Merge tag 'sound-5.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Lots of code development have been see in ASoC side as usual, while
  the continued development on memalloc helper and USB-audio low-
  latency support are found in the rest.

  Note that a few changes in the unusual places like arch/sh are
  included, which are a part of ASoC DAI format cleanups.

  ALSA core:

   - Continued memalloc helper updates and cleanups, now supporting
     non-coherent and non-contiguous pages

   - Fixes for races in mixer OSS layer

  ASoC:

   - A new version of the audio graph card which supports a wider range
     of systems

   - Several conversions to YAML DT bindings

   - Continuing cleanups to the SOF and Intel code

   - Move of the Cirrus DSP framework into drivers/firmware to allow for
     future use by non-audio DSPs

   - An overhaul of the cs42l42 driver, correcting many problems

   - DAI format terminology conversions over many drivers for cleanups

   - Support for AMD Vangogh and Yelow Cap, Cirrus CS35L41, Maxim
     MAX98520 and MAX98360A, Mediatek MT8195, Nuvoton NAU8821, nVidia
     Tegra210, NXP i.MX8ULP, Qualcomm AudioReach, Realtek ALC5682I-VS,
     RT5682S, and RT9120 and Rockchip RV1126 and RK3568

  USB-audio:

   - Continued improvements on low-latency playback

   - Quirks for Pioneer devices, Line6 HX-Stomp XL, Audient iD14

  HD-audio:

   - Reduce excessive udelay() calls on Intel platforms; this should
     reduce the CPU load with PulseAudio

   - Quirks for HP and Clevo laptops

  FireWire:

   - Support for meter information on MOTU"

* tag 'sound-5.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (513 commits)
  ALSA: usb-audio: Add quirk for Audient iD14
  ALSA: hda/realtek: Add quirk for Clevo PC70HS
  ALSA: usb-audio: Line6 HX-Stomp XL USB_ID for 48k-fixed quirk
  ALSA: usb-audio: Add registration quirk for JBL Quantum 400
  ASoC: rsnd: Fix an error handling path in 'rsnd_node_count()'
  ASoC: tlv320aic3x: Make aic3x_remove() return void
  ASoC: Intel: soc-acpi: use const for all uses of snd_soc_acpi_codecs
  ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs
  ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs
  ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s
  ASoC: Intel: sof_rt5682: detect codec variant in probe function
  ASoC: soc-acpi: add comp_ids field for machine driver matching
  ASoC: mediatek: mt8195: add mt8195-mt6359-rt1011-rt5682 bindings document
  ASoC: mediatek: mt8195: add machine driver with mt6359, rt1011 and rt5682
  ASoC: Stop dummy from overriding hwparams
  ASoC: topology: Change topology device to card device
  ASoC: topology: Use correct device for prints
  ASoC: topology: Check for dapm widget completeness
  ASoC: topology: Add header payload_size verification
  ASoC: core: Remove invalid snd_soc_component_set_jack call
  ...

1 files changed, 260 insertions, 0 deletions

diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
new file mode 100644
index 000000000000..ce9e5646d8f3
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
@@ -0,0 +1,260 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2021, Linaro Limited
+
+#include <linux/err.h>
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include "q6dsp-lpass-ports.h"
+#include "audioreach.h"
+#include "q6apm.h"
+
+#define AUDIOREACH_BE_PCM_BASE	16
+
+struct q6apm_lpass_dai_data {
+	struct q6apm_graph *graph[APM_PORT_MAX];
+	bool is_port_started[APM_PORT_MAX];
+	struct audioreach_module_config module_config[APM_PORT_MAX];
+};
+
+static int q6dma_set_channel_map(struct snd_soc_dai *dai,
+				 unsigned int tx_num, unsigned int *tx_ch_mask,
+				 unsigned int rx_num, unsigned int *rx_ch_mask)
+{
+
+	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct audioreach_module_config *cfg = &dai_data->module_config[dai->id];
+	int ch_mask;
+
+	switch (dai->id) {
+	case WSA_CODEC_DMA_TX_0:
+	case WSA_CODEC_DMA_TX_1:
+	case WSA_CODEC_DMA_TX_2:
+	case VA_CODEC_DMA_TX_0:
+	case VA_CODEC_DMA_TX_1:
+	case VA_CODEC_DMA_TX_2:
+	case TX_CODEC_DMA_TX_0:
+	case TX_CODEC_DMA_TX_1:
+	case TX_CODEC_DMA_TX_2:
+	case TX_CODEC_DMA_TX_3:
+	case TX_CODEC_DMA_TX_4:
+	case TX_CODEC_DMA_TX_5:
+		if (!tx_ch_mask) {
+			dev_err(dai->dev, "tx slot not found\n");
+			return -EINVAL;
+		}
+
+		if (tx_num > AR_PCM_MAX_NUM_CHANNEL) {
+			dev_err(dai->dev, "invalid tx num %d\n",
+				tx_num);
+			return -EINVAL;
+		}
+		ch_mask = *tx_ch_mask;
+
+		break;
+	case WSA_CODEC_DMA_RX_0:
+	case WSA_CODEC_DMA_RX_1:
+	case RX_CODEC_DMA_RX_0:
+	case RX_CODEC_DMA_RX_1:
+	case RX_CODEC_DMA_RX_2:
+	case RX_CODEC_DMA_RX_3:
+	case RX_CODEC_DMA_RX_4:
+	case RX_CODEC_DMA_RX_5:
+	case RX_CODEC_DMA_RX_6:
+	case RX_CODEC_DMA_RX_7:
+		/* rx */
+		if (!rx_ch_mask) {
+			dev_err(dai->dev, "rx slot not found\n");
+			return -EINVAL;
+		}
+		if (rx_num > APM_PORT_MAX_AUDIO_CHAN_CNT) {
+			dev_err(dai->dev, "invalid rx num %d\n",
+				rx_num);
+			return -EINVAL;
+		}
+		ch_mask = *rx_ch_mask;
+
+		break;
+	default:
+		dev_err(dai->dev, "%s: invalid dai id 0x%x\n",
+			__func__, dai->id);
+		return -EINVAL;
+	}
+
+	cfg->active_channels_mask = ch_mask;
+
+	return 0;
+}
+
+static int q6dma_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct audioreach_module_config *cfg = &dai_data->module_config[dai->id];
+
+	cfg->bit_width = params_width(params);
+	cfg->sample_rate = params_rate(params);
+	cfg->num_channels = params_channels(params);
+
+	return 0;
+}
+
+static void q6apm_lpass_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+{
+	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	int rc;
+
+	if (!dai_data->is_port_started[dai->id])
+		return;
+	rc = q6apm_graph_stop(dai_data->graph[dai->id]);
+	if (rc < 0)
+		dev_err(dai->dev, "fail to close APM port (%d)\n", rc);
+
+	q6apm_graph_close(dai_data->graph[dai->id]);
+	dai_data->is_port_started[dai->id] = false;
+}
+
+static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+{
+	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct audioreach_module_config *cfg = &dai_data->module_config[dai->id];
+	struct q6apm_graph *graph;
+	int graph_id = dai->id;
+	int rc;
+
+	/**
+	 * It is recommend to load DSP with source graph first and then sink
+	 * graph, so sequence for playback and capture will be different
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		graph = q6apm_graph_open(dai->dev, NULL, dai->dev, graph_id);
+		if (IS_ERR(graph)) {
+			dev_err(dai->dev, "Failed to open graph (%d)\n", graph_id);
+			rc = PTR_ERR(graph);
+			return rc;
+		}
+		dai_data->graph[graph_id] = graph;
+	}
+
+	cfg->direction = substream->stream;
+	rc = q6apm_graph_media_format_pcm(dai_data->graph[dai->id], cfg);
+
+	if (rc) {
+		dev_err(dai->dev, "Failed to set media format %d\n", rc);
+		return rc;
+	}
+
+	rc = q6apm_graph_prepare(dai_data->graph[dai->id]);
+	if (rc) {
+		dev_err(dai->dev, "Failed to prepare Graph %d\n", rc);
+		return rc;
+	}
+
+	rc = q6apm_graph_start(dai_data->graph[dai->id]);
+	if (rc < 0) {
+		dev_err(dai->dev, "fail to start APM port %x\n", dai->id);
+		return rc;
+	}
+	dai_data->is_port_started[dai->id] = true;
+
+	return 0;
+}
+
+static int q6apm_lpass_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
+{
+	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct q6apm_graph *graph;
+	int graph_id = dai->id;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		graph = q6apm_graph_open(dai->dev, NULL, dai->dev, graph_id);
+		if (IS_ERR(graph)) {
+			dev_err(dai->dev, "Failed to open graph (%d)\n", graph_id);
+			return PTR_ERR(graph);
+		}
+		dai_data->graph[graph_id] = graph;
+	}
+
+	return 0;
+}
+
+static int q6i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	struct audioreach_module_config *cfg = &dai_data->module_config[dai->id];
+
+	cfg->fmt = fmt;
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops q6dma_ops = {
+	.prepare	= q6apm_lpass_dai_prepare,
+	.startup	= q6apm_lpass_dai_startup,
+	.shutdown	= q6apm_lpass_dai_shutdown,
+	.set_channel_map  = q6dma_set_channel_map,
+	.hw_params        = q6dma_hw_params,
+};
+
+static const struct snd_soc_dai_ops q6i2s_ops = {
+	.prepare	= q6apm_lpass_dai_prepare,
+	.startup	= q6apm_lpass_dai_startup,
+	.shutdown	= q6apm_lpass_dai_shutdown,
+	.set_channel_map  = q6dma_set_channel_map,
+	.hw_params        = q6dma_hw_params,
+	.set_fmt	= q6i2s_set_fmt,
+};
+
+static const struct snd_soc_component_driver q6apm_lpass_dai_component = {
+	.name = "q6apm-be-dai-component",
+	.of_xlate_dai_name = q6dsp_audio_ports_of_xlate_dai_name,
+	.be_pcm_base = AUDIOREACH_BE_PCM_BASE,
+	.use_dai_pcm_id = true,
+};
+
+static int q6apm_lpass_dai_dev_probe(struct platform_device *pdev)
+{
+	struct q6dsp_audio_port_dai_driver_config cfg;
+	struct q6apm_lpass_dai_data *dai_data;
+	struct snd_soc_dai_driver *dais;
+	struct device *dev = &pdev->dev;
+	int num_dais;
+
+	dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL);
+	if (!dai_data)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, dai_data);
+
+	memset(&cfg, 0, sizeof(cfg));
+	cfg.q6i2s_ops = &q6i2s_ops;
+	cfg.q6dma_ops = &q6dma_ops;
+	dais = q6dsp_audio_ports_set_config(dev, &cfg, &num_dais);
+
+	return devm_snd_soc_register_component(dev, &q6apm_lpass_dai_component, dais, num_dais);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id q6apm_lpass_dai_device_id[] = {
+	{ .compatible = "qcom,q6apm-lpass-dais" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, q6apm_lpass_dai_device_id);
+#endif
+
+static struct platform_driver q6apm_lpass_dai_platform_driver = {
+	.driver = {
+		.name = "q6apm-lpass-dais",
+		.of_match_table = of_match_ptr(q6apm_lpass_dai_device_id),
+	},
+	.probe = q6apm_lpass_dai_dev_probe,
+};
+module_platform_driver(q6apm_lpass_dai_platform_driver);
+
+MODULE_DESCRIPTION("AUDIOREACH APM LPASS dai driver");
+MODULE_LICENSE("GPL");