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

[alsa-devel] [PATCH] ASoC: rt5514: expose Hotword Model control
by Hsin-Yu Chao 15 Sep '17

15 Sep '17

This change exposes mixer control 'Hotword Model' for switching hotword model in runtime. This new function requires updated firmware to load hotword model at address 0x4ff80000. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 52 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514.h | 3 +++ 2 files changed, 55 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a2722177470e..7482df8ff737 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -299,6 +299,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); struct snd_soc_codec *codec = rt5514->codec; const struct firmware *fw = NULL; + int ret = 0; if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) return 0; @@ -335,6 +336,27 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } + if (rt5514->model_buf && rt5514->model_len) { + ret = rt5514_spi_burst_write(0x4ff80000, + rt5514->model_buf, + ((rt5514->model_len / 8) + 1) * 8); + if (ret) { + dev_err(codec->dev, + "Model load failed %d\n", ret); + return ret; + } + } else { + request_firmware(&fw, RT5514_FIRMWARE3, + codec->dev); + if (fw) { + rt5514_spi_burst_write(0x4ff80000, + fw->data, + ((fw->size/8)+1)*8); + release_firmware(fw); + fw = NULL; + } + } + /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -349,6 +371,34 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } +static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, + const unsigned int __user *bytes, unsigned int size) +{ + 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; + int ret = 0; + + if (rt5514->model_buf || rt5514->model_len < size) { + if (rt5514->model_buf) + devm_kfree(codec->dev, rt5514->model_buf); + rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); + if (!rt5514->model_buf) { + ret = -ENOMEM; + goto done; + } + } + + /* Skips the TLV header. */ + bytes += 2; + + if (copy_from_user(rt5514->model_buf, bytes, size)) + ret = -EFAULT; +done: + rt5514->model_len = (ret ? 0 : size); + return ret; +} + 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), @@ -360,6 +410,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), + SND_SOC_BYTES_TLV("Hotword Model", 0x8504, + NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212a86d9..07475a9918d2 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -236,6 +236,7 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" +#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -262,6 +263,8 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; + u8 *model_buf; + unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.12.2

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[alsa-devel] [PATCH v3 00/11] Add STM32 DFSDM support
by Arnaud Pouliquen 11 Sep '17

11 Sep '17

Hello, I kept it as RFC as i reworked the global design (but also because not full validated...). Regarding Mark and Jonathan's comments, I decided to try an implementation with DMA support in IIO instead of handling it in ASOC. Solution seems more straightforward but as DMA cyclic mode and associated buffer block API is not part of IIO today, I still need some hacks... To be honest, I tried to integrate DMA cyclic management and associated buffer block Consumer API. but no enough expertise on IIO to success... :( Advantage vs V2: - No more data processing need in ASoC , done in IIO - ASoC uses IIO consumer interface ( except for buffer callback) Drawback : - Need to create a "SPI bus" trigger to allo to use Buffer consumer API => Need to find a solution to set this trigger by default in IIO audio driver - No standard buffer management - Implement DMA cyclic in DFSDM IIO driver. - Define specific API to handle buffer callback per period instead of callback per samples generated by iio_push_to_buffers_with_timestamp. V2-V3 delta: - New patches to support ASoC DMA codec in DT - ASoC: Add Bindins for DMIC codec driver. - ASoC: codec: add DT support in dmic codec. - New patches to allow in-kernel set of IIO buffer size and watermark - IIO: consumer: allow to set buffer sizes. - IIO DFSDM drivers - Split audio and ADC support in 2 drivers. - Implement DMA cyclic mode. - Add SPI bus Trigger for buffer management. - IIO sigma delta adc drivers - Suppress "simple and rename driver. - ASoC driver - Suppress DMA engine. - Suppress copy ops. - Use IIO consmumer interface to enable/Disable DFSDM. Regards, Arnaud Arnaud Pouliquen (11): iio: Add hardware consumer support IIO: Add DT bindings for sigma delta adc modulator IIO: ADC: add sigma delta modulator support IIO: add DT bindings for stm32 DFSDM filter IIO: ADC: add stm32 DFSDM support for Sigma delta ADC IIO: ADC: add stm32 DFSDM support for PDM microphone IIO: consumer: allow to set buffer sizes ASoC: Add Bindins for DMIC codec driver ASoC: codec: add DT support in dmic codec ASoC: add bindings for stm32 DFSDM filter ASoC: stm32: add DFSDM DAI support .../bindings/iio/adc/sigma-delta-modulator.txt | 13 + .../bindings/iio/adc/st,stm32-dfsdm-adc.txt | 120 ++++ Documentation/devicetree/bindings/sound/dmic.txt | 11 + .../devicetree/bindings/sound/st,stm32-adfsdm.txt | 41 ++ drivers/iio/Kconfig | 6 + drivers/iio/Makefile | 1 + drivers/iio/adc/Kconfig | 50 ++ drivers/iio/adc/Makefile | 4 + drivers/iio/adc/sd_adc_modulator.c | 98 +++ drivers/iio/adc/stm32-dfsdm-adc.c | 419 ++++++++++++ drivers/iio/adc/stm32-dfsdm-audio.c | 720 +++++++++++++++++++++ drivers/iio/adc/stm32-dfsdm-core.c | 658 +++++++++++++++++++ drivers/iio/adc/stm32-dfsdm.h | 372 +++++++++++ drivers/iio/buffer/industrialio-buffer-cb.c | 12 + drivers/iio/hw_consumer.c | 156 +++++ include/linux/iio/adc/stm32-dfsdm-audio.h | 41 ++ include/linux/iio/consumer.h | 13 + include/linux/iio/hw_consumer.h | 12 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/dmic.c | 8 + sound/soc/stm/Kconfig | 10 + sound/soc/stm/Makefile | 2 + sound/soc/stm/stm32_adfsdm.c | 362 +++++++++++ 24 files changed, 3131 insertions(+) create mode 100644 Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.txt create mode 100644 Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt create mode 100644 Documentation/devicetree/bindings/sound/dmic.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt create mode 100644 drivers/iio/adc/sd_adc_modulator.c create mode 100644 drivers/iio/adc/stm32-dfsdm-adc.c create mode 100644 drivers/iio/adc/stm32-dfsdm-audio.c create mode 100644 drivers/iio/adc/stm32-dfsdm-core.c create mode 100644 drivers/iio/adc/stm32-dfsdm.h create mode 100644 drivers/iio/hw_consumer.c create mode 100644 include/linux/iio/adc/stm32-dfsdm-audio.h create mode 100644 include/linux/iio/hw_consumer.h create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_adfsdm.c V2: Patch-set associated to this RFC proposes an implementation of the DFSDM features shared between ASoC and IIO frameworks. Patch-set is only a Skeleton of the drivers, so a base to discuss and validate a design. It contains minimum code to allow probing (with DT) and to expose the ASoC and IIO ABI. Hope that is sufficent in a first step to allow to focus on APIs. In this patch-set there are two new APIs used: - IIO in-kern API: based on hw_customer API proposed by Lars - ASOC <-> IIO API inspired by API defined for hdmi-codec for ASoC/DRM interconnect. API is dedicated to DFSDM only. Notice also that this design is based on following assumption: - Audio stream ABI interface is ASoC, no access to data through IIO ABI for PDM. - ASoC DMA should be used for audio transfer as designed for real time stream. - Need some runtime parameters exchange between ASoC and IIO due to the correlation between the sample frequency, the DFSDM decimation factor and the associated scaling. - "ASoC: dmaengine_pcm: add copy support" patch: I added a patch in ASoC that allows to implement a copy function to process data after DMA transfer. Requested, as DFSDM samples captured contain channel ID on 8-LSB bits and need also a potential rescale to present DAT on 24-bits. - "IIO: ADC: add sigma delta modulator support" patch: Simple dummy driver created to support external Sigma delta modulator. It is binded to DFSDM driver through hw_customer API. - 1.9.1

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[alsa-devel] [PATCH 23/25] ALSA/dummy: Replace tasklet with softirq hrtimer
by Anna-Maria Gleixner 08 Sep '17

08 Sep '17

From: Thomas Gleixner <tglx(a)linutronix.de> The tasklet is used to defer the execution of snd_pcm_period_elapsed() to the softirq context. Using the CLOCK_MONOTONIC_SOFT base invokes the timer callback in softirq context as well which renders the tasklet useless. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Anna-Maria Gleixner <anna-maria(a)linutronix.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org --- sound/drivers/dummy.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -376,17 +376,9 @@ struct dummy_hrtimer_pcm { ktime_t period_time; atomic_t running; struct hrtimer timer; - struct tasklet_struct tasklet; struct snd_pcm_substream *substream; }; -static void dummy_hrtimer_pcm_elapsed(unsigned long priv) -{ - struct dummy_hrtimer_pcm *dpcm = (struct dummy_hrtimer_pcm *)priv; - if (atomic_read(&dpcm->running)) - snd_pcm_period_elapsed(dpcm->substream); -} - static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer) { struct dummy_hrtimer_pcm *dpcm; @@ -394,7 +386,8 @@ static enum hrtimer_restart dummy_hrtime dpcm = container_of(timer, struct dummy_hrtimer_pcm, timer); if (!atomic_read(&dpcm->running)) return HRTIMER_NORESTART; - tasklet_schedule(&dpcm->tasklet); + + snd_pcm_period_elapsed(dpcm->substream); hrtimer_forward_now(timer, dpcm->period_time); return HRTIMER_RESTART; } @@ -421,7 +414,6 @@ static int dummy_hrtimer_stop(struct snd static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm) { hrtimer_cancel(&dpcm->timer); - tasklet_kill(&dpcm->tasklet); } static snd_pcm_uframes_t @@ -466,12 +458,10 @@ static int dummy_hrtimer_create(struct s if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC_SOFT, HRTIMER_MODE_REL); dpcm->timer.function = dummy_hrtimer_callback; dpcm->substream = substream; atomic_set(&dpcm->running, 0); - tasklet_init(&dpcm->tasklet, dummy_hrtimer_pcm_elapsed, - (unsigned long)dpcm); return 0; }

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[alsa-devel] [patch 0/4] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 08 Sep '17

08 Sep '17

Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. This code change doesn't do much except just refactor snd_soc_dapm_new_pcm function. Now the part of allocating memory for kcontrol is allocated to a new function and it is called only in the case it is needed. anish kumar (4): ASoC: dapm: fix error path in snd_soc_dapm_new_pcm ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm ASoC: dapm: Avoid creating kcontrol for params ASoC: dapm: Remove the redundant check sound/soc/soc-dapm.c | 123 ++++++++++++++++++++++++++++----------------------- 1 file changed, 67 insertions(+), 56 deletions(-) -- 2.10.1 (Apple Git-78)

1 1

[alsa-devel] [PATCH] ALSA: maestro3: Use common error handling code in two functions
by SF Markus Elfring 07 Sep '17

07 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 15:00:27 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/maestro3.c | 55 +++++++++++++++++++++++++--------------------------- 1 file changed, 26 insertions(+), 29 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index cafea6dc5c01..759c3f6ad7aa 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2622,22 +2622,18 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, err = request_firmware(&chip->assp_kernel_image, "ess/maestro3_assp_kernel.fw", &pci->dev); - if (err < 0) { - snd_m3_free(chip); - return err; - } + if (err) + goto free_chip; err = request_firmware(&chip->assp_minisrc_image, "ess/maestro3_assp_minisrc.fw", &pci->dev); - if (err < 0) { - snd_m3_free(chip); - return err; - } + if (err) + goto free_chip; + + err = pci_request_regions(pci, card->driver); + if (err) + goto free_chip; - if ((err = pci_request_regions(pci, card->driver)) < 0) { - snd_m3_free(chip); - return err; - } chip->iobase = pci_resource_start(pci, 0); /* just to be sure */ @@ -2655,8 +2651,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_m3_free(chip); - return -ENOMEM; + err = -ENOMEM; + goto free_chip; } chip->irq = pci->irq; @@ -2666,10 +2662,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, dev_warn(card->dev, "can't allocate apm buffer\n"); #endif - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_m3_free(chip); - return err; - } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err) + goto free_chip; if ((err = snd_m3_mixer(chip)) < 0) return err; @@ -2699,6 +2694,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, *chip_ret = chip; return 0; +free_chip: + snd_m3_free(chip); + return err; } /* @@ -2741,23 +2739,19 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) break; } - if ((err = snd_m3_create(card, pci, - external_amp[dev], - amp_gpio[dev], - &chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_m3_create(card, pci, external_amp[dev], amp_gpio[dev], &chip); + if (err) + goto free_card; + card->private_data = chip; sprintf(card->shortname, "ESS %s PCI", card->driver); sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, chip->iobase, chip->irq); - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err) + goto free_card; #if 0 /* TODO: not supported yet */ /* TODO enable MIDI IRQ and I/O */ @@ -2772,6 +2766,9 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) pci_set_drvdata(pci, card); dev++; return 0; +free_card: + snd_card_free(card); + return err; } static void snd_m3_remove(struct pci_dev *pci) -- 2.14.0

3 3

[alsa-devel] USB dB data
by Jack Bates 06 Sep '17

06 Sep '17

I confirmed with Wireshark that my USB sound card responds to GET_MIN with 0x0000 and GET_MAX with 0x6300 .. The audio device class definition [1] says the units are 1/256 dB (section 5.2.2.2.3: Mixer Control) ... So by my calculation, the min and max are 0 and 99 dB (0x6300 / 256 == 99) ... Why then does alsactl print: > range '0 - 99' > dbmin 0 > dbmax 38 (/usr/sbin/aslactl -f - store) I originally asked about this on the pulseaudio-discuss list [2] because it uses "dB data" [3]. The "0 - 99" makes sense -- it lines up with my calculation ... Where does the 38 come from? (I expect it's obvious, I just haven't found the explanation?) Background: I just got a Bose SoundLink Revolve ... When I connect it with USB, the volume is silent from 0 to 99% and deafening at 100% (using PulseAudio). Directly adjusting the volume with alsamixer works properly: It increases smoothly from 0 to 100% -- changes that are reflected in the PulseAudio volume as fine adjustments in between 99 and 100% (64587 to 65536 PulseAudio volume). From what I've read there's a common problem where especially USB devices report the wrong dB data (don't follow the spec [4]) -- this has previously been handled by patching the USB audio driver [5] ... However I haven't figured out how my device deviates from the spec (0x0000 to 0x6300 or 0 to 99 dB seem plausible) -- my understanding is incomplete: I haven't figured out where the value 38 comes from ... [1] http://www.usb.org/developers/docs/devclass_docs/audio10.pdf [2] https://lists.freedesktop.org/archives/pulseaudio-discuss/2017-August/02860… [3] http://0pointer.de/blog/projects/decibel-data [4] https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/637066 [5] https://bugs.freedesktop.org/show_bug.cgi?id=93193#c14

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[alsa-devel] [PATCH] ASoC: improve RT5514 dependencies
by Arnd Bergmann 05 Sep '17

05 Sep '17

With SND_SOC_RT5514_SPI=m and SND_SOC_RT5514=y, we get this link error: sound/soc/codecs/rt5514.o: In function `rt5514_dsp_voice_wake_up_put': rt5514.c:(.text+0x2068): undefined reference to `rt5514_spi_burst_write' rt5514.c:(.text+0x20c8): undefined reference to `rt5514_spi_burst_write' This adds another Kconfig symbol to work around the link error, forcing the main driver to be a loadable module as well if some other driver selects the I2C side to be built-in. Fixes: 2a18483a7fb4 ("ASoC: Intel: Add Kabylake machine driver for RT5514, RT5663 and MAX98927") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- This can now lead to a configuration with SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH=y and SND_SOC_RT5514=m. I don't know if that is a problem or not. An alternative approach would be to flip the dependency between rt5514_spi.ko and rt5514.ko and avoid having the rt5514_spi_burst_write symbol exported. --- sound/soc/codecs/Kconfig | 7 ++++++- sound/soc/intel/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 2 +- sound/soc/rockchip/Kconfig | 1 - 4 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 024ddc9938ed..115c6144ca2a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -117,7 +117,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C - select SND_SOC_RT5514 if I2C + select SND_SOC_RT5514_I2C if I2C select SND_SOC_RT5616 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -738,6 +738,11 @@ config SND_SOC_RT298 config SND_SOC_RT5514 tristate + default m if SND_SOC_RT5514_SPI=m + default SND_SOC_RT5514_I2C || SND_SOC_RT5514_SPI + +config SND_SOC_RT5514_I2C + tristate config SND_SOC_RT5514_SPI tristate diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b301bfff1c09..f5ef912190ed 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -259,7 +259,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH select SND_SOC_INTEL_SST select SND_SOC_INTEL_SKYLAKE select SND_SOC_RT5663 - select SND_SOC_RT5514 + select SND_SOC_RT5514_I2C select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI help diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 5c68797f36c4..6ccf73ae77ca 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -67,7 +67,7 @@ config SND_SOC_MT8173_RT5650_RT5514 tristate "ASoC Audio driver for MT8173 with RT5650 RT5514 codecs" depends on SND_SOC_MT8173 && I2C select SND_SOC_RT5645 - select SND_SOC_RT5514 + select SND_SOC_RT5514_I2C help This adds ASoC driver for Mediatek MT8173 boards with the RT5650 and RT5514 codecs. diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index c84487805876..c1139aecdfbe 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -65,7 +65,6 @@ config SND_SOC_RK3399_GRU_SOUND depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP && SPI select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98357A - select SND_SOC_RT5514 select SND_SOC_DA7219 select SND_SOC_RT5514_SPI help -- 2.9.0

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[alsa-devel] [PATCH v2 1/2] ASoC: codec: use enable pin to control dmic start and stop
by Lin Huang 05 Sep '17

05 Sep '17

From: huang lin <hl(a)rock-chips.com> on some board use enable pin to control dmic start and stop, so add this feature in dmic driver. Signed-off-by: Lin Huang <hl(a)rock-chips.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/dmic.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 010811e..d98233b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,7 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 - select SND_SOC_DMIC + select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 12e07f9..b88a1ee 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,8 @@ * */ +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/module.h> @@ -27,6 +29,34 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> +static int dmic_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); + + if (!dmic_en) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + gpiod_set_value(dmic_en, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(dmic_en, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops dmic_dai_ops = { + .trigger = dmic_daiops_trigger, +}; + static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -38,8 +68,23 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &dmic_dai_ops, }; +static int dmic_codec_probe(struct snd_soc_codec *codec) +{ + struct gpio_desc *dmic_en; + + dmic_en = devm_gpiod_get_optional(codec->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic_en)) + return PTR_ERR(dmic_en); + + snd_soc_codec_set_drvdata(codec, dmic_en); + + return 0; +} + static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, SND_SOC_NOPM, 0, 0), @@ -51,6 +96,7 @@ static const struct snd_soc_dapm_route intercon[] = { }; static const struct snd_soc_codec_driver soc_dmic = { + .probe = dmic_codec_probe, .component_driver = { .dapm_widgets = dmic_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets), -- 2.7.4

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[alsa-devel] [RFC PATCH 0/7] AC97 device/driver model revamp
by Robert Jarzmik 04 Sep '17

04 Sep '17

It all started in the pxa device-tree submission here : https://lkml.org/lkml/2016/2/25/965 It will be maintained in : git fetch https://github.com/rjarzmik/linux.git work/ac97 And now it transformed into this RFC, which would bring a ground for AC'97 devices closer to the linux device/driver model. This is just an RFC to see if we understand each other Mark. Amongst the driving ideas behind this RFC : - use device/driver model for AC'97 device/driver binding is based on vendor_id{1,2} - use auto-probe of the AC'97 bus to enumerate codecs and create devices - be compatible with previous platform_data model for codecs => this enables a smooth transition, where a codec (here wm9713) can be used either through platform_device or automatic discovery - struct snd_ac97 is not used This structure is really heavy, and doesn't represent an AC97 device, but rather an AC'97 codec internals IIUC. I think it's right place would be in include/sound/ac97/codec.h. This is also a problem for regmap adherence, hence the include/sound/ac97/compat.h. - a new AC'97 bus Kconfig is created This was done for a smooth transition ... let's see if it's a good idea. - split the AC'97 into : - the codec - the digital controller - the bus Amongst the things that are not all touched yet : - ac97_codec.c The generic AC'97 codec in sound/pci/ac97/ac97_codec.c has really a lot features, which are not at all PCI related. They could be transfered to sound/ac97, without the pci specific field. => this deserves a deep ahead thinking, which I'll do if the current approach is accepted by the communauty. The most important patch is 2/7 "ALSA: ac97: add an ac97 bus". This is the main evaluation point for the serie, the others are here so that I can test it all. Well, this is a long term effort, which might need a complete rewrite according to the comments it'll get. Let's expose it for comments and see how I can progress with it. Cheers. -- Robert Robert Jarzmik (7): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: wm9713: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support ARM: pxa: mioa701 remove wm9713 from platform devices ASoC: mioa701_wm9713: convert to new ac97 bus ASoC: add new ac97 bus support arch/arm/mach-pxa/mioa701.c | 1 - include/sound/ac97/codec.h | 98 ++++++++++++ include/sound/ac97/compat.h | 21 +++ include/sound/ac97/controller.h | 39 +++++ include/sound/ac97/regs.h | 262 +++++++++++++++++++++++++++++++ include/sound/ac97_codec.h | 239 +---------------------------- include/sound/pxa2xx-lib.h | 14 +- sound/Kconfig | 7 + sound/Makefile | 1 + sound/ac97/Kconfig | 9 ++ sound/ac97/Makefile | 5 + sound/ac97/ac97_core.h | 10 ++ sound/ac97/bus.c | 330 ++++++++++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 ++ sound/ac97/snd_ac97_compat.c | 104 +++++++++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 31 ++-- sound/soc/Kconfig | 4 + sound/soc/codecs/wm9713.c | 62 +++++++- sound/soc/pxa/Kconfig | 4 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/pxa2xx-ac97.c | 12 +- 22 files changed, 1000 insertions(+), 273 deletions(-) create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.1.4

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[alsa-devel] [PATCH v5 00/12] AC97 device/driver model revamp
by Robert Jarzmik 04 Sep '17

04 Sep '17

Hi Lars, Mark, Charles, Lee, This is a minor revision after v4, mainly reviews from Takashi, Charles. I extracted the diff from the v3 of sound/ac97 which is in [1] for easier spotting of what has globally changed. I tentatively kept your Ack Charles for patch 11/12, please shout if you disagree (there is an #ifdef added). I would still need a review from you, Lee, for the mfd patch (patch 6/12). Happy review. -- Robert Robert Jarzmik (12): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 ++++++++++----- drivers/mfd/Kconfig | 15 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 405 ++++++++++++++++++++++++ include/linux/mfd/wm97xx.h | 31 ++ include/sound/ac97/codec.h | 118 +++++++ include/sound/ac97/compat.h | 20 ++ include/sound/ac97/controller.h | 85 +++++ include/sound/ac97/regs.h | 262 ++++++++++++++++ include/sound/ac97_codec.h | 239 +------------- include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 16 + sound/ac97/bus.c | 539 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 108 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 37 ++- sound/arm/pxa2xx-ac97.c | 35 ++- sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/wm9705.c | 61 ++-- sound/soc/codecs/wm9712.c | 39 ++- sound/soc/codecs/wm9713.c | 39 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 46 ++- 29 files changed, 2011 insertions(+), 414 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.11.0 [1] sound diff from last revision (git diff submit/ac97-3..submit/ac97-4 sound/ac97 include/sound/ac97) diff --git a/include/sound/ac97/compat.h b/include/sound/ac97/compat.h index d876464bf7e4..1351cba40048 100644 --- a/include/sound/ac97/compat.h +++ b/include/sound/ac97/compat.h @@ -13,7 +13,6 @@ #define AC97_COMPAT_H #include <sound/ac97_codec.h> -#include <sound/soc.h> struct snd_ac97 *snd_ac97_compat_alloc(struct ac97_codec_device *adev); void snd_ac97_compat_release(struct snd_ac97 *ac97); diff --git a/include/sound/ac97/controller.h b/include/sound/ac97/controller.h index 18c699925a85..b36ecdd64f14 100644 --- a/include/sound/ac97/controller.h +++ b/include/sound/ac97/controller.h @@ -8,12 +8,13 @@ #ifndef AC97_CONTROLLER_H #define AC97_CONTROLLER_H +#include <linux/device.h> #include <linux/list.h> #define AC97_BUS_MAX_CODECS 4 #define AC97_SLOTS_AVAILABLE_ALL 0xf -struct device; +struct ac97_controller_ops; /** * struct ac97_controller - The AC97 controller of the AC-Link @@ -21,8 +22,8 @@ struct device; * @controllers: linked list of all existing controllers. * @adap: the shell device ac97-%d, ie. ac97 adapter * @nr: the number of the shell device - * @parent: the device providing the AC97 controller. * @slots_available: the mask of accessible/scanable codecs. + * @parent: the device providing the AC97 controller. * @codecs: the 4 possible AC97 codecs (NULL if none found). * @codecs_pdata: platform_data for each codec (NULL if no pdata). * @@ -34,8 +35,8 @@ struct ac97_controller { struct list_head controllers; struct device adap; int nr; - struct device *parent; unsigned short slots_available; + struct device *parent; struct ac97_codec_device *codecs[AC97_BUS_MAX_CODECS]; void *codecs_pdata[AC97_BUS_MAX_CODECS]; }; diff --git a/sound/ac97/ac97_core.h b/sound/ac97/ac97_core.h index 219fec0d52e0..08441a4fda7c 100644 --- a/sound/ac97/ac97_core.h +++ b/sound/ac97/ac97_core.h @@ -8,3 +8,9 @@ unsigned int snd_ac97_bus_scan_one(struct ac97_controller *ac97, unsigned int codec_num); + +static inline bool ac97_ids_match(unsigned int id1, unsigned int id2, + unsigned int mask) +{ + return (id1 & mask) == (id2 & mask); +} diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index af618a0e89cb..31f858eceffc 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -105,6 +105,7 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, return 0; err_free_codec: + put_device(&codec->dev); kfree(codec); ac97_ctrl->codecs[idx] = NULL; @@ -171,14 +172,10 @@ static int ac97_bus_reset(struct ac97_controller *ac97_ctrl) */ int snd_ac97_codec_driver_register(struct ac97_codec_driver *drv) { - int ret; - drv->driver.bus = &ac97_bus_type; - ret = driver_register(&drv->driver); - - return ret; + return driver_register(&drv->driver); } -EXPORT_SYMBOL(snd_ac97_codec_driver_register); +EXPORT_SYMBOL_GPL(snd_ac97_codec_driver_register); /** * snd_ac97_codec_driver_unregister - unregister an AC97 codec driver @@ -190,7 +187,7 @@ void snd_ac97_codec_driver_unregister(struct ac97_codec_driver *drv) { driver_unregister(&drv->driver); } -EXPORT_SYMBOL(snd_ac97_codec_driver_unregister); +EXPORT_SYMBOL_GPL(snd_ac97_codec_driver_unregister); /** * snd_ac97_codec_get_platdata - get platform_data @@ -208,7 +205,7 @@ void *snd_ac97_codec_get_platdata(const struct ac97_codec_device *adev) return ac97_ctrl->codecs_pdata[adev->num]; } -EXPORT_SYMBOL(snd_ac97_codec_get_platdata); +EXPORT_SYMBOL_GPL(snd_ac97_codec_get_platdata); static void ac97_ctrl_codecs_unregister(struct ac97_controller *ac97_ctrl) { @@ -359,7 +356,7 @@ struct ac97_controller *snd_ac97_controller_register( kfree(ac97_ctrl); return ERR_PTR(ret); } -EXPORT_SYMBOL(snd_ac97_controller_register); +EXPORT_SYMBOL_GPL(snd_ac97_controller_register); /** * snd_ac97_controller_unregister - unregister an ac97 controller @@ -370,7 +367,7 @@ void snd_ac97_controller_unregister(struct ac97_controller *ac97_ctrl) { ac97_del_adapter(ac97_ctrl); } -EXPORT_SYMBOL(snd_ac97_controller_unregister); +EXPORT_SYMBOL_GPL(snd_ac97_controller_unregister); #ifdef CONFIG_PM static int ac97_pm_runtime_suspend(struct device *dev) @@ -466,7 +463,7 @@ static int ac97_bus_match(struct device *dev, struct device_driver *drv) return false; do { - if ((id[i].id & id->mask) == (adev->vendor_id & id[i].mask)) + if (ac97_ids_match(id[i].id, adev->vendor_id, id[i].mask)) return true; } while (id[i++].id); diff --git a/sound/ac97/snd_ac97_compat.c b/sound/ac97/snd_ac97_compat.c index ac8d835c1513..61544e0d8de4 100644 --- a/sound/ac97/snd_ac97_compat.c +++ b/sound/ac97/snd_ac97_compat.c @@ -88,16 +88,19 @@ int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, { struct ac97_codec_device *adev = to_ac97_device(ac97->private_data); struct ac97_controller *actrl = adev->ac97_ctrl; + unsigned int scanned; if (try_warm) { compat_ac97_warm_reset(ac97); - if (snd_ac97_bus_scan_one(actrl, adev->num) == adev->vendor_id) + scanned = snd_ac97_bus_scan_one(actrl, adev->num); + if (ac97_ids_match(scanned, adev->vendor_id, id_mask)) return 1; } compat_ac97_reset(ac97); compat_ac97_warm_reset(ac97); - if (snd_ac97_bus_scan_one(actrl, adev->num) == adev->vendor_id) + scanned = snd_ac97_bus_scan_one(actrl, adev->num); + if (ac97_ids_match(scanned, adev->vendor_id, id_mask)) return 0; return -ENODEV;

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