Allows driver to be built on wider configs

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c30fedb..c6fe1e4 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -1,6 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" - depends on INTEL_SCU_IPC + depends on (INTEL_SCU_IPC || COMPILE_TEST) select SND_SOC_SN95031 select SND_SST_MFLD_PLATFORM help

This helps us to handle pcm and compress ops seperately and per dai

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform-pcm.c | 112 +++++++++++++++++------------- 1 files changed, 63 insertions(+), 49 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7c790f5..0d46005 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -230,19 +230,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) } /* end -- helper functions */

-static int sst_platform_open(struct snd_pcm_substream *substream) +static int sst_media_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + int ret_val = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; - int ret_val; - - pr_debug("sst_platform_open called\n"); - - snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); - ret_val = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret_val < 0) - return ret_val;

stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) @@ -251,50 +244,54 @@ static int sst_platform_open(struct snd_pcm_substream *substream)

/* get the sst ops */ mutex_lock(&sst_lock); - if (!sst) { + if (!sst || + !try_module_get(sst->dev->driver->owner)) { pr_err("no device available to run\n"); - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - if (!try_module_get(sst->dev->driver->owner)) { - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; + ret_val = -ENODEV; + goto out_ops; } stream->ops = sst->ops; mutex_unlock(&sst_lock);

stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; /* allocate memory for SST API set */ runtime->private_data = stream;

- return 0; + /* Make sure, that the period size is always even */ + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIODS, 2); + + return snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); +out_ops: + kfree(stream); + mutex_unlock(&sst_lock); + return ret_val; }

-static int sst_platform_close(struct snd_pcm_substream *substream) +static void sst_media_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id;

- pr_debug("sst_platform_close called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->ops->close(str_id); module_put(sst->dev->driver->owner); kfree(stream); - return ret_val; + return; }

-static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +static int sst_media_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id;

- pr_debug("sst_platform_pcm_prepare called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { @@ -316,6 +313,41 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) return ret_val; }

+static int sst_media_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + return 0; +} + +static int sst_media_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_soc_dai_ops sst_media_dai_ops = { + .startup = sst_media_open, + .shutdown = sst_media_close, + .prepare = sst_media_prepare, + .hw_params = sst_media_hw_params, + .hw_free = sst_media_hw_free, +}; + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (substream->pcm->internal) + return 0; + + runtime = substream->runtime; + runtime->hw = sst_platform_pcm_hw; + return 0; +} + static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -377,32 +409,14 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } - return stream->stream_info.buffer_ptr; -} - -static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - - return 0; -} - -static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); + return str_info->buffer_ptr; }

static struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, - .close = sst_platform_close, .ioctl = snd_pcm_lib_ioctl, - .prepare = sst_platform_pcm_prepare, .trigger = sst_platform_pcm_trigger, .pointer = sst_platform_pcm_pointer, - .hw_params = sst_platform_pcm_hw_params, - .hw_free = sst_platform_pcm_hw_free, };

static void sst_pcm_free(struct snd_pcm *pcm) @@ -413,15 +427,15 @@ static void sst_pcm_free(struct snd_pcm *pcm)

static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int retval = 0;

- pr_debug("sst_pcm_new called\n"); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + if (dai->driver->playback.channels_min || + dai->driver->capture.channels_min) { retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { pr_err("dma buffer allocationf fail\n");

The argument was called mad_substream which is no longer apt as older driver is not used anymore so rename as arg

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform-pcm.c | 10 +++++----- sound/soc/intel/sst-mfld-platform.h | 4 ++-- 2 files changed, 7 insertions(+), 7 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 0d46005..4528946 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -192,9 +192,9 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) return ret_val; }

-static void sst_period_elapsed(void *mad_substream) +static void sst_period_elapsed(void *arg) { - struct snd_pcm_substream *substream = mad_substream; + struct snd_pcm_substream *substream = arg; struct sst_runtime_stream *stream; int status;

@@ -218,7 +218,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) pr_debug("setting buffer ptr param\n"); sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; ret_val = stream->ops->device_control( @@ -255,7 +255,7 @@ static int sst_media_open(struct snd_pcm_substream *substream,

stream->stream_info.str_id = 0;

- stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; /* allocate memory for SST API set */ runtime->private_data = stream;

@@ -363,7 +363,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, pr_debug("sst: Trigger Start\n"); str_cmd = SST_SND_START; status = SST_PLATFORM_RUNNING; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c5e7dc..6d929c7 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -39,8 +39,8 @@ extern struct sst_device *sst;

struct pcm_stream_info { int str_id; - void *mad_substream; - void (*period_elapsed) (void *mad_substream); + void *arg; + void (*period_elapsed) (void *arg); unsigned long long buffer_ptr; int sfreq; };

Now the DSP is capable of reporting the delay, report it to upper layers

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform-pcm.c | 1 + sound/soc/intel/sst-mfld-platform.h | 1 + 2 files changed, 2 insertions(+), 0 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 4528946..80879e5 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -409,6 +409,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } + substream->runtime->delay = str_info->pcm_delay; return str_info->buffer_ptr; }

diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6d929c7..33a0a27 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -42,6 +42,7 @@ struct pcm_stream_info { void *arg; void (*period_elapsed) (void *arg); unsigned long long buffer_ptr; + unsigned long long pcm_delay; int sfreq; };

This will be used to update current driver as well as in support for the mrfld patches

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-dsp.h | 414 ++++++++++++++++++++++++++++++++++++++- 1 files changed, 403 insertions(+), 11 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h index 8d482d7..2c88785 100644 --- a/sound/soc/intel/sst-mfld-dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -3,7 +3,7 @@ /* * sst_mfld_dsp.h - Intel SST Driver for audio engine * - * Copyright (C) 2008-12 Intel Corporation + * Copyright (C) 2008-14 Intel Corporation * Authors: Vinod Koul vinod.koul@linux.intel.com * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * @@ -19,6 +19,142 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */

+#define SST_MAX_BIN_BYTES 1024 + +#define MAX_DBG_RW_BYTES 80 +#define MAX_NUM_SCATTER_BUFFERS 8 +#define MAX_LOOP_BACK_DWORDS 8 +/* IPC base address and mailbox, timestamp offsets */ +#define SST_MAILBOX_SIZE 0x0400 +#define SST_MAILBOX_SEND 0x0000 +#define SST_TIME_STAMP 0x1800 +#define SST_TIME_STAMP_MRFLD 0x800 +#define SST_RESERVED_OFFSET 0x1A00 +#define SST_SCU_LPE_MAILBOX 0x1000 +#define SST_LPE_SCU_MAILBOX 0x1400 +#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16) +#define PROCESS_MSG 0x80 + +/* Message ID's for IPC messages */ +/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */ + +/* I2L Firmware/Codec Download msgs */ +#define IPC_IA_PREP_LIB_DNLD 0x01 +#define IPC_IA_LIB_DNLD_CMPLT 0x02 +#define IPC_IA_GET_FW_VERSION 0x04 +#define IPC_IA_GET_FW_BUILD_INF 0x05 +#define IPC_IA_GET_FW_INFO 0x06 +#define IPC_IA_GET_FW_CTXT 0x07 +#define IPC_IA_SET_FW_CTXT 0x08 +#define IPC_IA_PREPARE_SHUTDOWN 0x31 +/* I2L Codec Config/control msgs */ +#define IPC_PREP_D3 0x10 +#define IPC_IA_SET_CODEC_PARAMS 0x10 +#define IPC_IA_GET_CODEC_PARAMS 0x11 +#define IPC_IA_SET_PPP_PARAMS 0x12 +#define IPC_IA_GET_PPP_PARAMS 0x13 +#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA +#define IPC_IA_ALG_PARAMS 0x1A +#define IPC_IA_TUNING_PARAMS 0x1B +#define IPC_IA_SET_RUNTIME_PARAMS 0x1C +#define IPC_IA_SET_PARAMS 0x1 +#define IPC_IA_GET_PARAMS 0x2 + +#define IPC_EFFECTS_CREATE 0xE +#define IPC_EFFECTS_DESTROY 0xF + +/* I2L Stream config/control msgs */ +#define IPC_IA_ALLOC_STREAM_MRFLD 0x2 +#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */ +#define IPC_IA_FREE_STREAM_MRFLD 0x03 +#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */ +#define IPC_IA_SET_STREAM_PARAMS 0x22 +#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12 +#define IPC_IA_GET_STREAM_PARAMS 0x23 +#define IPC_IA_PAUSE_STREAM 0x24 +#define IPC_IA_PAUSE_STREAM_MRFLD 0x4 +#define IPC_IA_RESUME_STREAM 0x25 +#define IPC_IA_RESUME_STREAM_MRFLD 0x5 +#define IPC_IA_DROP_STREAM 0x26 +#define IPC_IA_DROP_STREAM_MRFLD 0x07 +#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */ +#define IPC_IA_DRAIN_STREAM_MRFLD 0x8 +#define IPC_IA_CONTROL_ROUTING 0x29 +#define IPC_IA_VTSV_UPDATE_MODULES 0x20 +#define IPC_IA_VTSV_DETECTED 0x21 + +#define IPC_IA_START_STREAM_MRFLD 0X06 +#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */ + +#define IPC_IA_SET_GAIN_MRFLD 0x21 +/* Debug msgs */ +#define IPC_IA_DBG_MEM_READ 0x40 +#define IPC_IA_DBG_MEM_WRITE 0x41 +#define IPC_IA_DBG_LOOP_BACK 0x42 +#define IPC_IA_DBG_LOG_ENABLE 0x45 +#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47 + +/* L2I Firmware/Codec Download msgs */ +#define IPC_IA_FW_INIT_CMPLT 0x81 +#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01 +#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11 + +/* L2I Codec Config/control msgs */ +#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */ + +#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */ +#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */ +#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */ +#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */ +#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */ +#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */ + +#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */ +/* L2S messages */ +#define IPC_SC_DDR_LINK_UP 0xC0 +#define IPC_SC_DDR_LINK_DOWN 0xC1 +#define IPC_SC_SET_LPECLK_REQ 0xC2 +#define IPC_SC_SSP_BIT_BANG 0xC3 + +/* L2I Error reporting msgs */ +#define IPC_IA_MEM_ALLOC_FAIL 0xE0 +#define IPC_IA_PROC_ERR 0xE1 /* error in processing a + stream can be used by playback and + capture modules */ + +/* L2I Debug msgs */ +#define IPC_IA_PRINT_STRING 0xF0 + +/* Buffer under-run */ +#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B + +/* Mrfld specific defines: + * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR) + * received from FW, the format is: + * - IPC High: pvt_id is set to zero. Always short message. + * - msg_id is in lower 16-bits of IPC low payload. + * - pipe_id is in higher 16-bits of IPC low payload for period_elapsed. + * - error id is in higher 16-bits of IPC low payload for async errors. + */ +#define SST_ASYNC_DRV_ID 0 + +/* Command Response or Acknowledge message to any IPC message will have + * same message ID and stream ID information which is sent. + * There is no specific Ack message ID. The data field is used as response + * meaning. + */ +enum ackData { + IPC_ACK_SUCCESS = 0, + IPC_ACK_FAILURE, +}; + +enum ipc_ia_msg_id { + IPC_CMD = 1, /*!< Task Control message ID */ + IPC_SET_PARAMS = 2,/*!< Task Set param message ID */ + IPC_GET_PARAMS = 3, /*!< Task Get param message ID */ + IPC_INVALID = 0xFF, /*!<Task Get param message ID */ +}; + enum sst_codec_types { /* AUDIO/MUSIC CODEC Type Definitions */ SST_CODEC_TYPE_UNKNOWN = 0, @@ -35,14 +171,142 @@ enum stream_type { SST_STREAM_TYPE_MUSIC = 1, };

+struct ipc_dsp_hdr { + u16 mod_index_id:8; /*!< DSP Command ID specific to tasks */ + u16 pipe_id:8; /*!< instance of the module in the pipeline */ + u16 mod_id; /*!< Pipe_id */ + u16 cmd_id; /*!< Module ID = lpe_algo_types_t */ + u16 length; /*!< Length of the payload only */ +} __packed; + +union ipc_header_high { + struct { + u32 msg_id:8; /* Message ID - Max 256 Message Types */ + u32 task_id:4; /* Task ID associated with this comand */ + u32 drv_id:4; /* Identifier for the driver to track*/ + u32 rsvd1:8; /* Reserved */ + u32 result:4; /* Reserved */ + u32 res_rqd:1; /* Response rqd */ + u32 large:1; /* Large Message if large = 1 */ + u32 done:1; /* bit 30 - Done bit */ + u32 busy:1; /* bit 31 - busy bit*/ + } part; + u32 full; +} __packed; +/* IPC header */ +union ipc_header_mrfld { + struct { + u32 header_low_payload; + union ipc_header_high header_high; + } p; + u64 full; +} __packed; +/* CAUTION NOTE: All IPC message body must be multiple of 32 bits.*/ + +/* IPC Header */ +union ipc_header { + struct { + u32 msg_id:8; /* Message ID - Max 256 Message Types */ + u32 str_id:5; + u32 large:1; /* Large Message if large = 1 */ + u32 reserved:2; /* Reserved for future use */ + u32 data:14; /* Ack/Info for msg, size of msg in Mailbox */ + u32 done:1; /* bit 30 */ + u32 busy:1; /* bit 31 */ + } part; + u32 full; +} __packed; + +/* Firmware build info */ +struct sst_fw_build_info { + unsigned char date[16]; /* Firmware build date */ + unsigned char time[16]; /* Firmware build time */ +} __packed; + +/* Firmware Version info */ +struct snd_sst_fw_version { + u8 build; /* build number*/ + u8 minor; /* minor number*/ + u8 major; /* major number*/ + u8 type; /* build type */ +}; + +struct ipc_header_fw_init { + struct snd_sst_fw_version fw_version;/* Firmware version details */ + struct sst_fw_build_info build_info; + u16 result; /* Fw init result */ + u8 module_id; /* Module ID in case of error */ + u8 debug_info; /* Debug info from Module ID in case of fail */ +} __packed; + +struct snd_sst_tstamp { + u64 ring_buffer_counter; /* PB/CP: Bytes copied from/to DDR. */ + u64 hardware_counter; /* PB/CP: Bytes DMAed to/from SSP. */ + u64 frames_decoded; + u64 bytes_decoded; + u64 bytes_copied; + u32 sampling_frequency; + u32 channel_peak[8]; +} __packed; + +/* Stream type params struture for Alloc stream */ +struct snd_sst_str_type { + u8 codec_type; /* Codec type */ + u8 str_type; /* 1 = voice 2 = music */ + u8 operation; /* Playback or Capture */ + u8 protected_str; /* 0=Non DRM, 1=DRM */ + u8 time_slots; + u8 reserved; /* Reserved */ + u16 result; /* Result used for acknowledgment */ +} __packed; + +/* Library info structure */ +struct module_info { + u32 lib_version; + u32 lib_type;/*TBD- KLOCKWORK u8 lib_type;*/ + u32 media_type; + u8 lib_name[12]; + u32 lib_caps; + unsigned char b_date[16]; /* Lib build date */ + unsigned char b_time[16]; /* Lib build time */ +} __packed; + +/* Library slot info */ +struct lib_slot_info { + u8 slot_num; /* 1 or 2 */ + u8 reserved1; + u16 reserved2; + u32 iram_size; /* slot size in IRAM */ + u32 dram_size; /* slot size in DRAM */ + u32 iram_offset; /* starting offset of slot in IRAM */ + u32 dram_offset; /* starting offset of slot in DRAM */ +} __packed; + +struct snd_ppp_mixer_params { + __u32 type; /*Type of the parameter */ + __u32 size; + __u32 input_stream_bitmap; /*Input stream Bit Map*/ +} __packed; + +struct snd_sst_lib_download { + struct module_info lib_info; /* library info type, capabilities etc */ + struct lib_slot_info slot_info; /* slot info to be downloaded */ + u32 mod_entry_pt; +}; + +struct snd_sst_lib_download_info { + struct snd_sst_lib_download dload_lib; + u16 result; /* Result used for acknowledgment */ + u8 pvt_id; /* Private ID */ + u8 reserved; /* for alignment */ +}; struct snd_pcm_params { u8 num_chan; /* 1=Mono, 2=Stereo */ u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u8 use_offload_path; + u8 use_offload_path; /* 0-PCM using period elpased & ALSA interfaces + 1-PCM stream via compressed interface */ u8 reserved2; - u16 reserved3; + u32 sfreq; /* Sampling rate in Hz */ u8 channel_map[8]; } __packed;

@@ -76,6 +340,7 @@ struct snd_aac_params { struct snd_wma_params { u8 num_chan; /* 1=Mono, 2=Stereo */ u8 pcm_wd_sz; /* 16/24 - bit*/ + u16 reserved1; u32 brate; /* Use the hard coded value. */ u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ u32 channel_mask; /* Channel Mask */ @@ -101,26 +366,153 @@ struct sst_address_info { };

struct snd_sst_alloc_params_ext { - struct sst_address_info ring_buf_info[8]; - u8 sg_count; - u8 reserved; - u16 reserved2; - u32 frag_size; /*Number of samples after which period elapsed + __u16 sg_count; + __u16 reserved; + __u32 frag_size; /*Number of samples after which period elapsed message is sent valid only if path = 0*/ -} __packed; + struct sst_address_info ring_buf_info[8]; +};

struct snd_sst_stream_params { union snd_sst_codec_params uc; } __packed;

struct snd_sst_params { + u32 result; u32 stream_id; u8 codec; u8 ops; u8 stream_type; u8 device_type; + u8 task; struct snd_sst_stream_params sparams; struct snd_sst_alloc_params_ext aparams; };

+struct snd_sst_alloc_mrfld { + u16 codec_type; + u8 operation; + u8 sg_count; + struct sst_address_info ring_buf_info[8]; + u32 frag_size; + u32 ts; + struct snd_sst_stream_params codec_params; +} __packed; + +/* Alloc stream params structure */ +struct snd_sst_alloc_params { + struct snd_sst_str_type str_type; + struct snd_sst_stream_params stream_params; + struct snd_sst_alloc_params_ext alloc_params; +} __packed; + +/* Alloc stream response message */ +struct snd_sst_alloc_response { + struct snd_sst_str_type str_type; /* Stream type for allocation */ + struct snd_sst_lib_download lib_dnld; /* Valid only for codec dnld */ +}; + +/* Drop response */ +struct snd_sst_drop_response { + u32 result; + u32 bytes; +}; + +struct snd_sst_async_msg { + u32 msg_id; /* Async msg id */ + u32 payload[0]; +}; + +struct snd_sst_async_err_msg { + u32 fw_resp; /* Firmware Result */ + u32 lib_resp; /*Library result */ +} __packed; + +struct snd_sst_vol { + u32 stream_id; + s32 volume; + u32 ramp_duration; + u32 ramp_type; /* Ramp type, default=0 */ +}; + +/* Gain library parameters for mrfld + * based on DSP command spec v0.82 + */ +struct snd_sst_gain_v2 { + u16 gain_cell_num; /* num of gain cells to modify*/ + u8 cell_nbr_idx; /* instance index*/ + u8 cell_path_idx; /* pipe-id */ + u16 module_id; /*module id */ + u16 left_cell_gain; /* left gain value in dB*/ + u16 right_cell_gain; /* right gain value in dB*/ + u16 gain_time_const; /* gain time constant*/ +} __packed; + +struct snd_sst_mute { + u32 stream_id; + u32 mute; +}; + +struct snd_sst_runtime_params { + u8 type; + u8 str_id; + u8 size; + u8 rsvd; + void *addr; +} __packed; + +enum stream_param_type { + SST_SET_TIME_SLOT = 0, + SST_SET_CHANNEL_INFO = 1, + OTHERS = 2, /*reserved for future params*/ +}; + +/* CSV Voice call routing structure */ +struct snd_sst_control_routing { + u8 control; /* 0=start, 1=Stop */ + u8 reserved[3]; /* Reserved- for 32 bit alignment */ +}; + +struct ipc_post { + struct list_head node; + union ipc_header header; /* driver specific */ + bool is_large; + bool is_process_reply; + union ipc_header_mrfld mrfld_header; + char *mailbox_data; +}; + +struct snd_sst_ctxt_params { + u32 address; /* Physical Address in DDR where the context is stored */ + u32 size; /* size of the context */ +}; + +struct snd_sst_lpe_log_params { + u8 dbg_type; + u8 module_id; + u8 log_level; + u8 reserved; +} __packed; + +enum snd_sst_bytes_type { + SND_SST_BYTES_SET = 0x1, + SND_SST_BYTES_GET = 0x2, +}; + +struct snd_sst_bytes_v2 { + u8 type; + u8 ipc_msg; + u8 block; + u8 task_id; + u8 pipe_id; + u8 rsvd; + u16 len; + char bytes[0]; +}; + +#define MAX_VTSV_FILES 2 +struct snd_sst_vtsv_info { + struct sst_address_info vfiles[MAX_VTSV_FILES]; +} __packed; + #endif /* __SST_MFLD_DSP_H__ */

Tis will be used to add table based support for pcm front ends in subsequent patches

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform-pcm.c | 102 +++++++++++++++++++++---------- sound/soc/intel/sst-mfld-platform.h | 2 +- 2 files changed, 71 insertions(+), 33 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 80879e5..6e7bfb1 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -143,52 +143,90 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream) return state; }

+static void sst_fill_alloc_params(struct snd_pcm_substream *substream, + struct snd_sst_alloc_params_ext *alloc_param) +{ + unsigned int channels; + snd_pcm_uframes_t period_size; + ssize_t periodbytes; + ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + + channels = substream->runtime->channels; + period_size = substream->runtime->period_size; + periodbytes = samples_to_bytes(substream->runtime, period_size); + alloc_param->ring_buf_info[0].addr = buffer_addr; + alloc_param->ring_buf_info[0].size = buffer_bytes; + alloc_param->sg_count = 1; + alloc_param->reserved = 0; + alloc_param->frag_size = periodbytes * channels; + +} static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) + struct snd_sst_stream_params *param) { + param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param->uc.pcm_params.sfreq = substream->runtime->rate;

- param->num_chan = (u8) substream->runtime->channels; - param->pcm_wd_sz = substream->runtime->sample_bits; - param->reserved = 0; - param->sfreq = substream->runtime->rate; - param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); - param->period_count = substream->runtime->period_size; - param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); + /* PCM stream via ALSA interface */ + param->uc.pcm_params.use_offload_path = 0; + param->uc.pcm_params.reserved2 = 0; + memset(param->uc.pcm_params.channel_map, 0, sizeof(u8)); + +} +int sst_fill_stream_params(void *substream, + struct snd_sst_params *str_params, bool is_compress) +{ + struct snd_pcm_substream *pstream = NULL; + struct snd_compr_stream *cstream = NULL; + + if (is_compress == true) + cstream = (struct snd_compr_stream *)substream; + else + pstream = (struct snd_pcm_substream *)substream; + + str_params->stream_type = SST_STREAM_TYPE_MUSIC; + + /* For pcm streams */ + if (pstream) + str_params->ops = (u8)pstream->stream; + if (cstream) + str_params->ops = (u8)cstream->direction; + + return 0; }

-static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, + struct snd_soc_platform *platform) { struct sst_runtime_stream *stream = substream->runtime->private_data; - struct sst_pcm_params param = {0}; - struct sst_stream_params str_params = {0}; - int ret_val; + struct snd_sst_stream_params param = {{{0,},},}; + struct snd_sst_params str_params = {0}; + struct snd_sst_alloc_params_ext alloc_params = {0}; + int ret_val = 0;

/* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, &param); + sst_fill_alloc_params(substream, &alloc_params); substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; - str_params.codec = param.codec; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.device_type = substream->pcm->device + 1; - pr_debug("Playbck stream,Device %d\n", - substream->pcm->device); - } else { - str_params.ops = STREAM_OPS_CAPTURE; - str_params.device_type = SND_SST_DEVICE_CAPTURE; - pr_debug("Capture stream,Device %d\n", - substream->pcm->device); - } - ret_val = stream->ops->open(&str_params); - pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + str_params.aparams = alloc_params; + str_params.codec = SST_CODEC_TYPE_PCM; + + /* fill the device type and stream id to pass to SST driver */ + ret_val = sst_fill_stream_params(substream, &str_params, false); if (ret_val < 0) return ret_val;

- stream->stream_info.str_id = ret_val; - pr_debug("str id : %d\n", stream->stream_info.str_id); + stream->stream_info.str_id = str_params.stream_id; + + ret_val = stream->ops->open(&str_params); + if (ret_val <= 0) + return ret_val; + + return ret_val; }

@@ -300,8 +338,8 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, return ret_val; }

- ret_val = sst_platform_alloc_stream(substream); - if (ret_val < 0) + ret_val = sst_platform_alloc_stream(substream, dai->platform); + if (ret_val <= 0) return ret_val; snprintf(substream->pcm->id, sizeof(substream->pcm->id), "%d", stream->stream_info.str_id); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 33a0a27..aa5ddbb 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -125,7 +125,7 @@ struct compress_sst_ops { };

struct sst_ops { - int (*open) (struct sst_stream_params *str_param); + int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); int (*close) (unsigned int str_id); };

Merrifield DSP used various pipelines to identify the streams and processing modules. Add these defination in the pcm driver and also add a table for device entries to firmware pipeline id conversion

Signed-off-by: Vinod Koul vinod.koul@intel.com --- arch/x86/include/asm/platform_sst_audio.h | 78 ++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.h | 30 +++++++++ sound/soc/intel/sst-mfld-platform-pcm.c | 100 ++++++++++++++++++++++++++--- sound/soc/intel/sst-mfld-platform.h | 18 +++++ 4 files changed, 217 insertions(+), 9 deletions(-) create mode 100644 arch/x86/include/asm/platform_sst_audio.h create mode 100644 sound/soc/intel/sst-atom-controls.h

diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h new file mode 100644 index 0000000..0a4e140 --- /dev/null +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -0,0 +1,78 @@ +/* + * platform_sst_audio.h: sst audio platform data header file + * + * Copyright (C) 2012-14 Intel Corporation + * Author: Jeeja KP jeeja.kp@intel.com + * Omair Mohammed Abdullah omair.m.abdullah@intel.com + * Vinod Koul ,vinod.koul@intel.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; version 2 + * of the License. + */ +#ifndef _PLATFORM_SST_AUDIO_H_ +#define _PLATFORM_SST_AUDIO_H_ + +#include <linux/sfi.h> + +enum sst_audio_task_id_mrfld { + SST_TASK_ID_NONE = 0, + SST_TASK_ID_SBA = 1, + SST_TASK_ID_MEDIA = 3, + SST_TASK_ID_MAX = SST_TASK_ID_MEDIA, +}; + +/* Device IDs for Merrifield are Pipe IDs, + * ref: DSP spec v0.75 */ +enum sst_audio_device_id_mrfld { + /* Output pipeline IDs */ + PIPE_ID_OUT_START = 0x0, + PIPE_CODEC_OUT0 = 0x2, + PIPE_CODEC_OUT1 = 0x3, + PIPE_SPROT_LOOP_OUT = 0x4, + PIPE_MEDIA_LOOP1_OUT = 0x5, + PIPE_MEDIA_LOOP2_OUT = 0x6, + PIPE_VOIP_OUT = 0xC, + PIPE_PCM0_OUT = 0xD, + PIPE_PCM1_OUT = 0xE, + PIPE_PCM2_OUT = 0xF, + PIPE_MEDIA0_OUT = 0x12, + PIPE_MEDIA1_OUT = 0x13, +/* Input Pipeline IDs */ + PIPE_ID_IN_START = 0x80, + PIPE_CODEC_IN0 = 0x82, + PIPE_CODEC_IN1 = 0x83, + PIPE_SPROT_LOOP_IN = 0x84, + PIPE_MEDIA_LOOP1_IN = 0x85, + PIPE_MEDIA_LOOP2_IN = 0x86, + PIPE_VOIP_IN = 0x8C, + PIPE_PCM0_IN = 0x8D, + PIPE_PCM1_IN = 0x8E, + PIPE_MEDIA0_IN = 0x8F, + PIPE_MEDIA1_IN = 0x90, + PIPE_MEDIA2_IN = 0x91, + PIPE_RSVD = 0xFF, +}; + +/* The stream map for each platform consists of an array of the below + * stream map structure. + */ +struct sst_dev_stream_map { + u8 dev_num; /* device id */ + u8 subdev_num; /* substream */ + u8 direction; + u8 device_id; /* fw id */ + u8 task_id; /* fw task */ + u8 status; +}; + +struct sst_platform_data { + /* Intel software platform id*/ + struct sst_dev_stream_map *pdev_strm_map; + unsigned int strm_map_size; +}; + +int add_sst_platform_device(void); +#endif + diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h new file mode 100644 index 0000000..14063ab --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.h @@ -0,0 +1,30 @@ +/* + * Copyright (C) 2013-14 Intel Corp + * Author: Ramesh Babu ramesh.babu.koul@intel.com + * Omair M Abdullah omair.m.abdullah@intel.com + * Samreen Nilofer samreen.nilofer@intel.com + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SST_CONTROLS_V2_H__ +#define __SST_CONTROLS_V2_H__ + +enum { + MERR_DPCM_AUDIO = 0, + MERR_DPCM_COMPR, +}; + + +#endif diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 6e7bfb1..7de8788 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -1,7 +1,7 @@ /* * sst_mfld_platform.c - Intel MID Platform driver * - * Copyright (C) 2010-2013 Intel Corp + * Copyright (C) 2010-2014 Intel Corp * Author: Vinod Koul vinod.koul@intel.com * Author: Harsha Priya priya.harsha@intel.com * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -27,7 +27,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/compress_driver.h> +#include <asm/platform_sst_audio.h> #include "sst-mfld-platform.h" +#include "sst-atom-controls.h"

struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -92,6 +94,13 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { .fifo_size = SST_FIFO_SIZE, };

+static struct sst_dev_stream_map dpcm_strm_map[] = { + {0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */ + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0}, +}; + /* MFLD - MSIC */ static struct snd_soc_dai_driver sst_platform_dai[] = { { @@ -175,12 +184,36 @@ static void sst_fill_pcm_params(struct snd_pcm_substream *substream, memset(param->uc.pcm_params.channel_map, 0, sizeof(u8));

} + +static int sst_get_stream_mapping(int dev, int sdev, int dir, + struct sst_dev_stream_map *map, int size) +{ + int i; + + if (map == NULL) + return -EINVAL; + + + /* index 0 is not used in stream map */ + for (i = 1; i < size; i++) { + if ((map[i].dev_num == dev) && (map[i].direction == dir)) + return i; + } + return 0; +} + int sst_fill_stream_params(void *substream, - struct snd_sst_params *str_params, bool is_compress) + const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress) { + int map_size; + int index; + struct sst_dev_stream_map *map; struct snd_pcm_substream *pstream = NULL; struct snd_compr_stream *cstream = NULL;

+ map = ctx->pdata->pdev_strm_map; + map_size = ctx->pdata->strm_map_size; + if (is_compress == true) cstream = (struct snd_compr_stream *)substream; else @@ -189,11 +222,32 @@ int sst_fill_stream_params(void *substream, str_params->stream_type = SST_STREAM_TYPE_MUSIC;

/* For pcm streams */ - if (pstream) + if (pstream) { + index = sst_get_stream_mapping(pstream->pcm->device, + pstream->number, pstream->stream, + map, map_size); + if (index <= 0) + return -EINVAL; + + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + str_params->ops = (u8)pstream->stream; - if (cstream) - str_params->ops = (u8)cstream->direction; + } + + if (cstream) { + index = sst_get_stream_mapping(cstream->device->device, + 0, cstream->direction, + map, map_size); + if (index <= 0) + return -EINVAL; + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id;

+ str_params->ops = (u8)cstream->direction; + } return 0; }

@@ -206,6 +260,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, struct snd_sst_params str_params = {0}; struct snd_sst_alloc_params_ext alloc_params = {0}; int ret_val = 0; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);

/* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, &param); @@ -216,7 +271,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, str_params.codec = SST_CODEC_TYPE_PCM;

/* fill the device type and stream id to pass to SST driver */ - ret_val = sst_fill_stream_params(substream, &str_params, false); + ret_val = sst_fill_stream_params(substream, ctx, &str_params, false); if (ret_val < 0) return ret_val;

@@ -321,7 +376,22 @@ static void sst_media_close(struct snd_pcm_substream *substream, ret_val = stream->ops->close(str_id); module_put(sst->dev->driver->owner); kfree(stream); - return; +} + +static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform, + struct snd_pcm_substream *substream) +{ + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; + struct sst_runtime_stream *stream = + substream->runtime->private_data; + u32 str_id = stream->stream_info.str_id; + unsigned int pipe_id; + pipe_id = map[str_id].device_id; + + pr_debug("%s: got pipe_id = %#x for str_id = %d\n", + __func__, pipe_id, str_id); + return pipe_id; }

static int sst_media_prepare(struct snd_pcm_substream *substream, @@ -498,10 +568,22 @@ static const struct snd_soc_component_driver sst_component = {

static int sst_platform_probe(struct platform_device *pdev) { + struct sst_data *drv; int ret; + struct sst_platform_data *pdata = pdev->dev.platform_data; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (sst == NULL) { + pr_err("kzalloc failed\n"); + return -ENOMEM; + } + + pdata->pdev_strm_map = dpcm_strm_map; + pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map); + drv->pdata = pdata; + mutex_init(&drv->lock); + dev_set_drvdata(&pdev->dev, drv);

- pr_debug("sst_platform_probe called\n"); - sst = NULL; ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { pr_err("registering soc platform failed\n"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index aa5ddbb..33891a8 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -144,10 +144,28 @@ struct sst_device { char *name; struct device *dev; struct sst_ops *ops; + struct platform_device *pdev; struct compress_sst_ops *compr_ops; };

+struct sst_data; + void sst_set_stream_status(struct sst_runtime_stream *stream, int state); +struct sst_algo_int_control_v2 { + struct soc_mixer_control mc; + u16 module_id; /* module identifieer */ + u16 pipe_id; /* location info: pipe_id + instance_id */ + u16 instance_id; + unsigned int value; /* Value received is stored here */ +}; + +struct sst_data { + struct platform_device *pdev; + struct sst_platform_data *pdata; + struct mutex lock; +}; + int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); + #endif

As added in previosu patch along with stream to piep conversion si required for compressed audio too

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform-compress.c | 11 ++++++++--- sound/soc/intel/sst-mfld-platform.h | 7 +++---- 2 files changed, 11 insertions(+), 7 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 02abd19..29c059c 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, int retval; struct snd_sst_params str_params; struct sst_compress_cb cb; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);

stream = cstream->runtime->private_data; /* construct fw structure for this*/ memset(&str_params, 0, sizeof(str_params));

- str_params.ops = STREAM_OPS_PLAYBACK; - str_params.stream_type = SST_STREAM_TYPE_MUSIC; - str_params.device_type = SND_SST_DEVICE_COMPRESS; + /* fill the device type and stream id to pass to SST driver */ + retval = sst_fill_stream_params(cstream, ctx, &str_params, true); + pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval); + if (retval < 0) + return retval;

switch (params->codec.id) { case SND_AUDIOCODEC_MP3: { diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 33891a8..9dc962f 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -149,8 +149,10 @@ struct sst_device { };

struct sst_data; - void sst_set_stream_status(struct sst_runtime_stream *stream, int state); +int sst_fill_stream_params(void *substream, const struct sst_data *ctx, + struct snd_sst_params *str_params, bool is_compress); + struct sst_algo_int_control_v2 { struct soc_mixer_control mc; u16 module_id; /* module identifieer */ @@ -158,14 +160,11 @@ struct sst_algo_int_control_v2 { u16 instance_id; unsigned int value; /* Value received is stored here */ }; - struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; struct mutex lock; }; - int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); - #endif

Now that we have added code for managing DPS piplelines we need to add the code for DSPs FrontEnd and Backend dai.

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform-pcm.c | 113 +++++++++++++++++++++++-------- 1 files changed, 84 insertions(+), 29 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7de8788..3fcd35c 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -101,35 +101,6 @@ static struct sst_dev_stream_map dpcm_strm_map[] = { {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0}, };

-/* MFLD - MSIC */ -static struct snd_soc_dai_driver sst_platform_dai[] = { -{ - .name = "Headset-cpu-dai", - .id = 0, - .playback = { - .channels_min = SST_STEREO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 5, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Compress-cpu-dai", - .compress_dai = 1, - .playback = { - .channels_min = SST_STEREO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -};

/* helper functions */ void sst_set_stream_status(struct sst_runtime_stream *stream, @@ -444,6 +415,90 @@ static struct snd_soc_dai_ops sst_media_dai_ops = { .hw_free = sst_media_hw_free, };

+static struct snd_soc_dai_driver sst_platform_dai[] = { +{ + .name = "media-cpu-dai", + .ops = &sst_media_dai_ops, + .playback = { + .stream_name = "Headset Playback", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Headset Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "compress-cpu-dai", + .compress_dai = 1, + .playback = { + .stream_name = "Compress Playback", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +/*BE CPU Dais */ +{ + .name = "ssp0-port", + .playback = { + .stream_name = "ssp0 Tx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp0 Rx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "ssp1-port", + .playback = { + .stream_name = "ssp1 Tx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp1 Rx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "ssp2-port", + .playback = { + .stream_name = "ssp2 Tx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp2 Rx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +}; + static int sst_platform_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime;

The add the registers space for MRFLD DSP. The initialization is done in soc_probe of the platform. This will be used in subsequent patches to add platform widgets

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/Makefile | 3 +- sound/soc/intel/sst-atom-controls.c | 90 +++++++++ sound/soc/intel/sst-atom-controls.h | 318 +++++++++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform-pcm.c | 10 +- sound/soc/intel/sst-mfld-platform.h | 10 + 5 files changed, 429 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/sst-atom-controls.c

diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4bfca79..9480b5d 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -2,7 +2,8 @@ snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o snd-soc-sst-acpi-objs := sst-acpi.o

-snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o sst-mfld-platform-compress.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \ + sst-mfld-platform-compress.o sst-atom-controls.o snd-soc-mfld-machine-objs := mfld_machine.o

obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c new file mode 100644 index 0000000..f710888 --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.c @@ -0,0 +1,90 @@ +/* + * sst-atom-controls.c - Intel MID Platform driver DPCM ALSA controls for Mrfld + * + * Copyright (C) 2013-14 Intel Corp + * Author: Omair Mohammed Abdullah omair.m.abdullah@intel.com + * Vinod Koul vinod.koul@intel.com + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "sst-mfld-platform.h" +#include "sst-atom-controls.h" + +unsigned int sst_soc_read(struct snd_soc_platform *platform, + unsigned int reg) +{ + struct sst_data *drv = snd_soc_platform_get_drvdata(platform); + + pr_debug("%s: reg[%d] = %#x\n", __func__, reg, drv->widget[reg]); + BUG_ON(reg > (SST_NUM_WIDGETS - 1)); + return drv->widget[reg]; +} + +int sst_soc_write(struct snd_soc_platform *platform, + unsigned int reg, unsigned int val) +{ + struct sst_data *drv = snd_soc_platform_get_drvdata(platform); + + pr_debug("%s: reg[%d] = %#x\n", __func__, reg, val); + BUG_ON(reg > (SST_NUM_WIDGETS - 1)); + drv->widget[reg] = val; + return 0; +} + +unsigned int sst_reg_read(struct sst_data *drv, unsigned int reg, + unsigned int shift, unsigned int max) +{ + unsigned int mask = (1 << fls(max)) - 1; + + return (drv->widget[reg] >> shift) & mask; +} + +unsigned int sst_reg_write(struct sst_data *drv, unsigned int reg, + unsigned int shift, unsigned int max, unsigned int val) +{ + unsigned int mask = (1 << fls(max)) - 1; + + val &= mask; + val <<= shift; + drv->widget[reg] &= ~(mask << shift); + drv->widget[reg] |= val; + return val; +} + + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) +{ + int ret = 0; + struct sst_data *drv = snd_soc_platform_get_drvdata(platform); + + drv->byte_stream = devm_kzalloc(platform->dev, + SST_MAX_BIN_BYTES, GFP_KERNEL); + if (!drv->byte_stream) { + pr_err("%s: kzalloc failed\n", __func__); + return -ENOMEM; + } + drv->widget = devm_kzalloc(platform->dev, + SST_NUM_WIDGETS * sizeof(*drv->widget), + GFP_KERNEL); + if (!drv->widget) { + pr_err("%s: kzalloc failed\n", __func__); + return -ENOMEM; + } + + return ret; +} diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index 14063ab..8c35946 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -1,4 +1,6 @@ /* + * controls_v2.h - Intel MID Platform driver header file + * * Copyright (C) 2013-14 Intel Corp * Author: Ramesh Babu ramesh.babu.koul@intel.com * Omair M Abdullah omair.m.abdullah@intel.com @@ -25,6 +27,322 @@ enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_COMPR, }; +/* + * This section defines the map for the mixer widgets. + * + * Each mixer will be represented by single value and that value will have each + * bit corresponding to one input + * + * Each out_id will correspond to one mixer and one path. Each input will be + * represented by single bit in the register. + */ + +/* mixer register ids here */ +#define SST_MIX(x) (x) + +#define SST_MIX_CODEC0 SST_MIX(2) +#define SST_MIX_CODEC1 SST_MIX(3) +#define SST_MIX_LOOP0 SST_MIX(4) +#define SST_MIX_LOOP1 SST_MIX(5) +#define SST_MIX_LOOP2 SST_MIX(6) +#define SST_MIX_VOIP SST_MIX(12) +#define SST_MIX_PCM0 SST_MIX(13) +#define SST_MIX_PCM1 SST_MIX(14) +#define SST_MIX_PCM2 SST_MIX(15) + +#define SST_MIX_MEDIA0 SST_MIX(19) +#define SST_MIX_MEDIA1 SST_MIX(20) + +#define SST_NUM_MIX (SST_MIX_MEDIA1 + 1) + +#define SST_MIX_SWITCH (SST_NUM_MIX + 1) +#define SST_OUT_SWITCH (SST_NUM_MIX + 2) +#define SST_IN_SWITCH (SST_NUM_MIX + 3) +#define SST_MUX_REG (SST_NUM_MIX + 4) +#define SST_REG_LAST (SST_MUX_REG) + +/* last entry defines array size */ +#define SST_NUM_WIDGETS (SST_REG_LAST + 1) + +/* in each mixer register we will define one bit for each input */ +#define SST_MIX_IP(x) (x) + +#define SST_IP_CODEC0 SST_MIX_IP(2) +#define SST_IP_CODEC1 SST_MIX_IP(3) +#define SST_IP_LOOP0 SST_MIX_IP(4) +#define SST_IP_LOOP1 SST_MIX_IP(5) +#define SST_IP_LOOP2 SST_MIX_IP(6) +#define SST_IP_PROBE SST_MIX_IP(7) +#define SST_IP_VOIP SST_MIX_IP(12) +#define SST_IP_PCM0 SST_MIX_IP(13) +#define SST_IP_PCM1 SST_MIX_IP(14) +#define SST_IP_MEDIA0 SST_MIX_IP(17) +#define SST_IP_MEDIA1 SST_MIX_IP(18) +#define SST_IP_MEDIA2 SST_MIX_IP(19) +#define SST_IP_MEDIA3 SST_MIX_IP(20) + +#define SST_IP_LAST SST_IP_MEDIA3 + +#define SST_SWM_INPUT_COUNT (SST_IP_LAST + 1) +#define SST_CMD_SWM_MAX_INPUTS 6 + +#define SST_PATH_ID_SHIFT 8 +#define SST_DEFAULT_LOCATION_ID 0xFFFF +#define SST_DEFAULT_CELL_NBR 0xFF +#define SST_DEFAULT_MODULE_ID 0xFFFF + +/* + * Audio DSP Path Ids. Specified by the audio DSP FW + */ +enum sst_path_index { + SST_PATH_INDEX_CODEC_OUT0 = (0x02 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_OUT1 = (0x03 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_OUT = (0x04 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_OUT = (0x05 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_OUT = (0x06 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_OUT = (0x0C << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM0_OUT = (0x0D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_OUT = (0x0E << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM2_OUT = (0x0F << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_OUT = (0x12 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_OUT = (0x13 << SST_PATH_ID_SHIFT), + + + /* Start of input paths */ + SST_PATH_INDEX_CODEC_IN0 = (0x82 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_IN1 = (0x83 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_IN = (0x84 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_IN = (0x85 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_IN = (0x86 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_IN = (0x8C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_PCM0_IN = (0x8D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_IN = (0x8E << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_IN = (0x8F << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_IN = (0x90 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA2_IN = (0x91 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA3_IN = (0x9C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_RESERVED = (0xFF << SST_PATH_ID_SHIFT), +}; + +/* + * path IDs + */ +enum sst_swm_inputs { + SST_SWM_IN_CODEC0 = (SST_PATH_INDEX_CODEC_IN0 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_CODEC1 = (SST_PATH_INDEX_CODEC_IN1 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_VOIP = (SST_PATH_INDEX_VOIP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM0 = (SST_PATH_INDEX_PCM0_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM1 = (SST_PATH_INDEX_PCM1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA0 = (SST_PATH_INDEX_MEDIA0_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA1 = (SST_PATH_INDEX_MEDIA1_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA2 = (SST_PATH_INDEX_MEDIA2_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA3 = (SST_PATH_INDEX_MEDIA3_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR) +}; + +/* + * path IDs + */ +enum sst_swm_outputs { + SST_SWM_OUT_CODEC0 = (SST_PATH_INDEX_CODEC_OUT0 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_CODEC1 = (SST_PATH_INDEX_CODEC_OUT1 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_VOIP = (SST_PATH_INDEX_VOIP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM0 = (SST_PATH_INDEX_PCM0_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM1 = (SST_PATH_INDEX_PCM1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM2 = (SST_PATH_INDEX_PCM2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA0 = (SST_PATH_INDEX_MEDIA0_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_MEDIA1 = (SST_PATH_INDEX_MEDIA1_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR), +}; + +enum sst_ipc_msg { + SST_IPC_IA_CMD = 1, + SST_IPC_IA_SET_PARAMS, + SST_IPC_IA_GET_PARAMS, +}; + +enum sst_cmd_type { + SST_CMD_BYTES_SET = 1, + SST_CMD_BYTES_GET = 2, +}; + +enum sst_task { + SST_TASK_SBA = 1, + SST_TASK_MMX, +}; + +enum sst_type { + SST_TYPE_CMD = 1, + SST_TYPE_PARAMS, +}; + +enum sst_flag { + SST_FLAG_BLOCKED = 1, + SST_FLAG_NONBLOCK, +}; + +/* + * Enumeration for indexing the gain cells in VB_SET_GAIN DSP command + */ +enum sst_gain_index { + /* GAIN IDs for SB task start here */ + SST_GAIN_INDEX_CODEC_OUT0, + SST_GAIN_INDEX_CODEC_OUT1, + SST_GAIN_INDEX_CODEC_IN0, + SST_GAIN_INDEX_CODEC_IN1, + + SST_GAIN_INDEX_SPROT_LOOP_OUT, + SST_GAIN_INDEX_MEDIA_LOOP1_OUT, + SST_GAIN_INDEX_MEDIA_LOOP2_OUT, + + SST_GAIN_INDEX_PCM0_IN_LEFT, + SST_GAIN_INDEX_PCM0_IN_RIGHT, + + SST_GAIN_INDEX_PCM1_OUT_LEFT, + SST_GAIN_INDEX_PCM1_OUT_RIGHT, + SST_GAIN_INDEX_PCM1_IN_LEFT, + SST_GAIN_INDEX_PCM1_IN_RIGHT, + SST_GAIN_INDEX_PCM2_OUT_LEFT, + + SST_GAIN_INDEX_PCM2_OUT_RIGHT, + SST_GAIN_INDEX_VOIP_OUT, + SST_GAIN_INDEX_VOIP_IN, + + /* Gain IDs for MMX task start here */ + SST_GAIN_INDEX_MEDIA0_IN_LEFT, + SST_GAIN_INDEX_MEDIA0_IN_RIGHT, + SST_GAIN_INDEX_MEDIA1_IN_LEFT, + SST_GAIN_INDEX_MEDIA1_IN_RIGHT, + + SST_GAIN_INDEX_MEDIA2_IN_LEFT, + SST_GAIN_INDEX_MEDIA2_IN_RIGHT, + + SST_GAIN_INDEX_GAIN_END +};

+/* + * Audio DSP module IDs specified by FW spec + * TODO: Update with all modules + */ +enum sst_module_id { + SST_MODULE_ID_PCM = 0x0001, + SST_MODULE_ID_MP3 = 0x0002, + SST_MODULE_ID_MP24 = 0x0003, + SST_MODULE_ID_AAC = 0x0004, + SST_MODULE_ID_AACP = 0x0005, + SST_MODULE_ID_EAACP = 0x0006, + SST_MODULE_ID_WMA9 = 0x0007, + SST_MODULE_ID_WMA10 = 0x0008, + SST_MODULE_ID_WMA10P = 0x0009, + SST_MODULE_ID_RA = 0x000A, + SST_MODULE_ID_DDAC3 = 0x000B, + SST_MODULE_ID_TRUE_HD = 0x000C, + SST_MODULE_ID_HD_PLUS = 0x000D, + + SST_MODULE_ID_SRC = 0x0064, + SST_MODULE_ID_DOWNMIX = 0x0066, + SST_MODULE_ID_GAIN_CELL = 0x0067, + SST_MODULE_ID_SPROT = 0x006D, + SST_MODULE_ID_BASS_BOOST = 0x006E, + SST_MODULE_ID_STEREO_WDNG = 0x006F, + SST_MODULE_ID_AV_REMOVAL = 0x0070, + SST_MODULE_ID_MIC_EQ = 0x0071, + SST_MODULE_ID_SPL = 0x0072, + SST_MODULE_ID_ALGO_VTSV = 0x0073, + SST_MODULE_ID_NR = 0x0076, + SST_MODULE_ID_BWX = 0x0077, + SST_MODULE_ID_DRP = 0x0078, + SST_MODULE_ID_MDRP = 0x0079, + + SST_MODULE_ID_ANA = 0x007A, + SST_MODULE_ID_AEC = 0x007B, + SST_MODULE_ID_NR_SNS = 0x007C, + SST_MODULE_ID_SER = 0x007D, + SST_MODULE_ID_AGC = 0x007E, + + SST_MODULE_ID_CNI = 0x007F, + SST_MODULE_ID_CONTEXT_ALGO_AWARE = 0x0080, + SST_MODULE_ID_FIR_24 = 0x0081, + SST_MODULE_ID_IIR_24 = 0x0082, + + SST_MODULE_ID_ASRC = 0x0083, + SST_MODULE_ID_TONE_GEN = 0x0084, + SST_MODULE_ID_BMF = 0x0086, + SST_MODULE_ID_EDL = 0x0087, + SST_MODULE_ID_GLC = 0x0088, + + SST_MODULE_ID_FIR_16 = 0x0089, + SST_MODULE_ID_IIR_16 = 0x008A, + SST_MODULE_ID_DNR = 0x008B, + + SST_MODULE_ID_VIRTUALIZER = 0x008C, + SST_MODULE_ID_VISUALIZATION = 0x008D, + SST_MODULE_ID_LOUDNESS_OPTIMIZER = 0x008E, + SST_MODULE_ID_REVERBERATION = 0x008F, + + SST_MODULE_ID_CNI_TX = 0x0090, + SST_MODULE_ID_REF_LINE = 0x0091, + SST_MODULE_ID_VOLUME = 0x0092, + SST_MODULE_ID_FILT_DCR = 0x0094, + SST_MODULE_ID_SLV = 0x009A, + SST_MODULE_ID_NLF = 0x009B, + SST_MODULE_ID_TNR = 0x009C, + SST_MODULE_ID_WNR = 0x009D, + + SST_MODULE_ID_LOG = 0xFF00, + + SST_MODULE_ID_TASK = 0xFFFF, +}; + +enum sst_cmd { + SBA_IDLE = 14, + SBA_VB_SET_SPEECH_PATH = 26, + MMX_SET_GAIN = 33, + SBA_VB_SET_GAIN = 33, + FBA_VB_RX_CNI = 35, + MMX_SET_GAIN_TIMECONST = 36, + SBA_VB_SET_TIMECONST = 36, + SBA_VB_START = 85, + SBA_SET_SWM = 114, + SBA_SET_MDRP = 116, + SBA_HW_SET_SSP = 117, + SBA_SET_MEDIA_LOOP_MAP = 118, + SBA_SET_MEDIA_PATH = 119, + MMX_SET_MEDIA_PATH = 119, + SBA_VB_LPRO = 126, + SBA_VB_SET_FIR = 128, + SBA_VB_SET_IIR = 129, + SBA_SET_SSP_SLOT_MAP = 130, +}; + +enum sst_dsp_switch { + SST_SWITCH_OFF = 0, + SST_SWITCH_ON = 3, +}; + +enum sst_path_switch { + SST_PATH_OFF = 0, + SST_PATH_ON = 1, +}; + +enum sst_swm_state { + SST_SWM_OFF = 0, + SST_SWM_ON = 3, +};

#endif diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 3fcd35c..8df5aca 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -609,11 +609,19 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) return retval; }

-static struct snd_soc_platform_driver sst_soc_platform_drv = { +static int sst_soc_probe(struct snd_soc_platform *platform) +{ + return sst_dsp_init_v2_dpcm(platform); +} + +static struct snd_soc_platform_driver sst_soc_platform_drv = { + .probe = sst_soc_probe, .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, .pcm_free = sst_pcm_free, + .read = sst_soc_read, + .write = sst_soc_write, };

static const struct snd_soc_component_driver sst_component = { diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 9dc962f..dc60f86 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -149,6 +149,14 @@ struct sst_device { };

struct sst_data; + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform); +unsigned int sst_soc_read(struct snd_soc_platform *platform, unsigned int reg); +int sst_soc_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); +unsigned int sst_reg_read(struct sst_data *sst, unsigned int reg, + unsigned int shift, unsigned int max); +unsigned int sst_reg_write(struct sst_data *sst, unsigned int reg, + unsigned int shift, unsigned int max, unsigned int val); void sst_set_stream_status(struct sst_runtime_stream *stream, int state); int sst_fill_stream_params(void *substream, const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress); @@ -163,6 +171,8 @@ struct sst_algo_int_control_v2 { struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; + u32 *widget; + char *byte_stream; struct mutex lock; }; int sst_register_dsp(struct sst_device *sst);

This will be used by subsequent patches to send DSP cmds on DAPM widgets enable/disable

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform.h | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index dc60f86..5cfe947 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -63,7 +63,12 @@ enum sst_controls { SST_SND_BUFFER_POINTER = 0x05, SST_SND_STREAM_INIT = 0x06, SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, + SST_SET_RUNTIME_PARAMS = 0x08, + SST_SET_ALGO_PARAMS = 0x09, + SST_SET_BYTE_STREAM = 0x0A, + SST_GET_BYTE_STREAM = 0x0B, + + SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, };

enum sst_stream_ops { @@ -127,6 +132,7 @@ struct compress_sst_ops { struct sst_ops { int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); + int (*set_generic_params) (enum sst_controls cmd, void *arg); int (*close) (unsigned int str_id); };

When we have PCM (FE/BE) opened or DAPM widgets triggered we need power up/down DSP accordingly. The DSP will do ref count of these requests i.e. link these runtime_get/put calls of DSP

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform-pcm.c | 14 ++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 1 + 2 files changed, 15 insertions(+), 0 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 0fbb688..346f972 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -292,6 +292,16 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) return ret_val;

} + +static inline int power_up_sst(struct sst_runtime_stream *sst) +{ + return sst->ops->power(true); +} + +static inline int power_down_sst(struct sst_runtime_stream *sst) +{ + return sst->ops->power(false); +} /* end -- helper functions */

static int sst_media_open(struct snd_pcm_substream *substream, @@ -323,6 +333,8 @@ static int sst_media_open(struct snd_pcm_substream *substream, /* allocate memory for SST API set */ runtime->private_data = stream;

+ power_up_sst(stream); + /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, 2); @@ -342,6 +354,8 @@ static void sst_media_close(struct snd_pcm_substream *substream, int ret_val = 0, str_id;

stream = substream->runtime->private_data; + power_down_sst(stream); + str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->ops->close(str_id); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 5cfe947..4acdd32 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -134,6 +134,7 @@ struct sst_ops { int (*device_control) (int cmd, void *arg); int (*set_generic_params) (enum sst_controls cmd, void *arg); int (*close) (unsigned int str_id); + int (*power) (bool state); };

struct sst_runtime_stream {

This patchs adds all DAPM widgets and the event handlers for DSP expect the mixers. Since we are still discussing mixer update and is dependent upon component series

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-atom-controls.c | 237 +++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 4 + 2 files changed, 241 insertions(+), 0 deletions(-)

diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index 2938105..2e733f8 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -528,6 +528,161 @@ static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = { [SST_IP_MEDIA2] = SST_SWM_IN_MEDIA2, [SST_IP_MEDIA3] = SST_SWM_IN_MEDIA3, }; + +/** + * fill_swm_input - fill in the SWM input ids given the register + * + * The register value is a bit-field inicated which mixer inputs are ON. Use the + * lookup table to get the input-id and fill it in the structure. + */ +static int fill_swm_input(struct swm_input_ids *swm_input, unsigned int reg) +{ + uint i, is_set, nb_inputs = 0; + u16 input_loc_id; + + pr_debug("%s: reg: %#x\n", __func__, reg); + for (i = 0; i < SST_SWM_INPUT_COUNT; i++) { + is_set = reg & BIT(i); + if (!is_set) + continue; + + input_loc_id = swm_mixer_input_ids[i]; + SST_FILL_DESTINATION(2, swm_input->input_id, + input_loc_id, SST_DEFAULT_MODULE_ID); + nb_inputs++; + swm_input++; + pr_debug("input id: %#x, nb_inputs: %d\n", input_loc_id, nb_inputs); + + if (nb_inputs == SST_CMD_SWM_MAX_INPUTS) { + pr_warn("%s: SET_SWM cmd max inputs reached", __func__); + break; + } + } + return nb_inputs; +} + +static void sst_set_pipe_gain(struct sst_ids *ids, struct sst_data *drv, int mute) +{ + struct sst_gain_mixer_control *mc; + struct sst_gain_value *gv; + struct sst_module *gain = NULL; + + list_for_each_entry(gain, &ids->gain_list, node) { + struct snd_kcontrol *kctl = gain->kctl; + + pr_debug("control name=%s\n", kctl->id.name); + mc = (void *)kctl->private_value; + gv = mc->gain_val; + + sst_send_gain_cmd(drv, gv, mc->task_id, + mc->pipe_id | mc->instance_id, mc->module_id, mute); + } +} + +static int sst_swm_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct sst_cmd_set_swm cmd; + struct sst_data *drv = snd_soc_platform_get_drvdata(w->platform); + struct sst_ids *ids = w->priv; + bool set_mixer = false; + int val = drv->widget[ids->reg]; + + pr_debug("%s: widget = %s\n", __func__, w->name); + pr_debug("%s: reg[%d] = %#x\n", __func__, ids->reg, val); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMD: + set_mixer = true; + break; + case SND_SOC_DAPM_POST_REG: + if (w->power) + set_mixer = true; + break; + default: + set_mixer = false; + } + + if (set_mixer == false) + return 0; + + if (SND_SOC_DAPM_EVENT_ON(event) || + event == SND_SOC_DAPM_POST_REG) + cmd.switch_state = SST_SWM_ON; + else + cmd.switch_state = SST_SWM_OFF; + + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + /* MMX_SET_SWM == SBA_SET_SWM */ + cmd.header.command_id = SBA_SET_SWM; + + SST_FILL_DESTINATION(2, cmd.output_id, + ids->location_id, SST_DEFAULT_MODULE_ID); + cmd.nb_inputs = fill_swm_input(&cmd.input[0], val); + cmd.header.length = offsetof(struct sst_cmd_set_swm, input) - sizeof(struct sst_dsp_header) + + (cmd.nb_inputs * sizeof(cmd.input[0])); + + sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + ids->task_id, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); + return 0; +} + +/* SBA mixers - 16 inputs */ +#define SST_SBA_DECLARE_MIX_CONTROLS(kctl_name, mixer_reg) \ + static const struct snd_kcontrol_new kctl_name[] = { \ + SOC_SINGLE_EXT("codec_in0", mixer_reg, SST_IP_CODEC0, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("codec_in1", mixer_reg, SST_IP_CODEC1, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("sprot_loop_in", mixer_reg, SST_IP_LOOP0, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("media_loop1_in", mixer_reg, SST_IP_LOOP1, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("media_loop2_in", mixer_reg, SST_IP_LOOP2, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("pcm0_in", mixer_reg, SST_IP_PCM0, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("pcm1_in", mixer_reg, SST_IP_PCM1, 1, 0, \ + sst_mix_get, sst_mix_put), \ + } + +#define SST_SBA_MIXER_GRAPH_MAP(mix_name) \ + { mix_name, "codec_in0", "codec_in0" }, \ + { mix_name, "codec_in1", "codec_in1" }, \ + { mix_name, "sprot_loop_in", "sprot_loop_in" }, \ + { mix_name, "media_loop1_in", "media_loop1_in" }, \ + { mix_name, "media_loop2_in", "media_loop2_in" }, \ + { mix_name, "pcm0_in", "pcm0_in" }, \ + { mix_name, "pcm1_in", "pcm1_in" } + +#define SST_MMX_DECLARE_MIX_CONTROLS(kctl_name, mixer_reg) \ + static const struct snd_kcontrol_new kctl_name[] = { \ + SOC_SINGLE_EXT("media0_in", mixer_reg, SST_IP_MEDIA0, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("media1_in", mixer_reg, SST_IP_MEDIA1, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("media2_in", mixer_reg, SST_IP_MEDIA2, 1, 0, \ + sst_mix_get, sst_mix_put), \ + SOC_SINGLE_EXT("media3_in", mixer_reg, SST_IP_MEDIA3, 1, 0, \ + sst_mix_get, sst_mix_put), \ + } + +SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media0_controls, SST_MIX_MEDIA0); +SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media1_controls, SST_MIX_MEDIA1); + +/* 18 SBA mixers */ +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm0_controls, SST_MIX_PCM0); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm1_controls, SST_MIX_PCM1); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls, SST_MIX_PCM2); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls, SST_MIX_LOOP0); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls, SST_MIX_LOOP1); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls, SST_MIX_LOOP2); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls, SST_MIX_VOIP); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls, SST_MIX_CODEC0); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls, SST_MIX_CODEC1); + void sst_handle_vb_timer(struct snd_soc_platform *p, bool enable) { struct sst_cmd_generic cmd; @@ -751,6 +906,83 @@ static int sst_set_media_loop(struct snd_soc_dapm_widget *w, return 0; }

+static const struct snd_soc_dapm_widget sst_dapm_widgets[] = { + SST_AIF_IN("codec_in0", sst_set_be_modules), + SST_AIF_IN("codec_in1", sst_set_be_modules), + SST_AIF_OUT("codec_out0", sst_set_be_modules), + SST_AIF_OUT("codec_out1", sst_set_be_modules), + + /* Media Paths */ + /* MediaX IN paths are set via ALLOC, so no SET_MEDIA_PATH command */ + SST_PATH_INPUT("media0_in", SST_TASK_MMX, SST_SWM_IN_MEDIA0, sst_generic_modules_event), + SST_PATH_INPUT("media1_in", SST_TASK_MMX, SST_SWM_IN_MEDIA1, NULL), + SST_PATH_INPUT("media2_in", SST_TASK_MMX, SST_SWM_IN_MEDIA2, sst_set_media_path), + SST_PATH_INPUT("media3_in", SST_TASK_MMX, SST_SWM_IN_MEDIA3, NULL), + SST_PATH_OUTPUT("media0_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA0, sst_set_media_path), + SST_PATH_OUTPUT("media1_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA1, sst_set_media_path), + + /* SBA PCM Paths */ + SST_PATH_INPUT("pcm0_in", SST_TASK_SBA, SST_SWM_IN_PCM0, sst_set_media_path), + SST_PATH_INPUT("pcm1_in", SST_TASK_SBA, SST_SWM_IN_PCM1, sst_set_media_path), + SST_PATH_OUTPUT("pcm0_out", SST_TASK_SBA, SST_SWM_OUT_PCM0, sst_set_media_path), + SST_PATH_OUTPUT("pcm1_out", SST_TASK_SBA, SST_SWM_OUT_PCM1, sst_set_media_path), + SST_PATH_OUTPUT("pcm2_out", SST_TASK_SBA, SST_SWM_OUT_PCM2, sst_set_media_path), + + /* SBA Loops */ + SST_PATH_INPUT("sprot_loop_in", SST_TASK_SBA, SST_SWM_IN_SPROT_LOOP, NULL), + SST_PATH_INPUT("media_loop1_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP1, NULL), + SST_PATH_INPUT("media_loop2_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP2, NULL), + SST_PATH_MEDIA_LOOP_OUTPUT("sprot_loop_out", SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP, SST_FMT_MONO, sst_set_media_loop), + SST_PATH_MEDIA_LOOP_OUTPUT("media_loop1_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1, SST_FMT_MONO, sst_set_media_loop), + SST_PATH_MEDIA_LOOP_OUTPUT("media_loop2_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, SST_FMT_STEREO, sst_set_media_loop), + + /* Media Mixers */ +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"media0_in", NULL, "Compress Playback"}, + {"media1_in", NULL, "Headset Playback"}, + {"media2_in", NULL, "pcm0_out"}, + + {"media0_out mix 0", "media0_in", "media0_in"}, + {"media0_out mix 0", "media1_in", "media1_in"}, + {"media0_out mix 0", "media2_in", "media2_in"}, + {"media0_out mix 0", "media3_in", "media3_in"}, + {"media1_out mix 0", "media0_in", "media0_in"}, + {"media1_out mix 0", "media1_in", "media1_in"}, + {"media1_out mix 0", "media2_in", "media2_in"}, + {"media1_out mix 0", "media3_in", "media3_in"}, + + {"media0_out", NULL, "media0_out mix 0"}, + {"media1_out", NULL, "media1_out mix 0"}, + {"pcm0_in", NULL, "media0_out"}, + {"pcm1_in", NULL, "media1_out"}, + + {"Headset Capture", NULL, "pcm1_out"}, + {"Headset Capture", NULL, "pcm2_out"}, + {"pcm0_out", NULL, "pcm0_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("pcm0_out mix 0"), + {"pcm1_out", NULL, "pcm1_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("pcm1_out mix 0"), + {"pcm2_out", NULL, "pcm2_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("pcm2_out mix 0"), + + {"media_loop1_in", NULL, "media_loop1_out"}, + {"media_loop1_out", NULL, "media_loop1_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("media_loop1_out mix 0"), + {"media_loop2_in", NULL, "media_loop2_out"}, + {"media_loop2_out", NULL, "media_loop2_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("media_loop2_out mix 0"), + {"sprot_loop_in", NULL, "sprot_loop_out"}, + {"sprot_loop_out", NULL, "sprot_loop_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("sprot_loop_out mix 0"), + + {"codec_out0", NULL, "codec_out0 mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("codec_out0 mix 0"), + {"codec_out1", NULL, "codec_out1 mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("codec_out1 mix 0"), + +}; static const char * const slot_names[] = { "none", "slot 0", "slot 1", "slot 2", "slot 3", @@ -1060,6 +1292,11 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) return -ENOMEM; }

+ snd_soc_dapm_new_controls(&platform->dapm, sst_dapm_widgets, + ARRAY_SIZE(sst_dapm_widgets)); + snd_soc_dapm_add_routes(&platform->dapm, intercon, + ARRAY_SIZE(intercon)); + snd_soc_dapm_new_widgets(platform->dapm.card);

for (i = 0; i < SST_NUM_GAINS; i++) { sst_gains[i].mute = SST_GAIN_MUTE_DEFAULT; diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 4acdd32..fee03df 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -158,6 +158,10 @@ struct sst_device { struct sst_data;

int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform); +int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute); +void send_ssp_cmd(struct snd_soc_platform *platform, const char *id, bool enable); +void sst_handle_vb_timer(struct snd_soc_platform *platform, bool enable); + unsigned int sst_soc_read(struct snd_soc_platform *platform, unsigned int reg); int sst_soc_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); unsigned int sst_reg_read(struct sst_data *sst, unsigned int reg,

The is RFC patch for adding the platform mixer controls. This requires the dapm_set-get to be exported. Or changes required after component series is merged

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-atom-controls.c | 67 +++++++++++++++++++++++++++++++++++ 1 files changed, 67 insertions(+), 0 deletions(-)

diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index 2e733f8..81a24eb 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -66,6 +66,47 @@ unsigned int sst_reg_write(struct sst_data *drv, unsigned int reg, return val; }

+int sst_mix_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist;// = dapm_kcontrol_get_wlist(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct sst_data *drv = snd_soc_platform_get_drvdata(widget->platform); + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int val; + int connect; + struct snd_soc_dapm_update update; + + pr_debug("%s called set %#lx for %s\n", __func__, + ucontrol->value.integer.value[0], widget->name); + val = sst_reg_write(drv, mc->reg, mc->shift, mc->max, ucontrol->value.integer.value[0]); + connect = !!val; + + //dapm_kcontrol_set_value(kcontrol, val); + update.kcontrol = kcontrol; + update.reg = mc->reg; + update.mask = mask; + update.val = val; + + snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect, &update); + return 0; +} + +int sst_mix_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist;// = dapm_kcontrol_get_wlist(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct sst_data *drv = snd_soc_platform_get_drvdata(w->platform); + + ucontrol->value.integer.value[0] = !!sst_reg_read(drv, mc->reg, mc->shift, mc->max); + return 0; +} + static inline void sst_fill_byte_control(char *param, u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, @@ -937,6 +978,32 @@ static const struct snd_soc_dapm_widget sst_dapm_widgets[] = { SST_PATH_MEDIA_LOOP_OUTPUT("media_loop2_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, SST_FMT_STEREO, sst_set_media_loop),

/* Media Mixers */ + SST_SWM_MIXER("media0_out mix 0", SST_MIX_MEDIA0, SST_TASK_MMX, SST_SWM_OUT_MEDIA0, + sst_mix_media0_controls, sst_swm_mixer_event), + SST_SWM_MIXER("media1_out mix 0", SST_MIX_MEDIA1, SST_TASK_MMX, SST_SWM_OUT_MEDIA1, + sst_mix_media1_controls, sst_swm_mixer_event), + + /* SBA PCM mixers */ + SST_SWM_MIXER("pcm0_out mix 0", SST_MIX_PCM0, SST_TASK_SBA, SST_SWM_OUT_PCM0, + sst_mix_pcm0_controls, sst_swm_mixer_event), + SST_SWM_MIXER("pcm1_out mix 0", SST_MIX_PCM1, SST_TASK_SBA, SST_SWM_OUT_PCM1, + sst_mix_pcm1_controls, sst_swm_mixer_event), + SST_SWM_MIXER("pcm2_out mix 0", SST_MIX_PCM2, SST_TASK_SBA, SST_SWM_OUT_PCM2, + sst_mix_pcm2_controls, sst_swm_mixer_event), + + /* SBA Loop mixers */ + SST_SWM_MIXER("sprot_loop_out mix 0", SST_MIX_LOOP0, SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP, + sst_mix_sprot_l0_controls, sst_swm_mixer_event), + SST_SWM_MIXER("media_loop1_out mix 0", SST_MIX_LOOP1, SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1, + sst_mix_media_l1_controls, sst_swm_mixer_event), + SST_SWM_MIXER("media_loop2_out mix 0", SST_MIX_LOOP2, SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, + sst_mix_media_l2_controls, sst_swm_mixer_event), + + /* SBA Backend mixers */ + SST_SWM_MIXER("codec_out0 mix 0", SST_MIX_CODEC0, SST_TASK_SBA, SST_SWM_OUT_CODEC0, + sst_mix_codec0_controls, sst_swm_mixer_event), + SST_SWM_MIXER("codec_out1 mix 0", SST_MIX_CODEC1, SST_TASK_SBA, SST_SWM_OUT_CODEC1, + sst_mix_codec1_controls, sst_swm_mixer_event), };

static const struct snd_soc_dapm_route intercon[] = {