[alsa-devel] [PATCH v2 06/13] ASoC: Intel: add mrfld DSP defines
Subhransu S. Prusty
subhransu.s.prusty at intel.com
Thu Jul 10 06:44:50 CEST 2014
From: Vinod Koul <vinod.koul at intel.com>
We define the DSP commands,structures here which will be used to send the IPCs
Signed-off-by: Vinod Koul <vinod.koul at intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty at intel.com>
---
sound/soc/intel/Makefile | 3 +-
sound/soc/intel/sst-atom-controls.c | 41 +++++
sound/soc/intel/sst-atom-controls.h | 286 +++++++++++++++++++++++++++++++-
sound/soc/intel/sst-mfld-platform-pcm.c | 8 +-
sound/soc/intel/sst-mfld-platform.h | 3 +
5 files changed, 337 insertions(+), 4 deletions(-)
create mode 100644 sound/soc/intel/sst-atom-controls.c
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index 4bfca79a42ba..9480b5d40e02 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 000000000000..38731e3e5e95
--- /dev/null
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -0,0 +1,41 @@
+/*
+ * 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 at intel.com>
+ * Vinod Koul <vinod.koul at 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) {
+ 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 14063ab8c7c5..8554889c0694 100644
--- 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 at intel.com>
* Omair M Abdullah <omair.m.abdullah at 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 e5607cf198dc..eed4d2f080f6 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -619,7 +619,13 @@ 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 79dfee1fbdfe..fc703d8c1485 100644
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -155,6 +155,8 @@ 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);
@@ -169,6 +171,7 @@ 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);
--
1.9.0
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