Merge tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became again a busy development cycle.  There are few ALSA core
  updates (merely API cleanups and sparse fixes), with the majority of
  other changes are found in ASoC scene.

  Here are some highlights:

  ALSA core:
   - More helper macros for sparse warning fixes (e.g. bitwise types)
   - Slight optimization of PCM OSS locks
   - Make common handling for PCM / compress buffers (for SOF)

  ASoC:
   - Lots of code refactoring and modernization for (still ongoing)
     componentization works
   - Conversion of SND_SOC_ALL_CODECS to use imply
   - Continued refactoring and fixing of the Intel SOF/SST support,
     including the initial (but still incomplete) SoundWire support
   - SoundWire and more advanced clocking support for Realtek RT5682
   - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
     DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563
     and TLV320ADCX140

  HD-audio:
   - Optimizations in HDMI jack handling
   - A few new quirks and fixups for Realtek codecs

  USB-audio:
   - Delayed registration support
   - New quirks for Motu, Kingston, Presonus"

* tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits)
  ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
  Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
  ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups
  ALSA: hda/realtek - Set principled PC Beep configuration for ALC256
  ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256
  ALSA: hda/realtek - a fake key event is triggered by running shutup
  ALSA: hda: default enable CA0132 DSP support
  ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
  ASoC: tlv320adcx140: Remove undocumented property
  ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function
  ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
  ASoC: Intel: boards: add sof_sdw machine driver
  ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms
  ASoC: rt5682: move DAI clock registry to I2S mode
  ASoC: pxa: magician: convert to use i2c_new_client_device()
  ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities
  Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread
  ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
  ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks
  ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers
  ...

16 files changed, 97 insertions, 23 deletions

diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index 29f61053ab62..c5408c129f34 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -1,6 +1,6 @@
 # SPDX-License-Identifier: GPL-2.0-only
 menu "Audio support for Texas Instruments SoCs"
-depends on DMA_OMAP || TI_EDMA || COMPILE_TEST
+depends on DMA_OMAP || TI_EDMA || TI_K3_UDMA || COMPILE_TEST
 
 config SND_SOC_TI_EDMA_PCM
 	tristate
@@ -10,6 +10,10 @@ config SND_SOC_TI_SDMA_PCM
 	tristate
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 
+config SND_SOC_TI_UDMA_PCM
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
 comment "Texas Instruments DAI support for:"
 config SND_SOC_DAVINCI_ASP
 	tristate "daVinci Audio Serial Port (ASP) or McBSP support"
@@ -24,6 +28,7 @@ config SND_SOC_DAVINCI_MCASP
 	tristate "Multichannel Audio Serial Port (McASP) support"
 	select SND_SOC_TI_EDMA_PCM
 	select SND_SOC_TI_SDMA_PCM
+	select SND_SOC_TI_UDMA_PCM
 	help
 	  Say Y or M here if you want to have support for McASP IP found in
 	  various Texas Instruments SoCs like:
@@ -31,6 +36,7 @@ config SND_SOC_DAVINCI_MCASP
 	  - Sitara line of SoCs (AM335x, AM438x, etc)
 	  - DRA7x devices
 	  - Keystone devices
+	  - K3 devices (am654, j721e)
 
 config SND_SOC_DAVINCI_VCIF
 	tristate "daVinci Voice Interface (VCIF) support"
diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile
index 08c44d56ef3e..ea48c6679cc7 100644
--- a/sound/soc/ti/Makefile
+++ b/sound/soc/ti/Makefile
@@ -3,9 +3,11 @@
 # Platform drivers
 snd-soc-ti-edma-objs := edma-pcm.o
 snd-soc-ti-sdma-objs := sdma-pcm.o
+snd-soc-ti-udma-objs := udma-pcm.o
 
 obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o
 obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o
+obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o
 
 # CPU DAI drivers
 snd-soc-davinci-asp-objs := davinci-i2s.o
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index 8e2fb81ad05c..e17cd5e939f0 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -460,14 +460,14 @@ static void ams_delta_shutdown(struct snd_pcm_substream *substream)
 
 static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct snd_soc_dapm_context *dapm = &card->dapm;
 	int ret;
 	/* Codec is ready, now add/activate board specific controls */
 
 	/* Store a pointer to the codec structure for tty ldisc use */
-	cx20442_codec = rtd->codec_dai->component;
+	cx20442_codec = asoc_rtd_to_codec(rtd, 0)->component;
 
 	/* Add hook switch - can be used to control the codec from userspace
 	 * even if line discipline fails */
diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c
index 686b23d7a90d..2cfbeebdfb41 100644
--- a/sound/soc/ti/davinci-evm.c
+++ b/sound/soc/ti/davinci-evm.c
@@ -54,8 +54,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct snd_soc_card *soc_card = rtd->card;
 	int ret = 0;
 	unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *)
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index e1e937eb1dc1..734ffe925c4d 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -38,6 +38,7 @@
 
 #include "edma-pcm.h"
 #include "sdma-pcm.h"
+#include "udma-pcm.h"
 #include "davinci-mcasp.h"
 
 #define MCASP_MAX_AFIFO_DEPTH	64
@@ -1764,10 +1765,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of(
 	} else if (match) {
 		pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata),
 				     GFP_KERNEL);
-		if (!pdata) {
-			ret = -ENOMEM;
-			return pdata;
-		}
+		if (!pdata)
+			return NULL;
 	} else {
 		/* control shouldn't reach here. something is wrong */
 		ret = -EINVAL;
@@ -1875,6 +1874,7 @@ nodata:
 enum {
 	PCM_EDMA,
 	PCM_SDMA,
+	PCM_UDMA,
 };
 static const char *sdma_prefix = "ti,omap";
 
@@ -1912,6 +1912,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
 	dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp);
 	if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix)))
 		return PCM_SDMA;
+	else if (strstr(tmp, "udmap"))
+		return PCM_UDMA;
 
 	return PCM_EDMA;
 }
@@ -2371,6 +2373,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	case PCM_SDMA:
 		ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
 		break;
+	case PCM_UDMA:
+		ret = udma_pcm_platform_register(&pdev->dev);
+		break;
 	default:
 		dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret);
 	case -EPROBE_DEFER:
diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c
index c84650e4a7aa..ee4d3ef821a1 100644
--- a/sound/soc/ti/davinci-vcif.c
+++ b/sound/soc/ti/davinci-vcif.c
@@ -43,7 +43,7 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_vcif_dev *davinci_vcif_dev =
-			snd_soc_dai_get_drvdata(rtd->cpu_dai);
+			snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
 	u32 w;
 
@@ -62,7 +62,7 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_vcif_dev *davinci_vcif_dev =
-			snd_soc_dai_get_drvdata(rtd->cpu_dai);
+			snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
 	u32 w;
 
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 3ad2b6daf31e..a1672b479cb7 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -101,7 +101,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int err;
 
 	/* Set the codec system clock for DAC and ADC */
diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c
index 6d564ab5e437..61e45fea5dd8 100644
--- a/sound/soc/ti/omap-abe-twl6040.c
+++ b/sound/soc/ti/omap-abe-twl6040.c
@@ -46,7 +46,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
 	int clk_id, freq;
@@ -78,7 +78,7 @@ static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret = 0;
 
 	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
@@ -166,7 +166,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_card *card = rtd->card;
 	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
 	int hs_trim;
diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
index 1a3fe854e856..5a32b54bbf3b 100644
--- a/sound/soc/ti/omap-mcbsp-st.c
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -489,7 +489,7 @@ OMAP_MCBSP_ST_CONTROLS(3);
 
 int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
 
 	if (!mcbsp->st_data) {
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 302d5c493c29..3d41ca2238d4 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -737,7 +737,7 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream,
 		unsigned int packet_size)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
 	int words;
 
@@ -902,7 +902,7 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay(
 			struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
 	u16 fifo_use;
 	snd_pcm_sframes_t delay;
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
index d7ac4df6f2d9..f2dbadea33bb 100644
--- a/sound/soc/ti/omap-mcpdm.c
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -532,7 +532,7 @@ static const struct snd_soc_component_driver omap_mcpdm_component = {
 void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
 				    u8 rx1, u8 rx2)
 {
-	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2);
 }
diff --git a/sound/soc/ti/omap3pandora.c b/sound/soc/ti/omap3pandora.c
index 545f8dac9bd5..b04146311b31 100644
--- a/sound/soc/ti/omap3pandora.c
+++ b/sound/soc/ti/omap3pandora.c
@@ -32,8 +32,8 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret;
 
 	/* Set the codec system clock for DAC and ADC */
diff --git a/sound/soc/ti/osk5912.c b/sound/soc/ti/osk5912.c
index 1ca466bc4025..e01485cc51a1 100644
--- a/sound/soc/ti/osk5912.c
+++ b/sound/soc/ti/osk5912.c
@@ -39,7 +39,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int err;
 
 	/* Set the codec system clock for DAC and ADC */
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index fdb0dc85fe67..2a714a004163 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -103,7 +103,7 @@ static int rx51_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* Set the codec system clock for DAC and ADC */
 	return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c
new file mode 100644
index 000000000000..39830caaaf7c
--- /dev/null
+++ b/sound/soc/ti/udma-pcm.c
@@ -0,0 +1,43 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ *  Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com
+ *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "udma-pcm.h"
+
+static const struct snd_pcm_hardware udma_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
+				  SNDRV_PCM_INFO_INTERLEAVED,
+	.buffer_bytes_max	= SIZE_MAX,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= SZ_64K,
+	.periods_min		= 2,
+	.periods_max		= UINT_MAX,
+};
+
+static const struct snd_dmaengine_pcm_config udma_dmaengine_pcm_config = {
+	.pcm_hardware = &udma_pcm_hardware,
+	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+};
+
+int udma_pcm_platform_register(struct device *dev)
+{
+	return devm_snd_dmaengine_pcm_register(dev, &udma_dmaengine_pcm_config,
+					       0);
+}
+EXPORT_SYMBOL_GPL(udma_pcm_platform_register);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("UDMA PCM ASoC platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h
new file mode 100644
index 000000000000..54111e7312c1
--- /dev/null
+++ b/sound/soc/ti/udma-pcm.h
@@ -0,0 +1,18 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ *  Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ */
+
+#ifndef __UDMA_PCM_H__
+#define __UDMA_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_SOC_TI_UDMA_PCM)
+int udma_pcm_platform_register(struct device *dev);
+#else
+static inline int udma_pcm_platform_register(struct device *dev)
+{
+	return 0;
+}
+#endif /* CONFIG_SND_SOC_TI_UDMA_PCM */
+
+#endif /* __UDMA_PCM_H__ */