Merge tag 'sound-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "As shown in diffstat and logs, it was again a busy development cycle
  at this time, too. The most significant changes are still on-going
  refactoring / modernization works for ASoC core and drivers, but there
  are lots of other changes as well. Here we go, some highlights below:

  ASoC:

   - Quite a lot of cleanup / refactoring of ASoC core and APIs; most of
     them are systematic, but also including cleanups and modernization

   - A bulk of updates for some ASoC platforms, Freescale, sunxi and
     Intel SST/SOF

   - Initial support for Sound Open Firmware on i.MX8

   - Removal of deprecated w90x900 and nuc900 drivers

   - New support for Cirrus Logic CS47L15 and CS47L92, Freescale i.MX
     7ULP and 8MQ, Meson G12A and NXP UDA1334

  USB-audio:

   - More validations of descriptor units for hardening against bugs
     reported by fuzzers

   - PCM device assignment workaround for a past call-order change

   - Scarlett Gen2 mixer interface, a few more more quirks

  HD-audio:

   - Support for audio component with AMD/ATI and Nvidia HDMI codecs

   - Clean up HD-audio core and remove indirect access ops for Intel SOF

   - DMIC detection at probe; it would make systems automatically
     falling back to SST/SOF driver on devices that need DMIC handling.
     Needs a new Kconfig to set, and beware that it's still new and a
     bit experimental

  FireWire:

   - Lots of code refactoring and cleanups"

* tag 'sound-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (521 commits)
  ASoC: sdm845: remove unneeded semicolon
  ASoC: fsl_sai: Implement set_bclk_ratio
  ASoC: dmaengine: Replace strncpy() with strscpy_pad() for pcm->name
  ASoC: wcd9335: remove redundant use of ret variable
  ALSA: firewire-tascam: check intermediate state of clock status and retry
  ALSA: firewire-tascam: handle error code when getting current source of clock
  ASoC: hdmi-codec: Add an op to set callback function for plug event
  ASoC: rt5677: keep analog power register at SND_SOC_BIAS_OFF
  ASoC: rt5677: Remove magic number register writes
  ASoC: soc-core: self contained soc_unbind_aux_dev()
  ASoC: soc-core: add soc_unbind_aux_dev()
  ASoC: soc-core: self contained soc_bind_aux_dev()
  ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais()
  ASoC: soc-core: self contained soc_probe_link_dais()
  ASoC: soc-core: add new soc_link_init()
  ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai()
  ASoC: soc-core: self contained soc_remove_link_dais()
  ASoC: soc-core: self contained soc_remove_link_components()
  ASoC: soc-core: self contained soc_probe_link_components()
  ASoC: rt1308: make array pd static const, makes object smaller
  ...

1 files changed, 407 insertions, 0 deletions

diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
new file mode 100644
index 000000000000..1c7f63871c1d
--- /dev/null
+++ b/sound/soc/soc-dai.c
@@ -0,0 +1,407 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// soc-dai.c
+//
+// Copyright (C) 2019 Renesas Electronics Corp.
+// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+//
+
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+/**
+ * snd_soc_dai_set_sysclk - configure DAI system or master clock.
+ * @dai: DAI
+ * @clk_id: DAI specific clock ID
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the DAI master (MCLK) or system (SYSCLK) clocking.
+ */
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			   unsigned int freq, int dir)
+{
+	if (dai->driver->ops->set_sysclk)
+		return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);
+
+	return snd_soc_component_set_sysclk(dai->component, clk_id, 0,
+					    freq, dir);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
+
+/**
+ * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
+ * @dai: DAI
+ * @div_id: DAI specific clock divider ID
+ * @div: new clock divisor.
+ *
+ * Configures the clock dividers. This is used to derive the best DAI bit and
+ * frame clocks from the system or master clock. It's best to set the DAI bit
+ * and frame clocks as low as possible to save system power.
+ */
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+			   int div_id, int div)
+{
+	if (dai->driver->ops->set_clkdiv)
+		return dai->driver->ops->set_clkdiv(dai, div_id, div);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
+
+/**
+ * snd_soc_dai_set_pll - configure DAI PLL.
+ * @dai: DAI
+ * @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+			unsigned int freq_in, unsigned int freq_out)
+{
+	if (dai->driver->ops->set_pll)
+		return dai->driver->ops->set_pll(dai, pll_id, source,
+						 freq_in, freq_out);
+
+	return snd_soc_component_set_pll(dai->component, pll_id, source,
+					 freq_in, freq_out);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
+
+/**
+ * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio.
+ * @dai: DAI
+ * @ratio: Ratio of BCLK to Sample rate.
+ *
+ * Configures the DAI for a preset BCLK to sample rate ratio.
+ */
+int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	if (dai->driver->ops->set_bclk_ratio)
+		return dai->driver->ops->set_bclk_ratio(dai, ratio);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
+
+/**
+ * snd_soc_dai_set_fmt - configure DAI hardware audio format.
+ * @dai: DAI
+ * @fmt: SND_SOC_DAIFMT_* format value.
+ *
+ * Configures the DAI hardware format and clocking.
+ */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	if (dai->driver->ops->set_fmt == NULL)
+		return -ENOTSUPP;
+	return dai->driver->ops->set_fmt(dai, fmt);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
+
+/**
+ * snd_soc_xlate_tdm_slot - generate tx/rx slot mask.
+ * @slots: Number of slots in use.
+ * @tx_mask: bitmask representing active TX slots.
+ * @rx_mask: bitmask representing active RX slots.
+ *
+ * Generates the TDM tx and rx slot default masks for DAI.
+ */
+static int snd_soc_xlate_tdm_slot_mask(unsigned int slots,
+				       unsigned int *tx_mask,
+				       unsigned int *rx_mask)
+{
+	if (*tx_mask || *rx_mask)
+		return 0;
+
+	if (!slots)
+		return -EINVAL;
+
+	*tx_mask = (1 << slots) - 1;
+	*rx_mask = (1 << slots) - 1;
+
+	return 0;
+}
+
+/**
+ * snd_soc_dai_set_tdm_slot() - Configures a DAI for TDM operation
+ * @dai: The DAI to configure
+ * @tx_mask: bitmask representing active TX slots.
+ * @rx_mask: bitmask representing active RX slots.
+ * @slots: Number of slots in use.
+ * @slot_width: Width in bits for each slot.
+ *
+ * This function configures the specified DAI for TDM operation. @slot contains
+ * the total number of slots of the TDM stream and @slot_with the width of each
+ * slot in bit clock cycles. @tx_mask and @rx_mask are bitmasks specifying the
+ * active slots of the TDM stream for the specified DAI, i.e. which slots the
+ * DAI should write to or read from. If a bit is set the corresponding slot is
+ * active, if a bit is cleared the corresponding slot is inactive. Bit 0 maps to
+ * the first slot, bit 1 to the second slot and so on. The first active slot
+ * maps to the first channel of the DAI, the second active slot to the second
+ * channel and so on.
+ *
+ * TDM mode can be disabled by passing 0 for @slots. In this case @tx_mask,
+ * @rx_mask and @slot_width will be ignored.
+ *
+ * Returns 0 on success, a negative error code otherwise.
+ */
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+			     unsigned int tx_mask, unsigned int rx_mask,
+			     int slots, int slot_width)
+{
+	if (dai->driver->ops->xlate_tdm_slot_mask)
+		dai->driver->ops->xlate_tdm_slot_mask(slots,
+						      &tx_mask, &rx_mask);
+	else
+		snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
+
+	dai->tx_mask = tx_mask;
+	dai->rx_mask = rx_mask;
+
+	if (dai->driver->ops->set_tdm_slot)
+		return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask,
+						      slots, slot_width);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
+
+/**
+ * snd_soc_dai_set_channel_map - configure DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ *           0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ *           0~num-1 uses
+ *
+ * configure the relationship between channel number and TDM slot number.
+ */
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+				unsigned int tx_num, unsigned int *tx_slot,
+				unsigned int rx_num, unsigned int *rx_slot)
+{
+	if (dai->driver->ops->set_channel_map)
+		return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot,
+							 rx_num, rx_slot);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
+
+/**
+ * snd_soc_dai_get_channel_map - Get DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ *           0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ *           0~num-1 uses
+ */
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+				unsigned int *tx_num, unsigned int *tx_slot,
+				unsigned int *rx_num, unsigned int *rx_slot)
+{
+	if (dai->driver->ops->get_channel_map)
+		return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot,
+							 rx_num, rx_slot);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map);
+
+/**
+ * snd_soc_dai_set_tristate - configure DAI system or master clock.
+ * @dai: DAI
+ * @tristate: tristate enable
+ *
+ * Tristates the DAI so that others can use it.
+ */
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+	if (dai->driver->ops->set_tristate)
+		return dai->driver->ops->set_tristate(dai, tristate);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
+
+/**
+ * snd_soc_dai_digital_mute - configure DAI system or master clock.
+ * @dai: DAI
+ * @mute: mute enable
+ * @direction: stream to mute
+ *
+ * Mutes the DAI DAC.
+ */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
+			     int direction)
+{
+	if (dai->driver->ops->mute_stream)
+		return dai->driver->ops->mute_stream(dai, mute, direction);
+	else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
+		 dai->driver->ops->digital_mute)
+		return dai->driver->ops->digital_mute(dai, mute);
+	else
+		return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
+
+int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
+			  struct snd_pcm_substream *substream,
+			  struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int ret;
+
+	/* perform any topology hw_params fixups before DAI  */
+	if (rtd->dai_link->be_hw_params_fixup) {
+		ret = rtd->dai_link->be_hw_params_fixup(rtd, params);
+		if (ret < 0) {
+			dev_err(rtd->dev,
+				"ASoC: hw_params topology fixup failed %d\n",
+				ret);
+			return ret;
+		}
+	}
+
+	if (dai->driver->ops->hw_params) {
+		ret = dai->driver->ops->hw_params(substream, params, dai);
+		if (ret < 0) {
+			dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n",
+				dai->name, ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
+			 struct snd_pcm_substream *substream)
+{
+	if (dai->driver->ops->hw_free)
+		dai->driver->ops->hw_free(substream, dai);
+}
+
+int snd_soc_dai_startup(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	if (dai->driver->ops->startup)
+		ret = dai->driver->ops->startup(substream, dai);
+
+	return ret;
+}
+
+void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
+			 struct snd_pcm_substream *substream)
+{
+	if (dai->driver->ops->shutdown)
+		dai->driver->ops->shutdown(substream, dai);
+}
+
+int snd_soc_dai_prepare(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	if (dai->driver->ops->prepare)
+		ret = dai->driver->ops->prepare(substream, dai);
+
+	return ret;
+}
+
+int snd_soc_dai_trigger(struct snd_soc_dai *dai,
+			struct snd_pcm_substream *substream,
+			int cmd)
+{
+	int ret = 0;
+
+	if (dai->driver->ops->trigger)
+		ret = dai->driver->ops->trigger(substream, cmd, dai);
+
+	return ret;
+}
+
+int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
+				struct snd_pcm_substream *substream,
+				int cmd)
+{
+	int ret = 0;
+
+	if (dai->driver->ops->bespoke_trigger)
+		ret = dai->driver->ops->bespoke_trigger(substream, cmd, dai);
+
+	return ret;
+}
+
+snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
+				    struct snd_pcm_substream *substream)
+{
+	int delay = 0;
+
+	if (dai->driver->ops->delay)
+		delay = dai->driver->ops->delay(substream, dai);
+
+	return delay;
+}
+
+void snd_soc_dai_suspend(struct snd_soc_dai *dai)
+{
+	if (dai->driver->suspend)
+		dai->driver->suspend(dai);
+}
+
+void snd_soc_dai_resume(struct snd_soc_dai *dai)
+{
+	if (dai->driver->resume)
+		dai->driver->resume(dai);
+}
+
+int snd_soc_dai_probe(struct snd_soc_dai *dai)
+{
+	if (dai->driver->probe)
+		return dai->driver->probe(dai);
+	return 0;
+}
+
+int snd_soc_dai_remove(struct snd_soc_dai *dai)
+{
+	if (dai->driver->remove)
+		return dai->driver->remove(dai);
+	return 0;
+}
+
+int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
+			     struct snd_soc_pcm_runtime *rtd, int num)
+{
+	if (dai->driver->compress_new)
+		return dai->driver->compress_new(rtd, num);
+	return -ENOTSUPP;
+}
+
+/*
+ * snd_soc_dai_stream_valid() - check if a DAI supports the given stream
+ *
+ * Returns true if the DAI supports the indicated stream type.
+ */
+bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir)
+{
+	struct snd_soc_pcm_stream *stream;
+
+	if (dir == SNDRV_PCM_STREAM_PLAYBACK)
+		stream = &dai->driver->playback;
+	else
+		stream = &dai->driver->capture;
+
+	/* If the codec specifies any channels at all, it supports the stream */
+	return stream->channels_min;
+}