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[86.26.100.93]) by smtp.gmail.com with ESMTPSA id x7-v6sm7868034wrm.35.2018.05.18.05.58.51 (version=TLS1_2 cipher=ECDHE-RSA-AES128-SHA bits=128/128); Fri, 18 May 2018 05:58:52 -0700 (PDT) From: Srinivas Kandagatla To: broonie@kernel.org, linux-arm-msm@vger.kernel.org, alsa-devel@alsa-project.org, bgoswami@codeaurora.org Date: Fri, 18 May 2018 13:56:08 +0100 Message-Id: <20180518125610.26200-14-srinivas.kandagatla@linaro.org> X-Mailer: git-send-email 2.16.2 In-Reply-To: <20180518125610.26200-1-srinivas.kandagatla@linaro.org> References: <20180518125610.26200-1-srinivas.kandagatla@linaro.org> Cc: mark.rutland@arm.com, devicetree@vger.kernel.org, rohkumar@qti.qualcomm.com, gregkh@linuxfoundation.org, plai@codeaurora.org, lgirdwood@gmail.com, tiwai@suse.com, david.brown@linaro.org, robh+dt@kernel.org, Srinivas Kandagatla , linux-arm-kernel@lists.infradead.org, spatakok@qti.qualcomm.com, linux-kernel@vger.kernel.org Subject: [alsa-devel] [PATCH v9 13/15] ASoC: qdsp6: q6asm: Add q6asm dai driver X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.14 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , MIME-Version: 1.0 Errors-To: alsa-devel-bounces@alsa-project.org Sender: alsa-devel-bounces@alsa-project.org X-Virus-Scanned: ClamAV using ClamSMTP This patch adds support to q6asm dai driver which configures Q6ASM streams to pass pcm data. Signed-off-by: Srinivas Kandagatla Reviewed-and-tested-by: Rohit kumar Reviewed-by: Banajit Goswami --- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 624 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 629 insertions(+) create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index d3523a30d942..85bb7dd11fd9 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -62,6 +62,9 @@ config SND_SOC_QDSP6_ROUTING config SND_SOC_QDSP6_ASM tristate +config SND_SOC_QDSP6_ASM_DAI + tristate + config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" depends on QCOM_APR && HAS_DMA @@ -72,6 +75,7 @@ config SND_SOC_QDSP6 select SND_SOC_QDSP6_ADM select SND_SOC_QDSP6_ROUTING select SND_SOC_QDSP6_ASM + 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 bada1aa303c2..c33b3cacbea1 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -5,3 +5,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.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..349c6a883c63 --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -0,0 +1,624 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. +// Copyright (c) 2018, Linaro Limited + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#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 + +enum stream_state { + Q6ASM_STREAM_IDLE = 0, + Q6ASM_STREAM_STOPPED, + Q6ASM_STREAM_RUNNING, +}; + +struct q6asm_dai_rtd { + struct snd_pcm_substream *substream; + phys_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; +}; + +struct q6asm_dai_data { + long long int sid; +}; + +static struct snd_pcm_hardware q6asm_dai_hardware_capture = { + .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_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_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, \ + .probe = fe_dai_probe, \ + .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 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: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + 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_irq_pos += prtd->pcm_count; + snd_pcm_period_elapsed(substream); + if (prtd->state == Q6ASM_STREAM_RUNNING) + q6asm_write_async(prtd->audio_client, + prtd->pcm_count, 0, 0, NO_TIMESTAMP); + + 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); + + 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 snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME); + struct q6asm_dai_data *pdata; + int ret, i; + + pdata = snd_soc_component_get_drvdata(c); + 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_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) { + pr_err("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, FORMAT_LINEAR_PCM, + prtd->bits_per_sample); + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + ret = q6asm_open_read(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_stream_open(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; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = q6asm_media_format_block_multi_ch_pcm( + prtd->audio_client, 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, + runtime->rate, runtime->channels, + prtd->bits_per_sample); + + /* Queue the buffers */ + for (i = 0; i < runtime->periods; i++) + q6asm_read(prtd->audio_client); + + } + if (ret < 0) + pr_info("%s: CMD Format block failed\n", __func__); + + prtd->state = Q6ASM_STREAM_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: + 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 snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME); + struct q6asm_dai_rtd *prtd; + struct q6asm_dai_data *pdata; + struct device *dev = c->dev; + int ret = 0; + int stream_id; + + stream_id = cpu_dai->driver->id; + + pdata = snd_soc_component_get_drvdata(c); + if (!pdata) { + pr_err("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 (!prtd->audio_client) { + pr_info("%s: Could not allocate memory\n", __func__); + kfree(prtd); + return -ENOMEM; + } + + 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) + 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); + } + + 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); + 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_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_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME); + struct device *dev = c->dev; + + return dma_mmap_coherent(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_substream *psubstream, *csubstream; + struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_pcm *pcm = rtd->pcm; + struct device *dev; + int size, ret; + + dev = c->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 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 const struct snd_soc_dapm_route afe_pcm_routes[] = { + {"MM_DL1", NULL, "MultiMedia1 Playback" }, + {"MM_DL2", NULL, "MultiMedia2 Playback" }, + {"MM_DL3", NULL, "MultiMedia3 Playback" }, + {"MM_DL4", NULL, "MultiMedia4 Playback" }, + {"MM_DL5", NULL, "MultiMedia5 Playback" }, + {"MM_DL6", NULL, "MultiMedia6 Playback" }, + {"MM_DL7", NULL, "MultiMedia7 Playback" }, + {"MM_DL7", NULL, "MultiMedia8 Playback" }, + {"MultiMedia1 Capture", NULL, "MM_UL1"}, + {"MultiMedia2 Capture", NULL, "MM_UL2"}, + {"MultiMedia3 Capture", NULL, "MM_UL3"}, + {"MultiMedia4 Capture", NULL, "MM_UL4"}, + {"MultiMedia5 Capture", NULL, "MM_UL5"}, + {"MultiMedia6 Capture", NULL, "MM_UL6"}, + {"MultiMedia7 Capture", NULL, "MM_UL7"}, + {"MultiMedia8 Capture", NULL, "MM_UL8"}, + +}; + +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 = DRV_NAME, + .ops = &q6asm_dai_ops, + .pcm_new = q6asm_dai_pcm_new, + .pcm_free = q6asm_dai_pcm_free, + +}; + +static struct snd_soc_dai_driver q6asm_fe_dais[] = { + 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 q6asm_dai_bind(struct device *dev, struct device *master, void *data) +{ + struct device_node *node = dev->of_node; + struct of_phandle_args args; + struct q6asm_dai_data *pdata; + int rc; + + pdata = kzalloc(sizeof(struct q6asm_dai_data), 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); + + return snd_soc_register_component(dev, &q6asm_fe_dai_component, + q6asm_fe_dais, + ARRAY_SIZE(q6asm_fe_dais)); +} +static void q6asm_dai_unbind(struct device *dev, struct device *master, + void *data) +{ + struct q6asm_dai_data *pdata = dev_get_drvdata(dev); + + snd_soc_unregister_component(dev); + + kfree(pdata); + +} + +static const struct component_ops q6asm_dai_comp_ops = { + .bind = q6asm_dai_bind, + .unbind = q6asm_dai_unbind, +}; + +static int q6asm_dai_probe(struct platform_device *pdev) +{ + return component_add(&pdev->dev, &q6asm_dai_comp_ops); +} + +static int q6asm_dai_dev_remove(struct platform_device *pdev) +{ + component_del(&pdev->dev, &q6asm_dai_comp_ops); + return 0; +} + +static struct platform_driver q6asm_dai_platform_driver = { + .driver = { + .name = "q6asm-dai", + }, + .probe = q6asm_dai_probe, + .remove = q6asm_dai_dev_remove, +}; +module_platform_driver(q6asm_dai_platform_driver); + +MODULE_DESCRIPTION("Q6ASM dai driver"); +MODULE_LICENSE("GPL v2");