Merge tag 'sound-5.12-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "A relatively calm release at this time, and no massive code changes
  are found in the stats, while a wide range of code refactoring and
  cleanup have been done.

  Note that this update includes the tree-wide trivial changes for
  dropping the return value from ISA remove callbacks, too.

  Below lists up some highlight:

  ALSA Core:
   - Support for the software jack injection via debugfs
   - Fixes for sync_stop PCM operations

  HD-audio and USB-audio:
   - A few usual HD-audio device quirks
   - Updates for Tegra HD-audio
   - More quirks for Pioneer and other USB-audio devices
   - Stricter state checks at USB-audio disconnection

  ASoC:
   - Continued code refactoring, cleanup and fixes in ASoC core API
   - A KUnit testsuite for the topology code
   - Lots of ASoC Intel driver Realtek codec updates, quirk additions
     and fixes
   - Support for Ingenic JZ4760(B), Intel AlderLake-P, DT configured
     nVidia cards, Qualcomm lpass-rx-macro and lpass-tx-macro
   - Removal of obsolete SIRF prima/atlas, Txx9 and ZTE zx drivers

  Others:
   - Drop return value from ISA driver remove callback
   - Cleanup with DIV_ROUND_UP() macro
   - FireWire updates, HDSP output loopback support"

* tag 'sound-5.12-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (322 commits)
  ALSA: hda: intel-dsp-config: add Alder Lake support
  ASoC: soc-pcm: fix hw param limits calculation for multi-DAI
  ASoC: Intel: bytcr_rt5640: Add quirk for the Acer One S1002 tablet
  ASoC: Intel: bytcr_rt5651: Add quirk for the Jumper EZpad 7 tablet
  ASoC: Intel: bytcr_rt5640: Add quirk for the Voyo Winpad A15 tablet
  ASoC: Intel: bytcr_rt5640: Add quirk for the Estar Beauty HD MID 7316R tablet
  ASoC: soc-pcm: fix hwparams min/max init for dpcm
  ALSA: hda/realtek: Quirk for HP Spectre x360 14 amp setup
  ALSA: usb-audio: Add implicit fb quirk for BOSS GP-10
  ALSA: hda: Add another CometLake-H PCI ID
  ASoC: soc-pcm: add soc_pcm_hw_update_format()
  ASoC: soc-pcm: add soc_pcm_hw_update_chan()
  ASoC: soc-pcm: add soc_pcm_hw_update_rate()
  ASoC: wm_adsp: Remove unused control callback structure
  ASoC: SOF: relax ABI checks and avoid unnecessary warnings
  ASoC: codecs: lpass-tx-macro: add dapm widgets and route
  ASoC: codecs: lpass-tx-macro: add support for lpass tx macro
  ASoC: qcom: dt-bindings: add bindings for lpass tx macro codec
  ASoC: codecs: lpass-rx-macro: add iir widgets
  ASoC: codecs: lpass-rx-macro: add dapm widgets and route
  ...

1 files changed, 131 insertions, 8 deletions

diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
index f046987ee4cd..05bbacd0d174 100644
--- a/sound/soc/codecs/cpcap.c
+++ b/sound/soc/codecs/cpcap.c
@@ -16,6 +16,14 @@
 #include <sound/soc.h>
 #include <sound/tlv.h>
 
+/* Register 512 CPCAP_REG_VAUDIOC --- Audio Regulator and Bias Voltage */
+#define CPCAP_BIT_AUDIO_LOW_PWR           6
+#define CPCAP_BIT_AUD_LOWPWR_SPEED        5
+#define CPCAP_BIT_VAUDIOPRISTBY           4
+#define CPCAP_BIT_VAUDIO_MODE1            2
+#define CPCAP_BIT_VAUDIO_MODE0            1
+#define CPCAP_BIT_V_AUDIO_EN              0
+
 /* Register 513 CPCAP_REG_CC     --- CODEC */
 #define CPCAP_BIT_CDC_CLK2                15
 #define CPCAP_BIT_CDC_CLK1                14
@@ -221,6 +229,7 @@ struct cpcap_reg_info {
 };
 
 static const struct cpcap_reg_info cpcap_default_regs[] = {
+	{ CPCAP_REG_VAUDIOC, 0x003F, 0x0000 },
 	{ CPCAP_REG_CC, 0xFFFF, 0x0000 },
 	{ CPCAP_REG_CC, 0xFFFF, 0x0000 },
 	{ CPCAP_REG_CDI, 0xBFFF, 0x0000 },
@@ -1264,12 +1273,12 @@ static int cpcap_voice_hw_params(struct snd_pcm_substream *substream,
 
 	if (direction == SNDRV_PCM_STREAM_CAPTURE) {
 		mask = 0x0000;
-		mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0;
-		mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1;
-		mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2;
-		mask |= CPCAP_BIT_MIC2_TIMESLOT0;
-		mask |= CPCAP_BIT_MIC2_TIMESLOT1;
-		mask |= CPCAP_BIT_MIC2_TIMESLOT2;
+		mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0);
+		mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT1);
+		mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT2);
+		mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT0);
+		mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT1);
+		mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT2);
 		val = 0x0000;
 		if (channels >= 2)
 			val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0);
@@ -1371,8 +1380,121 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int cpcap_voice_set_mute(struct snd_soc_dai *dai,
-				int mute, int direction)
+
+/*
+ * Configure codec for voice call if requested.
+ *
+ * We can configure most with snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt()
+ * and snd_soc_dai_set_tdm_slot(). This function configures the rest of the
+ * cpcap related hardware as CPU is not involved in the voice call.
+ */
+static int cpcap_voice_call(struct cpcap_audio *cpcap, struct snd_soc_dai *dai,
+			    bool voice_call)
+{
+	int mask, err;
+
+	/* Modem to codec VAUDIO_MODE1 */
+	mask = BIT(CPCAP_BIT_VAUDIO_MODE1);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_VAUDIOC,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Clear MIC1_MUX for call */
+	mask = BIT(CPCAP_BIT_MIC1_MUX);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
+				 mask, voice_call ? 0 : mask);
+	if (err)
+		return err;
+
+	/* Set MIC2_MUX for call */
+	mask = BIT(CPCAP_BIT_MB_ON1L) | BIT(CPCAP_BIT_MB_ON1R) |
+		BIT(CPCAP_BIT_MIC2_MUX) | BIT(CPCAP_BIT_MIC2_PGA_EN);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Enable LDSP for call */
+	mask = BIT(CPCAP_BIT_A2_LDSP_L_EN) | BIT(CPCAP_BIT_A2_LDSP_R_EN);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXOA,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Enable CPCAP_BIT_PGA_CDC_EN for call */
+	mask = BIT(CPCAP_BIT_PGA_CDC_EN);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXCOA,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Unmute voice for call */
+	if (dai) {
+		err = snd_soc_dai_digital_mute(dai, !voice_call,
+					       SNDRV_PCM_STREAM_PLAYBACK);
+		if (err)
+			return err;
+	}
+
+	/* Set modem to codec mic CDC and HPF for call */
+	mask = BIT(CPCAP_BIT_MIC2_CDC_EN) | BIT(CPCAP_BIT_CDC_EN_RX) |
+	       BIT(CPCAP_BIT_AUDOHPF_1) | BIT(CPCAP_BIT_AUDOHPF_0) |
+	       BIT(CPCAP_BIT_AUDIHPF_1) | BIT(CPCAP_BIT_AUDIHPF_0);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CC,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Enable modem to codec CDC for call*/
+	mask = BIT(CPCAP_BIT_CDC_CLK_EN);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
+				 mask, voice_call ? mask : 0);
+
+	return err;
+}
+
+static int cpcap_voice_set_tdm_slot(struct snd_soc_dai *dai,
+				    unsigned int tx_mask, unsigned int rx_mask,
+				    int slots, int slot_width)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
+	int err, ts_mask, mask;
+	bool voice_call;
+
+	/*
+	 * Primitive test for voice call, probably needs more checks
+	 * later on for 16-bit calls detected, Bluetooth headset etc.
+	 */
+	if (tx_mask == 0 && rx_mask == 1 && slot_width == 8)
+		voice_call = true;
+	else
+		voice_call = false;
+
+	ts_mask = 0x7 << CPCAP_BIT_MIC2_TIMESLOT0;
+	ts_mask |= 0x7 << CPCAP_BIT_MIC1_RX_TIMESLOT0;
+
+	mask = (tx_mask & 0x7) << CPCAP_BIT_MIC2_TIMESLOT0;
+	mask |= (rx_mask & 0x7) << CPCAP_BIT_MIC1_RX_TIMESLOT0;
+
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
+				 ts_mask, mask);
+	if (err)
+		return err;
+
+	err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, slot_width * 1000);
+	if (err)
+		return err;
+
+	err = cpcap_voice_call(cpcap, dai, voice_call);
+	if (err)
+		return err;
+
+	return 0;
+}
+
+static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
@@ -1393,6 +1515,7 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = {
 	.hw_params	= cpcap_voice_hw_params,
 	.set_sysclk	= cpcap_voice_set_dai_sysclk,
 	.set_fmt	= cpcap_voice_set_dai_fmt,
+	.set_tdm_slot	= cpcap_voice_set_tdm_slot,
 	.mute_stream	= cpcap_voice_set_mute,
 	.no_capture_mute = 1,
 };