From 9a0d5113ac0ee513224ca2e5011b3a566de16207 Mon Sep 17 00:00:00 2001 From: Thomas Petazzoni Date: Thu, 30 Jan 2014 18:14:06 +0100 Subject: ASoC: kirkwood: enable Kirkwood driver for mvebu platforms The audio unit found in the Armada 370 SoC is similar to the one used in the Marvell Kirkwood and Marvell Dove SoCs. Therefore, this commit allows the Kirkwood audio driver to be built on mvebu platforms, and adds an additional compatible string to identify the Armada 370 variant of the audio unit. Signed-off-by: Thomas Petazzoni Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 78ed4a4..764a0ef 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST + depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3920a5e..9f84222 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -633,6 +633,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) static struct of_device_id mvebu_audio_of_match[] = { { .compatible = "marvell,kirkwood-audio" }, { .compatible = "marvell,dove-audio" }, + { .compatible = "marvell,armada370-audio" }, { } }; MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); -- cgit v1.1 From a355d67817eccdbe869020f8f5a3123f21f8106f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 12 Feb 2014 13:42:50 +0000 Subject: ASoC: Intel: Add a mfld prefix to Intel SST drivers. Resent with correct email for Mark. In order to differentiate the different Intel SST audio core drivers we need to rename the current drivers with a mfld prefix. This also includes renaming in the Makefile and Kconfig Acked-by: Vinod Koul Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 4 +- sound/soc/intel/Makefile | 4 +- sound/soc/intel/sst-mfld-dsp.h | 134 +++++++ sound/soc/intel/sst-mfld-platform.c | 725 ++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 153 ++++++++ sound/soc/intel/sst_dsp.h | 134 ------- sound/soc/intel/sst_platform.c | 725 ------------------------------------ sound/soc/intel/sst_platform.h | 153 -------- 8 files changed, 1016 insertions(+), 1016 deletions(-) create mode 100644 sound/soc/intel/sst-mfld-dsp.h create mode 100644 sound/soc/intel/sst-mfld-platform.c create mode 100644 sound/soc/intel/sst-mfld-platform.h delete mode 100644 sound/soc/intel/sst_dsp.h delete mode 100644 sound/soc/intel/sst_platform.c delete mode 100644 sound/soc/intel/sst_platform.h (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 61c10bf..4d9d0a5 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -2,12 +2,12 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC select SND_SOC_SN95031 - select SND_SST_PLATFORM + select SND_SST_MFLD_PLATFORM help This adds support for ASoC machine driver for Intel(R) MID Medfield 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". -config SND_SST_PLATFORM +config SND_SST_MFLD_PLATFORM tristate diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 6398833..eb899fc 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,5 +1,5 @@ -snd-soc-sst-platform-objs := sst_platform.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o snd-soc-mfld-machine-objs := mfld_machine.o -obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h new file mode 100644 index 0000000..3b63edc --- /dev/null +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -0,0 +1,134 @@ +#ifndef __SST_MFLD_DSP_H__ +#define __SST_MFLD_DSP_H__ +/* + * sst_mfld_dsp.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-12 Intel Corporation + * Authors: Vinod Koul + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +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, +}; + +enum stream_type { + SST_STREAM_TYPE_NONE = 0, + SST_STREAM_TYPE_MUSIC = 1, +}; + +struct snd_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u8 use_offload_path; + u8 reserved2; + u16 reserved3; + u8 channel_map[8]; +} __packed; + +/* MP3 Music Parameters Message */ +struct snd_mp3_params { + u16 codec; + 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*/ + u16 reserved2; /* Unused */ +} __packed; + +#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 { + u16 codec; + 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 */ + u16 reser2; + u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ + u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ + u8 reser1; + u16 reser3; +} __packed; + +/* WMA Music Parameters Message */ +struct snd_wma_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + 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 */ +} __packed; + +/* 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; +} __packed; + +/* Address and size info of a frame buffer */ +struct sst_address_info { + u32 addr; /* Address at IA */ + u32 size; /* Size of the buffer */ +}; + +struct snd_sst_alloc_params_ext { + struct sst_address_info ring_buf_info[8]; + u8 sg_count; + u8 reserved; + u16 reserved2; + u32 frag_size; /*Number of samples after which period elapsed + message is sent valid only if path = 0*/ +} __packed; + +struct snd_sst_stream_params { + union snd_sst_codec_params uc; +} __packed; + +struct snd_sst_params { + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct snd_sst_stream_params sparams; + struct snd_sst_alloc_params_ext aparams; +}; + +#endif /* __SST_MFLD_DSP_H__ */ diff --git a/sound/soc/intel/sst-mfld-platform.c b/sound/soc/intel/sst-mfld-platform.c new file mode 100644 index 0000000..840306c --- /dev/null +++ b/sound/soc/intel/sst-mfld-platform.c @@ -0,0 +1,725 @@ +/* + * sst_mfld_platform.c - Intel MID Platform driver + * + * Copyright (C) 2010-2013 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include +#include +#include "sst-mfld-platform.h" + +static struct sst_device *sst; +static DEFINE_MUTEX(sst_lock); + +int sst_register_dsp(struct sst_device *dev) +{ + if (WARN_ON(!dev)) + return -EINVAL; + if (!try_module_get(dev->dev->driver->owner)) + return -ENODEV; + mutex_lock(&sst_lock); + if (sst) { + pr_err("we already have a device %s\n", sst->name); + module_put(dev->dev->driver->owner); + mutex_unlock(&sst_lock); + return -EEXIST; + } + pr_debug("registering device %s\n", dev->name); + sst = dev; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_register_dsp); + +int sst_unregister_dsp(struct sst_device *dev) +{ + if (WARN_ON(!dev)) + return -EINVAL; + if (dev != sst) + return -EINVAL; + + mutex_lock(&sst_lock); + + if (!sst) { + mutex_unlock(&sst_lock); + return -EIO; + } + + module_put(sst->dev->driver->owner); + pr_debug("unreg %s\n", sst->name); + sst = NULL; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_unregister_dsp); + +static struct snd_pcm_hardware sst_platform_pcm_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .buffer_bytes_max = SST_MAX_BUFFER, + .period_bytes_min = SST_MIN_PERIOD_BYTES, + .period_bytes_max = SST_MAX_PERIOD_BYTES, + .periods_min = SST_MIN_PERIODS, + .periods_max = SST_MAX_PERIODS, + .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 = "Speaker-cpu-dai", + .id = 1, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra1-cpu-dai", + .id = 2, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_MONO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra2-cpu-dai", + .id = 3, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_STEREO, + .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, + }, +}, +}; + +static const struct snd_soc_component_driver sst_component = { + .name = "sst", +}; + +/* helper functions */ +static inline void sst_set_stream_status(struct sst_runtime_stream *stream, + int state) +{ + unsigned long flags; + spin_lock_irqsave(&stream->status_lock, flags); + stream->stream_status = state; + spin_unlock_irqrestore(&stream->status_lock, flags); +} + +static inline int sst_get_stream_status(struct sst_runtime_stream *stream) +{ + int state; + unsigned long flags; + + spin_lock_irqsave(&stream->status_lock, flags); + state = stream->stream_status; + spin_unlock_irqrestore(&stream->status_lock, flags); + return state; +} + +static void sst_fill_pcm_params(struct snd_pcm_substream *substream, + struct sst_pcm_params *param) +{ + + param->codec = SST_CODEC_TYPE_PCM; + 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); +} + +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +{ + 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; + + /* set codec params and inform SST driver the same */ + sst_fill_pcm_params(substream, ¶m); + substream->runtime->dma_area = substream->dma_buffer.area; + str_params.sparams = param; + str_params.codec = param.codec; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.device_type = substream->pcm->device + 1; + pr_debug("Playbck stream,Device %d\n", + substream->pcm->device); + } else { + str_params.ops = STREAM_OPS_CAPTURE; + str_params.device_type = SND_SST_DEVICE_CAPTURE; + pr_debug("Capture stream,Device %d\n", + substream->pcm->device); + } + ret_val = stream->ops->open(&str_params); + pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + if (ret_val < 0) + return ret_val; + + stream->stream_info.str_id = ret_val; + pr_debug("str id : %d\n", stream->stream_info.str_id); + return ret_val; +} + +static void sst_period_elapsed(void *mad_substream) +{ + struct snd_pcm_substream *substream = mad_substream; + struct sst_runtime_stream *stream; + int status; + + if (!substream || !substream->runtime) + return; + stream = substream->runtime->private_data; + if (!stream) + return; + status = sst_get_stream_status(stream); + if (status != SST_PLATFORM_RUNNING) + return; + snd_pcm_period_elapsed(substream); +} + +static int sst_platform_init_stream(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream = + substream->runtime->private_data; + int ret_val; + + pr_debug("setting buffer ptr param\n"); + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.period_elapsed = sst_period_elapsed; + stream->stream_info.mad_substream = substream; + stream->stream_info.buffer_ptr = 0; + stream->stream_info.sfreq = substream->runtime->rate; + ret_val = stream->ops->device_control( + SST_SND_STREAM_INIT, &stream->stream_info); + if (ret_val) + pr_err("control_set ret error %d\n", ret_val); + return ret_val; + +} +/* end -- helper functions */ + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sst_runtime_stream *stream; + int ret_val; + + pr_debug("sst_platform_open called\n"); + + snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); + ret_val = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret_val < 0) + return ret_val; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + spin_lock_init(&stream->status_lock); + + /* get the sst ops */ + mutex_lock(&sst_lock); + if (!sst) { + pr_err("no device available to run\n"); + mutex_unlock(&sst_lock); + kfree(stream); + return -ENODEV; + } + if (!try_module_get(sst->dev->driver->owner)) { + mutex_unlock(&sst_lock); + kfree(stream); + return -ENODEV; + } + stream->ops = sst->ops; + mutex_unlock(&sst_lock); + + stream->stream_info.str_id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; + /* allocate memory for SST API set */ + runtime->private_data = stream; + + return 0; +} + +static int sst_platform_close(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("sst_platform_close called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + if (str_id) + ret_val = stream->ops->close(str_id); + module_put(sst->dev->driver->owner); + kfree(stream); + return ret_val; +} + +static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("sst_platform_pcm_prepare called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + if (stream->stream_info.str_id) { + ret_val = stream->ops->device_control( + SST_SND_DROP, &str_id); + return ret_val; + } + + ret_val = sst_platform_alloc_stream(substream); + if (ret_val < 0) + return ret_val; + snprintf(substream->pcm->id, sizeof(substream->pcm->id), + "%d", stream->stream_info.str_id); + + ret_val = sst_platform_init_stream(substream); + if (ret_val) + return ret_val; + substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER; + return ret_val; +} + +static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int ret_val = 0, str_id; + struct sst_runtime_stream *stream; + int str_cmd, status; + + pr_debug("sst_platform_pcm_trigger called\n"); + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pr_debug("sst: Trigger Start\n"); + str_cmd = SST_SND_START; + status = SST_PLATFORM_RUNNING; + stream->stream_info.mad_substream = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("sst: in stop\n"); + str_cmd = SST_SND_DROP; + status = SST_PLATFORM_DROPPED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + pr_debug("sst: in pause\n"); + str_cmd = SST_SND_PAUSE; + status = SST_PLATFORM_PAUSED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + pr_debug("sst: in pause release\n"); + str_cmd = SST_SND_RESUME; + status = SST_PLATFORM_RUNNING; + break; + default: + return -EINVAL; + } + ret_val = stream->ops->device_control(str_cmd, &str_id); + if (!ret_val) + sst_set_stream_status(stream, status); + + return ret_val; +} + + +static snd_pcm_uframes_t sst_platform_pcm_pointer + (struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val, status; + struct pcm_stream_info *str_info; + + stream = substream->runtime->private_data; + status = sst_get_stream_status(stream); + if (status == SST_PLATFORM_INIT) + return 0; + str_info = &stream->stream_info; + ret_val = stream->ops->device_control( + SST_SND_BUFFER_POINTER, str_info); + if (ret_val) { + 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); +} + +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) +{ + pr_debug("sst_pcm_free called\n"); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + int retval = 0; + + pr_debug("sst_pcm_new called\n"); + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + SST_MIN_BUFFER, SST_MAX_BUFFER); + if (retval) { + pr_err("dma buffer allocationf fail\n"); + return retval; + } + } + return retval; +} + +/* compress stream operations */ +static void sst_compr_fragment_elapsed(void *arg) +{ + struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg; + + pr_debug("fragment elapsed by driver\n"); + if (cstream) + snd_compr_fragment_elapsed(cstream); +} + +static int sst_platform_compr_open(struct snd_compr_stream *cstream) +{ + + int ret_val = 0; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sst_runtime_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + spin_lock_init(&stream->status_lock); + + /* get the sst ops */ + if (!sst || !try_module_get(sst->dev->driver->owner)) { + pr_err("no device available to run\n"); + ret_val = -ENODEV; + goto out_ops; + } + stream->compr_ops = sst->compr_ops; + + stream->id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + runtime->private_data = stream; + return 0; +out_ops: + kfree(stream); + return ret_val; +} + +static int sst_platform_compr_free(struct snd_compr_stream *cstream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + stream = cstream->runtime->private_data; + /*need to check*/ + str_id = stream->id; + if (str_id) + ret_val = stream->compr_ops->close(str_id); + module_put(sst->dev->driver->owner); + kfree(stream); + pr_debug("%s: %d\n", __func__, ret_val); + return 0; +} + +static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct sst_runtime_stream *stream; + int retval; + struct snd_sst_params str_params; + struct sst_compress_cb cb; + + stream = cstream->runtime->private_data; + /* construct fw structure for this*/ + memset(&str_params, 0, sizeof(str_params)); + + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.stream_type = SST_STREAM_TYPE_MUSIC; + str_params.device_type = SND_SST_DEVICE_COMPRESS; + + switch (params->codec.id) { + case SND_AUDIOCODEC_MP3: { + str_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.mp3_params.pcm_wd_sz = 16; + break; + } + + case SND_AUDIOCODEC_AAC: { + str_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.aac_params.pcm_wd_sz = 16; + if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_ADTS; + else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_RAW; + else { + pr_err("Undefined format%d\n", params->codec.format); + return -EINVAL; + } + str_params.sparams.uc.aac_params.externalsr = + params->codec.sample_rate; + break; + } + + default: + pr_err("codec not supported, id =%d\n", params->codec.id); + return -EINVAL; + } + + str_params.aparams.ring_buf_info[0].addr = + virt_to_phys(cstream->runtime->buffer); + str_params.aparams.ring_buf_info[0].size = + cstream->runtime->buffer_size; + str_params.aparams.sg_count = 1; + str_params.aparams.frag_size = cstream->runtime->fragment_size; + + cb.param = cstream; + cb.compr_cb = sst_compr_fragment_elapsed; + + retval = stream->compr_ops->open(&str_params, &cb); + if (retval < 0) { + pr_err("stream allocation failed %d\n", retval); + return retval; + } + + stream->id = retval; + return 0; +} + +static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->control(cmd, stream->id); +} + +static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->tstamp(stream->id, tstamp); + tstamp->byte_offset = tstamp->copied_total % + (u32)cstream->runtime->buffer_size; + pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); + return 0; +} + +static int sst_platform_compr_ack(struct snd_compr_stream *cstream, + size_t bytes) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->bytes_written += bytes; + + return 0; +} + +static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_caps(caps); +} + +static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_codec_caps(codec); +} + +static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->set_metadata(stream->id, metadata); +} + +static struct snd_compr_ops sst_platform_compr_ops = { + + .open = sst_platform_compr_open, + .free = sst_platform_compr_free, + .set_params = sst_platform_compr_set_params, + .set_metadata = sst_platform_compr_set_metadata, + .trigger = sst_platform_compr_trigger, + .pointer = sst_platform_compr_pointer, + .ack = sst_platform_compr_ack, + .get_caps = sst_platform_compr_get_caps, + .get_codec_caps = sst_platform_compr_get_codec_caps, +}; + +static struct snd_soc_platform_driver sst_soc_platform_drv = { + .ops = &sst_platform_ops, + .compr_ops = &sst_platform_compr_ops, + .pcm_new = sst_pcm_new, + .pcm_free = sst_pcm_free, +}; + +static int sst_platform_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("sst_platform_probe called\n"); + sst = NULL; + ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); + if (ret) { + pr_err("registering soc platform failed\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &sst_component, + sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); + if (ret) { + pr_err("registering cpu dais failed\n"); + snd_soc_unregister_platform(&pdev->dev); + } + return ret; +} + +static int sst_platform_remove(struct platform_device *pdev) +{ + + snd_soc_unregister_component(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); + pr_debug("sst_platform_remove success\n"); + return 0; +} + +static struct platform_driver sst_platform_driver = { + .driver = { + .name = "sst-mfld-platform", + .owner = THIS_MODULE, + }, + .probe = sst_platform_probe, + .remove = sst_platform_remove, +}; + +module_platform_driver(sst_platform_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sst-mfld-platform"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h new file mode 100644 index 0000000..0c4e2dd --- /dev/null +++ b/sound/soc/intel/sst-mfld-platform.h @@ -0,0 +1,153 @@ +/* + * sst_mfld_platform.h - Intel MID Platform driver header file + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ + +#ifndef __SST_PLATFORMDRV_H__ +#define __SST_PLATFORMDRV_H__ + +#include "sst-mfld-dsp.h" + +#define SST_MONO 1 +#define SST_STEREO 2 +#define SST_MAX_CAP 5 + +#define SST_MAX_BUFFER (800*1024) +#define SST_MIN_BUFFER (800*1024) +#define SST_MIN_PERIOD_BYTES 32 +#define SST_MAX_PERIOD_BYTES SST_MAX_BUFFER +#define SST_MIN_PERIODS 2 +#define SST_MAX_PERIODS (1024*2) +#define SST_FIFO_SIZE 0 + +struct pcm_stream_info { + int str_id; + void *mad_substream; + void (*period_elapsed) (void *mad_substream); + unsigned long long buffer_ptr; + int sfreq; +}; + +enum sst_drv_status { + SST_PLATFORM_INIT = 1, + SST_PLATFORM_STARTED, + SST_PLATFORM_RUNNING, + SST_PLATFORM_PAUSED, + SST_PLATFORM_DROPPED, +}; + +enum sst_controls { + SST_SND_ALLOC = 0x00, + SST_SND_PAUSE = 0x01, + SST_SND_RESUME = 0x02, + SST_SND_DROP = 0x03, + SST_SND_FREE = 0x04, + SST_SND_BUFFER_POINTER = 0x05, + SST_SND_STREAM_INIT = 0x06, + SST_SND_START = 0x07, + SST_MAX_CONTROLS = 0x07, +}; + +enum sst_stream_ops { + STREAM_OPS_PLAYBACK = 0, + STREAM_OPS_CAPTURE, +}; + +enum sst_audio_device_type { + SND_SST_DEVICE_HEADSET = 1, + SND_SST_DEVICE_IHF, + SND_SST_DEVICE_VIBRA, + SND_SST_DEVICE_HAPTIC, + SND_SST_DEVICE_CAPTURE, + SND_SST_DEVICE_COMPRESS, +}; + +/* PCM Parameters */ +struct sst_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u32 ring_buffer_size; + u32 period_count; /* period elapsed in samples*/ + u32 ring_buffer_addr; +}; + +struct sst_stream_params { + u32 result; + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct sst_pcm_params sparams; +}; + +struct sst_compress_cb { + void *param; + void (*compr_cb)(void *param); +}; + +struct compress_sst_ops { + const char *name; + int (*open) (struct snd_sst_params *str_params, + struct sst_compress_cb *cb); + int (*control) (unsigned int cmd, unsigned int str_id); + int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); + int (*ack) (unsigned int str_id, unsigned long bytes); + int (*close) (unsigned int str_id); + int (*get_caps) (struct snd_compr_caps *caps); + int (*get_codec_caps) (struct snd_compr_codec_caps *codec); + int (*set_metadata) (unsigned int str_id, + struct snd_compr_metadata *mdata); + +}; + +struct sst_ops { + int (*open) (struct sst_stream_params *str_param); + int (*device_control) (int cmd, void *arg); + int (*close) (unsigned int str_id); +}; + +struct sst_runtime_stream { + int stream_status; + unsigned int id; + size_t bytes_written; + struct pcm_stream_info stream_info; + struct sst_ops *ops; + struct compress_sst_ops *compr_ops; + spinlock_t status_lock; +}; + +struct sst_device { + char *name; + struct device *dev; + struct sst_ops *ops; + struct compress_sst_ops *compr_ops; +}; + +int sst_register_dsp(struct sst_device *sst); +int sst_unregister_dsp(struct sst_device *sst); +#endif diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst_dsp.h deleted file mode 100644 index 0fce1de..0000000 --- a/sound/soc/intel/sst_dsp.h +++ /dev/null @@ -1,134 +0,0 @@ -#ifndef __SST_DSP_H__ -#define __SST_DSP_H__ -/* - * sst_dsp.h - Intel SST Driver for audio engine - * - * Copyright (C) 2008-12 Intel Corporation - * Authors: Vinod Koul - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * 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. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -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, -}; - -enum stream_type { - SST_STREAM_TYPE_NONE = 0, - SST_STREAM_TYPE_MUSIC = 1, -}; - -struct snd_pcm_params { - u16 codec; /* codec type */ - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u8 use_offload_path; - u8 reserved2; - u16 reserved3; - u8 channel_map[8]; -} __packed; - -/* MP3 Music Parameters Message */ -struct snd_mp3_params { - u16 codec; - 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*/ - u16 reserved2; /* Unused */ -} __packed; - -#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 { - u16 codec; - 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 */ - u16 reser2; - u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ - u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ - u8 reser1; - u16 reser3; -} __packed; - -/* WMA Music Parameters Message */ -struct snd_wma_params { - u16 codec; - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - 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 */ -} __packed; - -/* 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; -} __packed; - -/* Address and size info of a frame buffer */ -struct sst_address_info { - u32 addr; /* Address at IA */ - u32 size; /* Size of the buffer */ -}; - -struct snd_sst_alloc_params_ext { - struct sst_address_info ring_buf_info[8]; - u8 sg_count; - u8 reserved; - u16 reserved2; - u32 frag_size; /*Number of samples after which period elapsed - message is sent valid only if path = 0*/ -} __packed; - -struct snd_sst_stream_params { - union snd_sst_codec_params uc; -} __packed; - -struct snd_sst_params { - u32 stream_id; - u8 codec; - u8 ops; - u8 stream_type; - u8 device_type; - struct snd_sst_stream_params sparams; - struct snd_sst_alloc_params_ext aparams; -}; - -#endif /* __SST_DSP_H__ */ diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst_platform.c deleted file mode 100644 index f465a81..0000000 --- a/sound/soc/intel/sst_platform.c +++ /dev/null @@ -1,725 +0,0 @@ -/* - * sst_platform.c - Intel MID Platform driver - * - * Copyright (C) 2010-2013 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * 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. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * - */ -#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt - -#include -#include -#include -#include -#include -#include -#include -#include -#include "sst_platform.h" - -static struct sst_device *sst; -static DEFINE_MUTEX(sst_lock); - -int sst_register_dsp(struct sst_device *dev) -{ - if (WARN_ON(!dev)) - return -EINVAL; - if (!try_module_get(dev->dev->driver->owner)) - return -ENODEV; - mutex_lock(&sst_lock); - if (sst) { - pr_err("we already have a device %s\n", sst->name); - module_put(dev->dev->driver->owner); - mutex_unlock(&sst_lock); - return -EEXIST; - } - pr_debug("registering device %s\n", dev->name); - sst = dev; - mutex_unlock(&sst_lock); - return 0; -} -EXPORT_SYMBOL_GPL(sst_register_dsp); - -int sst_unregister_dsp(struct sst_device *dev) -{ - if (WARN_ON(!dev)) - return -EINVAL; - if (dev != sst) - return -EINVAL; - - mutex_lock(&sst_lock); - - if (!sst) { - mutex_unlock(&sst_lock); - return -EIO; - } - - module_put(sst->dev->driver->owner); - pr_debug("unreg %s\n", sst->name); - sst = NULL; - mutex_unlock(&sst_lock); - return 0; -} -EXPORT_SYMBOL_GPL(sst_unregister_dsp); - -static struct snd_pcm_hardware sst_platform_pcm_hw = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_DOUBLE | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_MMAP| - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_SYNC_START), - .buffer_bytes_max = SST_MAX_BUFFER, - .period_bytes_min = SST_MIN_PERIOD_BYTES, - .period_bytes_max = SST_MAX_PERIOD_BYTES, - .periods_min = SST_MIN_PERIODS, - .periods_max = SST_MAX_PERIODS, - .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 = "Speaker-cpu-dai", - .id = 1, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Vibra1-cpu-dai", - .id = 2, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_MONO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Vibra2-cpu-dai", - .id = 3, - .playback = { - .channels_min = SST_MONO, - .channels_max = SST_STEREO, - .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, - }, -}, -}; - -static const struct snd_soc_component_driver sst_component = { - .name = "sst", -}; - -/* helper functions */ -static inline void sst_set_stream_status(struct sst_runtime_stream *stream, - int state) -{ - unsigned long flags; - spin_lock_irqsave(&stream->status_lock, flags); - stream->stream_status = state; - spin_unlock_irqrestore(&stream->status_lock, flags); -} - -static inline int sst_get_stream_status(struct sst_runtime_stream *stream) -{ - int state; - unsigned long flags; - - spin_lock_irqsave(&stream->status_lock, flags); - state = stream->stream_status; - spin_unlock_irqrestore(&stream->status_lock, flags); - return state; -} - -static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) -{ - - param->codec = SST_CODEC_TYPE_PCM; - 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); -} - -static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) -{ - 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; - - /* set codec params and inform SST driver the same */ - sst_fill_pcm_params(substream, ¶m); - substream->runtime->dma_area = substream->dma_buffer.area; - str_params.sparams = param; - str_params.codec = param.codec; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.device_type = substream->pcm->device + 1; - pr_debug("Playbck stream,Device %d\n", - substream->pcm->device); - } else { - str_params.ops = STREAM_OPS_CAPTURE; - str_params.device_type = SND_SST_DEVICE_CAPTURE; - pr_debug("Capture stream,Device %d\n", - substream->pcm->device); - } - ret_val = stream->ops->open(&str_params); - pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); - if (ret_val < 0) - return ret_val; - - stream->stream_info.str_id = ret_val; - pr_debug("str id : %d\n", stream->stream_info.str_id); - return ret_val; -} - -static void sst_period_elapsed(void *mad_substream) -{ - struct snd_pcm_substream *substream = mad_substream; - struct sst_runtime_stream *stream; - int status; - - if (!substream || !substream->runtime) - return; - stream = substream->runtime->private_data; - if (!stream) - return; - status = sst_get_stream_status(stream); - if (status != SST_PLATFORM_RUNNING) - return; - snd_pcm_period_elapsed(substream); -} - -static int sst_platform_init_stream(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream = - substream->runtime->private_data; - int ret_val; - - pr_debug("setting buffer ptr param\n"); - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.period_elapsed = sst_period_elapsed; - stream->stream_info.mad_substream = substream; - stream->stream_info.buffer_ptr = 0; - stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->device_control( - SST_SND_STREAM_INIT, &stream->stream_info); - if (ret_val) - pr_err("control_set ret error %d\n", ret_val); - return ret_val; - -} -/* end -- helper functions */ - -static int sst_platform_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sst_runtime_stream *stream; - int ret_val; - - pr_debug("sst_platform_open called\n"); - - snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); - ret_val = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret_val < 0) - return ret_val; - - stream = kzalloc(sizeof(*stream), GFP_KERNEL); - if (!stream) - return -ENOMEM; - spin_lock_init(&stream->status_lock); - - /* get the sst ops */ - mutex_lock(&sst_lock); - if (!sst) { - pr_err("no device available to run\n"); - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - if (!try_module_get(sst->dev->driver->owner)) { - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - stream->ops = sst->ops; - mutex_unlock(&sst_lock); - - stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.mad_substream = substream; - /* allocate memory for SST API set */ - runtime->private_data = stream; - - return 0; -} - -static int sst_platform_close(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - pr_debug("sst_platform_close called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - if (str_id) - ret_val = stream->ops->close(str_id); - module_put(sst->dev->driver->owner); - kfree(stream); - return ret_val; -} - -static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - pr_debug("sst_platform_pcm_prepare called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - if (stream->stream_info.str_id) { - ret_val = stream->ops->device_control( - SST_SND_DROP, &str_id); - return ret_val; - } - - ret_val = sst_platform_alloc_stream(substream); - if (ret_val < 0) - return ret_val; - snprintf(substream->pcm->id, sizeof(substream->pcm->id), - "%d", stream->stream_info.str_id); - - ret_val = sst_platform_init_stream(substream); - if (ret_val) - return ret_val; - substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER; - return ret_val; -} - -static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - int ret_val = 0, str_id; - struct sst_runtime_stream *stream; - int str_cmd, status; - - pr_debug("sst_platform_pcm_trigger called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - pr_debug("sst: Trigger Start\n"); - str_cmd = SST_SND_START; - status = SST_PLATFORM_RUNNING; - stream->stream_info.mad_substream = substream; - break; - case SNDRV_PCM_TRIGGER_STOP: - pr_debug("sst: in stop\n"); - str_cmd = SST_SND_DROP; - status = SST_PLATFORM_DROPPED; - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - pr_debug("sst: in pause\n"); - str_cmd = SST_SND_PAUSE; - status = SST_PLATFORM_PAUSED; - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - pr_debug("sst: in pause release\n"); - str_cmd = SST_SND_RESUME; - status = SST_PLATFORM_RUNNING; - break; - default: - return -EINVAL; - } - ret_val = stream->ops->device_control(str_cmd, &str_id); - if (!ret_val) - sst_set_stream_status(stream, status); - - return ret_val; -} - - -static snd_pcm_uframes_t sst_platform_pcm_pointer - (struct snd_pcm_substream *substream) -{ - struct sst_runtime_stream *stream; - int ret_val, status; - struct pcm_stream_info *str_info; - - stream = substream->runtime->private_data; - status = sst_get_stream_status(stream); - if (status == SST_PLATFORM_INIT) - return 0; - str_info = &stream->stream_info; - ret_val = stream->ops->device_control( - SST_SND_BUFFER_POINTER, str_info); - if (ret_val) { - 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); -} - -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) -{ - pr_debug("sst_pcm_free called\n"); - snd_pcm_lib_preallocate_free_for_all(pcm); -} - -static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_pcm *pcm = rtd->pcm; - int retval = 0; - - pr_debug("sst_pcm_new called\n"); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - retval = snd_pcm_lib_preallocate_pages_for_all(pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - SST_MIN_BUFFER, SST_MAX_BUFFER); - if (retval) { - pr_err("dma buffer allocationf fail\n"); - return retval; - } - } - return retval; -} - -/* compress stream operations */ -static void sst_compr_fragment_elapsed(void *arg) -{ - struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg; - - pr_debug("fragment elapsed by driver\n"); - if (cstream) - snd_compr_fragment_elapsed(cstream); -} - -static int sst_platform_compr_open(struct snd_compr_stream *cstream) -{ - - int ret_val = 0; - struct snd_compr_runtime *runtime = cstream->runtime; - struct sst_runtime_stream *stream; - - stream = kzalloc(sizeof(*stream), GFP_KERNEL); - if (!stream) - return -ENOMEM; - - spin_lock_init(&stream->status_lock); - - /* get the sst ops */ - if (!sst || !try_module_get(sst->dev->driver->owner)) { - pr_err("no device available to run\n"); - ret_val = -ENODEV; - goto out_ops; - } - stream->compr_ops = sst->compr_ops; - - stream->id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - runtime->private_data = stream; - return 0; -out_ops: - kfree(stream); - return ret_val; -} - -static int sst_platform_compr_free(struct snd_compr_stream *cstream) -{ - struct sst_runtime_stream *stream; - int ret_val = 0, str_id; - - stream = cstream->runtime->private_data; - /*need to check*/ - str_id = stream->id; - if (str_id) - ret_val = stream->compr_ops->close(str_id); - module_put(sst->dev->driver->owner); - kfree(stream); - pr_debug("%s: %d\n", __func__, ret_val); - return 0; -} - -static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, - struct snd_compr_params *params) -{ - struct sst_runtime_stream *stream; - int retval; - struct snd_sst_params str_params; - struct sst_compress_cb cb; - - stream = cstream->runtime->private_data; - /* construct fw structure for this*/ - memset(&str_params, 0, sizeof(str_params)); - - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.stream_type = SST_STREAM_TYPE_MUSIC; - str_params.device_type = SND_SST_DEVICE_COMPRESS; - - switch (params->codec.id) { - case SND_AUDIOCODEC_MP3: { - str_params.codec = SST_CODEC_TYPE_MP3; - str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3; - str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in; - str_params.sparams.uc.mp3_params.pcm_wd_sz = 16; - break; - } - - case SND_AUDIOCODEC_AAC: { - str_params.codec = SST_CODEC_TYPE_AAC; - str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC; - str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in; - str_params.sparams.uc.aac_params.pcm_wd_sz = 16; - if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS) - str_params.sparams.uc.aac_params.bs_format = - AAC_BIT_STREAM_ADTS; - else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW) - str_params.sparams.uc.aac_params.bs_format = - AAC_BIT_STREAM_RAW; - else { - pr_err("Undefined format%d\n", params->codec.format); - return -EINVAL; - } - str_params.sparams.uc.aac_params.externalsr = - params->codec.sample_rate; - break; - } - - default: - pr_err("codec not supported, id =%d\n", params->codec.id); - return -EINVAL; - } - - str_params.aparams.ring_buf_info[0].addr = - virt_to_phys(cstream->runtime->buffer); - str_params.aparams.ring_buf_info[0].size = - cstream->runtime->buffer_size; - str_params.aparams.sg_count = 1; - str_params.aparams.frag_size = cstream->runtime->fragment_size; - - cb.param = cstream; - cb.compr_cb = sst_compr_fragment_elapsed; - - retval = stream->compr_ops->open(&str_params, &cb); - if (retval < 0) { - pr_err("stream allocation failed %d\n", retval); - return retval; - } - - stream->id = retval; - return 0; -} - -static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->control(cmd, stream->id); -} - -static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, - struct snd_compr_tstamp *tstamp) -{ - struct sst_runtime_stream *stream; - - stream = cstream->runtime->private_data; - stream->compr_ops->tstamp(stream->id, tstamp); - tstamp->byte_offset = tstamp->copied_total % - (u32)cstream->runtime->buffer_size; - pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); - return 0; -} - -static int sst_platform_compr_ack(struct snd_compr_stream *cstream, - size_t bytes) -{ - struct sst_runtime_stream *stream; - - stream = cstream->runtime->private_data; - stream->compr_ops->ack(stream->id, (unsigned long)bytes); - stream->bytes_written += bytes; - - return 0; -} - -static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream, - struct snd_compr_caps *caps) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->get_caps(caps); -} - -static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, - struct snd_compr_codec_caps *codec) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->get_codec_caps(codec); -} - -static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, - struct snd_compr_metadata *metadata) -{ - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->set_metadata(stream->id, metadata); -} - -static struct snd_compr_ops sst_platform_compr_ops = { - - .open = sst_platform_compr_open, - .free = sst_platform_compr_free, - .set_params = sst_platform_compr_set_params, - .set_metadata = sst_platform_compr_set_metadata, - .trigger = sst_platform_compr_trigger, - .pointer = sst_platform_compr_pointer, - .ack = sst_platform_compr_ack, - .get_caps = sst_platform_compr_get_caps, - .get_codec_caps = sst_platform_compr_get_codec_caps, -}; - -static struct snd_soc_platform_driver sst_soc_platform_drv = { - .ops = &sst_platform_ops, - .compr_ops = &sst_platform_compr_ops, - .pcm_new = sst_pcm_new, - .pcm_free = sst_pcm_free, -}; - -static int sst_platform_probe(struct platform_device *pdev) -{ - int ret; - - pr_debug("sst_platform_probe called\n"); - sst = NULL; - ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); - if (ret) { - pr_err("registering soc platform failed\n"); - return ret; - } - - ret = snd_soc_register_component(&pdev->dev, &sst_component, - sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); - if (ret) { - pr_err("registering cpu dais failed\n"); - snd_soc_unregister_platform(&pdev->dev); - } - return ret; -} - -static int sst_platform_remove(struct platform_device *pdev) -{ - - snd_soc_unregister_component(&pdev->dev); - snd_soc_unregister_platform(&pdev->dev); - pr_debug("sst_platform_remove success\n"); - return 0; -} - -static struct platform_driver sst_platform_driver = { - .driver = { - .name = "sst-platform", - .owner = THIS_MODULE, - }, - .probe = sst_platform_probe, - .remove = sst_platform_remove, -}; - -module_platform_driver(sst_platform_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); -MODULE_AUTHOR("Vinod Koul "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sst-platform"); diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst_platform.h deleted file mode 100644 index bee64fb..0000000 --- a/sound/soc/intel/sst_platform.h +++ /dev/null @@ -1,153 +0,0 @@ -/* - * sst_platform.h - Intel MID Platform driver header file - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul - * Author: Harsha Priya - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * 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. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * - */ - -#ifndef __SST_PLATFORMDRV_H__ -#define __SST_PLATFORMDRV_H__ - -#include "sst_dsp.h" - -#define SST_MONO 1 -#define SST_STEREO 2 -#define SST_MAX_CAP 5 - -#define SST_MAX_BUFFER (800*1024) -#define SST_MIN_BUFFER (800*1024) -#define SST_MIN_PERIOD_BYTES 32 -#define SST_MAX_PERIOD_BYTES SST_MAX_BUFFER -#define SST_MIN_PERIODS 2 -#define SST_MAX_PERIODS (1024*2) -#define SST_FIFO_SIZE 0 - -struct pcm_stream_info { - int str_id; - void *mad_substream; - void (*period_elapsed) (void *mad_substream); - unsigned long long buffer_ptr; - int sfreq; -}; - -enum sst_drv_status { - SST_PLATFORM_INIT = 1, - SST_PLATFORM_STARTED, - SST_PLATFORM_RUNNING, - SST_PLATFORM_PAUSED, - SST_PLATFORM_DROPPED, -}; - -enum sst_controls { - SST_SND_ALLOC = 0x00, - SST_SND_PAUSE = 0x01, - SST_SND_RESUME = 0x02, - SST_SND_DROP = 0x03, - SST_SND_FREE = 0x04, - SST_SND_BUFFER_POINTER = 0x05, - SST_SND_STREAM_INIT = 0x06, - SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, -}; - -enum sst_stream_ops { - STREAM_OPS_PLAYBACK = 0, - STREAM_OPS_CAPTURE, -}; - -enum sst_audio_device_type { - SND_SST_DEVICE_HEADSET = 1, - SND_SST_DEVICE_IHF, - SND_SST_DEVICE_VIBRA, - SND_SST_DEVICE_HAPTIC, - SND_SST_DEVICE_CAPTURE, - SND_SST_DEVICE_COMPRESS, -}; - -/* PCM Parameters */ -struct sst_pcm_params { - u16 codec; /* codec type */ - u8 num_chan; /* 1=Mono, 2=Stereo */ - u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u32 ring_buffer_size; - u32 period_count; /* period elapsed in samples*/ - u32 ring_buffer_addr; -}; - -struct sst_stream_params { - u32 result; - u32 stream_id; - u8 codec; - u8 ops; - u8 stream_type; - u8 device_type; - struct sst_pcm_params sparams; -}; - -struct sst_compress_cb { - void *param; - void (*compr_cb)(void *param); -}; - -struct compress_sst_ops { - const char *name; - int (*open) (struct snd_sst_params *str_params, - struct sst_compress_cb *cb); - int (*control) (unsigned int cmd, unsigned int str_id); - int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); - int (*ack) (unsigned int str_id, unsigned long bytes); - int (*close) (unsigned int str_id); - int (*get_caps) (struct snd_compr_caps *caps); - int (*get_codec_caps) (struct snd_compr_codec_caps *codec); - int (*set_metadata) (unsigned int str_id, - struct snd_compr_metadata *mdata); - -}; - -struct sst_ops { - int (*open) (struct sst_stream_params *str_param); - int (*device_control) (int cmd, void *arg); - int (*close) (unsigned int str_id); -}; - -struct sst_runtime_stream { - int stream_status; - unsigned int id; - size_t bytes_written; - struct pcm_stream_info stream_info; - struct sst_ops *ops; - struct compress_sst_ops *compr_ops; - spinlock_t status_lock; -}; - -struct sst_device { - char *name; - struct device *dev; - struct sst_ops *ops; - struct compress_sst_ops *compr_ops; -}; - -int sst_register_dsp(struct sst_device *sst); -int sst_unregister_dsp(struct sst_device *sst); -#endif -- cgit v1.1 From 74d04c3efbc4f10990e5c4218ad3f65bfdcf3c75 Mon Sep 17 00:00:00 2001 From: Thomas Petazzoni Date: Wed, 12 Feb 2014 18:20:56 +0100 Subject: sound: ASoC: add ASoC board driver for Armada 370 DB This commit adds a simple ASoC board driver fo the Armada 370 Development Board, which connects the audio unit of the Armada 370 SoC to the I2C-based CS42L51. For now, only the analog audio input and output through the CS42L51 are supported, but a followup patch adds S/PDIF support to this driver. Signed-off-by: Thomas Petazzoni Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 8 +++ sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 120 +++++++++++++++++++++++++++++++++++++ 3 files changed, 130 insertions(+) create mode 100644 sound/soc/kirkwood/armada-370-db.c (limited to 'sound') diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 764a0ef..2dc3ecf 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -6,6 +6,14 @@ config SND_KIRKWOOD_SOC the Kirkwood I2S interface. You will also need to select the audio interfaces to support below. +config SND_KIRKWOOD_SOC_ARMADA370_DB + tristate "SoC Audio support for Armada 370 DB" + depends on SND_KIRKWOOD_SOC && (ARCH_MVEBU || COMPILE_TEST) && I2C + select SND_SOC_CS42L51 + help + Say Y if you want to add support for SoC audio on + the Armada 370 Development Board. + config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 9e78138..7c1d8fe 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -4,6 +4,8 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o snd-soc-openrd-objs := kirkwood-openrd.o snd-soc-t5325-objs := kirkwood-t5325.o +snd-soc-armada-370-db-objs := armada-370-db.o +obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c new file mode 100644 index 0000000..977639b --- /dev/null +++ b/sound/soc/kirkwood/armada-370-db.c @@ -0,0 +1,120 @@ +/* + * Copyright (C) 2014 Marvell + * + * Thomas Petazzoni + * + * 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; either version 2 of the + * License, or (at your option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "../codecs/cs42l51.h" + +static int a370db_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; + unsigned int freq; + + switch (params_rate(params)) { + default: + case 44100: + freq = 11289600; + break; + case 48000: + freq = 12288000; + break; + case 96000: + freq = 24576000; + break; + } + + return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); +} + +static struct snd_soc_ops a370db_ops = { + .hw_params = a370db_hw_params, +}; + +static const struct snd_soc_dapm_widget a370db_dapm_widgets[] = { + SND_SOC_DAPM_HP("Out Jack", NULL), + SND_SOC_DAPM_LINE("In Jack", NULL), +}; + +static const struct snd_soc_dapm_route a370db_route[] = { + { "Out Jack", NULL, "HPL" }, + { "Out Jack", NULL, "HPR" }, + { "AIN1L", NULL, "In Jack" }, + { "AIN1L", NULL, "In Jack" }, +}; + +static struct snd_soc_dai_link a370db_dai[] = { +{ + .name = "CS42L51", + .stream_name = "analog", + .cpu_dai_name = "i2s", + .codec_dai_name = "cs42l51-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, + .ops = &a370db_ops, +}, +}; + +static struct snd_soc_card a370db = { + .name = "a370db", + .owner = THIS_MODULE, + .dai_link = a370db_dai, + .num_links = ARRAY_SIZE(a370db_dai), + .dapm_widgets = a370db_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(a370db_dapm_widgets), + .dapm_routes = a370db_route, + .num_dapm_routes = ARRAY_SIZE(a370db_route), +}; + +static int a370db_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &a370db; + + card->dev = &pdev->dev; + + a370db_dai[0].cpu_of_node = + of_parse_phandle(pdev->dev.of_node, + "marvell,audio-controller", 0); + a370db_dai[0].platform_of_node = a370db_dai[0].cpu_of_node; + + a370db_dai[0].codec_of_node = + of_parse_phandle(pdev->dev.of_node, + "marvell,audio-codec", 0); + + return devm_snd_soc_register_card(card->dev, card); +} + +static const struct of_device_id a370db_dt_ids[] = { + { .compatible = "marvell,a370db-audio" }, + { }, +}; + +static struct platform_driver a370db_driver = { + .driver = { + .name = "a370db-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(a370db_dt_ids), + }, + .probe = a370db_probe, +}; + +module_platform_driver(a370db_driver); + +MODULE_AUTHOR("Thomas Petazzoni "); +MODULE_DESCRIPTION("ALSA SoC a370db audio client"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:a370db-audio"); -- cgit v1.1 From 7b80300e749c2865fbfc23870d3b8f3186956fc0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 16 Feb 2014 10:04:43 +0800 Subject: ASoC: io: Remove SPI support All ASoC CODEC drivers that use SPI have now been converted to use regmap so we can delete SND_SOC_SPI, preventing any new users being added. Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index aa886cc..3a0d99e 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -106,13 +106,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, break; #endif -#if IS_ENABLED(CONFIG_REGMAP_SPI) - case SND_SOC_SPI: - codec->control_data = regmap_init_spi(to_spi_device(codec->dev), - &config); - break; -#endif - case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ codec->using_regmap = true; -- cgit v1.1 From 790aff62291eade029755713fb41c1143d4ddbc7 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 17 Feb 2014 13:32:06 +0000 Subject: ASoC: Intel: Add Intel SST audio DSP low level shim driver. Add support for Intel Smart Sound Technology (SST) audio DSPs. This driver provides the low level IO, reset, boot and IRQ management for Intel audio DSPs. These files make up the low level part of the SST audio driver stack and will be used by many Intel SST cores like Haswell, Broadwell and Baytrail. SST DSPs expose a memory mapped region (shim) for config and control. The shim layout is mostly shared without much modification across cores and this driver provides a uniform API to access the shim and to enable basic shim functions. It also provides functionality to abstract some shim functions for cores with different shim features. Signed-off-by: Liam Girdwood Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp-priv.h | 305 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-dsp.c | 324 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-dsp.h | 212 +++++++++++++++++++++++++++ 3 files changed, 841 insertions(+) create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h new file mode 100644 index 0000000..35de547 --- /dev/null +++ b/sound/soc/intel/sst-dsp-priv.h @@ -0,0 +1,305 @@ +/* + * Intel Smart Sound Technology + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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 __SOUND_SOC_SST_DSP_PRIV_H +#define __SOUND_SOC_SST_DSP_PRIV_H + +#include +#include +#include +#include + +struct sst_mem_block; +struct sst_module; +struct sst_fw; + +/* + * DSP Operations exported by platform Audio DSP driver. + */ +struct sst_ops { + /* DSP core boot / reset */ + void (*boot)(struct sst_dsp *); + void (*reset)(struct sst_dsp *); + + /* Shim IO */ + void (*write)(void __iomem *addr, u32 offset, u32 value); + u32 (*read)(void __iomem *addr, u32 offset); + void (*write64)(void __iomem *addr, u32 offset, u64 value); + u64 (*read64)(void __iomem *addr, u32 offset); + + /* DSP I/DRAM IO */ + void (*ram_read)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + void (*ram_write)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + + void (*dump)(struct sst_dsp *); + + /* IRQ handlers */ + irqreturn_t (*irq_handler)(int irq, void *context); + + /* SST init and free */ + int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata); + void (*free)(struct sst_dsp *sst); + + /* FW module parser/loader */ + int (*parse_fw)(struct sst_fw *sst_fw); +}; + +/* + * Audio DSP memory offsets and addresses. + */ +struct sst_addr { + u32 lpe_base; + u32 shim_offset; + u32 iram_offset; + void __iomem *lpe; + void __iomem *shim; + void __iomem *pci_cfg; + void __iomem *fw_ext; +}; + +/* + * Audio DSP Mailbox configuration. + */ +struct sst_mailbox { + void __iomem *in_base; + void __iomem *out_base; + size_t in_size; + size_t out_size; +}; + +/* + * Audio DSP Firmware data types. + */ +enum sst_data_type { + SST_DATA_M = 0, /* module block data */ + SST_DATA_P = 1, /* peristant data (text, data) */ + SST_DATA_S = 2, /* scratch data (usually buffers) */ +}; + +/* + * Audio DSP memory block types. + */ +enum sst_mem_type { + SST_MEM_IRAM = 0, + SST_MEM_DRAM = 1, + SST_MEM_ANY = 2, + SST_MEM_CACHE= 3, +}; + +/* + * Audio DSP Generic Firmware File. + * + * SST Firmware files can consist of 1..N modules. This generic structure is + * used to manage each firmware file and it's modules regardless of SST firmware + * type. A SST driver may load multiple FW files. + */ +struct sst_fw { + struct sst_dsp *dsp; + + /* base addresses of FW file data */ + dma_addr_t dmable_fw_paddr; /* physical address of fw data */ + void *dma_buf; /* virtual address of fw data */ + u32 size; /* size of fw data */ + + /* lists */ + struct list_head list; /* DSP list of FW */ + struct list_head module_list; /* FW list of modules */ + + void *private; /* core doesn't touch this */ +}; + +/* + * Audio DSP Generic Module data. + * + * This is used to dsecribe any sections of persistent (text and data) and + * scratch (buffers) of module data in ADSP memory space. + */ +struct sst_module_data { + + enum sst_mem_type type; /* destination memory type */ + enum sst_data_type data_type; /* type of module data */ + + u32 size; /* size in bytes */ + u32 offset; /* offset in FW file */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ +}; + +/* + * Audio DSP Generic Module Template. + * + * Used to define and register a new FW module. This data is extracted from + * FW module header information. + */ +struct sst_module_template { + u32 id; + u32 entry; /* entry point */ + struct sst_module_data s; /* scratch data */ + struct sst_module_data p; /* peristant data */ +}; + +/* + * Audio DSP Generic Module. + * + * Each Firmware file can consist of 1..N modules. A module can span multiple + * ADSP memory blocks. The simplest FW will be a file with 1 module. + */ +struct sst_module { + struct sst_dsp *dsp; + struct sst_fw *sst_fw; /* parent FW we belong too */ + + /* module configuration */ + u32 id; + u32 entry; /* module entry point */ + u32 offset; /* module offset in firmware file */ + u32 size; /* module size */ + struct sst_module_data s; /* scratch data */ + struct sst_module_data p; /* peristant data */ + + /* runtime */ + u32 usage_count; /* can be unloaded if count == 0 */ + void *private; /* core doesn't touch this */ + + /* lists */ + struct list_head block_list; /* Module list of blocks in use */ + struct list_head list; /* DSP list of modules */ + struct list_head list_fw; /* FW list of modules */ +}; + +/* + * SST Memory Block operations. + */ +struct sst_block_ops { + int (*enable)(struct sst_mem_block *block); + int (*disable)(struct sst_mem_block *block); +}; + +/* + * SST Generic Memory Block. + * + * SST ADP memory has multiple IRAM and DRAM blocks. Some ADSP blocks can be + * power gated. + */ +struct sst_mem_block { + struct sst_dsp *dsp; + struct sst_module *module; /* module that uses this block */ + + /* block config */ + u32 offset; /* offset from base */ + u32 size; /* block size */ + u32 index; /* block index 0..N */ + enum sst_mem_type type; /* block memory type IRAM/DRAM */ + struct sst_block_ops *ops; /* block operations, if any */ + + /* block status */ + enum sst_data_type data_type; /* data type held in this block */ + u32 bytes_used; /* bytes in use by modules */ + void *private; /* generic core does not touch this */ + int users; /* number of modules using this block */ + + /* block lists */ + struct list_head module_list; /* Module list of blocks */ + struct list_head list; /* Map list of free/used blocks */ +}; + +/* + * Generic SST Shim Interface. + */ +struct sst_dsp { + + /* runtime */ + struct sst_dsp_device *sst_dev; + spinlock_t spinlock; /* IPC locking */ + struct mutex mutex; /* DSP FW lock */ + struct device *dev; + void *thread_context; + int irq; + u32 id; + + /* list of free and used ADSP memory blocks */ + struct list_head used_block_list; + struct list_head free_block_list; + + /* operations */ + struct sst_ops *ops; + + /* debug FS */ + struct dentry *debugfs_root; + + /* base addresses */ + struct sst_addr addr; + + /* mailbox */ + struct sst_mailbox mailbox; + + /* SST FW files loaded and their modules */ + struct list_head module_list; + struct list_head fw_list; + + /* platform data */ + struct sst_pdata *pdata; + + /* DMA FW loading */ + struct sst_dma *dma; + bool fw_use_dma; +}; + +/* Size optimised DRAM/IRAM memcpy */ +static inline void sst_dsp_write(struct sst_dsp *sst, void *src, + u32 dest_offset, size_t bytes) +{ + sst->ops->ram_write(sst, sst->addr.lpe + dest_offset, src, bytes); +} + +static inline void sst_dsp_read(struct sst_dsp *sst, void *dest, + u32 src_offset, size_t bytes) +{ + sst->ops->ram_read(sst, dest, sst->addr.lpe + src_offset, bytes); +} + +static inline void *sst_dsp_get_thread_context(struct sst_dsp *sst) +{ + return sst->thread_context; +} + +/* Create/Free FW files - can contain multiple modules */ +struct sst_fw *sst_fw_new(struct sst_dsp *dsp, + const struct firmware *fw, void *private); +void sst_fw_free(struct sst_fw *sst_fw); +void sst_fw_free_all(struct sst_dsp *dsp); + +/* Create/Free firmware modules */ +struct sst_module *sst_module_new(struct sst_fw *sst_fw, + struct sst_module_template *template, void *private); +void sst_module_free(struct sst_module *sst_module); +int sst_module_insert(struct sst_module *sst_module); +int sst_module_remove(struct sst_module *sst_module); +int sst_module_insert_fixed_block(struct sst_module *module, + struct sst_module_data *data); +struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); + +/* allocate/free pesistent/scratch memory regions managed by drv */ +struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); +void sst_mem_block_free_scratch(struct sst_dsp *dsp, + struct sst_module *scratch); + +/* Register the DSPs memory blocks - would be nice to read from ACPI */ +struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, + u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, + void *private); +void sst_mem_block_unregister_all(struct sst_dsp *dsp); + +#endif diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c new file mode 100644 index 0000000..1888de54 --- /dev/null +++ b/sound/soc/intel/sst-dsp.c @@ -0,0 +1,324 @@ +/* + * Intel Smart Sound Technology (SST) DSP Core Driver + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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. + * + */ + +#include +#include +#include +#include +#include + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +#define CREATE_TRACE_POINTS +#include + +/* Public API */ +void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst->ops->write(sst->addr.shim, offset, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write); + +u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset) +{ + unsigned long flags; + u32 val; + + spin_lock_irqsave(&sst->spinlock, flags); + val = sst->ops->read(sst->addr.shim, offset); + spin_unlock_irqrestore(&sst->spinlock, flags); + + return val; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read); + +void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst->ops->write64(sst->addr.shim, offset, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write64); + +u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset) +{ + unsigned long flags; + u64 val; + + spin_lock_irqsave(&sst->spinlock, flags); + val = sst->ops->read64(sst->addr.shim, offset); + spin_unlock_irqrestore(&sst->spinlock, flags); + + return val; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read64); + +void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value) +{ + sst->ops->write(sst->addr.shim, offset, value); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write_unlocked); + +u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset) +{ + return sst->ops->read(sst->addr.shim, offset); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read_unlocked); + +void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value) +{ + sst->ops->write64(sst->addr.shim, offset, value); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write64_unlocked); + +u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset) +{ + return sst->ops->read64(sst->addr.shim, offset); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read64_unlocked); + +int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + bool change; + unsigned int old, new; + u32 ret; + + ret = sst_dsp_shim_read_unlocked(sst, offset); + + old = ret; + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write_unlocked(sst, offset, new); + + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits_unlocked); + +int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value) +{ + bool change; + u64 old, new; + + old = sst_dsp_shim_read64_unlocked(sst, offset); + + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write64_unlocked(sst, offset, new); + + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64_unlocked); + +int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sst->spinlock, flags); + change = sst_dsp_shim_update_bits_unlocked(sst, offset, mask, value); + spin_unlock_irqrestore(&sst->spinlock, flags); + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits); + +int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sst->spinlock, flags); + change = sst_dsp_shim_update_bits64_unlocked(sst, offset, mask, value); + spin_unlock_irqrestore(&sst->spinlock, flags); + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64); + +void sst_dsp_dump(struct sst_dsp *sst) +{ + sst->ops->dump(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_dump); + +void sst_dsp_reset(struct sst_dsp *sst) +{ + sst->ops->reset(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_reset); + +int sst_dsp_boot(struct sst_dsp *sst) +{ + sst->ops->boot(sst); + return 0; +} +EXPORT_SYMBOL_GPL(sst_dsp_boot); + +void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg) +{ + sst_dsp_shim_write_unlocked(dsp, SST_IPCX, msg | SST_IPCX_BUSY); + trace_sst_ipc_msg_tx(msg); +} +EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_tx); + +u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp) +{ + u32 msg; + + msg = sst_dsp_shim_read_unlocked(dsp, SST_IPCX); + trace_sst_ipc_msg_rx(msg); + + return msg; +} +EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_rx); + +int sst_dsp_mailbox_init(struct sst_dsp *sst, u32 inbox_offset, size_t inbox_size, + u32 outbox_offset, size_t outbox_size) +{ + sst->mailbox.in_base = sst->addr.lpe + inbox_offset; + sst->mailbox.out_base = sst->addr.lpe + outbox_offset; + sst->mailbox.in_size = inbox_size; + sst->mailbox.out_size = outbox_size; + return 0; +} +EXPORT_SYMBOL_GPL(sst_dsp_mailbox_init); + +void sst_dsp_outbox_write(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_outbox_write(bytes); + + memcpy_toio(sst->mailbox.out_base, message, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_outbox_wdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_outbox_write); + +void sst_dsp_outbox_read(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_outbox_read(bytes); + + memcpy_fromio(message, sst->mailbox.out_base, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_outbox_rdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_outbox_read); + +void sst_dsp_inbox_write(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_inbox_write(bytes); + + memcpy_toio(sst->mailbox.in_base, message, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_inbox_wdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_inbox_write); + +void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_inbox_read(bytes); + + memcpy_fromio(message, sst->mailbox.in_base, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_inbox_rdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); + +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) +{ + struct sst_dsp *sst; + int err; + + dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id); + + sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); + if (sst == NULL) + return NULL; + + spin_lock_init(&sst->spinlock); + mutex_init(&sst->mutex); + sst->dev = dev; + sst->thread_context = sst_dev->thread_context; + sst->sst_dev = sst_dev; + sst->id = pdata->id; + sst->irq = pdata->irq; + sst->ops = sst_dev->ops; + sst->pdata = pdata; + INIT_LIST_HEAD(&sst->used_block_list); + INIT_LIST_HEAD(&sst->free_block_list); + INIT_LIST_HEAD(&sst->module_list); + INIT_LIST_HEAD(&sst->fw_list); + + /* Initialise SST Audio DSP */ + if (sst->ops->init) { + err = sst->ops->init(sst, pdata); + if (err < 0) + return NULL; + } + + /* Register the ISR */ + err = request_threaded_irq(sst->irq, sst->ops->irq_handler, + sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); + if (err) + goto irq_err; + + return sst; + +irq_err: + if (sst->ops->free) + sst->ops->free(sst); + + return NULL; +} +EXPORT_SYMBOL_GPL(sst_dsp_new); + +void sst_dsp_free(struct sst_dsp *sst) +{ + free_irq(sst->irq, sst); + if (sst->ops->free) + sst->ops->free(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_free); + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood"); +MODULE_DESCRIPTION("Intel SST Core"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h new file mode 100644 index 0000000..0ce5c8d --- /dev/null +++ b/sound/soc/intel/sst-dsp.h @@ -0,0 +1,212 @@ +/* + * Intel Smart Sound Technology (SST) Core + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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 __SOUND_SOC_SST_DSP_H +#define __SOUND_SOC_SST_DSP_H + +#include +#include +#include + +/* SST Device IDs */ +#define SST_DEV_ID_LYNX_POINT 0x33C8 +#define SST_DEV_ID_WILDCAT_POINT 0x3438 + +/* Supported SST DMA Devices */ +#define SST_DMA_TYPE_DW 1 +#define SST_DMA_TYPE_MID 2 + +/* SST Shim register map + * The register naming can differ between products. Some products also + * contain extra functionality. + */ +#define SST_CSR 0x00 +#define SST_PISR 0x08 +#define SST_PIMR 0x10 +#define SST_ISRX 0x18 +#define SST_ISRD 0x20 +#define SST_IMRX 0x28 +#define SST_IMRD 0x30 +#define SST_IPCX 0x38 /* IPC IA -> SST */ +#define SST_IPCD 0x40 /* IPC SST -> IA */ +#define SST_ISRSC 0x48 +#define SST_ISRLPESC 0x50 +#define SST_IMRSC 0x58 +#define SST_IMRLPESC 0x60 +#define SST_IPCSC 0x68 +#define SST_IPCLPESC 0x70 +#define SST_CLKCTL 0x78 +#define SST_CSR2 0x80 +#define SST_LTRC 0xE0 +#define SST_HDMC 0xE8 +#define SST_DBGO 0xF0 + +#define SST_SHIM_SIZE 0x100 +#define SST_PWMCTRL 0x1000 + +/* SST Shim Register bits + * The register bit naming can differ between products. Some products also + * contain extra functionality. + */ + +/* CSR / CS */ +#define SST_CSR_RST (0x1 << 1) +#define SST_CSR_SBCS0 (0x1 << 2) +#define SST_CSR_SBCS1 (0x1 << 3) +#define SST_CSR_DCS(x) (x << 4) +#define SST_CSR_DCS_MASK (0x7 << 4) +#define SST_CSR_STALL (0x1 << 10) +#define SST_CSR_S0IOCS (0x1 << 21) +#define SST_CSR_S1IOCS (0x1 << 23) +#define SST_CSR_LPCS (0x1 << 31) + +/* ISRX / ISC */ +#define SST_ISRX_BUSY (0x1 << 1) +#define SST_ISRX_DONE (0x1 << 0) + +/* ISRD / ISD */ +#define SST_ISRD_BUSY (0x1 << 1) +#define SST_ISRD_DONE (0x1 << 0) + +/* IMRX / IMC */ +#define SST_IMRX_BUSY (0x1 << 1) +#define SST_IMRX_DONE (0x1 << 0) + +/* IPCX / IPCC */ +#define SST_IPCX_DONE (0x1 << 30) +#define SST_IPCX_BUSY (0x1 << 31) + +/* IPCD */ +#define SST_IPCD_DONE (0x1 << 30) +#define SST_IPCD_BUSY (0x1 << 31) + +/* CLKCTL */ +#define SST_CLKCTL_SMOS(x) (x << 24) +#define SST_CLKCTL_MASK (3 << 24) +#define SST_CLKCTL_DCPLCG (1 << 18) +#define SST_CLKCTL_SCOE1 (1 << 17) +#define SST_CLKCTL_SCOE0 (1 << 16) + +/* CSR2 / CS2 */ +#define SST_CSR2_SDFD_SSP0 (1 << 1) +#define SST_CSR2_SDFD_SSP1 (1 << 2) + +/* LTRC */ +#define SST_LTRC_VAL(x) (x << 0) + +/* HDMC */ +#define SST_HDMC_HDDA0(x) (x << 0) +#define SST_HDMC_HDDA1(x) (x << 7) + + +/* SST Vendor Defined Registers and bits */ +#define SST_VDRTCTL0 0xa0 +#define SST_VDRTCTL1 0xa4 +#define SST_VDRTCTL2 0xa8 +#define SST_VDRTCTL3 0xaC + +/* VDRTCTL0 */ +#define SST_VDRTCL0_DSRAMPGE_SHIFT 16 +#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) +#define SST_VDRTCL0_ISRAMPGE_SHIFT 6 +#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) + +struct sst_dsp; + +/* + * SST Device. + * + * This structure is populated by the SST core driver. + */ +struct sst_dsp_device { + /* Mandatory fields */ + struct sst_ops *ops; + irqreturn_t (*thread)(int irq, void *context); + void *thread_context; +}; + +/* + * SST Platform Data. + */ +struct sst_pdata { + /* ACPI data */ + u32 lpe_base; + u32 lpe_size; + u32 pcicfg_base; + u32 pcicfg_size; + int irq; + + /* Firmware */ + const char *fw_filename; + u32 fw_base; + u32 fw_size; + + /* DMA */ + u32 dma_base; + u32 dma_size; + int dma_engine; + + /* DSP */ + u32 id; + void *dsp; +}; + +/* Initialization */ +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); +void sst_dsp_free(struct sst_dsp *sst); + +/* SHIM Read / Write */ +void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value); +u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); +void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value); +u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value); + +/* SHIM Read / Write Unlocked for callers already holding sst lock */ +void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value); +u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); +void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value); +u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value); + +/* DSP reset & boot */ +void sst_dsp_reset(struct sst_dsp *sst); +int sst_dsp_boot(struct sst_dsp *sst); + +/* Msg IO */ +void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg); +u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp); + +/* Mailbox management */ +int sst_dsp_mailbox_init(struct sst_dsp *dsp, u32 inbox_offset, + size_t inbox_size, u32 outbox_offset, size_t outbox_size); +void sst_dsp_inbox_write(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_inbox_read(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_outbox_write(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_outbox_read(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_mailbox_dump(struct sst_dsp *dsp, size_t bytes); + +/* Debug */ +void sst_dsp_dump(struct sst_dsp *sst); + +#endif -- cgit v1.1 From c2f8783fa2d053a61059f6b784c917129fb3064b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 17 Feb 2014 13:32:07 +0000 Subject: ASoC: Intel: Add common SST driver loader on ACPI systems Most of the SST devices will be exposed as ACPI devices. It makes sense to avoid duplication of the driver enumeration logic and concentrate the functionality into a single ACPI SST enumeration file. Idea of this loader is to parse data we get from ACPI and to be able to load needed other SST drivers and ASoC machine driver runtime based on single ACPI ID what BIOS gives to us. Signed-off-by: Jarkko Nikula Signed-off-by: Liam Girdwood Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-acpi.c | 212 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 212 insertions(+) create mode 100644 sound/soc/intel/sst-acpi.c (limited to 'sound') diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c new file mode 100644 index 0000000..aba73ca --- /dev/null +++ b/sound/soc/intel/sst-acpi.c @@ -0,0 +1,212 @@ +/* + * Intel SST loader on ACPI systems + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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. + * + */ + +#include +#include +#include +#include + +#include "sst-dsp.h" + +#define SST_LPT_DSP_DMA_ADDR_OFFSET 0x0F0000 +#define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000 +#define SST_LPT_DSP_DMA_SIZE (1024 - 1) + +/* Descriptor for setting up SST platform data */ +struct sst_acpi_desc { + const char *drv_name; + /* Platform resource indexes. Must set to -1 if not used */ + int resindex_lpe_base; + int resindex_pcicfg_base; + int resindex_fw_base; + int irqindex_host_ipc; + int resindex_dma_base; + /* Unique number identifying the SST core on platform */ + int sst_id; + /* firmware file name */ + const char *fw_filename; + /* DMA only valid when resindex_dma_base != -1*/ + int dma_engine; + int dma_size; +}; + +/* Descriptor for SST ASoC machine driver */ +struct sst_acpi_mach { + const char *drv_name; + struct sst_acpi_desc *res_desc; +}; + +struct sst_acpi_priv { + struct platform_device *pdev_mach; + struct platform_device *pdev_pcm; + struct sst_pdata sst_pdata; + struct sst_acpi_desc *desc; +}; + +static int sst_acpi_probe(struct platform_device *pdev) +{ + const struct acpi_device_id *id; + struct device *dev = &pdev->dev; + struct sst_acpi_priv *sst_acpi; + struct sst_pdata *sst_pdata; + struct sst_acpi_mach *mach; + struct sst_acpi_desc *desc; + struct resource *mmio; + int ret = 0; + + sst_acpi = devm_kzalloc(dev, sizeof(*sst_acpi), GFP_KERNEL); + if (sst_acpi == NULL) + return -ENOMEM; + + id = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!id) + return -ENODEV; + + mach = (struct sst_acpi_mach *)id->driver_data; + desc = mach->res_desc; + sst_pdata = &sst_acpi->sst_pdata; + sst_pdata->id = desc->sst_id; + sst_pdata->fw_filename = desc->fw_filename; + sst_acpi->desc = desc; + + if (desc->resindex_dma_base >= 0) { + sst_pdata->dma_engine = desc->dma_engine; + sst_pdata->dma_base = desc->resindex_dma_base; + sst_pdata->dma_size = desc->dma_size; + } + + if (desc->irqindex_host_ipc >= 0) + sst_pdata->irq = platform_get_irq(pdev, desc->irqindex_host_ipc); + + if (desc->resindex_lpe_base >= 0) { + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_lpe_base); + if (mmio) { + sst_pdata->lpe_base = mmio->start; + sst_pdata->lpe_size = resource_size(mmio); + } + } + + if (desc->resindex_pcicfg_base >= 0) { + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_pcicfg_base); + if (mmio) { + sst_pdata->pcicfg_base = mmio->start; + sst_pdata->pcicfg_size = resource_size(mmio); + } + } + + if (desc->resindex_fw_base >= 0) { + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_fw_base); + if (mmio) { + sst_pdata->fw_base = mmio->start; + sst_pdata->fw_size = resource_size(mmio); + } + } + + /* register PCM and DAI driver */ + sst_acpi->pdev_pcm = + platform_device_register_data(dev, desc->drv_name, -1, + sst_pdata, sizeof(*sst_pdata)); + if (IS_ERR(sst_acpi->pdev_pcm)) + return PTR_ERR(sst_acpi->pdev_pcm); + + /* register machine driver */ + platform_set_drvdata(pdev, sst_acpi); + + sst_acpi->pdev_mach = + platform_device_register_data(dev, mach->drv_name, -1, + sst_pdata, sizeof(*sst_pdata)); + if (IS_ERR(sst_acpi->pdev_mach)) { + ret = PTR_ERR(sst_acpi->pdev_mach); + goto sst_err; + } + + return ret; + +sst_err: + platform_device_unregister(sst_acpi->pdev_pcm); + return ret; +} + +static int sst_acpi_remove(struct platform_device *pdev) +{ + struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + + platform_device_unregister(sst_acpi->pdev_mach); + platform_device_unregister(sst_acpi->pdev_pcm); + + return 0; +} + +static struct sst_acpi_desc sst_acpi_haswell_desc = { + .drv_name = "haswell-pcm-audio", + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_fw_base = -1, + .irqindex_host_ipc = 0, + .sst_id = SST_DEV_ID_LYNX_POINT, + .fw_filename = "intel/IntcSST1.bin", + .dma_engine = SST_DMA_TYPE_DW, + .resindex_dma_base = SST_LPT_DSP_DMA_ADDR_OFFSET, + .dma_size = SST_LPT_DSP_DMA_SIZE, +}; + +static struct sst_acpi_desc sst_acpi_broadwell_desc = { + .drv_name = "haswell-pcm-audio", + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_fw_base = -1, + .irqindex_host_ipc = 0, + .sst_id = SST_DEV_ID_WILDCAT_POINT, + .fw_filename = "intel/IntcSST2.bin", + .dma_engine = SST_DMA_TYPE_DW, + .resindex_dma_base = SST_WPT_DSP_DMA_ADDR_OFFSET, + .dma_size = SST_LPT_DSP_DMA_SIZE, +}; + +static struct sst_acpi_mach haswell_mach = { + .drv_name = "haswell-audio", + .res_desc = &sst_acpi_haswell_desc, +}; + +static struct sst_acpi_mach broadwell_mach = { + .drv_name = "broadwell-audio", + .res_desc = &sst_acpi_broadwell_desc, +}; + +static struct acpi_device_id sst_acpi_match[] = { + { "INT33C8", (unsigned long)&haswell_mach }, + { "INT3438", (unsigned long)&broadwell_mach }, + { } +}; +MODULE_DEVICE_TABLE(acpi, sst_acpi_match); + +static struct platform_driver sst_acpi_driver = { + .probe = sst_acpi_probe, + .remove = sst_acpi_remove, + .driver = { + .name = "sst-acpi", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(sst_acpi_match), + }, +}; +module_platform_driver(sst_acpi_driver); + +MODULE_AUTHOR("Jarkko Nikula "); +MODULE_DESCRIPTION("Intel SST loader on ACPI systems"); +MODULE_LICENSE("GPL v2"); -- cgit v1.1 From 30020472c354fbe4352b4b4d59bbc9a30aacf5c3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 17 Feb 2014 13:32:08 +0000 Subject: ASoC: Intel: Add Intel SST audio DSP Firmware loader. Provide services for Intel SST drivers to load SST modular firmware. SST Firmware can be made up of several modules. These modules can exist within any of the compatible SST memory blocks. Provide a generic memory block and firmware module manager that can be used with any SST firmware and core. Signed-off-by: Liam Girdwood Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 586 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 586 insertions(+) create mode 100644 sound/soc/intel/sst-firmware.c (limited to 'sound') diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c new file mode 100644 index 0000000..b6f9b5e --- /dev/null +++ b/sound/soc/intel/sst-firmware.c @@ -0,0 +1,586 @@ +/* + * Intel SST Firmware Loader + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +static void sst_memcpy32(void *dest, void *src, u32 bytes) +{ + u32 i; + + /* copy one 32 bit word at a time as 64 bit access is not supported */ + for (i = 0; i < bytes; i += 4) + memcpy_toio(dest + i, src + i, 4); +} + +/* create new generic firmware object */ +struct sst_fw *sst_fw_new(struct sst_dsp *dsp, + const struct firmware *fw, void *private) +{ + struct sst_fw *sst_fw; + int err; + + if (!dsp->ops->parse_fw) + return NULL; + + sst_fw = kzalloc(sizeof(*sst_fw), GFP_KERNEL); + if (sst_fw == NULL) + return NULL; + + sst_fw->dsp = dsp; + sst_fw->private = private; + sst_fw->size = fw->size; + + err = dma_coerce_mask_and_coherent(dsp->dev, DMA_BIT_MASK(32)); + if (err < 0) { + kfree(sst_fw); + return NULL; + } + + /* allocate DMA buffer to store FW data */ + sst_fw->dma_buf = dma_alloc_coherent(dsp->dev, sst_fw->size, + &sst_fw->dmable_fw_paddr, GFP_DMA); + if (!sst_fw->dma_buf) { + dev_err(dsp->dev, "error: DMA alloc failed\n"); + kfree(sst_fw); + return NULL; + } + + /* copy FW data to DMA-able memory */ + memcpy((void *)sst_fw->dma_buf, (void *)fw->data, fw->size); + + /* call core specific FW paser to load FW data into DSP */ + err = dsp->ops->parse_fw(sst_fw); + if (err < 0) { + dev_err(dsp->dev, "error: parse fw failed %d\n", err); + goto parse_err; + } + + mutex_lock(&dsp->mutex); + list_add(&sst_fw->list, &dsp->fw_list); + mutex_unlock(&dsp->mutex); + + return sst_fw; + +parse_err: + dma_free_coherent(dsp->dev, sst_fw->size, + sst_fw->dma_buf, + sst_fw->dmable_fw_paddr); + kfree(sst_fw); + return NULL; +} +EXPORT_SYMBOL_GPL(sst_fw_new); + +/* free single firmware object */ +void sst_fw_free(struct sst_fw *sst_fw) +{ + struct sst_dsp *dsp = sst_fw->dsp; + + mutex_lock(&dsp->mutex); + list_del(&sst_fw->list); + mutex_unlock(&dsp->mutex); + + dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf, + sst_fw->dmable_fw_paddr); + kfree(sst_fw); +} +EXPORT_SYMBOL_GPL(sst_fw_free); + +/* free all firmware objects */ +void sst_fw_free_all(struct sst_dsp *dsp) +{ + struct sst_fw *sst_fw, *t; + + mutex_lock(&dsp->mutex); + list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) { + + list_del(&sst_fw->list); + dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf, + sst_fw->dmable_fw_paddr); + kfree(sst_fw); + } + mutex_unlock(&dsp->mutex); +} +EXPORT_SYMBOL_GPL(sst_fw_free_all); + +/* create a new SST generic module from FW template */ +struct sst_module *sst_module_new(struct sst_fw *sst_fw, + struct sst_module_template *template, void *private) +{ + struct sst_dsp *dsp = sst_fw->dsp; + struct sst_module *sst_module; + + sst_module = kzalloc(sizeof(*sst_module), GFP_KERNEL); + if (sst_module == NULL) + return NULL; + + sst_module->id = template->id; + sst_module->dsp = dsp; + sst_module->sst_fw = sst_fw; + + memcpy(&sst_module->s, &template->s, sizeof(struct sst_module_data)); + memcpy(&sst_module->p, &template->p, sizeof(struct sst_module_data)); + + INIT_LIST_HEAD(&sst_module->block_list); + + mutex_lock(&dsp->mutex); + list_add(&sst_module->list, &dsp->module_list); + mutex_unlock(&dsp->mutex); + + return sst_module; +} +EXPORT_SYMBOL_GPL(sst_module_new); + +/* free firmware module and remove from available list */ +void sst_module_free(struct sst_module *sst_module) +{ + struct sst_dsp *dsp = sst_module->dsp; + + mutex_lock(&dsp->mutex); + list_del(&sst_module->list); + mutex_unlock(&dsp->mutex); + + kfree(sst_module); +} +EXPORT_SYMBOL_GPL(sst_module_free); + +static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type, + u32 offset) +{ + struct sst_mem_block *block; + + list_for_each_entry(block, &dsp->free_block_list, list) { + if (block->type == type && block->offset == offset) + return block; + } + + return NULL; +} + +static int block_alloc_contiguous(struct sst_module *module, + struct sst_module_data *data, u32 offset, int size) +{ + struct list_head tmp = LIST_HEAD_INIT(tmp); + struct sst_dsp *dsp = module->dsp; + struct sst_mem_block *block; + + while (size > 0) { + block = find_block(dsp, data->type, offset); + if (!block) { + list_splice(&tmp, &dsp->free_block_list); + return -ENOMEM; + } + + list_move_tail(&block->list, &tmp); + offset += block->size; + size -= block->size; + } + + list_splice(&tmp, &dsp->used_block_list); + return 0; +} + +/* allocate free DSP blocks for module data - callers hold locks */ +static int block_alloc(struct sst_module *module, + struct sst_module_data *data) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_mem_block *block, *tmp; + int ret = 0; + + if (data->size == 0) + return 0; + + /* find first free whole blocks that can hold module */ + list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { + + /* ignore blocks with wrong type */ + if (block->type != data->type) + continue; + + if (data->size > block->size) + continue; + + data->offset = block->offset; + block->data_type = data->data_type; + block->bytes_used = data->size % block->size; + list_add(&block->module_list, &module->block_list); + list_move(&block->list, &dsp->used_block_list); + dev_dbg(dsp->dev, " *module %d added block %d:%d\n", + module->id, block->type, block->index); + return 0; + } + + /* then find free multiple blocks that can hold module */ + list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { + + /* ignore blocks with wrong type */ + if (block->type != data->type) + continue; + + /* do we span > 1 blocks */ + if (data->size > block->size) { + ret = block_alloc_contiguous(module, data, + block->offset + block->size, + data->size - block->size); + if (ret == 0) + return ret; + } + } + + /* not enough free block space */ + return -ENOMEM; +} + +/* remove module from memory - callers hold locks */ +static void block_module_remove(struct sst_module *module) +{ + struct sst_mem_block *block, *tmp; + struct sst_dsp *dsp = module->dsp; + int err; + + /* disable each block */ + list_for_each_entry(block, &module->block_list, module_list) { + + if (block->ops && block->ops->disable) { + err = block->ops->disable(block); + if (err < 0) + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + } + } + + /* mark each block as free */ + list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + list_del(&block->module_list); + list_move(&block->list, &dsp->free_block_list); + } +} + +/* prepare the memory block to receive data from host - callers hold locks */ +static int block_module_prepare(struct sst_module *module) +{ + struct sst_mem_block *block; + int ret = 0; + + /* enable each block so that's it'e ready for module P/S data */ + list_for_each_entry(block, &module->block_list, module_list) { + + if (block->ops && block->ops->enable) + ret = block->ops->enable(block); + if (ret < 0) { + dev_err(module->dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + goto err; + } + } + return ret; + +err: + list_for_each_entry(block, &module->block_list, module_list) { + if (block->ops && block->ops->disable) + block->ops->disable(block); + } + return ret; +} + +/* allocate memory blocks for static module addresses - callers hold locks */ +static int block_alloc_fixed(struct sst_module *module, + struct sst_module_data *data) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_mem_block *block, *tmp; + u32 end = data->offset + data->size, block_end; + int err; + + /* only IRAM/DRAM blocks are managed */ + if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM) + return 0; + + /* are blocks already attached to this module */ + list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + + /* force compacting mem blocks of the same data_type */ + if (block->data_type != data->data_type) + continue; + + block_end = block->offset + block->size; + + /* find block that holds section */ + if (data->offset >= block->offset && end < block_end) + return 0; + + /* does block span more than 1 section */ + if (data->offset >= block->offset && data->offset < block_end) { + + err = block_alloc_contiguous(module, data, + block->offset + block->size, + data->size - block->size + data->offset - block->offset); + if (err < 0) + return -ENOMEM; + + /* module already owns blocks */ + return 0; + } + } + + /* find first free blocks that can hold section in free list */ + list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { + block_end = block->offset + block->size; + + /* find block that holds section */ + if (data->offset >= block->offset && end < block_end) { + + /* add block */ + block->data_type = data->data_type; + list_move(&block->list, &dsp->used_block_list); + list_add(&block->module_list, &module->block_list); + return 0; + } + + /* does block span more than 1 section */ + if (data->offset >= block->offset && data->offset < block_end) { + + err = block_alloc_contiguous(module, data, + block->offset + block->size, + data->size - block->size); + if (err < 0) + return -ENOMEM; + + /* add block */ + block->data_type = data->data_type; + list_move(&block->list, &dsp->used_block_list); + list_add(&block->module_list, &module->block_list); + return 0; + } + + } + + return -ENOMEM; +} + +/* Load fixed module data into DSP memory blocks */ +int sst_module_insert_fixed_block(struct sst_module *module, + struct sst_module_data *data) +{ + struct sst_dsp *dsp = module->dsp; + int ret; + + mutex_lock(&dsp->mutex); + + /* alloc blocks that includes this section */ + ret = block_alloc_fixed(module, data); + if (ret < 0) { + dev_err(dsp->dev, + "error: no free blocks for section at offset 0x%x size 0x%x\n", + data->offset, data->size); + mutex_unlock(&dsp->mutex); + return -ENOMEM; + } + + /* prepare DSP blocks for module copy */ + ret = block_module_prepare(module); + if (ret < 0) { + dev_err(dsp->dev, "error: fw module prepare failed\n"); + goto err; + } + + /* copy partial module data to blocks */ + sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size); + + mutex_unlock(&dsp->mutex); + return ret; + +err: + block_module_remove(module); + mutex_unlock(&dsp->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block); + +/* Unload entire module from DSP memory */ +int sst_block_module_remove(struct sst_module *module) +{ + struct sst_dsp *dsp = module->dsp; + + mutex_lock(&dsp->mutex); + block_module_remove(module); + mutex_unlock(&dsp->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(sst_block_module_remove); + +/* register a DSP memory block for use with FW based modules */ +struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, + u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, + void *private) +{ + struct sst_mem_block *block; + + block = kzalloc(sizeof(*block), GFP_KERNEL); + if (block == NULL) + return NULL; + + block->offset = offset; + block->size = size; + block->index = index; + block->type = type; + block->dsp = dsp; + block->private = private; + block->ops = ops; + + mutex_lock(&dsp->mutex); + list_add(&block->list, &dsp->free_block_list); + mutex_unlock(&dsp->mutex); + + return block; +} +EXPORT_SYMBOL_GPL(sst_mem_block_register); + +/* unregister all DSP memory blocks */ +void sst_mem_block_unregister_all(struct sst_dsp *dsp) +{ + struct sst_mem_block *block, *tmp; + + mutex_lock(&dsp->mutex); + + /* unregister used blocks */ + list_for_each_entry_safe(block, tmp, &dsp->used_block_list, list) { + list_del(&block->list); + kfree(block); + } + + /* unregister free blocks */ + list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { + list_del(&block->list); + kfree(block); + } + + mutex_unlock(&dsp->mutex); +} +EXPORT_SYMBOL_GPL(sst_mem_block_unregister_all); + +/* allocate scratch buffer blocks */ +struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp) +{ + struct sst_module *sst_module, *scratch; + struct sst_mem_block *block, *tmp; + u32 block_size; + int ret = 0; + + scratch = kzalloc(sizeof(struct sst_module), GFP_KERNEL); + if (scratch == NULL) + return NULL; + + mutex_lock(&dsp->mutex); + + /* calculate required scratch size */ + list_for_each_entry(sst_module, &dsp->module_list, list) { + if (scratch->s.size > sst_module->s.size) + scratch->s.size = scratch->s.size; + else + scratch->s.size = sst_module->s.size; + } + + dev_dbg(dsp->dev, "scratch buffer required is %d bytes\n", + scratch->s.size); + + /* init scratch module */ + scratch->dsp = dsp; + scratch->s.type = SST_MEM_DRAM; + scratch->s.data_type = SST_DATA_S; + INIT_LIST_HEAD(&scratch->block_list); + + /* check free blocks before looking at used blocks for space */ + if (!list_empty(&dsp->free_block_list)) + block = list_first_entry(&dsp->free_block_list, + struct sst_mem_block, list); + else + block = list_first_entry(&dsp->used_block_list, + struct sst_mem_block, list); + block_size = block->size; + + /* allocate blocks for module scratch buffers */ + dev_dbg(dsp->dev, "allocating scratch blocks\n"); + ret = block_alloc(scratch, &scratch->s); + if (ret < 0) { + dev_err(dsp->dev, "error: can't alloc scratch blocks\n"); + goto err; + } + + /* assign the same offset of scratch to each module */ + list_for_each_entry(sst_module, &dsp->module_list, list) + sst_module->s.offset = scratch->s.offset; + + mutex_unlock(&dsp->mutex); + return scratch; + +err: + list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list) + list_del(&block->module_list); + mutex_unlock(&dsp->mutex); + return NULL; +} +EXPORT_SYMBOL_GPL(sst_mem_block_alloc_scratch); + +/* free all scratch blocks */ +void sst_mem_block_free_scratch(struct sst_dsp *dsp, + struct sst_module *scratch) +{ + struct sst_mem_block *block, *tmp; + + mutex_lock(&dsp->mutex); + + list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list) + list_del(&block->module_list); + + mutex_unlock(&dsp->mutex); +} +EXPORT_SYMBOL_GPL(sst_mem_block_free_scratch); + +/* get a module from it's unique ID */ +struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id) +{ + struct sst_module *module; + + mutex_lock(&dsp->mutex); + + list_for_each_entry(module, &dsp->module_list, list) { + if (module->id == id) { + mutex_unlock(&dsp->mutex); + return module; + } + } + + mutex_unlock(&dsp->mutex); + return NULL; +} +EXPORT_SYMBOL_GPL(sst_module_get_from_id); -- cgit v1.1 From ddfa40b1586f8c7c6bb8bb9dd398cf656c98e6ee Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 17 Feb 2014 13:32:10 +0000 Subject: ASoC: Intel: Add build support for Intel SST DSP core. This adds kernel build support for Intel SST core audio. Signed-off-by: Liam Girdwood Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 12 ++++++++++++ sound/soc/intel/Makefile | 7 +++++++ 2 files changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4d9d0a5..b0a07e4 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -11,3 +11,15 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate + +config SND_SOC_INTEL_SST + tristate "ASoC support for Intel(R) Smart Sound Technology" + select SND_SOC_INTEL_SST_ACPI if ACPI + help + This adds support for Intel(R) Smart Sound Technology (SST). + Say Y if you have such a device + If unsure select "N". + +config SND_SOC_INTEL_SST_ACPI + tristate + diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index eb899fc..cf47100 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,5 +1,12 @@ +# Core support +snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o +snd-soc-sst-acpi-objs := sst-acpi.o + snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o snd-soc-mfld-machine-objs := mfld_machine.o obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o + +obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o +obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o -- cgit v1.1 From 2280e90efae92060f6fa717fb1afa9861bd2b520 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 18 Feb 2014 10:41:50 +0000 Subject: ASoC: Intel: Add GFP_KERNEL flag to firmware DMA buffer. Add GFP_KERNEL when allocating firmware DMA buffer. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index b6f9b5e..31cd154 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -65,7 +65,7 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, /* allocate DMA buffer to store FW data */ sst_fw->dma_buf = dma_alloc_coherent(dsp->dev, sst_fw->size, - &sst_fw->dmable_fw_paddr, GFP_DMA); + &sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL); if (!sst_fw->dma_buf) { dev_err(dsp->dev, "error: DMA alloc failed\n"); kfree(sst_fw); -- cgit v1.1 From b47bd099fbb3412e34e8915c7a06bc3dbe73bc01 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 18 Feb 2014 11:43:16 +0000 Subject: ASoC: Intel: Rename SST trace event header to be less generic. The Intel audio DSP SST trace event header has been renamed from sst.h to intel-sst.h in order to avoid any confusion with any future Samoa Standard Time drivers ;) Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 1888de54..e0ad2e5 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -24,7 +24,7 @@ #include "sst-dsp-priv.h" #define CREATE_TRACE_POINTS -#include +#include /* Public API */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) -- cgit v1.1 From 0c49a8c9f9fafdf4345259b9b50513905afcf860 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 18 Feb 2014 16:42:02 +0200 Subject: ASoC: Intel: Move extended fw base and size fields in struct sst_pdata Move fw_base and fw_size fields in struct sst_pdata under ACPI data for clarifying that these are not related to firmware file but for platform specific extended firmware area reserved by the BIOS. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 0ce5c8d..d134359 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -147,12 +147,12 @@ struct sst_pdata { u32 lpe_size; u32 pcicfg_base; u32 pcicfg_size; + u32 fw_base; + u32 fw_size; int irq; /* Firmware */ const char *fw_filename; - u32 fw_base; - u32 fw_size; /* DMA */ u32 dma_base; -- cgit v1.1 From e5161d7987f14338ad0a3cf376b9bb6838416183 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 19 Feb 2014 10:30:38 +0200 Subject: ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing We originally thought to request SST audio DSP firmware during the SST platform driver initialization. However plain request_firmware doesn't work in driver probe paths if userspace is not ready to handle it. For instance when drivers are built-in. Implementing asynchronous firmware request in SST platform driver initialization complicates code needlessly since it anyway will fail if firmware is missing. This is more simple to handle by requesting firmware asynchronously in sst_acpi_probe() and register SST platform only after firmware is loaded. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-acpi.c | 56 ++++++++++++++++++++++++++++++++-------------- sound/soc/intel/sst-dsp.h | 2 +- 2 files changed, 40 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index aba73ca..64154a7 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -16,6 +16,7 @@ #include #include +#include #include #include @@ -56,6 +57,32 @@ struct sst_acpi_priv { struct sst_acpi_desc *desc; }; +static void sst_acpi_fw_cb(const struct firmware *fw, void *context) +{ + struct platform_device *pdev = context; + struct device *dev = &pdev->dev; + struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; + struct sst_acpi_desc *desc = sst_acpi->desc; + + sst_pdata->fw = fw; + if (!fw) { + dev_err(dev, "Cannot load firmware %s\n", desc->fw_filename); + return; + } + + /* register PCM and DAI driver */ + sst_acpi->pdev_pcm = + platform_device_register_data(dev, desc->drv_name, -1, + sst_pdata, sizeof(*sst_pdata)); + if (IS_ERR(sst_acpi->pdev_pcm)) { + dev_err(dev, "Cannot register device %s. Error %d\n", + desc->drv_name, (int)PTR_ERR(sst_acpi->pdev_pcm)); + } + + return; +} + static int sst_acpi_probe(struct platform_device *pdev) { const struct acpi_device_id *id; @@ -79,7 +106,6 @@ static int sst_acpi_probe(struct platform_device *pdev) desc = mach->res_desc; sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; - sst_pdata->fw_filename = desc->fw_filename; sst_acpi->desc = desc; if (desc->resindex_dma_base >= 0) { @@ -118,37 +144,33 @@ static int sst_acpi_probe(struct platform_device *pdev) } } - /* register PCM and DAI driver */ - sst_acpi->pdev_pcm = - platform_device_register_data(dev, desc->drv_name, -1, - sst_pdata, sizeof(*sst_pdata)); - if (IS_ERR(sst_acpi->pdev_pcm)) - return PTR_ERR(sst_acpi->pdev_pcm); - - /* register machine driver */ platform_set_drvdata(pdev, sst_acpi); + /* register machine driver */ sst_acpi->pdev_mach = platform_device_register_data(dev, mach->drv_name, -1, sst_pdata, sizeof(*sst_pdata)); - if (IS_ERR(sst_acpi->pdev_mach)) { - ret = PTR_ERR(sst_acpi->pdev_mach); - goto sst_err; - } + if (IS_ERR(sst_acpi->pdev_mach)) + return PTR_ERR(sst_acpi->pdev_mach); - return ret; + /* continue SST probing after firmware is loaded */ + ret = request_firmware_nowait(THIS_MODULE, true, desc->fw_filename, + dev, GFP_KERNEL, pdev, sst_acpi_fw_cb); + if (ret) + platform_device_unregister(sst_acpi->pdev_mach); -sst_err: - platform_device_unregister(sst_acpi->pdev_pcm); return ret; } static int sst_acpi_remove(struct platform_device *pdev) { struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; platform_device_unregister(sst_acpi->pdev_mach); - platform_device_unregister(sst_acpi->pdev_pcm); + if (!IS_ERR_OR_NULL(sst_acpi->pdev_pcm)) + platform_device_unregister(sst_acpi->pdev_pcm); + release_firmware(sst_pdata->fw); return 0; } diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index d134359..3730fd3 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -152,7 +152,7 @@ struct sst_pdata { int irq; /* Firmware */ - const char *fw_filename; + const struct firmware *fw; /* DMA */ u32 dma_base; -- cgit v1.1 From 7bcd84878ebd87168682b117e1cb9dd8a55a1a8b Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 19 Feb 2014 14:06:23 +0000 Subject: ASoC: Intel: Fix sparse warnings for firmware loader Sparse gives us the following warnings on sst-firmware.c CHECK sound/soc/intel/sst-firmware.c sound/soc/intel/sst-firmware.c:39:34: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:39:34: expected void volatile [noderef] *dst sound/soc/intel/sst-firmware.c:39:34: got void * sound/soc/intel/sst-firmware.c:417:36: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:417:36: expected void *dest sound/soc/intel/sst-firmware.c:417:36: got void [noderef] * sound/soc/intel/sst-firmware.c:430:5: warning: symbol 'sst_block_module_remove' was not declared. Should it be static? and CC [M] sound/soc/intel/sst-dsp.o sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] * sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] * sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] * This patch removes these warnings Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp-priv.h | 7 +++++-- sound/soc/intel/sst-firmware.c | 2 +- 2 files changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index 35de547..fa2c780 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -41,8 +41,10 @@ struct sst_ops { u64 (*read64)(void __iomem *addr, u32 offset); /* DSP I/DRAM IO */ - void (*ram_read)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); - void (*ram_write)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + void (*ram_read)(struct sst_dsp *sst, void *dest, void __iomem *src, + size_t bytes); + void (*ram_write)(struct sst_dsp *sst, void __iomem *dest, void *src, + size_t bytes); void (*dump)(struct sst_dsp *); @@ -295,6 +297,7 @@ struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); void sst_mem_block_free_scratch(struct sst_dsp *dsp, struct sst_module *scratch); +int sst_block_module_remove(struct sst_module *module); /* Register the DSPs memory blocks - would be nice to read from ACPI */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 31cd154..dee7eb5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void sst_memcpy32(void *dest, void *src, u32 bytes) +static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { u32 i; -- cgit v1.1 From 6dda27cbbd1d2d2ac4833236f5fd8a81c14d200a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 19 Feb 2014 16:35:58 +0200 Subject: ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform Initial implementation of this driver focused only matching SST ACPI ID with single machine driver and same firmware file per platform. It was known restriction to be improved incrementally. This patch is now changing this that SST ACPI ID refers purely to platform specific data which refers to machine drivers on this platform, not vice versa. Matching machine driver is found by looking at ACPI ID which would best match with the driver. Typically this would be the ACPI ID of audio codec but is not tied to it. This patch also changes that DSP firmware name is machine not platform specific. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-acpi.c | 84 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 58 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 64154a7..c7e36c9 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -26,9 +26,20 @@ #define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000 #define SST_LPT_DSP_DMA_SIZE (1024 - 1) +/* Descriptor for SST ASoC machine driver */ +struct sst_acpi_mach { + /* ACPI ID for the matching machine driver. Audio codec for instance */ + const u8 id[ACPI_ID_LEN]; + /* machine driver name */ + const char *drv_name; + /* firmware file name */ + const char *fw_filename; +}; + /* Descriptor for setting up SST platform data */ struct sst_acpi_desc { const char *drv_name; + struct sst_acpi_mach *machines; /* Platform resource indexes. Must set to -1 if not used */ int resindex_lpe_base; int resindex_pcicfg_base; @@ -37,24 +48,17 @@ struct sst_acpi_desc { int resindex_dma_base; /* Unique number identifying the SST core on platform */ int sst_id; - /* firmware file name */ - const char *fw_filename; /* DMA only valid when resindex_dma_base != -1*/ int dma_engine; int dma_size; }; -/* Descriptor for SST ASoC machine driver */ -struct sst_acpi_mach { - const char *drv_name; - struct sst_acpi_desc *res_desc; -}; - struct sst_acpi_priv { struct platform_device *pdev_mach; struct platform_device *pdev_pcm; struct sst_pdata sst_pdata; struct sst_acpi_desc *desc; + struct sst_acpi_mach *mach; }; static void sst_acpi_fw_cb(const struct firmware *fw, void *context) @@ -64,10 +68,11 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; struct sst_acpi_desc *desc = sst_acpi->desc; + struct sst_acpi_mach *mach = sst_acpi->mach; sst_pdata->fw = fw; if (!fw) { - dev_err(dev, "Cannot load firmware %s\n", desc->fw_filename); + dev_err(dev, "Cannot load firmware %s\n", mach->fw_filename); return; } @@ -83,6 +88,28 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) return; } +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_acpi_mach *sst_acpi_find_machine( + struct sst_acpi_mach *machines) +{ + struct sst_acpi_mach *mach; + bool found = false; + + for (mach = machines; mach->id[0]; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + static int sst_acpi_probe(struct platform_device *pdev) { const struct acpi_device_id *id; @@ -102,8 +129,13 @@ static int sst_acpi_probe(struct platform_device *pdev) if (!id) return -ENODEV; - mach = (struct sst_acpi_mach *)id->driver_data; - desc = mach->res_desc; + desc = (struct sst_acpi_desc *)id->driver_data; + mach = sst_acpi_find_machine(desc->machines); + if (mach == NULL) { + dev_err(dev, "No matching ASoC machine driver found\n"); + return -ENODEV; + } + sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; sst_acpi->desc = desc; @@ -154,7 +186,7 @@ static int sst_acpi_probe(struct platform_device *pdev) return PTR_ERR(sst_acpi->pdev_mach); /* continue SST probing after firmware is loaded */ - ret = request_firmware_nowait(THIS_MODULE, true, desc->fw_filename, + ret = request_firmware_nowait(THIS_MODULE, true, mach->fw_filename, dev, GFP_KERNEL, pdev, sst_acpi_fw_cb); if (ret) platform_device_unregister(sst_acpi->pdev_mach); @@ -175,45 +207,45 @@ static int sst_acpi_remove(struct platform_device *pdev) return 0; } +static struct sst_acpi_mach haswell_machines[] = { + { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" }, + {} +}; + static struct sst_acpi_desc sst_acpi_haswell_desc = { .drv_name = "haswell-pcm-audio", + .machines = haswell_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = 1, .resindex_fw_base = -1, .irqindex_host_ipc = 0, .sst_id = SST_DEV_ID_LYNX_POINT, - .fw_filename = "intel/IntcSST1.bin", .dma_engine = SST_DMA_TYPE_DW, .resindex_dma_base = SST_LPT_DSP_DMA_ADDR_OFFSET, .dma_size = SST_LPT_DSP_DMA_SIZE, }; +static struct sst_acpi_mach broadwell_machines[] = { + { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" }, + {} +}; + static struct sst_acpi_desc sst_acpi_broadwell_desc = { .drv_name = "haswell-pcm-audio", + .machines = broadwell_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = 1, .resindex_fw_base = -1, .irqindex_host_ipc = 0, .sst_id = SST_DEV_ID_WILDCAT_POINT, - .fw_filename = "intel/IntcSST2.bin", .dma_engine = SST_DMA_TYPE_DW, .resindex_dma_base = SST_WPT_DSP_DMA_ADDR_OFFSET, .dma_size = SST_LPT_DSP_DMA_SIZE, }; -static struct sst_acpi_mach haswell_mach = { - .drv_name = "haswell-audio", - .res_desc = &sst_acpi_haswell_desc, -}; - -static struct sst_acpi_mach broadwell_mach = { - .drv_name = "broadwell-audio", - .res_desc = &sst_acpi_broadwell_desc, -}; - static struct acpi_device_id sst_acpi_match[] = { - { "INT33C8", (unsigned long)&haswell_mach }, - { "INT3438", (unsigned long)&broadwell_mach }, + { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, + { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, { } }; MODULE_DEVICE_TABLE(acpi, sst_acpi_match); -- cgit v1.1 From afd954900af84bf66a0dd72b2648a2c15fb68f25 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 19 Feb 2014 16:45:57 +0000 Subject: ASoC: Intel: Fix build for sst-dsp.c on PPC architecture Disable build on non X86 architectures except for compile testing. This fixes the following build errors on PPC and adds an option for testing the build on other architectures as suggested by Mark Brown :- sound/soc/intel/sst-dsp.c: In function 'sst_dsp_outbox_write': sound/soc/intel/sst-dsp.c:218:2: error: implicit declaration of function 'memcpy_toio' [-Werror=implicit-function-declaration] memcpy_toio(sst->mailbox.out_base, message, bytes); ^ sound/soc/intel/sst-dsp.c: In function 'sst_dsp_outbox_read': sound/soc/intel/sst-dsp.c:231:2: error: implicit declaration of function 'memcpy_fromio' [-Werror=implicit-function-declaration] memcpy_fromio(message, sst->mailbox.out_base, bytes); ^ Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 1 + sound/soc/intel/sst-dsp.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b0a07e4..dd048fe 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -15,6 +15,7 @@ config SND_SST_MFLD_PLATFORM config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI + depends on (X86 || COMPILE_TEST) help This adds support for Intel(R) Smart Sound Technology (SST). Say Y if you have such a device diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index e0ad2e5..6e22c12 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -19,6 +19,7 @@ #include #include #include +#include #include "sst-dsp.h" #include "sst-dsp-priv.h" -- cgit v1.1 From 6415e307b1d270e4e5f7ee7bd9aac4ac4f1daf65 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:09:52 +0100 Subject: ASoC: lm4857: Use SOC_ENUM_SINGLE_EXT_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 0e5743e..4f048db 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -101,8 +101,7 @@ static const char *lm4857_mode[] = { "Headphone", }; -static const struct soc_enum lm4857_mode_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode); +static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode); static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN"), -- cgit v1.1 From a0628934d6d3fcca5588fd9617270f63c5387f1b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:12:32 +0100 Subject: ASoC: max98088: Use SOC_*_ENUM_SINGLE_DECL() Just replace with the helper macros. No functional change at all. Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 39 +++++++++++++++++++-------------------- 1 file changed, 19 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ee660e2..25ce135 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -597,28 +597,27 @@ static const unsigned int max98088_exmode_values[] = { 0x00, 0x43, 0x10, 0x20, 0x30, 0x40, 0x11, 0x22, 0x32 }; -static const struct soc_enum max98088_exmode_enum = - SOC_VALUE_ENUM_SINGLE(M98088_REG_41_SPKDHP, 0, 127, - ARRAY_SIZE(max98088_exmode_texts), - max98088_exmode_texts, - max98088_exmode_values); +static SOC_VALUE_ENUM_SINGLE_DECL(max98088_exmode_enum, + M98088_REG_41_SPKDHP, 0, 127, + max98088_exmode_texts, + max98088_exmode_values); static const char *max98088_ex_thresh[] = { /* volts PP */ "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; -static const struct soc_enum max98088_ex_thresh_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_42_SPKDHP_THRESH, 0, 8, - max98088_ex_thresh), -}; +static SOC_ENUM_SINGLE_DECL(max98088_ex_thresh_enum, + M98088_REG_42_SPKDHP_THRESH, 0, + max98088_ex_thresh); static const char *max98088_fltr_mode[] = {"Voice", "Music" }; -static const struct soc_enum max98088_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 7, 2, max98088_fltr_mode), -}; +static SOC_ENUM_SINGLE_DECL(max98088_filter_mode_enum, + M98088_REG_18_DAI1_FILTERS, 7, + max98088_fltr_mode); static const char *max98088_extmic_text[] = { "None", "MIC1", "MIC2" }; -static const struct soc_enum max98088_extmic_enum = - SOC_ENUM_SINGLE(M98088_REG_48_CFG_MIC, 0, 3, max98088_extmic_text); +static SOC_ENUM_SINGLE_DECL(max98088_extmic_enum, + M98088_REG_48_CFG_MIC, 0, + max98088_extmic_text); static const struct snd_kcontrol_new max98088_extmic_mux = SOC_DAPM_ENUM("External MIC Mux", max98088_extmic_enum); @@ -626,12 +625,12 @@ static const struct snd_kcontrol_new max98088_extmic_mux = static const char *max98088_dai1_fltr[] = { "Off", "fc=258/fs=16k", "fc=500/fs=16k", "fc=258/fs=8k", "fc=500/fs=8k", "fc=200"}; -static const struct soc_enum max98088_dai1_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 0, 6, max98088_dai1_fltr), -}; -static const struct soc_enum max98088_dai1_adc_filter_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 4, 6, max98088_dai1_fltr), -}; +static SOC_ENUM_SINGLE_DECL(max98088_dai1_dac_filter_enum, + M98088_REG_18_DAI1_FILTERS, 0, + max98088_dai1_fltr); +static SOC_ENUM_SINGLE_DECL(max98088_dai1_adc_filter_enum, + M98088_REG_18_DAI1_FILTERS, 4, + max98088_dai1_fltr); static int max98088_mic1pre_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From af1f0a50823a3eb8bb7a11731c02b77d145fff70 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:15:26 +0100 Subject: ASoC: max98095: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 50 ++++++++++++++++++++++++--------------------- 1 file changed, 27 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 3ba1170..ddbb416 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -560,25 +560,27 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, } static const char * const max98095_fltr_mode[] = { "Voice", "Music" }; -static const struct soc_enum max98095_dai1_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 7, 2, max98095_fltr_mode), -}; -static const struct soc_enum max98095_dai2_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 7, 2, max98095_fltr_mode), -}; +static SOC_ENUM_SINGLE_DECL(max98095_dai1_filter_mode_enum, + M98095_02E_DAI1_FILTERS, 7, + max98095_fltr_mode); +static SOC_ENUM_SINGLE_DECL(max98095_dai2_filter_mode_enum, + M98095_038_DAI2_FILTERS, 7, + max98095_fltr_mode); static const char * const max98095_extmic_text[] = { "None", "MIC1", "MIC2" }; -static const struct soc_enum max98095_extmic_enum = - SOC_ENUM_SINGLE(M98095_087_CFG_MIC, 0, 3, max98095_extmic_text); +static SOC_ENUM_SINGLE_DECL(max98095_extmic_enum, + M98095_087_CFG_MIC, 0, + max98095_extmic_text); static const struct snd_kcontrol_new max98095_extmic_mux = SOC_DAPM_ENUM("External MIC Mux", max98095_extmic_enum); static const char * const max98095_linein_text[] = { "INA", "INB" }; -static const struct soc_enum max98095_linein_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 6, 2, max98095_linein_text); +static SOC_ENUM_SINGLE_DECL(max98095_linein_enum, + M98095_086_CFG_LINE, 6, + max98095_linein_text); static const struct snd_kcontrol_new max98095_linein_mux = SOC_DAPM_ENUM("Linein Input Mux", max98095_linein_enum); @@ -586,24 +588,26 @@ static const struct snd_kcontrol_new max98095_linein_mux = static const char * const max98095_line_mode_text[] = { "Stereo", "Differential"}; -static const struct soc_enum max98095_linein_mode_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 7, 2, max98095_line_mode_text); +static SOC_ENUM_SINGLE_DECL(max98095_linein_mode_enum, + M98095_086_CFG_LINE, 7, + max98095_line_mode_text); -static const struct soc_enum max98095_lineout_mode_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 4, 2, max98095_line_mode_text); +static SOC_ENUM_SINGLE_DECL(max98095_lineout_mode_enum, + M98095_086_CFG_LINE, 4, + max98095_line_mode_text); static const char * const max98095_dai_fltr[] = { "Off", "Elliptical-HPF-16k", "Butterworth-HPF-16k", "Elliptical-HPF-8k", "Butterworth-HPF-8k", "Butterworth-HPF-Fs/240"}; -static const struct soc_enum max98095_dai1_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 0, 6, max98095_dai_fltr), -}; -static const struct soc_enum max98095_dai2_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 0, 6, max98095_dai_fltr), -}; -static const struct soc_enum max98095_dai3_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_042_DAI3_FILTERS, 0, 6, max98095_dai_fltr), -}; +static SOC_ENUM_SINGLE_DECL(max98095_dai1_dac_filter_enum, + M98095_02E_DAI1_FILTERS, 0, + max98095_dai_fltr); +static SOC_ENUM_SINGLE_DECL(max98095_dai2_dac_filter_enum, + M98095_038_DAI2_FILTERS, 0, + max98095_dai_fltr); +static SOC_ENUM_SINGLE_DECL(max98095_dai3_dac_filter_enum, + M98095_042_DAI3_FILTERS, 0, + max98095_dai_fltr); static int max98095_mic1pre_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From 1e42c3e426b3f7bc61ba338dd6507a293108117c Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 20 Feb 2014 21:48:42 +0000 Subject: ASoC: Intel: Add support for Haswell/Broadwell DSP Add support for low level differentiation functions for Haswell and Broadwell SST DSPs. This includes suppoprt for DSP boot and reset, DSP firmware module parsing and DSP memory block map initialisation. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.c | 60 +++++ sound/soc/intel/sst-dsp.h | 10 + sound/soc/intel/sst-haswell-dsp.c | 517 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 587 insertions(+) create mode 100644 sound/soc/intel/sst-haswell-dsp.c (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 6e22c12..0c129fd 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -27,6 +27,66 @@ #define CREATE_TRACE_POINTS #include +/* Internal generic low-level SST IO functions - can be overidden */ +void sst_shim32_write(void __iomem *addr, u32 offset, u32 value) +{ + writel(value, addr + offset); +} +EXPORT_SYMBOL_GPL(sst_shim32_write); + +u32 sst_shim32_read(void __iomem *addr, u32 offset) +{ + return readl(addr + offset); +} +EXPORT_SYMBOL_GPL(sst_shim32_read); + +void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) +{ + memcpy_toio(addr + offset, &value, sizeof(value)); +} +EXPORT_SYMBOL_GPL(sst_shim32_write64); + +u64 sst_shim32_read64(void __iomem *addr, u32 offset) +{ + u64 val; + + memcpy_fromio(&val, addr + offset, sizeof(val)); + return val; +} +EXPORT_SYMBOL_GPL(sst_shim32_read64); + +static inline void _sst_memcpy_toio_32(volatile u32 __iomem *dest, + u32 *src, size_t bytes) +{ + int i, words = bytes >> 2; + + for (i = 0; i < words; i++) + writel(src[i], dest + i); +} + +static inline void _sst_memcpy_fromio_32(u32 *dest, + const volatile __iomem u32 *src, size_t bytes) +{ + int i, words = bytes >> 2; + + for (i = 0; i < words; i++) + dest[i] = readl(src + i); +} + +void sst_memcpy_toio_32(struct sst_dsp *sst, + void __iomem *dest, void *src, size_t bytes) +{ + _sst_memcpy_toio_32(dest, src, bytes); +} +EXPORT_SYMBOL_GPL(sst_memcpy_toio_32); + +void sst_memcpy_fromio_32(struct sst_dsp *sst, void *dest, + void __iomem *src, size_t bytes) +{ + _sst_memcpy_fromio_32(dest, src, bytes); +} +EXPORT_SYMBOL_GPL(sst_memcpy_fromio_32); + /* Public API */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) { diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 3730fd3..608418c 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -189,6 +189,16 @@ u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset); int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, u64 mask, u64 value); +/* Internal generic low-level SST IO functions - can be overidden */ +void sst_shim32_write(void __iomem *addr, u32 offset, u32 value); +u32 sst_shim32_read(void __iomem *addr, u32 offset); +void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value); +u64 sst_shim32_read64(void __iomem *addr, u32 offset); +void sst_memcpy_toio_32(struct sst_dsp *sst, + void __iomem *dest, void *src, size_t bytes); +void sst_memcpy_fromio_32(struct sst_dsp *sst, + void *dest, void __iomem *src, size_t bytes); + /* DSP reset & boot */ void sst_dsp_reset(struct sst_dsp *sst); int sst_dsp_boot(struct sst_dsp *sst); diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c new file mode 100644 index 0000000..12f7317 --- /dev/null +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -0,0 +1,517 @@ +/* + * Intel Haswell SST DSP driver + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" +#include "sst-haswell-ipc.h" + +#include + +#define SST_HSW_FW_SIGNATURE_SIZE 4 +#define SST_HSW_FW_SIGN "$SST" +#define SST_HSW_FW_LIB_SIGN "$LIB" + +#define SST_WPT_SHIM_OFFSET 0xFB000 +#define SST_LP_SHIM_OFFSET 0xE7000 +#define SST_WPT_IRAM_OFFSET 0xA0000 +#define SST_LP_IRAM_OFFSET 0x80000 + +#define SST_SHIM_PM_REG 0x84 + +#define SST_HSW_IRAM 1 +#define SST_HSW_DRAM 2 +#define SST_HSW_REGS 3 + +struct dma_block_info { + __le32 type; /* IRAM/DRAM */ + __le32 size; /* Bytes */ + __le32 ram_offset; /* Offset in I/DRAM */ + __le32 rsvd; /* Reserved field */ +} __attribute__((packed)); + +struct fw_module_info { + __le32 persistent_size; + __le32 scratch_size; +} __attribute__((packed)); + +struct fw_header { + unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* FW signature */ + __le32 file_size; /* size of fw minus this header */ + __le32 modules; /* # of modules */ + __le32 file_format; /* version of header format */ + __le32 reserved[4]; +} __attribute__((packed)); + +struct fw_module_header { + unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* module signature */ + __le32 mod_size; /* size of module */ + __le32 blocks; /* # of blocks */ + __le16 padding; + __le16 type; /* codec type, pp lib */ + __le32 entry_point; + struct fw_module_info info; +} __attribute__((packed)); + +static void hsw_free(struct sst_dsp *sst); + +static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, + struct fw_module_header *module) +{ + struct dma_block_info *block; + struct sst_module *mod; + struct sst_module_data block_data; + struct sst_module_template template; + int count; + void __iomem *ram; + + /* TODO: allowed module types need to be configurable */ + if (module->type != SST_HSW_MODULE_BASE_FW + && module->type != SST_HSW_MODULE_PCM_SYSTEM + && module->type != SST_HSW_MODULE_PCM + && module->type != SST_HSW_MODULE_PCM_REFERENCE + && module->type != SST_HSW_MODULE_PCM_CAPTURE + && module->type != SST_HSW_MODULE_LPAL) + return 0; + + dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n", + module->signature, module->mod_size, + module->blocks, module->type); + dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point); + dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n", + module->info.persistent_size, module->info.scratch_size); + + memset(&template, 0, sizeof(template)); + template.id = module->type; + template.entry = module->entry_point; + template.p.size = module->info.persistent_size; + template.p.type = SST_MEM_DRAM; + template.p.data_type = SST_DATA_P; + template.s.size = module->info.scratch_size; + template.s.type = SST_MEM_DRAM; + template.s.data_type = SST_DATA_S; + + mod = sst_module_new(fw, &template, NULL); + if (mod == NULL) + return -ENOMEM; + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + + if (block->size <= 0) { + dev_err(dsp->dev, + "error: block %d size invalid\n", count); + sst_module_free(mod); + return -EINVAL; + } + + switch (block->type) { + case SST_HSW_IRAM: + ram = dsp->addr.lpe; + block_data.offset = + block->ram_offset + dsp->addr.iram_offset; + block_data.type = SST_MEM_IRAM; + break; + case SST_HSW_DRAM: + ram = dsp->addr.lpe; + block_data.offset = block->ram_offset; + block_data.type = SST_MEM_DRAM; + break; + default: + dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", + block->type, count); + sst_module_free(mod); + return -EINVAL; + } + + block_data.size = block->size; + block_data.data_type = SST_DATA_M; + block_data.data = (void *)block + sizeof(*block); + block_data.data_offset = block_data.data - fw->dma_buf; + + dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + "size 0x%x ==> ram %p offset 0x%x\n", + count, block->type, block->size, ram, + block->ram_offset); + + sst_module_insert_fixed_block(mod, &block_data); + + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +static int hsw_parse_fw_image(struct sst_fw *sst_fw) +{ + struct fw_header *header; + struct sst_module *scratch; + struct fw_module_header *module; + struct sst_dsp *dsp = sst_fw->dsp; + struct sst_hsw *hsw = sst_fw->private; + int ret, count; + + /* Read the header information from the data pointer */ + header = (struct fw_header *)sst_fw->dma_buf; + + /* verify FW */ + if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) || + (sst_fw->size != header->file_size + sizeof(*header))) { + dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n"); + return -EINVAL; + } + + dev_dbg(dsp->dev, "header size=0x%x modules=0x%x fmt=0x%x size=%zu\n", + header->file_size, header->modules, + header->file_format, sizeof(*header)); + + /* parse each module */ + module = (void *)sst_fw->dma_buf + sizeof(*header); + for (count = 0; count < header->modules; count++) { + + /* module */ + ret = hsw_parse_module(dsp, sst_fw, module); + if (ret < 0) { + dev_err(dsp->dev, "error: invalid module %d\n", count); + return ret; + } + module = (void *)module + sizeof(*module) + module->mod_size; + } + + /* allocate persistent/scratch mem regions */ + scratch = sst_mem_block_alloc_scratch(dsp); + if (scratch == NULL) + return -ENOMEM; + + sst_hsw_set_scratch_module(hsw, scratch); + + return 0; +} + +static irqreturn_t hsw_irq(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + u32 isr; + int ret = IRQ_NONE; + + spin_lock(&sst->spinlock); + + /* Interrupt arrived, check src */ + isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX); + if (isr & SST_ISRX_DONE) { + trace_sst_irq_done(isr, + sst_dsp_shim_read_unlocked(sst, SST_IMRX)); + + /* Mask Done interrupt before return */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_DONE, SST_IMRX_DONE); + ret = IRQ_WAKE_THREAD; + } + + if (isr & SST_ISRX_BUSY) { + trace_sst_irq_busy(isr, + sst_dsp_shim_read_unlocked(sst, SST_IMRX)); + + /* Mask Busy interrupt before return */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_BUSY, SST_IMRX_BUSY); + ret = IRQ_WAKE_THREAD; + } + + spin_unlock(&sst->spinlock); + return ret; +} + +static void hsw_boot(struct sst_dsp *sst) +{ + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + + /* stall DSP core, set clk to 192/96Mhz */ + sst_dsp_shim_update_bits_unlocked(sst, + SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, + SST_CSR_STALL | SST_CSR_DCS(4)); + + /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + + /* disable DMA finish function for SSP0 & SSP1 */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, + SST_CSR2_SDFD_SSP1); + + /* enable DMA engine 0,1 all channels to access host memory */ + sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC, + SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff), + SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff)); + + /* disable all clock gating */ + writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* set DSP to RUN */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0); +} + +static void hsw_reset(struct sst_dsp *sst) +{ + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL); + + /* keep in reset for 10ms */ + mdelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +} + +struct sst_adsp_memregion { + u32 start; + u32 end; + int blocks; + enum sst_mem_type type; +}; + +/* lynx point ADSP mem regions */ +static const struct sst_adsp_memregion lp_region[] = { + {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ + {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ + {0x80000, 0xE0000, 12, SST_MEM_IRAM}, /* I-SRAM - 12 * 32kB */ +}; + +/* wild cat point ADSP mem regions */ +static const struct sst_adsp_memregion wpt_region[] = { + {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ + {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ + {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */ + {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */ +}; + +static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + /* ADSP DRAM & IRAM */ + sst->addr.lpe_base = pdata->lpe_base; + sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); + if (!sst->addr.lpe) + return -ENODEV; + + /* ADSP PCI MMIO config space */ + sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size); + if (!sst->addr.pci_cfg) { + iounmap(sst->addr.lpe); + return -ENODEV; + } + + /* SST Shim */ + sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset; + return 0; +} + +static u32 hsw_block_get_bit(struct sst_mem_block *block) +{ + u32 bit = 0, shift = 0; + + switch (block->type) { + case SST_MEM_DRAM: + shift = 16; + break; + case SST_MEM_IRAM: + shift = 6; + break; + default: + return 0; + } + + bit = 1 << (block->index + shift); + + return bit; +} + +/* enable 32kB memory block - locks held by caller */ +static int hsw_block_enable(struct sst_mem_block *block) +{ + struct sst_dsp *sst = block->dsp; + u32 bit, val; + + if (block->users++ > 0) + return 0; + + dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + bit = hsw_block_get_bit(block); + writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* wait 18 DSP clock ticks */ + udelay(10); + + return 0; +} + +/* disable 32kB memory block - locks held by caller */ +static int hsw_block_disable(struct sst_mem_block *block) +{ + struct sst_dsp *sst = block->dsp; + u32 bit, val; + + if (--block->users > 0) + return 0; + + dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + bit = hsw_block_get_bit(block); + writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + + return 0; +} + +static struct sst_block_ops sst_hsw_ops = { + .enable = hsw_block_enable, + .disable = hsw_block_disable, +}; + +static int hsw_enable_shim(struct sst_dsp *sst) +{ + int tries = 10; + u32 reg; + + /* enable shim */ + reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG); + writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG); + + /* check that ADSP shim is enabled */ + while (tries--) { + reg = sst_dsp_shim_read_unlocked(sst, SST_CSR); + if (reg != 0xffffffff) + return 0; + + msleep(1); + } + + return -ENODEV; +} + +static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + const struct sst_adsp_memregion *region; + struct device *dev; + int ret = -ENODEV, i, j, region_count; + u32 offset, size; + + dev = sst->dev; + + switch (sst->id) { + case SST_DEV_ID_LYNX_POINT: + region = lp_region; + region_count = ARRAY_SIZE(lp_region); + sst->addr.iram_offset = SST_LP_IRAM_OFFSET; + sst->addr.shim_offset = SST_LP_SHIM_OFFSET; + break; + case SST_DEV_ID_WILDCAT_POINT: + region = wpt_region; + region_count = ARRAY_SIZE(wpt_region); + sst->addr.iram_offset = SST_WPT_IRAM_OFFSET; + sst->addr.shim_offset = SST_WPT_SHIM_OFFSET; + break; + default: + dev_err(dev, "error: failed to get mem resources\n"); + return ret; + } + + ret = hsw_acpi_resource_map(sst, pdata); + if (ret < 0) { + dev_err(dev, "error: failed to map resources\n"); + return ret; + } + + /* enable the DSP SHIM */ + ret = hsw_enable_shim(sst); + if (ret < 0) { + dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n"); + return ret; + } + + ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + /* Enable Interrupt from both sides */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0); + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD, + (0x3 | 0x1 << 16 | 0x3 << 21), 0x0); + + /* register DSP memory blocks - ideally we should get this from ACPI */ + for (i = 0; i < region_count; i++) { + offset = region[i].start; + size = (region[i].end - region[i].start) / region[i].blocks; + + /* register individual memory blocks */ + for (j = 0; j < region[i].blocks; j++) { + sst_mem_block_register(sst, offset, size, + region[i].type, &sst_hsw_ops, j, sst); + offset += size; + } + } + + /* set default power gating mask */ + writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0); + + return 0; +} + +static void hsw_free(struct sst_dsp *sst) +{ + sst_mem_block_unregister_all(sst); + iounmap(sst->addr.lpe); + iounmap(sst->addr.pci_cfg); +} + +struct sst_ops haswell_ops = { + .reset = hsw_reset, + .boot = hsw_boot, + .write = sst_shim32_write, + .read = sst_shim32_read, + .write64 = sst_shim32_write64, + .read64 = sst_shim32_read64, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .irq_handler = hsw_irq, + .init = hsw_init, + .free = hsw_free, + .parse_fw = hsw_parse_fw_image, +}; -- cgit v1.1 From 22981243589c3468481f5cfd04176233d2c7be4b Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 20 Feb 2014 21:48:43 +0000 Subject: ASoC: Intel: Add Haswell/Broadwell IPC Add support for Haswell and Broadwell DSP IPC. This is used by the DSP platform PCM driver to configure the DSP for audio operations. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 1785 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 488 ++++++++++ 2 files changed, 2273 insertions(+) create mode 100644 sound/soc/intel/sst-haswell-ipc.c create mode 100644 sound/soc/intel/sst-haswell-ipc.h (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c new file mode 100644 index 0000000..668d486 --- /dev/null +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -0,0 +1,1785 @@ +/* + * Intel SST Haswell/Broadwell IPC Support + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sst-haswell-ipc.h" +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +/* Global Message - Generic */ +#define IPC_GLB_TYPE_SHIFT 24 +#define IPC_GLB_TYPE_MASK (0x1f << IPC_GLB_TYPE_SHIFT) +#define IPC_GLB_TYPE(x) (x << IPC_GLB_TYPE_SHIFT) + +/* Global Message - Reply */ +#define IPC_GLB_REPLY_SHIFT 0 +#define IPC_GLB_REPLY_MASK (0x1f << IPC_GLB_REPLY_SHIFT) +#define IPC_GLB_REPLY_TYPE(x) (x << IPC_GLB_REPLY_TYPE_SHIFT) + +/* Stream Message - Generic */ +#define IPC_STR_TYPE_SHIFT 20 +#define IPC_STR_TYPE_MASK (0xf << IPC_STR_TYPE_SHIFT) +#define IPC_STR_TYPE(x) (x << IPC_STR_TYPE_SHIFT) +#define IPC_STR_ID_SHIFT 16 +#define IPC_STR_ID_MASK (0xf << IPC_STR_ID_SHIFT) +#define IPC_STR_ID(x) (x << IPC_STR_ID_SHIFT) + +/* Stream Message - Reply */ +#define IPC_STR_REPLY_SHIFT 0 +#define IPC_STR_REPLY_MASK (0x1f << IPC_STR_REPLY_SHIFT) + +/* Stream Stage Message - Generic */ +#define IPC_STG_TYPE_SHIFT 12 +#define IPC_STG_TYPE_MASK (0xf << IPC_STG_TYPE_SHIFT) +#define IPC_STG_TYPE(x) (x << IPC_STG_TYPE_SHIFT) +#define IPC_STG_ID_SHIFT 10 +#define IPC_STG_ID_MASK (0x3 << IPC_STG_ID_SHIFT) +#define IPC_STG_ID(x) (x << IPC_STG_ID_SHIFT) + +/* Stream Stage Message - Reply */ +#define IPC_STG_REPLY_SHIFT 0 +#define IPC_STG_REPLY_MASK (0x1f << IPC_STG_REPLY_SHIFT) + +/* Debug Log Message - Generic */ +#define IPC_LOG_OP_SHIFT 20 +#define IPC_LOG_OP_MASK (0xf << IPC_LOG_OP_SHIFT) +#define IPC_LOG_OP_TYPE(x) (x << IPC_LOG_OP_SHIFT) +#define IPC_LOG_ID_SHIFT 16 +#define IPC_LOG_ID_MASK (0xf << IPC_LOG_ID_SHIFT) +#define IPC_LOG_ID(x) (x << IPC_LOG_ID_SHIFT) + +/* IPC message timeout (msecs) */ +#define IPC_TIMEOUT_MSECS 300 +#define IPC_BOOT_MSECS 200 +#define IPC_MSG_WAIT 0 +#define IPC_MSG_NOWAIT 1 + +/* Firmware Ready Message */ +#define IPC_FW_READY (0x1 << 29) +#define IPC_STATUS_MASK (0x3 << 30) + +#define IPC_EMPTY_LIST_SIZE 8 +#define IPC_MAX_STREAMS 4 + +/* Mailbox */ +#define IPC_MAX_MAILBOX_BYTES 256 + +/* Global Message - Types and Replies */ +enum ipc_glb_type { + IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */ + IPC_GLB_PERFORMANCE_MONITOR = 1, /* Performance monitoring actions */ + IPC_GLB_ALLOCATE_STREAM = 3, /* Request to allocate new stream */ + IPC_GLB_FREE_STREAM = 4, /* Request to free stream */ + IPC_GLB_GET_FW_CAPABILITIES = 5, /* Retrieves firmware capabilities */ + IPC_GLB_STREAM_MESSAGE = 6, /* Message directed to stream or its stages */ + /* Request to store firmware context during D0->D3 transition */ + IPC_GLB_REQUEST_DUMP = 7, + /* Request to restore firmware context during D3->D0 transition */ + IPC_GLB_RESTORE_CONTEXT = 8, + IPC_GLB_GET_DEVICE_FORMATS = 9, /* Set device format */ + IPC_GLB_SET_DEVICE_FORMATS = 10, /* Get device format */ + IPC_GLB_SHORT_REPLY = 11, + IPC_GLB_ENTER_DX_STATE = 12, + IPC_GLB_GET_MIXER_STREAM_INFO = 13, /* Request mixer stream params */ + IPC_GLB_DEBUG_LOG_MESSAGE = 14, /* Message to or from the debug logger. */ + IPC_GLB_REQUEST_TRANSFER = 16, /* < Request Transfer for host */ + IPC_GLB_MAX_IPC_MESSAGE_TYPE = 17, /* Maximum message number */ +}; + +enum ipc_glb_reply { + IPC_GLB_REPLY_SUCCESS = 0, /* The operation was successful. */ + IPC_GLB_REPLY_ERROR_INVALID_PARAM = 1, /* Invalid parameter was passed. */ + IPC_GLB_REPLY_UNKNOWN_MESSAGE_TYPE = 2, /* Uknown message type was resceived. */ + IPC_GLB_REPLY_OUT_OF_RESOURCES = 3, /* No resources to satisfy the request. */ + IPC_GLB_REPLY_BUSY = 4, /* The system or resource is busy. */ + IPC_GLB_REPLY_PENDING = 5, /* The action was scheduled for processing. */ + IPC_GLB_REPLY_FAILURE = 6, /* Critical error happened. */ + IPC_GLB_REPLY_INVALID_REQUEST = 7, /* Request can not be completed. */ + IPC_GLB_REPLY_STAGE_UNINITIALIZED = 8, /* Processing stage was uninitialized. */ + IPC_GLB_REPLY_NOT_FOUND = 9, /* Required resource can not be found. */ + IPC_GLB_REPLY_SOURCE_NOT_STARTED = 10, /* Source was not started. */ +}; + +/* Stream Message - Types */ +enum ipc_str_operation { + IPC_STR_RESET = 0, + IPC_STR_PAUSE = 1, + IPC_STR_RESUME = 2, + IPC_STR_STAGE_MESSAGE = 3, + IPC_STR_NOTIFICATION = 4, + IPC_STR_MAX_MESSAGE +}; + +/* Stream Stage Message Types */ +enum ipc_stg_operation { + IPC_STG_GET_VOLUME = 0, + IPC_STG_SET_VOLUME, + IPC_STG_SET_WRITE_POSITION, + IPC_STG_SET_FX_ENABLE, + IPC_STG_SET_FX_DISABLE, + IPC_STG_SET_FX_GET_PARAM, + IPC_STG_SET_FX_SET_PARAM, + IPC_STG_SET_FX_GET_INFO, + IPC_STG_MUTE_LOOPBACK, + IPC_STG_MAX_MESSAGE +}; + +/* Stream Stage Message Types For Notification*/ +enum ipc_stg_operation_notify { + IPC_POSITION_CHANGED = 0, + IPC_STG_GLITCH, + IPC_STG_MAX_NOTIFY +}; + +enum ipc_glitch_type { + IPC_GLITCH_UNDERRUN = 1, + IPC_GLITCH_DECODER_ERROR, + IPC_GLITCH_DOUBLED_WRITE_POS, + IPC_GLITCH_MAX +}; + +/* Debug Control */ +enum ipc_debug_operation { + IPC_DEBUG_ENABLE_LOG = 0, + IPC_DEBUG_DISABLE_LOG = 1, + IPC_DEBUG_REQUEST_LOG_DUMP = 2, + IPC_DEBUG_NOTIFY_LOG_DUMP = 3, + IPC_DEBUG_MAX_DEBUG_LOG +}; + +/* Firmware Ready */ +struct sst_hsw_ipc_fw_ready { + u32 inbox_offset; + u32 outbox_offset; + u32 inbox_size; + u32 outbox_size; + u32 fw_info_size; + u8 fw_info[1]; +} __attribute__((packed)); + +struct ipc_message { + struct list_head list; + u32 header; + + /* direction wrt host CPU */ + char tx_data[IPC_MAX_MAILBOX_BYTES]; + size_t tx_size; + char rx_data[IPC_MAX_MAILBOX_BYTES]; + size_t rx_size; + + wait_queue_head_t waitq; + bool pending; + bool complete; + bool wait; + int errno; +}; + +struct sst_hsw_stream; +struct sst_hsw; + +/* Stream infomation */ +struct sst_hsw_stream { + /* configuration */ + struct sst_hsw_ipc_stream_alloc_req request; + struct sst_hsw_ipc_stream_alloc_reply reply; + struct sst_hsw_ipc_stream_free_req free_req; + + /* Mixer info */ + u32 mute_volume[SST_HSW_NO_CHANNELS]; + u32 mute[SST_HSW_NO_CHANNELS]; + + /* runtime info */ + struct sst_hsw *hsw; + int host_id; + bool commited; + bool running; + + /* Notification work */ + struct work_struct notify_work; + u32 header; + + /* Position info from DSP */ + struct sst_hsw_ipc_stream_set_position wpos; + struct sst_hsw_ipc_stream_get_position rpos; + struct sst_hsw_ipc_stream_glitch_position glitch; + + /* Volume info */ + struct sst_hsw_ipc_volume_req vol_req; + + /* driver callback */ + u32 (*notify_position)(struct sst_hsw_stream *stream, void *data); + void *pdata; + + struct list_head node; +}; + +/* FW log ring information */ +struct sst_hsw_log_stream { + dma_addr_t dma_addr; + unsigned char *dma_area; + unsigned char *ring_descr; + int pages; + int size; + + /* Notification work */ + struct work_struct notify_work; + wait_queue_head_t readers_wait_q; + struct mutex rw_mutex; + + u32 last_pos; + u32 curr_pos; + u32 reader_pos; + + /* fw log config */ + u32 config[SST_HSW_FW_LOG_CONFIG_DWORDS]; + + struct sst_hsw *hsw; +}; + +/* SST Haswell IPC data */ +struct sst_hsw { + struct device *dev; + struct sst_dsp *dsp; + struct platform_device *pdev_pcm; + + /* FW config */ + struct sst_hsw_ipc_fw_ready fw_ready; + struct sst_hsw_ipc_fw_version version; + struct sst_module *scratch; + bool fw_done; + + /* stream */ + struct list_head stream_list; + + /* global mixer */ + struct sst_hsw_ipc_stream_info_reply mixer_info; + enum sst_hsw_volume_curve curve_type; + u32 curve_duration; + u32 mute[SST_HSW_NO_CHANNELS]; + u32 mute_volume[SST_HSW_NO_CHANNELS]; + + /* DX */ + struct sst_hsw_ipc_dx_reply dx; + + /* boot */ + wait_queue_head_t boot_wait; + bool boot_complete; + bool shutdown; + + /* IPC messaging */ + struct list_head tx_list; + struct list_head rx_list; + struct list_head empty_list; + wait_queue_head_t wait_txq; + struct task_struct *tx_thread; + struct kthread_worker kworker; + struct kthread_work kwork; + bool pending; + struct ipc_message *msg; + + /* FW log stream */ + struct sst_hsw_log_stream log_stream; +}; + +#define CREATE_TRACE_POINTS +#include + +static inline u32 msg_get_global_type(u32 msg) +{ + return (msg & IPC_GLB_TYPE_MASK) >> IPC_GLB_TYPE_SHIFT; +} + +static inline u32 msg_get_global_reply(u32 msg) +{ + return (msg & IPC_GLB_REPLY_MASK) >> IPC_GLB_REPLY_SHIFT; +} + +static inline u32 msg_get_stream_type(u32 msg) +{ + return (msg & IPC_STR_TYPE_MASK) >> IPC_STR_TYPE_SHIFT; +} + +static inline u32 msg_get_stage_type(u32 msg) +{ + return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; +} + +static inline u32 msg_set_stage_type(u32 msg, u32 type) +{ + return (msg & ~IPC_STG_TYPE_MASK) + + (type << IPC_STG_TYPE_SHIFT); +} + +static inline u32 msg_get_stream_id(u32 msg) +{ + return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; +} + +static inline u32 msg_get_notify_reason(u32 msg) +{ + return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; +} + +u32 create_channel_map(enum sst_hsw_channel_config config) +{ + switch (config) { + case SST_HSW_CHANNEL_CONFIG_MONO: + return (0xFFFFFFF0 | SST_HSW_CHANNEL_CENTER); + case SST_HSW_CHANNEL_CONFIG_STEREO: + return (0xFFFFFF00 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_RIGHT << 4)); + case SST_HSW_CHANNEL_CONFIG_2_POINT_1: + return (0xFFFFF000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_RIGHT << 4) + | (SST_HSW_CHANNEL_LFE << 8 )); + case SST_HSW_CHANNEL_CONFIG_3_POINT_0: + return (0xFFFFF000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_CENTER << 4) + | (SST_HSW_CHANNEL_RIGHT << 8)); + case SST_HSW_CHANNEL_CONFIG_3_POINT_1: + return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_CENTER << 4) + | (SST_HSW_CHANNEL_RIGHT << 8) + | (SST_HSW_CHANNEL_LFE << 12)); + case SST_HSW_CHANNEL_CONFIG_QUATRO: + return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_RIGHT << 4) + | (SST_HSW_CHANNEL_LEFT_SURROUND << 8) + | (SST_HSW_CHANNEL_RIGHT_SURROUND << 12)); + case SST_HSW_CHANNEL_CONFIG_4_POINT_0: + return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_CENTER << 4) + | (SST_HSW_CHANNEL_RIGHT << 8) + | (SST_HSW_CHANNEL_CENTER_SURROUND << 12)); + case SST_HSW_CHANNEL_CONFIG_5_POINT_0: + return (0xFFF00000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_CENTER << 4) + | (SST_HSW_CHANNEL_RIGHT << 8) + | (SST_HSW_CHANNEL_LEFT_SURROUND << 12) + | (SST_HSW_CHANNEL_RIGHT_SURROUND << 16)); + case SST_HSW_CHANNEL_CONFIG_5_POINT_1: + return (0xFF000000 | SST_HSW_CHANNEL_CENTER + | (SST_HSW_CHANNEL_LEFT << 4) + | (SST_HSW_CHANNEL_RIGHT << 8) + | (SST_HSW_CHANNEL_LEFT_SURROUND << 12) + | (SST_HSW_CHANNEL_RIGHT_SURROUND << 16) + | (SST_HSW_CHANNEL_LFE << 20)); + case SST_HSW_CHANNEL_CONFIG_DUAL_MONO: + return (0xFFFFFF00 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_LEFT << 4)); + default: + return 0xFFFFFFFF; + } +} + +static struct sst_hsw_stream *get_stream_by_id(struct sst_hsw *hsw, + int stream_id) +{ + struct sst_hsw_stream *stream; + + list_for_each_entry(stream, &hsw->stream_list, node) { + if (stream->reply.stream_hw_id == stream_id) + return stream; + } + + return NULL; +} + +static void ipc_shim_dbg(struct sst_hsw *hsw, const char *text) +{ + struct sst_dsp *sst = hsw->dsp; + u32 isr, ipcd, imrx, ipcx; + + ipcx = sst_dsp_shim_read_unlocked(sst, SST_IPCX); + isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX); + ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD); + imrx = sst_dsp_shim_read_unlocked(sst, SST_IMRX); + + dev_err(hsw->dev, "ipc: --%s-- ipcx 0x%8.8x isr 0x%8.8x ipcd 0x%8.8x imrx 0x%8.8x\n", + text, ipcx, isr, ipcd, imrx); +} + +/* locks held by caller */ +static struct ipc_message *msg_get_empty(struct sst_hsw *hsw) +{ + struct ipc_message *msg = NULL; + + if (!list_empty(&hsw->empty_list)) { + msg = list_first_entry(&hsw->empty_list, struct ipc_message, + list); + list_del(&msg->list); + } + + return msg; +} + +static void ipc_tx_msgs(struct kthread_work *work) +{ + struct sst_hsw *hsw = + container_of(work, struct sst_hsw, kwork); + struct ipc_message *msg; + unsigned long flags; + u32 ipcx; + + spin_lock_irqsave(&hsw->dsp->spinlock, flags); + + if (list_empty(&hsw->tx_list) || hsw->pending) { + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + return; + } + + /* if the DSP is busy we will TX messages after IRQ */ + ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX); + if (ipcx & SST_IPCX_BUSY) { + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + return; + } + + msg = list_first_entry(&hsw->tx_list, struct ipc_message, list); + + list_move(&msg->list, &hsw->rx_list); + + /* send the message */ + sst_dsp_outbox_write(hsw->dsp, msg->tx_data, msg->tx_size); + sst_dsp_ipc_msg_tx(hsw->dsp, msg->header | SST_IPCX_BUSY); + + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); +} + +/* locks held by caller */ +static void tx_msg_reply_complete(struct sst_hsw *hsw, struct ipc_message *msg) +{ + msg->complete = true; + trace_ipc_reply("completed", msg->header); + + if (!msg->wait) + list_add_tail(&msg->list, &hsw->empty_list); + else + wake_up(&msg->waitq); +} + +static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg, + void *rx_data) +{ + unsigned long flags; + int ret; + + /* wait for DSP completion (in all cases atm inc pending) */ + ret = wait_event_timeout(msg->waitq, msg->complete, + msecs_to_jiffies(IPC_TIMEOUT_MSECS)); + + spin_lock_irqsave(&hsw->dsp->spinlock, flags); + if (ret == 0) { + ipc_shim_dbg(hsw, "message timeout"); + + trace_ipc_error("error message timeout for", msg->header); + ret = -ETIMEDOUT; + } else { + + /* copy the data returned from DSP */ + if (msg->rx_size) + memcpy(rx_data, msg->rx_data, msg->rx_size); + ret = msg->errno; + } + + list_add_tail(&msg->list, &hsw->empty_list); + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + return ret; +} + +static int ipc_tx_message(struct sst_hsw *hsw, u32 header, void *tx_data, + size_t tx_bytes, void *rx_data, size_t rx_bytes, int wait) +{ + struct ipc_message *msg; + unsigned long flags; + + spin_lock_irqsave(&hsw->dsp->spinlock, flags); + + msg = msg_get_empty(hsw); + if (msg == NULL) { + spin_unlock(&hsw->dsp->spinlock); + return -EBUSY; + } + + if (tx_bytes) + memcpy(msg->tx_data, tx_data, tx_bytes); + + msg->header = header; + msg->tx_size = tx_bytes; + msg->rx_size = rx_bytes; + msg->wait = wait; + msg->errno = 0; + msg->pending = false; + msg->complete = false; + + list_add_tail(&msg->list, &hsw->tx_list); + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + + queue_kthread_work(&hsw->kworker, &hsw->kwork); + + if (wait) + return tx_wait_done(hsw, msg, rx_data); + else + return 0; +} + +static inline int ipc_tx_message_wait(struct sst_hsw *hsw, u32 header, + void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes) +{ + return ipc_tx_message(hsw, header, tx_data, tx_bytes, rx_data, + rx_bytes, 1); +} + +static inline int ipc_tx_message_nowait(struct sst_hsw *hsw, u32 header, + void *tx_data, size_t tx_bytes) +{ + return ipc_tx_message(hsw, header, tx_data, tx_bytes, NULL, 0, 0); +} + +static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) +{ + struct sst_hsw_ipc_fw_ready fw_ready; + u32 offset; + + offset = (header & 0x1FFFFFFF) << 3; + + dev_dbg(hsw->dev, "ipc: DSP is ready 0x%8.8x offset %d\n", + header, offset); + + /* copy data from the DSP FW ready offset */ + sst_dsp_read(hsw->dsp, &fw_ready, offset, sizeof(fw_ready)); + + sst_dsp_mailbox_init(hsw->dsp, fw_ready.inbox_offset, + fw_ready.inbox_size, fw_ready.outbox_offset, + fw_ready.outbox_size); + + hsw->boot_complete = true; + wake_up(&hsw->boot_wait); + + dev_dbg(hsw->dev, " mailbox upstream 0x%x - size 0x%x\n", + fw_ready.inbox_offset, fw_ready.inbox_size); + dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n", + fw_ready.outbox_offset, fw_ready.outbox_size); +} + +static void hsw_notification_work(struct work_struct *work) +{ + struct sst_hsw_stream *stream = container_of(work, + struct sst_hsw_stream, notify_work); + struct sst_hsw_ipc_stream_glitch_position *glitch = &stream->glitch; + struct sst_hsw_ipc_stream_get_position *pos = &stream->rpos; + struct sst_hsw *hsw = stream->hsw; + u32 reason; + + reason = msg_get_notify_reason(stream->header); + + switch (reason) { + case IPC_STG_GLITCH: + trace_ipc_notification("DSP stream under/overrun", + stream->reply.stream_hw_id); + sst_dsp_inbox_read(hsw->dsp, glitch, sizeof(*glitch)); + + dev_err(hsw->dev, "glitch %d pos 0x%x write pos 0x%x\n", + glitch->glitch_type, glitch->present_pos, + glitch->write_pos); + break; + + case IPC_POSITION_CHANGED: + trace_ipc_notification("DSP stream position changed for", + stream->reply.stream_hw_id); + sst_dsp_inbox_read(hsw->dsp, pos, sizeof(&pos)); + + if (stream->notify_position) + stream->notify_position(stream, stream->pdata); + + break; + default: + dev_err(hsw->dev, "error: unknown notification 0x%x\n", + stream->header); + break; + } + + /* tell DSP that notification has been handled */ + sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD, + SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); + + /* unmask busy interrupt */ + sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); +} + +static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) +{ + struct ipc_message *msg; + + /* clear reply bits & status bits */ + header &= ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK); + + if (list_empty(&hsw->rx_list)) { + dev_err(hsw->dev, "error: rx list empty but received 0x%x\n", + header); + return NULL; + } + + list_for_each_entry(msg, &hsw->rx_list, list) { + if (msg->header == header) + return msg; + } + + return NULL; +} + +static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg) +{ + struct sst_hsw_stream *stream; + u32 header = msg->header & ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK); + u32 stream_id = msg_get_stream_id(header); + u32 stream_msg = msg_get_stream_type(header); + + stream = get_stream_by_id(hsw, stream_id); + if (stream == NULL) + return; + + switch (stream_msg) { + case IPC_STR_STAGE_MESSAGE: + case IPC_STR_NOTIFICATION: + case IPC_STR_RESET: + break; + case IPC_STR_PAUSE: + stream->running = false; + trace_ipc_notification("stream paused", + stream->reply.stream_hw_id); + break; + case IPC_STR_RESUME: + stream->running = true; + trace_ipc_notification("stream running", + stream->reply.stream_hw_id); + break; + } +} + +static int hsw_process_reply(struct sst_hsw *hsw, u32 header) +{ + struct ipc_message *msg; + u32 reply = msg_get_global_reply(header); + + trace_ipc_reply("processing -->", header); + + msg = reply_find_msg(hsw, header); + if (msg == NULL) { + trace_ipc_error("error: can't find message header", header); + return -EIO; + } + + /* first process the header */ + switch (reply) { + case IPC_GLB_REPLY_PENDING: + trace_ipc_pending_reply("received", header); + msg->pending = true; + hsw->pending = true; + return 1; + case IPC_GLB_REPLY_SUCCESS: + if (msg->pending) { + trace_ipc_pending_reply("completed", header); + sst_dsp_inbox_read(hsw->dsp, msg->rx_data, + msg->rx_size); + hsw->pending = false; + } else { + /* copy data from the DSP */ + sst_dsp_outbox_read(hsw->dsp, msg->rx_data, + msg->rx_size); + } + break; + /* these will be rare - but useful for debug */ + case IPC_GLB_REPLY_UNKNOWN_MESSAGE_TYPE: + trace_ipc_error("error: unknown message type", header); + msg->errno = -EBADMSG; + break; + case IPC_GLB_REPLY_OUT_OF_RESOURCES: + trace_ipc_error("error: out of resources", header); + msg->errno = -ENOMEM; + break; + case IPC_GLB_REPLY_BUSY: + trace_ipc_error("error: reply busy", header); + msg->errno = -EBUSY; + break; + case IPC_GLB_REPLY_FAILURE: + trace_ipc_error("error: reply failure", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_STAGE_UNINITIALIZED: + trace_ipc_error("error: stage uninitialized", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_NOT_FOUND: + trace_ipc_error("error: reply not found", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_SOURCE_NOT_STARTED: + trace_ipc_error("error: source not started", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_INVALID_REQUEST: + trace_ipc_error("error: invalid request", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_ERROR_INVALID_PARAM: + trace_ipc_error("error: invalid parameter", header); + msg->errno = -EINVAL; + break; + default: + trace_ipc_error("error: unknown reply", header); + msg->errno = -EINVAL; + break; + } + + /* update any stream states */ + hsw_stream_update(hsw, msg); + + /* wake up and return the error if we have waiters on this message ? */ + list_del(&msg->list); + tx_msg_reply_complete(hsw, msg); + + return 1; +} + +static int hsw_stream_message(struct sst_hsw *hsw, u32 header) +{ + u32 stream_msg, stream_id, stage_type; + struct sst_hsw_stream *stream; + int handled = 0; + + stream_msg = msg_get_stream_type(header); + stream_id = msg_get_stream_id(header); + stage_type = msg_get_stage_type(header); + + stream = get_stream_by_id(hsw, stream_id); + if (stream == NULL) + return handled; + + stream->header = header; + + switch (stream_msg) { + case IPC_STR_STAGE_MESSAGE: + dev_err(hsw->dev, "error: stage msg not implemented 0x%8.8x\n", + header); + break; + case IPC_STR_NOTIFICATION: + schedule_work(&stream->notify_work); + break; + default: + /* handle pending message complete request */ + handled = hsw_process_reply(hsw, header); + break; + } + + return handled; +} + +static int hsw_log_message(struct sst_hsw *hsw, u32 header) +{ + u32 operation = (header & IPC_LOG_OP_MASK) >> IPC_LOG_OP_SHIFT; + struct sst_hsw_log_stream *stream = &hsw->log_stream; + int ret = 1; + + if (operation != IPC_DEBUG_REQUEST_LOG_DUMP) { + dev_err(hsw->dev, + "error: log msg not implemented 0x%8.8x\n", header); + return 0; + } + + mutex_lock(&stream->rw_mutex); + stream->last_pos = stream->curr_pos; + sst_dsp_inbox_read( + hsw->dsp, &stream->curr_pos, sizeof(stream->curr_pos)); + mutex_unlock(&stream->rw_mutex); + + schedule_work(&stream->notify_work); + + return ret; +} + +static int hsw_process_notification(struct sst_hsw *hsw) +{ + struct sst_dsp *sst = hsw->dsp; + u32 type, header; + int handled = 1; + + header = sst_dsp_shim_read_unlocked(sst, SST_IPCD); + type = msg_get_global_type(header); + + trace_ipc_request("processing -->", header); + + /* FW Ready is a special case */ + if (!hsw->boot_complete && header & IPC_FW_READY) { + hsw_fw_ready(hsw, header); + return handled; + } + + switch (type) { + case IPC_GLB_GET_FW_VERSION: + case IPC_GLB_ALLOCATE_STREAM: + case IPC_GLB_FREE_STREAM: + case IPC_GLB_GET_FW_CAPABILITIES: + case IPC_GLB_REQUEST_DUMP: + case IPC_GLB_GET_DEVICE_FORMATS: + case IPC_GLB_SET_DEVICE_FORMATS: + case IPC_GLB_ENTER_DX_STATE: + case IPC_GLB_GET_MIXER_STREAM_INFO: + case IPC_GLB_MAX_IPC_MESSAGE_TYPE: + case IPC_GLB_RESTORE_CONTEXT: + case IPC_GLB_SHORT_REPLY: + dev_err(hsw->dev, "error: message type %d header 0x%x\n", + type, header); + break; + case IPC_GLB_STREAM_MESSAGE: + handled = hsw_stream_message(hsw, header); + break; + case IPC_GLB_DEBUG_LOG_MESSAGE: + handled = hsw_log_message(hsw, header); + break; + default: + dev_err(hsw->dev, "error: unexpected type %d hdr 0x%8.8x\n", + type, header); + break; + } + + return handled; +} + +static irqreturn_t hsw_irq_thread(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + struct sst_hsw *hsw = sst_dsp_get_thread_context(sst); + u32 ipcx, ipcd; + int handled; + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + + ipcx = sst_dsp_ipc_msg_rx(hsw->dsp); + ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD); + + /* reply message from DSP */ + if (ipcx & SST_IPCX_DONE) { + + /* Handle Immediate reply from DSP Core */ + handled = hsw_process_reply(hsw, ipcx); + + if (handled) { + /* clear DONE bit - tell DSP we have completed */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, + SST_IPCX_DONE, 0); + + /* unmask Done interrupt */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_DONE, 0); + } + } + + /* new message from DSP */ + if (ipcd & SST_IPCD_BUSY) { + + /* Handle Notification and Delayed reply from DSP Core */ + handled = hsw_process_notification(hsw); + + /* clear BUSY bit and set DONE bit - accept new messages */ + if (handled) { + sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, + SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); + + /* unmask busy interrupt */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_BUSY, 0); + } + } + + spin_unlock_irqrestore(&sst->spinlock, flags); + + /* continue to send any remaining messages... */ + queue_kthread_work(&hsw->kworker, &hsw->kwork); + + return IRQ_HANDLED; +} + +int sst_hsw_fw_get_version(struct sst_hsw *hsw, + struct sst_hsw_ipc_fw_version *version) +{ + int ret; + + ret = ipc_tx_message_wait(hsw, IPC_GLB_TYPE(IPC_GLB_GET_FW_VERSION), + NULL, 0, version, sizeof(*version)); + if (ret < 0) + dev_err(hsw->dev, "error: get version failed\n"); + + return ret; +} + +/* Mixer Controls */ +int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel) +{ + int ret; + + ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, + &stream->mute_volume[channel]); + if (ret < 0) + return ret; + + ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", + stream->reply.stream_hw_id, channel); + return ret; + } + + stream->mute[channel] = 1; + return 0; +} + +int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel) + +{ + int ret; + + stream->mute[channel] = 0; + ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, + stream->mute_volume[channel]); + if (ret < 0) { + dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", + stream->reply.stream_hw_id, channel); + return ret; + } + + return 0; +} + +int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel, u32 *volume) +{ + if (channel > 1) + return -EINVAL; + + sst_dsp_read(hsw->dsp, volume, + stream->reply.volume_register_address[channel], sizeof(volume)); + + return 0; +} + +int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u64 curve_duration, + enum sst_hsw_volume_curve curve) +{ + /* curve duration in steps of 100ns */ + stream->vol_req.curve_duration = curve_duration; + stream->vol_req.curve_type = curve; + + return 0; +} + +/* stream volume */ +int sst_hsw_stream_set_volume(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) +{ + struct sst_hsw_ipc_volume_req *req; + u32 header; + int ret; + + trace_ipc_request("set stream volume", stream->reply.stream_hw_id); + + if (channel > 1) + return -EINVAL; + + if (stream->mute[channel]) { + stream->mute_volume[channel] = volume; + return 0; + } + + header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | + IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); + header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); + header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT); + header |= (stage_id << IPC_STG_ID_SHIFT); + + req = &stream->vol_req; + req->channel = channel; + req->target_volume = volume; + + ret = ipc_tx_message_wait(hsw, header, req, sizeof(*req), NULL, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: set stream volume failed\n"); + return ret; + } + + return 0; +} + +int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) +{ + int ret; + + ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, + &hsw->mute_volume[channel]); + if (ret < 0) + return ret; + + ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", + channel); + return ret; + } + + hsw->mute[channel] = 1; + return 0; +} + +int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) +{ + int ret; + + ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, + hsw->mixer_info.volume_register_address[channel]); + if (ret < 0) { + dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", + channel); + return ret; + } + + hsw->mute[channel] = 0; + return 0; +} + +int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, + u32 *volume) +{ + if (channel > 1) + return -EINVAL; + + sst_dsp_read(hsw->dsp, volume, + hsw->mixer_info.volume_register_address[channel], + sizeof(*volume)); + + return 0; +} + +int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, + u64 curve_duration, enum sst_hsw_volume_curve curve) +{ + /* curve duration in steps of 100ns */ + hsw->curve_duration = curve_duration; + hsw->curve_type = curve; + + return 0; +} + +/* global mixer volume */ +int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, + u32 volume) +{ + struct sst_hsw_ipc_volume_req req; + u32 header; + int ret; + + trace_ipc_request("set mixer volume", volume); + + /* set both at same time ? */ + if (channel == 2) { + if (hsw->mute[0] && hsw->mute[1]) { + hsw->mute_volume[0] = hsw->mute_volume[1] = volume; + return 0; + } else if (hsw->mute[0]) + req.channel = 1; + else if (hsw->mute[1]) + req.channel = 0; + else + req.channel = 0xffffffff; + } else { + /* set only 1 channel */ + if (hsw->mute[channel]) { + hsw->mute_volume[channel] = volume; + return 0; + } + req.channel = channel; + } + + header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | + IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); + header |= (hsw->mixer_info.mixer_hw_id << IPC_STR_ID_SHIFT); + header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT); + header |= (stage_id << IPC_STG_ID_SHIFT); + + req.curve_duration = hsw->curve_duration; + req.curve_type = hsw->curve_type; + req.target_volume = volume; + + ret = ipc_tx_message_wait(hsw, header, &req, sizeof(req), NULL, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: set mixer volume failed\n"); + return ret; + } + + return 0; +} + +/* Stream API */ +struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, + u32 (*notify_position)(struct sst_hsw_stream *stream, void *data), + void *data) +{ + struct sst_hsw_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (stream == NULL) + return NULL; + + list_add(&stream->node, &hsw->stream_list); + stream->notify_position = notify_position; + stream->pdata = data; + stream->hsw = hsw; + stream->host_id = id; + + /* work to process notification messages */ + INIT_WORK(&stream->notify_work, hsw_notification_work); + + return stream; +} + +int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) +{ + u32 header; + int ret = 0; + + /* dont free DSP streams that are not commited */ + if (!stream->commited) + goto out; + + trace_ipc_request("stream free", stream->host_id); + + stream->free_req.stream_id = stream->reply.stream_hw_id; + header = IPC_GLB_TYPE(IPC_GLB_FREE_STREAM); + + ret = ipc_tx_message_wait(hsw, header, &stream->free_req, + sizeof(stream->free_req), NULL, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: free stream %d failed\n", + stream->free_req.stream_id); + return -EAGAIN; + } + + trace_hsw_stream_free_req(stream, &stream->free_req); + +out: + list_del(&stream->node); + kfree(stream); + + return ret; +} + +int sst_hsw_stream_set_bits(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, enum sst_hsw_bitdepth bits) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set bits\n"); + return -EINVAL; + } + + stream->request.format.bitdepth = bits; + return 0; +} + +int sst_hsw_stream_set_channels(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, int channels) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set channels\n"); + return -EINVAL; + } + + /* stereo is only supported atm */ + if (channels != 2) + return -EINVAL; + + stream->request.format.ch_num = channels; + return 0; +} + +int sst_hsw_stream_set_rate(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, int rate) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set rate\n"); + return -EINVAL; + } + + stream->request.format.frequency = rate; + return 0; +} + +int sst_hsw_stream_set_map_config(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 map, + enum sst_hsw_channel_config config) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set map\n"); + return -EINVAL; + } + + stream->request.format.map = map; + stream->request.format.config = config; + return 0; +} + +int sst_hsw_stream_set_style(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, enum sst_hsw_interleaving style) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set style\n"); + return -EINVAL; + } + + stream->request.format.style = style; + return 0; +} + +int sst_hsw_stream_set_valid(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 bits) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set valid bits\n"); + return -EINVAL; + } + + stream->request.format.valid_bit = bits; + return 0; +} + +/* Stream Configuration */ +int sst_hsw_stream_format(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + enum sst_hsw_stream_path_id path_id, + enum sst_hsw_stream_type stream_type, + enum sst_hsw_stream_format format_id) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set format\n"); + return -EINVAL; + } + + stream->request.path_id = path_id; + stream->request.stream_type = stream_type; + stream->request.format_id = format_id; + + trace_hsw_stream_alloc_request(stream, &stream->request); + + return 0; +} + +int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 ring_pt_address, u32 num_pages, + u32 ring_size, u32 ring_offset, u32 ring_first_pfn) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for buffer\n"); + return -EINVAL; + } + + stream->request.ringinfo.ring_pt_address = ring_pt_address; + stream->request.ringinfo.num_pages = num_pages; + stream->request.ringinfo.ring_size = ring_size; + stream->request.ringinfo.ring_offset = ring_offset; + stream->request.ringinfo.ring_first_pfn = ring_first_pfn; + + trace_hsw_stream_buffer(stream); + + return 0; +} + +int sst_hsw_stream_set_module_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id, + u32 entry_point) +{ + struct sst_hsw_module_map *map = &stream->request.map; + + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set module\n"); + return -EINVAL; + } + + /* only support initial module atm */ + map->module_entries_count = 1; + map->module_entries[0].module_id = module_id; + map->module_entries[0].entry_point = entry_point; + + return 0; +} + +int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 offset, u32 size) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set pmem\n"); + return -EINVAL; + } + + stream->request.persistent_mem.offset = offset; + stream->request.persistent_mem.size = size; + + return 0; +} + +int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 offset, u32 size) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set smem\n"); + return -EINVAL; + } + + stream->request.scratch_mem.offset = offset; + stream->request.scratch_mem.size = size; + + return 0; +} + +int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) +{ + struct sst_hsw_ipc_stream_alloc_req *str_req = &stream->request; + struct sst_hsw_ipc_stream_alloc_reply *reply = &stream->reply; + u32 header; + int ret; + + trace_ipc_request("stream alloc", stream->host_id); + + header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM); + + ret = ipc_tx_message_wait(hsw, header, str_req, sizeof(*str_req), + reply, sizeof(*reply)); + if (ret < 0) { + dev_err(hsw->dev, "error: stream commit failed\n"); + return ret; + } + + stream->commited = 1; + trace_hsw_stream_alloc_reply(stream); + + return 0; +} + +/* Stream Information - these calls could be inline but we want the IPC + ABI to be opaque to client PCM drivers to cope with any future ABI changes */ +int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->reply.stream_hw_id; +} + +int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->reply.mixer_hw_id; +} + +u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->reply.read_position_register_address; +} + +u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->reply.presentation_position_register_address; +} + +u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 channel) +{ + if (channel >= 2) + return 0; + + return stream->reply.peak_meter_register_address[channel]; +} + +u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 channel) +{ + if (channel >= 2) + return 0; + + return stream->reply.volume_register_address[channel]; +} + +int sst_hsw_mixer_get_info(struct sst_hsw *hsw) +{ + struct sst_hsw_ipc_stream_info_reply *reply; + u32 header; + int ret; + + reply = &hsw->mixer_info; + header = IPC_GLB_TYPE(IPC_GLB_GET_MIXER_STREAM_INFO); + + trace_ipc_request("get global mixer info", 0); + + ret = ipc_tx_message_wait(hsw, header, NULL, 0, reply, sizeof(*reply)); + if (ret < 0) { + dev_err(hsw->dev, "error: get stream info failed\n"); + return ret; + } + + trace_hsw_mixer_info_reply(reply); + + return 0; +} + +/* Send stream command */ +static int sst_hsw_stream_operations(struct sst_hsw *hsw, int type, + int stream_id, int wait) +{ + u32 header; + + header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | IPC_STR_TYPE(type); + header |= (stream_id << IPC_STR_ID_SHIFT); + + if (wait) + return ipc_tx_message_wait(hsw, header, NULL, 0, NULL, 0); + else + return ipc_tx_message_nowait(hsw, header, NULL, 0); +} + +/* Stream ALSA trigger operations */ +int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int wait) +{ + int ret; + + trace_ipc_request("stream pause", stream->reply.stream_hw_id); + + ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, + stream->reply.stream_hw_id, wait); + if (ret < 0) + dev_err(hsw->dev, "error: failed to pause stream %d\n", + stream->reply.stream_hw_id); + + return ret; +} + +int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int wait) +{ + int ret; + + trace_ipc_request("stream resume", stream->reply.stream_hw_id); + + ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, + stream->reply.stream_hw_id, wait); + if (ret < 0) + dev_err(hsw->dev, "error: failed to resume stream %d\n", + stream->reply.stream_hw_id); + + return ret; +} + +int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) +{ + int ret, tries = 10; + + /* dont reset streams that are not commited */ + if (!stream->commited) + return 0; + + /* wait for pause to complete before we reset the stream */ + while (stream->running && tries--) + msleep(1); + if (!tries) { + dev_err(hsw->dev, "error: reset stream %d still running\n", + stream->reply.stream_hw_id); + return -EINVAL; + } + + trace_ipc_request("stream reset", stream->reply.stream_hw_id); + + ret = sst_hsw_stream_operations(hsw, IPC_STR_RESET, + stream->reply.stream_hw_id, 1); + if (ret < 0) + dev_err(hsw->dev, "error: failed to reset stream %d\n", + stream->reply.stream_hw_id); + return ret; +} + +/* Stream pointer positions */ +int sst_hsw_get_dsp_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->rpos.position; +} + +int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 position) +{ + u32 header; + int ret; + + trace_stream_write_position(stream->reply.stream_hw_id, position); + + header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | + IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); + header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); + header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); + header |= (stage_id << IPC_STG_ID_SHIFT); + stream->wpos.position = position; + + ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, + sizeof(stream->wpos)); + if (ret < 0) + dev_err(hsw->dev, "error: stream %d set position %d failed\n", + stream->reply.stream_hw_id, position); + + return ret; +} + +/* physical BE config */ +int sst_hsw_device_set_config(struct sst_hsw *hsw, + enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, + enum sst_hsw_device_mode mode, u32 clock_divider) +{ + struct sst_hsw_ipc_device_config_req config; + u32 header; + int ret; + + trace_ipc_request("set device config", dev); + + config.ssp_interface = dev; + config.clock_frequency = mclk; + config.mode = mode; + config.clock_divider = clock_divider; + + trace_hsw_device_config_req(&config); + + header = IPC_GLB_TYPE(IPC_GLB_SET_DEVICE_FORMATS); + + ret = ipc_tx_message_wait(hsw, header, &config, sizeof(config), + NULL, 0); + if (ret < 0) + dev_err(hsw->dev, "error: set device formats failed\n"); + + return ret; +} +EXPORT_SYMBOL_GPL(sst_hsw_device_set_config); + +/* DX Config */ +int sst_hsw_dx_set_state(struct sst_hsw *hsw, + enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx) +{ + u32 header, state_; + int ret; + + header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE); + state_ = state; + + trace_ipc_request("PM enter Dx state", state); + + ret = ipc_tx_message_wait(hsw, header, &state_, sizeof(state_), + dx, sizeof(dx)); + if (ret < 0) { + dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state); + return ret; + } + + dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", + dx->entries_no, state); + + memcpy(&hsw->dx, dx, sizeof(*dx)); + return 0; +} + +/* Used to save state into hsw->dx_reply */ +int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, + u32 *offset, u32 *size, u32 *source) +{ + struct sst_hsw_ipc_dx_memory_item *dx_mem; + struct sst_hsw_ipc_dx_reply *dx_reply; + int entry_no; + + dx_reply = &hsw->dx; + entry_no = dx_reply->entries_no; + + trace_ipc_request("PM get Dx state", entry_no); + + if (item >= entry_no) + return -EINVAL; + + dx_mem = &dx_reply->mem_info[item]; + *offset = dx_mem->offset; + *size = dx_mem->size; + *source = dx_mem->source; + + return 0; +} + +static int msg_empty_list_init(struct sst_hsw *hsw) +{ + int i; + + hsw->msg = kzalloc(sizeof(struct ipc_message) * + IPC_EMPTY_LIST_SIZE, GFP_KERNEL); + if (hsw->msg == NULL) + return -ENOMEM; + + for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + init_waitqueue_head(&hsw->msg[i].waitq); + list_add(&hsw->msg[i].list, &hsw->empty_list); + } + + return 0; +} + +void sst_hsw_set_scratch_module(struct sst_hsw *hsw, + struct sst_module *scratch) +{ + hsw->scratch = scratch; +} + +struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw) +{ + return hsw->dsp; +} + +static struct sst_dsp_device hsw_dev = { + .thread = hsw_irq_thread, + .ops = &haswell_ops, +}; + +int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) +{ + struct sst_hsw_ipc_fw_version version; + struct sst_hsw *hsw; + struct sst_fw *hsw_sst_fw; + int ret; + + dev_dbg(dev, "initialising Audio DSP IPC\n"); + + hsw = devm_kzalloc(dev, sizeof(*hsw), GFP_KERNEL); + if (hsw == NULL) + return -ENOMEM; + + hsw->dev = dev; + INIT_LIST_HEAD(&hsw->stream_list); + INIT_LIST_HEAD(&hsw->tx_list); + INIT_LIST_HEAD(&hsw->rx_list); + INIT_LIST_HEAD(&hsw->empty_list); + init_waitqueue_head(&hsw->boot_wait); + init_waitqueue_head(&hsw->wait_txq); + + ret = msg_empty_list_init(hsw); + if (ret < 0) + goto list_err; + + /* start the IPC message thread */ + init_kthread_worker(&hsw->kworker); + hsw->tx_thread = kthread_run(kthread_worker_fn, + &hsw->kworker, + dev_name(hsw->dev)); + if (IS_ERR(hsw->tx_thread)) { + ret = PTR_ERR(hsw->tx_thread); + dev_err(hsw->dev, "error: failed to create message TX task\n"); + goto list_err; + } + init_kthread_work(&hsw->kwork, ipc_tx_msgs); + + hsw_dev.thread_context = hsw; + + /* init SST shim */ + hsw->dsp = sst_dsp_new(dev, &hsw_dev, pdata); + if (hsw->dsp == NULL) { + ret = -ENODEV; + goto list_err; + } + + /* keep the DSP in reset state for base FW loading */ + sst_dsp_reset(hsw->dsp); + + hsw_sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw); + + if (hsw_sst_fw == NULL) { + ret = -ENODEV; + dev_err(dev, "error: failed to load firmware\n"); + goto fw_err; + } + + /* wait for DSP boot completion */ + sst_dsp_boot(hsw->dsp); + ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, + msecs_to_jiffies(IPC_BOOT_MSECS)); + if (ret == 0) { + ret = -EIO; + dev_err(hsw->dev, "error: ADSP boot timeout\n"); + goto boot_err; + } + + /* get the FW version */ + sst_hsw_fw_get_version(hsw, &version); + dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n", + version.type, version.major, version.minor, version.build); + + /* get the globalmixer */ + ret = sst_hsw_mixer_get_info(hsw); + if (ret < 0) { + dev_err(hsw->dev, "error: failed to get stream info\n"); + goto boot_err; + } + + pdata->dsp = hsw; + return 0; + +boot_err: + sst_dsp_reset(hsw->dsp); + sst_fw_free(hsw_sst_fw); +fw_err: + sst_dsp_free(hsw->dsp); + kfree(hsw->msg); +list_err: + return ret; +} +EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); + +void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) +{ + struct sst_hsw *hsw = pdata->dsp; + + sst_dsp_reset(hsw->dsp); + sst_fw_free_all(hsw->dsp); + sst_dsp_free(hsw->dsp); + kfree(hsw->scratch); + kfree(hsw->msg); +} +EXPORT_SYMBOL_GPL(sst_hsw_dsp_free); diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h new file mode 100644 index 0000000..d517929 --- /dev/null +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -0,0 +1,488 @@ +/* + * Intel SST Haswell/Broadwell IPC Support + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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_HASWELL_IPC_H +#define __SST_HASWELL_IPC_H + +#include +#include +#include + +#define SST_HSW_NO_CHANNELS 2 +#define SST_HSW_MAX_DX_REGIONS 14 + +#define SST_HSW_FW_LOG_CONFIG_DWORDS 12 +#define SST_HSW_GLOBAL_LOG 15 + +/** + * Upfront defined maximum message size that is + * expected by the in/out communication pipes in FW. + */ +#define SST_HSW_IPC_MAX_PAYLOAD_SIZE 400 +#define SST_HSW_MAX_INFO_SIZE 64 +#define SST_HSW_BUILD_HASH_LENGTH 40 + +struct sst_hsw; +struct sst_hsw_stream; +struct sst_hsw_log_stream; +struct sst_pdata; +struct sst_module; +extern struct sst_ops haswell_ops; + +/* Stream Allocate Path ID */ +enum sst_hsw_stream_path_id { + SST_HSW_STREAM_PATH_SSP0_OUT = 0, + SST_HSW_STREAM_PATH_SSP0_IN = 1, + SST_HSW_STREAM_PATH_MAX_PATH_ID = 2, +}; + +/* Stream Allocate Stream Type */ +enum sst_hsw_stream_type { + SST_HSW_STREAM_TYPE_RENDER = 0, + SST_HSW_STREAM_TYPE_SYSTEM = 1, + SST_HSW_STREAM_TYPE_CAPTURE = 2, + SST_HSW_STREAM_TYPE_LOOPBACK = 3, + SST_HSW_STREAM_TYPE_MAX_STREAM_TYPE = 4, +}; + +/* Stream Allocate Stream Format */ +enum sst_hsw_stream_format { + SST_HSW_STREAM_FORMAT_PCM_FORMAT = 0, + SST_HSW_STREAM_FORMAT_MP3_FORMAT = 1, + SST_HSW_STREAM_FORMAT_AAC_FORMAT = 2, + SST_HSW_STREAM_FORMAT_MAX_FORMAT_ID = 3, +}; + +/* Device ID */ +enum sst_hsw_device_id { + SST_HSW_DEVICE_SSP_0 = 0, + SST_HSW_DEVICE_SSP_1 = 1, +}; + +/* Device Master Clock Frequency */ +enum sst_hsw_device_mclk { + SST_HSW_DEVICE_MCLK_OFF = 0, + SST_HSW_DEVICE_MCLK_FREQ_6_MHZ = 1, + SST_HSW_DEVICE_MCLK_FREQ_12_MHZ = 2, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ = 3, +}; + +/* Device Clock Master */ +enum sst_hsw_device_mode { + SST_HSW_DEVICE_CLOCK_SLAVE = 0, + SST_HSW_DEVICE_CLOCK_MASTER = 1, +}; + +/* DX Power State */ +enum sst_hsw_dx_state { + SST_HSW_DX_STATE_D0 = 0, + SST_HSW_DX_STATE_D1 = 1, + SST_HSW_DX_STATE_D3 = 3, + SST_HSW_DX_STATE_MAX = 3, +}; + +/* Audio stream stage IDs */ +enum sst_hsw_fx_stage_id { + SST_HSW_STAGE_ID_WAVES = 0, + SST_HSW_STAGE_ID_DTS = 1, + SST_HSW_STAGE_ID_DOLBY = 2, + SST_HSW_STAGE_ID_BOOST = 3, + SST_HSW_STAGE_ID_MAX_FX_ID +}; + +/* DX State Type */ +enum sst_hsw_dx_type { + SST_HSW_DX_TYPE_FW_IMAGE = 0, + SST_HSW_DX_TYPE_MEMORY_DUMP = 1 +}; + +/* Volume Curve Type*/ +enum sst_hsw_volume_curve { + SST_HSW_VOLUME_CURVE_NONE = 0, + SST_HSW_VOLUME_CURVE_FADE = 1 +}; + +/* Sample ordering */ +enum sst_hsw_interleaving { + SST_HSW_INTERLEAVING_PER_CHANNEL = 0, + SST_HSW_INTERLEAVING_PER_SAMPLE = 1, +}; + +/* Channel indices */ +enum sst_hsw_channel_index { + SST_HSW_CHANNEL_LEFT = 0, + SST_HSW_CHANNEL_CENTER = 1, + SST_HSW_CHANNEL_RIGHT = 2, + SST_HSW_CHANNEL_LEFT_SURROUND = 3, + SST_HSW_CHANNEL_CENTER_SURROUND = 3, + SST_HSW_CHANNEL_RIGHT_SURROUND = 4, + SST_HSW_CHANNEL_LFE = 7, + SST_HSW_CHANNEL_INVALID = 0xF, +}; + +/* List of supported channel maps. */ +enum sst_hsw_channel_config { + SST_HSW_CHANNEL_CONFIG_MONO = 0, /* mono only. */ + SST_HSW_CHANNEL_CONFIG_STEREO = 1, /* L & R. */ + SST_HSW_CHANNEL_CONFIG_2_POINT_1 = 2, /* L, R & LFE; PCM only. */ + SST_HSW_CHANNEL_CONFIG_3_POINT_0 = 3, /* L, C & R; MP3 & AAC only. */ + SST_HSW_CHANNEL_CONFIG_3_POINT_1 = 4, /* L, C, R & LFE; PCM only. */ + SST_HSW_CHANNEL_CONFIG_QUATRO = 5, /* L, R, Ls & Rs; PCM only. */ + SST_HSW_CHANNEL_CONFIG_4_POINT_0 = 6, /* L, C, R & Cs; MP3 & AAC only. */ + SST_HSW_CHANNEL_CONFIG_5_POINT_0 = 7, /* L, C, R, Ls & Rs. */ + SST_HSW_CHANNEL_CONFIG_5_POINT_1 = 8, /* L, C, R, Ls, Rs & LFE. */ + SST_HSW_CHANNEL_CONFIG_DUAL_MONO = 9, /* One channel replicated in two. */ + SST_HSW_CHANNEL_CONFIG_INVALID, +}; + +/* List of supported bit depths. */ +enum sst_hsw_bitdepth { + SST_HSW_DEPTH_8BIT = 8, + SST_HSW_DEPTH_16BIT = 16, + SST_HSW_DEPTH_24BIT = 24, /* Default. */ + SST_HSW_DEPTH_32BIT = 32, + SST_HSW_DEPTH_INVALID = 33, +}; + +enum sst_hsw_module_id { + SST_HSW_MODULE_BASE_FW = 0x0, + SST_HSW_MODULE_MP3 = 0x1, + SST_HSW_MODULE_AAC_5_1 = 0x2, + SST_HSW_MODULE_AAC_2_0 = 0x3, + SST_HSW_MODULE_SRC = 0x4, + SST_HSW_MODULE_WAVES = 0x5, + SST_HSW_MODULE_DOLBY = 0x6, + SST_HSW_MODULE_BOOST = 0x7, + SST_HSW_MODULE_LPAL = 0x8, + SST_HSW_MODULE_DTS = 0x9, + SST_HSW_MODULE_PCM_CAPTURE = 0xA, + SST_HSW_MODULE_PCM_SYSTEM = 0xB, + SST_HSW_MODULE_PCM_REFERENCE = 0xC, + SST_HSW_MODULE_PCM = 0xD, + SST_HSW_MODULE_BLUETOOTH_RENDER_MODULE = 0xE, + SST_HSW_MODULE_BLUETOOTH_CAPTURE_MODULE = 0xF, + SST_HSW_MAX_MODULE_ID, +}; + +enum sst_hsw_performance_action { + SST_HSW_PERF_START = 0, + SST_HSW_PERF_STOP = 1, +}; + +/* SST firmware module info */ +struct sst_hsw_module_info { + u8 name[SST_HSW_MAX_INFO_SIZE]; + u8 version[SST_HSW_MAX_INFO_SIZE]; +} __attribute__((packed)); + +/* Module entry point */ +struct sst_hsw_module_entry { + enum sst_hsw_module_id module_id; + u32 entry_point; +} __attribute__((packed)); + +/* Module map - alignement matches DSP */ +struct sst_hsw_module_map { + u8 module_entries_count; + struct sst_hsw_module_entry module_entries[1]; +} __attribute__((packed)); + +struct sst_hsw_memory_info { + u32 offset; + u32 size; +} __attribute__((packed)); + +struct sst_hsw_fx_enable { + struct sst_hsw_module_map module_map; + struct sst_hsw_memory_info persistent_mem; +} __attribute__((packed)); + +struct sst_hsw_get_fx_param { + u32 parameter_id; + u32 param_size; +} __attribute__((packed)); + +struct sst_hsw_perf_action { + u32 action; +} __attribute__((packed)); + +struct sst_hsw_perf_data { + u64 timestamp; + u64 cycles; + u64 datatime; +} __attribute__((packed)); + +/* FW version */ +struct sst_hsw_ipc_fw_version { + u8 build; + u8 minor; + u8 major; + u8 type; + u8 fw_build_hash[SST_HSW_BUILD_HASH_LENGTH]; + u32 fw_log_providers_hash; +} __attribute__((packed)); + +/* Stream ring info */ +struct sst_hsw_ipc_stream_ring { + u32 ring_pt_address; + u32 num_pages; + u32 ring_size; + u32 ring_offset; + u32 ring_first_pfn; +} __attribute__((packed)); + +/* Debug Dump Log Enable Request */ +struct sst_hsw_ipc_debug_log_enable_req { + struct sst_hsw_ipc_stream_ring ringinfo; + u32 config[SST_HSW_FW_LOG_CONFIG_DWORDS]; +} __attribute__((packed)); + +/* Debug Dump Log Reply */ +struct sst_hsw_ipc_debug_log_reply { + u32 log_buffer_begining; + u32 log_buffer_size; +} __attribute__((packed)); + +/* Stream glitch position */ +struct sst_hsw_ipc_stream_glitch_position { + u32 glitch_type; + u32 present_pos; + u32 write_pos; +} __attribute__((packed)); + +/* Stream get position */ +struct sst_hsw_ipc_stream_get_position { + u32 position; + u32 fw_cycle_count; +} __attribute__((packed)); + +/* Stream set position */ +struct sst_hsw_ipc_stream_set_position { + u32 position; + u32 end_of_buffer; +} __attribute__((packed)); + +/* Stream Free Request */ +struct sst_hsw_ipc_stream_free_req { + u8 stream_id; + u8 reserved[3]; +} __attribute__((packed)); + +/* Set Volume Request */ +struct sst_hsw_ipc_volume_req { + u32 channel; + u32 target_volume; + u64 curve_duration; + u32 curve_type; +} __attribute__((packed)); + +/* Device Configuration Request */ +struct sst_hsw_ipc_device_config_req { + u32 ssp_interface; + u32 clock_frequency; + u32 mode; + u16 clock_divider; + u16 reserved; +} __attribute__((packed)); + +/* Audio Data formats */ +struct sst_hsw_audio_data_format_ipc { + u32 frequency; + u32 bitdepth; + u32 map; + u32 config; + u32 style; + u8 ch_num; + u8 valid_bit; + u8 reserved[2]; +} __attribute__((packed)); + +/* Stream Allocate Request */ +struct sst_hsw_ipc_stream_alloc_req { + u8 path_id; + u8 stream_type; + u8 format_id; + u8 reserved; + struct sst_hsw_audio_data_format_ipc format; + struct sst_hsw_ipc_stream_ring ringinfo; + struct sst_hsw_module_map map; + struct sst_hsw_memory_info persistent_mem; + struct sst_hsw_memory_info scratch_mem; + u32 number_of_notifications; +} __attribute__((packed)); + +/* Stream Allocate Reply */ +struct sst_hsw_ipc_stream_alloc_reply { + u32 stream_hw_id; + u32 mixer_hw_id; // returns rate ???? + u32 read_position_register_address; + u32 presentation_position_register_address; + u32 peak_meter_register_address[SST_HSW_NO_CHANNELS]; + u32 volume_register_address[SST_HSW_NO_CHANNELS]; +} __attribute__((packed)); + +/* Get Mixer Stream Info */ +struct sst_hsw_ipc_stream_info_reply { + u32 mixer_hw_id; + u32 peak_meter_register_address[SST_HSW_NO_CHANNELS]; + u32 volume_register_address[SST_HSW_NO_CHANNELS]; +} __attribute__((packed)); + +/* DX State Request */ +struct sst_hsw_ipc_dx_req { + u8 state; + u8 reserved[3]; +} __attribute__((packed)); + +/* DX State Reply Memory Info Item */ +struct sst_hsw_ipc_dx_memory_item { + u32 offset; + u32 size; + u32 source; +} __attribute__((packed)); + +/* DX State Reply */ +struct sst_hsw_ipc_dx_reply { + u32 entries_no; + struct sst_hsw_ipc_dx_memory_item mem_info[SST_HSW_MAX_DX_REGIONS]; +} __attribute__((packed)); + +struct sst_hsw_ipc_fw_version; + +/* SST Init & Free */ +struct sst_hsw *sst_hsw_new(struct device *dev, const u8 *fw, size_t fw_length, + u32 fw_offset); +void sst_hsw_free(struct sst_hsw *hsw); +int sst_hsw_fw_get_version(struct sst_hsw *hsw, + struct sst_hsw_ipc_fw_version *version); +u32 create_channel_map(enum sst_hsw_channel_config config); + +/* Stream Mixer Controls - */ +int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel); +int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel); + +int sst_hsw_stream_set_volume(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); +int sst_hsw_stream_get_volume(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); + +int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u64 curve_duration, + enum sst_hsw_volume_curve curve); + +/* Global Mixer Controls - */ +int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); +int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); + +int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, + u32 volume); +int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, + u32 *volume); + +int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, + u64 curve_duration, enum sst_hsw_volume_curve curve); + +/* Stream API */ +struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, + u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), + void *data); + +int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream); + +/* Stream Configuration */ +int sst_hsw_stream_format(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + enum sst_hsw_stream_path_id path_id, + enum sst_hsw_stream_type stream_type, + enum sst_hsw_stream_format format_id); + +int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 ring_pt_address, u32 num_pages, + u32 ring_size, u32 ring_offset, u32 ring_first_pfn); + +int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream); + +int sst_hsw_stream_set_valid(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 bits); +int sst_hsw_stream_set_rate(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int rate); +int sst_hsw_stream_set_bits(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + enum sst_hsw_bitdepth bits); +int sst_hsw_stream_set_channels(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, int channels); +int sst_hsw_stream_set_map_config(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 map, + enum sst_hsw_channel_config config); +int sst_hsw_stream_set_style(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + enum sst_hsw_interleaving style); +int sst_hsw_stream_set_module_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id, + u32 entry_point); +int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 offset, u32 size); +int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 offset, u32 size); +int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 channel); +u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 channel); +int sst_hsw_mixer_get_info(struct sst_hsw *hsw); + +/* Stream ALSA trigger operations */ +int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int wait); +int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int wait); +int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream); + +/* Stream pointer positions */ +int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 *position); +int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 *position); +int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 position); +int sst_hsw_get_dsp_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); + +/* HW port config */ +int sst_hsw_device_set_config(struct sst_hsw *hsw, + enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, + enum sst_hsw_device_mode mode, u32 clock_divider); + +/* DX Config */ +int sst_hsw_dx_set_state(struct sst_hsw *hsw, + enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); +int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, + u32 *offset, u32 *size, u32 *source); + +/* init */ +int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); +void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata); +struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); +void sst_hsw_set_scratch_module(struct sst_hsw *hsw, + struct sst_module *scratch); + +#endif -- cgit v1.1 From d37797705d959c21e2f846ac73c2e17303bff936 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 20 Feb 2014 21:48:44 +0000 Subject: ASoC: Intel: Add Haswell and Broadwell PCM platform driver Add the Haswell and Broadwell PCM DSP platform driver. This driver uses the IPC driver for communication with the SST DSP. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 872 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 872 insertions(+) create mode 100644 sound/soc/intel/sst-haswell-pcm.c (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c new file mode 100644 index 0000000..0a32dd1 --- /dev/null +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -0,0 +1,872 @@ +/* + * Intel SST Haswell/Broadwell PCM Support + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sst-haswell-ipc.h" +#include "sst-dsp-priv.h" +#include "sst-dsp.h" + +#define HSW_PCM_COUNT 6 +#define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */ + +/* simple volume table */ +static const u32 volume_map[] = { + HSW_VOLUME_MAX >> 30, + HSW_VOLUME_MAX >> 29, + HSW_VOLUME_MAX >> 28, + HSW_VOLUME_MAX >> 27, + HSW_VOLUME_MAX >> 26, + HSW_VOLUME_MAX >> 25, + HSW_VOLUME_MAX >> 24, + HSW_VOLUME_MAX >> 23, + HSW_VOLUME_MAX >> 22, + HSW_VOLUME_MAX >> 21, + HSW_VOLUME_MAX >> 20, + HSW_VOLUME_MAX >> 19, + HSW_VOLUME_MAX >> 18, + HSW_VOLUME_MAX >> 17, + HSW_VOLUME_MAX >> 16, + HSW_VOLUME_MAX >> 15, + HSW_VOLUME_MAX >> 14, + HSW_VOLUME_MAX >> 13, + HSW_VOLUME_MAX >> 12, + HSW_VOLUME_MAX >> 11, + HSW_VOLUME_MAX >> 10, + HSW_VOLUME_MAX >> 9, + HSW_VOLUME_MAX >> 8, + HSW_VOLUME_MAX >> 7, + HSW_VOLUME_MAX >> 6, + HSW_VOLUME_MAX >> 5, + HSW_VOLUME_MAX >> 4, + HSW_VOLUME_MAX >> 3, + HSW_VOLUME_MAX >> 2, + HSW_VOLUME_MAX >> 1, + HSW_VOLUME_MAX >> 0, +}; + +#define HSW_PCM_PERIODS_MAX 64 +#define HSW_PCM_PERIODS_MIN 2 + +static const struct snd_pcm_hardware hsw_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE, + .periods_min = HSW_PCM_PERIODS_MIN, + .periods_max = HSW_PCM_PERIODS_MAX, + .buffer_bytes_max = HSW_PCM_PERIODS_MAX * PAGE_SIZE, +}; + +/* private data for each PCM DSP stream */ +struct hsw_pcm_data { + int dai_id; + struct sst_hsw_stream *stream; + u32 volume[2]; + struct snd_pcm_substream *substream; + struct snd_compr_stream *cstream; + unsigned int wpos; + struct mutex mutex; +}; + +/* private data for the driver */ +struct hsw_priv_data { + /* runtime DSP */ + struct sst_hsw *hsw; + + /* page tables */ + unsigned char *pcm_pg[HSW_PCM_COUNT][2]; + + /* DAI data */ + struct hsw_pcm_data pcm[HSW_PCM_COUNT]; +}; + +static inline u32 hsw_mixer_to_ipc(unsigned int value) +{ + if (value >= ARRAY_SIZE(volume_map)) + return volume_map[0]; + else + return volume_map[value]; +} + +static inline unsigned int hsw_ipc_to_mixer(u32 value) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(volume_map); i++) { + if (volume_map[i] >= value) + return i; + } + + return i - 1; +} + +static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(platform); + struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct sst_hsw *hsw = pdata->hsw; + u32 volume; + + mutex_lock(&pcm_data->mutex); + + if (!pcm_data->stream) { + pcm_data->volume[0] = + hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + pcm_data->volume[1] = + hsw_mixer_to_ipc(ucontrol->value.integer.value[1]); + mutex_unlock(&pcm_data->mutex); + return 0; + } + + if (ucontrol->value.integer.value[0] == + ucontrol->value.integer.value[1]) { + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 2, volume); + } else { + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 0, volume); + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 1, volume); + } + + mutex_unlock(&pcm_data->mutex); + return 0; +} + +static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(platform); + struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct sst_hsw *hsw = pdata->hsw; + u32 volume; + + mutex_lock(&pcm_data->mutex); + + if (!pcm_data->stream) { + ucontrol->value.integer.value[0] = + hsw_ipc_to_mixer(pcm_data->volume[0]); + ucontrol->value.integer.value[1] = + hsw_ipc_to_mixer(pcm_data->volume[1]); + mutex_unlock(&pcm_data->mutex); + return 0; + } + + sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 0, &volume); + ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume); + sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 1, &volume); + ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume); + mutex_unlock(&pcm_data->mutex); + + return 0; +} + +static int hsw_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct sst_hsw *hsw = pdata->hsw; + u32 volume; + + if (ucontrol->value.integer.value[0] == + ucontrol->value.integer.value[1]) { + + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + sst_hsw_mixer_set_volume(hsw, 0, 2, volume); + + } else { + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + sst_hsw_mixer_set_volume(hsw, 0, 0, volume); + + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]); + sst_hsw_mixer_set_volume(hsw, 0, 1, volume); + } + + return 0; +} + +static int hsw_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct sst_hsw *hsw = pdata->hsw; + unsigned int volume = 0; + + sst_hsw_mixer_get_volume(hsw, 0, 0, &volume); + ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume); + + sst_hsw_mixer_get_volume(hsw, 0, 1, &volume); + ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume); + + return 0; +} + +/* TLV used by both global and stream volumes */ +static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1); + +/* System Pin has no volume control */ +static const struct snd_kcontrol_new hsw_volume_controls[] = { + /* Global DSP volume */ + SOC_DOUBLE_EXT_TLV("Master Playback Volume", 0, 0, 8, + ARRAY_SIZE(volume_map) -1, 0, + hsw_volume_get, hsw_volume_put, hsw_vol_tlv), + /* Offload 0 volume */ + SOC_DOUBLE_EXT_TLV("Media0 Playback Volume", 1, 0, 8, + ARRAY_SIZE(volume_map), 0, + hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), + /* Offload 1 volume */ + SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8, + ARRAY_SIZE(volume_map), 0, + hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), + /* Loopback volume */ + SOC_DOUBLE_EXT_TLV("Loopback Capture Volume", 3, 0, 8, + ARRAY_SIZE(volume_map), 0, + hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), + /* Mic Capture volume */ + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, + ARRAY_SIZE(volume_map), 0, + hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), +}; + +/* Create DMA buffer page table for DSP */ +static int create_adsp_page_table(struct hsw_priv_data *pdata, + struct snd_soc_pcm_runtime *rtd, + unsigned char *dma_area, size_t size, int pcm, int stream) +{ + int i, pages; + + if (size % PAGE_SIZE) + pages = (size / PAGE_SIZE) + 1; + else + pages = size / PAGE_SIZE; + + dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dma_area, size, pages); + + for (i = 0; i < pages; i++) { + u32 idx = (((i << 2) + i)) >> 1; + u32 pfn = (virt_to_phys(dma_area + i * PAGE_SIZE)) >> PAGE_SHIFT; + u32 *pg_table; + + dev_dbg(rtd->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn); + + pg_table = (u32*)(pdata->pcm_pg[pcm][stream] + idx); + + if (i & 1) + *pg_table |= (pfn << 4); + else + *pg_table |= pfn; + } + + return 0; +} + +/* this may get called several times by oss emulation */ +static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw *hsw = pdata->hsw; + struct sst_module *module_data; + struct sst_dsp *dsp; + enum sst_hsw_stream_type stream_type; + enum sst_hsw_stream_path_id path_id; + u32 rate, bits, map, pages, module_id; + u8 channels; + int ret; + + /* stream direction */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + path_id = SST_HSW_STREAM_PATH_SSP0_OUT; + else + path_id = SST_HSW_STREAM_PATH_SSP0_IN; + + /* DSP stream type depends on DAI ID */ + switch (rtd->cpu_dai->id) { + case 0: + stream_type = SST_HSW_STREAM_TYPE_SYSTEM; + module_id = SST_HSW_MODULE_PCM_SYSTEM; + break; + case 1: + case 2: + stream_type = SST_HSW_STREAM_TYPE_RENDER; + module_id = SST_HSW_MODULE_PCM; + break; + case 3: + /* path ID needs to be OUT for loopback */ + stream_type = SST_HSW_STREAM_TYPE_LOOPBACK; + path_id = SST_HSW_STREAM_PATH_SSP0_OUT; + module_id = SST_HSW_MODULE_PCM_REFERENCE; + break; + case 4: + stream_type = SST_HSW_STREAM_TYPE_CAPTURE; + module_id = SST_HSW_MODULE_PCM_CAPTURE; + break; + default: + dev_err(rtd->dev, "error: invalid DAI ID %d\n", + rtd->cpu_dai->id); + return -EINVAL; + }; + + ret = sst_hsw_stream_format(hsw, pcm_data->stream, + path_id, stream_type, SST_HSW_STREAM_FORMAT_PCM_FORMAT); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set format %d\n", ret); + return ret; + } + + rate = params_rate(params); + ret = sst_hsw_stream_set_rate(hsw, pcm_data->stream, rate); + if (ret < 0) { + dev_err(rtd->dev, "error: could not set rate %d\n", rate); + return ret; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + bits = SST_HSW_DEPTH_16BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = SST_HSW_DEPTH_24BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32); + break; + default: + dev_err(rtd->dev, "error: invalid format %d\n", + params_format(params)); + return -EINVAL; + } + + ret = sst_hsw_stream_set_bits(hsw, pcm_data->stream, bits); + if (ret < 0) { + dev_err(rtd->dev, "error: could not set bits %d\n", bits); + return ret; + } + + /* we only support stereo atm */ + channels = params_channels(params); + if (channels != 2) { + dev_err(rtd->dev, "error: invalid channels %d\n", channels); + return -EINVAL; + } + + map = create_channel_map(SST_HSW_CHANNEL_CONFIG_STEREO); + sst_hsw_stream_set_map_config(hsw, pcm_data->stream, + map, SST_HSW_CHANNEL_CONFIG_STEREO); + + ret = sst_hsw_stream_set_channels(hsw, pcm_data->stream, channels); + if (ret < 0) { + dev_err(rtd->dev, "error: could not set channels %d\n", + channels); + return ret; + } + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) { + dev_err(rtd->dev, "error: could not allocate %d bytes for PCM %d\n", + params_buffer_bytes(params), ret); + return ret; + } + + ret = create_adsp_page_table(pdata, rtd, runtime->dma_area, + runtime->dma_bytes, rtd->cpu_dai->id, substream->stream); + if (ret < 0) + return ret; + + sst_hsw_stream_set_style(hsw, pcm_data->stream, + SST_HSW_INTERLEAVING_PER_CHANNEL); + + if (runtime->dma_bytes % PAGE_SIZE) + pages = (runtime->dma_bytes / PAGE_SIZE) + 1; + else + pages = runtime->dma_bytes / PAGE_SIZE; + + ret = sst_hsw_stream_buffer(hsw, pcm_data->stream, + virt_to_phys(pdata->pcm_pg[rtd->cpu_dai->id][substream->stream]), + pages, runtime->dma_bytes, 0, + (u32)(virt_to_phys(runtime->dma_area) >> PAGE_SHIFT)); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set DMA buffer %d\n", ret); + return ret; + } + + dsp = sst_hsw_get_dsp(hsw); + + module_data = sst_module_get_from_id(dsp, module_id); + if (module_data == NULL) { + dev_err(rtd->dev, "error: failed to get module config\n"); + return -EINVAL; + } + + /* we use hardcoded memory offsets atm, will be updated for new FW */ + if (stream_type == SST_HSW_STREAM_TYPE_CAPTURE) { + sst_hsw_stream_set_module_info(hsw, pcm_data->stream, + SST_HSW_MODULE_PCM_CAPTURE, module_data->entry); + sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, + 0x449400, 0x4000); + sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, + 0x400000, 0); + } else { /* stream_type == SST_HSW_STREAM_TYPE_SYSTEM */ + sst_hsw_stream_set_module_info(hsw, pcm_data->stream, + SST_HSW_MODULE_PCM_SYSTEM, module_data->entry); + + sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, + module_data->offset, module_data->size); + sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, + 0x44d400, 0x3800); + + sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, + module_data->offset, module_data->size); + sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, + 0x400000, 0); + } + + ret = sst_hsw_stream_commit(hsw, pcm_data->stream); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to commit stream %d\n", ret); + return ret; + } + + ret = sst_hsw_stream_pause(hsw, pcm_data->stream, 1); + if (ret < 0) + dev_err(rtd->dev, "error: failed to pause %d\n", ret); + + return 0; +} + +static int hsw_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw *hsw = pdata->hsw; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_hsw_stream_resume(hsw, pcm_data->stream, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_hsw_stream_pause(hsw, pcm_data->stream, 0); + break; + default: + break; + } + + return 0; +} + +static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) +{ + struct hsw_pcm_data *pcm_data = data; + struct snd_pcm_substream *substream = pcm_data->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + u32 pos; + + pos = frames_to_bytes(runtime, + (runtime->control->appl_ptr % runtime->buffer_size)); + + dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + + /* let alsa know we have play a period */ + snd_pcm_period_elapsed(substream); + return pos; +} + +static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw *hsw = pdata->hsw; + snd_pcm_uframes_t offset; + + offset = bytes_to_frames(runtime, + sst_hsw_get_dsp_position(hsw, pcm_data->stream)); + + dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n", + frames_to_bytes(runtime, (u32)offset)); + return offset; +} + +static int hsw_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data; + struct sst_hsw *hsw = pdata->hsw; + + pcm_data = &pdata->pcm[rtd->cpu_dai->id]; + + mutex_lock(&pcm_data->mutex); + + snd_soc_pcm_set_drvdata(rtd, pcm_data); + pcm_data->substream = substream; + + snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware); + + pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id, + hsw_notify_pointer, pcm_data); + if (pcm_data->stream == NULL) { + dev_err(rtd->dev, "error: failed to create stream\n"); + mutex_unlock(&pcm_data->mutex); + return -EINVAL; + } + + /* Set previous saved volume */ + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 0, pcm_data->volume[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 1, pcm_data->volume[1]); + + mutex_unlock(&pcm_data->mutex); + return 0; +} + +static int hsw_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw *hsw = pdata->hsw; + int ret; + + mutex_lock(&pcm_data->mutex); + ret = sst_hsw_stream_reset(hsw, pcm_data->stream); + if (ret < 0) { + dev_dbg(rtd->dev, "error: reset stream failed %d\n", ret); + goto out; + } + + ret = sst_hsw_stream_free(hsw, pcm_data->stream); + if (ret < 0) { + dev_dbg(rtd->dev, "error: free stream failed %d\n", ret); + goto out; + } + pcm_data->stream = NULL; + +out: + mutex_unlock(&pcm_data->mutex); + return ret; +} + +static struct snd_pcm_ops hsw_pcm_ops = { + .open = hsw_pcm_open, + .close = hsw_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hsw_pcm_hw_params, + .hw_free = hsw_pcm_hw_free, + .trigger = hsw_pcm_trigger, + .pointer = hsw_pcm_pointer, + .mmap = snd_pcm_lib_default_mmap, +}; + +static void hsw_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + rtd->card->dev, + hsw_pcm_hardware.buffer_bytes_max, + hsw_pcm_hardware.buffer_bytes_max); + if (ret) { + dev_err(rtd->dev, "dma buffer allocation failed %d\n", + ret); + return ret; + } + } + + return ret; +} + +#define HSW_FORMATS \ + (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver hsw_dais[] = { + { + .name = "System Pin", + .playback = { + .stream_name = "System Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, + { + /* PCM */ + .name = "Offload0 Pin", + .playback = { + .stream_name = "Offload0 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = HSW_FORMATS, + }, + }, + { + /* PCM */ + .name = "Offload1 Pin", + .playback = { + .stream_name = "Offload1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = HSW_FORMATS, + }, + }, + { + .name = "Loopback Pin", + .capture = { + .stream_name = "Loopback Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = HSW_FORMATS, + }, + }, + { + .name = "Capture Pin", + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = HSW_FORMATS, + }, + }, +}; + +static const struct snd_soc_dapm_widget widgets[] = { + + /* Backend DAIs */ + SND_SOC_DAPM_AIF_IN("SSP0 CODEC IN", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SSP0 CODEC OUT", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SSP1 BT IN", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SSP1 BT OUT", NULL, 0, SND_SOC_NOPM, 0, 0), + + /* Global Playback Mixer */ + SND_SOC_DAPM_MIXER("Playback VMixer", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route graph[] = { + + /* Playback Mixer */ + {"Playback VMixer", NULL, "System Playback"}, + {"Playback VMixer", NULL, "Offload0 Playback"}, + {"Playback VMixer", NULL, "Offload1 Playback"}, + + {"SSP0 CODEC OUT", NULL, "Playback VMixer"}, + + {"Analog Capture", NULL, "SSP0 CODEC IN"}, +}; + +static int hsw_pcm_probe(struct snd_soc_platform *platform) +{ + struct sst_pdata *pdata = dev_get_platdata(platform->dev); + struct hsw_priv_data *priv_data; + int i; + + if (!pdata) + return -ENODEV; + + priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); + priv_data->hsw = pdata->dsp; + snd_soc_platform_set_drvdata(platform, priv_data); + + /* allocate DSP buffer page tables */ + for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { + + mutex_init(&priv_data->pcm[i].mutex); + + /* playback */ + if (hsw_dais[i].playback.channels_min) { + priv_data->pcm_pg[i][0] = kzalloc(PAGE_SIZE, GFP_DMA); + if (priv_data->pcm_pg[i][0] == NULL) + goto err; + } + + /* capture */ + if (hsw_dais[i].capture.channels_min) { + priv_data->pcm_pg[i][1] = kzalloc(PAGE_SIZE, GFP_DMA); + if (priv_data->pcm_pg[i][1] == NULL) + goto err; + } + } + + return 0; + +err: + for (;i >= 0; i--) { + if (hsw_dais[i].playback.channels_min) + kfree(priv_data->pcm_pg[i][0]); + if (hsw_dais[i].capture.channels_min) + kfree(priv_data->pcm_pg[i][1]); + } + return -ENOMEM; +} + +static int hsw_pcm_remove(struct snd_soc_platform *platform) +{ + struct hsw_priv_data *priv_data = + snd_soc_platform_get_drvdata(platform); + int i; + + for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { + if (hsw_dais[i].playback.channels_min) + kfree(priv_data->pcm_pg[i][0]); + if (hsw_dais[i].capture.channels_min) + kfree(priv_data->pcm_pg[i][1]); + } + + return 0; +} + +static struct snd_soc_platform_driver hsw_soc_platform = { + .probe = hsw_pcm_probe, + .remove = hsw_pcm_remove, + .ops = &hsw_pcm_ops, + .pcm_new = hsw_pcm_new, + .pcm_free = hsw_pcm_free, + .controls = hsw_volume_controls, + .num_controls = ARRAY_SIZE(hsw_volume_controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = graph, + .num_dapm_routes = ARRAY_SIZE(graph), +}; + +static const struct snd_soc_component_driver hsw_dai_component = { + .name = "haswell-dai", +}; + +static int hsw_pcm_dev_probe(struct platform_device *pdev) +{ + struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + int ret; + + ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata); + if (ret < 0) + return -ENODEV; + + ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform); + if (ret < 0) + goto err_plat; + + ret = snd_soc_register_component(&pdev->dev, &hsw_dai_component, + hsw_dais, ARRAY_SIZE(hsw_dais)); + if (ret < 0) + goto err_comp; + + return 0; + +err_comp: + snd_soc_unregister_platform(&pdev->dev); +err_plat: + sst_hsw_dsp_free(&pdev->dev, sst_pdata); + return 0; +} + +static int hsw_pcm_dev_remove(struct platform_device *pdev) +{ + struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + sst_hsw_dsp_free(&pdev->dev, sst_pdata); + + return 0; +} + +static struct platform_driver hsw_pcm_driver = { + .driver = { + .name = "haswell-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = hsw_pcm_dev_probe, + .remove = hsw_pcm_dev_remove, +}; +module_platform_driver(hsw_pcm_driver); + +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Haswell/Lynxpoint + Broadwell/Wildcatpoint PCM"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:haswell-pcm-audio"); -- cgit v1.1 From 5e4482fcb119d61f4bef226c442634cdd2618b31 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 20 Feb 2014 21:48:46 +0000 Subject: ASoC: Intel: Add build support for Haswell ADSP Build the Haswell/Broadwell PCM, IPC and DSP drivers. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 ++ sound/soc/intel/Makefile | 6 ++++++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index dd048fe..4f1ac8f 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -24,3 +24,5 @@ config SND_SOC_INTEL_SST config SND_SOC_INTEL_SST_ACPI tristate +config SND_SOC_INTEL_HASWELL + tristate diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index cf47100..4c08b21 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -10,3 +10,9 @@ obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o + +# Platform Support +snd-soc-sst-haswell-pcm-objs := \ + sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o + +obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += snd-soc-sst-haswell-pcm.o -- cgit v1.1 From 90931b9eaed9aaf772784a93da320cf10713effa Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 20 Feb 2014 21:48:47 +0000 Subject: ASoC: Intel: Add Haswell Machine support Add support for Haswell based machines with SST DSP audio. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 11 +++ sound/soc/intel/Makefile | 5 + sound/soc/intel/haswell.c | 241 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 257 insertions(+) create mode 100644 sound/soc/intel/haswell.c (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4f1ac8f..ce5a692 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -26,3 +26,14 @@ config SND_SOC_INTEL_SST_ACPI config SND_SOC_INTEL_HASWELL tristate + +config SND_SOC_INTEL_HASWELL_MACH + tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_HASWELL + select SND_SOC_RT5640 + help + This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell + Ultrabook platforms. + Say Y if you have such a device + If unsure select "N". \ No newline at end of file diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4c08b21..1c18815 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -16,3 +16,8 @@ snd-soc-sst-haswell-pcm-objs := \ sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += snd-soc-sst-haswell-pcm.o + +# Machine support +snd-soc-sst-haswell-objs := haswell.o + +obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c new file mode 100644 index 0000000..0d61197 --- /dev/null +++ b/sound/soc/intel/haswell.c @@ -0,0 +1,241 @@ +/* + * Intel Haswell Lynxpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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. + * + */ + +#include +#include +#include +#include +#include +#include + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt5640.h" + +/* Haswell ULT platforms have a Headphone and Mic jack */ +static const struct snd_soc_dapm_widget haswell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route haswell_rt5640_map[] = { + + {"Headphones", NULL, "HPOR"}, + {"Headphones", NULL, "HPOL"}, + {"IN2P", NULL, "Mic"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int haswell_ssp0_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); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int haswell_rt5640_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; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec DAI configuration\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + /* set correct codec filter for DAI format and clock config */ + snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000); + + return ret; +} + +static struct snd_soc_ops haswell_rt5640_ops = { + .hw_params = haswell_rt5640_hw_params, +}; + +static int haswell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *haswell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(haswell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "failed to set device config\n"); + return ret; + } + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Mic"); + + return 0; +} + +static struct snd_soc_dai_link haswell_rt5640_dais[] = { + /* Front End DAI links */ + { + .name = "System", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = haswell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Capture", + .stream_name = "Capture", + .cpu_dai_name = "Capture Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT33CA:00", + .codec_dai_name = "rt5640-aif1", + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = haswell_ssp0_fixup, + .ops = &haswell_rt5640_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ +static struct snd_soc_card haswell_rt5640 = { + .name = "haswell-rt5640", + .owner = THIS_MODULE, + .dai_link = haswell_rt5640_dais, + .num_links = ARRAY_SIZE(haswell_rt5640_dais), + .dapm_widgets = haswell_widgets, + .num_dapm_widgets = ARRAY_SIZE(haswell_widgets), + .dapm_routes = haswell_rt5640_map, + .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map), + .fully_routed = true, +}; + +static int haswell_audio_probe(struct platform_device *pdev) +{ + haswell_rt5640.dev = &pdev->dev; + + return snd_soc_register_card(&haswell_rt5640); +} + +static int haswell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&haswell_rt5640); + return 0; +} + +static struct platform_driver haswell_audio = { + .probe = haswell_audio_probe, + .remove = haswell_audio_remove, + .driver = { + .name = "haswell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(haswell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:haswell-audio"); -- cgit v1.1 From 26d04ca8c46e3af510f0bbbe4d0b1aca8e18b393 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 11:00:13 +0100 Subject: ASoC: lm49453: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index e19490c..d6f391a 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -213,15 +213,13 @@ static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; -static const struct soc_enum lm49453_adcl_enum = - SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, - ARRAY_SIZE(lm49453_adcl_mux_text), - lm49453_adcl_mux_text); - -static const struct soc_enum lm49453_adcr_enum = - SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, - ARRAY_SIZE(lm49453_adcr_mux_text), - lm49453_adcr_mux_text); +static SOC_ENUM_SINGLE_DECL(lm49453_adcl_enum, + LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + lm49453_adcl_mux_text); + +static SOC_ENUM_SINGLE_DECL(lm49453_adcr_enum, + LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + lm49453_adcr_mux_text); static const struct snd_kcontrol_new lm49453_adcl_mux_control = SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum); -- cgit v1.1 From 56b2f349137bfdd23e498f12a97fe3d6139c097b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 20 Feb 2014 09:06:30 +0900 Subject: ASoC: io: Remove SND_SOC_I2C Now that all users have been converted to regmap we can eliminate the ASoC level wrapper for I2C I/O reducing the amount of duplicated functionality. Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 3a0d99e..add99e2 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -99,13 +99,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, config.val_bits = data_bits; switch (control) { -#if IS_ENABLED(CONFIG_REGMAP_I2C) - case SND_SOC_I2C: - codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), - &config); - break; -#endif - case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ codec->using_regmap = true; -- cgit v1.1 From cb29d7b9ef7faf95e27d90362a5e7694c5479093 Mon Sep 17 00:00:00 2001 From: xiangxiao Date: Sun, 23 Feb 2014 14:40:44 +0800 Subject: ASoC: add data field into snd_soc_jack_gpio so callback could get the context data as needed Signed-off-by: xiangxiao Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 23d43da..7200602 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -250,7 +250,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) report = 0; if (gpio->jack_status_check) - report = gpio->jack_status_check(); + report = gpio->jack_status_check(gpio->data); snd_soc_jack_report(jack, report, gpio->report); } -- cgit v1.1 From f1adf5be51a952d06760d8b38c55e209bbf7054e Mon Sep 17 00:00:00 2001 From: xiangxiao Date: Sun, 23 Feb 2014 14:44:52 +0800 Subject: ASoC: delay the initial jack detect by debounce_time so the hardware could get time to initialize and debounce Signed-off-by: xiangxiao Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7200602..b903f82 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -342,7 +342,8 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpio_export(gpios[i].gpio, false); /* Update initial jack status */ - snd_soc_jack_gpio_detect(&gpios[i]); + schedule_delayed_work(&gpios[i].work, + msecs_to_jiffies(gpios[i].debounce_time)); } return 0; -- cgit v1.1 From 6ef20de726bd68b68a925cc63f12e0ed655c6b56 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Feb 2014 17:26:56 +0200 Subject: ASoC: Intel: Add Baytrail SST ID and Baytrail specific register bits While the SHIM register addresses in Baytrail are the same than Haswell and Broadwell their register size is 64-bit and some bits are different. This patch adds the SST device ID for Baytrail and Baytrail specific SHIM bit definitions. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 608418c..74052b5 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -24,6 +24,7 @@ /* SST Device IDs */ #define SST_DEV_ID_LYNX_POINT 0x33C8 #define SST_DEV_ID_WILDCAT_POINT 0x3438 +#define SST_DEV_ID_BYT 0x0F28 /* Supported SST DMA Devices */ #define SST_DMA_TYPE_DW 1 @@ -72,10 +73,15 @@ #define SST_CSR_S0IOCS (0x1 << 21) #define SST_CSR_S1IOCS (0x1 << 23) #define SST_CSR_LPCS (0x1 << 31) +#define SST_BYT_CSR_RST (0x1 << 0) +#define SST_BYT_CSR_VECTOR_SEL (0x1 << 1) +#define SST_BYT_CSR_STALL (0x1 << 2) +#define SST_BYT_CSR_PWAITMODE (0x1 << 3) /* ISRX / ISC */ #define SST_ISRX_BUSY (0x1 << 1) #define SST_ISRX_DONE (0x1 << 0) +#define SST_BYT_ISRX_REQUEST (0x1 << 1) /* ISRD / ISD */ #define SST_ISRD_BUSY (0x1 << 1) @@ -84,14 +90,19 @@ /* IMRX / IMC */ #define SST_IMRX_BUSY (0x1 << 1) #define SST_IMRX_DONE (0x1 << 0) +#define SST_BYT_IMRX_REQUEST (0x1 << 1) /* IPCX / IPCC */ #define SST_IPCX_DONE (0x1 << 30) #define SST_IPCX_BUSY (0x1 << 31) +#define SST_BYT_IPCX_DONE ((u64)0x1 << 62) +#define SST_BYT_IPCX_BUSY ((u64)0x1 << 63) /* IPCD */ #define SST_IPCD_DONE (0x1 << 30) #define SST_IPCD_BUSY (0x1 << 31) +#define SST_BYT_IPCD_DONE ((u64)0x1 << 62) +#define SST_BYT_IPCD_BUSY ((u64)0x1 << 63) /* CLKCTL */ #define SST_CLKCTL_SMOS(x) (x << 24) -- cgit v1.1 From f746966377d08aea60a1f21a6387855409524b9b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Feb 2014 17:26:57 +0200 Subject: ASoC: Intel: Add Intel Baytrail SST DSP support This adds basic functionality for Baytrail SST DSP initialization and firmware loading. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-dsp.c | 372 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-dsp-priv.h | 1 + 2 files changed, 373 insertions(+) create mode 100644 sound/soc/intel/sst-baytrail-dsp.c (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c new file mode 100644 index 0000000..a50bf7f --- /dev/null +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -0,0 +1,372 @@ +/* + * Intel Baytrail SST DSP driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" +#include "sst-baytrail-ipc.h" + +#define SST_BYT_FW_SIGNATURE_SIZE 4 +#define SST_BYT_FW_SIGN "$SST" + +#define SST_BYT_IRAM_OFFSET 0xC0000 +#define SST_BYT_DRAM_OFFSET 0x100000 +#define SST_BYT_SHIM_OFFSET 0x140000 + +enum sst_ram_type { + SST_BYT_IRAM = 1, + SST_BYT_DRAM = 2, + SST_BYT_CACHE = 3, +}; + +struct dma_block_info { + enum sst_ram_type type; /* IRAM/DRAM */ + u32 size; /* Bytes */ + u32 ram_offset; /* Offset in I/DRAM */ + u32 rsvd; /* Reserved field */ +}; + +struct fw_header { + unsigned char signature[SST_BYT_FW_SIGNATURE_SIZE]; + u32 file_size; /* size of fw minus this header */ + u32 modules; /* # of modules */ + u32 file_format; /* version of header format */ + u32 reserved[4]; +}; + +struct sst_byt_fw_module_header { + unsigned char signature[SST_BYT_FW_SIGNATURE_SIZE]; + u32 mod_size; /* size of module */ + u32 blocks; /* # of blocks */ + u32 type; /* codec type, pp lib */ + u32 entry_point; +}; + +static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, + struct sst_byt_fw_module_header *module) +{ + struct dma_block_info *block; + struct sst_module *mod; + struct sst_module_data block_data; + struct sst_module_template template; + int count; + + memset(&template, 0, sizeof(template)); + template.id = module->type; + template.entry = module->entry_point; + template.p.type = SST_MEM_DRAM; + template.p.data_type = SST_DATA_P; + template.s.type = SST_MEM_DRAM; + template.s.data_type = SST_DATA_S; + + mod = sst_module_new(fw, &template, NULL); + if (mod == NULL) + return -ENOMEM; + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + + if (block->size <= 0) { + dev_err(dsp->dev, "block %d size invalid\n", count); + return -EINVAL; + } + + switch (block->type) { + case SST_BYT_IRAM: + block_data.offset = block->ram_offset + + dsp->addr.iram_offset; + block_data.type = SST_MEM_IRAM; + break; + case SST_BYT_DRAM: + block_data.offset = block->ram_offset + + dsp->addr.dram_offset; + block_data.type = SST_MEM_DRAM; + break; + case SST_BYT_CACHE: + block_data.offset = block->ram_offset + + (dsp->addr.fw_ext - dsp->addr.lpe); + block_data.type = SST_MEM_CACHE; + break; + default: + dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n", + block->type, count); + return -EINVAL; + } + + block_data.size = block->size; + block_data.data_type = SST_DATA_M; + block_data.data = (void *)block + sizeof(*block); + + sst_module_insert_fixed_block(mod, &block_data); + + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +static int sst_byt_parse_fw_image(struct sst_fw *sst_fw) +{ + struct fw_header *header; + struct sst_byt_fw_module_header *module; + struct sst_dsp *dsp = sst_fw->dsp; + int ret, count; + + /* Read the header information from the data pointer */ + header = (struct fw_header *)sst_fw->dma_buf; + + /* verify FW */ + if ((strncmp(header->signature, SST_BYT_FW_SIGN, 4) != 0) || + (sst_fw->size != header->file_size + sizeof(*header))) { + /* Invalid FW signature */ + dev_err(dsp->dev, "Invalid FW sign/filesize mismatch\n"); + return -EINVAL; + } + + dev_dbg(dsp->dev, + "header sign=%4s size=0x%x modules=0x%x fmt=0x%x size=%zu\n", + header->signature, header->file_size, header->modules, + header->file_format, sizeof(*header)); + + module = (void *)sst_fw->dma_buf + sizeof(*header); + for (count = 0; count < header->modules; count++) { + /* module */ + ret = sst_byt_parse_module(dsp, sst_fw, module); + if (ret < 0) { + dev_err(dsp->dev, "invalid module %d\n", count); + return ret; + } + module = (void *)module + sizeof(*module) + module->mod_size; + } + + return 0; +} + +static void sst_byt_dump_shim(struct sst_dsp *sst) +{ + int i; + u64 reg; + + for (i = 0; i <= 0xF0; i += 8) { + reg = sst_dsp_shim_read64_unlocked(sst, i); + if (reg) + dev_dbg(sst->dev, "shim 0x%2.2x value 0x%16.16llx\n", + i, reg); + } + + for (i = 0x00; i <= 0xff; i += 4) { + reg = readl(sst->addr.pci_cfg + i); + if (reg) + dev_dbg(sst->dev, "pci 0x%2.2x value 0x%8.8x\n", + i, (u32)reg); + } +} + +static irqreturn_t sst_byt_irq(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + u64 isrx; + irqreturn_t ret = IRQ_NONE; + + spin_lock(&sst->spinlock); + + isrx = sst_dsp_shim_read64_unlocked(sst, SST_ISRX); + if (isrx & SST_ISRX_DONE) { + /* ADSP has processed the message request from IA */ + sst_dsp_shim_update_bits64_unlocked(sst, SST_IPCX, + SST_BYT_IPCX_DONE, 0); + ret = IRQ_WAKE_THREAD; + } + if (isrx & SST_BYT_ISRX_REQUEST) { + /* mask message request from ADSP and do processing later */ + sst_dsp_shim_update_bits64_unlocked(sst, SST_IMRX, + SST_BYT_IMRX_REQUEST, + SST_BYT_IMRX_REQUEST); + ret = IRQ_WAKE_THREAD; + } + + spin_unlock(&sst->spinlock); + + return ret; +} + +static void sst_byt_boot(struct sst_dsp *sst) +{ + int tries = 10; + + /* release stall and wait to unstall */ + sst_dsp_shim_update_bits64(sst, SST_CSR, SST_BYT_CSR_STALL, 0x0); + while (tries--) { + if (!(sst_dsp_shim_read64(sst, SST_CSR) & + SST_BYT_CSR_PWAITMODE)) + break; + msleep(100); + } + if (tries < 0) { + dev_err(sst->dev, "unable to start DSP\n"); + sst_byt_dump_shim(sst); + } +} + +static void sst_byt_reset(struct sst_dsp *sst) +{ + /* put DSP into reset, set reset vector and stall */ + sst_dsp_shim_update_bits64(sst, SST_CSR, + SST_BYT_CSR_RST | SST_BYT_CSR_VECTOR_SEL | SST_BYT_CSR_STALL, + SST_BYT_CSR_RST | SST_BYT_CSR_VECTOR_SEL | SST_BYT_CSR_STALL); + + udelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits64(sst, SST_CSR, SST_BYT_CSR_RST, 0); +} + +struct sst_adsp_memregion { + u32 start; + u32 end; + int blocks; + enum sst_mem_type type; +}; + +/* BYT test stuff */ +static const struct sst_adsp_memregion byt_region[] = { + {0xC0000, 0x100000, 8, SST_MEM_IRAM}, /* I-SRAM - 8 * 32kB */ + {0x100000, 0x140000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ +}; + +static int sst_byt_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + sst->addr.lpe_base = pdata->lpe_base; + sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); + if (!sst->addr.lpe) + return -ENODEV; + + /* ADSP PCI MMIO config space */ + sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size); + if (!sst->addr.pci_cfg) { + iounmap(sst->addr.lpe); + return -ENODEV; + } + + /* SST Extended FW allocation */ + sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size); + if (!sst->addr.fw_ext) { + iounmap(sst->addr.pci_cfg); + iounmap(sst->addr.lpe); + return -ENODEV; + } + + /* SST Shim */ + sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset; + + sst_dsp_mailbox_init(sst, SST_BYT_MAILBOX_OFFSET + 0x204, + SST_BYT_IPC_MAX_PAYLOAD_SIZE, + SST_BYT_MAILBOX_OFFSET, + SST_BYT_IPC_MAX_PAYLOAD_SIZE); + + sst->irq = pdata->irq; + + return 0; +} + +static int sst_byt_init(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + const struct sst_adsp_memregion *region; + struct device *dev; + int ret = -ENODEV, i, j, region_count; + u32 offset, size; + + dev = sst->dev; + + switch (sst->id) { + case SST_DEV_ID_BYT: + region = byt_region; + region_count = ARRAY_SIZE(byt_region); + sst->addr.iram_offset = SST_BYT_IRAM_OFFSET; + sst->addr.dram_offset = SST_BYT_DRAM_OFFSET; + sst->addr.shim_offset = SST_BYT_SHIM_OFFSET; + break; + default: + dev_err(dev, "failed to get mem resources\n"); + return ret; + } + + ret = sst_byt_resource_map(sst, pdata); + if (ret < 0) { + dev_err(dev, "failed to map resources\n"); + return ret; + } + + /* + * save the physical address of extended firmware block in the first + * 4 bytes of the mailbox + */ + memcpy_toio(sst->addr.lpe + SST_BYT_MAILBOX_OFFSET, + &pdata->fw_base, sizeof(u32)); + + ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + /* enable Interrupt from both sides */ + sst_dsp_shim_update_bits64(sst, SST_IMRX, 0x3, 0x0); + sst_dsp_shim_update_bits64(sst, SST_IMRD, 0x3, 0x0); + + /* register DSP memory blocks - ideally we should get this from ACPI */ + for (i = 0; i < region_count; i++) { + offset = region[i].start; + size = (region[i].end - region[i].start) / region[i].blocks; + + /* register individual memory blocks */ + for (j = 0; j < region[i].blocks; j++) { + sst_mem_block_register(sst, offset, size, + region[i].type, NULL, j, sst); + offset += size; + } + } + + return 0; +} + +static void sst_byt_free(struct sst_dsp *sst) +{ + sst_mem_block_unregister_all(sst); + iounmap(sst->addr.lpe); + iounmap(sst->addr.pci_cfg); + iounmap(sst->addr.fw_ext); +} + +struct sst_ops sst_byt_ops = { + .reset = sst_byt_reset, + .boot = sst_byt_boot, + .write = sst_shim32_write, + .read = sst_shim32_read, + .write64 = sst_shim32_write64, + .read64 = sst_shim32_read64, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .irq_handler = sst_byt_irq, + .init = sst_byt_init, + .free = sst_byt_free, + .parse_fw = sst_byt_parse_fw_image, +}; diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index fa2c780..fe8e81aa 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -66,6 +66,7 @@ struct sst_addr { u32 lpe_base; u32 shim_offset; u32 iram_offset; + u32 dram_offset; void __iomem *lpe; void __iomem *shim; void __iomem *pci_cfg; -- cgit v1.1 From f7d01fd6754c1a257af46ec465d946132c7d004d Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Feb 2014 17:26:58 +0200 Subject: ASoC: Intel: Add Intel Baytrail SST DSP IPC support Add support for Baytrail SST DSP IPC. This provides mechanism to communicate with the DSP firmware. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-ipc.c | 866 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 +++ 2 files changed, 935 insertions(+) create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c new file mode 100644 index 0000000..8c91a68 --- /dev/null +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -0,0 +1,866 @@ +/* + * Intel Baytrail SST IPC Support + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sst-baytrail-ipc.h" +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +/* IPC message timeout */ +#define IPC_TIMEOUT_MSECS 300 +#define IPC_BOOT_MSECS 200 + +#define IPC_EMPTY_LIST_SIZE 8 + +/* IPC header bits */ +#define IPC_HEADER_MSG_ID_MASK 0xff +#define IPC_HEADER_MSG_ID(x) ((x) & IPC_HEADER_MSG_ID_MASK) +#define IPC_HEADER_STR_ID_SHIFT 8 +#define IPC_HEADER_STR_ID_MASK 0x1f +#define IPC_HEADER_STR_ID(x) (((x) & 0x1f) << IPC_HEADER_STR_ID_SHIFT) +#define IPC_HEADER_LARGE_SHIFT 13 +#define IPC_HEADER_LARGE(x) (((x) & 0x1) << IPC_HEADER_LARGE_SHIFT) +#define IPC_HEADER_DATA_SHIFT 16 +#define IPC_HEADER_DATA_MASK 0x3fff +#define IPC_HEADER_DATA(x) (((x) & 0x3fff) << IPC_HEADER_DATA_SHIFT) + +/* mask for differentiating between notification and reply message */ +#define IPC_NOTIFICATION (0x1 << 7) + +/* I2L Stream config/control msgs */ +#define IPC_IA_ALLOC_STREAM 0x20 +#define IPC_IA_FREE_STREAM 0x21 +#define IPC_IA_PAUSE_STREAM 0x24 +#define IPC_IA_RESUME_STREAM 0x25 +#define IPC_IA_DROP_STREAM 0x26 +#define IPC_IA_START_STREAM 0x30 + +/* notification messages */ +#define IPC_IA_FW_INIT_CMPLT 0x81 +#define IPC_SST_PERIOD_ELAPSED 0x97 + +/* IPC messages between host and ADSP */ +struct sst_byt_address_info { + u32 addr; + u32 size; +} __packed; + +struct sst_byt_str_type { + u8 codec_type; + u8 str_type; + u8 operation; + u8 protected_str; + u8 time_slots; + u8 reserved; + u16 result; +} __packed; + +struct sst_byt_pcm_params { + u8 num_chan; + u8 pcm_wd_sz; + u8 use_offload_path; + u8 reserved; + u32 sfreq; + u8 channel_map[8]; +} __packed; + +struct sst_byt_frames_info { + u16 num_entries; + u16 rsrvd; + u32 frag_size; + struct sst_byt_address_info ring_buf_info[8]; +} __packed; + +struct sst_byt_alloc_params { + struct sst_byt_str_type str_type; + struct sst_byt_pcm_params pcm_params; + struct sst_byt_frames_info frame_info; +} __packed; + +struct sst_byt_alloc_response { + struct sst_byt_str_type str_type; + u8 reserved[88]; +} __packed; + +struct sst_byt_start_stream_params { + u32 byte_offset; +} __packed; + +struct sst_byt_tstamp { + u64 ring_buffer_counter; + u64 hardware_counter; + u64 frames_decoded; + u64 bytes_decoded; + u64 bytes_copied; + u32 sampling_frequency; + u32 channel_peak[8]; +} __packed; + +/* driver internal IPC message structure */ +struct ipc_message { + struct list_head list; + u64 header; + + /* direction wrt host CPU */ + char tx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE]; + size_t tx_size; + char rx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE]; + size_t rx_size; + + wait_queue_head_t waitq; + bool complete; + bool wait; + int errno; +}; + +struct sst_byt_stream; +struct sst_byt; + +/* stream infomation */ +struct sst_byt_stream { + struct list_head node; + + /* configuration */ + struct sst_byt_alloc_params request; + struct sst_byt_alloc_response reply; + + /* runtime info */ + struct sst_byt *byt; + int str_id; + bool commited; + bool running; + + /* driver callback */ + u32 (*notify_position)(struct sst_byt_stream *stream, void *data); + void *pdata; +}; + +/* SST Baytrail IPC data */ +struct sst_byt { + struct device *dev; + struct sst_dsp *dsp; + + /* stream */ + struct list_head stream_list; + + /* boot */ + wait_queue_head_t boot_wait; + bool boot_complete; + + /* IPC messaging */ + struct list_head tx_list; + struct list_head rx_list; + struct list_head empty_list; + wait_queue_head_t wait_txq; + struct task_struct *tx_thread; + struct kthread_worker kworker; + struct kthread_work kwork; + struct ipc_message *msg; +}; + +static inline u64 sst_byt_header(int msg_id, int data, bool large, int str_id) +{ + u64 header; + + header = IPC_HEADER_MSG_ID(msg_id) | + IPC_HEADER_STR_ID(str_id) | + IPC_HEADER_LARGE(large) | + IPC_HEADER_DATA(data) | + SST_BYT_IPCX_BUSY; + + return header; +} + +static inline u16 sst_byt_header_msg_id(u64 header) +{ + return header & IPC_HEADER_MSG_ID_MASK; +} + +static inline u8 sst_byt_header_str_id(u64 header) +{ + return (header >> IPC_HEADER_STR_ID_SHIFT) & IPC_HEADER_STR_ID_MASK; +} + +static inline u16 sst_byt_header_data(u64 header) +{ + return (header >> IPC_HEADER_DATA_SHIFT) & IPC_HEADER_DATA_MASK; +} + +static struct sst_byt_stream *sst_byt_get_stream(struct sst_byt *byt, + int stream_id) +{ + struct sst_byt_stream *stream; + + list_for_each_entry(stream, &byt->stream_list, node) { + if (stream->str_id == stream_id) + return stream; + } + + return NULL; +} + +static void sst_byt_ipc_shim_dbg(struct sst_byt *byt, const char *text) +{ + struct sst_dsp *sst = byt->dsp; + u64 isr, ipcd, imrx, ipcx; + + ipcx = sst_dsp_shim_read64_unlocked(sst, SST_IPCX); + isr = sst_dsp_shim_read64_unlocked(sst, SST_ISRX); + ipcd = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); + imrx = sst_dsp_shim_read64_unlocked(sst, SST_IMRX); + + dev_err(byt->dev, + "ipc: --%s-- ipcx 0x%llx isr 0x%llx ipcd 0x%llx imrx 0x%llx\n", + text, ipcx, isr, ipcd, imrx); +} + +/* locks held by caller */ +static struct ipc_message *sst_byt_msg_get_empty(struct sst_byt *byt) +{ + struct ipc_message *msg = NULL; + + if (!list_empty(&byt->empty_list)) { + msg = list_first_entry(&byt->empty_list, + struct ipc_message, list); + list_del(&msg->list); + } + + return msg; +} + +static void sst_byt_ipc_tx_msgs(struct kthread_work *work) +{ + struct sst_byt *byt = + container_of(work, struct sst_byt, kwork); + struct ipc_message *msg; + u64 ipcx; + unsigned long flags; + + spin_lock_irqsave(&byt->dsp->spinlock, flags); + if (list_empty(&byt->tx_list)) { + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + return; + } + + /* if the DSP is busy we will TX messages after IRQ */ + ipcx = sst_dsp_shim_read64_unlocked(byt->dsp, SST_IPCX); + if (ipcx & SST_BYT_IPCX_BUSY) { + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + return; + } + + msg = list_first_entry(&byt->tx_list, struct ipc_message, list); + + list_move(&msg->list, &byt->rx_list); + + /* send the message */ + if (msg->header & IPC_HEADER_LARGE(true)) + sst_dsp_outbox_write(byt->dsp, msg->tx_data, msg->tx_size); + sst_dsp_shim_write64_unlocked(byt->dsp, SST_IPCX, msg->header); + + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); +} + +static inline void sst_byt_tx_msg_reply_complete(struct sst_byt *byt, + struct ipc_message *msg) +{ + msg->complete = true; + + if (!msg->wait) + list_add_tail(&msg->list, &byt->empty_list); + else + wake_up(&msg->waitq); +} + +static int sst_byt_tx_wait_done(struct sst_byt *byt, struct ipc_message *msg, + void *rx_data) +{ + unsigned long flags; + int ret; + + /* wait for DSP completion */ + ret = wait_event_timeout(msg->waitq, msg->complete, + msecs_to_jiffies(IPC_TIMEOUT_MSECS)); + + spin_lock_irqsave(&byt->dsp->spinlock, flags); + if (ret == 0) { + list_del(&msg->list); + sst_byt_ipc_shim_dbg(byt, "message timeout"); + + ret = -ETIMEDOUT; + } else { + + /* copy the data returned from DSP */ + if (msg->rx_size) + memcpy(rx_data, msg->rx_data, msg->rx_size); + ret = msg->errno; + } + + list_add_tail(&msg->list, &byt->empty_list); + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + return ret; +} + +static int sst_byt_ipc_tx_message(struct sst_byt *byt, u64 header, + void *tx_data, size_t tx_bytes, + void *rx_data, size_t rx_bytes, int wait) +{ + unsigned long flags; + struct ipc_message *msg; + + spin_lock_irqsave(&byt->dsp->spinlock, flags); + + msg = sst_byt_msg_get_empty(byt); + if (msg == NULL) { + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + return -EBUSY; + } + + msg->header = header; + msg->tx_size = tx_bytes; + msg->rx_size = rx_bytes; + msg->wait = wait; + msg->errno = 0; + msg->complete = false; + + if (tx_bytes) { + /* msg content = lower 32-bit of the header + data */ + *(u32 *)msg->tx_data = (u32)(header & (u32)-1); + memcpy(msg->tx_data + sizeof(u32), tx_data, tx_bytes); + msg->tx_size += sizeof(u32); + } + + list_add_tail(&msg->list, &byt->tx_list); + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + + queue_kthread_work(&byt->kworker, &byt->kwork); + + if (wait) + return sst_byt_tx_wait_done(byt, msg, rx_data); + else + return 0; +} + +static inline int sst_byt_ipc_tx_msg_wait(struct sst_byt *byt, u64 header, + void *tx_data, size_t tx_bytes, + void *rx_data, size_t rx_bytes) +{ + return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes, + rx_data, rx_bytes, 1); +} + +static inline int sst_byt_ipc_tx_msg_nowait(struct sst_byt *byt, u64 header, + void *tx_data, size_t tx_bytes) +{ + return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes, + NULL, 0, 0); +} + +static struct ipc_message *sst_byt_reply_find_msg(struct sst_byt *byt, + u64 header) +{ + struct ipc_message *msg = NULL, *_msg; + u64 mask; + + /* match reply to message sent based on msg and stream IDs */ + mask = IPC_HEADER_MSG_ID_MASK | + IPC_HEADER_STR_ID_MASK << IPC_HEADER_STR_ID_SHIFT; + header &= mask; + + if (list_empty(&byt->rx_list)) { + dev_err(byt->dev, + "ipc: rx list is empty but received 0x%llx\n", header); + goto out; + } + + list_for_each_entry(_msg, &byt->rx_list, list) { + if ((_msg->header & mask) == header) { + msg = _msg; + break; + } + } + +out: + return msg; +} + +static void sst_byt_stream_update(struct sst_byt *byt, struct ipc_message *msg) +{ + struct sst_byt_stream *stream; + u64 header = msg->header; + u8 stream_id = sst_byt_header_str_id(header); + u8 stream_msg = sst_byt_header_msg_id(header); + + stream = sst_byt_get_stream(byt, stream_id); + if (stream == NULL) + return; + + switch (stream_msg) { + case IPC_IA_DROP_STREAM: + case IPC_IA_PAUSE_STREAM: + case IPC_IA_FREE_STREAM: + stream->running = false; + break; + case IPC_IA_START_STREAM: + case IPC_IA_RESUME_STREAM: + stream->running = true; + break; + } +} + +static int sst_byt_process_reply(struct sst_byt *byt, u64 header) +{ + struct ipc_message *msg; + + msg = sst_byt_reply_find_msg(byt, header); + if (msg == NULL) + return 1; + + if (header & IPC_HEADER_LARGE(true)) { + msg->rx_size = sst_byt_header_data(header); + sst_dsp_inbox_read(byt->dsp, msg->rx_data, msg->rx_size); + } + + /* update any stream states */ + sst_byt_stream_update(byt, msg); + + list_del(&msg->list); + /* wake up */ + sst_byt_tx_msg_reply_complete(byt, msg); + + return 1; +} + +static void sst_byt_fw_ready(struct sst_byt *byt, u64 header) +{ + dev_dbg(byt->dev, "ipc: DSP is ready 0x%llX\n", header); + + byt->boot_complete = true; + wake_up(&byt->boot_wait); +} + +static int sst_byt_process_notification(struct sst_byt *byt, + unsigned long *flags) +{ + struct sst_dsp *sst = byt->dsp; + struct sst_byt_stream *stream; + u64 header; + u8 msg_id, stream_id; + int handled = 1; + + header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); + msg_id = sst_byt_header_msg_id(header); + + switch (msg_id) { + case IPC_SST_PERIOD_ELAPSED: + stream_id = sst_byt_header_str_id(header); + stream = sst_byt_get_stream(byt, stream_id); + if (stream && stream->running && stream->notify_position) { + spin_unlock_irqrestore(&sst->spinlock, *flags); + stream->notify_position(stream, stream->pdata); + spin_lock_irqsave(&sst->spinlock, *flags); + } + break; + case IPC_IA_FW_INIT_CMPLT: + sst_byt_fw_ready(byt, header); + break; + } + + return handled; +} + +static irqreturn_t sst_byt_irq_thread(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + struct sst_byt *byt = sst_dsp_get_thread_context(sst); + u64 header; + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + + header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); + if (header & SST_BYT_IPCD_BUSY) { + if (header & IPC_NOTIFICATION) { + /* message from ADSP */ + sst_byt_process_notification(byt, &flags); + } else { + /* reply from ADSP */ + sst_byt_process_reply(byt, header); + } + /* + * clear IPCD BUSY bit and set DONE bit. Tell DSP we have + * processed the message and can accept new. Clear data part + * of the header + */ + sst_dsp_shim_update_bits64_unlocked(sst, SST_IPCD, + SST_BYT_IPCD_DONE | SST_BYT_IPCD_BUSY | + IPC_HEADER_DATA(IPC_HEADER_DATA_MASK), + SST_BYT_IPCD_DONE); + /* unmask message request interrupts */ + sst_dsp_shim_update_bits64_unlocked(sst, SST_IMRX, + SST_BYT_IMRX_REQUEST, 0); + } + + spin_unlock_irqrestore(&sst->spinlock, flags); + + /* continue to send any remaining messages... */ + queue_kthread_work(&byt->kworker, &byt->kwork); + + return IRQ_HANDLED; +} + +/* stream API */ +struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id, + u32 (*notify_position)(struct sst_byt_stream *stream, void *data), + void *data) +{ + struct sst_byt_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (stream == NULL) + return NULL; + + list_add(&stream->node, &byt->stream_list); + stream->notify_position = notify_position; + stream->pdata = data; + stream->byt = byt; + stream->str_id = id; + + return stream; +} + +int sst_byt_stream_set_bits(struct sst_byt *byt, struct sst_byt_stream *stream, + int bits) +{ + stream->request.pcm_params.pcm_wd_sz = bits; + return 0; +} + +int sst_byt_stream_set_channels(struct sst_byt *byt, + struct sst_byt_stream *stream, u8 channels) +{ + stream->request.pcm_params.num_chan = channels; + return 0; +} + +int sst_byt_stream_set_rate(struct sst_byt *byt, struct sst_byt_stream *stream, + unsigned int rate) +{ + stream->request.pcm_params.sfreq = rate; + return 0; +} + +/* stream sonfiguration */ +int sst_byt_stream_type(struct sst_byt *byt, struct sst_byt_stream *stream, + int codec_type, int stream_type, int operation) +{ + stream->request.str_type.codec_type = codec_type; + stream->request.str_type.str_type = stream_type; + stream->request.str_type.operation = operation; + stream->request.str_type.time_slots = 0xc; + + return 0; +} + +int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream, + uint32_t buffer_addr, uint32_t buffer_size) +{ + stream->request.frame_info.num_entries = 1; + stream->request.frame_info.ring_buf_info[0].addr = buffer_addr; + stream->request.frame_info.ring_buf_info[0].size = buffer_size; + /* calculate bytes per 4 ms fragment */ + stream->request.frame_info.frag_size = + stream->request.pcm_params.sfreq * + stream->request.pcm_params.num_chan * + stream->request.pcm_params.pcm_wd_sz / 8 * + 4 / 1000; + return 0; +} + +int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + struct sst_byt_alloc_params *str_req = &stream->request; + struct sst_byt_alloc_response *reply = &stream->reply; + u64 header; + int ret; + + header = sst_byt_header(IPC_IA_ALLOC_STREAM, + sizeof(*str_req) + sizeof(u32), + true, stream->str_id); + ret = sst_byt_ipc_tx_msg_wait(byt, header, str_req, sizeof(*str_req), + reply, sizeof(*reply)); + if (ret < 0) { + dev_err(byt->dev, "ipc: error stream commit failed\n"); + return ret; + } + + stream->commited = true; + + return 0; +} + +int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + u64 header; + int ret = 0; + + if (!stream->commited) + goto out; + + header = sst_byt_header(IPC_IA_FREE_STREAM, 0, false, stream->str_id); + ret = sst_byt_ipc_tx_msg_wait(byt, header, NULL, 0, NULL, 0); + if (ret < 0) { + dev_err(byt->dev, "ipc: free stream %d failed\n", + stream->str_id); + return -EAGAIN; + } + + stream->commited = false; +out: + list_del(&stream->node); + kfree(stream); + + return ret; +} + +static int sst_byt_stream_operations(struct sst_byt *byt, int type, + int stream_id, int wait) +{ + struct sst_byt_start_stream_params start_stream; + u64 header; + void *tx_msg = NULL; + size_t size = 0; + + if (type != IPC_IA_START_STREAM) { + header = sst_byt_header(type, 0, false, stream_id); + } else { + start_stream.byte_offset = 0; + header = sst_byt_header(IPC_IA_START_STREAM, + sizeof(start_stream) + sizeof(u32), + true, stream_id); + tx_msg = &start_stream; + size = sizeof(start_stream); + } + + if (wait) + return sst_byt_ipc_tx_msg_wait(byt, header, + tx_msg, size, NULL, 0); + else + return sst_byt_ipc_tx_msg_nowait(byt, header, tx_msg, size); +} + +/* stream ALSA trigger operations */ +int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + int ret; + + ret = sst_byt_stream_operations(byt, IPC_IA_START_STREAM, + stream->str_id, 0); + if (ret < 0) + dev_err(byt->dev, "ipc: error failed to start stream %d\n", + stream->str_id); + + return ret; +} + +int sst_byt_stream_stop(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + int ret; + + /* don't stop streams that are not commited */ + if (!stream->commited) + return 0; + + ret = sst_byt_stream_operations(byt, IPC_IA_DROP_STREAM, + stream->str_id, 0); + if (ret < 0) + dev_err(byt->dev, "ipc: error failed to stop stream %d\n", + stream->str_id); + return ret; +} + +int sst_byt_stream_pause(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + int ret; + + ret = sst_byt_stream_operations(byt, IPC_IA_PAUSE_STREAM, + stream->str_id, 0); + if (ret < 0) + dev_err(byt->dev, "ipc: error failed to pause stream %d\n", + stream->str_id); + + return ret; +} + +int sst_byt_stream_resume(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + int ret; + + ret = sst_byt_stream_operations(byt, IPC_IA_RESUME_STREAM, + stream->str_id, 0); + if (ret < 0) + dev_err(byt->dev, "ipc: error failed to resume stream %d\n", + stream->str_id); + + return ret; +} + +int sst_byt_get_dsp_position(struct sst_byt *byt, + struct sst_byt_stream *stream, int buffer_size) +{ + struct sst_dsp *sst = byt->dsp; + struct sst_byt_tstamp fw_tstamp; + u8 str_id = stream->str_id; + u32 tstamp_offset; + + tstamp_offset = SST_BYT_TIMESTAMP_OFFSET + str_id * sizeof(fw_tstamp); + memcpy_fromio(&fw_tstamp, + sst->addr.lpe + tstamp_offset, sizeof(fw_tstamp)); + + return do_div(fw_tstamp.ring_buffer_counter, buffer_size); +} + +static int msg_empty_list_init(struct sst_byt *byt) +{ + struct ipc_message *msg; + int i; + + byt->msg = kzalloc(sizeof(*msg) * IPC_EMPTY_LIST_SIZE, GFP_KERNEL); + if (byt->msg == NULL) + return -ENOMEM; + + for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + init_waitqueue_head(&byt->msg[i].waitq); + list_add(&byt->msg[i].list, &byt->empty_list); + } + + return 0; +} + +struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt) +{ + return byt->dsp; +} + +static struct sst_dsp_device byt_dev = { + .thread = sst_byt_irq_thread, + .ops = &sst_byt_ops, +}; + +int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) +{ + struct sst_byt *byt; + struct sst_fw *byt_sst_fw; + int err; + + dev_dbg(dev, "initialising Byt DSP IPC\n"); + + byt = devm_kzalloc(dev, sizeof(*byt), GFP_KERNEL); + if (byt == NULL) + return -ENOMEM; + + byt->dev = dev; + INIT_LIST_HEAD(&byt->stream_list); + INIT_LIST_HEAD(&byt->tx_list); + INIT_LIST_HEAD(&byt->rx_list); + INIT_LIST_HEAD(&byt->empty_list); + init_waitqueue_head(&byt->boot_wait); + init_waitqueue_head(&byt->wait_txq); + + err = msg_empty_list_init(byt); + if (err < 0) + goto list_err; + + /* start the IPC message thread */ + init_kthread_worker(&byt->kworker); + byt->tx_thread = kthread_run(kthread_worker_fn, + &byt->kworker, + dev_name(byt->dev)); + if (IS_ERR(byt->tx_thread)) { + err = PTR_ERR(byt->tx_thread); + dev_err(byt->dev, "error failed to create message TX task\n"); + goto list_err; + } + init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs); + + byt_dev.thread_context = byt; + + /* init SST shim */ + byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); + if (byt->dsp == NULL) { + err = -ENODEV; + goto list_err; + } + + /* keep the DSP in reset state for base FW loading */ + sst_dsp_reset(byt->dsp); + + byt_sst_fw = sst_fw_new(byt->dsp, pdata->fw, byt); + if (byt_sst_fw == NULL) { + err = -ENODEV; + dev_err(dev, "error: failed to load firmware\n"); + goto fw_err; + } + + /* wait for DSP boot completion */ + sst_dsp_boot(byt->dsp); + err = wait_event_timeout(byt->boot_wait, byt->boot_complete, + msecs_to_jiffies(IPC_BOOT_MSECS)); + if (err == 0) { + err = -EIO; + dev_err(byt->dev, "ipc: error DSP boot timeout\n"); + goto boot_err; + } + + pdata->dsp = byt; + + return 0; + +boot_err: + sst_dsp_reset(byt->dsp); + sst_fw_free(byt_sst_fw); +fw_err: + sst_dsp_free(byt->dsp); + kfree(byt->msg); +list_err: + kfree(byt); + return err; +} +EXPORT_SYMBOL_GPL(sst_byt_dsp_init); + +void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata) +{ + struct sst_byt *byt = pdata->dsp; + + sst_dsp_reset(byt->dsp); + sst_fw_free_all(byt->dsp); + sst_dsp_free(byt->dsp); + kfree(byt->msg); +} +EXPORT_SYMBOL_GPL(sst_byt_dsp_free); diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h new file mode 100644 index 0000000..f172b64 --- /dev/null +++ b/sound/soc/intel/sst-baytrail-ipc.h @@ -0,0 +1,69 @@ +/* + * Intel Baytrail SST IPC Support + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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_BYT_IPC_H +#define __SST_BYT_IPC_H + +#include + +struct sst_byt; +struct sst_byt_stream; +struct sst_pdata; +extern struct sst_ops sst_byt_ops; + + +#define SST_BYT_MAILBOX_OFFSET 0x144000 +#define SST_BYT_TIMESTAMP_OFFSET (SST_BYT_MAILBOX_OFFSET + 0x800) + +/** + * Upfront defined maximum message size that is + * expected by the in/out communication pipes in FW. + */ +#define SST_BYT_IPC_MAX_PAYLOAD_SIZE 200 + +/* stream API */ +struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id, + uint32_t (*get_write_position)(struct sst_byt_stream *stream, + void *data), + void *data); + +/* stream configuration */ +int sst_byt_stream_set_bits(struct sst_byt *byt, struct sst_byt_stream *stream, + int bits); +int sst_byt_stream_set_channels(struct sst_byt *byt, + struct sst_byt_stream *stream, u8 channels); +int sst_byt_stream_set_rate(struct sst_byt *byt, struct sst_byt_stream *stream, + unsigned int rate); +int sst_byt_stream_type(struct sst_byt *byt, struct sst_byt_stream *stream, + int codec_type, int stream_type, int operation); +int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream, + uint32_t buffer_addr, uint32_t buffer_size); +int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream); +int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream); + +/* stream ALSA trigger operations */ +int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream); +int sst_byt_stream_stop(struct sst_byt *byt, struct sst_byt_stream *stream); +int sst_byt_stream_pause(struct sst_byt *byt, struct sst_byt_stream *stream); +int sst_byt_stream_resume(struct sst_byt *byt, struct sst_byt_stream *stream); + +int sst_byt_get_dsp_position(struct sst_byt *byt, + struct sst_byt_stream *stream, int buffer_size); + +/* init */ +int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata); +void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata); +struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt); + +#endif -- cgit v1.1 From aef1311a7d7e77408611cdf143d32bb1709527b6 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Feb 2014 17:26:59 +0200 Subject: ASoC: Intel: Add Intel Baytrail SST PCM platform driver This adds the Baytrail SST DSP PCM platform driver. It registers itself with the ALSA SoC layer and uses Intel Baytrail SST DSP IPC for DSP control. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-pcm.c | 422 +++++++++++++++++++++++++++++++++++++ 1 file changed, 422 insertions(+) create mode 100644 sound/soc/intel/sst-baytrail-pcm.c (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c new file mode 100644 index 0000000..6d101f3 --- /dev/null +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -0,0 +1,422 @@ +/* + * Intel Baytrail SST PCM Support + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include "sst-baytrail-ipc.h" +#include "sst-dsp-priv.h" +#include "sst-dsp.h" + +#define BYT_PCM_COUNT 2 + +static const struct snd_pcm_hardware sst_byt_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FORMAT_S24_LE, + .period_bytes_min = 384, + .period_bytes_max = 48000, + .periods_min = 2, + .periods_max = 250, + .buffer_bytes_max = 96000, +}; + +/* private data for each PCM DSP stream */ +struct sst_byt_pcm_data { + struct sst_byt_stream *stream; + struct snd_pcm_substream *substream; + struct mutex mutex; +}; + +/* private data for the driver */ +struct sst_byt_priv_data { + /* runtime DSP */ + struct sst_byt *byt; + + /* DAI data */ + struct sst_byt_pcm_data pcm[BYT_PCM_COUNT]; +}; + +/* this may get called several times by oss emulation */ +static int sst_byt_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + u32 rate, bits; + u8 channels; + int ret, playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + dev_dbg(rtd->dev, "PCM: hw_params, pcm_data %p\n", pcm_data); + + ret = sst_byt_stream_type(byt, pcm_data->stream, + 1, 1, !playback); + if (ret < 0) { + dev_err(rtd->dev, "failed to set stream format %d\n", ret); + return ret; + } + + rate = params_rate(params); + ret = sst_byt_stream_set_rate(byt, pcm_data->stream, rate); + if (ret < 0) { + dev_err(rtd->dev, "could not set rate %d\n", rate); + return ret; + } + + bits = snd_pcm_format_width(params_format(params)); + ret = sst_byt_stream_set_bits(byt, pcm_data->stream, bits); + if (ret < 0) { + dev_err(rtd->dev, "could not set formats %d\n", + params_rate(params)); + return ret; + } + + channels = (u8)(params_channels(params) & 0xF); + ret = sst_byt_stream_set_channels(byt, pcm_data->stream, channels); + if (ret < 0) { + dev_err(rtd->dev, "could not set channels %d\n", + params_rate(params)); + return ret; + } + + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + + ret = sst_byt_stream_buffer(byt, pcm_data->stream, + substream->dma_buffer.addr, + params_buffer_bytes(params)); + if (ret < 0) { + dev_err(rtd->dev, "PCM: failed to set DMA buffer %d\n", ret); + return ret; + } + + ret = sst_byt_stream_commit(byt, pcm_data->stream); + if (ret < 0) { + dev_err(rtd->dev, "PCM: failed stream commit %d\n", ret); + return ret; + } + + return 0; +} + +static int sst_byt_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->dev, "PCM: hw_free\n"); + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + + dev_dbg(rtd->dev, "PCM: trigger %d\n", cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + sst_byt_stream_start(byt, pcm_data->stream); + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_byt_stream_resume(byt, pcm_data->stream); + break; + case SNDRV_PCM_TRIGGER_STOP: + sst_byt_stream_stop(byt, pcm_data->stream); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_byt_stream_pause(byt, pcm_data->stream); + break; + default: + break; + } + + return 0; +} + +static u32 byt_notify_pointer(struct sst_byt_stream *stream, void *data) +{ + struct sst_byt_pcm_data *pcm_data = data; + struct snd_pcm_substream *substream = pcm_data->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + u32 pos; + + pos = frames_to_bytes(runtime, + (runtime->control->appl_ptr % + runtime->buffer_size)); + + dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + + snd_pcm_period_elapsed(substream); + return pos; +} + +static snd_pcm_uframes_t sst_byt_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + snd_pcm_uframes_t offset; + int pos; + + pos = sst_byt_get_dsp_position(byt, pcm_data->stream, + snd_pcm_lib_buffer_bytes(substream)); + offset = bytes_to_frames(runtime, pos); + + dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n", + frames_to_bytes(runtime, (u32)offset)); + return offset; +} + +static int sst_byt_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + + dev_dbg(rtd->dev, "PCM: open\n"); + + pcm_data = &pdata->pcm[rtd->cpu_dai->id]; + mutex_lock(&pcm_data->mutex); + + snd_soc_pcm_set_drvdata(rtd, pcm_data); + pcm_data->substream = substream; + + snd_soc_set_runtime_hwparams(substream, &sst_byt_pcm_hardware); + + pcm_data->stream = sst_byt_stream_new(byt, rtd->cpu_dai->id + 1, + byt_notify_pointer, pcm_data); + if (pcm_data->stream == NULL) { + dev_err(rtd->dev, "failed to create stream\n"); + mutex_unlock(&pcm_data->mutex); + return -EINVAL; + } + + mutex_unlock(&pcm_data->mutex); + return 0; +} + +static int sst_byt_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + int ret; + + dev_dbg(rtd->dev, "PCM: close\n"); + + mutex_lock(&pcm_data->mutex); + ret = sst_byt_stream_free(byt, pcm_data->stream); + if (ret < 0) { + dev_dbg(rtd->dev, "Free stream fail\n"); + goto out; + } + pcm_data->stream = NULL; + +out: + mutex_unlock(&pcm_data->mutex); + return ret; +} + +static int sst_byt_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->dev, "PCM: mmap\n"); + return snd_pcm_lib_default_mmap(substream, vma); +} + +static struct snd_pcm_ops sst_byt_pcm_ops = { + .open = sst_byt_pcm_open, + .close = sst_byt_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sst_byt_pcm_hw_params, + .hw_free = sst_byt_pcm_hw_free, + .trigger = sst_byt_pcm_trigger, + .pointer = sst_byt_pcm_pointer, + .mmap = sst_byt_pcm_mmap, +}; + +static void sst_byt_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int sst_byt_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + size_t size; + int ret = 0; + + ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + size = sst_byt_pcm_hardware.buffer_bytes_max; + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + rtd->card->dev, + size, size); + if (ret) { + dev_err(rtd->dev, "dma buffer allocation failed %d\n", + ret); + return ret; + } + } + + return ret; +} + +static struct snd_soc_dai_driver byt_dais[] = { + { + .name = "Front-cpu-dai", + .playback = { + .stream_name = "System Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S16_LE, + }, + }, + { + .name = "Mic1-cpu-dai", + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, +}; + +static int sst_byt_pcm_probe(struct snd_soc_platform *platform) +{ + struct sst_pdata *plat_data = dev_get_platdata(platform->dev); + struct sst_byt_priv_data *priv_data; + int i; + + if (!plat_data) + return -ENODEV; + + priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), + GFP_KERNEL); + priv_data->byt = plat_data->dsp; + snd_soc_platform_set_drvdata(platform, priv_data); + + for (i = 0; i < ARRAY_SIZE(byt_dais); i++) + mutex_init(&priv_data->pcm[i].mutex); + + return 0; +} + +static int sst_byt_pcm_remove(struct snd_soc_platform *platform) +{ + return 0; +} + +static struct snd_soc_platform_driver byt_soc_platform = { + .probe = sst_byt_pcm_probe, + .remove = sst_byt_pcm_remove, + .ops = &sst_byt_pcm_ops, + .pcm_new = sst_byt_pcm_new, + .pcm_free = sst_byt_pcm_free, +}; + +static const struct snd_soc_component_driver byt_dai_component = { + .name = "byt-dai", +}; + +static int sst_byt_pcm_dev_probe(struct platform_device *pdev) +{ + struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + int ret; + + ret = sst_byt_dsp_init(&pdev->dev, sst_pdata); + if (ret < 0) + return -ENODEV; + + ret = snd_soc_register_platform(&pdev->dev, &byt_soc_platform); + if (ret < 0) + goto err_plat; + + ret = snd_soc_register_component(&pdev->dev, &byt_dai_component, + byt_dais, ARRAY_SIZE(byt_dais)); + if (ret < 0) + goto err_comp; + + return 0; + +err_comp: + snd_soc_unregister_platform(&pdev->dev); +err_plat: + sst_byt_dsp_free(&pdev->dev, sst_pdata); + return ret; +} + +static int sst_byt_pcm_dev_remove(struct platform_device *pdev) +{ + struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + sst_byt_dsp_free(&pdev->dev, sst_pdata); + + return 0; +} + +static struct platform_driver sst_byt_pcm_driver = { + .driver = { + .name = "baytrail-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = sst_byt_pcm_dev_probe, + .remove = sst_byt_pcm_dev_remove, +}; +module_platform_driver(sst_byt_pcm_driver); + +MODULE_AUTHOR("Jarkko Nikula"); +MODULE_DESCRIPTION("Baytrail PCM"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:baytrail-pcm-audio"); -- cgit v1.1 From 6439c8ad1edcca1051936de9f624c5ab9ebe03f7 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Feb 2014 17:27:00 +0200 Subject: ASoC: Intel: Add machine driver for Baytrail SST with RT5640 codec Add machine driver for Baytrail SST DSP platform with RT5640 audio codec. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 194 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 194 insertions(+) create mode 100644 sound/soc/intel/byt-rt5640.c (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c new file mode 100644 index 0000000..038547b --- /dev/null +++ b/sound/soc/intel/byt-rt5640.c @@ -0,0 +1,194 @@ +/* + * Intel Baytrail SST RT5640 machine driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../codecs/rt5640.h" + +#include "sst-dsp.h" + +static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { + {"IN2P", NULL, "Headset Mic"}, + {"IN2N", NULL, "Headset Mic"}, + {"DMIC1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, +}; + +static const struct snd_kcontrol_new byt_rt5640_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int byt_rt5640_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; + unsigned int fmt; + int ret; + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(codec_dai->dev, + "can't set codec DAI configuration %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set codec clock %d\n", ret); + return ret; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1, + params_rate(params) * 64, + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret); + return ret; + } + return 0; +} + +static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = runtime->card; + + card->dapm.idle_bias_off = true; + + ret = snd_soc_add_card_controls(card, byt_rt5640_controls, + ARRAY_SIZE(byt_rt5640_controls)); + if (ret) { + dev_err(card->dev, "unable to add card controls\n"); + return ret; + } + + snd_soc_dapm_ignore_suspend(dapm, "HPOL"); + snd_soc_dapm_ignore_suspend(dapm, "HPOR"); + + snd_soc_dapm_ignore_suspend(dapm, "SPOLP"); + snd_soc_dapm_ignore_suspend(dapm, "SPOLN"); + snd_soc_dapm_ignore_suspend(dapm, "SPORP"); + snd_soc_dapm_ignore_suspend(dapm, "SPORN"); + + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Headphone"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Int Mic"); + + snd_soc_dapm_sync(dapm); + return ret; +} + +static struct snd_soc_ops byt_rt5640_ops = { + .hw_params = byt_rt5640_hw_params, +}; + +static struct snd_soc_dai_link byt_rt5640_dais[] = { + { + .name = "Baytrail Audio", + .stream_name = "Audio", + .cpu_dai_name = "Front-cpu-dai", + .codec_dai_name = "rt5640-aif1", + .codec_name = "i2c-10EC5640:00", + .platform_name = "baytrail-pcm-audio", + .init = byt_rt5640_init, + .ignore_suspend = 1, + .ops = &byt_rt5640_ops, + }, + { + .name = "Baytrail Voice", + .stream_name = "Voice", + .cpu_dai_name = "Mic1-cpu-dai", + .codec_dai_name = "rt5640-aif1", + .codec_name = "i2c-10EC5640:00", + .platform_name = "baytrail-pcm-audio", + .init = NULL, + .ignore_suspend = 1, + .ops = &byt_rt5640_ops, + }, +}; + +static struct snd_soc_card byt_rt5640_card = { + .name = "byt-rt5640", + .dai_link = byt_rt5640_dais, + .num_links = ARRAY_SIZE(byt_rt5640_dais), + .dapm_widgets = byt_rt5640_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets), + .dapm_routes = byt_rt5640_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map), +}; + +static int byt_rt5640_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &byt_rt5640_card; + struct device *dev = &pdev->dev; + + card->dev = &pdev->dev; + dev_set_drvdata(dev, card); + return snd_soc_register_card(card); +} + +static int byt_rt5640_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver byt_rt5640_audio = { + .probe = byt_rt5640_probe, + .remove = byt_rt5640_remove, + .driver = { + .name = "byt-rt5640", + .owner = THIS_MODULE, + }, +}; +module_platform_driver(byt_rt5640_audio) + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); +MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:byt-rt5640"); -- cgit v1.1 From e0298612147f668b55644e230340237b7c1a991d Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Feb 2014 17:27:01 +0200 Subject: ASoC: Intel: Add Baytrail SST and byt-rt5640 machine driver probing Add Baytrail SST descriptor with the byt-rt5640 machine driver to sst-acpi loader. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-acpi.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index c7e36c9..0bb4316 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -243,9 +243,26 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { .dma_size = SST_LPT_DSP_DMA_SIZE, }; +static struct sst_acpi_mach baytrail_machines[] = { + { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, + {} +}; + +static struct sst_acpi_desc sst_acpi_baytrail_desc = { + .drv_name = "baytrail-pcm-audio", + .machines = baytrail_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_fw_base = 2, + .irqindex_host_ipc = 5, + .sst_id = SST_DEV_ID_BYT, + .resindex_dma_base = -1, +}; + static struct acpi_device_id sst_acpi_match[] = { { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, + { "80860F28", (unsigned long)&sst_acpi_baytrail_desc }, { } }; MODULE_DEVICE_TABLE(acpi, sst_acpi_match); -- cgit v1.1 From 20df8d03a7a6d71cc8c1fe54cf2c69b2d416423f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Feb 2014 17:27:02 +0200 Subject: ASoC: Intel: Add build support for Baytrail SST Enable build support for Baytrail SST DSP platform and byt-rt5640 machine drivers. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 14 +++++++++++++- sound/soc/intel/Makefile | 5 +++++ 2 files changed, 18 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ce5a692..274af16 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -27,6 +27,9 @@ config SND_SOC_INTEL_SST_ACPI config SND_SOC_INTEL_HASWELL tristate +config SND_SOC_INTEL_BAYTRAIL + tristate + config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS @@ -36,4 +39,13 @@ config SND_SOC_INTEL_HASWELL_MACH This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell Ultrabook platforms. Say Y if you have such a device - If unsure select "N". \ No newline at end of file + If unsure select "N". + +config SND_SOC_INTEL_BYT_RT5640_MACH + tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_BAYTRAIL + select SND_SOC_RT5640 + help + This adds audio driver for Intel Baytrail platform based boards + with the RT5640 audio codec. diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 1c18815..edeb79ae 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -14,10 +14,15 @@ obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o # Platform Support snd-soc-sst-haswell-pcm-objs := \ sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o +snd-soc-sst-baytrail-pcm-objs := \ + sst-baytrail-ipc.o sst-baytrail-pcm.o sst-baytrail-dsp.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += snd-soc-sst-haswell-pcm.o +obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o # Machine support snd-soc-sst-haswell-objs := haswell.o +snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o -- cgit v1.1 From 5069e5c93ca73eab185d3bb338a1362275f9ea70 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 25 Feb 2014 16:37:47 +0200 Subject: ASoC: Intel: sst-acpi: Fix Oops in case of missing firmware I swear I tested missing firmware in commit e5161d7987f1 ("ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing"). Unfortunately same wasn't done in commit 6dda27cbbd1d ("ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform") which will cause NULL pointer dereference in sst_acpi_fw_cb() when printing the error since sst_acpi->mach is not set. Fix this obvious error by setting the sst_acpi->mach in sst_acpi_probe(). Reported-by: Mika Westerberg Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-acpi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 0bb4316..5d06eec 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -139,6 +139,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; sst_acpi->desc = desc; + sst_acpi->mach = mach; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; -- cgit v1.1 From 951e9bb1fa589177183af1696ecfd4e4d8d37cbf Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 21 Feb 2014 11:52:40 +0300 Subject: ASoC: Intel: sst-firmware: missing curly braces (harmless) There were some curly braces intended here, but the code actually works the same either way so it's not a bug. Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index dee7eb5..f768710 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -293,7 +293,7 @@ static int block_module_prepare(struct sst_module *module) /* enable each block so that's it'e ready for module P/S data */ list_for_each_entry(block, &module->block_list, module_list) { - if (block->ops && block->ops->enable) + if (block->ops && block->ops->enable) { ret = block->ops->enable(block); if (ret < 0) { dev_err(module->dsp->dev, @@ -301,6 +301,7 @@ static int block_module_prepare(struct sst_module *module) block->type, block->index); goto err; } + } } return ret; -- cgit v1.1 From f9da9e434d9dad684aec159a74b9c8436d4faf5a Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 25 Feb 2014 11:32:50 +0300 Subject: ASoC: intel: restore IRQs on error This should be spin_unlock_irqrestore() instead of spin_unlock() Fixes: 22981243589c ('ASoC: Intel: Add Haswell/Broadwell IPC') Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 668d486..552aebf 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -527,7 +527,7 @@ static int ipc_tx_message(struct sst_hsw *hsw, u32 header, void *tx_data, msg = msg_get_empty(hsw); if (msg == NULL) { - spin_unlock(&hsw->dsp->spinlock); + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); return -EBUSY; } -- cgit v1.1 From 31d632f95567a84e1344aa110249a8346c35d2ec Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 25 Feb 2014 11:33:44 +0300 Subject: ASoC: intel: incorrect sizeof() This should be sizeof(pos) instead of sizeof(&pos). Most likely they are both 8 bytes though so it doesn't often make a difference in real life. Fixes: 22981243589c ('ASoC: Intel: Add Haswell/Broadwell IPC') Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 552aebf..1f1576a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -617,7 +617,7 @@ static void hsw_notification_work(struct work_struct *work) case IPC_POSITION_CHANGED: trace_ipc_notification("DSP stream position changed for", stream->reply.stream_hw_id); - sst_dsp_inbox_read(hsw->dsp, pos, sizeof(&pos)); + sst_dsp_inbox_read(hsw->dsp, pos, sizeof(pos)); if (stream->notify_position) stream->notify_position(stream, stream->pdata); -- cgit v1.1 From 52be4776cace06bf3d3df85fa490e61421824051 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 25 Feb 2014 15:17:27 +0200 Subject: ASoC: Intel: byt-rt5640: Update internal mic and speaker kcontrol names Use more sensible kcontrol names than "Int Mic" and "Ext Spk". Speakers especially are usually integrated devices in sales models. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwoood Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 038547b..672fedb 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -29,27 +29,27 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), }; static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"IN2P", NULL, "Headset Mic"}, {"IN2N", NULL, "Headset Mic"}, - {"DMIC1", NULL, "Int Mic"}, + {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, - {"Ext Spk", NULL, "SPOLP"}, - {"Ext Spk", NULL, "SPOLN"}, - {"Ext Spk", NULL, "SPORP"}, - {"Ext Spk", NULL, "SPORN"}, + {"Speaker", NULL, "SPOLP"}, + {"Speaker", NULL, "SPOLN"}, + {"Speaker", NULL, "SPORP"}, + {"Speaker", NULL, "SPORN"}, }; static const struct snd_kcontrol_new byt_rt5640_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Int Mic"), - SOC_DAPM_PIN_SWITCH("Ext Spk"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), }; static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, @@ -113,8 +113,8 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headphone"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "Int Mic"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Internal Mic"); snd_soc_dapm_sync(dapm); return ret; -- cgit v1.1 From 95c40d4b0eb47d82a8841364627a35ad6fad720b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 25 Feb 2014 15:17:28 +0200 Subject: ASoC: Intel: byt-rt5640: Use init time DAI format Setting static DAI format has been supported in the soc-core quite some time now so there is no need to set it runtime in machine driver. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwoood Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 672fedb..eff97c8 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -57,19 +57,8 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int fmt; int ret; - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; - - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - dev_err(codec_dai->dev, - "can't set codec DAI configuration %d\n", ret); - return ret; - } - ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, params_rate(params) * 256, SND_SOC_CLOCK_IN); @@ -132,6 +121,8 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .codec_dai_name = "rt5640-aif1", .codec_name = "i2c-10EC5640:00", .platform_name = "baytrail-pcm-audio", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .init = byt_rt5640_init, .ignore_suspend = 1, .ops = &byt_rt5640_ops, @@ -143,6 +134,8 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .codec_dai_name = "rt5640-aif1", .codec_name = "i2c-10EC5640:00", .platform_name = "baytrail-pcm-audio", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .init = NULL, .ignore_suspend = 1, .ops = &byt_rt5640_ops, -- cgit v1.1 From a6cf8f7b53fff6b5e3463793aa9885e133e7ef86 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 26 Feb 2014 15:31:46 +0200 Subject: ASoC: Intel: Baytrail: Fix implicit declaration of function 'memcpy_fromio' MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Some kernel configurations can cause following build error: sound/soc/intel/sst-baytrail-ipc.c: In function ‘sst_byt_get_dsp_position’: sound/soc/intel/sst-baytrail-ipc.c:744:2: error: implicit declaration of function ‘memcpy_fromio’ [-Werror=implicit-function-declaration] memcpy_fromio(&fw_tstamp, ^ cc1: some warnings being treated as errors Fix this by including explicitly. Reported-by: kbuild test robot Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-ipc.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 8c91a68..c12e194 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include "sst-baytrail-ipc.h" -- cgit v1.1 From 9aa8210d40c2140daf655c5299557bd68362399a Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Thu, 27 Feb 2014 17:49:51 +0800 Subject: ASoC: io: Clean up snd_soc_codec_set_cache_io() Now that all users have been converted to regmap and the config.reg_bits and config.val_bits can be setted by each user through regmap core API. So these two params are redundant here. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index add99e2..18353f1 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -88,16 +88,11 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control) { - struct regmap_config config; int ret; - memset(&config, 0, sizeof(config)); codec->write = hw_write; codec->read = hw_read; - config.reg_bits = addr_bits; - config.val_bits = data_bits; - switch (control) { case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ -- cgit v1.1 From fe2265e4833a3ebea426d748b5ecf8d3ff31edc8 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Thu, 27 Feb 2014 17:49:52 +0800 Subject: ASoC: core: Set the default I/O up try regmap. For most CODEC drivers which the REGMAP is used, the soc_probe_codec() will do the stuff work of snd_soc_codec_set_cache_io(), which the CODEC drivers' ASoC probe will do too, and almost at the same time. This patch set the default I/O up try regmap, and then the CODEC drivers' stuff work of snd_soc_codec_set_cache_io() will be redundant, while if one CODEC driver needed to set it's own I/O, then it can rewrite the default ones. Then could we just discard the snd_soc_codec_set_cache_io() from the CODEC drivers' ASoC probe to simplify the code. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50..93854f0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1137,6 +1137,10 @@ static int soc_probe_codec(struct snd_soc_card *card, codec->dapm.idle_bias_off = driver->idle_bias_off; + /* Set the default I/O up try regmap */ + if (dev_get_regmap(codec->dev, NULL)) + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { @@ -1150,10 +1154,6 @@ static int soc_probe_codec(struct snd_soc_card *card, codec->name); } - /* If the driver didn't set I/O up try regmap */ - if (!codec->write && dev_get_regmap(codec->dev, NULL)) - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (driver->controls) snd_soc_add_codec_controls(codec, driver->controls, driver->num_controls); -- cgit v1.1 From 30519cb8d2ecb7f0f0cdc42d709da0d9f7a04bcb Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Thu, 27 Feb 2014 17:49:53 +0800 Subject: ASoC: sgtl5000: Simplify ASoC probe code Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0fcbe90..c8c37e4 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1350,14 +1350,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) int ret; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - /* setup i2c data ops */ - codec->control_data = sgtl5000->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = sgtl5000_enable_regulators(codec); if (ret) return ret; -- cgit v1.1 From d4179c1deafd216b9358f76f5f399220cb8451ab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 4 Mar 2014 16:54:58 +0800 Subject: ASoC: da732x: Replace hw_read usage with snd_soc_read() Pre-merge code was using direct hw_read() calls as an out of framework way of doing volatile register I/O when not using regmap. This has never functioned correctly in mainline due to the regmap conversion, the hw_read() implementation still does caching. In order to facilitate removal of the subsystem level I/O code convert to use snd_soc_read(), there should be no functional impact. Signed-off-by: Mark Brown Acked-by: Adam Thomson --- sound/soc/codecs/da732x.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b65..8053e0e 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1301,9 +1301,9 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); /* Check DAC offset sign */ - sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + sign[DA732X_HPL_DAC] = (snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO); - sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + sign[DA732X_HPR_DAC] = (snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO); /* Binary search DAC offset values (both channels at once) */ @@ -1320,10 +1320,10 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); - if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + if ((snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC]) offset[DA732X_HPL_DAC] &= ~step; - if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + if ((snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC]) offset[DA732X_HPR_DAC] &= ~step; @@ -1364,9 +1364,9 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); /* Check output offset sign */ - sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) & + sign[DA732X_HPL_AMP] = snd_soc_read(codec, DA732X_REG_HPL) & DA732X_HP_OUT_COMPO; - sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) & + sign[DA732X_HPR_AMP] = snd_soc_read(codec, DA732X_REG_HPR) & DA732X_HP_OUT_COMPO; snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP | @@ -1387,10 +1387,10 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); - if ((codec->hw_read(codec, DA732X_REG_HPL) & + if ((snd_soc_read(codec, DA732X_REG_HPL) & DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP]) offset[DA732X_HPL_AMP] &= ~step; - if ((codec->hw_read(codec, DA732X_REG_HPR) & + if ((snd_soc_read(codec, DA732X_REG_HPR) & DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP]) offset[DA732X_HPR_AMP] &= ~step; -- cgit v1.1 From 9202c377390f2708dece910f2e066a6308a38abc Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 5 Mar 2014 14:11:57 +0300 Subject: ASoC: Baytrail: fix error handling in sst_byt_dsp_init() Calling "kfree(byt)" is a double free because that was allocated with devm_kzalloc(). There were a couple places which leak "byt->msg". That memory is allocated in msg_empty_list_init(). Fixes: f7d01fd6754c ('ASoC: Intel: Add Intel Baytrail SST DSP IPC support') Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-ipc.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index c12e194..d0eaeee 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -796,7 +796,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) err = msg_empty_list_init(byt); if (err < 0) - goto list_err; + return -ENOMEM; /* start the IPC message thread */ init_kthread_worker(&byt->kworker); @@ -806,7 +806,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) if (IS_ERR(byt->tx_thread)) { err = PTR_ERR(byt->tx_thread); dev_err(byt->dev, "error failed to create message TX task\n"); - goto list_err; + goto err_free_msg; } init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs); @@ -816,7 +816,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); if (byt->dsp == NULL) { err = -ENODEV; - goto list_err; + goto err_free_msg; } /* keep the DSP in reset state for base FW loading */ @@ -848,9 +848,9 @@ boot_err: sst_fw_free(byt_sst_fw); fw_err: sst_dsp_free(byt->dsp); +err_free_msg: kfree(byt->msg); -list_err: - kfree(byt); + return err; } EXPORT_SYMBOL_GPL(sst_byt_dsp_init); -- cgit v1.1 From f69f41e1a2568f2ebdcf021fe216c1e9ba24cc1f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 6 Mar 2014 14:56:04 +0000 Subject: ASoC: Intel: Check Haswell IPC process_reply/notification return value. Check the return value for error when processing replies and notifications. The patch 22981243589c: "ASoC: Intel: Add Haswell/Broadwell IPC" from > Feb 20, 2014, leads to the following imaginary static checker warning: > > sound/soc/intel/sst-haswell-ipc.c:898 hsw_irq_thread() > warn: this is always true. > > sound/soc/intel/sst-haswell-ipc.c > 895 /* Handle Immediate reply from DSP Core */ > 896 handled = hsw_process_reply(hsw, ipcx); > ^^^^^^^^^^^^^^^^^ > Returns 1 on success/error and -EIO on error. > > 897 > 898 if (handled) { > 899 /* clear DONE bit - tell DSP we have completed */ > 900 sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, > 901 SST_IPCX_DONE, 0); > 902 > 903 /* unmask Done interrupt */ > 904 sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, > 905 SST_IMRX_DONE, 0); > 906 } > Reported-by: Dan Carpenter Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 1f1576a..f46bb4d 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -895,7 +895,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) /* Handle Immediate reply from DSP Core */ handled = hsw_process_reply(hsw, ipcx); - if (handled) { + if (handled > 0) { /* clear DONE bit - tell DSP we have completed */ sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, SST_IPCX_DONE, 0); @@ -913,7 +913,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) handled = hsw_process_notification(hsw); /* clear BUSY bit and set DONE bit - accept new messages */ - if (handled) { + if (handled > 0) { sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); -- cgit v1.1 From dd6646bcfa9c811518bfcdbd13e4f73b749646bb Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 6 Mar 2014 14:56:05 +0000 Subject: ASoC: Intel: Use .dai_fmt for setting Haswell BE format. Update the Haswell driver to use .dai_fmt in DAI link to set the format. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/haswell.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 0d61197..54345a2 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -69,14 +69,6 @@ static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec DAI configuration\n"); - return ret; - } - ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, SND_SOC_CLOCK_IN); @@ -188,6 +180,8 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .no_pcm = 1, .codec_name = "i2c-INT33CA:00", .codec_dai_name = "rt5640-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ignore_suspend = 1, .ignore_pmdown_time = 1, .be_hw_params_fixup = haswell_ssp0_fixup, -- cgit v1.1 From a8282136a1b811edb95b3c0e1d9664510afaa307 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 6 Mar 2014 14:56:06 +0000 Subject: ASoC: Intel: Clean up indentation for Haswell machine driver/Kconfig Clean up some indentation. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 10 +++++----- sound/soc/intel/sst-haswell-dsp.c | 2 +- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 274af16..4577b69 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -31,15 +31,15 @@ config SND_SOC_INTEL_BAYTRAIL tristate config SND_SOC_INTEL_HASWELL_MACH - tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help - This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell + This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell Ultrabook platforms. - Say Y if you have such a device - If unsure select "N". + Say Y if you have such a device + If unsure select "N". config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 12f7317..f5ebf36 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -503,7 +503,7 @@ static void hsw_free(struct sst_dsp *sst) struct sst_ops haswell_ops = { .reset = hsw_reset, - .boot = hsw_boot, + .boot = hsw_boot, .write = sst_shim32_write, .read = sst_shim32_read, .write64 = sst_shim32_write64, -- cgit v1.1 From 5d6be5aa6becc750c5c2aa0ef8f7209ce19aa328 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 11 Mar 2014 12:43:20 +0800 Subject: ASoC: codec: Simplify ASoC probe code. For some CODEC drivers like who act as the MFDs children are ignored by this patch. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 10 +--------- sound/soc/codecs/adau1373.c | 7 ------- sound/soc/codecs/adav80x.c | 7 ------- sound/soc/codecs/ak4535.c | 9 --------- sound/soc/codecs/ak4641.c | 8 -------- sound/soc/codecs/ak4642.c | 8 -------- sound/soc/codecs/ak4671.c | 12 +----------- sound/soc/codecs/alc5623.c | 7 ------- sound/soc/codecs/alc5632.c | 8 -------- sound/soc/codecs/cs4270.c | 9 --------- sound/soc/codecs/cs42l51.c | 6 ------ sound/soc/codecs/cs42l52.c | 9 +-------- sound/soc/codecs/cs42l73.c | 11 +---------- sound/soc/codecs/da7210.c | 8 -------- sound/soc/codecs/da7213.c | 8 -------- sound/soc/codecs/da732x.c | 13 ++----------- sound/soc/codecs/da9055.c | 8 -------- sound/soc/codecs/isabelle.c | 16 ---------------- sound/soc/codecs/lm49453.c | 17 ----------------- sound/soc/codecs/max9768.c | 5 ----- sound/soc/codecs/max98088.c | 6 ------ sound/soc/codecs/max98090.c | 8 -------- sound/soc/codecs/max98095.c | 6 ------ sound/soc/codecs/max9850.c | 8 -------- sound/soc/codecs/ml26124.c | 10 ---------- sound/soc/codecs/rt5631.c | 9 --------- sound/soc/codecs/rt5640.c | 8 -------- sound/soc/codecs/sn95031.c | 2 -- sound/soc/codecs/ssm2518.c | 10 ---------- sound/soc/codecs/ssm2602.c | 7 ------- sound/soc/codecs/sta32x.c | 14 -------------- sound/soc/codecs/sta529.c | 10 ---------- sound/soc/codecs/tlv320aic23.c | 8 -------- sound/soc/codecs/tlv320aic26.c | 2 -- sound/soc/codecs/tlv320aic32x4.c | 2 -- sound/soc/codecs/tlv320aic3x.c | 6 ------ sound/soc/codecs/wm2000.c | 2 -- sound/soc/codecs/wm2200.c | 7 ------- sound/soc/codecs/wm5100.c | 7 ------- sound/soc/codecs/wm8510.c | 10 +--------- sound/soc/codecs/wm8523.c | 7 ------- sound/soc/codecs/wm8580.c | 6 ------ sound/soc/codecs/wm8711.c | 6 ------ sound/soc/codecs/wm8728.c | 11 +---------- sound/soc/codecs/wm8731.c | 7 ------- sound/soc/codecs/wm8737.c | 6 ------ sound/soc/codecs/wm8741.c | 6 ------ sound/soc/codecs/wm8750.c | 6 ------ sound/soc/codecs/wm8753.c | 7 ------- sound/soc/codecs/wm8770.c | 6 ------ sound/soc/codecs/wm8776.c | 6 ------ sound/soc/codecs/wm8804.c | 8 -------- sound/soc/codecs/wm8900.c | 8 +------- sound/soc/codecs/wm8903.c | 10 +--------- sound/soc/codecs/wm8904.c | 9 --------- sound/soc/codecs/wm8940.c | 6 ------ sound/soc/codecs/wm8955.c | 8 -------- sound/soc/codecs/wm8960.c | 6 ------ sound/soc/codecs/wm8961.c | 7 ------- sound/soc/codecs/wm8962.c | 7 ------- sound/soc/codecs/wm8971.c | 6 ------ sound/soc/codecs/wm8974.c | 6 ------ sound/soc/codecs/wm8978.c | 8 +------- sound/soc/codecs/wm8983.c | 6 ------ sound/soc/codecs/wm8985.c | 7 ------- sound/soc/codecs/wm8988.c | 8 -------- sound/soc/codecs/wm8990.c | 8 -------- sound/soc/codecs/wm8991.c | 8 -------- sound/soc/codecs/wm8993.c | 7 ------- sound/soc/codecs/wm8995.c | 7 ------- sound/soc/codecs/wm8996.c | 12 +----------- sound/soc/codecs/wm9081.c | 11 +---------- sound/soc/codecs/wm9090.c | 10 ---------- 73 files changed, 13 insertions(+), 562 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 9381a76..6844d0b 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -322,14 +322,6 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ad193x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } /* default setting for ad193x */ @@ -347,7 +339,7 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); - return ret; + return 0; } static struct snd_soc_codec_driver soc_codec_dev_ad193x = { diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index eb836ed..db5c303 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1376,15 +1376,8 @@ static int adau1373_probe(struct snd_soc_codec *codec) struct adau1373_platform_data *pdata = codec->dev->platform_data; bool lineout_differential = false; unsigned int val; - int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } - if (pdata) { if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) return -EINVAL; diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f78b27a..8d79c3f 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -798,15 +798,8 @@ static struct snd_soc_dai_driver adav80x_dais[] = { static int adav80x_probe(struct snd_soc_codec *codec) { - int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } - /* Force PLLs on for SYSCLK output */ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 684fe91..30e2978 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -388,15 +388,6 @@ static int ak4535_resume(struct snd_soc_codec *codec) static int ak4535_probe(struct snd_soc_codec *codec) { - struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ak4535->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 94cbe50..a7b7d985 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -519,14 +519,6 @@ static int ak4641_resume(struct snd_soc_codec *codec) static int ak4641_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* power on device */ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f646c6..92655cc 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -465,14 +465,6 @@ static int ak4642_resume(struct snd_soc_codec *codec) static int ak4642_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index deb2b44..998fa0c 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -613,17 +613,7 @@ static struct snd_soc_dai_driver ak4671_dai = { static int ak4671_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return ret; + return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int ak4671_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index ed50625..09f7e77 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -904,13 +904,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - codec->control_data = alc5623->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - alc5623_reset(codec); /* power on device */ diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index d885056..ec071a6 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1063,14 +1063,6 @@ static int alc5632_probe(struct snd_soc_codec *codec) struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = alc5632->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 83c835d..3920e62 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -506,15 +506,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; - /* Tell ASoC what kind of I/O to use to read the registers. ASoC will - * then do the I2C transactions itself. - */ - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - return ret; - } - /* Disable auto-mute. This feature appears to be buggy. In some * situations, auto-mute will not deactivate when it should, so we want * this feature disabled by default. An application (e.g. alsactl) can diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 3eab460..a0c6060d 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -489,12 +489,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec) { int ret, reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* * DAC configuration * - Use signal processor diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0bac6d5..4bd59ce 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1115,14 +1115,7 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) static int cs42l52_probe(struct snd_soc_codec *codec) { struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); - int ret; - codec->control_data = cs42l52->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } regcache_cache_only(cs42l52->regmap, true); cs42l52_add_mic_controls(codec); @@ -1134,7 +1127,7 @@ static int cs42l52_probe(struct snd_soc_codec *codec) cs42l52->sysclk = CS42L52_DEFAULT_CLK; cs42l52->config.format = CS42L52_DEFAULT_FORMAT; - return ret; + return 0; } static int cs42l52_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6..b9aa009 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1348,17 +1348,8 @@ static int cs42l73_resume(struct snd_soc_codec *codec) static int cs42l73_probe(struct snd_soc_codec *codec) { - int ret; struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); - codec->control_data = cs42l73->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Set Charge Pump Frequency */ @@ -1371,7 +1362,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec) cs42l73->mclksel = CS42L73_CLKID_MCLK1; cs42l73->mclk = 0; - return ret; + return 0; } static int cs42l73_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e62e294..a5838ba 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1071,17 +1071,9 @@ static struct snd_soc_dai_driver da7210_dai = { static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); - int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ da7210->master = 0; /* This will be set from set_fmt() */ diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 0c77e7a..110f4dd 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1384,17 +1384,9 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, static int da7213_probe(struct snd_soc_codec *codec) { - int ret; struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); struct da7213_platform_data *pdata = da7213->pdata; - codec->control_data = da7213->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Default to using ALC auto offset calibration mode. */ snd_soc_update_bits(codec, DA7213_ALC_CTRL1, DA7213_ALC_CALIB_MODE_MAN, 0); diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 8053e0e..fdefc4b 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1516,23 +1516,14 @@ static int da732x_probe(struct snd_soc_codec *codec) { struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret = 0; da732x->codec = codec; dapm->idle_bias_off = false; - codec->control_data = da732x->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to register codec.\n"); - goto err; - } - da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -err: - return ret; + + return 0; } static int da732x_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a4..f0a371d 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1381,16 +1381,8 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec, static int da9055_probe(struct snd_soc_codec *codec) { - int ret; struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); - codec->control_data = da9055->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Enable all Gain Ramps */ snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5839048..087b3cb 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1082,23 +1082,7 @@ static struct snd_soc_dai_driver isabelle_dai[] = { }, }; -static int isabelle_probe(struct snd_soc_codec *codec) -{ - int ret = 0; - - codec->control_data = dev_get_regmap(codec->dev, NULL); - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_isabelle = { - .probe = isabelle_probe, .set_bias_level = isabelle_set_bias_level, .controls = isabelle_snd_controls, .num_controls = ARRAY_SIZE(isabelle_snd_controls), diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index e19490c..069cb03 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1409,22 +1409,6 @@ static int lm49453_resume(struct snd_soc_codec *codec) return 0; } -static int lm49453_probe(struct snd_soc_codec *codec) -{ - struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); - int ret = 0; - - codec->control_data = lm49453->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - return 0; -} - /* power down chip */ static int lm49453_remove(struct snd_soc_codec *codec) { @@ -1433,7 +1417,6 @@ static int lm49453_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { - .probe = lm49453_probe, .remove = lm49453_remove, .suspend = lm49453_suspend, .resume = lm49453_resume, diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 31f9156..ec481fc 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -135,11 +135,6 @@ static int max9768_probe(struct snd_soc_codec *codec) struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = max9768->regmap; - ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP); - if (ret) - return ret; - if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) { ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM); if (ret) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ee660e2..64965005 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1915,12 +1915,6 @@ static int max98088_probe(struct snd_soc_codec *codec) regcache_mark_dirty(max98088->regmap); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* initialize private data */ max98088->sysclk = (unsigned)-1; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d..4ac3b67 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2195,14 +2195,6 @@ static int max98090_probe(struct snd_soc_codec *codec) max98090->codec = codec; - codec->control_data = max98090->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Reset the codec, the DSP core, and disable all interrupts */ max98090_reset(max98090); diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 3ba1170..2a9bfef 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2234,12 +2234,6 @@ static int max98095_probe(struct snd_soc_codec *codec) struct i2c_client *client; int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 82757eb..4fdf5aa 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -312,14 +312,6 @@ static int max9850_resume(struct snd_soc_codec *codec) static int max9850_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* enable zero-detect */ snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1); /* enable slew-rate control */ diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 185fa3bc..b9f21fe 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -586,16 +586,6 @@ static int ml26124_resume(struct snd_soc_codec *codec) static int ml26124_probe(struct snd_soc_codec *codec) { - int ret; - struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); - codec->control_data = priv->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Software Reset */ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 912c9cb..73fcd3c 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1585,15 +1585,6 @@ static int rt5631_probe(struct snd_soc_codec *codec) { struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); unsigned int val; - int ret; - - codec->control_data = rt5631->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } val = rt5631_read_index(codec, RT5631_ADDA_MIXER_INTL_REG3); if (val & 0x0002) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb411..074cff1 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1943,16 +1943,8 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, static int rt5640_probe(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - int ret; rt5640->codec = codec; - codec->control_data = rt5640->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } codec->dapm.idle_bias_off = 1; rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 13045f2..193760e 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -825,8 +825,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index cc8debc..cf4a657 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -648,16 +648,6 @@ static struct snd_soc_dai_driver ssm2518_dai = { static int ssm2518_probe(struct snd_soc_codec *codec) { - struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ssm2518->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index af76bbd..9fe4d48 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -573,13 +573,6 @@ static int ssm260x_probe(struct snd_soc_codec *codec) struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = ssm2602->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 06edb39..1cab6f6 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -872,16 +872,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) return ret; } - /* Tell ASoC what kind of I/O to use to read the registers. ASoC will - * then do the I2C transactions itself. - */ - codec->control_data = sta32x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - goto err; - } - /* Chip documentation explicitly requires that the reset values * of reserved register bits are left untouched. * Write the register default value to cache for reserved registers, @@ -946,10 +936,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return 0; - -err: - regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - return ret; } static int sta32x_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 40c07be..30aae61 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -322,16 +322,6 @@ static struct snd_soc_dai_driver sta529_dai = { static int sta529_probe(struct snd_soc_codec *codec) { - struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = sta529->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 7b4cfef..46b8a50 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -559,14 +559,6 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Reset codec */ snd_soc_write(codec, TLV320AIC23_RESET, 0); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 94a658f..8037bea 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -295,8 +295,6 @@ static int aic26_probe(struct snd_soc_codec *codec) struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, reg; - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - aic26->codec = codec; /* Reset the codec to power on defaults */ diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c6bd7e7..1d9b117 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -614,8 +614,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec) struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 470fbfb..b183510 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1344,12 +1344,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; aic3x->disable_nb[i].aic3x = aic3x; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 8ae5027..83a2c87 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -786,8 +786,6 @@ static int wm2000_probe(struct snd_soc_codec *codec) { struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP); - /* This will trigger a transition to standby mode by default */ wm2000_anc_set_mode(wm2000); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 57ba315..5129d91 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1556,15 +1556,8 @@ static int wm2200_probe(struct snd_soc_codec *codec) int ret; wm2200->codec = codec; - codec->control_data = wm2200->regmap; codec->dapm.bias_level = SND_SOC_BIAS_OFF; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4e3e31a..bac848f 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2337,13 +2337,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) int ret, i; wm5100->codec = codec; - codec->control_data = wm5100->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 7df7d45..1c1e328 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -589,20 +589,12 @@ static int wm8510_resume(struct snd_soc_codec *codec) static int wm8510_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); - return ret; - } - wm8510_reset(codec); /* power on device */ wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 74d106d..e6116af 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -392,18 +392,11 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - int ret; wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0]; wm8523->rate_constraint.count = ARRAY_SIZE(wm8523->rate_constraint_list); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Change some default settings - latch VU and enable ZC */ snd_soc_update_bits(codec, WM8523_DAC_GAINR, WM8523_DACR_VU, WM8523_DACR_VU); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989a..7558c83 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -869,12 +869,6 @@ static int wm8580_probe(struct snd_soc_codec *codec) struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); if (ret != 0) { diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index d99f948..ef6cbc7 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -367,12 +367,6 @@ static int wm8711_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8711_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index cd89033..bac7fc2 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -228,19 +228,10 @@ static int wm8728_resume(struct snd_soc_codec *codec) static int wm8728_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", - ret); - return ret; - } - /* power on device */ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } static int wm8728_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0297203..2c95b63 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -583,13 +583,6 @@ static int wm8731_probe(struct snd_soc_codec *codec) struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; - codec->control_data = wm8731->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) wm8731->supplies[i].supply = wm8731_supply_names[i]; diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 2f167a8..3693479 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -570,12 +570,6 @@ static int wm8737_probe(struct snd_soc_codec *codec) struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); if (ret != 0) { diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 2895c8d..ecf4fcf 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -429,12 +429,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_get; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err_enable; - } - ret = wm8741_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 78616a6..33990b6 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -702,12 +702,6 @@ static int wm8750_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8750_reset(codec); if (ret < 0) { printk(KERN_ERR "wm8750: failed to reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da9..0d1670b 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1471,13 +1471,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work); - codec->control_data = wm8753->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8753_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 89a18d8..32e7363 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -580,12 +580,6 @@ static int wm8770_probe(struct snd_soc_codec *codec) wm8770 = snd_soc_codec_get_drvdata(codec); wm8770->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies), wm8770->supplies); if (ret) { diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ef82467..70952ce 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -430,12 +430,6 @@ static int wm8776_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8776_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9bc8206..448a943 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -546,14 +546,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804 = snd_soc_codec_get_drvdata(codec); - codec->control_data = wm8804->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e98bc70..637be63 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1178,13 +1178,7 @@ static int wm8900_resume(struct snd_soc_codec *codec) static int wm8900_probe(struct snd_soc_codec *codec) { - int ret = 0, reg; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } + int reg; reg = snd_soc_read(codec, WM8900_REG_ID); if (reg != 0x8900) { diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index eebcb1d..cf2f49f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1897,21 +1897,13 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903) static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int ret; wm8903->codec = codec; - codec->control_data = wm8903->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 53bbfac..817dddc 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2047,9 +2047,6 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = wm8904->regmap; switch (wm8904->devtype) { case WM8904: @@ -2063,12 +2060,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) return -EINVAL; } - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - wm8904_handle_pdata(codec); wm8904_add_widgets(codec); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b404c26..1cdabaf 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -712,12 +712,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8940_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 82c8ba9..a94d185 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -896,14 +896,6 @@ static int wm8955_probe(struct snd_soc_codec *codec) struct wm8955_pdata *pdata = dev_get_platdata(codec->dev); int ret, i; - codec->control_data = wm8955->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8955->supplies); i++) wm8955->supplies[i].supply = wm8955_supply_names[i]; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index f156010..d04e9ca 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -976,12 +976,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level = wm8960_set_bias_level_capless; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8960_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 900328e..db84507 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -836,15 +836,8 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Enable class W */ reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B); reg |= WM8961_CP_DYN_PWR_MASK; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 97db3b4..1d556c9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3400,13 +3400,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) bool dmicclk, dmicdat; wm8962->codec = codec; - codec->control_data = wm8962->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } wm8962->disable_nb[0].notifier_call = wm8962_regulator_event_0; wm8962->disable_nb[1].notifier_call = wm8962_regulator_event_1; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 67aba78..09b7b42 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -648,12 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); - return ret; - } - INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); wm8971_workq = create_workqueue("wm8971"); if (wm8971_workq == NULL) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 15f45c7..ea0de26 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -593,12 +593,6 @@ static int wm8974_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8974_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index d8fc531..13de368 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -975,19 +975,13 @@ static const int update_reg[] = { static int wm8978_probe(struct snd_soc_codec *codec) { struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); - int ret = 0, i; + int i; /* * Set default system clock to PLL, it is more precise, this is also the * default hardware setting */ wm8978->sysclk = WM8978_PLL; - codec->control_data = wm8978->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* * Set the update bit in all registers, that have one. This way all diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index aa41ba0..84aa093 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1000,12 +1000,6 @@ static int wm8983_probe(struct snd_soc_codec *codec) int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 271b517..64e211c 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1000,13 +1000,6 @@ static int wm8985_probe(struct snd_soc_codec *codec) int ret; wm8985 = snd_soc_codec_get_drvdata(codec); - codec->control_data = wm8985->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } for (i = 0; i < ARRAY_SIZE(wm8985->supplies); i++) wm8985->supplies[i].supply = wm8985_supply_names[i]; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index a55e1c2..424bbf7 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -810,16 +810,8 @@ static int wm8988_resume(struct snd_soc_codec *codec) static int wm8988_probe(struct snd_soc_codec *codec) { - struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); int ret = 0; - codec->control_data = wm8988->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8988_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 0ccd4d8..1487625 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1292,14 +1292,6 @@ static int wm8990_resume(struct snd_soc_codec *codec) */ static int wm8990_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret); - return ret; - } - wm8990_reset(codec); /* charge output caps */ diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 32d2195..844cc4a 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1248,14 +1248,6 @@ static int wm8991_remove(struct snd_soc_codec *codec) static int wm8991_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a..1674a1f 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1493,13 +1493,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->hubs_data.dcs_codes_r = -2; wm8993->hubs_data.series_startup = 1; - codec->control_data = wm8993->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Latch volume update bits and default ZC on */ snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME, WM8993_DAC_VU, WM8993_DAC_VU); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4300caf..9fd76c9c 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2047,13 +2047,6 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; - codec->control_data = wm8995->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) wm8995->supplies[i].supply = wm8995_supply_names[i]; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1a7655b..5406643 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2628,14 +2628,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->dcs_done); init_completion(&wm8996->fll_lock); - codec->control_data = wm8996->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err; - } - if (wm8996->pdata.num_retune_mobile_cfgs) wm8996_retune_mobile_pdata(codec); else @@ -2674,13 +2666,11 @@ static int wm8996_probe(struct snd_soc_codec *codec) } else { dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + return ret; } } return 0; - -err: - return ret; } static int wm8996_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 0982c1d..cda185d 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1265,15 +1265,6 @@ static struct snd_soc_dai_driver wm9081_dai = { static int wm9081_probe(struct snd_soc_codec *codec) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = wm9081->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* Enable zero cross by default */ snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, @@ -1288,7 +1279,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm9081_eq_controls)); } - return ret; + return 0; } static int wm9081_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a07fe16..8793417 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -522,16 +522,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { - struct wm9090_priv *wm9090 = dev_get_drvdata(codec->dev); - int ret; - - codec->control_data = wm9090->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Configure some defaults; they will be written out when we * bring the bias up. */ -- cgit v1.1 From 092eba937d948a76ff55825922eff4df010f6a17 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 11 Mar 2014 12:43:21 +0800 Subject: ASoC: io: New signature for snd_soc_codec_set_cache_io() Now that all users have been converted to regmap and the config.reg_bits and config.val_bits can be setted by each user through regmap core API. So these two params are redundant here. Since the only control type that left is SND_SOC_REGMAP, so remove it. Drop the control params and add struct regmap *regmap to simplify the code. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 3 +-- sound/soc/codecs/cq93vc.c | 3 +-- sound/soc/codecs/mc13783.c | 4 ++-- sound/soc/codecs/si476x.c | 6 +++-- sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/wm5102.c | 4 +--- sound/soc/codecs/wm5110.c | 3 +-- sound/soc/codecs/wm8350.c | 4 +--- sound/soc/codecs/wm8400.c | 3 +-- sound/soc/codecs/wm8994.c | 3 +-- sound/soc/codecs/wm8997.c | 4 +--- sound/soc/soc-core.c | 2 +- sound/soc/soc-io.c | 47 +++++++++++++++++---------------------- 13 files changed, 35 insertions(+), 52 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 647a72c..773b533 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1327,8 +1327,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; - codec->control_data = pm860x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, pm860x->regmap); if (ret) return ret; diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 43737a27..1e25c7a 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -138,9 +138,8 @@ static int cq93vc_probe(struct snd_soc_codec *codec) struct davinci_vc *davinci_vc = codec->dev->platform_data; davinci_vc->cq93vc.codec = codec; - codec->control_data = davinci_vc->regmap; - snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, davinci_vc->regmap); /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 582c2bb..fc28b20 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -614,8 +614,8 @@ static int mc13783_probe(struct snd_soc_codec *codec) struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, + dev_get_regmap(codec->dev->parent, NULL)); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index fa2b8e0..244c097 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include @@ -209,8 +210,9 @@ out: static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + struct regmap *regmap = dev_get_regmap(codec->dev->parent, NULL); + + return snd_soc_codec_set_cache_io(codec, regmap); } static struct snd_soc_dai_ops si476x_dai_ops = { diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4f35839..64afda7 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -122,7 +122,6 @@ struct tlv320dac33_priv { unsigned int uthr; enum dac33_state state; - enum snd_soc_control_type control_type; void *control_data; }; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ce9c8e1..5613d0e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1758,9 +1758,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2c3c962..66d3ad4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1588,10 +1588,9 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; priv->core.arizona->dapm = &codec->dapm; - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index a183dcf..757256b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1505,9 +1505,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - codec->control_data = wm8350->regmap; - - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, wm8350->regmap); /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 48dc7d2..939baf8 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1310,10 +1310,9 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, priv); priv->wm8400 = wm8400; - codec->control_data = wm8400->regmap; priv->codec = codec; - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, wm8400->regmap); ret = devm_regulator_bulk_get(wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b9be9cb..32cc83e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3985,9 +3985,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) int ret, i; wm8994->hubs.codec = codec; - codec->control_data = control->regmap; - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, control->regmap); mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 555115e..e3d1522 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1052,9 +1052,7 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec) struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad2dd14..6510a8e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1139,7 +1139,7 @@ static int soc_probe_codec(struct snd_soc_card *card, /* Set the default I/O up try regmap */ if (dev_get_regmap(codec->dev, NULL)) - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, NULL); if (driver->probe) { ret = driver->probe(codec); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 18353f1..8aa0869 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -69,9 +69,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) * snd_soc_codec_set_cache_io: Set up standard I/O functions. * * @codec: CODEC to configure. - * @addr_bits: Number of bits of register address data. - * @data_bits: Number of bits of data per register. - * @control: Control bus used. + * @map: Register map to write to * * Register formats are frequently shared between many I2C and SPI * devices. In order to promote code reuse the ASoC core provides @@ -85,41 +83,36 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) * volatile registers. */ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control) + struct regmap *regmap) { int ret; + /* Device has made its own regmap arrangements */ + if (!regmap) + codec->control_data = dev_get_regmap(codec->dev, NULL); + else + codec->control_data = regmap; + + if (IS_ERR(codec->control_data)) + return PTR_ERR(codec->control_data); + codec->write = hw_write; codec->read = hw_read; - switch (control) { - case SND_SOC_REGMAP: - /* Device has made its own regmap arrangements */ - codec->using_regmap = true; - if (!codec->control_data) - codec->control_data = dev_get_regmap(codec->dev, NULL); - - if (codec->control_data) { - ret = regmap_get_val_bytes(codec->control_data); - /* Errors are legitimate for non-integer byte - * multiples */ - if (ret > 0) - codec->val_bytes = ret; - } - break; - - default: - return -EINVAL; - } + ret = regmap_get_val_bytes(codec->control_data); + /* Errors are legitimate for non-integer byte + * multiples */ + if (ret > 0) + codec->val_bytes = ret; + + codec->using_regmap = true; - return PTR_ERR_OR_ZERO(codec->control_data); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); #else int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control) + struct regmap *regmap) { return -ENOTSUPP; } -- cgit v1.1 From a32c17b87c17f5e2e68edcf4d163ee42f9490652 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 11 Mar 2014 12:43:22 +0800 Subject: ASoC: core: Fix check before setting default I/O up try regmap Since the CODEC driver could specify its own I/O(read and write) while registering the CODEC for some reason, maybe the MFDs is used, etc. So just do check it, if they are not specified by CODEC driver then try to set up the default regmap I/O if regmap is used. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6510a8e..cbddbd5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1137,9 +1137,15 @@ static int soc_probe_codec(struct snd_soc_card *card, codec->dapm.idle_bias_off = driver->idle_bias_off; - /* Set the default I/O up try regmap */ - if (dev_get_regmap(codec->dev, NULL)) - snd_soc_codec_set_cache_io(codec, NULL); + if (!codec->write && dev_get_regmap(codec->dev, NULL)) { + /* Set the default I/O up try regmap */ + ret = snd_soc_codec_set_cache_io(codec, NULL); + if (ret < 0) { + dev_err(codec->dev, + "Failed to set cache I/O: %d\n", ret); + goto err_probe; + } + } if (driver->probe) { ret = driver->probe(codec); -- cgit v1.1 From e95d73c437a09e7febea18f8e998f958ef6d7a72 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Mar 2014 15:27:29 +0100 Subject: ASoC: ams-delta: Fix compile error snd_soc_dapm_mutex_unlock() wants a pointer to the DAPM context, not the CODEC. Fixes: 03510ca07 ("ASoC: ams-delta: Update locking around use of DAPM pin API") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 5750de1..14718cd 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -327,7 +327,7 @@ static void cx81801_close(struct tty_struct *tty) snd_soc_dapm_sync_unlocked(dapm); - snd_soc_dapm_mutex_unlock(codec); + snd_soc_dapm_mutex_unlock(dapm); } /* Line discipline .hangup() */ -- cgit v1.1 From 7b2655b409211d619d99a1bb325797dee22a1c8b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Mar 2014 15:27:36 +0100 Subject: ASoC: snappercl15: Convert to table based DAPM setup Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/cirrus/snappercl15.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 29238a7..5b68b10 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -65,18 +65,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; -} - static struct snd_soc_dai_link snappercl15_dai = { .name = "tlv320aic23", .stream_name = "AIC23", @@ -84,7 +72,6 @@ static struct snd_soc_dai_link snappercl15_dai = { .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", .platform_name = "ep93xx-i2s", - .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, .ops = &snappercl15_ops, @@ -95,6 +82,11 @@ static struct snd_soc_card snd_soc_snappercl15 = { .owner = THIS_MODULE, .dai_link = &snappercl15_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int snappercl15_probe(struct platform_device *pdev) -- cgit v1.1 From 0fbd44ab772890cf5dcda87311575e99d2c000c1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Mar 2014 15:27:38 +0100 Subject: ASoC: pxa: magician: Convert to table based DAPM and control setup Use table based setup to register the controls and DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/magician.c | 34 ++++++++++++---------------------- 1 file changed, 12 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 41ab6678..259e048 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -41,9 +41,8 @@ static int magician_hp_switch; static int magician_spk_switch = 1; static int magician_in_sel = MAGICIAN_MIC; -static void magician_ext_control(struct snd_soc_codec *codec) +static void magician_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; snd_soc_dapm_mutex_lock(dapm); @@ -75,10 +74,9 @@ static void magician_ext_control(struct snd_soc_codec *codec) static int magician_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ - magician_ext_control(codec); + magician_ext_control(&rtd->card->dapm); return 0; } @@ -277,13 +275,13 @@ static int magician_get_hp(struct snd_kcontrol *kcontrol, static int magician_set_hp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (magician_hp_switch == ucontrol->value.integer.value[0]) return 0; magician_hp_switch = ucontrol->value.integer.value[0]; - magician_ext_control(codec); + magician_ext_control(&card->dapm); return 1; } @@ -297,13 +295,13 @@ static int magician_get_spk(struct snd_kcontrol *kcontrol, static int magician_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (magician_spk_switch == ucontrol->value.integer.value[0]) return 0; magician_spk_switch = ucontrol->value.integer.value[0]; - magician_ext_control(codec); + magician_ext_control(&card->dapm); return 1; } @@ -400,7 +398,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; /* NC codec pins */ snd_soc_dapm_nc_pin(dapm, "VOUTLHP"); @@ -410,19 +407,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "VINL"); snd_soc_dapm_nc_pin(dapm, "VINR"); - /* Add magician specific controls */ - err = snd_soc_add_codec_controls(codec, uda1380_magician_controls, - ARRAY_SIZE(uda1380_magician_controls)); - if (err < 0) - return err; - - /* Add magician specific widgets */ - snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - - /* Set up magician specific audio path interconnects */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -456,6 +440,12 @@ static struct snd_soc_card snd_soc_card_magician = { .dai_link = magician_dai, .num_links = ARRAY_SIZE(magician_dai), + .controls = uda1380_magician_controls, + .num_controls = ARRAY_SIZE(uda1380_magician_controls), + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *magician_snd_device; -- cgit v1.1 From 079942abb389e6d4197db03299ad13297afd812a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Mar 2014 15:27:39 +0100 Subject: ASoC: pxa: tosa: Convert to table based DAPM and control setup Use table based setup to register the controls and DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/tosa.c | 35 +++++++++++++---------------------- 1 file changed, 13 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index cead165..4a956d1 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -44,9 +44,8 @@ static int tosa_jack_func; static int tosa_spk_func; -static void tosa_ext_control(struct snd_soc_codec *codec) +static void tosa_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; snd_soc_dapm_mutex_lock(dapm); @@ -82,10 +81,9 @@ static void tosa_ext_control(struct snd_soc_codec *codec) static int tosa_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ - tosa_ext_control(codec); + tosa_ext_control(&rtd->card->dapm); return 0; } @@ -104,13 +102,13 @@ static int tosa_get_jack(struct snd_kcontrol *kcontrol, static int tosa_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (tosa_jack_func == ucontrol->value.integer.value[0]) return 0; tosa_jack_func = ucontrol->value.integer.value[0]; - tosa_ext_control(codec); + tosa_ext_control(&card->dapm); return 1; } @@ -124,13 +122,13 @@ static int tosa_get_spk(struct snd_kcontrol *kcontrol, static int tosa_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (tosa_spk_func == ucontrol->value.integer.value[0]) return 0; tosa_spk_func = ucontrol->value.integer.value[0]; - tosa_ext_control(codec); + tosa_ext_control(&card->dapm); return 1; } @@ -191,24 +189,10 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "MONOOUT"); - /* add tosa specific controls */ - err = snd_soc_add_codec_controls(codec, tosa_controls, - ARRAY_SIZE(tosa_controls)); - if (err < 0) - return err; - - /* add tosa specific widgets */ - snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets, - ARRAY_SIZE(tosa_dapm_widgets)); - - /* set up tosa specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -239,6 +223,13 @@ static struct snd_soc_card tosa = { .owner = THIS_MODULE, .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), + + .controls = tosa_controls, + .num_controls = ARRAY_SIZE(tosa_controls), + .dapm_widgets = tosa_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int tosa_probe(struct platform_device *pdev) -- cgit v1.1 From f410d5c953cf3c11629f138c5a2c3d3f40c61b5d Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 10 Mar 2014 10:38:40 -0600 Subject: ASoC: Intel: don't select RT5640 if !I2C MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The rt5640 driver won't compile without I2C enabled. Hence, the Intel Haswell and Baytrail+RT5640 ASoC drivers must also depend on I2C, since these select RT5640. This solves: sound/soc/codecs/rt5640.c:2220:1: warning: data definition has no type or storage class [enabled by default] sound/soc/codecs/rt5640.c:2220:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/rt5640.c:2220:1: warning: parameter names (without types) in function declaration [enabled by default] sound/soc/codecs/rt5640.c:2210:26: warning: ‘rt5640_i2c_driver’ defined but not used [-Wunused-variable] Reported-by: Jim Davis Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4577b69..3c81b38 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -32,7 +32,7 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -43,7 +43,7 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help -- cgit v1.1 From b92af2b8844c964fc7e1905f797a8d780c7212de Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Mar 2014 15:27:37 +0100 Subject: ASoC: s6105-ipcam: Convert to table based DAPM setup Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. While we are at it also remove the snd_soc_dapm_enable_pin() in the init callback, since pins are enabled by default. Also drop the snd_soc_dapm_sync() calls, since they are ignored by the core anyway until the card has been fully instantiated. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/s6000/s6105-ipcam.c | 28 ++++++++++------------------ 1 file changed, 10 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 945e8ab..0b21d1d 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -104,8 +104,8 @@ static int output_type_get(struct snd_kcontrol *kcontrol, static int output_type_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = kcontrol->private_data; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = kcontrol->private_data; + struct snd_soc_dapm_context *dapm = &card->dapm; unsigned int val = (ucontrol->value.enumerated.item[0] != 0); char *differential = "Audio Out Differential"; char *stereo = "Audio Out Stereo"; @@ -137,13 +137,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add s6105 specific widgets */ - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); - - /* Set up s6105 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + struct snd_soc_card *card = rtd->card; /* not present */ snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); @@ -157,17 +151,10 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "RLOUT"); snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Audio In"); - /* must correspond to audio_out_mux.private_value initializer */ - snd_soc_dapm_disable_pin(dapm, "Audio Out Differential"); - snd_soc_dapm_sync(dapm); - snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo"); - - snd_soc_dapm_sync(dapm); + snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential"); - snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec)); + snd_ctl_add(card->snd_card, snd_ctl_new1(&audio_out_mux, card)); return 0; } @@ -190,6 +177,11 @@ static struct snd_soc_card snd_soc_card_s6105 = { .owner = THIS_MODULE, .dai_link = &s6105_dai, .num_links = 1, + + .dapm_widgets = aic3x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct s6000_snd_platform_data s6105_snd_data __initdata = { -- cgit v1.1 From e00447fafbf7daf2cd49205b97e63d9734068a4f Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 11 Mar 2014 12:57:32 +0200 Subject: ASoC: tlv320aic31xx: Add basic codec driver implementation This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs. The driver is currently missing support at least for mini DSP features and jack detection. I have tested the driver only on TLV320AIC3111, but based on the data sheets TLV320AIC3100, TLV320AIC3110, and TLV320AIC3120 should work Ok too. The base for the implementation was taken from: git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx -branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1295 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++++++ 4 files changed, 1559 insertions(+) create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..66f6c53 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -73,6 +73,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C @@ -361,6 +362,9 @@ config SND_SOC_TLV320AIC26 tristate depends on SPI +config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC32X4 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1deeb20..ff1775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -64,6 +64,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o +snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o @@ -194,6 +195,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o +obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 0000000..e60e37b --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1295 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha + * + * Based on ground work by: Ajit Kulkarni + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + * The TLV320AIC31xx series of audio codec is a low-power, highly integrated + * high performance codec which provides a stereo DAC, a mono ADC, + * and mono/stereo Class-D speaker driver. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tlv320aic31xx.h" + +static const struct reg_default aic31xx_reg_defaults[] = { + { AIC31XX_CLKMUX, 0x00 }, + { AIC31XX_PLLPR, 0x11 }, + { AIC31XX_PLLJ, 0x04 }, + { AIC31XX_PLLDMSB, 0x00 }, + { AIC31XX_PLLDLSB, 0x00 }, + { AIC31XX_NDAC, 0x01 }, + { AIC31XX_MDAC, 0x01 }, + { AIC31XX_DOSRMSB, 0x00 }, + { AIC31XX_DOSRLSB, 0x80 }, + { AIC31XX_NADC, 0x01 }, + { AIC31XX_MADC, 0x01 }, + { AIC31XX_AOSR, 0x80 }, + { AIC31XX_IFACE1, 0x00 }, + { AIC31XX_DATA_OFFSET, 0x00 }, + { AIC31XX_IFACE2, 0x00 }, + { AIC31XX_BCLKN, 0x01 }, + { AIC31XX_DACSETUP, 0x14 }, + { AIC31XX_DACMUTE, 0x0c }, + { AIC31XX_LDACVOL, 0x00 }, + { AIC31XX_RDACVOL, 0x00 }, + { AIC31XX_ADCSETUP, 0x00 }, + { AIC31XX_ADCFGA, 0x80 }, + { AIC31XX_ADCVOL, 0x00 }, + { AIC31XX_HPDRIVER, 0x04 }, + { AIC31XX_SPKAMP, 0x06 }, + { AIC31XX_DACMIXERROUTE, 0x00 }, + { AIC31XX_LANALOGHPL, 0x7f }, + { AIC31XX_RANALOGHPR, 0x7f }, + { AIC31XX_LANALOGSPL, 0x7f }, + { AIC31XX_RANALOGSPR, 0x7f }, + { AIC31XX_HPLGAIN, 0x02 }, + { AIC31XX_HPRGAIN, 0x02 }, + { AIC31XX_SPLGAIN, 0x00 }, + { AIC31XX_SPRGAIN, 0x00 }, + { AIC31XX_MICBIAS, 0x00 }, + { AIC31XX_MICPGA, 0x80 }, + { AIC31XX_MICPGAPI, 0x00 }, + { AIC31XX_MICPGAMI, 0x00 }, +}; + +static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_PAGECTL: /* regmap implementation requires this */ + case AIC31XX_RESET: /* always clears after write */ + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return true; + } + return false; +} + +static bool aic31xx_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return false; + } + return true; +} + +static const struct regmap_range_cfg aic31xx_ranges[] = { + { + .range_min = 0, + .range_max = 12 * 128, + .selector_reg = AIC31XX_PAGECTL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +struct regmap_config aic31xx_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = aic31xx_writeable, + .volatile_reg = aic31xx_volatile, + .reg_defaults = aic31xx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = aic31xx_ranges, + .num_ranges = ARRAY_SIZE(aic31xx_ranges), + .max_register = 12 * 128, +}; + +#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = { + "HPVDD", + "SPRVDD", + "SPLVDD", + "AVDD", + "IOVDD", + "DVDD", +}; + +struct aic31xx_disable_nb { + struct notifier_block nb; + struct aic31xx_priv *aic31xx; +}; + +struct aic31xx_priv { + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct aic31xx_pdata pdata; + struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + unsigned int sysclk; + int rate_div_line; +}; + +struct aic31xx_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; +}; + +/* ADC dividers can be disabled by cofiguring them to 0 */ +static const struct aic31xx_rate_divs aic31xx_divs[] = { + /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ + /* 8k rate */ + {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, + {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, + /* 11.025k rate */ + {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, + {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, + /* 16k rate */ + {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, + {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, + /* 22.05k rate */ + {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, + {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, + /* 32k rate */ + {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, + {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, + /* 44.1k rate */ + {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, + {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, + /* 48k rate */ + {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, + {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, + /* 88.2k rate */ + {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, + {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, + /* 96k rate */ + {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, + {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, + /* 176.4k rate */ + {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, + {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, + /* 192k rate */ + {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, + {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, +}; + +static const char * const ldac_in_text[] = { + "Off", "Left Data", "Right Data", "Mono" +}; + +static const char * const rdac_in_text[] = { + "Off", "Right Data", "Left Data", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text); + +static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text); + +static const char * const mic_select_text[] = { + "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm" +}; + +static const +SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); + +static const +SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0); +static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); + +/* + * controls to be exported to the user space + */ +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, + AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv), + + SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, + adc_fgain_tlv), + + SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1), + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL, + 0, -24, 40, 6, 0, adc_cgain_tlv), + + SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, + 119, 0, mic_pga_tlv), + + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic311x_snd_controls[] = { + SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv), + + SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic310x_snd_controls[] = { + SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + 2, 1, 0), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + 3, 3, 0, class_D_drv_tlv), + + SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new ldac_in_control = + SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum); + +static const struct snd_kcontrol_new rdac_in_control = + SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); + +int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, + unsigned int mask, unsigned int wbits, int sleep, + int count) +{ + unsigned int bits; + int counter = count; + int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { + usleep_range(sleep, sleep * 2); + ret = regmap_read(aic31xx->regmap, reg, &bits); + counter--; + } + if ((bits & mask) != wbits) { + dev_err(aic31xx->dev, + "%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n", + __func__, reg, bits, wbits, ret, mask, + (count - counter) * sleep); + ret = -1; + } + return ret; +} + +#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg)) + +static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec); + unsigned int reg = AIC31XX_DACFLAG1; + unsigned int mask; + + switch (WIDGET_BIT(w->reg, w->shift)) { + case WIDGET_BIT(AIC31XX_DACSETUP, 7): + mask = AIC31XX_LDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_DACSETUP, 6): + mask = AIC31XX_RDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 7): + mask = AIC31XX_HPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 6): + mask = AIC31XX_HPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 7): + mask = AIC31XX_SPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 6): + mask = AIC31XX_SPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_ADCSETUP, 7): + mask = AIC31XX_ADCPWRSTATUS_MASK; + reg = AIC31XX_ADCFLAG; + break; + default: + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", + w->name, __func__); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); + case SND_SOC_DAPM_POST_PMD: + return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100); + default: + dev_dbg(w->codec->dev, + "Unhandled dapm widget event %d from %s\n", + event, w->name); + } + return 0; +} + +static const struct snd_kcontrol_new left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + +static const struct snd_kcontrol_new p_term_mic1lp = + SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1rp = + SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum); + +static const struct snd_kcontrol_new m_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum); + +static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0); + +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, + aic31xx->pdata.micbias_vg << + AIC31XX_MICBIAS_SHIFT); + dev_dbg(codec->dev, "%s: turned on\n", __func__); + break; + case SND_SOC_DAPM_PRE_PMD: + /* turn mic bias off */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, 0); + dev_dbg(codec->dev, "%s: turned off\n", __func__); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Left Input", + SND_SOC_NOPM, 0, 0, &ldac_in_control), + SND_SOC_DAPM_MUX("DAC Right Input", + SND_SOC_NOPM, 0, 0, &rdac_in_control), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback", + AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback", + AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + left_output_switches, + ARRAY_SIZE(left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + right_output_switches, + ARRAY_SIZE(right_output_switches)), + + SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpl_switch), + SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpr_switch), + + /* Output drivers */ + SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lp), + SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1rp), + SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lm), + + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, + &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ + SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, + 7, 1, NULL, 0), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), +}; + +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { + /* AIC3111 and AIC3110 have stereo class-D amplifier */ + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spr_switch), + SND_SOC_DAPM_OUTPUT("SPL"), + SND_SOC_DAPM_OUTPUT("SPR"), +}; + +/* AIC3100 and AIC3120 have only mono class-D amplifier */ +static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_OUTPUT("SPK"), +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { + /* DAC Input Routing */ + {"DAC Left Input", "Left Data", "DAC IN"}, + {"DAC Left Input", "Right Data", "DAC IN"}, + {"DAC Left Input", "Mono", "DAC IN"}, + {"DAC Right Input", "Left Data", "DAC IN"}, + {"DAC Right Input", "Right Data", "DAC IN"}, + {"DAC Right Input", "Mono", "DAC IN"}, + {"DAC Left", NULL, "DAC Left Input"}, + {"DAC Right", NULL, "DAC Right Input"}, + + /* Mic input */ + {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"}, + {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"}, + {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"}, + + {"ADC", NULL, "MIC_GAIN_CTL"}, + + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From MIC1LP", "MIC1LP"}, + {"Output Left", "From MIC1RP", "MIC1RP"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From MIC1RP", "MIC1RP"}, + + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +aic311x_audio_map[] = { + /* SP L path */ + {"Speaker Left", "Switch", "Output Left"}, + {"SPL ClassD", NULL, "Speaker Left"}, + {"SPL", NULL, "SPL ClassD"}, + + /* SP R path */ + {"Speaker Right", "Switch", "Output Right"}, + {"SPR ClassD", NULL, "Speaker Right"}, + {"SPR", NULL, "SPR ClassD"}, +}; + +static const struct snd_soc_dapm_route +aic310x_audio_map[] = { + /* SP L path */ + {"Speaker", "Switch", "Output Left"}, + {"SPK ClassD", NULL, "Speaker"}, + {"SPK", NULL, "SPK ClassD"}, +}; + +static int aic31xx_add_controls(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) + ret = snd_soc_add_codec_controls( + codec, aic311x_snd_controls, + ARRAY_SIZE(aic311x_snd_controls)); + else + ret = snd_soc_add_codec_controls( + codec, aic310x_snd_controls, + ARRAY_SIZE(aic310x_snd_controls)); + + return ret; +} + +static int aic31xx_add_widgets(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, aic311x_dapm_widgets, + ARRAY_SIZE(aic311x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map, + ARRAY_SIZE(aic311x_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic310x_dapm_widgets, + ARRAY_SIZE(aic310x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map, + ARRAY_SIZE(aic310x_audio_map)); + if (ret) + return ret; + } + + return 0; +} + +static int aic31xx_setup_pll(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_n = 0; + int i; + + /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, + AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK); + + for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { + if (aic31xx_divs[i].rate == params_rate(params) && + aic31xx_divs[i].mclk == aic31xx->sysclk) + break; + } + + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + __func__, params_rate(params)); + return -EINVAL; + } + + /* PLL configuration */ + snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, + (aic31xx_divs[i].p_val << 4) | 0x01); + snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); + + snd_soc_write(codec, AIC31XX_PLLDMSB, + aic31xx_divs[i].pll_d >> 8); + snd_soc_write(codec, AIC31XX_PLLDLSB, + aic31xx_divs[i].pll_d & 0xff); + + /* DAC dividers configuration */ + snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].ndac); + snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].mdac); + + snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8); + snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff); + + /* ADC dividers configuration. Write reset value 1 if not used. */ + snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1); + snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1); + + snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); + + /* Bit clock divider configuration. */ + bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) + / snd_soc_params_to_frame_size(params); + if (bclk_n == 0) { + dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", + __func__); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_BCLKN, + AIC31XX_PLL_MASK, bclk_n); + + aic31xx->rate_div_line = i; + + dev_dbg(codec->dev, + "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", + aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, + aic31xx_divs[i].p_val, aic31xx_divs[i].dosr, + aic31xx_divs[i].ndac, aic31xx_divs[i].mdac, + aic31xx_divs[i].aosr, aic31xx_divs[i].nadc, + aic31xx_divs[i].madc, bclk_n); + + return 0; +} + +static int aic31xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + u8 data = 0; + + dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", + __func__, params_format(params), params_width(params), + params_rate(params)); + + switch (params_width(params)) { + case 16: + break; + case 20: + data = (AIC31XX_WORD_LEN_20BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 24: + data = (AIC31XX_WORD_LEN_24BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 32: + data = (AIC31XX_WORD_LEN_32BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + default: + dev_err(codec->dev, "%s: Unsupported format %d\n", + __func__, params_format(params)); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATALEN_MASK, + data); + + return aic31xx_setup_pll(codec, params); +} + +static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, + AIC31XX_DACMUTE_MASK); + } else { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, 0x0); + } + + return 0; +} + +static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 iface_reg1 = 0; + u8 iface_reg3 = 0; + u8 dsp_a_val = 0; + + dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER; + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + dsp_a_val = 0x1; + case SND_SOC_DAIFMT_DSP_B: + /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface_reg3 |= AIC31XX_BCLKINV_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + iface_reg1 |= (AIC31XX_DSP_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATATYPE_MASK | + AIC31XX_IFACE1_MASTER_MASK, + iface_reg1); + snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET, + AIC31XX_DATA_OFFSET_MASK, + dsp_a_val); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BCLKINV_MASK, + iface_reg3); + + return 0; +} + +static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", + __func__, clk_id, freq, dir); + + for (i = 0; aic31xx_divs[i].mclk != freq; i++) { + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; + } + } + + /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK, + clk_id << AIC31XX_PLL_CLKIN_SHIFT); + + aic31xx->sysclk = freq; + return 0; +} + +static int aic31xx_regulator_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct aic31xx_disable_nb *disable_nb = + container_of(nb, struct aic31xx_disable_nb, nb); + struct aic31xx_priv *aic31xx = disable_nb->aic31xx; + + if (event & REGULATOR_EVENT_DISABLE) { + /* + * Put codec to reset and as at least one of the + * supplies was disabled. + */ + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + + regcache_mark_dirty(aic31xx->regmap); + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); + } + + return 0; +} + +static void aic31xx_clk_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 on = AIC31XX_PM_MASK; + + dev_dbg(codec->dev, "codec clock -> on (rate %d)\n", + aic31xx_divs[aic31xx->rate_div_line].rate); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on); + mdelay(10); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].nadc) + snd_soc_update_bits(codec, AIC31XX_NADC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].madc) + snd_soc_update_bits(codec, AIC31XX_MADC, mask, on); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on); +} + +static void aic31xx_clk_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 off = 0; + + dev_dbg(codec->dev, "codec clock -> off\n"); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off); + snd_soc_update_bits(codec, AIC31XX_MADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off); +} + +static int aic31xx_power_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret) + return ret; + + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) { + gpio_set_value(aic31xx->pdata.gpio_reset, 1); + udelay(100); + } + regcache_cache_only(aic31xx->regmap, false); + ret = regcache_sync(aic31xx->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + regcache_cache_only(aic31xx->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + return ret; + } + return 0; +} + +static int aic31xx_power_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + regcache_cache_only(aic31xx->regmap, true); + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + + return ret; +} + +static int aic31xx_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, + codec->dapm.bias_level, level); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_clk_on(codec); + break; + case SND_SOC_BIAS_STANDBY: + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + aic31xx_power_on(codec); + break; + case SND_SOC_BIAS_PREPARE: + aic31xx_clk_off(codec); + break; + default: + BUG(); + } + break; + case SND_SOC_BIAS_OFF: + aic31xx_power_off(codec); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int aic31xx_suspend(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic31xx_resume(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic31xx_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(aic31xx->dev, "## %s\n", __func__); + + aic31xx = snd_soc_codec_get_drvdata(codec); + codec->control_data = aic31xx->regmap; + + aic31xx->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret != 0) { + dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { + aic31xx->disable_nb[i].nb.notifier_call = + aic31xx_regulator_event; + aic31xx->disable_nb[i].aic31xx = aic31xx; + ret = regulator_register_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + if (ret) { + dev_err(codec->dev, + "Failed to request regulator notifier: %d\n", + ret); + return ret; + } + } + + regcache_cache_only(aic31xx->regmap, true); + regcache_mark_dirty(aic31xx->regmap); + + ret = aic31xx_add_controls(codec); + if (ret) + return ret; + + ret = aic31xx_add_widgets(codec); + + return ret; +} + +static int aic31xx_codec_remove(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + /* power down chip */ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + regulator_unregister_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { + .probe = aic31xx_codec_probe, + .remove = aic31xx_codec_remove, + .suspend = aic31xx_suspend, + .resume = aic31xx_resume, + .set_bias_level = aic31xx_set_bias_level, + .controls = aic31xx_snd_controls, + .num_controls = ARRAY_SIZE(aic31xx_snd_controls), + .dapm_widgets = aic31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), + .dapm_routes = aic31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), +}; + +static struct snd_soc_dai_ops aic31xx_dai_ops = { + .hw_params = aic31xx_hw_params, + .set_sysclk = aic31xx_set_dai_sysclk, + .set_fmt = aic31xx_set_dai_fmt, + .digital_mute = aic31xx_dac_mute, +}; + +static struct snd_soc_dai_driver aic31xx_dai_driver[] = { + { + .name = "tlv320aic31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + .symmetric_rates = 1, + } +}; + +#if defined(CONFIG_OF) +static const struct of_device_id tlv320aic31xx_of_match[] = { + { .compatible = "ti,tlv320aic310x" }, + { .compatible = "ti,tlv320aic311x" }, + { .compatible = "ti,tlv320aic3100" }, + { .compatible = "ti,tlv320aic3110" }, + { .compatible = "ti,tlv320aic3120" }, + { .compatible = "ti,tlv320aic3111" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match); + +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ + struct device_node *np = aic31xx->dev->of_node; + unsigned int value = MICBIAS_2_0V; + int ret; + + of_property_read_u32(np, "ai31xx-micbias-vg", &value); + switch (value) { + case MICBIAS_2_0V: + case MICBIAS_2_5V: + case MICBIAS_AVDDV: + aic31xx->pdata.micbias_vg = value; + break; + default: + dev_err(aic31xx->dev, + "Bad ai31xx-micbias-vg value %d DT\n", + value); + aic31xx->pdata.micbias_vg = MICBIAS_2_0V; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if (ret > 0) + aic31xx->pdata.gpio_reset = ret; +} +#else /* CONFIG_OF */ +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ +} +#endif /* CONFIG_OF */ + +void aic31xx_device_init(struct aic31xx_priv *aic31xx) +{ + int ret, i; + + dev_set_drvdata(aic31xx->dev, aic31xx); + + if (dev_get_platdata(aic31xx->dev)) + memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev), + sizeof(aic31xx->pdata)); + else if (aic31xx->dev->of_node) + aic31xx_pdata_from_of(aic31xx); + + if (aic31xx->pdata.gpio_reset) { + ret = devm_gpio_request_one(aic31xx->dev, + aic31xx->pdata.gpio_reset, + GPIOF_OUT_INIT_HIGH, + "aic31xx-reset-pin"); + if (ret < 0) { + dev_err(aic31xx->dev, "not able to acquire gpio\n"); + return; + } + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + aic31xx->supplies[i].supply = aic31xx_supply_names[i]; + + ret = devm_regulator_bulk_get(aic31xx->dev, + ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret != 0) + dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + +} + +static int aic31xx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic31xx_priv *aic31xx; + int ret; + const struct regmap_config *regmap_config; + + dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__, + id->name, (int) id->driver_data); + + regmap_config = &aic31xx_i2c_regmap; + + aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL); + if (aic31xx == NULL) + return -ENOMEM; + + aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); + + if (IS_ERR(aic31xx->regmap)) { + ret = PTR_ERR(aic31xx->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + aic31xx->dev = &i2c->dev; + + aic31xx->pdata.codec_type = id->driver_data; + + aic31xx_device_init(aic31xx); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); + + return ret; +} + +static int aic31xx_i2c_remove(struct i2c_client *i2c) +{ + struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); + + kfree(aic31xx); + return 0; +} + +static const struct i2c_device_id aic31xx_i2c_id[] = { + { "tlv320aic310x", AIC3100 }, + { "tlv320aic311x", AIC3110 }, + { "tlv320aic3100", AIC3100 }, + { "tlv320aic3110", AIC3110 }, + { "tlv320aic3120", AIC3120 }, + { "tlv320aic3111", AIC3111 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); + +static struct i2c_driver aic31xx_i2c_driver = { + .driver = { + .name = "tlv320aic31xx-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + }, + .probe = aic31xx_i2c_probe, + .remove = (aic31xx_i2c_remove), + .id_table = aic31xx_i2c_id, +}; + +module_i2c_driver(aic31xx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver"); +MODULE_AUTHOR("Jyri Sarha"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h new file mode 100644 index 0000000..52ed57c --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -0,0 +1,258 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2013 Texas Instruments, Inc. + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ +#ifndef _TLV320AIC31XX_H +#define _TLV320AIC31XX_H + +#define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 + +#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + + +#define AIC31XX_STEREO_CLASS_D_BIT 0x1 +#define AIC31XX_MINIDSP_BIT 0x2 + +enum aic31xx_type { + AIC3100 = 0, + AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, + AIC3120 = AIC31XX_MINIDSP_BIT, + AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), +}; + +struct aic31xx_pdata { + enum aic31xx_type codec_type; + unsigned int gpio_reset; + int micbias_vg; +}; + +/* Page Control Register */ +#define AIC31XX_PAGECTL 0x00 + +/* Page 0 Registers */ +/* Software reset register */ +#define AIC31XX_RESET 0x01 +/* OT FLAG register */ +#define AIC31XX_OT_FLAG 0x03 +/* Clock clock Gen muxing, Multiplexers*/ +#define AIC31XX_CLKMUX 0x04 +/* PLL P and R-VAL register */ +#define AIC31XX_PLLPR 0x05 +/* PLL J-VAL register */ +#define AIC31XX_PLLJ 0x06 +/* PLL D-VAL MSB register */ +#define AIC31XX_PLLDMSB 0x07 +/* PLL D-VAL LSB register */ +#define AIC31XX_PLLDLSB 0x08 +/* DAC NDAC_VAL register*/ +#define AIC31XX_NDAC 0x0B +/* DAC MDAC_VAL register */ +#define AIC31XX_MDAC 0x0C +/* DAC OSR setting register 1, MSB value */ +#define AIC31XX_DOSRMSB 0x0D +/* DAC OSR setting register 2, LSB value */ +#define AIC31XX_DOSRLSB 0x0E +#define AIC31XX_MINI_DSP_INPOL 0x10 +/* Clock setting register 8, PLL */ +#define AIC31XX_NADC 0x12 +/* Clock setting register 9, PLL */ +#define AIC31XX_MADC 0x13 +/* ADC Oversampling (AOSR) Register */ +#define AIC31XX_AOSR 0x14 +/* Clock setting register 9, Multiplexers */ +#define AIC31XX_CLKOUTMUX 0x19 +/* Clock setting register 10, CLOCKOUT M divider value */ +#define AIC31XX_CLKOUTMVAL 0x1A +/* Audio Interface Setting Register 1 */ +#define AIC31XX_IFACE1 0x1B +/* Audio Data Slot Offset Programming */ +#define AIC31XX_DATA_OFFSET 0x1C +/* Audio Interface Setting Register 2 */ +#define AIC31XX_IFACE2 0x1D +/* Clock setting register 11, BCLK N Divider */ +#define AIC31XX_BCLKN 0x1E +/* Audio Interface Setting Register 3, Secondary Audio Interface */ +#define AIC31XX_IFACESEC1 0x1F +/* Audio Interface Setting Register 4 */ +#define AIC31XX_IFACESEC2 0x20 +/* Audio Interface Setting Register 5 */ +#define AIC31XX_IFACESEC3 0x21 +/* I2C Bus Condition */ +#define AIC31XX_I2C 0x22 +/* ADC FLAG */ +#define AIC31XX_ADCFLAG 0x24 +/* DAC Flag Registers */ +#define AIC31XX_DACFLAG1 0x25 +#define AIC31XX_DACFLAG2 0x26 +/* Sticky Interrupt flag (overflow) */ +#define AIC31XX_OFFLAG 0x27 +/* Sticy DAC Interrupt flags */ +#define AIC31XX_INTRDACFLAG 0x2C +/* Sticy ADC Interrupt flags */ +#define AIC31XX_INTRADCFLAG 0x2D +/* DAC Interrupt flags 2 */ +#define AIC31XX_INTRDACFLAG2 0x2E +/* ADC Interrupt flags 2 */ +#define AIC31XX_INTRADCFLAG2 0x2F +/* INT1 interrupt control */ +#define AIC31XX_INT1CTRL 0x30 +/* INT2 interrupt control */ +#define AIC31XX_INT2CTRL 0x31 +/* GPIO1 control */ +#define AIC31XX_GPIO1 0x33 + +#define AIC31XX_DACPRB 0x3C +/* ADC Instruction Set Register */ +#define AIC31XX_ADCPRB 0x3D +/* DAC channel setup register */ +#define AIC31XX_DACSETUP 0x3F +/* DAC Mute and volume control register */ +#define AIC31XX_DACMUTE 0x40 +/* Left DAC channel digital volume control */ +#define AIC31XX_LDACVOL 0x41 +/* Right DAC channel digital volume control */ +#define AIC31XX_RDACVOL 0x42 +/* Headset detection */ +#define AIC31XX_HSDETECT 0x43 +/* ADC Digital Mic */ +#define AIC31XX_ADCSETUP 0x51 +/* ADC Digital Volume Control Fine Adjust */ +#define AIC31XX_ADCFGA 0x52 +/* ADC Digital Volume Control Coarse Adjust */ +#define AIC31XX_ADCVOL 0x53 + + +/* Page 1 Registers */ +/* Headphone drivers */ +#define AIC31XX_HPDRIVER 0x9F +/* Class-D Speakear Amplifier */ +#define AIC31XX_SPKAMP 0xA0 +/* HP Output Drivers POP Removal Settings */ +#define AIC31XX_HPPOP 0xA1 +/* Output Driver PGA Ramp-Down Period Control */ +#define AIC31XX_SPPGARAMP 0xA2 +/* DAC_L and DAC_R Output Mixer Routing */ +#define AIC31XX_DACMIXERROUTE 0xA3 +/* Left Analog Vol to HPL */ +#define AIC31XX_LANALOGHPL 0xA4 +/* Right Analog Vol to HPR */ +#define AIC31XX_RANALOGHPR 0xA5 +/* Left Analog Vol to SPL */ +#define AIC31XX_LANALOGSPL 0xA6 +/* Right Analog Vol to SPR */ +#define AIC31XX_RANALOGSPR 0xA7 +/* HPL Driver */ +#define AIC31XX_HPLGAIN 0xA8 +/* HPR Driver */ +#define AIC31XX_HPRGAIN 0xA9 +/* SPL Driver */ +#define AIC31XX_SPLGAIN 0xAA +/* SPR Driver */ +#define AIC31XX_SPRGAIN 0xAB +/* HP Driver Control */ +#define AIC31XX_HPCONTROL 0xAC +/* MIC Bias Control */ +#define AIC31XX_MICBIAS 0xAE +/* MIC PGA*/ +#define AIC31XX_MICPGA 0xAF +/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ +#define AIC31XX_MICPGAPI 0xB0 +/* ADC Input Selection for M-Terminal */ +#define AIC31XX_MICPGAMI 0xB1 +/* Input CM Settings */ +#define AIC31XX_MICPGACM 0xB2 + +/* Bits, masks and shifts */ + +/* AIC31XX_CLKMUX */ +#define AIC31XX_PLL_CLKIN_MASK 0x0c +#define AIC31XX_PLL_CLKIN_SHIFT 2 +#define AIC31XX_PLL_CLKIN_MCLK 0 +#define AIC31XX_CODEC_CLKIN_MASK 0x03 +#define AIC31XX_CODEC_CLKIN_SHIFT 0 +#define AIC31XX_CODEC_CLKIN_PLL 3 +#define AIC31XX_CODEC_CLKIN_BCLK 1 + +/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC, + AIC31XX_BCLKN */ +#define AIC31XX_PLL_MASK 0x7f +#define AIC31XX_PM_MASK 0x80 + +/* AIC31XX_IFACE1 */ +#define AIC31XX_WORD_LEN_16BITS 0x00 +#define AIC31XX_WORD_LEN_20BITS 0x01 +#define AIC31XX_WORD_LEN_24BITS 0x02 +#define AIC31XX_WORD_LEN_32BITS 0x03 +#define AIC31XX_IFACE1_DATALEN_MASK 0x30 +#define AIC31XX_IFACE1_DATALEN_SHIFT (4) +#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0 +#define AIC31XX_IFACE1_DATATYPE_SHIFT (6) +#define AIC31XX_I2S_MODE 0x00 +#define AIC31XX_DSP_MODE 0x01 +#define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC31XX_LEFT_JUSTIFIED_MODE 0x03 +#define AIC31XX_IFACE1_MASTER_MASK 0x0C +#define AIC31XX_BCLK_MASTER 0x08 +#define AIC31XX_WCLK_MASTER 0x04 + +/* AIC31XX_DATA_OFFSET */ +#define AIC31XX_DATA_OFFSET_MASK 0xFF + +/* AIC31XX_IFACE2 */ +#define AIC31XX_BCLKINV_MASK 0x08 +#define AIC31XX_BDIVCLK_MASK 0x03 +#define AIC31XX_DAC2BCLK 0x00 +#define AIC31XX_DACMOD2BCLK 0x01 +#define AIC31XX_ADC2BCLK 0x02 +#define AIC31XX_ADCMOD2BCLK 0x03 + +/* AIC31XX_ADCFLAG */ +#define AIC31XX_ADCPWRSTATUS_MASK 0x40 + +/* AIC31XX_DACFLAG1 */ +#define AIC31XX_LDACPWRSTATUS_MASK 0x80 +#define AIC31XX_RDACPWRSTATUS_MASK 0x08 +#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20 +#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02 +#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10 +#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01 + +/* AIC31XX_INTRDACFLAG */ +#define AIC31XX_HPSCDETECT_MASK 0x80 +#define AIC31XX_BUTTONPRESS_MASK 0x20 +#define AIC31XX_HSPLUG_MASK 0x10 +#define AIC31XX_LDRCTHRES_MASK 0x08 +#define AIC31XX_RDRCTHRES_MASK 0x04 +#define AIC31XX_DACSINT_MASK 0x02 +#define AIC31XX_DACAINT_MASK 0x01 + +/* AIC31XX_INT1CTRL */ +#define AIC31XX_HSPLUGDET_MASK 0x80 +#define AIC31XX_BUTTONPRESSDET_MASK 0x40 +#define AIC31XX_DRCTHRES_MASK 0x20 +#define AIC31XX_AGCNOISE_MASK 0x10 +#define AIC31XX_OC_MASK 0x08 +#define AIC31XX_ENGINE_MASK 0x04 + +/* AIC31XX_DACSETUP */ +#define AIC31XX_SOFTSTEP_MASK 0x03 + +/* AIC31XX_DACMUTE */ +#define AIC31XX_DACMUTE_MASK 0x0C + +/* AIC31XX_MICBIAS */ +#define AIC31XX_MICBIAS_MASK 0x03 +#define AIC31XX_MICBIAS_SHIFT 0 + +#endif /* _TLV320AIC31XX_H */ -- cgit v1.1