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Re: [alsa-devel] a7839e96 (PNP: increase max resources) breaks my ALSA intel8x0 sound card
by Robert Hancock 27 Feb '08

27 Feb '08

Avuton Olrich wrote: > With v2.6.24 my second ALSA sound device stopped working. > > After bisection it says this was the offending commit. > > a7839e960675b549f06209d18283d5cee2ce9261 is first bad commit > commit a7839e960675b549f06209d18283d5cee2ce9261 > Author: Zhao Yakui <yakui.zhao(a)intel.com> > Date: Wed Nov 28 16:21:21 2007 -0800 > > PNP: increase the maximum number of resources > > On some systems the number of resources(IO,MEM) returnedy by PNP device is > greater than the PNP constant, for example motherboard devices. It brings > that some resources can't be reserved and resource confilicts. This will > cause PCI resources are assigned wrongly in some systems, and cause hang. > This is a regression since we deleted ACPI motherboard driver and use PNP > system driver. > > [akpm(a)linux-foundation.org: fix text and coding-style a bit] > Signed-off-by: Li Shaohua <shaohua.li(a)intel.com> > Signed-off-by: Zhao Yakui <yakui.zhao(a)intel.com> > Cc: Bjorn Helgaas <bjorn.helgaas(a)hp.com> > Cc: Thomas Renninger <trenn(a)suse.de> > Cc: <stable(a)kernel.org> > Signed-off-by: Andrew Morton <akpm(a)linux-foundation.org> > Signed-off-by: Linus Torvalds <torvalds(a)linux-foundation.org> > > The audio device is 00:1b.0 (see my lspci -vvv output), the other > audio device works fine. > > http://avuton.googlepages.com/v2.6-before (dmesg revision before) > http://avuton.googlepages.com/v2.6-after (dmesg broken revision) > http://avuton.googlepages.com/lspci-vvv > http://avuton.googlepages.com/config (from the broken revision) > http://avuton.googlepages.com/iomem > http://avuton.googlepages.com/ioports Here's why the driver fails to load: [ 31.133060] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 [ 31.133141] PCI: Unable to reserve mem region #1:4000@febf8000 for device 0000:00:1b.0 [ 31.133197] ACPI: PCI interrupt for device 0000:00:1b.0 disabled [ 31.133244] HDA Intel: probe of 0000:00:1b.0 failed with error -16 The iomem location of the HDA controller conflicts with this reservation by the BIOS: [ 22.906654] system 00:08: iomem range 0xfebfa000-0xfebfac00 has been reserved There was a patch floating around to ignore PnPACPI reservations which conflict with PCI BARs, which appears to be what's happening in this case. That patch originally worked for any board, but was later made specific to a certain Supermicro motherboard which had the sata_nv controller MMIO regions marked as reserved, preventing the driver from loading. We may need a more general solution. See: https://bugzilla.redhat.com/show_bug.cgi?id=313491

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[alsa-devel] HDMI Multichannel LPCM/RAW Support/Status
by Dustin Kerstein 27 Feb '08

27 Feb '08

I was just curious if there is any planned support for multichannel LPCM/Raw audio for newer HDMI based chipsets. There are several matx boards with HDMI 1.1 and above compliance, which means they should be able to pass 8-channel LPCM audio. Also, in HDMI 1.3 and above, Bit Stream passing is supported, which actually should be easier, bandwidth wise, than LPCM. Multichannel PCM (above 2 channels) was never an issue because SPDIF lacked the bandwidth. After talking with some people in irc it seems as though the # of channels setting in Alsa is only one bit. Thanks. Dustin

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Re: [alsa-devel] PATCH for alsa-info.sh - using mktemp for instead of hardcoded temp dir.
by Takashi Iwai 27 Feb '08

27 Feb '08

At Thu, 28 Feb 2008 02:42:20 +1000, Travis Place wrote: > > On Thu, 28 Feb 2008 2:39:16 am you wrote: > > At Thu, 28 Feb 2008 00:36:52 +1000, > > > > Travis Place wrote: > > > Email subject should explain well enough. > > > > I see some problems with this change: > > - it doesn't work if /tmp/alsainfo doesn't exist > > - temp directory isn't removed if --no-upload is given > > > > I think you should avoid --tempdir=xxx with mktemp. > > > > The output file name is another question. > > IMO, it'd be better to pass the output file name via a command line > > option, and put it on the home directory (e.g. ~/alsa-info.out) as > > default rather than /tmp. > > well, /tmp/alsainfo/ will exist, as we create it in the script.. its > $ALSAINFODIR .. the temp dir is then created inside that. It's created too late. And, if we create /tmp/alsainfo, there is little merit of mktemp as /tmp/alsainfo could be a symlink. Simply TEMPDIR should be what mktemp creates, and clean it up after quitting. > As for not being cleaned up, if --no-upload is specified, ill check that now.. > > Also, is outputting the resulting file to ~/ wise ? It's at least safer than the fixed filename on /tmp. I don't insist on this option, though. Anyway, --output-file=xxx would be a nice addition. > should this occur even if the result is uploaded, or only on > --no-upload ? I don't think it's needed when uploaded (except for any debugging reason). Just my $0.02. thanks, Takashi

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[alsa-devel] PATCH for alsa-info.sh - using mktemp for instead of hardcoded temp dir.
by Travis Place 27 Feb '08

27 Feb '08

Email subject should explain well enough. -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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[alsa-devel] new driver: channel alignment (was: 30mS interval problem)
by William Juul 27 Feb '08

27 Feb '08

Hello I am new to ALSA and trying to write a new driver for a ADC connected with PCM to an AVR32 on a NGW100 reference card provided by Atmel. I have the problem that different channels from the PCM stream comes in different positions in the WAV file. For example when I disable all but the first PCM channel on the ADC, I can see with an oscilloscope that there is only data in the first timeslot on the PCM data stream. But the data from this channel can occur in any of the 4 channels in the WAV file. This is consistent throughout the one capture/file, but in the next capture the data will occur in another channel in the WAV-file. How can I make sure that the data from one particular channel always occur in a fixed position in the WAV file? Best regards William Juul

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[alsa-devel] [ALC262 patch_realtek.c] Add Fujitsu Lifebook E8410 to quirk table
by Tony Vroon 27 Feb '08

27 Feb '08

Good afternoon, Could you please consider the following patch for 2.6.25-rc4. The BIOS autoprobe doesn't do a very good job setting this card up. Card details: 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 10cf:142d (and verbose codec information attached) Regards, Tony V. LINX sysadmin

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Re: [alsa-devel] Ends up with write error (Ning Zhang)
by Ning Zhang 27 Feb '08

27 Feb '08

Hi all, There is no response so far, maybe I am not clear. So in sum, here are the questions: 1. In meddle-level driver, how comes the runtime->control->avail_min? I mean which function set the value, and according what to set the value? Does it have some relationship to the hardware configuration in the device driver( i.e snd_pcm_hardware )? 2. Should I use the snd_pcm_period_elapsed function even I don't use the interrupt mechanism? If yes, In which callback function should I call it(snd_pcm_period_elapsed)? 3,Can anybody do me a favor to explain the mechanism of the middle-level driver or give me some hints? Thanks in advance. BRs, David > > ------------------------------ > > Message: 7 > Date: Wed, 20 Feb 2008 16:30:54 +0800 > From: "Ning Zhang" <ning.zhang06(a)gmail.com> > Subject: [alsa-devel] Ends up with write error > To: alsa-devel <alsa-devel(a)alsa-project.org> > Message-ID: > <48a1cba90802200030v65794b96j7c956406fdc0df9c(a)mail.gmail.com> > Content-Type: text/plain; charset=GB2312 > > Hi all, > I am writing a ALSA driver for my ARM 926 platform. The ARM communicates > with the DSP by accessing a 128-byte shared memory. I don't use IRQ > mechanism.(In fact, I am not sure the platform has this mechanism) > Here is some code slices: > > static struct snd_pcm_hardware snd_arch_playback_hw = { > > .info = (SNDRV_PCM_INFO_MMAP | > > SNDRV_PCM_INFO_INTERLEAVED | > > SNDRV_PCM_INFO_BLOCK_TRANSFER), > > .formats = SNDRV_PCM_FMTBIT_S16_LE, > > .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ > > SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ > > SNDRV_PCM_RATE_48000), > > .rate_min = 8000, > > .rate_max = 48000, > > .channels_min = 2, > > .channels_max = 2, > > .buffer_bytes_max = 128, > > .period_bytes_min = 2, > > .period_bytes_max = 128, > > .periods_min = 1, > > .periods_max = 64, > > }; > > > > static snd_pcm_uframes_t snd_arch_pcm_pointer(struct snd_pcm_substream > > *substream) > > { ? > > /* > > * get the current hardware pointer > > */ > > current_ptr = arch_fw_read_dsp_reg(chip, AFE_RWADDR) & AFE_RD_ADDR; > > > return bytes_to_frames(runtime, current_ptr); > > } > > static int snd_arch_pcm_playback_copy(struct snd_pcm_substream *substream, > > int channel, > > snd_pcm_uframes_t pos, > > void __user *src, > > snd_pcm_uframes_t count) > > { ?? > > > > count = frames_to_bytes(runtime, count); > > pos = frames_to_bytes(runtime, pos); > > chip->bufpos = pos; > > ?. > > spin_lock(chip->reg_lock); > > if ( count > 0 && count <= MAX_BUF_SIZE ) { > > if ( copy_from_user(chip->bufptr, src, count) ) { > > snd_printk(KERN_ERR "can not copy user data!\n"); > > return -EFAULT; > > } > > if ( arch_fill_rx_buf(chip, > > (signed short*)(chip->bufptr), > count) < 0 ) { > > snd_printk(KERN_ERR "fill rx buffer failed!\n"); > > return -1; > > } > > ????? > > } > spin_unlock(chip->reg_lock); > > return 0; > > } > > The problem is my driver can make sounds, but ends up a write > error. After > much debugging, I think there might be some problem in the > interworking between > middle-level driver and the device driver. > > I found the runtime->control->avail_min=32, and the > runtime->buffer_size and runtime->period_size are both 32. So at first > middle-level driver call snd_pcm_lib_write1 transfers 32 frames(128 bytes) > to the device driver's copy callback function and trigger function. Then > when middle-level driver wants to transfer another 32 frames, it calls the > pointer callback function and gets the position, because at this time the > whole bytes has not been processed completely, so the pointer function's > return value must less than 32, and the avail must less than > runtime->control_->avail_min, and goes to this flow and ends up a write > error. > > > _______________________________________________________________________________ > > if (((avail < runtime->control->avail_min && size > avail) || > > (size >= runtime->xfer_align && avail < > runtime->xfer_align))) { > > ?????... > > long tout; > > init_waitqueue_entry(&wait, current); > > add_wait_queue(&runtime->sleep, &wait); > > while (1) { > > if (signal_pending(current)) { > > state = SIGNALED; > > break; > > } > > set_current_state(TASK_INTERRUPTIBLE); > > snd_pcm_stream_unlock_irq(substream); > > tout = schedule_timeout(10 * HZ); > > snd_pcm_stream_lock_irq(substream); > > if (tout == 0) { > > if (runtime->status->state != > SNDRV_PCM_STATE_PREPARED && > > runtime->status->state != > SNDRV_PCM_STATE_PAUSED) { > > state = > runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED; > > break; > > } > > } > > switch (runtime->status->state) { > > > ????.. > > : > > state = ERROR; > > goto _end_loop; > > > > default: > > break; > > } > > avail = > snd_pcm_playback_avail(runtime); > > if (avail >= > runtime->control->avail_min) > { > > state = READY; > > break; > > } > > } > > _end_loop: > > remove_wait_queue(&runtime->sleep, &wait); > > > > switch (state) { > > ??. > > case EXPIRED: > > snd_printd("playback write error (DMA > or > IRQ trouble?)\n"); > > err = -EIO; > > goto _end_unlock; > > ??? > > } > > } > > This code is a part of snd_pcm_lib_write1() in sound/core/pcm_lib.c > > I think that I misunderstood the mechanism of middle-level driver. Could > you please correct me? > > What should the pointer callback function return? What's it for? > > How comes the runtime->control->avail_min? > > Because I don't use the IRQ, so should I use the snd_pcm_period_elapsed > function? And where? > > ALSA Version is 1.0.14rc1 and I use alsa-lib-1.0.15. > > Can someone comment on this and give me some hints to solve this > problem? Thanks > in advance. > > BRs, > > David > > ------------------------------ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > End of Alsa-devel Digest, Vol 12, Issue 69 > ****************************************** >

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Re: [alsa-devel] Via VT82xx VT1708 ICE1724HT and higher
by Demian Martin 27 Feb '08

27 Feb '08

Pavel: Thanks for the insight. I thought I was doing something wrong since no one else has noticed. I tried your suggestion to use plughw:0,1 and the verbose output looks fine, the data rate shifts fine (I'm watching the L/R clock on the input to the DAC on an external DAC) but no audible sound comes out the converter unless it's a 16 bit 44.1 file. The 24 bit stuff is not getting converted into something useful to the DAC. Should I check to see if the data clock is 48X the LR clock? What info do you need from me to look deeper? I can supply a sample clip of a 176.4 KHz recording from Reference Recordings if it would be helpful. I'm also testing with the samples from Linn and Gimmel. I looked at the Plug plugin but have not been able to figure out how to configure it for my needs. There is way too much marginal information on the web and little that directs me to an optimum setting. -Demian From: Pavel Hofman <pavel.hofman(a)insite.cz> Subject: Re: [alsa-devel] Via VT82xx VT1708 ICE1724HT and higher sample rate/bit depth files through SPDIF To: Demian Martin <demianm_1(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Message-ID: <47C423E4.4060900(a)insite.cz> Content-Type: text/plain; charset=windows-1252; format=flowed Hi Martin, > > First- sample rate > > Even though the documents claim that it will play at 176.4 KHz and 192KHz > whenever I play a file at these rates it down converts to ? the rate (88200 > and 96000). I tested this with the ICE1724HT (Juli@) which has almost the > same core, I never could get 176.4KHz at the SPDIF output. I confirmed that > the ICE1724 will play the file at 176.4KHz in Windows so its not a hardware > limitation. I tried to do the same with the VT1708 (on a Via CX700 chipset) > and the Windows driver wouldn?t let me at anything other than 48000. I spent a few days on this issue and confirm that ICE1724HT cannot output 176.4kHz from its internal SPDIF transmitter, WHEN clocked by its internal clock circuits. A scope hooked to corresponding pins/outputs revealed the following: All I2S lines output correct 176.4kHz signal, i.e. D/A codecs receive proper signal. However, I did not find a way to force ICE1724 to output 176.4kHz, it always drops to 88.2kHz, no matter what MT01/MT02 combination is. The same happens with windows drivers of my Prodigy 192. The sequence is: 44.1, 48, 88.2, 96, 88.2, 192 I have not seen any manufacturer of Envy-based cards revealing this deficiency. The "preliminary" datasheet we all use mentions only that the chip forces the 128x mode when switched to 176.4kHz. I believe the output of 88.2kHz is an undocumented feature or perhaps even a bug of the chip. Juli is a card whose manufacturer may be aware of the problem. It is not clocked by ICE1724 internal clocks fed by the two crystals. Instead, the crystals also provide signal to external circuits generating clock signals of required frequency, feeding the chip in slave mode. The circuitry is controlled by ice1724 GPIOs. This way the ICE1724 is not aware of its clocking frequency and cannot thus mis-configure its SPDIF transmitter. The windows driver does output proper 176.4kHz. I am working on Juli driver and have been analyzing this beast for a few nights. The special clocking is already working at home :) > Second, the only way I can get the 24 bit files out in a form that the card > can play is to pass them through dmix. But I need to reconfigure dmix for > each sample rate. And I?m not convinced that it isn?t resampling twice, both > at the input and the output. Or how to configure the buffers etc. to prevent > underruns even when the only conversion is to pass a 24 bit file to the > card. ICE cards use 32bits format. Try aplay -v -D plughw:0 24bit.wav and you will see the conversion to 32 bits done by the plug plugin, no dmix involved. > > I?m trying to build a high quality music server that will pass the files to > either the internal or an external DAC at the bit depth and sample rate they > were recorded at. I have tried to understand the various plugs and settings > but it seems they are all focused on mixing various streams. I want to send > the files unmixed to the output. I can?t figure out what set of plugs and > settings in asound.conf would enable this. Check the plug plugin. Regards, Pavel Hofman. -----------------

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[alsa-devel] ATI HD 2600 Sound?
by Isaac Manuel 26 Feb '08

26 Feb '08

HP Pavilion HDX9200 Laptop PC Ubuntu 7.10 (Gutsy Gibbon) Linux Kernel 2.6.22-14-generic Sound only play through headphones. *aplay -l* **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 0: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 *sudo lspci -v *Audio device: ATI Technologies Inc Unknown device aa08 Subsystem: Hewlett-Packard Company Unknown device 30d4 Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at e8310000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Capabilities: [58] Express Legacy Endpoint IRQ 0 Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- * **sudo lspci -nv* 0403: 1002:aa08 Subsystem: 103c:30d4 Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at e8310000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Capabilities: [58] Express Legacy Endpoint IRQ 0 Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable-

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[alsa-devel] hdsp firmware
by Karl Grill 26 Feb '08

26 Feb '08

Hi all, as a first step towards implementing support for the hdsp rpm, I've uploaded a patch to the alsa-firmware branch as bug report #3810. Besides the patch, I've put up the firmware binaries in .tar.gz format. This contains replacements for the rev.11 firmware for the digiface/multiface and also for the rpm. The latter is of little use right now, but I'd be very happy if anybody could try those for the digiface/multiface and report if it works, so I know if my assumptions on the drivers are correct. best regards kg PS. sorry I didn't file this report in the firmware section, but the web form there doesn't give me any usable choice for Category yet insists that I supply one.

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