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

[alsa-devel] [PATCH] sound/isa/sc6000.c: add missing sc6000_setup_board
by Julia Lawall 19 Apr '12

19 Apr '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> There seems to be no interaction between *port and the requested region, so the clearing of *vport should take place if the request_region fails. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not tested. sc6000_setup_board seems rather complicated, so my reasoning could be completely wrong. sound/isa/sc6000.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index d97d0f3..57679cb 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -594,7 +594,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) snd_printk(KERN_ERR PFX "I/O port cannot be iomaped.\n"); err = -EBUSY; - goto err_unmap1; + goto err_unmap0; } /* to make it marked as used */ @@ -678,9 +678,10 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) return 0; err_unmap2: - sc6000_setup_board(*vport, 0); release_region(mss_port[dev], 4); err_unmap1: + sc6000_setup_board(*vport, 0); +err_unmap0: release_region(port[dev], 0x10); err_exit: snd_card_free(card);

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[alsa-devel] [PATCH 0/5] ASoC: sh: fsi-codec
by Kuninori Morimoto 19 Apr '12

19 Apr '12

Hi Mark, Liam Cc: Paul, Rafael These patches adds fsi-codec support for SuperH board. Kuninori Morimoto (5): ASoC: fsi: add fsi-codec card support ASoC: fsi: remove fsi-ak4642 card driver ASoC: fsi: remove fsi-hdmi card driver ASoC: fsi: remove fsi-da7210 card driver ASoC: fsi: add FSI-WM8978 support Current FSI-xxx cards are using fsi-xxx.c driver. But these were very similar. To reduce useless code, #1 adds common card driver as "fsi-codec", and #2-#4 switch to use it. And #5 adds new card support Please note that these patches are based on latest linus/master. I guess current mark/for-3.4 will get conflict Best regards --- Kuninori Morimoto

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[alsa-devel] ALSA: Updating Sample Rates in sound/core/pcm_native.c
by Jeremy Torres 19 Apr '12

19 Apr '12

Hello, I have updated pcm.h in the linux kernel to allow for new sample rates. For example, I have added 128000 and 144000 sample rates. Per the documentation in pcm.h, I also updated pcm_native.c (the rates table). However when I try and use the "default" device for the sample rates: 1. For 128000: plug conversion occurs for 128K, but the slave is configured to 96K; 2. For 144000: plug conversion is attempted at the 176400 rate; Note that: a) I have snd_pcm_hw_resample set to FALSE in my application; however, I've been using aplay/arecord to test as well with repeatable results. b) I do not have an asoundrc setup for plug conversion, etc. I only have the default /usr/share/alsa/alsa.conf; c) When I try the the device "hw:0,0", I get a (proper) warning that the sample rate requested is not being honored (i.e., 96K instead of 128K). I have placed debug in pcm_lib.c, and pcm_native.c and my sample rate constraints are set. My understanding is if I define a specific rate and it can be matched in the "rates" table, it should be honored. Environment Info: ALSA lib/driver version 1.0.20 SOC Driver Any help would be greatly appreciated! Regards, Jeremy

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Re: [alsa-devel] No sound on Quanta KN1 with kernel 3.4
by Takashi Iwai 19 Apr '12

19 Apr '12

At Sat, 14 Apr 2012 22:52:09 +0200, Uros Vampl wrote: > > On 14.04.12 21:25, Takashi Iwai wrote: > > At Sat, 14 Apr 2012 19:10:30 +0200, > > Uros Vampl wrote: > > > > > > Hi, > > > > > > as I wrote on the -users list, I don't have sound anymore in kernel 3.4. > > > I was told to post the output of alsa-info.sh here, so here I am. The > > > requested output is attached. > > > > Could you give the alsa-info.sh output on the working kernel? > > In that way, it becomes easier to spot out what caused the problem. > > Attached. Could you try the patch below? Also, which output is missing? Is it a speaker output or a line-out jack? Basically it's a bug of BIOS that doesn't give the proper association number for the corresponding pin 0x0f. The patch is just to make less restrictive. But, if it's a laptop, the corresponding pin is likely a speaker output, so it'd be better to fix the pin-default value itself via pinfix table. thanks, Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7a8fcc4..83f0b13 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4957,8 +4957,6 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, if (!(wid_caps & AC_WCAP_STEREO)) if (!cfg->mono_out_pin) cfg->mono_out_pin = nid; - if (!assoc) - continue; if (!assoc_line_out) assoc_line_out = assoc; else if (assoc_line_out != assoc)

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[alsa-devel] [PATCH v2] ASoC: dapm - Add API call to query valid DAPM paths.
by Liam Girdwood 18 Apr '12

18 Apr '12

In preparation for ASoC DSP support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink 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 DSP core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- Changes since V1:- Now uses new DAPM mutex. Tracepoint used for path debug. include/sound/soc-dapm.h | 5 ++ include/trace/events/asoc.h | 80 ++++++++++++++++++++++++++++ sound/soc/soc-dapm.c | 122 +++++++++++++++++++++++++++++++++++++++---- 3 files changed, 197 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index bea0c86..e3833d9 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -321,6 +321,7 @@ struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; struct regulator; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -403,6 +404,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); +/* dapm path query */ +int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ab26f8a..6d8efb1 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -7,6 +7,8 @@ #include <linux/ktime.h> #include <linux/tracepoint.h> +#define DAPM_DIRECT "(direct)" + struct snd_soc_jack; struct snd_soc_codec; struct snd_soc_platform; @@ -241,6 +243,84 @@ TRACE_EVENT(snd_soc_dapm_walk_done, (int)__entry->path_checks, (int)__entry->neighbour_checks) ); +TRACE_EVENT(snd_soc_dapm_output_path, + + TP_PROTO(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_path *path), + + TP_ARGS(widget, path), + + TP_STRUCT__entry( + __string( wname, widget->name ) + __string( pname, path->name ? path->name : DAPM_DIRECT) + __string( psname, path->sink->name ) + __field( int, path_sink ) + __field( int, path_connect ) + ), + + TP_fast_assign( + __assign_str(wname, widget->name); + __assign_str(pname, path->name ? path->name : DAPM_DIRECT); + __assign_str(psname, path->sink->name); + __entry->path_connect = path->connect; + __entry->path_sink = (int)path->sink; + ), + + TP_printk("%c%s -> %s -> %s\n", + (int) __entry->path_sink && + (int) __entry->path_connect ? '*' : ' ', + __get_str(wname), __get_str(pname), __get_str(psname)) +); + +TRACE_EVENT(snd_soc_dapm_input_path, + + TP_PROTO(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_path *path), + + TP_ARGS(widget, path), + + TP_STRUCT__entry( + __string( wname, widget->name ) + __string( pname, path->name ? path->name : DAPM_DIRECT) + __string( psname, path->source->name ) + __field( int, path_source ) + __field( int, path_connect ) + ), + + TP_fast_assign( + __assign_str(wname, widget->name); + __assign_str(pname, path->name ? path->name : DAPM_DIRECT); + __assign_str(psname, path->source->name); + __entry->path_connect = path->connect; + __entry->path_source = (int)path->source; + ), + + TP_printk("%c%s <- %s <- %s\n", + (int) __entry->path_source && + (int) __entry->path_connect ? '*' : ' ', + __get_str(wname), __get_str(pname), __get_str(psname)) +); + +TRACE_EVENT(snd_soc_dapm_connected, + + TP_PROTO(int paths, int stream), + + TP_ARGS(paths, stream), + + TP_STRUCT__entry( + __field( int, paths ) + __field( int, stream ) + ), + + TP_fast_assign( + __entry->paths = paths; + __entry->stream = stream; + ), + + TP_printk("%s: found %d paths\n", + __entry->stream ? "capture" : "playback", __entry->paths) +); + TRACE_EVENT(snd_soc_jack_irq, TP_PROTO(const char *name), diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d7ee73a..214323f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -705,11 +705,51 @@ 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; @@ -765,9 +805,23 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + trace_snd_soc_dapm_output_path(widget, path); + if (path->sink && path->connect) { 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); } } @@ -780,7 +834,8 @@ 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; @@ -848,9 +903,23 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + trace_snd_soc_dapm_input_path(widget, path); + if (path->source && path->connect) { 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->sink); + 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); } } @@ -859,6 +928,39 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dai: the soc DAI. + * @stream: stream direction. + * @list: list of active widgets for this stream. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial stream 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_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list) +{ + struct snd_soc_card *card = dai->card; + int paths; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm_reset(card); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = is_connected_output_ep(dai->playback_widget, list); + else + paths = is_connected_input_ep(dai->playback_widget, list); + + trace_snd_soc_dapm_connected(paths, stream); + dapm_clear_walk(&card->dapm); + mutex_unlock(&card->dapm_mutex); + + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -915,9 +1017,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - 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; } @@ -940,7 +1042,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); 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 { @@ -956,7 +1058,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); 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 { @@ -1558,9 +1660,9 @@ 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%s in %d out %d", -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: samsung: Hook up AIF2 to the CODEC on Littlemill
by Mark Brown 18 Apr '12

18 Apr '12

Connect the WM1250-EV1 baseband simulator on Littlemill systems up to the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider range of use cases to be represented. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/littlemill.c | 82 ++++++++++++++++++++++++++++++++++------ 1 file changed, 70 insertions(+), 12 deletions(-) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index e741685..71bff63 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -23,10 +23,10 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; int ret; - if (dapm->dev != codec_dai->dev) + if (dapm->dev != aif1_dai->dev) return 0; switch (level) { @@ -36,7 +36,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, * then do so now, otherwise these are noops. */ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK2, 32768, sample_rate * 512); if (ret < 0) { @@ -44,7 +44,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, return ret; } - ret = snd_soc_dai_set_sysclk(codec_dai, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_FLL1, sample_rate * 512, SND_SOC_CLOCK_IN); @@ -66,25 +66,58 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; int ret; - if (dapm->dev != codec_dai->dev) + if (dapm->dev != aif1_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) { - pr_err("Failed to switch away from FLL: %d\n", ret); + pr_err("Failed to switch away from FLL2: %d\n", ret); return ret; } - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, 0, 0, 0); if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); + pr_err("Failed to stop FLL2: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL1: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL1: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_PREPARE: + ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, + WM8994_FLL_SRC_BCLK, 64 * 8000, + 8000 * 256); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_FLL2, + 8000 * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); return ret; } break; @@ -131,6 +164,14 @@ static struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + static struct snd_soc_dai_link littlemill_dai[] = { { .name = "CPU", @@ -143,6 +184,17 @@ static struct snd_soc_dai_link littlemill_dai[] = { | SND_SOC_DAIFMT_CBM_CFM, .ops = &littlemill_ops, }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm8994-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, }; static struct snd_soc_dapm_widget widgets[] = { @@ -169,10 +221,16 @@ static struct snd_soc_jack littlemill_headset; static int littlemill_late_probe(struct snd_soc_card *card) { struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) return ret; -- 1.7.10

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[alsa-devel] [PATCH] configure.in: look for ncurses pkg-config first
by Mike Frysinger 18 Apr '12

18 Apr '12

Newer ncurses installs pkg-config files, and these are much easier to manage when cross-compiling, so default to searching for those. If they can't be found, we fall back to the existing xxx-config. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- configure.in | 64 +++++++++++++++++++++++++++++++++++++-------------------- 1 files changed, 41 insertions(+), 23 deletions(-) diff --git a/configure.in b/configure.in index 2208a95..9404df0 100644 --- a/configure.in +++ b/configure.in @@ -29,6 +29,7 @@ AC_PROG_INSTALL AC_PROG_MKDIR_P AC_PROG_LN_S AC_PROG_SED +PKG_PROG_PKG_CONFIG AM_PATH_ALSA(1.0.24) AC_CHECK_FUNC([snd_ctl_elem_add_enumerated], , [AC_ERROR([No user enum control support in alsa-lib])]) @@ -130,34 +131,52 @@ if test x$alsamixer = xtrue; then NCURSESLIBSUFFIX="" CURSES_NLS="no" if test "$curseslib" = "ncursesw" -o $ "$curseslib" = "auto" -a "$USE_NLS" = "yes" $; then - AC_CHECK_PROG([ncursesw5_config], [ncursesw5-config], [yes]) - if test "$ncursesw5_config" = "yes"; then - CURSESINC="<ncurses.h>" - CURSESLIB=`ncursesw5-config --libs` - CURSESLIBDIR=`ncursesw5-config --libdir` - CURSES_CFLAGS=`ncursesw5-config --cflags` - curseslib="ncursesw" - else - AC_CHECK_LIB(ncursesw, initscr, - [ CURSESINC='<ncurses.h>'; CURSESLIB='-lncursesw'; curseslib="ncursesw"]) - fi + dnl First try out pkg-config, then fall back to old config scripts. + PKG_CHECK_MODULES([NCURSESW], [ncursesw], [ + CURSESINC="<ncurses.h>" + CURSESLIB="${NCURSESW_LIBS}" + CURSESLIBDIR= + CURSES_CFLAGS="${NCURSESW_CFLAGS}" + curseslib="ncursesw" + ], [ + AC_CHECK_PROG([ncursesw5_config], [ncursesw5-config], [yes]) + if test "$ncursesw5_config" = "yes"; then + CURSESINC="<ncurses.h>" + CURSESLIB=`ncursesw5-config --libs` + CURSESLIBDIR=`ncursesw5-config --libdir` + CURSES_CFLAGS=`ncursesw5-config --cflags` + curseslib="ncursesw" + else + AC_CHECK_LIB(ncursesw, initscr, + [ CURSESINC='<ncurses.h>'; CURSESLIB='-lncursesw'; curseslib="ncursesw"]) + fi + ]) if test -n "$CURSESINC"; then NCURSESLIBSUFFIX="w" CURSES_NLS="yes" fi fi if test "$curseslib" = "ncurses" -o "$curseslib" = "auto"; then - AC_CHECK_PROG([ncurses5_config], [ncurses5-config], [yes]) - if test "$ncurses5_config" = "yes"; then - CURSESINC="<ncurses.h>" - CURSESLIB=`ncurses5-config --libs` - CURSESLIBDIR=`ncurses5-config --libdir` - CURSES_CFLAGS=`ncurses5-config --cflags` - curseslib="ncurses" - else - AC_CHECK_LIB(ncurses, initscr, - [ CURSESINC='<ncurses.h>'; CURSESLIB='-lncurses'; curseslib="ncurses"]) - fi + dnl First try out pkg-config, then fall back to old config scripts. + PKG_CHECK_MODULES([NCURSES], [ncurses], [ + CURSESINC="<ncurses.h>" + CURSESLIB="${NCURSES_LIBS}" + CURSESLIBDIR= + CURSES_CFLAGS="${NCURSES_CFLAGS}" + curseslib="ncurses" + ], [ + AC_CHECK_PROG([ncurses5_config], [ncurses5-config], [yes]) + if test "$ncurses5_config" = "yes"; then + CURSESINC="<ncurses.h>" + CURSESLIB=`ncurses5-config --libs` + CURSESLIBDIR=`ncurses5-config --libdir` + CURSES_CFLAGS=`ncurses5-config --cflags` + curseslib="ncurses" + else + AC_CHECK_LIB(ncurses, initscr, + [ CURSESINC='<ncurses.h>'; CURSESLIB='-lncurses'; curseslib="ncurses"]) + fi + ]) fi if test "$curseslib" = "curses" -o "$curseslib" = "auto"; then AC_CHECK_LIB(curses, initscr, @@ -287,7 +306,6 @@ SAVE_UTIL_VERSION AC_SUBST(LIBRT) dnl Check for systemd -PKG_PROG_PKG_CONFIG AC_ARG_WITH([systemdsystemunitdir], AS_HELP_STRING([--with-systemdsystemunitdir=DIR], [Directory for systemd service files]), [], [with_systemdsystemunitdir=$($PKG_CONFIG --variable=systemdsystemunitdir systemd)]) -- 1.7.8.5

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[alsa-devel] [PATCH] configure.in: use AS_HELP_STRING everywhere
by Mike Frysinger 18 Apr '12

18 Apr '12

This macro takes care of spacing for us. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- configure.in | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/configure.in b/configure.in index 9404df0..e7bb4ca 100644 --- a/configure.in +++ b/configure.in @@ -76,7 +76,7 @@ CURSES_CFLAGS="" alsamixer= if test "$have_mixer" = "yes"; then AC_ARG_ENABLE(alsamixer, - [ --disable-alsamixer Disable alsamixer compilation], + AS_HELP_STRING([--disable-alsamixer], [Disable alsamixer compilation]), [case "${enableval}" in yes) alsamixer=true ;; no) alsamixer=false ;; @@ -87,7 +87,7 @@ AM_CONDITIONAL(ALSAMIXER, test x$alsamixer = xtrue) dnl Disable alsaconf AC_ARG_ENABLE(alsaconf, - [ --disable-alsaconf Disable alsaconf packaging], + AS_HELP_STRING([--disable-alsaconf], [Disable alsaconf packaging]), [case "${enableval}" in yes) alsaconf=true ;; no) alsaconf=false ;; @@ -97,7 +97,7 @@ AM_CONDITIONAL(ALSACONF, test x$alsaconf = xtrue) dnl Disable alsaloop AC_ARG_ENABLE(alsaloop, - [ --disable-alsaloop Disable alsaloop packaging], + AS_HELP_STRING([--disable-alsaloop], [Disable alsaloop packaging]), [case "${enableval}" in yes) alsaloop=true ;; no) alsaloop=false ;; @@ -124,7 +124,7 @@ dnl Checks for header files. AC_HEADER_STDC if test x$alsamixer = xtrue; then AC_ARG_WITH(curses, - [ --with-curses libname Specify the curses library to use (default=auto)], + AS_HELP_STRING([--with-curses=libname], [Specify the curses library to use (default=auto)]), curseslib="$withval", curseslib="auto") CURSESLIBDIR="" @@ -283,7 +283,7 @@ mydatadir="$dir/alsa" AC_DEFINE_UNQUOTED(DATADIR, "$mydatadir", [directory containing alsa configuration]) AC_ARG_WITH(testsound, - [ --with-testsound=file give the path of test sound file for alsaconf], + AS_HELP_STRING([--with-testsound=file], [give the path of test sound file for alsaconf]), TESTSOUND="$withval", TESTSOUND="$dir/test.wav") AC_SUBST(TESTSOUND) -- 1.7.8.5

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[alsa-devel] Fast Track Ultra mixer - opinions anyone?
by Felix Homann 18 Apr '12

18 Apr '12

Hi, over the weekend I tried to prepare a "set-everything-to-zero-by-default" mixer patch. And failed... I could get the controls to be defaulted to zero by quirking the volume range, but then there would not have been any sound at all, even after adjusting the levels. So without diving any deeper in this I decided to put some work in the mixer controls. You can find a patch to play with attached to this mail. I've left a couple of commented quirks in mixer.c so you can see, where simple changes can be made. But working on this patch raised a couple of questions I would like hear your opinions on: ** Effects ** Essentially, with the attached patch you can control everything but the effect program. The switch should not be too hard to implement. But before I put any more work on it I would like to ask wether it makes sense at all. IMO, the effect section on the FTUs really sucks. I can't believe that anybody can really use it in a reasonable way. So should we really add another 40 controls to the mixer which can cause a lot of pain when misadjusted (i.e. anything else but 0)? For those reasons I would prefer not to put effect support in the kernel. What do you think? ** dB values ** Currently alsamixer does not report any dB changes when you change a control value. I don't know how to fix it. Anyone else? ** Naming scheme again ** Once again, is everyone pleased with the current naming scheme? Looking forward for your suggestions. Regards, Felix

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[alsa-devel] [RFC PATCH 0/4] snd-ice1712: Add Philips PSC724 Ultimate Edge
by Ondrej Zary 18 Apr '12

18 Apr '12

Hello, this is first attempt to add Philips PSC724 Ultimate Edge sound card support to snd-ice1712 driver. Unlike other snd-ice1712 subdrivers, the codec code is splitted into separate modules and thus reusable. Working: all analog outputs (front, rear, center+lfe, headphone) and inputs (front mic, rear mic, line, cd, aux), mixer, headphone jack detection Currently untested: SPDIF, real multichannel sounds, power management Buggy: switching AGC mode in mixer causes alsamixer to crash - looks like it does not like notifications about control activating/deactivating

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