uint8_t)0x0; *(uint8_t*)0x2000702a = (uint8_t)0x0; *(uint8_t*)0x2000702b = (uint8_t)0x0; *(uint8_t*)0x2000702c = (uint8_t)0x0; *(uint8_t*)0x2000702d = (uint8_t)0x0; *(uint8_t*)0x2000702e = (uint8_t)0x0; *(uint8_t*)0x2000702f = (uint8_t)0x0; *(uint8_t*)0x20007030 = (uint8_t)0x0; *(uint8_t*)0x20007031 = (uint8_t)0x0; *(uint8_t*)0x20007032 = (uint8_t)0x0; r[203] = syscall(SYS_ioctl, r[2], 0x40605346ul, 0x20006fd7ul, 0, 0, 0); *(uint8_t*)0x20005000 = (uint8_t)0x3ff; *(uint8_t*)0x20005001 = (uint8_t)0x2e; *(uint8_t*)0x20005002 = (uint8_t)0x1; *(uint8_t*)0x20005003 = (uint8_t)0x8; *(uint32_t*)0x2000500c = (uint32_t)0x4; *(uint8_t*)0x20005010 = (uint8_t)0x8001; *(uint8_t*)0x20005011 = (uint8_t)0x2; *(uint8_t*)0x20005012 = (uint8_t)0xfffffffffffffffa; *(uint8_t*)0x20005013 = (uint8_t)0xffff; *(uint8_t*)0x2000501c = (uint8_t)0x5; *(uint8_t*)0x2000501d = (uint8_t)0x2; *(uint8_t*)0x2000501e = (uint8_t)0x5; *(uint8_t*)0x2000501f = (uint8_t)0x0; *(uint8_t*)0x20005020 = (uint8_t)0x5; *(uint8_t*)0x20005021 = (uint8_t)0x100000000; *(uint8_t*)0x20005022 = (uint8_t)0x7; *(uint8_t*)0x20005023 = (uint8_t)0xfffffffffffff2fb; *(uint32_t*)0x2000502c = (uint32_t)0x8; *(uint8_t*)0x20005030 = (uint8_t)0x80; *(uint8_t*)0x20005031 = (uint8_t)0x485; *(uint8_t*)0x20005032 = (uint8_t)0x4; *(uint8_t*)0x20005033 = (uint8_t)0x4; *(uint8_t*)0x2000503c = (uint8_t)0x9; *(uint8_t*)0x2000503d = (uint8_t)0x7; *(uint8_t*)0x2000503e = (uint8_t)0x20; *(uint8_t*)0x2000503f = (uint8_t)0x0; *(uint8_t*)0x20005040 = (uint8_t)0x20; *(uint8_t*)0x20005041 = (uint8_t)0x9; *(uint8_t*)0x20005042 = (uint8_t)0x2; *(uint8_t*)0x20005043 = (uint8_t)0x2; *(uint64_t*)0x20005058 = (uint64_t)0x0; *(uint64_t*)0x20005060 = (uint64_t)0x0; *(uint8_t*)0x20005068 = (uint8_t)0x1c8; *(uint8_t*)0x20005069 = (uint8_t)0xfffffffffffffff7; *(uint8_t*)0x2000506a = (uint8_t)0x1ff; *(uint8_t*)0x2000506b = (uint8_t)0x9f9f; *(uint32_t*)0x2000507c = (uint32_t)0x90; *(uint32_t*)0x20005080 = (uint32_t)0x40; *(uint32_t*)0x20005084 = (uint32_t)0x0; *(uint8_t*)0x20005088 = (uint8_t)0x0; *(uint8_t*)0x20005089 = (uint8_t)0x4; *(uint8_t*)0x2000508a = (uint8_t)0x1; *(uint8_t*)0x2000508b = (uint8_t)0xffffffff80000001; *(uint32_t*)0x20005094 = (uint32_t)0x9; *(uint8_t*)0x20005098 = (uint8_t)0x8; *(uint8_t*)0x20005099 = (uint8_t)0x7; *(uint8_t*)0x2000509a = (uint8_t)0xfff; *(uint8_t*)0x2000509b = (uint8_t)0x8; *(uint32_t*)0x200050a8 = (uint32_t)0xbd1e; r[253] = syscall(SYS_write, r[2], 0x20005000ul, 0x95cul, 0, 0, 0); return 0; } I am on commit 30f05309bde49295e02e45c7e615f73aa4e0ccc2 (Jan 20) + the following pending patch from Takashi: diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index f845ecf..656d9a9 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -90,7 +90,7 @@ static int snd_hrtimer_start(struct snd_timer *t) struct snd_hrtimer *stime = t->private_data; atomic_set(&stime->running, 0); - hrtimer_cancel(&stime->hrt); + hrtimer_try_to_cancel(&stime->hrt); hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution), HRTIMER_MODE_REL); atomic_set(&stime->running, 1); @@ -101,6 +101,7 @@ static int snd_hrtimer_stop(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; atomic_set(&stime->running, 0); + hrtimer_try_to_cancel(&stime->hrt); return 0; }

2 1

[alsa-devel] [PATCH v2 1/3] ALSA: mts64: whitespace fixes
by Sudip Mukherjee 30 Jan '16

30 Jan '16

checkpatch was complaining about trailing whitespace, multiple blank lines, space before start of a line and a required space before opening brace. This patch takes care of these issues related to whitespace changes and there was no other change. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v2: combined 3 patches of v1 series into this patch. sound/drivers/mts64.c | 112 ++++++++++++++++++++++++-------------------------- 1 file changed, 54 insertions(+), 58 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 2a008a9..f7d938d 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1,14 +1,14 @@ -/* +/* * ALSA Driver for Ego Systems Inc. (ESI) Miditerminal 4140 * Copyright (c) 2006 by Matthias König <mk(a)phasorlab.de> * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * @@ -38,7 +38,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -static struct platform_device *platform_devices[SNDRV_CARDS]; +static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; module_param_array(index, int, NULL, S_IRUGO); @@ -92,7 +92,7 @@ static int snd_mts64_create(struct snd_card *card, *rchip = NULL; mts = kzalloc(sizeof(struct mts64), GFP_KERNEL); - if (mts == NULL) + if (mts == NULL) return -ENOMEM; /* Init chip specific data */ @@ -118,9 +118,9 @@ static int snd_mts64_create(struct snd_card *card, /* Control Bits */ #define MTS64_CTL_READOUT 0x08 /* enable readout */ -#define MTS64_CTL_WRITE_CMD 0x06 -#define MTS64_CTL_WRITE_DATA 0x02 -#define MTS64_CTL_STROBE 0x01 +#define MTS64_CTL_WRITE_CMD 0x06 +#define MTS64_CTL_WRITE_DATA 0x02 +#define MTS64_CTL_STROBE 0x01 /* Command */ #define MTS64_CMD_RESET 0xfe @@ -130,7 +130,7 @@ static int snd_mts64_create(struct snd_card *card, #define MTS64_CMD_SMPTE_STOP 0xef #define MTS64_CMD_SMPTE_FPS_24 0xe3 #define MTS64_CMD_SMPTE_FPS_25 0xe2 -#define MTS64_CMD_SMPTE_FPS_2997 0xe4 +#define MTS64_CMD_SMPTE_FPS_2997 0xe4 #define MTS64_CMD_SMPTE_FPS_30D 0xe1 #define MTS64_CMD_SMPTE_FPS_30 0xe0 #define MTS64_CMD_COM_OPEN 0xf8 /* setting the communication mode */ @@ -159,7 +159,6 @@ static void mts64_write_command(struct parport *p, u8 c); static void mts64_write_data(struct parport *p, u8 c); static void mts64_write_midi(struct mts64 *mts, u8 c, int midiport); - /* Enables the readout procedure * * Before we can read a midi byte from the device, we have to set @@ -171,10 +170,10 @@ static void mts64_enable_readout(struct parport *p) c = parport_read_control(p); c |= MTS64_CTL_READOUT; - parport_write_control(p, c); + parport_write_control(p, c); } -/* Disables readout +/* Disables readout * * Readout is disabled by clearing bit 3 of control */ @@ -201,9 +200,9 @@ static int mts64_device_ready(struct parport *p) for (i = 0; i < 0xffff; ++i) { c = parport_read_status(p); c &= MTS64_STAT_BSY; - if (c != 0) + if (c != 0) return 1; - } + } return 0; } @@ -234,7 +233,7 @@ static int mts64_device_init(struct parport *p) return -EIO; } -/* +/* * Opens the device (set communication mode) */ static int mts64_device_open(struct mts64 *mts) @@ -248,7 +247,7 @@ static int mts64_device_open(struct mts64 *mts) return 0; } -/* +/* * Close device (clear communication mode) */ static int mts64_device_close(struct mts64 *mts) @@ -265,14 +264,14 @@ static int mts64_device_close(struct mts64 *mts) } /* map hardware port to substream number - * + * * When reading a byte from the device, the device tells us * on what port the byte is. This HW port has to be mapped to * the midiport (substream number). * substream 0-3 are Midiports 1-4 * substream 4 is SMPTE Timecode * The mapping is done by the table: - * HW | 0 | 1 | 2 | 3 | 4 + * HW | 0 | 1 | 2 | 3 | 4 * SW | 0 | 1 | 4 | 2 | 3 */ static u8 mts64_map_midi_input(u8 c) @@ -282,10 +281,9 @@ static u8 mts64_map_midi_input(u8 c) return map[c]; } - /* Probe parport for device * - * Do we have a Miditerminal 4140 on parport? + * Do we have a Miditerminal 4140 on parport? * Returns: * 0 device found * -ENODEV no device @@ -298,13 +296,13 @@ static int mts64_probe(struct parport *p) mts64_write_command(p, MTS64_CMD_PROBE); msleep(50); - + c = mts64_read(p); c &= 0x00ff; - if (c != MTS64_CMD_PROBE) + if (c != MTS64_CMD_PROBE) return -ENODEV; - else + else return 0; } @@ -330,7 +328,7 @@ static u16 mts64_read(struct parport *p) /* Read a byte from device * * Note, that readout mode has to be enabled. - * readout procedure is as follows: + * readout procedure is as follows: * - Write number of the Bit to read to DATA * - Read STATUS * - Bit 5 of STATUS indicates if Bit is set @@ -348,10 +346,10 @@ static u8 mts64_read_char(struct parport *p) parport_write_data(p, i); c >>= 1; status = parport_read_status(p); - if (status & MTS64_STAT_BIT_SET) + if (status & MTS64_STAT_BIT_SET) c |= 0x80; } - + return c; } @@ -369,9 +367,9 @@ static void mts64_smpte_start(struct parport *p, u8 seconds, u8 frames, u8 idx) { - static u8 fps[5] = { MTS64_CMD_SMPTE_FPS_24, + static u8 fps[5] = { MTS64_CMD_SMPTE_FPS_24, MTS64_CMD_SMPTE_FPS_25, - MTS64_CMD_SMPTE_FPS_2997, + MTS64_CMD_SMPTE_FPS_2997, MTS64_CMD_SMPTE_FPS_30D, MTS64_CMD_SMPTE_FPS_30 }; @@ -405,7 +403,7 @@ static void mts64_write_command(struct parport *p, u8 c) parport_write_control(p, MTS64_CTL_WRITE_CMD); } -/* Write a data byte to device +/* Write a data byte to device */ static void mts64_write_data(struct parport *p, u8 c) { @@ -652,7 +650,6 @@ static struct snd_kcontrol_new mts64_ctl_smpte_fps = { .put = snd_mts64_ctl_smpte_fps_put }; - static int snd_mts64_ctl_create(struct snd_card *card, struct mts64 *mts) { @@ -669,7 +666,7 @@ static int snd_mts64_ctl_create(struct snd_card *card, for (i = 0; control[i]; ++i) { err = snd_ctl_add(card, snd_ctl_new1(control[i], mts)); if (err < 0) { - snd_printd("Cannot create control: %s\n", + snd_printd("Cannot create control: %s\n", control[i]->name); return err; } @@ -688,8 +685,8 @@ static int snd_mts64_rawmidi_open(struct snd_rawmidi_substream *substream) struct mts64 *mts = substream->rmidi->private_data; if (mts->open_count == 0) { - /* We don't need a spinlock here, because this is just called - if the device has not been opened before. + /* We don't need a spinlock here, because this is just called + if the device has not been opened before. So there aren't any IRQs from the device */ mts64_device_open(mts); @@ -746,8 +743,8 @@ static void snd_mts64_rawmidi_input_trigger(struct snd_rawmidi_substream *substr if (up) mts->mode[substream->number] |= MTS64_MODE_INPUT_TRIGGERED; else - mts->mode[substream->number] &= ~MTS64_MODE_INPUT_TRIGGERED; - + mts->mode[substream->number] &= ~MTS64_MODE_INPUT_TRIGGERED; + spin_unlock_irqrestore(&mts->lock, flags); } @@ -771,42 +768,42 @@ static int snd_mts64_rawmidi_create(struct snd_card *card) struct snd_rawmidi_substream *substream; struct list_head *list; int err; - - err = snd_rawmidi_new(card, CARD_NAME, 0, - MTS64_NUM_OUTPUT_PORTS, - MTS64_NUM_INPUT_PORTS, + + err = snd_rawmidi_new(card, CARD_NAME, 0, + MTS64_NUM_OUTPUT_PORTS, + MTS64_NUM_INPUT_PORTS, &rmidi); - if (err < 0) + if (err < 0) return err; rmidi->private_data = mts; strcpy(rmidi->name, CARD_NAME); rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | - SNDRV_RAWMIDI_INFO_INPUT | - SNDRV_RAWMIDI_INFO_DUPLEX; + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; mts->rmidi = rmidi; /* register rawmidi ops */ - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mts64_rawmidi_output_ops); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mts64_rawmidi_input_ops); /* name substreams */ /* output */ - list_for_each(list, + list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { substream = list_entry(list, struct snd_rawmidi_substream, list); sprintf(substream->name, "Miditerminal %d", substream->number+1); } /* input */ - list_for_each(list, + list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) { substream = list_entry(list, struct snd_rawmidi_substream, list); mts->midi_input_substream[substream->number] = substream; - switch(substream->number) { + switch (substream->number) { case MTS64_SMPTE_SUBSTREAM: strcpy(substream->name, "Miditerminal SMPTE"); break; @@ -840,7 +837,7 @@ static void snd_mts64_interrupt(void *private) if (status & MTS64_STAT_PORT) { mts->current_midi_input_port = mts64_map_midi_input(data); } else { - if (mts->current_midi_input_port == -1) + if (mts->current_midi_input_port == -1) goto __out; substream = mts->midi_input_substream[mts->current_midi_input_port]; if (mts->mode[substream->number] & MTS64_MODE_INPUT_TRIGGERED) @@ -860,7 +857,7 @@ static int snd_mts64_probe_port(struct parport *p) 0, NULL); if (!pardev) return -EIO; - + if (parport_claim(pardev)) { parport_unregister_device(pardev); return -EIO; @@ -944,7 +941,7 @@ static int snd_mts64_probe(struct platform_device *pdev) if (dev >= SNDRV_CARDS) return -ENODEV; - if (!enable[dev]) + if (!enable[dev]) return -ENOENT; if ((err = snd_mts64_probe_port(p)) < 0) return err; @@ -957,7 +954,7 @@ static int snd_mts64_probe(struct platform_device *pdev) } strcpy(card->driver, DRIVER_NAME); strcpy(card->shortname, "ESI " CARD_NAME); - sprintf(card->longname, "%s at 0x%lx, irq %i", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); pardev = parport_register_device(p, /* port */ @@ -980,7 +977,7 @@ static int snd_mts64_probe(struct platform_device *pdev) } card->private_data = mts; card->private_free = snd_mts64_card_private_free; - + if ((err = snd_mts64_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); goto __err; @@ -1024,7 +1021,6 @@ static int snd_mts64_remove(struct platform_device *pdev) return 0; } - static struct platform_driver snd_mts64_driver = { .probe = snd_mts64_probe, .remove = snd_mts64_remove, @@ -1045,7 +1041,7 @@ static void snd_mts64_unregister_all(void) platform_device_unregister(platform_devices[i]); platform_devices[i] = NULL; } - } + } platform_driver_unregister(&snd_mts64_driver); parport_unregister_driver(&mts64_parport_driver); } -- 1.9.1

1 2

[alsa-devel] [PATCH v2] emu10k1: correctly handling failed thread creation
by Insu Yun 29 Jan '16

29 Jan '16

Since kthread_create can be failed, it needs to check whether error occurred and return error code. Signed-off-by: Insu Yun <wuninsu(a)gmail.com> --- sound/pci/emu10k1/emu10k1_main.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 28e2f8b..8914534 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1141,6 +1141,14 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) emu->emu1010.firmware_thread = kthread_create(emu1010_firmware_thread, emu, "emu1010_firmware"); + if (IS_ERR(emu->emu1010.firmware_thread)) { + err = PTR_ERR(emu->emu1010.firmware_thread); + emu->emu1010.firmware_thread = NULL; + dev_info(emu->card->dev, + "emu1010: Creating thread failed\n"); + return err; + } + wake_up_process(emu->emu1010.firmware_thread); } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: add cs4271 and cs4272 support
by Felipe F. Tonello 29 Jan '16

29 Jan '16

add cs4271 and cs42727 support for fsl-asoc-card Signed-off-by: Felipe F. Tonello <eu(a)felipetonello.com> --- .../devicetree/bindings/sound/imx-audio-cs427x.txt | 47 ++++++++++++++++++++++ sound/soc/fsl/Kconfig | 4 +- sound/soc/fsl/fsl-asoc-card.c | 7 ++++ 3 files changed, 56 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-cs427x.txt diff --git a/Documentation/devicetree/bindings/sound/imx-audio-cs427x.txt b/Documentation/devicetree/bindings/sound/imx-audio-cs427x.txt new file mode 100644 index 000000000000..295f60b19418 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-cs427x.txt @@ -0,0 +1,47 @@ +Freescale i.MX audio complex with CS4271 or CS4272 codec + +Required properties: + + - compatible : "fsl,imx-audio-cs427x" + + - model : The user-visible name of this sound complex + + - ssi-controller : The phandle of the i.MX SSI controller + + - audio-codec : The phandle of the CS4271 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, CS427x + pins, and the jacks on the board: + + Board connectors: + * Mic Jack + * Headphone Jack + + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) + + - mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx6-rex-cs427x", + "fsl,imx-audio-cs427x"; + model = "audio-cs427x"; + ssi-controller = <&ssi1>; + audio-codec = <&cs4271>; + audio-routing = + "Mic Jack", "AINA", + "Mic Jack", "AINB", + "Headphone Jack", "AOUTA+", + "Headphone Jack", "AOUTA-", + "Headphone Jack", "AOUTB+", + "Headphone Jack", "AOUTB-"; + mux-int-port = <1>; + mux-ext-port = <3>; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 14dfdee05fd5..35aabf9dc503 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -292,8 +292,8 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_SSI help ALSA SoC Audio support with ASRC feature for Freescale SoCs that have - ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 - and SGTL5000. + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888, + CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card. endif # SND_IMX_SOC diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 562b3bd22d9a..56b6b07d1922 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -28,6 +28,8 @@ #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" +#define CS427x_SYSCLK_MCLK 0 + #define RX 0 #define TX 1 @@ -528,6 +530,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { + codec_dai_name = "cs4271-hifi"; + priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; @@ -681,6 +687,7 @@ fail: static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-cs427x", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, -- 2.7.0

4 3

[alsa-devel] [PATCH v2 1/3] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format
by Lars-Peter Clausen 29 Jan '16

29 Jan '16

The DAI link format should be specified for the whole link rather than just one component on the link. So move the format specification for the HDMI audio link from the CPU component to the link itself. Since the sh-mobile-hdmi DAI driver doesn't implement the set_fmt() callback in this case there is no functional difference between only specifying the the format for the CPU side or for the whole link, but the later it will allow us to remove support for just specifying the format for one component. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1. --- arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 6d949f1..75de26c 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1040,9 +1040,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", -- 1.8.0

5 15

[alsa-devel] [PATCH] usb: Add native DSD support for PS Audio NuWave DAC
by Jurgen Kramer 29 Jan '16

29 Jan '16

This patch adds native DSD support for the PS Audio NuWave DAC. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 75e9ecb..f437527 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1296,6 +1296,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */ + case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.5.0

2 1

[alsa-devel] [PATCH] usb: Fix OPPO HA-1 vendor ID
by Jurgen Kramer 29 Jan '16

29 Jan '16

In my patch adding native DSD support for the Oppo HA-1, the wrong vendor ID got through. This patch fixes the vendor ID and aligns the comment. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 16364d9..75e9ecb 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1287,7 +1287,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ - case USB_ID(0x22d8, 0x0416): /* OPPO HA-1*/ + case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.5.0

2 1