July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: Make snd_pcm_debug_name usable outside pcm_lib
by linux＠audioscience.com 24 Jul '11

24 Jul '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Formatting a PCM name is useful for module debug too. Add snd_prefix when making function public. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- include/sound/pcm.h | 12 ++++++++++++ sound/core/pcm_lib.c | 10 +++++----- sound/pci/asihpi/asihpi.c | 21 --------------------- 3 files changed, 17 insertions(+), 26 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e1bad11..c83c754 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -507,6 +507,18 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream); void snd_pcm_vma_notify_data(void *client, void *data); int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area); + +#if defined CONFIG_SND_DEBUG +void snd_pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len); +#else +static inline void +snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) +{ + *buf = 0; +} +#endif + /* * PCM library */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f134130..f82329d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -128,7 +128,7 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } -static void pcm_debug_name(struct snd_pcm_substream *substream, +void snd_pcm_debug_name(struct snd_pcm_substream *substream, char *name, size_t len) { snprintf(name, len, "pcmC%dD%d%c:%d", @@ -168,7 +168,7 @@ static void xrun(struct snd_pcm_substream *substream) snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); snd_printd(KERN_DEBUG "XRUN: %s\n", name); dump_stack_on_xrun(substream); } @@ -243,7 +243,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream) return; if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) return; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { entry = &log->entries[idx]; if (entry->period_size == 0) @@ -319,7 +319,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (pos >= runtime->buffer_size) { if (printk_ratelimit()) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); snd_printd(KERN_ERR "BUG: %s, pos = %ld, " "buffer size = %ld, period size = %ld\n", @@ -364,7 +364,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (xrun_debug(substream, in_interrupt ? XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); snd_printd("%s_update: %s: pos=%u/%u/%u, " "hwptr=%ld/%ld/%ld/%ld\n", in_interrupt ? "period" : "hwptr", diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index b941d25..eae62eb 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -41,31 +41,10 @@ #include <sound/tlv.h> #include <sound/hwdep.h> - MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. <support(a)audioscience.com>"); MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); -#if defined CONFIG_SND_DEBUG -/* copied from pcm_lib.c, hope later patch will make that version public -and this copy can be removed */ -static inline void -snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) -{ - snprintf(buf, size, "pcmC%dD%d%c:%d", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p', - substream->number); -} -#else -static inline void -snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) -{ - *buf = 0; -} -#endif - #if defined CONFIG_SND_DEBUG_VERBOSE /** * snd_printddd - very verbose debug printk -- 1.7.0.4

3 4

Re: [alsa-devel] [arch-general] Panic - no sound devices after upgrade
by Adrian Knoth 23 Jul '11

23 Jul '11

On 07/23/11 16:21, Fons Adriaensen wrote: Note: I suggest to narrow the CC list a bit and better move the discussion to alsa-devel. At least, let's drop LKML and arch. >>> Seems to have solved the problem :-) :-) >> Good to know. I'll wait for your final confirmation and then commit it. > > One problem: the gain settings have changed, now having a range of 0..64. > How do these values map to the internal ones (0..0xffff) ? I *hope* there > is value corresponding to EXACTLY 0x8000 (gain = 1) !!! For the simple ALSA mixer, the driver does gain = ucontrol->value.integer.value[0]*UNITY_GAIN/64; UNITY_GAIN is 32768 (0x8000), so if you select 64 on the ALSA side, it gives you exactly what you need. Or in other words: faders to 100% in alsamixer, that's it. With hdspmixer or any other software that's using the full mixer, you can still use the fine-grained 65k steps. > BTW, why was this change made? No idea. I reckon faberman was tired of stepping through 65k mixer levels in alsamixer. ;) (disclaimer: I never used the old driver, so I don't know how it was before) Cheers

1 0

[alsa-devel] Sending headset insertion events to user space on bootup
by Brad Rubin 23 Jul '11

23 Jul '11

Hi all, I’m having an issue passing events when a headset is inserted upon bootup. I’m sending a headset insertion event to user space through snd_soc_jack_report, but user space is not yet ready to receive events and therefore misses the event. Since the kernel driver doesn’t resend this event, then there’s no way for user space to see it. User space is still booting up when the event is sent. Does anyone know of a solution so user space can receive headset insertion events on bootup? Is is possible to either allow the kernel to postpone sending events until user space is ready or to store events in some sort of cache so that user space can access them? Would it be possible to use ALSA to store the headset status in a device node? Thanks, Brad -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 1

[alsa-devel] ASoC Peek individual register debugfs
by Patrick Lai 23 Jul '11

23 Jul '11

Hi, Does latest soc-core.c provide debugfs interface for peeking individual register? If not, I plan to add this interface. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 3

[alsa-devel] ABE routing table without TWL6040
by hong zhang 23 Jul '11

23 Jul '11

List, My custom OMAP4 board does not have TWL6040 device. What I do is 1. comment out TWL6040 in arch/arm/mach-omap2/Kconfig 2. comment out twl6040 inside sdp4430_dai[] at sound/soc/omap/sdp4430.c I find ABE routing table is no longer useful because of changes in steps 1 and 2. run "arecord -f S16_LE -r 48000 -c 2 -D plughw:0,0 try.wav" returns error: Recording WAVE 'try.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo try.wav: No such file or directory dmesg shows dmic_trigger is called. 1. Could anyone please give me hint where I should look into? 2. Try Legacy DMIC to turn off ABE by comment out ABE in sound/soc/omap/kconfig but "No Sound card Found" when boot. Thanks. ---henry

1 0

[alsa-devel] [PATCH] sound, hda_codec: Fix compilation warnings
by Borislav Petkov 22 Jul '11

22 Jul '11

From: Borislav Petkov <borislav.petkov(a)amd.com> Convert macro to an inline function to shut up gcc: sound/pci/hda/patch_sigmatel.c: In function `stac92xx_init´: sound/pci/hda/patch_sigmatel.c:4385:3: warning: statement with no effect sound/pci/hda/patch_sigmatel.c: In function `stac92xx_resume´: sound/pci/hda/patch_sigmatel.c:4925:3: warning: statement with no effect sound/pci/hda/hda_codec.c: In function `snd_hda_mixer_amp_switch_put´: sound/pci/hda/patch_realtek.c:4176:2: warning: statement with no effect sound/pci/hda/hda_codec.c:2320:2: warning: statement with no effect sound/pci/hda/patch_realtek.c: In function `alc_resume´: sound/pci/hda/patch_realtek.c:4558:2: warning: statement with no effect sound/pci/hda/patch_realtek.c: In function `alc269_resume´: sound/pci/hda/patch_realtek.c:14878:2: warning: statement with no effect Signed-off-by: Borislav Petkov <borislav.petkov(a)amd.com> --- sound/pci/hda/hda_codec.h | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 59c9730..79ede92 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1005,8 +1005,12 @@ int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) return codec->patch_ops.check_power_status(codec, nid); return 0; } -#else -#define hda_call_check_power_status(codec, nid) 0 +#else +static inline +int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + return 0; +} #endif /* -- 1.7.4.rc2

2 1

[alsa-devel] additional problem
by David Henderson 22 Jul '11

22 Jul '11

Hi gang! I've successfully been able to compile the alsa-utils package with the "--with-alsa-inc-prefix=/opt/staging/alsa/var/share/include --with-alsa-prefix=/opt/staging/alsa/lib", but the problem I'm having now is that the compiled binaries are looking for those directories during run-time and not just compile-time. Does anyone have any thoughts on using those directories for package creation, but that the software doesn't use the '/opt/staging/alsa' prefix during run-time? Thanks, Dave

4 25

[alsa-devel] [PATCH V3 0/3] ASoC for Alchemy Au1000/1500/1100
by Manuel Lauss 22 Jul '11

22 Jul '11

Hello, These 3 patches implement ASoC drivers for the AC97 and I2S controllers found on early Alchemy chips. They are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V2: - implemented changes after feedback from Lars-Peter Clausen: * split patch 1 in two, one for the ASoC drivers, and a separate for DB1000 machine code. * get rid of automatic dma device registration * tidied the I2S/AC97 sources - mark sound/mips/au1x00.c as DEPRECATED instead of removing it outright. Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (3): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver arch/mips/alchemy/devboards/db1x00/platform.c | 48 +++ arch/mips/include/asm/mach-au1x00/au1000.h | 31 -- sound/mips/Kconfig | 5 +- sound/soc/au1x/Kconfig | 28 ++ sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 375 ++++++++++++++++++++++++ sound/soc/au1x/db1000.c | 75 +++++ sound/soc/au1x/dma.c | 381 +++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 348 ++++++++++++++++++++++ sound/soc/au1x/psc.h | 22 +- 10 files changed, 1282 insertions(+), 41 deletions(-) create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 3

[alsa-devel] [PATCH] regulator: Make core more chatty about some errors
by Mark Brown 22 Jul '11

22 Jul '11

Prevent some head scratching by making the core log about some rare but possible errors with invalid voltage ranges and modes being set. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/regulator/core.c | 11 +++++++++-- 1 files changed, 9 insertions(+), 2 deletions(-) diff --git a/drivers/regulator/core.c b/drivers/regulator/core.c index f59821f..3700d09 100644 --- a/drivers/regulator/core.c +++ b/drivers/regulator/core.c @@ -151,8 +151,11 @@ static int regulator_check_voltage(struct regulator_dev *rdev, if (*min_uV < rdev->constraints->min_uV) *min_uV = rdev->constraints->min_uV; - if (*min_uV > *max_uV) + if (*min_uV > *max_uV) { + rdev_err(rdev, "unsupportable voltage range: %d-%duV\n", + min_uV, max_uV); return -EINVAL; + } return 0; } @@ -205,8 +208,11 @@ static int regulator_check_current_limit(struct regulator_dev *rdev, if (*min_uA < rdev->constraints->min_uA) *min_uA = rdev->constraints->min_uA; - if (*min_uA > *max_uA) + if (*min_uA > *max_uA) { + rdev_err(rdev, "unsupportable current range: %d-%duA\n", + min_uA, max_uA); return -EINVAL; + } return 0; } @@ -221,6 +227,7 @@ static int regulator_mode_constrain(struct regulator_dev *rdev, int *mode) case REGULATOR_MODE_STANDBY: break; default: + rdev_err(rdev, "invalid mode %x specified\n", *mode); return -EINVAL; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2] ASoC: dapm - Add API call to query the widgets of valid DAPM paths.
by Liam Girdwood 22 Jul '11

22 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets and return the path widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the Dynamic PCM core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. This modifies the existing is_connected_input_ep() and is_connected_output_ep() by adding a new parameter to store any valid widgets in a list. We also add some extra debug to provide further insight into the DAPM path and widget discovery. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++ sound/soc/soc-dapm.c | 184 +++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 178 insertions(+), 12 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..198997d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,11 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fbfcda0..4205a97 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -605,11 +605,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -638,15 +678,38 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { + if (path->weak) continue; if (path->walked) continue; - if (path->sink && path->connect) { + + if (path->sink) { + + dev_dbg(widget->dapm->dev," %c : %s -> %s -> %s\n", + path->sink && path->connect ? '*' : ' ', + widget->name, path->name, path->sink->name); + + if (!path->connect) + continue; + path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -657,7 +720,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -691,21 +755,117 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { + if (path->weak) continue; if (path->walked) continue; - if (path->source && path->connect) { + if (path->source) { + + dev_dbg(widget->dapm->dev," %c : %s <- %s <- %s\n", + path->source && path->connect ? '*' : ' ', + widget->name, path->name, path->source->name); + + if (!path->connect) + continue; + path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->source); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } return con; } +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + paths = is_connected_output_ep(root, list); + dev_dbg(dapm->dev, "Playback: found %d paths from %s\n", paths, root->name); + + return paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Capture: checking paths from %s\n",root->name); + paths = is_connected_input_ep(root, list); + dev_dbg(dapm->dev, "Capture: found %d paths from %s\n", paths, root->name); + + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial widget specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream) +{ + struct snd_soc_dapm_widget *w; + enum snd_soc_dapm_type type; + int paths; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + type = snd_soc_dapm_aif_in; + else + type = snd_soc_dapm_aif_out; + + /* get stream root widget AIF from stream string and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (w->id != type) + continue; + + if (strstr(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget for %s not found\n", stream_name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_walk(dapm); + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -732,9 +892,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -745,7 +905,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) int in; if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -759,7 +919,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) int out; if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1315,9 +1475,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", -- 1.7.4.1

2 3