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

[alsa-devel] [PATCH 0/9] AC97 device/driver model revamp
by Robert Jarzmik 19 Sep '17

19 Sep '17

It all started in the pxa device-tree submission here : https://lkml.org/lkml/2016/2/25/965 Last submission was a RFC in here: http://www.gossamer-threads.com/lists/linux/kernel/2446863 It will be maintained in : git fetch https://github.com/rjarzmik/linux.git work/ac97 The driving ideas are still the same, and I put them in [1] for memory. This is the first post RFC submission. In order to make a full demonstration of the framework, wm97xx was converted to an MFD, see [2] for the "why". As this serie is a complete wm9713 change towards the new AC97 bus, here is the patch organization : - 1/9, 2/9, 3/9 and 5/9: the new AC97 bus, for Mark and Takashi to review - 4/9: wm9713 conversion, this one has ugly #ifdefs, should be reviewed more carefully - 6/9: prerequisite for final 9/9, for Marek and Sebastian to approve => this one is independent and can flow through Sebastian's tree in this next-cycle if accepted - 7/9: prerequisite for final 9/9, for Mark/Dmitry => this one is also independent and can flow in this cycle if accepted - 8/9: prerequisite for final 9/9, for Lee to review => this one has a dependency on 2/9, so it probably can't fit in this next-cycle - 9/9: this last one depends on all the patches before, so it can't fit in this next-cycle As a sum-up, I'd like to push for review for this -next cycle at least the patches 1, 2, 3, 5, 6 and 9. Happy review. Robert Jarzmik (9): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: wm9713: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support power_supply: wm97xx_battery: use power_supply_get_drvdata Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 247 +++++++++++----- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 282 ++++++++++++++++++ drivers/power/supply/wm97xx_battery.c | 25 +- include/linux/mfd/wm97xx.h | 31 ++ include/sound/ac97/codec.h | 115 +++++++ include/sound/ac97/compat.h | 21 ++ 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 | 10 + sound/ac97/bus.c | 510 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 105 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 39 ++- sound/soc/Kconfig | 4 + sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/wm9713.c | 37 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 22 +- 28 files changed, 1738 insertions(+), 382 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 -- 2.1.4 [1] RFC PATCH reminder: 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. [2] As a corollary, wm97xx is changed into a MFD device, which at least spawns: - the touchscreen device - the ac97 codec - the problem to solve was stated in http://www.gossamer-threads.com/lists/linux/kernel/2510113#2510113

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[alsa-devel] [PATCH 0/3] Fix capture devices functionality on TM2
by Jaechul Lee 19 Sep '17

19 Sep '17

Hello, This patchset makes capture functionality working on TM2. Capture devices can't be detected because I2S doesn't have appropriate dai name. Therefore, TM2 will select unexpected cpu dai which is named "i2s-samsung-sec" that doesn't have a capture functionality. In samsung I2S driver, it tries to register two components without specific name. The driver finally has components having same name: dai name as well. As a result, dai_link doesn't have enough information to select cpu_dai properly. Jaechul Lee (3): ASoC: samsung: i2s: Use specific name for i2s dais ASoC: samsung: Use 'samsung-i2s' cpu_dai for dai_links ASoC: samsung: Fix invalid argument when devm_gpiod_get is called sound/soc/samsung/i2s.c | 6 ++++++ sound/soc/samsung/tm2_wm5110.c | 7 ++++--- 2 files changed, 10 insertions(+), 3 deletions(-) -- 2.14.1

3 7

[alsa-devel] [PATCH 0/2] Add ACPI table for the ts3a227e driver
by Thierry Escande 19 Sep '17

19 Sep '17

Hi, This series adds ACPI match table for the ts3a227e driver. The ts3a227e is an Autonomous Audio Accessory Detection and Configuration Switch used to detect audio accessory features like jack plugging and unplugging, mic presence, and button pressed events. This series also fixes a null pointer dereference occurring in the TI max9890 driver when the ts3a227e hardware is present and its driver probed. The jack detection was enabled in the ts3a227e driver before the jack structure was allocated, causing the null pointer access. Regards, Thierry Fang, Yang A (1): ASoC: ts3a227e: add acpi table Thierry Escande (1): ASoc: cht_bsw_max98090_ti: Fix jack initialization sound/soc/codecs/ts3a227e.c | 10 ++++++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 52 +++++++++++++++++----------- 2 files changed, 41 insertions(+), 21 deletions(-) -- 2.7.4

3 6

[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

5 35

[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; }

4 13

[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

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[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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