Message-Id: <E1fKn1x-00077V-7q@debutante>
Date:   Mon, 21 May 2018 16:47:25 +0100
From:   Mark Brown <broonie@...nel.org>
To:     Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Cc:     Mark Brown <broonie@...nel.org>, broonie@...nel.org,
        linux-arm-msm@...r.kernel.org, alsa-devel@...a-project.org,
        bgoswami@...eaurora.org, mark.rutland@....com,
        devicetree@...r.kernel.org, rohkumar@....qualcomm.com,
        gregkh@...uxfoundation.org, plai@...eaurora.org,
        lgirdwood@...il.com, tiwai@...e.com, david.brown@...aro.org,
        robh+dt@...nel.org, linux-arm-kernel@...ts.infradead.org,
        spatakok@....qualcomm.com, linux-kernel@...r.kernel.org,
        alsa-devel@...a-project.org
Subject: Applied "ASoC: qdsp6: q6routing: Add q6routing driver" to the asoc tree

The patch

   ASoC: qdsp6: q6routing: Add q6routing driver

has been applied to the asoc tree at

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From e3a33673e845f609977970d147d5f8e094c52c73 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Date: Fri, 18 May 2018 13:56:04 +0100
Subject: [PATCH] ASoC: qdsp6: q6routing: Add q6routing driver

This patch adds support to q6 routing driver which configures route
between ASM and AFE module using ADM apis.

This driver uses dapm widgets to setup the matrix between AFE ports and
ASM streams.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@...eaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@...eaurora.org>
Signed-off-by: Mark Brown <broonie@...nel.org>
---
 sound/soc/qcom/Kconfig           |   4 +
 sound/soc/qcom/qdsp6/Makefile    |   1 +
 sound/soc/qcom/qdsp6/q6routing.c | 400 +++++++++++++++++++++++++++++++
 sound/soc/qcom/qdsp6/q6routing.h |   9 +
 4 files changed, 414 insertions(+)
 create mode 100644 sound/soc/qcom/qdsp6/q6routing.c
 create mode 100644 sound/soc/qcom/qdsp6/q6routing.h

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 941774abd94f..43f9ed85efa8 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -53,6 +53,9 @@ config SND_SOC_QDSP6_AFE
 config SND_SOC_QDSP6_ADM
 	tristate
 
+config SND_SOC_QDSP6_ROUTING
+	tristate
+
 config SND_SOC_QDSP6_ASM
 	tristate
 
@@ -63,6 +66,7 @@ config SND_SOC_QDSP6
 	select SND_SOC_QDSP6_CORE
 	select SND_SOC_QDSP6_AFE
 	select SND_SOC_QDSP6_ADM
+	select SND_SOC_QDSP6_ROUTING
 	select SND_SOC_QDSP6_ASM
 	help
 	 To add support for MSM QDSP6 Soc Audio.
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 01d9dcf3375c..0e8e2febb7ec 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -2,4 +2,5 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
 obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
 obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
 obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
+obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
 obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
new file mode 100644
index 000000000000..1d7a4088a435
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -0,0 +1,400 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_platform.h>
+#include <linux/bitops.h>
+#include <linux/component.h>
+#include <linux/mutex.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/asound.h>
+#include <sound/pcm_params.h>
+#include "q6afe.h"
+#include "q6asm.h"
+#include "q6adm.h"
+#include "q6routing.h"
+
+#define DRV_NAME "q6routing-component"
+
+struct session_data {
+	int state;
+	int port_id;
+	int path_type;
+	int app_type;
+	int acdb_id;
+	int sample_rate;
+	int bits_per_sample;
+	int channels;
+	int perf_mode;
+	int numcopps;
+	int fedai_id;
+	unsigned long copp_map;
+	struct q6copp *copps[MAX_COPPS_PER_PORT];
+};
+
+struct msm_routing_data {
+	struct session_data sessions[MAX_SESSIONS];
+	struct session_data port_data[AFE_MAX_PORTS];
+	struct device *dev;
+	struct mutex lock;
+};
+
+static struct msm_routing_data *routing_data;
+
+/**
+ * q6routing_stream_open() - Register a new stream for route setup
+ *
+ * @fedai_id: Frontend dai id.
+ * @perf_mode: Performance mode.
+ * @stream_id: ASM stream id to map.
+ * @stream_type: Direction of stream
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6routing_stream_open(int fedai_id, int perf_mode,
+			   int stream_id, int stream_type)
+{
+	int j, topology, num_copps = 0;
+	struct route_payload payload;
+	struct q6copp *copp;
+	int copp_idx;
+	struct session_data *session, *pdata;
+
+	if (!routing_data) {
+		pr_err("Routing driver not yet ready\n");
+		return -EINVAL;
+	}
+
+	session = &routing_data->sessions[stream_id - 1];
+	pdata = &routing_data->port_data[session->port_id];
+
+	mutex_lock(&routing_data->lock);
+	session->fedai_id = fedai_id;
+
+	session->path_type = pdata->path_type;
+	session->sample_rate = pdata->sample_rate;
+	session->channels = pdata->channels;
+	session->bits_per_sample = pdata->bits_per_sample;
+
+	payload.num_copps = 0; /* only RX needs to use payload */
+	topology = NULL_COPP_TOPOLOGY;
+	copp = q6adm_open(routing_data->dev, session->port_id,
+			      session->path_type, session->sample_rate,
+			      session->channels, topology, perf_mode,
+			      session->bits_per_sample, 0, 0);
+
+	if (!copp) {
+		mutex_unlock(&routing_data->lock);
+		return -EINVAL;
+	}
+
+	copp_idx = q6adm_get_copp_id(copp);
+	set_bit(copp_idx, &session->copp_map);
+	session->copps[copp_idx] = copp;
+
+	for_each_set_bit(j, &session->copp_map, MAX_COPPS_PER_PORT) {
+		payload.port_id[num_copps] = session->port_id;
+		payload.copp_idx[num_copps] = j;
+		num_copps++;
+	}
+
+	if (num_copps) {
+		payload.num_copps = num_copps;
+		payload.session_id = stream_id;
+		q6adm_matrix_map(routing_data->dev, session->path_type,
+				 payload, perf_mode);
+	}
+	mutex_unlock(&routing_data->lock);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(q6routing_stream_open);
+
+static struct session_data *get_session_from_id(struct msm_routing_data *data,
+						int fedai_id)
+{
+	int i;
+
+	for (i = 0; i < MAX_SESSIONS; i++) {
+		if (fedai_id == data->sessions[i].fedai_id)
+			return &data->sessions[i];
+	}
+
+	return NULL;
+}
+/**
+ * q6routing_stream_close() - Deregister a stream
+ *
+ * @fedai_id: Frontend dai id.
+ * @stream_type: Direction of stream
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+void q6routing_stream_close(int fedai_id, int stream_type)
+{
+	struct session_data *session;
+	int idx;
+
+	session = get_session_from_id(routing_data, fedai_id);
+	if (!session)
+		return;
+
+	for_each_set_bit(idx, &session->copp_map, MAX_COPPS_PER_PORT) {
+		if (session->copps[idx]) {
+			q6adm_close(routing_data->dev, session->copps[idx]);
+			session->copps[idx] = NULL;
+		}
+	}
+
+	session->fedai_id = -1;
+	session->copp_map = 0;
+}
+EXPORT_SYMBOL_GPL(q6routing_stream_close);
+
+static int msm_routing_get_audio_mixer(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_context *dapm =
+	    snd_soc_dapm_kcontrol_dapm(kcontrol);
+	struct soc_mixer_control *mc =
+	    (struct soc_mixer_control *)kcontrol->private_value;
+	int session_id = mc->shift;
+	struct snd_soc_component *c = snd_soc_dapm_to_component(dapm);
+	struct msm_routing_data *priv = dev_get_drvdata(c->dev);
+	struct session_data *session = &priv->sessions[session_id];
+
+	if (session->port_id == mc->reg)
+		ucontrol->value.integer.value[0] = 1;
+	else
+		ucontrol->value.integer.value[0] = 0;
+
+	return 0;
+}
+
+static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_context *dapm =
+				    snd_soc_dapm_kcontrol_dapm(kcontrol);
+	struct snd_soc_component *c = snd_soc_dapm_to_component(dapm);
+	struct msm_routing_data *data = dev_get_drvdata(c->dev);
+	struct soc_mixer_control *mc =
+		    (struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_dapm_update *update = NULL;
+	int be_id = mc->reg;
+	int session_id = mc->shift;
+	struct session_data *session = &data->sessions[session_id];
+
+	if (ucontrol->value.integer.value[0]) {
+		session->port_id = be_id;
+		snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update);
+	} else {
+		session->port_id = -1;
+		snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update);
+	}
+
+	return 1;
+}
+
+static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
+	SOC_SINGLE_EXT("MultiMedia1", HDMI_RX,
+		       MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0,
+		       msm_routing_get_audio_mixer,
+		       msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia2", HDMI_RX,
+		       MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0,
+		       msm_routing_get_audio_mixer,
+		       msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia3", HDMI_RX,
+		       MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0,
+		       msm_routing_get_audio_mixer,
+		       msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia4", HDMI_RX,
+		       MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0,
+		       msm_routing_get_audio_mixer,
+		       msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia5", HDMI_RX,
+		       MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0,
+		       msm_routing_get_audio_mixer,
+		       msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia6", HDMI_RX,
+		       MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0,
+		       msm_routing_get_audio_mixer,
+		       msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia7", HDMI_RX,
+		       MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0,
+		       msm_routing_get_audio_mixer,
+		       msm_routing_put_audio_mixer),
+	SOC_SINGLE_EXT("MultiMedia8", HDMI_RX,
+		       MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0,
+		       msm_routing_get_audio_mixer,
+		       msm_routing_put_audio_mixer),
+};
+
+static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
+	/* Frontend AIF */
+	SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0),
+
+	/* Mixer definitions */
+	SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0,
+			   hdmi_mixer_controls,
+			   ARRAY_SIZE(hdmi_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"HDMI Mixer", "MultiMedia1", "MM_DL1"},
+	{"HDMI Mixer", "MultiMedia2", "MM_DL2"},
+	{"HDMI Mixer", "MultiMedia3", "MM_DL3"},
+	{"HDMI Mixer", "MultiMedia4", "MM_DL4"},
+	{"HDMI Mixer", "MultiMedia5", "MM_DL5"},
+	{"HDMI Mixer", "MultiMedia6", "MM_DL6"},
+	{"HDMI Mixer", "MultiMedia7", "MM_DL7"},
+	{"HDMI Mixer", "MultiMedia8", "MM_DL8"},
+	{"HDMI_RX", NULL, "HDMI Mixer"},
+};
+
+static int routing_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_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+	struct msm_routing_data *data = dev_get_drvdata(c->dev);
+	unsigned int be_id = rtd->cpu_dai->id;
+	struct session_data *session;
+	int path_type;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		path_type = ADM_PATH_PLAYBACK;
+	else
+		path_type = ADM_PATH_LIVE_REC;
+
+	if (be_id > AFE_MAX_PORTS)
+		return -EINVAL;
+
+	session = &data->port_data[be_id];
+
+	mutex_lock(&data->lock);
+
+	session->path_type = path_type;
+	session->sample_rate = params_rate(params);
+	session->channels = params_channels(params);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+			session->bits_per_sample = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+			session->bits_per_sample = 24;
+		break;
+	default:
+		break;
+	}
+
+	mutex_unlock(&data->lock);
+	return 0;
+}
+
+static struct snd_pcm_ops q6pcm_routing_ops = {
+	.hw_params = routing_hw_params,
+};
+
+static int msm_routing_probe(struct snd_soc_component *c)
+{
+	int i;
+
+	for (i = 0; i < MAX_SESSIONS; i++)
+		routing_data->sessions[i].port_id = -1;
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver msm_soc_routing_component = {
+	.ops = &q6pcm_routing_ops,
+	.probe = msm_routing_probe,
+	.name = DRV_NAME,
+	.dapm_widgets = msm_qdsp6_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
+};
+
+static int q6routing_dai_bind(struct device *dev, struct device *master,
+			      void *data)
+{
+	routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL);
+	if (!routing_data)
+		return -ENOMEM;
+
+	routing_data->dev = dev;
+
+	mutex_init(&routing_data->lock);
+	dev_set_drvdata(dev, routing_data);
+
+	return snd_soc_register_component(dev, &msm_soc_routing_component,
+					  NULL, 0);
+}
+
+static void q6routing_dai_unbind(struct device *dev, struct device *master,
+				 void *d)
+{
+	struct msm_routing_data *data = dev_get_drvdata(dev);
+
+	snd_soc_unregister_component(dev);
+
+	kfree(data);
+
+	routing_data = NULL;
+}
+
+static const struct component_ops q6routing_dai_comp_ops = {
+	.bind   = q6routing_dai_bind,
+	.unbind = q6routing_dai_unbind,
+};
+
+static int q6pcm_routing_probe(struct platform_device *pdev)
+{
+	return component_add(&pdev->dev, &q6routing_dai_comp_ops);
+}
+
+static int q6pcm_routing_remove(struct platform_device *pdev)
+{
+	component_del(&pdev->dev, &q6routing_dai_comp_ops);
+	return 0;
+}
+
+static struct platform_driver q6pcm_routing_platform_driver = {
+	.driver = {
+		.name = "q6routing",
+	},
+	.probe = q6pcm_routing_probe,
+	.remove = q6pcm_routing_remove,
+};
+module_platform_driver(q6pcm_routing_platform_driver);
+
+MODULE_DESCRIPTION("Q6 Routing platform");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6routing.h b/sound/soc/qcom/qdsp6/q6routing.h
new file mode 100644
index 000000000000..35514e651130
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.h
@@ -0,0 +1,9 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef _Q6_PCM_ROUTING_H
+#define _Q6_PCM_ROUTING_H
+
+int q6routing_stream_open(int fedai_id, int perf_mode,
+			   int stream_id, int stream_type);
+void q6routing_stream_close(int fedai_id, int stream_type);
+
+#endif /*_Q6_PCM_ROUTING_H */
-- 
2.17.0

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