Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/sst-mfld-platform-pcm.c | 112 +++++++++++++++++++++++-------- 1 files changed, 83 insertions(+), 29 deletions(-)

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7c790f5..f4b85e5 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -92,35 +92,6 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { .fifo_size = SST_FIFO_SIZE, };

-/* 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, @@ -316,6 +287,89 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) return ret_val; }

+static struct snd_soc_dai_driver sst_platform_dai[] = { +{ + .name = "media-cpu-dai", + .playback = { + .stream_name = "Media 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 = "Media 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_pcm_trigger(struct snd_pcm_substream *substream, int cmd) {

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/Kconfig | 17 + sound/soc/intel/board/Makefile | 4 + sound/soc/intel/board/merr_dpcm_wm8958.c | 619 ++++++++++++++++++++++++++++++ 3 files changed, 640 insertions(+), 0 deletions(-) create mode 100644 sound/soc/intel/board/Makefile create mode 100644 sound/soc/intel/board/merr_dpcm_wm8958.c

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 3c81b38..422c32d 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -49,3 +49,20 @@ config SND_SOC_INTEL_BYT_RT5640_MACH help This adds audio driver for Intel Baytrail platform based boards with the RT5640 audio codec. + +config SND_SOC_INTEL_MRFLD_WM8958_MACH + tristate "SOC Machine Audio driver for Intel Merrifield MID platform" + depends on X86 + select SND_SOC_WM8994 + select MFD_CORE + select MFD_WM8994 + select REGULATOR_WM8994 + select SND_SST_MFLD_PLATFORM + select SND_SST_MACHINE + select SND_INTEL_SST + default n + help + This adds support for ASoC machine driver for Intel(R) MID Merrifield platform + used as alsa device in audio substem in Intel(R) MID devices + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/board/Makefile b/sound/soc/intel/board/Makefile new file mode 100644 index 0000000..fdfbecf --- /dev/null +++ b/sound/soc/intel/board/Makefile @@ -0,0 +1,4 @@ +# Merrifield board +snd-merr-dpcm-wm8958-objs := merr_dpcm_wm8958.o + +obj-$(CONFIG_SND_SOC_INTEL_MRFLD_WM8958_MACH) += snd-merr-dpcm-wm8958.o diff --git a/sound/soc/intel/board/merr_dpcm_wm8958.c b/sound/soc/intel/board/merr_dpcm_wm8958.c new file mode 100644 index 0000000..4c17746 --- /dev/null +++ b/sound/soc/intel/board/merr_dpcm_wm8958.c @@ -0,0 +1,619 @@ +/* + * merr_dpcm_wm8958.c - ASoc DPCM Machine driver for Intel Merrfield MID platform + * + * Copyright (C) 2013-14 Intel Corp + * Author: 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/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/async.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <linux/input.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include <linux/mfd/wm8994/pdata.h> +#include "../../codecs/wm8994.h" +#include "../platform-libs/controls_v2.h" + +/* Codec PLL output clk rate */ +#define CODEC_SYSCLK_RATE 24576000 +/* Input clock to codec at MCLK1 PIN */ +#define CODEC_IN_MCLK1_RATE 19200000 +/* Input clock to codec at MCLK2 PIN */ +#define CODEC_IN_MCLK2_RATE 32768 +/* define to select between MCLK1 and MCLK2 input to codec as its clock */ +#define CODEC_IN_MCLK1 1 +#define CODEC_IN_MCLK2 2 + +struct mrfld_8958_mc_private { + struct snd_soc_jack jack; + int jack_retry; +}; + + +static inline struct snd_soc_codec *mrfld_8958_get_codec(struct snd_soc_card *card) +{ + bool found = false; + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strstr(codec->name, "wm8994-codec")) { + pr_debug("codec was %s", codec->name); + continue; + } else { + found = true; + break; + } + } + if (found == false) { + pr_warn("%s: cant find codec", __func__); + return NULL; + } + return codec; +} + +/* TODO: find better way of doing this */ +static struct snd_soc_dai *find_codec_dai(struct snd_soc_card *card, const char *dai_name) +{ + int i; + for (i = 0; i < card->num_rtd; i++) { + if (!strcmp(card->rtd[i].codec_dai->name, dai_name)) + return card->rtd[i].codec_dai; + } + pr_err("%s: unable to find codec dai\n", __func__); + /* this should never occur */ + WARN_ON(1); + return NULL; +} + +/* Function to switch the input clock for codec, When audio is in + * progress input clock to codec will be through MCLK1 which is 19.2MHz + * while in off state input clock to codec will be through 32KHz through + * MCLK2 + * card : Sound card structure + * src : Input clock source to codec + */ +static int mrfld_8958_set_codec_clk(struct snd_soc_card *card, int src) +{ + struct snd_soc_dai *aif1_dai = find_codec_dai(card, "wm8994-aif1"); + int ret; + + if (!aif1_dai) + return -ENODEV; + + switch (src) { + case CODEC_IN_MCLK1: + /* Turn ON the PLL to generate required sysclk rate + * from MCLK1 */ + ret = snd_soc_dai_set_pll(aif1_dai, + WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + CODEC_IN_MCLK1_RATE, CODEC_SYSCLK_RATE); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + /* Switch to MCLK1 input */ + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_FLL1, + CODEC_SYSCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set codec sysclk configuration %d\n", + ret); + return ret; + } + break; + case CODEC_IN_MCLK2: + /* Switch to MCLK2 */ + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch to MCLK2: %d", ret); + return ret; + } + /* Turn off PLL for MCLK1 */ + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop the FLL: %d", ret); + return ret; + } + break; + default: + return -EINVAL; + } + return 0; +} + +static int mrfld_wm8958_set_clk_fmt(struct snd_soc_dai *codec_dai) +{ + unsigned int fmt; + int ret = 0; + struct snd_soc_card *card = codec_dai->card; + + pr_err("setting snd_soc_dai_set_tdm_slot\n"); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0, 0, 4, SNDRV_PCM_FORMAT_S24_LE); + if (ret < 0) { + pr_err("can't set codec pcm format %d\n", ret); + return ret; + } + + /* WM8958 slave Mode */ + fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err("can't set codec DAI configuration %d\n", ret); + return ret; + } + + /* FIXME: move this to SYS_CLOCK event handler when codec driver + * dependency is clean. + */ + /* Switch to 19.2MHz MCLK1 input clock for codec */ + ret = mrfld_8958_set_codec_clk(card, CODEC_IN_MCLK1); + + return ret; +} + +static int mrfld_8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + if (!strcmp(codec_dai->name, "wm8994-aif1")) + return mrfld_wm8958_set_clk_fmt(codec_dai); + return 0; +} + +static int mrfld_wm8958_compr_set_params(struct snd_compr_stream *cstream) +{ + return 0; +} + +static const struct snd_soc_pcm_stream mrfld_wm8958_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int merr_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + pr_debug("Invoked %s for dailink %s\n", __func__, rtd->dai_link->name); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int mrfld_8958_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *aif1_dai = find_codec_dai(card, "wm8994-aif1"); + int ret = 0; + + if (!aif1_dai) + return -ENODEV; + + if (dapm->dev != aif1_dai->dev) + return 0; + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + + ret = mrfld_wm8958_set_clk_fmt(aif1_dai); + break; + default: + break; + } + pr_debug("%s card(%s)->bias_level %u\n", __func__, card->name, + card->dapm.bias_level); + return ret; +} + +static int mrfld_8958_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *aif1_dai = find_codec_dai(card, "wm8994-aif1"); + int ret = 0; + + if (!aif1_dai) + return -ENODEV; + + if (dapm->dev != aif1_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + /* We are in stabdba down so */ + /* Switch to 32KHz MCLK2 input clock for codec + */ + ret = mrfld_8958_set_codec_clk(card, CODEC_IN_MCLK2); + break; + default: + break; + } + card->dapm.bias_level = level; + pr_debug("%s card(%s)->bias_level %u\n", __func__, card->name, + card->dapm.bias_level); + return ret; +} + +static const struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static const struct snd_soc_dapm_route map[] = { + { "Headphones", NULL, "HPOUT1L" }, + { "Headphones", NULL, "HPOUT1R" }, + + /* saltbay uses 2 DMICs, other configs may use more so change below + * accordingly + */ + { "DMIC1DAT", NULL, "DMIC" }, + { "DMIC2DAT", NULL, "DMIC" }, + /*{ "DMIC3DAT", NULL, "DMIC" },*/ + /*{ "DMIC4DAT", NULL, "DMIC" },*/ + + /* MICBIAS2 is connected as Bias for AMIC so we link it + * here. Also AMIC wires up to IN1LP pin. + * DMIC is externally connected to 1.8V rail, so no link rqd. + */ + { "AMIC", NULL, "MICBIAS2" }, + { "IN1LP", NULL, "AMIC" }, + + /* SWM map link the SWM outs to codec AIF */ + { "AIF1 Playback", NULL, "ssp2 Tx"}, + { "ssp2 Tx", NULL, "codec_out0"}, + { "ssp2 Tx", NULL, "codec_out1"}, + { "codec_in0", NULL, "ssp2 Rx" }, + { "codec_in1", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture"}, + +}; + +static const struct wm8958_micd_rate micdet_rates[] = { + { 32768, true, 1, 4 }, + { 32768, false, 1, 1 }, + { 44100 * 256, true, 7, 10 }, + { 44100 * 256, false, 7, 10 }, +}; + +static int mrfld_8958_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + unsigned int fmt; + struct snd_soc_codec *codec; + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *aif1_dai = find_codec_dai(card, "wm8994-aif1"); + struct mrfld_8958_mc_private *ctx = snd_soc_card_get_drvdata(card); + + if (!aif1_dai) + return -ENODEV; + + pr_debug("Entry %s\n", __func__); + + ret = snd_soc_dai_set_tdm_slot(aif1_dai, 0, 0, 4, SNDRV_PCM_FORMAT_S24_LE); + if (ret < 0) { + pr_err("can't set codec pcm format %d\n", ret); + return ret; + } + + /* WM8958 slave Mode */ + fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(aif1_dai, fmt); + if (ret < 0) { + pr_err("can't set codec DAI configuration %d\n", ret); + return ret; + } + + mrfld_8958_set_bias_level(card, &card->dapm, SND_SOC_BIAS_OFF); + card->dapm.idle_bias_off = true; + + /* these pins are not used in SB config so mark as nc + * + * LINEOUT1, 2 + * IN1R + * DMICDAT2 + */ + snd_soc_dapm_nc_pin(&card->dapm, "DMIC2DAT"); + snd_soc_dapm_nc_pin(&card->dapm, "LINEOUT1P"); + snd_soc_dapm_nc_pin(&card->dapm, "LINEOUT1N"); + snd_soc_dapm_nc_pin(&card->dapm, "LINEOUT2P"); + snd_soc_dapm_nc_pin(&card->dapm, "LINEOUT2N"); + snd_soc_dapm_nc_pin(&card->dapm, "IN1RN"); + snd_soc_dapm_nc_pin(&card->dapm, "IN1RP"); + + /* Force enable VMID to avoid cold latency constraints */ + snd_soc_dapm_force_enable_pin(&card->dapm, "VMID"); + snd_soc_dapm_sync(&card->dapm); + + codec = mrfld_8958_get_codec(card); + if (!codec) { + pr_err("%s: we didnt find the codec pointer!\n", __func__); + return 0; + } + + ctx->jack_retry = 0; + ret = snd_soc_jack_new(codec, "Intel MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1, + &ctx->jack); + if (ret) { + pr_err("jack creation failed\n"); + return ret; + } + + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_1, KEY_MEDIA); + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + + wm8958_mic_detect(codec, &ctx->jack, NULL, NULL, NULL, NULL); + + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_FILTERS_1, WM8994_AIF1DAC1_MUTE, 0); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_FILTERS_1, WM8994_AIF1DAC2_MUTE, 0); + + /* Micbias1 is always off, so for pm optimizations make sure the micbias1 + * discharge bit is set to floating to avoid discharge in disable state + */ + snd_soc_update_bits(codec, WM8958_MICBIAS1, WM8958_MICB1_DISCH, 0); + + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; +static int mrfld_8958_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops mrfld_8958_ops = { + .startup = mrfld_8958_startup, +}; + +static struct snd_soc_ops mrfld_8958_be_ssp2_ops = { + .hw_params = mrfld_8958_hw_params, +}; +static struct snd_soc_compr_ops mrfld_compr_ops = { + .set_params = mrfld_wm8958_compr_set_params, +}; + +struct snd_soc_dai_link mrfld_8958_msic_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Merrifield Audio Port", + .stream_name = "Saltbay Audio", + .cpu_dai_name = "Headset-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "sst-platform", + .init = mrfld_8958_init, + .ignore_suspend = 1, + .dynamic = 1, + .ops = &mrfld_8958_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Merrifield Compress Port", + .stream_name = "Saltbay Compress", + .platform_name = "sst-platform", + .cpu_dai_name = "Compress-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .init = mrfld_8958_init, + .compr_ops = &mrfld_compr_ops, + }, + + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-platform", + .no_pcm = 1, + .codec_dai_name = "wm8994-aif1", + .codec_name = "wm8994-codec", + .be_hw_params_fixup = merr_codec_fixup, + .ignore_suspend = 1, + .ops = &mrfld_8958_be_ssp2_ops, + }, +}; + +#ifdef CONFIG_PM_SLEEP +static int snd_mrfld_8958_prepare(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_codec *codec; + struct snd_soc_dapm_context *dapm; + + pr_debug("In %s\n", __func__); + + codec = mrfld_8958_get_codec(card); + if (!codec) { + pr_err("%s: couldn't find the codec pointer!\n", __func__); + return -EAGAIN; + } + + pr_debug("found codec %s\n", codec->name); + dapm = &codec->dapm; + + snd_soc_dapm_disable_pin(dapm, "VMID"); + snd_soc_dapm_sync(dapm); + + snd_soc_suspend(dev); + return 0; +} + +static void snd_mrfld_8958_complete(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_codec *codec; + struct snd_soc_dapm_context *dapm; + + pr_debug("In %s\n", __func__); + + codec = mrfld_8958_get_codec(card); + if (!codec) { + pr_err("%s: couldn't find the codec pointer!\n", __func__); + return; + } + + pr_debug("found codec %s\n", codec->name); + dapm = &codec->dapm; + + snd_soc_dapm_force_enable_pin(dapm, "VMID"); + snd_soc_dapm_sync(dapm); + + snd_soc_resume(dev); + return; +} + +static int snd_mrfld_8958_poweroff(struct device *dev) +{ + pr_debug("In %s\n", __func__); + snd_soc_poweroff(dev); + return 0; +} +#else +#define snd_mrfld_8958_prepare NULL +#define snd_mrfld_8958_complete NULL +#define snd_mrfld_8958_poweroff NULL +#endif + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mrfld = { + .name = "wm8958-audio", + .dai_link = mrfld_8958_msic_dailink, + .num_links = ARRAY_SIZE(mrfld_8958_msic_dailink), + .set_bias_level = mrfld_8958_set_bias_level, + .set_bias_level_post = mrfld_8958_set_bias_level_post, + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = map, + .num_dapm_routes = ARRAY_SIZE(map), +}; + +static int snd_mrfld_8958_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct mrfld_8958_mc_private *drv; + + pr_debug("Entry %s\n", __func__); + + drv = kzalloc(sizeof(*drv), GFP_ATOMIC); + if (!drv) { + pr_err("allocation failed\n"); + return -ENOMEM; + } + + /* register the soc card */ + snd_soc_card_mrfld.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_mrfld, drv); + ret_val = snd_soc_register_card(&snd_soc_card_mrfld); + if (ret_val) { + pr_err("snd_soc_register_card failed %d\n", ret_val); + goto unalloc; + } + platform_set_drvdata(pdev, &snd_soc_card_mrfld); + pr_info("%s successful\n", __func__); + return ret_val; + +unalloc: + kfree(drv); + return ret_val; +} + +static int snd_mrfld_8958_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + struct mrfld_8958_mc_private *drv = snd_soc_card_get_drvdata(soc_card); + + pr_debug("In %s\n", __func__); + kfree(drv); + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +const struct dev_pm_ops snd_mrfld_8958_mc_pm_ops = { + .prepare = snd_mrfld_8958_prepare, + .complete = snd_mrfld_8958_complete, + .poweroff = snd_mrfld_8958_poweroff, +}; + +static struct platform_driver snd_mrfld_8958_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "mrfld_wm8958", + .pm = &snd_mrfld_8958_mc_pm_ops, + }, + .probe = snd_mrfld_8958_mc_probe, + .remove = snd_mrfld_8958_mc_remove, +}; + +static int snd_mrfld_8958_driver_init(void) +{ + pr_info("Merrifield Machine Driver mrfld_wm8958 registerd\n"); + return platform_driver_register(&snd_mrfld_8958_mc_driver); +} + +static void snd_mrfld_8958_driver_exit(void) +{ + pr_debug("In %s\n", __func__); + platform_driver_unregister(&snd_mrfld_8958_mc_driver); +} +late_initcall(snd_mrfld_8958_driver_init); +module_exit(snd_mrfld_8958_driver_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) Merrifield MID Machine driver"); +MODULE_AUTHOR("Vinod Koul vinod.koul@linux.intel.com"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mrfld_wm8958");

This patch adds DPCM based supprt for managing mrfld platform

Signed-off-by: Vinod Koul vinod.koul@intel.com --- sound/soc/intel/Kconfig | 3 + sound/soc/intel/platform-libs/controls_v2.h | 754 +++++++++++++ sound/soc/intel/platform-libs/controls_v2_dpcm.c | 1315 ++++++++++++++++++++++ sound/soc/intel/platform-libs/sst_widgets.h | 378 +++++++ sound/soc/intel/platform_ipc_v2.h | 694 ++++++++++++ sound/soc/intel/sst-mfld-platform-compress.c | 8 +- sound/soc/intel/sst-mfld-platform-pcm.c | 384 ++++++- sound/soc/intel/sst-mfld-platform.h | 2 + 8 files changed, 3480 insertions(+), 58 deletions(-) create mode 100644 sound/soc/intel/platform-libs/controls_v2.h create mode 100644 sound/soc/intel/platform-libs/controls_v2_dpcm.c create mode 100644 sound/soc/intel/platform-libs/sst_widgets.h create mode 100644 sound/soc/intel/platform_ipc_v2.h

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 422c32d..1254eb8 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -66,3 +66,6 @@ config SND_SOC_INTEL_MRFLD_WM8958_MACH used as alsa device in audio substem in Intel(R) MID devices Say Y if you have such a device If unsure select "N". + +config SND_INTEL_SST + tristate diff --git a/sound/soc/intel/platform-libs/controls_v2.h b/sound/soc/intel/platform-libs/controls_v2.h new file mode 100644 index 0000000..7988d81 --- /dev/null +++ b/sound/soc/intel/platform-libs/controls_v2.h @@ -0,0 +1,754 @@ +/* + * 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 + * 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, +}; +/* + * 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_MODEM SST_MIX(0) +#define SST_MIX_BT SST_MIX(1) +#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_PROBE SST_MIX(7) +#define SST_MIX_HF_SNS SST_MIX(8) +#define SST_MIX_HF SST_MIX(9) +#define SST_MIX_SPEECH SST_MIX(10) +#define SST_MIX_RXSPEECH SST_MIX(11) +#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_AWARE SST_MIX(16) +#define SST_MIX_VAD SST_MIX(17) +#define SST_MIX_FM SST_MIX(18) + +#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) + +#define SST_BT_FM_MUX_SHIFT 0 +#define SST_VOICE_MODE_SHIFT 1 +#define SST_BT_MODE_SHIFT 2 + +/* in each mixer register we will define one bit for each input */ +#define SST_MIX_IP(x) (x) + +#define SST_IP_MODEM SST_MIX_IP(0) +#define SST_IP_BT SST_MIX_IP(1) +#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_SIDETONE SST_MIX_IP(8) +#define SST_IP_TXSPEECH SST_MIX_IP(9) +#define SST_IP_SPEECH SST_MIX_IP(10) +#define SST_IP_TONE SST_MIX_IP(11) +#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_LOW_PCM0 SST_MIX_IP(15) +#define SST_IP_FM SST_MIX_IP(16) +#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_MODEM_OUT = (0x00 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_BT_OUT = (0x01 << SST_PATH_ID_SHIFT), + 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_PROBE_OUT = (0x07 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_HF_SNS_OUT = (0x08 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_VOICE_UPLINK_REF2 = (0x08 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_HF_OUT = (0x09 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_VOICE_UPLINK_REF1 = (0x09 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPEECH_OUT = (0x0A << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_VOICE_UPLINK = (0x0A << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_RX_SPEECH_OUT = (0x0B << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_VOICE_DOWNLINK = (0x0B << 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_AWARE_OUT = (0x10 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_VAD_OUT = (0x11 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_OUT = (0x12 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_OUT = (0x13 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_FM_OUT = (0x14 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_PROBE1_PIPE_OUT = (0x15 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE2_PIPE_OUT = (0x16 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE3_PIPE_OUT = (0x17 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE4_PIPE_OUT = (0x18 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE5_PIPE_OUT = (0x19 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE6_PIPE_OUT = (0x1A << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE7_PIPE_OUT = (0x1B << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE8_PIPE_OUT = (0x1C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SIDETONE_OUT = (0x1D << SST_PATH_ID_SHIFT), + + /* Start of input paths */ + SST_PATH_INDEX_MODEM_IN = (0x80 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_BT_IN = (0x81 << SST_PATH_ID_SHIFT), + 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_PROBE_IN = (0x87 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_SIDETONE_IN = (0x88 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_TX_SPEECH_IN = (0x89 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_SPEECH_IN = (0x8A << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_TONE_IN = (0x8B << 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_FM_IN = (0x92 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_PROBE1_PIPE_IN = (0x93 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE2_PIPE_IN = (0x94 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE3_PIPE_IN = (0x95 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE4_PIPE_IN = (0x96 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE5_PIPE_IN = (0x97 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE6_PIPE_IN = (0x98 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE7_PIPE_IN = (0x99 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PROBE8_PIPE_IN = (0x9A << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA3_IN = (0x9C << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_LOW_PCM0_IN = (0x9D << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_RESERVED = (0xFF << SST_PATH_ID_SHIFT), +}; + +/* + * switch matrix input path IDs + */ +enum sst_swm_inputs { + SST_SWM_IN_MODEM = (SST_PATH_INDEX_MODEM_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_BT = (SST_PATH_INDEX_BT_IN | SST_DEFAULT_CELL_NBR), + 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_PROBE = (SST_PATH_INDEX_PROBE_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_SIDETONE = (SST_PATH_INDEX_SIDETONE_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_TXSPEECH = (SST_PATH_INDEX_TX_SPEECH_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_SPEECH = (SST_PATH_INDEX_SPEECH_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_TONE = (SST_PATH_INDEX_TONE_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_FM = (SST_PATH_INDEX_FM_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA3 = (SST_PATH_INDEX_MEDIA3_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_LOW_PCM0 = (SST_PATH_INDEX_LOW_PCM0_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR) +}; + +/* + * switch matrix output path IDs + */ +enum sst_swm_outputs { + SST_SWM_OUT_MODEM = (SST_PATH_INDEX_MODEM_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_BT = (SST_PATH_INDEX_BT_OUT | SST_DEFAULT_CELL_NBR), + 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_PROBE = (SST_PATH_INDEX_PROBE_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_HF_SNS = (SST_PATH_INDEX_HF_SNS_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_HF = (SST_PATH_INDEX_HF_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_SPEECH = (SST_PATH_INDEX_SPEECH_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_RXSPEECH = (SST_PATH_INDEX_RX_SPEECH_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_AWARE = (SST_PATH_INDEX_AWARE_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_VAD = (SST_PATH_INDEX_VAD_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_FM = (SST_PATH_INDEX_FM_OUT | SST_DEFAULT_CELL_NBR), + 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_FBA_UL, + SST_TASK_MMX, + SST_TASK_AWARE, + SST_TASK_FBA_DL, +}; + +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_MODEM_OUT, + SST_GAIN_INDEX_MODEM_IN, + SST_GAIN_INDEX_BT_OUT, + SST_GAIN_INDEX_BT_IN, + SST_GAIN_INDEX_FM_OUT, + + SST_GAIN_INDEX_FM_IN, + 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_RX_SPEECH_OUT, + SST_GAIN_INDEX_TX_SPEECH_IN, + + SST_GAIN_INDEX_SPEECH_OUT, + SST_GAIN_INDEX_SPEECH_IN, + SST_GAIN_INDEX_HF_OUT, + SST_GAIN_INDEX_HF_SNS_OUT, + SST_GAIN_INDEX_TONE_IN, + + SST_GAIN_INDEX_SIDETONE_IN, + SST_GAIN_INDEX_PROBE_OUT, + SST_GAIN_INDEX_PROBE_IN, + 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, + SST_GAIN_INDEX_AWARE_OUT, + SST_GAIN_INDEX_VAD_OUT, + + /* Gain IDs for FBA task start here */ + SST_GAIN_INDEX_VOICE_UL, + + /* 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, + FBA_VB_ANA = 37, + FBA_VB_SET_FIR = 38, + FBA_VB_SET_IIR = 39, + SBA_VB_START_TONE = 41, + SBA_VB_STOP_TONE = 42, + FBA_VB_AEC = 47, + FBA_VB_NR_UL = 48, + FBA_VB_AGC = 49, + FBA_VB_WNR = 52, + FBA_VB_SLV = 53, + FBA_VB_NR_DL = 55, + SBA_PROBE = 66, + MMX_PROBE = 66, + FBA_VB_SET_BIQUAD_D_C = 69, + FBA_VB_DUAL_BAND_COMP = 70, + FBA_VB_SNS = 72, + FBA_VB_SER = 78, + FBA_VB_TX_CNI = 80, + SBA_VB_START = 85, + FBA_VB_SET_REF_LINE = 94, + FBA_VB_SET_DELAY_LINE = 95, + FBA_VB_BWX = 104, + FBA_VB_GMM = 105, + FBA_VB_GLC = 107, + FBA_VB_BMF = 111, + FBA_VB_DNR = 113, + MMX_SET_SWM = 114, + 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, + FBA_VB_TNR_UL = 119, + FBA_VB_TNR_DL = 121, + FBA_VB_NLF = 125, + SBA_VB_LPRO = 126, + FBA_VB_MDRP = 127, + SBA_VB_SET_FIR = 128, + SBA_VB_SET_IIR = 129, + SBA_SET_SSP_SLOT_MAP = 130, + AWARE_ENV_CLASS_PARAMS = 130, + VAD_ENV_CLASS_PARAMS = 2049, +}; + +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, +}; + +#define SST_FILL_LOCATION_IDS(dst, cell_idx, pipe_id) do { \ + dst.location_id.p.cell_nbr_idx = (cell_idx); \ + dst.location_id.p.path_id = (pipe_id); \ + } while (0) +#define SST_FILL_LOCATION_ID(dst, loc_id) (\ + dst.location_id.f = (loc_id)) +#define SST_FILL_MODULE_ID(dst, mod_id) (\ + dst.module_id = (mod_id)) + +#define SST_FILL_DESTINATION1(dst, id) do { \ + SST_FILL_LOCATION_ID(dst, (id) & 0xFFFF); \ + SST_FILL_MODULE_ID(dst, ((id) & 0xFFFF0000) >> 16); \ + } while (0) +#define SST_FILL_DESTINATION2(dst, loc_id, mod_id) do { \ + SST_FILL_LOCATION_ID(dst, loc_id); \ + SST_FILL_MODULE_ID(dst, mod_id); \ + } while (0) +#define SST_FILL_DESTINATION3(dst, cell_idx, path_id, mod_id) do { \ + SST_FILL_LOCATION_IDS(dst, cell_idx, path_id); \ + SST_FILL_MODULE_ID(dst, mod_id); \ + } while (0) + +#define SST_FILL_DESTINATION(level, dst, ...) \ + SST_FILL_DESTINATION##level(dst, __VA_ARGS__) +#define SST_FILL_DEFAULT_DESTINATION(dst) \ + SST_FILL_DESTINATION(2, dst, SST_DEFAULT_LOCATION_ID, SST_DEFAULT_MODULE_ID) + +struct sst_destination_id { + union sst_location_id { + struct { + u8 cell_nbr_idx; /* module index */ + u8 path_id; /* pipe_id */ + } __packed p; /* part */ + u16 f; /* full */ + } __packed location_id; + u16 module_id; +} __packed; + +struct sst_dsp_header { + struct sst_destination_id dst; + u16 command_id; + u16 length; +} __packed; + +/* + * + * Common Commands + * + */ +struct sst_cmd_generic { + struct sst_dsp_header header; +} __packed; + +struct swm_input_ids { + struct sst_destination_id input_id; +} __packed; + +struct sst_cmd_set_swm { + struct sst_dsp_header header; + struct sst_destination_id output_id; + u16 switch_state; + u16 nb_inputs; + struct swm_input_ids input[SST_CMD_SWM_MAX_INPUTS]; +} __packed; + +struct sst_cmd_set_media_path { + struct sst_dsp_header header; + u16 switch_state; +} __packed; + +struct pcm_cfg { + u8 s_length:2; + u8 rate:3; + u8 format:3; +} __packed; + +struct sst_cmd_set_speech_path { + struct sst_dsp_header header; + u16 switch_state; + struct { + u16 rsvd:8; + struct pcm_cfg cfg; + } config; +} __packed; + +struct gain_cell { + struct sst_destination_id dest; + s16 cell_gain_left; + s16 cell_gain_right; + u16 gain_time_constant; +} __packed; + +#define NUM_GAIN_CELLS 1 +struct sst_cmd_set_gain_dual { + struct sst_dsp_header header; + u16 gain_cell_num; + struct gain_cell cell_gains[NUM_GAIN_CELLS]; +} __packed; + +struct sst_cmd_set_params { + struct sst_destination_id dst; + u16 command_id; + char params[0]; +} __packed; + +/* + * + * Media (MMX) commands + * + */ + +/* + * + * SBA commands + * + */ +struct sst_cmd_sba_vb_start { + struct sst_dsp_header header; +} __packed; + +union sba_media_loop_params { + struct { + u16 rsvd:8; + struct pcm_cfg cfg; + } part; + u16 full; +} __packed; + +struct sst_cmd_sba_set_media_loop_map { + struct sst_dsp_header header; + u16 switch_state; + union sba_media_loop_params param; + u16 map; +} __packed; + +struct sst_cmd_tone_stop { + struct sst_dsp_header header; + u16 switch_state; +} __packed; + +enum sst_ssp_mode { + SSP_MODE_MASTER = 0, + SSP_MODE_SLAVE = 1, +}; + +enum sst_ssp_pcm_mode { + SSP_PCM_MODE_NORMAL = 0, + SSP_PCM_MODE_NETWORK = 1, +}; + +enum sst_ssp_duplex { + SSP_DUPLEX = 0, + SSP_RX = 1, + SSP_TX = 2, +}; + +enum sst_ssp_fs_frequency { + SSP_FS_8_KHZ = 0, + SSP_FS_16_KHZ = 1, + SSP_FS_44_1_KHZ = 2, + SSP_FS_48_KHZ = 3, +}; + +enum sst_ssp_fs_polarity { + SSP_FS_ACTIVE_LOW = 0, + SSP_FS_ACTIVE_HIGH = 1, +}; + +enum sst_ssp_protocol { + SSP_MODE_PCM = 0, + SSP_MODE_I2S = 1, +}; + +enum sst_ssp_port_id { + SSP_MODEM = 0, + SSP_BT = 1, + SSP_FM = 2, + SSP_CODEC = 3, +}; + +struct sst_cmd_sba_hw_set_ssp { + struct sst_dsp_header header; + u16 selection; /* 0:SSP0(def), 1:SSP1, 2:SSP2 */ + + u16 switch_state; + + u16 nb_bits_per_slots:6; /* 0-32 bits, 24 (def) */ + u16 nb_slots:4; /* 0-8: slots per frame */ + u16 mode:3; /* 0:Master, 1: Slave */ + u16 duplex:3; + + u16 active_tx_slot_map:8; /* Bit map, 0:off, 1:on */ + u16 reserved1:8; + + u16 active_rx_slot_map:8; /* Bit map 0: Off, 1:On */ + u16 reserved2:8; + + u16 frame_sync_frequency; + + u16 frame_sync_polarity:8; + u16 data_polarity:8; + + u16 frame_sync_width; /* 1 to N clocks */ + u16 ssp_protocol:8; + u16 start_delay:8; /* Start delay in terms of clock ticks */ +} __packed; + +#define SST_MAX_TDM_SLOTS 8 + +struct sst_param_sba_ssp_slot_map { + struct sst_dsp_header header; + + u16 param_id; + u16 param_len; + u16 ssp_index; + + u8 rx_slot_map[SST_MAX_TDM_SLOTS]; + u8 tx_slot_map[SST_MAX_TDM_SLOTS]; +} __packed; + +enum { + SST_PROBE_EXTRACTOR = 0, + SST_PROBE_INJECTOR = 1, +}; + +struct sst_cmd_probe { + struct sst_dsp_header header; + + u16 switch_state; + struct sst_destination_id probe_dst; + + u16 shared_mem:1; + u16 probe_in:1; + u16 probe_out:1; + u16 rsvd_1:13; + + u16 rsvd_2:5; + u16 probe_mode:2; + u16 rsvd_3:1; + struct pcm_cfg cfg; + + u16 sm_buf_id; + + u16 gain[6]; + u16 rsvd_4[9]; +} __packed; + +struct sst_probe_config { + const char *name; + u16 loc_id; + u16 mod_id; + u8 task_id; + struct pcm_cfg cfg; +}; + +int sst_mix_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int sst_mix_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int sst_vtsv_enroll_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int sst_vtsv_enroll_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +#endif diff --git a/sound/soc/intel/platform-libs/controls_v2_dpcm.c b/sound/soc/intel/platform-libs/controls_v2_dpcm.c new file mode 100644 index 0000000..12facc3 --- /dev/null +++ b/sound/soc/intel/platform-libs/controls_v2_dpcm.c @@ -0,0 +1,1315 @@ +/* + * controls_v2_dpcm.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 "../platform_ipc_v2.h" +#include "../sst_platform.h" +#include "../sst_platform_pvt.h" +#include "controls_v2.h" +#include "sst_widgets.h" + +static inline void sst_fill_byte_control(char *param, + u8 ipc_msg, u8 block, + u8 task_id, u8 pipe_id, + u16 len, void *cmd_data) +{ + + struct snd_sst_bytes_v2 *byte_data = (struct snd_sst_bytes_v2 *)param; + byte_data->type = SST_CMD_BYTES_SET; + byte_data->ipc_msg = ipc_msg; + byte_data->block = block; + byte_data->task_id = task_id; + byte_data->pipe_id = pipe_id; + + if (len > SST_MAX_BIN_BYTES - sizeof(*byte_data)) { + pr_err("%s: command length too big (%u)", __func__, len); + len = SST_MAX_BIN_BYTES - sizeof(*byte_data); + WARN_ON(1); /* this happens only if code is wrong */ + } + byte_data->len = len; + memcpy(byte_data->bytes, cmd_data, len); + print_hex_dump_bytes("writing to lpe: ", DUMP_PREFIX_OFFSET, + byte_data, len + sizeof(*byte_data)); +} + +static int sst_fill_and_send_cmd_unlocked(struct sst_data *sst, + u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, + void *cmd_data, u16 len) +{ + sst_fill_byte_control(sst->byte_stream, ipc_msg, block, task_id, pipe_id, + len, cmd_data); + return sst_dsp->ops->set_generic_params(SST_SET_BYTE_STREAM, + sst->byte_stream); +} + +/** + * sst_fill_and_send_cmd - generate the IPC message and send it to the FW + * @ipc_msg: type of IPC (CMD, SET_PARAMS, GET_PARAMS) + * @cmd_data: the IPC payload + */ +static int sst_fill_and_send_cmd(struct sst_data *sst, + u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, + void *cmd_data, u16 len) +{ + int ret; + + mutex_lock(&sst->lock); + ret = sst_fill_and_send_cmd_unlocked(sst, ipc_msg, block, task_id, pipe_id, + cmd_data, len); + mutex_unlock(&sst->lock); + + return ret; +} + +/* + * slot map value is a bitfield where each bit represents a FW channel + * + * 3 2 1 0 # 0 = codec0, 1 = codec1 + * RLRLRLRL # 3, 4 = reserved + * + * e.g. slot 0 rx map = 00001100b -> data from slot 0 goes into codec_in1 L,R + */ +static u8 sst_ssp_slot_map[SST_MAX_TDM_SLOTS] = { + 0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default rx map */ +}; + +/* + * channel map value is a bitfield where each bit represents a slot + * + * 76543210 # 0 = slot 0, 1 = slot 1 + * + * e.g. codec1_0 tx map = 00000101b -> data from codec_out1_0 goes into slot 0, 2 + */ +static u8 sst_ssp_channel_map[SST_MAX_TDM_SLOTS] = { + 0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default tx map */ +}; + +static void sst_send_slot_map(struct sst_data *sst) +{ + struct sst_param_sba_ssp_slot_map cmd; + + pr_debug("Enter: %s\n", __func__); + + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + cmd.header.command_id = SBA_SET_SSP_SLOT_MAP; + cmd.header.length = sizeof(struct sst_param_sba_ssp_slot_map) + - sizeof(struct sst_dsp_header); + + cmd.param_id = SBA_SET_SSP_SLOT_MAP; + cmd.param_len = sizeof(cmd.rx_slot_map) + sizeof(cmd.tx_slot_map) + sizeof(cmd.ssp_index); + cmd.ssp_index = SSP_CODEC; + + memcpy(cmd.rx_slot_map, &sst_ssp_slot_map[0], sizeof(cmd.rx_slot_map)); + memcpy(cmd.tx_slot_map, &sst_ssp_channel_map[0], sizeof(cmd.tx_slot_map)); + + sst_fill_and_send_cmd(sst, SST_IPC_IA_SET_PARAMS, SST_FLAG_BLOCKED, + SST_TASK_SBA, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); +} + +int sst_slot_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct sst_enum *e = (struct sst_enum *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = e->max; + + if (uinfo->value.enumerated.item > e->max- 1) + uinfo->value.enumerated.item = e->max- 1; + strcpy(uinfo->value.enumerated.name, + e->texts[uinfo->value.enumerated.item]); + return 0; +} + +/** + * sst_slot_get - get the status of the interleaver/deinterleaver control + * + * Searches the map where the control status is stored, and gets the + * channel/slot which is currently set for this enumerated control. Since it is + * an enumerated control, there is only one possible value. + */ +static int sst_slot_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sst_enum *e = (void *)kcontrol->private_value; + unsigned int ctl_no = e->reg; + unsigned int is_tx = e->tx; + unsigned int val, mux; + u8 *map = is_tx ? sst_ssp_channel_map : sst_ssp_slot_map; + + val = 1 << ctl_no; + /* search which slot/channel has this bit set - there should be only one */ + for (mux = e->max; mux > 0; mux--) + if (map[mux - 1] & val) + break; + + ucontrol->value.enumerated.item[0] = mux; + pr_debug("%s: %s - %s map = %#x\n", __func__, is_tx ? "tx channel" : "rx slot", + e->texts[mux], mux ? map[mux - 1] : -1); + return 0; +} + +/** + * sst_slot_put - set the status of interleaver/deinterleaver control + * + * (de)interleaver controls are defined in opposite sense to be user-friendly + * + * Instead of the enum value being the value written to the register, it is the + * register address; and the kcontrol number (register num) is the value written + * to the register. This is so that there can be only one value for each + * slot/channel since there is only one control for each slot/channel. + * + * This means that whenever an enum is set, we need to clear the bit + * for that kcontrol_no for all the interleaver OR deinterleaver registers + */ +static int sst_slot_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_enum *e = (void *)kcontrol->private_value; + int i; + unsigned int ctl_no = e->reg; + unsigned int is_tx = e->tx; + unsigned int slot_channel_no; + unsigned int val, mux; + + u8 *map = is_tx ? sst_ssp_channel_map : sst_ssp_slot_map; + + val = 1 << ctl_no; + mux = ucontrol->value.enumerated.item[0]; + if (mux > e->max- 1) + return -EINVAL; + + /* first clear all registers of this bit */ + for (i = 0; i < e->max; i++) + map[i] &= ~val; + + if (mux == 0) /* kctl set to 'none' */ + return 0; + + /* offset by one to take "None" into account */ + slot_channel_no = mux - 1; + map[slot_channel_no] |= val; + + pr_debug("%s: %s %s map = %#x\n", __func__, is_tx ? "tx channel" : "rx slot", + e->texts[mux], map[slot_channel_no]); + + if (e->w && e->w->power) + sst_send_slot_map(sst); + return 0; +} + +/* assumes a boolean mux */ +static inline bool get_mux_state(struct sst_data *sst, unsigned int reg, unsigned int shift) +{ + return (sst_reg_read(sst, reg, shift, 1) == 1); +} + +static void sst_send_algo_cmd(struct sst_data *sst, + struct sst_algo_control *bc) +{ + int len; + struct sst_cmd_set_params *cmd; + + /* bc->max includes sizeof algos + length field */ + len = sizeof(cmd->dst) + sizeof(cmd->command_id) + bc->max; + + cmd = kzalloc(len, GFP_KERNEL); + if (cmd == NULL) { + pr_err("Failed to send cmd, kzalloc failed\n"); + return; + } + + SST_FILL_DESTINATION(2, cmd->dst, bc->pipe_id, bc->module_id); + cmd->command_id = bc->cmd_id; + memcpy(cmd->params, bc->params, bc->max); + + sst_fill_and_send_cmd(sst, SST_IPC_IA_SET_PARAMS, SST_FLAG_BLOCKED, + bc->task_id, 0, cmd, len); + kfree(cmd); +} + +/** + * sst_find_and_send_pipe_algo - send all the algo parameters for a pipe + * + * The algos which are in each pipeline are sent to the firmware one by one + */ +static void sst_find_and_send_pipe_algo(struct sst_data *sst, + const char *pipe, struct sst_ids *ids) +{ + struct sst_algo_control *bc; + struct module *algo = NULL; + + pr_debug("Enter: %s, widget=%s\n", __func__, pipe); + + list_for_each_entry(algo, &ids->algo_list, node) { + bc = (void *)algo->kctl->private_value; + + pr_debug("Found algo control name=%s pipe=%s\n", algo->kctl->id.name, pipe); + sst_send_algo_cmd(sst, bc); + } +} + +int sst_algo_bytes_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct sst_algo_control *bc = (void *)kcontrol->private_value; + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = bc->max; + + /* allocate space to cache the algo parameters in the driver */ + if (bc->params == NULL) { + bc->params = devm_kzalloc(platform->dev, bc->max, GFP_KERNEL); + if (bc->params == NULL) { + pr_err("kzalloc failed\n"); + return -ENOMEM; + } + } + return 0; +} + +static int sst_algo_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sst_algo_control *bc = (void *)kcontrol->private_value; + + switch (bc->type) { + case SST_ALGO_PARAMS: + if (bc->params) + memcpy(ucontrol->value.bytes.data, bc->params, bc->max); + break; + case SST_ALGO_BYPASS: + ucontrol->value.integer.value[0] = bc->bypass ? 1 : 0; + pr_debug("%s: bypass %d\n", __func__, bc->bypass); + break; + default: + pr_err("Invalid Input- algo type:%d\n", bc->type); + return -EINVAL; + + } + return 0; +} + +static int sst_algo_control_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_algo_control *bc = (void *)kcontrol->private_value; + + pr_debug("in %s control_name=%s\n", __func__, kcontrol->id.name); + switch (bc->type) { + case SST_ALGO_PARAMS: + if (bc->params) + memcpy(bc->params, ucontrol->value.bytes.data, bc->max); + break; + case SST_ALGO_BYPASS: + bc->bypass = !!ucontrol->value.integer.value[0]; + break; + default: + pr_err("Invalid Input- algo type:%ld\n", ucontrol->value.integer.value[0]); + return -EINVAL; + } + /*if pipe is enabled, need to send the algo params from here */ + if (bc->w && bc->w->power) + sst_send_algo_cmd(sst, bc); + + return 0; +} + +static int sst_gain_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = mc->stereo ? 2 : 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + return 0; +} + +/** + * sst_send_gain_cmd - send the gain algorithm IPC to the FW + * @gv: the stored value of gain (also contains rampduration) + * @mute: flag that indicates whether this was called from the + * digital_mute callback or directly. If called from the + * digital_mute callback, module will be muted/unmuted based on this + * flag. The flag is always 0 if called directly. + * + * The user-set gain value is sent only if the user-controllable 'mute' control + * is OFF (indicated by gv->mute). Otherwise, the mute value (MIN value) is + * sent. + */ +static void sst_send_gain_cmd(struct sst_data *sst, struct sst_gain_value *gv, + u16 task_id, u16 loc_id, u16 module_id, int mute) +{ + struct sst_cmd_set_gain_dual cmd; + pr_debug("%s\n", __func__); + + cmd.header.command_id = MMX_SET_GAIN; + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + cmd.gain_cell_num = 1; + + if (mute || gv->mute) { + cmd.cell_gains[0].cell_gain_left = SST_GAIN_MIN_VALUE; + cmd.cell_gains[0].cell_gain_right = SST_GAIN_MIN_VALUE; + } else { + cmd.cell_gains[0].cell_gain_left = gv->l_gain; + cmd.cell_gains[0].cell_gain_right = gv->r_gain; + } + SST_FILL_DESTINATION(2, cmd.cell_gains[0].dest, + loc_id, module_id); + cmd.cell_gains[0].gain_time_constant = gv->ramp_duration; + + cmd.header.length = sizeof(struct sst_cmd_set_gain_dual) + - sizeof(struct sst_dsp_header); + + sst_fill_and_send_cmd(sst, SST_IPC_IA_SET_PARAMS, SST_FLAG_BLOCKED, + task_id, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); +} + +static int sst_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value; + struct sst_gain_value *gv = mc->gain_val; + + switch (mc->type) { + case SST_GAIN_TLV: + ucontrol->value.integer.value[0] = gv->l_gain; + ucontrol->value.integer.value[1] = gv->r_gain; + break; + case SST_GAIN_MUTE: + ucontrol->value.integer.value[0] = gv->mute ? 1 : 0; + break; + case SST_GAIN_RAMP_DURATION: + ucontrol->value.integer.value[0] = gv->ramp_duration; + break; + default: + pr_err("Invalid Input- gain type:%d\n", mc->type); + return -EINVAL; + }; + return 0; +} + +static int sst_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value; + struct sst_gain_value *gv = mc->gain_val; + + switch (mc->type) { + case SST_GAIN_TLV: + gv->l_gain = ucontrol->value.integer.value[0]; + gv->r_gain = ucontrol->value.integer.value[1]; + pr_debug("%s: %s: Volume %d, %d\n", __func__, mc->pname, gv->l_gain, gv->r_gain); + break; + case SST_GAIN_MUTE: + gv->mute = !!ucontrol->value.integer.value[0]; + pr_debug("%s: %s: Mute %d\n", __func__, mc->pname, gv->mute); + break; + case SST_GAIN_RAMP_DURATION: + gv->ramp_duration = ucontrol->value.integer.value[0]; + pr_debug("%s: %s: RampDuration %d\n", __func__, mc->pname, gv->ramp_duration); + break; + default: + pr_err("Invalid Input- gain type:%d\n", mc->type); + return -EINVAL; + }; + + if (mc->w && mc->w->power) + sst_send_gain_cmd(sst, gv, mc->task_id, + mc->pipe_id | mc->instance_id, mc->module_id, 0); + return 0; +} + +static void sst_set_pipe_gain(struct sst_ids *ids, struct sst_data *sst, int mute); + +static void sst_send_pipe_module_params(struct snd_soc_dapm_widget *w) +{ + struct sst_data *sst = snd_soc_platform_get_drvdata(w->platform); + struct sst_ids *ids = w->priv; + + sst_find_and_send_pipe_algo(sst, w->name, ids); + sst_set_pipe_gain(ids, sst, 0); +} + +static int sst_generic_modules_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + sst_send_pipe_module_params(w); + return 0; +} + +static const DECLARE_TLV_DB_SCALE(sst_gain_tlv_common, SST_GAIN_MIN_VALUE * 10, 10, 0); + +/* Look up table to convert MIXER SW bit regs to SWM inputs */ +static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = { + [SST_IP_CODEC0] = SST_SWM_IN_CODEC0, + [SST_IP_CODEC1] = SST_SWM_IN_CODEC1, + [SST_IP_LOOP0] = SST_SWM_IN_SPROT_LOOP, + [SST_IP_LOOP1] = SST_SWM_IN_MEDIA_LOOP1, + [SST_IP_LOOP2] = SST_SWM_IN_MEDIA_LOOP2, + [SST_IP_PCM0] = SST_SWM_IN_PCM0, + [SST_IP_PCM1] = SST_SWM_IN_PCM1, + [SST_IP_MEDIA0] = SST_SWM_IN_MEDIA0, + [SST_IP_MEDIA1] = SST_SWM_IN_MEDIA1, + [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 *sst, int mute) +{ + struct sst_gain_mixer_control *mc; + struct sst_gain_value *gv; + struct 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(sst, 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 *sst = snd_soc_platform_get_drvdata(w->platform); + struct sst_ids *ids = w->priv; + bool set_mixer = false; + int val = sst->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(sst, 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_aware_controls, SST_MIX_AWARE); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_vad_controls, SST_MIX_VAD); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_hf_sns_controls, SST_MIX_HF_SNS); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_hf_controls, SST_MIX_HF); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_speech_controls, SST_MIX_SPEECH); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_rxspeech_controls, SST_MIX_RXSPEECH); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls, SST_MIX_CODEC0); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls, SST_MIX_CODEC1); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_bt_controls, SST_MIX_BT); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_fm_controls, SST_MIX_FM); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_modem_controls, SST_MIX_MODEM); + +void sst_handle_vb_timer(struct snd_soc_platform *p, bool enable) +{ + struct sst_cmd_generic cmd; + struct sst_data *sst = snd_soc_platform_get_drvdata(p); + static int timer_usage; + + if (enable) + cmd.header.command_id = SBA_VB_START; + else + cmd.header.command_id = SBA_IDLE; + pr_debug("%s: enable=%u, usage=%d\n", __func__, enable, timer_usage); + + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + cmd.header.length = 0; + + if (enable) + sst_dsp->ops->power(true); + + mutex_lock(&sst->lock); + if (enable) + timer_usage++; + else + timer_usage--; + + /* Send the command only if this call is the first enable or last + * disable + */ + if ((enable && (timer_usage == 1)) || + (!enable && (timer_usage == 0))) + sst_fill_and_send_cmd_unlocked(sst, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + SST_TASK_SBA, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); + mutex_unlock(&sst->lock); + + if (!enable) + sst_dsp->ops->power(false); +} + +#define SST_SSP_CODEC_MUX 0 +#define SST_SSP_CODEC_DOMAIN 0 + +static const int sst_ssp_mux_shift[SST_NUM_SSPS] = { + [SST_SSP2] = -1, +}; + +static const int sst_ssp_domain_shift[SST_NUM_SSPS][SST_MAX_SSP_MUX] = { + [SST_SSP2][0] = -1, +}; + +/** + * sst_ssp_config - contains SSP configuration for different UCs + * + * The 3-D array contains SSP configuration for different SSPs for different + * domains (e.g. NB, WB), as well as muxed SSPs. + * + * The first dimension has SSP number + * The second dimension has SSP Muxing (e.g. BT/FM muxed on same SSP) + * The third dimension has SSP domains (e.g. NB/WB for BT) + */ +static const struct sst_ssp_config +sst_ssp_configs[SST_NUM_SSPS][SST_MAX_SSP_MUX][SST_MAX_SSP_DOMAINS] = { + [SST_SSP2] = { + [SST_SSP_CODEC_MUX] = { + [SST_SSP_CODEC_DOMAIN] = { + .ssp_id = SSP_CODEC, + .bits_per_slot = 24, + .slots = 4, + .ssp_mode = SSP_MODE_MASTER, + .pcm_mode = SSP_PCM_MODE_NETWORK, + .duplex = SSP_DUPLEX, + .ssp_protocol = SSP_MODE_PCM, + .fs_width = 1, + .fs_frequency = SSP_FS_48_KHZ, + .active_slot_map = 0xF, + .start_delay = 0, + }, + }, + }, +}; + +#define SST_SSP_CFG(wssp_no) \ + (const struct sst_ssp_cfg){ .ssp_config = &sst_ssp_configs[wssp_no], \ + .ssp_number = wssp_no, \ + .mux_shift = &sst_ssp_mux_shift[wssp_no], \ + .domain_shift = &sst_ssp_domain_shift[wssp_no], } + +void send_ssp_cmd(struct snd_soc_platform *platform, const char *id, bool enable) +{ + struct sst_cmd_sba_hw_set_ssp cmd; + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + unsigned int domain, mux; + int domain_shift, mux_shift, ssp_no; + const struct sst_ssp_config *config; + const struct sst_ssp_cfg *ssp; + + + pr_err("Enter:%s, enable=%d port_name=%s\n", __func__, enable, id); + + if (strcmp(id, "ssp2-port") == 0) + ssp_no = SST_SSP2; + else + return; + + ssp = &SST_SSP_CFG(ssp_no); + + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + cmd.header.command_id = SBA_HW_SET_SSP; + cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp) + - sizeof(struct sst_dsp_header); + mux_shift = *ssp->mux_shift; + mux = (mux_shift == -1) ? 0 : get_mux_state(sst, SST_MUX_REG, mux_shift); + domain_shift = (*ssp->domain_shift)[mux]; + domain = (domain_shift == -1) ? 0 : get_mux_state(sst, SST_MUX_REG, domain_shift); + + config = &(*ssp->ssp_config)[mux][domain]; + pr_debug("%s: ssp_id: %u, mux: %d, domain: %d\n", __func__, + config->ssp_id, mux, domain); + + if (enable) + cmd.switch_state = SST_SWITCH_ON; + else + cmd.switch_state = SST_SWITCH_OFF; + + cmd.selection = config->ssp_id; + cmd.nb_bits_per_slots = config->bits_per_slot; + cmd.nb_slots = config->slots; + cmd.mode = config->ssp_mode | (config->pcm_mode << 1); + cmd.duplex = config->duplex; + cmd.active_tx_slot_map = config->active_slot_map; + cmd.active_rx_slot_map = config->active_slot_map; + cmd.frame_sync_frequency = config->fs_frequency; + cmd.frame_sync_polarity = SSP_FS_ACTIVE_HIGH; + cmd.data_polarity = 1; + cmd.frame_sync_width = config->fs_width; + cmd.ssp_protocol = config->ssp_protocol; + cmd.start_delay = config->start_delay; + cmd.reserved1 = cmd.reserved2 = 0xFF; + + sst_fill_and_send_cmd(sst, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + SST_TASK_SBA, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); +} + +static int sst_set_be_modules(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct sst_data *sst = snd_soc_platform_get_drvdata(w->platform); + + pr_debug("Enter: %s, widget=%s\n", __func__, w->name); + + if (SND_SOC_DAPM_EVENT_ON(event)) { + sst_send_slot_map(sst); + sst_send_pipe_module_params(w); + } + return 0; +} + +static int sst_set_media_path(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct sst_cmd_set_media_path cmd; + struct sst_data *sst = snd_soc_platform_get_drvdata(w->platform); + struct sst_ids *ids = w->priv; + + pr_debug("%s: widget=%s\n", __func__, w->name); + pr_debug("%s: task=%u, location=%#x\n", __func__, + ids->task_id, ids->location_id); + + if (SND_SOC_DAPM_EVENT_ON(event)) + cmd.switch_state = SST_PATH_ON; + else + cmd.switch_state = SST_PATH_OFF; + + SST_FILL_DESTINATION(2, cmd.header.dst, + ids->location_id, SST_DEFAULT_MODULE_ID); + + /* MMX_SET_MEDIA_PATH == SBA_SET_MEDIA_PATH */ + cmd.header.command_id = MMX_SET_MEDIA_PATH; + cmd.header.length = sizeof(struct sst_cmd_set_media_path) + - sizeof(struct sst_dsp_header); + + sst_fill_and_send_cmd(sst, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + ids->task_id, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); + + if (SND_SOC_DAPM_EVENT_ON(event)) + sst_send_pipe_module_params(w); + return 0; +} + +static int sst_set_media_loop(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct sst_cmd_sba_set_media_loop_map cmd; + struct sst_data *sst = snd_soc_platform_get_drvdata(w->platform); + struct sst_ids *ids = w->priv; + + pr_debug("Enter:%s, widget=%s\n", __func__, w->name); + if (SND_SOC_DAPM_EVENT_ON(event)) + cmd.switch_state = SST_SWITCH_ON; + else + cmd.switch_state = SST_SWITCH_OFF; + + SST_FILL_DESTINATION(2, cmd.header.dst, + ids->location_id, SST_DEFAULT_MODULE_ID); + + cmd.header.command_id = SBA_SET_MEDIA_LOOP_MAP; + cmd.header.length = sizeof(struct sst_cmd_sba_set_media_loop_map) + - sizeof(struct sst_dsp_header); + cmd.param.part.cfg.rate = 2; /* 48khz */ + + cmd.param.part.cfg.format = ids->format; /* stereo/Mono */ + cmd.param.part.cfg.s_length = 1; /* 24bit left justified*/ + cmd.map = 0; /* Algo sequence: Gain - DRP - FIR - IIR */ + + sst_fill_and_send_cmd(sst, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + SST_TASK_SBA, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); + if (SND_SOC_DAPM_EVENT_ON(event)) + sst_send_pipe_module_params(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 */ + 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[] = { + {"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", + "slot 4", "slot 5", "slot 6", "slot 7", /* not supported by FW */ +}; + +static const char * const channel_names[] = { + "none", + "codec_out0_0", "codec_out0_1", "codec_out1_0", "codec_out1_1", + "codec_out2_0", "codec_out2_1", "codec_out3_0", "codec_out3_1", /* not supported by FW */ +}; + +#define SST_INTERLEAVER(xpname, slot_name, slotno) \ + SST_SSP_SLOT_CTL(xpname, "interleaver", slot_name, slotno, true, \ + channel_names, sst_slot_get, sst_slot_put) + +#define SST_DEINTERLEAVER(xpname, channel_name, channel_no) \ + SST_SSP_SLOT_CTL(xpname, "deinterleaver", channel_name, channel_no, false, \ + slot_names, sst_slot_get, sst_slot_put) + +static const struct snd_kcontrol_new sst_slot_controls[] = { + SST_INTERLEAVER("codec_out", "slot 0", 0), + SST_INTERLEAVER("codec_out", "slot 1", 1), + SST_INTERLEAVER("codec_out", "slot 2", 2), + SST_INTERLEAVER("codec_out", "slot 3", 3), + SST_DEINTERLEAVER("codec_in", "codec_in0_0", 0), + SST_DEINTERLEAVER("codec_in", "codec_in0_1", 1), + SST_DEINTERLEAVER("codec_in", "codec_in1_0", 2), + SST_DEINTERLEAVER("codec_in", "codec_in1_1", 3), +}; + +/* Gain helper with min/max set */ +#define SST_GAIN(name, path_id, task_id, instance, gain_var) \ + SST_GAIN_KCONTROLS(name, "gain", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE, \ + SST_GAIN_TC_MIN, SST_GAIN_TC_MAX, \ + sst_gain_get, sst_gain_put, \ + SST_MODULE_ID_GAIN_CELL, path_id, instance, task_id, \ + sst_gain_tlv_common, gain_var) + +#define SST_VOLUME(name, path_id, task_id, instance, gain_var) \ + SST_GAIN_KCONTROLS(name, "volume", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE, \ + SST_GAIN_TC_MIN, SST_GAIN_TC_MAX, \ + sst_gain_get, sst_gain_put, \ + SST_MODULE_ID_VOLUME, path_id, instance, task_id, \ + sst_gain_tlv_common, gain_var) + +#define SST_NUM_GAINS 36 +static struct sst_gain_value sst_gains[SST_NUM_GAINS]; + +static const struct snd_kcontrol_new sst_gain_controls[] = { + SST_GAIN("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[0]), + SST_GAIN("media1_in", SST_PATH_INDEX_MEDIA1_IN, SST_TASK_MMX, 0, &sst_gains[1]), + SST_GAIN("media2_in", SST_PATH_INDEX_MEDIA2_IN, SST_TASK_MMX, 0, &sst_gains[2]), + SST_GAIN("media3_in", SST_PATH_INDEX_MEDIA3_IN, SST_TASK_MMX, 0, &sst_gains[3]), + + SST_GAIN("pcm0_in", SST_PATH_INDEX_PCM0_IN, SST_TASK_SBA, 0, &sst_gains[4]), + SST_GAIN("pcm1_in", SST_PATH_INDEX_PCM1_IN, SST_TASK_SBA, 0, &sst_gains[5]), + SST_GAIN("pcm1_out", SST_PATH_INDEX_PCM1_OUT, SST_TASK_SBA, 0, &sst_gains[7]), + SST_GAIN("pcm2_out", SST_PATH_INDEX_PCM2_OUT, SST_TASK_SBA, 0, &sst_gains[8]), + + SST_GAIN("codec_in0", SST_PATH_INDEX_CODEC_IN0, SST_TASK_SBA, 0, &sst_gains[20]), + SST_GAIN("codec_in1", SST_PATH_INDEX_CODEC_IN1, SST_TASK_SBA, 0, &sst_gains[21]), + SST_GAIN("codec_out0", SST_PATH_INDEX_CODEC_OUT0, SST_TASK_SBA, 0, &sst_gains[22]), + SST_GAIN("codec_out1", SST_PATH_INDEX_CODEC_OUT1, SST_TASK_SBA, 0, &sst_gains[23]), + SST_GAIN("media_loop1_out", SST_PATH_INDEX_MEDIA_LOOP1_OUT, SST_TASK_SBA, 0, &sst_gains[30]), + SST_GAIN("media_loop2_out", SST_PATH_INDEX_MEDIA_LOOP2_OUT, SST_TASK_SBA, 0, &sst_gains[31]), + SST_GAIN("sprot_loop_out", SST_PATH_INDEX_SPROT_LOOP_OUT, SST_TASK_SBA, 0, &sst_gains[32]), + SST_VOLUME("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[33]), +}; + +static const struct snd_kcontrol_new sst_algo_controls[] = { + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "fir", 272, SST_MODULE_ID_FIR_24, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "iir", 300, SST_MODULE_ID_IIR_24, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "mdrp", 286, SST_MODULE_ID_MDRP, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "fir", 272, SST_MODULE_ID_FIR_24, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "iir", 300, SST_MODULE_ID_IIR_24, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "mdrp", 286, SST_MODULE_ID_MDRP, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP), + SST_ALGO_KCONTROL_BYTES("sprot_loop_out", "lpro", 192, SST_MODULE_ID_SPROT, + SST_PATH_INDEX_SPROT_LOOP_OUT, 0, SST_TASK_SBA, SBA_VB_LPRO), + SST_ALGO_KCONTROL_BYTES("codec_in0", "dcr", 52, SST_MODULE_ID_FILT_DCR, + SST_PATH_INDEX_CODEC_IN0, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("codec_in1", "dcr", 52, SST_MODULE_ID_FILT_DCR, + SST_PATH_INDEX_CODEC_IN1, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + +}; + +static inline bool is_sst_dapm_widget(struct snd_soc_dapm_widget *w) +{ + if ((w->id == snd_soc_dapm_pga) || + (w->id == snd_soc_dapm_aif_in) || + (w->id == snd_soc_dapm_aif_out) || + (w->id == snd_soc_dapm_input) || + (w->id == snd_soc_dapm_output) || + (w->id == snd_soc_dapm_mixer)) + return true; + else + return false; +} + +/** + * sst_send_pipe_gains - send gains for the front-end DAIs + * + * The gains in the pipes connected to the front-ends are muted/unmuted + * automatically via the digital_mute() DAPM callback. This function sends the + * gains for the front-end pipes. + */ +int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) +{ + struct snd_soc_platform *platform = dai->platform; + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_path *p = NULL; + + pr_debug("%s: enter, dai-name=%s dir=%d\n", __func__, dai->name, stream); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + pr_debug("Stream name=%s\n", dai->playback_widget->name); + w = dai->playback_widget; + list_for_each_entry(p, &w->sinks, list_source) { + if (p->connected && !p->connected(w, p->sink)) + continue; + + if (p->connect && p->sink->power && is_sst_dapm_widget(p->sink)) { + struct sst_ids *ids = p->sink->priv; + + pr_debug("send gains for widget=%s\n", p->sink->name); + sst_set_pipe_gain(ids, sst, mute); + } + } + } else { + pr_debug("Stream name=%s\n", dai->capture_widget->name); + w = dai->capture_widget; + list_for_each_entry(p, &w->sources, list_sink) { + if (p->connected && !p->connected(w, p->sink)) + continue; + + if (p->connect && p->source->power && is_sst_dapm_widget(p->source)) { + struct sst_ids *ids = p->source->priv; + + pr_debug("send gain for widget=%s\n", p->source->name); + sst_set_pipe_gain(ids, sst, mute); + } + } + } + return 0; +} + +/** + * sst_fill_module_list - populate the list of modules/gains for a pipe + * + * + * Fills the widget pointer in the kcontrol private data, and also fills the + * kcontrol pointer in the widget private data. + * + * Widget pointer is used to send the algo/gain in the .put() handler if the + * widget is powerd on. + * + * Kcontrol pointer is used to send the algo/gain in the widget power ON/OFF + * event handler. Each widget (pipe) has multiple algos stored in the algo_list. + */ +static int sst_fill_module_list(struct snd_kcontrol *kctl, + struct snd_soc_dapm_widget *w, int type) +{ + struct module *module = NULL; + struct sst_ids *ids = w->priv; + + module = devm_kzalloc(w->platform->dev, sizeof(*module), GFP_KERNEL); + if (!module) { + pr_err("kzalloc block failed\n"); + return -ENOMEM; + } + + if (type == SST_MODULE_GAIN) { + struct sst_gain_mixer_control *mc = (void *)kctl->private_value; + + mc->w = w; + module->kctl = kctl; + list_add_tail(&module->node, &ids->gain_list); + } else if (type == SST_MODULE_ALGO) { + struct sst_algo_control *bc = (void *)kctl->private_value; + + bc->w = w; + module->kctl = kctl; + list_add_tail(&module->node, &ids->algo_list); + } + + return 0; +} + +/** + * sst_fill_widget_module_info - fill list of gains/algos for the pipe + * @widget: pipe modelled as a DAPM widget + * + * Fill the list of gains/algos for the widget by looking at all the card + * controls and comparing the name of the widget with the first part of control + * name. First part of control name contains the pipe name (widget name). + */ +static int sst_fill_widget_module_info(struct snd_soc_dapm_widget *w, + struct snd_soc_platform *platform) +{ + struct snd_kcontrol *kctl; + int index, ret = 0; + struct snd_card *card = platform->card->snd_card; + char *idx; + + down_read(&card->controls_rwsem); + + list_for_each_entry(kctl, &card->controls, list) { + idx = strstr(kctl->id.name, " "); + if (idx == NULL) + continue; + index = strlen(kctl->id.name) - strlen(idx); + if (strstr(kctl->id.name, "volume") && + !strncmp(kctl->id.name, w->name, index)) + ret = sst_fill_module_list(kctl, w, SST_MODULE_GAIN); + else if (strstr(kctl->id.name, "params") && + !strncmp(kctl->id.name, w->name, index)) + ret = sst_fill_module_list(kctl, w, SST_MODULE_ALGO); + else if (strstr(kctl->id.name, "mute") && + !strncmp(kctl->id.name, w->name, index)) { + struct sst_gain_mixer_control *mc = (void *)kctl->private_value; + mc->w = w; + } else if (strstr(kctl->id.name, "interleaver") && + !strncmp(kctl->id.name, w->name, index)) { + struct sst_enum *e = (void *)kctl->private_value; + e->w = w; + } else if (strstr(kctl->id.name, "deinterleaver") && + !strncmp(kctl->id.name, w->name, index)) { + struct sst_enum *e = (void *)kctl->private_value; + e->w = w; + } + if (ret < 0) { + up_read(&card->controls_rwsem); + return ret; + } + } + up_read(&card->controls_rwsem); + return 0; +} + +/** + * sst_fill_linked_widgets - fill the parent pointer for the linked widget + */ +static void sst_fill_linked_widgets(struct snd_soc_platform *platform, + struct sst_ids *ids) +{ + struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_context *dapm = &platform->dapm; + + unsigned int len = strlen(ids->parent_wname); + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!strncmp(ids->parent_wname, w->name, len)) { + ids->parent_w = w; + break; + } + } +} + +/** + * sst_map_modules_to_pipe - fill algo/gains list for all pipes + */ +static int sst_map_modules_to_pipe(struct snd_soc_platform *platform) +{ + struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_context *dapm = &platform->dapm; + int ret = 0; + + list_for_each_entry(w, &dapm->card->widgets, list) { + if (w->platform && is_sst_dapm_widget(w) && (w->priv)) { + struct sst_ids *ids = w->priv; + + pr_debug("widget type=%d name=%s\n", w->id, w->name); + INIT_LIST_HEAD(&ids->algo_list); + INIT_LIST_HEAD(&ids->gain_list); + ret = sst_fill_widget_module_info(w, platform); + if (ret < 0) + return ret; + /* fill linked widgets */ + if (ids->parent_wname != NULL) + sst_fill_linked_widgets(platform, ids); + } + } + return 0; +} + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) +{ + int i, ret = 0; + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + + sst->byte_stream = devm_kzalloc(platform->dev, + SST_MAX_BIN_BYTES, GFP_KERNEL); + if (!sst->byte_stream) { + pr_err("%s: kzalloc failed\n", __func__); + return -ENOMEM; + } + sst->widget = devm_kzalloc(platform->dev, + SST_NUM_WIDGETS * sizeof(*sst->widget), + GFP_KERNEL); + if (!sst->widget) { + pr_err("%s: kzalloc failed\n", __func__); + 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; + sst_gains[i].l_gain = SST_GAIN_VOLUME_DEFAULT; + sst_gains[i].r_gain = SST_GAIN_VOLUME_DEFAULT; + sst_gains[i].ramp_duration = SST_GAIN_RAMP_DURATION_DEFAULT; + } + + snd_soc_add_platform_controls(platform, sst_gain_controls, + ARRAY_SIZE(sst_gain_controls)); + + snd_soc_add_platform_controls(platform, sst_algo_controls, + ARRAY_SIZE(sst_algo_controls)); + snd_soc_add_platform_controls(platform, sst_slot_controls, + ARRAY_SIZE(sst_slot_controls)); + + ret = sst_map_modules_to_pipe(platform); + + return ret; +} diff --git a/sound/soc/intel/platform-libs/sst_widgets.h b/sound/soc/intel/platform-libs/sst_widgets.h new file mode 100644 index 0000000..25deae6 --- /dev/null +++ b/sound/soc/intel/platform-libs/sst_widgets.h @@ -0,0 +1,378 @@ +/* + * sst_widgets.h - Intel helpers to generate FW widgets + * + * Copyright (C) 2013-14 Intel Corp + * Author: Omair Mohammed Abdullah omair.m.abdullah@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_WIDGETS_H__ +#define __SST_WIDGETS_H__ + +#include <sound/soc.h> +#include <sound/tlv.h> + +#define SST_MODULE_GAIN 1 +#define SST_MODULE_ALGO 2 + +#define SST_FMT_MONO 0 +#define SST_FMT_STEREO 3 + +/* physical SSP numbers */ +enum { + SST_SSP0 = 0, + SST_SSP1, + SST_SSP2, + SST_SSP_LAST = SST_SSP2, +}; + +#define SST_NUM_SSPS (SST_SSP_LAST + 1) /* physical SSPs */ +#define SST_MAX_SSP_MUX 2 /* single SSP muxed between pipes */ +#define SST_MAX_SSP_DOMAINS 2 /* domains present in each pipe */ + +struct module { + struct snd_kcontrol *kctl; + struct list_head node; +}; + +struct sst_ssp_config { + u8 ssp_id; + u8 bits_per_slot; + u8 slots; + u8 ssp_mode; + u8 pcm_mode; + u8 duplex; + u8 ssp_protocol; + u8 fs_frequency; + u8 active_slot_map; + u8 start_delay; + u16 fs_width; +}; + +struct sst_ssp_cfg { + const u8 ssp_number; + const int *mux_shift; + const int (*domain_shift)[SST_MAX_SSP_MUX]; + const struct sst_ssp_config (*ssp_config)[SST_MAX_SSP_MUX][SST_MAX_SSP_DOMAINS]; +}; + +struct sst_ids { + u16 location_id; + u16 module_id; + u8 task_id; + u8 format; + u8 reg; + const char *parent_wname; + struct snd_soc_dapm_widget *parent_w; + struct list_head algo_list; + struct list_head gain_list; + const struct sst_pcm_format *pcm_fmt; +}; + + +#define SST_AIF_IN(wname, wevent) \ +{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \ +} + +#define SST_AIF_OUT(wname, wevent) \ +{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \ +} + +#define SST_INPUT(wname, wevent) \ +{ .id = snd_soc_dapm_input, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \ +} + +#define SST_OUTPUT(wname, wevent) \ +{ .id = snd_soc_dapm_output, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \ +} + +#define SST_DAPM_OUTPUT(wname, wloc_id, wtask_id, wformat, wevent) \ +{ .id = snd_soc_dapm_output, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .location_id = wloc_id, .task_id = wtask_id,\ + .pcm_fmt = wformat, } \ +} + +#define SST_PATH(wname, wtask, wloc_id, wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \ + .kcontrol_news = NULL, .num_kcontrols = 0, \ + .event = wevent, .event_flags = wflags, \ + .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, } \ +} + +#define SST_LINKED_PATH(wname, wtask, wloc_id, linked_wname, wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \ + .kcontrol_news = NULL, .num_kcontrols = 0, \ + .event = wevent, .event_flags = wflags, \ + .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \ + .parent_wname = linked_wname} \ +} + +#define SST_PATH_MEDIA_LOOP(wname, wtask, wloc_id, wformat, wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \ + .kcontrol_news = NULL, .num_kcontrols = 0, \ + .event = wevent, .event_flags = wflags, \ + .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \ + .format = wformat,} \ +} + +/* output is triggered before input */ +#define SST_PATH_INPUT(name, task_id, loc_id, event) \ + SST_PATH(name, task_id, loc_id, event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) + +#define SST_PATH_LINKED_INPUT(name, task_id, loc_id, linked_wname, event) \ + SST_LINKED_PATH(name, task_id, loc_id, linked_wname, event, \ + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) + +#define SST_PATH_OUTPUT(name, task_id, loc_id, event) \ + SST_PATH(name, task_id, loc_id, event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD) + +#define SST_PATH_LINKED_OUTPUT(name, task_id, loc_id, linked_wname, event) \ + SST_LINKED_PATH(name, task_id, loc_id, linked_wname, event, \ + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD) + +#define SST_PATH_MEDIA_LOOP_OUTPUT(name, task_id, loc_id, format, event) \ + SST_PATH_MEDIA_LOOP(name, task_id, loc_id, format, event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD) + + +#define SST_SWM_MIXER(wname, wreg, wtask, wloc_id, wcontrols, wevent) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \ + .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols),\ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD | \ + SND_SOC_DAPM_POST_REG, \ + .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \ + .reg = wreg } \ +} + +enum sst_gain_kcontrol_type { + SST_GAIN_TLV, + SST_GAIN_MUTE, + SST_GAIN_RAMP_DURATION, +}; + +struct sst_gain_mixer_control { + bool stereo; + enum sst_gain_kcontrol_type type; + struct sst_gain_value *gain_val; + int max; + int min; + u16 instance_id; + u16 module_id; + u16 pipe_id; + u16 task_id; + char pname[44]; + struct snd_soc_dapm_widget *w; +}; + +struct sst_gain_value { + u16 ramp_duration; + s16 l_gain; + s16 r_gain; + bool mute; +}; + +#define SST_GAIN_VOLUME_DEFAULT (-1440) +#define SST_GAIN_RAMP_DURATION_DEFAULT 5 /* timeconstant */ +#define SST_GAIN_MUTE_DEFAULT true + +#define SST_GAIN_KCONTROL_TLV(xname, xhandler_get, xhandler_put, \ + xmod, xpipe, xinstance, xtask, tlv_array, xgain_val, \ + xmin, xmax, xpname) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = sst_gain_ctl_info,\ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct sst_gain_mixer_control) \ + { .stereo = true, .max = xmax, .min = xmin, .type = SST_GAIN_TLV, \ + .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\ + .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname} + +#define SST_GAIN_KCONTROL_INT(xname, xhandler_get, xhandler_put, \ + xmod, xpipe, xinstance, xtask, xtype, xgain_val, \ + xmin, xmax, xpname) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sst_gain_ctl_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct sst_gain_mixer_control) \ + { .stereo = false, .max = xmax, .min = xmin, .type = xtype, \ + .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\ + .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname} + +#define SST_GAIN_KCONTROL_BOOL(xname, xhandler_get, xhandler_put,\ + xmod, xpipe, xinstance, xtask, xgain_val, xpname) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_bool_ext, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct sst_gain_mixer_control) \ + { .stereo = false, .type = SST_GAIN_MUTE, \ + .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\ + .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname} + +#define SST_CONTROL_NAME(xpname, xmname, xinstance, xtype) \ + xpname " " xmname " " #xinstance " " xtype + +#define SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, xtype, xsubmodule) \ + xpname " " xmname " " #xinstance " " xtype " " xsubmodule + +/* + * 3 Controls for each Gain module + * e.g. - pcm0_in gain 0 volume + * - pcm0_in gain 0 rampduration + * - pcm0_in gain 0 mute + */ +#define SST_GAIN_KCONTROLS(xpname, xmname, xmin_gain, xmax_gain, xmin_tc, xmax_tc, \ + xhandler_get, xhandler_put, \ + xmod, xpipe, xinstance, xtask, tlv_array, xgain_val) \ + { SST_GAIN_KCONTROL_INT(SST_CONTROL_NAME(xpname, xmname, xinstance, "rampduration"), \ + xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, SST_GAIN_RAMP_DURATION, \ + xgain_val, xmin_tc, xmax_tc, xpname) }, \ + { SST_GAIN_KCONTROL_BOOL(SST_CONTROL_NAME(xpname, xmname, xinstance, "mute"), \ + xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, \ + xgain_val, xpname) } ,\ + { SST_GAIN_KCONTROL_TLV(SST_CONTROL_NAME(xpname, xmname, xinstance, "volume"), \ + xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, tlv_array, \ + xgain_val, xmin_gain, xmax_gain, xpname) } + +#define SST_GAIN_TC_MIN 5 +#define SST_GAIN_TC_MAX 5000 +#define SST_GAIN_MIN_VALUE -1440 /* in 0.1 DB units */ +#define SST_GAIN_MAX_VALUE 360 + +enum sst_algo_kcontrol_type { + SST_ALGO_PARAMS, + SST_ALGO_BYPASS, +}; + +struct sst_algo_control { + enum sst_algo_kcontrol_type type; + int max; + u16 module_id; + u16 pipe_id; + u16 task_id; + u16 cmd_id; + bool bypass; + unsigned char *params; + struct snd_soc_dapm_widget *w; +}; + +/* size of the control = size of params + size of length field */ +#define SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, xmod, xtask, xcmd) \ + (struct sst_algo_control){ \ + .max = xcount + sizeof(u16), .type = xtype, .module_id = xmod, \ + .pipe_id = xpipe, .task_id = xtask, .cmd_id = xcmd, \ + } + +#define SST_ALGO_KCONTROL(xname, xcount, xmod, xpipe, \ + xtask, xcmd, xtype, xinfo, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = xinfo, .get = xget, .put = xput, \ + .private_value = (unsigned long)& \ + SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, \ + xmod, xtask, xcmd), \ +} + +#define SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, \ + xpipe, xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "params"), \ + xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS, \ + sst_algo_bytes_ctl_info, \ + sst_algo_control_get, sst_algo_control_set) + +#define SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask) \ + SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "bypass"), \ + 0, xmod, xpipe, xtask, 0, SST_ALGO_BYPASS, \ + snd_soc_info_bool_ext, \ + sst_algo_control_get, sst_algo_control_set) + +#define SST_ALGO_BYPASS_PARAMS(xpname, xmname, xcount, xmod, xpipe, \ + xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask), \ + SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, xpipe, xinstance, xtask, xcmd) + +#define SST_COMBO_ALGO_KCONTROL_BYTES(xpname, xmname, xsubmod, xcount, xmod, \ + xpipe, xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL(SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, "params", \ + xsubmod), \ + xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS, \ + sst_algo_bytes_ctl_info, \ + sst_algo_control_get, sst_algo_control_set) + + +struct sst_enum { + bool tx; + unsigned short reg; + unsigned int max; + const char * const *texts; + struct snd_soc_dapm_widget *w; +}; + +/* only 4 slots/channels supported atm */ +#define SST_SSP_SLOT_ENUM(s_ch_no, is_tx, xtexts) \ + (struct sst_enum){ .reg = s_ch_no, .tx = is_tx, .max = 4+1, .texts = xtexts, } + +#define SST_SLOT_CTL_NAME(xpname, xmname, s_ch_name) \ + xpname " " xmname " " s_ch_name + +#define SST_SSP_SLOT_CTL(xpname, xmname, s_ch_name, s_ch_no, is_tx, xtexts, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = SST_SLOT_CTL_NAME(xpname, xmname, s_ch_name), \ + .info = sst_slot_enum_info, \ + .get = xget, .put = xput, \ + .private_value = (unsigned long)&SST_SSP_SLOT_ENUM(s_ch_no, is_tx, xtexts), \ +} + +#define SST_MUX_CTL_NAME(xpname, xinstance) \ + xpname " " #xinstance + +#define SST_SSP_MUX_ENUM(xreg, xshift, xtexts) \ + (struct soc_enum){ .reg = xreg, .texts = xtexts, .shift_l = xshift, \ + .shift_r = xshift, .items = ARRAY_SIZE(xtexts), } + +#define SST_SSP_MUX_CTL(xpname, xinstance, xreg, xshift, xtexts, xget, xput) \ + SOC_DAPM_ENUM_EXT(SST_MUX_CTL_NAME(xpname, xinstance), \ + SST_SSP_MUX_ENUM(xreg, xshift, xtexts), \ + xget, xput) + +struct sst_probe_value { + unsigned int val; + const struct soc_enum *p_enum; +}; + +#define SST_PROBE_CTL_NAME(dir, num, type) \ + dir #num " " type + +#define SST_PROBE_ENUM(xname, xenum, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sst_probe_enum_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct sst_probe_value) \ + { .val = 0, .p_enum = &xenum } } + +#endif diff --git a/sound/soc/intel/platform_ipc_v2.h b/sound/soc/intel/platform_ipc_v2.h new file mode 100644 index 0000000..a718f9c --- /dev/null +++ b/sound/soc/intel/platform_ipc_v2.h @@ -0,0 +1,694 @@ +/* +* platform_ipc_v2.h - Intel MID Platform driver FW IPC definitions +* +* Copyright (C) 2008-10 Intel Corporation +* Author: Vinod Koul vinod.koul@intel.com +* Harsha Priya priya.harsha@intel.com +* Dharageswari R dharageswari.r@intel.com +* KP Jeeja jeeja.kp@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. +* +* You should have received a copy of the GNU General Public License along +* with this program; if not, write to the Free Software Foundation, Inc., +* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. +* +* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +* +* This driver exposes the audio engine functionalities to the ALSA +* and middleware. +* This file has definitions shared between the firmware and driver +*/ +#ifndef __PLATFORM_IPC_V2_H__ +#define __PLATFORM_IPC_V2_H__ + +#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 0x680 +#define SST_TIME_STAMP_BYT 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_SST_VB_RESET 0x28 +#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. + */ + +/* SCU IPC for resetting & power gating the LPE through SCU */ +#define IPC_SCU_LPE_RESET 0xA3 + +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, + SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ + SST_CODEC_TYPE_MP3, + SST_CODEC_TYPE_MP24, + SST_CODEC_TYPE_AAC, + SST_CODEC_TYPE_AACP, + SST_CODEC_TYPE_eAACP, + SST_CODEC_TYPE_WMA9, + SST_CODEC_TYPE_WMA10, + SST_CODEC_TYPE_WMA10P, + SST_CODEC_TYPE_RA, + SST_CODEC_TYPE_DDAC3, + SST_CODEC_TYPE_STEREO_TRUE_HD, + SST_CODEC_TYPE_STEREO_HD_PLUS, + + /* VOICE CODEC Type Definitions */ + SST_CODEC_TYPE_VOICE_PCM = 0x21, /* Pass through voice codec */ +}; + +enum sst_algo_types { + SST_ALGO_SRC = 0x64, + SST_ALGO_MIXER = 0x65, + SST_ALGO_DOWN_MIXER = 0x66, + SST_ALGO_VTSV = 0x73, + SST_ALGO_AUDCLASSIFIER = 0x80, + SST_ALGO_VOLUME_CONTROL = 0x92, + SST_ALGO_GEQ = 0x99, +}; + +enum stream_type { + SST_STREAM_TYPE_NONE = 0, + SST_STREAM_TYPE_MUSIC = 1, + SST_STREAM_TYPE_NORMAL = 2, + SST_STREAM_TYPE_PROBE = 3, + SST_STREAM_TYPE_LONG_PB = 4, + SST_STREAM_TYPE_LOW_LATENCY = 5, +}; + +enum sst_error_codes { + /* Error code,response to msgId: Description */ + /* Common error codes */ + SST_SUCCESS = 0, /* Success */ + SST_ERR_INVALID_STREAM_ID = 1, + SST_ERR_INVALID_MSG_ID = 2, + SST_ERR_INVALID_STREAM_OP = 3, + SST_ERR_INVALID_PARAMS = 4, + SST_ERR_INVALID_CODEC = 5, + SST_ERR_INVALID_MEDIA_TYPE = 6, + SST_ERR_STREAM_ERR = 7, + + /* IPC specific error codes */ + SST_IPC_ERR_CALL_BACK_NOT_REGD = 8, + SST_IPC_ERR_STREAM_NOT_ALLOCATED = 9, + SST_IPC_ERR_STREAM_ALLOC_FAILED = 10, + SST_IPC_ERR_GET_STREAM_FAILED = 11, + SST_ERR_MOD_NOT_AVAIL = 12, + SST_ERR_MOD_DNLD_RQD = 13, + SST_ERR_STREAM_STOPPED = 14, + SST_ERR_STREAM_IN_USE = 15, + + /* Capture specific error codes */ + SST_CAP_ERR_INCMPLTE_CAPTURE_MSG = 16, + SST_CAP_ERR_CAPTURE_FAIL = 17, + SST_CAP_ERR_GET_DDR_NEW_SGLIST = 18, + SST_CAP_ERR_UNDER_RUN = 19, + SST_CAP_ERR_OVERFLOW = 20, + + /* Playback specific error codes*/ + SST_PB_ERR_INCMPLTE_PLAY_MSG = 21, + SST_PB_ERR_PLAY_FAIL = 22, + SST_PB_ERR_GET_DDR_NEW_SGLIST = 23, + + /* Codec manager specific error codes */ + SST_LIB_ERR_LIB_DNLD_REQUIRED = 24, + SST_LIB_ERR_LIB_NOT_SUPPORTED = 25, + + /* Library manager specific error codes */ + SST_SCC_ERR_PREP_DNLD_FAILED = 26, + SST_SCC_ERR_LIB_DNLD_RES_FAILED = 27, + /* Scheduler specific error codes */ + SST_SCH_ERR_FAIL = 28, + + /* DMA specific error codes */ + SST_DMA_ERR_NO_CHNL_AVAILABLE = 29, + SST_DMA_ERR_INVALID_INPUT_PARAMS = 30, + SST_DMA_ERR_CHNL_ALREADY_SUSPENDED = 31, + SST_DMA_ERR_CHNL_ALREADY_STARTED = 32, + SST_DMA_ERR_CHNL_NOT_ENABLED = 33, + SST_DMA_ERR_TRANSFER_FAILED = 34, + + SST_SSP_ERR_ALREADY_ENABLED = 35, + SST_SSP_ERR_ALREADY_DISABLED = 36, + SST_SSP_ERR_NOT_INITIALIZED = 37, + SST_SSP_ERR_SRAM_NO_DMA_DATA = 38, + + /* Other error codes */ + SST_ERR_MOD_INIT_FAIL = 39, + + /* FW init error codes */ + SST_RDR_ERR_IO_DEV_SEL_NOT_ALLOWED = 40, + SST_RDR_ERR_ROUTE_ALREADY_STARTED = 41, + SST_RDR_ERR_IO_DEV_SEL_FAILED = 42, + SST_RDR_PREP_CODEC_DNLD_FAILED = 43, + + /* Memory debug error codes */ + SST_ERR_DBG_MEM_READ_FAIL = 44, + SST_ERR_DBG_MEM_WRITE_FAIL = 45, + SST_ERR_INSUFFICIENT_INPUT_SG_LIST = 46, + SST_ERR_INSUFFICIENT_OUTPUT_SG_LIST = 47, + + SST_ERR_BUFFER_NOT_AVAILABLE = 48, + SST_ERR_BUFFER_NOT_ALLOCATED = 49, + SST_ERR_INVALID_REGION_TYPE = 50, + SST_ERR_NULL_PTR = 51, + SST_ERR_INVALID_BUFFER_SIZE = 52, + SST_ERR_INVALID_BUFFER_INDEX = 53, + + /*IIPC specific error codes */ + SST_IIPC_QUEUE_FULL = 54, + SST_IIPC_ERR_MSG_SND_FAILED = 55, + SST_PB_ERR_UNDERRUN_OCCURED = 56, + SST_RDR_INSUFFICIENT_MIXER_BUFFER = 57, + SST_INVALID_TIME_SLOTS = 58, +}; + +enum dbg_mem_data_type { + /* Data type of debug read/write */ + DATA_TYPE_U32, + DATA_TYPE_U16, + DATA_TYPE_U8, +}; + +enum dbg_type { + NO_DEBUG = 0, + SRAM_DEBUG, + PTI_DEBUG, +}; + +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; + +struct ipc_dsp_effects_info { + u16 cmd_id; + u16 length; + u16 sel_pos; + u16 sel_algo_id; + u16 cpu_load; /* CPU load indication */ + u16 memory_usage; /* Data Memory usage */ + u32 flags; /* effect engine caps/requirements flags */ +} __packed; + +struct ipc_effect_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 */ +} __packed; + +struct ipc_effect_payload { + struct ipc_effect_dsp_hdr dsp_hdr; + char *data; +}; + +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; + +/* SST to IA memory read debug message */ +struct ipc_sst_ia_dbg_mem_rw { + u16 num_bytes;/* Maximum of MAX_DBG_RW_BYTES */ + u16 data_type;/* enum: dbg_mem_data_type */ + u32 address; /* Memory address of data memory of data_type */ + u8 rw_bytes[MAX_DBG_RW_BYTES];/* Maximum of 64 bytes can be RW */ +} __packed; + +struct ipc_sst_ia_dbg_loop_back { + u16 num_dwords; /* Maximum of MAX_DBG_RW_BYTES */ + u16 increment_val;/* Increments dwords by this value, 0- no increment */ + u32 lpbk_dwords[MAX_LOOP_BACK_DWORDS];/* Maximum of 8 dwords loopback */ +} __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*/ + u8 use_offload_path; /* 0-PCM using period elpased & ALSA interfaces + 1-PCM stream via compressed interface */ + u8 reserved2; + u32 sfreq; /* Sampling rate in Hz */ + u8 channel_map[8]; +} __packed; + +/* MP3 Music Parameters Message */ +struct snd_mp3_params { + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 crc_check; /* crc_check - disable (0) or enable (1) */ + u8 reserved1; /* unused*/ +}; + +#define AAC_BIT_STREAM_ADTS 0 +#define AAC_BIT_STREAM_ADIF 1 +#define AAC_BIT_STREAM_RAW 2 + +/* AAC Music Parameters Message */ +struct snd_aac_params { + u8 num_chan; /* 1=Mono, 2=Stereo*/ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */ + u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ + u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ + u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ + u8 reser1; + u16 reser2; +}; + +/* WMA Music Parameters Message */ +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 */ + u16 format_tag; /* Format Tag */ + u16 block_align; /* packet size */ + u16 wma_encode_opt;/* Encoder option */ + u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */ + u8 reserved; /* reserved */ +}; + +/* Codec params struture */ +union snd_sst_codec_params { + struct snd_pcm_params pcm_params; + struct snd_mp3_params mp3_params; + struct snd_aac_params aac_params; + struct snd_wma_params wma_params; +}; + +/* Address and size info of a frame buffer in DDR */ +struct sst_address_info { + __u32 addr; /* Address at IA */ + __u32 size; /* Size of the buffer */ +} __packed; + +/* Additional params for Alloc struct*/ +struct snd_sst_alloc_params_ext { + __u16 sg_count; + __u16 reserved; + __u32 frag_size; /*Number of samples after which period elapsed + message is sent valid only if path = 0*/ + 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 /* __PLATFORMDRV_IPC_V2_H__ */ diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 02abd19..ca58f4c 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -105,9 +105,11 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, /* 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-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index f4b85e5..836d9a6 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -28,6 +28,7 @@ #include <sound/soc.h> #include <sound/compress_driver.h> #include "sst-mfld-platform.h" +#include "platform-libs/controls_v2.h"

struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -92,6 +93,39 @@ 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, SST_DEV_MAP_IN_USE}, + {MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, SST_DEV_MAP_IN_USE}, + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, SST_DEV_MAP_IN_USE}, +}; + +static int sst_platform_media_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) { + struct snd_sst_runtime_params params_data; + int channels = slots; + + /* registering with SST driver to get access to SST APIs to use */ + if (!sst_dsp) { + pr_err("sst: DSP not registered\n"); + return -EIO; + } + params_data.type = SST_SET_CHANNEL_INFO; + params_data.str_id = SND_SST_DEVICE_IHF; + params_data.size = sizeof(channels); + params_data.addr = &channels; + return sst_dsp->ops->set_generic_params(SST_SET_RUNTIME_PARAMS, + (void *)&params_data); +} + +static int sst_media_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + + pr_debug("%s: enter, mute=%d dai-name=%s dir=%d\n", __func__, mute, dai->name, stream); + + return sst_send_pipe_gains(dai, stream, mute); +}

/* helper functions */ void sst_set_stream_status(struct sst_runtime_stream *stream, @@ -114,45 +148,163 @@ 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; + + pr_debug("period_size = %d\n", alloc_param->frag_size); + pr_debug("ring_buf_addr = 0x%x\n", alloc_param->ring_buf_info[0].addr); +} static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) -{ - - 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); + 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; + + /* 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)); + pr_debug("sfreq= %d, wd_sz = %d\n", + param->uc.pcm_params.sfreq, param->uc.pcm_params.pcm_wd_sz); + +} + +static int sst_get_stream_mapping(int dev, int sdev, int dir, + struct sst_dev_stream_map *map, int size, u8 pipe_id, + const struct sst_lowlatency_deepbuff *ll_db) +{ + int index; + + if (map == NULL) + return -EINVAL; + + pr_debug("dev %d sdev %d dir %d\n", dev, sdev, dir); + + /* index 0 is not used in stream map */ + for (index = 1; index < size; index++) { + if ((map[index].dev_num == dev) && + (map[index].subdev_num == sdev) && + (map[index].direction == dir)) { + if (map[index].status == SST_DEV_MAP_IN_USE) { + return index; + } else if (map[index].status == SST_DEV_MAP_FREE) { + map[index].status = SST_DEV_MAP_IN_USE; + + pr_debug("%s: pipe_id 0%x index %d", __func__, + map[index].device_id, index); + + return index; + } + } + } + return 0; }

-static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +int sst_fill_stream_params(void *substream, + 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 + pstream = (struct snd_pcm_substream *)substream; + + str_params->stream_type = SST_STREAM_TYPE_MUSIC; + + /* For pcm streams */ + if (pstream) { + index = sst_get_stream_mapping(pstream->pcm->device, + pstream->number, pstream->stream, + map, map_size, ctx->pipe_id, &ctx->ll_db); + 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; + + pr_debug("str_id = %d, device_type = 0x%x, task = %d", + str_params->stream_id, str_params->device_type, + str_params->task); + + str_params->ops = (u8)pstream->stream; + } + + if (cstream) { + /* FIXME: Add support for subdevice number in + * snd_compr_stream */ + index = sst_get_stream_mapping(cstream->device->device, + 0, cstream->direction, + map, map_size, ctx->pipe_id, &ctx->ll_db); + 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; + pr_debug("compress str_id = %d, device_type = 0x%x, task = %d", + str_params->stream_id, str_params->device_type, + str_params->task); + + str_params->ops = (u8)cstream->direction; + } + return 0; +} + +#define CALC_PERIODTIME(period_size, rate) (((period_size) * 1000) / (rate)) + +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; + 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); 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); - } + str_params.aparams = alloc_params; + str_params.codec = SST_CODEC_TYPE_PCM; + + ctx->ll_db.period_time = CALC_PERIODTIME(substream->runtime->period_size, + substream->runtime->rate); + + /* fill the device type and stream id to pass to SST driver */ + ret_val = sst_fill_stream_params(substream, ctx, &str_params, false); + pr_debug("platform prepare: fill stream params ret_val = 0x%x\n", ret_val); + if (ret_val < 0) + return ret_val; + + stream->stream_info.str_id = str_params.stream_id; + ret_val = stream->ops->open(&str_params); pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); if (ret_val < 0) @@ -242,25 +394,55 @@ static int sst_platform_open(struct snd_pcm_substream *substream) /* allocate memory for SST API set */ runtime->private_data = stream;

- return 0; +static void sst_free_stream_in_use(struct sst_dev_stream_map *map, int str_id) +{ + if (map[str_id].dev_num == MERR_DPCM_AUDIO) { + /* Do nothing in capture for audio device */ + if ((map[str_id].dev_num == MERR_DPCM_AUDIO) && + (map[str_id].direction == SNDRV_PCM_STREAM_CAPTURE)) + return; + if ((map[str_id].task_id == SST_TASK_ID_MEDIA) && + (map[str_id].status == SST_DEV_MAP_IN_USE)) { + pr_debug("str_id %d device_id 0x%x\n", str_id, map[str_id].device_id); + map[str_id].status = SST_DEV_MAP_FREE; + map[str_id].device_id = PIPE_RSVD; + } + } + return; }

static int sst_platform_close(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; + struct sst_data *ctx = snd_soc_platform_get_drvdata(dai->platform);

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); + sst_free_stream_in_use(ctx->pdata->pdev_strm_map, str_id); module_put(sst->dev->driver->owner); kfree(stream); - return ret_val; +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_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; @@ -274,8 +456,8 @@ static int sst_platform_pcm_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); @@ -287,18 +469,78 @@ 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) +{ + pr_debug("%s\n", __func__); + + 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 int sst_enable_ssp(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pr_debug("In %s :dai=%s pb=%d cp= %d dai_active=%d id=%d\n", __func__, + dai->name, dai->playback_active, dai->capture_active, dai->active, dai->id); + if (!dai->active) { + sst_handle_vb_timer(dai->platform, true); + send_ssp_cmd(dai->platform, dai->name, 1); + } + return 0; +} + +static void sst_disable_ssp(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pr_debug("In %s :dai=%s pb=%d cp= %d dai_active=%d id=%d\n", __func__, + dai->name, dai->playback_active, dai->capture_active, dai->active, dai->id); + if (!dai->active) { + send_ssp_cmd(dai->platform, dai->name, 0); + sst_handle_vb_timer(dai->platform, false); + } +} + +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, + .set_tdm_slot = sst_platform_media_set_tdm_slot, + .mute_stream = sst_media_digital_mute, +}; + +static struct snd_soc_dai_ops sst_compr_dai_ops = { + .mute_stream = sst_media_digital_mute, +}; + +static struct snd_soc_dai_ops sst_be_dai_ops = { + .startup = sst_enable_ssp, + .shutdown = sst_disable_ssp, +}; + static struct snd_soc_dai_driver sst_platform_dai[] = { { .name = "media-cpu-dai", + .ops = &sst_media_dai_ops, .playback = { - .stream_name = "Media 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 = "Media Capture", + .stream_name = "Headset Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, @@ -308,6 +550,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { { .name = "compress-cpu-dai", .compress_dai = 1, + .ops = &sst_compr_dai_ops, .playback = { .stream_name = "Compress Playback", .channels_min = SST_STEREO, @@ -370,6 +613,27 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { }, };

+static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai_link = rtd->dai_link; + + pr_debug("sst_platform_open called:%s\n", dai_link->cpu_dai_name); + if (substream->pcm->internal) + return 0; + runtime = substream->runtime; + runtime->hw = sst_platform_pcm_hw; + return 0; +} + +static int sst_platform_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai_link = rtd->dai_link; + pr_debug("sst_platform_close called:%s\n", dai_link->cpu_dai_name); + return 0; +} static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -377,6 +641,8 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, struct sst_runtime_stream *stream; int str_cmd, status;

+ if (substream->pcm->internal) + return 0; pr_debug("sst_platform_pcm_trigger called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; @@ -431,32 +697,16 @@ 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); + substream->runtime->delay = str_info->pcm_delay; + 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) @@ -485,11 +735,21 @@ 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) +{ + pr_debug("Enter:%s\n", __func__); + 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 = { @@ -499,10 +759,24 @@ static const struct snd_soc_component_driver sst_component = {

static int sst_platform_probe(struct platform_device *pdev) { + struct sst_data *sst; int ret; + struct sst_platform_data *pdata = pdev->dev.platform_data;

pr_debug("sst_platform_probe called\n"); - sst = NULL; + sst = devm_kzalloc(&pdev->dev, sizeof(*sst), 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); + sst_pdev = &pdev->dev; + sst->pdata = pdata; + mutex_init(&sst->lock); + dev_set_drvdata(&pdev->dev, sst); + 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 6c5e7dc..67b375b 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -126,6 +126,7 @@ struct compress_sst_ops { struct sst_ops { int (*open) (struct sst_stream_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); };

@@ -143,6 +144,7 @@ struct sst_device { char *name; struct device *dev; struct sst_ops *ops; + struct platform_device *pdev; struct compress_sst_ops *compr_ops; };