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

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Set constraint for 24 msb
by Peter Ujfalusi 15 Jan '12

15 Jan '12

OMAP McPDM supports 24 msb in the 32 bit sample. Make sure that applications get this information. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 0e25df4..50d758a 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -265,6 +265,8 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, mutex_lock(&mcpdm->mutex); + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + if (!dai->active) { /* Enable watch dog for ES above ES 1.0 to avoid saturation */ if (omap_rev() != OMAP4430_REV_ES1_0) { -- 1.7.8.3

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[alsa-devel] [PATCH] alsamixer: Confirm that a drastic volume change is intentional.
by Kusanagi Kouichi 14 Jan '12

14 Jan '12

Home key has been disabled because it might at least surprise a user unless volume is already nearly full. Not only home but 9, 8 and so on might also surprise a user. So instead of disableing key, show confirmation dialog for drastic volume change. Signed-off-by: Kusanagi Kouichi <slash(a)ac.auone-net.jp> --- alsamixer/mixer_widget.c | 52 ++++++++++++++++++++++++++++++++++++++------ alsamixer/volume_mapping.c | 3 ++ 2 files changed, 48 insertions(+), 7 deletions(-) diff --git a/alsamixer/mixer_widget.c b/alsamixer/mixer_widget.c index caaf777..d01a6ec 100644 --- a/alsamixer/mixer_widget.c +++ b/alsamixer/mixer_widget.c @@ -207,6 +207,7 @@ static void show_help(void) _("Up/Down Change volume"), _("+ - Change volume"), _("Page Up/Dn Change volume in big steps"), + _("Home Set volume to 100%"), _("End Set volume to 0%"), _("0-9 Set volume to 0%-90%"), _("Q W E Increase left/both/right volumes"), @@ -305,9 +306,10 @@ static void change_enum_relative(struct control *control, int delta) snd_mixer_selem_set_enum_item(control->elem, i, new_index); } -static void change_volume_to_percent(struct control *control, int value, unsigned int channels) +static int change_volume_to_percent(struct control *control, int value, unsigned int channels) { int (*set_func)(snd_mixer_elem_t *, snd_mixer_selem_channel_id_t, double, int); + int err = 0; if (!(control->flags & HAS_VOLUME_1)) channels = LEFT; @@ -316,9 +318,10 @@ static void change_volume_to_percent(struct control *control, int value, unsigne else set_func = set_normalized_capture_volume; if (channels & LEFT) - set_func(control->elem, control->volume_channels[0], value / 100.0, 0); - if (channels & RIGHT) - set_func(control->elem, control->volume_channels[1], value / 100.0, 0); + err = set_func(control->elem, control->volume_channels[0], value / 100.0, 0); + if (channels & RIGHT && err <= 0) + err = set_func(control->elem, control->volume_channels[1], value / 100.0, 0); + return err; } static double clamp_volume(double v) @@ -371,7 +374,44 @@ static void change_control_to_percent(int value, unsigned int channels) if (control->flags & TYPE_ENUM) change_enum_to_percent(control, value); else - change_volume_to_percent(control, value, channels); + if (change_volume_to_percent(control, -value, channels)) { + char buf[50]; + const char * const text[] = { buf, " (Y/n)" }; + const struct widget *active_widget; + + snprintf(buf, sizeof buf, "Do you really change volume to %d%%?", value); + show_text(text, 2, ""); + active_widget = get_active_widget(); + wtimeout(active_widget->window, -1); + while (1) { + switch (wgetch(active_widget->window)) { +#ifdef KEY_RESIZE + case KEY_RESIZE: + window_size_changed(); + continue; +#endif + case 'Y': + case 'y': + case '\n': + case KEY_ENTER: + change_volume_to_percent(control, value, channels); + break; + case 'N': + case 'n': + case 27: + case KEY_CANCEL: + case KEY_CLOSE: + case KEY_EXIT: + break; + default: + continue; + } + break; + } + active_widget->handle_key(KEY_ENTER); + } else + change_volume_to_percent(control, value, channels); + display_controls(); } @@ -536,12 +576,10 @@ static void on_handle_key(int key) case KEY_NPAGE: change_control_relative(-5, LEFT | RIGHT); break; -#if 0 case KEY_BEG: case KEY_HOME: change_control_to_percent(100, LEFT | RIGHT); break; -#endif case KEY_LL: case KEY_END: change_control_to_percent(0, LEFT | RIGHT); diff --git a/alsamixer/volume_mapping.c b/alsamixer/volume_mapping.c index 1c0d7c4..bd160ed 100644 --- a/alsamixer/volume_mapping.c +++ b/alsamixer/volume_mapping.c @@ -133,6 +133,9 @@ static int set_normalized_volume(snd_mixer_elem_t *elem, double min_norm; int err; + if (volume < 0) + return -volume - get_normalized_volume(elem, channel, ctl_dir) >= 0.5; + err = get_dB_range[ctl_dir](elem, &min, &max); if (err < 0 || min >= max) { err = get_raw_range[ctl_dir](elem, &min, &max); -- 1.7.8.3

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[alsa-devel] [PATCH 0/2] cuse: implement mmap/munmap
by Miklos Szeredi 13 Jan '12

13 Jan '12

These patches implement mmap support for CUSE (character device in userspace). The immediate application for this is to get old games requiring OSS/mmap support to work with OSS Proxy, written by Tejun. So firstly this is a call for testers; please try this out and report any problems. Three pieces are needed: kernel: git://git.kernel.org/pub/scm/linux/kernel/git/mszeredi/fuse.git for-next fuse: git://fuse.git.sourceforge.net/gitroot/fuse/fuse master osspd: git://fuse.git.sourceforge.net/gitroot/fuse/osspd master Comments about the interface and implementation in general (not related to OSS proxy) are very welcome as well. Thanks, Miklos --- Miklos Szeredi (1): fuse: create fuse_conn_operations Tejun Heo (1): cuse: implement memory mapping --- fs/fuse/cuse.c | 426 +++++++++++++++++++++++++++++++++++++++++++++++++- fs/fuse/dev.c | 155 ++----------------- fs/fuse/fuse_i.h | 35 ++++- fs/fuse/inode.c | 167 +++++++++++++++++++- include/linux/fuse.h | 25 +++ 5 files changed, 656 insertions(+), 152 deletions(-)

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[alsa-devel] Read and write different buffer sizes
by Elias Kokkinis 13 Jan '12

13 Jan '12

Hello all, I would like to ask if it is possible in ALSA to define different buffer sizes in the input and output streams. I am using PCM sound and I want to read 2048 samples, do some processing involving overlap-add and then keep 1024 of them and put them to the output. And so on... So I want to have the same sample rate and everything but different buffer sizes. Is this possible? I have tried an implementation of this, but I think the output for some reason still outputs 2048 samples and the audio is choppy, like having 1024 samples of audio and 1024 of silence. I don't know if I am making myself clear here, maybe it's the 10 hours of work up to now, but anyway any help/comments would be greatly appreciated. Elias Kokkinis

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Re: [alsa-devel] Trying to understand alsa
by Andrew Eikum 13 Jan '12

13 Jan '12

On Thu, Jan 12, 2012 at 11:16:47PM +0000, Jonathan Andrews wrote: > On Thu, 2012-01-12 at 16:12 -0600, Andrew Eikum wrote: > > On Thu, Jan 12, 2012 at 09:44:11PM +0000, Jonathan Andrews wrote: > > > I have an application that works using 512 sample packets of 22050Hz 16 > > > bit mono audio. The 'receiver' takes many audio streams from a network > > > via UDP, at the moment it pipes them into pulse. > > > > > > Can alsa buffer audio. At the moment every time I and set an audio > > > buffer size I get a negative response from > > > snd_pcm_hw_params_set_buffer_size . I'm somewhat confused about the > > > units alsa uses ... > > > > > > > You don't want to over-specify your requirements. You require a buffer > > size of "at least" 3 * 512 frames. So use set_buffer_size_min(). > > I cant find any reference to "set_buffer_size_min" in the ALSA API > documentation I have or the link you provided ? > snd_pcm_hw_params_set_buffer_time_near() is the closest and that seems > to take an argument in useconds. > ? > http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html > The doxygen docs are split strangely. The function is documented here: http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___h_w___params.… I find keeping the alsa-lib source on hand for quick grepping is often helpful. Andrew

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Re: [alsa-devel] Dell N5110 HDA External Speaker is mute
by Takashi Iwai 13 Jan '12

13 Jan '12

At Thu, 12 Jan 2012 11:51:12 +0000, Cefn Hoile wrote: > > Thanks, Takashi. > > I don't know what tools and invocations to use test out the GPIO > values you describe. I think you're saying I shouldn't bother toggling > things through HDA Analyzer as there's nothing obviously wrong in that > part of the configuration (as reported by alsa-info.sh which is > attached as requested). > > I noticed in HD-Audio.txt a section which mentions a similar problem > and IDT codecs specifically (below). Is there an IRC or mailing list > where people could talk me through the basics of trying different > GPIO/EAPD, Power Enable or Pin Sense manipulations. I wouldn't know > how, yet. If I succeed I'll report back. > > Hopefully this will be worth it, as I think the N5110 is a pretty > popular model with a common card, and probably a bunch of Dell > hardware could get fixed if it could be figured out. You can change the GPIO setup via hda-verb on the fly. For example, to set GPIO bit 0, try hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x01 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 where the output direction is assumed, and only GPIO 0 is set. For GPIO bit 1, hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x02 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 and 0x04 for GPIO bit 2. Your device have up to three GPIO bits. If none of above doesn't any change, try to set DATA 0 instead of 1, and/or DIR 0 instead of 1. Takashi > QUOTING FROM HD-Audio.txt >>>>>>> > > If you are masochistic enough to debug the driver problem, note the > following: > > - The speaker (and the headphone, too) output often requires the > external amplifier. This can be set usually via EAPD verb or a > certain GPIO. If the codec pin supports EAPD, you have a better > chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly > it's either GPIO0 or GPIO1) may turn on/off EAPD. > - Some Realtek codecs require special vendor-specific coefficients to > turn on the amplifier. See patch_realtek.c. > - IDT codecs may have extra power-enable/disable controls on each > analog pin. See patch_sigmatel.c. > - Very rare but some devices don't accept the pin-detection verb until > triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the > codec-communication stall. Some examples are found in > patch_realtek.c. > > <<<<<<<< > > On 12 January 2012 11:25, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 12 Jan 2012 10:56:24 +0000, > > Cefn Hoile wrote: > >> > >> In case anything has changed in my config since the original info run, > >> here is the same script run right now... > >> > >> http://www.alsa-project.org/db/?f=d0a1fd23339878916b611c88e64e4868b45445ce > > > > Could you rather attach? Then it'll be archived and searchable. > > > > Anyway, I don't see any obvious problem in the output. > > If the pin assignment is correct, the routing and amp setups should be > > OK with the current status. > > > > Maybe it's a missing GPIO setup. Try to adjust GPIO mask, direction > > and value bits. > > > > > > Takashi

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[alsa-devel] [PATCH] Alsa-utils: Fix typo causing some volumes not to be initialized
by David Henningsson 13 Jan '12

13 Jan '12

When I was reading the source code, I stumbled over this typo causing Center, LFE, and Surround speakers not to be initialized. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- alsactl/init/default | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/alsactl/init/default b/alsactl/init/default index af64cb1..5fb0113 100644 --- a/alsactl/init/default +++ b/alsactl/init/default @@ -124,7 +124,7 @@ CTL{name}="Front Speaker Playback Switch",CTL{do_search}=="1", \ CTL{values}="on" CTL{reset}="mixer" -CTL{name}="Front Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" +CTL{name}="Surround Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" # if master volume control is present, turn speaker volume to max ENV{has_pmaster_vol}=="true",CTL{write}=="0dB",GOTO="" ENV{has_pmaster_vol}=="true",CTL{write}=="100%",GOTO="" @@ -134,7 +134,7 @@ CTL{name}="Surround Speaker Playback Switch",CTL{do_search}=="1", \ CTL{values}="on" CTL{reset}="mixer" -CTL{name}="Front Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" +CTL{name}="Center Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" # if master volume control is present, turn speaker volume to max ENV{has_pmaster_vol}=="true",CTL{write}=="0dB",GOTO="" ENV{has_pmaster_vol}=="true",CTL{write}=="100%",GOTO="" @@ -144,7 +144,7 @@ CTL{name}="Center Speaker Playback Switch",CTL{do_search}=="1", \ CTL{values}="on" CTL{reset}="mixer" -CTL{name}="Front Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" +CTL{name}="LFE Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" # if master volume control is present, turn speaker volume to max ENV{has_pmaster_vol}=="true",CTL{write}=="0dB",GOTO="" ENV{has_pmaster_vol}=="true",CTL{write}=="100%",GOTO="" -- 1.7.5.4

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[alsa-devel] [PATCH] ALSA: HDA: Use LPIB position fix for Macbook Pro 7, 1
by David Henningsson 13 Jan '12

13 Jan '12

Several users have reported "choppy" audio under the 3.2 kernel, and that changing position_fix to 1 has resolved their problem. The chip is an nVidia Corporation MCP89 High Definition Audio, [10de:0d94] (rev a2). Cc: stable(a)kernel.org (v3.2+) BugLink: https://bugs.launchpad.net/bugs/909419 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bd447b6..2e1c063 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2500,6 +2500,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS 1101HA", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), + SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), -- 1.7.5.4

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[alsa-devel] duplicated test in sound/soc/codecs/sgtl5000.c
by Julia Lawall 13 Jan '12

13 Jan '12

There is a duplicated test in the function sgtl5000_restore_regs in the file sound/soc/codecs/sgtl5000.c: /* this regs depends on the others */ if (reg == SGTL5000_CHIP_ANA_POWER || reg == SGTL5000_CHIP_CLK_CTRL || reg == SGTL5000_CHIP_LINREG_CTRL || reg == SGTL5000_CHIP_LINE_OUT_CTRL || reg == SGTL5000_CHIP_CLK_CTRL) continue; I don't know if some constant other than SGTL5000_CHIP_CLK_CTRL should be tested, or if the second test should just be deleted. julia

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[alsa-devel] Todays GIT system of alsa
by Donn Washburn 13 Jan '12

13 Jan '12

Hey Group; I wish to let you know that todays GIT set of files fixed openSuSEs 11.4 and 12.1 problem that I have been having. That was a loss of pots/adjustments and mixed mute controls on kmix, gmixer and pavucontrol. All of mixers work. The system required snd-hda-intel module. The system is a ASUS M4A87TD Mother board and sound VIA VT1818 High Definition Audio 8 Channel Codec. lsmod | less snd_pcm_oss 42358 0 snd_mixer_oss 18292 1 snd_pcm_oss snd_seq 49875 0 snd_seq_device 14530 1 snd_seq snd_hda_intel 29539 10 snd_hda_codec 105907 3 snd_hda_codec_hdmi,snd_hda_codec_via,snd_hda_intel snd_hwdep 13604 1 snd_hda_codec snd_pcm 81432 7 snd_pcm_oss,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_timer 25525 5 snd_seq,snd_pcm snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_codec_hdmi, snd_hda-codec_via,snd_hda_soundcore 15165 1 snd Thanks again -- 73 de Donn Washburn 307 Savoy Street Email:" n5xwb(a)comcast.net " Sugar Land, TX 77478 LL# 1.281.242.3256 Ham Callsign N5XWB HAMs : " n5xwb(a)arrl.net " VoIP via Skype:n5xwbg BMWMOA #:4146 Ambassador " http://counter.li.org " #279316

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