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

[alsa-devel] softvol master has preference over virtual master in amixer
by Raymond Yau 21 Oct '12

21 Oct '12

does softvol plugin "master" has any preference over vmaster in amixer ? http://www.alsa-project.org/db/?f=44b26f5cd37f33bfef2871ff1fd250301bdb5364 control.32 { iface MIXER name 'Master Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.42 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } Card hw:0 'Intel'/'HDA Intel at 0xfeaf8000 irq 43' Mixer name: 'Realtek ALC887-VD' Components: 'HDA:10ec0887,10438445,00100302' Controls : 42 Simple ctrls : 22 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [191.00dB] [on] Front Right: Playback 255 [100%] [191.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] https://launchpadlibrarian.net/119782219/Symptom_PulseAudioLog.txt Oct 15 10:49:11 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:11 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:11 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:11 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument

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[alsa-devel] [PATCH 1/5] sound/isa/opti9xx/miro.c: eliminate possible double free
by Julia Lawall 21 Oct '12

21 Oct '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> snd_miro_probe is a static function that is only called twice in the file that defines it. At each call site, its argument is freed using snd_card_free. Thus, there is no need for snd_miro_probe to call snd_card_free on its argument on any of its error exit paths. Because snd_card_free both reads the fields of its argument and kfrees its argments, the results of the second snd_card_free should be unpredictable. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ identifier f,free,a; parameter list[n] ps; type T; expression e; @@ f(ps,T a,...) { ... when any when != a = e if(...) { ... free(a); ... return ...; } ... when any } @@ identifier r.f,r.free; expression x,a; expression list[r.n] xs; @@ * x = f(xs,a,...); if (...) { ... free(a); ... return ...; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not tested. sound/isa/opti9xx/miro.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 3d1afb6..4a7ff4e 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1286,7 +1286,6 @@ static int __devinit snd_miro_probe(struct snd_card *card) error = snd_card_miro_aci_detect(card, miro); if (error < 0) { - snd_card_free(card); snd_printk(KERN_ERR "unable to detect aci chip\n"); return -ENODEV; }

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[alsa-devel] [PATCH 1/1] Add USB audio quirk supporting Roland VG-99 advanced mode
by pete.leigh＠gmail.com 21 Oct '12

21 Oct '12

From: Pete Leigh <pete.leigh(a)gmail.com> Signed-off-by: Pete Leigh <pete.leigh(a)gmail.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..ffeabf8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1457,6 +1457,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Advanced mode of the Roland VG-99, with MIDI and 24-bit PCM at 44.1 + * kHz. In standard mode, the device has ID 0582:00b3, and offers + * 16-bit PCM at 44.1 kHz with no MIDI. + */ + USB_DEVICE(0x0582, 0x00b2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "VG-99", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* Roland SonicCell */ USB_DEVICE(0x0582, 0x00c2), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.11.7

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[alsa-devel] [PATCH 1/1] USB audio quirk for Roland VG-99 advanced mode
by pete.leigh＠gmail.com 21 Oct '12

21 Oct '12

From: Pete Leigh <pete.leigh(a)gmail.com> Without this quirk the VG-99 will work in standard mode (set under USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out but no midi, and is not recognised when set to advanced mode. After applying this, I can also use the VG-99 in advanced mode: 24 24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far untested. In standard mode, the device appears with ID 0x00b3, so the behaviour isn't affected by this quirk. Thanks to Clemens Ladisch for simplifying and correcting my initial attempt! Signed-off-by: Pete Leigh <pete.leigh(a)gmail.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..ffeabf8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1457,6 +1457,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Advanced mode of the Roland VG-99, with MIDI and 24-bit PCM at 44.1 + * kHz. In standard mode, the device has ID 0582:00b3, and offers + * 16-bit PCM at 44.1 kHz with no MIDI. + */ + USB_DEVICE(0x0582, 0x00b2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "VG-99", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* Roland SonicCell */ USB_DEVICE(0x0582, 0x00c2), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.11.7

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[alsa-devel] [PATCH - my module 1/1] USB audio quirk for Roland VG-99 advanced mode
by pete.leigh＠gmail.com 21 Oct '12

21 Oct '12

From: Pete Leigh <pete.leigh(a)gmail.com> Without this quirk the VG-99 will work in standard mode (set under USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out but no midi, and is not recognised when set to advanced mode. After applying this, I can also use the VG-99 in advanced mode: 24 24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far untested. In standard mode, the device appears with ID 0x00b3, so the behaviour isn't affected by this quirk. Thanks to Clemens Ladisch for simplifying and correcting my initial attempt! Signed-off-by: Pete Leigh <pete.leigh(a)gmail.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..ffeabf8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1457,6 +1457,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Advanced mode of the Roland VG-99, with MIDI and 24-bit PCM at 44.1 + * kHz. In standard mode, the device has ID 0582:00b3, and offers + * 16-bit PCM at 44.1 kHz with no MIDI. + */ + USB_DEVICE(0x0582, 0x00b2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "VG-99", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* Roland SonicCell */ USB_DEVICE(0x0582, 0x00c2), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.11.7

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[alsa-devel] [PATCH] ALSA: aloop - do not reschedule timer if deleted
by Omair Mohammed Abdullah 21 Oct '12

21 Oct '12

If the timer is deleted while the timer handler is running, it may get rescheduled and an unnecessary period elapsed will be sent. Add a flag to reschedule the timer only if it has not been stopped. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/drivers/aloop.c | 21 ++++++++++++++++++--- 1 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 0fe6d64..fec4e17 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -121,6 +121,7 @@ struct loopback_pcm { unsigned long last_jiffies; struct timer_list timer; spinlock_t timer_lock; + bool timer_stopped; }; static struct platform_device *devices[SNDRV_CARDS]; @@ -166,12 +167,11 @@ static inline unsigned int get_rate_shift(struct loopback_pcm *dpcm) return get_setup(dpcm)->rate_shift; } -static void loopback_timer_start(struct loopback_pcm *dpcm) +static void do_timer_start(struct loopback_pcm *dpcm) { unsigned long tick; unsigned int rate_shift = get_rate_shift(dpcm); - spin_lock(&dpcm->timer_lock); if (rate_shift != dpcm->pcm_rate_shift) { dpcm->pcm_rate_shift = rate_shift; dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); @@ -184,6 +184,13 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; dpcm->timer.expires = jiffies + tick; add_timer(&dpcm->timer); +} + +static inline void loopback_timer_start(struct loopback_pcm *dpcm) +{ + spin_lock(&dpcm->timer_lock); + dpcm->timer_stopped = false; + do_timer_start(dpcm); spin_unlock(&dpcm->timer_lock); } @@ -192,6 +199,7 @@ static inline void loopback_timer_stop(struct loopback_pcm *dpcm) spin_lock(&dpcm->timer_lock); del_timer(&dpcm->timer); dpcm->timer.expires = 0; + dpcm->timer_stopped = true; spin_unlock(&dpcm->timer_lock); } @@ -540,7 +548,13 @@ static void loopback_timer_function(unsigned long data) running = loopback_pos_update(dpcm->cable); if (running & (1 << dpcm->substream->stream)) { - loopback_timer_start(dpcm); + spin_lock(&dpcm->timer_lock); + if (dpcm->timer_stopped) { + spin_unlock(&dpcm->timer_lock); + return; + } + do_timer_start(dpcm); + spin_unlock(&dpcm->timer_lock); if (dpcm->period_update_pending) { dpcm->period_update_pending = 0; snd_pcm_period_elapsed(dpcm->substream); @@ -670,6 +684,7 @@ static int loopback_open(struct snd_pcm_substream *substream) } dpcm->loopback = loopback; dpcm->substream = substream; + dpcm->timer_stopped = true; setup_timer(&dpcm->timer, loopback_timer_function, (unsigned long)dpcm); spin_lock_init(&dpcm->timer_lock); -- 1.7.4.1

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[alsa-devel] [PATCH] ALSA: snd-usb: remove unused variable in init_pitch_v2()
by Wei Yongjun 21 Oct '12

21 Oct '12

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> The variable ep is initialized but never used otherwise, so remove the unused variable. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/usb/pcm.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 55e19e1..f77b87a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -171,11 +171,8 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[1]; - unsigned int ep; int err; - ep = get_endpoint(alts, 0)->bEndpointAddress; - data[0] = 1; if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,

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[alsa-devel] calling snd_ctl_notify from mixer ctl_put
by Ondrej Zary 21 Oct '12

21 Oct '12

Hello, I want to enable/disable some mixer controls based on state of another control (switching between limiter and AGC mode). So I (re)set the SNDRV_CTL_ELEM_ACCESS_INACTIVE flag in kctl->vd[0].access and then call snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &kctl->id); Everyting from ctl_put() mixer callback. When changing the control value in alsamixer, it crashes with: alsamixer: simple.c:948: snd_mixer_selem_get_enum_item: Assertion `(elem)->type == SND_MIXER_ELEM_SIMPLE' failed. When the snd_ctl_notify() call is commented-out, it does not crash. There is snd_ctl_activate_id() function in sound/core/control.c (not used by any driver, btw.) which does the same, protected by card->controls_rwsem. However, it locks up on that semaphore when called from the mixer callback. Oxygen/Xonar WM8776 driver is doing the same thing - I wonder if I'm doing something wrong or if that driver crashes alsamixer too? Is there any way to do this properly? -- Ondrej Zary

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Re: [alsa-devel] usb-audio: Reloop Play support (TI TUSB3200AC)
by Daniel Mack 20 Oct '12

20 Oct '12

On 20.10.2012 16:24, Didier Villevalois wrote: > OK this quirk works perfectly! (hooray!!!) > > /* Reloop Play */ > { > USB_DEVICE(0x200c, 0x100b), > .bInterfaceClass = USB_CLASS_PER_INTERFACE, > .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { > .vendor_name = "Reloop", > .product_name = "Play", > .ifnum = QUIRK_ANY_INTERFACE, > .type = QUIRK_COMPOSITE, > .data = &(const struct snd_usb_audio_quirk[]) { > { > .ifnum = 0, > .type = QUIRK_AUDIO_STANDARD_MIXER, > }, > { > .ifnum = 1, > .type = QUIRK_AUDIO_FIXED_ENDPOINT, > .data = &(const struct audioformat) { > .formats = SNDRV_PCM_FMTBIT_S24_3LE, > .channels = 4, > .iface = 1, > .altsetting = 1, > .altset_idx = 1, > .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, > .endpoint = 0x01, > .rates = SNDRV_PCM_RATE_44100 | > SNDRV_PCM_RATE_48000, > .rate_min = 44100, > .rate_max = 48000, > .nr_rates = 2, > .rate_table = (unsigned int[]) { > 44100, 48000 > } > } > }, > { > .ifnum = -1 > } > } > } > }, > > Does anyone have additional comments to do before I make a patch ? Let's ask Clemens - the quirks master - for comments :) > Also are there some special ALSA rules or straight signed-off > git-formated patches over the kernel tree are OK ? That's ok. The patch should apply on top of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git though. > Thanks again for your help! I'm really happy to have a working > device! :D Good work! I just really wonder what that updated firmware is all about ... Daniel

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[alsa-devel] [PATCH 0/9] Fixes for RME MADI cards
by Adrian Knoth 20 Oct '12

20 Oct '12

Hi! I've spent another afternoon at IOSONO to fix a series of bugs in the hdspm driver. Andre Schramm and me have tested almost all patches on a box with two PCIe MADI cards and one PCI MADI card. The DDS/Varispeed patch has only been tested on my RayDAT, the sync_in detection for AES32 has not been tested at all, though the documentation is pretty clear that the previous bit position was wrong. If there are AES(32) owners with two or more cards out there, please test and report back. The typo and the coding style patches have not been tested, since they're trivial. Cheers Adrian Knoth (9): ALSA: hdspm - Allow DDS/Varispeed to be set from userspace ALSA: hdspm - Report external rate in slave mode on PCI MADI ALSA: hdspm - Fix sync check reporting on all RME HDSPM cards ALSA: hdspm - Fix reported autosync_sample_rate ALSA: hdspm - Also report autosync_sample_rate on MADI and MADIface ALSA: hdspm - Fix sync_in reporting on RME MADI cards ALSA: hdspm - Fix sync_in detection on AES/AES32 ALSA: hdspm - Fix typo in kcontrol element on RME MADI cards ALSA: hdspm - Fix coding style in CTL_ELEM macros sound/pci/rme9652/hdspm.c | 232 ++++++++++++++++++++++++++------------------- 1 file changed, 137 insertions(+), 95 deletions(-) -- 1.7.10.4

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