October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Clevo P170HM / Sager NP8170 audio
by Andrew Mahone 04 Dec '11

04 Dec '11

This laptop is detected as ALC892 audio (and by the codec ID this seems to be valid), but none of the configurations for it quite match this device. I've been working on documenting the connections with HDA Analyzer. alsa-info output: http://www.alsa-project.org/db/?f=b86e4c6447867fbd129a36d8a63e2f5caa836c52 The laptop has built-in 5.1, and 5.1 or 7.1 output via jacks. DAC NIDs: 0x02 front 0x03 rear 0x04 clfe 0x05 side internal mixers: 0x0c front 0x0d rear 0x0e clfe 0x0f side internal speaker pins: 0x14 front 0x15 "rear" (front of laptop) 0x16 clfe external ports: fixed route: 0x17 side can select any output: 0x18 mic / output 0x1a line-in / output 0x1b headphone I've been experimenting with patch_realtek.c to add the extra DAC NID, and have had some success with a 6-channel mode for the built-in speakers and an 8-channel mode that mutes all internal speakers and sets the jacks up for output. I still need to manually connect pins 0x18, 0x1a, 0x1b to the correct internal mixers for 7.1 to work, and headphone plugin does not mute internal speakers in 5.1 mode. I have not added a plain stereo mode for internal speakers, although it appears that i can probably configure the codec to mix the front channels to center and LFE. Are there any suggestions for what to do next? My current state as a patch against the linux-3.0.y repository is at http://pastie.org/2400649 -- Andrew Mahone andrew DOT mahone AT gmail DOT com Q: How many voters does it take to change a lightbulb? A: None - if voting could change anything they'd make it illegal

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Re: [alsa-devel] [patch][saa7134] do not change mute state for capturing audio
by Mauro Carvalho Chehab 03 Dec '11

03 Dec '11

Em 24-07-2011 14:45, Stas Sergeev escreveu: > 23.07.2011 19:09, Mauro Carvalho Chehab wrote: >> > In this case, it will not be autounmuted after tuning. >> Hard to tell about your solution without seeing a patch. > OK, it turns out the automute code is already there, > but it doesn't work. The driver for some reasons > starts the scan on initialization, finds the carrier: > --- > saa7134[0]/audio: found PAL main sound carrier @ 6.000 MHz [3969/324] > --- > and, because of that, disables the automute. If the > real mute is not enabled at that point, you get the > white noise right away. The automute code works fine. Maybe you have a strong interference at the default tuning frequency, leading into saa7134 miss-detection. For saa7134 driver: 1) There is an audio carrier; 2) an application wants to listen audio; 3) the device is at automute mode. So, when there's an audio carrier, it will unmute the audio at streaming start. The driver is doing the _right_ thing by letting the audio to flow to PA, when it starts the capture. Unfortunately, on your specific device, starting audio capture also enables audio at the audio output pin. So, you're noticing a problem. People with a saa7134 device without alsa stream won't notice it (old devices). People with alsa stream without anything connected to the LINE OUT people aren't noticing it, if PA is not copying the saa7134 PCM IN stream to the sound card PCM out device (the default, for PA). So, only people that has saa7134 with alsa stream that opted to wire the saa7134 device to the sound card, and with a strong interference at the default tunning frequency (400 MHz) would notice a problem. > Since I have no idea why it finds some carrier, I can't > fix that in any way. Or, maybe, not to call the scan > on driver init? What will that break? Analog tuners need to be tuned at the device init on a high frequency according with their datasheets, otherwise the PLL may fail. > Anyway, as long as the automute code is broken, > we should either start fixing it, or fix PA, or fix mplayer... As I always said, the fix should be at PA, as it makes no sense to start capturing at saa7134 without first configuring it. So, or PA should be converted into a V4L2-aware application, in order to properly init the device (with seems to be the wrong approach) or it _SHOULD_NOT_ automatically enable capture on those devices, as this may cause undesired side effects, on the devices that have the capture pin directly wired to the output pin, witch seems to be the case of your device. PS.: it seems that you've removed Lennart/alsa people from the c/c. I'm re-adding them, as I'm expecting a fix from their side. > Dunno. I wonder how come so many bugs left unfixed > for so long, resulting in a white noise to people... You're the first one that reported it, and the code is there for _years_. So, this is not a commonly noticed problem at all. Mauro.

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Re: [alsa-devel] DM365 McBSP as Master
by Appalayagari Sreedhar 30 Nov '11

30 Nov '11

Hi Vladimir, Thank you very much. i will do the same. on further probing and testing. i found that if i open the device O_WRONLY and O_RDONLY mode it works. if i try to open the device using the O_RDWR mode it fails. is there any configuration settings i need to check. -Sreedhar. On 10/17/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: > Hi Sreedhar, > > You should always CC the alsa-devel channel list for any alsa related > questions instead of writing directly to TI support and me. But you can > CC both me and TI people. > In that case your questions will not go unanswered. > > > Appalayagari Sreedhar wrote: >> Hi Vladimir, >> >> I need your help to solve one issue. >> >> I have developed a audio driver in ALSA model, the audio ic is >> interfaced with the TI DM365 processor MCBSP ports. I am working with >> the 2.6.32 kernel. >> >> I am able to play the audio using the aplay command. but when i try >> to open the device using /dev/dsp i am getting the -ve value. it fails >> to open the device. >> > I believe that you've enabled OSS emulation for alsa drivers: > CONFIG_SND_PCM_OSS=y > and read carefully Documentation/sound/alsa/OSS-Emulation.txt guide. > > Here you didn't mention how you are trying to open /dev/dsp and the > error log sample. > If you'll read the mentioned guide then you'll find some hints about PCM > mode changes via proc interface. Probably that will shine some light on > your issue. > >> when i check the >> >> >>>> ls /dev/dsp >>>> >> >> it is available in the device directory. >> >> I have enabled the Sound OSS related options in the kernel menuconfig. >> with the same options i am able to open the device for the tlv320aic3x >> device. >> >> Could you please suggest where i am doing wrong? and help to solve >> this problem. >> >> One more observation : >> >> after doing aplay i am able to open the device using the /dev/dsp. >> >> if try to open the device using the /dev/dsp before aplay i am getting >> the error. >> >> could you please tell me where i am doing. >> > It seems that aplay configures the codec and mcbsp interface calling > it's TRIGGER_START but your own application does not. Once configured > the pcm streams can pass normally using /dev/dsp. Not usre but seems > that the problem in inability to call TRIGGER. > > Regards, > Vladimir >> I have posted the queries in ti e2e website, please check them below. >> i did not receive any proper response till now so i am contacting you. >> >> http://e2e.ti.com/support/embedded/f/354/p/138037/504805.aspx >> >> Thank you in advance. >> Sreedhar. >> >> On 9/23/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: >> >>> Hi Sreedhar, >>> >>> Appalayagari Sreedhar wrote: >>> >>>> Hi Vladimir Barinov, >>>> >>>> Now I am able to make it work McBSP as master and also as slave. >>>> We decided to configure the McBSP as Slave and Codec as Master. >>>> >>>> I have one question, Since the Clocks are generated by the Codec. >>>> will there be any Channel Swap in the audio? >>>> >>>> If so how to handle this channel swap mechanism. >>>> >>>> >>> Yes, It possible to get channels swap on any external codec. >>> Unfortunately I don't know any strong method for this issue. I just >>> experience similar problem on TI's PMIC audio and get rid from EVERY >>> excessive clock in system to be able to fix it. I mean you have to stop >>> all related clocks in proper sequence during audio pause/stop. When I >>> did it the channel swap was fixed. >>> >>> Regards, >>> Vladimir >>> >>>> Thank you, >>>> Sreedhar. >>>> >>>> >>>> >>>> >>>> On 9/20/11, Appalayagari Sreedhar <sreecha01(a)gmail.com> wrote: >>>> >>>> >>>>> Hi Vladimir Barinov, >>>>> >>>>> Thank you very much. I will check as you suggested. >>>>> Thanks once again you are only person from whom i got the support since >>>>> 5 >>>>> days. >>>>> >>>>> Thanks once again. >>>>> >>>>> Thank you, >>>>> Sreedhar. >>>>> >>>>> On 9/20/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: >>>>> >>>>> >>>>>> Sreedhar, >>>>>> >>>>>> 1) verify that mux registers are configured properly >>>>>> 2) startup with codec as master since the current code tested in this >>>>>> mode and usually msbsp can't provide variety of clocks to match all >>>>>> samplerates. >>>>>> 3) check that you can read/write codec registers with success (i2c >>>>>> works >>>>>> find and the codec is not in reset state - check your h/w) >>>>>> Probe MCLK by scope and find that signal is ON, otherwise you didn't >>>>>> make codec setup (i2c communication or codec doesn't work) >>>>>> >>>>>> After above you can start moving to MCBSP as master if you really >>>>>> need >>>>>> it. >>>>>> >>>>>> Regards, >>>>>> Vladimir >>>>>> >>>>>> Appalayagari Sreedhar wrote: >>>>>> >>>>>> >>>>>>> Hi Vladimir Barinov, >>>>>>> >>>>>>> I am working with DM365 evm board. I have interfaced conexant audio >>>>>>> ic to the McBSP. >>>>>>> >>>>>>> I need to do I2S data transfer. I need to program the McBSP as master >>>>>>> and conexant audio IC as slave. >>>>>>> >>>>>>> i did the macro changes in the davinci-evm.c file in AUDIO_FORMAT >>>>>>> like >>>>>>> this. >>>>>>> >>>>>>> #define AUDIO_FORMAT (SND_SOC_DAIFMT_I2S | \ >>>>>>> SND_SOC_DAIFMT_CBS_CFS) >>>>>>> >>>>>>> I am using the McBSP internal clock for McBSP. I printed the McBSP >>>>>>> registers and found that SCLKME - 0 and CLKSM - 1. by reading the >>>>>>> registers. >>>>>>> >>>>>>> when i run the aplay command i am not viewing any clocks when i probe >>>>>>> the signals using the CRO. >>>>>>> >>>>>>> Could you please tell me where i am doing wrong? >>>>>>> >>>>>>> Please find my queries in the following thread in the ti e2e >>>>>>> comminity. >>>>>>> >>>>>>> http://e2e.ti.com/support/embedded/f/354/p/134788/485628.aspx#485628 >>>>>>> >>>>>>> Thank you, Sreedhar. >>>>>>> >>>>>>> >>>>>>> >>> > >

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[alsa-devel] Default device XRUNs with perriods=2
by Alan Horstmann 29 Nov '11

29 Nov '11

Hi, I am trying to debug/fix some problems with Portaudio on Linux platform, and would appreciate assistance from Alsa experts. The 'default' pcm on the test systems here (Suse 9 and SLED10), is standard for the configuration and includes dmix and plug conversion, with fixed period of 1024 at 48000. Pulseaudio is not on these systems. When playing data (using a polling mechanism) at 48000 both 3 (or more) and 2 periods of buffer work fine; with the fixed period of 1024 the lowest latency is 2 * 1024 / 48000 = 42.6ms. However, at 44100, 2 periods does not run properly at all, but 3 does. The periods in this case are (940, 941) due to the rate conversion. What I see (captured on another machine) from start of the stream is one 21.3ms period of good audio, a discontinuity followed by some variable amount of out-of-sync audio, and a section of silence. I believe that the discontinuity corresponds to the point at which snd_pcm_poll_descriptors_revents() returns POLLERR and portaudio detects the POLLERR and restarts the stream, so that after a certain delay there is another 21.3ms of audio, POLLERR, now no out-of-sync audio, just silence, before it all repeats. So it stutters only outputting 21.3ms bursts of audio. The CPU usage during this is about 2%. I am suspecting some mis-configuration, or is it wrong to expect this to work at 2 periods due to the rate conversion etc? Any pointers etc would be really appreciated. This is the info dump:- ACCESS: MMAP_INTERLEAVED FORMAT: FLOAT_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: 64 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (21333 21334) PERIOD_SIZE: (940 941) PERIOD_BYTES: (7520 7528) PERIODS: 2 BUFFER_TIME: (42653 42654) BUFFER_SIZE: 1881 BUFFER_BYTES: 15048 TICK_TIME: 0 start_mode: DATA xrun_mode: STOP tstamp_mode: MMAP period_step: 1 sleep_min: 0 avail_min: 940 xfer_align: 1 silence_threshold: 0 silence_size: 4235635449541951488 boundary: 4235635449541951488 stream : PLAYBACK access : MMAP_INTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 1881 period_size : 940 period_time : 21333 tick_time : 0 tstamp_mode : MMAP period_step : 1 sleep_min : 0 avail_min : 940 xfer_align : 1 start_threshold : 940 stop_threshold : 1881 silence_threshold: 0 silence_size : 4235635449541951488 boundary : 4235635449541951488 Regards Alan

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[alsa-devel] [PATCH 0/5] RFC for snd-usb: rework usb endpoint logic
by Daniel Mack 18 Nov '11

18 Nov '11

It's been a while since the last round of patches for this topic, so here is a new version. In contrast to earlier versions, I did another bunch of cleanips and can now use input and output simultaniously. The patches that follow are structured in a way that should make reviews easier, as they first implement a new streaming model while leaving the old one working, and then switch to it in a separate patch. This way, the series is also fully bisectable. The problems that remain is a regression in handling data streams with feedback endpoints in full duplex mode and a hard lockup when changing audio parameters while two streams are running. I guess the latter is caused by missing locking. I didn't sign-off the patches on purpose, as I would really like to get them reviewed before they go in. Can people have a look and state whether the whole idea is at all sane? Thanks a lot, Daniel Daniel Mack (5): ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: set MAX_URBS to 16 ALSA: snd-usb: add support for implicit feedback sound/usb/card.c | 6 +- sound/usb/card.h | 56 ++- sound/usb/endpoint.c | 1347 ++++++++++++++++++++++++-------------------------- sound/usb/endpoint.h | 31 +- sound/usb/pcm.c | 412 +++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 1 + 7 files changed, 1081 insertions(+), 803 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH v3 0/3] ARM: mxs: add recording support for saif
by Dong Aisheng 09 Nov '11

09 Nov '11

Changes since v2: * separate clkmux code into another patch according to Uwe * add lock according to Wolfram * Other minus fixes suggested by Uwe and Wolfram. * remove Wolfram's fix saif clock setting patch which is in v2 series since Wolfram will reform the clock code and send it out himself. For test purpose, user still needs that patch. People can get it from v2 series. The patch is: [PATCH v2 3/3] arm: mxs: disable clock-gates when setting saif-clocks The patches are based on imx-features branch since commit: f4f01e31835f. Changes since v1: The main changes are move mach-specific code(clkmux in DIGCTL) from saif driver to mach-specific layer based on Wolfram's suggestion. Note that the last patch is a RFC patch and sent out for testing since without that patch the saif may not work. Dong Aisheng (3): ARM: mxs: add saif clkmux functions ARM: mx28evk: add platform data for saif ARM: mx28evk: set a initial clock rate for saif arch/arm/mach-mxs/clock-mx28.c | 67 +++++++++++++++++++++++ arch/arm/mach-mxs/devices-mx28.h | 3 +- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 +- arch/arm/mach-mxs/include/mach/common.h | 2 + arch/arm/mach-mxs/include/mach/devices-common.h | 4 +- arch/arm/mach-mxs/include/mach/digctl.h | 21 +++++++ arch/arm/mach-mxs/mach-mx28evk.c | 16 +++++- 7 files changed, 112 insertions(+), 6 deletions(-) create mode 100644 arch/arm/mach-mxs/include/mach/digctl.h

4 7

[alsa-devel] [RFC] hda - is_jack_detectable()
by Raymond Yau 09 Nov '11

09 Nov '11

Refer to HDA Specification 7.3.3.31 Configuration Default Misc[3:0] is a bit field used to indicate other information about the jack. Currently, only bit 0 is defined. If bit 0 is set, it indicates that the jack has no presence detect capability, so even if a Pin Complex indicates that the codec hardware supports the presence detect functionality on the jack, the external circuitry is not capable of supporting the functionality. The bit definitions for the Miscfield are in Table 106. Do we need to change is_jack_detectable() when user setup BIOS to use AC97 Front Audio Panel which set AC_DEFCFG_MISC_NO_PRESENCE bit ? +#define get_defcfg_misc(cfg) \ + ((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) { return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) && + (!(get_defcfg_misc(codec,nid) & AC_DEFCFG_MISC_NO_PRESENCE)) && (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP); } or add another fuction is_defcfg_misc_no_presence() to indicate jack has no presence detect capability ?

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[alsa-devel] [PATCH v2 RESEND] ASoC: wm9081: Use snd_soc_update_bits for read-modify-write
by Axel Lin 05 Nov '11

05 Nov '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- RESEND: I just found I doesn't set my mail client properly to show correct sender, so here is a resend. v2: For Initial cold start ( change bias level from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY ), the v1 does not set VMID_SEL[1:0] bits correctly for Normal bias enable & soft start off. In v2 we also clear VMID_SEL[1:0] bits for Normal bias enable & soft start off. We did clear VMID_SEL[1:0] bits in SND_SOC_BIAS_OFF, thus we know the VMID_SEL[1:0] bits status is 0b00 in this case. sound/soc/codecs/wm9081.c | 82 +++++++++++++++++++------------------------- 1 files changed, 35 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 66092ef..9c38ddb 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -766,84 +766,73 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x2; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x2); /* Normal bias current */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, 0); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg &= ~WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, 0); /* Select startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC | WM9081_BIAS_ENA); /* VMID 2*4k; Soft VMID ramp enable */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg |= WM9081_VMID_RAMP | 0x6; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); - + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | 0x6, + WM9081_VMID_RAMP | 0x6); mdelay(100); /* Normal bias enable & soft start off */ - reg |= WM9081_BIAS_ENA; - reg &= ~WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_BIAS_ENA | + WM9081_VMID_SEL_MASK, + WM9081_BIAS_ENA); /* Standard bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, 0); } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x04; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_VMID_SEL_MASK, 0x04); /* Standby bias current on */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, + WM9081_STBY_BIAS_ENA); break; case SND_SOC_BIAS_OFF: /* Startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, WM9081_BIAS_SRC); /* Disable VMID and biases with soft ramping */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); - reg |= WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_VMID_SEL_MASK | + WM9081_BIAS_ENA, WM9081_VMID_RAMP); /* Actively discharge LINEOUT */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg |= WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, + WM9081_LINEOUT_DISCH); break; } @@ -1242,11 +1231,10 @@ static int wm9081_probe(struct snd_soc_codec *codec) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); - snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, - reg | WM9081_SPKPGAZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, + WM9081_LINEOUTZC, WM9081_LINEOUTZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, + WM9081_SPKPGAZC, WM9081_SPKPGAZC); snd_soc_add_controls(codec, wm9081_snd_controls, ARRAY_SIZE(wm9081_snd_controls)); -- 1.7.5.4

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[alsa-devel] [PATCH 0/2] OMAP2+: DMA: fix src/dst position reporting
by Peter Ujfalusi 03 Nov '11

03 Nov '11

Hello, If the user asks for the sDMA current position before the first data has been transmitted (before the first DMA request has been generated), the reported position is not valid: src position: CSAC is uninitialized dst position: CDAC is 0 The return values in both case considered invalid. This sitation can be identified by checking if the CDAC register is 0 (it is initialized to 0 in omap_dam_start call). In this case return the programmed source/destination address. The affected omap_get_dma_src_pos/omap_get_dma_dst_pos functions are used by the audio stack mainly for checking the current position of the audio stream. Regards, Peter --- Peter Ujfalusi (2): OMAP2+: DMA: Workaround for invalid source position OMAP2+: DMA: Workaround for invalid destination position arch/arm/plat-omap/dma.c | 25 ++++++++++++++++++++++--- 1 files changed, 22 insertions(+), 3 deletions(-) -- 1.7.7.1

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[alsa-devel] [PATCH v8 3/3] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 01 Nov '11

01 Nov '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for AUX1, AUX2 and INPGA (Moved here from DAPM patch) Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 77 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0ebcbd5..ff084a6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -421,26 +447,42 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; @@ -451,14 +493,23 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], @@ -514,12 +565,21 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, @@ -528,9 +588,20 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"ADC Right", NULL, "INPGA Right"}, /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, @@ -887,6 +958,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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