linux_dsm_epyc7002/sound/soc/soc-core.c

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// SPDX-License-Identifier: GPL-2.0+
//
// soc-core.c -- ALSA SoC Audio Layer
//
// Copyright 2005 Wolfson Microelectronics PLC.
// Copyright 2005 Openedhand Ltd.
// Copyright (C) 2010 Slimlogic Ltd.
// Copyright (C) 2010 Texas Instruments Inc.
//
// Author: Liam Girdwood <lrg@slimlogic.co.uk>
// with code, comments and ideas from :-
// Richard Purdie <richard@openedhand.com>
//
// TODO:
// o Add hw rules to enforce rates, etc.
// o More testing with other codecs/machines.
// o Add more codecs and platforms to ensure good API coverage.
// o Support TDM on PCM and I2S
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/bitops.h>
#include <linux/debugfs.h>
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
#include <linux/platform_device.h>
#include <linux/pinctrl/consumer.h>
#include <linux/ctype.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 15:04:11 +07:00
#include <linux/slab.h>
#include <linux/of.h>
#include <linux/of_graph.h>
#include <linux/dmi.h>
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
#include <sound/core.h>
#include <sound/jack.h>
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
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: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-25 18:12:49 +07:00
#include <sound/soc-dpcm.h>
#include <sound/soc-topology.h>
#include <sound/soc-link.h>
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
#include <sound/initval.h>
#define CREATE_TRACE_POINTS
#include <trace/events/asoc.h>
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
#define NAME_SIZE 32
static DEFINE_MUTEX(client_mutex);
static LIST_HEAD(component_list);
static LIST_HEAD(unbind_card_list);
#define for_each_component(component) \
list_for_each_entry(component, &component_list, list)
/*
* This is used if driver don't need to have CPU/Codec/Platform
* dai_link. see soc.h
*/
struct snd_soc_dai_link_component null_dailink_component[0];
EXPORT_SYMBOL_GPL(null_dailink_component);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
/*
* This is a timeout to do a DAPM powerdown after a stream is closed().
* It can be used to eliminate pops between different playback streams, e.g.
* between two audio tracks.
*/
static int pmdown_time = 5000;
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
static ssize_t pmdown_time_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
return sprintf(buf, "%ld\n", rtd->pmdown_time);
}
static ssize_t pmdown_time_set(struct device *dev,
struct device_attribute *attr,
const char *buf, size_t count)
{
struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
int ret;
ret = kstrtol(buf, 10, &rtd->pmdown_time);
if (ret)
return ret;
return count;
}
static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set);
static struct attribute *soc_dev_attrs[] = {
&dev_attr_pmdown_time.attr,
NULL
};
static umode_t soc_dev_attr_is_visible(struct kobject *kobj,
struct attribute *attr, int idx)
{
struct device *dev = kobj_to_dev(kobj);
struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
if (!rtd)
return 0;
if (attr == &dev_attr_pmdown_time.attr)
return attr->mode; /* always visible */
return rtd->num_codecs ? attr->mode : 0; /* enabled only with codec */
}
static const struct attribute_group soc_dapm_dev_group = {
.attrs = soc_dapm_dev_attrs,
.is_visible = soc_dev_attr_is_visible,
};
static const struct attribute_group soc_dev_group = {
.attrs = soc_dev_attrs,
.is_visible = soc_dev_attr_is_visible,
};
static const struct attribute_group *soc_dev_attr_groups[] = {
&soc_dapm_dev_group,
&soc_dev_group,
NULL
};
#ifdef CONFIG_DEBUG_FS
struct dentry *snd_soc_debugfs_root;
EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
static void soc_init_component_debugfs(struct snd_soc_component *component)
{
if (!component->card->debugfs_card_root)
return;
if (component->debugfs_prefix) {
char *name;
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);
}
snd_soc_dapm_debugfs_init(snd_soc_component_get_dapm(component),
component->debugfs_root);
}
static void soc_cleanup_component_debugfs(struct snd_soc_component *component)
{
if (!component->debugfs_root)
return;
debugfs_remove_recursive(component->debugfs_root);
component->debugfs_root = NULL;
}
static int dai_list_show(struct seq_file *m, void *v)
{
struct snd_soc_component *component;
struct snd_soc_dai *dai;
mutex_lock(&client_mutex);
for_each_component(component)
for_each_component_dais(component, dai)
seq_printf(m, "%s\n", dai->name);
mutex_unlock(&client_mutex);
return 0;
}
DEFINE_SHOW_ATTRIBUTE(dai_list);
static int component_list_show(struct seq_file *m, void *v)
{
struct snd_soc_component *component;
mutex_lock(&client_mutex);
for_each_component(component)
seq_printf(m, "%s\n", component->name);
mutex_unlock(&client_mutex);
return 0;
}
DEFINE_SHOW_ATTRIBUTE(component_list);
static void soc_init_card_debugfs(struct snd_soc_card *card)
{
card->debugfs_card_root = debugfs_create_dir(card->name,
snd_soc_debugfs_root);
debugfs_create_u32("dapm_pop_time", 0644, card->debugfs_card_root,
&card->pop_time);
snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root);
}
static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
{
debugfs_remove_recursive(card->debugfs_card_root);
card->debugfs_card_root = NULL;
}
static void snd_soc_debugfs_init(void)
{
snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL);
debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL,
&dai_list_fops);
debugfs_create_file("components", 0444, snd_soc_debugfs_root, NULL,
&component_list_fops);
}
static void snd_soc_debugfs_exit(void)
{
debugfs_remove_recursive(snd_soc_debugfs_root);
}
#else
static inline void soc_init_component_debugfs(
struct snd_soc_component *component)
{
}
static inline void soc_cleanup_component_debugfs(
struct snd_soc_component *component)
{
}
static inline void soc_init_card_debugfs(struct snd_soc_card *card)
{
}
static inline void soc_cleanup_card_debugfs(struct snd_soc_card *card)
{
}
static inline void snd_soc_debugfs_init(void)
{
}
static inline void snd_soc_debugfs_exit(void)
{
}
#endif
static int snd_soc_rtd_add_component(struct snd_soc_pcm_runtime *rtd,
struct snd_soc_component *component)
{
struct snd_soc_component *comp;
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
int i;
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
for_each_rtd_components(rtd, i, comp) {
/* already connected */
if (comp == component)
return 0;
}
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
/* see for_each_rtd_components */
rtd->components[rtd->num_components] = component;
rtd->num_components++;
return 0;
}
struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd,
const char *driver_name)
{
struct snd_soc_component *component;
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
int i;
if (!driver_name)
return NULL;
/*
* NOTE
*
* snd_soc_rtdcom_lookup() will find component from rtd by using
* specified driver name.
* But, if many components which have same driver name are connected
* to 1 rtd, this function will return 1st found component.
*/
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
for_each_rtd_components(rtd, i, component) {
const char *component_name = component->driver->name;
if (!component_name)
continue;
if ((component_name == driver_name) ||
strcmp(component_name, driver_name) == 0)
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
return component;
}
return NULL;
}
EXPORT_SYMBOL_GPL(snd_soc_rtdcom_lookup);
struct snd_soc_component
*snd_soc_lookup_component_nolocked(struct device *dev, const char *driver_name)
{
struct snd_soc_component *component;
struct snd_soc_component *found_component;
found_component = NULL;
for_each_component(component) {
if ((dev == component->dev) &&
(!driver_name ||
(driver_name == component->driver->name) ||
(strcmp(component->driver->name, driver_name) == 0))) {
found_component = component;
break;
}
}
return found_component;
}
EXPORT_SYMBOL_GPL(snd_soc_lookup_component_nolocked);
struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
const char *driver_name)
{
struct snd_soc_component *component;
mutex_lock(&client_mutex);
component = snd_soc_lookup_component_nolocked(dev, driver_name);
mutex_unlock(&client_mutex);
return component;
}
EXPORT_SYMBOL_GPL(snd_soc_lookup_component);
struct snd_soc_pcm_runtime
*snd_soc_get_pcm_runtime(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_link)
{
struct snd_soc_pcm_runtime *rtd;
for_each_card_rtds(card, rtd) {
if (rtd->dai_link == dai_link)
return rtd;
}
dev_dbg(card->dev, "ASoC: failed to find rtd %s\n", dai_link->name);
return NULL;
}
EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
/*
* Power down the audio subsystem pmdown_time msecs after close is called.
* This is to ensure there are no pops or clicks in between any music tracks
* due to DAPM power cycling.
*/
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int playback = SNDRV_PCM_STREAM_PLAYBACK;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
dev_dbg(rtd->dev,
"ASoC: pop wq checking: %s status: %s waiting: %s\n",
codec_dai->driver->playback.stream_name,
snd_soc_dai_stream_active(codec_dai, playback) ?
"active" : "inactive",
rtd->pop_wait ? "yes" : "no");
/* are we waiting on this codec DAI stream */
if (rtd->pop_wait == 1) {
rtd->pop_wait = 0;
snd_soc_dapm_stream_event(rtd, playback,
SND_SOC_DAPM_STREAM_STOP);
}
mutex_unlock(&rtd->card->pcm_mutex);
}
EXPORT_SYMBOL_GPL(snd_soc_close_delayed_work);
static void soc_release_rtd_dev(struct device *dev)
{
/* "dev" means "rtd->dev" */
kfree(dev);
}
static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd)
{
if (!rtd)
return;
list_del(&rtd->list);
if (delayed_work_pending(&rtd->delayed_work))
flush_delayed_work(&rtd->delayed_work);
ASoC: soc-pcm: remove soc_pcm_private_free() soc-topology adds extra dai_link by using snd_soc_add_dai_link(), and removes it by snd_soc_romove_dai_link(). This snd_soc_add/remove_dai_link() and/or its related functions are unbalanced before, and now, these are balance-uped. But, it finds the random operation issue, and it is reported by Pierre-Louis. When card was released, topology will call snd_soc_remove_dai_link() via (A). static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; /* This should be called before snd_card_free() */ (A) soc_remove_link_components(card); /* free the ALSA card at first; this syncs with pending operations */ if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } /* remove and free each DAI */ (X) soc_remove_link_dais(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... } At (A), topology calls snd_soc_remove_dai_link(). Then topology rtd, and its related all data are freed. Next, (B) is called, and then, pcm->private_free = soc_pcm_private_free() is called. static void soc_pcm_private_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; /* need to sync the delayed work before releasing resources */ flush_delayed_work(&rtd->delayed_work); snd_soc_pcm_component_free(rtd); } Here, it gets rtd via pcm->private_data. But, topology related rtd are already freed at (A). Normal sound card has no damage, becase it frees rtd at (C). These are finalizing rtd related data. Thus, these should be called when rtd was freed, not sound card was freed. It is very natural and understandable. In other words, pcm->private_free = soc_pcm_private_free() is no longer needed. Extra issue is that there is zero chance to call soc_remove_dai() for topology related dai at (X). Because (A) removes rtd connection from card too, and, (X) is based on card connected rtd. This means, (X) need to be called before (C) (= for normal sound) and (A) (= for topology). Now, I want to focus this patch which is the reason why snd_card_free() = (B) is located there. commit 4efda5f2130da033aeedc5b3205569893b910de2 ("ASoC: Fix use-after-free at card unregistration") Original snd_card_free() was called last of this function. But moved to top to avoid use-after-free issue. The issue was happen at soc_pcm_free() which was pcm->private_free, today it is updated/renamed to soc_pcm_private_free(). In other words, (B) need to be called before (C) (= for normal sound) and (A) (= for topology), because it needs (not yet freed) rtd. But, (A) need to be called before (B), because it needs card->snd_card pointer. If we call flush_delayed_work() and snd_soc_pcm_component_free() (= same as soc_pcm_private_free()) when rtd was freed (= (C), (A)), there is no reason to call snd_card_free() at top of this function. It can be called end of this function, again. But, in such case, it will likely break unbind again, as Takashi-san reported. When unbind is performed in a busy state, the code may release still-in-use resources. At least we need to call snd_card_disconnect_sync() at the first place. The final code will be... static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; if (card->snd_card) (Z) snd_card_disconnect_sync(card->snd_card); (X) soc_remove_link_dais(card); (A) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } } To avoid release still-in-use resources, call snd_card_disconnect_sync() at (Z). (X) is needed for both non-topology and topology. topology removes rtd via (A), and non topology removes rtd via (C). snd_card_free() is no longer related to use-after-free issue. Thus, locating (B) is no problem. Fixes: df95a16d2a9626 ("ASoC: soc-core: fix RIP warning on card removal") Fixes: bc7a9091e5b927 ("ASoC: soc-core: add soc_unbind_dai_link()") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/87o8xax88g.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-11-18 08:51:11 +07:00
snd_soc_pcm_component_free(rtd);
/*
* we don't need to call kfree() for rtd->dev
* see
* soc_release_rtd_dev()
*
* We don't need rtd->dev NULL check, because
* it is alloced *before* rtd.
* see
* soc_new_pcm_runtime()
*/
device_unregister(rtd->dev);
}
static void close_delayed_work(struct work_struct *work) {
struct snd_soc_pcm_runtime *rtd =
container_of(work, struct snd_soc_pcm_runtime,
delayed_work.work);
if (rtd->close_delayed_work_func)
rtd->close_delayed_work_func(rtd);
}
2015-11-18 14:34:11 +07:00
static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
struct snd_soc_card *card, struct snd_soc_dai_link *dai_link)
{
struct snd_soc_pcm_runtime *rtd;
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
struct snd_soc_component *component;
struct device *dev;
int ret;
int stream;
2015-11-18 14:34:11 +07:00
/*
* for rtd->dev
*/
dev = kzalloc(sizeof(struct device), GFP_KERNEL);
if (!dev)
return NULL;
dev->parent = card->dev;
dev->release = soc_release_rtd_dev;
dev->groups = soc_dev_attr_groups;
dev_set_name(dev, "%s", dai_link->name);
ret = device_register(dev);
if (ret < 0) {
put_device(dev); /* soc_release_rtd_dev */
return NULL;
}
/*
* for rtd
*/
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
rtd = devm_kzalloc(dev,
sizeof(*rtd) +
sizeof(*component) * (dai_link->num_cpus +
dai_link->num_codecs +
dai_link->num_platforms),
GFP_KERNEL);
2015-11-18 14:34:11 +07:00
if (!rtd)
goto free_rtd;
2015-11-18 14:34:11 +07:00
rtd->dev = dev;
INIT_LIST_HEAD(&rtd->list);
for_each_pcm_streams(stream) {
INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients);
INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients);
}
dev_set_drvdata(dev, rtd);
INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
/*
* for rtd->dais
*/
rtd->dais = devm_kcalloc(dev, dai_link->num_cpus + dai_link->num_codecs,
treewide: kzalloc() -> kcalloc() The kzalloc() function has a 2-factor argument form, kcalloc(). This patch replaces cases of: kzalloc(a * b, gfp) with: kcalloc(a * b, gfp) as well as handling cases of: kzalloc(a * b * c, gfp) with: kzalloc(array3_size(a, b, c), gfp) as it's slightly less ugly than: kzalloc_array(array_size(a, b), c, gfp) This does, however, attempt to ignore constant size factors like: kzalloc(4 * 1024, gfp) though any constants defined via macros get caught up in the conversion. Any factors with a sizeof() of "unsigned char", "char", and "u8" were dropped, since they're redundant. The Coccinelle script used for this was: // Fix redundant parens around sizeof(). @@ type TYPE; expression THING, E; @@ ( kzalloc( - (sizeof(TYPE)) * E + sizeof(TYPE) * E , ...) | kzalloc( - (sizeof(THING)) * E + sizeof(THING) * E , ...) ) // Drop single-byte sizes and redundant parens. @@ expression COUNT; typedef u8; typedef __u8; @@ ( kzalloc( - sizeof(u8) * (COUNT) + COUNT , ...) | kzalloc( - sizeof(__u8) * (COUNT) + COUNT , ...) | kzalloc( - sizeof(char) * (COUNT) + COUNT , ...) | kzalloc( - sizeof(unsigned char) * (COUNT) + COUNT , ...) | kzalloc( - sizeof(u8) * COUNT + COUNT , ...) | kzalloc( - sizeof(__u8) * COUNT + COUNT , ...) | kzalloc( - sizeof(char) * COUNT + COUNT , ...) | kzalloc( - sizeof(unsigned char) * COUNT + COUNT , ...) ) // 2-factor product with sizeof(type/expression) and identifier or constant. @@ type TYPE; expression THING; identifier COUNT_ID; constant COUNT_CONST; @@ ( - kzalloc + kcalloc ( - sizeof(TYPE) * (COUNT_ID) + COUNT_ID, sizeof(TYPE) , ...) | - kzalloc + kcalloc ( - sizeof(TYPE) * COUNT_ID + COUNT_ID, sizeof(TYPE) , ...) | - kzalloc + kcalloc ( - sizeof(TYPE) * (COUNT_CONST) + COUNT_CONST, sizeof(TYPE) , ...) | - kzalloc + kcalloc ( - sizeof(TYPE) * COUNT_CONST + COUNT_CONST, sizeof(TYPE) , ...) | - kzalloc + kcalloc ( - sizeof(THING) * (COUNT_ID) + COUNT_ID, sizeof(THING) , ...) | - kzalloc + kcalloc ( - sizeof(THING) * COUNT_ID + COUNT_ID, sizeof(THING) , ...) | - kzalloc + kcalloc ( - sizeof(THING) * (COUNT_CONST) + COUNT_CONST, sizeof(THING) , ...) | - kzalloc + kcalloc ( - sizeof(THING) * COUNT_CONST + COUNT_CONST, sizeof(THING) , ...) ) // 2-factor product, only identifiers. @@ identifier SIZE, COUNT; @@ - kzalloc + kcalloc ( - SIZE * COUNT + COUNT, SIZE , ...) // 3-factor product with 1 sizeof(type) or sizeof(expression), with // redundant parens removed. @@ expression THING; identifier STRIDE, COUNT; type TYPE; @@ ( kzalloc( - sizeof(TYPE) * (COUNT) * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | kzalloc( - sizeof(TYPE) * (COUNT) * STRIDE + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | kzalloc( - sizeof(TYPE) * COUNT * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | kzalloc( - sizeof(TYPE) * COUNT * STRIDE + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | kzalloc( - sizeof(THING) * (COUNT) * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | kzalloc( - sizeof(THING) * (COUNT) * STRIDE + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | kzalloc( - sizeof(THING) * COUNT * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | kzalloc( - sizeof(THING) * COUNT * STRIDE + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) ) // 3-factor product with 2 sizeof(variable), with redundant parens removed. @@ expression THING1, THING2; identifier COUNT; type TYPE1, TYPE2; @@ ( kzalloc( - sizeof(TYPE1) * sizeof(TYPE2) * COUNT + array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2)) , ...) | kzalloc( - sizeof(TYPE1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2)) , ...) | kzalloc( - sizeof(THING1) * sizeof(THING2) * COUNT + array3_size(COUNT, sizeof(THING1), sizeof(THING2)) , ...) | kzalloc( - sizeof(THING1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(THING1), sizeof(THING2)) , ...) | kzalloc( - sizeof(TYPE1) * sizeof(THING2) * COUNT + array3_size(COUNT, sizeof(TYPE1), sizeof(THING2)) , ...) | kzalloc( - sizeof(TYPE1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(TYPE1), sizeof(THING2)) , ...) ) // 3-factor product, only identifiers, with redundant parens removed. @@ identifier STRIDE, SIZE, COUNT; @@ ( kzalloc( - (COUNT) * STRIDE * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | kzalloc( - COUNT * (STRIDE) * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | kzalloc( - COUNT * STRIDE * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | kzalloc( - (COUNT) * (STRIDE) * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | kzalloc( - COUNT * (STRIDE) * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | kzalloc( - (COUNT) * STRIDE * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | kzalloc( - (COUNT) * (STRIDE) * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | kzalloc( - COUNT * STRIDE * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) ) // Any remaining multi-factor products, first at least 3-factor products, // when they're not all constants... @@ expression E1, E2, E3; constant C1, C2, C3; @@ ( kzalloc(C1 * C2 * C3, ...) | kzalloc( - (E1) * E2 * E3 + array3_size(E1, E2, E3) , ...) | kzalloc( - (E1) * (E2) * E3 + array3_size(E1, E2, E3) , ...) | kzalloc( - (E1) * (E2) * (E3) + array3_size(E1, E2, E3) , ...) | kzalloc( - E1 * E2 * E3 + array3_size(E1, E2, E3) , ...) ) // And then all remaining 2 factors products when they're not all constants, // keeping sizeof() as the second factor argument. @@ expression THING, E1, E2; type TYPE; constant C1, C2, C3; @@ ( kzalloc(sizeof(THING) * C2, ...) | kzalloc(sizeof(TYPE) * C2, ...) | kzalloc(C1 * C2 * C3, ...) | kzalloc(C1 * C2, ...) | - kzalloc + kcalloc ( - sizeof(TYPE) * (E2) + E2, sizeof(TYPE) , ...) | - kzalloc + kcalloc ( - sizeof(TYPE) * E2 + E2, sizeof(TYPE) , ...) | - kzalloc + kcalloc ( - sizeof(THING) * (E2) + E2, sizeof(THING) , ...) | - kzalloc + kcalloc ( - sizeof(THING) * E2 + E2, sizeof(THING) , ...) | - kzalloc + kcalloc ( - (E1) * E2 + E1, E2 , ...) | - kzalloc + kcalloc ( - (E1) * (E2) + E1, E2 , ...) | - kzalloc + kcalloc ( - E1 * E2 + E1, E2 , ...) ) Signed-off-by: Kees Cook <keescook@chromium.org>
2018-06-13 04:03:40 +07:00
sizeof(struct snd_soc_dai *),
2015-11-18 14:34:11 +07:00
GFP_KERNEL);
if (!rtd->dais)
goto free_rtd;
/*
* dais = [][][][][][][][][][][][][][][][][][]
* ^cpu_dais ^codec_dais
* |--- num_cpus ---|--- num_codecs --|
* see
* asoc_rtd_to_cpu()
* asoc_rtd_to_codec()
*/
rtd->num_cpus = dai_link->num_cpus;
rtd->num_codecs = dai_link->num_codecs;
rtd->card = card;
rtd->dai_link = dai_link;
rtd->num = card->num_rtd++;
/* see for_each_card_rtds */
2015-11-18 14:34:11 +07:00
list_add_tail(&rtd->list, &card->rtd_list);
return rtd;
free_rtd:
soc_free_pcm_runtime(rtd);
return NULL;
2015-11-18 14:34:11 +07:00
}
static void snd_soc_flush_all_delayed_work(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd;
for_each_card_rtds(card, rtd)
flush_delayed_work(&rtd->delayed_work);
}
#ifdef CONFIG_PM_SLEEP
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
/* powers down audio subsystem for suspend */
int snd_soc_suspend(struct device *dev)
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
{
struct snd_soc_card *card = dev_get_drvdata(dev);
struct snd_soc_component *component;
2015-11-18 14:34:11 +07:00
struct snd_soc_pcm_runtime *rtd;
int playback = SNDRV_PCM_STREAM_PLAYBACK;
2015-11-18 14:34:11 +07:00
int i;
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
/* 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
* suspend before that's finished - wait for it to complete.
*/
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0);
/* we're going to block userspace touching us until resume completes */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot);
/* mute any active DACs */
for_each_card_rtds(card, rtd) {
struct snd_soc_dai *dai;
2015-11-18 14:34:11 +07:00
if (rtd->dai_link->ignore_suspend)
continue;
for_each_rtd_dais(rtd, i, dai) {
if (snd_soc_dai_stream_active(dai, playback))
snd_soc_dai_digital_mute(dai, 1, playback);
}
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
}
/* suspend all pcms */
for_each_card_rtds(card, rtd) {
2015-11-18 14:34:11 +07:00
if (rtd->dai_link->ignore_suspend)
continue;
2015-11-18 14:34:11 +07:00
snd_pcm_suspend_all(rtd->pcm);
}
snd_soc_card_suspend_pre(card);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
/* close any waiting streams */
snd_soc_flush_all_delayed_work(card);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
for_each_card_rtds(card, rtd) {
int stream;
2015-11-18 14:34:11 +07:00
if (rtd->dai_link->ignore_suspend)
continue;
for_each_pcm_streams(stream)
snd_soc_dapm_stream_event(rtd, stream,
SND_SOC_DAPM_STREAM_SUSPEND);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
}
/* Recheck all endpoints too, their state is affected by suspend */
dapm_mark_endpoints_dirty(card);
snd_soc_dapm_sync(&card->dapm);
/* suspend all COMPONENTs */
for_each_card_rtds(card, rtd) {
if (rtd->dai_link->ignore_suspend)
continue;
for_each_rtd_components(rtd, i, component) {
struct snd_soc_dapm_context *dapm =
snd_soc_component_get_dapm(component);
/*
* ignore if component was already suspended
*/
if (snd_soc_component_is_suspended(component))
continue;
/*
* If there are paths active then the COMPONENT will be
* held with bias _ON and should not be suspended.
*/
switch (snd_soc_dapm_get_bias_level(dapm)) {
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
case SND_SOC_BIAS_STANDBY:
/*
* If the COMPONENT is capable of idle
* bias off then being in STANDBY
* means it's doing something,
* otherwise fall through.
*/
if (dapm->idle_bias_off) {
dev_dbg(component->dev,
"ASoC: idle_bias_off CODEC on over suspend\n");
break;
}
/* fall through */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
case SND_SOC_BIAS_OFF:
snd_soc_component_suspend(component);
if (component->regmap)
regcache_mark_dirty(component->regmap);
/* deactivate pins to sleep state */
pinctrl_pm_select_sleep_state(component->dev);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
break;
default:
dev_dbg(component->dev,
"ASoC: COMPONENT is on over suspend\n");
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
break;
}
}
}
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
snd_soc_card_suspend_post(card);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_suspend);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
/*
* deferred resume work, so resume can complete before we finished
* setting our codec back up, which can be very slow on I2C
*/
static void soc_resume_deferred(struct work_struct *work)
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
struct snd_soc_card *card =
container_of(work, struct snd_soc_card,
deferred_resume_work);
2015-11-18 14:34:11 +07:00
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_component *component;
2015-11-18 14:34:11 +07:00
int i;
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
/*
* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
* so userspace apps are blocked from touching us
*/
dev_dbg(card->dev, "ASoC: starting resume work\n");
/* Bring us up into D2 so that DAPM starts enabling things */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2);
snd_soc_card_resume_pre(card);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
for_each_card_components(card, component) {
if (snd_soc_component_is_suspended(component))
snd_soc_component_resume(component);
}
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
for_each_card_rtds(card, rtd) {
int stream;
2015-11-18 14:34:11 +07:00
if (rtd->dai_link->ignore_suspend)
continue;
for_each_pcm_streams(stream)
snd_soc_dapm_stream_event(rtd, stream,
SND_SOC_DAPM_STREAM_RESUME);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
}
/* unmute any active DACs */
for_each_card_rtds(card, rtd) {
struct snd_soc_dai *dai;
int playback = SNDRV_PCM_STREAM_PLAYBACK;
2015-11-18 14:34:11 +07:00
if (rtd->dai_link->ignore_suspend)
continue;
for_each_rtd_dais(rtd, i, dai) {
if (snd_soc_dai_stream_active(dai, playback))
snd_soc_dai_digital_mute(dai, 0, playback);
}
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
}
snd_soc_card_resume_post(card);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
dev_dbg(card->dev, "ASoC: resume work completed\n");
/* Recheck all endpoints too, their state is affected by suspend */
dapm_mark_endpoints_dirty(card);
snd_soc_dapm_sync(&card->dapm);
/* userspace can access us now we are back as we were before */
snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0);
}
/* powers up audio subsystem after a suspend */
int snd_soc_resume(struct device *dev)
{
struct snd_soc_card *card = dev_get_drvdata(dev);
struct snd_soc_component *component;
/* If the card is not initialized yet there is nothing to do */
if (!card->instantiated)
return 0;
/* activate pins from sleep state */
for_each_card_components(card, component)
if (snd_soc_component_active(component))
pinctrl_pm_select_default_state(component->dev);
dev_dbg(dev, "ASoC: Scheduling resume work\n");
if (!schedule_work(&card->deferred_resume_work))
dev_err(dev, "ASoC: resume work item may be lost\n");
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_resume);
static void soc_resume_init(struct snd_soc_card *card)
{
/* deferred resume work */
INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
}
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
#else
#define snd_soc_suspend NULL
#define snd_soc_resume NULL
static inline void soc_resume_init(struct snd_soc_card *card)
{
}
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
#endif
static struct device_node
*soc_component_to_node(struct snd_soc_component *component)
{
struct device_node *of_node;
of_node = component->dev->of_node;
if (!of_node && component->dev->parent)
of_node = component->dev->parent->of_node;
return of_node;
}
static int snd_soc_is_matching_component(
const struct snd_soc_dai_link_component *dlc,
struct snd_soc_component *component)
{
struct device_node *component_of_node;
if (!dlc)
return 0;
component_of_node = soc_component_to_node(component);
if (dlc->of_node && component_of_node != dlc->of_node)
return 0;
if (dlc->name && strcmp(component->name, dlc->name))
return 0;
return 1;
}
static struct snd_soc_component *soc_find_component(
const struct snd_soc_dai_link_component *dlc)
{
struct snd_soc_component *component;
lockdep_assert_held(&client_mutex);
/*
* NOTE
*
* It returns *1st* found component, but some driver
* has few components by same of_node/name
* ex)
* CPU component and generic DMAEngine component
*/
for_each_component(component)
if (snd_soc_is_matching_component(dlc, component))
return component;
return NULL;
}
/**
* snd_soc_find_dai - Find a registered DAI
*
* @dlc: name of the DAI or the DAI driver and optional component info to match
*
* This function will search all registered components and their DAIs to
* find the DAI of the same name. The component's of_node and name
* should also match if being specified.
*
* Return: pointer of DAI, or NULL if not found.
*/
struct snd_soc_dai *snd_soc_find_dai(
const struct snd_soc_dai_link_component *dlc)
{
struct snd_soc_component *component;
struct snd_soc_dai *dai;
lockdep_assert_held(&client_mutex);
/* Find CPU DAI from registered DAIs */
for_each_component(component) {
if (!snd_soc_is_matching_component(dlc, component))
continue;
for_each_component_dais(component, dai) {
if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)
&& (!dai->driver->name
|| strcmp(dai->driver->name, dlc->dai_name)))
continue;
return dai;
}
}
return NULL;
}
EXPORT_SYMBOL_GPL(snd_soc_find_dai);
static int soc_dai_link_sanity_check(struct snd_soc_card *card,
struct snd_soc_dai_link *link)
{
int i;
struct snd_soc_dai_link_component *cpu, *codec, *platform;
for_each_link_codecs(link, i, codec) {
/*
* Codec must be specified by 1 of name or OF node,
* not both or neither.
*/
if (!!codec->name == !!codec->of_node) {
dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n",
link->name);
return -EINVAL;
}
/* Codec DAI name must be specified */
if (!codec->dai_name) {
dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n",
link->name);
return -EINVAL;
}
/*
* Defer card registration if codec component is not added to
* component list.
*/
if (!soc_find_component(codec)) {
dev_dbg(card->dev,
"ASoC: codec component %s not found for link %s\n",
codec->name, link->name);
return -EPROBE_DEFER;
}
}
for_each_link_platforms(link, i, platform) {
/*
* Platform may be specified by either name or OF node, but it
* can be left unspecified, then no components will be inserted
* in the rtdcom list
*/
if (!!platform->name == !!platform->of_node) {
dev_err(card->dev,
"ASoC: Neither/both platform name/of_node are set for %s\n",
link->name);
return -EINVAL;
}
/*
* Defer card registration if platform component is not added to
* component list.
*/
if (!soc_find_component(platform)) {
dev_dbg(card->dev,
"ASoC: platform component %s not found for link %s\n",
platform->name, link->name);
return -EPROBE_DEFER;
}
}
for_each_link_cpus(link, i, cpu) {
/*
* CPU device may be specified by either name or OF node, but
* can be left unspecified, and will be matched based on DAI
* name alone..
*/
if (cpu->name && cpu->of_node) {
dev_err(card->dev,
"ASoC: Neither/both cpu name/of_node are set for %s\n",
link->name);
return -EINVAL;
}
/*
* Defer card registration if cpu dai component is not added to
* component list.
*/
if ((cpu->of_node || cpu->name) &&
!soc_find_component(cpu)) {
dev_dbg(card->dev,
"ASoC: cpu component %s not found for link %s\n",
cpu->name, link->name);
return -EPROBE_DEFER;
}
/*
* At least one of CPU DAI name or CPU device name/node must be
* specified
*/
if (!cpu->dai_name &&
!(cpu->name || cpu->of_node)) {
dev_err(card->dev,
"ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n",
link->name);
return -EINVAL;
}
}
return 0;
}
/**
* snd_soc_remove_pcm_runtime - Remove a pcm_runtime from card
* @card: The ASoC card to which the pcm_runtime has
* @rtd: The pcm_runtime to remove
*
* This function removes a pcm_runtime from the ASoC card.
*/
void snd_soc_remove_pcm_runtime(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd)
{
lockdep_assert_held(&client_mutex);
/* release machine specific resources */
snd_soc_link_exit(rtd);
/*
* Notify the machine driver for extra destruction
*/
snd_soc_card_remove_dai_link(card, rtd->dai_link);
soc_free_pcm_runtime(rtd);
}
EXPORT_SYMBOL_GPL(snd_soc_remove_pcm_runtime);
/**
* snd_soc_add_pcm_runtime - Add a pcm_runtime dynamically via dai_link
* @card: The ASoC card to which the pcm_runtime is added
* @dai_link: The DAI link to find pcm_runtime
*
* This function adds a pcm_runtime ASoC card by using dai_link.
*
* Note: Topology can use this API to add pcm_runtime when probing the
* topology component. And machine drivers can still define static
* DAI links in dai_link array.
*/
int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_link)
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
{
2015-11-18 14:34:11 +07:00
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_dai_link_component *codec, *platform, *cpu;
struct snd_soc_component *component;
int i, ret;
lockdep_assert_held(&client_mutex);
/*
* Notify the machine driver for extra initialization
*/
ret = snd_soc_card_add_dai_link(card, dai_link);
if (ret < 0)
return ret;
if (dai_link->ignore)
return 0;
dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name);
ret = soc_dai_link_sanity_check(card, dai_link);
if (ret < 0)
return ret;
rtd = soc_new_pcm_runtime(card, dai_link);
if (!rtd)
return -ENOMEM;
for_each_link_cpus(dai_link, i, cpu) {
asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai(cpu);
if (!asoc_rtd_to_cpu(rtd, i)) {
dev_info(card->dev, "ASoC: CPU DAI %s not registered\n",
cpu->dai_name);
goto _err_defer;
}
snd_soc_rtd_add_component(rtd, asoc_rtd_to_cpu(rtd, i)->component);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
}
/* Find CODEC from registered CODECs */
for_each_link_codecs(dai_link, i, codec) {
asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec);
if (!asoc_rtd_to_codec(rtd, i)) {
dev_info(card->dev, "ASoC: CODEC DAI %s not registered\n",
codec->dai_name);
2015-11-18 14:34:11 +07:00
goto _err_defer;
}
snd_soc_rtd_add_component(rtd, asoc_rtd_to_codec(rtd, i)->component);
}
/* Find PLATFORM from registered PLATFORMs */
for_each_link_platforms(dai_link, i, platform) {
for_each_component(component) {
if (!snd_soc_is_matching_component(platform, component))
continue;
snd_soc_rtd_add_component(rtd, component);
}
}
return 0;
2015-11-18 14:34:11 +07:00
_err_defer:
snd_soc_remove_pcm_runtime(card, rtd);
return -EPROBE_DEFER;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
}
EXPORT_SYMBOL_GPL(snd_soc_add_pcm_runtime);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
static int soc_init_pcm_runtime(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_soc_component *component;
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
int ret, num, i;
/* set default power off timeout */
rtd->pmdown_time = pmdown_time;
/* do machine specific initialization */
ret = snd_soc_link_init(rtd);
if (ret < 0)
return ret;
if (dai_link->dai_fmt) {
ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
if (ret)
return ret;
}
/* add DPCM sysfs entries */
soc_dpcm_debugfs_add(rtd);
num = rtd->num;
/*
* most drivers will register their PCMs using DAI link ordering but
* topology based drivers can use the DAI link id field to set PCM
* device number and then use rtd + a base offset of the BEs.
*/
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
for_each_rtd_components(rtd, i, component) {
if (!component->driver->use_dai_pcm_id)
continue;
if (rtd->dai_link->no_pcm)
num += component->driver->be_pcm_base;
else
num = rtd->dai_link->id;
}
/* create compress_device if possible */
ret = snd_soc_dai_compress_new(cpu_dai, rtd, num);
if (ret != -ENOTSUPP) {
if (ret < 0)
dev_err(card->dev, "ASoC: can't create compress %s\n",
dai_link->stream_name);
return ret;
}
/* create the pcm */
ret = soc_new_pcm(rtd, num);
if (ret < 0) {
dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
dai_link->stream_name, ret);
return ret;
}
return snd_soc_pcm_dai_new(rtd);
}
static void soc_set_name_prefix(struct snd_soc_card *card,
struct snd_soc_component *component)
{
struct device_node *of_node = soc_component_to_node(component);
const char *str;
int ret, i;
for (i = 0; i < card->num_configs; i++) {
struct snd_soc_codec_conf *map = &card->codec_conf[i];
if (snd_soc_is_matching_component(&map->dlc, component)) {
component->name_prefix = map->name_prefix;
return;
}
}
/*
* If there is no configuration table or no match in the table,
* check if a prefix is provided in the node
*/
ret = of_property_read_string(of_node, "sound-name-prefix", &str);
if (ret < 0)
return;
component->name_prefix = str;
}
static void soc_remove_component(struct snd_soc_component *component,
int probed)
{
if (!component->card)
return;
if (probed)
snd_soc_component_remove(component);
/* For framework level robustness */
snd_soc_component_set_jack(component, NULL, NULL);
list_del_init(&component->card_list);
snd_soc_dapm_free(snd_soc_component_get_dapm(component));
soc_cleanup_component_debugfs(component);
component->card = NULL;
snd_soc_component_module_put_when_remove(component);
}
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_dai *dai;
int probed = 0;
int ret;
if (!strcmp(component->name, "snd-soc-dummy"))
return 0;
if (component->card) {
if (component->card != card) {
dev_err(component->dev,
"Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n",
card->name, component->card->name);
return -ENODEV;
}
return 0;
}
ret = snd_soc_component_module_get_when_probe(component);
if (ret < 0)
return ret;
component->card = card;
soc_set_name_prefix(card, component);
soc_init_component_debugfs(component);
snd_soc_dapm_init(dapm, card, component);
ret = snd_soc_dapm_new_controls(dapm,
component->driver->dapm_widgets,
component->driver->num_dapm_widgets);
if (ret != 0) {
dev_err(component->dev,
"Failed to create new controls %d\n", ret);
goto err_probe;
}
for_each_component_dais(component, dai) {
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;
}
}
ret = snd_soc_component_probe(component);
if (ret < 0) {
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);
probed = 1;
/*
* machine specific init
* see
* snd_soc_component_set_aux()
*/
ret = snd_soc_component_init(component);
if (ret < 0)
goto err_probe;
ret = snd_soc_add_component_controls(component,
component->driver->controls,
component->driver->num_controls);
if (ret < 0)
goto err_probe;
ret = snd_soc_dapm_add_routes(dapm,
component->driver->dapm_routes,
component->driver->num_dapm_routes);
if (ret < 0) {
if (card->disable_route_checks) {
dev_info(card->dev,
"%s: disable_route_checks set, ignoring errors on add_routes\n",
__func__);
} else {
dev_err(card->dev,
"%s: snd_soc_dapm_add_routes failed: %d\n",
__func__, ret);
goto err_probe;
}
}
/* see for_each_card_components */
list_add(&component->card_list, &card->component_dev_list);
err_probe:
if (ret < 0)
soc_remove_component(component, probed);
return ret;
}
static void soc_remove_link_dais(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd;
int order;
for_each_comp_order(order) {
for_each_card_rtds(card, rtd) {
/* remove all rtd connected DAIs in good order */
snd_soc_pcm_dai_remove(rtd, order);
}
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
}
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
static int soc_probe_link_dais(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd;
int order, ret;
for_each_comp_order(order) {
for_each_card_rtds(card, rtd) {
dev_dbg(card->dev,
"ASoC: probe %s dai link %d late %d\n",
card->name, rtd->num, order);
/* probe all rtd connected DAIs in good order */
ret = snd_soc_pcm_dai_probe(rtd, order);
if (ret)
return ret;
}
}
return 0;
}
static void soc_remove_link_components(struct snd_soc_card *card)
{
struct snd_soc_component *component;
struct snd_soc_pcm_runtime *rtd;
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
int i, order;
for_each_comp_order(order) {
for_each_card_rtds(card, rtd) {
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
for_each_rtd_components(rtd, i, component) {
if (component->driver->remove_order != order)
continue;
soc_remove_component(component, 1);
}
}
}
}
static int soc_probe_link_components(struct snd_soc_card *card)
{
struct snd_soc_component *component;
struct snd_soc_pcm_runtime *rtd;
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
int i, ret, order;
for_each_comp_order(order) {
for_each_card_rtds(card, rtd) {
ASoC: soc-core: remove snd_soc_rtdcom_list Current ALSA SoC is using struct snd_soc_rtdcom_list to connecting component to rtd by using list_head. struct snd_soc_rtdcom_list { struct snd_soc_component *component; struct list_head list; /* rtd::component_list */ }; struct snd_soc_pcm_runtime { ... struct list_head component_list; /* list of connected components */ ... }; The CPU/Codec/Platform component which will be connected to rtd (a) is indicated via dai_link at snd_soc_add_pcm_runtime() int snd_soc_add_pcm_runtime(...) { ... /* Find CPU from registered CPUs */ rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); ... (a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component); ... /* Find CODEC from registered CODECs */ (b) for_each_link_codecs(dai_link, i, codec) { rtd->codec_dais[i] = snd_soc_find_dai(codec); ... (a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component); } ... /* Find PLATFORM from registered PLATFORMs */ (b) for_each_link_platforms(dai_link, i, platform) { for_each_component(component) { ... (a) snd_soc_rtdcom_add(rtd, component); } } } It shows, it is possible to know how many components will be connected to rtd by using dai_link->num_cpus dai_link->num_codecs dai_link->num_platforms If so, we can use component pointer array instead of list_head, in such case, code can be more simple. This patch removes struct snd_soc_rtdcom_list that is only of temporary value, and convert to pointer array. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-01-10 09:35:21 +07:00
for_each_rtd_components(rtd, i, component) {
if (component->driver->probe_order != order)
continue;
ret = soc_probe_component(card, component);
if (ret < 0)
return ret;
}
}
}
return 0;
}
static void soc_unbind_aux_dev(struct snd_soc_card *card)
{
struct snd_soc_component *component, *_component;
for_each_card_auxs_safe(card, component, _component) {
/* for snd_soc_component_init() */
snd_soc_component_set_aux(component, NULL);
list_del(&component->card_aux_list);
}
}
static int soc_bind_aux_dev(struct snd_soc_card *card)
{
struct snd_soc_component *component;
struct snd_soc_aux_dev *aux;
int i;
for_each_card_pre_auxs(card, i, aux) {
/* codecs, usually analog devices */
component = soc_find_component(&aux->dlc);
if (!component)
return -EPROBE_DEFER;
/* for snd_soc_component_init() */
snd_soc_component_set_aux(component, aux);
/* see for_each_card_auxs */
list_add(&component->card_aux_list, &card->aux_comp_list);
}
return 0;
}
static int soc_probe_aux_devices(struct snd_soc_card *card)
{
struct snd_soc_component *component;
int order;
int ret;
for_each_comp_order(order) {
for_each_card_auxs(card, component) {
if (component->driver->probe_order != order)
continue;
ret = soc_probe_component(card, component);
if (ret < 0)
return ret;
}
}
return 0;
}
static void soc_remove_aux_devices(struct snd_soc_card *card)
{
struct snd_soc_component *comp, *_comp;
int order;
for_each_comp_order(order) {
for_each_card_auxs_safe(card, comp, _comp) {
if (comp->driver->remove_order == order)
soc_remove_component(comp, 1);
}
}
}
/**
* snd_soc_runtime_set_dai_fmt() - Change DAI link format for a ASoC runtime
* @rtd: The runtime for which the DAI link format should be changed
* @dai_fmt: The new DAI link format
*
* This function updates the DAI link format for all DAIs connected to the DAI
* link for the specified runtime.
*
* Note: For setups with a static format set the dai_fmt field in the
* corresponding snd_dai_link struct instead of using this function.
*
* Returns 0 on success, otherwise a negative error code.
*/
int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
unsigned int dai_fmt)
{
struct snd_soc_dai *cpu_dai;
struct snd_soc_dai *codec_dai;
unsigned int inv_dai_fmt;
unsigned int i;
int ret;
for_each_rtd_codec_dais(rtd, i, codec_dai) {
ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
if (ret != 0 && ret != -ENOTSUPP) {
dev_warn(codec_dai->dev,
"ASoC: Failed to set DAI format: %d\n", ret);
return ret;
}
}
/*
* Flip the polarity for the "CPU" end of a CODEC<->CODEC link
* the component which has non_legacy_dai_naming is Codec
*/
inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK;
switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
break;
case SND_SOC_DAIFMT_CBM_CFS:
inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM;
break;
case SND_SOC_DAIFMT_CBS_CFM:
inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS;
break;
case SND_SOC_DAIFMT_CBS_CFS:
inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
break;
}
for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
unsigned int fmt = dai_fmt;
if (cpu_dai->component->driver->non_legacy_dai_naming)
fmt = inv_dai_fmt;
ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
if (ret != 0 && ret != -ENOTSUPP) {
dev_warn(cpu_dai->dev,
"ASoC: Failed to set DAI format: %d\n", ret);
return ret;
}
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt);
#ifdef CONFIG_DMI
/*
* If a DMI filed contain strings in this blacklist (e.g.
* "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken
* as invalid and dropped when setting the card long name from DMI info.
*/
static const char * const dmi_blacklist[] = {
"To be filled by OEM",
"TBD by OEM",
"Default String",
"Board Manufacturer",
"Board Vendor Name",
"Board Product Name",
NULL, /* terminator */
};
/*
* Trim special characters, and replace '-' with '_' since '-' is used to
* separate different DMI fields in the card long name. Only number and
* alphabet characters and a few separator characters are kept.
*/
static void cleanup_dmi_name(char *name)
{
int i, j = 0;
for (i = 0; name[i]; i++) {
if (isalnum(name[i]) || (name[i] == '.')
|| (name[i] == '_'))
name[j++] = name[i];
else if (name[i] == '-')
name[j++] = '_';
}
name[j] = '\0';
}
/*
* Check if a DMI field is valid, i.e. not containing any string
* in the black list.
*/
static int is_dmi_valid(const char *field)
{
int i = 0;
while (dmi_blacklist[i]) {
if (strstr(field, dmi_blacklist[i]))
return 0;
i++;
}
return 1;
}
/*
* Append a string to card->dmi_longname with character cleanups.
*/
static void append_dmi_string(struct snd_soc_card *card, const char *str)
{
char *dst = card->dmi_longname;
size_t dst_len = sizeof(card->dmi_longname);
size_t len;
len = strlen(dst);
snprintf(dst + len, dst_len - len, "-%s", str);
len++; /* skip the separator "-" */
if (len < dst_len)
cleanup_dmi_name(dst + len);
}
/**
* snd_soc_set_dmi_name() - Register DMI names to card
* @card: The card to register DMI names
* @flavour: The flavour "differentiator" for the card amongst its peers.
*
* An Intel machine driver may be used by many different devices but are
* difficult for userspace to differentiate, since machine drivers ususally
* use their own name as the card short name and leave the card long name
* blank. To differentiate such devices and fix bugs due to lack of
* device-specific configurations, this function allows DMI info to be used
* as the sound card long name, in the format of
* "vendor-product-version-board"
* (Character '-' is used to separate different DMI fields here).
* This will help the user space to load the device-specific Use Case Manager
* (UCM) configurations for the card.
*
* Possible card long names may be:
* DellInc.-XPS139343-01-0310JH
* ASUSTeKCOMPUTERINC.-T100TA-1.0-T100TA
* Circuitco-MinnowboardMaxD0PLATFORM-D0-MinnowBoardMAX
*
* This function also supports flavoring the card longname to provide
* the extra differentiation, like "vendor-product-version-board-flavor".
*
* We only keep number and alphabet characters and a few separator characters
* in the card long name since UCM in the user space uses the card long names
* as card configuration directory names and AudoConf cannot support special
* charactors like SPACE.
*
* Returns 0 on success, otherwise a negative error code.
*/
int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
{
const char *vendor, *product, *product_version, *board;
if (card->long_name)
return 0; /* long name already set by driver or from DMI */
/* make up dmi long name as: vendor-product-version-board */
vendor = dmi_get_system_info(DMI_BOARD_VENDOR);
if (!vendor || !is_dmi_valid(vendor)) {
dev_warn(card->dev, "ASoC: no DMI vendor name!\n");
return 0;
}
snprintf(card->dmi_longname, sizeof(card->dmi_longname), "%s", vendor);
cleanup_dmi_name(card->dmi_longname);
product = dmi_get_system_info(DMI_PRODUCT_NAME);
if (product && is_dmi_valid(product)) {
append_dmi_string(card, product);
/*
* some vendors like Lenovo may only put a self-explanatory
* name in the product version field
*/
product_version = dmi_get_system_info(DMI_PRODUCT_VERSION);
if (product_version && is_dmi_valid(product_version))
append_dmi_string(card, product_version);
}
board = dmi_get_system_info(DMI_BOARD_NAME);
if (board && is_dmi_valid(board)) {
if (!product || strcasecmp(board, product))
append_dmi_string(card, board);
} else if (!product) {
/* fall back to using legacy name */
dev_warn(card->dev, "ASoC: no DMI board/product name!\n");
return 0;
}
/* Add flavour to dmi long name */
if (flavour)
append_dmi_string(card, flavour);
/* set the card long name */
card->long_name = card->dmi_longname;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name);
#endif /* CONFIG_DMI */
static void soc_check_tplg_fes(struct snd_soc_card *card)
{
struct snd_soc_component *component;
const struct snd_soc_component_driver *comp_drv;
struct snd_soc_dai_link *dai_link;
int i;
for_each_component(component) {
/* does this component override BEs ? */
if (!component->driver->ignore_machine)
continue;
/* for this machine ? */
if (!strcmp(component->driver->ignore_machine,
card->dev->driver->name))
goto match;
if (strcmp(component->driver->ignore_machine,
dev_name(card->dev)))
continue;
match:
/* machine matches, so override the rtd data */
for_each_card_prelinks(card, i, dai_link) {
/* ignore this FE */
if (dai_link->dynamic) {
dai_link->ignore = true;
continue;
}
dev_dbg(card->dev, "info: override BE DAI link %s\n",
card->dai_link[i].name);
/* override platform component */
if (!dai_link->platforms) {
dev_err(card->dev, "init platform error");
continue;
}
dai_link->platforms->name = component->name;
/* convert non BE into BE */
if (!dai_link->no_pcm) {
dai_link->no_pcm = 1;
if (dai_link->dpcm_playback)
dev_warn(card->dev,
"invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n",
dai_link->name);
if (dai_link->dpcm_capture)
dev_warn(card->dev,
"invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n",
dai_link->name);
/* convert normal link into DPCM one */
if (!(dai_link->dpcm_playback ||
dai_link->dpcm_capture)) {
dai_link->dpcm_playback = !dai_link->capture_only;
dai_link->dpcm_capture = !dai_link->playback_only;
}
}
/*
* override any BE fixups
* see
* snd_soc_link_be_hw_params_fixup()
*/
dai_link->be_hw_params_fixup =
component->driver->be_hw_params_fixup;
/*
* most BE links don't set stream name, so set it to
* dai link name if it's NULL to help bind widgets.
*/
if (!dai_link->stream_name)
dai_link->stream_name = dai_link->name;
}
/* Inform userspace we are using alternate topology */
if (component->driver->topology_name_prefix) {
/* topology shortname created? */
if (!card->topology_shortname_created) {
comp_drv = component->driver;
snprintf(card->topology_shortname, 32, "%s-%s",
comp_drv->topology_name_prefix,
card->name);
card->topology_shortname_created = true;
}
/* use topology shortname */
card->name = card->topology_shortname;
}
}
}
#define soc_setup_card_name(name, name1, name2, norm) \
__soc_setup_card_name(name, sizeof(name), name1, name2, norm)
static void __soc_setup_card_name(char *name, int len,
const char *name1, const char *name2,
int normalization)
{
int i;
snprintf(name, len, "%s", name1 ? name1 : name2);
if (!normalization)
return;
/*
* Name normalization
*
* The driver name is somewhat special, as it's used as a key for
* searches in the user-space.
*
* ex)
* "abcd??efg" -> "abcd__efg"
*/
for (i = 0; i < len; i++) {
switch (name[i]) {
case '_':
case '-':
case '\0':
break;
default:
if (!isalnum(name[i]))
name[i] = '_';
break;
}
}
}
static void soc_cleanup_card_resources(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd, *n;
ASoC: soc-core: remove soc_remove_dai_links() It is easy to read code if it is cleanly using paired function/naming, like start <-> stop, register <-> unregister, etc, etc. But, current ALSA SoC code is very random, unbalance, not paired, etc. It is easy to create bug at the such code, and it will be difficult to debug. soc_cleanup_card_resources() (a) which is paired function of snd_soc_instantiate_card() (A) is calling soc_remove_dai_links() (*) to remove card related resources, but it is breaking add/remove balance (B)(b)(C)(c)(D)(d), in other words these should be called from soc_cleanup_card_resources() (a) from balance point of view. More headacke is that it is using original removing method for dai_link even though we already have snd_soc_remove_dai_link() which is the function for it (d). This patch removes snd_soc_remove_dai_links() and balance up code. static void soc_remove_dai_links(...) { ... (b) soc_remove_link_dais(card); (c) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) { ... /* it should use snd_soc_remove_dai_link() here */ (d) list_del(&link->list); } } (a) static int soc_cleanup_card_resources(...) { ... /* remove and free each DAI */ (*) soc_remove_dai_links(card); ... } (A) static int snd_soc_instantiate_card(struct snd_soc_card *card) { ... /* add predefined DAI links to the list */ for_each_card_prelinks(card, i, dai_link) (B) snd_soc_add_dai_link(card, dai_link); ... /* probe all components used by DAI links on this card */ (C) ret = soc_probe_link_components(card); ... /* probe all DAI links on this card */ (D) ret = soc_probe_link_dais(card); ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/875zl7bu1r.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-10-02 12:22:40 +07:00
ASoC: soc-pcm: remove soc_pcm_private_free() soc-topology adds extra dai_link by using snd_soc_add_dai_link(), and removes it by snd_soc_romove_dai_link(). This snd_soc_add/remove_dai_link() and/or its related functions are unbalanced before, and now, these are balance-uped. But, it finds the random operation issue, and it is reported by Pierre-Louis. When card was released, topology will call snd_soc_remove_dai_link() via (A). static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; /* This should be called before snd_card_free() */ (A) soc_remove_link_components(card); /* free the ALSA card at first; this syncs with pending operations */ if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } /* remove and free each DAI */ (X) soc_remove_link_dais(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... } At (A), topology calls snd_soc_remove_dai_link(). Then topology rtd, and its related all data are freed. Next, (B) is called, and then, pcm->private_free = soc_pcm_private_free() is called. static void soc_pcm_private_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; /* need to sync the delayed work before releasing resources */ flush_delayed_work(&rtd->delayed_work); snd_soc_pcm_component_free(rtd); } Here, it gets rtd via pcm->private_data. But, topology related rtd are already freed at (A). Normal sound card has no damage, becase it frees rtd at (C). These are finalizing rtd related data. Thus, these should be called when rtd was freed, not sound card was freed. It is very natural and understandable. In other words, pcm->private_free = soc_pcm_private_free() is no longer needed. Extra issue is that there is zero chance to call soc_remove_dai() for topology related dai at (X). Because (A) removes rtd connection from card too, and, (X) is based on card connected rtd. This means, (X) need to be called before (C) (= for normal sound) and (A) (= for topology). Now, I want to focus this patch which is the reason why snd_card_free() = (B) is located there. commit 4efda5f2130da033aeedc5b3205569893b910de2 ("ASoC: Fix use-after-free at card unregistration") Original snd_card_free() was called last of this function. But moved to top to avoid use-after-free issue. The issue was happen at soc_pcm_free() which was pcm->private_free, today it is updated/renamed to soc_pcm_private_free(). In other words, (B) need to be called before (C) (= for normal sound) and (A) (= for topology), because it needs (not yet freed) rtd. But, (A) need to be called before (B), because it needs card->snd_card pointer. If we call flush_delayed_work() and snd_soc_pcm_component_free() (= same as soc_pcm_private_free()) when rtd was freed (= (C), (A)), there is no reason to call snd_card_free() at top of this function. It can be called end of this function, again. But, in such case, it will likely break unbind again, as Takashi-san reported. When unbind is performed in a busy state, the code may release still-in-use resources. At least we need to call snd_card_disconnect_sync() at the first place. The final code will be... static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; if (card->snd_card) (Z) snd_card_disconnect_sync(card->snd_card); (X) soc_remove_link_dais(card); (A) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } } To avoid release still-in-use resources, call snd_card_disconnect_sync() at (Z). (X) is needed for both non-topology and topology. topology removes rtd via (A), and non topology removes rtd via (C). snd_card_free() is no longer related to use-after-free issue. Thus, locating (B) is no problem. Fixes: df95a16d2a9626 ("ASoC: soc-core: fix RIP warning on card removal") Fixes: bc7a9091e5b927 ("ASoC: soc-core: add soc_unbind_dai_link()") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/87o8xax88g.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-11-18 08:51:11 +07:00
if (card->snd_card)
snd_card_disconnect_sync(card->snd_card);
snd_soc_dapm_shutdown(card);
/* remove and free each DAI */
ASoC: soc-core: remove soc_remove_dai_links() It is easy to read code if it is cleanly using paired function/naming, like start <-> stop, register <-> unregister, etc, etc. But, current ALSA SoC code is very random, unbalance, not paired, etc. It is easy to create bug at the such code, and it will be difficult to debug. soc_cleanup_card_resources() (a) which is paired function of snd_soc_instantiate_card() (A) is calling soc_remove_dai_links() (*) to remove card related resources, but it is breaking add/remove balance (B)(b)(C)(c)(D)(d), in other words these should be called from soc_cleanup_card_resources() (a) from balance point of view. More headacke is that it is using original removing method for dai_link even though we already have snd_soc_remove_dai_link() which is the function for it (d). This patch removes snd_soc_remove_dai_links() and balance up code. static void soc_remove_dai_links(...) { ... (b) soc_remove_link_dais(card); (c) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) { ... /* it should use snd_soc_remove_dai_link() here */ (d) list_del(&link->list); } } (a) static int soc_cleanup_card_resources(...) { ... /* remove and free each DAI */ (*) soc_remove_dai_links(card); ... } (A) static int snd_soc_instantiate_card(struct snd_soc_card *card) { ... /* add predefined DAI links to the list */ for_each_card_prelinks(card, i, dai_link) (B) snd_soc_add_dai_link(card, dai_link); ... /* probe all components used by DAI links on this card */ (C) ret = soc_probe_link_components(card); ... /* probe all DAI links on this card */ (D) ret = soc_probe_link_dais(card); ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/875zl7bu1r.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-10-02 12:22:40 +07:00
soc_remove_link_dais(card);
ASoC: soc-pcm: remove soc_pcm_private_free() soc-topology adds extra dai_link by using snd_soc_add_dai_link(), and removes it by snd_soc_romove_dai_link(). This snd_soc_add/remove_dai_link() and/or its related functions are unbalanced before, and now, these are balance-uped. But, it finds the random operation issue, and it is reported by Pierre-Louis. When card was released, topology will call snd_soc_remove_dai_link() via (A). static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; /* This should be called before snd_card_free() */ (A) soc_remove_link_components(card); /* free the ALSA card at first; this syncs with pending operations */ if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } /* remove and free each DAI */ (X) soc_remove_link_dais(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... } At (A), topology calls snd_soc_remove_dai_link(). Then topology rtd, and its related all data are freed. Next, (B) is called, and then, pcm->private_free = soc_pcm_private_free() is called. static void soc_pcm_private_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; /* need to sync the delayed work before releasing resources */ flush_delayed_work(&rtd->delayed_work); snd_soc_pcm_component_free(rtd); } Here, it gets rtd via pcm->private_data. But, topology related rtd are already freed at (A). Normal sound card has no damage, becase it frees rtd at (C). These are finalizing rtd related data. Thus, these should be called when rtd was freed, not sound card was freed. It is very natural and understandable. In other words, pcm->private_free = soc_pcm_private_free() is no longer needed. Extra issue is that there is zero chance to call soc_remove_dai() for topology related dai at (X). Because (A) removes rtd connection from card too, and, (X) is based on card connected rtd. This means, (X) need to be called before (C) (= for normal sound) and (A) (= for topology). Now, I want to focus this patch which is the reason why snd_card_free() = (B) is located there. commit 4efda5f2130da033aeedc5b3205569893b910de2 ("ASoC: Fix use-after-free at card unregistration") Original snd_card_free() was called last of this function. But moved to top to avoid use-after-free issue. The issue was happen at soc_pcm_free() which was pcm->private_free, today it is updated/renamed to soc_pcm_private_free(). In other words, (B) need to be called before (C) (= for normal sound) and (A) (= for topology), because it needs (not yet freed) rtd. But, (A) need to be called before (B), because it needs card->snd_card pointer. If we call flush_delayed_work() and snd_soc_pcm_component_free() (= same as soc_pcm_private_free()) when rtd was freed (= (C), (A)), there is no reason to call snd_card_free() at top of this function. It can be called end of this function, again. But, in such case, it will likely break unbind again, as Takashi-san reported. When unbind is performed in a busy state, the code may release still-in-use resources. At least we need to call snd_card_disconnect_sync() at the first place. The final code will be... static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; if (card->snd_card) (Z) snd_card_disconnect_sync(card->snd_card); (X) soc_remove_link_dais(card); (A) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } } To avoid release still-in-use resources, call snd_card_disconnect_sync() at (Z). (X) is needed for both non-topology and topology. topology removes rtd via (A), and non topology removes rtd via (C). snd_card_free() is no longer related to use-after-free issue. Thus, locating (B) is no problem. Fixes: df95a16d2a9626 ("ASoC: soc-core: fix RIP warning on card removal") Fixes: bc7a9091e5b927 ("ASoC: soc-core: add soc_unbind_dai_link()") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/87o8xax88g.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-11-18 08:51:11 +07:00
soc_remove_link_components(card);
ASoC: soc-core: remove soc_remove_dai_links() It is easy to read code if it is cleanly using paired function/naming, like start <-> stop, register <-> unregister, etc, etc. But, current ALSA SoC code is very random, unbalance, not paired, etc. It is easy to create bug at the such code, and it will be difficult to debug. soc_cleanup_card_resources() (a) which is paired function of snd_soc_instantiate_card() (A) is calling soc_remove_dai_links() (*) to remove card related resources, but it is breaking add/remove balance (B)(b)(C)(c)(D)(d), in other words these should be called from soc_cleanup_card_resources() (a) from balance point of view. More headacke is that it is using original removing method for dai_link even though we already have snd_soc_remove_dai_link() which is the function for it (d). This patch removes snd_soc_remove_dai_links() and balance up code. static void soc_remove_dai_links(...) { ... (b) soc_remove_link_dais(card); (c) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) { ... /* it should use snd_soc_remove_dai_link() here */ (d) list_del(&link->list); } } (a) static int soc_cleanup_card_resources(...) { ... /* remove and free each DAI */ (*) soc_remove_dai_links(card); ... } (A) static int snd_soc_instantiate_card(struct snd_soc_card *card) { ... /* add predefined DAI links to the list */ for_each_card_prelinks(card, i, dai_link) (B) snd_soc_add_dai_link(card, dai_link); ... /* probe all components used by DAI links on this card */ (C) ret = soc_probe_link_components(card); ... /* probe all DAI links on this card */ (D) ret = soc_probe_link_dais(card); ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/875zl7bu1r.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-10-02 12:22:40 +07:00
for_each_card_rtds_safe(card, rtd, n)
snd_soc_remove_pcm_runtime(card, rtd);
ASoC: soc-core: remove soc_remove_dai_links() It is easy to read code if it is cleanly using paired function/naming, like start <-> stop, register <-> unregister, etc, etc. But, current ALSA SoC code is very random, unbalance, not paired, etc. It is easy to create bug at the such code, and it will be difficult to debug. soc_cleanup_card_resources() (a) which is paired function of snd_soc_instantiate_card() (A) is calling soc_remove_dai_links() (*) to remove card related resources, but it is breaking add/remove balance (B)(b)(C)(c)(D)(d), in other words these should be called from soc_cleanup_card_resources() (a) from balance point of view. More headacke is that it is using original removing method for dai_link even though we already have snd_soc_remove_dai_link() which is the function for it (d). This patch removes snd_soc_remove_dai_links() and balance up code. static void soc_remove_dai_links(...) { ... (b) soc_remove_link_dais(card); (c) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) { ... /* it should use snd_soc_remove_dai_link() here */ (d) list_del(&link->list); } } (a) static int soc_cleanup_card_resources(...) { ... /* remove and free each DAI */ (*) soc_remove_dai_links(card); ... } (A) static int snd_soc_instantiate_card(struct snd_soc_card *card) { ... /* add predefined DAI links to the list */ for_each_card_prelinks(card, i, dai_link) (B) snd_soc_add_dai_link(card, dai_link); ... /* probe all components used by DAI links on this card */ (C) ret = soc_probe_link_components(card); ... /* probe all DAI links on this card */ (D) ret = soc_probe_link_dais(card); ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/875zl7bu1r.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-10-02 12:22:40 +07:00
/* remove auxiliary devices */
soc_remove_aux_devices(card);
soc_unbind_aux_dev(card);
snd_soc_dapm_free(&card->dapm);
soc_cleanup_card_debugfs(card);
/* remove the card */
snd_soc_card_remove(card);
ASoC: soc-pcm: remove soc_pcm_private_free() soc-topology adds extra dai_link by using snd_soc_add_dai_link(), and removes it by snd_soc_romove_dai_link(). This snd_soc_add/remove_dai_link() and/or its related functions are unbalanced before, and now, these are balance-uped. But, it finds the random operation issue, and it is reported by Pierre-Louis. When card was released, topology will call snd_soc_remove_dai_link() via (A). static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; /* This should be called before snd_card_free() */ (A) soc_remove_link_components(card); /* free the ALSA card at first; this syncs with pending operations */ if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } /* remove and free each DAI */ (X) soc_remove_link_dais(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... } At (A), topology calls snd_soc_remove_dai_link(). Then topology rtd, and its related all data are freed. Next, (B) is called, and then, pcm->private_free = soc_pcm_private_free() is called. static void soc_pcm_private_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; /* need to sync the delayed work before releasing resources */ flush_delayed_work(&rtd->delayed_work); snd_soc_pcm_component_free(rtd); } Here, it gets rtd via pcm->private_data. But, topology related rtd are already freed at (A). Normal sound card has no damage, becase it frees rtd at (C). These are finalizing rtd related data. Thus, these should be called when rtd was freed, not sound card was freed. It is very natural and understandable. In other words, pcm->private_free = soc_pcm_private_free() is no longer needed. Extra issue is that there is zero chance to call soc_remove_dai() for topology related dai at (X). Because (A) removes rtd connection from card too, and, (X) is based on card connected rtd. This means, (X) need to be called before (C) (= for normal sound) and (A) (= for topology). Now, I want to focus this patch which is the reason why snd_card_free() = (B) is located there. commit 4efda5f2130da033aeedc5b3205569893b910de2 ("ASoC: Fix use-after-free at card unregistration") Original snd_card_free() was called last of this function. But moved to top to avoid use-after-free issue. The issue was happen at soc_pcm_free() which was pcm->private_free, today it is updated/renamed to soc_pcm_private_free(). In other words, (B) need to be called before (C) (= for normal sound) and (A) (= for topology), because it needs (not yet freed) rtd. But, (A) need to be called before (B), because it needs card->snd_card pointer. If we call flush_delayed_work() and snd_soc_pcm_component_free() (= same as soc_pcm_private_free()) when rtd was freed (= (C), (A)), there is no reason to call snd_card_free() at top of this function. It can be called end of this function, again. But, in such case, it will likely break unbind again, as Takashi-san reported. When unbind is performed in a busy state, the code may release still-in-use resources. At least we need to call snd_card_disconnect_sync() at the first place. The final code will be... static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; if (card->snd_card) (Z) snd_card_disconnect_sync(card->snd_card); (X) soc_remove_link_dais(card); (A) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } } To avoid release still-in-use resources, call snd_card_disconnect_sync() at (Z). (X) is needed for both non-topology and topology. topology removes rtd via (A), and non topology removes rtd via (C). snd_card_free() is no longer related to use-after-free issue. Thus, locating (B) is no problem. Fixes: df95a16d2a9626 ("ASoC: soc-core: fix RIP warning on card removal") Fixes: bc7a9091e5b927 ("ASoC: soc-core: add soc_unbind_dai_link()") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/87o8xax88g.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
2019-11-18 08:51:11 +07:00
if (card->snd_card) {
snd_card_free(card->snd_card);
card->snd_card = NULL;
}
}
static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
{
if (card->instantiated) {
card->instantiated = false;
snd_soc_flush_all_delayed_work(card);
soc_cleanup_card_resources(card);
if (!unregister)
list_add(&card->list, &unbind_card_list);
} else {
if (unregister)
list_del(&card->list);
}
}
static int snd_soc_bind_card(struct snd_soc_card *card)
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
{
2015-11-18 14:34:11 +07:00
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_component *component;
struct snd_soc_dai_link *dai_link;
int ret, i;
mutex_lock(&client_mutex);
mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
snd_soc_dapm_init(&card->dapm, card, NULL);
/* check whether any platform is ignore machine FE and using topology */
soc_check_tplg_fes(card);
/* bind aux_devs too */
ret = soc_bind_aux_dev(card);
if (ret < 0)
goto probe_end;
/* add predefined DAI links to the list */
card->num_rtd = 0;
for_each_card_prelinks(card, i, dai_link) {
ret = snd_soc_add_pcm_runtime(card, dai_link);
if (ret < 0)
goto probe_end;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
/* card bind complete so register a sound card */
ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
card->owner, 0, &card->snd_card);
if (ret < 0) {
dev_err(card->dev,
"ASoC: can't create sound card for card %s: %d\n",
card->name, ret);
goto probe_end;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
}
soc_init_card_debugfs(card);
soc_resume_init(card);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
ret = snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets,
card->num_dapm_widgets);
if (ret < 0)
goto probe_end;
ret = snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets,
card->num_of_dapm_widgets);
if (ret < 0)
goto probe_end;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
/* initialise the sound card only once */
ret = snd_soc_card_probe(card);
if (ret < 0)
goto probe_end;
/* probe all components used by DAI links on this card */
ret = soc_probe_link_components(card);
if (ret < 0) {
dev_err(card->dev,
"ASoC: failed to instantiate card %d\n", ret);
goto probe_end;
}
/* probe auxiliary components */
ret = soc_probe_aux_devices(card);
if (ret < 0) {
dev_err(card->dev,
"ASoC: failed to probe aux component %d\n", ret);
goto probe_end;
}
/* probe all DAI links on this card */
ret = soc_probe_link_dais(card);
if (ret < 0) {
dev_err(card->dev,
"ASoC: failed to instantiate card %d\n", ret);
goto probe_end;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
}
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
for_each_card_rtds(card, rtd) {
ret = soc_init_pcm_runtime(card, rtd);
if (ret < 0)
goto probe_end;
}
ASoC: dapm: Implement and instantiate DAI widgets In order to allow us to do smarter things with DAI links create DAPM widgets which directly represent the DAIs in the DAPM graph. These are automatically created from the DAIs as we probe the card with references held in both directions between the widget and the DAI. The widgets are not made available for direct instantiation by drivers, they are created automatically from the DAIs. Drivers should be updated to create stream routes using DAPM maps rather than by annotating AIF and DAC widgets with streams. In order to ease transition to this model from existing drivers we automatically create DAPM routes between the DAI widgets and the existing stream widgets which are started and stopped by the DAI widgets, though the old stream handling mechanism is still in place. This also has the nice effect of removing non-DAPM devices as any device with a DAI acquires a widget automatically which will allow future simplifications to the core DAPM logic. The intention is that in future the AIF and DAI widgets will gain the ability to interact such that we are able to manage activity on individual channels independantly rather than powering up and down the entire AIF as we do currently. Currently we only generate these for CODECs, mostly as I have no systems with non-CODEC DAPM to integrate with. It should be a simple matter of programming to add the additional hookup for these. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-02-17 10:37:51 +07:00
snd_soc_dapm_link_dai_widgets(card);
snd_soc_dapm_connect_dai_link_widgets(card);
ASoC: dapm: Implement and instantiate DAI widgets In order to allow us to do smarter things with DAI links create DAPM widgets which directly represent the DAIs in the DAPM graph. These are automatically created from the DAIs as we probe the card with references held in both directions between the widget and the DAI. The widgets are not made available for direct instantiation by drivers, they are created automatically from the DAIs. Drivers should be updated to create stream routes using DAPM maps rather than by annotating AIF and DAC widgets with streams. In order to ease transition to this model from existing drivers we automatically create DAPM routes between the DAI widgets and the existing stream widgets which are started and stopped by the DAI widgets, though the old stream handling mechanism is still in place. This also has the nice effect of removing non-DAPM devices as any device with a DAI acquires a widget automatically which will allow future simplifications to the core DAPM logic. The intention is that in future the AIF and DAI widgets will gain the ability to interact such that we are able to manage activity on individual channels independantly rather than powering up and down the entire AIF as we do currently. Currently we only generate these for CODECs, mostly as I have no systems with non-CODEC DAPM to integrate with. It should be a simple matter of programming to add the additional hookup for these. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-02-17 10:37:51 +07:00
ret = snd_soc_add_card_controls(card, card->controls,
card->num_controls);
if (ret < 0)
goto probe_end;
ret = snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
card->num_dapm_routes);
if (ret < 0) {
if (card->disable_route_checks) {
dev_info(card->dev,
"%s: disable_route_checks set, ignoring errors on add_routes\n",
__func__);
} else {
dev_err(card->dev,
"%s: snd_soc_dapm_add_routes failed: %d\n",
__func__, ret);
goto probe_end;
}
}
ret = snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes,
card->num_of_dapm_routes);
if (ret < 0)
goto probe_end;
/* try to set some sane longname if DMI is available */
snd_soc_set_dmi_name(card, NULL);
soc_setup_card_name(card->snd_card->shortname,
card->name, NULL, 0);
soc_setup_card_name(card->snd_card->longname,
card->long_name, card->name, 0);
soc_setup_card_name(card->snd_card->driver,
card->driver_name, card->name, 1);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
if (card->components) {
/* the current implementation of snd_component_add() accepts */
/* multiple components in the string separated by space, */
/* but the string collision (identical string) check might */
/* not work correctly */
ret = snd_component_add(card->snd_card, card->components);
if (ret < 0) {
dev_err(card->dev, "ASoC: %s snd_component_add() failed: %d\n",
card->name, ret);
goto probe_end;
}
}
ret = snd_soc_card_late_probe(card);
if (ret < 0)
goto probe_end;
snd_soc_dapm_new_widgets(card);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
ret = snd_card_register(card->snd_card);
if (ret < 0) {
dev_err(card->dev, "ASoC: failed to register soundcard %d\n",
ret);
goto probe_end;
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
card->instantiated = 1;
dapm_mark_endpoints_dirty(card);
snd_soc_dapm_sync(&card->dapm);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
/* deactivate pins to sleep state */
for_each_card_components(card, component)
if (!snd_soc_component_active(component))
pinctrl_pm_select_sleep_state(component->dev);
probe_end:
if (ret < 0)
soc_cleanup_card_resources(card);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
mutex_unlock(&card->mutex);
mutex_unlock(&client_mutex);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
return ret;
}
/* probes a new socdev */
static int soc_probe(struct platform_device *pdev)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
struct snd_soc_card *card = platform_get_drvdata(pdev);
/*
* no card, so machine driver should be registering card
* we should not be here in that case so ret error
*/
if (!card)
return -EINVAL;
dev_warn(&pdev->dev,
"ASoC: machine %s should use snd_soc_register_card()\n",
card->name);
/* Bodge while we unpick instantiation */
card->dev = &pdev->dev;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
return snd_soc_register_card(card);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
}
/* removes a socdev */
static int soc_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
snd_soc_unregister_card(card);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
return 0;
}
int snd_soc_poweroff(struct device *dev)
{
struct snd_soc_card *card = dev_get_drvdata(dev);
struct snd_soc_component *component;
if (!card->instantiated)
return 0;
/*
* Flush out pmdown_time work - we actually do want to run it
* now, we're shutting down so no imminent restart.
*/
snd_soc_flush_all_delayed_work(card);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
snd_soc_dapm_shutdown(card);
/* deactivate pins to sleep state */
for_each_card_components(card, component)
pinctrl_pm_select_sleep_state(component->dev);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_poweroff);
const struct dev_pm_ops snd_soc_pm_ops = {
.suspend = snd_soc_suspend,
.resume = snd_soc_resume,
.freeze = snd_soc_suspend,
.thaw = snd_soc_resume,
.poweroff = snd_soc_poweroff,
.restore = snd_soc_resume,
};
EXPORT_SYMBOL_GPL(snd_soc_pm_ops);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
/* ASoC platform driver */
static struct platform_driver soc_driver = {
.driver = {
.name = "soc-audio",
.pm = &snd_soc_pm_ops,
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
},
.probe = soc_probe,
.remove = soc_remove,
};
/**
* snd_soc_cnew - create new control
* @_template: control template
* @data: control private data
* @long_name: control long name
* @prefix: control name prefix
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
*
* Create a new mixer control from a template control.
*
* Returns 0 for success, else error.
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
void *data, const char *long_name,
const char *prefix)
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
{
struct snd_kcontrol_new template;
struct snd_kcontrol *kcontrol;
char *name = NULL;
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
memcpy(&template, _template, sizeof(template));
template.index = 0;
if (!long_name)
long_name = template.name;
if (prefix) {
name = kasprintf(GFP_KERNEL, "%s %s", prefix, long_name);
if (!name)
return NULL;
template.name = name;
} else {
template.name = long_name;
}
kcontrol = snd_ctl_new1(&template, data);
kfree(name);
return kcontrol;
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
}
EXPORT_SYMBOL_GPL(snd_soc_cnew);
static int snd_soc_add_controls(struct snd_card *card, struct device *dev,
const struct snd_kcontrol_new *controls, int num_controls,
const char *prefix, void *data)
{
int err, i;
for (i = 0; i < num_controls; i++) {
const struct snd_kcontrol_new *control = &controls[i];
err = snd_ctl_add(card, snd_soc_cnew(control, data,
control->name, prefix));
if (err < 0) {
dev_err(dev, "ASoC: Failed to add %s: %d\n",
control->name, err);
return err;
}
}
return 0;
}
/**
* snd_soc_add_component_controls - Add an array of controls to a component.
*
* @component: Component to add controls to
* @controls: Array of controls to add
* @num_controls: Number of elements in the array
*
* Return: 0 for success, else error.
*/
int snd_soc_add_component_controls(struct snd_soc_component *component,
const struct snd_kcontrol_new *controls, unsigned int num_controls)
{
struct snd_card *card = component->card->snd_card;
return snd_soc_add_controls(card, component->dev, controls,
num_controls, component->name_prefix, component);
}
EXPORT_SYMBOL_GPL(snd_soc_add_component_controls);
/**
* snd_soc_add_card_controls - add an array of controls to a SoC card.
* Convenience function to add a list of controls.
*
* @soc_card: SoC card to add controls to
* @controls: array of controls to add
* @num_controls: number of elements in the array
*
* Return 0 for success, else error.
*/
int snd_soc_add_card_controls(struct snd_soc_card *soc_card,
const struct snd_kcontrol_new *controls, int num_controls)
{
struct snd_card *card = soc_card->snd_card;
return snd_soc_add_controls(card, soc_card->dev, controls, num_controls,
NULL, soc_card);
}
EXPORT_SYMBOL_GPL(snd_soc_add_card_controls);
/**
* snd_soc_add_dai_controls - add an array of controls to a DAI.
* Convienience function to add a list of controls.
*
* @dai: DAI to add controls to
* @controls: array of controls to add
* @num_controls: number of elements in the array
*
* Return 0 for success, else error.
*/
int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
const struct snd_kcontrol_new *controls, int num_controls)
{
struct snd_card *card = dai->component->card->snd_card;
return snd_soc_add_controls(card, dai->dev, controls, num_controls,
NULL, dai);
}
EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls);
/**
* snd_soc_register_card - Register a card with the ASoC core
*
* @card: Card to register
*
*/
int snd_soc_register_card(struct snd_soc_card *card)
{
if (!card->name || !card->dev)
return -EINVAL;
dev_set_drvdata(card->dev, card);
INIT_LIST_HEAD(&card->widgets);
INIT_LIST_HEAD(&card->paths);
INIT_LIST_HEAD(&card->dapm_list);
INIT_LIST_HEAD(&card->aux_comp_list);
INIT_LIST_HEAD(&card->component_dev_list);
INIT_LIST_HEAD(&card->list);
2015-11-18 14:34:11 +07:00
INIT_LIST_HEAD(&card->rtd_list);
INIT_LIST_HEAD(&card->dapm_dirty);
INIT_LIST_HEAD(&card->dobj_list);
card->instantiated = 0;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
mutex_init(&card->mutex);
mutex_init(&card->dapm_mutex);
mutex_init(&card->pcm_mutex);
ASoC: dpcm: prevent snd_soc_dpcm use after free The dpcm get from fe_clients/be_clients may be free before use Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Use irq spin lock version, since the lock may be used in interrupts. possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-08 12:05:53 +07:00
spin_lock_init(&card->dpcm_lock);
return snd_soc_bind_card(card);
}
EXPORT_SYMBOL_GPL(snd_soc_register_card);
/**
* snd_soc_unregister_card - Unregister a card with the ASoC core
*
* @card: Card to unregister
*
*/
int snd_soc_unregister_card(struct snd_soc_card *card)
{
mutex_lock(&client_mutex);
snd_soc_unbind_card(card, true);
mutex_unlock(&client_mutex);
dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_card);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
/*
* Simplify DAI link configuration by removing ".-1" from device names
* and sanitizing names.
*/
static char *fmt_single_name(struct device *dev, int *id)
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
{
char *found, name[NAME_SIZE];
int id1, id2;
if (dev_name(dev) == NULL)
return NULL;
strlcpy(name, dev_name(dev), NAME_SIZE);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
/* are we a "%s.%d" name (platform and SPI components) */
found = strstr(name, dev->driver->name);
if (found) {
/* get ID */
if (sscanf(&found[strlen(dev->driver->name)], ".%d", id) == 1) {
/* discard ID from name if ID == -1 */
if (*id == -1)
found[strlen(dev->driver->name)] = '\0';
}
} else {
/* I2C component devices are named "bus-addr" */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
char tmp[NAME_SIZE];
/* create unique ID number from I2C addr and bus */
*id = ((id1 & 0xffff) << 16) + id2;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
/* sanitize component name for DAI link creation */
snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name,
name);
strlcpy(name, tmp, NAME_SIZE);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
} else
*id = 0;
}
return devm_kstrdup(dev, name, GFP_KERNEL);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
}
/*
* Simplify DAI link naming for single devices with multiple DAIs by removing
* any ".-1" and using the DAI name (instead of device name).
*/
static inline char *fmt_multiple_name(struct device *dev,
struct snd_soc_dai_driver *dai_drv)
{
if (dai_drv->name == NULL) {
dev_err(dev,
"ASoC: error - multiple DAI %s registered with no name\n",
dev_name(dev));
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
return NULL;
}
return devm_kstrdup(dev, dai_drv->name, GFP_KERNEL);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
}
void snd_soc_unregister_dai(struct snd_soc_dai *dai)
{
dev_dbg(dai->dev, "ASoC: Unregistered DAI '%s'\n", dai->name);
list_del(&dai->list);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
/**
* snd_soc_register_dai - Register a DAI dynamically & create its widgets
*
* @component: The component the DAIs are registered for
* @dai_drv: DAI driver to use for the DAI
* @legacy_dai_naming: if %true, use legacy single-name format;
* if %false, use multiple-name format;
*
* Topology can use this API to register DAIs when probing a component.
* These DAIs's widgets will be freed in the card cleanup and the DAIs
* will be freed in the component cleanup.
*/
struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component,
struct snd_soc_dai_driver *dai_drv,
bool legacy_dai_naming)
{
struct device *dev = component->dev;
struct snd_soc_dai *dai;
dev_dbg(dev, "ASoC: dynamically register DAI %s\n", dev_name(dev));
lockdep_assert_held(&client_mutex);
dai = devm_kzalloc(dev, sizeof(*dai), GFP_KERNEL);
if (dai == NULL)
return NULL;
/*
* Back in the old days when we still had component-less DAIs,
* instead of having a static name, component-less DAIs would
* inherit the name of the parent device so it is possible to
* register multiple instances of the DAI. We still need to keep
* the same naming style even though those DAIs are not
* component-less anymore.
*/
if (legacy_dai_naming &&
(dai_drv->id == 0 || dai_drv->name == NULL)) {
dai->name = fmt_single_name(dev, &dai->id);
} else {
dai->name = fmt_multiple_name(dev, dai_drv);
if (dai_drv->id)
dai->id = dai_drv->id;
else
dai->id = component->num_dai;
}
if (!dai->name)
return NULL;
dai->component = component;
dai->dev = dev;
dai->driver = dai_drv;
/* see for_each_component_dais */
list_add_tail(&dai->list, &component->dai_list);
component->num_dai++;
dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name);
return dai;
}
/**
* snd_soc_unregister_dai - Unregister DAIs from the ASoC core
*
* @component: The component for which the DAIs should be unregistered
*/
static void snd_soc_unregister_dais(struct snd_soc_component *component)
{
struct snd_soc_dai *dai, *_dai;
for_each_component_dais_safe(component, dai, _dai)
snd_soc_unregister_dai(dai);
}
/**
* snd_soc_register_dais - Register a DAI with the ASoC core
*
* @component: The component the DAIs are registered for
* @dai_drv: DAI driver to use for the DAIs
* @count: Number of DAIs
*/
static int snd_soc_register_dais(struct snd_soc_component *component,
struct snd_soc_dai_driver *dai_drv,
size_t count)
{
struct snd_soc_dai *dai;
unsigned int i;
int ret;
for (i = 0; i < count; i++) {
dai = snd_soc_register_dai(component, dai_drv + i, count == 1 &&
!component->driver->non_legacy_dai_naming);
if (dai == NULL) {
ret = -ENOMEM;
goto err;
}
}
return 0;
err:
snd_soc_unregister_dais(component);
return ret;
}
#define ENDIANNESS_MAP(name) \
(SNDRV_PCM_FMTBIT_##name##LE | SNDRV_PCM_FMTBIT_##name##BE)
static u64 endianness_format_map[] = {
ENDIANNESS_MAP(S16_),
ENDIANNESS_MAP(U16_),
ENDIANNESS_MAP(S24_),
ENDIANNESS_MAP(U24_),
ENDIANNESS_MAP(S32_),
ENDIANNESS_MAP(U32_),
ENDIANNESS_MAP(S24_3),
ENDIANNESS_MAP(U24_3),
ENDIANNESS_MAP(S20_3),
ENDIANNESS_MAP(U20_3),
ENDIANNESS_MAP(S18_3),
ENDIANNESS_MAP(U18_3),
ENDIANNESS_MAP(FLOAT_),
ENDIANNESS_MAP(FLOAT64_),
ENDIANNESS_MAP(IEC958_SUBFRAME_),
};
/*
* Fix up the DAI formats for endianness: codecs don't actually see
* the endianness of the data but we're using the CPU format
* definitions which do need to include endianness so we ensure that
* codec DAIs always have both big and little endian variants set.
*/
static void convert_endianness_formats(struct snd_soc_pcm_stream *stream)
{
int i;
for (i = 0; i < ARRAY_SIZE(endianness_format_map); i++)
if (stream->formats & endianness_format_map[i])
stream->formats |= endianness_format_map[i];
}
static void snd_soc_try_rebind_card(void)
{
struct snd_soc_card *card, *c;
list_for_each_entry_safe(card, c, &unbind_card_list, list)
if (!snd_soc_bind_card(card))
list_del(&card->list);
}
static void snd_soc_del_component_unlocked(struct snd_soc_component *component)
{
struct snd_soc_card *card = component->card;
snd_soc_unregister_dais(component);
if (card)
snd_soc_unbind_card(card, false);
list_del(&component->list);
}
int snd_soc_add_component(struct device *dev,
struct snd_soc_component *component,
const struct snd_soc_component_driver *component_driver,
struct snd_soc_dai_driver *dai_drv,
int num_dai)
{
const char *name = fmt_single_name(dev, &component->id);
int ret;
int i;
if (!name) {
dev_err(dev, "ASoC: Failed to allocate name\n");
return -ENOMEM;
}
mutex_lock(&client_mutex);
ret = snd_soc_component_initialize(component, component_driver,
dev, name);
if (ret)
goto err_free;
if (component_driver->endianness) {
for (i = 0; i < num_dai; i++) {
convert_endianness_formats(&dai_drv[i].playback);
convert_endianness_formats(&dai_drv[i].capture);
}
}
ret = snd_soc_register_dais(component, dai_drv, num_dai);
if (ret < 0) {
dev_err(dev, "ASoC: Failed to register DAIs: %d\n", ret);
goto err_cleanup;
}
if (!component->driver->write && !component->driver->read) {
if (!component->regmap)
component->regmap = dev_get_regmap(component->dev,
NULL);
if (component->regmap)
snd_soc_component_setup_regmap(component);
}
/* see for_each_component */
list_add(&component->list, &component_list);
err_cleanup:
if (ret < 0)
snd_soc_del_component_unlocked(component);
err_free:
mutex_unlock(&client_mutex);
if (ret == 0)
snd_soc_try_rebind_card();
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_add_component);
int snd_soc_register_component(struct device *dev,
const struct snd_soc_component_driver *component_driver,
struct snd_soc_dai_driver *dai_drv,
int num_dai)
{
struct snd_soc_component *component;
component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL);
if (!component)
return -ENOMEM;
return snd_soc_add_component(dev, component, component_driver,
dai_drv, num_dai);
}
EXPORT_SYMBOL_GPL(snd_soc_register_component);
/**
* snd_soc_unregister_component_by_driver - Unregister component using a given driver
* from the ASoC core
*
* @dev: The device to unregister
* @component_driver: The component driver to unregister
*/
void snd_soc_unregister_component_by_driver(struct device *dev,
const struct snd_soc_component_driver *component_driver)
{
struct snd_soc_component *component;
if (!component_driver)
return;
mutex_lock(&client_mutex);
component = snd_soc_lookup_component_nolocked(dev, component_driver->name);
if (!component)
goto out;
snd_soc_del_component_unlocked(component);
out:
mutex_unlock(&client_mutex);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_component_by_driver);
/**
* snd_soc_unregister_component - Unregister all related component
* from the ASoC core
*
* @dev: The device to unregister
*/
void snd_soc_unregister_component(struct device *dev)
{
struct snd_soc_component *component;
mutex_lock(&client_mutex);
while (1) {
component = snd_soc_lookup_component_nolocked(dev, NULL);
if (!component)
break;
snd_soc_del_component_unlocked(component);
}
mutex_unlock(&client_mutex);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
/* Retrieve a card's name from device tree */
int snd_soc_of_parse_card_name(struct snd_soc_card *card,
const char *propname)
{
struct device_node *np;
int ret;
if (!card->dev) {
pr_err("card->dev is not set before calling %s\n", __func__);
return -EINVAL;
}
np = card->dev->of_node;
ret = of_property_read_string_index(np, propname, 0, &card->name);
/*
* EINVAL means the property does not exist. This is fine providing
* card->name was previously set, which is checked later in
* snd_soc_register_card.
*/
if (ret < 0 && ret != -EINVAL) {
dev_err(card->dev,
"ASoC: Property '%s' could not be read: %d\n",
propname, ret);
return ret;
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name);
static const struct snd_soc_dapm_widget simple_widgets[] = {
SND_SOC_DAPM_MIC("Microphone", NULL),
SND_SOC_DAPM_LINE("Line", NULL),
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_SPK("Speaker", NULL),
};
int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
const char *propname)
{
struct device_node *np = card->dev->of_node;
struct snd_soc_dapm_widget *widgets;
const char *template, *wname;
int i, j, num_widgets, ret;
num_widgets = of_property_count_strings(np, propname);
if (num_widgets < 0) {
dev_err(card->dev,
"ASoC: Property '%s' does not exist\n", propname);
return -EINVAL;
}
if (num_widgets & 1) {
dev_err(card->dev,
"ASoC: Property '%s' length is not even\n", propname);
return -EINVAL;
}
num_widgets /= 2;
if (!num_widgets) {
dev_err(card->dev, "ASoC: Property '%s's length is zero\n",
propname);
return -EINVAL;
}
widgets = devm_kcalloc(card->dev, num_widgets, sizeof(*widgets),
GFP_KERNEL);
if (!widgets) {
dev_err(card->dev,
"ASoC: Could not allocate memory for widgets\n");
return -ENOMEM;
}
for (i = 0; i < num_widgets; i++) {
ret = of_property_read_string_index(np, propname,
2 * i, &template);
if (ret) {
dev_err(card->dev,
"ASoC: Property '%s' index %d read error:%d\n",
propname, 2 * i, ret);
return -EINVAL;
}
for (j = 0; j < ARRAY_SIZE(simple_widgets); j++) {
if (!strncmp(template, simple_widgets[j].name,
strlen(simple_widgets[j].name))) {
widgets[i] = simple_widgets[j];
break;
}
}
if (j >= ARRAY_SIZE(simple_widgets)) {
dev_err(card->dev,
"ASoC: DAPM widget '%s' is not supported\n",
template);
return -EINVAL;
}
ret = of_property_read_string_index(np, propname,
(2 * i) + 1,
&wname);
if (ret) {
dev_err(card->dev,
"ASoC: Property '%s' index %d read error:%d\n",
propname, (2 * i) + 1, ret);
return -EINVAL;
}
widgets[i].name = wname;
}
card->of_dapm_widgets = widgets;
card->num_of_dapm_widgets = num_widgets;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
int snd_soc_of_get_slot_mask(struct device_node *np,
const char *prop_name,
unsigned int *mask)
{
u32 val;
const __be32 *of_slot_mask = of_get_property(np, prop_name, &val);
int i;
if (!of_slot_mask)
return 0;
val /= sizeof(u32);
for (i = 0; i < val; i++)
if (be32_to_cpup(&of_slot_mask[i]))
*mask |= (1 << i);
return val;
}
EXPORT_SYMBOL_GPL(snd_soc_of_get_slot_mask);
int snd_soc_of_parse_tdm_slot(struct device_node *np,
unsigned int *tx_mask,
unsigned int *rx_mask,
unsigned int *slots,
unsigned int *slot_width)
{
u32 val;
int ret;
if (tx_mask)
snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", tx_mask);
if (rx_mask)
snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", rx_mask);
if (of_property_read_bool(np, "dai-tdm-slot-num")) {
ret = of_property_read_u32(np, "dai-tdm-slot-num", &val);
if (ret)
return ret;
if (slots)
*slots = val;
}
if (of_property_read_bool(np, "dai-tdm-slot-width")) {
ret = of_property_read_u32(np, "dai-tdm-slot-width", &val);
if (ret)
return ret;
if (slot_width)
*slot_width = val;
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_tdm_slot);
void snd_soc_of_parse_node_prefix(struct device_node *np,
struct snd_soc_codec_conf *codec_conf,
struct device_node *of_node,
const char *propname)
{
const char *str;
int ret;
ret = of_property_read_string(np, propname, &str);
if (ret < 0) {
/* no prefix is not error */
return;
}
codec_conf->dlc.of_node = of_node;
codec_conf->name_prefix = str;
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_node_prefix);
int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
const char *propname)
{
struct device_node *np = card->dev->of_node;
int num_routes;
struct snd_soc_dapm_route *routes;
int i, ret;
num_routes = of_property_count_strings(np, propname);
if (num_routes < 0 || num_routes & 1) {
dev_err(card->dev,
"ASoC: Property '%s' does not exist or its length is not even\n",
propname);
return -EINVAL;
}
num_routes /= 2;
if (!num_routes) {
dev_err(card->dev, "ASoC: Property '%s's length is zero\n",
propname);
return -EINVAL;
}
treewide: devm_kzalloc() -> devm_kcalloc() The devm_kzalloc() function has a 2-factor argument form, devm_kcalloc(). This patch replaces cases of: devm_kzalloc(handle, a * b, gfp) with: devm_kcalloc(handle, a * b, gfp) as well as handling cases of: devm_kzalloc(handle, a * b * c, gfp) with: devm_kzalloc(handle, array3_size(a, b, c), gfp) as it's slightly less ugly than: devm_kcalloc(handle, array_size(a, b), c, gfp) This does, however, attempt to ignore constant size factors like: devm_kzalloc(handle, 4 * 1024, gfp) though any constants defined via macros get caught up in the conversion. Any factors with a sizeof() of "unsigned char", "char", and "u8" were dropped, since they're redundant. Some manual whitespace fixes were needed in this patch, as Coccinelle really liked to write "=devm_kcalloc..." instead of "= devm_kcalloc...". The Coccinelle script used for this was: // Fix redundant parens around sizeof(). @@ expression HANDLE; type TYPE; expression THING, E; @@ ( devm_kzalloc(HANDLE, - (sizeof(TYPE)) * E + sizeof(TYPE) * E , ...) | devm_kzalloc(HANDLE, - (sizeof(THING)) * E + sizeof(THING) * E , ...) ) // Drop single-byte sizes and redundant parens. @@ expression HANDLE; expression COUNT; typedef u8; typedef __u8; @@ ( devm_kzalloc(HANDLE, - sizeof(u8) * (COUNT) + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(__u8) * (COUNT) + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(char) * (COUNT) + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(unsigned char) * (COUNT) + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(u8) * COUNT + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(__u8) * COUNT + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(char) * COUNT + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(unsigned char) * COUNT + COUNT , ...) ) // 2-factor product with sizeof(type/expression) and identifier or constant. @@ expression HANDLE; type TYPE; expression THING; identifier COUNT_ID; constant COUNT_CONST; @@ ( - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * (COUNT_ID) + COUNT_ID, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * COUNT_ID + COUNT_ID, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * (COUNT_CONST) + COUNT_CONST, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * COUNT_CONST + COUNT_CONST, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * (COUNT_ID) + COUNT_ID, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * COUNT_ID + COUNT_ID, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * (COUNT_CONST) + COUNT_CONST, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * COUNT_CONST + COUNT_CONST, sizeof(THING) , ...) ) // 2-factor product, only identifiers. @@ expression HANDLE; identifier SIZE, COUNT; @@ - devm_kzalloc + devm_kcalloc (HANDLE, - SIZE * COUNT + COUNT, SIZE , ...) // 3-factor product with 1 sizeof(type) or sizeof(expression), with // redundant parens removed. @@ expression HANDLE; expression THING; identifier STRIDE, COUNT; type TYPE; @@ ( devm_kzalloc(HANDLE, - sizeof(TYPE) * (COUNT) * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE) * (COUNT) * STRIDE + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE) * COUNT * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE) * COUNT * STRIDE + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING) * (COUNT) * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING) * (COUNT) * STRIDE + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING) * COUNT * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING) * COUNT * STRIDE + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) ) // 3-factor product with 2 sizeof(variable), with redundant parens removed. @@ expression HANDLE; expression THING1, THING2; identifier COUNT; type TYPE1, TYPE2; @@ ( devm_kzalloc(HANDLE, - sizeof(TYPE1) * sizeof(TYPE2) * COUNT + array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING1) * sizeof(THING2) * COUNT + array3_size(COUNT, sizeof(THING1), sizeof(THING2)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(THING1), sizeof(THING2)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE1) * sizeof(THING2) * COUNT + array3_size(COUNT, sizeof(TYPE1), sizeof(THING2)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(TYPE1), sizeof(THING2)) , ...) ) // 3-factor product, only identifiers, with redundant parens removed. @@ expression HANDLE; identifier STRIDE, SIZE, COUNT; @@ ( devm_kzalloc(HANDLE, - (COUNT) * STRIDE * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - COUNT * (STRIDE) * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - COUNT * STRIDE * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - (COUNT) * (STRIDE) * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - COUNT * (STRIDE) * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - (COUNT) * STRIDE * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - (COUNT) * (STRIDE) * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - COUNT * STRIDE * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) ) // Any remaining multi-factor products, first at least 3-factor products, // when they're not all constants... @@ expression HANDLE; expression E1, E2, E3; constant C1, C2, C3; @@ ( devm_kzalloc(HANDLE, C1 * C2 * C3, ...) | devm_kzalloc(HANDLE, - (E1) * E2 * E3 + array3_size(E1, E2, E3) , ...) | devm_kzalloc(HANDLE, - (E1) * (E2) * E3 + array3_size(E1, E2, E3) , ...) | devm_kzalloc(HANDLE, - (E1) * (E2) * (E3) + array3_size(E1, E2, E3) , ...) | devm_kzalloc(HANDLE, - E1 * E2 * E3 + array3_size(E1, E2, E3) , ...) ) // And then all remaining 2 factors products when they're not all constants, // keeping sizeof() as the second factor argument. @@ expression HANDLE; expression THING, E1, E2; type TYPE; constant C1, C2, C3; @@ ( devm_kzalloc(HANDLE, sizeof(THING) * C2, ...) | devm_kzalloc(HANDLE, sizeof(TYPE) * C2, ...) | devm_kzalloc(HANDLE, C1 * C2 * C3, ...) | devm_kzalloc(HANDLE, C1 * C2, ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * (E2) + E2, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * E2 + E2, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * (E2) + E2, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * E2 + E2, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - (E1) * E2 + E1, E2 , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - (E1) * (E2) + E1, E2 , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - E1 * E2 + E1, E2 , ...) ) Signed-off-by: Kees Cook <keescook@chromium.org>
2018-06-13 04:07:58 +07:00
routes = devm_kcalloc(card->dev, num_routes, sizeof(*routes),
GFP_KERNEL);
if (!routes) {
dev_err(card->dev,
"ASoC: Could not allocate DAPM route table\n");
return -EINVAL;
}
for (i = 0; i < num_routes; i++) {
ret = of_property_read_string_index(np, propname,
2 * i, &routes[i].sink);
if (ret) {
dev_err(card->dev,
"ASoC: Property '%s' index %d could not be read: %d\n",
propname, 2 * i, ret);
return -EINVAL;
}
ret = of_property_read_string_index(np, propname,
(2 * i) + 1, &routes[i].source);
if (ret) {
dev_err(card->dev,
"ASoC: Property '%s' index %d could not be read: %d\n",
propname, (2 * i) + 1, ret);
return -EINVAL;
}
}
card->num_of_dapm_routes = num_routes;
card->of_dapm_routes = routes;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing);
unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
const char *prefix,
struct device_node **bitclkmaster,
struct device_node **framemaster)
{
int ret, i;
char prop[128];
unsigned int format = 0;
int bit, frame;
const char *str;
struct {
char *name;
unsigned int val;
} of_fmt_table[] = {
{ "i2s", SND_SOC_DAIFMT_I2S },
{ "right_j", SND_SOC_DAIFMT_RIGHT_J },
{ "left_j", SND_SOC_DAIFMT_LEFT_J },
{ "dsp_a", SND_SOC_DAIFMT_DSP_A },
{ "dsp_b", SND_SOC_DAIFMT_DSP_B },
{ "ac97", SND_SOC_DAIFMT_AC97 },
{ "pdm", SND_SOC_DAIFMT_PDM},
{ "msb", SND_SOC_DAIFMT_MSB },
{ "lsb", SND_SOC_DAIFMT_LSB },
};
if (!prefix)
prefix = "";
/*
* check "dai-format = xxx"
* or "[prefix]format = xxx"
* SND_SOC_DAIFMT_FORMAT_MASK area
*/
ret = of_property_read_string(np, "dai-format", &str);
if (ret < 0) {
snprintf(prop, sizeof(prop), "%sformat", prefix);
ret = of_property_read_string(np, prop, &str);
}
if (ret == 0) {
for (i = 0; i < ARRAY_SIZE(of_fmt_table); i++) {
if (strcmp(str, of_fmt_table[i].name) == 0) {
format |= of_fmt_table[i].val;
break;
}
}
}
/*
* check "[prefix]continuous-clock"
* SND_SOC_DAIFMT_CLOCK_MASK area
*/
snprintf(prop, sizeof(prop), "%scontinuous-clock", prefix);
if (of_property_read_bool(np, prop))
format |= SND_SOC_DAIFMT_CONT;
else
format |= SND_SOC_DAIFMT_GATED;
/*
* check "[prefix]bitclock-inversion"
* check "[prefix]frame-inversion"
* SND_SOC_DAIFMT_INV_MASK area
*/
snprintf(prop, sizeof(prop), "%sbitclock-inversion", prefix);
bit = !!of_get_property(np, prop, NULL);
snprintf(prop, sizeof(prop), "%sframe-inversion", prefix);
frame = !!of_get_property(np, prop, NULL);
switch ((bit << 4) + frame) {
case 0x11:
format |= SND_SOC_DAIFMT_IB_IF;
break;
case 0x10:
format |= SND_SOC_DAIFMT_IB_NF;
break;
case 0x01:
format |= SND_SOC_DAIFMT_NB_IF;
break;
default:
/* SND_SOC_DAIFMT_NB_NF is default */
break;
}
/*
* check "[prefix]bitclock-master"
* check "[prefix]frame-master"
* SND_SOC_DAIFMT_MASTER_MASK area
*/
snprintf(prop, sizeof(prop), "%sbitclock-master", prefix);
bit = !!of_get_property(np, prop, NULL);
if (bit && bitclkmaster)
*bitclkmaster = of_parse_phandle(np, prop, 0);
snprintf(prop, sizeof(prop), "%sframe-master", prefix);
frame = !!of_get_property(np, prop, NULL);
if (frame && framemaster)
*framemaster = of_parse_phandle(np, prop, 0);
switch ((bit << 4) + frame) {
case 0x11:
format |= SND_SOC_DAIFMT_CBM_CFM;
break;
case 0x10:
format |= SND_SOC_DAIFMT_CBM_CFS;
break;
case 0x01:
format |= SND_SOC_DAIFMT_CBS_CFM;
break;
default:
format |= SND_SOC_DAIFMT_CBS_CFS;
break;
}
return format;
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt);
int snd_soc_get_dai_id(struct device_node *ep)
{
struct snd_soc_component *component;
struct snd_soc_dai_link_component dlc;
int ret;
dlc.of_node = of_graph_get_port_parent(ep);
dlc.name = NULL;
/*
* For example HDMI case, HDMI has video/sound port,
* but ALSA SoC needs sound port number only.
* Thus counting HDMI DT port/endpoint doesn't work.
* Then, it should have .of_xlate_dai_id
*/
ret = -ENOTSUPP;
mutex_lock(&client_mutex);
component = soc_find_component(&dlc);
if (component)
ret = snd_soc_component_of_xlate_dai_id(component, ep);
mutex_unlock(&client_mutex);
of_node_put(dlc.of_node);
device property: Fix usecount for of_graph_get_port_parent() Fix inconsistent use of of_graph_get_port_parent() where asoc_simple_card_parse_graph_dai() does of_node_get() before calling it while other callers do not. We can fix this by not trashing the node passed to of_graph_get_port_parent(). Let's also make sure the callers have correct refcounts and remove related incorrect of_node_put() calls for of_for_each_phandle as that's done by of_phandle_iterator_next() except when we break out of the loop early. Let's fix both issues with a single patch to avoid kobject refcounts getting messed up more if two patches are merged separately. Otherwise strange issues can happen caused by memory corruption caused by too many kobject_del() calls such as: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:747 ... (___might_sleep) (__mutex_lock) (mutex_lock_nested) (kernfs_remove) (kobject_del) (kobject_put) (of_get_next_parent) (of_graph_get_port_parent) (asoc_simple_card_parse_graph_dai [snd_soc_simple_card_utils]) (asoc_graph_card_probe [snd_soc_audio_graph_card]) Fixes: 0ef472a973eb ("of_graph: add of_graph_get_port_parent()") Fixes: 2692c1c63c29 ("ASoC: add audio-graph-card support") Fixes: 1689333f8311 ("ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai()") Signed-off-by: Tony Lindgren <tony@atomide.com> Reviewed-by: Rob Herring <robh@kernel.org> Tested-by: Antonio Borneo <borneo.antonio@gmail.com> Tested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-07-28 15:23:15 +07:00
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_get_dai_id);
int snd_soc_get_dai_name(struct of_phandle_args *args,
const char **dai_name)
{
struct snd_soc_component *pos;
struct device_node *component_of_node;
int ret = -EPROBE_DEFER;
mutex_lock(&client_mutex);
for_each_component(pos) {
component_of_node = soc_component_to_node(pos);
if (component_of_node != args->np)
continue;
ret = snd_soc_component_of_xlate_dai_name(pos, args, dai_name);
if (ret == -ENOTSUPP) {
struct snd_soc_dai *dai;
int id = -1;
switch (args->args_count) {
case 0:
id = 0; /* same as dai_drv[0] */
break;
case 1:
id = args->args[0];
break;
default:
/* not supported */
break;
}
if (id < 0 || id >= pos->num_dai) {
ret = -EINVAL;
continue;
}
ret = 0;
/* find target DAI */
for_each_component_dais(pos, dai) {
if (id == 0)
break;
id--;
}
*dai_name = dai->driver->name;
if (!*dai_name)
*dai_name = pos->name;
} else if (ret) {
/*
* if another error than ENOTSUPP is returned go on and
* check if another component is provided with the same
* node. This may happen if a device provides several
* components
*/
continue;
}
break;
}
mutex_unlock(&client_mutex);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_get_dai_name);
int snd_soc_of_get_dai_name(struct device_node *of_node,
const char **dai_name)
{
struct of_phandle_args args;
int ret;
ret = of_parse_phandle_with_args(of_node, "sound-dai",
"#sound-dai-cells", 0, &args);
if (ret)
return ret;
ret = snd_soc_get_dai_name(&args, dai_name);
of_node_put(args.np);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name);
/*
* snd_soc_of_put_dai_link_codecs - Dereference device nodes in the codecs array
* @dai_link: DAI link
*
* Dereference device nodes acquired by snd_soc_of_get_dai_link_codecs().
*/
void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link)
{
struct snd_soc_dai_link_component *component;
int index;
for_each_link_codecs(dai_link, index, component) {
if (!component->of_node)
break;
of_node_put(component->of_node);
component->of_node = NULL;
}
}
EXPORT_SYMBOL_GPL(snd_soc_of_put_dai_link_codecs);
/*
* snd_soc_of_get_dai_link_codecs - Parse a list of CODECs in the devicetree
* @dev: Card device
* @of_node: Device node
* @dai_link: DAI link
*
* Builds an array of CODEC DAI components from the DAI link property
* 'sound-dai'.
* The array is set in the DAI link and the number of DAIs is set accordingly.
* The device nodes in the array (of_node) must be dereferenced by calling
* snd_soc_of_put_dai_link_codecs() on @dai_link.
*
* Returns 0 for success
*/
int snd_soc_of_get_dai_link_codecs(struct device *dev,
struct device_node *of_node,
struct snd_soc_dai_link *dai_link)
{
struct of_phandle_args args;
struct snd_soc_dai_link_component *component;
char *name;
int index, num_codecs, ret;
/* Count the number of CODECs */
name = "sound-dai";
num_codecs = of_count_phandle_with_args(of_node, name,
"#sound-dai-cells");
if (num_codecs <= 0) {
if (num_codecs == -ENOENT)
dev_err(dev, "No 'sound-dai' property\n");
else
dev_err(dev, "Bad phandle in 'sound-dai'\n");
return num_codecs;
}
treewide: devm_kzalloc() -> devm_kcalloc() The devm_kzalloc() function has a 2-factor argument form, devm_kcalloc(). This patch replaces cases of: devm_kzalloc(handle, a * b, gfp) with: devm_kcalloc(handle, a * b, gfp) as well as handling cases of: devm_kzalloc(handle, a * b * c, gfp) with: devm_kzalloc(handle, array3_size(a, b, c), gfp) as it's slightly less ugly than: devm_kcalloc(handle, array_size(a, b), c, gfp) This does, however, attempt to ignore constant size factors like: devm_kzalloc(handle, 4 * 1024, gfp) though any constants defined via macros get caught up in the conversion. Any factors with a sizeof() of "unsigned char", "char", and "u8" were dropped, since they're redundant. Some manual whitespace fixes were needed in this patch, as Coccinelle really liked to write "=devm_kcalloc..." instead of "= devm_kcalloc...". The Coccinelle script used for this was: // Fix redundant parens around sizeof(). @@ expression HANDLE; type TYPE; expression THING, E; @@ ( devm_kzalloc(HANDLE, - (sizeof(TYPE)) * E + sizeof(TYPE) * E , ...) | devm_kzalloc(HANDLE, - (sizeof(THING)) * E + sizeof(THING) * E , ...) ) // Drop single-byte sizes and redundant parens. @@ expression HANDLE; expression COUNT; typedef u8; typedef __u8; @@ ( devm_kzalloc(HANDLE, - sizeof(u8) * (COUNT) + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(__u8) * (COUNT) + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(char) * (COUNT) + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(unsigned char) * (COUNT) + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(u8) * COUNT + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(__u8) * COUNT + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(char) * COUNT + COUNT , ...) | devm_kzalloc(HANDLE, - sizeof(unsigned char) * COUNT + COUNT , ...) ) // 2-factor product with sizeof(type/expression) and identifier or constant. @@ expression HANDLE; type TYPE; expression THING; identifier COUNT_ID; constant COUNT_CONST; @@ ( - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * (COUNT_ID) + COUNT_ID, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * COUNT_ID + COUNT_ID, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * (COUNT_CONST) + COUNT_CONST, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * COUNT_CONST + COUNT_CONST, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * (COUNT_ID) + COUNT_ID, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * COUNT_ID + COUNT_ID, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * (COUNT_CONST) + COUNT_CONST, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * COUNT_CONST + COUNT_CONST, sizeof(THING) , ...) ) // 2-factor product, only identifiers. @@ expression HANDLE; identifier SIZE, COUNT; @@ - devm_kzalloc + devm_kcalloc (HANDLE, - SIZE * COUNT + COUNT, SIZE , ...) // 3-factor product with 1 sizeof(type) or sizeof(expression), with // redundant parens removed. @@ expression HANDLE; expression THING; identifier STRIDE, COUNT; type TYPE; @@ ( devm_kzalloc(HANDLE, - sizeof(TYPE) * (COUNT) * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE) * (COUNT) * STRIDE + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE) * COUNT * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE) * COUNT * STRIDE + array3_size(COUNT, STRIDE, sizeof(TYPE)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING) * (COUNT) * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING) * (COUNT) * STRIDE + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING) * COUNT * (STRIDE) + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING) * COUNT * STRIDE + array3_size(COUNT, STRIDE, sizeof(THING)) , ...) ) // 3-factor product with 2 sizeof(variable), with redundant parens removed. @@ expression HANDLE; expression THING1, THING2; identifier COUNT; type TYPE1, TYPE2; @@ ( devm_kzalloc(HANDLE, - sizeof(TYPE1) * sizeof(TYPE2) * COUNT + array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING1) * sizeof(THING2) * COUNT + array3_size(COUNT, sizeof(THING1), sizeof(THING2)) , ...) | devm_kzalloc(HANDLE, - sizeof(THING1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(THING1), sizeof(THING2)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE1) * sizeof(THING2) * COUNT + array3_size(COUNT, sizeof(TYPE1), sizeof(THING2)) , ...) | devm_kzalloc(HANDLE, - sizeof(TYPE1) * sizeof(THING2) * (COUNT) + array3_size(COUNT, sizeof(TYPE1), sizeof(THING2)) , ...) ) // 3-factor product, only identifiers, with redundant parens removed. @@ expression HANDLE; identifier STRIDE, SIZE, COUNT; @@ ( devm_kzalloc(HANDLE, - (COUNT) * STRIDE * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - COUNT * (STRIDE) * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - COUNT * STRIDE * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - (COUNT) * (STRIDE) * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - COUNT * (STRIDE) * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - (COUNT) * STRIDE * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - (COUNT) * (STRIDE) * (SIZE) + array3_size(COUNT, STRIDE, SIZE) , ...) | devm_kzalloc(HANDLE, - COUNT * STRIDE * SIZE + array3_size(COUNT, STRIDE, SIZE) , ...) ) // Any remaining multi-factor products, first at least 3-factor products, // when they're not all constants... @@ expression HANDLE; expression E1, E2, E3; constant C1, C2, C3; @@ ( devm_kzalloc(HANDLE, C1 * C2 * C3, ...) | devm_kzalloc(HANDLE, - (E1) * E2 * E3 + array3_size(E1, E2, E3) , ...) | devm_kzalloc(HANDLE, - (E1) * (E2) * E3 + array3_size(E1, E2, E3) , ...) | devm_kzalloc(HANDLE, - (E1) * (E2) * (E3) + array3_size(E1, E2, E3) , ...) | devm_kzalloc(HANDLE, - E1 * E2 * E3 + array3_size(E1, E2, E3) , ...) ) // And then all remaining 2 factors products when they're not all constants, // keeping sizeof() as the second factor argument. @@ expression HANDLE; expression THING, E1, E2; type TYPE; constant C1, C2, C3; @@ ( devm_kzalloc(HANDLE, sizeof(THING) * C2, ...) | devm_kzalloc(HANDLE, sizeof(TYPE) * C2, ...) | devm_kzalloc(HANDLE, C1 * C2 * C3, ...) | devm_kzalloc(HANDLE, C1 * C2, ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * (E2) + E2, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(TYPE) * E2 + E2, sizeof(TYPE) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * (E2) + E2, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - sizeof(THING) * E2 + E2, sizeof(THING) , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - (E1) * E2 + E1, E2 , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - (E1) * (E2) + E1, E2 , ...) | - devm_kzalloc + devm_kcalloc (HANDLE, - E1 * E2 + E1, E2 , ...) ) Signed-off-by: Kees Cook <keescook@chromium.org>
2018-06-13 04:07:58 +07:00
component = devm_kcalloc(dev,
num_codecs, sizeof(*component),
GFP_KERNEL);
if (!component)
return -ENOMEM;
dai_link->codecs = component;
dai_link->num_codecs = num_codecs;
/* Parse the list */
for_each_link_codecs(dai_link, index, component) {
ret = of_parse_phandle_with_args(of_node, name,
"#sound-dai-cells",
index, &args);
if (ret)
goto err;
component->of_node = args.np;
ret = snd_soc_get_dai_name(&args, &component->dai_name);
if (ret < 0)
goto err;
}
return 0;
err:
snd_soc_of_put_dai_link_codecs(dai_link);
dai_link->codecs = NULL;
dai_link->num_codecs = 0;
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_link_codecs);
static int __init snd_soc_init(void)
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
{
snd_soc_debugfs_init();
snd_soc_util_init();
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
return platform_driver_register(&soc_driver);
}
module_init(snd_soc_init);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
static void __exit snd_soc_exit(void)
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
{
snd_soc_util_exit();
snd_soc_debugfs_exit();
platform_driver_unregister(&soc_driver);
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
}
module_exit(snd_soc_exit);
/* Module information */
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
[ALSA] ASoC: core code This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 23:31:09 +07:00
MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:soc-audio");