September 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] ALSA: rawmidi: cleanup the get next midi device ioctl
by Dan Carpenter 09 Sep '10

09 Sep '10

I'm doing an audit to find integer overflows and my static checker complained that in the original code "device + 1" could overflow. The overflow is harmless, but it's still worth cleaning up. The other thing that I noticed is that if you pass in a device which is higher than SNDRV_RAWMIDI_DEVICES then it doesn't return an error code but just tells you that the next device is "device + 1". I have rewritten it to just return -EINVAL if you pass in a bogus value that's either too high or too low. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index eb68326..f944180 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -829,8 +829,12 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)argp)) return -EFAULT; + if (device < 0) + return -EINVAL; + if (device > SNDRV_RAWMIDI_DEVICES) + return -EINVAL; mutex_lock(&register_mutex); - device = device < 0 ? 0 : device + 1; + device++; while (device < SNDRV_RAWMIDI_DEVICES) { if (snd_rawmidi_search(card, device)) break;

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[alsa-devel] cmipci surround sound front-left plays on lfe
by Tom Corner 09 Sep '10

09 Sep '10

I've been going around and around on this. The surround sound 5.1 is playing fine except channel 0 (front-left) is playing on the woofer. I have a Terratec Aureon fun - CMI8738-MC6. I can redirect the output on 0 to right-front successfully using .asoundrc and "ttable.0.1 1". I have gotten desperate and have tested this card in Windows and was able to get it working right there. I have been using the file www_lynnemusic_com_surround_test.wma through mplayer and speaker-test to to test the setup. I am using pulseaudio, but I reconfigured my system without pulseaudio and get the same result. Any help would be much appreciated. Tom Corner

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[alsa-devel] USB Audio device syncpipe error
by 6275 kson 09 Sep '10

09 Sep '10

Hello, I am working on a usb audio device and I'm having trouble getting it to show up as an output device in Ubuntu's sound preferences. The device is working well with other OS's. My syslog shows this error. 'cannot submit syncpipe for urb 0, error -90: internal error' If someone could help me diagnose the source of the problem I would appreciate it very much. Here are some more details. The device operates at high speed but uses USB Audio Class 1.0. The output endpoint is Isochronous asynchronous and I am using a feedback endpoint with a 32ms polling interval. The device works with Ubuntu when running at full speed. (Input and Output show up in the sound preferences). At High Speed only the Input shows up in the sound preferences. _From my usb analyzer log I never see a request for the feedback endpoint when running at HS, while I do see the request at FS. Cheers,

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[alsa-devel] [PATCH] ALSA: HDA: Add fixup pins for Ideapad Y550
by David Henningsson 09 Sep '10

09 Sep '10

By adding the subwoofer as a speaker pin, it is treated correctly when auto-muting. I'm a little unsure of whether it should go to stable as well, because it needs yesterday's commit bb35febd to make sense (in fact, I developed parts of that patch in parallel). Any opinions? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH] ALSA: HDA: Add fixup pins for Ideapad Y550
by David Henningsson 09 Sep '10

09 Sep '10

By adding the subwoofer as a speaker pin, it is treated correctly when auto-muting. BugLink: https://launchpad.net/bugs/611803 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++++++++- 1 files changed, 25 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f11a9ca..0c25d22 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18963,6 +18963,26 @@ static void alc662_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC662_FIXUP_IDEAPAD, +}; + +static const struct alc_fixup alc662_fixups[] = { + [ALC662_FIXUP_IDEAPAD] = { + .pins = (const struct alc_pincfg[]) { + { 0x17, 0x99130112 }, /* subwoofer */ + { } + } + }, +}; + +static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), + {} +}; + + + static int patch_alc662(struct hda_codec *codec) { struct alc_spec *spec; @@ -18995,6 +19015,7 @@ static int patch_alc662(struct hda_codec *codec) } if (board_config == ALC662_AUTO) { + alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 1); /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) { @@ -19053,8 +19074,11 @@ static int patch_alc662(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; - if (board_config == ALC662_AUTO) + if (board_config == ALC662_AUTO) { spec->init_hook = alc662_auto_init; + alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 0); + } + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc662_loopbacks; -- 1.7.1 --------------090907070406080900090002--

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[PATCH] ALSA: hda - Clear left-over hp_pins in
by Takashi Iwai 08 Sep '10

08 Sep '10

snd_hda_parse_pin_def_config() In snd_hda_parse_def_config(), some unused values may remain in hp_pins[] array during the headphone-reassignment workaround. This patch clears the unused array members. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2980c27..bfdde7b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4558,6 +4558,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, memmove(sequences_hp + i, sequences_hp + i + 1, sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); } + memset(cfg->hp_pins + cfg->hp_outs, 0, + sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs)); } /* sort by sequence */ -- 1.7.2.2 ================ -- ~Nirbheek Chauhan Gentoo GNOME+Mozilla Team

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[alsa-devel] [PATCH] OMAP: McBSP: Fix static function warning
by G, Manjunath Kondaiah 08 Sep '10

08 Sep '10

This patch fixes sparse warning due non declaration of static function sound/soc/omap/omap-mcbsp.c:783:5: warning: symbol 'omap_mcbsp_st_info_volsw' was not declared. Should it be static? Signed-off-by: G, Manjunath Kondaiah <manjugk(a)ti.com> Cc: alsa-devel(a)alsa-project.org Cc: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Tony Lindgren <tony(a)atomide.com> Cc: Nishanth Menon <nm(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 86f2139..5768381 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -780,7 +780,7 @@ struct snd_soc_dai omap_mcbsp_dai[] = { EXPORT_SYMBOL_GPL(omap_mcbsp_dai); -int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, +static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct soc_mixer_control *mc = -- 1.7.0.4

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[alsa-devel] [PATCH 0/4] ASoC: rx1950-related fixes
by Vasily Khoruzhick 08 Sep '10

08 Sep '10

Some cosmetical fixes for rx1950 and uda1380 ASoC drivers

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[alsa-devel] [patch 0/3] ASoC: t5325 audio support
by Arnaud Patard 08 Sep '10

08 Sep '10

Hi, This patchset is the first try to get merged audio support for HP t5325 thinclient. This board is using a ALC5621 so the major part of this series is made of the ALC5623 driver. It's supporting ALC5621/22/23 but has been tested only with a ALC5621. Arnaud

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[alsa-devel] [patch] sound/oss: potential integer overflow
by Dan Carpenter 08 Sep '10

08 Sep '10

We don't want "pre_event_timeout" to be negative because that would result in a stack traces in dmesg when we schedule a negative timeout. In the original code "HZ * val" could overflow so I just moved the check for negative below the multiply. Also the code in snd_seq_oss_ioctl() deliberately set the timeout to -1 which is wrong. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index e85789e..de61295 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -1507,9 +1507,9 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a case SNDCTL_MIDI_PRETIME: if (get_user(val, p)) return -EFAULT; + val = (HZ * val) / 10; if (val < 0) val = 0; - val = (HZ * val) / 10; pre_event_timeout = val; break; diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c index 782b3b8..73acacd 100644 --- a/sound/oss/midibuf.c +++ b/sound/oss/midibuf.c @@ -380,9 +380,9 @@ int MIDIbuf_ioctl(int dev, struct file *file, case SNDCTL_MIDI_PRETIME: if (get_user(val, (int __user *)arg)) return -EFAULT; + val = (HZ * val) / 10; if (val < 0) val = 0; - val = (HZ * val) / 10; parms[dev].prech_timeout = val; return put_user(val, (int __user *)arg); diff --git a/sound/core/seq/oss/seq_oss_ioctl.c b/sound/core/seq/oss/seq_oss_ioctl.c index 5ac701c..ae2a39f 100644 --- a/sound/core/seq/oss/seq_oss_ioctl.c +++ b/sound/core/seq/oss/seq_oss_ioctl.c @@ -191,10 +191,9 @@ snd_seq_oss_ioctl(struct seq_oss_devinfo *dp, unsigned int cmd, unsigned long ca return 0; if (get_user(val, p)) return -EFAULT; - if (val <= 0) - val = -1; - else - val = (HZ * val) / 10; + val = (HZ * val) / 10; + if (val < 0) + val = 0; dp->readq->pre_event_timeout = val; return put_user(val, p) ? -EFAULT : 0;

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