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[alsa-devel] McBSP troubles
by Sergey Lapin 21 Nov '09

21 Nov '09

Hi, all! I have problems making any sound with current linux-omap master. ASoC repository did not change anything either. I use slightly modified (removed all GPIOs/jacks and different clock names because we use OMAP3 instead of OMAP2 n810 uses) sound/soc/omap/n810.c, with McBSP2. Codec is tlv320aic32 and it is well supplied with 13 MHz clock, so should work happily. Old 2.6.28-omap1 kernel + modifications works well with this, but it lacks some fundamental features recently added to mainline and linux-omap kernels, so we considered upgrading. Almost all stuff on custom board works, except for sound, which I fight for several weeks already, so I look for any help with this. I use OMAP3525 SoC. Debugging output of mcbsp driver is the following: gnome5:~# aplay -c 2 -f S16_LE /dev/urandom [ 4593.488159] disabling LINE_OUT [ 4593.491302] disabling DMIC [ 4593.494018] Disabling ASSYM pin Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo [ 4593.620086] omap-mcbsp omap-mcbsp.2: Configuring McBSP2 phys_base: 0x49022000 [ 4593.678375] gnome5_jack_event [ 4593.735015] omap-mcbsp omap-mcbsp.2: **** McBSP2 regs **** [ 4593.740539] omap-mcbsp omap-mcbsp.2: DRR2: 0xd907da5a [ 4593.745697] omap-mcbsp omap-mcbsp.2: DRR1: 0x0000 [ 4593.750518] omap-mcbsp omap-mcbsp.2: DXR2: 0x0000 [ 4593.755340] omap-mcbsp omap-mcbsp.2: DXR1: 0x0000 [ 4593.760162] omap-mcbsp omap-mcbsp.2: SPCR2: 0x02f5 [ 4593.764984] omap-mcbsp omap-mcbsp.2: SPCR1: 0x0030 [ 4593.769805] omap-mcbsp omap-mcbsp.2: RCR2: 0x8041 [ 4593.774627] omap-mcbsp omap-mcbsp.2: RCR1: 0x0040 [ 4593.779449] omap-mcbsp omap-mcbsp.2: XCR2: 0x8041 [ 4593.784271] omap-mcbsp omap-mcbsp.2: XCR1: 0x0040 [ 4593.789093] omap-mcbsp omap-mcbsp.2: SRGR2: 0x001f [ 4593.793884] omap-mcbsp omap-mcbsp.2: SRGR1: 0x0f00 [ 4593.798706] omap-mcbsp omap-mcbsp.2: PCR0: 0x000f [ 4593.803527] omap-mcbsp omap-mcbsp.2: *********************** [ 4603.811187] ALSA sound/core/pcm_lib.c:1708: playback write error (DMA or IRQ trouble?) [ 4614.818878] ALSA sound/core/pcm_lib.c:1708: playback write error (DMA or IRQ trouble?) aplay: pcm_write:1442: write error: And no sound at all. Codec is supplied properly with clock, and programmed normally, so I think that's McBSP trouble, but I might be wrong. Clock is supplied properly. So, what is changed and needs updating? Is it McBSP or codec? I see no signals using oscilloscope on _FSX and _CLKX. Any ideas? S.

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[alsa-devel] Sigmatel STAC 9228
by Alonso Quiroz H. 21 Nov '09

21 Nov '09

Hey guys!, this is my first mail to the list. Excuse me if this is not the better way to send my request. I have a Dell XPS M1530 laptop, with a Sigmatel soundcard... well, read it for yourselves: http://pastebin.ca/1679794 By default (in Ubuntu 9.10) the card doesn't work in 5.1 mode, and that is my problem... by default it doesn't show the option "Surround 5.1" I have tried a lot of combinations of model=XXXXX in /etc/modprobe.d/alsa-base.conf, starting by the ones in /usr/share/doc/alsa-base/driver/HD-Audio-Models.txt. The better result is with "3stack": It 's possible to select the 5.1 option, but I don't get sound at all. The weird thing is that de volume control of Pulse shows bars moving. Thanks in advance. Sorry about my english. Alonso Quiroz H.

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[alsa-devel] RFC: ice1712 virtual devices
by Arno Schuring 21 Nov '09

21 Nov '09

Hello list, This is basically a resend of http://thread.gmane.org/gmane.linux.alsa.devel/59481/focus=59672 , which fixed the front: device of ice1712 cards to accept two-channel input. Currently, the front: device is exposed through the route plugin, which requires all clients to mmap all 10 channels, even though the front device is supposed to be a stereo device. This patch changes the front: device definition such that it matches the definition of iec958 in the same file. Additionally, I'm tempted to remove the surround* definitions because the chip does not really offer surround-style multichannel: it basically just offers multiple stereo channels, and does not provide any channel mapping beyond stereo. Finally, I'm also experimenting with the dshare plugin to allow applications to access the iec958: and front: devices simultaneously. Can anyone point me to a working example for this? From reading the alsa-lib documentation, it is not clear to me how I should nest the different plugins. Many thanks, Arno Schuring -- diff --git a/src/conf/cards/ICE1712.conf b/src/conf/cards/ICE1712.conf index 01e50d2..d7acb81 100644 --- a/src/conf/cards/ICE1712.conf +++ b/src/conf/cards/ICE1712.conf @@ -32,12 +32,28 @@ ICE1712.pcm.front.0 { @args.CARD { type string } - type route - ttable.0.0 1 - ttable.1.1 1 - slave.pcm { - type hw - card $CARD + type asym + playback.pcm { + type route + ttable.0.0 1 + ttable.1.1 1 + slave.pcm { + type hw + card $CARD + } + slave.format S32_LE + slave.channels 10 + } + capture.pcm { + type route + ttable.0.0 1 + ttable.1.1 1 + slave.pcm { + type hw + card $CARD + } + slave.format S32_LE + slave.channels 12 } }

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[alsa-devel] opti-miro: fix missing semicolon
by Krzysztof Helt 20 Nov '09

20 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Fix the mistake introduced by me earlier patch. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- I have borrowed the Miro PCM20 card. I can test the driver while making patches. diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b8170ad..1776103 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1277,7 +1277,7 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) if (mpu_port < 0) { snd_card_free(card); snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY + return -EBUSY; } } if (miro->irq == SNDRV_AUTO_IRQ) { ---------------------------------------------------------------------- KONKURS! Kliknij i wygraj luksusowe auto na weekend lub jedn� z 650 innych nagr�d od L’Oreal Men Expert!! >> http://link.interia.pl/f2428

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[alsa-devel] ALSA: hda - add possibility to choose speakers configuration for 4930g
by Łukasz Wojniłowicz 20 Nov '09

20 Nov '09

As in wrote title patch adds possibility to choose speakers configuration for 4930g. This feature was available in ALSA 1.0.19 (or even 1.0.20) but disappeared in 1.0.21

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Insaen 20 Nov '09

20 Nov '09

I own a SB x-fi fatality incl. the midi input box witch i realy would like to use for my bass guitar and use the record options, i run the card on ubuntu 9.04 and its running well but can't realy set the jack system right for software like rosegarden or lmms or any other music editing software. I know it's got something to do with my alsa config?? is there a forum about this toppick?Simon says have a nice day..2-;

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[alsa-devel] ALSA: hda - 4930g switch lfe with side in mixer
by Łukasz Wojniłowicz 20 Nov '09

20 Nov '09

Built-in sub-woofer can now be controlled by lfe slider instead of side slider on Acer Aspire 5930g

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[alsa-devel] [PATCH] sound: usbmidi: Use hweight16
by Akinobu Mita 20 Nov '09

20 Nov '09

Use hweight16 instead of Brian Kernighan's/Peter Wegner's method Signed-off-by: Akinobu Mita <akinobu.mita(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org Cc: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/usbmidi.c | 13 ++----------- 1 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 0eff19c..e5b0689 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1062,15 +1062,6 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, return 0; } -static unsigned int snd_usbmidi_count_bits(unsigned int x) -{ - unsigned int bits; - - for (bits = 0; x; ++bits) - x &= x - 1; - return bits; -} - /* * Frees an output endpoint. * May be called when ep hasn't been initialized completely. @@ -1914,8 +1905,8 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, out_ports = 0; in_ports = 0; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { - out_ports += snd_usbmidi_count_bits(endpoints[i].out_cables); - in_ports += snd_usbmidi_count_bits(endpoints[i].in_cables); + out_ports += hweight16(endpoints[i].out_cables); + in_ports += hweight16(endpoints[i].in_cables); } err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports); if (err < 0) { -- 1.6.5.1

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[alsa-devel] Mute lfe and side for 4930g when pluging in headphones
by Łukasz Wojniłowicz 20 Nov '09

20 Nov '09

Sorry that I'm not sending this patch by git but it takes for me too long to clone alsa-kernel.git repository at 15 KB download rate. Thanks for your understanding

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[alsa-devel] [PATCH 1/4] ASoC: DaVinci: i2s, reduce underruns by combining into 1 element
by Troy Kisky 19 Nov '09

19 Nov '09

Allow the left and right 16 bit samples to be shifted out as 1 32 bit sample. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- This applies to Marks for-2.6.33 branch Sorry about the prior bad patch base. arch/arm/mach-davinci/include/mach/asp.h | 6 ++ sound/soc/davinci/davinci-i2s.c | 74 ++++++++++++++++++++++-------- 2 files changed, 61 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 18e4ce3..019c647 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -51,6 +51,12 @@ struct snd_platform_data { u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ unsigned int codec_fmt; + /* + * Allowing this is more efficient and eliminates left and right swaps + * caused by underruns, but will swap the left and right channels + * when compared to previous behavior. + */ + unsigned enable_channel_combine:1; /* McASP specific fields */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d336786..b2a5372 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,6 +97,23 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + struct davinci_mcbsp_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; @@ -105,6 +122,27 @@ struct davinci_mcbsp_dev { int mode; u32 pcr; struct clk *clk; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned enable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -344,6 +382,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -373,29 +413,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - dma_params->acnt = dma_params->data_type; + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && dev->enable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; - - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -510,7 +545,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) + dev->enable_channel_combine = pdata->enable_channel_combine; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; -- 1.5.6.3

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