On 5/1/2018 5:08 AM, Srinivas Kandagatla wrote:
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@xxxxxxxxxx>
Reviewed-and-tested-by: Rohit kumar <rohitkr@xxxxxxxxxxxxxx>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6routing.c | 393 +++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6routing.h | 9 +
4 files changed, 407 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..72863672f11f
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -0,0 +1,393 @@
+// 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 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;
+ 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_idx = 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_idx < 0) {
+ mutex_unlock(&routing_data->lock);
+ return -EINVAL;
+ }
+
+ set_bit(copp_idx, &session->copp_map);
+
+ 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)
+ q6adm_close(routing_data->dev, session->port_id,
+ session->perf_mode, idx);
+
+ 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,
At some point in future, when FE DAI IDs become more than "31",
SOC_DOUBLE_EXT would need to be used with "reg" marked with SND_SOC_NOPM
and FE DAI and BE DAI IDs passed in shift_left and shift_right. This is
to avoid having any "shift" operation issue in DAPM code, when reg is
not marked with SND_SOC_NOPM, and shift value is more than 31.
However, for now, its OK.
Acked-by: Banajit Goswami <bgoswami@xxxxxxxxxxxxxx
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