February 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 0/7] MFD: twl6040: Conversion to i2c driver
by Peter Ujfalusi 09 Feb '12

09 Feb '12

Hello, Changes since v1 (RFC series): - make use of regmap in the converted twl6040 driver [1] - use module_i2c_driver to reduce line of code - Use regulator_bulk functions for now in the twl6040 mfd driver. [2] [1] Initially I'm not using the register cache offered by regmap, it will come as a followup series [2] When the driver supports more power levels, the bulk_enable/disable will be removed, and the VIO/V2V1 supplies are going to be handled separately. As of now it is better to enable them with the bulk call. Still in the to-do list: Implement proper regulator support on SDP4430, and PandaBoard for the needed supplies by the twl6040. The supply source is the same on both boards: TWL6040 VDD2V1 <- TWL6030 V2V1 SMPS TWL6040 VDDVIO <- TWL6030 V1V8 SMPS There seams to be no control implemented in the regulator driver for these SMPS regulators, so this need to be done first. The reason that I'm sending this series (even when I know that the proper regulator support is missing for SDP4430/PandaBoard) is to get feedback for the rest of the series, so I can include the fixes in the v3 series. ------ Intro mail from the original series: This series will convert the twl6040 MFD driver to an i2c driver. Compared to older twl4030/5030/TPS the twl6040 is a standalone audio IC. It is better if the twl6040-core (and all of it's child devices) does not depend on the twl-core since it has nothing to with it. With this conversion the dependency on twl can be dropped from the twl6040 driver stack (core, vibra, audio). Between the first and second patch the audio will not probe on OMAP4, but I felt it is better this way at least for the first RFC series to not have too big change within one patch. The two patch can be squashed together later if no objections. Regards, Peter --- Peter Ujfalusi (7): MFD: twl-core: Detach twl6040 from the pmic mfd driver MFD: twl6040: Convert to i2c driver, and separate it from twl core ASoC: twl6040: Remove dependency on twl4030 from Kconfig OMAP: 4430sdp: Correct fixed regulator device ID OMAP: sdp4430: Add fixed regulator for twl6040 needs OMAP: omap4panda: Add fixed regulator for twl6040 needs MFD: TWL6040: Add regulator support for VIO, V2V1 supplies arch/arm/mach-omap2/board-4430sdp.c | 77 ++++++++++++++-- arch/arm/mach-omap2/board-generic.c | 2 +- arch/arm/mach-omap2/board-omap4panda.c | 75 ++++++++++++++-- arch/arm/mach-omap2/twl-common.c | 37 +++++++- arch/arm/mach-omap2/twl-common.h | 10 +-- drivers/input/misc/Kconfig | 1 - drivers/input/misc/twl6040-vibra.c | 4 +- drivers/mfd/Kconfig | 2 +- drivers/mfd/twl-core.c | 58 +++++++----- drivers/mfd/twl6040-core.c | 155 +++++++++++++++++++++---------- include/linux/i2c/twl.h | 12 --- include/linux/mfd/twl6040.h | 29 ++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/twl6040.c | 3 +- 14 files changed, 345 insertions(+), 122 deletions(-) -- 1.7.8.4

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Re: [alsa-devel] [PATCH] [trivial] twl4030: Fix typo in twl4030.c
by Peter Ujfalusi 09 Feb '12

09 Feb '12

Hi, On 02/08/2012 01:53 PM, Masanari Iida wrote: > Correct spelling "memroy" to "memory" in > sound/soc/codecs/twl4030.c > > Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> > --- > sound/soc/codecs/twl4030.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c > index 3039ba2..d7eee0d 100644 > --- a/sound/soc/codecs/twl4030.c > +++ b/sound/soc/codecs/twl4030.c > @@ -2221,7 +2221,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) > > twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); > if (twl4030 == NULL) { > - printk("Can not allocate memroy\n"); > + printk("Can not allocate memory\n"); > return -ENOMEM; > } > snd_soc_codec_set_drvdata(codec, twl4030); Could you please resend the patch with the ASoC maintainers on CC (Liam, Mark), and the alsa-devel list. Thank you. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com>

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 08 Feb '12

08 Feb '12

The following changes since commit 43b6cec27e1e50a1de3eff47e66e502f3fe7e66e: ASoC: wm_hubs: Correct line input to line output 2 paths (2012-02-01 23:55:56 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-3.3 for you to fetch changes up to a7c4183be2d6a7da8c97a9b671b5f3aed321127e: ASoC: wm8994: Disable line output discharge prior to ramping VMID (2012-02-08 19:52:00 +0000) ---------------------------------------------------------------- A few small WM8994 updates to go on top of the previous lot of things that were sent. They collide with some -next work so I'd really like to get them into 3.3-rc3 if possible to merge back up into the -next code. All driver specific and unexciting in the grand scheme of things. ---------------------------------------------------------------- Mark Brown (3): ASoC: wm8994: Enabling VMID should take a runtime PM reference ASoC: wm8994: Fix typo in VMID ramp setting ASoC: wm8994: Disable line output discharge prior to ramping VMID sound/soc/codecs/wm8994.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH] ASoC: core: Convert single bit wide bitfields into bools
by Mark Brown 08 Feb '12

08 Feb '12

It's what they are, we've just been using these for historical reasons. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 12 ++++++------ include/sound/soc-dapm.h | 26 +++++++++++++------------- include/sound/soc.h | 32 ++++++++++++++++---------------- 3 files changed, 35 insertions(+), 35 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1a2b08c..53f9a1f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -208,7 +208,7 @@ struct snd_soc_dai_driver { /* DAI capabilities */ struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; - unsigned int symmetric_rates:1; + bool symmetric_rates; /* probe ordering - for components with runtime dependencies */ int probe_order; @@ -230,13 +230,13 @@ struct snd_soc_dai { struct snd_soc_dai_driver *driver; /* DAI runtime info */ - unsigned int capture_active:1; /* stream is in use */ - unsigned int playback_active:1; /* stream is in use */ - unsigned int symmetric_rates:1; + bool capture_active; /* stream is in use */ + bool playback_active; /* stream is in use */ + bool symmetric_rates; struct snd_pcm_runtime *runtime; unsigned int active; - unsigned char pop_wait:1; - unsigned char probed:1; + bool pop_wait; + bool probed; /* DAI DMA data */ void *playback_dma_data; diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index db8435a..99661c0 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -456,9 +456,9 @@ struct snd_soc_dapm_path { struct snd_kcontrol *kcontrol; /* status */ - u32 connect:1; /* source and sink widgets are connected */ - u32 walked:1; /* path has been walked */ - u32 weak:1; /* path ignored for power management */ + bool connect; /* source and sink widgets are connected */ + bool walked; /* path has been walked */ + bool weak; /* path ignored for power management */ int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -488,16 +488,16 @@ struct snd_soc_dapm_widget { unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ unsigned int off_val; /* off state value */ - unsigned char power:1; /* block power status */ - unsigned char invert:1; /* invert the power bit */ - unsigned char active:1; /* active stream on DAC, ADC's */ - unsigned char connected:1; /* connected codec pin */ - unsigned char new:1; /* cnew complete */ - unsigned char ext:1; /* has external widgets */ - unsigned char force:1; /* force state */ - unsigned char ignore_suspend:1; /* kept enabled over suspend */ - unsigned char new_power:1; /* power from this run */ - unsigned char power_checked:1; /* power checked this run */ + bool power; /* block power status */ + bool invert; /* invert the power bit */ + bool active; /* active stream on DAC, ADC's */ + bool connected; /* connected codec pin */ + bool new; /* cnew complete */ + bool ext; /* has external widgets */ + bool force; /* force state */ + bool ignore_suspend; /* kept enabled over suspend */ + bool new_power; /* power from this run */ + bool power_checked; /* power checked this run */ int subseq; /* sort within widget type */ int (*power_check)(struct snd_soc_dapm_widget *w); diff --git a/include/sound/soc.h b/include/sound/soc.h index d1e7a6a..466b1f3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -571,14 +571,14 @@ struct snd_soc_codec { /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; - unsigned int cache_bypass:1; /* Suppress access to the cache */ - unsigned int suspended:1; /* Codec is in suspend PM state */ - unsigned int probed:1; /* Codec has been probed */ - unsigned int ac97_registered:1; /* Codec has been AC97 registered */ - unsigned int ac97_created:1; /* Codec has been created by SoC */ - unsigned int sysfs_registered:1; /* codec has been sysfs registered */ - unsigned int cache_init:1; /* codec cache has been initialized */ - unsigned int using_regmap:1; /* using regmap access */ + bool cache_bypass; /* Suppress access to the cache */ + bool suspended; /* Codec is in suspend PM state */ + bool probed; /* Codec has been probed */ + bool ac97_registered; /* Codec has been AC97 registered */ + bool ac97_created; /* Codec has been created by SoC */ + bool sysfs_registered; /* codec has been sysfs registered */ + bool cache_init; /* codec cache has been initialized */ + bool using_regmap; /* using regmap access */ u32 cache_only; /* Suppress writes to hardware */ u32 cache_sync; /* Cache needs to be synced to hardware */ @@ -598,7 +598,7 @@ struct snd_soc_codec { /* dapm */ struct snd_soc_dapm_context dapm; - unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ + bool ignore_pmdown_time; /* pmdown_time is ignored at stop */ #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_codec_root; @@ -712,8 +712,8 @@ struct snd_soc_platform { struct device *dev; struct snd_soc_platform_driver *driver; - unsigned int suspended:1; /* platform is suspended */ - unsigned int probed:1; + bool suspended; /* platform is suspended */ + bool probed; struct snd_soc_card *card; struct list_head list; @@ -737,13 +737,13 @@ struct snd_soc_dai_link { unsigned int dai_fmt; /* format to set on init */ /* Keep DAI active over suspend */ - unsigned int ignore_suspend:1; + bool ignore_suspend; /* Symmetry requirements */ - unsigned int symmetric_rates:1; + bool symmetric_rates; /* pmdown_time is ignored at stop */ - unsigned int ignore_pmdown_time:1; + bool ignore_pmdown_time; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); @@ -876,8 +876,8 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; - unsigned int complete:1; - unsigned int dev_registered:1; + bool complete; + bool dev_registered; long pmdown_time; -- 1.7.9.rc1

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[alsa-devel] [PATCH] ASoC: wm8962: Support mono playback and record
by Mark Brown 08 Feb '12

08 Feb '12

Make sure we generate enough BCLKs for I2S style modes by always having a minimum of two channels worth of clocks for the BCLK. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0f6bd06..85f71a7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2536,6 +2536,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, int adctl3 = 0; wm8962->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + wm8962->bclk *= 2; + wm8962->lrclk = params_rate(params); for (i = 0; i < ARRAY_SIZE(sr_vals); i++) { @@ -2913,14 +2916,14 @@ static struct snd_soc_dai_driver wm8962_dai = { .name = "wm8962", .playback = { .stream_name = "Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8962_RATES, .formats = WM8962_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8962_RATES, .formats = WM8962_FORMATS, -- 1.7.9.rc1

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[alsa-devel] [RFC/PATCH] ALSA: PCM - Add PCM creation API for ASoC dynamic PCMs.
by Liam Girdwood 08 Feb '12

08 Feb '12

The new ASoC dynamic PCM core needs to create PCMs and substreams that are for use by internal ASoC drivers only and not visible to userspace for direct IO. These new PCMs are similar to regular PCMs expect they have no device nodes or procfs entries. The ASoC component drivers use them in exactly the same way as regular PCMs for PCM and DAI operations. The intention is that a dynamic PCM based driver will register both regular PCMs and dynamic PCMs. The regular PCMs will be used for all IO with userspace however the dynamic PCMs will be used by the driver to route digital audio through numerous back end DAI links (with potentially a DSP providing different hw_params, DAI formats based on the regular front end PCM params) to devices like CODECs, MODEMs, Bluetooth, FM, DMICs, etc This patch adds a new snd_pcm_new_dynamic() API call to create the dynamic PCM without device nodes or procfs. It also adds snd_pcm_new_stream_dynamic() and snd_pcm_dev_register_dynamic() which are used to create the dynamic PCM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/pcm.h | 3 + sound/core/pcm.c | 145 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 148 insertions(+), 0 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0cf91b2..1597f89 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -475,6 +475,9 @@ extern const struct file_operations snd_pcm_f_ops[2]; int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm **rpcm); +int snd_pcm_new_dynamic(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, + struct snd_pcm ** rpcm); int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count); int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8928ca87..8e2a429 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -41,6 +41,7 @@ static DEFINE_MUTEX(register_mutex); static int snd_pcm_free(struct snd_pcm *pcm); static int snd_pcm_dev_free(struct snd_device *device); static int snd_pcm_dev_register(struct snd_device *device); +static int snd_pcm_dev_register_dynamic(struct snd_device *device); static int snd_pcm_dev_disconnect(struct snd_device *device); static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) @@ -697,6 +698,60 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) EXPORT_SYMBOL(snd_pcm_new_stream); /** + * snd_pcm_new_stream_dynamic - create a new dynamic PCM stream + * @pcm: the pcm instance + * @stream: the stream direction, SNDRV_PCM_STREAM_XXX + * @substream_count: the number of substreams + * + * Creates a new dynamic stream for the pcm. Dynamic PCM streams have no procfs + * entries or device entries and are hence not directly controllable by + * userspace. + * The corresponding stream on the pcm must have been empty before + * calling this, i.e. zero must be given to the argument of + * snd_pcm_new(). + * + * Returns zero if successful, or a negative error code on failure. + */ +static int snd_pcm_new_stream_dynamic(struct snd_pcm *pcm, int stream, + int substream_count) +{ + int idx; + struct snd_pcm_str *pstr = &pcm->streams[stream]; + struct snd_pcm_substream *substream, *prev; + + pstr->stream = stream; + pstr->pcm = pcm; + pstr->substream_count = substream_count; + + prev = NULL; + for (idx = 0, prev = NULL; idx < substream_count; idx++) { + substream = kzalloc(sizeof(*substream), GFP_KERNEL); + if (substream == NULL) { + snd_printk(KERN_ERR "Cannot allocate BE PCM substream\n"); + return -ENOMEM; + } + substream->pcm = pcm; + substream->pstr = pstr; + substream->number = idx; + substream->stream = stream; + sprintf(substream->name, "subdevice #%i", idx); + substream->buffer_bytes_max = UINT_MAX; + if (prev == NULL) + pstr->substream = substream; + else + prev->next = substream; + + substream->group = &substream->self_group; + spin_lock_init(&substream->self_group.lock); + INIT_LIST_HEAD(&substream->self_group.substreams); + list_add_tail(&substream->link_list, &substream->self_group.substreams); + atomic_set(&substream->mmap_count, 0); + prev = substream; + } + return 0; +} + +/** * snd_pcm_new - create a new PCM instance * @card: the card instance * @id: the id string @@ -735,6 +790,7 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device, } pcm->card = card; pcm->device = device; + if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { @@ -758,6 +814,72 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device, EXPORT_SYMBOL(snd_pcm_new); +/** + * snd_pcm_new_dynamic - create a new dynamic PCM instance for ASoC BE DAI link + * @card: the card instance + * @id: the id string + * @device: the device index (zero based - shared with normal PCMs) + * @playback_count: the number of substreams for playback + * @capture_count: the number of substreams for capture + * @rpcm: the pointer to store the new pcm instance + * + * Creates a new dynamic PCM instance with no userspace device or procfs + * entries. This is used by ASoC Back End PCMs in order to create a PCM that + * will only be used internally by kernel drivers. i.e. it cannot be opened + * by userspace. It provides existing ASoC components drivers with a substream + * and access to any private data. + * + * The pcm operators have to be set afterwards to the new instance + * via snd_pcm_set_ops(). + * + * Returns zero if successful, or a negative error code on failure. + */ +int snd_pcm_new_dynamic(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, + struct snd_pcm ** rpcm) +{ + struct snd_pcm *pcm; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_pcm_dev_free, + .dev_register = snd_pcm_dev_register_dynamic, + .dev_disconnect = snd_pcm_dev_disconnect, + }; + + if (snd_BUG_ON(!card)) + return -ENXIO; + if (rpcm) + *rpcm = NULL; + pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + if (pcm == NULL) { + snd_printk(KERN_ERR "Cannot allocate dynamic PCM\n"); + return -ENOMEM; + } + pcm->card = card; + pcm->device = device; + + if (id) + strlcpy(pcm->id, id, sizeof(pcm->id)); + if ((err = snd_pcm_new_stream_dynamic(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { + snd_pcm_free(pcm); + return err; + } + if ((err = snd_pcm_new_stream_dynamic(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count)) < 0) { + snd_pcm_free(pcm); + return err; + } + + if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { + snd_pcm_free(pcm); + return err; + } + if (rpcm) + *rpcm = pcm; + return 0; +} + +EXPORT_SYMBOL(snd_pcm_new_dynamic); + static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; @@ -1035,6 +1157,29 @@ static int snd_pcm_dev_register(struct snd_device *device) return 0; } +static int snd_pcm_dev_register_dynamic(struct snd_device *device) +{ + int err; + struct snd_pcm_notify *notify; + struct snd_pcm *pcm; + + if (snd_BUG_ON(!device || !device->device_data)) + return -ENXIO; + pcm = device->device_data; + mutex_lock(&register_mutex); + err = snd_pcm_add(pcm); + if (err) { + mutex_unlock(&register_mutex); + return err; + } + + list_for_each_entry(notify, &snd_pcm_notify_list, list) + notify->n_register(pcm); + + mutex_unlock(&register_mutex); + return 0; +} + static int snd_pcm_dev_disconnect(struct snd_device *device) { struct snd_pcm *pcm = device->device_data; -- 1.7.5.4

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Re: [alsa-devel] FW: [PATCH v3] ASoC: Add support for TI LM49453 Audio codec
by M R Swami Reddy 08 Feb '12

08 Feb '12

> -----Original Message----- > From: Mark Brown [mailto:broonie@opensource.wolfsonmicro.com] > Sent: Tuesday, February 07, 2012 10:34 PM > To: Reddy, MR Swami > Cc: Girdwood, Liam; alsa-devel(a)alsa-project.org > Subject: Re: [PATCH v3] ASoC: Add support for TI LM49453 Audio codec > > On Mon, Feb 06, 2012 at 06:20:26AM -0800, Reddy, MR Swami wrote: > >> Changes made in v3: >> o Updated the lm49453_set_dai_pll() as per review comments in v2 patch. >> o Removed pll disable code in _set_dai_pll(). > > This doesn't really seem to address the issue at all - you still have the problems with the set_pll() function not doing anything it's supposed to do with the input and output frequencies, and now there's no way to disable the PLL. OK. So its ideal to remove the _set_dai_pll(), as its not doing anything here. Thanks Swami

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[alsa-devel] [RESEND PATCH] davinci: CQ93VC: remove machine specific header file inclusion from codec driver
by Manjunath Hadli 08 Feb '12

08 Feb '12

remove unnecessary inclusion of machine specific header file mach/dm365.h from cq93vc.c voice codec driver which comes in the way of platform code consolidation. Signed-off-by: Manjunath Hadli <manjunath.hadli(a)ti.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> Cc: <alsa-devel(a)alsa-project.org> Cc: <linux-kernel(a)vger.kernel.org> --- sound/soc/codecs/cq93vc.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd0..ff4eb7a 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -38,8 +38,6 @@ #include <sound/soc.h> #include <sound/initval.h> -#include <mach/dm365.h> - static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, unsigned int reg) { -- 1.6.2.4

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[alsa-devel] [PATCH 1/15] sound/soc/mxs/mxs-saif.c: add missing iounmap
by Julia Lawall 08 Feb '12

08 Feb '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Add missing iounmap in error handling code, in a case where the function already preforms iounmap on some other execution path. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e; statement S,S1; int ret; @@ e = \(ioremap\|ioremap_nocache\)(...) ... when != iounmap(e) if (<+...e...+>) S ... when any when != iounmap(e) *if (...) { ... when != iounmap(e) return ...; } ... when any iounmap(e); // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/mxs/mxs-saif.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 049e543..5ee0adb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -680,7 +680,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); - goto failed_ioremap; + goto failed_get_resource; } saif->dma_param.chan_num = dmares->start; @@ -739,6 +739,7 @@ failed_register: failed_get_irq2: free_irq(saif->irq, saif); failed_get_irq1: +failed_get_resource: iounmap(saif->base); failed_ioremap: release_mem_region(iores->start, resource_size(iores));

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[alsa-devel] concept of substream...
by Vinod Koul 08 Feb '12

08 Feb '12

Hi, I was trying to understand the concept of substream. So can we have a device which can have support say 5 substreams and these are controlled (start/stop) independent of each other. At any instance we can have any number of substreams active and runtime we should be able to activate/deactivate a particular substream. Is that doable with this concept or should i look at something else? -- ~Vinod

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