ASoC: Intel: add mrfld DSP defines

We define the DSP commands,structures here which will be used to send the IPCs

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index 7acbfc4..f841786 100644 (file)
--- 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 (file)
index 0000000..ace3c4a
--- /dev/null
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -0,0 +1,39 @@
+/*
+ *  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"
+
+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)
+               return -ENOMEM;
+
+       return ret;
+}

diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index 14063ab..8554889 100644 (file)
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -1,4 +1,6 @@
 /*
+ *  sst-atom-controls.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>
@@ -18,13 +20,293 @@
  *
  */
 
-#ifndef __SST_CONTROLS_V2_H__
-#define __SST_CONTROLS_V2_H__
+#ifndef __SST_ATOM_CONTROLS_H__
+#define __SST_ATOM_CONTROLS_H__
 
 enum {
        MERR_DPCM_AUDIO = 0,
        MERR_DPCM_COMPR,
 };
 
+/* define a bit for each mixer 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 a89ff7e..8e1e9bc 100644 (file)
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -550,7 +550,13 @@ 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,

diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h
index d4c28b8..faaba10 100644 (file)
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -143,6 +143,8 @@ struct sst_device {
 };
 
 struct sst_data;
+
+int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform);
 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);
@@ -157,6 +159,7 @@ struct sst_algo_int_control_v2 {
 struct sst_data {
        struct platform_device *pdev;
        struct sst_platform_data *pdata;
+       char *byte_stream;
        struct mutex lock;
 };
 int sst_register_dsp(struct sst_device *sst);