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

[alsa-devel] big latency and buffering question
by Achim Herrmann 02 Oct '09

02 Oct '09

Hi all, I'm completely new to sound and ALSA development and I have to develop an application which is capturing periodically a small amount of audio data (e.g. 64 frames), doing some processing and doing playback of the result. I started with ALSA example "pcm.c" to see if ALSA can meet these requirements. During investigation i found out, that up to a choosen buffer size of 140ms, generated sine sound is fine. Below 140ms, sound is stuttering, independent of call method I'm using. For buffer size of 140ms I found out, that target buffer size is 6174 frames (at 44.1kHz sample rate), period size is 3087 frames and loop inside write_and_poll function is calculated every 70 ms and 3087 frames are written. Think this is doublebuffering then, right? But if I now choose a buffersize of 139 ms or lower, target buffer size and period size are always the same. So no double buffering anymore and sound is stuttering. So, currently I'm far away from goal. I'd like to have a buffer size of e.g. 128 frames (2ms). Thus, period size should be 64 frames and loop should be calculated every 1ms. (If necessary I would use Linux-RT) If I choose a buffersize of 2ms, sound is far away from being sine :-) - Can I use ALSA for a application described in first sentence above? - What do I have to change in pcm.c to be able to create a sine signal using a buffersize of e.g. 128 frames and a period size of 64 frames? Any hints are appreciated Thanks in advance Achim

1 1

[alsa-devel] [PATCH] ASoC: Improve the debugfs hierarchy
by Peter Ujfalusi 02 Oct '09

02 Oct '09

Change the way the debugfs entries are created: If the codec->dev is valid, than use: debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/ if the codec->dev is NULL: debugfs/asoc/{codec->name}/ as root for the debugfs entries. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- The patch applies on top of sound-2.6 topic/asoc commit 88439ac793934a47f47ad285656b63d09f5937c8 ASoC: add support for multiple cards/codecs in debugfs sound/soc/soc-core.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e4ab36d..1dec9d2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1256,8 +1256,12 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { char codec_root[128]; - snprintf(codec_root, sizeof(codec_root), - "%s-%s", dev_name(codec->socdev->dev), codec->name); + if (codec->dev) + snprintf(codec_root, sizeof(codec_root), + "%s.%s", codec->name, dev_name(codec->dev)); + else + snprintf(codec_root, sizeof(codec_root), + "%s", codec->name); codec->debugfs_codec_root = debugfs_create_dir(codec_root, debugfs_root); -- 1.6.5.rc2

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 02 Oct '09

02 Oct '09

The following changes since commit c877c25170e2655d519b29e91d6c91d5d1a72a6f: Takashi Iwai (1): ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: fix kconfig order of Blackfin drivers Mark Brown (1): Merge remote branch 'takashi/fix/asoc' into for-2.6.32 sound/soc/blackfin/Kconfig | 98 ++++++++++++++++++++++---------------------- 1 files changed, 49 insertions(+), 49 deletions(-)

3 3

[alsa-devel] [PATCH 06/34] don't use __devexit_p to wrap hal2_remove
by Uwe Kleine-König 02 Oct '09

02 Oct '09

The function hal2_remove is defined using __exit, so don't use __devexit_p but __exit_p to wrap it. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/mips/hal2.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index c52691c..72523a7 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -926,7 +926,7 @@ static int __exit hal2_remove(struct platform_device *pdev) static struct platform_driver hal2_driver = { .probe = hal2_probe, - .remove = __devexit_p(hal2_remove), + .remove = __exit_p(hal2_remove), .driver = { .name = "sgihal2", .owner = THIS_MODULE, -- 1.6.4.3

2 5

[alsa-devel] [PATCH] Don't check if period size is equal to buffer size
by Lubomir Rintel 02 Oct '09

02 Oct '09

At least some drivers seem to allow a single period and those which don't would correct us anyway. Moreover, Pulseaudio with tsched sets this frequently. --- aplay/aplay.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index c7c82a1..4fa36cb 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1044,11 +1044,6 @@ static void set_params(void) } snd_pcm_hw_params_get_period_size(params, &chunk_size, 0); snd_pcm_hw_params_get_buffer_size(params, &buffer_size); - if (chunk_size == buffer_size) { - error(_("Can't use period equal to buffer size (%lu == %lu)"), - chunk_size, buffer_size); - exit(EXIT_FAILURE); - } snd_pcm_sw_params_current(handle, swparams); if (avail_min < 0) n = chunk_size; -- 1.6.2.5

3 2

[alsa-devel] [PATCH] sound: Make keywest_driver static
by Jean Delvare 02 Oct '09

02 Oct '09

From: Jean Delvare <khali(a)linux-fr.org> Subject: sound: Make keywest_driver static I can't see any reason for struct i2c_driver keywest_driver to not be static. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/ppc/keywest.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-2.6.32-rc1.orig/sound/ppc/keywest.c 2009-10-01 09:45:18.000000000 +0200 +++ linux-2.6.32-rc1/sound/ppc/keywest.c 2009-10-01 18:05:06.000000000 +0200 @@ -93,7 +93,7 @@ static const struct i2c_device_id keywes { } }; -struct i2c_driver keywest_driver = { +static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, -- Jean Delvare

2 1

[alsa-devel] [PATCH] sscape: remove MIDI instances counting with limit ULONG_MAX
by Krzysztof Helt 02 Oct '09

02 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, remove redundant "sscape:" prefix in messages printed by snd_printk and convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sscape.c | 85 +++++++++++++--------------------------------------- 1 files changed, 21 insertions(+), 64 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465c..ece2bb9 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -149,9 +149,8 @@ struct soundscape { /* * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL + * its firmware. */ - spinlock_t fwlock; unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +163,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +173,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR + "Failed to allocate %lu bytes for DMA\n", + size); return NULL; } } @@ -797,37 +793,13 @@ static int mpu401_open(struct snd_mpu401 * mpu) int err; if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); + snd_printk(KERN_ERR "MIDI disabled, please load firmware\n"); err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); } return err; } -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); -} - /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ @@ -845,8 +817,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -908,21 +878,20 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, err = snd_wss_pcm(chip, 0, &pcm); if (err < 0) { - snd_printk(KERN_ERR "sscape: No PCM device " - "for AD1845 chip\n"); + snd_printk(KERN_ERR "No PCM device for AD1845 chip\n"); goto _error; } err = snd_wss_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR "sscape: No mixer device " + snd_printk(KERN_ERR "No mixer device " "for AD1845 chip\n"); goto _error; } if (chip->hardware != WSS_HW_AD1848) { err = snd_wss_timer(chip, 0, NULL); if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " + snd_printk(KERN_ERR "No timer device " "for AD1845 chip\n"); goto _error; } @@ -932,7 +901,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, err = snd_ctl_add(card, snd_ctl_new1(&midi_mixer_ctl, chip)); if (err < 0) { - snd_printk(KERN_ERR "sscape: Could not create " + snd_printk(KERN_ERR "Could not create " "MIDI mixer control\n"); goto _error; } @@ -968,15 +937,14 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR - "sscape: can't grab port 0x%lx\n", port[dev]); + snd_printk(KERN_ERR "can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; if (sscape->type == SSCAPE_VIVO) { wss_res = request_region(wss_port[dev], 4, "SoundScape"); if (!wss_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%lx\n", + snd_printk(KERN_ERR "can't grab port 0x%lx\n", wss_port[dev]); err = -EBUSY; goto _release_region; @@ -988,18 +956,18 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ err = request_dma(dma[dev], "SoundScape"); if (err < 0) { - snd_printk(KERN_ERR "sscape: can't grab DMA %d\n", dma[dev]); + snd_printk(KERN_ERR "can't grab DMA %d\n", dma[dev]); goto _release_region; } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1030,13 +998,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + snd_printk(KERN_ERR "Invalid IRQ %d\n", irq[dev]); return -ENXIO; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,7 +1041,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + snd_printk(KERN_ERR "No AD1845 device at 0x%lx, IRQ %d\n", wss_port[dev], irq[dev]); goto _release_dma; } @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "Failed to register sound card\n"); goto _release_card; } -- 1.6.0.3 --------------------------------------------------------------- 2012 rok wcale nie musi by� tragiczny! Kliknij >>> http://link.interia.pl/f238b

2 1

[alsa-devel] Patches for alsactl init db
by Dan Chen 02 Oct '09

02 Oct '09

Hi, Attached are two patches against alsa-utils.git HEAD that create an intel8x0 init db file for ICH4. This is necessary from bug reports in Debian and Ubuntu that state that External Amplifier must be muted to have audible (analog) playback by default on the Sony VAIO VGN-T350P. Thanks, Dan

3 5

[alsa-devel] Help! Playing multiple audio stream using alsa libraries
by Arjun Sambamoorthy 02 Oct '09

02 Oct '09

Hello all, I am trying to play two audio streams using alsa libraries at the same time, but I am not able to do so. I could play a single audio stream though. I get bi-directional G711 MU Law RTP streams and I am using the sample playback capture code from the tutorial section for playing it. I am using the Non blocking and interleaved mode for playing the streams. The alsa library plays both the audio streams for a few seconds but it is incomprehensible and after few seconds I get the error "write to audio interface failed (broken pipe)". I also tried opening two pcm handles for each stream to the same device ("default"), but the second handle is unusable gives the same error "write to audio interface failed (broken pipe)". I have the same problem playing two audio streams to the /dev/dsp device. I am pretty sure I am not doing it the right way. Any help is highly appreciated or pointer to links that explains how to play multiple streams using alsa. Thanks Arjunen

1 1

[alsa-devel] [PATCH 1/3] ASoC: DaVinci: i2s, reduce underruns by combining into 1 element
by Troy Kisky 01 Oct '09

01 Oct '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 Kevin's temp/asoc branch --- 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..a3d2aa1 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 disable_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 12a6c54..081b2d4 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, +}; + static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { .name = "I2S PCM Stereo out", }; @@ -113,6 +130,27 @@ struct davinci_mcbsp_dev { u32 pcr; struct clk *clk; struct davinci_pcm_dma_params *dma_params[2]; + /* + * 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 disable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -359,6 +397,8 @@ 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); @@ -388,27 +428,22 @@ 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; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && !dev->disable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; + 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); @@ -524,7 +559,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) + dev->disable_channel_combine = pdata->disable_channel_combine; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; -- 1.5.6.3

6 28