diff options
Diffstat (limited to 'sound/soc/qcom/qdsp6/q6asm-dai.c')
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6asm-dai.c | 916 |
1 files changed, 658 insertions, 258 deletions
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 5b986b74dd36..709b4f3318ff 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -2,9 +2,11 @@ // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited +#include <dt-bindings/sound/qcom,q6asm.h> #include <linux/init.h> #include <linux/err.h> #include <linux/module.h> +#include <linux/of.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> @@ -12,9 +14,9 @@ #include <sound/pcm.h> #include <linux/spinlock.h> #include <sound/compress_driver.h> +#include <asm/div64.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" @@ -37,9 +39,9 @@ #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) -#define Q6ASM_DAI_TX_RX 0 -#define Q6ASM_DAI_TX 1 -#define Q6ASM_DAI_RX 2 + +#define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) +#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) enum stream_state { Q6ASM_STREAM_IDLE = 0, @@ -50,31 +52,40 @@ enum stream_state { struct q6asm_dai_rtd { struct snd_pcm_substream *substream; struct snd_compr_stream *cstream; - struct snd_compr_params codec_param; + 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; + uint64_t bytes_sent; + uint64_t bytes_received; + uint64_t copied_total; uint16_t bits_per_sample; + snd_pcm_uframes_t queue_ptr; 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 struct snd_pcm_hardware q6asm_dai_hardware_capture = { - .info = (SNDRV_PCM_INFO_MMAP | +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_NO_REWINDS | SNDRV_PCM_INFO_SYNC_APPLPTR | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), @@ -94,10 +105,11 @@ static struct snd_pcm_hardware q6asm_dai_hardware_capture = { .fifo_size = 0, }; -static struct snd_pcm_hardware q6asm_dai_hardware_playback = { - .info = (SNDRV_PCM_INFO_MMAP | +static const 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_NO_REWINDS | SNDRV_PCM_INFO_SYNC_APPLPTR | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -119,8 +131,13 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = { #define Q6ASM_FEDAI_DRIVER(num) { \ .playback = { \ .stream_name = "MultiMedia"#num" Playback", \ - .rates = (SNDRV_PCM_RATE_8000_192000| \ - SNDRV_PCM_RATE_KNOT), \ + .rates = (SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_12000 | \ + SNDRV_PCM_RATE_24000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000), \ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE), \ .channels_min = 1, \ @@ -130,8 +147,9 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = { }, \ .capture = { \ .stream_name = "MultiMedia"#num" Capture", \ - .rates = (SNDRV_PCM_RATE_8000_48000| \ - SNDRV_PCM_RATE_KNOT), \ + .rates = (SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_12000 | \ + SNDRV_PCM_RATE_24000), \ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE), \ .channels_min = 1, \ @@ -143,18 +161,6 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = { .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, @@ -171,34 +177,25 @@ static const struct snd_compr_codec_caps q6asm_compr_caps = { }; static void event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + 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->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); + q6asm_read(prtd->audio_client, prtd->stream_id); break; default: @@ -206,35 +203,36 @@ static void event_handler(uint32_t opcode, uint32_t token, } } -static int q6asm_dai_prepare(struct snd_pcm_substream *substream) +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 = substream->private_data; + struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream); 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; + struct device *dev = component->dev; int ret, i; - pdata = snd_soc_component_get_drvdata(c); + pdata = snd_soc_component_get_drvdata(component); if (!pdata) return -EINVAL; if (!prtd || !prtd->audio_client) { - pr_err("%s: private data null or audio client freed\n", + 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) { + if (prtd->state == Q6ASM_STREAM_RUNNING) { /* clear the previous setup if any */ - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + 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); + prtd->state = Q6ASM_STREAM_STOPPED; } ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client, @@ -243,58 +241,93 @@ static int q6asm_dai_prepare(struct snd_pcm_substream *substream) prtd->periods); if (ret < 0) { - pr_err("Audio Start: Buffer Allocation failed rc = %d\n", + 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, FORMAT_LINEAR_PCM, - prtd->bits_per_sample); + 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, FORMAT_LINEAR_PCM, - prtd->bits_per_sample); + ret = q6asm_open_read(prtd->audio_client, prtd->stream_id, + 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; + 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) { - pr_err("%s: stream reg failed ret:%d\n", __func__, ret); - return 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, runtime->rate, - runtime->channels, NULL, + 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, - runtime->rate, runtime->channels, - prtd->bits_per_sample); + 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); + q6asm_read(prtd->audio_client, prtd->stream_id); } if (ret < 0) - pr_info("%s: CMD Format block failed\n", __func__); + dev_info(dev, "%s: CMD Format block failed\n", __func__); + else + prtd->state = Q6ASM_STREAM_RUNNING; - prtd->state = Q6ASM_STREAM_RUNNING; + return ret; - return 0; +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_ack(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; + int i, ret = 0, avail_periods; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && prtd->state == Q6ASM_STREAM_RUNNING) { + avail_periods = (runtime->control->appl_ptr - prtd->queue_ptr)/runtime->period_size; + for (i = 0; i < avail_periods; i++) { + ret = q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); + + if (ret < 0) { + dev_err(component->dev, "Error queuing playback buffer %d\n", ret); + return ret; + } + prtd->queue_ptr += runtime->period_size; + } + } + + return ret; } -static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int cmd) +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; @@ -304,15 +337,18 @@ static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int 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); + 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, CMD_EOS); + 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, CMD_PAUSE); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_PAUSE); break; default: ret = -EINVAL; @@ -322,23 +358,23 @@ static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static int q6asm_dai_open(struct snd_pcm_substream *substream) +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 = 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 snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(soc_prtd, 0); struct q6asm_dai_rtd *prtd; struct q6asm_dai_data *pdata; - struct device *dev = c->dev; + struct device *dev = component->dev; int ret = 0; int stream_id; stream_id = cpu_dai->driver->id; - pdata = snd_soc_component_get_drvdata(c); + pdata = snd_soc_component_get_drvdata(component); if (!pdata) { - pr_err("Drv data not found ..\n"); + dev_err(dev, "Drv data not found ..\n"); return -EINVAL; } @@ -351,27 +387,25 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream) (q6asm_cb)event_handler, prtd, stream_id, LEGACY_PCM_MODE); if (IS_ERR(prtd->audio_client)) { - pr_info("%s: Could not allocate memory\n", __func__); + 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) - 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"); + dev_info(dev, "snd_pcm_hw_constraint_integer failed\n"); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = snd_pcm_hw_constraint_minmax(runtime, @@ -379,50 +413,53 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream) 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); + 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); + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 480); if (ret < 0) { - pr_err("constraint for period bytes step ret = %d\n", + 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); + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 480); if (ret < 0) { - pr_err("constraint for buffer bytes step ret = %d\n", + 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 (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback); + runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max; + } else { + snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_capture); + runtime->dma_bytes = q6asm_dai_hardware_capture.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) +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 = substream->private_data; + struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream); struct q6asm_dai_rtd *prtd = runtime->private_data; if (prtd->audio_client) { if (prtd->state) - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_cmd(prtd->audio_client, prtd->stream_id, + CMD_CLOSE); q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); @@ -435,34 +472,24 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream) return 0; } -static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_pcm_substream *substream) +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; + snd_pcm_uframes_t ptr; - if (prtd->pcm_irq_pos >= prtd->pcm_size) - prtd->pcm_irq_pos = 0; - - return bytes_to_frames(runtime, (prtd->pcm_irq_pos)); -} + ptr = q6asm_get_hw_pointer(prtd->audio_client, substream->stream) * runtime->period_size; + if (ptr) + return ptr - 1; -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); + return 0; } -static int q6asm_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +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; @@ -482,61 +509,93 @@ static int q6asm_dai_hw_params(struct snd_pcm_substream *substream, 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 void compress_event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + 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; + + guard(spinlock_irqsave)(&prtd->lock); switch (opcode) { case ASM_CLIENT_EVENT_CMD_RUN_DONE: - spin_lock_irqsave(&prtd->lock, flags); if (!prtd->bytes_sent) { - q6asm_write_async(prtd->audio_client, prtd->pcm_count, - 0, 0, NO_TIMESTAMP); + 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: - prtd->state = Q6ASM_STREAM_STOPPED; + 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; + } break; case ASM_CLIENT_EVENT_DATA_WRITE_DONE: - spin_lock_irqsave(&prtd->lock, flags); - prtd->copied_total += prtd->pcm_count; + 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); + if (prtd->state != Q6ASM_STREAM_RUNNING) 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 (avail >= prtd->pcm_count) { - q6asm_write_async(prtd->audio_client, - prtd->pcm_count, 0, 0, NO_TIMESTAMP); - prtd->bytes_sent += prtd->pcm_count; + 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; } - spin_unlock_irqrestore(&prtd->lock, flags); + if (prtd->notify_on_drain && is_last_buffer) + q6asm_cmd_nowait(prtd->audio_client, + prtd->stream_id, CMD_EOS); + break; default: @@ -544,19 +603,19 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, } } -static int q6asm_dai_compr_open(struct snd_compr_stream *stream) +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_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct snd_compr_runtime *runtime = stream->runtime; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct q6asm_dai_data *pdata; - struct device *dev = c->dev; + 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(c); + pdata = snd_soc_component_get_drvdata(component); if (!pdata) { dev_err(dev, "Drv data not found ..\n"); return -EINVAL; @@ -566,14 +625,17 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) 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 (!prtd->audio_client) { + if (IS_ERR(prtd->audio_client)) { dev_err(dev, "Could not allocate memory\n"); - kfree(prtd); - return -ENOMEM; + ret = PTR_ERR(prtd->audio_client); + goto free_prtd; } size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * @@ -582,7 +644,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) &prtd->dma_buffer); if (ret) { dev_err(dev, "Cannot allocate buffer(s)\n"); - return ret; + goto free_client; } if (pdata->sid < 0) @@ -595,17 +657,32 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) 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_compr_stream *stream) +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, CMD_CLOSE); + 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, @@ -619,21 +696,197 @@ static int q6asm_dai_compr_free(struct snd_compr_stream *stream) return 0; } -static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, +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; - struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); int dir = stream->direction; struct q6asm_dai_data *pdata; - struct device *dev = c->dev; + struct device *dev = component->dev; int ret; - memcpy(&prtd->codec_param, params, sizeof(*params)); - - pdata = snd_soc_component_get_drvdata(c); + pdata = snd_soc_component_get_drvdata(component); if (!pdata) return -EINVAL; @@ -646,15 +899,15 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, 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, params->codec.id, - prtd->bits_per_sample); + 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; + goto open_err; } } @@ -663,7 +916,15 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, prtd->session_id, dir); if (ret) { dev_err(dev, "Stream reg failed ret:%d\n", ret); - return ret; + goto q6_err; + } + + 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); + goto q6_err; } ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, @@ -672,15 +933,74 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, if (ret < 0) { dev_err(dev, "Buffer Mapping failed ret:%d\n", ret); - return -ENOMEM; + ret = -ENOMEM; + goto q6_err; } prtd->state = Q6ASM_STREAM_RUNNING; return 0; + +q6_err: + q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); + +open_err: + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + return ret; +} + +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_compr_stream *stream, int cmd) +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; @@ -690,15 +1010,26 @@ static int q6asm_dai_compr_trigger(struct snd_compr_stream *stream, int 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); + 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, CMD_EOS); + 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, CMD_PAUSE); + 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; @@ -708,52 +1039,108 @@ static int q6asm_dai_compr_trigger(struct snd_compr_stream *stream, int cmd) return ret; } -static int q6asm_dai_compr_pointer(struct snd_compr_stream *stream, - struct snd_compr_tstamp *tstamp) +static int q6asm_dai_compr_pointer(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_tstamp64 *tstamp) { struct snd_compr_runtime *runtime = stream->runtime; struct q6asm_dai_rtd *prtd = runtime->private_data; - unsigned long flags; + uint64_t temp_copied_total; - spin_lock_irqsave(&prtd->lock, flags); + guard(spinlock_irqsave)(&prtd->lock); tstamp->copied_total = prtd->copied_total; - tstamp->byte_offset = prtd->copied_total % prtd->pcm_size; - - spin_unlock_irqrestore(&prtd->lock, flags); + temp_copied_total = tstamp->copied_total; + tstamp->byte_offset = do_div(temp_copied_total, prtd->pcm_size); return 0; } -static int q6asm_dai_compr_ack(struct snd_compr_stream *stream, - size_t count) +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; + uint64_t avail, bytes_in_flight = 0; + void *dstn; + size_t copy; + u32 app_pointer; + uint64_t bytes_received; + uint64_t temp_bytes_received; + + bytes_received = prtd->bytes_received; + temp_bytes_received = 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); + temp_bytes_received = bytes_received; + } + + app_pointer = do_div(temp_bytes_received, 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; + } + + guard(spinlock_irqsave)(&prtd->lock); + + 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; - spin_lock_irqsave(&prtd->lock, flags); - prtd->bytes_received += count; - spin_unlock_irqrestore(&prtd->lock, flags); + /* 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; + } return count; } -static int q6asm_dai_compr_mmap(struct snd_compr_stream *stream, - struct vm_area_struct *vma) +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 snd_soc_pcm_runtime *rtd = stream->private_data; - struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); - struct device *dev = c->dev; + 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_compr_stream *stream, +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; @@ -761,13 +1148,18 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream, 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 = 1; + 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_compr_stream *stream, +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) { @@ -781,77 +1173,65 @@ static int q6asm_dai_compr_get_codec_caps(struct snd_compr_stream *stream, return 0; } -static struct snd_compr_ops q6asm_dai_compr_ops = { +static const 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, - .ack = q6asm_dai_compr_ack, + .copy = q6asm_compr_copy, }; -static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) +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_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; - } - } + size_t size = q6asm_dai_hardware_playback.buffer_bytes_max; - 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; + return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + component->dev, size); } -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_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, - .ops = &q6asm_dai_ops, - .pcm_new = q6asm_dai_pcm_new, - .pcm_free = q6asm_dai_pcm_free, - .compr_ops = &q6asm_dai_compr_ops, + .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, + .ack = q6asm_dai_ack, + .pointer = q6asm_dai_pointer, + .pcm_construct = q6asm_dai_pcm_new, + .compress_ops = &q6asm_dai_compress_ops, + .dapm_widgets = q6asm_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets), + .legacy_dai_naming = 1, }; -static struct snd_soc_dai_driver q6asm_fe_dais[] = { +static struct snd_soc_dai_driver q6asm_fe_dais_template[] = { Q6ASM_FEDAI_DRIVER(1), Q6ASM_FEDAI_DRIVER(2), Q6ASM_FEDAI_DRIVER(3), @@ -862,24 +1242,41 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = { Q6ASM_FEDAI_DRIVER(8), }; +static const struct snd_soc_dai_ops q6asm_dai_ops = { + .compress_new = snd_soc_new_compress, +}; + static int of_q6asm_parse_dai_data(struct device *dev, struct q6asm_dai_data *pdata) { - static struct snd_soc_dai_driver *dai_drv; + struct snd_soc_dai_driver *dai_drv; struct snd_soc_pcm_stream empty_stream; struct device_node *node; - int ret, id, dir; + 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) { + if (ret || id >= MAX_SESSIONS || id < 0) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } - dai_drv = &q6asm_fe_dais[id]; + dai_drv = &pdata->dais[idx++]; + *dai_drv = q6asm_fe_dais_template[id]; ret = of_property_read_u32(node, "direction", &dir); if (ret) @@ -891,7 +1288,7 @@ static int of_q6asm_parse_dai_data(struct device *dev, dai_drv->playback = empty_stream; if (of_property_read_bool(node, "is-compress-dai")) - dai_drv->compress_new = snd_soc_new_compress; + dai_drv->ops = &q6asm_dai_ops; } return 0; @@ -917,18 +1314,21 @@ static int q6asm_dai_probe(struct platform_device *pdev) dev_set_drvdata(dev, pdata); - of_q6asm_parse_dai_data(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, - q6asm_fe_dais, - ARRAY_SIZE(q6asm_fe_dais)); + 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 = { |
