August 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: s3c24xx platform: Fix s3c2410_dma_started called at wrong time
by Shine Liu 25 Aug '09

25 Aug '09

s3c24xx dma has the auto reload feature, when the the trnasfer is done, CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So the transmission is repeated. IRQ is issued while auto reload occurs. We change the DISRC and DCON[19:0] in the ISR, but at this time, the auto reload has been performed already. The first block is being re-transmitted by the DMA. So we need rewrite the DISRC and DCON[19:0] for the next block immediatly after the this block has been started to be transported. The function s3c2410_dma_started() is for this perpose, which is called in the form of "s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger(). But it is not correct. DMA transmission won't start until DMA REQ signal arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in s3c24xx_i2s_trigger(). In the current framework, s3c24xx_pcm_trigger() is always called before s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in this function. However, s3c2410_dma_started() is dma related, to call this function we should provide the channel number, which is given by substream->runtime->private_data->params->channel. The private_data points to a struct s3c24xx_runtime_data object, which is define in s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c To move the struct s3c24xx_runtime_data defination to s3c24xx-pcm.h is one solution. If not moved, the patch shold be like: Signed-off-by: Shine Liu <shinel(a)foxmail.com> ---------------------------------------------------------------- --- a/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-20 16:06:14.000000000 +0800 @@ -28,4 +28,6 @@ extern struct snd_soc_platform s3c24xx_soc_platform; extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream); + #endif --- a/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-20 15:56:46.000000000 +0800 @@ -66,6 +66,28 @@ struct s3c24xx_pcm_dma_params *params; }; +/* s3c24xx_pcm_dma_started + * + * To load the second dma buffer immediately after the + * first dma buffer has been started transferring, so that + * in the dma autoreload situation, the first data block + * won't be re-transferred after the first irq issued. + * + * This function is called in the s3c24xx-i2s-trigger() of + * s3c24xx-i2s.c, just after the dma transfer is really started. + * + * It's ugly here. Need a elegant way to fix this bug. + * FIXME + * + */ + +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); + +} + /* s3c24xx_pcm_enqueue * * place a dma buffer onto the queue for the dma system @@ -255,7 +277,6 @@ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: --- a/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-20 16:08:06.000000000 +0800 @@ -296,6 +296,9 @@ s3c24xx_snd_rxctrl(1); else s3c24xx_snd_txctrl(1); + + /* DMA transfer is really started, equeue the next buffer */ + s3c24xx_pcm_dma_started(substream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND:

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 25 Aug '09

25 Aug '09

The following changes since commit c6ea2af76ab79b82a0bfed313e4c2b68b10c2bc1: Takashi Iwai (1): ASoC: Remove unneeded inclusion of linux/regulator/consumer.h 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 Janusz Krzysztofik (1): ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 Jarkko Nikula (3): OMAP: McBSP: Use textual values in DMA operating mode sysfs files ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop Mark Brown (4): ASoC: Add input and output AIF widgets ASoC: WM8993 digital mixing support ASoC: Select core DMA when building for S3C64xx Merge branch 'topic/digital-mixing' into for-2.6.32 arch/arm/plat-omap/dma.c | 10 +++ arch/arm/plat-omap/include/mach/mcbsp.h | 2 - arch/arm/plat-omap/mcbsp.c | 132 ++++++++++++------------------ include/sound/soc-dapm.h | 8 ++ sound/soc/codecs/wm8993.c | 88 ++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 13 +--- sound/soc/s3c24xx/Kconfig | 1 + sound/soc/soc-dapm.c | 58 ++++++++++---- 8 files changed, 199 insertions(+), 113 deletions(-)

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[alsa-devel] [PATCH 0/3] Few patches to recent OMAP McBSP and ASoC changes
by Jarkko Nikula 25 Aug '09

25 Aug '09

Hi I would like to add these tree patches to recent set: [PATCHv5 00/20] OMAP ASoC changes in DMA utilization http://mailman.alsa-project.org/pipermail/alsa-devel/2009-August/020619.html These are actually independent patches but probably easier to handle as with the set above since some of the patches are touching both ASoC and OMAP subsystems. First patch is the most important since it changes the user space API for McBSP DMA operating mode selection. It would be more difficult to change it later if software starts to use current numeric value based operating mode selection in sysfs files. -- Jarkko

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[alsa-devel] multiple devices or one device w/ multiple subdevices?
by pl bossart 25 Aug '09

25 Aug '09

Hi, I have a piece of custom hardware that provides 7 independent PCM outputs. This is not intended for multichannel playback, the content on each output can be completely different; playback will not be synchronized and will start separately. The only restriction is that all outputs rely on the same baserate and share the same bitclock, i.e. all the outputs work at 48kHz. _From the mailling list archives, subdevices seem to be relevant when doing hw mixing or when lots of streams are handled. None of this applies to my case. Also it looks like PulseAudio knows nothing about subdevices in its detection loop, meaning the user would need to manually add some load-module commands in the PulseAudio configuration files. And last but not least, I didn't see very many lines of code (and no documentation) that showed how subdevices are registered/handled, etc. My conclusion is that multiple devices would seem like a much better option. But I wanted to check with the community that I am on the right track. Thanks for your feedback! - P

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[alsa-devel] [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
by Janusz Krzysztofik 25 Aug '09

25 Aug '09

Implement DMA channel self linking on OMAP1510 using AUTO_INIT and REPEAT flags of the DMA CCR register. Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Note: In its current form, enable_lnk() seems not very consistent (ie. result of a conditional assignment is then overwritten unconditionally). As rather different from OMAP1510 that I am familiar with, I am not quite sure how support for other OMAP models should really look like. So I decided to leave it as is and put my bit in a separate block, adding an additional FIXME comment. --- linux-2.6.31-rc5/arch/arm/plat-omap/dma.c.orig 2009-08-07 13:38:02.000000000 +0200 +++ linux-2.6.31-rc5/arch/arm/plat-omap/dma.c 2009-08-10 09:01:41.000000000 +0200 @@ -653,8 +653,21 @@ static inline void enable_lnk(int lch) { u32 l; + if (omap_dma_in_1510_mode()) { + /* only self linking is supported */ + if (dma_chan[lch].next_lch == lch) { + /* set AUTO_INIT | REPEAT bits in CCR */ + l = dma_read(CCR(lch)); + l |= 3 << 8; + dma_write(l, CCR(lch)); + } + return; + } + l = dma_read(CLNK_CTRL(lch)); + /* FIXME: + * this conditional assignment is then overwritten unconditionally */ if (cpu_class_is_omap1()) l &= ~(1 << 14); @@ -675,13 +688,21 @@ static inline void disable_lnk(int lch) { u32 l; - l = dma_read(CLNK_CTRL(lch)); + if (omap_dma_in_1510_mode()) + l = dma_read(CCR(lch)); + else + l = dma_read(CLNK_CTRL(lch)); /* Disable interrupts */ if (cpu_class_is_omap1()) { dma_write(0, CICR(lch)); - /* Set the STOP_LNK bit */ - l |= 1 << 14; + if (omap_dma_in_1510_mode()) { + /* Unset AUTO_INIT & REPEAT bits */ + l &= ~(3 << 8); + } else { + /* Set the STOP_LNK bit */ + l |= 1 << 14; + } } if (cpu_class_is_omap2()) { @@ -690,7 +711,10 @@ static inline void disable_lnk(int lch) l &= ~(1 << 15); } - dma_write(l, CLNK_CTRL(lch)); + if (omap_dma_in_1510_mode()) + dma_write(l, CCR(lch)); + else + dma_write(l, CLNK_CTRL(lch)); dma_chan[lch].flags &= ~OMAP_DMA_ACTIVE; } @@ -928,7 +952,7 @@ void omap_start_dma(int lch) { u32 l; - if (!omap_dma_in_1510_mode() && dma_chan[lch].next_lch != -1) { + if (dma_chan[lch].next_lch != -1) { int next_lch, cur_lch; char dma_chan_link_map[OMAP_DMA4_LOGICAL_DMA_CH_COUNT]; @@ -979,7 +1003,7 @@ void omap_stop_dma(int lch) { u32 l; - if (!omap_dma_in_1510_mode() && dma_chan[lch].next_lch != -1) { + if (dma_chan[lch].next_lch != -1) { int next_lch, cur_lch = lch; char dma_chan_link_map[OMAP_DMA4_LOGICAL_DMA_CH_COUNT]; @@ -1130,7 +1154,7 @@ int omap_dma_running(void) */ void omap_dma_link_lch(int lch_head, int lch_queue) { - if (omap_dma_in_1510_mode()) { + if (omap_dma_in_1510_mode() && (lch_head != lch_queue)) { printk(KERN_ERR "DMA linking is not supported in 1510 mode\n"); BUG(); return; @@ -1152,7 +1176,7 @@ EXPORT_SYMBOL(omap_dma_link_lch); */ void omap_dma_unlink_lch(int lch_head, int lch_queue) { - if (omap_dma_in_1510_mode()) { + if (omap_dma_in_1510_mode() && (lch_head != lch_queue)) { printk(KERN_ERR "DMA linking is not supported in 1510 mode\n"); BUG(); return; @@ -1822,7 +1846,7 @@ static int omap1_dma_handle_ch(int ch) if (unlikely(csr & OMAP_DMA_DROP_IRQ)) printk(KERN_WARNING "DMA synchronization event drop occurred " "with device %d\n", dma_chan[ch].dev_id); - if (likely(csr & OMAP_DMA_BLOCK_IRQ)) + if (likely(csr & OMAP_DMA_BLOCK_IRQ) && (dma_chan[ch].next_lch != ch)) dma_chan[ch].flags &= ~OMAP_DMA_ACTIVE; if (likely(dma_chan[ch].callback != NULL)) dma_chan[ch].callback(ch, csr, dma_chan[ch].data);

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Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Aug '09

24 Aug '09

2009/8/24 Takashi Iwai <tiwai(a)suse.de> > > At Mon, 24 Aug 2009 17:10:10 +0400, > Denis Kup wrote: > > > > Sorry, I was very busy last two months. We don't finish with this theme. > > What about my patch? > > Please check with the latest sound GIT tree (or alsa-driver snapshot) > whether it still work and repost the patch. There have been lots of > changes in HD-audio driver code, so an old patch might not work any > more. > > > thanks, > > Takashi I tested patch with alsa-driver-1.0.20.40.g13311.579.gcf341. All works as it should. It's diff between patched version( patch_realtek_new.c ) and unpatched: dener@linux-wpg9:~/Documents/1> diff ./patch_realtek_new.c ./patch_realtek_old.c 223d222 < ALC888_ACER_ASPIRE_7730G, 8341,8366d8339 < static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { < {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, < {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, < {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, < { } /* end */ < }; < < static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) < { < unsigned int present; < < present = snd_hda_codec_read(codec, 0x15, 0, < AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; < snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, < HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); < snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, < HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); < } < < static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, < unsigned int res) < { < if ((res >> 26) == ALC880_HP_EVENT) < alc888_acer_aspire_7730G_front_automute(codec); < } < 8558d8530 < [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", 8605d8576 < SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), 9094,9107d9064 < }, < [ALC888_ACER_ASPIRE_7730G] = { < .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, < .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, < .num_dacs = ARRAY_SIZE(alc883_dac_nids), < .dac_nids = alc883_dac_nids, < .dig_out_nid = ALC883_DIGOUT_NID, < .dig_in_nid = ALC883_DIGIN_NID, < .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), < .channel_mode = alc883_3ST_6ch_modes, < .need_dac_fix = 1, < .input_mux = &alc883_capture_source, < .unsol_event = alc888_acer_aspire_7730G_unsol_event, < .init_hook = alc888_acer_aspire_7730G_front_automute,

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[alsa-devel] [PATCHv4 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 24 Aug '09

24 Aug '09

Hello again guys, Here is version 4 of these changes. No real changes from previous versions. The only thing is that now I've rebased on top of Takashi's tree: http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=summary as there are pending things there which Jarkko has pointed on previous series. BR, Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 78 ++++++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 511 insertions(+), 14 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Aug '09

24 Aug '09

2009/6/24 Denis Kup <dener.kup(a)gmail.com>: > 2009/6/24 Takashi Iwai <tiwai(a)suse.de>: >> At Wed, 24 Jun 2009 16:09:58 +0400, >> Denis Kup wrote: >>> >>> Hey, when i first installed alsa to my laptop, there no sound from >>> subwoofer, jacks were map very strange (2 surrounds for ex.). I try >>> all models from the list, but all don't support my subwoofer. After a >>> weak of hard work i made a patch that enables ACER_ASPIRE_7730G model >>> for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal >>> frontal speakers and internal subwoofer.You can see acer.com for more >>> info about hardware and jacks. >>> >>> It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: >> >> Can you try Emilio's patch recently posted (and pass >> model=acer-aspire-6530g) with the very latest alsa-driver-snapshot? >> This essentially includes your changes. It's in: >> From: Emilio López <buhitoescolar(a)gmail.com> >> Subject: Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G >> Date: Tue, 23 Jun 2009 22:32:45 -0300 >> Message-ID: <d54969c50906231832w48ce7315uc5cf8b3f81534c76(a)mail.gmail.com> >> >>> After changing file i install driver, and it automatically detect its >>> model (maybe there are old settings from previous install, so i dont >>> 100% sure that it was detected). With this settings subwoofer works >>> perfect, internal stereo and subwoofer mute when headphone are plugged >>> all works, except 1 thing: subwoofer uses center channel instead LFE. >>> To fix it i use such asound.conf file: >> ... >>> Is there any chance to swap Center and LFE in patch_realtek.c? >> >> Unlikely as ALC888* doesn't support the channel-swap feature (although >> some other ALC codecs do). >> >> >> thanks, >> >> Takashi >> > > You are talking about that > http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html > ?? Its right link to drivers http://www.alsa-project.org/snapshot/ ?? > And there is easier way to apply patches than manually adding strings > to files? > I manage with installing. I use alsa-driver-snapshot.tar.gz http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ and this Amilo changes http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html . But this patch has smaller functionality against i have. There is advantages and disadvantages of both patches: My patch: + 1) when black jack plugged, internal speakers and subwoofer mute + 2) there is 3 maps for jacks: (black/pink/blue) 2ch: front/ext mic/ (something i dont know, it should be line in), 4ch: front/ ext mic/ side 6ch: front/ CLFE / side; (its changes in channel mode option in mixer) + 3) internal mic works + 4) there are separate boost mic channels for internal and external mics - 5) internal subwoofer controls with Center channel in mixer ? 6) i dont know how line in works (i will check it when reinstall to my patch) ? 7) i dont know how spdif works, because i have not any spdif speakers in house Amilo patch: - 1) when black jack pluged, only internal speaker mutes, subwoofer continue to work (it has no meaning when i listen to music in headphones) - 2) there is only 1 map for jacks: front/ext mic/ (something i dont know, it should be line in) - 3) there no side and center channels - 3) internal mic dont work - 4) there no line in in input source menu in mixer + 5) Subwoofer controls with LFE channel in mixer ? 7) i dont know how spdif works, because i have not any spdif speakers in house

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[alsa-devel] [RFC] [PATCH 3/3] ASoC: OMAP: Enhance OMAP1510 DMA progress software counter
by Janusz Krzysztofik 24 Aug '09

24 Aug '09

Enhance period index accuracy, particularly just before buffer rewind, by making use of DMA interrupt status flags in addition to simply counting interrupts This patch applies on top of patch 2 from this series: [RFC][PATCH 2/3] ASoC: OMAP: Make use of DMA channel self linking on OMAP1510 Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c.orig 2009-08-10 08:31:08.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c 2009-08-10 08:55:09.000000000 +0200 @@ -68,13 +68,28 @@ static void omap_pcm_dma_irq(int ch, u16 * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); - if (prtd->period_index >= 0) { - if (++prtd->period_index == runtime->periods) { + if (stat & OMAP_DMA_LAST_IRQ) { + /* last period of a buffer has been started */ + if (prtd->period_index == (runtime->periods - 1)) { + /* we are in sync, do nothing */ + spin_unlock_irqrestore(&prtd->lock, flags); + return; + } else if (prtd->period_index >= 0) { + /* possible IRQ loss, update the pointer */ + prtd->period_index = runtime->periods - 1; + } + } else if (prtd->period_index >= 0) { + /* playback in progress - increment the counter, + * check for end of buffer */ + if ((++prtd->period_index >= runtime->periods) || + (stat & OMAP_DMA_BLOCK_IRQ)) { + /* end of buffer reached, loop back */ prtd->period_index = 0; } } spin_unlock_irqrestore(&prtd->lock, flags); - } + } else if (stat & OMAP_DMA_LAST_IRQ) + return; snd_pcm_period_elapsed(substream); } @@ -175,7 +190,8 @@ static int omap_pcm_prepare(struct snd_p dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | + (cpu_is_omap1510() ? OMAP_DMA_LAST_IRQ : 0)); return 0; }

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[alsa-devel] [PATCH 1/2] ASoC: Add input and output AIF widgets
by Mark Brown 24 Aug '09

24 Aug '09

Currently DAPM interfaces with the audio streams to and from the processor at the DAC and ADC widgets. As the digital capabilities of parts increases this is becoming a less and less able to meet the needs of parts. To meet the needs of these devices create new widgets interfacing with the TDM bus but not integrated into any other functionality. Audio can then be routed to and from these widgets using existing routing widgets. A slot number is provided in the definition but this is currently not used yet. This is intended to support devices which can use more than one TDM slot on a single interface. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 8 ++++++ sound/soc/soc-dapm.c | 58 ++++++++++++++++++++++++++++++++------------- 2 files changed, 49 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 35814ce..3388405 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -137,6 +137,12 @@ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ +#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } +#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} @@ -312,6 +318,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_aif_in, /* audio interface input */ + snd_soc_dapm_aif_out, /* audio interface output */ }; /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8846131..a225e5a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,17 +55,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_micbias] = 2, - [snd_soc_dapm_mic] = 3, - [snd_soc_dapm_mux] = 4, - [snd_soc_dapm_value_mux] = 4, - [snd_soc_dapm_dac] = 5, - [snd_soc_dapm_mixer] = 6, - [snd_soc_dapm_mixer_named_ctl] = 6, - [snd_soc_dapm_pga] = 7, - [snd_soc_dapm_adc] = 8, - [snd_soc_dapm_hp] = 9, - [snd_soc_dapm_spk] = 9, - [snd_soc_dapm_post] = 10, + [snd_soc_dapm_aif_in] = 3, + [snd_soc_dapm_aif_out] = 3, + [snd_soc_dapm_mic] = 4, + [snd_soc_dapm_mux] = 5, + [snd_soc_dapm_value_mux] = 5, + [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_mixer] = 7, + [snd_soc_dapm_mixer_named_ctl] = 7, + [snd_soc_dapm_pga] = 8, + [snd_soc_dapm_adc] = 9, + [snd_soc_dapm_hp] = 10, + [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_post] = 11, }; static int dapm_down_seq[] = { @@ -81,8 +83,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_micbias] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_value_mux] = 9, - [snd_soc_dapm_supply] = 10, - [snd_soc_dapm_post] = 11, + [snd_soc_dapm_aif_in] = 10, + [snd_soc_dapm_aif_out] = 10, + [snd_soc_dapm_supply] = 11, + [snd_soc_dapm_post] = 12, }; static void pop_wait(u32 pop_time) @@ -224,6 +228,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: p->connect = 1; break; /* does effect routing - dynamically connected */ @@ -497,8 +503,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (widget->id == snd_soc_dapm_supply) return 0; - if (widget->id == snd_soc_dapm_adc && widget->active) - return 1; + switch (widget->id) { + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + if (widget->active) + return 1; + default: + break; + } if (widget->connected) { /* connected pin ? */ @@ -537,8 +549,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return 0; /* active stream ? */ - if (widget->id == snd_soc_dapm_dac && widget->active) - return 1; + switch (widget->id) { + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + if (widget->active) + return 1; + default: + break; + } if (widget->connected) { /* connected pin ? */ @@ -1052,6 +1070,8 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: if (w->name) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); @@ -1342,6 +1362,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, case snd_soc_dapm_pre: case snd_soc_dapm_post: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: list_add(&path->list, &codec->dapm_paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -1444,9 +1466,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) dapm_new_mux(codec, w); break; case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: -- 1.6.3.3

1 1

[alsa-devel] No Sound - X-Fi CA0110 IBG - Ubuntu 910 with Kernel 2.6.31-6
by David Favor 24 Aug '09

24 Aug '09

root@home1:~# uname -r 2.6.31-6-generic root@home1:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Generic [HD-Audio Generic], device 0: CA0110 Analog [CA0110 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Generic [HD-Audio Generic], device 1: CA0110 Digital [CA0110 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 root@home1:~# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.20. root@home1:~# cat /proc/asound/card0/codec#1 Codec: Creative CA0110-IBG Address: 1 Function Id: 0x1 Vendor Id: 0x1102000a Subsystem Id: 0x11021007 Revision Id: 0x100000 No Modem Function Group found Default PCM: N/A Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=255, o=255, i=255, unsolicited=1, wake=1 root@home1:~# lsmod | grep snd snd_hda_codec_ca0110 7868 1 snd_hda_intel 26600 6 snd_hda_codec 69276 2 snd_hda_codec_ca0110,snd_hda_intel snd_pcm 75360 4 snd_hda_intel,snd_hda_codec snd_timer 22244 1 snd_pcm snd 59204 15 snd_hda_codec_ca0110,snd_hda_intel,snd_hda_codec,snd_pcm,snd_timer soundcore 7264 1 snd snd_page_alloc 9156 2 snd_hda_intel,snd_pcm root@home1:~# lspci -vvnn 0d:00.0 Audio device [0403]: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG [1102:0009] Subsystem: Creative Labs Device [1102:0018] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 26 Region 0: Memory at fbefc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel I find no mention anywhere of anyone getting this card to work. Someone let me know if you have a working card or if I should toss this one. Thanks.

3 4

[alsa-devel] [PATCH] hda: move open coded tricks into get_wcaps_channels()
by Wu Fengguang 24 Aug '09

24 Aug '09

Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_local.h | 10 ++++++++++ sound/pci/hda/hda_proc.c | 4 +--- sound/pci/hda/patch_atihdmi.c | 3 +-- 3 files changed, 12 insertions(+), 5 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_local.h 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_local.h 2009-08-24 09:49:44.000000000 +0800 @@ -410,6 +410,16 @@ static inline u32 get_wcaps(struct hda_c /* get the widget type from widget capability bits */ #define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT) +static inline unsigned int get_wcaps_channels(u32 wcaps) +{ + unsigned int chans; + + chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = ((chans << 1) | 1) + 1; + + return chans; +} + u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); --- sound-2.6.orig/sound/pci/hda/hda_proc.c 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_proc.c 2009-08-24 09:49:44.000000000 +0800 @@ -515,9 +515,7 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, get_wid_type_name(wid_type), wid_caps); if (wid_caps & AC_WCAP_STEREO) { - unsigned int chans; - chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13; - chans = ((chans << 1) | 1) + 1; + unsigned int chans = get_wcaps_channels(wid_caps); if (chans == 2) snd_iprintf(buffer, " Stereo"); else --- sound-2.6.orig/sound/pci/hda/patch_atihdmi.c 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_atihdmi.c 2009-08-24 09:49:44.000000000 +0800 @@ -141,8 +141,7 @@ static int atihdmi_build_pcms(struct hda /* FIXME: we must check ELD and change the PCM parameters dynamically */ chans = get_wcaps(codec, CVT_NID); - chans = (chans & AC_WCAP_CHAN_CNT_EXT) >> 13; - chans = ((chans << 1) | 1) + 1; + chans = get_wcaps_channels(chans); info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; return 0;

2 1

[PATCH] ASoC: add missing inclusion of debugfs.h
by Takashi Iwai 24 Aug '09

24 Aug '09

To fix compile errors. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2f5295d..7b74547 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -37,6 +37,7 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/jiffies.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 1.6.3.3

1 0

[alsa-devel] [PATCH] ali5451: remove dead code
by Bartlomiej Zolnierkiewicz 23 Aug '09

23 Aug '09

From: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> Subject: [PATCH] ali5451: remove dead code Remove code covered by #if/endif 0 and #ifdef/endif CODEC_RESET (CODEC_RESET is never defined). Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 65 -------------------------------------------- 1 file changed, 65 deletions(-) Index: b/sound/pci/ali5451/ali5451.c =================================================================== --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -477,45 +477,6 @@ static int snd_ali_reset_5451(struct snd return 0; } -#ifdef CODEC_RESET - -static int snd_ali_reset_codec(struct snd_ali *codec) -{ - struct pci_dev *pci_dev; - unsigned char bVal; - unsigned int dwVal; - unsigned short wCount, wReg; - - pci_dev = codec->pci_m1533; - - pci_read_config_dword(pci_dev, 0x7c, &dwVal); - pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000); - udelay(5000); - pci_read_config_dword(pci_dev, 0x7c, &dwVal); - pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff); - udelay(5000); - - bVal = inb(ALI_REG(codec,ALI_SCTRL)); - bVal |= 0x02; - outb(ALI_REG(codec,ALI_SCTRL),bVal); - udelay(5000); - bVal = inb(ALI_REG(codec,ALI_SCTRL)); - bVal &= 0xfd; - outb(ALI_REG(codec,ALI_SCTRL),bVal); - udelay(15000); - - wCount = 200; - while (wCount--) { - wReg = snd_ali_codec_read(codec->ac97, AC97_POWERDOWN); - if ((wReg & 0x000f) == 0x000f) - return 0; - udelay(5000); - } - return -1; -} - -#endif - /* * ALI 5451 Controller */ @@ -560,22 +521,6 @@ static void snd_ali_disable_address_inte outl(gc, ALI_REG(codec, ALI_GC_CIR)); } -#if 0 /* not used */ -static void snd_ali_enable_voice_irq(struct snd_ali *codec, - unsigned int channel) -{ - unsigned int mask; - struct snd_ali_channel_control *pchregs = &(codec->chregs); - - snd_ali_printk("enable_voice_irq channel=%d\n",channel); - - mask = 1 << (channel & 0x1f); - pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten)); - pchregs->data.ainten |= mask; - outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten)); -} -#endif - static void snd_ali_disable_voice_irq(struct snd_ali *codec, unsigned int channel) { @@ -676,16 +621,6 @@ static void snd_ali_free_channel_pcm(str } } -#if 0 /* not used */ -static void snd_ali_start_voice(struct snd_ali *codec, unsigned int channel) -{ - unsigned int mask = 1 << (channel & 0x1f); - - snd_ali_printk("start_voice: channel=%d\n",channel); - outl(mask, ALI_REG(codec,codec->chregs.regs.start)); -} -#endif - static void snd_ali_stop_voice(struct snd_ali *codec, unsigned int channel) { unsigned int mask = 1 << (channel & 0x1f);

2 1

[alsa-devel] [PATCH] ali5451: fix timeout handling in snd_ali_{codecs, timer}_ready()
by Bartlomiej Zolnierkiewicz 23 Aug '09

23 Aug '09

From: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> Subject: [PATCH] ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready() Modify loops in such way that the register value is checked also after the timeout condition, just in case the heavy interrupt load etc. caused the thread to sleep for the time period exceeding the timeout value. While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready(). Reported-by: Jack Byer <ojbyer(a)usa.net> Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) Index: b/sound/pci/ali5451/ali5451.c =================================================================== --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -314,8 +314,11 @@ static int snd_ali_codec_ready(struct sn res = snd_ali_5451_peek(codec,port); if (!(res & 0x8000)) return 0; + if (!time_after_eq(end_time, jiffies)) + break; schedule_timeout_uninterruptible(1); - } while (time_after_eq(end_time, jiffies)); + } while (1); + snd_ali_5451_poke(codec, port, res & ~0x8000); snd_printdd("ali_codec_ready: codec is not ready.\n "); return -EIO; @@ -327,15 +330,17 @@ static int snd_ali_stimer_ready(struct s unsigned long dwChk1,dwChk2; dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER); - dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); end_time = jiffies + msecs_to_jiffies(250); do { dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); if (dwChk2 != dwChk1) return 0; + if (!time_after_eq(end_time, jiffies)) + break; schedule_timeout_uninterruptible(1); - } while (time_after_eq(end_time, jiffies)); + } while (1); + snd_printk(KERN_ERR "ali_stimer_read: stimer is not ready.\n"); return -EIO; }

2 3

[alsa-devel] [PATCH] ASoC: free socdev if init_card() fails in wm9705_soc_probe()
by Roel Kluin 23 Aug '09

23 Aug '09

Free socdev if snd_soc_init_card() fails. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- It doesn't seem to happen in snd_soc_init_card(), but please review. diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index fa88b46..e7d2840 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -406,7 +406,7 @@ static int wm9705_soc_probe(struct platform_device *pdev) ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register card\n"); - goto pcm_err; + goto reset_err; } return 0;

2 1

Re: [alsa-devel] ALSA synchronization
by arif setiawan 23 Aug '09

23 Aug '09

Hi, We are working on audio processing (encryption-decryption) and audio will be sent using radio. Our configuration is : Audio source - ALSA audio in - encryption - ALSA audio out - transmission over radio (now we still use audio cable) - ALSA audio in - decryption - ALSA audio out - speaker. So data will be sent through radio channel and its converted to analog form by ALSA PCM during playback. We want to solve synchronization issue by using data header so that we can decrypt correct data block. So far we still unable to verify that data we received in the decryption side is similar to data that we sent from encryption box. Supposed that we filled buffer in encryption box with some values, when we check buffer in decryption box, we can't get same data. Our guess that data somehow changed in digital-analog/analog-digital conversion or we might make mistakes in filling the buffer so data is not compatible with ALSA PCM. Any suggestion to solve this problem? Thank you, Best Regards, Arif

3 3

[alsa-devel] [PATCH] Convert WM97xx-battery to dev_pm_ops
by Marek Vasut 22 Aug '09

22 Aug '09

Hi, Anton, are you fine with this one? Mark, in case Anton is ok, would you mind applying? Thanks! Cheers! Marek Vasut

1 0

[alsa-devel] ALSA synchronization
by arif setiawan 21 Aug '09

21 Aug '09

Hi all, We are developing a AES based speech encryption module using beagleboard. We have two beagleboard: one for encryption and other for decryption. We use audio-in/audio-out for sending speech. We have problem with synchronization of audio stream from encryption board to decryption board. Since AES is block cipher, the buffer content received by decryption board must be the same as sent from encryption board. Right now, we want to use some bytes as data header, but we don't now how to make this work. Any idea how to solve synchronization in ALSA? Best Regards, Arif

2 1

[alsa-devel] [PATCH] various acer-mic-internal bugs solution
by epic 21 Aug '09

21 Aug '09

Several mic related bugs maybe solved with this patch or a derived one 1 -My internal-mic was not working and front jack mic sound was terrible (useless for headset mic). 2 -Detected offset level from inputs {line, mic}, this offset level increases with specific [boost] and/or [capture(NID 0x23)] gain. * I just played around several days compiling and testing alsa with my acer-alc268, checked all data communication with alc268, pin config etc. and tested different values for offset in pins. * Found internal-mic comes alive with a 50% offset v_reference and mic sound is much better. *This patch applies/solves only for alc268 with option model=acer in alsa-base.conf, but i think the others acer-alcxxx bugs maybe solved in the same way. changing PIN_VREF80 to PIN_VREF50 at NID 0x18 mic NID 0x19 internal-mic NID 0x1a line-in (not tested) *Don't know what can happen to other acer-alcxxx, so i only patched acer-alc268. But this patch solved my mic-internal issue and most important for me now i can use a headset mic. *It remains a level offset, lower than before, but mic is usable, the level is showed in the pictures. *Pictures: input with two channels and a positive offset in left channel (red), negative offset in rigth channel (blue) http://imagebin.ca/view/Y8TyujP.html working internal microphone ok http://imagebin.ca/view/nzTzeX.html Realtek datasheet 1.3 is available for alc268 2.6.26-2-686 #1 SMP i686 GNU/Linux debian/squeeze description: Computer product: Aspire 7720 vendor: Acer version: V1.17 TAGS: alc268 offset level internal mic microphone alsa hda intel realtek Signed-off-by: EliasTG <epicadocan a.t gmail.com> upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Tue Aug 18 21:27:39 UTC 2009 !!Linux Distribution !!------------------ Debian GNU/Linux squeeze/sid \n \l !!Kernel Information !!------------------ Kernel release: 2.6.26-2-686 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.20 Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 11 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1025:011e !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 snd-hda-intel: index=0 model=test !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : 0,-1,-1,-1,-1,-1,-1,-1 model : test,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC268 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0268 Subsystem Id: 0x1025011e Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=04, enabled=1 Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19840: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a30941: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0281304e: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4017952d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000010: OUT Pin Default 0x02451130: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Processing Coefficient: 0x03 Coefficient Index: 0x01 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x15 0x15] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x13 Codec: Conexant ID 2c06 Address: 1 Function Id: 0x2 Vendor Id: 0x14f12c06 Subsystem Id: 0x1025011e Revision Id: 0x100000 Modem Function Group: 0x2 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Aug 18 22:02 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Aug 18 22:02 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 Aug 18 22:02 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 24 Aug 18 22:02 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Aug 18 22:02 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Aug 18 05:24 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Aug 18 05:24 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0x9b300000 irq 11' Mixer name : 'Realtek ALC268' Components : 'HDA:10ec0268,1025011e,00100003 HDA:14f12c06,1025011e,00100000' Controls : 12 Simple ctrls : 9 Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 44 [69%] [-20.00dB] [on] Front Right: Playback 44 [69%] [-20.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 185 [73%] [-14.00dB] Front Right: Playback 185 [73%] [-14.00dB] Simple mixer control 'Line In Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 2 [100%] Front Right: 2 [100%] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 21 [68%] [15.00dB] [on] Front Right: Capture 21 [68%] [15.00dB] [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 12 Mono: Front Left: Playback 0 [0%] [-24.00dB] [off] Front Right: Playback 0 [0%] [-24.00dB] [off] Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 0 [0%] [-30.00dB] Front Right: Capture 0 [0%] [-30.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Internal Mic' 'Line' Item0: 'Mic' Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 2 [100%] Front Right: 2 [100%] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 44 value.1 44 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Master Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Mic Boost' value.0 2 value.1 2 } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Internal Mic Boost' value.0 2 value.1 2 } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Line In Boost' value.0 0 value.1 0 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1650 comment.dbmax 3000 iface MIXER name 'Capture Volume' value.0 21 value.1 21 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.8 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Internal Mic' comment.item.2 Line iface MIXER name 'Input Source' value Mic } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 12' comment.dbmin -2400 comment.dbmax 0 iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 } control.10 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Beep Playback Switch' value.0 false value.1 false } control.11 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 185 value.1 185 } control.12 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 120' comment.tlv '0000000100000008fffff44800000032' comment.dbmin -3000 comment.dbmax 3000 iface MIXER name 'Digital Capture Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_hwdep usb_storage nvidia ppdev parport_pc lp parport ipv6 sbp2 loop snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_dummy snd_seq_oss snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device mxl5005s zl10353 snd arc4 ecb crypto_blkcipher soundcore joydev dvb_usb_ce6230 dvb_usb iwl3945 snd_page_alloc dvb_core uvcvideo serio_raw mac80211 i2c_i801 acer_wmi firmware_class videodev v4l1_compat cfg80211 i2c_core psmouse pcspkr led_class evdev video output battery button ac wmi ext2 mbcache usbhid hid ff_memless ide_cd_mod cdrom ide_pci_generic sd_mod ata_generic uhci_hcd ahci sdhci libata ricoh_mmc mmc_core ohci1394 scsi_mod dock ieee1394 piix ide_core tg3 ehci_hcd usbcore intel_agp agpgart thermal processor fan thermal_sys dm_mirror dm_log dm_mod !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x12 0x411111f0 0x13 0x411111f0 0x14 0x02211020 0x15 0x99130110 0x16 0x411111f0 0x18 0x02a19840 0x19 0x99a30941 0x1a 0x0281304e 0x1c 0x411111f0 0x1d 0x4017952d 0x1e 0x02451130 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D1/init_pin_configs: 0x73 0x016a10f0 /sys/class/sound/hwC0D1/driver_pin_configs: /sys/class/sound/hwC0D1/user_pin_configs: /sys/class/sound/hwC0D1/init_verbs: Codec: Realtek ALC268 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0268 Subsystem Id: 0x1025011e Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=04, enabled=1 Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19840: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a30941: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0281304e: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4017952d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000010: OUT Pin Default 0x02451130: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Processing Coefficient: 0xa948 Coefficient Index: 0x02 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x15 0x15] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x13 [617100.073588] hda_codec:patch_alc268 here we go! v2 [617100.079773] hda_codec:snd_hda_codec_write: 0x170e01 [617100.079856] hda_codec:snd_hda_codec_read: 0x1f000d --> 0x0 [617100.079954] hda_codec:snd_hda_codec_read: 0x100f0000 --> 0x14f12c06 [617100.080028] hda_codec:snd_hda_codec_read: 0x100f0001 --> 0x0 [617100.080105] hda_codec:snd_hda_codec_read: 0x100f0002 --> 0x100000 [617100.080170] hda_codec:snd_hda_codec_read: 0x100f0004 --> 0x20001 [617100.080257] hda_codec:snd_hda_codec_read: 0x102f0005 --> 0x102 [617100.080333] hda_codec:snd_hda_codec_read: 0x102f0004 --> 0x700004 [617100.080398] hda_codec:snd_hda_codec_read: 0x170f0009 --> 0xf00000 [617100.080464] hda_codec:snd_hda_codec_read: 0x171f0009 --> 0x100100 [617100.080548] hda_codec:snd_hda_codec_read: 0x172f0009 --> 0x0 [617100.080625] hda_codec:snd_hda_codec_read: 0x173f0009 --> 0x400100 [617100.080689] hda_codec:snd_hda_codec_read: 0x173f1c00 --> 0x16a10f0 [617100.080756] hda_codec:snd_hda_codec_read: 0x102f2000 --> 0x1025011e [617100.080762] hda_codec:snd_hda_codec_write: 0x10270500 [617100.110915] hda_codec:snd_hda_codec_read: 0x102f0500 --> 0x0 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f000a --> 0xe0560 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f000b --> 0x1 [617100.114318] hda_codec:snd_hda_codec_read: 0x8f000a --> 0x60160 [617100.114318] hda_codec:snd_hda_codec_read: 0x8f000b --> 0x1 [617100.114318] hda_codec:snd_hda_codec_write: 0x17ff00 [617100.114318] hda_codec:snd_hda_codec_write: 0x23b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x33b000 [617100.114318] hda_codec:snd_hda_codec_write: 0xe37000 [617100.114318] hda_codec:snd_hda_codec_write: 0xe70100 [617100.114318] hda_codec:snd_hda_codec_write: 0xf37000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1037000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470740 [617100.114318] hda_codec:snd_hda_codec_write: 0x15707c0 [617100.114318] hda_codec:snd_hda_codec_write: 0x1670740 [617100.114318] hda_codec:snd_hda_codec_write: 0x1870722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1970722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1c70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1d70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x143b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x153b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x163b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x183b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a3b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x1d37000 [617100.114318] hda_codec:snd_hda_codec_write: 0xf37180 [617100.114318] hda_codec:snd_hda_codec_write: 0x1037180 [617100.114318] hda_codec:snd_hda_codec_write: 0x2370100 [617100.114318] hda_codec:snd_hda_codec_write: 0x233b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x2470100 [617100.114318] hda_codec:snd_hda_codec_write: 0x243b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470c02 [617100.114318] hda_codec:snd_hda_codec_write: 0x1570c02 [617100.114318] hda_codec:snd_hda_codec_write: 0x1270720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1370720 [617100.114318] hda_codec:snd_hda_codec_write: 0x14707c0 [617100.114318] hda_codec:snd_hda_codec_write: 0x1570740 [617100.114318] hda_codec:snd_hda_codec_write: 0x1870722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a70722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470884 [617100.114318] hda_codec:snd_hda_codec_read: 0x14f0900 --> 0x0 [617100.114318] hda_codec:snd_hda_codec_read: 0x14ba000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x15ba000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x15b8000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f0012 --> 0x34040 [617100.114318] hda_codec:snd_hda_codec_read: 0x23f0012 --> 0x80051f0b [617100.135224] hda_codec:snd_hda_codec_read: 0x14f0900 --> 0x80000000 [617100.162363] hda_codec:snd_hda_codec_read: 0x18f000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x19f000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x1af000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x2ba000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x23a036 [617100.162363] hda_codec:snd_hda_codec_read: 0x2b8000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x239036 [617100.162363] hda_codec:snd_hda_codec_read: 0x3ba000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x33a036 [617100.162363] hda_codec:snd_hda_codec_read: 0x3b8000 --> 0x0 [617100.162377] hda_codec:snd_hda_codec_write: 0x339036 [617100.162387] hda_codec:snd_hda_codec_write: 0x143a000 [617100.162491] hda_codec:snd_hda_codec_read: 0x14b8000 --> 0x80 [617100.162498] hda_codec:snd_hda_codec_write: 0x1439000 [617100.162557] hda_codec:snd_hda_codec_read: 0x18b2000 --> 0x0 [617100.162621] hda_codec:snd_hda_codec_read: 0x18b0000 --> 0x0 [617100.162702] hda_codec:snd_hda_codec_read: 0x19b2000 --> 0x0 [617100.162789] hda_codec:snd_hda_codec_read: 0x19b0000 --> 0x0 [617100.162869] hda_codec:snd_hda_codec_read: 0x1ab2000 --> 0x0 [617100.162955] hda_codec:snd_hda_codec_read: 0x1ab0000 --> 0x0 [617100.163035] hda_codec:snd_hda_codec_read: 0x23ba000 --> 0x0 [617100.163122] hda_codec:snd_hda_codec_read: 0x23b8000 --> 0x0 [617100.163128] hda_codec:snd_hda_codec_write: 0x2339006 [617100.163136] hda_codec:snd_hda_codec_write: 0x233a080 [617100.163239] hda_codec:snd_hda_codec_read: 0x1db2000 --> 0x0 [617100.163306] hda_codec:snd_hda_codec_read: 0x1db0000 --> 0x0 [617100.163385] hda_codec:snd_hda_codec_read: 0xfb2001 --> 0x80 [617100.163452] hda_codec:snd_hda_codec_read: 0xfb0001 --> 0x80 [617100.163538] hda_codec:snd_hda_codec_read: 0x10b2001 --> 0x80 [617100.163615] hda_codec:snd_hda_codec_read: 0x10b0001 --> 0x80 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000a --> 0xe0560 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000b --> 0x1 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000d --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0012 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0004 --> 0x20023 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0011 --> 0x40000004 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1600 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1700 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1800 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1900 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1a00 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1500 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0009 --> 0x1d [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0012 --> 0x34040 [617165.253398] hda_codec:snd_hda_codec_read: 0x2ba000 --> 0x36 [617165.253398] hda_codec:snd_hda_codec_read: 0x2b8000 --> 0x36 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0600 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f000a --> 0xe0560 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f000b --> 0x1 [617165.253398] hda_codec:snd_hda_codec_read: 0x3f0009 --> 0x1d [617165.253398] hda_codec:snd_hda_codec_read: 0x3f0012 --> 0x34040 [617165.253398] hda_codec:snd_hda_codec_read: 0x3ba000 --> 0x36 [617165.255878] hda_codec:snd_hda_codec_read: 0x3b8000 --> 0x36 [617165.255942] hda_codec:snd_hda_codec_read: 0x3f0600 --> 0x0 [617165.256009] hda_codec:snd_hda_codec_read: 0x3f000a --> 0xe0560 [617165.256095] hda_codec:snd_hda_codec_read: 0x3f000b --> 0x1 [617165.256174] hda_codec:snd_hda_codec_read: 0x4f0009 --> 0xf00000 [617165.256239] hda_codec:snd_hda_codec_read: 0x5f0009 --> 0xf00000 [617165.256316] hda_codec:snd_hda_codec_read: 0x6f0009 --> 0x211 [617165.256383] hda_codec:snd_hda_codec_read: 0x6f0600 --> 0x0 [617165.256447] hda_codec:snd_hda_codec_read: 0x6f0d00 --> 0x0 [617165.256524] hda_codec:snd_hda_codec_read: 0x6f000a --> 0x1e05e0 [617165.256590] hda_codec:snd_hda_codec_read: 0x6f000b --> 0x1 [617165.256676] hda_codec:snd_hda_codec_read: 0x7f0009 --> 0x100111 [617165.256754] hda_codec:snd_hda_codec_read: 0x7f000e --> 0x1 [617165.256818] hda_codec:snd_hda_codec_read: 0x7f0200 --> 0x24 [617165.256884] hda_codec:snd_hda_codec_read: 0x7f0600 --> 0x0 [617165.256961] hda_codec:snd_hda_codec_read: 0x7f0400 --> 0x0 [617165.257025] hda_codec:snd_hda_codec_read: 0x7f000a --> 0x60160 [617165.257092] hda_codec:snd_hda_codec_read: 0x7f000b --> 0x1 [617165.257171] hda_codec:snd_hda_codec_read: 0x8f0009 --> 0x100111 [617165.257235] hda_codec:snd_hda_codec_read: 0x8f000e --> 0x1 [617165.257321] hda_codec:snd_hda_codec_read: 0x8f0200 --> 0x23 [617165.257398] hda_codec:snd_hda_codec_read: 0x8f0600 --> 0x0 [617165.257464] hda_codec:snd_hda_codec_read: 0x8f0400 --> 0x0 [617165.257529] hda_codec:snd_hda_codec_read: 0x8f000a --> 0x60160 [617165.257605] hda_codec:snd_hda_codec_read: 0x8f000b --> 0x1 [617165.257674] hda_codec:snd_hda_codec_read: 0x9f0009 --> 0xf00000 [617165.257759] hda_codec:snd_hda_codec_read: 0xaf0009 --> 0xf00000 [617165.257836] hda_codec:snd_hda_codec_read: 0xbf0009 --> 0xf00000 [617165.257903] hda_codec:snd_hda_codec_read: 0xcf0009 --> 0xf00000 [617165.257980] hda_codec:snd_hda_codec_read: 0xdf0009 --> 0xf00000 [617165.258047] hda_codec:snd_hda_codec_read: 0xef0009 --> 0x20010a [617165.258112] hda_codec:snd_hda_codec_read: 0xef000e --> 0x1 [617165.258188] hda_codec:snd_hda_codec_read: 0xef0200 --> 0x2 [617165.258255] hda_codec:snd_hda_codec_read: 0xef000d --> 0x80000000 [617165.258320] hda_codec:snd_hda_codec_read: 0xeb0000 --> 0x0 [617165.258397] hda_codec:snd_hda_codec_read: 0xff0009 --> 0x20010b [617165.258464] hda_codec:snd_hda_codec_read: 0xff000e --> 0x2 [617165.258528] hda_codec:snd_hda_codec_read: 0xff0200 --> 0x1d02 [617165.258605] hda_codec:snd_hda_codec_read: 0xff000d --> 0x80000000 [617165.258672] hda_codec:snd_hda_codec_read: 0xfb2000 --> 0x0 [617165.258756] hda_codec:snd_hda_codec_read: 0xfb0000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0xfb2001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0xfb0001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f0009 --> 0x20010b [617165.259833] hda_codec:snd_hda_codec_read: 0x10f000e --> 0x3 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f0200 --> 0x21d03 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f000d --> 0x80000000 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2002 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0002 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x11f0009 --> 0xf00000 [617165.259833] hda_codec:snd_hda_codec_read: 0x12f0009 --> 0x400001 [617165.259868] hda_codec:snd_hda_codec_read: 0x12f000c --> 0x20 [617165.259945] hda_codec:snd_hda_codec_read: 0x12f1c00 --> 0x411111f0 [617165.260012] hda_codec:snd_hda_codec_read: 0x12f0700 --> 0x20 [617165.260089] hda_codec:snd_hda_codec_read: 0x13f0009 --> 0x400001 [617165.260155] hda_codec:snd_hda_codec_read: 0x13f000c --> 0x20 [617165.260220] hda_codec:snd_hda_codec_read: 0x13f1c00 --> 0x411111f0 [617165.260298] hda_codec:snd_hda_codec_read: 0x13f0700 --> 0x20 [617165.260364] hda_codec:snd_hda_codec_read: 0x14f0009 --> 0x40018d [617165.260449] hda_codec:snd_hda_codec_read: 0x14f000e --> 0x1 [617165.260526] hda_codec:snd_hda_codec_read: 0x14f0200 --> 0xf [617165.260590] hda_codec:snd_hda_codec_read: 0x14f0012 --> 0x80000000 [617165.260657] hda_codec:snd_hda_codec_read: 0x14ba000 --> 0x0 [617165.260734] hda_codec:snd_hda_codec_read: 0x14b8000 --> 0x0 [617165.260798] hda_codec:snd_hda_codec_read: 0x14f000c --> 0x1003c [617165.260865] hda_codec:snd_hda_codec_read: 0x14f0c00 --> 0x2 [617165.260949] hda_codec:snd_hda_codec_read: 0x14f1c00 --> 0x2211020 [617165.261027] hda_codec:snd_hda_codec_read: 0x14f0700 --> 0xc0 [617165.261094] hda_codec:snd_hda_codec_read: 0x14f0800 --> 0x84 [617165.261179] hda_codec:snd_hda_codec_read: 0x15f0009 --> 0x40018d [617165.261256] hda_codec:snd_hda_codec_read: 0x15f000e --> 0x1 [617165.261322] hda_codec:snd_hda_codec_read: 0x15f0200 --> 0x10 [617165.261399] hda_codec:snd_hda_codec_read: 0x15f0012 --> 0x80000000 [617165.261464] hda_codec:snd_hda_codec_read: 0x15ba000 --> 0x80 [617165.261530] hda_codec:snd_hda_codec_read: 0x15b8000 --> 0x80 [617165.261615] hda_codec:snd_hda_codec_read: 0x15f000c --> 0x1003c [617165.261692] hda_codec:snd_hda_codec_read: 0x15f0c00 --> 0x2 [617165.261756] hda_codec:snd_hda_codec_read: 0x15f1c00 --> 0x99130110 [617165.261845] hda_codec:snd_hda_codec_read: 0x15f0700 --> 0x40 [617165.261921] hda_codec:snd_hda_codec_read: 0x15f0800 --> 0x0 [617165.262004] hda_codec:snd_hda_codec_read: 0x16f0009 --> 0x40010c [617165.262071] hda_codec:snd_hda_codec_read: 0x16f000e --> 0x1 [617165.262156] hda_codec:snd_hda_codec_read: 0x16f0200 --> 0xe [617165.262232] hda_codec:snd_hda_codec_read: 0x16f0012 --> 0x80000000 [617165.262299] hda_codec:snd_hda_codec_read: 0x16b8000 --> 0x80 [617165.262386] hda_codec:snd_hda_codec_read: 0x16f000c --> 0x10 [617165.262462] hda_codec:snd_hda_codec_read: 0x16f1c00 --> 0x411111f0 [617165.262551] hda_codec:snd_hda_codec_read: 0x16f0700 --> 0x40 [617165.262628] hda_codec:snd_hda_codec_read: 0x17f0009 --> 0xf00000 [617165.262693] hda_codec:snd_hda_codec_read: 0x18f0009 --> 0x40018f [617165.262780] hda_codec:snd_hda_codec_read: 0x18f000e --> 0x1 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0200 --> 0x2 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f000d --> 0x4f0200 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b2000 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b0000 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0012 --> 0x80000000 [617165.263834] hda_codec:snd_hda_codec_read: 0x18ba000 --> 0x80 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b8000 --> 0x80 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f000c --> 0x3734 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f1c00 --> 0x2a19840 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0700 --> 0x22 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0800 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x19f0009 --> 0x40008b [617165.263834] hda_codec:snd_hda_codec_read: 0x19f000d --> 0x4f0200 [617165.263838] hda_codec:snd_hda_codec_read: 0x19b2000 --> 0x0 [617165.263904] hda_codec:snd_hda_codec_read: 0x19b0000 --> 0x0 [617165.263969] hda_codec:snd_hda_codec_read: 0x19f000c --> 0x3724 [617165.264046] hda_codec:snd_hda_codec_read: 0x19f1c00 --> 0x99a30941 [617165.264114] hda_codec:snd_hda_codec_read: 0x19f0700 --> 0x22 [617165.264199] hda_codec:snd_hda_codec_read: 0x19f0800 --> 0x0 [617165.264276] hda_codec:snd_hda_codec_read: 0x1af0009 --> 0x40018f [617165.264343] hda_codec:snd_hda_codec_read: 0x1af000e --> 0x1 [617165.264429] hda_codec:snd_hda_codec_read: 0x1af0200 --> 0x2 [617165.264505] hda_codec:snd_hda_codec_read: 0x1af000d --> 0x4f0200 [617165.264570] hda_codec:snd_hda_codec_read: 0x1ab2000 --> 0x0 [617165.264657] hda_codec:snd_hda_codec_read: 0x1ab0000 --> 0x0 [617165.264733] hda_codec:snd_hda_codec_read: 0x1af0012 --> 0x80000000 [617165.264800] hda_codec:snd_hda_codec_read: 0x1aba000 --> 0x80 [617165.264884] hda_codec:snd_hda_codec_read: 0x1ab8000 --> 0x80 [617165.264961] hda_codec:snd_hda_codec_read: 0x1af000c --> 0x3734 [617165.265026] hda_codec:snd_hda_codec_read: 0x1af1c00 --> 0x281304e [617165.265114] hda_codec:snd_hda_codec_read: 0x1af0700 --> 0x22 [617165.265191] hda_codec:snd_hda_codec_read: 0x1af0800 --> 0x0 [617165.265258] hda_codec:snd_hda_codec_read: 0x1bf0009 --> 0xf00000 [617165.265335] hda_codec:snd_hda_codec_read: 0x1cf0009 --> 0x400001 [617165.265400] hda_codec:snd_hda_codec_read: 0x1cf000c --> 0x20 [617165.265467] hda_codec:snd_hda_codec_read: 0x1cf1c00 --> 0x411111f0 [617165.265545] hda_codec:snd_hda_codec_read: 0x1cf0700 --> 0x20 [617165.265610] hda_codec:snd_hda_codec_read: 0x1df0009 --> 0x400000 [617165.265697] hda_codec:snd_hda_codec_read: 0x1df000c --> 0x20 [617165.265773] hda_codec:snd_hda_codec_read: 0x1df1c00 --> 0x4017952d [617165.265842] hda_codec:snd_hda_codec_read: 0x1df0700 --> 0x20 [617165.265927] hda_codec:snd_hda_codec_read: 0x1ef0009 --> 0x400380 [617165.266003] hda_codec:snd_hda_codec_read: 0x1ef000e --> 0x1 [617165.266070] hda_codec:snd_hda_codec_read: 0x1ef0200 --> 0x6 [617165.266146] hda_codec:snd_hda_codec_read: 0x1ef000c --> 0x10 [617165.266210] hda_codec:snd_hda_codec_read: 0x1ef1c00 --> 0x2451130 [617165.266278] hda_codec:snd_hda_codec_read: 0x1ef0700 --> 0x40 [617165.266355] hda_codec:snd_hda_codec_read: 0x1ef0800 --> 0x0 [617165.266420] hda_codec:snd_hda_codec_read: 0x1ff0009 --> 0xf00000 [617165.266507] hda_codec:snd_hda_codec_read: 0x20f0009 --> 0xf00040 [617165.266584] hda_codec:snd_hda_codec_read: 0x20f0010 --> 0xa00 [617165.266651] hda_codec:snd_hda_codec_read: 0x20c0000 --> 0x3 [617165.266738] hda_codec:snd_hda_codec_read: 0x20d0000 --> 0x1 [617165.266815] hda_codec:snd_hda_codec_read: 0x21f0009 --> 0xf00000 [617165.267835] hda_codec:snd_hda_codec_read: 0x22f0009 --> 0xf00000 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0009 --> 0x30010d [617165.267835] hda_codec:snd_hda_codec_read: 0x23f000e --> 0x7 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0200 --> 0x1c1a1918 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0204 --> 0x121514 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0012 --> 0x80051f0b [617165.267835] hda_codec:snd_hda_codec_read: 0x23ba000 --> 0x80 [617165.267835] hda_codec:snd_hda_codec_read: 0x23b8000 --> 0x6 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0100 --> 0x0 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0009 --> 0x30010d [617165.267835] hda_codec:snd_hda_codec_read: 0x24f000e --> 0x7 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0200 --> 0x1c1a1918 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0204 --> 0x131514 [617165.267870] hda_codec:snd_hda_codec_read: 0x24f0012 --> 0x80051f0b [617165.267947] hda_codec:snd_hda_codec_read: 0x24ba000 --> 0x0 [617165.268012] hda_codec:snd_hda_codec_read: 0x24b8000 --> 0x0 [617165.268098] hda_codec:snd_hda_codec_read: 0x24f0100 --> 0x0 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: Acer Incorporated [ALI] Device [1025:011e] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 11 Region 0: Memory at 9b300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

2 1

[alsa-devel] [PATCH v3 4/4] sh: add FSI driver support for ms7724se
by Kuninori Morimoto 21 Aug '09

21 Aug '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o remove un-used dma-sh.h I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 arch/sh/boards/mach-se/7724/setup.c | 89 +++++++++++++++++++++++++++++++++++ 1 files changed, 89 insertions(+), 0 deletions(-) diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 9162081..eb144f6 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -22,6 +22,7 @@ #include <linux/usb/r8a66597.h> #include <video/sh_mobile_lcdc.h> #include <media/sh_mobile_ceu.h> +#include <sound/sh_fsi.h> #include <asm/io.h> #include <asm/heartbeat.h> #include <asm/sh_eth.h> @@ -246,6 +247,65 @@ static struct platform_device ceu1_device = { }, }; +/* FSI */ +/* + * FSI-A use external clock which came from ak464x. + * So, we should change parent of fsi + */ +#define FCLKACR 0xa4150008 +static void fsimck_init(struct clk *clk) +{ + u32 status = ctrl_inl(clk->enable_reg); + + /* use external clock */ + status &= ~0x000000ff; + status |= 0x00000080; + ctrl_outl(status, clk->enable_reg); +} + +static struct clk_ops fsimck_clk_ops = { + .init = fsimck_init, +}; + +static struct clk fsimcka_clk = { + .name = "fsimcka_clk", + .id = -1, + .ops = &fsimck_clk_ops, + .enable_reg = (void __iomem *)FCLKACR, + .rate = 0, /* unknown */ +}; + +struct sh_fsi_platform_info fsi_info = { + .porta_flags = SH_FSI_BRS_INV | + SH_FSI_OUT_SLAVE_MODE | + SH_FSI_IN_SLAVE_MODE | + SH_FSI_OFMT(PCM) | + SH_FSI_IFMT(PCM), +}; + +static struct resource fsi_resources[] = { + [0] = { + .name = "FSI", + .start = 0xFE3C0000, + .end = 0xFE3C021d, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = 108, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device fsi_device = { + .name = "sh_fsi", + .id = 0, + .num_resources = ARRAY_SIZE(fsi_resources), + .resource = fsi_resources, + .dev = { + .platform_data = &fsi_info, + }, +}; + /* KEYSC in SoC (Needs SW33-2 set to ON) */ static struct sh_keysc_info keysc_info = { .mode = SH_KEYSC_MODE_1, @@ -351,6 +411,7 @@ static struct platform_device *ms7724se_devices[] __initdata = { &keysc_device, &sh_eth_device, &sh7724_usb0_host_device, + &fsi_device, }; #define EEPROM_OP 0xBA206000 @@ -418,11 +479,13 @@ static void __init sh_eth_init(void) static int __init devices_setup(void) { u16 sw = ctrl_inw(SW4140); /* select camera, monitor */ + struct clk *fsia_clk; /* Reset Release */ ctrl_outw(ctrl_inw(FPGA_OUT) & ~((1 << 1) | /* LAN */ (1 << 6) | /* VIDEO DAC */ + (1 << 7) | /* AK4643 */ (1 << 12) | /* USB0 */ (1 << 14)), /* RMII */ FPGA_OUT); @@ -558,6 +621,32 @@ static int __init devices_setup(void) gpio_request(GPIO_FN_KEYOUT1, NULL); gpio_request(GPIO_FN_KEYOUT0, NULL); + /* enable FSI */ + gpio_request(GPIO_FN_FSIMCKB, NULL); + gpio_request(GPIO_FN_FSIMCKA, NULL); + gpio_request(GPIO_FN_FSIOASD, NULL); + gpio_request(GPIO_FN_FSIIABCK, NULL); + gpio_request(GPIO_FN_FSIIALRCK, NULL); + gpio_request(GPIO_FN_FSIOABCK, NULL); + gpio_request(GPIO_FN_FSIOALRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOAO, NULL); + gpio_request(GPIO_FN_FSIIBSD, NULL); + gpio_request(GPIO_FN_FSIOBSD, NULL); + gpio_request(GPIO_FN_FSIIBBCK, NULL); + gpio_request(GPIO_FN_FSIIBLRCK, NULL); + gpio_request(GPIO_FN_FSIOBBCK, NULL); + gpio_request(GPIO_FN_FSIOBLRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOBO, NULL); + gpio_request(GPIO_FN_FSIIASD, NULL); + + /* change parent of FSI A */ + fsia_clk = clk_get(NULL, "fsia_clk"); + clk_register(&fsimcka_clk); + clk_set_parent(fsia_clk, &fsimcka_clk); + clk_set_rate(fsia_clk, 11000); + clk_set_rate(&fsimcka_clk, 11000); + clk_put(fsia_clk); + /* * enable SH-Eth * -- 1.6.0.4

2 1

[alsa-devel] [PATCH v3 3/4] ASoC: Add FSI-AK4642 sound support for SuperH
by Kuninori Morimoto 21 Aug '09

21 Aug '09

This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o remove un-used ak4642_setup_data I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 sound/soc/sh/Kconfig | 8 +++ sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-ak4642.c | 107 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 117 insertions(+), 0 deletions(-) create mode 100644 sound/soc/sh/fsi-ak4642.c diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 01943a1..9154b43 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -40,4 +40,12 @@ config SND_SH7760_AC97 This option enables generic sound support for the first AC97 unit of the SH7760. +config SND_FSI_AK4642 + bool "FSI-AK4642 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_AK4642 + help + This option enables generic sound support for the + FSI - AK4642 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 9fbcc4a..a699787 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -12,5 +12,7 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o +snd-soc-fsi-ak4642-objs := fsi-ak4642.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o +obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c new file mode 100644 index 0000000..c7af097 --- /dev/null +++ b/sound/soc/sh/fsi-ak4642.c @@ -0,0 +1,107 @@ +/* + * FSI-AK464x sound support for ms7724se + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include <../sound/soc/codecs/ak4642.h> + +static struct snd_soc_dai_link fsi_dai_link = { + .name = "AK4642", + .stream_name = "AK4642", + .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .codec_dai = &ak4642_dai, + .ops = NULL, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_dai_link, + .num_links = 1, +}; + +static struct snd_soc_device fsi_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_ak4642, +}; + +#define AK4642_BUS 0 +#define AK4642_ADR 0x12 +static int ak4642_add_i2c_device(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = AK4642_ADR; + strlcpy(info.type, "ak4642", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(AK4642_BUS); + if (!adapter) { + printk(KERN_DEBUG "can't get i2c adapter\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_DEBUG "can't add i2c device\n"); + return -ENODEV; + } + + return 0; +} + +static struct platform_device *fsi_snd_device; + +static int __init fsi_ak4642_init(void) +{ + int ret = -ENOMEM; + + ak4642_add_i2c_device(); + + fsi_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_snd_device) + goto out; + + platform_set_drvdata(fsi_snd_device, + &fsi_snd_devdata); + fsi_snd_devdata.dev = &fsi_snd_device->dev; + ret = platform_device_add(fsi_snd_device); + + if (ret) + platform_device_put(fsi_snd_device); + +out: + return ret; +} + +static void __exit fsi_ak4642_exit(void) +{ + platform_device_unregister(fsi_snd_device); +} + +module_init(fsi_ak4642_init); +module_exit(fsi_ak4642_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

2 1

[alsa-devel] [PATCH v3 2/4] ASoC: Add ak4642/ak4643 codec support
by Kuninori Morimoto 21 Aug '09

21 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o add ak4643 in i2c table o remove un-used hw_read o remove un-used ak4642_setup_data I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 502 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 20 ++ 4 files changed, 528 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 167a5ce..0edca93 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4642 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -92,6 +93,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4642 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fbab43b..fb4af28 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4642-objs := ak4642.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -54,6 +55,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c new file mode 100644 index 0000000..e057c7b --- /dev/null +++ b/sound/soc/codecs/ak4642.c @@ -0,0 +1,502 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is not tested. + * AK4643 is tested. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak4642.h" + +#define AK4642_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK4642_CACHEREGNUM 0x25 + +struct snd_soc_codec_device soc_codec_dev_ak4642; + +/* codec private data */ +struct ak4642_priv { + struct snd_soc_codec codec; + unsigned int sysclk; +}; + +static struct snd_soc_codec *ak4642_codec; + +/* + * ak4642 register cache + */ +static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak4642 register cache + */ +static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write ak4642 register cache + */ +static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK4642 register space + */ +static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4642 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak4642_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak4642_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + r |= ak4642_write(codec, i, cache[i]); + + return r; +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Sampling Frequency: 44.1kHz + * Digital Volume: −8dB + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + * + * Example code use 0x39, 0x79 value for 0x01 address, + * But we need MCKO (0x02) bit now + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x0f, 0x09); + ak4642_write(codec, 0x0e, 0x19); + ak4642_write(codec, 0x09, 0x91); + ak4642_write(codec, 0x0c, 0x91); + ak4642_write(codec, 0x0a, 0x28); + ak4642_write(codec, 0x0d, 0x28); + ak4642_write(codec, 0x00, 0x64); + ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ + ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Sampling Frequency:44.1kHz + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x02, 0x05); + ak4642_write(codec, 0x06, 0x3c); + ak4642_write(codec, 0x08, 0xe1); + ak4642_write(codec, 0x0b, 0x00); + ak4642_write(codec, 0x07, 0x21); + ak4642_write(codec, 0x00, 0x41); + ak4642_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak4642_write(codec, 0x01, 0x3b); + ak4642_write(codec, 0x01, 0x0b); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x0e, 0x11); + ak4642_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x10, 0x00); + ak4642_write(codec, 0x07, 0x01); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *ak4642 = codec->private_data; + + ak4642->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, +}; + +struct snd_soc_dai ak4642_dai = { + .name = "AK4642", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4642_dai); + +static int ak4642_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak4642_sync(codec); + return 0; +} + +/* + * initialise the AK4642 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4642_init(struct ak4642_priv *ak4642) +{ + struct snd_soc_codec *codec = &ak4642->codec; + int ret = 0; + + if (ak4642_codec) { + dev_err(codec->dev, "Another ak4642 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4642; + codec->name = "AK4642"; + codec->owner = THIS_MODULE; + codec->read = ak4642_read_reg_cache; + codec->write = ak4642_write; + codec->dai = &ak4642_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); + codec->reg_cache = kmemdup(ak4642_reg, + sizeof(ak4642_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + ak4642_dai.dev = codec->dev; + ak4642_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto reg_cache_err; + } + + ret = snd_soc_register_dai(&ak4642_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto reg_cache_err; + } + + /* + * clock setting + * + * Audio I/F Format: MSB justified (ADC & DAC) + * BICK frequency at Master Mode: 64fs + * Input Master Clock Select at PLL Mode: 11.2896MHz + * MCKO: Enable + * Sampling Frequency: 44.1kHz + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p89. + * + * please fix-me + */ + ak4642_write(codec, 0x01, 0x08); + ak4642_write(codec, 0x04, 0x4a); + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x01, 0x0b); + + return ret; + +reg_cache_err: + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4642_priv *ak4642; + struct snd_soc_codec *codec; + int ret; + + ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + if (!ak4642) + return -ENOMEM; + + codec = &ak4642->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, ak4642); + codec->control_data = i2c; + + ret = ak4642_init(ak4642); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK4642\n"); + + return ret; +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + struct ak4642_priv *ak4642 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&ak4642_dai); + snd_soc_unregister_codec(&ak4642->codec); + kfree(ak4642->codec.reg_cache); + kfree(ak4642); + ak4642_codec = NULL; + + return 0; +} + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", 0 }, + { "ak4643", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "AK4642 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +#endif + +static int ak4642_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret; + + if (!ak4642_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4642_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to register card\n"); + goto card_err; + } + + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; + +} + +/* power down chip */ +static int ak4642_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4642 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); + +static int __init ak4642_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ak4642_i2c_driver); +#endif + return ret; + +} +module_init(ak4642_modinit); + +static void __exit ak4642_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4642_i2c_driver); +#endif + +} +module_exit(ak4642_exit); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h new file mode 100644 index 0000000..e476833 --- /dev/null +++ b/sound/soc/codecs/ak4642.h @@ -0,0 +1,20 @@ +/* + * ak4642.h -- AK4642 Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on ak4535.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AK4642_H +#define _AK4642_H + +extern struct snd_soc_dai ak4642_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4642; + +#endif -- 1.6.0.4

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[alsa-devel] [PATCH] pyalsa: fix integer overflow in alsaseq.c
by Chris Coleman 21 Aug '09

21 Aug '09

Hello alsa-devel, I've been using the alsaseq python module and I found a bug. Sometimes the SEQ_* constants have extremely large and incorrect values. For example, 25769803811 instead of 35. The lower 32-bits are always correct. Obviously, I'm running a 64-bit operating system. The problem is that the `value` member of the `ConstantObject` structure is an `unsigned int` whereas it should be a `long`. I've attached a patch. It's against the latest released version, 1.0.20. Regards, Chris Coleman Signed-off-by: Chris Coleman <chris.coleman83 at gmail.com>

2 3

[alsa-devel] [patch 0/4] ASoC S3C24XX/Simtec support
by Ben Dooks 21 Aug '09

21 Aug '09

re-worked against ASoC git branch Will be away from until 26th, any comments will be sorted when back. --

2 6