Merge tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became again a busy development cycle.  There are few ALSA core
  updates (merely API cleanups and sparse fixes), with the majority of
  other changes are found in ASoC scene.

  Here are some highlights:

  ALSA core:
   - More helper macros for sparse warning fixes (e.g. bitwise types)
   - Slight optimization of PCM OSS locks
   - Make common handling for PCM / compress buffers (for SOF)

  ASoC:
   - Lots of code refactoring and modernization for (still ongoing)
     componentization works
   - Conversion of SND_SOC_ALL_CODECS to use imply
   - Continued refactoring and fixing of the Intel SOF/SST support,
     including the initial (but still incomplete) SoundWire support
   - SoundWire and more advanced clocking support for Realtek RT5682
   - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
     DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563
     and TLV320ADCX140

  HD-audio:
   - Optimizations in HDMI jack handling
   - A few new quirks and fixups for Realtek codecs

  USB-audio:
   - Delayed registration support
   - New quirks for Motu, Kingston, Presonus"

* tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits)
  ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
  Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
  ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups
  ALSA: hda/realtek - Set principled PC Beep configuration for ALC256
  ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256
  ALSA: hda/realtek - a fake key event is triggered by running shutup
  ALSA: hda: default enable CA0132 DSP support
  ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
  ASoC: tlv320adcx140: Remove undocumented property
  ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function
  ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
  ASoC: Intel: boards: add sof_sdw machine driver
  ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms
  ASoC: rt5682: move DAI clock registry to I2S mode
  ASoC: pxa: magician: convert to use i2c_new_client_device()
  ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities
  Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread
  ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
  ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks
  ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers
  ...

1 files changed, 37 insertions, 0 deletions

diff --git a/sound/usb/format.c b/sound/usb/format.c
index 9f5cb4ed3a0c..50e1874c847c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -247,6 +247,36 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 	return 0;
 }
 
+
+/*
+ * Presonus Studio 1810c supports a limited set of sampling
+ * rates per altsetting but reports the full set each time.
+ * If we don't filter out the unsupported rates and attempt
+ * to configure the card, it will hang refusing to do any
+ * further audio I/O until a hard reset is performed.
+ *
+ * The list of supported rates per altsetting (set of available
+ * I/O channels) is described in the owner's manual, section 2.2.
+ */
+static bool s1810c_valid_sample_rate(struct audioformat *fp,
+				     unsigned int rate)
+{
+	switch (fp->altsetting) {
+	case 1:
+		/* All ADAT ports available */
+		return rate <= 48000;
+	case 2:
+		/* Half of ADAT ports available */
+		return (rate == 88200 || rate == 96000);
+	case 3:
+		/* Analog I/O only (no S/PDIF nor ADAT) */
+		return rate >= 176400;
+	default:
+		return false;
+	}
+	return false;
+}
+
 /*
  * Helper function to walk the array of sample rate triplets reported by
  * the device. The problem is that we need to parse whole array first to
@@ -283,6 +313,12 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
 		}
 
 		for (rate = min; rate <= max; rate += res) {
+
+			/* Filter out invalid rates on Presonus Studio 1810c */
+			if (chip->usb_id == USB_ID(0x0194f, 0x010c) &&
+			    !s1810c_valid_sample_rate(fp, rate))
+				goto skip_rate;
+
 			if (fp->rate_table)
 				fp->rate_table[nr_rates] = rate;
 			if (!fp->rate_min || rate < fp->rate_min)
@@ -297,6 +333,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
 				break;
 			}
 
+skip_rate:
 			/* avoid endless loop */
 			if (res == 0)
 				break;