Merge tag 'sound-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We had again a busy development cycle with many new drivers as well as
  lots of core improvements / cleanups. Let's go for highlights:

  ALSA core:

   - PCM locking scheme was refactored for reducing a global rwlock

   - PCM suspend is handled in the device type PM ops now; lots of
     explicit calls were reduced by this action

   - Cleanups about PCM buffer preallocation calls

   - Kill NULL device object in memory allocations

   - Lots of procfs API cleanups

  ASoC core:

   - Support for only powering up channels that are actively being used

   - Cleanups / fixes of topology API

  ASoC drivers:

   - MediaTek BTCVSD for a Bluetooth radio chip, which is the first such
     driver we've had upstream!

   - Quite a few improvements to simplify the generic card drivers,
     especially the merge of the SCU cards into the main generic drivers

   - Lots of fixes for probing on Intel systems to follow more standard
     styles

   - A big refresh and cleanup of the Samsung drivers

   - New drivers: Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341
     and CS35L26, Google ChromeOS embedded controllers, Ingenic JZ4725B,
     MediaTek BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328,
     Spreadtrum DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM
     formatters

  ALSA drivers:

   - Improvements of Tegra HD-audio controller driver for supporting new
     chips

   - HD-audio codec quirks for ALC294 S4 resume, ASUS laptop, Chrome
     headset button support and Dell workstations

   - Improved DSD support on USB-audio

   - Quirk for MOTU MicroBook II USB-audio

   - Support for Fireface UCX support and Solid State Logic Duende
     Classic/Mini"

* tag 'sound-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (461 commits)
  ALSA: usb-audio: Add quirk for MOTU MicroBook II
  ASoC: stm32: i2s: skip useless write in slave mode
  ASoC: stm32: i2s: fix race condition in irq handler
  ASoC: stm32: i2s: remove useless callback
  ASoC: stm32: i2s: fix dma configuration
  ASoC: stm32: i2s: fix stream count management
  ASoC: stm32: i2s: fix 16 bit format support
  ASoC: stm32: i2s: fix IRQ clearing
  ASoC: qcom: Kconfig: fix dependency for sdm845
  ASoC: Intel: Boards: Add Maxim98373 support
  ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address
  ALSA: firewire-motu: fix construction of PCM frame for capture direction
  ALSA: bebob: use more identical mod_alias for Saffire Pro 10 I/O against Liquid Saffire 56
  ALSA: hda: Extend i915 component bind timeout
  ASoC: wm_adsp: Improve logging messages
  ASoC: wm_adsp: Add support for multiple compressed buffers
  ASoC: wm_adsp: Refactor compress stream initialisation
  ASoC: wm_adsp: Reorder some functions for improved clarity
  ASoC: wm_adsp: Factor out stripping padding from ADSP data
  ASoC: cs35l36: Fix an IS_ERR() vs NULL checking bug
  ...

1 files changed, 118 insertions, 0 deletions

diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-hostless.c b/sound/soc/mediatek/mt8183/mt8183-dai-hostless.c
new file mode 100644
index 000000000000..1667ad352d34
--- /dev/null
+++ b/sound/soc/mediatek/mt8183/mt8183-dai-hostless.c
@@ -0,0 +1,118 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// MediaTek ALSA SoC Audio DAI Hostless Control
+//
+// Copyright (c) 2018 MediaTek Inc.
+// Author: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
+
+#include "mt8183-afe-common.h"
+
+/* dai component */
+static const struct snd_soc_dapm_route mtk_dai_hostless_routes[] = {
+	/* Hostless ADDA Loopback */
+	{"ADDA_DL_CH1", "ADDA_UL_CH1", "Hostless LPBK DL"},
+	{"ADDA_DL_CH1", "ADDA_UL_CH2", "Hostless LPBK DL"},
+	{"ADDA_DL_CH2", "ADDA_UL_CH1", "Hostless LPBK DL"},
+	{"ADDA_DL_CH2", "ADDA_UL_CH2", "Hostless LPBK DL"},
+	{"Hostless LPBK UL", NULL, "ADDA Capture"},
+
+	/* Hostless Speech */
+	{"ADDA_DL_CH1", "PCM_1_CAP_CH1", "Hostless Speech DL"},
+	{"ADDA_DL_CH2", "PCM_1_CAP_CH1", "Hostless Speech DL"},
+	{"ADDA_DL_CH2", "PCM_1_CAP_CH2", "Hostless Speech DL"},
+	{"ADDA_DL_CH1", "PCM_2_CAP_CH1", "Hostless Speech DL"},
+	{"ADDA_DL_CH2", "PCM_2_CAP_CH1", "Hostless Speech DL"},
+	{"ADDA_DL_CH2", "PCM_2_CAP_CH2", "Hostless Speech DL"},
+	{"PCM_1_PB_CH1", "ADDA_UL_CH1", "Hostless Speech DL"},
+	{"PCM_1_PB_CH2", "ADDA_UL_CH2", "Hostless Speech DL"},
+	{"PCM_2_PB_CH1", "ADDA_UL_CH1", "Hostless Speech DL"},
+	{"PCM_2_PB_CH2", "ADDA_UL_CH2", "Hostless Speech DL"},
+
+	{"Hostless Speech UL", NULL, "PCM 1 Capture"},
+	{"Hostless Speech UL", NULL, "PCM 2 Capture"},
+	{"Hostless Speech UL", NULL, "ADDA Capture"},
+};
+
+/* dai ops */
+static int mtk_dai_hostless_startup(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
+{
+	struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
+
+	return snd_soc_set_runtime_hwparams(substream, afe->mtk_afe_hardware);
+}
+
+static const struct snd_soc_dai_ops mtk_dai_hostless_ops = {
+	.startup = mtk_dai_hostless_startup,
+};
+
+/* dai driver */
+#define MTK_HOSTLESS_RATES (SNDRV_PCM_RATE_8000_48000 |\
+			   SNDRV_PCM_RATE_88200 |\
+			   SNDRV_PCM_RATE_96000 |\
+			   SNDRV_PCM_RATE_176400 |\
+			   SNDRV_PCM_RATE_192000)
+
+#define MTK_HOSTLESS_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			     SNDRV_PCM_FMTBIT_S24_LE |\
+			     SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = {
+	{
+		.name = "Hostless LPBK DAI",
+		.id = MT8183_DAI_HOSTLESS_LPBK,
+		.playback = {
+			.stream_name = "Hostless LPBK DL",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MTK_HOSTLESS_RATES,
+			.formats = MTK_HOSTLESS_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Hostless LPBK UL",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MTK_HOSTLESS_RATES,
+			.formats = MTK_HOSTLESS_FORMATS,
+		},
+		.ops = &mtk_dai_hostless_ops,
+	},
+	{
+		.name = "Hostless Speech DAI",
+		.id = MT8183_DAI_HOSTLESS_SPEECH,
+		.playback = {
+			.stream_name = "Hostless Speech DL",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MTK_HOSTLESS_RATES,
+			.formats = MTK_HOSTLESS_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Hostless Speech UL",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = MTK_HOSTLESS_RATES,
+			.formats = MTK_HOSTLESS_FORMATS,
+		},
+		.ops = &mtk_dai_hostless_ops,
+	},
+};
+
+int mt8183_dai_hostless_register(struct mtk_base_afe *afe)
+{
+	struct mtk_base_afe_dai *dai;
+
+	dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
+
+	list_add(&dai->list, &afe->sub_dais);
+
+	dai->dai_drivers = mtk_dai_hostless_driver;
+	dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver);
+
+	dai->dapm_routes = mtk_dai_hostless_routes;
+	dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes);
+
+	return 0;
+}