December 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

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[alsa-devel] [RFC] virtual master control for HD-audio
by Takashi Iwai 06 May '08

06 May '08

Hi, another thing I'd like to push into the next kernel is the virtual master volume control. As I posted in earlier posts, it adds virtual controls that have several slave controls with the same types, e.g. Front, Surround, Center, LFE, etc. Then these are adjusted simultaneously via the master control. It'd be appreciated if some one can test the patches with HD-audio h/w that has no master control yet (e.g. some STAC codecs). Note that this won't add the master volumes if there are no real volume controls. Some codecs have really no volume control, and this won't help for such devices. Two (and one for driver) patches will follow after this. Takashi

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[alsa-devel] Altec Lansing FX5051 mixer
by Jeremy Nickurak 04 Apr '08

04 Apr '08

I recently purchased an Altec Lansing USB Surround speaker set. (It's a set of 5.1 speakers, with a built-in USB sound card). I've got about a week before the return policy expires and I'd have to return it. The mixer doesn't work at all w/linux kernel 2.6.22, everything is locked at 100% volume, which is pretty unhelpful, as it's the only way to adjust the volume. (no analog controls). Also, the digital control panel locks up when the speakers power up, further preventing any access to the volume controls. The device works out-of-the-box without any drivers under XP, so I have to so-far assume that alsa's usb-audio is missing something. Any thoughts as to what could be wrong? I'm more than happy to prod the device and submit any information I can extract about it, or even potentially opening the device up for remote access if a reputable developer is interested in examining it. -- Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =-

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[alsa-devel] ALSA driver for Native Instruments sound hardware
by Daniel Mack 23 Mar '08

23 Mar '08

Hi, some months after the last posting, we finally managed to finish a new version of the ALSA driver for Native Instruments' external USB sound cards such as "Rig Kontrol 2", "Kore Kontroller", "Audio Kontrol 1" and "Audio 8 DJ". Compared to the last release, the following things were changed: - the streaming logic was completely reworked so that multiple substreams are handled correctly - a Linux input device layer was added to support the knobs, buttons and pedals on "Rig Kontrol 2" and "AudioKontrol 1" - workqueues were entirely removed from the code - most data structures are now allocated within one big struct to avoid multiple, smaller allocations - many more cleanups according to Clemens' reply on my last posting[1] The driver was properly tested with all the mentioned hardware on both record and playback channels. The patchset should apply to the current mecurial alsa-kernel tree. Please let me know what you think. Best regards, Daniel [1] http://article.gmane.org/gmane.linux.alsa.devel/34324/ Signed-off-by: Daniel Mack <daniel(a)caiaq.org> Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de>

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Re: [alsa-devel] hdsp rpm support
by kgrill＠chello.at 25 Feb '08

25 Feb '08

I started to work on this about a year ago, but had to stop because my son took the card to Istanbul where he spent a year studying. I'll get back to it in the next days, and depending on how much has changed in the meantime I may get something ready by the end of January. regards kg

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[alsa-devel] Notebook Toshiba satellite x200-21g with realtek snd-hda-intel audio device still not supported (codec ALC268)
by Daniele (Mastro) 19 Feb '08

19 Feb '08

hi all i'm new to this mailing list i've bought a laptop and i would like to help you supporting my sound card what works: 2 speaker, low volume (it should be a 5+1 audio card) what doesn't -) low volume -) inoperating headphone jack -) not working mic -) no subwoofer -) may be some other stuff not tested ( SPDIF? hardware midi?) with speaker-test -t wav -c 6 only front left/right channel works my kernel is a debian kernel Linux Luffy 2.6.23-1-686-bigmem #1 SMP Wed Dec 5 02:42:05 UTC 2007 i686 GNU/Linux alsa 1.0.15 drivers (compiled with debian-way: module-assistant a-i alsa-source) i've already tried all the option i could think about auto (default) toshiba 3stacks 3stacks-6ch 3stacks-6ch-dig neither one has better/worst behavior I've checked the module has toke those option by cat /sys/module/snd_hda_intel/parameters/model anyway I've read the ALSA-Configuration.txt shipped with kernel and for Codec ALC268 it only has 4 options ALC268 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops auto auto-config reading BIOS (default) I've googled a bit and i found out that the Toshiba A205 has completely different problem.. the sound doesn't work with it.. reading the changelog from 1.0.14 to 1.0.15 (alsa) it say: Added the proper model=toshiba for Toshiba A305 with ALC268 codec. so i think my audio card is an unknown model the laptop is brand new and i only found this model in Italy anyway i think that the same audio card is shipped with the Toshiba Satellite x205-S7483 witch is an american laptop some info about the audio card: lspci -n 00:1b.0 0403: 8086:284b (rev 03) lspci -vvv 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Unknown device ff00 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f6600000 (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] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) cat /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff0a 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 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] 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: [0x40 0x40] 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 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo 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 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: 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 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT 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: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP 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 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: 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: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN 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: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN 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 0x083734: IN OUT Detect Pin Default 0x0281103e: [Jack] Line In at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono 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: [0x1f 0x1f] 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: [0x8a 0x8a] Connection: 7 0x18 0x19 0x1a* 0x1c 0x14 0x15 0x13 many of this think doesn't mean a thing to me but i will be glad to help you if you ask.. i don't mind compiling some other driver/kernel/tool for this just ask me one last note: searching the internet i've found this page: https://wiki.ubuntu.com/Gutsy_Intel_HD_Audio_Controller if you go down you find that the Toshiba Satellite x205-S7483 has the exact same problem here. I think they have the same card searching the Realtek web site i didn't find any 6 channel audio card with the ALC268 codec may be this mean the ALC268 is not the right codec for my audio card or this is a really new model/heavily Toshiba modified device i've already filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659 please ask me if you need any more info or want me to do some test thanks you all for your work ps: i've subscribed to vger mailing list, what's the right mailing list for alsa-devel? -- Daniele Segato

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[alsa-devel] underruns and strange code in pcm_rate.c (and patch)
by Stas Sergeev 01 Feb '08

01 Feb '08

Hi. I spent 4 week-ends debugging the constant underruns and lock-ups with the libao-based programs. Now I've finally tracked the problem to some strange code in pcm_rate.c, namely snd_pcm_rate_poll_descriptors(). I am not sure what was the reason behind altering the "avail_min" every time. Looks like some heuristic was intended, say "see what amount of fragment part was written now, and make sure that amount of free space will be available next time, after poll". Whatever, this code gives underruns. It can increase avail_min by almost period_size. libao sets avail_min==period_size, so we can end up with avail_min==period_size*2. libao sets buffer_size==stop_threshold==period_size*2. So we end up with avail_min~=stop_threshold, which gives underruns. Since I don't know how this code was indended to work, I just removed it, and everything works perfectly now (also had to patch libao to check the returned rate). The patch is attached, any comments? Signed-off-by: Stas Sergeev <stsp(a)aknet.ru>

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[alsa-devel] Realtek ALC883 support on Albatron KI690-AM2 motherboard
by Andrew Paprocki 13 Jan '08

13 Jan '08

Hello everyone, I'm currently trying to get the ALC883 on the Albatron KI690-AM2 motherboard working under ALSA and I'm running into problems with some of the ports. From looking at the patch_realtek.c file inside the intel_hda driver, it appears there are many variants of hookups to this chipset which all require some manual configuration in the driver. I've tried what appears to be the closest match (ALC883_6ST_DIG or 6stack-dig), but it still does not enable all of the ports. The motherboard webpage is here: http://www.albatron.com.tw/English/product/mb/pro_detail.asp?rlink=Specific… The board has two audio stacks on the back panel. Facing the jack inputs, the left-hand stack contains: <S/PDIF output> <TOSLINK w/ green cover> Next to that stack is a larger 6-stack with the following configuration: <blue> <black> <green> <pink> <grey> <TOSLINK w/ pink cover> The box is running a stock debian 2.6.18-4-486 kernel. The kernel outputs this at startup if the "6stack-dig" model is not explicitly supplied: hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... hda-intel: Invalid position buffer, using LPIB read method instead. When using the "6stack-dig" mode, ALSA sees two devices: card 0: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 2: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 When attempting to use ALSA to open the first device in raw mode to output to each stereo pair separately, 5 pairs are detected, and the outputs appear as follows... Pair 1: <green> (working) Pair 2: <pink> (working) Pair 3: <black> (working) Pair 4: (no audio) Pair 5: (no audio) No sound can be heard from the blue or grey jacks in any configuration. I'm wondering if a special entry is needed in patch_realtek.c to handle this board's usage of the ALC883. I am willing to try changing the code and testing if someone can point me to what might need to be done. 01:05.2 Audio device: ATI Technologies Inc Unknown device 7919 Subsystem: Albatron Corp. Unknown device 5000 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 4 bytes Interrupt: pin B routed to IRQ 3 Region 0: Memory at fdffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Thanks, -Andrew

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[alsa-devel] Hotplug plugin
by Erik Slagter 13 Jan '08

13 Jan '08

Sorry for crossposting, but this is a problem I am not quite sure might be solved by users or developers. Also I am not too lazy to do a little development myself, if that appears to be necessary. This is my setup: - internal snd-hda soundcard connected to speakers in the bathroom playing 24x7 music; - jvc amplifier in living room with integrated USB sound device. The USB sound device in the amplifier vanishes when the amplifier is powered down. So... this is my problem. I want to have 24x7 music on the internal soundcard and on the amplifier. When the amplifier is switched off, no audio should be send there (of course) but playback should resume as soon as the amplifier is switched on and the USB device re-appears. I created a "tee" construction using asound.conf which adresses both soundcards, this works. At least... until the amplifer is switched off and the USB device disappears. Then the sound playing application gets an error and quits. I tried these approaches for a solution: - pulseaudio, using hal-detection and "combined" device: doesn't work because a removed device is never reinserted into the combined device; - alsa extplug: easy to create, no worries about the "slave" device, you get it for free, but I abandoned the idea as there is probably no way to recreate the slave pcm while the "owner" remains active. - alsa ioplug, attempt 1: there is probably no way to to address and recreate the "slave" (which I open myself in the plugin using snd_pcm_open) in such a way that it is completely transparent to the application that it is talking to the "slave" (not a "real" alsa slave) instead of the ioplug plugin - alsa ioplug, attempt 2: I tried to make the callback functions of the ioplug call the callbacks of the "slave" directly, but that probably isn't going to work either (too much callbacks in the pcm struct that have no equivalent in the ioplug struct) - alsa ioplug, attempt 3: open and write to the hardware devices directly. Abonded it as well because this cannot practically be done, as you'd need parts copies of header files from the kernel and alsalib for hidden data structures. Anyone ideas to solve this in a neat way? (preferrably using an alsa plugin; for the moment I am using a self-made external program that alsa dumps it sound to (file plugin) and uses two alsa-player processes to get the audio to the devices)

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[alsa-devel] Disabling buffer fill level preprocessing by ALSA
by Lennart Poettering 12 Jan '08

12 Jan '08

Hi! In PulseAudio I want to schedule on my own when I need to write audio data into the device and when not. To achieve that I want to be notified via poll() whenever a period boundary is passed (i.e. when an IRQ happens), but only then. That's different from the usual mode where you are notified via poll() whether there is space in the playback buffer that needs to be filled up. On OSS the mmap() mode enables a mode like I described above. After enabling mmap() the application can decide by itself what it considers full and what empty in the dma buffer, and use GETOPTR to query the playback position. poll() on the OSS fd will directly reflect the sound card IRQs and is not influenced if you ever wrote data to device or not. I assume that I can enable a mode like that with one of the SW params. But quite frankly the docs for it are not enlighening at all. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] intel-hd on an intel G33 motherboard
by De Fuuss 11 Jan '08

11 Jan '08

Hi, Maybe I'm missing something, but I've failed to get even the slightest noise out of the onboard sound card from a Intel G33 motherboard. I tried the alsa drivers from the alsa-driver-1.0.15 package as well as the the alsa drivers provided by kernel 2.6.23 (built as modules), to no avail. The sound chip is identified as a Intel HDA / Realtek ALC888 on the Intel G33 ICH9 motherboard. After reading some of the documentation from the ALSA site and googling through half the internet I haven't found out yet if this specific sound card is supported by ALSA and if some module options must be passed (such as options snd-intel-hda model=.....). Could somebody be so kind as to point me into the right direction? Thank you. Best regards, Patrick _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[alsa-devel] [PATCH] ASoC drivers for the Freescale MPC8610 SoC
by Timur Tabi 10 Jan '08

10 Jan '08

This patch adds ALSA SoC device drivers for the Freescale MPC8610 SoC and the MPC8610-HPCD reference board. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch applies on top of git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa.git. Documentation/powerpc/booting-without-of.txt | 40 ++ arch/powerpc/boot/dts/mpc8610_hpcd.dts | 107 ++++ arch/powerpc/configs/mpc8610_hpcd_defconfig | 171 ++++++- arch/powerpc/platforms/86xx/mpc8610_hpcd.c | 18 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/fsl/Kconfig | 21 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl_dma.c | 819 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_dma.h | 149 +++++ sound/soc/fsl/fsl_ssi.c | 614 +++++++++++++++++++ sound/soc/fsl/fsl_ssi.h | 224 +++++++ sound/soc/fsl/mpc8610_hpcd.c | 621 +++++++++++++++++++ 13 files changed, 2789 insertions(+), 4 deletions(-) create mode 100644 sound/soc/fsl/Kconfig create mode 100644 sound/soc/fsl/Makefile create mode 100644 sound/soc/fsl/fsl_dma.c create mode 100644 sound/soc/fsl/fsl_dma.h create mode 100644 sound/soc/fsl/fsl_ssi.c create mode 100644 sound/soc/fsl/fsl_ssi.h create mode 100644 sound/soc/fsl/mpc8610_hpcd.c diff --git a/Documentation/powerpc/booting-without-of.txt b/Documentation/powerpc/booting-without-of.txt index e9a3cb1..826af91 100644 --- a/Documentation/powerpc/booting-without-of.txt +++ b/Documentation/powerpc/booting-without-of.txt @@ -53,6 +53,7 @@ Table of Contents j) CFI or JEDEC memory-mapped NOR flash k) Global Utilities Block l) Xilinx IP cores + p) Freescale Synchronous Serial Interface VII - Specifying interrupt information for devices 1) interrupts property @@ -2514,6 +2515,45 @@ platforms are moved over to use the flattened-device-tree model. Requred properties: - current-speed : Baud rate of uartlite + p) Freescale Synchronous Serial Interface + + The SSI is a serial device that communicates with audio codecs. It can + be programmed in AC97, I2S, left-justified, or right-justified modes. + + Required properties: + - compatible : compatible list, containing "fsl,ssi" + - cell-index : the SSI, <0> = SSI1, <1> = SSI2, and so on + - reg : offset and length of the register set for the device + - interrupts : <a b> where a is the interrupt number and b is a + field that represents an encoding of the sense and + level information for the interrupt. This should be + encoded based on the information in section 2) + depending on the type of interrupt controller you + have. + - interrupt-parent : the phandle for the interrupt controller that + services interrupts for this device. + - fsl,mode : the operating mode for the SSI interface + "i2s-slave" - I2S mode, SSI is clock slave + "i2s-master" - I2S mode, SSI is clock master + "lj-slave" - left-justified mode, SSI is clock slave + "lj-master" - l.j. mode, SSI is clock master + "rj-slave" - right-justified mode, SSI is clock slave + "rj-master" - r.j., SSI is clock master + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master + + Optional properties: + - codec : child node that defines an audio codec connected + to this SSI + + Child 'codec' node required properties: + - compatible : compatible list, contains the name of the codec + + Child 'codec' node optional properties: + - bus-frequency : The frequency of the input clock, which typically + comes from an on-board dedicated oscillator. + + More devices will be defined as this spec matures. VII - Specifying interrupt information for devices diff --git a/arch/powerpc/boot/dts/mpc8610_hpcd.dts b/arch/powerpc/boot/dts/mpc8610_hpcd.dts index 966edf1..4f12ede 100644 --- a/arch/powerpc/boot/dts/mpc8610_hpcd.dts +++ b/arch/powerpc/boot/dts/mpc8610_hpcd.dts @@ -103,6 +103,113 @@ reg = <e0000 1000>; fsl,has-rstcr; }; + + ssi@16000 { + compatible = "fsl,ssi"; + cell-index = <0>; + reg = <16000 100>; + interrupt-parent = <&mpic>; + interrupts = <3e 2>; + fsl,mode = "i2s-slave"; + codec { + compatible = "cirrus,cs4270"; + /* MCLK source is a stand-alone oscillator */ + bus-frequency = <bb8000>; + }; + }; + + ssi@16100 { + compatible = "fsl,ssi"; + cell-index = <1>; + reg = <16100 100>; + interrupt-parent = <&mpic>; + interrupts = <3f 2>; + }; + + dma@21300 { + #address-cells = <1>; + #size-cells = <1>; + compatible = "fsl,mpc8610-dma", "fsl,mpc8540-dma"; + cell-index = <0>; + reg = <21300 4>; /* DMA general status register */ + ranges = <0 21100 200>; + + dma-channel@0 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <0>; + reg = <0 80>; + interrupt-parent = <&mpic>; + interrupts = <14 2>; + }; + dma-channel@1 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <1>; + reg = <80 80>; + interrupt-parent = <&mpic>; + interrupts = <15 2>; + }; + dma-channel@2 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <2>; + reg = <100 80>; + interrupt-parent = <&mpic>; + interrupts = <16 2>; + }; + dma-channel@3 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <3>; + reg = <180 80>; + interrupt-parent = <&mpic>; + interrupts = <17 2>; + }; + }; + + dma@c300 { + #address-cells = <1>; + #size-cells = <1>; + compatible = "fsl,mpc8610-dma", "fsl,mpc8540-dma"; + cell-index = <1>; + reg = <c300 4>; /* DMA general status register */ + ranges = <0 c100 200>; + + dma-channel@0 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <0>; + reg = <0 80>; + interrupt-parent = <&mpic>; + interrupts = <3c 2>; + }; + dma-channel@1 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <1>; + reg = <80 80>; + interrupt-parent = <&mpic>; + interrupts = <3d 2>; + }; + dma-channel@2 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <2>; + reg = <100 80>; + interrupt-parent = <&mpic>; + interrupts = <3e 2>; + }; + dma-channel@3 { + compatible = "fsl,mpc8610-dma-channel", + "fsl,mpc8540-dma-channel"; + cell-index = <3>; + reg = <180 80>; + interrupt-parent = <&mpic>; + interrupts = <3f 2>; + }; + }; + }; pci@e0008000 { diff --git a/arch/powerpc/configs/mpc8610_hpcd_defconfig b/arch/powerpc/configs/mpc8610_hpcd_defconfig index 0483211..fb68886 100644 --- a/arch/powerpc/configs/mpc8610_hpcd_defconfig +++ b/arch/powerpc/configs/mpc8610_hpcd_defconfig @@ -684,7 +684,7 @@ CONFIG_SERIAL_8250_RSA=y CONFIG_SERIAL_CORE=y CONFIG_SERIAL_CORE_CONSOLE=y # CONFIG_SERIAL_JSM is not set -CONFIG_SERIAL_OF_PLATFORM=y +# CONFIG_SERIAL_OF_PLATFORM is not set CONFIG_UNIX98_PTYS=y # CONFIG_LEGACY_PTYS is not set # CONFIG_IPMI_HANDLER is not set @@ -699,7 +699,60 @@ CONFIG_UNIX98_PTYS=y # CONFIG_RAW_DRIVER is not set # CONFIG_TCG_TPM is not set CONFIG_DEVPORT=y -# CONFIG_I2C is not set +CONFIG_I2C=y +CONFIG_I2C_BOARDINFO=y +# CONFIG_I2C_CHARDEV is not set + +# +# I2C Algorithms +# +# CONFIG_I2C_ALGOBIT is not set +# CONFIG_I2C_ALGOPCF is not set +# CONFIG_I2C_ALGOPCA is not set + +# +# I2C Hardware Bus support +# +# CONFIG_I2C_ALI1535 is not set +# CONFIG_I2C_ALI1563 is not set +# CONFIG_I2C_ALI15X3 is not set +# CONFIG_I2C_AMD756 is not set +# CONFIG_I2C_AMD8111 is not set +# CONFIG_I2C_I801 is not set +# CONFIG_I2C_I810 is not set +# CONFIG_I2C_PIIX4 is not set +CONFIG_I2C_MPC=y +# CONFIG_I2C_NFORCE2 is not set +# CONFIG_I2C_OCORES is not set +# CONFIG_I2C_PARPORT_LIGHT is not set +# CONFIG_I2C_PROSAVAGE is not set +# CONFIG_I2C_SAVAGE4 is not set +# CONFIG_I2C_SIMTEC is not set +# CONFIG_I2C_SIS5595 is not set +# CONFIG_I2C_SIS630 is not set +# CONFIG_I2C_SIS96X is not set +# CONFIG_I2C_TAOS_EVM is not set +# CONFIG_I2C_VIA is not set +# CONFIG_I2C_VIAPRO is not set +# CONFIG_I2C_VOODOO3 is not set + +# +# Miscellaneous I2C Chip support +# +# CONFIG_SENSORS_DS1337 is not set +# CONFIG_SENSORS_DS1374 is not set +# CONFIG_DS1682 is not set +# CONFIG_SENSORS_EEPROM is not set +# CONFIG_SENSORS_PCF8574 is not set +# CONFIG_SENSORS_PCA9539 is not set +# CONFIG_SENSORS_PCF8591 is not set +# CONFIG_SENSORS_M41T00 is not set +# CONFIG_SENSORS_MAX6875 is not set +# CONFIG_SENSORS_TSL2550 is not set +# CONFIG_I2C_DEBUG_CORE is not set +# CONFIG_I2C_DEBUG_ALGO is not set +# CONFIG_I2C_DEBUG_BUS is not set +# CONFIG_I2C_DEBUG_CHIP is not set # # SPI support @@ -746,7 +799,119 @@ CONFIG_DUMMY_CONSOLE=y # # Sound # -# CONFIG_SOUND is not set +CONFIG_SOUND=y + +# +# Advanced Linux Sound Architecture +# +CONFIG_SND=y +CONFIG_SND_TIMER=y +CONFIG_SND_PCM=y +# CONFIG_SND_SEQUENCER is not set +CONFIG_SND_OSSEMUL=y +CONFIG_SND_MIXER_OSS=y +CONFIG_SND_PCM_OSS=y +# CONFIG_SND_PCM_OSS_PLUGINS is not set +# CONFIG_SND_DYNAMIC_MINORS is not set +# CONFIG_SND_SUPPORT_OLD_API is not set +CONFIG_SND_VERBOSE_PROCFS=y +# CONFIG_SND_VERBOSE_PRINTK is not set +# CONFIG_SND_DEBUG is not set + +# +# Generic devices +# +# CONFIG_SND_DUMMY is not set +# CONFIG_SND_MTPAV is not set +# CONFIG_SND_SERIAL_U16550 is not set +# CONFIG_SND_MPU401 is not set + +# +# PCI devices +# +# CONFIG_SND_AD1889 is not set +# CONFIG_SND_ALS300 is not set +# CONFIG_SND_ALS4000 is not set +# CONFIG_SND_ALI5451 is not set +# CONFIG_SND_ATIIXP is not set +# CONFIG_SND_ATIIXP_MODEM is not set +# CONFIG_SND_AU8810 is not set +# CONFIG_SND_AU8820 is not set +# CONFIG_SND_AU8830 is not set +# CONFIG_SND_AZT3328 is not set +# CONFIG_SND_BT87X is not set +# CONFIG_SND_CA0106 is not set +# CONFIG_SND_CMIPCI is not set +# CONFIG_SND_CS4281 is not set +# CONFIG_SND_CS46XX is not set +# CONFIG_SND_CS5530 is not set +# CONFIG_SND_DARLA20 is not set +# CONFIG_SND_GINA20 is not set +# CONFIG_SND_LAYLA20 is not set +# CONFIG_SND_DARLA24 is not set +# CONFIG_SND_GINA24 is not set +# CONFIG_SND_LAYLA24 is not set +# CONFIG_SND_MONA is not set +# CONFIG_SND_MIA is not set +# CONFIG_SND_ECHO3G is not set +# CONFIG_SND_INDIGO is not set +# CONFIG_SND_INDIGOIO is not set +# CONFIG_SND_INDIGODJ is not set +# CONFIG_SND_EMU10K1 is not set +# CONFIG_SND_EMU10K1X is not set +# CONFIG_SND_ENS1370 is not set +# CONFIG_SND_ENS1371 is not set +# CONFIG_SND_ES1938 is not set +# CONFIG_SND_ES1968 is not set +# CONFIG_SND_FM801 is not set +# CONFIG_SND_HDA_INTEL is not set +# CONFIG_SND_HDSP is not set +# CONFIG_SND_HDSPM is not set +# CONFIG_SND_ICE1712 is not set +# CONFIG_SND_ICE1724 is not set +# CONFIG_SND_INTEL8X0 is not set +# CONFIG_SND_INTEL8X0M is not set +# CONFIG_SND_KORG1212 is not set +# CONFIG_SND_MAESTRO3 is not set +# CONFIG_SND_MIXART is not set +# CONFIG_SND_NM256 is not set +# CONFIG_SND_PCXHR is not set +# CONFIG_SND_RIPTIDE is not set +# CONFIG_SND_RME32 is not set +# CONFIG_SND_RME96 is not set +# CONFIG_SND_RME9652 is not set +# CONFIG_SND_SONICVIBES is not set +# CONFIG_SND_TRIDENT is not set +# CONFIG_SND_VIA82XX is not set +# CONFIG_SND_VIA82XX_MODEM is not set +# CONFIG_SND_VX222 is not set +# CONFIG_SND_YMFPCI is not set + +# +# ALSA PowerMac devices +# + +# +# ALSA PowerPC devices +# + +# +# System on Chip audio support +# +CONFIG_SND_SOC=y + +# +# SoC Audio support for SuperH +# + +# +# ALSA SoC audio for Freescale SOCs +# +CONFIG_SND_SOC_MPC8610=y +CONFIG_SND_SOC_MPC8610_HPCD=y +CONFIG_SND_SOC_CS4270=y +CONFIG_SND_SOC_CS4270_VD33_ERRATA=y + CONFIG_HID_SUPPORT=y CONFIG_HID=y # CONFIG_HID_DEBUG is not set diff --git a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c index 6390895..6e1bde3 100644 --- a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c +++ b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c @@ -34,9 +34,27 @@ #include <asm/mpic.h> +#include <asm/of_platform.h> #include <sysdev/fsl_pci.h> #include <sysdev/fsl_soc.h> +static struct of_device_id mpc8610_ids[] = { + { .type = "soc", }, + {} +}; + +static int __init mpc8610_declare_of_platform_devices(void) +{ + if (!machine_is(mpc86xx_hpcd)) + return 0; + + /* Without this call, the SSI device driver won't get probed. */ + of_platform_bus_probe(NULL, mpc8610_ids, NULL); + + return 0; +} +device_initcall(mpc8610_declare_of_platform_devices); + void __init mpc86xx_hpcd_init_irq(void) { diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 97b2552..43bbc60 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -28,6 +28,7 @@ source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/fsl/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 3041403..4869c9a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig new file mode 100644 index 0000000..4a5bbd2 --- /dev/null +++ b/sound/soc/fsl/Kconfig @@ -0,0 +1,21 @@ +menu "ALSA SoC audio for Freescale SOCs" + +config SND_SOC_MPC8610 + bool "ALSA SoC support for the MPC8610 SOC" + depends on SND_SOC #&& MPC8610_HPCD + default y #if MPC8610 + help + Say Y if you want to add support for codecs attached to the SSI + device on an MPC8610. + +config SND_SOC_MPC8610_HPCD + # ALSA SoC support for Freescale MPC8610HPCD + bool "ALSA SoC support for the Freescale MPC8610 HPCD board" + depends on SND_SOC_MPC8610 + select SND_SOC_CS4270 + select SND_SOC_CS4270_VD33_ERRATA + default y #if MPC8610_HPCD + help + Say Y if you want to enable audio on the Freescale MPC8610 HPCD. + +endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile new file mode 100644 index 0000000..62f680a --- /dev/null +++ b/sound/soc/fsl/Makefile @@ -0,0 +1,6 @@ +# MPC8610 HPCD Machine Support +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o + +# MPC8610 Platform Support +obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o + diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c new file mode 100644 index 0000000..6b86be0 --- /dev/null +++ b/sound/soc/fsl/fsl_dma.c @@ -0,0 +1,819 @@ +/* + * Freescale DMA ALSA SoC PCM driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + * + * This driver implements ASoC support for the Elo DMA controller, which is + * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, + * the PCM driver is what handles the DMA buffer. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> +#include <linux/delay.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/io.h> + +#include "fsl_dma.h" + +/* + * The formats that the DMA controller supports, which is anything + * that is 8, 16, or 32 bits. + */ +#define FSLDMA_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_U24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_S32_BE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_U32_BE) + +#define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS) + +/* DMA global data. This structure is used by fsl_dma_open() to determine + * which DMA channels to assign to a substream. Unfortunately, ASoC V1 does + * not allow the machine driver to provide this information to the PCM + * driver in advance, and there's no way to differentiate between the two + * DMA controllers. So for now, this driver only supports one SSI device + * using two DMA channels. We cannot support multiple DMA devices. + * + * ssi_stx_phys: bus address of SSI STX register + * ssi_srx_phys: bus address of SSI SRX register + * dma_channel: pointer to the DMA channel's registers + * irq: IRQ for this DMA channel + * assigned: set to 1 if that DMA channel is assigned to a substream + */ +static struct { + dma_addr_t ssi_stx_phys; + dma_addr_t ssi_srx_phys; + struct ccsr_dma_channel __iomem *dma_channel[2]; + unsigned int irq[2]; + unsigned int assigned[2]; +} dma_global_data; + +/* + * The number of DMA links to use. Two is the bare minimum, but if you + * have really small links you might need more. + */ +#define NUM_DMA_LINKS 2 + +/* Per-substream DMA data + * + * Each substream has a 1-to-1 association with a DMA channel. + * + * The link[] array is first because it needs to be aligned on a 32-byte + * boundary, so putting it first will ensure alignment without padding the + * structure. + * + * link[]: array of link descriptors + * controller_id: which DMA controller (0, 1, ...) + * channel_id: which DMA channel on the controller (0, 1, 2, ...) + * dma_channel: pointer to the DMA channel's registers + * irq: IRQ for this DMA channel + * substream: pointer to the substream object, needed by the ISR + * ssi_sxx_phys: bus address of the STX or SRX register to use + * ld_buf_phys: physical address of the LD buffer + * current_link: index into link[] of the link currently being processed + * dma_buf_phys: physical address of the DMA buffer + * dma_buf_next: physical address of the next period to process + * dma_buf_end: physical address of the byte after the end of the DMA buffer + * period_size: the size of a single period + */ +struct fsl_dma_private { + struct fsl_dma_link_descriptor link[NUM_DMA_LINKS]; + unsigned int controller_id; + unsigned int channel_id; + struct ccsr_dma_channel __iomem *dma_channel; + unsigned int irq; + struct snd_pcm_substream *substream; + dma_addr_t ssi_sxx_phys; + dma_addr_t ld_buf_phys; + unsigned int current_link; + dma_addr_t dma_buf_phys; + dma_addr_t dma_buf_next; + dma_addr_t dma_buf_end; + size_t period_size; + unsigned int num_periods; +}; + +/* + * Define the hardware characteristics for the PCM hardware. + * + * The PCM hardware is the Freescale DMA controller. This structure defines + * the capabilities of that hardware. + * + * Since the sampling rate and data format are not controlled by the DMA + * controller, we specify no limits for those values. The only exception is + * period_bytes_min, which is set to a reasonably low value to prevent the + * DMA controller from generating too many interrupts per second. + * + * Since each link descriptor has a 32-bit byte count field, we set + * period_bytes_max to the largest 32-bit number. We also have no maximum + * number of periods. + */ +static const struct snd_pcm_hardware fsl_dma_hardware = { + + .info = SNDRV_PCM_INFO_INTERLEAVED, + .formats = FSLDMA_PCM_FORMATS, + .rates = FSLDMA_PCM_RATES, + .rate_min = 5512, + .rate_max = 192000, + .period_bytes_min = 512, /* A reasonable limit */ + .period_bytes_max = (u32) -1, + .periods_min = NUM_DMA_LINKS, + .periods_max = (unsigned int) -1, + .buffer_bytes_max = 128 * 1024, /* A reasonable limit */ +}; + +/* + * Tell ALSA that the DMA transfer has aborted + * + * This function should be called by the ISR whenever the DMA controller + * halts data transfer. + */ +static void fsl_dma_abort_stream(struct snd_pcm_substream *substream) +{ + unsigned long flags; + + snd_pcm_stream_lock_irqsave(substream, flags); + + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + + snd_pcm_stream_unlock_irqrestore(substream, flags); +} + +/* + * Update the link descriptor pointers to point to the next period buffer. +*/ +static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) +{ + struct fsl_dma_link_descriptor *link = + &dma_private->link[dma_private->current_link]; + + /* Update our link descriptors to point to the next period */ + if (dma_private->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->source_addr = + cpu_to_be32(dma_private->dma_buf_next); + else + link->dest_addr = + cpu_to_be32(dma_private->dma_buf_next); + + /* Update our variables for next time */ + dma_private->dma_buf_next += dma_private->period_size; + + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + dma_private->dma_buf_next = dma_private->dma_buf_phys; + + if (++dma_private->current_link >= NUM_DMA_LINKS) + dma_private->current_link = 0; +} + +/* + * Interrupt handler for the DMA controller + */ +static irqreturn_t fsl_dma_isr(int irq, void *dev_id) +{ + struct fsl_dma_private *dma_private = dev_id; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + irqreturn_t ret = IRQ_NONE; + u32 sr, sr2 = 0; + + /* We got an interrupt, so read the status register to see what we + were interrupted for. + */ + sr = in_be32(&dma_channel->sr); + + if (sr & CCSR_DMA_SR_TE) { + dev_err(dma_private->substream->pcm->card->dev, + "DMA transmit error (controller=%u channel=%u irq=%u\n", + dma_private->controller_id, + dma_private->channel_id, irq); + fsl_dma_abort_stream(dma_private->substream); + sr2 |= CCSR_DMA_SR_TE; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_CH) + ret = IRQ_HANDLED; + + if (sr & CCSR_DMA_SR_PE) { + dev_err(dma_private->substream->pcm->card->dev, + "DMA%u programming error (channel=%u irq=%u)\n", + dma_private->controller_id, + dma_private->channel_id, irq); + fsl_dma_abort_stream(dma_private->substream); + sr2 |= CCSR_DMA_SR_PE; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_EOLNI) { + sr2 |= CCSR_DMA_SR_EOLNI; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_CB) + ret = IRQ_HANDLED; + + if (sr & CCSR_DMA_SR_EOSI) { + struct snd_pcm_substream *substream = dma_private->substream; + + /* Tell ALSA we completed a period. */ + snd_pcm_period_elapsed(substream); + + /* + * Update our link descriptors to point to the next period. We + * only need to do this if the number of periods is not equal to + * the number of links. + */ + if (dma_private->num_periods != NUM_DMA_LINKS) + fsl_dma_update_pointers(dma_private); + + sr2 |= CCSR_DMA_SR_EOSI; + ret = IRQ_HANDLED; + } + + if (sr & CCSR_DMA_SR_EOLSI) { + sr2 |= CCSR_DMA_SR_EOLSI; + ret = IRQ_HANDLED; + } + + /* Clear the bits that we set */ + if (sr2) + out_be32(&dma_channel->sr, sr2); + + return ret; +} + +/* + * Initialize this PCM driver. + * + * This function is called when the codec driver calls snd_soc_new_pcms(), + * once for each .dai_link in the machine driver's snd_soc_machine + * structure. + */ +static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + static u64 fsl_dma_dmamask = 0xffffffff; + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &fsl_dma_dmamask; + + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = fsl_dma_dmamask; + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[0].substream->dma_buffer); + if (ret) { + dev_err(card->dev, + "Can't allocate playback DMA buffer (size=%u)\n", + fsl_dma_hardware.buffer_bytes_max); + return -ENOMEM; + } + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[1].substream->dma_buffer); + if (ret) { + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + dev_err(card->dev, + "Can't allocate capture DMA buffer (size=%u)\n", + fsl_dma_hardware.buffer_bytes_max); + return -ENOMEM; + } + + return 0; +} + +/* + * Open a new stream. + * + * Each stream has its own DMA buffer. + */ +static int fsl_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private; + dma_addr_t ld_buf_phys; + unsigned int channel; + int ret = 0; + + /* + * Reject any DMA buffer whose size is not a multiple of the period + * size. We need to make sure that the DMA buffer can be evenly divided + * into periods. + */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(substream->pcm->card->dev, "invalid buffer size\n"); + return ret; + } + + channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + if (dma_global_data.assigned[channel]) { + dev_err(substream->pcm->card->dev, + "DMA channel already assigned\n"); + return -EBUSY; + } + + dma_private = dma_alloc_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); + if (!dma_private) { + dev_err(substream->pcm->card->dev, + "can't allocate DMA private data\n"); + return -ENOMEM; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_private->ssi_sxx_phys = dma_global_data.ssi_stx_phys; + else + dma_private->ssi_sxx_phys = dma_global_data.ssi_srx_phys; + + dma_private->dma_channel = dma_global_data.dma_channel[channel]; + dma_private->irq = dma_global_data.irq[channel]; + dma_private->substream = substream; + dma_private->ld_buf_phys = ld_buf_phys; + dma_private->dma_buf_phys = substream->dma_buffer.addr; + + /* We only support one DMA controller for now */ + dma_private->controller_id = 0; + dma_private->channel_id = channel; + + ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); + if (ret) { + dev_err(substream->pcm->card->dev, + "can't register ISR for IRQ %u (ret=%i)\n", + dma_private->irq, ret); + dma_free_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), + dma_private, dma_private->ld_buf_phys); + return ret; + } + + dma_global_data.assigned[channel] = 1; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); + runtime->private_data = dma_private; + + return 0; +} + +/* + * Allocate the DMA buffer and the DMA link descriptors. + * + * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link + * descriptors that ping-pong from one period to the next. For example, if + * there are six periods and two link descriptors, this is how they look + * before playback starts: + * + * The last link descriptor + * ____________ points back to the first + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * | | + * V V + * _________________________________________ + * | | | | | | | The DMA buffer is + * | | | | | | | divided into 6 parts + * |______|______|______|______|______|______| + * + * and here's how they look after the first period is finished playing: + * + * ____________ + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * |______________ + * | | + * V V + * _________________________________________ + * | | | | | | | + * | | | | | | | + * |______|______|______|______|______|______| + * + * The first link descriptor now points to the third period. The DMA + * controller is currently playing the second period. When it finishes, it + * will jump back to the first descriptor and play the third period. + * + * There are four reasons we do this: + * + * 1. The only way to get the DMA controller to automatically restart the + * transfer when it gets to the end of the buffer is to use chaining + * mode. Basic direct mode doesn't offer that feature. + * 2. We need to receive an interrupt at the end of every period. The DMA + * controller can generate an interrupt at the end of every link transfer + * (aka segment). Making each period into a DMA segment will give us the + * interrupts we need. + * 3. By creating only two link descriptors, regardless of the number of + * periods, we do not need to reallocate the link descriptors if the + * number of periods changes. + * 4. All of the audio data is still stored in a single, contiguous DMA + * buffer, which is what ALSA expects. We're just dividing it into + * contiguous parts, and creating a link descriptor for each one. + * + * Note that due to a quirk of the SSI's STX register, the target address + * for the DMA operations depends on the sample size. So we don't program + * the dest_addr (for playback -- source_addr for capture) fields in the + * link descriptors here. We do that in fsl_dma_prepare() + */ +static int fsl_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + + dma_addr_t temp_addr; /* Pointer to next period */ + u64 temp_link; /* Pointer to next link descriptor */ + u32 mr; /* Temporary variable for MR register */ + + unsigned int i; + + /* Get all the parameters we need */ + size_t buffer_size = params_buffer_bytes(hw_params); + size_t period_size = params_period_bytes(hw_params); + + /* Initialize our DMA tracking variables */ + dma_private->period_size = period_size; + dma_private->num_periods = params_periods(hw_params); + dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; + dma_private->dma_buf_next = dma_private->dma_buf_phys + + (NUM_DMA_LINKS * period_size); + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + dma_private->dma_buf_next = dma_private->dma_buf_phys; + + /* + * Initialize each link descriptor. + * + * The actual address in STX0 (destination for playback, source for + * capture) is based on the sample size, but we don't know the sample + * size in this function, so we'll have to adjust that later. See + * comments in fsl_dma_prepare(). + * + * The DMA controller does not have a cache, so the CPU does not + * need to tell it to flush its cache. However, the DMA + * controller does need to tell the CPU to flush its cache. + * That's what the SNOOP bit does. + * + * Also, even though the DMA controller supports 36-bit addressing, for + * simplicity we currently support only 32-bit addresses for the audio + * buffer itself. + */ + temp_addr = substream->dma_buffer.addr; + temp_link = dma_private->ld_buf_phys + + sizeof(struct fsl_dma_link_descriptor); + + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + link->count = cpu_to_be32(period_size); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->next = cpu_to_be64(temp_link); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->source_addr = cpu_to_be32(temp_addr); + else + link->dest_addr = cpu_to_be32(temp_addr); + + temp_addr += period_size; + temp_link += sizeof(struct fsl_dma_link_descriptor); + } + /* The last link descriptor points to the first */ + dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + + /* Tell the DMA controller where the first link descriptor is */ + out_be32(&dma_channel->clndar, + CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); + out_be32(&dma_channel->eclndar, + CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); + + /* The manual says the BCR must be clear before enabling EMP */ + out_be32(&dma_channel->bcr, 0); + + /* Program the mode register for interrupts, external master control, + and source/destination hold. Also clear the Channel Abort bit. + */ + mr = in_be32(&dma_channel->mr) & + ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); + + /* + * We want External Master Start and External Master Pause enabled, + * because the SSI is controlling the DMA controller. We want the DMA + * controller to be set up in advance, and then we signal only the SSI + * to start transfering. + * + * We want End-Of-Segment Interrupts enabled, because this will generate + * an interrupt at the end of each segment (each link descriptor + * represents one segment). Each DMA segment is the same thing as an + * ALSA period, so this is how we get an interrupt at the end of every + * period. + * + * We want Error Interrupt enabled, so that we can get an error if + * the DMA controller is mis-programmed somehow. + */ + mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + CCSR_DMA_MR_EMS_EN; + + /* For playback, we want the destination address to be held. For + capture, set the source address to be held. */ + mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; + + out_be32(&dma_channel->mr, mr); + + return 0; +} + +/* + * Prepare the DMA registers for playback. + * + * This function is called after the specifics of the audio data are known, + * i.e. snd_pcm_runtime is initialized. + * + * In this function, we finish programming the registers of the DMA + * controller that are dependent on the sample size. + * + * One of the drawbacks with big-endian is that when copying integers of + * different sizes to a fixed-sized register, the address to which the + * integer must be copied is dependent on the size of the integer. + * + * For example, if P is the address of a 32-bit register, and X is a 32-bit + * integer, then X should be copied to address P. However, if X is a 16-bit + * integer, then it should be copied to P+2. If X is an 8-bit register, + * then it should be copied to P+3. + * + * So for playback of 8-bit samples, the DMA controller must transfer single + * bytes from the DMA buffer to the last byte of the STX0 register, i.e. + * offset by 3 bytes. For 16-bit samples, the offset is two bytes. + * + * For 24-bit samples, the offset is 1 byte. However, the DMA controller + * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4, + * and 8 bytes at a time). So we do not support packed 24-bit samples. + * 24-bit data must be padded to 32 bits. + */ +static int fsl_dma_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + u32 mr; + unsigned int i; + dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ + unsigned int frame_size; /* Number of bytes per frame */ + + ssi_sxx_phys = dma_private->ssi_sxx_phys; + + mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | + CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); + + switch (runtime->sample_bits) { + case 8: + mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; + ssi_sxx_phys += 3; + break; + case 16: + mr |= CCSR_DMA_MR_DAHTS_2 | CCSR_DMA_MR_SAHTS_2; + ssi_sxx_phys += 2; + break; + case 32: + mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4; + break; + default: + dev_err(substream->pcm->card->dev, + "unsupported sample size %u\n", runtime->sample_bits); + return -EINVAL; + } + + frame_size = runtime->frame_bits / 8; + /* + * BWC should always be a multiple of the frame size. BWC determines + * how many bytes are sent/received before the DMA controller checks the + * SSI to see if it needs to stop. For playback, the transmit FIFO can + * hold three frames, so we want to send two frames at a time. For + * capture, the receive FIFO is triggered when it contains one frame, so + * we want to receive one frame at a time. + */ + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mr |= CCSR_DMA_MR_BWC(2 * frame_size); + else + mr |= CCSR_DMA_MR_BWC(frame_size); + + out_be32(&dma_channel->mr, mr); + + /* + * Program the address of the DMA transfer to/from the SSI. + */ + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->dest_addr = cpu_to_be32(ssi_sxx_phys); + else + link->source_addr = cpu_to_be32(ssi_sxx_phys); + } + + return 0; +} + +/* + * fsl_dma_pointer - 'pointer' PCM operation callback function + * + * This function is called by ALSA when ALSA wants to know where in the + * stream buffer the hardware currently is. + * + * For playback, the SAR register contains the physical address of the most + * recent DMA transfer. For capture, the value is in the DAR register. + * + * The base address of the buffer is stored in the source_addr field of the + * first link descriptor. + */ +static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + dma_addr_t position; + snd_pcm_uframes_t frames; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + position = in_be32(&dma_channel->sar); + else + position = in_be32(&dma_channel->dar); + + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); + + /* + * If the current address is just past the end of the buffer, wrap it + * around. + */ + if (frames == runtime->buffer_size) + frames = 0; + + return frames; +} + +/* + * Release the resources allocated in fsl_dma_hw_params() and + * de-program the registers. This function can be called multiple times. + */ +static int fsl_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + + if (dma_private) { + struct ccsr_dma_channel __iomem *dma_channel; + + dma_channel = dma_private->dma_channel; + + dump_dma_regs(__FUNCTION__, __LINE__, dma_channel); + dump_lds(dma_private); + + /* Stop the DMA */ + out_be32(&dma_channel->mr, CCSR_DMA_MR_CA); + out_be32(&dma_channel->mr, 0); + + /* Reset all the other registers */ + out_be32(&dma_channel->sr, -1); + out_be32(&dma_channel->clndar, 0); + out_be32(&dma_channel->eclndar, 0); + out_be32(&dma_channel->satr, 0); + out_be32(&dma_channel->sar, 0); + out_be32(&dma_channel->datr, 0); + out_be32(&dma_channel->dar, 0); + out_be32(&dma_channel->bcr, 0); + out_be32(&dma_channel->nlndar, 0); + out_be32(&dma_channel->enlndar, 0); + } + + return 0; +} + +/* + * Close the stream. + */ +static int fsl_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_dma_private *dma_private = runtime->private_data; + int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + if (dma_private) { + if (dma_private->irq) + free_irq(dma_private->irq, dma_private); + + if (dma_private->ld_buf_phys) { + dma_unmap_single(substream->pcm->dev, + dma_private->ld_buf_phys, + sizeof(dma_private->link), DMA_TO_DEVICE); + } + + /* Deallocate the fsl_dma_private structure */ + dma_free_coherent(substream->pcm->dev, + sizeof(struct fsl_dma_private), + dma_private, dma_private->ld_buf_phys); + substream->runtime->private_data = NULL; + } + + dma_global_data.assigned[dir] = 0; + + return 0; +} + +/* + * Remove this PCM driver. + */ +static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) { + substream = pcm->streams[i].substream; + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + } +} + +static struct snd_pcm_ops fsl_dma_ops = { + .open = fsl_dma_open, + .close = fsl_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsl_dma_hw_params, + .hw_free = fsl_dma_hw_free, + .prepare = fsl_dma_prepare, + .pointer = fsl_dma_pointer, +}; + +struct snd_soc_platform fsl_soc_platform = { + .name = "fsl-dma", + .pcm_ops = &fsl_dma_ops, + .pcm_new = fsl_dma_new, + .pcm_free = fsl_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(fsl_soc_platform); + +int fsl_dma_configure(struct fsl_dma_info *dma_info) +{ + static int initialized; + + /* We only support one DMA controller for now */ + if (initialized) + return 0; + + dma_global_data.ssi_stx_phys = dma_info->ssi_stx_phys; + dma_global_data.ssi_srx_phys = dma_info->ssi_srx_phys; + dma_global_data.dma_channel[0] = dma_info->dma_channel[0]; + dma_global_data.dma_channel[1] = dma_info->dma_channel[1]; + dma_global_data.irq[0] = dma_info->dma_irq[0]; + dma_global_data.irq[1] = dma_info->dma_irq[1]; + dma_global_data.assigned[0] = 0; + dma_global_data.assigned[1] = 0; + + initialized = 1; + return 1; +} +EXPORT_SYMBOL_GPL(fsl_dma_configure); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale FSL Elo DMA ASoC PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h new file mode 100644 index 0000000..430a6ce --- /dev/null +++ b/sound/soc/fsl/fsl_dma.h @@ -0,0 +1,149 @@ +/* + * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC + * + * 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 _MPC8610_PCM_H +#define _MPC8610_PCM_H + +struct ccsr_dma { + u8 res0[0x100]; + struct ccsr_dma_channel { + __be32 mr; /* Mode register */ + __be32 sr; /* Status register */ + __be32 eclndar; /* Current link descriptor extended addr reg */ + __be32 clndar; /* Current link descriptor address register */ + __be32 satr; /* Source attributes register */ + __be32 sar; /* Source address register */ + __be32 datr; /* Destination attributes register */ + __be32 dar; /* Destination address register */ + __be32 bcr; /* Byte count register */ + __be32 enlndar; /* Next link descriptor extended address reg */ + __be32 nlndar; /* Next link descriptor address register */ + u8 res1[4]; + __be32 eclsdar; /* Current list descriptor extended addr reg */ + __be32 clsdar; /* Current list descriptor address register */ + __be32 enlsdar; /* Next list descriptor extended address reg */ + __be32 nlsdar; /* Next list descriptor address register */ + __be32 ssr; /* Source stride register */ + __be32 dsr; /* Destination stride register */ + u8 res2[0x38]; + } channel[4]; + __be32 dgsr; +}; + +#define CCSR_DMA_MR_BWC_DISABLED 0x0F000000 +#define CCSR_DMA_MR_BWC_SHIFT 24 +#define CCSR_DMA_MR_BWC_MASK 0x0F000000 +#define CCSR_DMA_MR_BWC(x) \ + ((ilog2(x) << CCSR_DMA_MR_BWC_SHIFT) & CCSR_DMA_MR_BWC_MASK) +#define CCSR_DMA_MR_EMP_EN 0x00200000 +#define CCSR_DMA_MR_EMS_EN 0x00040000 +#define CCSR_DMA_MR_DAHTS_MASK 0x00030000 +#define CCSR_DMA_MR_DAHTS_1 0x00000000 +#define CCSR_DMA_MR_DAHTS_2 0x00010000 +#define CCSR_DMA_MR_DAHTS_4 0x00020000 +#define CCSR_DMA_MR_DAHTS_8 0x00030000 +#define CCSR_DMA_MR_SAHTS_MASK 0x0000C000 +#define CCSR_DMA_MR_SAHTS_1 0x00000000 +#define CCSR_DMA_MR_SAHTS_2 0x00004000 +#define CCSR_DMA_MR_SAHTS_4 0x00008000 +#define CCSR_DMA_MR_SAHTS_8 0x0000C000 +#define CCSR_DMA_MR_DAHE 0x00002000 +#define CCSR_DMA_MR_SAHE 0x00001000 +#define CCSR_DMA_MR_SRW 0x00000400 +#define CCSR_DMA_MR_EOSIE 0x00000200 +#define CCSR_DMA_MR_EOLNIE 0x00000100 +#define CCSR_DMA_MR_EOLSIE 0x00000080 +#define CCSR_DMA_MR_EIE 0x00000040 +#define CCSR_DMA_MR_XFE 0x00000020 +#define CCSR_DMA_MR_CDSM_SWSM 0x00000010 +#define CCSR_DMA_MR_CA 0x00000008 +#define CCSR_DMA_MR_CTM 0x00000004 +#define CCSR_DMA_MR_CC 0x00000002 +#define CCSR_DMA_MR_CS 0x00000001 + +#define CCSR_DMA_SR_TE 0x00000080 +#define CCSR_DMA_SR_CH 0x00000020 +#define CCSR_DMA_SR_PE 0x00000010 +#define CCSR_DMA_SR_EOLNI 0x00000008 +#define CCSR_DMA_SR_CB 0x00000004 +#define CCSR_DMA_SR_EOSI 0x00000002 +#define CCSR_DMA_SR_EOLSI 0x00000001 + +/* ECLNDAR takes bits 32-36 of the CLNDAR register */ +static inline u32 CCSR_DMA_ECLNDAR_ADDR(u64 x) +{ + return (x >> 32) & 0xf; +} + +#define CCSR_DMA_CLNDAR_ADDR(x) ((x) & 0xFFFFFFFE) +#define CCSR_DMA_CLNDAR_EOSIE 0x00000008 + +/* SATR and DATR, combined */ +#define CCSR_DMA_ATR_PBATMU 0x20000000 +#define CCSR_DMA_ATR_TFLOWLVL_0 0x00000000 +#define CCSR_DMA_ATR_TFLOWLVL_1 0x06000000 +#define CCSR_DMA_ATR_TFLOWLVL_2 0x08000000 +#define CCSR_DMA_ATR_TFLOWLVL_3 0x0C000000 +#define CCSR_DMA_ATR_PCIORDER 0x02000000 +#define CCSR_DMA_ATR_SME 0x01000000 +#define CCSR_DMA_ATR_NOSNOOP 0x00040000 +#define CCSR_DMA_ATR_SNOOP 0x00050000 +#define CCSR_DMA_ATR_ESAD_MASK 0x0000000F + +/** + * List Descriptor for extended chaining mode DMA operations. + * + * The CLSDAR register points to the first (in a linked-list) List + * Descriptor. Each object must be aligned on a 32-byte boundary. Each + * list descriptor points to a linked-list of link Descriptors. + */ +struct fsl_dma_list_descriptor { + __be64 next; /* Address of next list descriptor */ + __be64 first_link; /* Address of first link descriptor */ + __be32 source; /* Source stride */ + __be32 dest; /* Destination stride */ + u8 res[8]; /* Reserved */ +} __attribute__ ((aligned(32), packed)); + +/** + * Link Descriptor for basic and extended chaining mode DMA operations. + * + * A Link Descriptor points to a single DMA buffer. Each link descriptor + * must be aligned on a 32-byte boundary. + */ +struct fsl_dma_link_descriptor { + __be32 source_attr; /* Programmed into SATR register */ + __be32 source_addr; /* Programmed into SAR register */ + __be32 dest_attr; /* Programmed into DATR register */ + __be32 dest_addr; /* Programmed into DAR register */ + __be64 next; /* Address of next link descriptor */ + __be32 count; /* Byte count */ + u8 res[4]; /* Reserved */ +} __attribute__ ((aligned(32), packed)); + +/* DMA information needed to create a snd_soc_cpu_dai object + * + * ssi_stx_phys: bus address of SSI STX register to use + * ssi_srx_phys: bus address of SSI SRX register to use + * dma[0]: points to the DMA channel to use for playback + * dma[1]: points to the DMA channel to use for capture + * dma_irq[0]: IRQ of the DMA channel to use for playback + * dma_irq[1]: IRQ of the DMA channel to use for capture + */ +struct fsl_dma_info { + dma_addr_t ssi_stx_phys; + dma_addr_t ssi_srx_phys; + struct ccsr_dma_channel __iomem *dma_channel[2]; + unsigned int dma_irq[2]; +}; + +extern struct snd_soc_platform fsl_soc_platform; + +int fsl_dma_configure(struct fsl_dma_info *dma_info); + +#endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c new file mode 100644 index 0000000..a219e58 --- /dev/null +++ b/sound/soc/fsl/fsl_ssi.c @@ -0,0 +1,614 @@ +/* + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <asm/immap_86xx.h> + +#include "fsl_ssi.h" + +/* + * The sample rates supported by the I2S + * + * This driver currently only supports the SSI running in I2S slave mode, + * which means the codec determines the sample rate. Therefore, we tell + * ALSA that we support all rates and let the codec driver decide what rates + * are really supported. + */ +#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS) + +/* + * The audio formats supported by the SSI + * + * This driver currently only supports the SSI running in I2S slave mode. + * + * The SSI has a limitation in that the samples must be in the same byte + * order as the host CPU. This is because when multiple bytes are written + * to the STX register, the bytes and bits must be written in the same + * order. The STX is a shift register, so all the bits need to be aligned + * (bit-endianness must match byte-endianness). Processors typically write + * the bits within a byte in the same order that the bytes of a word are + * written in. So if the host CPU is big-endian, then only big-endian + * samples will be written to STX properly. + */ +#ifdef __BIG_ENDIAN +#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S24_BE) +#else +#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) +#endif + +/* SSI information. + * + * name: short name for this device ("SSI0", "SSI1", etc) + * ssi: pointer to the SSI's registers + * ssi_phys: physical address of the SSI registers + * irq: IRQ of this SSI + * dev: struct device pointer + * playback: the number of playback streams opened + * capture: the number of capture streams opened + */ +struct fsl_ssi_private { + char name[8]; + struct ccsr_ssi __iomem *ssi; + dma_addr_t ssi_phys; + unsigned int irq; + struct device *dev; + unsigned int playback; + unsigned int capture; + struct snd_soc_cpu_dai cpu_dai; + struct device_attribute dev_attr; + + struct { + unsigned int rfrc; + unsigned int tfrc; + unsigned int cmdau; + unsigned int cmddu; + unsigned int rxt; + unsigned int rdr1; + unsigned int rdr0; + unsigned int tde1; + unsigned int tde0; + unsigned int roe1; + unsigned int roe0; + unsigned int tue1; + unsigned int tue0; + unsigned int tfs; + unsigned int rfs; + unsigned int tls; + unsigned int rls; + unsigned int rff1; + unsigned int rff0; + unsigned int tfe1; + unsigned int tfe0; + } stats; +}; + +/* + * SSI interrupt handler. + * + * Do we even want an SSI interrupt handler? Maybe for error handling, but + * buffer processing is handled via DMA interrupts. + */ +static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) +{ + struct fsl_ssi_private *ssi_private = dev_id; + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + irqreturn_t ret = IRQ_NONE; + __be32 sisr; + __be32 sisr2 = 0; + + /* We got an interrupt, so read the status register to see what we + were interrupted for. We mask it with the Interrupt Enable register + so that we only check for events that we're interested in. + */ + sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier); + + if (sisr & CCSR_SSI_SISR_RFRC) { + ssi_private->stats.rfrc++; + sisr2 |= CCSR_SSI_SISR_RFRC; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFRC) { + ssi_private->stats.tfrc++; + sisr2 |= CCSR_SSI_SISR_TFRC; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_CMDAU) { + ssi_private->stats.cmdau++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_CMDDU) { + ssi_private->stats.cmddu++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RXT) { + ssi_private->stats.rxt++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RDR1) { + ssi_private->stats.rdr1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RDR0) { + ssi_private->stats.rdr0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TDE1) { + ssi_private->stats.tde1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TDE0) { + ssi_private->stats.tde0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_ROE1) { + ssi_private->stats.roe1++; + sisr2 |= CCSR_SSI_SISR_ROE1; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_ROE0) { + ssi_private->stats.roe0++; + sisr2 |= CCSR_SSI_SISR_ROE0; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TUE1) { + ssi_private->stats.tue1++; + sisr2 |= CCSR_SSI_SISR_TUE1; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TUE0) { + ssi_private->stats.tue0++; + sisr2 |= CCSR_SSI_SISR_TUE0; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFS) { + ssi_private->stats.tfs++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFS) { + ssi_private->stats.rfs++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TLS) { + ssi_private->stats.tls++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RLS) { + ssi_private->stats.rls++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFF1) { + ssi_private->stats.rff1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_RFF0) { + ssi_private->stats.rff0++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFE1) { + ssi_private->stats.tfe1++; + ret = IRQ_HANDLED; + } + + if (sisr & CCSR_SSI_SISR_TFE0) { + ssi_private->stats.tfe0++; + ret = IRQ_HANDLED; + } + + /* Clear the bits that we set */ + if (sisr2) + out_be32(&ssi->sisr, sisr2); + + return ret; +} + +/* + * Startup + * + * This is the first function called when a stream is opened. + * + * If this is the first stream open, then grab the IRQ and program most of + * the SSI registers. + */ +static int fsl_ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + /* + * If this is the first stream opened, then request the IRQ + * and initialize the SSI registers. + */ + if (!ssi_private->playback && !ssi_private->capture) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int ret; + + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, + ssi_private->name, ssi_private); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not claim irq %u\n", ssi_private->irq); + return ret; + } + + /* + * Section 16.5 of the MPC8610 reference manual says that the + * SSI needs to be disabled before updating the registers we set + * here. + */ + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + /* + * Program the SSI into I2S Slave Non-Network Synchronous mode. + * Also enable the transmit and receive FIFO. + * + * FIXME: Little-endian samples require a different shift dir + */ + clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, + CCSR_SSI_SCR_TFR_CLK_DIS | + CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); + + out_be32(&ssi->stcr, + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | + CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | + CCSR_SSI_STCR_TSCKP); + + out_be32(&ssi->srcr, + CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | + CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | + CCSR_SSI_SRCR_RSCKP); + + /* + * The DC and PM bits are only used if the SSI is the clock + * master. + */ + + /* 4. Enable the interrupts and DMA requests */ + out_be32(&ssi->sier, + CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | + CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | + CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | + CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | + CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN); + + /* + * Set the watermark for transmit FIFI 0 and receive FIFO 0. We + * don't use FIFO 1. Since the SSI only supports stereo, the + * watermark should never be an odd number. + */ + out_be32(&ssi->sfcsr, + CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2)); + + /* + * We keep the SSI disabled because if we enable it, then the + * DMA controller will start. It's not supposed to start until + * the SCR.TE (or SCR.RE) bit is set, but it does anyway. The + * DMA controller will transfer one "BWC" of data (i.e. the + * amount of data that the MR.BWC bits are set to). The reason + * this is bad is because at this point, the PCM driver has not + * finished initializing the DMA controller. + */ + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssi_private->playback++; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ssi_private->capture++; + + return 0; +} + +/* + * Prepare the SSI. + * + * Most of the SSI registers have been programmed in the startup function, + * but the word length must be programmed here. Unfortunately, programming + * the SxCCR.WL bits requires the SSI to be temporarily disabled. This can + * cause a problem with supporting simultaneous playback and capture. If + * the SSI is already playing a stream, then that stream may be temporarily + * stopped when you start capture. + * + * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the + * clock master. + */ +static int fsl_ssi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 wl; + + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + + setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + return 0; +} + +/* + * Start and stop the DMA transfer. + * + * This function is called by ALSA to start, stop, pause, and resume the DMA + * transfer of data. + * + * The SSI has put the DMA in external master start and pause mode, which + * means the SSI completely controls the flow of data. + */ +static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + } else { + setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + + /* + * I have no idea why we need a delay here. Without it, + * the DMA won't start (or the SSI won't tell the DMA to + * start). + */ + msleep(1); + } + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); + else + clrbits32(&ssi->scr, CCSR_SSI_SCR_RE); + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSI if there + * are no other substreams open. + */ +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssi_private->playback--; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ssi_private->capture--; + + /* + * If this is the last active substream, disable the SSI and release + * the IRQ. + */ + if (!ssi_private->playback && !ssi_private->capture) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + + free_irq(ssi_private->irq, ssi_private); + } +} + +/* Set the clock frequency and direction + * + * This function is called by the machine driver to tell us what the clock + * frequency and direction are. + * + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN), + * and we don't care about the frequency. Return an error if the direction + * is not SND_SOC_CLOCK_IN. + */ +static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + + return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; +} + +/* Set the serial format. + * + * This function is called by the machine driver to tell us what serial + * format to use. + * + * Currently, we only support I2S mode. Return an error if the format is + * not SND_SOC_DAIFMT_I2S. + */ +static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +{ + return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; +} + +/* + * Template CPU DAI for the SSI + */ +static struct snd_soc_cpu_dai fsl_ssi_dai_template = { + .playback = { + /* The SSI does not support monaural audio. */ + .channels_min = 2, + .channels_max = 2, + .rates = FSLSSI_I2S_RATES, + .formats = FSLSSI_I2S_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = FSLSSI_I2S_RATES, + .formats = FSLSSI_I2S_FORMATS, + }, + .ops = { + .startup = fsl_ssi_startup, + .prepare = fsl_ssi_prepare, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_trigger, + }, + .dai_ops = { + .set_sysclk = fsl_ssi_set_sysclk, + .set_fmt = fsl_ssi_set_fmt, + }, +}; + +static ssize_t fsl_sysfs_ssi_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct fsl_ssi_private *ssi_private = + container_of(attr, struct fsl_ssi_private, dev_attr); + ssize_t length; + + length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc); + length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc); + length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau); + length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu); + length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt); + length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1); + length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0); + length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1); + length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0); + length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1); + length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0); + length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1); + length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0); + length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs); + length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs); + length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls); + length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls); + length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1); + length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0); + length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1); + length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0); + + return length; +} + +/* Create a snd_soc_cpu_dai structure + * + * This function is called by the machine driver to create a snd_soc_cpu_dai + * structure. A private data structure, with SSI hardware information, is + * appended to it. + */ +struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +{ + struct snd_soc_cpu_dai *fsl_ssi_dai; + struct fsl_ssi_private *ssi_private; + int ret = 0; + struct device_attribute *dev_attr; + + ssi_private = kzalloc(sizeof(struct fsl_ssi_private), GFP_KERNEL); + if (!ssi_private) { + dev_err(ssi_info->dev, "could not allocate DAI object\n"); + return NULL; + } + memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, + sizeof(struct snd_soc_cpu_dai)); + + fsl_ssi_dai = &ssi_private->cpu_dai; + dev_attr = &ssi_private->dev_attr; + + sprintf(ssi_private->name, "ssi%u", (u8) ssi_info->id); + ssi_private->ssi = ssi_info->ssi; + ssi_private->ssi_phys = ssi_info->ssi_phys; + ssi_private->irq = ssi_info->irq; + ssi_private->dev = ssi_info->dev; + + ssi_private->dev->driver_data = fsl_ssi_dai; + + /* Initialize the the device_attribute structure */ + dev_attr->attr.name = "ssi-stats"; + dev_attr->attr.mode = S_IRUGO; + dev_attr->show = fsl_sysfs_ssi_show; + + ret = device_create_file(ssi_private->dev, dev_attr); + if (ret) { + dev_err(ssi_info->dev, "could not create sysfs %s file\n", + ssi_private->dev_attr.attr.name); + kfree(fsl_ssi_dai); + return NULL; + } + + fsl_ssi_dai->private_data = ssi_private; + fsl_ssi_dai->name = ssi_private->name; + fsl_ssi_dai->id = ssi_info->id; + + return fsl_ssi_dai; +} +EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); + +void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +{ + struct fsl_ssi_private *ssi_private = + container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); + + device_remove_file(ssi_private->dev, &ssi_private->dev_attr); + + kfree(ssi_private); +} +EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h new file mode 100644 index 0000000..f6e33af --- /dev/null +++ b/sound/soc/fsl/fsl_ssi.h @@ -0,0 +1,224 @@ +/* + * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 SoC + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#ifndef _MPC8610_I2S_H +#define _MPC8610_I2S_H + +/* SSI Register Map */ +struct ccsr_ssi { + __be32 stx0; /* 0x.0000 - SSI Transmit Data Register 0 */ + __be32 stx1; /* 0x.0004 - SSI Transmit Data Register 1 */ + __be32 srx0; /* 0x.0008 - SSI Receive Data Register 0 */ + __be32 srx1; /* 0x.000C - SSI Receive Data Register 1 */ + __be32 scr; /* 0x.0010 - SSI Control Register */ + __be32 sisr; /* 0x.0014 - SSI Interrupt Status Register Mixed */ + __be32 sier; /* 0x.0018 - SSI Interrupt Enable Register */ + __be32 stcr; /* 0x.001C - SSI Transmit Configuration Register */ + __be32 srcr; /* 0x.0020 - SSI Receive Configuration Register */ + __be32 stccr; /* 0x.0024 - SSI Transmit Clock Control Register */ + __be32 srccr; /* 0x.0028 - SSI Receive Clock Control Register */ + __be32 sfcsr; /* 0x.002C - SSI FIFO Control/Status Register */ + __be32 str; /* 0x.0030 - SSI Test Register */ + __be32 sor; /* 0x.0034 - SSI Option Register */ + __be32 sacnt; /* 0x.0038 - SSI AC97 Control Register */ + __be32 sacadd; /* 0x.003C - SSI AC97 Command Address Register */ + __be32 sacdat; /* 0x.0040 - SSI AC97 Command Data Register */ + __be32 satag; /* 0x.0044 - SSI AC97 Tag Register */ + __be32 stmsk; /* 0x.0048 - SSI Transmit Time Slot Mask Register */ + __be32 srmsk; /* 0x.004C - SSI Receive Time Slot Mask Register */ + __be32 saccst; /* 0x.0050 - SSI AC97 Channel Status Register */ + __be32 saccen; /* 0x.0054 - SSI AC97 Channel Enable Register */ + __be32 saccdis; /* 0x.0058 - SSI AC97 Channel Disable Register */ +}; + +#define CCSR_SSI_SCR_RFR_CLK_DIS 0x00000800 +#define CCSR_SSI_SCR_TFR_CLK_DIS 0x00000400 +#define CCSR_SSI_SCR_TCH_EN 0x00000100 +#define CCSR_SSI_SCR_SYS_CLK_EN 0x00000080 +#define CCSR_SSI_SCR_I2S_MODE_MASK 0x00000060 +#define CCSR_SSI_SCR_I2S_MODE_NORMAL 0x00000000 +#define CCSR_SSI_SCR_I2S_MODE_MASTER 0x00000020 +#define CCSR_SSI_SCR_I2S_MODE_SLAVE 0x00000040 +#define CCSR_SSI_SCR_SYN 0x00000010 +#define CCSR_SSI_SCR_NET 0x00000008 +#define CCSR_SSI_SCR_RE 0x00000004 +#define CCSR_SSI_SCR_TE 0x00000002 +#define CCSR_SSI_SCR_SSIEN 0x00000001 + +#define CCSR_SSI_SISR_RFRC 0x01000000 +#define CCSR_SSI_SISR_TFRC 0x00800000 +#define CCSR_SSI_SISR_CMDAU 0x00040000 +#define CCSR_SSI_SISR_CMDDU 0x00020000 +#define CCSR_SSI_SISR_RXT 0x00010000 +#define CCSR_SSI_SISR_RDR1 0x00008000 +#define CCSR_SSI_SISR_RDR0 0x00004000 +#define CCSR_SSI_SISR_TDE1 0x00002000 +#define CCSR_SSI_SISR_TDE0 0x00001000 +#define CCSR_SSI_SISR_ROE1 0x00000800 +#define CCSR_SSI_SISR_ROE0 0x00000400 +#define CCSR_SSI_SISR_TUE1 0x00000200 +#define CCSR_SSI_SISR_TUE0 0x00000100 +#define CCSR_SSI_SISR_TFS 0x00000080 +#define CCSR_SSI_SISR_RFS 0x00000040 +#define CCSR_SSI_SISR_TLS 0x00000020 +#define CCSR_SSI_SISR_RLS 0x00000010 +#define CCSR_SSI_SISR_RFF1 0x00000008 +#define CCSR_SSI_SISR_RFF0 0x00000004 +#define CCSR_SSI_SISR_TFE1 0x00000002 +#define CCSR_SSI_SISR_TFE0 0x00000001 + +#define CCSR_SSI_SIER_RFRC_EN 0x01000000 +#define CCSR_SSI_SIER_TFRC_EN 0x00800000 +#define CCSR_SSI_SIER_RDMAE 0x00400000 +#define CCSR_SSI_SIER_RIE 0x00200000 +#define CCSR_SSI_SIER_TDMAE 0x00100000 +#define CCSR_SSI_SIER_TIE 0x00080000 +#define CCSR_SSI_SIER_CMDAU_EN 0x00040000 +#define CCSR_SSI_SIER_CMDDU_EN 0x00020000 +#define CCSR_SSI_SIER_RXT_EN 0x00010000 +#define CCSR_SSI_SIER_RDR1_EN 0x00008000 +#define CCSR_SSI_SIER_RDR0_EN 0x00004000 +#define CCSR_SSI_SIER_TDE1_EN 0x00002000 +#define CCSR_SSI_SIER_TDE0_EN 0x00001000 +#define CCSR_SSI_SIER_ROE1_EN 0x00000800 +#define CCSR_SSI_SIER_ROE0_EN 0x00000400 +#define CCSR_SSI_SIER_TUE1_EN 0x00000200 +#define CCSR_SSI_SIER_TUE0_EN 0x00000100 +#define CCSR_SSI_SIER_TFS_EN 0x00000080 +#define CCSR_SSI_SIER_RFS_EN 0x00000040 +#define CCSR_SSI_SIER_TLS_EN 0x00000020 +#define CCSR_SSI_SIER_RLS_EN 0x00000010 +#define CCSR_SSI_SIER_RFF1_EN 0x00000008 +#define CCSR_SSI_SIER_RFF0_EN 0x00000004 +#define CCSR_SSI_SIER_TFE1_EN 0x00000002 +#define CCSR_SSI_SIER_TFE0_EN 0x00000001 + +#define CCSR_SSI_STCR_TXBIT0 0x00000200 +#define CCSR_SSI_STCR_TFEN1 0x00000100 +#define CCSR_SSI_STCR_TFEN0 0x00000080 +#define CCSR_SSI_STCR_TFDIR 0x00000040 +#define CCSR_SSI_STCR_TXDIR 0x00000020 +#define CCSR_SSI_STCR_TSHFD 0x00000010 +#define CCSR_SSI_STCR_TSCKP 0x00000008 +#define CCSR_SSI_STCR_TFSI 0x00000004 +#define CCSR_SSI_STCR_TFSL 0x00000002 +#define CCSR_SSI_STCR_TEFS 0x00000001 + +#define CCSR_SSI_SRCR_RXEXT 0x00000400 +#define CCSR_SSI_SRCR_RXBIT0 0x00000200 +#define CCSR_SSI_SRCR_RFEN1 0x00000100 +#define CCSR_SSI_SRCR_RFEN0 0x00000080 +#define CCSR_SSI_SRCR_RFDIR 0x00000040 +#define CCSR_SSI_SRCR_RXDIR 0x00000020 +#define CCSR_SSI_SRCR_RSHFD 0x00000010 +#define CCSR_SSI_SRCR_RSCKP 0x00000008 +#define CCSR_SSI_SRCR_RFSI 0x00000004 +#define CCSR_SSI_SRCR_RFSL 0x00000002 +#define CCSR_SSI_SRCR_REFS 0x00000001 + +/* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR 0x00020000 +#define CCSR_SSI_SxCCR_WL_SHIFT 13 +#define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 +#define CCSR_SSI_SxCCR_WL(x) \ + (((((x) / 2) - 1) << CCSR_SSI_SxCCR_WL_SHIFT) & CCSR_SSI_SxCCR_WL_MASK) +#define CCSR_SSI_SxCCR_DC_SHIFT 8 +#define CCSR_SSI_SxCCR_DC_MASK 0x00001F00 +#define CCSR_SSI_SxCCR_DC(x) \ + ((((x) - 1) << CCSR_SSI_SxCCR_DC_SHIFT) & CCSR_SSI_SxCCR_DC_MASK) +#define CCSR_SSI_SxCCR_PM_SHIFT 0 +#define CCSR_SSI_SxCCR_PM_MASK 0x000000FF +#define CCSR_SSI_SxCCR_PM(x) \ + ((((x) - 1) << CCSR_SSI_SxCCR_PM_SHIFT) & CCSR_SSI_SxCCR_PM_MASK) + +/* + * The xFCNT bits are read-only, and the xFWM bits are read/write. Use the + * CCSR_SSI_SFCSR_xFCNTy() macros to read the FIFO counters, and use the + * CCSR_SSI_SFCSR_xFWMy() macros to set the watermarks. + */ +#define CCSR_SSI_SFCSR_RFCNT1_SHIFT 28 +#define CCSR_SSI_SFCSR_RFCNT1_MASK 0xF0000000 +#define CCSR_SSI_SFCSR_RFCNT1(x) \ + (((x) & CCSR_SSI_SFCSR_RFCNT1_MASK) >> CCSR_SSI_SFCSR_RFCNT1_SHIFT) +#define CCSR_SSI_SFCSR_TFCNT1_SHIFT 24 +#define CCSR_SSI_SFCSR_TFCNT1_MASK 0x0F000000 +#define CCSR_SSI_SFCSR_TFCNT1(x) \ + (((x) & CCSR_SSI_SFCSR_TFCNT1_MASK) >> CCSR_SSI_SFCSR_TFCNT1_SHIFT) +#define CCSR_SSI_SFCSR_RFWM1_SHIFT 20 +#define CCSR_SSI_SFCSR_RFWM1_MASK 0x00F00000 +#define CCSR_SSI_SFCSR_RFWM1(x) \ + (((x) << CCSR_SSI_SFCSR_RFWM1_SHIFT) & CCSR_SSI_SFCSR_RFWM1_MASK) +#define CCSR_SSI_SFCSR_TFWM1_SHIFT 16 +#define CCSR_SSI_SFCSR_TFWM1_MASK 0x000F0000 +#define CCSR_SSI_SFCSR_TFWM1(x) \ + (((x) << CCSR_SSI_SFCSR_TFWM1_SHIFT) & CCSR_SSI_SFCSR_TFWM1_MASK) +#define CCSR_SSI_SFCSR_RFCNT0_SHIFT 12 +#define CCSR_SSI_SFCSR_RFCNT0_MASK 0x0000F000 +#define CCSR_SSI_SFCSR_RFCNT0(x) \ + (((x) & CCSR_SSI_SFCSR_RFCNT0_MASK) >> CCSR_SSI_SFCSR_RFCNT0_SHIFT) +#define CCSR_SSI_SFCSR_TFCNT0_SHIFT 8 +#define CCSR_SSI_SFCSR_TFCNT0_MASK 0x00000F00 +#define CCSR_SSI_SFCSR_TFCNT0(x) \ + (((x) & CCSR_SSI_SFCSR_TFCNT0_MASK) >> CCSR_SSI_SFCSR_TFCNT0_SHIFT) +#define CCSR_SSI_SFCSR_RFWM0_SHIFT 4 +#define CCSR_SSI_SFCSR_RFWM0_MASK 0x000000F0 +#define CCSR_SSI_SFCSR_RFWM0(x) \ + (((x) << CCSR_SSI_SFCSR_RFWM0_SHIFT) & CCSR_SSI_SFCSR_RFWM0_MASK) +#define CCSR_SSI_SFCSR_TFWM0_SHIFT 0 +#define CCSR_SSI_SFCSR_TFWM0_MASK 0x0000000F +#define CCSR_SSI_SFCSR_TFWM0(x) \ + (((x) << CCSR_SSI_SFCSR_TFWM0_SHIFT) & CCSR_SSI_SFCSR_TFWM0_MASK) + +#define CCSR_SSI_STR_TEST 0x00008000 +#define CCSR_SSI_STR_RCK2TCK 0x00004000 +#define CCSR_SSI_STR_RFS2TFS 0x00002000 +#define CCSR_SSI_STR_RXSTATE(x) (((x) >> 8) & 0x1F) +#define CCSR_SSI_STR_TXD2RXD 0x00000080 +#define CCSR_SSI_STR_TCK2RCK 0x00000040 +#define CCSR_SSI_STR_TFS2RFS 0x00000020 +#define CCSR_SSI_STR_TXSTATE(x) ((x) & 0x1F) + +#define CCSR_SSI_SOR_CLKOFF 0x00000040 +#define CCSR_SSI_SOR_RX_CLR 0x00000020 +#define CCSR_SSI_SOR_TX_CLR 0x00000010 +#define CCSR_SSI_SOR_INIT 0x00000008 +#define CCSR_SSI_SOR_WAIT_SHIFT 1 +#define CCSR_SSI_SOR_WAIT_MASK 0x00000006 +#define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT) +#define CCSR_SSI_SOR_SYNRST 0x00000001 + +/* Instantiation data for an SSI interface + * + * This structure contains all the information that the the SSI driver needs + * to instantiate an SSI interface with ALSA. The machine driver should + * create this structure, fill it in, call fsl_ssi_create_dai(), and then + * delete the structure. + * + * id: which SSI this is (0, 1, etc. ) + * ssi: pointer to the SSI's registers + * ssi_phys: physical address of the SSI registers + * irq: IRQ of this SSI + * dev: struct device, used to create the sysfs statistics file +*/ +struct fsl_ssi_info { + unsigned int id; + struct ccsr_ssi __iomem *ssi; + dma_addr_t ssi_phys; + unsigned int irq; + struct device *dev; +}; + +struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); +void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); + +#endif + diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c new file mode 100644 index 0000000..5421a5c --- /dev/null +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -0,0 +1,621 @@ +/** + * Freescale MPC8610HPCD ALSA SoC Fabric driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <asm/of_device.h> +#include <asm/of_platform.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <asm/immap_86xx.h> + +#include "../codecs/cs4270.h" +#include "fsl_dma.h" +#include "fsl_ssi.h" + +/* + * Fabric-specific ASoC device data + * + * This structure contains data for a single sound platform device on an + * MPC8610 HPCD. Some of the data is taken from the device tree. + */ +struct mpc8610hpcd_data { + struct snd_soc_device sound_devdata; + struct snd_soc_dai_link dai; + struct snd_soc_machine machine; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + struct ccsr_guts __iomem *guts; + struct ccsr_ssi __iomem *ssi; + unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ + unsigned int ssi_irq; + unsigned int dma_id; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_irq[2]; + struct ccsr_dma_channel __iomem *dma[2]; + unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ +}; + +/* + * Initalize the board + * + * This function is called when platform_device_add() is called. It is used + * to initialize the board-specific hardware. + * + * Here we program the DMACR and PMUXCR registers. + */ +int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) +{ + struct mpc8610hpcd_data *machine_data = sound_device->dev.platform_data; + + /* Program the signal routing between the SSI and the DMA */ + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); + + guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, + machine_data->dma_channel_id[0], 0); + guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, + machine_data->dma_channel_id[1], 0); + + guts_set_pmuxcr_dma(machine_data->guts, 1, 0, 0); + guts_set_pmuxcr_dma(machine_data->guts, 1, 3, 0); + guts_set_pmuxcr_dma(machine_data->guts, 0, 3, 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); + break; + case 1: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); + break; + } + + return 0; +} + +/* + * Program the board based on various hardware parameters + */ +static int mpc8610hpcd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct mpc8610hpcd_data *machine_data = rtd->socdev->dev->platform_data; + int ret = 0; + + /* Tell the CPU driver what the serial protocol is. */ + if (cpu_dai->dai_ops.set_fmt) { + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; + } + } + + /* Tell the codec driver what the serial protocol is. */ + if (codec_dai->dai_ops.set_fmt) { + ret = codec_dai->dai_ops.set_fmt(codec_dai, + machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; + } + } + + /* + * Tell the CPU driver what the clock frequency is, and whether it's a + * slave or master. + */ + if (cpu_dai->dai_ops.set_sysclk) { + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0, + machine_data->clk_frequency, + machine_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; + } + } + + /* + * Tell the codec driver what the MCLK frequency is, and whether it's + * a slave or master. + */ + if (codec_dai->dai_ops.set_sysclk) { + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; + } + } + + return 0; +} + +/* + * Remove the sound device + * + * This function is called to remove the sound device for one SSI. We + * de-program the DMACR and PMUXCR register. + */ +int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) +{ + struct mpc8610hpcd_data *machine_data = sound_device->dev.platform_data; + + /* Restore the signal routing */ + + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[0], 0); + guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + machine_data->dma_channel_id[1], 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + case 1: + clrsetbits_be32(&machine_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + } + + return 0; +} + +/* + * ASoC fabric driver operations + */ +static struct snd_soc_ops mpc8610hpcd_ops = { + .startup = mpc8610hpcd_startup, +}; + +/* + * ASoC machine + */ +static struct snd_soc_machine mpc8610_hpcd_machine = { + .probe = mpc8610_hpcd_machine_probe, + .remove = mpc8610_hpcd_machine_remove, + .name = "MPC8610 HPCD", + .num_links = 1, +}; + +/* + * Probe function for the fabric driver driver. + * + * This function gets called when an SSI node is found in the device tree. + * + * Although this is a fabric driver, the SSI node is the "master" node with + * respect to audio hardware connections. Therefore, we create a new ASoC + * device for each new SSI node that has a codec attached. + * + * FIXME: Currently, we only support one DMA controller, so if there are + * multiple SSI nodes with codecs, only the first will be supported. + * + * FIXME: Even if we did support multiple DMA controllers, we have no + * mechanism for assigning DMA controllers and channels to the individual + * SSI devices. We also probably aren't compatible with the generic Elo DMA + * device driver. + */ +static int mpc8610_hpcd_probe(struct of_device *ofdev, + const struct of_device_id *match) +{ + struct device_node *np = ofdev->node; + struct device_node *codec_np = NULL; + struct device_node *guts_np = NULL; + struct device_node *dma_np = NULL; + struct device_node *dma_channel_np = NULL; + const char *sprop; + const u32 *iprop; + struct resource res; + struct platform_device *sound_device = NULL; + struct mpc8610hpcd_data *machine_data; + struct fsl_ssi_info ssi_info; + struct fsl_dma_info dma_info; + int ret = -ENODEV; + + machine_data = kzalloc(sizeof(struct mpc8610hpcd_data), GFP_KERNEL); + if (!machine_data) + return -ENOMEM; + + memset(&ssi_info, 0, sizeof(ssi_info)); + memset(&dma_info, 0, sizeof(dma_info)); + + ssi_info.dev = &ofdev->dev; + + /* + * We are only interested in SSIs with a codec child node in them, so + * let's make sure this SSI has one. + */ + while ((codec_np = of_get_next_child(np, codec_np)) != NULL) { + if (strcmp(codec_np->name, "codec") == 0) { + /* Most drivers forget the final of_node_put() call */ + of_node_put(codec_np); + break; + } + } + + if (!codec_np) + goto error; + + /* The MPC8610 HPCD only knows about the CS4270 codec, so reject + anything else. */ + if (!of_device_is_compatible(codec_np, "cirrus,cs4270")) + goto error; + + /* Get the device ID */ + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) { + dev_err(&ofdev->dev, "cell-index property not found\n"); + ret = -EINVAL; + goto error; + } + machine_data->ssi_id = *iprop; + ssi_info.id = *iprop; + + /* Get the serial format and clock direction. */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop) { + dev_err(&ofdev->dev, "fsl,mode property not found\n"); + ret = -EINVAL; + goto error; + } + + if (strcasecmp(sprop, "i2s-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + + /* In i2s-slave mode, the codec has its own clock source, so + we need to get the frequency from the device tree and pass + it to the codec driver. */ + iprop = of_get_property(codec_np, "bus-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&ofdev->dev, "codec bus-frequency property " + "is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + machine_data->clk_frequency = *iprop; + } else if (strcasecmp(sprop, "i2s-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "lj-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "lj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "rj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "rj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "ac97-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "ac97-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else { + dev_err(&ofdev->dev, + "unrecognized fsl,mode property \"%s\"\n", sprop); + ret = -EINVAL; + goto error; + } + + if (!machine_data->clk_frequency) { + dev_err(&ofdev->dev, "unknown clock frequency\n"); + ret = -EINVAL; + goto error; + } + + /* Read the SSI information from the device tree */ + ret = of_address_to_resource(np, 0, &res); + if (ret) { + dev_err(&ofdev->dev, "could not obtain SSI address\n"); + goto error; + } + if (!res.start) { + dev_err(&ofdev->dev, "invalid SSI address\n"); + goto error; + } + ssi_info.ssi_phys = res.start; + + machine_data->ssi = ioremap(ssi_info.ssi_phys, sizeof(struct ccsr_ssi)); + if (!machine_data->ssi) { + dev_err(&ofdev->dev, "could not map SSI address %x\n", + ssi_info.ssi_phys); + ret = -EINVAL; + goto error; + } + ssi_info.ssi = machine_data->ssi; + + + /* Get the IRQ of the SSI */ + machine_data->ssi_irq = irq_of_parse_and_map(np, 0); + if (!machine_data->ssi_irq) { + dev_err(&ofdev->dev, "could not get SSI IRQ\n"); + ret = -EINVAL; + goto error; + } + ssi_info.irq = machine_data->ssi_irq; + + + /* Map the global utilities registers. */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); + if (!guts_np) { + dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); + ret = -EINVAL; + goto error; + } + machine_data->guts = of_iomap(guts_np, 0); + of_node_put(guts_np); + if (!machine_data->guts) { + dev_err(&ofdev->dev, "could not map GUTS\n"); + ret = -EINVAL; + goto error; + } + + /* Find the DMA channels to use. For now, we always use the first DMA + controller. */ + for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") { + iprop = of_get_property(dma_np, "cell-index", NULL); + if (iprop && (*iprop == 0)) { + of_node_put(dma_np); + break; + } + } + if (!dma_np) { + dev_err(&ofdev->dev, "could not find DMA node\n"); + ret = -EINVAL; + goto error; + } + machine_data->dma_id = *iprop; + + /* + * Find the DMA channels to use. For now, we always use DMA channel 0 + * for playback, and DMA channel 1 for capture. + */ + while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) { + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + /* Is it DMA channel 0? */ + if (iprop && (*iprop == 0)) { + /* dma_channel[0] and dma_irq[0] are for playback */ + dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0); + dma_info.dma_irq[0] = + irq_of_parse_and_map(dma_channel_np, 0); + machine_data->dma_channel_id[0] = *iprop; + continue; + } + if (iprop && (*iprop == 1)) { + /* dma_channel[1] and dma_irq[1] are for capture */ + dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0); + dma_info.dma_irq[1] = + irq_of_parse_and_map(dma_channel_np, 0); + machine_data->dma_channel_id[1] = *iprop; + continue; + } + } + if (!dma_info.dma_channel[0] || !dma_info.dma_channel[1] || + !dma_info.dma_irq[0] || !dma_info.dma_irq[1]) { + dev_err(&ofdev->dev, "could not find DMA channels\n"); + ret = -EINVAL; + goto error; + } + + dma_info.ssi_stx_phys = ssi_info.ssi_phys + + offsetof(struct ccsr_ssi, stx0); + dma_info.ssi_srx_phys = ssi_info.ssi_phys + + offsetof(struct ccsr_ssi, srx0); + + /* We have the DMA information, so tell the DMA driver what it is */ + if (!fsl_dma_configure(&dma_info)) { + dev_err(&ofdev->dev, "could not instantiate DMA device\n"); + ret = -EBUSY; + goto error; + } + + /* + * Initialize our DAI data structure. We should probably get this + * information from the device tree. + */ + machine_data->dai.name = "CS4270"; + machine_data->dai.stream_name = "CS4270"; + + machine_data->dai.cpu_dai = fsl_ssi_create_dai(&ssi_info); + machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */ + machine_data->dai.ops = &mpc8610hpcd_ops; + + mpc8610_hpcd_machine.dai_link = &machine_data->dai; + + /* Allocate a new audio platform device structure */ + sound_device = platform_device_alloc("soc-audio", -1); + if (!sound_device) { + dev_err(&ofdev->dev, "platform device allocation failed\n"); + ret = -ENOMEM; + goto error; + } + + machine_data->sound_devdata.machine = &mpc8610_hpcd_machine; + machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; + machine_data->sound_devdata.platform = &fsl_soc_platform; + + sound_device->dev.platform_data = machine_data; + + + /* Set the platform device and ASoC device to point to each other */ + platform_set_drvdata(sound_device, &machine_data->sound_devdata); + + machine_data->sound_devdata.dev = &sound_device->dev; + + + /* Tell ASoC to probe us. This will call mpc8610_hpcd_machine.probe(), + if it exists. */ + ret = platform_device_add(sound_device); + + if (ret) { + dev_err(&ofdev->dev, "platform device add failed\n"); + goto error; + } + + dev_set_drvdata(&ofdev->dev, sound_device); + + return 0; + +error: + if (sound_device) + platform_device_unregister(sound_device); + + if (machine_data->dai.cpu_dai) + fsl_ssi_destroy_dai(machine_data->dai.cpu_dai); + + if (ssi_info.ssi) + iounmap(ssi_info.ssi); + + if (ssi_info.irq) + irq_dispose_mapping(ssi_info.irq); + + if (dma_info.dma_channel[0]) + iounmap(dma_info.dma_channel[0]); + + if (dma_info.dma_channel[1]) + iounmap(dma_info.dma_channel[1]); + + if (dma_info.dma_irq[0]) + irq_dispose_mapping(dma_info.dma_irq[0]); + + if (dma_info.dma_irq[1]) + irq_dispose_mapping(dma_info.dma_irq[1]); + + if (machine_data->guts) + iounmap(machine_data->guts); + + kfree(machine_data); + + return ret; +} + +static int mpc8610_hpcd_remove(struct of_device *ofdev) +{ + struct platform_device *sound_device = dev_get_drvdata(&ofdev->dev); + struct mpc8610hpcd_data *machine_data = sound_device->dev.platform_data; + + platform_device_unregister(sound_device); + + if (machine_data->dai.cpu_dai) + fsl_ssi_destroy_dai(machine_data->dai.cpu_dai); + + if (machine_data->ssi) + iounmap(machine_data->ssi); + + if (machine_data->dma[0]) + iounmap(machine_data->dma[0]); + + if (machine_data->dma[1]) + iounmap(machine_data->dma[1]); + + if (machine_data->dma_irq[0]) + irq_dispose_mapping(machine_data->dma_irq[0]); + + if (machine_data->dma_irq[1]) + irq_dispose_mapping(machine_data->dma_irq[1]); + + if (machine_data->guts) + iounmap(machine_data->guts); + + kfree(machine_data); + sound_device->dev.platform_data = NULL; + + dev_set_drvdata(&ofdev->dev, NULL); + + return 0; +} + +static struct of_device_id mpc8610_hpcd_match[] = { + { + .compatible = "fsl,ssi", + }, + {} +}; +MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); + +static struct of_platform_driver mpc8610_hpcd_of_driver = { + .owner = THIS_MODULE, + .name = "mpc8610_hpcd", + .match_table = mpc8610_hpcd_match, + .probe = mpc8610_hpcd_probe, + .remove = mpc8610_hpcd_remove, +}; + +/** + * Fabric driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init mpc8610hpcd_init(void) +{ + int ret; + + printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); + + ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); + + if (ret) + printk(KERN_ERR + "mpc8610-hpcd: failed to register platform driver\n"); + + return ret; +} + +/** + * Fabric driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit mpc8610hpcd_exit(void) +{ + of_unregister_platform_driver(&mpc8610_hpcd_of_driver); +} + +module_init(mpc8610hpcd_init); +module_exit(mpc8610hpcd_exit); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); +MODULE_LICENSE("GPL"); -- 1.5.2.4

12 88

[alsa-devel] [RFC] alsa-kernel clean-ups
by Takashi Iwai 08 Jan '08

08 Jan '08

Hi, ALSA kernel tree has lots of unused and rotten codes. I'd like to clean up this. Here we go... 1. Indirect control element access This is what I already suggested. This hasn't been used (we have no proper API in alsa-lib), and 32/64bit wrapper doesn't work at all. Also, the idea to copy such a big data area via ioctl is bad. If we need a big matrix mixer, let make each matrix element accessible individually. Copying the whole matrix at each time isn't efficient. 2. PCM xfer_align parameter This sw_params parameter has never been used in a sane manner, and no one understands what this does exactly. The current implementation looks also buggy because it allows write of shorter size than xfer_align. So, if you do partial writes, the write isn't actually aligned at all. Removing this parameter will make some pcm_lib_* code more readable (and less buggy). 3. PCM sleep_min and tick The "tick" in PCM is set (again) via sw_params. And, nobody uses this feature at all except for a command line option of aplay. (This is literally "nobody", as I checked alsa-lib API calls in all programs in some major distros.) Above all, if we need finer wake-ups for the position update, it's basically an issue that the driver should solve, not tuned by each application. 4. sound/driver.h This header file exists only for some hacks to adapt alsa-driver tree. It's useless for building in the kernel. Let's move a few lines in it to sound/core.h and remove it. For building the modules on alsa-driver external tree, we can simply add #include "adriver.h" to each build stub (alsa-driver/*/*.c) before inclusion of alsa-kernel codes. I'll post some patches to clean the above thing up in the following posts. Please speak up if you have objections. Takashi

3 16

[alsa-devel] snd_pcm_status_get_tstamp()
by Lennart Poettering 07 Jan '08

07 Jan '08

Hi! The documentation for snd_pcm_status_get_tstamp() is not really enlightening. After a call to snd_pcm_status() what data does the tstamp field actually carry? Some kind of system clock timestamp when the data in the status struct was valid, or some kind of sound card clock timestamp that is relative to the point in time where the device was opened/started? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

3 4

[alsa-devel] Status of snd_pcm_rewind()
by Lennart Poettering 07 Jan '08

07 Jan '08

Hi! What's the status of snd_pcm_rewind()? Is it available on all devices? I assume not (at least dmix doesn't...). But I guess I can safely assume it is available when mmap is used? Is there any API function to test if it is available? I am working on adding "glitch-free" support to PulseAudio. I.e. have a huge playback buffer (> 2s), and schedule playback based on system timers instead of sound card interrupts, thus allowing us to dynamically adjust the inetrrupt frequency to the clients that are connected, and thus save power. I can enable this technique only if I can rewrite the playback buffer whenever I need to, thus I need to in advance if snd_pcm_rewind() will work or not. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 3

[alsa-devel] Conexant CX2051 on HP Spartan
by Claudio Matsuoka 07 Jan '08

07 Jan '08

Hi, I'm working with Herton Krzesinski on a HP Spartan laptop which has a Conexant CX2051 configured with internal and external microphones connected to two different ADC nodes (0x14 and 0x15 in http://helllabs.org/tmp/hp-spartan-ng.png). We were able to get input from one or other ADC, but we weren't able to switch from one capture source to another (switching between 0x17 and 0x1d would be simple, but HP chose to route the external and internal mics to two different ADCs). What is the recommended way to handle this kind of setup?

2 1

[alsa-devel] [PATCH] Another EMU0404 Board ID
by Veli-Matti Valtonen 07 Jan '08

07 Jan '08

Hey, This is based on pseudo-random playing around with the capabilities. With ca0102 this card gives no output atall, ca0108 appears to work fine, so it rather looks similar to the EMU1010b/EMU1010 changes. Some other people seem to have succeeded in using this aswell: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3496 diff -u -r alsa-driver-hg20071222/alsa-kernel/pci/emu10k1/emu10k1_main.c alsa-driver-hg20071222-0404b/alsa-kernel/pci/emu10k1/emu10k1_main.c --- alsa-driver-hg20071222/alsa-kernel/pci/emu10k1/emu10k1_main.c 2007-12-22 03:00:09.000000000 +0200 +++ alsa-driver-hg20071222-0404b/alsa-kernel/pci/emu10k1/emu10k1_main.c 2007-12-23 23:01:12.807047391 +0200 @@ -1419,6 +1419,14 @@ .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */ + /* EMU0404b */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, + .driver = "Audigy2", .name = "E-mu 0404b [4002]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ /* Tested by James(a)superbug.co.uk 20-3-2007. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, .driver = "Audigy2", .name = "E-mu 0404 [4002]",

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[alsa-devel] es18xx: Enable wavetable input from ESS chips
by Krzysztof Helt 07 Jan '08

07 Jan '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch enables wavetable chips ES689/ES69X connected to ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal from the wavetable is active. It has no effect if there is no ESS wavetable chip present. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch has a gotcha. In theory, the OPL synthesis or wavetable input should use FM DAC depending on the active clock signal from the wavetable chip. Unfortunately, the default state of the wavetable chip from ESS (ES689/ES690/ES692) is to send clock signal all the time. This blocks OPL3 signal if the wavetable chip is present. This problem can be solved with sysex command 4 send to the wavetable chip. It sets the wavetable chip to auto-deactivate mode which switches the wavetable off if there are 5 seconds without midi input (according to the es1879 datasheet). I tested this patch on the es1868 chip without the wavetable. The OPL3 synthesis worked (no clock signal from the wavetable). I tested it on the card with es1868 chip and es690 wavetable. The OPL3 synthesis did not work, the wavetable worked. Both the OPL3 and wavetable worked if the sysex command 4 was sent to the wavetable chip (e.g. by amidi tool). Is it ok to add this sysex command sending to the driver? How? The wavetable clock signal can be detected (there is status bit which says if wavetable is active). Another question is about ES938 chip (bass, treble and 3D control) on the card. This chip is controlled by sysex midi commands. It can be detected (by reading settings with sysex commands). Is it possible to add this to the alsamixer or external tool for setting these controls is recommended? Regards, Krzysztof diff -urp linux-2.6.24/sound/isa/es18xx.c linux-new/sound/isa/es18xx.c --- linux-2.6.24/sound/isa/es18xx.c 2007-12-25 23:23:03.000000000 +0100 +++ linux-new/sound/isa/es18xx.c 2007-12-26 16:30:08.000000000 +0100 @@ -1442,6 +1442,8 @@ static int __devinit snd_es18xx_initiali snd_es18xx_write(chip, 0xB2, 0x50); /* Enable MPU and hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x42); + /* Enable ESS wavetable input */ + snd_es18xx_mixer_bits(chip, 0x48, 0x10, 0x10); } else { int irqmask, dma1mask, dma2mask; ---------------------------------------------------------------------- Rozdajemy nagrody! Wygraj samochod! Sprawdz >> http://link.interia.pl/f1cb8

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[alsa-devel] [PATCH] [ALSA-KERNEL] Add IEC958 digital out support to AD1984 for Lenovo Thinkpads T61/X61
by Jerone Young 07 Jan '08

07 Jan '08

This patch adds IEC958 digital out support for the AD1984 sound card. This card can be found in Lenovo Thinkapds T61/X61. The digital out is not located on the Thinkpad, but optional docking station (it's coxial digital out). I've add this support as it is done the exact same way for the AD1983 & AD1884. I have tested this patch with my Lenovo Thinkpad T61 hooked up to a docking station (that has the digital coxial) and then run to my home theater reciever. Works like a charm :-) Patch is attached and also in this email. Signed-off-by: Jerone Young <jerone(a)gmail.com> diff -r 1227a1c12325 pci/hda/patch_analog.c --- a/pci/hda/patch_analog.c Mon Dec 24 14:40:56 2007 +0100 +++ b/pci/hda/patch_analog.c Thu Dec 27 17:41:13 2007 -0600 @@ -3095,6 +3095,16 @@ static struct snd_kcontrol_new ad1984_th .get = ad198x_mux_enum_get, .put = ad198x_mux_enum_put, }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, { } /* end */ }; @@ -3197,7 +3207,7 @@ static int patch_ad1984(struct hda_codec codec->patch_ops.build_pcms = ad1984_build_pcms; break; case AD1984_THINKPAD: - spec->multiout.dig_out_nid = 0; + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; spec->input_mux = &ad1984_thinkpad_capture_source; spec->mixers[0] = ad1984_thinkpad_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;

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[alsa-devel] ICE1724 chipset and envy24control
by Norval Watson 07 Jan '08

07 Jan '08

Hi, I just got an M-Audio Audiophile 192 soundcard with ICE1724 chipset. Works out of the box on debian sid. Is there any way to control this card with envy24control utility from alsa-tools? envy24control is written for ICE1712 chipset. Is there any project to develop control utility for ICE1724 card? Seasons greetings, Norval Watson Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

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[alsa-devel] function snd_mixer_load failed: Inappropriate ioctl for device
by Salatiel Filho 07 Jan '08

07 Jan '08

*** please let me in the to or cc fields. I am not on the list *** I am having a big trouble to make sound work on my ARM machine. I am running debian 4. # uname -a Linux LS-GL7D6 2.6.16.57-arm2 #7 Wed Dec 12 18:14:51 BRT 2007 armv5tejl GNU/Linux # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Dec 12 2007 for kernel 2.6.16.57-arm2. Modules load successfully # lsmod Module Size Used by snd_pcm_oss 47680 0 snd_mixer_oss 15040 1 snd_pcm_oss snd_usb_audio 79904 0 snd_pcm 75877 2 snd_pcm_oss,snd_usb_audio snd_timer 20164 1 snd_pcm snd_page_alloc 8264 1 snd_pcm snd_usb_lib 15808 1 snd_usb_audio snd_rawmidi 21376 1 snd_usb_lib snd_seq_device 7116 1 snd_rawmidi snd_hwdep 7460 1 snd_usb_audio snd 54368 9 snd_pcm_oss,snd_mixer_oss,snd_usb_audio,snd_pcm,snd_timer,snd_usb_lib,snd_rawmidi,snd_seq_device,snd_hwdep soundcore 7684 1 snd nfs 102044 0 lockd 61780 1 nfs sunrpc 142808 3 nfs,lockd dm_mod 48176 0 usbhid 38628 0 # cat /proc/asound/cards 0 [default ]: USB-Audio - C-Media USB Headphone Set C-Media USB Headphone Set at usb-ehci_platform.4523-1, full speed # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Dec 12 2007 for kernel 2.6.16.57-arm2. but everytime i try to use alsamixer i get: # alsamixer alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device # aplay test.wav Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: xrun:1090: read/write error, state = PREPARED # strace alsamixer ... stat64("/usr//share/alsa/alsa.conf", {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 open("/usr//share/alsa/alsa.conf", O_RDONLY) = 3 fstat64(3, {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40017000 read(3, "#\n# ALSA library configuration "..., 4096) = 4096 read(3, "r\n\t\t\t\tname defaults.pcm.device\n\t"..., 4096) = 4096 read(3, "{\n\t\t\t@func refer\n\t\t\tname default"..., 4096) = 419 read(3, "", 4096) = 0 read(3, "", 4096) = 0 close(3) = 0 munmap(0x40017000, 4096) = 0 access("/etc/asound.conf", R_OK) = -1 ENOENT (No such file or directory) access("/root/.asoundrc", R_OK) = -1 ENOENT (No such file or directory) open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbec2da64) = 0 ioctl(3, UI_DEV_CREATE, 0xbec2db2c) = 0 close(3) = 0 stat64("/usr//share/alsa/alsa.conf", {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbec2da44) = 0 fcntl64(3, F_GETFL) = 0x2 (flags O_RDWR) fcntl64(3, F_SETFL, O_RDWR|O_NONBLOCK) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0xbec2dac0) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0xbec2dac0) = 0 ioctl(3, USBDEVFS_CONNECTINFO, 0xbec2d86c) = 0 ioctl(3, USBDEVFS_IOCTL, 0xbec2d42c) = -1 ENOTTY (Inappropriate ioctl for device) fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 2), ...}) = 0 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40017000 write(1, "\n", 1 ) = 1 write(2, "alsamixer: function snd_mixer_lo"..., 74alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device ) = 74 munmap(0x40017000, 4096) = 0 io_submit(0x2, 0, 0x8 <unfinished ... exit status 2> any help? The same card works ok on my x86 machine using . So the problem is not the card but something else. -- []'s Salatiel "O maior prazer do inteligente é bancar o idiota diante de um idiota que banca o inteligente".

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by Takashi Iwai 02 Jan '08

02 Jan '08

At Thu, 20 Dec 2007 10:13:29 -0800, John Stile wrote: > > On Thu, 2007-12-20 at 12:09 +0100, Takashi Iwai wrote: > > At Thu, 20 Dec 2007 00:37:46 +0000, > > forumjunk(a)comcast.net wrote: > > > > > > > > > For VIA platforms, to avoid the driver compile error in pci.h, reconfigure your kernel to remove SMP support, and set the architecture type to PC-compatible. > > > > > > (you already know this, I'm sure): > > > Recompile the kernel (as a paranoid, I always trash /lib/modules/<kernel_version> before making modules_install). Install the kernel, of course, and reconfigure your boot loader and/or initrd and reboot. > > > > > > There are still several fixes that need to be made to get all of this to compile cleanly, but I haven't got the permanent fixes nailed down yet. I'm almost positive that these are alsa-driver configure errors or omissions. I can't (yet) find a kernel config that will avoid them. The fixes below are harmless, and they will allow you to get the driver compiled. > > > > My guess is that the error of configure script is the only point we > > need to fix. > > > > > Add this definition to the top section of acore/memory_wrapper.c after the line that reads #include <linux/version.h> > > > > > > #define VMALLOC_VMADDR(x) ((unsigned long)(x)) > > > > > > Change these lines (45 and 46 or close to that) of acore/memory_wrapper.c from: > > > > > > pmd = pmd_offset(pgd, lpage); > > > pte = pte_offset(pmd, lpage); > > > > > > to > > > > > > pmd = pmd_offset((pud_t *)pgd, lpage); > > > pte = pte_offset_kernel(pmd, lpage); > > > > > > The driver will then compile cleanly (at least with no errors or warnings relating to any of these changes) > > > > Don't do this. This part shouldn't be compiled at all for recent > > kernels. This is, again, the problem of configure script that > > detected the codes wrongly. > > > > > > So, could you guys show config.log? Otherwise it's hard to know > > what's really wrong. > > > > > > thanks, > I recompiled the kernel without SMP support, and as PC-compatible (per > forumjunk(a)comcast.net). In config.log, vmalloc_to_page shows a warning > starting at line 540. "function cpuid_count causes a warning: pointer > targets in passing argument 4 of 'native_cpuid' differ in signedness". > --------------------------------------- > The config.log Thanks. Does the patch below fix the probelm? Run hgcompile after applying it. Takashi diff -r 414b1156a8b3 configure.in --- a/configure.in Tue Dec 18 16:09:20 2007 +0100 +++ b/configure.in Fri Dec 21 12:10:53 2007 +0100 @@ -1230,7 +1230,7 @@ esac dnl set ia32 (X86) case "$processor" in - i?86*|k?|crusoe|mcyrixiii|mwinchip|viac3_2|mcore2|geodegx*) + i?86*|k?|crusoe|mcyrixiii|mwinchip|viac3_2|mcore2|geodegx*|x86_32) if $KCC -mpreferred-stack-boundary=2 -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then c_opts="-mpreferred-stack-boundary=2 $c_opts" CONFIG_X86=y

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[alsa-devel] Getting the AICA device to work properly
by Adrian McMenamin 31 Dec '07

31 Dec '07

I have built mpg123 for SH4 (using uclibc for a buildroot/busybox environment): ./configure --host=sh4-linux CC="/home/adrian/buildroot/build_sh4/staging_dir/bin/sh4-linux-uclibc-gcc" --with-cpu=generic_fpu --prefix=/tftpboot/192.168.61.55/usr LDFLAGS="-L/tftpboot/192.168.61.55/lib" --oldincludedir=/tftpboot/192.168.61.55/usr/include -with-audio=alsa --sysconfdir=/tftpboot/192.168.61.55/etc --enable-modules=no --enable-static --enable-debug When I try to play sound I get this: $ mpg123 -v ms.mp3 [getlopt.c:115] debug: getsingleopt: -v [getlopt.c:115] debug: getsingleopt: ms.mp3 [audio.c:501] debug: opening normal audio/file [libmpg123.c:190] debug: set flags to 0x40 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3 version 1.0.1; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes [frame.c:464] debug: frame_gapless_init: from 0 to 0 samples [libmpg123.c:99] debug: cpu opt setting Decoder: generic [frame.c:141] debug: frame 0x435650 buffer [frame.c:242] debug: frame 0x435650 buffer done [tabinit.c:85] debug: MMX decode tables [tabinit.c:107] debug: MMX decode tables done [audio.c:279] debug: audio_capabilities ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' [mpg123.c:813] debug: Going to play ms.mp3 [mpg123.c:483] debug: OK... going to finally open. [frame.c:464] debug: frame_gapless_init: from 0 to 0 samples [readers.c:620] error: Cannot file ms.mp3: No such file or directory [mpg123.c:495] error: Cannot open ms.mp3: File access error. (code 22) [playlist.c:116] debug: going to free() the playlist [playlist.c:120] debug: free()ing entry 0 [playlist.c:126] debug: free()d the playlist [audio.c:441] debug: exit output [audio.c:524] debug: closing output [audio.c:148] debug: closing output module The main problem appears to be that ALSA lib has no knowledge of the installed (snd-aica) sound module - eg $ aplay -L null Discard all samples (playback) or generate zero samples (capture) How do I fix this? I don't think I had this problem before because my alsalib set up was somewhat banjaxed - things like aplay segfaulted - but the oss smulation layer worked well. When I use the emulation layer now (rebuilding mpg123) I now get: $ mpg123 ./ms.mp3 [getlopt.c:115] debug: getsingleopt: ./ms.mp3 [audio.c:501] debug: opening normal audio/file [oss.c:180] error: Can't open default sound device! [audio.c:505] error: failed to open audio device [mpg123.c:666] error: Failed to initialize output, goodbye.

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[alsa-devel] Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)
by Michael Kühn 29 Dec '07

29 Dec '07

Hey folks, i'm the owner of a Acer 6492 Notebook. Details of this soundchip are at the bottom of this e-mail. I want to support you for developing a version of ALSA that support this type of intel HDA audio device. So tell me, how i can help you. (Maybe with testing developer version?). My OS is Gentoo Linux, and i used the latest ALSA driver (just unmask the alsa-driver package). Hope it's ok. :) Thanks for reply. P.S.: Details of the audiochip can found here: http://nopaste.com/p/aS3FUn5pqb

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[alsa-devel] ALC268 "test" model
by Marco Schuster 28 Dec '07

28 Dec '07

Hi all, I try to use a patch from J. Woithe to put a test model into the ALC268 driver [1] (enabling all pins etc.); unfortunately it gives a lot of compiling errors ([2]) which I am unable to understand as I don't have sufficient knowledge of the internal API of HDA codec. Can someone please look over it and correct the mistakes? Marco Schuster [1]: http://harddisk.is-a-geek.org/patch.txt [2]: http://harddisk.is-a-geek.org/compiling-errors.txt

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