From ea5edfe2f1ce5b2254a5ec4c1bb224fac48c3153 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 4 Aug 2014 15:04:19 +0530 Subject: ASoC: Intel: Fix to use byte control interface Using a byte control interface instead of generic_params ioctl. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform.h | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c6a42c..cc3a088 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -63,9 +63,7 @@ enum sst_controls { SST_SND_BUFFER_POINTER = 0x05, SST_SND_STREAM_INIT = 0x06, SST_SND_START = 0x07, - SST_SET_BYTE_STREAM = 0x100A, - SST_GET_BYTE_STREAM = 0x100B, - SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, + SST_MAX_CONTROLS = 0x07, }; enum sst_stream_ops { @@ -129,7 +127,7 @@ struct compress_sst_ops { struct sst_ops { int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); - int (*set_generic_params)(enum sst_controls cmd, void *arg); + int (*send_byte_stream)(struct snd_sst_bytes_v2 *bytes); int (*close) (unsigned int str_id); }; -- cgit v1.1 From 5981c2d6db2ef16d96ee4d1c4d3ddff4ad9d8ebc Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 4 Aug 2014 15:04:20 +0530 Subject: ASoC: Intel: mfld-pcm: Use function instead of ioctl Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 21 +++++++++------------ sound/soc/intel/sst-mfld-platform.h | 19 ++++++------------- 2 files changed, 15 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 706212a..42766a5 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -314,8 +314,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->device_control( - SST_SND_STREAM_INIT, &stream->stream_info); + ret_val = stream->ops->stream_init(&stream->stream_info); if (ret_val) pr_err("control_set ret error %d\n", ret_val); return ret_val; @@ -403,8 +402,7 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, 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); + ret_val = stream->ops->stream_drop(str_id); return ret_val; } @@ -461,7 +459,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, { int ret_val = 0, str_id; struct sst_runtime_stream *stream; - int str_cmd, status; + int status; pr_debug("sst_platform_pcm_trigger called\n"); stream = substream->runtime->private_data; @@ -469,29 +467,29 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, 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.arg = substream; + ret_val = stream->ops->stream_start(str_id); break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); - str_cmd = SST_SND_DROP; status = SST_PLATFORM_DROPPED; + ret_val = stream->ops->stream_drop(str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: pr_debug("sst: in pause\n"); - str_cmd = SST_SND_PAUSE; status = SST_PLATFORM_PAUSED; + ret_val = stream->ops->stream_pause(str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pr_debug("sst: in pause release\n"); - str_cmd = SST_SND_RESUME; status = SST_PLATFORM_RUNNING; + ret_val = stream->ops->stream_pause_release(str_id); break; default: return -EINVAL; } - ret_val = stream->ops->device_control(str_cmd, &str_id); + if (!ret_val) sst_set_stream_status(stream, status); @@ -511,8 +509,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->ops->device_control( - SST_SND_BUFFER_POINTER, str_info); + ret_val = stream->ops->stream_read_tstamp(str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); return ret_val; diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index cc3a088..2d6e65b 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -54,18 +54,6 @@ enum sst_drv_status { 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, @@ -126,7 +114,12 @@ struct compress_sst_ops { struct sst_ops { int (*open) (struct snd_sst_params *str_param); - int (*device_control) (int cmd, void *arg); + int (*stream_init) (struct pcm_stream_info *str_info); + int (*stream_start) (int str_id); + int (*stream_drop) (int str_id); + int (*stream_pause) (int str_id); + int (*stream_pause_release) (int str_id); + int (*stream_read_tstamp) (struct pcm_stream_info *str_info); int (*send_byte_stream)(struct snd_sst_bytes_v2 *bytes); int (*close) (unsigned int str_id); }; -- cgit v1.1 From b12b087c8715286b8759016f1d5c36cac0bb37f6 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 4 Aug 2014 15:04:21 +0530 Subject: ASoC: Intel: mfld-pcm: Change sst_ops prototypes to take dev parameter sst_ops need to use the sst driver context. So pass sst device as argument, which can be used to retrieve sst context. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 19 +++++++++---------- sound/soc/intel/sst-mfld-platform.h | 18 +++++++++--------- 2 files changed, 18 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 42766a5..a89ff7e 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -277,7 +277,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, stream->stream_info.str_id = str_params.stream_id; - ret_val = stream->ops->open(&str_params); + ret_val = stream->ops->open(sst->dev, &str_params); if (ret_val <= 0) return ret_val; @@ -314,13 +314,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->stream_init(&stream->stream_info); + ret_val = stream->ops->stream_init(sst->dev, &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_media_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -372,7 +371,7 @@ static void sst_media_close(struct snd_pcm_substream *substream, stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) - ret_val = stream->ops->close(str_id); + ret_val = stream->ops->close(sst->dev, str_id); module_put(sst->dev->driver->owner); kfree(stream); } @@ -402,7 +401,7 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->ops->stream_drop(str_id); + ret_val = stream->ops->stream_drop(sst->dev, str_id); return ret_val; } @@ -469,22 +468,22 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, pr_debug("sst: Trigger Start\n"); status = SST_PLATFORM_RUNNING; stream->stream_info.arg = substream; - ret_val = stream->ops->stream_start(str_id); + ret_val = stream->ops->stream_start(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); status = SST_PLATFORM_DROPPED; - ret_val = stream->ops->stream_drop(str_id); + ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: pr_debug("sst: in pause\n"); status = SST_PLATFORM_PAUSED; - ret_val = stream->ops->stream_pause(str_id); + ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pr_debug("sst: in pause release\n"); status = SST_PLATFORM_RUNNING; - ret_val = stream->ops->stream_pause_release(str_id); + ret_val = stream->ops->stream_pause_release(sst->dev, str_id); break; default: return -EINVAL; @@ -509,7 +508,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->ops->stream_read_tstamp(str_info); + ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); return ret_val; diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 2d6e65b..d4c28b8 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -113,15 +113,15 @@ struct compress_sst_ops { }; struct sst_ops { - int (*open) (struct snd_sst_params *str_param); - int (*stream_init) (struct pcm_stream_info *str_info); - int (*stream_start) (int str_id); - int (*stream_drop) (int str_id); - int (*stream_pause) (int str_id); - int (*stream_pause_release) (int str_id); - int (*stream_read_tstamp) (struct pcm_stream_info *str_info); - int (*send_byte_stream)(struct snd_sst_bytes_v2 *bytes); - int (*close) (unsigned int str_id); + int (*open) (struct device *dev, struct snd_sst_params *str_param); + int (*stream_init) (struct device *dev, struct pcm_stream_info *str_info); + int (*stream_start) (struct device *dev, int str_id); + int (*stream_drop) (struct device *dev, int str_id); + int (*stream_pause) (struct device *dev, int str_id); + int (*stream_pause_release) (struct device *dev, int str_id); + int (*stream_read_tstamp) (struct device *dev, struct pcm_stream_info *str_info); + int (*send_byte_stream)(struct device *dev, struct snd_sst_bytes_v2 *bytes); + int (*close) (struct device *dev, unsigned int str_id); }; struct sst_runtime_stream { -- cgit v1.1 From d8499c9b4b03ca88d7c7b4094cb09471658df7c2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 4 Aug 2014 15:15:55 +0530 Subject: ASoC: Intel: add mrfld DSP defines We define the DSP commands,structures here which will be used to send the IPCs Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 3 +- sound/soc/intel/sst-atom-controls.c | 39 +++++ sound/soc/intel/sst-atom-controls.h | 286 +++++++++++++++++++++++++++++++- sound/soc/intel/sst-mfld-platform-pcm.c | 8 +- sound/soc/intel/sst-mfld-platform.h | 3 + 5 files changed, 335 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/sst-atom-controls.c (limited to 'sound/soc') diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 7acbfc4..f841786 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -2,7 +2,8 @@ snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o snd-soc-sst-acpi-objs := sst-acpi.o -snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o sst-mfld-platform-compress.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \ + sst-mfld-platform-compress.o sst-atom-controls.o snd-soc-mfld-machine-objs := mfld_machine.o obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c new file mode 100644 index 0000000..ace3c4a --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.c @@ -0,0 +1,39 @@ +/* + * sst-atom-controls.c - Intel MID Platform driver DPCM ALSA controls for Mrfld + * + * Copyright (C) 2013-14 Intel Corp + * Author: Omair Mohammed Abdullah + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include "sst-mfld-platform.h" +#include "sst-atom-controls.h" + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) +{ + int ret = 0; + struct sst_data *drv = snd_soc_platform_get_drvdata(platform); + + drv->byte_stream = devm_kzalloc(platform->dev, + SST_MAX_BIN_BYTES, GFP_KERNEL); + if (!drv->byte_stream) + return -ENOMEM; + + return ret; +} diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index 14063ab..8554889 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -1,4 +1,6 @@ /* + * sst-atom-controls.h - Intel MID Platform driver header file + * * Copyright (C) 2013-14 Intel Corp * Author: Ramesh Babu * Omair M Abdullah @@ -18,13 +20,293 @@ * */ -#ifndef __SST_CONTROLS_V2_H__ -#define __SST_CONTROLS_V2_H__ +#ifndef __SST_ATOM_CONTROLS_H__ +#define __SST_ATOM_CONTROLS_H__ enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_COMPR, }; +/* define a bit for each mixer input */ +#define SST_MIX_IP(x) (x) + +#define SST_IP_CODEC0 SST_MIX_IP(2) +#define SST_IP_CODEC1 SST_MIX_IP(3) +#define SST_IP_LOOP0 SST_MIX_IP(4) +#define SST_IP_LOOP1 SST_MIX_IP(5) +#define SST_IP_LOOP2 SST_MIX_IP(6) +#define SST_IP_PROBE SST_MIX_IP(7) +#define SST_IP_VOIP SST_MIX_IP(12) +#define SST_IP_PCM0 SST_MIX_IP(13) +#define SST_IP_PCM1 SST_MIX_IP(14) +#define SST_IP_MEDIA0 SST_MIX_IP(17) +#define SST_IP_MEDIA1 SST_MIX_IP(18) +#define SST_IP_MEDIA2 SST_MIX_IP(19) +#define SST_IP_MEDIA3 SST_MIX_IP(20) + +#define SST_IP_LAST SST_IP_MEDIA3 + +#define SST_SWM_INPUT_COUNT (SST_IP_LAST + 1) +#define SST_CMD_SWM_MAX_INPUTS 6 + +#define SST_PATH_ID_SHIFT 8 +#define SST_DEFAULT_LOCATION_ID 0xFFFF +#define SST_DEFAULT_CELL_NBR 0xFF +#define SST_DEFAULT_MODULE_ID 0xFFFF + +/* + * Audio DSP Path Ids. Specified by the audio DSP FW + */ +enum sst_path_index { + SST_PATH_INDEX_CODEC_OUT0 = (0x02 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_OUT1 = (0x03 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_OUT = (0x04 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_OUT = (0x05 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_OUT = (0x06 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_OUT = (0x0C << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM0_OUT = (0x0D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_OUT = (0x0E << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM2_OUT = (0x0F << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_OUT = (0x12 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_OUT = (0x13 << SST_PATH_ID_SHIFT), + + + /* Start of input paths */ + SST_PATH_INDEX_CODEC_IN0 = (0x82 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_IN1 = (0x83 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_IN = (0x84 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_IN = (0x85 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_IN = (0x86 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_IN = (0x8C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_PCM0_IN = (0x8D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_IN = (0x8E << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_IN = (0x8F << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_IN = (0x90 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA2_IN = (0x91 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA3_IN = (0x9C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_RESERVED = (0xFF << SST_PATH_ID_SHIFT), +}; + +/* + * path IDs + */ +enum sst_swm_inputs { + SST_SWM_IN_CODEC0 = (SST_PATH_INDEX_CODEC_IN0 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_CODEC1 = (SST_PATH_INDEX_CODEC_IN1 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_VOIP = (SST_PATH_INDEX_VOIP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM0 = (SST_PATH_INDEX_PCM0_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM1 = (SST_PATH_INDEX_PCM1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA0 = (SST_PATH_INDEX_MEDIA0_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA1 = (SST_PATH_INDEX_MEDIA1_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA2 = (SST_PATH_INDEX_MEDIA2_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA3 = (SST_PATH_INDEX_MEDIA3_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR) +}; + +/* + * path IDs + */ +enum sst_swm_outputs { + SST_SWM_OUT_CODEC0 = (SST_PATH_INDEX_CODEC_OUT0 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_CODEC1 = (SST_PATH_INDEX_CODEC_OUT1 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_VOIP = (SST_PATH_INDEX_VOIP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM0 = (SST_PATH_INDEX_PCM0_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM1 = (SST_PATH_INDEX_PCM1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM2 = (SST_PATH_INDEX_PCM2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA0 = (SST_PATH_INDEX_MEDIA0_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_MEDIA1 = (SST_PATH_INDEX_MEDIA1_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR), +}; + +enum sst_ipc_msg { + SST_IPC_IA_CMD = 1, + SST_IPC_IA_SET_PARAMS, + SST_IPC_IA_GET_PARAMS, +}; + +enum sst_cmd_type { + SST_CMD_BYTES_SET = 1, + SST_CMD_BYTES_GET = 2, +}; + +enum sst_task { + SST_TASK_SBA = 1, + SST_TASK_MMX, +}; + +enum sst_type { + SST_TYPE_CMD = 1, + SST_TYPE_PARAMS, +}; + +enum sst_flag { + SST_FLAG_BLOCKED = 1, + SST_FLAG_NONBLOCK, +}; + +/* + * Enumeration for indexing the gain cells in VB_SET_GAIN DSP command + */ +enum sst_gain_index { + /* GAIN IDs for SB task start here */ + SST_GAIN_INDEX_CODEC_OUT0, + SST_GAIN_INDEX_CODEC_OUT1, + SST_GAIN_INDEX_CODEC_IN0, + SST_GAIN_INDEX_CODEC_IN1, + + SST_GAIN_INDEX_SPROT_LOOP_OUT, + SST_GAIN_INDEX_MEDIA_LOOP1_OUT, + SST_GAIN_INDEX_MEDIA_LOOP2_OUT, + + SST_GAIN_INDEX_PCM0_IN_LEFT, + SST_GAIN_INDEX_PCM0_IN_RIGHT, + + SST_GAIN_INDEX_PCM1_OUT_LEFT, + SST_GAIN_INDEX_PCM1_OUT_RIGHT, + SST_GAIN_INDEX_PCM1_IN_LEFT, + SST_GAIN_INDEX_PCM1_IN_RIGHT, + SST_GAIN_INDEX_PCM2_OUT_LEFT, + + SST_GAIN_INDEX_PCM2_OUT_RIGHT, + SST_GAIN_INDEX_VOIP_OUT, + SST_GAIN_INDEX_VOIP_IN, + + /* Gain IDs for MMX task start here */ + SST_GAIN_INDEX_MEDIA0_IN_LEFT, + SST_GAIN_INDEX_MEDIA0_IN_RIGHT, + SST_GAIN_INDEX_MEDIA1_IN_LEFT, + SST_GAIN_INDEX_MEDIA1_IN_RIGHT, + + SST_GAIN_INDEX_MEDIA2_IN_LEFT, + SST_GAIN_INDEX_MEDIA2_IN_RIGHT, + + SST_GAIN_INDEX_GAIN_END +}; + +/* + * Audio DSP module IDs specified by FW spec + * TODO: Update with all modules + */ +enum sst_module_id { + SST_MODULE_ID_PCM = 0x0001, + SST_MODULE_ID_MP3 = 0x0002, + SST_MODULE_ID_MP24 = 0x0003, + SST_MODULE_ID_AAC = 0x0004, + SST_MODULE_ID_AACP = 0x0005, + SST_MODULE_ID_EAACP = 0x0006, + SST_MODULE_ID_WMA9 = 0x0007, + SST_MODULE_ID_WMA10 = 0x0008, + SST_MODULE_ID_WMA10P = 0x0009, + SST_MODULE_ID_RA = 0x000A, + SST_MODULE_ID_DDAC3 = 0x000B, + SST_MODULE_ID_TRUE_HD = 0x000C, + SST_MODULE_ID_HD_PLUS = 0x000D, + + SST_MODULE_ID_SRC = 0x0064, + SST_MODULE_ID_DOWNMIX = 0x0066, + SST_MODULE_ID_GAIN_CELL = 0x0067, + SST_MODULE_ID_SPROT = 0x006D, + SST_MODULE_ID_BASS_BOOST = 0x006E, + SST_MODULE_ID_STEREO_WDNG = 0x006F, + SST_MODULE_ID_AV_REMOVAL = 0x0070, + SST_MODULE_ID_MIC_EQ = 0x0071, + SST_MODULE_ID_SPL = 0x0072, + SST_MODULE_ID_ALGO_VTSV = 0x0073, + SST_MODULE_ID_NR = 0x0076, + SST_MODULE_ID_BWX = 0x0077, + SST_MODULE_ID_DRP = 0x0078, + SST_MODULE_ID_MDRP = 0x0079, + + SST_MODULE_ID_ANA = 0x007A, + SST_MODULE_ID_AEC = 0x007B, + SST_MODULE_ID_NR_SNS = 0x007C, + SST_MODULE_ID_SER = 0x007D, + SST_MODULE_ID_AGC = 0x007E, + + SST_MODULE_ID_CNI = 0x007F, + SST_MODULE_ID_CONTEXT_ALGO_AWARE = 0x0080, + SST_MODULE_ID_FIR_24 = 0x0081, + SST_MODULE_ID_IIR_24 = 0x0082, + + SST_MODULE_ID_ASRC = 0x0083, + SST_MODULE_ID_TONE_GEN = 0x0084, + SST_MODULE_ID_BMF = 0x0086, + SST_MODULE_ID_EDL = 0x0087, + SST_MODULE_ID_GLC = 0x0088, + + SST_MODULE_ID_FIR_16 = 0x0089, + SST_MODULE_ID_IIR_16 = 0x008A, + SST_MODULE_ID_DNR = 0x008B, + + SST_MODULE_ID_VIRTUALIZER = 0x008C, + SST_MODULE_ID_VISUALIZATION = 0x008D, + SST_MODULE_ID_LOUDNESS_OPTIMIZER = 0x008E, + SST_MODULE_ID_REVERBERATION = 0x008F, + + SST_MODULE_ID_CNI_TX = 0x0090, + SST_MODULE_ID_REF_LINE = 0x0091, + SST_MODULE_ID_VOLUME = 0x0092, + SST_MODULE_ID_FILT_DCR = 0x0094, + SST_MODULE_ID_SLV = 0x009A, + SST_MODULE_ID_NLF = 0x009B, + SST_MODULE_ID_TNR = 0x009C, + SST_MODULE_ID_WNR = 0x009D, + + SST_MODULE_ID_LOG = 0xFF00, + + SST_MODULE_ID_TASK = 0xFFFF, +}; + +enum sst_cmd { + SBA_IDLE = 14, + SBA_VB_SET_SPEECH_PATH = 26, + MMX_SET_GAIN = 33, + SBA_VB_SET_GAIN = 33, + FBA_VB_RX_CNI = 35, + MMX_SET_GAIN_TIMECONST = 36, + SBA_VB_SET_TIMECONST = 36, + SBA_VB_START = 85, + SBA_SET_SWM = 114, + SBA_SET_MDRP = 116, + SBA_HW_SET_SSP = 117, + SBA_SET_MEDIA_LOOP_MAP = 118, + SBA_SET_MEDIA_PATH = 119, + MMX_SET_MEDIA_PATH = 119, + SBA_VB_LPRO = 126, + SBA_VB_SET_FIR = 128, + SBA_VB_SET_IIR = 129, + SBA_SET_SSP_SLOT_MAP = 130, +}; + +enum sst_dsp_switch { + SST_SWITCH_OFF = 0, + SST_SWITCH_ON = 3, +}; + +enum sst_path_switch { + SST_PATH_OFF = 0, + SST_PATH_ON = 1, +}; + +enum sst_swm_state { + SST_SWM_OFF = 0, + SST_SWM_ON = 3, +}; #endif diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index a89ff7e..8e1e9bc 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -550,7 +550,13 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) return retval; } -static struct snd_soc_platform_driver sst_soc_platform_drv = { +static int sst_soc_probe(struct snd_soc_platform *platform) +{ + return sst_dsp_init_v2_dpcm(platform); +} + +static struct snd_soc_platform_driver sst_soc_platform_drv = { + .probe = sst_soc_probe, .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index d4c28b8..faaba10 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -143,6 +143,8 @@ struct sst_device { }; struct sst_data; + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform); void sst_set_stream_status(struct sst_runtime_stream *stream, int state); int sst_fill_stream_params(void *substream, const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress); @@ -157,6 +159,7 @@ struct sst_algo_int_control_v2 { struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; + char *byte_stream; struct mutex lock; }; int sst_register_dsp(struct sst_device *sst); -- cgit v1.1 From 81c7cfd1b22a0ee5e40efef72ec2cd17dbf12e6d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:18 +0200 Subject: ASoC: Move debugfs registration to the component level The debugfs registration is mostly identical between platforms and CODECs. This patches consolidates the two implementations at the component level. Unfortunately there are still a couple of CODEC specific debugfs files that are related to legacy ASoC IO that need to be registered. For this a new callback is added to the component struct that will be initialized when a CODEC is registered and will be used to register the CODEC specific files. Once there are no drivers left using legacy IO this can be removed again. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 122 ++++++++++++++++++++++----------------------------- 1 file changed, 52 insertions(+), 70 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4bfd4a..79371a7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -270,79 +270,56 @@ static const struct file_operations codec_reg_fops = { .llseek = default_llseek, }; -static struct dentry *soc_debugfs_create_dir(struct dentry *parent, - const char *fmt, ...) +static void soc_init_component_debugfs(struct snd_soc_component *component) { - struct dentry *de; - va_list ap; - char *s; + if (component->debugfs_prefix) { + char *name; - va_start(ap, fmt); - s = kvasprintf(GFP_KERNEL, fmt, ap); - va_end(ap); + name = kasprintf(GFP_KERNEL, "%s:%s", + component->debugfs_prefix, component->name); + if (name) { + component->debugfs_root = debugfs_create_dir(name, + component->card->debugfs_card_root); + kfree(name); + } + } else { + component->debugfs_root = debugfs_create_dir(component->name, + component->card->debugfs_card_root); + } - if (!s) - return NULL; + if (!component->debugfs_root) { + dev_warn(component->dev, + "ASoC: Failed to create component debugfs directory\n"); + return; + } - de = debugfs_create_dir(s, parent); - kfree(s); + snd_soc_dapm_debugfs_init(snd_soc_component_get_dapm(component), + component->debugfs_root); - return de; + if (component->init_debugfs) + component->init_debugfs(component); } -static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +static void soc_cleanup_component_debugfs(struct snd_soc_component *component) { - struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root; + debugfs_remove_recursive(component->debugfs_root); +} - codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, - "codec:%s", - codec->component.name); - if (!codec->debugfs_codec_root) { - dev_warn(codec->dev, - "ASoC: Failed to create codec debugfs directory\n"); - return; - } +static void soc_init_codec_debugfs(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root, + debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root, + debugfs_create_bool("cache_only", 0444, codec->component.debugfs_root, &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->debugfs_codec_root, + codec->component.debugfs_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) dev_warn(codec->dev, "ASoC: Failed to create codec register debugfs file\n"); - - snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); -} - -static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ - debugfs_remove_recursive(codec->debugfs_codec_root); -} - -static void soc_init_platform_debugfs(struct snd_soc_platform *platform) -{ - struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root; - - platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, - "platform:%s", - platform->component.name); - if (!platform->debugfs_platform_root) { - dev_warn(platform->dev, - "ASoC: Failed to create platform debugfs directory\n"); - return; - } - - snd_soc_dapm_debugfs_init(&platform->component.dapm, - platform->debugfs_platform_root); -} - -static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) -{ - debugfs_remove_recursive(platform->debugfs_platform_root); } static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, @@ -474,19 +451,15 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card) #else -static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ -} +#define soc_init_codec_debugfs NULL -static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +static inline void soc_init_component_debugfs( + struct snd_soc_component *component) { } -static inline void soc_init_platform_debugfs(struct snd_soc_platform *platform) -{ -} - -static inline void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +static inline void soc_cleanup_component_debugfs( + struct snd_soc_component *component) { } @@ -1026,7 +999,7 @@ static int soc_remove_platform(struct snd_soc_platform *platform) /* Make sure all DAPM widgets are freed */ snd_soc_dapm_free(&platform->component.dapm); - soc_cleanup_platform_debugfs(platform); + soc_cleanup_component_debugfs(&platform->component); platform->probed = 0; module_put(platform->dev->driver->owner); @@ -1046,7 +1019,7 @@ static void soc_remove_codec(struct snd_soc_codec *codec) /* Make sure all DAPM widgets are freed */ snd_soc_dapm_free(&codec->dapm); - soc_cleanup_codec_debugfs(codec); + soc_cleanup_component_debugfs(&codec->component); codec->probed = 0; list_del(&codec->card_list); module_put(codec->dev->driver->owner); @@ -1187,7 +1160,7 @@ static int soc_probe_codec(struct snd_soc_card *card, if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; - soc_init_codec_debugfs(codec); + soc_init_component_debugfs(&codec->component); if (driver->dapm_widgets) { ret = snd_soc_dapm_new_controls(&codec->dapm, @@ -1242,7 +1215,7 @@ static int soc_probe_codec(struct snd_soc_card *card, return 0; err_probe: - soc_cleanup_codec_debugfs(codec); + soc_cleanup_component_debugfs(&codec->component); module_put(codec->dev->driver->owner); return ret; @@ -1262,7 +1235,7 @@ static int soc_probe_platform(struct snd_soc_card *card, if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; - soc_init_platform_debugfs(platform); + soc_init_component_debugfs(&platform->component); if (driver->dapm_widgets) snd_soc_dapm_new_controls(&platform->component.dapm, @@ -1302,7 +1275,7 @@ static int soc_probe_platform(struct snd_soc_card *card, return 0; err_probe: - soc_cleanup_platform_debugfs(platform); + soc_cleanup_component_debugfs(&platform->component); module_put(platform->dev->driver->owner); return ret; @@ -4266,6 +4239,10 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, if (platform_drv->read) platform->component.read = snd_soc_platform_drv_read; +#ifdef CONFIG_DEBUG_FS + platform->component.debugfs_prefix = "platform"; +#endif + mutex_lock(&client_mutex); snd_soc_component_add_unlocked(&platform->component); list_add(&platform->list, &platform_list); @@ -4455,6 +4432,11 @@ int snd_soc_register_codec(struct device *dev, codec->component.val_bytes = codec_drv->reg_word_size; mutex_init(&codec->mutex); +#ifdef CONFIG_DEBUG_FS + codec->component.init_debugfs = soc_init_codec_debugfs; + codec->component.debugfs_prefix = "codec"; +#endif + if (!codec->component.write) { if (codec_drv->get_regmap) regmap = codec_drv->get_regmap(dev); -- cgit v1.1 From f1d45cc3ae96a6173129b2c164c216272faa5fc0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:19 +0200 Subject: ASoC: Consolidate platform and CODEC probe/remove The platform and CODEC probe and remove code is now largely identical. This patch consolidates it at the component level. The resulting code is slightly larger due to all the boiler plate code setting up the indirection for the table based control and DAPM registration. Once all drivers have been update to no longer use the snd_soc_codec_driver and snd_soc_platform_driver specific fields for this the indirection can be removed again. This patch contains two noteworthy hacks that are only meant to be temporary to be able to update drivers and the core in separate incremental patches. The first hack is related to that some DPCM platforms expect that the DAPM widgets for the DAIs of a snd_soc_component are created in the DAPM context of the snd_soc_platform that has the same parent device. For handling this the steal_sibling_dai_widgets attribute is introduced. It gets set for snd_soc_platforms that register DAPM elements. When creating the DAI widgets for a component this flag is checked and if it is found on one of the siblings the component will not create any DAI widgets in its own DAPM context. If the attribute is set on a platform it will look for siblings components and create DAI widgets for them in its own context. The fix for this will be to update the offending drivers to only register a single component rather than two. The second hack deals with the fact that the ASoC card suspend and resume code still needs a list of CODECs that have been registered for the card. To handle this the generic probe and remove path have a check to see if the component is CODEC and if yes add/remove it to the card's CODEC list. While it is possible to clean up the suspend/resume code to not need the CODEC list anymore this is a bit of a chicken and egg problem since it will become easier to clean up the suspend/resume code once there is a unified component layer. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 335 ++++++++++++++++++---------------- sound/soc/soc-generic-dmaengine-pcm.c | 4 +- 2 files changed, 177 insertions(+), 162 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 79371a7..b833cc6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -985,44 +985,20 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return 0; } -static int soc_remove_platform(struct snd_soc_platform *platform) +static void soc_remove_component(struct snd_soc_component *component) { - int ret; - - if (platform->driver->remove) { - ret = platform->driver->remove(platform); - if (ret < 0) - dev_err(platform->dev, "ASoC: failed to remove %d\n", - ret); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->component.dapm); - - soc_cleanup_component_debugfs(&platform->component); - platform->probed = 0; - module_put(platform->dev->driver->owner); - - return 0; -} - -static void soc_remove_codec(struct snd_soc_codec *codec) -{ - int err; + /* This is a HACK and will be removed soon */ + if (component->codec) + list_del(&component->codec->card_list); - if (codec->driver->remove) { - err = codec->driver->remove(codec); - if (err < 0) - dev_err(codec->dev, "ASoC: failed to remove %d\n", err); - } + if (component->remove) + component->remove(component); - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&codec->dapm); + snd_soc_dapm_free(snd_soc_component_get_dapm(component)); - soc_cleanup_component_debugfs(&codec->component); - codec->probed = 0; - list_del(&codec->card_list); - module_put(codec->dev->driver->owner); + soc_cleanup_component_debugfs(component); + component->probed = 0; + module_put(component->dev->driver->owner); } static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) @@ -1086,25 +1062,24 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, int i; /* remove the platform */ - if (platform && platform->probed && - platform->driver->remove_order == order) { - soc_remove_platform(platform); - } + if (platform && platform->component.probed && + platform->component.driver->remove_order == order) + soc_remove_component(&platform->component); /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { codec = rtd->codec_dais[i]->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + if (codec && codec->component.probed && + codec->component.driver->remove_order == order) + soc_remove_component(&codec->component); } /* remove any CPU-side CODEC */ if (cpu_dai) { codec = cpu_dai->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + if (codec && codec->component.probed && + codec->component.driver->remove_order == order) + soc_remove_component(&codec->component); } } @@ -1146,137 +1121,108 @@ static void soc_set_name_prefix(struct snd_soc_card *card, } } -static int soc_probe_codec(struct snd_soc_card *card, - struct snd_soc_codec *codec) +static int soc_probe_component(struct snd_soc_card *card, + struct snd_soc_component *component) { - int ret = 0; - const struct snd_soc_codec_driver *driver = codec->driver; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct snd_soc_component *dai_component, *component2; struct snd_soc_dai *dai; + int ret; - codec->component.card = card; - codec->dapm.card = card; - soc_set_name_prefix(card, &codec->component); + component->card = card; + dapm->card = card; + soc_set_name_prefix(card, component); - if (!try_module_get(codec->dev->driver->owner)) + if (!try_module_get(component->dev->driver->owner)) return -ENODEV; - soc_init_component_debugfs(&codec->component); + soc_init_component_debugfs(component); - if (driver->dapm_widgets) { - ret = snd_soc_dapm_new_controls(&codec->dapm, - driver->dapm_widgets, - driver->num_dapm_widgets); + if (component->dapm_widgets) { + ret = snd_soc_dapm_new_controls(dapm, component->dapm_widgets, + component->num_dapm_widgets); if (ret != 0) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to create new controls %d\n", ret); goto err_probe; } } - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) { - ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + /* + * This is rather ugly, but certain platforms expect that the DAPM + * widgets for the DAIs for components with the same parent device are + * created in the platforms DAPM context. Until that is fixed we need to + * keep this. + */ + if (component->steal_sibling_dai_widgets) { + dai_component = NULL; + list_for_each_entry(component2, &component_list, list) { + if (component == component2) + continue; - if (ret != 0) { - dev_err(codec->dev, - "Failed to create DAI widgets %d\n", ret); - goto err_probe; + if (component2->dev == component->dev && + !list_empty(&component2->dai_list)) { + dai_component = component2; + break; + } } - } - - codec->dapm.idle_bias_off = driver->idle_bias_off; - - if (driver->probe) { - ret = driver->probe(codec); - if (ret < 0) { - dev_err(codec->dev, - "ASoC: failed to probe CODEC %d\n", ret); - goto err_probe; + } else { + dai_component = component; + list_for_each_entry(component2, &component_list, list) { + if (component2->dev == component->dev && + component2->steal_sibling_dai_widgets) { + dai_component = NULL; + break; + } } - WARN(codec->dapm.idle_bias_off && - codec->dapm.bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias with idle_bias_off==1\n", - codec->component.name); } - if (driver->controls) - snd_soc_add_codec_controls(codec, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, - driver->num_dapm_routes); - - /* mark codec as probed and add to card codec list */ - codec->probed = 1; - list_add(&codec->card_list, &card->codec_dev_list); - list_add(&codec->dapm.list, &card->dapm_list); - - return 0; - -err_probe: - soc_cleanup_component_debugfs(&codec->component); - module_put(codec->dev->driver->owner); - - return ret; -} - -static int soc_probe_platform(struct snd_soc_card *card, - struct snd_soc_platform *platform) -{ - int ret = 0; - const struct snd_soc_platform_driver *driver = platform->driver; - struct snd_soc_component *component; - struct snd_soc_dai *dai; - - platform->component.card = card; - platform->component.dapm.card = card; - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - - soc_init_component_debugfs(&platform->component); - - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&platform->component.dapm, - driver->dapm_widgets, driver->num_dapm_widgets); - - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(component, &component_list, list) { - if (component->dev != platform->dev) - continue; - list_for_each_entry(dai, &component->dai_list, list) - snd_soc_dapm_new_dai_widgets(&platform->component.dapm, - dai); + if (dai_component) { + list_for_each_entry(dai, &dai_component->dai_list, list) { + snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret != 0) { + dev_err(component->dev, + "Failed to create DAI widgets %d\n", + ret); + goto err_probe; + } + } } - platform->component.dapm.idle_bias_off = 1; - - if (driver->probe) { - ret = driver->probe(platform); + if (component->probe) { + ret = component->probe(component); if (ret < 0) { - dev_err(platform->dev, - "ASoC: failed to probe platform %d\n", ret); + dev_err(component->dev, + "ASoC: failed to probe component %d\n", ret); goto err_probe; } + + WARN(dapm->idle_bias_off && + dapm->bias_level != SND_SOC_BIAS_OFF, + "codec %s can not start from non-off bias with idle_bias_off==1\n", + component->name); } - if (driver->controls) - snd_soc_add_platform_controls(platform, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&platform->component.dapm, - driver->dapm_routes, driver->num_dapm_routes); + if (component->controls) + snd_soc_add_component_controls(component, component->controls, + component->num_controls); + if (component->dapm_routes) + snd_soc_dapm_add_routes(dapm, component->dapm_routes, + component->num_dapm_routes); - /* mark platform as probed and add to card platform list */ - platform->probed = 1; - list_add(&platform->component.dapm.list, &card->dapm_list); + component->probed = 1; + list_add(&dapm->list, &card->dapm_list); + + /* This is a HACK and will be removed soon */ + if (component->codec) + list_add(&component->codec->card_list, &card->codec_dev_list); return 0; err_probe: - soc_cleanup_component_debugfs(&platform->component); - module_put(platform->dev->driver->owner); + soc_cleanup_component_debugfs(component); + module_put(component->dev->driver->owner); return ret; } @@ -1334,33 +1280,36 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_component *component; int i, ret; /* probe the CPU-side component, if it is a CODEC */ - if (cpu_dai->codec && - !cpu_dai->codec->probed && - cpu_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, cpu_dai->codec); - if (ret < 0) - return ret; + if (rtd->cpu_dai->codec) { + component = &rtd->cpu_dai->codec->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); + if (ret < 0) + return ret; + } } /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->codec->probed && - rtd->codec_dais[i]->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, rtd->codec_dais[i]->codec); + component = &rtd->codec_dais[i]->codec->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); if (ret < 0) return ret; } } /* probe the platform */ - if (!platform->probed && - platform->driver->probe_order == order) { - ret = soc_probe_platform(card, platform); + if (!platform->component.probed && + platform->component.driver->probe_order == order) { + ret = soc_probe_component(card, &platform->component); if (ret < 0) return ret; } @@ -1647,12 +1596,12 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->codec->probed) { + if (rtd->codec->component.probed) { dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); return -EBUSY; } - ret = soc_probe_codec(card, rtd->codec); + ret = soc_probe_component(card, &rtd->codec->component); if (ret < 0) return ret; @@ -1681,8 +1630,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (codec && codec->probed) - soc_remove_codec(codec); + if (codec && codec->component.probed) + soc_remove_component(&codec->component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -4198,6 +4147,20 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); +static int snd_soc_platform_drv_probe(struct snd_soc_component *component) +{ + struct snd_soc_platform *platform = snd_soc_component_to_platform(component); + + return platform->driver->probe(platform); +} + +static void snd_soc_platform_drv_remove(struct snd_soc_component *component) +{ + struct snd_soc_platform *platform = snd_soc_component_to_platform(component); + + platform->driver->remove(platform); +} + static int snd_soc_platform_drv_write(struct snd_soc_component *component, unsigned int reg, unsigned int val) { @@ -4234,6 +4197,24 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; + if (platform_drv->controls) { + platform->component.controls = platform_drv->controls; + platform->component.num_controls = platform_drv->num_controls; + } + if (platform_drv->dapm_widgets) { + platform->component.dapm_widgets = platform_drv->dapm_widgets; + platform->component.num_dapm_widgets = platform_drv->num_dapm_widgets; + platform->component.steal_sibling_dai_widgets = true; + } + if (platform_drv->dapm_routes) { + platform->component.dapm_routes = platform_drv->dapm_routes; + platform->component.num_dapm_routes = platform_drv->num_dapm_routes; + } + + if (platform_drv->probe) + platform->component.probe = snd_soc_platform_drv_probe; + if (platform_drv->remove) + platform->component.remove = snd_soc_platform_drv_remove; if (platform_drv->write) platform->component.write = snd_soc_platform_drv_write; if (platform_drv->read) @@ -4363,6 +4344,20 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) stream->formats |= codec_format_map[i]; } +static int snd_soc_codec_drv_probe(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return codec->driver->probe(codec); +} + +static void snd_soc_codec_drv_remove(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + codec->driver->remove(codec); +} + static int snd_soc_codec_drv_write(struct snd_soc_component *component, unsigned int reg, unsigned int val) { @@ -4411,12 +4406,30 @@ int snd_soc_register_codec(struct device *dev, return -ENOMEM; codec->component.dapm_ptr = &codec->dapm; + codec->component.codec = codec; ret = snd_soc_component_initialize(&codec->component, &codec_drv->component_driver, dev); if (ret) goto err_free; + if (codec_drv->controls) { + codec->component.controls = codec_drv->controls; + codec->component.num_controls = codec_drv->num_controls; + } + if (codec_drv->dapm_widgets) { + codec->component.dapm_widgets = codec_drv->dapm_widgets; + codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets; + } + if (codec_drv->dapm_routes) { + codec->component.dapm_routes = codec_drv->dapm_routes; + codec->component.num_dapm_routes = codec_drv->num_dapm_routes; + } + + if (codec_drv->probe) + codec->component.probe = snd_soc_codec_drv_probe; + if (codec_drv->remove) + codec->component.remove = snd_soc_codec_drv_remove; if (codec_drv->write) codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6307f85..b329b84 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -336,10 +336,12 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = { }; static const struct snd_soc_platform_driver dmaengine_pcm_platform = { + .component_driver = { + .probe_order = SND_SOC_COMP_ORDER_LATE, + }, .ops = &dmaengine_pcm_ops, .pcm_new = dmaengine_pcm_new, .pcm_free = dmaengine_pcm_free, - .probe_order = SND_SOC_COMP_ORDER_LATE, }; static const char * const dmaengine_pcm_dma_channel_names[] = { -- cgit v1.1 From 93c3ce76ccced3a8718149e8734ccaa931e9a1f1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:20 +0200 Subject: ASoC: Make rtd->codec optional There are some place in the ASoC core that expect rtd->codec to be non NULL (mainly CODEC specific sysfs files). With componentization going forward rtd->codec might be NULL in some cases. This patch prepares the core for this by not registering CODEC specific sysfs files if rtd->codec is NULL. sysfs file removal does not need to be conditionalized as it handles the removal of non-existing files just fine. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b833cc6..1c705c2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1261,17 +1261,21 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, } rtd->dev_registered = 1; - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); + if (rtd->codec) { + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(rtd->dev); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec dapm sysfs entries: %d\n", + ret); - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", ret); + /* add codec sysfs entries */ + ret = device_create_file(rtd->dev, &dev_attr_codec_reg); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec sysfs files: %d\n", + ret); + } return 0; } -- cgit v1.1 From 61aca5646b736a794d40de29a197144db3f0c5ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:21 +0200 Subject: ASoC: Add component level probe/remove support Now that we have a unified probe and remove path make sure to call them for all components. soc_{probe,remove}_component are responsible for setting up the DAPM context for the component, initialize the component prefix, manage the debugfs entries as well as do the registration of table based controls and DAPM elements. They also call the component drivers probe and remove callbacks. This patch makes these things available for generic snd_soc_component drivers rather than only having them for snd_soc_codec and snd_soc_platform drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 42 ++++++++++++++++++++++++------------------ 1 file changed, 24 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c705c2..08fd85e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1058,7 +1058,7 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_codec *codec; + struct snd_soc_component *component; int i; /* remove the platform */ @@ -1068,18 +1068,17 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { - codec = rtd->codec_dais[i]->codec; - if (codec && codec->component.probed && - codec->component.driver->remove_order == order) - soc_remove_component(&codec->component); + component = rtd->codec_dais[i]->component; + if (component->probed && + component->driver->remove_order == order) + soc_remove_component(component); } /* remove any CPU-side CODEC */ if (cpu_dai) { - codec = cpu_dai->codec; - if (codec && codec->component.probed && - codec->component.driver->remove_order == order) - soc_remove_component(&codec->component); + if (cpu_dai->component->probed && + cpu_dai->component->driver->remove_order == order) + soc_remove_component(cpu_dai->component); } } @@ -1289,19 +1288,17 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, int i, ret; /* probe the CPU-side component, if it is a CODEC */ - if (rtd->cpu_dai->codec) { - component = &rtd->cpu_dai->codec->component; - if (!component->probed && - component->driver->probe_order == order) { - ret = soc_probe_component(card, component); - if (ret < 0) - return ret; - } + component = rtd->cpu_dai->component; + if (!component->probed && + component->driver->probe_order == order) { + ret = soc_probe_component(card, component); + if (ret < 0) + return ret; } /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { - component = &rtd->codec_dais[i]->codec->component; + component = rtd->codec_dais[i]->component; if (!component->probed && component->driver->probe_order == order) { ret = soc_probe_component(card, component); @@ -4042,6 +4039,8 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->dev = dev; component->driver = driver; + component->probe = component->driver->probe; + component->remove = component->driver->remove; if (!component->dapm_ptr) component->dapm_ptr = &component->dapm; @@ -4055,6 +4054,13 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, if (driver->stream_event) dapm->stream_event = snd_soc_component_stream_event; + component->controls = driver->controls; + component->num_controls = driver->num_controls; + component->dapm_widgets = driver->dapm_widgets; + component->num_dapm_widgets = driver->num_dapm_widgets; + component->dapm_routes = driver->dapm_routes; + component->num_dapm_routes = driver->num_dapm_routes; + INIT_LIST_HEAD(&component->dai_list); mutex_init(&component->io_mutex); -- cgit v1.1 From 65d9361f0cb50a20641802ee3075145d72e4409c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:22 +0200 Subject: ASoC: Move AUX dev support to the component level This patch makes it possible to register arbitrary components as a AUX dev for a card. This was previously only possible for CODEC components. With componentization having made it possible for components to have DAPM contexts and controls there is no reason why AUX devs should be artificially limited to snd_soc_codec devices. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 48 ++++++++++++++++++++++++++++++++++++------------ 1 file changed, 36 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 08fd85e..08c04f4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -860,6 +860,23 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; +static struct snd_soc_component *soc_find_component( + const struct device_node *of_node, const char *name) +{ + struct snd_soc_component *component; + + list_for_each_entry(component, &component_list, list) { + if (of_node) { + if (component->dev->of_node == of_node) + return component; + } else if (strcmp(component->name, name) == 0) { + return component; + } + } + + return NULL; +} + static struct snd_soc_codec *soc_find_codec( const struct device_node *codec_of_node, const char *codec_name) @@ -1577,17 +1594,24 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *codecname = aux_dev->codec_name; + const char *name = aux_dev->codec_name; - rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname); - if (!rtd->codec) { + rtd->component = soc_find_component(aux_dev->codec_of_node, name); + if (!rtd->component) { if (aux_dev->codec_of_node) - codecname = of_node_full_name(aux_dev->codec_of_node); + name = of_node_full_name(aux_dev->codec_of_node); - dev_err(card->dev, "ASoC: %s not registered\n", codecname); + dev_err(card->dev, "ASoC: %s not registered\n", name); return -EPROBE_DEFER; } + /* + * Some places still reference rtd->codec, so we have to keep that + * initialized if the component is a CODEC. Once all those references + * have been removed, this code can be removed as well. + */ + rtd->codec = rtd->component->codec; + return 0; } @@ -1597,18 +1621,18 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->codec->component.probed) { - dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); + if (rtd->component->probed) { + dev_err(rtd->dev, "ASoC: codec already probed\n"); return -EBUSY; } - ret = soc_probe_component(card, &rtd->codec->component); + ret = soc_probe_component(card, rtd->component); if (ret < 0) return ret; /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(&rtd->codec->dapm); + ret = aux_dev->init(snd_soc_component_get_dapm(rtd->component)); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", aux_dev->name, ret); @@ -1622,7 +1646,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) static void soc_remove_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_component *component = rtd->component; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1631,8 +1655,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (codec && codec->component.probed) - soc_remove_component(&codec->component); + if (component && component->probed) + soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) -- cgit v1.1 From 57bf772687700e206c760ba2e4097f78bde97887 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:23 +0200 Subject: ASoC: Pass component instead of DAPM context to AUX dev init callback Given that the component is the containing structure it makes more sense to pass the component rather than the DAPM context to the AUX dev init callback. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 6 ++++-- sound/soc/soc-core.c | 2 +- 2 files changed, 5 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 9902efc..a054826 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -228,10 +228,12 @@ static struct snd_soc_dai_link speyside_dai[] = { }, }; -static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) +static int speyside_wm9081_init(struct snd_soc_component *component) { + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, + return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, MCLK_AUDIO_RATE, 0); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 08c04f4..4393bc3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1632,7 +1632,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(snd_soc_component_get_dapm(rtd->component)); + ret = aux_dev->init(rtd->component); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", aux_dev->name, ret); -- cgit v1.1 From 70090bbb8b7d7da7a6f64969b43a61c493c560ff Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:24 +0200 Subject: ASoC: Move component->probed check into soc_{remove,probe}_component() Having the check in a centralized place makes the code a bit cleaner and shorter. Note: There is a slight semantic change in this patch. soc_probe_aux_dev() will no longer return -EBUSY if the AUX dev has already been probed before. This is fine though since it will simply do nothing in that case and return success. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4393bc3..2fbfbfc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1004,6 +1004,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) static void soc_remove_component(struct snd_soc_component *component) { + if (!component->probed) + return; + /* This is a HACK and will be removed soon */ if (component->codec) list_del(&component->codec->card_list); @@ -1079,22 +1082,19 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, int i; /* remove the platform */ - if (platform && platform->component.probed && - platform->component.driver->remove_order == order) + if (platform && platform->component.driver->remove_order == order) soc_remove_component(&platform->component); /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { component = rtd->codec_dais[i]->component; - if (component->probed && - component->driver->remove_order == order) + if (component->driver->remove_order == order) soc_remove_component(component); } /* remove any CPU-side CODEC */ if (cpu_dai) { - if (cpu_dai->component->probed && - cpu_dai->component->driver->remove_order == order) + if (cpu_dai->component->driver->remove_order == order) soc_remove_component(cpu_dai->component); } } @@ -1145,6 +1145,9 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_dai *dai; int ret; + if (component->probed) + return 0; + component->card = card; dapm->card = card; soc_set_name_prefix(card, component); @@ -1306,8 +1309,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, /* probe the CPU-side component, if it is a CODEC */ component = rtd->cpu_dai->component; - if (!component->probed && - component->driver->probe_order == order) { + if (component->driver->probe_order == order) { ret = soc_probe_component(card, component); if (ret < 0) return ret; @@ -1316,8 +1318,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { component = rtd->codec_dais[i]->component; - if (!component->probed && - component->driver->probe_order == order) { + if (component->driver->probe_order == order) { ret = soc_probe_component(card, component); if (ret < 0) return ret; @@ -1325,8 +1326,7 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, } /* probe the platform */ - if (!platform->component.probed && - platform->component.driver->probe_order == order) { + if (platform->component.driver->probe_order == order) { ret = soc_probe_component(card, &platform->component); if (ret < 0) return ret; @@ -1621,11 +1621,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->component->probed) { - dev_err(rtd->dev, "ASoC: codec already probed\n"); - return -EBUSY; - } - ret = soc_probe_component(card, rtd->component); if (ret < 0) return ret; -- cgit v1.1 From ffbd7dd72bd3ad9bcae9190788c858e57f1e8e4e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:25 +0200 Subject: ASoC: Cleanup DAI module reference counting Currently when a DAI has no CODEC associated to it the reference on the module containing the DAI driver is increased when the DAI is probed and decrease when the DAI is removed. For DAIs with CODECs the module reference count was already incremented when the CODEC is probed. Now that all components have their module reference count incremented when they are probed and all DAIs do have a component it is possible to remove the module reference counting on DAI probe and removal. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fbfbfc..4dc2876 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1067,8 +1067,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) cpu_dai->name, err); } cpu_dai->probed = 0; - if (!cpu_dai->codec) - module_put(cpu_dai->dev->driver->owner); } } @@ -1422,18 +1420,12 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - if (!cpu_dai->codec) { - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; - } - if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: failed to probe CPU DAI %s: %d\n", cpu_dai->name, ret); - module_put(cpu_dai->dev->driver->owner); return ret; } } -- cgit v1.1 From e60cd14f0bf6c004cd7032a24a036ba32d56e08a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:26 +0200 Subject: ASoC: Consolidate CPU and CODEC DAI removal CPU and CODEC DAI works exactly the same way. There is already a helper function for CODEC DAI removal, use that one as well for CPU DAI removal. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4dc2876..5f6f978 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1021,28 +1021,27 @@ static void soc_remove_component(struct snd_soc_component *component) module_put(component->dev->driver->owner); } -static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) +static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (codec_dai && codec_dai->probed && - codec_dai->driver->remove_order == order) { - if (codec_dai->driver->remove) { - err = codec_dai->driver->remove(codec_dai); + if (dai && dai->probed && + dai->driver->remove_order == order) { + if (dai->driver->remove) { + err = dai->driver->remove(dai); if (err < 0) - dev_err(codec_dai->dev, + dev_err(dai->dev, "ASoC: failed to remove %s: %d\n", - codec_dai->name, err); + dai->name, err); } - codec_dai->probed = 0; + dai->probed = 0; } } static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, err; + int i; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1054,20 +1053,9 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* remove the CODEC DAI */ for (i = 0; i < rtd->num_codecs; i++) - soc_remove_codec_dai(rtd->codec_dais[i], order); + soc_remove_dai(rtd->codec_dais[i], order); - /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed && - cpu_dai->driver->remove_order == order) { - if (cpu_dai->driver->remove) { - err = cpu_dai->driver->remove(cpu_dai); - if (err < 0) - dev_err(cpu_dai->dev, - "ASoC: failed to remove %s: %d\n", - cpu_dai->name, err); - } - cpu_dai->probed = 0; - } + soc_remove_dai(rtd->cpu_dai, order); } static void soc_remove_link_components(struct snd_soc_card *card, int num, -- cgit v1.1 From 14621c7e5e72200ec021a7580121130ce7f2ff22 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:27 +0200 Subject: ASoC: Consolidate CPU and CODEC DAI lookup The lookup of CPU and CODEC DAIs is fairly similar and can easily be consolidated into a single helper function. There are two main differences in the current implementation of the CPU and CODEC DAI lookup: 1) CPU DAIs can be looked up by the DAI name alone and do not necessarily require a component name/of_node. 2) The CODEC DAI search only considers DAIs from CODEC components. For 1) the new helper function will allow to lookup DAIs without providing a component name or of_node, but since snd_soc_register_card() already rejects CODEC DAI link components without neither a of_node or a name we'll never get into the situation where we try to lookup a CODEC DAI without a name/of_node. For 2) the new helper function just always considers all components. Componentization is now at a point where it is possible to register a CODEC as a snd_soc_component rather than a snd_soc_codec, by considering DAIs from all components it is possible to use such a CODEC in a DAI link. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 72 ++++++++++++++-------------------------------------- 1 file changed, 19 insertions(+), 53 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5f6f978..140f43f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -877,35 +877,23 @@ static struct snd_soc_component *soc_find_component( return NULL; } -static struct snd_soc_codec *soc_find_codec( - const struct device_node *codec_of_node, - const char *codec_name) +static struct snd_soc_dai *snd_soc_find_dai( + const struct snd_soc_dai_link_component *dlc) { - struct snd_soc_codec *codec; + struct snd_soc_component *component; + struct snd_soc_dai *dai; - list_for_each_entry(codec, &codec_list, list) { - if (codec_of_node) { - if (codec->dev->of_node != codec_of_node) - continue; - } else { - if (strcmp(codec->component.name, codec_name)) + /* Find CPU DAI from registered DAIs*/ + list_for_each_entry(component, &component_list, list) { + if (dlc->of_node && component->dev->of_node != dlc->of_node) + continue; + if (dlc->name && strcmp(dev_name(component->dev), dlc->name)) + continue; + list_for_each_entry(dai, &component->dai_list, list) { + if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)) continue; - } - - return codec; - } - - return NULL; -} - -static struct snd_soc_dai *soc_find_codec_dai(struct snd_soc_codec *codec, - const char *codec_dai_name) -{ - struct snd_soc_dai *codec_dai; - list_for_each_entry(codec_dai, &codec->component.dai_list, list) { - if (!strcmp(codec_dai->name, codec_dai_name)) { - return codec_dai; + return dai; } } @@ -916,33 +904,19 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_component *component; struct snd_soc_dai_link_component *codecs = dai_link->codecs; + struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_platform *platform; - struct snd_soc_dai *cpu_dai; const char *platform_name; int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); - /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(component, &component_list, list) { - if (dai_link->cpu_of_node && - component->dev->of_node != dai_link->cpu_of_node) - continue; - if (dai_link->cpu_name && - strcmp(dev_name(component->dev), dai_link->cpu_name)) - continue; - list_for_each_entry(cpu_dai, &component->dai_list, list) { - if (dai_link->cpu_dai_name && - strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; - - rtd->cpu_dai = cpu_dai; - } - } - + cpu_dai_component.name = dai_link->cpu_name; + cpu_dai_component.of_node = dai_link->cpu_of_node; + cpu_dai_component.dai_name = dai_link->cpu_dai_name; + rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); @@ -953,15 +927,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* Find CODEC from registered CODECs */ for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_codec *codec; - codec = soc_find_codec(codecs[i].of_node, codecs[i].name); - if (!codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - codecs[i].name); - return -EPROBE_DEFER; - } - - codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name); + codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); -- cgit v1.1 From 886f5692253de1a9509f5cb708432b2157afb57c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:28 +0200 Subject: ASoC: Automatically initialize regmap for all components So far regmap is only automatically initialized for CODECs. Now that we have the infrastructure in place to let components have DAPM widgets and controls that want to use the generic regmap based IO also make sure to automatically initialize regmap for all components. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 35 +++++++++++++++++------------------ sound/soc/soc-io.c | 28 ---------------------------- 2 files changed, 17 insertions(+), 46 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 140f43f..96f2866 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4032,8 +4032,23 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, return 0; } +static void snd_soc_component_init_regmap(struct snd_soc_component *component) +{ + if (!component->regmap) + component->regmap = dev_get_regmap(component->dev, NULL); + if (component->regmap) { + int val_bytes = regmap_get_val_bytes(component->regmap); + /* Errors are legitimate for non-integer byte multiples */ + if (val_bytes > 0) + component->val_bytes = val_bytes; + } +} + static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { + if (!component->write && !component->read) + snd_soc_component_init_regmap(component); + list_add(&component->list, &component_list); } @@ -4371,7 +4386,6 @@ int snd_soc_register_codec(struct device *dev, { struct snd_soc_codec *codec; struct snd_soc_dai *dai; - struct regmap *regmap; int ret, i; dev_dbg(dev, "codec register %s\n", dev_name(dev)); @@ -4425,23 +4439,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.debugfs_prefix = "codec"; #endif - if (!codec->component.write) { - if (codec_drv->get_regmap) - regmap = codec_drv->get_regmap(dev); - else - regmap = dev_get_regmap(dev, NULL); - - if (regmap) { - ret = snd_soc_component_init_io(&codec->component, - regmap); - if (ret) { - dev_err(codec->dev, - "Failed to set cache I/O:%d\n", - ret); - goto err_cleanup; - } - } - } + if (codec_drv->get_regmap) + codec->component.regmap = codec_drv->get_regmap(dev); for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 7767fbd..9b39390 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -271,31 +271,3 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, return snd_soc_component_write(&platform->component, reg, val); } EXPORT_SYMBOL_GPL(snd_soc_platform_write); - -/** - * snd_soc_component_init_io() - Initialize regmap IO - * - * @component: component to initialize - * @regmap: regmap instance to use for IO operations - * - * Return: 0 on success, a negative error code otherwise - */ -int snd_soc_component_init_io(struct snd_soc_component *component, - struct regmap *regmap) -{ - int ret; - - if (!regmap) - return -EINVAL; - - ret = regmap_get_val_bytes(regmap); - /* Errors are legitimate for non-integer byte - * multiples */ - if (ret > 0) - component->val_bytes = ret; - - component->regmap = regmap; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_component_init_io); -- cgit v1.1 From 75af7c081982d76cef0daf26e96b5d1e8cb9d631 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:29 +0200 Subject: ASoC: Remove support for legacy snd_soc_platform IO There were never any actual users of this in upstream and by we have with regmap a replacement in place, which should be used by new drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 22 ---------------------- 1 file changed, 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96f2866..2d7a9ec 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4151,24 +4151,6 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } -static int snd_soc_platform_drv_write(struct snd_soc_component *component, - unsigned int reg, unsigned int val) -{ - struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - - return platform->driver->write(platform, reg, val); -} - -static int snd_soc_platform_drv_read(struct snd_soc_component *component, - unsigned int reg, unsigned int *val) -{ - struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - - *val = platform->driver->read(platform, reg); - - return 0; -} - /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -4205,10 +4187,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; - if (platform_drv->write) - platform->component.write = snd_soc_platform_drv_write; - if (platform_drv->read) - platform->component.read = snd_soc_platform_drv_read; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; -- cgit v1.1 From c5599b87a8317738a541d8893cb327df5d04b007 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 19 Aug 2014 15:51:30 +0200 Subject: ASoC: Replace list_empty(&card->codec_dev_list) with !card->instantiated With componentization we no longer necessarily need a snd_soc_codec struct for a card. Instead of checking if the card's CODEC list is empty just use card->instantiated to check if the card has been instantiated yet. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2d7a9ec..c36983a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -552,10 +552,8 @@ int snd_soc_suspend(struct device *dev) struct snd_soc_codec *codec; int i, j; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* Due to the resume being scheduled into a workqueue we could @@ -808,10 +806,8 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* activate pins from sleep state */ -- cgit v1.1 From 5819c2fa55d4a6eaf7fe025a393dce98fc4b2116 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 24 Aug 2014 15:36:55 +0200 Subject: ASoC: Restore idle_bias_off initialization This was accidentally lost in commit f1d45cc3ae96 ("ASoC: Consolidate platform and CODEC probe/remove"). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c36983a..4196826 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4010,6 +4010,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; + dapm->idle_bias_off = true; if (driver->seq_notifier) dapm->seq_notifier = snd_soc_component_seq_notifier; if (driver->stream_event) @@ -4399,6 +4400,7 @@ int snd_soc_register_codec(struct device *dev, codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.codec = codec; + codec->dapm.idle_bias_off = codec_drv->idle_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) -- cgit v1.1 From 06cb1eb3de5c905da60ab91dbf99aaf96a43d043 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 21 Aug 2014 18:20:49 +0530 Subject: ASoC: mfld-compress: Use dedicated function instead of ioctl Also pass sst device as an argument to function pointer prototypes of compr_ops. This will be used to derive sst driver context. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-compress.c | 38 +++++++++++++++++++++------- sound/soc/intel/sst-mfld-platform.h | 27 ++++++++++++-------- 2 files changed, 46 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 29c059c..59467775 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -86,7 +86,7 @@ static int sst_platform_compr_free(struct snd_compr_stream *cstream) /*need to check*/ str_id = stream->id; if (str_id) - ret_val = stream->compr_ops->close(str_id); + ret_val = stream->compr_ops->close(sst->dev, str_id); module_put(sst->dev->driver->owner); kfree(stream); pr_debug("%s: %d\n", __func__, ret_val); @@ -158,7 +158,7 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, cb.drain_cb_param = cstream; cb.drain_notify = sst_drain_notify; - retval = stream->compr_ops->open(&str_params, &cb); + retval = stream->compr_ops->open(sst->dev, &str_params, &cb); if (retval < 0) { pr_err("stream allocation failed %d\n", retval); return retval; @@ -170,10 +170,30 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, 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); + struct sst_runtime_stream *stream = cstream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (stream->compr_ops->stream_start) + return stream->compr_ops->stream_start(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_STOP: + if (stream->compr_ops->stream_drop) + return stream->compr_ops->stream_drop(sst->dev, stream->id); + case SND_COMPR_TRIGGER_DRAIN: + if (stream->compr_ops->stream_drain) + return stream->compr_ops->stream_drain(sst->dev, stream->id); + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + if (stream->compr_ops->stream_partial_drain) + return stream->compr_ops->stream_partial_drain(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (stream->compr_ops->stream_pause) + return stream->compr_ops->stream_pause(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (stream->compr_ops->stream_pause_release) + return stream->compr_ops->stream_pause_release(sst->dev, stream->id); + default: + return -EINVAL; + } } static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, @@ -182,7 +202,7 @@ static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream; stream = cstream->runtime->private_data; - stream->compr_ops->tstamp(stream->id, tstamp); + stream->compr_ops->tstamp(sst->dev, 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); @@ -195,7 +215,7 @@ static int sst_platform_compr_ack(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream; stream = cstream->runtime->private_data; - stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->compr_ops->ack(sst->dev, stream->id, (unsigned long)bytes); stream->bytes_written += bytes; return 0; @@ -225,7 +245,7 @@ static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream = cstream->runtime->private_data; - return stream->compr_ops->set_metadata(stream->id, metadata); + return stream->compr_ops->set_metadata(sst->dev, stream->id, metadata); } struct snd_compr_ops sst_platform_compr_ops = { diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index faaba10..0c5b943d 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -99,17 +99,24 @@ struct sst_compress_cb { 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, + int (*open)(struct device *dev, + struct snd_sst_params *str_params, struct sst_compress_cb *cb); + int (*stream_start)(struct device *dev, unsigned int str_id); + int (*stream_drop)(struct device *dev, unsigned int str_id); + int (*stream_drain)(struct device *dev, unsigned int str_id); + int (*stream_partial_drain)(struct device *dev, unsigned int str_id); + int (*stream_pause)(struct device *dev, unsigned int str_id); + int (*stream_pause_release)(struct device *dev, unsigned int str_id); + + int (*tstamp)(struct device *dev, unsigned int str_id, + struct snd_compr_tstamp *tstamp); + int (*ack)(struct device *dev, unsigned int str_id, + unsigned long bytes); + int (*close)(struct device *dev, 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)(struct device *dev, unsigned int str_id, struct snd_compr_metadata *mdata); - }; struct sst_ops { -- cgit v1.1 From b792346fa8660a22a06f118cebe47709f507914f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 28 Aug 2014 14:07:11 +0300 Subject: ASoC: Remove unused cache_only from struct snd_soc_codec There are no real users for cache_only in "struct snd_soc_codec" so remove it and needless debugfs node. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4196826..1b422c5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -311,8 +311,6 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->component.debugfs_root, - &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->component.debugfs_root, -- cgit v1.1 From bd033808e2b160bab61cfe18b0ecb4ccc7809516 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 20 Aug 2014 13:08:47 +0200 Subject: ASoC: sst-haswell-pcm: Alloc state struct in driver probe() Resource allocations should happen in driver probe callback rather than in snd_soc_platform probe functions. Especially if the resource is device managed. The snd_soc_* probe/remove functions are mainly intended to be used for things that require the component to be already bound to a card. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 24 +++++++++++++----------- 1 file changed, 13 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 61bf6da..1de0958 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -778,20 +778,11 @@ static const struct snd_soc_dapm_route graph[] = { static int hsw_pcm_probe(struct snd_soc_platform *platform) { + struct hsw_priv_data *priv_data = snd_soc_platform_get_drvdata(platform); struct sst_pdata *pdata = dev_get_platdata(platform->dev); - struct hsw_priv_data *priv_data; - struct device *dma_dev; + struct device *dma_dev = pdata->dma_dev; int i, ret = 0; - if (!pdata) - return -ENODEV; - - dma_dev = pdata->dma_dev; - - 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++) { @@ -863,12 +854,23 @@ static const struct snd_soc_component_driver hsw_dai_component = { static int hsw_pcm_dev_probe(struct platform_device *pdev) { struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + struct hsw_priv_data *priv_data; int ret; + if (!sst_pdata) + return -EINVAL; + + priv_data = devm_kzalloc(&pdev->dev, sizeof(*priv_data), GFP_KERNEL); + if (!priv_data) + return -ENOMEM; + ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata); if (ret < 0) return -ENODEV; + priv_data->hsw = sst_pdata->dsp; + platform_set_drvdata(pdev, priv_data); + ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform); if (ret < 0) goto err_plat; -- cgit v1.1 From 923976a30b36ce0970e88f53ed2f2b5b61aeeb73 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 20 Aug 2014 13:08:48 +0200 Subject: ASoC: sst-haswell-pcm: Move controls and DAPM elements to component The sst-haswell-pcm driver registers both a snd_soc_component and a snd_soc_platform and expects that the DAPM widgets for the DAIs registered by component are added to the DAPM context of the platform. This requires us to have a hack in the ASoC core which does so. Moving the DAPM elements over to the component allows us to remove this hack. While we are at it also move the controls over to the component. The controls don't need the platform for anything other than snd_soc_platform_get_drvdata(), this can easily be replaced by snd_soc_component_get_drvdata(). As the long term goal is to register only a single component this is a step in the right direction. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 32 +++++++++++++++----------------- 1 file changed, 15 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 1de0958..33fc5c3 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -138,11 +138,10 @@ static inline unsigned int hsw_ipc_to_mixer(u32 value) static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); 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; @@ -176,11 +175,10 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); 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; @@ -208,8 +206,8 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, static int hsw_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); - struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -233,8 +231,8 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol, static int hsw_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); - struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct sst_hsw *hsw = pdata->hsw; unsigned int volume = 0; @@ -839,16 +837,16 @@ static struct snd_soc_platform_driver hsw_soc_platform = { .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", + .name = "haswell-dai", + .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 int hsw_pcm_dev_probe(struct platform_device *pdev) -- cgit v1.1 From 0634814fe0f29a46c44386a03f259f99c983bf7e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 20 Aug 2014 13:08:49 +0200 Subject: ASoC: Remove table based DAPM/control setup support from snd_soc_platform_driver There are no users left and new users should rather use the component_driver struct embedded in the snd_soc_platform_driver struct to do this. E.g.: static const struct snd_soc_platform_driver foobar_driver = { .component_driver = { .dapm_widgets = ..., .num_dapm_widgets = ..., ..., }, ... }; instead of static const struct snd_soc_platform_driver foobar_driver = { .dapm_widgets = ..., .num_dapm_widgets = ..., ... }; This also allows us to remove the steal_sibling_dai_widgets hack. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 58 ++++++---------------------------------------------- 1 file changed, 6 insertions(+), 52 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b422c5..052f59c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1087,7 +1087,6 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - struct snd_soc_component *dai_component, *component2; struct snd_soc_dai *dai; int ret; @@ -1114,44 +1113,12 @@ static int soc_probe_component(struct snd_soc_card *card, } } - /* - * This is rather ugly, but certain platforms expect that the DAPM - * widgets for the DAIs for components with the same parent device are - * created in the platforms DAPM context. Until that is fixed we need to - * keep this. - */ - if (component->steal_sibling_dai_widgets) { - dai_component = NULL; - list_for_each_entry(component2, &component_list, list) { - if (component == component2) - continue; - - if (component2->dev == component->dev && - !list_empty(&component2->dai_list)) { - dai_component = component2; - break; - } - } - } else { - dai_component = component; - list_for_each_entry(component2, &component_list, list) { - if (component2->dev == component->dev && - component2->steal_sibling_dai_widgets) { - dai_component = NULL; - break; - } - } - } - - if (dai_component) { - list_for_each_entry(dai, &dai_component->dai_list, list) { - snd_soc_dapm_new_dai_widgets(dapm, dai); - if (ret != 0) { - dev_err(component->dev, - "Failed to create DAI widgets %d\n", - ret); - goto err_probe; - } + list_for_each_entry(dai, &component->dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret != 0) { + dev_err(component->dev, + "Failed to create DAI widgets %d\n", ret); + goto err_probe; } } @@ -4164,19 +4131,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; - if (platform_drv->controls) { - platform->component.controls = platform_drv->controls; - platform->component.num_controls = platform_drv->num_controls; - } - if (platform_drv->dapm_widgets) { - platform->component.dapm_widgets = platform_drv->dapm_widgets; - platform->component.num_dapm_widgets = platform_drv->num_dapm_widgets; - platform->component.steal_sibling_dai_widgets = true; - } - if (platform_drv->dapm_routes) { - platform->component.dapm_routes = platform_drv->dapm_routes; - platform->component.num_dapm_routes = platform_drv->num_dapm_routes; - } if (platform_drv->probe) platform->component.probe = snd_soc_platform_drv_probe; -- cgit v1.1 From 9cca023e5c5c13486d48d47a46564c359af9ae73 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Sep 2014 11:40:40 +0200 Subject: ASoC: wm8{350,753,971}: Use snd_soc_dapm_to_codec() instead of dapm->codec The CODEC struct in the snd_soc_dapm_context struct is deprecated and scheduled for removal. Use the snd_soc_dapm_to_codec() function instead. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3dfdcc4..628ec77 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -212,7 +212,7 @@ static void wm8350_pga_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1, *out2 = &wm8350_data->out2; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e54e097..21ca3a9 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1433,7 +1433,7 @@ static void wm8753_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8753_set_bias_level(codec, dapm->bias_level); } diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 0499cd4..39ddb9b 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -615,7 +615,7 @@ static void wm8971_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8971_set_bias_level(codec, codec->dapm.bias_level); } -- cgit v1.1 From a761f87f367a2a172cbc62d0e88eabe175d349a8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Sep 2014 11:40:41 +0200 Subject: ASoC: rx51: Use snd_soc_dapm_to_codec() instead of dapm->codec The CODEC struct in the snd_soc_dapm_context struct is deprecated and scheduled for removal. Use the snd_soc_dapm_to_codec() function instead. Signed-off-by: Lars-Peter Clausen Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/rx51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 943922c..b10ae80 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -168,7 +168,7 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, static int rx51_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - struct snd_soc_codec *codec = w->dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); if (SND_SOC_DAPM_EVENT_ON(event)) tpa6130a2_stereo_enable(codec, 1); -- cgit v1.1 From 0bd2ac3dae74ee25c5ea171cb572731c7a89c248 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Sep 2014 11:40:42 +0200 Subject: ASoC: Remove CODEC pointer from snd_soc_dapm_context The only remaining user of the CODEC pointer in the DAPM struct is to initialize the CODEC pointer in the widget struct. The later is scheduled for removal, but has still a few users left. For now use dapm->component->codec to initialize it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 - sound/soc/soc-dapm.c | 2 +- 2 files changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 052f59c..8d45eec 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4351,7 +4351,6 @@ int snd_soc_register_codec(struct device *dev, if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - codec->dapm.codec = codec; codec->dapm.idle_bias_off = codec_drv->idle_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352..1f1e965 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3107,7 +3107,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->codec; + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); -- cgit v1.1 From b2d9de549c30170eed5691d369cf16680e0ce03a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 3 Oct 2014 15:32:40 +0300 Subject: ASoC: dapm: Fix NULL pointer dereference when registering card with widgets Commit 0bd2ac3dae74 ("ASoC: Remove CODEC pointer from snd_soc_dapm_context") introduced regression to snd_soc_dapm_new_controls() when registering a card with card->dapm_widgets set. Call chain is: snd_soc_register_card() -> snd_soc_instantiate_card() -> snd_soc_dapm_new_controls() -> snd_soc_dapm_new_control() Null pointer dereference occurs since card->dapm context doesn't have associated component. Fix this by setting widget codec pointer conditionally. Signed-off-by: Jarkko Nikula Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1f1e965..231deb2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3107,7 +3107,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->component->codec; + if (dapm->component) + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); -- cgit v1.1