February 2008 - Alsa-devel - mailman.alsa-project.org

[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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[alsa-devel] Zoom H4 hand held studio.
by Joseph klitzke 26 Feb '08

26 Feb '08

Greetings I am inquiring into the development of a driver for the Zoom H4 Hand held recorder. Is there any work being done on the device? If not is it being considered. Is there anything I can do to get it started? In short the H4 can 4 track recorder and a usb audio device with effects, a great piece of recording gear. Thanks Joseph Klitzke

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[alsa-devel] snd-usb-audio syncronisation problem
by Adrian Stancu 26 Feb '08

26 Feb '08

Hi, I'm using an E-MU 0404 USB interface with the snd-usb-audio module from alsa 1.0.16, operating in isochronous asynchronous mode. This interface was producing very annoying click/pop sounds at regular intervals during playback. Eventually I discovered that the syncronisation mechanism telling the driver to send audio data faster or slower to match the interface's consumption speed does not work as expected, for some reason. I did a very dirty hack within the source of this driver, simulating the interface asking for data to be sent a little bit quicker, and this almost fixed the problem *for me*. I really wish to debug this issue further, to find what actually does not work as it should in this feed-back mechanism. However this was the first time I dealt with alsa source code, so I do not have much experience. Therefore, I really need help with debugging this problem and maybe finding an universal fix. Thanks in advance, Adi

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[alsa-devel] HDA-Intel - Patch to support RV7xx HDMI Audio
by Yang, Libin 26 Feb '08

26 Feb '08

Hi Takashi, This patch is to add R7xx HDMI audio support. Would you please accept it? Thanks a lot! Signed-off-by: Libin Yang <Libin.yang(a)amd.com> --- alsa-driver-1.0.16/alsa-kernel/pci/hda/hda_intel.c.orig 2008-02-13 11:57:11.000000000 -0500 +++ alsa-driver-1.0.16/alsa-kernel/pci/hda/hda_intel.c 2008-02-13 12:02:39.000000000 -0500 @@ -2015,6 +2015,9 @@ { 0x1002, 0xaa20, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV635 HDMI */ { 0x1002, 0xaa28, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV620 HDMI */ { 0x1002, 0xaa30, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV770 HDMI */ + { 0x1002, 0xaa38, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV730 HDMI */ + { 0x1002, 0xaa40, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV710 HDMI */ + { 0x1002, 0xaa48, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV740 HDMI */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ Thanks & Regards, Libin

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[alsa-devel] Via VT82xx VT1708 ICE1724HT and higher sample rate/bit depth files through SPDIF
by Demian Martin 26 Feb '08

26 Feb '08

I have several problems trying to play files higher than 48000 KHz and with bit depth of 24 bits. I submitted this to the users and got a resounding silence. 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 ½ 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 wouldnt let me at anything other than 48000. 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 Im not convinced that it isnt 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. Im 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 cant figure out what set of plugs and settings in asound.conf would enable this.

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[alsa-devel] [PATCH 1/3] ASoC: Fix TLV320AIC3X PLL divider table for 64 kHz rate
by Mark Brown 26 Feb '08

26 Feb '08

From: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 710e028..569ecac 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -681,8 +681,8 @@ static const struct aic3x_rate_divs aic3x_divs[] = { {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ - {22579200, 96000, 96000, 0x1, 8, 7075}, - {33868800, 96000, 96000, 0x1, 5, 8049}, + {22579200, 64000, 96000, 0x1, 8, 7075}, + {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, -- 1.5.4.3

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