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Message-Id: <20171205221812.25641-13-srinivas.kandagatla@linaro.org>
Date: Tue, 5 Dec 2017 22:18:09 +0000
From: srinivas.kandagatla@...aro.org
To: broonie@...nel.org, bgoswami@...eaurora.org
Cc: alsa-devel@...a-project.org, tiwai@...e.com, plai@...eaurora.org,
kwestfie@...eaurora.org, linux-kernel@...r.kernel.org,
linux-arm-msm@...r.kernel.org, lkasam@....qualcomm.com,
Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Subject: [PATCH v2 12/15] ASoC: qcom: qdsp6: Add support to q6asm dai driver
From: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
This patch adds support to q6asm dai driver which configures Q6ASM streams
to pass pcm data.
Currently the driver only exposes 2 playback streams for hdmi playback
support, it can be easily extended to add all 8 streams.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
---
sound/soc/qcom/Kconfig | 6 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6asm-dai.c | 534 +++++++++++++++++++++++++++++++++++++++
3 files changed, 541 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 003ce182691c..ecd1e4ba834d 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -68,6 +68,11 @@ config SND_SOC_QDSP6_AFE_DAI
tristate
default n
+config SND_SOC_QDSP6_ASM_DAI
+ tristate
+ default n
+
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
select SND_SOC_QDSP6_AFE
@@ -76,6 +81,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_CORE
select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_AFE_DAI
+ select SND_SOC_QDSP6_ASM_DAI
help
To add support for MSM QDSP6 Soc Audio.
This will enable sound soc platform specific
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index bd8bd02bf09e..03576a442fb5 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -4,3 +4,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o
+obj-$(CONFIG_SND_SOC_QDSP6_ASM_DAI) += q6asm-dai.o
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
new file mode 100644
index 000000000000..709c5de230fa
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -0,0 +1,534 @@
+/* SPDX-License-Identifier: GPL-2.0
+* Copyright (c) 2011-2016, The Linux Foundation
+* Copyright (c) 2017, Linaro Limited
+*/
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/of_device.h>
+#include <sound/pcm_params.h>
+#include "q6asm.h"
+#include "q6routing.h"
+#include "common.h"
+
+#define PLAYBACK_MIN_NUM_PERIODS 2
+#define PLAYBACK_MAX_NUM_PERIODS 8
+#define PLAYBACK_MAX_PERIOD_SIZE 65536
+#define PLAYBACK_MIN_PERIOD_SIZE 128
+
+enum stream_state {
+ IDLE = 0,
+ STOPPED,
+ RUNNING,
+};
+
+struct q6asm_dai_rtd {
+ struct snd_pcm_substream *substream;
+ dma_addr_t phys;
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ unsigned int periods;
+ uint16_t bits_per_sample;
+ uint16_t source; /* Encoding source bit mask */
+
+ struct audio_client *audio_client;
+ uint16_t session_id;
+
+ enum stream_state state;
+ bool set_channel_map;
+ char channel_map[8];
+};
+
+struct q6asm_dai_data {
+ u64 sid;
+};
+
+static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = (PLAYBACK_MAX_NUM_PERIODS *
+ PLAYBACK_MAX_PERIOD_SIZE),
+ .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
+ .periods_min = PLAYBACK_MIN_NUM_PERIODS,
+ .periods_max = PLAYBACK_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+ 88200, 96000, 176400, 192000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode, uint32_t token,
+ uint32_t *payload, void *priv)
+{
+ struct q6asm_dai_rtd *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+
+ switch (opcode) {
+ case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ break;
+ case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+ prtd->state = STOPPED;
+ break;
+ case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ snd_pcm_period_elapsed(substream);
+ if (prtd->state == RUNNING)
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+
+ break;
+ }
+ default:
+ break;
+ }
+}
+
+static int q6asm_dai_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct q6asm_dai_data *pdata;
+ int ret;
+
+ pdata = dev_get_drvdata(soc_prtd->platform->dev);
+ if (!pdata)
+ return -EINVAL;
+
+ if (!prtd || !prtd->audio_client) {
+ pr_err("%s: private data null or audio client freed\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ if (prtd->state) {
+ /* clear the previous setup if any */
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_unmap_memory_regions(substream->stream,
+ prtd->audio_client);
+ q6routing_dereg_phy_stream(soc_prtd->dai_link->id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ }
+
+ ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
+ prtd->phys,
+ (prtd->pcm_size / prtd->periods),
+ prtd->periods);
+
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
+ ret);
+ return -ENOMEM;
+ }
+
+ ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
+ if (ret < 0) {
+ pr_err("%s: q6asm_open_write failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return -ENOMEM;
+ }
+
+ prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+ ret = q6routing_reg_phy_stream(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
+ prtd->session_id, substream->stream);
+ if (ret) {
+ pr_err("%s: stream reg failed ret:%d\n", __func__, ret);
+ return ret;
+ }
+
+ ret = q6asm_media_format_block_multi_ch_pcm(
+ prtd->audio_client, runtime->rate,
+ runtime->channels, !prtd->set_channel_map,
+ prtd->channel_map, prtd->bits_per_sample);
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ prtd->state = RUNNING;
+
+ return 0;
+}
+
+static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ prtd->state = STOPPED;
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct q6asm_dai_rtd *prtd;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = soc_prtd->platform->dev;
+ int ret = 0;
+
+ pdata = dev_get_drvdata(dev);
+ if (!pdata) {
+ pr_err("Platform data not found ..\n");
+ return -EINVAL;
+ }
+
+ prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(dev,
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+
+// prtd->audio_client->dev = dev;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ runtime->hw = q6asm_dai_hardware_playback;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
+ PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes min max ret = %d\n",
+ ret);
+ }
+ }
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (ret < 0) {
+ pr_err("constraint for period bytes step ret = %d\n",
+ ret);
+ }
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes step ret = %d\n",
+ ret);
+ }
+
+ prtd->set_channel_map = false;
+ runtime->private_data = prtd;
+
+ snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback);
+
+ runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max;
+
+
+ if (pdata->sid < 0)
+ prtd->phys = substream->dma_buffer.addr;
+ else
+ prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int q6asm_dai_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ if (prtd->audio_client) {
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_unmap_memory_regions(substream->stream,
+ prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ }
+ q6routing_dereg_phy_stream(soc_prtd->dai_link->id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ kfree(prtd);
+ return 0;
+}
+
+static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int q6asm_dai_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct snd_card *card = soc_prtd->card->snd_card;
+
+ return dma_mmap_coherent(card->dev, vma,
+ runtime->dma_area, runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static int q6asm_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ prtd->pcm_size = params_buffer_bytes(params);
+ prtd->periods = params_periods(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ prtd->bits_per_sample = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ prtd->bits_per_sample = 24;
+ break;
+ }
+
+ return 0;
+}
+
+static struct snd_pcm_ops q6asm_dai_ops = {
+ .open = q6asm_dai_open,
+ .hw_params = q6asm_dai_hw_params,
+ .close = q6asm_dai_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = q6asm_dai_prepare,
+ .trigger = q6asm_dai_trigger,
+ .pointer = q6asm_dai_pointer,
+ .mmap = q6asm_dai_mmap,
+};
+
+static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ struct snd_pcm_substream *substream;
+ struct snd_card *card = rtd->card->snd_card;
+ struct device *dev = card->dev;
+ struct device_node *node = dev->of_node;
+ struct q6asm_dai_data *pdata = dev_get_drvdata(rtd->platform->dev);
+ struct of_phandle_args args;
+
+ int size, ret = 0;
+
+ ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
+ if (ret < 0)
+ pdata->sid = -1;
+ else
+ pdata->sid = args.args[0];
+
+
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ size = q6asm_dai_hardware_playback.buffer_bytes_max;
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+ &substream->dma_buffer);
+ if (ret) {
+ dev_err(dev, "Cannot allocate buffer(s)\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+ substream = pcm->streams[i].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+static struct snd_soc_platform_driver q6asm_soc_platform = {
+ .ops = &q6asm_dai_ops,
+ .pcm_new = q6asm_dai_pcm_new,
+ .pcm_free = q6asm_dai_pcm_free,
+
+};
+
+static const struct snd_soc_dapm_route afe_pcm_routes[] = {
+ {"MM_DL1", NULL, "MultiMedia1 Playback" },
+ {"MM_DL2", NULL, "MultiMedia2 Playback" },
+
+};
+
+static int fe_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_dapm_context *dapm;
+
+ dapm = snd_soc_component_get_dapm(dai->component);
+ snd_soc_dapm_add_routes(dapm, afe_pcm_routes,
+ ARRAY_SIZE(afe_pcm_routes));
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver q6asm_fe_dai_component = {
+ .name = "q6asm-fe-dai",
+};
+
+static struct snd_soc_dai_driver q6asm_fe_dais[] = {
+ {
+ .playback = {
+ .stream_name = "MultiMedia1 Playback",
+ .rates = (SNDRV_PCM_RATE_8000_192000|
+ SNDRV_PCM_RATE_KNOT),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "MM_DL1",
+ .probe = fe_dai_probe,
+ .id = MSM_FRONTEND_DAI_MULTIMEDIA1,
+ },
+ {
+ .playback = {
+ .stream_name = "MultiMedia2 Playback",
+ .rates = (SNDRV_PCM_RATE_8000_192000|
+ SNDRV_PCM_RATE_KNOT),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "MM_DL2",
+ .probe = fe_dai_probe,
+ .id = MSM_FRONTEND_DAI_MULTIMEDIA2,
+ },
+};
+
+static int q6asm_dai_probe(struct platform_device *pdev)
+{
+ struct q6asm_dai_data *pdata;
+ struct device *dev = &pdev->dev;
+ int rc;
+
+ pdata = devm_kzalloc(dev, sizeof(struct q6asm_dai_data), GFP_KERNEL);
+ if (!pdata)
+ return -ENOMEM;
+
+
+ dev_set_drvdata(dev, pdata);
+
+ rc = snd_soc_register_platform(dev, &q6asm_soc_platform);
+ if (rc) {
+ dev_err(&pdev->dev, "err_dai_platform\n");
+ return rc;
+ }
+
+ rc = snd_soc_register_component(dev, &q6asm_fe_dai_component,
+ q6asm_fe_dais,
+ ARRAY_SIZE(q6asm_fe_dais));
+ if (rc)
+ dev_err(dev, "err_dai_component\n");
+
+ return rc;
+
+}
+
+static int q6asm_dai_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver q6asm_dai_driver = {
+ .driver = {
+ .name = "q6asm_dai",
+ .owner = THIS_MODULE,
+ },
+ .probe = q6asm_dai_probe,
+ .remove = q6asm_dai_remove,
+};
+
+module_platform_driver(q6asm_dai_driver);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
--
2.15.0
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