Commit 01d7584c authored by Liam Girdwood's avatar Liam Girdwood Committed by Mark Brown
Browse files

ASoC: dpcm: Add Dynamic PCM core operations. 



The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.

Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.

e.g. pcm:0,0 routing digital data to 2 external codecs.

FE pcm:0,0  ----> BE (McBSP.0) ----> CODEC 0
             +--> BE (McPDM.0) ----> CODEC 1

e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.

FE pcm:0,0 ---
             +--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---

The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.

DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.

Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.

This patch adds the core DPCM code and contains :-

 o The FE and BE PCM operations.
 o FE and BE DAI link support.
 o FE and BE PCM creation.
 o BE support API.
 o BE and FE link management.

Signed-off-by: default avatarLiam Girdwood <lrg@ti.com>
Signed-off-by: default avatarMark Brown <broonie@opensource.wolfsonmicro.com>
parent f20c2cb9

include/sound/soc-dpcm.h

0 → 100644
+132 −0
Original line number Diff line number Diff line 
/*

 * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support

 *

 * Author:		Liam Girdwood <lrg@ti.com>

 *

 * This program is free software; you can redistribute it and/or modify

 * it under the terms of the GNU General Public License version 2 as

 * published by the Free Software Foundation.

 */



#ifndef __LINUX_SND_SOC_DPCM_H

#define __LINUX_SND_SOC_DPCM_H



#include <linux/list.h>

#include <sound/pcm.h>



struct snd_soc_pcm_runtime;



/*

 * Types of runtime_update to perform. e.g. originated from FE PCM ops

 * or audio route changes triggered by muxes/mixers.

 */

enum snd_soc_dpcm_update {

	SND_SOC_DPCM_UPDATE_NO	= 0,

	SND_SOC_DPCM_UPDATE_BE,

	SND_SOC_DPCM_UPDATE_FE,

};



/*

 * Dynamic PCM Frontend -> Backend link management states.

 */

enum snd_soc_dpcm_link_state {

	SND_SOC_DPCM_LINK_STATE_NEW	= 0,	/* newly created link */

	SND_SOC_DPCM_LINK_STATE_FREE,		/* link to be dismantled */

};



/*

 * Dynamic PCM Frontend -> Backend link PCM states.

 */

enum snd_soc_dpcm_state {

	SND_SOC_DPCM_STATE_NEW	= 0,

	SND_SOC_DPCM_STATE_OPEN,

	SND_SOC_DPCM_STATE_HW_PARAMS,

	SND_SOC_DPCM_STATE_PREPARE,

	SND_SOC_DPCM_STATE_START,

	SND_SOC_DPCM_STATE_STOP,

	SND_SOC_DPCM_STATE_PAUSED,

	SND_SOC_DPCM_STATE_SUSPEND,

	SND_SOC_DPCM_STATE_HW_FREE,

	SND_SOC_DPCM_STATE_CLOSE,

};



/*

 * Dynamic PCM trigger ordering. Triggering flexibility is required as some

 * DSPs require triggering before/after their CPU platform and DAIs.

 *

 * i.e. some clients may want to manually order this call in their PCM

 * trigger() whilst others will just use the regular core ordering.

 */

enum snd_soc_dpcm_trigger {

	SND_SOC_DPCM_TRIGGER_PRE		= 0,

	SND_SOC_DPCM_TRIGGER_POST,

};



/*

 * Dynamic PCM link

 * This links together a FE and BE DAI at runtime and stores the link

 * state information and the hw_params configuration.

 */

struct snd_soc_dpcm {

	/* FE and BE DAIs*/

	struct snd_soc_pcm_runtime *be;

	struct snd_soc_pcm_runtime *fe;



	/* link state */

	enum snd_soc_dpcm_link_state state;



	/* list of BE and FE for this DPCM link */

	struct list_head list_be;

	struct list_head list_fe;



	/* hw params for this link - may be different for each link */

	struct snd_pcm_hw_params hw_params;

};



/*

 * Dynamic PCM runtime data.

 */

struct snd_soc_dpcm_runtime {

	struct list_head be_clients;

	struct list_head fe_clients;



	int users;

	struct snd_pcm_runtime *runtime;

	struct snd_pcm_hw_params hw_params;



	/* state and update */

	enum snd_soc_dpcm_update runtime_update;

	enum snd_soc_dpcm_state state;

};



/* can this BE stop and free */

int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,

		struct snd_soc_pcm_runtime *be, int stream);



/* can this BE perform a hw_params() */

int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,

		struct snd_soc_pcm_runtime *be, int stream);



/* is the current PCM operation for this FE ? */

int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream);



/* is the current PCM operation for this BE ? */

int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,

		struct snd_soc_pcm_runtime *be, int stream);



/* get the substream for this BE */

struct snd_pcm_substream *

	snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream);



/* get the BE runtime state */

enum snd_soc_dpcm_state

	snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream);



/* set the BE runtime state */

void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,

	enum snd_soc_dpcm_state state);



/* internal use only */

int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);



#endif

include/sound/soc.h

+17 −0
Original line number Diff line number Diff line
@@ -287,6 +287,7 @@ struct snd_soc_jack_zone; 
struct snd_soc_jack_pin;

struct snd_soc_cache_ops;

#include <sound/soc-dapm.h>

#include <sound/soc-dpcm.h>



#ifdef CONFIG_GPIOLIB

struct snd_soc_jack_gpio;
@@ -782,23 +783,36 @@ struct snd_soc_dai_link { 
	const char *cpu_dai_name;

	const struct device_node *cpu_dai_of_node;

	const char *codec_dai_name;

	int be_id;	/* optional ID for machine driver BE identification */



	const struct snd_soc_pcm_stream *params;



	unsigned int dai_fmt;           /* format to set on init */



	enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */



	/* Keep DAI active over suspend */

	unsigned int ignore_suspend:1;



	/* Symmetry requirements */

	unsigned int symmetric_rates:1;



	/* Do not create a PCM for this DAI link (Backend link) */

	unsigned int no_pcm:1;



	/* This DAI link can route to other DAI links at runtime (Frontend)*/

	unsigned int dynamic:1;



	/* pmdown_time is ignored at stop */

	unsigned int ignore_pmdown_time:1;



	/* codec/machine specific init - e.g. add machine controls */

	int (*init)(struct snd_soc_pcm_runtime *rtd);



	/* optional hw_params re-writing for BE and FE sync */

	int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,

			struct snd_pcm_hw_params *params);



	/* machine stream operations */

	struct snd_soc_ops *ops;

};
@@ -930,6 +944,9 @@ struct snd_soc_pcm_runtime { 


	unsigned int dev_registered:1;



	/* Dynamic PCM BE runtime data */

	struct snd_soc_dpcm_runtime dpcm[2];



	long pmdown_time;



	/* runtime devices */

sound/soc/soc-core.c

+5 −0
Original line number Diff line number Diff line
@@ -39,6 +39,7 @@ 
#include <sound/pcm.h>

#include <sound/pcm_params.h>

#include <sound/soc.h>

#include <sound/soc-dpcm.h>

#include <sound/initval.h>



#define CREATE_TRACE_POINTS
@@ -1161,6 +1162,10 @@ static int soc_post_component_init(struct snd_soc_card *card, 
	rtd->dev->init_name = name;

	dev_set_drvdata(rtd->dev, rtd);

	mutex_init(&rtd->pcm_mutex);

	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);

	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);

	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);

	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);

	ret = device_add(rtd->dev);

	if (ret < 0) {

		dev_err(card->dev,

sound/soc/soc-pcm.c

+1216 −30

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