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

Re: [alsa-devel] [PATCH - add Keith McMillen Stringport to quirks tableadd quirk for Keith McMillen StringPort
by Takashi Iwai 05 Aug '11

05 Aug '11

At Thu, 4 Aug 2011 12:25:56 -0700, msp(a)ucsd.edu wrote: > > From: Miller Puckette <msp(a)ucsd.edu> > > > Signed-off-by: Miller Puckette <msp(a)ucsd.edu> Applied. Thanks. Takashi > diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h > index dba0b7f..4d4f865 100644 > --- a/sound/usb/quirks-table.h > +++ b/sound/usb/quirks-table.h > @@ -2417,6 +2417,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), > .idProduct = 0x1020, > }, > > +/* KeithMcMillen Stringport */ > +{ > + USB_DEVICE(0x1f38, 0x0001), > + .bInterfaceClass = USB_CLASS_AUDIO, > +}, > + > /* Miditech devices */ > { > USB_DEVICE(0x4752, 0x0011), > -- > 1.7.2.2 >

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[PATCH] ALSA: snd-usb-caiaq: Correct offset fields of outbound iso_frame_desc
by Daniel Mack 05 Aug '11

05 Aug '11

This fixes faulty outbount packets in case the inbound packets received from the hardware are fragmented and contain bogus input iso frames. The bug has been there for ages, but for some strange reasons, it was only triggered by newer machines in 64bit mode. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: William Light <wrl(a)illest.net> Reported-by: Pedro Ribeiro <pedrib(a)gmail.com> Cc: stable(a)kernel.org --- sound/usb/caiaq/audio.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index d0d493c..aa52b3e 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -614,6 +614,7 @@ static void read_completed(struct urb *urb) struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; + size_t offset = 0; if (urb->status || !info) return; @@ -634,7 +635,8 @@ static void read_completed(struct urb *urb) len = urb->iso_frame_desc[outframe].actual_length; out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; - out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; + out->iso_frame_desc[outframe].offset = offset; + offset += len; if (len > 0) { spin_lock(&dev->spinlock); @@ -650,7 +652,7 @@ static void read_completed(struct urb *urb) } if (send_it) { - out->number_of_packets = FRAMES_PER_URB; + out->number_of_packets = outframe; out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } -- 1.7.6 --------------000700020004070602000804--

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[alsa-devel] Suspend/Resume failure of Alsa but OSS emulation still works
by Shane Volpe 05 Aug '11

05 Aug '11

My embedded system's audio is not resuming after the system resumes from suspend to RAM, below is a detailed discription of the problem and what I have done, to date, to identify the source of the problem. I have tested the following scenario with several audio applications such as Alsa aplay and ogg123: If I have an audio player (aplay or ogg123) using Alsa and I suspend my system to RAM when the system wakes up (resumes) there is no sound. If I kill the audio player and start it back up the sound returns so it seems the act of opening and closing Alsa fixes the failure. During the "no sound state" I also opened alsamixer and adjusted (successfully) several alsa settings including volume but the sound still did not resume until I restarted the player. I also configured one player to use the OSS (emulated by Alsa). When I used the emulated OSS audio driver (/dev/dsp) the sound did resume after the system woke up (resumed) from memory so everything WORKED! I then configured the player back to use the native Alsa driver and again there was no sound after resume. Has anyone seen this issue before or does anyone have additional tests or debugging ideas they can recommend to further narrow down the issue. I'm very comfortable with modifying and compiling the Kernel so please do not limit your suggestions to user-space tweaks. Embedded System Information: Processor: ARM/Xscale (PXA270) Audio Codec: AC97 (UCB1400) alsa lib version: 1.0.24.1 Linux Kernel Version: 2.6.39.2 Regards, Shane

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[alsa-devel] [PATCH] ASoC: OMAP4: mcpdm: Fix IRQ handling
by Peter Ujfalusi 05 Aug '11

05 Aug '11

Use the correct variable holding the IRQ reason flags to check the reason for the interrupt. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello Liam, this patch as well with my previous series for McPDM can go for 3.1, since this is fixing the interrupt handler function. Regards, Peter sound/soc/omap/mcpdm.c | 14 ++++++-------- 1 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index 1594fe0..9746a49 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -283,41 +283,39 @@ static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) /* Acknowledge irq event */ omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); - if (irq & MCPDM_DN_IRQ_FULL) { + if (irq_status & MCPDM_DN_IRQ_FULL) { dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); omap_mcpdm_reset_playback(1); omap_mcpdm_playback_open(mcpdm_irq->downlink); omap_mcpdm_reset_playback(0); } - if (irq & MCPDM_DN_IRQ_EMPTY) { + if (irq_status & MCPDM_DN_IRQ_EMPTY) { dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); omap_mcpdm_reset_playback(1); omap_mcpdm_playback_open(mcpdm_irq->downlink); omap_mcpdm_reset_playback(0); } - if (irq & MCPDM_DN_IRQ) { + if (irq_status & MCPDM_DN_IRQ) dev_dbg(mcpdm_irq->dev, "DN write request\n"); - } - if (irq & MCPDM_UP_IRQ_FULL) { + if (irq_status & MCPDM_UP_IRQ_FULL) { dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); omap_mcpdm_reset_capture(1); omap_mcpdm_capture_open(mcpdm_irq->uplink); omap_mcpdm_reset_capture(0); } - if (irq & MCPDM_UP_IRQ_EMPTY) { + if (irq_status & MCPDM_UP_IRQ_EMPTY) { dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); omap_mcpdm_reset_capture(1); omap_mcpdm_capture_open(mcpdm_irq->uplink); omap_mcpdm_reset_capture(0); } - if (irq & MCPDM_UP_IRQ) { + if (irq_status & MCPDM_UP_IRQ) dev_dbg(mcpdm_irq->dev, "UP write request\n"); - } return IRQ_HANDLED; } -- 1.7.6

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[alsa-devel] USB mic delay problem on OMAP4 PandaBoard
by ABDUL MOIZ 05 Aug '11

05 Aug '11

Hi, I am working on OMAP4 platform (Panda Board) using ubuntu 10.10(kernel: 2.6.35) and recording audio with USB mic and getting the following problem: When I am recording with the arecord utility as shown below: *$ arecord -Dplughw:1,0 -f cd foo.wav * audio is recorded with XRUN. So we have tested by changing buffer size and period size as shown below, *$ arecord -Dplughw:1,0 -f cd --buffer-size=65536 --period-size=128 foo.wav * and we are getting good recorded audio quality. I found from other posts that buffer-size of 65536 is very large and it should work with size of only 1024. same usb mic works without xruns on my desktop with buffer size 1024. The problem is that the application has to be told to use insanely large buffers in order to avoid xruns. I suspect the problem is either in the kernel (possibly just a misconfiguration) or in hardware. So Can anyone suggest, how to solve the above problem? Regards, Abdul Moiz

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[alsa-devel] snd soc spi read/write
by Scott Jiang 05 Aug '11

05 Aug '11

Hi Mark, My version is linux 3.0, I found snd_soc_spi_read/write didn't work at 16-8 bit mode. The addr is 16bit, the function compares reg with reg_cache_size. This certainly causes failure. In the former version, reg &= 0xff seems right. So how to use snd_soc_16_8_write now? Regards, Scott

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Re: [alsa-devel] [PATCH 1/3] irq: If an IRQ is a GPIO, request and configure it
by Stephen Warren 05 Aug '11

05 Aug '11

Rob Herring wrote at Thursday, August 04, 2011 7:55 PM: > On 08/04/2011 06:00 PM, Stephen Warren wrote: > > Many IRQs are associated with GPIO pins. When the pin is used as an IRQ, > > it can't be used as anything else; it should be requested. Enhance the > > core interrupt code to call gpio_request() and gpio_direction_input() for > > any IRQ that is also a GPIO. This prevents duplication of these calls in > > each driver that uses an IRQ. ... > > diff --git a/kernel/irq/manage.c b/kernel/irq/manage.c ... > > @@ -978,6 +979,16 @@ __setup_irq(unsigned int irq, struct irq_desc *desc, struct irqaction *new) > > old = *old_ptr; > > } while (old); > > shared = 1; > > + } else { > > + gpio = irq_to_gpio(irq); > > If you read the documentation for gpio, it is not recommended to use > irq_to_gpio. There's only a handful of users. Part of the problem is it > is platform specific and the gpio core cannot convert irq to gpio > number It seems like that's a soluble problem though? I was thinking about adding a to_gpio function to struct irq_chip, as the inverse of struct gpio_chip's to_irq. Then, presumably any platform would be able to convert back from IRQ to GPIO, provided the platform called a new __irq_to_gpio from the platform-specific gpio_to_irq, just like most gpio_to_irq implementations defer to __gpio_to_irq. I ended up not doing that in this patchset, since Tegra's gpio_to_irq function already works for the IRQs/GPIOs that were relevant for my testing, and I wanted to post a simple patch first to driver discussion. > Here is the relevant section: > > Non-error values returned from irq_to_gpio() would most commonly be used > with gpio_get_value(), for example to initialize or update driver state > when the IRQ is edge-triggered. Note that some platforms don't support > this reverse mapping, so you should avoid using it. -- nvpublic

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[alsa-devel] Has OMAP4 officially installed AIC31xx driver yet?
by hong zhang 04 Aug '11

04 Aug '11

Liam, I am wondering if OMAP4 has officially installed AIC31xx driver or not. If yes, where can I get it? See following , I have one aic31xx driver but it has problems to be second sound card if use sdp4430_snd_device = platform_device_alloc("soc-audio", -1); in sdp4430.c omap4_snd_device = platform_device_alloc("soc-audio", 1); in omap4_panda_aic31xx.c or sdp4430_snd_device = platform_device_alloc("soc-audio", 0); in sdp4430.c omap4_snd_device = platform_device_alloc("soc-audio", 1); in omap4_panda_aic31xx.c when mounting such as Freeing init memory: 200K init: ureadahead main process (796) terminated with status 5 Unable to handle kernel paging request at virtual address 72656767 pgd = dc3b4000 [72656767] *pgd=00000000 Internal error: Oops: 5 [#1] PREEMPT SMP last sysfs file: /sys/devices/virtual/block/ram11/removable Modules linked in: CPU: 0 Not tainted (2.6.35.7-ctbu-5-ga320686-dirty #19) PC is at snd_soc_info_enum_double+0x5c/0x6c LR is at snd_ctl_elem_info_user+0xc0/0x1d8 pc : [<c0324204>] lr : [<c0311bd0>] psr: 80000113 sp : dc3c1d18 ip : dc3c1d30 fp : dc3c1d24 r10: dc040dc0 r9 : dc9f297c r8 : 00000000 r7 : dcae1b80 r6 : c1105511 r5 : dc3c1d30 r4 : dc9f2800 r3 : 00000000 r2 : 72656767 r1 : 00000000 r0 : dc3c1d88 Flags: Nzcv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 9c3b404a DAC: 00000015 Thanks. ---Henry

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Re: [alsa-devel] Regression in 3.0-git after commit c82693db52beced0419cecf09a3c81adfe95a544 - sound does not work
by Takashi Iwai 04 Aug '11

04 Aug '11

At Thu, 04 Aug 2011 13:19:17 -0500, Larry Finger wrote: > > On 08/04/2011 12:55 PM, Takashi Iwai wrote: > > > > Thanks. Through a quick glance at these information, the problem > > looks like that both newly appearing "Headphone" and "Speaker" volumes > > are set to zero. Usually alsactl invoked by udev will set appropriate > > values for new elements, but in your case, it didn't work by some > > reason. > > > > Try to adjust them. (Note that you don't have to rebuild the kernel > > just for this test; pass model=auto option to snd-hda-intel module, > > then it'll switch to that mode.) > > By adding that option to /etc/modprobe.d/99-local.conf, and removing my patch, I > got new controls with the mixer. After fiddling with them a bit, I got working > sound. The issue is fixed. OK, thanks for a quick check! BTW, are you using systemd or sysvinit? The culprit might be the missing call of "alsactl init" at the startup, and it depends on the start-up system, of course. Takashi

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Re: [alsa-devel] Regression in 3.0-git after commit c82693db52beced0419cecf09a3c81adfe95a544 - sound does not work
by Takashi Iwai 04 Aug '11

04 Aug '11

At Thu, 04 Aug 2011 11:07:13 -0500, Larry Finger wrote: > > On 08/04/2011 12:17 AM, Takashi Iwai wrote: > > > > And reverting the above fixes your problem, right? > > Then please give alsa-info.sh outputs on both working and non-working > > kernels (run with --no-upload option). > > Yes, reverting that one hunk fixes my problem. > > The alsa-info.sh outputs for the working and nonworking cases are attached. Thanks. Through a quick glance at these information, the problem looks like that both newly appearing "Headphone" and "Speaker" volumes are set to zero. Usually alsactl invoked by udev will set appropriate values for new elements, but in your case, it didn't work by some reason. Try to adjust them. (Note that you don't have to rebuild the kernel just for this test; pass model=auto option to snd-hda-intel module, then it'll switch to that mode.) thanks, Takashi

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