diff options
Diffstat (limited to '')
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6asm-dai.c | 1383 |
1 files changed, 1383 insertions, 0 deletions
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c new file mode 100644 index 000000000..84cf190aa --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -0,0 +1,1383 @@ +// 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/slab.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <linux/spinlock.h> +#include <sound/compress_driver.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 "q6dsp-errno.h" + +#define DRV_NAME "q6asm-fe-dai" + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 8 +#define PLAYBACK_MAX_PERIOD_SIZE 65536 +#define PLAYBACK_MIN_PERIOD_SIZE 128 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 8 +#define CAPTURE_MAX_PERIOD_SIZE 4096 +#define CAPTURE_MIN_PERIOD_SIZE 320 +#define SID_MASK_DEFAULT 0xF + +/* Default values used if user space does not set */ +#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024) +#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) +#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) +#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) + +#define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) +#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) + +enum stream_state { + Q6ASM_STREAM_IDLE = 0, + Q6ASM_STREAM_STOPPED, + Q6ASM_STREAM_RUNNING, +}; + +struct q6asm_dai_rtd { + struct snd_pcm_substream *substream; + struct snd_compr_stream *cstream; + struct snd_codec codec; + struct snd_dma_buffer dma_buffer; + spinlock_t lock; + phys_addr_t phys; + unsigned int pcm_size; + unsigned int pcm_count; + unsigned int pcm_irq_pos; /* IRQ position */ + unsigned int periods; + unsigned int bytes_sent; + unsigned int bytes_received; + unsigned int copied_total; + uint16_t bits_per_sample; + uint16_t source; /* Encoding source bit mask */ + struct audio_client *audio_client; + uint32_t next_track_stream_id; + bool next_track; + uint32_t stream_id; + uint16_t session_id; + enum stream_state state; + uint32_t initial_samples_drop; + uint32_t trailing_samples_drop; + bool notify_on_drain; +}; + +struct q6asm_dai_data { + struct snd_soc_dai_driver *dais; + int num_dais; + long long int sid; +}; + +static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | + 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_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 4, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * + CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware q6asm_dai_hardware_playback = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | + 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, +}; + +#define Q6ASM_FEDAI_DRIVER(num) { \ + .playback = { \ + .stream_name = "MultiMedia"#num" 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, \ + }, \ + .capture = { \ + .stream_name = "MultiMedia"#num" Capture", \ + .rates = (SNDRV_PCM_RATE_8000_48000| \ + SNDRV_PCM_RATE_KNOT), \ + .formats = (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE), \ + .channels_min = 1, \ + .channels_max = 4, \ + .rate_min = 8000, \ + .rate_max = 48000, \ + }, \ + .name = "MultiMedia"#num, \ + .id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \ + } + +/* 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 const struct snd_compr_codec_caps q6asm_compr_caps = { + .num_descriptors = 1, + .descriptor[0].max_ch = 2, + .descriptor[0].sample_rates = { 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000, 88200, + 96000, 176400, 192000 }, + .descriptor[0].num_sample_rates = 13, + .descriptor[0].bit_rate[0] = 320, + .descriptor[0].bit_rate[1] = 128, + .descriptor[0].num_bitrates = 2, + .descriptor[0].profiles = 0, + .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO, + .descriptor[0].formats = 0, +}; + +static void event_handler(uint32_t opcode, uint32_t token, + void *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: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); + break; + case ASM_CLIENT_EVENT_CMD_EOS_DONE: + prtd->state = Q6ASM_STREAM_STOPPED; + break; + case ASM_CLIENT_EVENT_DATA_WRITE_DONE: { + prtd->pcm_irq_pos += prtd->pcm_count; + snd_pcm_period_elapsed(substream); + if (prtd->state == Q6ASM_STREAM_RUNNING) + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); + + break; + } + case ASM_CLIENT_EVENT_DATA_READ_DONE: + prtd->pcm_irq_pos += prtd->pcm_count; + snd_pcm_period_elapsed(substream); + if (prtd->state == Q6ASM_STREAM_RUNNING) + q6asm_read(prtd->audio_client, prtd->stream_id); + + break; + default: + break; + } +} + +static int q6asm_dai_prepare(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream); + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct q6asm_dai_data *pdata; + struct device *dev = component->dev; + int ret, i; + + pdata = snd_soc_component_get_drvdata(component); + if (!pdata) + return -EINVAL; + + if (!prtd || !prtd->audio_client) { + dev_err(dev, "%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, prtd->stream_id, CMD_CLOSE); + q6asm_unmap_memory_regions(substream->stream, + prtd->audio_client); + q6routing_stream_close(soc_prtd->dai_link->id, + substream->stream); + } + + ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client, + prtd->phys, + (prtd->pcm_size / prtd->periods), + prtd->periods); + + if (ret < 0) { + dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n", + ret); + return -ENOMEM; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, + FORMAT_LINEAR_PCM, + 0, prtd->bits_per_sample, false); + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + ret = q6asm_open_read(prtd->audio_client, prtd->stream_id, + FORMAT_LINEAR_PCM, + prtd->bits_per_sample); + } + + if (ret < 0) { + dev_err(dev, "%s: q6asm_open_write failed\n", __func__); + goto open_err; + } + + prtd->session_id = q6asm_get_session_id(prtd->audio_client); + ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE, + prtd->session_id, substream->stream); + if (ret) { + dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret); + goto routing_err; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = q6asm_media_format_block_multi_ch_pcm( + prtd->audio_client, prtd->stream_id, + runtime->rate, runtime->channels, NULL, + prtd->bits_per_sample); + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client, + prtd->stream_id, + runtime->rate, + runtime->channels, + prtd->bits_per_sample); + + /* Queue the buffers */ + for (i = 0; i < runtime->periods; i++) + q6asm_read(prtd->audio_client, prtd->stream_id); + + } + if (ret < 0) + dev_info(dev, "%s: CMD Format block failed\n", __func__); + else + prtd->state = Q6ASM_STREAM_RUNNING; + + return ret; + +routing_err: + q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); +open_err: + q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + + return ret; +} + +static int q6asm_dai_trigger(struct snd_soc_component *component, + 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: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id, + 0, 0, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + prtd->state = Q6ASM_STREAM_STOPPED; + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_EOS); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_PAUSE); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int q6asm_dai_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0); + struct q6asm_dai_rtd *prtd; + struct q6asm_dai_data *pdata; + struct device *dev = component->dev; + int ret = 0; + int stream_id; + + stream_id = cpu_dai->driver->id; + + pdata = snd_soc_component_get_drvdata(component); + if (!pdata) { + dev_err(dev, "Drv 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, + LEGACY_PCM_MODE); + if (IS_ERR(prtd->audio_client)) { + dev_info(dev, "%s: Could not allocate memory\n", __func__); + ret = PTR_ERR(prtd->audio_client); + kfree(prtd); + return ret; + } + + /* DSP expects stream id from 1 */ + prtd->stream_id = 1; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = q6asm_dai_hardware_playback; + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + runtime->hw = q6asm_dai_hardware_capture; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_sample_rates); + if (ret < 0) + dev_info(dev, "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) + dev_info(dev, "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) { + dev_err(dev, "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) { + dev_err(dev, "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) { + dev_err(dev, "constraint for buffer bytes step ret = %d\n", + ret); + } + + 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_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream); + struct q6asm_dai_rtd *prtd = runtime->private_data; + + if (prtd->audio_client) { + if (prtd->state) + q6asm_cmd(prtd->audio_client, prtd->stream_id, + CMD_CLOSE); + + q6asm_unmap_memory_regions(substream->stream, + prtd->audio_client); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + } + q6routing_stream_close(soc_prtd->dai_link->id, + substream->stream); + kfree(prtd); + return 0; +} + +static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_soc_component *component, + 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_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = component->dev; + + return dma_mmap_coherent(dev, vma, + runtime->dma_area, runtime->dma_addr, + runtime->dma_bytes); +} + +static int q6asm_dai_hw_params(struct snd_soc_component *component, + 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 void compress_event_handler(uint32_t opcode, uint32_t token, + void *payload, void *priv) +{ + struct q6asm_dai_rtd *prtd = priv; + struct snd_compr_stream *substream = prtd->cstream; + unsigned long flags; + u32 wflags = 0; + uint64_t avail; + uint32_t bytes_written, bytes_to_write; + bool is_last_buffer = false; + + switch (opcode) { + case ASM_CLIENT_EVENT_CMD_RUN_DONE: + spin_lock_irqsave(&prtd->lock, flags); + if (!prtd->bytes_sent) { + q6asm_stream_remove_initial_silence(prtd->audio_client, + prtd->stream_id, + prtd->initial_samples_drop); + + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); + prtd->bytes_sent += prtd->pcm_count; + } + + spin_unlock_irqrestore(&prtd->lock, flags); + break; + + case ASM_CLIENT_EVENT_CMD_EOS_DONE: + spin_lock_irqsave(&prtd->lock, flags); + if (prtd->notify_on_drain) { + if (substream->partial_drain) { + /* + * Close old stream and make it stale, switch + * the active stream now! + */ + q6asm_cmd_nowait(prtd->audio_client, + prtd->stream_id, + CMD_CLOSE); + /* + * vaild stream ids start from 1, So we are + * toggling this between 1 and 2. + */ + prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1); + } + + snd_compr_drain_notify(prtd->cstream); + prtd->notify_on_drain = false; + + } else { + prtd->state = Q6ASM_STREAM_STOPPED; + } + spin_unlock_irqrestore(&prtd->lock, flags); + break; + + case ASM_CLIENT_EVENT_DATA_WRITE_DONE: + spin_lock_irqsave(&prtd->lock, flags); + + bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT; + prtd->copied_total += bytes_written; + snd_compr_fragment_elapsed(substream); + + if (prtd->state != Q6ASM_STREAM_RUNNING) { + spin_unlock_irqrestore(&prtd->lock, flags); + break; + } + + avail = prtd->bytes_received - prtd->bytes_sent; + if (avail > prtd->pcm_count) { + bytes_to_write = prtd->pcm_count; + } else { + if (substream->partial_drain || prtd->notify_on_drain) + is_last_buffer = true; + bytes_to_write = avail; + } + + if (bytes_to_write) { + if (substream->partial_drain && is_last_buffer) { + wflags |= ASM_LAST_BUFFER_FLAG; + q6asm_stream_remove_trailing_silence(prtd->audio_client, + prtd->stream_id, + prtd->trailing_samples_drop); + } + + q6asm_write_async(prtd->audio_client, prtd->stream_id, + bytes_to_write, 0, 0, wflags); + + prtd->bytes_sent += bytes_to_write; + } + + if (prtd->notify_on_drain && is_last_buffer) + q6asm_cmd_nowait(prtd->audio_client, + prtd->stream_id, CMD_EOS); + + spin_unlock_irqrestore(&prtd->lock, flags); + break; + + default: + break; + } +} + +static int q6asm_dai_compr_open(struct snd_soc_component *component, + struct snd_compr_stream *stream) +{ + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct snd_compr_runtime *runtime = stream->runtime; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct q6asm_dai_data *pdata; + struct device *dev = component->dev; + struct q6asm_dai_rtd *prtd; + int stream_id, size, ret; + + stream_id = cpu_dai->driver->id; + pdata = snd_soc_component_get_drvdata(component); + if (!pdata) { + dev_err(dev, "Drv data not found ..\n"); + return -EINVAL; + } + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + return -ENOMEM; + + /* DSP expects stream id from 1 */ + prtd->stream_id = 1; + + prtd->cstream = stream; + prtd->audio_client = q6asm_audio_client_alloc(dev, + (q6asm_cb)compress_event_handler, + prtd, stream_id, LEGACY_PCM_MODE); + if (IS_ERR(prtd->audio_client)) { + dev_err(dev, "Could not allocate memory\n"); + ret = PTR_ERR(prtd->audio_client); + goto free_prtd; + } + + size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * + COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, + &prtd->dma_buffer); + if (ret) { + dev_err(dev, "Cannot allocate buffer(s)\n"); + goto free_client; + } + + if (pdata->sid < 0) + prtd->phys = prtd->dma_buffer.addr; + else + prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32); + + snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer); + spin_lock_init(&prtd->lock); + runtime->private_data = prtd; + + return 0; + +free_client: + q6asm_audio_client_free(prtd->audio_client); +free_prtd: + kfree(prtd); + + return ret; +} + +static int q6asm_dai_compr_free(struct snd_soc_component *component, + struct snd_compr_stream *stream) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = stream->private_data; + + if (prtd->audio_client) { + if (prtd->state) { + q6asm_cmd(prtd->audio_client, prtd->stream_id, + CMD_CLOSE); + if (prtd->next_track_stream_id) { + q6asm_cmd(prtd->audio_client, + prtd->next_track_stream_id, + CMD_CLOSE); + } + } + + snd_dma_free_pages(&prtd->dma_buffer); + q6asm_unmap_memory_regions(stream->direction, + prtd->audio_client); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + } + q6routing_stream_close(rtd->dai_link->id, stream->direction); + kfree(prtd); + + return 0; +} + +static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_codec *codec, + int stream_id) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct q6asm_flac_cfg flac_cfg; + struct q6asm_wma_cfg wma_cfg; + struct q6asm_alac_cfg alac_cfg; + struct q6asm_ape_cfg ape_cfg; + unsigned int wma_v9 = 0; + struct device *dev = component->dev; + int ret; + union snd_codec_options *codec_options; + struct snd_dec_flac *flac; + struct snd_dec_wma *wma; + struct snd_dec_alac *alac; + struct snd_dec_ape *ape; + + codec_options = &(prtd->codec.options); + + memcpy(&prtd->codec, codec, sizeof(*codec)); + + switch (codec->id) { + case SND_AUDIOCODEC_FLAC: + + memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg)); + flac = &codec_options->flac_d; + + flac_cfg.ch_cfg = codec->ch_in; + flac_cfg.sample_rate = codec->sample_rate; + flac_cfg.stream_info_present = 1; + flac_cfg.sample_size = flac->sample_size; + flac_cfg.min_blk_size = flac->min_blk_size; + flac_cfg.max_blk_size = flac->max_blk_size; + flac_cfg.max_frame_size = flac->max_frame_size; + flac_cfg.min_frame_size = flac->min_frame_size; + + ret = q6asm_stream_media_format_block_flac(prtd->audio_client, + stream_id, + &flac_cfg); + if (ret < 0) { + dev_err(dev, "FLAC CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + + case SND_AUDIOCODEC_WMA: + wma = &codec_options->wma_d; + + memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg)); + + wma_cfg.sample_rate = codec->sample_rate; + wma_cfg.num_channels = codec->ch_in; + wma_cfg.bytes_per_sec = codec->bit_rate / 8; + wma_cfg.block_align = codec->align; + wma_cfg.bits_per_sample = prtd->bits_per_sample; + wma_cfg.enc_options = wma->encoder_option; + wma_cfg.adv_enc_options = wma->adv_encoder_option; + wma_cfg.adv_enc_options2 = wma->adv_encoder_option2; + + if (wma_cfg.num_channels == 1) + wma_cfg.channel_mask = 4; /* Mono Center */ + else if (wma_cfg.num_channels == 2) + wma_cfg.channel_mask = 3; /* Stereo FL/FR */ + else + return -EINVAL; + + /* check the codec profile */ + switch (codec->profile) { + case SND_AUDIOPROFILE_WMA9: + wma_cfg.fmtag = 0x161; + wma_v9 = 1; + break; + + case SND_AUDIOPROFILE_WMA10: + wma_cfg.fmtag = 0x166; + break; + + case SND_AUDIOPROFILE_WMA9_PRO: + wma_cfg.fmtag = 0x162; + break; + + case SND_AUDIOPROFILE_WMA9_LOSSLESS: + wma_cfg.fmtag = 0x163; + break; + + case SND_AUDIOPROFILE_WMA10_LOSSLESS: + wma_cfg.fmtag = 0x167; + break; + + default: + dev_err(dev, "Unknown WMA profile:%x\n", + codec->profile); + return -EIO; + } + + if (wma_v9) + ret = q6asm_stream_media_format_block_wma_v9( + prtd->audio_client, stream_id, + &wma_cfg); + else + ret = q6asm_stream_media_format_block_wma_v10( + prtd->audio_client, stream_id, + &wma_cfg); + if (ret < 0) { + dev_err(dev, "WMA9 CMD failed:%d\n", ret); + return -EIO; + } + break; + + case SND_AUDIOCODEC_ALAC: + memset(&alac_cfg, 0x0, sizeof(alac_cfg)); + alac = &codec_options->alac_d; + + alac_cfg.sample_rate = codec->sample_rate; + alac_cfg.avg_bit_rate = codec->bit_rate; + alac_cfg.bit_depth = prtd->bits_per_sample; + alac_cfg.num_channels = codec->ch_in; + + alac_cfg.frame_length = alac->frame_length; + alac_cfg.pb = alac->pb; + alac_cfg.mb = alac->mb; + alac_cfg.kb = alac->kb; + alac_cfg.max_run = alac->max_run; + alac_cfg.compatible_version = alac->compatible_version; + alac_cfg.max_frame_bytes = alac->max_frame_bytes; + + switch (codec->ch_in) { + case 1: + alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO; + break; + case 2: + alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO; + break; + } + ret = q6asm_stream_media_format_block_alac(prtd->audio_client, + stream_id, + &alac_cfg); + if (ret < 0) { + dev_err(dev, "ALAC CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + + case SND_AUDIOCODEC_APE: + memset(&ape_cfg, 0x0, sizeof(ape_cfg)); + ape = &codec_options->ape_d; + + ape_cfg.sample_rate = codec->sample_rate; + ape_cfg.num_channels = codec->ch_in; + ape_cfg.bits_per_sample = prtd->bits_per_sample; + + ape_cfg.compatible_version = ape->compatible_version; + ape_cfg.compression_level = ape->compression_level; + ape_cfg.format_flags = ape->format_flags; + ape_cfg.blocks_per_frame = ape->blocks_per_frame; + ape_cfg.final_frame_blocks = ape->final_frame_blocks; + ape_cfg.total_frames = ape->total_frames; + ape_cfg.seek_table_present = ape->seek_table_present; + + ret = q6asm_stream_media_format_block_ape(prtd->audio_client, + stream_id, + &ape_cfg); + if (ret < 0) { + dev_err(dev, "APE CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + + default: + break; + } + + return 0; +} + +static int q6asm_dai_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = stream->private_data; + int dir = stream->direction; + struct q6asm_dai_data *pdata; + struct device *dev = component->dev; + int ret; + + pdata = snd_soc_component_get_drvdata(component); + if (!pdata) + return -EINVAL; + + if (!prtd || !prtd->audio_client) { + dev_err(dev, "private data null or audio client freed\n"); + return -EINVAL; + } + + prtd->periods = runtime->fragments; + prtd->pcm_count = runtime->fragment_size; + prtd->pcm_size = runtime->fragments * runtime->fragment_size; + prtd->bits_per_sample = 16; + + if (dir == SND_COMPRESS_PLAYBACK) { + ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id, + params->codec.profile, prtd->bits_per_sample, + true); + + if (ret < 0) { + dev_err(dev, "q6asm_open_write failed\n"); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + return ret; + } + } + + prtd->session_id = q6asm_get_session_id(prtd->audio_client); + ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE, + prtd->session_id, dir); + if (ret) { + dev_err(dev, "Stream reg failed ret:%d\n", ret); + return ret; + } + + ret = __q6asm_dai_compr_set_codec_params(component, stream, + ¶ms->codec, + prtd->stream_id); + if (ret) { + dev_err(dev, "codec param setup failed ret:%d\n", ret); + return ret; + } + + ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, + (prtd->pcm_size / prtd->periods), + prtd->periods); + + if (ret < 0) { + dev_err(dev, "Buffer Mapping failed ret:%d\n", ret); + return -ENOMEM; + } + + prtd->state = Q6ASM_STREAM_RUNNING; + + return 0; +} + +static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + int ret = 0; + + switch (metadata->key) { + case SNDRV_COMPRESS_ENCODER_PADDING: + prtd->trailing_samples_drop = metadata->value[0]; + break; + case SNDRV_COMPRESS_ENCODER_DELAY: + prtd->initial_samples_drop = metadata->value[0]; + if (prtd->next_track_stream_id) { + ret = q6asm_open_write(prtd->audio_client, + prtd->next_track_stream_id, + prtd->codec.id, + prtd->codec.profile, + prtd->bits_per_sample, + true); + if (ret < 0) { + dev_err(component->dev, "q6asm_open_write failed\n"); + return ret; + } + ret = __q6asm_dai_compr_set_codec_params(component, stream, + &prtd->codec, + prtd->next_track_stream_id); + if (ret < 0) { + dev_err(component->dev, "q6asm_open_write failed\n"); + return ret; + } + + ret = q6asm_stream_remove_initial_silence(prtd->audio_client, + prtd->next_track_stream_id, + prtd->initial_samples_drop); + prtd->next_track_stream_id = 0; + + } + + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int q6asm_dai_compr_trigger(struct snd_soc_component *component, + struct snd_compr_stream *stream, int cmd) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id, + 0, 0, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + prtd->state = Q6ASM_STREAM_STOPPED; + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_EOS); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_PAUSE); + break; + case SND_COMPR_TRIGGER_NEXT_TRACK: + prtd->next_track = true; + prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1); + break; + case SND_COMPR_TRIGGER_DRAIN: + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + prtd->notify_on_drain = true; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int q6asm_dai_compr_pointer(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + unsigned long flags; + + spin_lock_irqsave(&prtd->lock, flags); + + tstamp->copied_total = prtd->copied_total; + tstamp->byte_offset = prtd->copied_total % prtd->pcm_size; + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int q6asm_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *stream, char __user *buf, + size_t count) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + unsigned long flags; + u32 wflags = 0; + int avail, bytes_in_flight = 0; + void *dstn; + size_t copy; + u32 app_pointer; + u32 bytes_received; + + bytes_received = prtd->bytes_received; + + /** + * Make sure that next track data pointer is aligned at 32 bit boundary + * This is a Mandatory requirement from DSP data buffers alignment + */ + if (prtd->next_track) + bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count); + + app_pointer = bytes_received/prtd->pcm_size; + app_pointer = bytes_received - (app_pointer * prtd->pcm_size); + dstn = prtd->dma_buffer.area + app_pointer; + + if (count < prtd->pcm_size - app_pointer) { + if (copy_from_user(dstn, buf, count)) + return -EFAULT; + } else { + copy = prtd->pcm_size - app_pointer; + if (copy_from_user(dstn, buf, copy)) + return -EFAULT; + if (copy_from_user(prtd->dma_buffer.area, buf + copy, + count - copy)) + return -EFAULT; + } + + spin_lock_irqsave(&prtd->lock, flags); + + bytes_in_flight = prtd->bytes_received - prtd->copied_total; + + if (prtd->next_track) { + prtd->next_track = false; + prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count); + prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count); + } + + prtd->bytes_received = bytes_received + count; + + /* Kick off the data to dsp if its starving!! */ + if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) { + uint32_t bytes_to_write = prtd->pcm_count; + + avail = prtd->bytes_received - prtd->bytes_sent; + + if (avail < prtd->pcm_count) + bytes_to_write = avail; + + q6asm_write_async(prtd->audio_client, prtd->stream_id, + bytes_to_write, 0, 0, wflags); + prtd->bytes_sent += bytes_to_write; + } + + spin_unlock_irqrestore(&prtd->lock, flags); + + return count; +} + +static int q6asm_dai_compr_mmap(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct vm_area_struct *vma) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct device *dev = component->dev; + + return dma_mmap_coherent(dev, vma, + prtd->dma_buffer.area, prtd->dma_buffer.addr, + prtd->dma_buffer.bytes); +} + +static int q6asm_dai_compr_get_caps(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_caps *caps) +{ + caps->direction = SND_COMPRESS_PLAYBACK; + caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE; + caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; + caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; + caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; + caps->num_codecs = 5; + caps->codecs[0] = SND_AUDIOCODEC_MP3; + caps->codecs[1] = SND_AUDIOCODEC_FLAC; + caps->codecs[2] = SND_AUDIOCODEC_WMA; + caps->codecs[3] = SND_AUDIOCODEC_ALAC; + caps->codecs[4] = SND_AUDIOCODEC_APE; + + return 0; +} + +static int q6asm_dai_compr_get_codec_caps(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_codec_caps *codec) +{ + switch (codec->codec) { + case SND_AUDIOCODEC_MP3: + *codec = q6asm_compr_caps; + break; + default: + break; + } + + return 0; +} + +static struct snd_compress_ops q6asm_dai_compress_ops = { + .open = q6asm_dai_compr_open, + .free = q6asm_dai_compr_free, + .set_params = q6asm_dai_compr_set_params, + .set_metadata = q6asm_dai_compr_set_metadata, + .pointer = q6asm_dai_compr_pointer, + .trigger = q6asm_dai_compr_trigger, + .get_caps = q6asm_dai_compr_get_caps, + .get_codec_caps = q6asm_dai_compr_get_codec_caps, + .mmap = q6asm_dai_compr_mmap, + .copy = q6asm_compr_copy, +}; + +static int q6asm_dai_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm_substream *psubstream, *csubstream; + struct snd_pcm *pcm = rtd->pcm; + struct device *dev; + int size, ret; + + dev = component->dev; + size = q6asm_dai_hardware_playback.buffer_bytes_max; + psubstream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (psubstream) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, + &psubstream->dma_buffer); + if (ret) { + dev_err(dev, "Cannot allocate buffer(s)\n"); + return ret; + } + } + + csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + if (csubstream) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, + &csubstream->dma_buffer); + if (ret) { + dev_err(dev, "Cannot allocate buffer(s)\n"); + if (psubstream) + snd_dma_free_pages(&psubstream->dma_buffer); + return ret; + } + } + + return 0; +} + +static void q6asm_dai_pcm_free(struct snd_soc_component *component, + 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 const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_component_driver q6asm_fe_dai_component = { + .name = DRV_NAME, + .open = q6asm_dai_open, + .hw_params = q6asm_dai_hw_params, + .close = q6asm_dai_close, + .prepare = q6asm_dai_prepare, + .trigger = q6asm_dai_trigger, + .pointer = q6asm_dai_pointer, + .mmap = q6asm_dai_mmap, + .pcm_construct = q6asm_dai_pcm_new, + .pcm_destruct = q6asm_dai_pcm_free, + .compress_ops = &q6asm_dai_compress_ops, + .dapm_widgets = q6asm_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets), +}; + +static struct snd_soc_dai_driver q6asm_fe_dais_template[] = { + Q6ASM_FEDAI_DRIVER(1), + Q6ASM_FEDAI_DRIVER(2), + Q6ASM_FEDAI_DRIVER(3), + Q6ASM_FEDAI_DRIVER(4), + Q6ASM_FEDAI_DRIVER(5), + Q6ASM_FEDAI_DRIVER(6), + Q6ASM_FEDAI_DRIVER(7), + Q6ASM_FEDAI_DRIVER(8), +}; + +static int of_q6asm_parse_dai_data(struct device *dev, + struct q6asm_dai_data *pdata) +{ + struct snd_soc_dai_driver *dai_drv; + struct snd_soc_pcm_stream empty_stream; + struct device_node *node; + int ret, id, dir, idx = 0; + + + pdata->num_dais = of_get_child_count(dev->of_node); + if (!pdata->num_dais) { + dev_err(dev, "No dais found in DT\n"); + return -EINVAL; + } + + pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv), + GFP_KERNEL); + if (!pdata->dais) + return -ENOMEM; + + memset(&empty_stream, 0, sizeof(empty_stream)); + + for_each_child_of_node(dev->of_node, node) { + ret = of_property_read_u32(node, "reg", &id); + if (ret || id >= MAX_SESSIONS || id < 0) { + dev_err(dev, "valid dai id not found:%d\n", ret); + continue; + } + + dai_drv = &pdata->dais[idx++]; + *dai_drv = q6asm_fe_dais_template[id]; + + ret = of_property_read_u32(node, "direction", &dir); + if (ret) + continue; + + if (dir == Q6ASM_DAI_RX) + dai_drv->capture = empty_stream; + else if (dir == Q6ASM_DAI_TX) + dai_drv->playback = empty_stream; + + if (of_property_read_bool(node, "is-compress-dai")) + dai_drv->compress_new = snd_soc_new_compress; + } + + return 0; +} + +static int q6asm_dai_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct device_node *node = dev->of_node; + struct of_phandle_args args; + struct q6asm_dai_data *pdata; + int rc; + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + rc = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args); + if (rc < 0) + pdata->sid = -1; + else + pdata->sid = args.args[0] & SID_MASK_DEFAULT; + + dev_set_drvdata(dev, pdata); + + rc = of_q6asm_parse_dai_data(dev, pdata); + if (rc) + return rc; + + return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component, + pdata->dais, pdata->num_dais); +} + +#ifdef CONFIG_OF +static const struct of_device_id q6asm_dai_device_id[] = { + { .compatible = "qcom,q6asm-dais" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6asm_dai_device_id); +#endif + +static struct platform_driver q6asm_dai_platform_driver = { + .driver = { + .name = "q6asm-dai", + .of_match_table = of_match_ptr(q6asm_dai_device_id), + }, + .probe = q6asm_dai_probe, +}; +module_platform_driver(q6asm_dai_platform_driver); + +MODULE_DESCRIPTION("Q6ASM dai driver"); +MODULE_LICENSE("GPL v2"); |