Commit 7f04f3ed authored by Linus Torvalds's avatar Linus Torvalds
Browse files

Merge tag 'sound-5.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound 

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes over several drivers, but all are driver-
  specific and nothing looks scary.

  Slightly large changes are seen in ASoC qcom driver for the bugs that
  were revealed by the recent ASoC core change to report the invalid
  register access errors. Also ASoC fsl got a slight intensive change
  for the distortion fix.

  Others are only trivial fixes or device-specific quirks"

* tag 'sound-5.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (25 commits)
  ALSA: hda: avoid reset of sdo_limit
  ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion
  ALSA: usb-audio: ignore broken processing/extension unit
  ASoC: intel: Fix memleak in sst_media_open
  ASoC: wm8994: Avoid attempts to read unreadable registers
  ASoC: msm8916-wcd-analog: fix register Interrupt offset
  ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
  ALSA: hda/realtek: Add model alc298-samsung-headphone
  ALSA: usb-audio: Update documentation comment for MS2109 quirk
  ALSA: isa: fix spelling mistakes in the comments
  ALSA: usb-audio: Add capture support for Saffire 6 (USB 1.1)
  ALSA: hda/realtek: Add quirk for Samsung Galaxy Flex Book
  ASoC: q6routing: add dummy register read/write function
  ASoC: q6afe-dai: mark all widgets registers as SND_SOC_NOPM
  ASoC: Make soc_component_read() returning an error code again
  ASoC: amd: Replacing component->name with codec_dai->name.
  ASoC: fsl: Fix unused variable warning
  ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n
  ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n
  ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n
  ...
parents 43d387a4 b90b925f

sound/hda/hdac_bus.c

+12 −0
Original line number Diff line number Diff line
@@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, 
	INIT_LIST_HEAD(&bus->hlink_list);

	init_waitqueue_head(&bus->rirb_wq);

	bus->irq = -1;



	/*

	 * Default value of '8' is as per the HD audio specification (Rev 1.0a).

	 * Following relation is used to derive STRIPE control value.

	 *  For sample rate <= 48K:

	 *   { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }

	 *  For sample rate > 48K:

	 *   { ((num_channels * bits_per_sample * rate/48000) /

	 *	number of SDOs) >= 8 }

	 */

	bus->sdo_limit = 8;



	return 0;

}

EXPORT_SYMBOL_GPL(snd_hdac_bus_init);

sound/hda/hdac_controller.c

+0 −11
Original line number Diff line number Diff line
@@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) 


	bus->chip_init = true;



	/*

	 * Default value of '8' is as per the HD audio specification (Rev 1.0a).

	 * Following relation is used to derive STRIPE control value.

	 *  For sample rate <= 48K:

	 *   { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }

	 *  For sample rate > 48K:

	 *   { ((num_channels * bits_per_sample * rate/48000) /

	 *	number of SDOs) >= 8 }

	 */

	bus->sdo_limit = 8;



	return true;

}

EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip);

sound/isa/sscape.c

+3 −3
Original line number Diff line number Diff line
@@ -308,7 +308,7 @@ static inline int verify_mpu401(const struct snd_mpu401 *mpu) 
}



/*

 * This is apparently the standard way to initailise an MPU-401

 * This is apparently the standard way to initialise an MPU-401

 */

static inline void initialise_mpu401(const struct snd_mpu401 *mpu)

{
@@ -339,7 +339,7 @@ static void soundscape_free(struct snd_card *c) 
}



/*

 * Tell the SoundScape to begin a DMA tranfer using the given channel.

 * Tell the SoundScape to begin a DMA transfer using the given channel.

 * All locking issues are left to the caller.

 */

static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
@@ -803,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 *mpu) 
}



/*

 * Initialse an MPU-401 subdevice for MIDI support on the SoundScape.

 * Initialise an MPU-401 subdevice for MIDI support on the SoundScape.

 */

static int create_mpu401(struct snd_card *card, int devnum,

			 unsigned long port, int irq)

sound/pci/hda/patch_realtek.c

+3 −0
Original line number Diff line number Diff line
@@ -7694,6 +7694,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 
	SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),

	SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),

	SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),

	SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),

	SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),

	SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),

	SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),

	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
@@ -7955,6 +7957,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { 
	{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},

	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},

	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},

	{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},

	{}

};

#define ALC225_STANDARD_PINS \

sound/soc/amd/acp3x-rt5682-max9836.c

+1 −1
Original line number Diff line number Diff line
@@ -138,7 +138,7 @@ static int acp3x_1015_hw_params(struct snd_pcm_substream *substream, 
	srate = params_rate(params);



	for_each_rtd_codec_dais(rtd, i, codec_dai) {

		if (strcmp(codec_dai->component->name, "rt1015-aif"))

		if (strcmp(codec_dai->name, "rt1015-aif"))

			continue;

		ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64);

		if (ret < 0)