Merge tag 'sound-5.11-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Lots of changes (slightly more code increase than usual) at this time,
  while most of code changes are ASoC driver-specific.

  Here are some highlights:

  Core:

   - The new auxiliary bus implementation for Intel DSP, which will be
     used by other drivers as well

   - Lots of ASoC core cleanups and refactoring

   - UBSAN and KCSAN fixes in rawmidi, sequencer and a few others

   - Compress-offload API enhancement for the pause during draining

  HD- and USB-audio:

   - Enhancements of the USB-audio implicit feedback support, including
     better full-duplex operations

   - Continued CA0132 improvements and fixes

   - A few new quirk entries, HDMI audio fixes

  ASoC:

   - Support for boot time selection of Intel DSP firmware, which should
     help distros/users testing new stuff more easily; the kconfig was
     moved to boot time option, too

   - Some basic DPCM support in audio graph card

   - Removal of old pre-DT Freescale drivers

   - Support for Allwinner H6 I2S, Analog Devices ADAU1372, Intel
     Alderlake-S, GMediatek MT8192, NXP i.MX HDMI and XCVR, Realtek
     RT715, Qualcomm SM8250 and simple GPIO based muxes"

* tag 'sound-5.11-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (445 commits)
  ALSA: pcm: oss: Fix potential out-of-bounds shift
  ALSA: usb-audio: Fix potential out-of-bounds shift
  ALSA: hda/ca0132 - Add ZxR surround DAC setup.
  ALSA: hda/ca0132 - Add 8051 PLL write helper functions.
  ALSA: hda/hdmi: packet buffer index must be set before reading value
  ASoC: SOF: imx: update kernel-doc description
  ASoC: mediatek: mt8183: delete some unreachable code
  ASoC: mediatek: mt8183: add PM ops to machine drivers
  ASoC: topology: Fix wrong size check
  ASoC: topology: Add missing size check
  ASoC: SOF: Intel: hda: fix the condition passed to sof_dev_dbg_or_err
  ASoC: SOF: modify the SOF_DBG flags
  ASoC: SOF: Intel: hda: remove duplicated status dump
  ASoC: rt1015p: delay 300ms after SDB pulling high for calibration
  ASoC: rt1015p: move SDB control from trigger to DAPM
  ASoC: wm_adsp: remove "ctl" from list on error in wm_adsp_create_control()
  ALSA: usb-audio: Fix control 'access overflow' errors from chmap
  ALSA: hda/hdmi: always print pin NIDs as hexadecimal
  ALSA: hda/realtek - Add supported for more Lenovo ALC285 Headset Button
  ALSA: hda/ca0132 - Remove now unnecessary DSP setup functions.
  ...

1 files changed, 109 insertions, 18 deletions

diff --git a/sound/usb/format.c b/sound/usb/format.c
index 3bfead393aa3..9ebc5d202c87 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -16,7 +16,6 @@
 #include "card.h"
 #include "quirks.h"
 #include "helper.h"
-#include "debug.h"
 #include "clock.h"
 #include "format.h"
 
@@ -40,6 +39,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
 	case UAC_VERSION_1:
 	default: {
 		struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
+		if (format >= 64)
+			return 0; /* invalid format */
 		sample_width = fmt->bBitResolution;
 		sample_bytes = fmt->bSubframeSize;
 		format = 1ULL << format;
@@ -165,6 +166,23 @@ static int set_fixed_rate(struct audioformat *fp, int rate, int rate_bits)
 	return 0;
 }
 
+/* set up rate_min, rate_max and rates from the rate table */
+static void set_rate_table_min_max(struct audioformat *fp)
+{
+	unsigned int rate;
+	int i;
+
+	fp->rate_min = INT_MAX;
+	fp->rate_max = 0;
+	fp->rates = 0;
+	for (i = 0; i < fp->nr_rates; i++) {
+		rate = fp->rate_table[i];
+		fp->rate_min = min(fp->rate_min, rate);
+		fp->rate_max = max(fp->rate_max, rate);
+		fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+	}
+}
+
 /*
  * parse the format descriptor and stores the possible sample rates
  * on the audioformat table (audio class v1).
@@ -199,7 +217,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 			return -ENOMEM;
 
 		fp->nr_rates = 0;
-		fp->rate_min = fp->rate_max = 0;
 		for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) {
 			unsigned int rate = combine_triple(&fmt[idx]);
 			if (!rate)
@@ -218,18 +235,15 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 			     chip->usb_id == USB_ID(0x041e, 0x4068)))
 				rate = 8000;
 
-			fp->rate_table[fp->nr_rates] = rate;
-			if (!fp->rate_min || rate < fp->rate_min)
-				fp->rate_min = rate;
-			if (!fp->rate_max || rate > fp->rate_max)
-				fp->rate_max = rate;
-			fp->rates |= snd_pcm_rate_to_rate_bit(rate);
-			fp->nr_rates++;
+			fp->rate_table[fp->nr_rates++] = rate;
 		}
 		if (!fp->nr_rates) {
-			hwc_debug("All rates were zero. Skipping format!\n");
+			usb_audio_info(chip,
+				       "%u:%d: All rates were zero\n",
+				       fp->iface, fp->altsetting);
 			return -EINVAL;
 		}
+		set_rate_table_min_max(fp);
 	} else {
 		/* continuous rates */
 		fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
@@ -335,8 +349,6 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
 {
 	int i, nr_rates = 0;
 
-	fp->rates = fp->rate_min = fp->rate_max = 0;
-
 	for (i = 0; i < nr_triplets; i++) {
 		int min = combine_quad(&data[2 + 12 * i]);
 		int max = combine_quad(&data[6 + 12 * i]);
@@ -372,12 +384,6 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
 
 			if (fp->rate_table)
 				fp->rate_table[nr_rates] = rate;
-			if (!fp->rate_min || rate < fp->rate_min)
-				fp->rate_min = rate;
-			if (!fp->rate_max || rate > fp->rate_max)
-				fp->rate_max = rate;
-			fp->rates |= snd_pcm_rate_to_rate_bit(rate);
-
 			nr_rates++;
 			if (nr_rates >= MAX_NR_RATES) {
 				usb_audio_err(chip, "invalid uac2 rates\n");
@@ -417,6 +423,85 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip,
 	return -ENODEV;
 }
 
+/* check whether the given altsetting is supported for the already set rate */
+static bool check_valid_altsetting_v2v3(struct snd_usb_audio *chip, int iface,
+					int altsetting)
+{
+	struct usb_device *dev = chip->dev;
+	__le64 raw_data = 0;
+	u64 data;
+	int err;
+
+	/* we assume 64bit is enough for any altsettings */
+	if (snd_BUG_ON(altsetting >= 64 - 8))
+		return false;
+
+	err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
+			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+			      UAC2_AS_VAL_ALT_SETTINGS << 8,
+			      iface, &raw_data, sizeof(raw_data));
+	if (err < 0)
+		return false;
+
+	data = le64_to_cpu(raw_data);
+	/* first byte contains the bitmap size */
+	if ((data & 0xff) * 8 < altsetting)
+		return false;
+	if (data & (1ULL << (altsetting + 8)))
+		return true;
+
+	return false;
+}
+
+/*
+ * Validate each sample rate with the altsetting
+ * Rebuild the rate table if only partial values are valid
+ */
+static int validate_sample_rate_table_v2v3(struct snd_usb_audio *chip,
+					   struct audioformat *fp,
+					   int clock)
+{
+	struct usb_device *dev = chip->dev;
+	unsigned int *table;
+	unsigned int nr_rates;
+	int i, err;
+
+	table = kcalloc(fp->nr_rates, sizeof(*table), GFP_KERNEL);
+	if (!table)
+		return -ENOMEM;
+
+	/* clear the interface altsetting at first */
+	usb_set_interface(dev, fp->iface, 0);
+
+	nr_rates = 0;
+	for (i = 0; i < fp->nr_rates; i++) {
+		err = snd_usb_set_sample_rate_v2v3(chip, fp, clock,
+						   fp->rate_table[i]);
+		if (err < 0)
+			continue;
+
+		if (check_valid_altsetting_v2v3(chip, fp->iface, fp->altsetting))
+			table[nr_rates++] = fp->rate_table[i];
+	}
+
+	if (!nr_rates) {
+		usb_audio_dbg(chip,
+			      "No valid sample rate available for %d:%d, assuming a firmware bug\n",
+			      fp->iface, fp->altsetting);
+		nr_rates = fp->nr_rates; /* continue as is */
+	}
+
+	if (fp->nr_rates == nr_rates) {
+		kfree(table);
+		return 0;
+	}
+
+	kfree(fp->rate_table);
+	fp->rate_table = table;
+	fp->nr_rates = nr_rates;
+	return 0;
+}
+
 /*
  * parse the format descriptor and stores the possible sample rates
  * on the audioformat table (audio class v2 and v3).
@@ -509,6 +594,12 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip,
 	 * allocated, so the rates will be stored */
 	parse_uac2_sample_rate_range(chip, fp, nr_triplets, data);
 
+	ret = validate_sample_rate_table_v2v3(chip, fp, clock);
+	if (ret < 0)
+		goto err_free;
+
+	set_rate_table_min_max(fp);
+
 err_free:
 	kfree(data);
 err: