Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became wide and scattered updates all over the sound tree as
  diffstat shows: lots of (still ongoing) refactoring works in ASoC,
  fixes and cleanups caught by static analysis, inclusive term
  conversions as well as lots of new drivers. Below are highlights:

  ASoC core:
   - API cleanups and conversions to the unified mute_stream() call
   - Simplify I/O helper functions
   - Use helper macros to retrieve RTD from substreams

  ASoC drivers:
   - Lots of fixes and cleanups in Intel ASoC drivers
   - Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
     Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
     nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
     boards, TI J721e EVM

  ALSA core:
   - Minor code refacotring for SG-buffer handling

  HD-audio:
   - Generalization of mute-LED handling with LED classdev
   - Intel silent stream support for HDMI
   - Device-specific fixes: CA0132, Loongson-3

  Others:
   - Usual USB- and HD-audio quirks for various devices
   - Fixes for echoaudio DMA position handling
   - Various documents and trivial fixes for sparse warnings
   - Conversion to adopt inclusive terms"

* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
  ALSA: pci: delete repeated words in comments
  ALSA: isa: delete repeated words in comments
  ALSA: hda/tegra: Add 100us dma stop delay
  ALSA: hda: Add dma stop delay variable
  ASoC: hda/tegra: Set buffer alignment to 128 bytes
  ALSA: seq: oss: Serialize ioctls
  ALSA: hda/hdmi: Add quirk to force connectivity
  ALSA: usb-audio: add startech usb audio dock name
  ALSA: usb-audio: Add support for Lenovo ThinkStation P620
  Revert "ALSA: hda: call runtime_allow() for all hda controllers"
  ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
  ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
  ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
  ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
  ALSA: docs: fix typo
  ALSA: doc: use correct config variable name
  ASoC: core: Two step component registration
  ASoC: core: Simplify snd_soc_component_initialize declaration
  ASoC: core: Relocate and expose snd_soc_component_initialize
  ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
  ...

1 files changed, 45 insertions, 20 deletions

diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index 69f3af4524ab..ff42f629b035 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -61,8 +61,13 @@ static int sst_fill_and_send_cmd_unlocked(struct sst_data *drv,
 
 /**
  * sst_fill_and_send_cmd - generate the IPC message and send it to the FW
- * @ipc_msg:	type of IPC (CMD, SET_PARAMS, GET_PARAMS)
- * @cmd_data:	the IPC payload
+ * @drv: sst_data
+ * @ipc_msg: type of IPC (CMD, SET_PARAMS, GET_PARAMS)
+ * @block: block index
+ * @task_id: task index
+ * @pipe_id: pipe index
+ * @cmd_data: the IPC payload
+ * @len: length of data to be sent
  */
 static int sst_fill_and_send_cmd(struct sst_data *drv,
 				 u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id,
@@ -78,7 +83,7 @@ static int sst_fill_and_send_cmd(struct sst_data *drv,
 	return ret;
 }
 
-/**
+/*
  * tx map value is a bitfield where each bit represents a FW channel
  *
  *			3 2 1 0		# 0 = codec0, 1 = codec1
@@ -90,7 +95,7 @@ static u8 sst_ssp_tx_map[SST_MAX_TDM_SLOTS] = {
 	0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default rx map */
 };
 
-/**
+/*
  * rx map value is a bitfield where each bit represents a slot
  *
  *			  76543210	# 0 = slot 0, 1 = slot 1
@@ -101,7 +106,7 @@ static u8 sst_ssp_rx_map[SST_MAX_TDM_SLOTS] = {
 	0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default tx map */
 };
 
-/**
+/*
  * NOTE: this is invoked with lock held
  */
 static int sst_send_slot_map(struct sst_data *drv)
@@ -145,7 +150,8 @@ static int sst_slot_enum_info(struct snd_kcontrol *kcontrol,
 
 /**
  * sst_slot_get - get the status of the interleaver/deinterleaver control
- *
+ * @kcontrol: control pointer
+ * @ucontrol: User data
  * Searches the map where the control status is stored, and gets the
  * channel/slot which is currently set for this enumerated control. Since it is
  * an enumerated control, there is only one possible value.
@@ -197,7 +203,8 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol
 
 /**
  * sst_slot_put - set the status of interleaver/deinterleaver control
- *
+ * @kcontrol: control pointer
+ * @ucontrol: User data
  * (de)interleaver controls are defined in opposite sense to be user-friendly
  *
  * Instead of the enum value being the value written to the register, it is the
@@ -280,7 +287,9 @@ static int sst_send_algo_cmd(struct sst_data *drv,
 
 /**
  * sst_find_and_send_pipe_algo - send all the algo parameters for a pipe
- *
+ * @drv: sst_data
+ * @pipe: string identifier
+ * @ids: list of algorithms
  * The algos which are in each pipeline are sent to the firmware one by one
  *
  * Called with lock held
@@ -379,11 +388,15 @@ static int sst_gain_ctl_info(struct snd_kcontrol *kcontrol,
 
 /**
  * sst_send_gain_cmd - send the gain algorithm IPC to the FW
- * @gv:		the stored value of gain (also contains rampduration)
- * @mute:	flag that indicates whether this was called from the
- *		digital_mute callback or directly. If called from the
- *		digital_mute callback, module will be muted/unmuted based on this
- *		flag. The flag is always 0 if called directly.
+ * @drv: sst_data
+ * @gv:the stored value of gain (also contains rampduration)
+ * @task_id: task index
+ * @loc_id: location/position index
+ * @module_id: module index
+ * @mute: flag that indicates whether this was called from the
+ *  digital_mute callback or directly. If called from the
+ *  digital_mute callback, module will be muted/unmuted based on this
+ *  flag. The flag is always 0 if called directly.
  *
  * Called with sst_data.lock held
  *
@@ -544,9 +557,12 @@ static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = {
 
 /**
  * fill_swm_input - fill in the SWM input ids given the register
+ * @cmpnt: ASoC component
+ * @swm_input: array of swm_input_ids
+ * @reg: the register value is a bit-field inicated which mixer inputs are ON.
  *
- * The register value is a bit-field inicated which mixer inputs are ON. Use the
- * lookup table to get the input-id and fill it in the structure.
+ * Use the lookup table to get the input-id and fill it in the
+ * structure.
  */
 static int fill_swm_input(struct snd_soc_component *cmpnt,
 		struct swm_input_ids *swm_input, unsigned int reg)
@@ -577,7 +593,7 @@ static int fill_swm_input(struct snd_soc_component *cmpnt,
 }
 
 
-/**
+/*
  * called with lock held
  */
 static int sst_set_pipe_gain(struct sst_ids *ids,
@@ -707,7 +723,7 @@ SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls);
 SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls);
 SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls);
 SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls);
-SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(__maybe_unused sst_mix_voip_controls);
 SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls);
 SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls);
 SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_modem_controls);
@@ -881,7 +897,7 @@ int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-/**
+/*
  * sst_ssp_config - contains SSP configuration for media UC
  * this can be overwritten by set_dai_xxx APIs
  */
@@ -1300,6 +1316,9 @@ static bool is_sst_dapm_widget(struct snd_soc_dapm_widget *w)
 
 /**
  * sst_send_pipe_gains - send gains for the front-end DAIs
+ * @dai: front-end dai
+ * @stream: direction
+ * @mute: boolean indicating mute status
  *
  * The gains in the pipes connected to the front-ends are muted/unmuted
  * automatically via the digital_mute() DAPM callback. This function sends the
@@ -1357,7 +1376,9 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute)
 
 /**
  * sst_fill_module_list - populate the list of modules/gains for a pipe
- *
+ * @kctl: kcontrol pointer
+ * @w: dapm widget
+ * @type: widget type
  *
  * Fills the widget pointer in the kcontrol private data, and also fills the
  * kcontrol pointer in the widget private data.
@@ -1403,7 +1424,8 @@ static int sst_fill_module_list(struct snd_kcontrol *kctl,
 
 /**
  * sst_fill_widget_module_info - fill list of gains/algos for the pipe
- * @widget:	pipe modelled as a DAPM widget
+ * @w: pipe modeled as a DAPM widget
+ * @component: ASoC component
  *
  * Fill the list of gains/algos for the widget by looking at all the card
  * controls and comparing the name of the widget with the first part of control
@@ -1463,6 +1485,8 @@ static int sst_fill_widget_module_info(struct snd_soc_dapm_widget *w,
 
 /**
  * sst_fill_linked_widgets - fill the parent pointer for the linked widget
+ * @component: ASoC component
+ * @ids: sst_ids array
  */
 static void sst_fill_linked_widgets(struct snd_soc_component *component,
 						struct sst_ids *ids)
@@ -1480,6 +1504,7 @@ static void sst_fill_linked_widgets(struct snd_soc_component *component,
 
 /**
  * sst_map_modules_to_pipe - fill algo/gains list for all pipes
+ * @component: ASoC component
  */
 static int sst_map_modules_to_pipe(struct snd_soc_component *component)
 {