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-rw-r--r--include/sound/sof/dai-intel.h4
-rw-r--r--include/sound/sof/dai.h10
-rw-r--r--sound/soc/amd/acp-rt5645.c2
-rw-r--r--sound/soc/sof/intel/hda-dai.c82
-rw-r--r--sound/soc/sof/intel/hda.c6
-rw-r--r--sound/soc/sof/sof-audio.c4
-rw-r--r--sound/soc/sof/sof-of-dev.c21
-rw-r--r--sound/soc/sof/sof-priv.h6
-rw-r--r--sound/soc/sof/topology.c24
9 files changed, 143 insertions, 16 deletions
diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h
index 136adf6686e2..7a266f41983c 100644
--- a/include/sound/sof/dai-intel.h
+++ b/include/sound/sof/dai-intel.h
@@ -48,6 +48,10 @@
#define SOF_DAI_INTEL_SSP_CLKCTRL_FS_KA BIT(4)
/* bclk idle */
#define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_IDLE_HIGH BIT(5)
+/* mclk early start */
+#define SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_ES BIT(6)
+/* bclk early start */
+#define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_ES BIT(7)
/* DMIC max. four controllers for eight microphone channels */
#define SOF_DAI_INTEL_DMIC_NUM_CTRL 4
diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h
index 6bb403e8c5ee..9625f47557b8 100644
--- a/include/sound/sof/dai.h
+++ b/include/sound/sof/dai.h
@@ -50,6 +50,13 @@
#define SOF_DAI_FMT_INV_MASK 0x0f00
#define SOF_DAI_FMT_CLOCK_PROVIDER_MASK 0xf000
+/* DAI_CONFIG flags */
+#define SOF_DAI_CONFIG_FLAGS_MASK 0x3
+#define SOF_DAI_CONFIG_FLAGS_NONE (0 << 0) /**< DAI_CONFIG sent without stage information */
+#define SOF_DAI_CONFIG_FLAGS_HW_PARAMS (1 << 0) /**< DAI_CONFIG sent during hw_params stage */
+#define SOF_DAI_CONFIG_FLAGS_HW_FREE (2 << 0) /**< DAI_CONFIG sent during hw_free stage */
+#define SOF_DAI_CONFIG_FLAGS_RFU (3 << 0) /**< not used, reserved for future use */
+
/** \brief Types of DAI */
enum sof_ipc_dai_type {
SOF_DAI_INTEL_NONE = 0, /**< None */
@@ -69,7 +76,8 @@ struct sof_ipc_dai_config {
/* physical protocol and clocking */
uint16_t format; /**< SOF_DAI_FMT_ */
- uint16_t reserved16; /**< alignment */
+ uint8_t group_id; /**< group ID, 0 means no group (ABI 3.17) */
+ uint8_t flags; /**< SOF_DAI_CONFIG_FLAGS_ (ABI 3.19) */
/* reserved for future use */
uint32_t reserved[8];
diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c
index d6ba94677ac2..6d5c547a32de 100644
--- a/sound/soc/amd/acp-rt5645.c
+++ b/sound/soc/amd/acp-rt5645.c
@@ -91,7 +91,7 @@ static int cz_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-static struct snd_soc_ops cz_aif1_ops = {
+static const struct snd_soc_ops cz_aif1_ops = {
.hw_params = cz_aif1_hw_params,
};
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 59d6750c6a20..dfd2df0b1bc3 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -440,6 +440,11 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = {
#endif
+/* only one flag used so far to harden hw_params/hw_free/trigger/prepare */
+struct ssp_dai_dma_data {
+ bool setup;
+};
+
static int ssp_dai_setup_or_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai,
bool setup)
{
@@ -469,22 +474,95 @@ static int ssp_dai_setup_or_free(struct snd_pcm_substream *substream, struct snd
return hda_ctrl_dai_widget_free(w);
}
+static int ssp_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct ssp_dai_dma_data *dma_data;
+
+ dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL);
+ if (!dma_data)
+ return -ENOMEM;
+
+ snd_soc_dai_set_dma_data(dai, substream, dma_data);
+
+ return 0;
+}
+
+static int ssp_dai_setup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai,
+ bool setup)
+{
+ struct ssp_dai_dma_data *dma_data;
+ int ret = 0;
+
+ dma_data = snd_soc_dai_get_dma_data(dai, substream);
+ if (!dma_data) {
+ dev_err(dai->dev, "%s: failed to get dma_data\n", __func__);
+ return -EIO;
+ }
+
+ if (dma_data->setup != setup) {
+ ret = ssp_dai_setup_or_free(substream, dai, setup);
+ if (!ret)
+ dma_data->setup = setup;
+ }
+ return ret;
+}
+
static int ssp_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- return ssp_dai_setup_or_free(substream, dai, true);
+ /* params are ignored for now */
+ return ssp_dai_setup(substream, dai, true);
+}
+
+static int ssp_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ /*
+ * the SSP will only be reconfigured during resume operations and
+ * not in case of xruns
+ */
+ return ssp_dai_setup(substream, dai, true);
+}
+
+static int ssp_dai_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ if (cmd != SNDRV_PCM_TRIGGER_SUSPEND)
+ return 0;
+
+ return ssp_dai_setup(substream, dai, false);
}
static int ssp_dai_hw_free(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
- return ssp_dai_setup_or_free(substream, dai, false);
+ return ssp_dai_setup(substream, dai, false);
+}
+
+static void ssp_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct ssp_dai_dma_data *dma_data;
+
+ dma_data = snd_soc_dai_get_dma_data(dai, substream);
+ if (!dma_data) {
+ dev_err(dai->dev, "%s: failed to get dma_data\n", __func__);
+ return;
+ }
+ snd_soc_dai_set_dma_data(dai, substream, NULL);
+ kfree(dma_data);
}
static const struct snd_soc_dai_ops ssp_dai_ops = {
+ .startup = ssp_dai_startup,
.hw_params = ssp_dai_hw_params,
+ .prepare = ssp_dai_prepare,
+ .trigger = ssp_dai_trigger,
.hw_free = ssp_dai_hw_free,
+ .shutdown = ssp_dai_shutdown,
};
/*
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 066e90506ae6..1463f3de01bc 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -71,6 +71,9 @@ int hda_ctrl_dai_widget_setup(struct snd_soc_dapm_widget *w)
return ret;
}
+ /* set HW_PARAMS flag */
+ config->flags = FIELD_PREP(SOF_DAI_CONFIG_FLAGS_MASK, SOF_DAI_CONFIG_FLAGS_HW_PARAMS);
+
/* send DAI_CONFIG IPC */
ret = sof_ipc_tx_message(sdev->ipc, config->hdr.cmd, config, config->hdr.size,
&reply, sizeof(reply));
@@ -107,6 +110,9 @@ int hda_ctrl_dai_widget_free(struct snd_soc_dapm_widget *w)
config = &sof_dai->dai_config[sof_dai->current_config];
+ /* set HW_FREE flag */
+ config->flags = FIELD_PREP(SOF_DAI_CONFIG_FLAGS_MASK, SOF_DAI_CONFIG_FLAGS_HW_FREE);
+
ret = sof_ipc_tx_message(sdev->ipc, config->hdr.cmd, config, config->hdr.size,
&reply, sizeof(reply));
if (ret < 0)
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index c4cabe26b157..262cb3ad4674 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -8,6 +8,7 @@
// Author: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
//
+#include <linux/bitfield.h>
#include "sof-audio.h"
#include "ops.h"
@@ -55,6 +56,9 @@ static int sof_dai_config_setup(struct snd_sof_dev *sdev, struct snd_sof_dai *da
return -EINVAL;
}
+ /* set NONE flag to clear all previous settings */
+ config->flags = FIELD_PREP(SOF_DAI_CONFIG_FLAGS_MASK, SOF_DAI_CONFIG_FLAGS_NONE);
+
ret = sof_ipc_tx_message(sdev->ipc, config->hdr.cmd, config, config->hdr.size,
&reply, sizeof(reply));
diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c
index f0f819a46456..885430a42226 100644
--- a/sound/soc/sof/sof-of-dev.c
+++ b/sound/soc/sof/sof-of-dev.c
@@ -7,12 +7,21 @@
#include <linux/firmware.h>
#include <linux/module.h>
+#include <linux/moduleparam.h>
#include <linux/pm_runtime.h>
#include <sound/sof.h>
#include "ops.h"
#include "imx/imx-ops.h"
+static char *fw_path;
+module_param(fw_path, charp, 0444);
+MODULE_PARM_DESC(fw_path, "alternate path for SOF firmware.");
+
+static char *tplg_path;
+module_param(tplg_path, charp, 0444);
+MODULE_PARM_DESC(tplg_path, "alternate path for SOF topology.");
+
/* platform specific devices */
#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8)
static struct sof_dev_desc sof_of_imx8qxp_desc = {
@@ -87,9 +96,15 @@ static int sof_of_probe(struct platform_device *pdev)
sof_pdata->dev = &pdev->dev;
sof_pdata->fw_filename = desc->default_fw_filename;
- /* TODO: read alternate fw and tplg filenames from DT */
- sof_pdata->fw_filename_prefix = sof_pdata->desc->default_fw_path;
- sof_pdata->tplg_filename_prefix = sof_pdata->desc->default_tplg_path;
+ if (fw_path)
+ sof_pdata->fw_filename_prefix = fw_path;
+ else
+ sof_pdata->fw_filename_prefix = sof_pdata->desc->default_fw_path;
+
+ if (tplg_path)
+ sof_pdata->tplg_filename_prefix = tplg_path;
+ else
+ sof_pdata->tplg_filename_prefix = sof_pdata->desc->default_tplg_path;
/* set callback to be called on successful device probe to enable runtime_pm */
sof_pdata->sof_probe_complete = sof_of_probe_complete;
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index 1289e2efeb62..4e5bab838cbf 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -24,6 +24,12 @@
#define SOF_DBG_ENABLE_TRACE BIT(0)
#define SOF_DBG_RETAIN_CTX BIT(1) /* prevent DSP D3 on FW exception */
#define SOF_DBG_VERIFY_TPLG BIT(2) /* verify topology during load */
+#define SOF_DBG_DYNAMIC_PIPELINES_OVERRIDE BIT(3) /* 0: use topology token
+ * 1: override topology
+ */
+#define SOF_DBG_DYNAMIC_PIPELINES_ENABLE BIT(4) /* 0: use static pipelines
+ * 1: use dynamic pipelines
+ */
#define SOF_DBG_DUMP_REGS BIT(0)
#define SOF_DBG_DUMP_MBOX BIT(1)
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 44d60081bc26..534f004f6162 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -1759,9 +1759,14 @@ static int sof_widget_load_pipeline(struct snd_soc_component *scomp, int index,
goto err;
}
- dev_dbg(scomp->dev, "pipeline %s: period %d pri %d mips %d core %d frames %d\n",
+ if (sof_core_debug & SOF_DBG_DYNAMIC_PIPELINES_OVERRIDE)
+ swidget->dynamic_pipeline_widget = sof_core_debug &
+ SOF_DBG_DYNAMIC_PIPELINES_ENABLE;
+
+ dev_dbg(scomp->dev, "pipeline %s: period %d pri %d mips %d core %d frames %d dynamic %d\n",
swidget->widget->name, pipeline->period, pipeline->priority,
- pipeline->period_mips, pipeline->core, pipeline->frames_per_sched);
+ pipeline->period_mips, pipeline->core, pipeline->frames_per_sched,
+ swidget->dynamic_pipeline_widget);
swidget->private = pipeline;
@@ -2374,13 +2379,14 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
}
ret = sof_widget_load_dai(scomp, index, swidget, tw, dai);
- if (ret == 0) {
- sof_connect_dai_widget(scomp, w, tw, dai);
- list_add(&dai->list, &sdev->dai_list);
- swidget->private = dai;
- } else {
+ if (!ret)
+ ret = sof_connect_dai_widget(scomp, w, tw, dai);
+ if (ret < 0) {
kfree(dai);
+ break;
}
+ list_add(&dai->list, &sdev->dai_list);
+ swidget->private = dai;
break;
case snd_soc_dapm_mixer:
ret = sof_widget_load_mixer(scomp, index, swidget, tw);
@@ -2825,12 +2831,12 @@ static int sof_link_ssp_load(struct snd_soc_component *scomp, int index,
config[i].ssp.rx_slots = le32_to_cpu(hw_config[i].rx_slots);
config[i].ssp.tx_slots = le32_to_cpu(hw_config[i].tx_slots);
- dev_dbg(scomp->dev, "tplg: config SSP%d fmt 0x%x mclk %d bclk %d fclk %d width (%d)%d slots %d mclk id %d quirks %d\n",
+ dev_dbg(scomp->dev, "tplg: config SSP%d fmt %#x mclk %d bclk %d fclk %d width (%d)%d slots %d mclk id %d quirks %d clks_control %#x\n",
config[i].dai_index, config[i].format,
config[i].ssp.mclk_rate, config[i].ssp.bclk_rate,
config[i].ssp.fsync_rate, config[i].ssp.sample_valid_bits,
config[i].ssp.tdm_slot_width, config[i].ssp.tdm_slots,
- config[i].ssp.mclk_id, config[i].ssp.quirks);
+ config[i].ssp.mclk_id, config[i].ssp.quirks, config[i].ssp.clks_control);
/* validate SSP fsync rate and channel count */
if (config[i].ssp.fsync_rate < 8000 || config[i].ssp.fsync_rate > 192000) {