Messages in this thread | | | Subject | Re: [alsa-devel] [PATCH v3 15/25] ASoC: qcom: qdsp6: Add support to q6asm dai driver | From | Srinivas Kandagatla <> | Date | Thu, 22 Feb 2018 11:16:26 +0000 |
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Thanks for your review Rohit,
On 21/02/18 11:14, Rohit Kumar wrote: > > > On 2/13/2018 10:28 PM, srinivas.kandagatla@linaro.org wrote: >> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> >> >> This patch adds support to q6asm dai driver which configures Q6ASM >> streams >> to pass pcm data. >> >> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.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 Makes sense, I will add them.
>> + 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 Yep, I will do that in next version.
>> + 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 Sure!
>> + >> + 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;
Okay, I will take closer look at sdm845 and other socs, and make it more generic in next version.
>> + 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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