Merge tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Many updates in this development cycle are found in ASoC where it got
  a wide range of changes for the continued refactoring.

  Some highlights are below.

  ASoC:

   - Continued refactoring work by Morimoto-san toward the full
     componentization; the changes are seen allover the places

   - Support for force disconnecting muxes in DAPM

   - Continued development of ASoC Intel SOF stuff

   - New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
     CX2072X, Realtek RT1011 and RT1308

  HD-audio:

   - More fixes and adjustments for ASoC SOF HD-audio

   - Fix for resume problem on some Realtek codecs

  USB-audio:

   - A few fixes for the issues reported by syzbot USB fuzzer

   - Fix for UAC2 extension unit parser

   - Quirks for Line6 Helix, Emgaic Unitor 8

  FireWire:

   - Lots of code refactoring and fixes in most of its components"

* tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (626 commits)
  ALSA: firewire-lib: code refactoring for local variables
  ALSA: firewire-lib: code refactoring for post operation to data block counter
  ALSA: firewire-lib: code refactoring for error path of parser for CIP header
  ALSA: firewire-lib: fix different data block counter between probed event and transferred isochronous packet
  ALSA: firewire-lib: fix initial value of data block count for IR context without CIP_DBC_IS_END_EVENT
  ALSA: firewire-lib/fireface: fix initial value of data block counter for IR context with CIP_NO_HEADER
  ALSA: firewire-lib: fix invalid length of rx packet payload for tracepoint events
  ALSA: usb-audio: fix Line6 Helix audio format rates
  firewire-motu: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: firewire-digi00x: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: dice: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: oxfw: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: fireworks: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: bebob: fix wrong reference count for stream functionality at error path of rawmidi interface
  ASoC: SOF: Intel: implement runtime idle for CNL/APL
  ASoC: SOF: add runtime idle callback
  ASoC: hdac_hdmi: report codec link up/down status to bus
  ASoC: SOF: debug: fix possible memory leak in sof_dfsentry_write()
  ASoC: sunxi: sun50i-codec-analog: Add earpiece
  ASoC: rt5665: remove redundant assignment to variable idx
  ...

1 files changed, 76 insertions, 15 deletions

diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index ca568b9bf0f2..f1104d7d6426 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -53,6 +53,9 @@ struct pcm3168a_priv {
 	unsigned long sysclk;
 	unsigned int adc_fmt;
 	unsigned int dac_fmt;
+	int tdm_slots;
+	u32 tdm_mask[2];
+	int slot_width;
 };
 
 static const char *const pcm3168a_roll_off[] = { "Sharp", "Slow" };
@@ -384,6 +387,47 @@ static int pcm3168a_set_dai_fmt_adc(struct snd_soc_dai *dai,
 	return pcm3168a_set_dai_fmt(dai, format, false);
 }
 
+static int pcm3168a_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 pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
+
+	if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) {
+		dev_err(component->dev,
+			"Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n",
+			tx_mask, rx_mask, slots);
+		return -EINVAL;
+	}
+
+	if (slot_width &&
+	    (slot_width != 16 && slot_width != 24 && slot_width != 32 )) {
+		dev_err(component->dev, "Unsupported slot_width %d\n",
+			slot_width);
+		return -EINVAL;
+	}
+
+	if (pcm3168a->tdm_slots && pcm3168a->tdm_slots != slots) {
+		dev_err(component->dev, "Not matching slots %d vs %d\n",
+			pcm3168a->tdm_slots, slots);
+		return -EINVAL;
+	}
+
+	if (pcm3168a->slot_width && pcm3168a->slot_width != slot_width) {
+		dev_err(component->dev, "Not matching slot_width %d vs %d\n",
+			pcm3168a->slot_width, slot_width);
+		return -EINVAL;
+	}
+
+	pcm3168a->tdm_slots = slots;
+	pcm3168a->slot_width = slot_width;
+	pcm3168a->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+	pcm3168a->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
+
+	return 0;
+}
+
 static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params,
 			     struct snd_soc_dai *dai)
@@ -393,11 +437,10 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 	bool tx, master_mode;
 	u32 val, mask, shift, reg;
 	unsigned int rate, fmt, ratio, max_ratio;
-	unsigned int chan;
-	int i, min_frame_size;
+	unsigned int tdm_slots;
+	int i, slot_width;
 
 	rate = params_rate(params);
-	chan = params_channels(params);
 
 	ratio = pcm3168a->sysclk / rate;
 
@@ -428,30 +471,46 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	min_frame_size = params_width(params) * 2;
-	switch (min_frame_size) {
-	case 32:
+	if (pcm3168a->slot_width)
+		slot_width = pcm3168a->slot_width;
+	else
+		slot_width = params_width(params);
+
+	switch (slot_width) {
+	case 16:
 		if (master_mode || (fmt != PCM3168A_FMT_RIGHT_J)) {
-			dev_err(component->dev, "32-bit frames are supported only for slave mode using right justified\n");
+			dev_err(component->dev, "16-bit slots are supported only for slave mode using right justified\n");
 			return -EINVAL;
 		}
 		fmt = PCM3168A_FMT_RIGHT_J_16;
 		break;
-	case 48:
+	case 24:
 		if (master_mode || (fmt & PCM3168A_FMT_DSP_MASK)) {
-			dev_err(component->dev, "48-bit frames not supported in master mode, or slave mode using DSP\n");
+			dev_err(component->dev, "24-bit slots not supported in master mode, or slave mode using DSP\n");
 			return -EINVAL;
 		}
 		break;
-	case 64:
+	case 32:
 		break;
 	default:
-		dev_err(component->dev, "unsupported frame size: %d\n", min_frame_size);
+		dev_err(component->dev, "unsupported frame size: %d\n", slot_width);
 		return -EINVAL;
 	}
 
-	/* for TDM */
-	if (chan > 2) {
+	if (pcm3168a->tdm_slots)
+		tdm_slots = pcm3168a->tdm_slots;
+	else
+		tdm_slots = params_channels(params);
+
+	/*
+	 * Switch the codec to TDM mode when more than 2 TDM slots are needed
+	 * for the stream.
+	 * If pcm3168a->tdm_slots is not set or set to more than 2 (8/6 usually)
+	 * then DIN1/DOUT1 is used in TDM mode.
+	 * If pcm3168a->tdm_slots is set to 2 then DIN1/2/3/4 and DOUT1/2/3 is
+	 * used in normal mode, no need to switch to TDM modes.
+	 */
+	if (tdm_slots > 2) {
 		switch (fmt) {
 		case PCM3168A_FMT_I2S:
 		case PCM3168A_FMT_DSP_A:
@@ -551,14 +610,16 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
 	.set_fmt	= pcm3168a_set_dai_fmt_dac,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params,
-	.digital_mute	= pcm3168a_digital_mute
+	.digital_mute	= pcm3168a_digital_mute,
+	.set_tdm_slot	= pcm3168a_set_tdm_slot,
 };
 
 static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = {
 	.startup	= pcm3168a_startup,
 	.set_fmt	= pcm3168a_set_dai_fmt_adc,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
-	.hw_params	= pcm3168a_hw_params
+	.hw_params	= pcm3168a_hw_params,
+	.set_tdm_slot	= pcm3168a_set_tdm_slot,
 };
 
 static struct snd_soc_dai_driver pcm3168a_dais[] = {