July 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 00/18] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 14 Jul '17

14 Jul '17

The Cirrus Logic CS47L35, CS47L85, CS47L90/91 codecs are complex audio SoC devices. In addition to the core audio capability they have onboard GPIO, regulators, DSPs and interrupt controller and a large register map space accessed over SPI or I2C. This family of codecs is based around common IP blocks and they are managed by a set of common drivers referred to as "Madera". Summary of main changes since V1: - dt bindings moved to separate patches - pdata struct descriptions fixed to be kerneldoc format - now re-uses the arizona micsupp and ldo1 regulators - various fixes and improvements to the pinctrl driver - gpio driver can now interoperate with the pinctrl driver - some changes to the way Kconfig options are selected Richard Fitzgerald (18): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs dt-bindings: mfd: Add bindings for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: arizona-micsupp: Add support for Cirrus Logic Madera codecs regulator: arizona-ldo1: Add support for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs dt-bindings: pinctrl: Add bindings for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs dt-bindings: gpio: Add bindings for GPIO on Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs dt-bindings: sound: Add bindings for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 .../devicetree/bindings/gpio/gpio-madera.txt | 27 + Documentation/devicetree/bindings/mfd/madera.txt | 96 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 102 + .../bindings/regulator/arizona-regulator.txt | 3 +- Documentation/devicetree/bindings/sound/madera.txt | 66 + MAINTAINERS | 24 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 196 + drivers/irqchip/Kconfig | 3 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 349 + drivers/mfd/Kconfig | 50 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1688 ++++ drivers/mfd/cs47l85-tables.c | 3169 +++++++ drivers/mfd/cs47l90-tables.c | 2830 +++++++ drivers/mfd/madera-core.c | 635 ++ drivers/mfd/madera-i2c.c | 131 + drivers/mfd/madera-spi.c | 132 + drivers/mfd/madera.h | 52 + drivers/pinctrl/Kconfig | 1 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/cirrus/Kconfig | 15 + drivers/pinctrl/cirrus/Makefile | 11 + drivers/pinctrl/cirrus/pinctrl-cs47l35.c | 46 + drivers/pinctrl/cirrus/pinctrl-cs47l85.c | 60 + drivers/pinctrl/cirrus/pinctrl-cs47l90.c | 58 + drivers/pinctrl/cirrus/pinctrl-madera.c | 1106 +++ drivers/pinctrl/cirrus/pinctrl-madera.h | 40 + drivers/regulator/Kconfig | 15 +- drivers/regulator/arizona-ldo1.c | 82 +- drivers/regulator/arizona-micsupp.c | 71 +- include/dt-bindings/sound/madera.h | 18 + include/linux/irqchip/irq-madera-pdata.h | 19 + include/linux/irqchip/irq-madera.h | 96 + include/linux/mfd/madera/core.h | 180 + include/linux/mfd/madera/pdata.h | 61 + include/linux/mfd/madera/registers.h | 8832 ++++++++++++++++++++ include/sound/madera-pdata.h | 58 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1747 ++++ sound/soc/codecs/cs47l85.c | 2706 ++++++ sound/soc/codecs/cs47l90.c | 2645 ++++++ sound/soc/codecs/madera.c | 4430 ++++++++++ sound/soc/codecs/madera.h | 470 ++ 47 files changed, 32363 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/gpio-madera.txt create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/cirrus/Kconfig create mode 100644 drivers/pinctrl/cirrus/Makefile create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l35.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l85.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l90.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.h create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 1.9.1

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Re: [alsa-devel] [PATCH] ASoC: fsl_asrc: constify snd_soc_dai_ops structure
by Joe Perches 14 Jul '17

14 Jul '17

On Fri, 2017-07-14 at 12:02 +0100, Mark Brown wrote: > On Thu, Jul 13, 2017 at 11:18:11AM -0700, Joe Perches wrote: > > > I don't it's better. > > It's not that confusing if the 0/n patch cover letter is cc'd > > to all the appropriate mailing lists and all the [1..n]/n > > patches are sent with in-reply-to of the cover letter and > > send to the maintainers and appropriate mailing lists. > > With large serieses like Gustavo is sending the CC list can easily hit > the points where mailing lists start blocking it, and the individual > pathces really do need to go to the relevant people so they have sight > of them. I agree and that's what I wrote.

1 0

[alsa-devel] [PATCH 0/2] Patches for salsa-lib
by Felix Hädicke 14 Jul '17

14 Jul '17

Hi, Here are two little patches for salsa-lib. I would be happy if these could be included in the next release. And I saw that in the released salsa-lib 0.1.6 tarball, the generated headers asoundlib.h, version.h and recipe.h are included. I guess this is by mistake? This caused unexpected behaviour for me, because when performing an out-of-tree build, these are used instead of the generated headers in the build directory. Regards, Felix -- 2.13.2

2 9

Re: [alsa-devel] [PATCH] ASoC: rt5514: Support the DSP recording continuously after the hotwording triggered
by Mark Brown 13 Jul '17

13 Jul '17

On Thu, Jul 13, 2017 at 02:01:51AM +0000, Oder Chiou wrote: > > > +static irqreturn_t rt5514_spi_irq(int irq, void *data) { > > > + struct rt5514_dsp *rt5514_dsp = data; > > > + > > > + schedule_work(&rt5514_dsp->start_work); > > > + > > > + return IRQ_HANDLED; > > > +} > > Why not just use a threaded interrupt handler for the start_work? > > That'd get you into process context. > Because of the SPI related function the work thread should be manipulated > by the interrupt handler function. If the SPI related function is > implemented to the interrupt handler directly, it will cause the runtime > error while the interrupt handler is triggered. Why will this cause an error?

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Add hdmi id for a Geminilake variant
by Subhransu S. Prusty 12 Jul '17

12 Jul '17

Few GLK platform variants report a different vendor id. Add it. Also add the missing check for GLK in is_haswell_plus(). Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v1 -> v2: Fix a typo in commit message update the commit message to add glk check in is_haswell_plus() sound/pci/hda/patch_hdmi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 76c85f08bea6..d549f35f39d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -53,9 +53,11 @@ #define is_skylake(codec) ((codec)->core.vendor_id == 0x80862809) #define is_broxton(codec) ((codec)->core.vendor_id == 0x8086280a) #define is_kabylake(codec) ((codec)->core.vendor_id == 0x8086280b) +#define is_geminilake(codec) (((codec)->core.vendor_id == 0x8086280d) || \ + ((codec)->core.vendor_id == 0x80862800)) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec)) + || is_kabylake(codec)) || is_geminilake(codec) #define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) @@ -3790,6 +3792,7 @@ static int patch_via_hdmi(struct hda_codec *codec) HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), +HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: hda - Add hdmi id for a Geminilake variant
by Subhransu S. Prusty 12 Jul '17

12 Jul '17

Few GLK platform variants report a different vendor nid. Add it. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 76c85f08bea6..d549f35f39d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -53,9 +53,11 @@ #define is_skylake(codec) ((codec)->core.vendor_id == 0x80862809) #define is_broxton(codec) ((codec)->core.vendor_id == 0x8086280a) #define is_kabylake(codec) ((codec)->core.vendor_id == 0x8086280b) +#define is_geminilake(codec) (((codec)->core.vendor_id == 0x8086280d) || \ + ((codec)->core.vendor_id == 0x80862800)) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec)) + || is_kabylake(codec)) || is_geminilake(codec) #define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) @@ -3790,6 +3792,7 @@ static int patch_via_hdmi(struct hda_codec *codec) HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), +HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rt5514: Support the DSP recording continuously after the hotwording triggered
by Oder Chiou 12 Jul '17

12 Jul '17

The patch uses the IRQ to copy the PCM data to userspace continuously after the hotwording triggered from DSP. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 147 +++++++++++++++++++++++------------------- sound/soc/codecs/rt5514-spi.h | 7 +- 2 files changed, 84 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 7ed62e8..5884850 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -39,13 +39,12 @@ static struct spi_device *rt5514_spi; struct rt5514_dsp { struct device *dev; + struct work_struct start_work; 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; + size_t buf_size, get_size, dma_offset; }; static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = { @@ -80,6 +79,8 @@ static void rt5514_spi_copy_work(struct work_struct *work) container_of(work, struct rt5514_dsp, copy_work.work); struct snd_pcm_runtime *runtime; size_t period_bytes, truncated_bytes = 0; + unsigned int cur_wp, remain_data; + u8 buf[8]; mutex_lock(&rt5514_dsp->dma_lock); if (!rt5514_dsp->substream) { @@ -90,8 +91,22 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime = rt5514_dsp->substream->runtime; 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->get_size >= rt5514_dsp->buf_size) { + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, + sizeof(buf)); + cur_wp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + remain_data = (cur_wp >= rt5514_dsp->buf_rp) ? + (cur_wp - rt5514_dsp->buf_rp) : + ((rt5514_dsp->buf_limit - rt5514_dsp->buf_rp) + + (cur_wp - rt5514_dsp->buf_base)); + + if (remain_data < period_bytes) { + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + goto done; + } + } if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) { rt5514_spi_burst_read(rt5514_dsp->buf_rp, @@ -112,24 +127,57 @@ static void rt5514_spi_copy_work(struct work_struct *work) 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->buf_rp = rt5514_dsp->buf_base + period_bytes - + truncated_bytes; } + rt5514_dsp->get_size += period_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); + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + done: mutex_unlock(&rt5514_dsp->dma_lock); } +static void rt5514_spi_start_work(struct work_struct *work) +{ + struct rt5514_dsp *rt5514_dsp = + container_of(work, struct rt5514_dsp, start_work); + u8 buf[8]; + + rt5514_dsp->get_size = 0; + rt5514_dsp->dma_offset = 0; + + /** + * The address area x1800XXXX is the register address, and it cannot + * support spi burst read perfectly. So we use the spi burst read + * individually to make sure the data correctly. + */ + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; + + schedule_delayed_work(&rt5514_dsp->copy_work, 0); +} + /* PCM for streaming audio from the DSP buffer */ static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream) { @@ -170,59 +218,6 @@ static int rt5514_spi_hw_free(struct snd_pcm_substream *substream) 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); - u8 buf[8]; - - rt5514_dsp->dma_offset = 0; - rt5514_dsp->dsp_offset = 0; - - /** - * The address area x1800XXXX is the register address, and it cannot - * support spi burst read perfectly. So we use the spi burst read - * individually to make sure the data correctly. - */ - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_RP, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - rt5514_spi_burst_read(RT5514_BUFFER_VOICE_SIZE, (u8 *)&buf, - sizeof(buf)); - rt5514_dsp->buf_size = buf[0] | buf[1] << 8 | buf[2] << 16 | - buf[3] << 24; - - return 0; -} - -static int rt5514_spi_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct rt5514_dsp *rt5514_dsp = - snd_soc_platform_get_drvdata(rtd->platform); - - if (cmd == SNDRV_PCM_TRIGGER_START) { - 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) { @@ -238,25 +233,45 @@ static const 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, - .trigger = rt5514_spi_trigger, - .prepare = rt5514_spi_prepare, .pointer = rt5514_spi_pcm_pointer, .mmap = snd_pcm_lib_mmap_vmalloc, .page = snd_pcm_lib_get_vmalloc_page, }; +static irqreturn_t rt5514_spi_irq(int irq, void *data) +{ + struct rt5514_dsp *rt5514_dsp = data; + + schedule_work(&rt5514_dsp->start_work); + + return IRQ_HANDLED; +} + static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) { struct rt5514_dsp *rt5514_dsp; + int ret; rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), GFP_KERNEL); rt5514_dsp->dev = &rt5514_spi->dev; mutex_init(&rt5514_dsp->dma_lock); + INIT_WORK(&rt5514_dsp->start_work, rt5514_spi_start_work); INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work); snd_soc_platform_set_drvdata(platform, rt5514_dsp); + if (rt5514_spi->irq) { + ret = devm_request_threaded_irq(&rt5514_spi->dev, + rt5514_spi->irq, NULL, rt5514_spi_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5514-spi", + rt5514_dsp); + if (ret) + dev_err(&rt5514_spi->dev, + "%s Failed to reguest IRQ: %d\n", __func__, + ret); + } + return 0; } diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h index f69b1cd..a6434ee 100644 --- a/sound/soc/codecs/rt5514-spi.h +++ b/sound/soc/codecs/rt5514-spi.h @@ -17,10 +17,9 @@ */ #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 +#define RT5514_BUFFER_VOICE_BASE 0x18000200 +#define RT5514_BUFFER_VOICE_LIMIT 0x18000204 +#define RT5514_BUFFER_VOICE_WP 0x1800020c /* SPI Command */ enum { -- 2.7.4

2 1

[alsa-devel] Slow/piecewise send of sysex?
by Ville Oikarinen 12 Jul '17

12 Jul '17

Hello. I just created this question as a ticket https://github.com/opensrc/alsa/issues/24 but was instructed to ask here. So: I'm using amidi --send-hex to send data to a Commodore 64 via a CLab MIDI interface. The problem is that even with 100 bytes the probability of a buffer overrun is about 30%. I didn't find a way to throttle the send speed. I also tried sending the data in smaller chunks, but I cannot get amidi to send anything after the first chunk. I suspect it refuses to send anything if the data doesn't start with F0 i.e. sysex dump start. If I am right about that assumption, as a workaround I could encapsulate each chunk as a full sysex dump, but that requires extra logic on the receiving side and it also wastes bandwidth, so I don't really wish to do that. Primarily I would prefer a way to throttle the speed, but if I cannot have that, I would like to be able to send the data a few bytes at a time in a simple shell loop. - Ville Oikarinen

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[alsa-devel] [PATCH v2] ASoC: samsung: i2s: Support more resolution rates
by Jaechul Lee 12 Jul '17

12 Jul '17

This driver can support more frequencies over 96KHz. There are no reasons to limit the frequency range below 96KHz. If codecs/amps or something else can't support higher resolution rates, the constraints would be set rates properly because each drivers have its own limits. I added the 'pcm_rates' field to the dai_data to be set rates by the compatibilities. As a result, rates will be set each devices respectively. For example of exynos5433, rates will be set from 8KHz to 192KHz. Signed-off-by: Jaechul Lee <jcsing.lee(a)samsung.com> --- v2: - changed the name of variable to pcm_rates. - removed duplicated code. - modified commit message. --- sound/soc/samsung/i2s.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d4ccc5..c9f87f7bae99 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -50,6 +50,7 @@ struct samsung_i2s_variant_regs { struct samsung_i2s_dai_data { u32 quirks; + unsigned int pcm_rates; const struct samsung_i2s_variant_regs *i2s_variant_regs; }; @@ -1076,13 +1077,13 @@ static const struct snd_soc_component_driver samsung_i2s_component = { .name = "samsung-i2s", }; -#define SAMSUNG_I2S_RATES SNDRV_PCM_RATE_8000_96000 - #define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) +static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, + const struct samsung_i2s_dai_data *i2s_dai_data, + bool sec) { struct i2s_dai *i2s; @@ -1101,13 +1102,13 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.resume = i2s_resume; i2s->i2s_dai_drv.playback.channels_min = 1; i2s->i2s_dai_drv.playback.channels_max = 2; - i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES; + i2s->i2s_dai_drv.playback.rates = i2s_dai_data->pcm_rates; i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; if (!sec) { i2s->i2s_dai_drv.capture.channels_min = 1; i2s->i2s_dai_drv.capture.channels_max = 2; - i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; + i2s->i2s_dai_drv.capture.rates = i2s_dai_data->pcm_rates; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; } return i2s; @@ -1242,7 +1243,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) i2s_dai_data = (struct samsung_i2s_dai_data *) platform_get_device_id(pdev)->driver_data; - pri_dai = i2s_alloc_dai(pdev, false); + pri_dai = i2s_alloc_dai(pdev, i2s_dai_data, false); if (!pri_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_pri\n"); return -ENOMEM; @@ -1316,7 +1317,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) goto err_disable_clk; if (quirks & QUIRK_SEC_DAI) { - sec_dai = i2s_alloc_dai(pdev, true); + sec_dai = i2s_alloc_dai(pdev, i2s_dai_data, true); if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); ret = -ENOMEM; @@ -1452,29 +1453,34 @@ static const struct samsung_i2s_variant_regs i2sv5_i2s1_regs = { static const struct samsung_i2s_dai_data i2sv3_dai_type = { .quirks = QUIRK_NO_MUXPSR, + .pcm_rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv3_regs, }; static const struct samsung_i2s_dai_data i2sv5_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_IDMA, + .pcm_rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv3_regs, }; static const struct samsung_i2s_dai_data i2sv6_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_TDM | QUIRK_SUPPORTS_IDMA, + .pcm_rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv6_regs, }; static const struct samsung_i2s_dai_data i2sv7_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_TDM, + .pcm_rates = SNDRV_PCM_RATE_8000_192000, .i2s_variant_regs = &i2sv7_regs, }; static const struct samsung_i2s_dai_data i2sv5_dai_type_i2s1 = { .quirks = QUIRK_PRI_6CHAN | QUIRK_NEED_RSTCLR, + .pcm_rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv5_i2s1_regs, }; -- 2.11.0

5 4

[alsa-devel] [PATCH - alsa-lib 1/1] snd_user_file: avoid use wordexp
by Natanael Copa 12 Jul '17

12 Jul '17

As suggested in POSIX[1], wordexp might execute the shell. If the libc implementation does so, it will break the firefox sandbox which does not allow exec. This happened on Alpine Linux with musl libc[2]. Since we cannot guarantee that the system wordexp implementation does not execute shell, we cannot really use it, and need to implement the ~/ expansion ourselves. Generally, wordexp is a large attack vector and it is better to avoid it since only tilde expansion is needed. [1]: http://pubs.opengroup.org/onlinepubs/9699919799/functions/wordexp.html#tag_… [2]: http://bugs.alpinelinux.org/issues/7454#note-2 Signed-off-by: Natanael Copa <ncopa(a)alpinelinux.org> diff --git a/configure.ac b/configure.ac index 26e5d125..076c6bab 100644 --- a/configure.ac +++ b/configure.ac @@ -304,7 +304,7 @@ fi AC_SUBST(ALSA_DEPLIBS) dnl Check for headers -AC_CHECK_HEADERS([wordexp.h endian.h sys/endian.h sys/shm.h]) +AC_CHECK_HEADERS([endian.h sys/endian.h sys/shm.h]) dnl Check for resmgr support... AC_MSG_CHECKING(for resmgr support) diff --git a/src/userfile.c b/src/userfile.c index 72779da4..0e3f5fae 100644 --- a/src/userfile.c +++ b/src/userfile.c @@ -21,6 +21,11 @@ #include <config.h> #include <string.h> #include <errno.h> +#include <sys/types.h> +#include <unistd.h> +#include <pwd.h> +#include <stdio.h> +#include <stdlib.h> /** * \brief Get the full file name @@ -28,46 +33,58 @@ * \param result The pointer to store the resultant file name * \return 0 if successful, or a negative error code * - * Parses the given file name with POSIX-Shell-like expansion and - * stores the first matchine one. The returned string is strdup'ed. + * Parses the given file name with POSIX-Shell-like expansion for ~/. + * The returned string is strdup'ed. */ -#ifdef HAVE_WORDEXP_H -#include <wordexp.h> #include <assert.h> int snd_user_file(const char *file, char **result) { - wordexp_t we; int err; - + size_t len; + char *buf = NULL; + assert(file && result); - err = wordexp(file, &we, WRDE_NOCMD); - switch (err) { - case WRDE_NOSPACE: - wordfree(&we); - return -ENOMEM; - case 0: - if (we.we_wordc == 1) - break; - wordfree(&we); - /* fall thru */ - default: - return -EINVAL; + *result = NULL; + + /* expand ~/ if needed */ + if (file[0] == '~' && file[1] == '/') { + const char *home = getenv("HOME"); + if (home == NULL) { + struct passwd pwent, *p = NULL; + uid_t id = getuid(); + size_t bufsize = 1024; + + buf = malloc(bufsize); + if (buf == NULL) + goto out; + + while ((err = getpwuid_r(id, &pwent, buf, bufsize, &p)) == ERANGE) { + char *newbuf; + bufsize += 1024; + if (bufsize < 1024) + break; + newbuf = realloc(buf, bufsize); + if (newbuf == NULL) + goto out; + buf = newbuf; + } + home = err ? "" : pwent.pw_dir; + } + len = strlen(home) + strlen(&file[2]) + 2; + *result = malloc(len); + if (*result) + snprintf(*result, len, "%s/%s", home, &file[2]); + } else { + *result = strdup(file); } - *result = strdup(we.we_wordv[0]); - wordfree(&we); + +out: + if (buf) + free(buf); + if (*result == NULL) return -ENOMEM; return 0; } -#else /* !HAVE_WORDEXP_H */ -/* just copy the string - would be nicer to expand by ourselves, though... */ -int snd_user_file(const char *file, char **result) -{ - *result = strdup(file); - if (! *result) - return -ENOMEM; - return 0; -} -#endif /* HAVE_WORDEXP_H */ -- 2.13.2

1 0

[alsa-devel] Clock Drift between hardware sound card and snd_aloop
by Rob Nertney 12 Jul '17

12 Jul '17

Hi all, I have successfully built a sound capture card that runs at a given rate off its own clock. I receive the stream and do some post-processing on it, and I want to present this modified data to the user as a capture pcm. I initially thought that snd_aloop would be the best idea, as I can pipe my post-processed data directly into it. However, the driver is apparently calculating exact timing of what "should" be the rate based off of CPU ticks/jiffies and the requested period/sample rate/etc in the requested output stream. In my actual hardware I see cycle slips. Underruns and overruns based off of minor adjustments to the card's rate. Is snd_aloop the appropriate driver to be using? Or should I create a new card, similar to snd_aloop, which is more specifically interrupted (and associated pointer locations, etc. modified) from the hardware card as well? It seems like I should be able to synchronize snd_aloop's timing with my hardware clock, but I haven't found anywhere in the API to achieve this. How else can I avoid this clock drift? Thanks, Rob

2 1

[alsa-devel] [PATCH] WIP:asoc:qdsp6: fix semicolon.cocci warnings
by Julia Lawall 11 Jul '17

11 Jul '17

Remove unneeded semicolons. Generated by: scripts/coccinelle/misc/semicolon.cocci Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- tree: https://github.com/rrnayak/linux genpd-performance-state head: 3f1a0555478c46892595728fcb4a2c7ae7e1f7da commit: b4f8ada6fb36369cf0e45421dde636d79e845554 [48/114] WIP:asoc:qdsp6: add adm driver :::::: branch date: 2 hours ago :::::: commit date: 8 weeks ago q6adm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) --- a/sound/soc/qcom/qdsp6v2/q6adm.c +++ b/sound/soc/qcom/qdsp6v2/q6adm.c @@ -806,7 +806,7 @@ int adm_arrange_mch_map(struct adm_cmd_d default: goto non_mch_path; break; - }; + } if ((open->dev_num_channel > 2) && multi_ch_maps[idx].set_channel_map) memcpy(open->dev_channel_mapping, @@ -1019,10 +1019,10 @@ int adm_matrix_map(int path, struct rout route->hdr.pkt_size = cmd_size; route->hdr.src_svc = 0; route->hdr.src_domain = APR_DOMAIN_APPS; - route->hdr.src_port = 0; /* Ignored */; + route->hdr.src_port = 0; /* Ignored */ route->hdr.dest_svc = APR_SVC_ADM; route->hdr.dest_domain = APR_DOMAIN_ADSP; - route->hdr.dest_port = 0; /* Ignored */; + route->hdr.dest_port = 0; /* Ignored */ route->hdr.token = 0; if (path == ADM_PATH_COMPRESSED_RX) { pr_debug("%s: ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x%x\n",

1 0

[alsa-devel] [PATCH] ASoC: pxa: SND_PXA2XX_SOC should depend on HAS_DMA
by Geert Uytterhoeven 11 Jul '17

11 Jul '17

If NO_DMA=y: ERROR: "bad_dma_ops" [sound/soc/pxa/snd-soc-pxa2xx.ko] undefined! ERROR: "bad_dma_ops" [sound/arm/snd-pxa2xx-lib.ko] undefined! ERROR: "dma_common_mmap" [sound/arm/snd-pxa2xx-lib.ko] undefined! Add a dependency on HAS_DMA to fix this. Fixes: 73d7ee2e831f106c ("ASoC: pxa: add COMPILE_TEST on SND_PXA2XX_SOC") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e46edc0549..484ab3c2ad672fc8 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,7 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA || COMPILE_TEST + depends on HAS_DMA select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to -- 2.7.4

3 2

[alsa-devel] Applied "ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static" to the asoc tree
by Mark Brown 11 Jul '17

11 Jul '17

The patch ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static 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 fc2b74460d96ec28110a581d6e58d15c27304d16 Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Tue, 11 Jul 2017 15:08:55 +0100 Subject: [PATCH] ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static Don't populate array hdmi_codec_eld_spk_alloc_bits on the stack but make it static. Makes the object code smaller by over 260 bytes: Before: text data bss dec hex filename 10882 3384 64 14330 37fa sound/soc/codecs/hdmi-codec.o After: text data bss dec hex filename 10557 3440 64 14061 36ed sound/soc/codecs/hdmi-codec.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 7686a80861f1..509ab513b4b2 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -326,7 +326,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc) { int i; - const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { + static const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { [0] = FL | FR, [1] = LFE, [2] = FC, [3] = RL | RR, [4] = RC, [5] = FLC | FRC, [6] = RLC | RRC, }; -- 2.13.2

1 0

[alsa-devel] [PATCH] ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static
by Colin King 11 Jul '17

11 Jul '17

From: Colin Ian King <colin.king(a)canonical.com> Don't populate array hdmi_codec_eld_spk_alloc_bits on the stack but make it static. Makes the object code smaller by over 260 bytes: Before: text data bss dec hex filename 10882 3384 64 14330 37fa sound/soc/codecs/hdmi-codec.o After: text data bss dec hex filename 10557 3440 64 14061 36ed sound/soc/codecs/hdmi-codec.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 7686a80861f1..509ab513b4b2 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -326,7 +326,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc) { int i; - const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { + static const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { [0] = FL | FR, [1] = LFE, [2] = FC, [3] = RL | RR, [4] = RC, [5] = FLC | FRC, [6] = RLC | RRC, }; -- 2.11.0

1 0

[alsa-devel] New codec device ID for ALC1220
by Kailang 11 Jul '17

11 Jul '17

Hi Takashi, ALC1220 has another device ID. Attach patch was add 0x1168 device ID support. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH][-next] ASoC: rt5663: add in missing loop counter to avoid infinite loop
by Colin King 10 Jul '17

10 Jul '17

From: Colin Ian King <colin.king(a)canonical.com> The loop counter 'count' is missing an increment, so we could end up with an infinite loop. Add in a pre-increment to count to fix this. Detected by CoverityScan, CID#1450120 ("Logically Dead Code") Fixes: 1a425dbf1a10 ("ASoC: rt5663: Update the calibration funciton") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5663.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 82edbcf39caa..5b3c50c641d8 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3044,7 +3044,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) else break; - if (count > 200) + if (++count > 200) break; } -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Move the auto disable DSP function to set_bias_level()
by Oder Chiou 10 Jul '17

10 Jul '17

Move the auto disable DSP function from the widget event "DAPM Pre" to the set_bias_level() function to avoid the unexpected behavior. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 49 +++++++++++++++++------------------------------ 1 file changed, 18 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1b6796c..a71e630 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -471,35 +471,6 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } -static int rt5514_pre_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 rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - /** - * If the DSP is enabled in start of recording, the DSP - * should be disabled, and sync back to normal recording - * settings to make sure recording properly. - */ - if (rt5514->dsp_enabled) { - rt5514->dsp_enabled = 0; - regmap_multi_reg_write(rt5514->i2c_regmap, - rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch)); - regcache_mark_dirty(rt5514->regmap); - regcache_sync(rt5514->regmap); - } - break; - - default: - return 0; - } - - return 0; -} - static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1L"), @@ -607,8 +578,6 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Audio Interface */ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), - - SND_SOC_DAPM_PRE("DAPM Pre", rt5514_pre_event), }; static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { @@ -975,6 +944,24 @@ static int rt5514_set_bias_level(struct snd_soc_codec *codec, } break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* + * If the DSP is enabled in start of recording, the DSP + * should be disabled, and sync back to normal recording + * settings to make sure recording properly. + */ + if (rt5514->dsp_enabled) { + rt5514->dsp_enabled = 0; + regmap_multi_reg_write(rt5514->i2c_regmap, + rt5514_i2c_patch, + ARRAY_SIZE(rt5514_i2c_patch)); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->regmap); + } + } + break; + default: break; } -- 2.7.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: rt5663: Add the manual offset field to compensate the DC offset
by Oder Chiou 10 Jul '17

10 Jul '17

The patch adds the manual offset field in the devicetree to compensate the DC offset that will be different between the PCB layout. It only can be measured by the real production. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5663.txt | 5 ++ include/sound/rt5663.h | 20 ++++++++ sound/soc/codecs/rt5663.c | 55 ++++++++++++++++++++-- sound/soc/codecs/rt5663.h | 2 + 4 files changed, 77 insertions(+), 5 deletions(-) create mode 100644 include/sound/rt5663.h diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 70eaeae..f7c7cc5 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -12,6 +12,11 @@ Required properties: Optional properties: +- "realtek,dc_offset_l_manual" +- "realtek,dc_offset_r_manual" + Based on the different PCB layout, add the manual offset value to + compensate the DC offset for each L and R channel. + Pins on the device (for linking into audio routes) for RT5663: * IN1P diff --git a/include/sound/rt5663.h b/include/sound/rt5663.h new file mode 100644 index 0000000..be6f1b2 --- /dev/null +++ b/include/sound/rt5663.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt5663.h -- Platform data for RT5663 + * + * Copyright 2017 Realtek Semiconductor Corp. + * + * 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 __LINUX_SND_RT5663_H +#define __LINUX_SND_RT5663_H + +struct rt5663_platform_data { + unsigned int dc_offset_l_manual; + unsigned int dc_offset_r_manual; +}; + +#endif + diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 82edbcf..52ecca4 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -40,6 +40,7 @@ enum { struct rt5663_priv { struct snd_soc_codec *codec; + struct rt5663_platform_data pdata; struct regmap *regmap; struct delayed_work jack_detect_work; struct snd_soc_jack *hs_jack; @@ -57,6 +58,10 @@ struct rt5663_priv { int jack_type; }; +static const struct reg_sequence rt5663_patch_list[] = { + { 0x002a, 0x8080 }, +}; + static const struct reg_default rt5663_v2_reg[] = { { 0x0000, 0x0000 }, { 0x0001, 0xc8c8 }, @@ -476,7 +481,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0023, 0x0039 }, { 0x0026, 0xc0c0 }, { 0x0029, 0x8080 }, - { 0x002a, 0xa0a0 }, + { 0x002a, 0x8080 }, { 0x002c, 0x000c }, { 0x002d, 0x0000 }, { 0x0040, 0x0808 }, @@ -1958,15 +1963,11 @@ static const struct snd_kcontrol_new rt5663_adda_r_mix[] = { static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = { SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, RT5663_M_DAC_L1_STO_L_SHIFT, 1, 1), - SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, - RT5663_M_DAC_R1_STO_L_SHIFT, 1, 1), }; static const struct snd_kcontrol_new rt5663_sto1_dac_r_mix[] = { SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, RT5663_M_DAC_L1_STO_R_SHIFT, 1, 1), - SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, - RT5663_M_DAC_R1_STO_R_SHIFT, 1, 1), }; /* Out Switch */ @@ -3126,9 +3127,20 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) usleep_range(10000, 10005); } +static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) +{ + device_property_read_u32(dev, "realtek,dc_offset_l_manual", + &rt5663->pdata.dc_offset_l_manual); + device_property_read_u32(dev, "realtek,dc_offset_r_manual", + &rt5663->pdata.dc_offset_r_manual); + + return 0; +} + static int rt5663_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt5663_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5663_priv *rt5663; int ret; unsigned int val; @@ -3142,6 +3154,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5663); + if (pdata) + rt5663->pdata = *pdata; + else + rt5663_parse_dp(rt5663, &i2c->dev); + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); if (IS_ERR(regmap)) { ret = PTR_ERR(regmap); @@ -3190,6 +3207,34 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_write(rt5663->regmap, RT5663_RESET, 0); dev_dbg(&i2c->dev, "calibrate done\n"); + switch (rt5663->codec_ver) { + case CODEC_VER_1: + break; + case CODEC_VER_0: + ret = regmap_register_patch(rt5663->regmap, rt5663_patch_list, + ARRAY_SIZE(rt5663_patch_list)); + if (ret != 0) + dev_warn(&i2c->dev, + "Failed to apply regmap patch: %d\n", ret); + break; + default: + dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); + } + + if (rt5663->pdata.dc_offset_l_manual) { + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, + rt5663->pdata.dc_offset_l_manual >> 16); + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3, + rt5663->pdata.dc_offset_l_manual & 0xffff); + } + + if (rt5663->pdata.dc_offset_r_manual) { + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5, + rt5663->pdata.dc_offset_r_manual >> 16); + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6, + rt5663->pdata.dc_offset_r_manual & 0xffff); + } + /* GPIO1 as IRQ */ regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, RT5663_GP1_PIN_MASK, RT5663_GP1_PIN_IRQ); diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h index 4621812..c5a9b69 100644 --- a/sound/soc/codecs/rt5663.h +++ b/sound/soc/codecs/rt5663.h @@ -12,6 +12,8 @@ #ifndef __RT5663_H__ #define __RT5663_H__ +#include <sound/rt5663.h> + /* Info */ #define RT5663_RESET 0x0000 #define RT5663_VENDOR_ID 0x00fd -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix default dma_buffer_size
by Subhransu S. Prusty 10 Jul '17

10 Jul '17

If the dma_buffer_size is not defined in topology, fix it to 2ms default value to make backward compatible. Fixes: f6e6ab1d16ec ("ASoC: Intel: Skylake: Fix dma buffer size calculation") Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index eca85827dbd2..fb2f1f603f3c 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -540,6 +540,14 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, cpr_mconfig->gtw_cfg.dma_buffer_size = mconfig->dma_buffer_size * dma_io_buf; + /* fallback to 2ms default value */ + if (!cpr_mconfig->gtw_cfg.dma_buffer_size) { + if (mconfig->hw_conn_type == SKL_CONN_SOURCE) + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->obs; + else + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->ibs; + } + cpr_mconfig->cpr_feature_mask = 0; cpr_mconfig->gtw_cfg.config_length = 0; -- 1.9.1

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[alsa-devel] ALSA BLE MIDI
by happy musicmaker 10 Jul '17

10 Jul '17

Is there development for Bluetooth (BLE) MIDI under Linux ? More and more devices now support BLE MIDI. I have an BLE USB adapter that works fine with many MIDI BLE devices but the devices don't shop up in ALSA (Ubuntu 17.04). It seems Linux only supports audio devices.

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[alsa-devel] MEDION sound devices informations
by fulgor fulgor 10 Jul '17

10 Jul '17

I have run the script alsa-info.SH www.alsa-project.org/db/?f=6fb68d0201f421b41fe65677fde2ec5b1c9d4429 thanks for all your work

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[alsa-devel] [PATCH] ALSA: pcm: Simplify check for dma_mmap_coherent() availability
by Takashi Iwai 10 Jul '17

10 Jul '17

We check the availability of dma_mmap_coherent() in hw_support_mmap() but with an ugly ifdef of lots of arch-checks. Now we have a nice CONFIG_ARCH_NO_COHERENT_DMA_MMAP kconfig, and this can be used together with CONFIG_HAS_DMA check for a cleaner and more comprehensive check. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b26c7e61d600..2a730f38e375 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -238,10 +238,8 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) { if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - /* check architectures that return -EINVAL from dma_mmap_coherent() */ - /* FIXME: this should be some global flag */ -#if defined(CONFIG_C6X) || defined(CONFIG_FRV) || defined(CONFIG_MN10300) ||\ - defined(CONFIG_PARISC) || defined(CONFIG_XTENSA) + /* architecture supports dma_mmap_coherent()? */ +#if defined(CONFIG_ARCH_NO_COHERENT_DMA_MMAP) || !defined(CONFIG_HAS_DMA) if (!substream->ops->mmap && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return false; -- 2.13.2

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[alsa-devel] [PATCH] ALSA: pcm: Protect call to dma_mmap_coherent() by check for HAS_DMA
by Geert Uytterhoeven 10 Jul '17

10 Jul '17

If NO_DMA=y: sound/core/pcm_native.o: In function `snd_pcm_lib_default_mmap': pcm_native.c:(.text+0x144c): undefined reference to `bad_dma_ops' pcm_native.c:(.text+0x1474): undefined reference to `dma_common_mmap' Add a check for HAS_DMA to fix this. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b3d5bed750297ec2..e540a22aa3d8988a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3502,7 +3502,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, } #endif /* CONFIG_GENERIC_ALLOCATOR */ #ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */ - if (!substream->ops->page && + if (IS_ENABLED(CONFIG_HAS_DMA) && !substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, area, -- 2.7.4

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[alsa-devel] arecord cannot interrupt streaming via CTRL-C when using voc_pcm_wirte()
by Wischer, Timo (ADITG/ESB) 10 Jul '17

10 Jul '17

Hi all, please merge following patch: >From fb58aa47b8715924667a0a4383ce9a3615378413 Mon Sep 17 00:00:00 2001 From: Srikanth Krishnakar <Srikanth_Krishnakar(a)mentor.com> Date: Mon, 19 Dec 2016 14:04:39 +0530 Subject: alsa-utils: interrupt streaming via signal in voc_pcm_write arecord (alsa-utils v1.1.2) cannot interrupt streaming via CTRL-C. Fixed the issue by properly handling 'in_aborting' flag in appropriate functions. Signed-off-by: Anant Agrawal <Anant_Agrawal(a)mentor.com> Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Signed-off-by: Srikanth Krishnakar <Srikanth_Krishnakar(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 2da7dda..cdaa1a8 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2161,7 +2161,7 @@ static ssize_t voc_pcm_write(u_char *data, size_t count) ssize_t result = count, r; size_t size; - while (count > 0) { + while (count > 0 && !in_aborting) { size = count; if (size > chunk_bytes - buffer_pos) size = chunk_bytes - buffer_pos; -- 2.7.4 Best regards Timo Wischer Advanced Driver Information Technology GmbH Software Group I (ADITG/ESB) Robert-Bosch-Str. 200 31139 Hildesheim Germany Tel. +49 5121 49 6938 Fax +49 5121 49 6999 twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschäftsführung: Wilhelm Grabow, Ken Yaguchi

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