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Message-ID: <6ea65838-f01a-ecbf-70cd-0dc9299eae80@codeaurora.org>
Date: Wed, 21 Feb 2018 16:44:24 +0530
From: Rohit Kumar <rohitkr@...eaurora.org>
To: srinivas.kandagatla@...aro.org, andy.gross@...aro.org,
broonie@...nel.org, linux-arm-msm@...r.kernel.org,
alsa-devel@...a-project.org
Cc: mark.rutland@....com, devicetree@...r.kernel.org,
bgoswami@...eaurora.org, rohkumar@....qualcomm.com,
linux-kernel@...r.kernel.org, plai@...eaurora.org, tiwai@...e.com,
lgirdwood@...il.com, david.brown@...aro.org, robh+dt@...nel.org,
spatakok@....qualcomm.com, linux-soc@...r.kernel.org,
linux-arm-kernel@...ts.infradead.org
Subject: Re: [alsa-devel] [PATCH v3 15/25] ASoC: qcom: qdsp6: Add support to
q6asm dai driver
On 2/13/2018 10:28 PM, srinivas.kandagatla@...aro.org wrote:
> From: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
>
> This patch adds support to q6asm dai driver which configures Q6ASM streams
> to pass pcm data.
>
> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
[..]
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
> new file mode 100644
> index 000000000000..7c5e94b2ced4
> --- /dev/null
> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
> @@ -0,0 +1,621 @@
> +// 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>
[..]
> +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:
Need to add support for V2 version of opcodes
> + q6asm_write_async(prtd->audio_client,
> + prtd->pcm_count, 0, 0, NO_TIMESTAMP);
> + break;
> + case ASM_CLIENT_EVENT_CMD_EOS_DONE:
> + prtd->state = Q6ASM_STREAM_STOPPED;
> + break;
> + case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
> + prtd->pcm
[..]
> +
> +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;
below two cases can be combined with START if no change
> + 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 = Q6ASM_STREAM_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 snd_soc_dai *cpu_dai = soc_prtd->cpu_dai;
> +
> + struct q6asm_dai_rtd *prtd;
> + struct q6asm_dai_data *pdata;
> + struct device *dev = soc_prtd->platform->dev;
> + int ret = 0;
> + int stream_id;
> +
> + stream_id = cpu_dai->driver->id;
> +
> + pdata = q6asm_get_dai_data(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,
> + (q6asm_cb)event_handler, prtd, stream_id);
> + if (!prtd->audio_client) {
> + pr_info("%s: Could not allocate memory\n", __func__);
> + kfree(prtd);
> + return -ENOMEM;
> + }
> +
> +// prtd->audio_client->dev = dev;
cleanup this
> +
> + 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);
[..]
> +
> +static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
> +{
> + struct snd_pcm_substream *substream;
> + struct of_phandle_args args;
> + struct device_node *node;
> + struct q6asm_dai_data *pdata;
> + struct snd_pcm *pcm = rtd->pcm;
> + struct device *dev;
> + int size, ret;
> +
> + dev = rtd->platform->dev->parent;
> + node = dev->of_node;
> + pdata = q6asm_get_dai_data(rtd->platform->dev);
> +
> + ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
> + if (ret < 0)
> + pdata->sid = -1;
> + else
> + pdata->sid = args.args[0];
> +
> +
> +
iommus for sdm845 is 16bit value. we need to have sid_mask which is 0x1
in sdm845. We need to mask sid with 0x1 to get proper sid.
pdata->sid &= 0x1;
> + 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;
> +}
> +
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