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

Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 08 Jul '09

08 Jul '09

At Tue, 7 Jul 2009 11:49:57 +0200, Andreas Nüßlein wrote: > > On Tuesday 07 July 2009 08:00:19 you wrote: > > At Mon, 6 Jul 2009 22:14:56 +0200, > > > > Andreas Nüßlein wrote: > > > On Monday 06 July 2009 21:26:18 you wrote: > > > > At Mon, 06 Jul 2009 11:16:35 -0600, > > > > > > > > Sean Burke wrote: > > > > > Scríobh Takashi Iwai: > > > > > > At Mon, 6 Jul 2009 17:53:46 +0200, > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > >>> The missing pin configuration initialization was already fixed by > > > > > >>> the driver overriding it after checking PCI SSID (which is > > > > > >>> different from the codec SSID). So, this should be no problem. > > > > > >>> > > > > > >>> However, the reason why the analog output doesn't work might be > > > > > >>> different from that. There might be something else missing, but > > > > > >>> I don't know. > > > > > >>> > > > > > >>> > > > > > >>> Takashi > > > > > >> > > > > > >> oh =( > > > > > >> > > > > > >> > > > > > >> hmm.. anything i can do? would it help if i tried changing values > > > > > >> randomly with hda-analyzer.py? > > > > > > > > > > > > Well, did the driver without my change work more or less with > > > > > > the analog audio, or have you never gotten the analog output? > > > > > > You can use the generic parser (i.e. the state without cirrus > > > > > > patch) by passing model=generic option to snd-hda-intel. > > > > > > > > > > For my part, nothing worked with the generic driver. I can't confirm > > > > > digital out, but I can confirm that the kfree error is gone. What > > > > > options are open for figuring out what remains? > > > > > > > > Easy things to test are GPIO bits. Run hda-verb like > > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > or > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > > etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > > > > corresponds to each GPIO. In many case, GPIO0 or GPIO1 corresponds to > > > > the amplifier (EAPD) bit. > > > > Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > > > > turn on/off the GPIO bits by SET_GPIO_DATA. Running > > > > hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > > > > will show the current GPIO data bits. Or you can check it in codec#* > > > > proc file. > > > > > > > > > > > > Takashi > > > > > > w000000000000000000000000000000000t! =) > > > > > > takashi, thank you _so_ much! > > > > > > after running all 4 of those: > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > or > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > i suddendly had sound!! :D :D :D :D :D :D :D > > > only via speakers though - there is no sound via headphones right now. > > > > > > mixer channels: > > > - Master (with Mutebutton), PCM and Front (also with Mute) all work =) > > > - i don't know what surround would do (or it's extra switch) > > > - headphones-volumes and mute button don't affect the speakers, which is > > > good =) > > > > > > > > > is there a way to reset what i did with hda-verb, so that i can figure > > > out which combination it was exactly? > > > > You can just change the value 0x0f to a different value. > > At least, you can try commands like > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > and check the speaker output at each time. > > Also, check the GPIO direction, > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > ... > > > > Regarding the headphone: is the speaker muted when you plug in the > > headphone? If not, it's likely an issue of the jack detection. If > > the speaker is muted but no headphone output, it's a missing > > initialization (or wrong GPIO setup). > > > > > > Takashi > > > so here are my findings so far: > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > > is sufficient to turn on the right functionality for the speakers; meaning: > - 'front speaker'-channel actually controls the two front speakers within the > macbookpro - even controlling left and right channel seperately works =) > - 'surround' mute-switch and volume-control (why are those not within on > thing btw?) also work and control a third speaker in the mac, which > really seems responsibly for surround :) > > headphones do not yet work, however i made a diff of > "/proc/asound/card0/codec#0" with and without something plugged in: OK, so actually the driver works as expected, but the hardware doesn't behave as expected. For muting the speaker, something else (like toggling that GPIO) is needed. The mystery about the silent HP output still remains, though. Takashi

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[alsa-devel] [PATCH 2.6.31-rc2] Missing volume controls for Intel HDA (ALC269/EeePC)
by Darren Salt 08 Jul '09

08 Jul '09

There is a regression, introduced in aa202455eec51699e44f658530728162cefa1307 (in alsa-kernel) which I noticed when trying to use the headphone socket on my EeeCPC 901: the output was *very* quiet, practically silent. This patch corrects the control types to that which was obviously intended in the referenced commit. Signed-Off-By: Darren Salt <linux(a)youmustbejoking.demon.co.uk> --- a/sound/pci/hda/patch_realtek.c 2009-07-05 17:30:41.000000000 +0100 +++ b/sound/pci/hda/patch_realtek.c 2009-07-08 14:50:38.000000000 +0100 @@ -12878,9 +12878,9 @@ static struct snd_kcontrol_new alc269_eeepc_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; -- | Darren Salt | linux at youmustbejoking | nr. Ashington, | Doon | using Debian GNU/Linux | or ds ,demon,co,uk | Northumberland | Army | <URL:http://www.youmustbejoking.demon.co.uk/progs.packages.html> I have one illusion, and that's that I have no illusions.

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[alsa-devel] Problems with snd-hda-intel and Asus M2NPV-VM. Please help!
by VDR User 08 Jul '09

08 Jul '09

I posted this to the alsa-user mailing list but since then I've read that the problem may actually be a bug in alsa so I've decided to forward the posting here as well. Thanks for your consideration. Best regards, Derek ---------- Forwarded message ---------- From: VDR User <user.vdr(a)gmail.com> Date: Wed, Jun 24, 2009 at 11:08 AM Subject: Problems with snd-hda-intel and Asus M2NPV-VM. Please help! To: "mailing list: alsa" <alsa-user(a)lists.sourceforge.net> Greetings. I've been trying for several days now to get the internal spdif working and am posting here as a last resort in hopes someone will know what's wrong. I'll post system details at the end. My mainboard has 3 1/8" plugs on the back and an internal spdif_out connector. If I plug say headphones into the green connector on the back, I hear sound in them. I don't actually want to use analog sound at all. Instead I've plugged the internal spdif_out+ground into my video cards spdif input. Then I have the dvi output from the video card going to an hdmi input on my tv. Note, I have the same setup (except for different mainboard) in another box and have used it to test the video card, which worked fine. The problem is spdif doesn't seem to be working at all. I've eliminated everything else down to either a problem in alsa somewhere, or possibly my xine settings (though I doubt it since I used the same settings from my working box). For alsa I've tried loading the snd_hda_intel driver with "model=3stack", "model=3stack position_fix=1", and "model=3stack position_fix=1 enable_msi=1". None of these worked, I still get nothing on spdif out. I've also tried (un)muting tons of combinations in alsamixer. Could the problem be my kernel version? Alsa version? libasound? Any ideas at all guys?? Any and all help is GREATLY appreciated! Best regards, Derek System details: Please note there is no desktop/windows manager. There is also no keyboard/mouse plugged in. This box is a dedicated htpc running VDR and connected directly to a tv. I'm using PuTTy SSH from my Windows desktop for maintenance. debian testing kernel 2.6.30 asus m2npv-vm mainboard (tried both firmware 0901 and 1401) - uses mcp51 chipset asus EN8400GS Silent 512MB pci-e video card (with spdif input connector) vdr-1.7.8 xine-0.9.3 xine-vdpau r273 ii alsa-base 1.0.20+dfsg-1 ALSA driver configuration files ii alsa-utils 1.0.20-1 ALSA utilities ii libesd-alsa0 0.2.41-4 Enlightened Sound Daemon (ALSA) - Shared lib ii libsdl1.2debian-alsa 1.2.13-4+b1 Simple DirectMedia Layer (with X11 and ALSA htpc:~$ sudo alsactl init Unknown hardware: "HDA-Intel" "Analog Devices AD1986A" "HDA:11d41986,104381b3,00100500" "0x1043" "0x81cb" Hardware is initialized using a guess method htpc:~$ cat /proc/asound/cards 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe024000 irq 22 htpc:~$ cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer alsamixer shows: Card: HDA NVidia Chip: Analog Devices AD1986A Master Headphon PCM Front Surround Center LFE Line CD Mic Mic Boos IEC958 IEC958 D Aux Mono Beep Channel Stereo D (everything is unmuted except stereo downmix) htpc:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 htpc:~$ aplay -L default:CARD=NVidia HDA NVidia, AD198x Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, AD198x Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) htpc:~$ cat /proc/asound/card0/codec#0 Codec: Analog Devices AD1986A Address: 0 Function Id: 0x1 Vendor Id: 0x11d41986 Subsystem Id: 0x104381b3 Revision Id: 0x100500 No Modem Function Group found Default PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 GPIO: io=0, o=1, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital Converter: stream=5, channel=0 Digital: Enabled GenLevel Digital category: 0x2 PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Delay: 3 samples Connection: 2 0x01* 0x06 Node 0x03 [Audio Output] wcaps 0x44d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Processing caps: benign=1, ncoeff=70 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=12, channel=0 Power: setting=D0, actual=D0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x07 [Audio Mixer] wcaps 0x200101: Stereo Connection: 8 0x03 0x09 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x08 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Connection: 2 0x04 0x05 Node 0x0a [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x07* 0x04 0x05 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x07* 0x04 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x04* 0x07 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x05* 0x08 Node 0x0e [Audio Selector] wcaps 0x300100: Mono Connection: 2 0x08* 0x11 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 8 0x1f 0x20 0x1d 0x1d 0x27* 0x28 0x29 0x2a Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x20 0x1c* 0x1f Node 0x11 [Audio Selector] wcaps 0x300941: Stereo R/L Connection: 2 0x0f* 0x2b Processing caps: benign=1, ncoeff=0 Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x08 0x08] Connection: 8 0x11* 0x22 0x00 0x21 0x10 0x07 0x08 0x23 Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x11 Node 0x14 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Connection: 1 0x23 Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Connection: 1 0x22 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x21 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x10 Node 0x18 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x80] Connection: 2 0x19* 0x24 Node 0x19 [Beep Generator Widget] wcaps 0x700000: Mono Node 0x1a [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x0000001f: OUT HP Detect Trigger ImpSense Pin Default 0x02214021: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x0a Node 0x1b [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x0001001f: OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x01014011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x1c [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x91 0x91] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out R/L Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01019015: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x1, Sequence = 0x5 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x1e [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x03] Pincap 0x00000010: OUT Pin Default 0x501700f0: [N/A] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a190f0: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x018130f0: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000027: IN Detect Trigger ImpSense Pin Default 0x509700f0: [N/A] Aux at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x993310f0: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x50b700f0: [N/A] Telephony at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x90f700f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x014510f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: setting=D0, actual=D0 Connection: 8 0x07 0x08 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x27 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x1d Node 0x28 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x20 Node 0x29 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1d 0x20 Node 0x2a [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x1f 0x1d 0x20 Node 0x2b [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0f

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[alsa-devel] [Patch] hda-intel: Add quirks for RTL888 & RV630/M76 based MSI GX710
by William Weston 08 Jul '09

08 Jul '09

Hello, This patch is for yet another MSI laptop that needs the quirk to init the autosense on the headphone jack properly. Best Regards, --ww diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf4b78a..4301680 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9111,6 +9111,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x42cd, "MSI GX710", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), @@ -9124,6 +9125,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0xaa08, "MSI GX710", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),

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[alsa-devel] [PATCH]HDA: Disable AMD SB600 64bit address support only
by Andiry Brienza 08 Jul '09

08 Jul '09

HDA driver disabled HD audio 64bit address support for all AMD SB600/SB700/SB800 platforms with commit 09240cf429505891d6123ce14a29f58f2a60121e due to one SB600 issue reported by community, but we do not see the similar issue on SB700/SB800 platforms. This patch is to refine the workaround for SB600 only. Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> diff -ruN a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c --- a/sound/pci/hda/hda_intel.c 2009-07-01 17:14:23.000000000 +0800 +++ b/sound/pci/hda/hda_intel.c 2009-07-08 10:24:35.000000000 +0800 @@ -2322,9 +2322,19 @@ gcap = azx_readw(chip, GCAP); snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); - /* ATI chips seems buggy about 64bit DMA addresses */ - if (chip->driver_type == AZX_DRIVER_ATI) - gcap &= ~ICH6_GCAP_64OK; + /* disable SB600 64bit support for safety */ + if ((chip->driver_type == AZX_DRIVER_ATI) || + (chip->driver_type == AZX_DRIVER_ATIHDMI)) { + struct pci_dev *p_smbus; + p_smbus = pci_get_device(PCI_VENDOR_ID_ATI, + PCI_DEVICE_ID_ATI_SBX00_SMBUS, + NULL); + if (p_smbus) { + if (p_smbus->revision < 0x30) + gcap &= ~ICH6_GCAP_64OK; + pci_dev_put(p_smbus); + } + } /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))

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[alsa-devel] [PATCH] sound dapm: fix check for external widgets bug (based on linux-2.6.30.1)
by Rongrong Cao 07 Jul '09

07 Jul '09

From: Rongrong Cao rrcao(a)ambarella.com This patch is based on linux-2.6.30.1 In SOC DAPM layer of SOUND subsystem, when add signal route (in the function snd_soc_dapm_add_route() ), the original code has wrong logic when dapm layer check each widget whether an external one. --------------------------------------------------------------------------------------------------------- --- linux-2.6.30.1/sound/soc/soc-dapm.c.orig 2009-07-06 19:48:43.000000000 +0800 +++ linux-2.6.30.1/sound/soc/soc-dapm.c 2009-07-06 19:03:15.000000000 +0800 @@ -983,8 +983,8 @@ static int snd_soc_dapm_add_route(struct if (wsink->id == snd_soc_dapm_input) { if (wsource->id == snd_soc_dapm_micbias || wsource->id == snd_soc_dapm_mic || - wsink->id == snd_soc_dapm_line || - wsink->id == snd_soc_dapm_output) + wsource->id == snd_soc_dapm_line || + wsource->id == snd_soc_dapm_output) wsink->ext = 1; } if (wsource->id == snd_soc_dapm_output) { --------------------------------------------------------------------------------------------------------- Signed-off-by: Rongrong Cao rrcao(a)ambarella.com ******************** Best regards! Allen (Rongrong) Cao Ambarella Shanghai Ltd. Tel: (86-21) 50942311 Fax: (86-21) 50942321 ---------------------------------------------------------------------------------------------------------------------------------- This email message (including attachments) contains Ambarella Proprietary and/or Confidential Information and is intended only for the addressee. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email immediately and destroy all copies of the original message. ----------------------------------------------------------------------------------------------------------------------------------

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[alsa-devel] ALSA SoC driver playback twice as fast as desired
by Louis Lee 07 Jul '09

07 Jul '09

I've been developing an ALSA SoC driver for the past 3 weeks. The codec is a wolfsom WM8973 on a Freescale i.MX27 development platform. I've added SPI support to the existing (using i.MX27 platform SPI API -- not exactly portable) codec in the GIT repository. In addition I wrote up the DAI (I2S over SSI bus) and the PCM portion of the driver which handles DMA transfers. I written the driver to reduce the number of factors to reduce the number of factors that would be causing any problems. The files being played are all encoded for 48kHz. The PLLs, MCLK and BCLK settings are all set to assume a sample rate of 48KHz. Even the file is a mono file. However, when attempting to play the file back using SoX or aplay a self generated sine file made to play for 5 seconds always plays twice as fast. The windows chimes file for instance (which is a 2 x 16 bit audio file) when played as a single channel file -- essentially doubling the play time (0.6 to 1.2 seconds) -- will end up playing properly. The timer displayed by SoX is correct for that file. However, when the file is played back it is obviously being pumped through too fast. I'm out of ideas as to where to look. The DAI interface is set as a I2S slave. The clock is mastered by the Codec (WM8974). I've set up min_channels and max_channels to be 1 whereever possible. On a scope the clocking seems to be at the right speed. Why is it sound is playing twice as fast? Any ideas? Any help is appreciated. WM8974 is a mono codec. The SSI bus on the i.MX27 is set to I2S slave mode with a 16 bit word length. The word length on the codec is set to 16 bits. The audio interface on the codec is set to I2S, sample rate coded to do only 48kHz. THANK YOU _________________________________________________________________ We are your photos. Share us now with Windows Live Photos. http://go.microsoft.com/?linkid=9666047

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[alsa-devel] [PATCH] cmi8330: Allow MPU-401-less operation
by Ondrej Zary 07 Jul '09

07 Jul '09

Adding MPU-401 support to cmi8330 driver could cause a regression (non-working sound) on a system where there is no free IRQ for the MPU-401 device (which is not very uncommon as this card requires two separate IRQs plus a third one for MPU-401). When MPU-401 PnP configuration fails (mostly because of unavailable IRQ), just ignore MPU-401 and continue without it. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-test/sound/isa/cmi8330.c 2009-07-06 22:39:24.000000000 +0200 +++ linux-2.6.30-pentium/sound/isa/cmi8330.c 2009-07-06 22:39:09.000000000 +0200 @@ -389,12 +389,12 @@ pdev = acard->mpu; err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR "MPU-401 PnP configure failure\n"); - return -EBUSY; + if (err < 0) + snd_printk(KERN_ERR "MPU-401 PnP configure failure: will be disabled\n"); + else { + mpuport[dev] = pnp_port_start(pdev, 0); + mpuirq[dev] = pnp_irq(pdev, 0); } - mpuport[dev] = pnp_port_start(pdev, 0); - mpuirq[dev] = pnp_irq(pdev, 0); return 0; } #endif -- Ondrej Zary

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Re: [alsa-devel] Intel HDA using VIA vt1708s
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 7 Jul 2009 15:49:30 +0200, Chrissi k wrote: > > Nope, no luck recording anything. changing the input source doesn't seem to > change anything at all. When I do "arecord..." using "stereo mix", I get a > peak from the external mic-in, same goes for "mic" and "front mic". > I also noticed, the external mic-in does not monitor, e.g. I can't directly > hear the signal, although it should play, as "mic" and "front" are both turned > on in the playback tab. > > here's an alsa-info, taken during recording. Thanks. Looks like the driver doesn't work well for selecting the capture source. Will check tomorrow. (Meanwhile you can change the connection selection of the widget 0x17 via hda-verb to choose the capture source manually.) Takashi

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[alsa-devel] Wrong sample rate user space -> kernel space?
by Daniel Mack 07 Jul '09

07 Jul '09

On a PXA embedded board, the sample rate propagation from user space to kernel space seems to be wrong. When I start aplay (v1.0.19) from userspace like this ... # aplay -f dat /dev/urandom Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo ... params_rate(params) in my snd_soc_ops .hw_params callback returns 88200, not 48000. Can anyone reproduce this? I guess this is unrelated to PXA and ASoC. Thanks, Daniel

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[alsa-devel] Third time: Unlink PCM resource ?
by Werner Van Belle 07 Jul '09

07 Jul '09

Hello, This is the third time I tried to get information on the same topic but there has been no response. Since it is crucial for me to get this to work, I really would appreciate your input. If necssary I can provide further documentation. The problem is as follows: if a process which opened a DSP device forks of a new process, then this child will have the same handle towards the DSP. What I now want to do is to release the handle in the child without closing the device for the parent as well. So essentially the sequence below illustrates the problem if one is unable to 'unlink' a dsp device. a- parent process creates and open dsp device b- parent process forks of a child 1. Child 1 has access to the same dsp device c- parent process closes the DSP device d- client process can no longer write to the DSP device (these are already odd semantics, but okay, can live with it) e- parent process forks of a second child. Child 2 has no dsp device f- Child 2 tries to open the DSP device. This fails because the device, although closed, is not released yet by Child 1. Is this normal behavior ? Is there some rigid documentation on the closing and sharing semantics of pcm devices ? Is there a way to unlink a device without closing it ? If this is the wrong mailinglist, then who should I contact regarding these erratic semantics ? Wkr, -- Dr. Werner Van Belle http://werner.yellowcouch.org/

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[alsa-devel] [PATCH] [RFC 1/13] Intel SST drivers documentation
by Vinod Koul 07 Jul '09

07 Jul '09

This is the first patch of the series. This patch contains the documentation describing the new audio drivers for upcoming Intel MID platform using Atom Processor which uses DSP core for audio processing named Intel smart sound Technology (SST). There are two drivers involved. One driver is SST driver which governs the DSP engine and the other is MAD (MID Audio driver) driver that governs three different sound cards. The intel SST driver enables offload architecture where by the encoded files like mp3, aac and wma can be directly decoded, post processed and rendered by the DSP engine. More details to follow in the following patches The interfaces exposed are documented here as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: Documentation/sound/alsa/intel_sst.txt --- Documentation/sound/alsa/intel_sst.txt | 426 ++++++++++++++++++++++++++++++++ 1 files changed, 426 insertions(+), 0 deletions(-) create mode 100644 Documentation/sound/alsa/intel_sst.txt diff --git a/Documentation/sound/alsa/intel_sst.txt b/Documentation/sound/alsa/intel_sst.txt new file mode 100644 index 0000000..e5d6346 --- /dev/null +++ b/Documentation/sound/alsa/intel_sst.txt @@ -0,0 +1,426 @@ + Intel Smart Sound Technology API documentation + ============================================== + + Vinod Koul <vinod.koul(a)intel.com> + Harsha Priya <priya.harsha(a)intel.com> + +General +------- + +Intel SST is a new hardware component that is capable of decoding and +rendering. This hardware takes in encoded data, decodes and renders the same. +The activity on the CPU is much reduced thus reducing the C0 residency time. +This enables low power playback and increases the battery life. This concept +is made use of in Intel Moblin Internet devices. Intel SST driver is the one +that enables this hardware for use. The firmare file is required to be placed +in /lib/firmware for the driver to download the binary to hardware. + +There are 3 sound cards that Intel SST hardware supports and enables. +Intel SST is the only hardware in the platform that can render to sound card. +These sound cards are exposed and mapped through Intel MID driver which is an +ALSA driver. This driver uses Intel SST driver to send data and controls to the +hardware. + +To make use of the low power playback feature, the application or framework +needs to open a Intel SST driver handle and send the encoded stream +like MP3, AAC or WMA through this driver to play out. If its a PCM stream +the application or framework should use ALSA framework. The PCM stream will +be handled by Intel MID driver which will inturn make use of Intel SST driver +to send out pcm data to sound card. + +To enable the applications or framework to send encoded data, the Intel SST +driver exposes few IOCTLS. The following documentation provides how to use the +Intel SST driver interface. + +This documentation provides the interfaces for rendering audio streams and +tones from the operating system and applications to an audio codec, and +ultimately to the speaker or headphones and capturing/recording +audio streams from input devices + +Interfaces supported: +--------------------- +Intel SST driver supports the following set of file operations listed below for +device /dev/intel_sst: + * open + * close + * mmap + * munmap + * read + * write + * readv + * writev + * ioctl + +Intel SST driver supports the following set of file operations for +device /dev/intel_sst_ctl: + * open + * close + * ioctl +The description of each the above file operations are provided in the following +sections. + +The /dev/intel_sst device can be used by framework/application to send encoded +data + +The /dev/intel_sst_ctl device needs to be opened only by a privilaged +application that can control the policy decisions of the system. (AKA Audio +manager). + +open: +~~~~~ +Application can open the character device through /dev/intel_sst or +/dev/intel_sst_ctl. Open call through /dev/intel_sst will succeed only when a +stream can be allocated for the application. But the stream allocation happens +when the application invokes ioctl system call with SNDRV_SST_STREAM_SET_PARAMS +IO Control to set the stream parameters. Each handle is specific to a stream +only and one stream cannot have multiple handles. +For multiple streams, multiple handles are required to be opened. +/dev/intel_sst_ctl is expected to be used only by Audio Manager. +Audio Manager is also assumed to be running as a privileged process with its +own UID. There is no stream associated when opening with /dev/intel_sst_ctl. +Multiple handles for /dev/intel_sst are not allowed, and only ONE handle is +allowed for Audio Manager. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + - EACCES Max handles reached + - ENODEV Device is not working + - EBUSY Firmware load fail + - EINVAL Invalid params, Major or minor number +close: +~~~~~~ +Appication can close the character device through standard linux close call +when it completes the playing or recoding The driver will clean up the stream +related data structures. The driver will free/close the streams allocated. +In case the stream consists of a mmapped buffer then that will be unmapped. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + - EBADF Invalid/Corrupted handle + - ENODEV Device is not working + - EBUSY Device is busy, should retry after delay + +mmap: +~~~~~ +Applicatiion can call this standard linux call to get mmaped buffers. This call +needs to be called only after stream alloation, else the call would fail +The optimal size of buffer which can be requested using this call is provided +by SNDRV_SST_DRIVER_INFO ioctl. SST Driver will allocate and return the memory +which needs to be used in SNDRV_SST_MMAP_PLAY/ SNDRV_SST_MMAP_CAPTURE. +This call is not required to be called in case the application wants to send +user buffers for playback/capture.Please note that a stream is required to be +mmapped once in lifetime of stream and unmapped once before close. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + - EBADF Invalid handle + - EINVAL Invalid arguments + - ENOMEM Memory allocation failed, + retry with smaller size + - EBUSY Device is busy, should retry after delay +munmap: +~~~~~~~ +Application should call this the standard Linux munmap system call to free any + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + - EBADF Invalid handle + - EINVAL Invalid arguments, address ranges + - EBUSY Device is busy, should retry after delay +write/writev: +~~~~~~~~~~~~ +The application needs to use this system call when it needs to send buffer(s) +containing frames to play audio/speech to the output devices such as +speakers. It is mandatory to invoke SNSDRV_SST_STREAM_SET_PARAMS +ioctl to set up the parameters for the streams before invoking this +system call. This system call will return error if it +is called for a mmapped stream as its expected that mmapped stream uses +SNDRV_SST_MMAP_PLAY ioctl. The buffer doesnt have any max +frames/byte/word length constraints but needs a minimal of one frame size. +For optimization it is recommended that buffer size be a multiple of 20KB along +the frame boundaries. +This call will return when +1) SNDRV_SST_STREAM_DROP is invoked using another application thread +2) The SST firmware has the watermark level for the PCM samples reached +3) The stream encounters an error + Possible Return Values: + ~~~~~~~~~~~~~~~~~~~~~~~ + +ve Bytes written successfully + - EBADF Invalid handle + - EINVAL Invalid arguments + - EBADRQC Invalid request for the stream + - EIO FW error in write +read/readv: +~~~~~~~~~~~ +The application can call this system call when it needs to capture buffer(s) of +recorded audio/speech from the microphone or any other capture device. It is +mandatory to invoke SNDRV_SST_SET_PARAMS ioctl to set up the parameters for the +streams before invoking this system call. This system call will return error if +it is called for a mmapped stream as its expected that mmapped stream used +SNDRV_SST_MMAP_CAPTURE ioctl. The buffer doesnt have any max frames/byte/word +length constraints but needs minimal of one frame size. For optimization its +recommended that buffer size be multiple of 20KB. +The read() will return when +1) SNDRV_SST_STREAM_DROP is invoked using another application thread +2) The SST firmware identifies the watermark level for the PCM samples is reached +3) The stream encounters an error + Possible Return Values: + ~~~~~~~~~~~~~~~~~~~~~~~ + +ve Bytes read successfully. + - EBADF Invalid handle + - EINVAL Invalid arguments + - EBADRQC Invalid request for the stream + - EIO FW error in read +ioctl: +~~~~~~ +SST driver supports a set of IO controls for each device, /dev/intel_sst and +/dev/intel_sst_ctl. All the IOCTLS are blocked till the corresponding operation +is complete. The IO controls are; + +ioctls for /dev/intel_sst: +~~~~~~~~~~~~~~~~~~~~~~~~~~ +SNDRV_SST_STREAM_SET_PARAMS - Set the stream parameter. If sent first + time, the stream will be allocated and be kept + in initialized state. It is required to send + START IO control to start playing. +SNDRV_SST_STREAM_DROP - Stop the stream dropping all the frames. + ALSA IOCTL reused. +SNDRV_SST_STREAM_PAUSE - Pause the stream, no samples are lost + ALSA IOCTL reused. +SNDRV_SST_STREAM_RESUME - Resume from pause, no samples are lost + ALSA IOCTL reused. +SNDRV_SST_STREAM_GET_TSTAMP - Get the time stamp +SNDRV_SST_STREAM_START - Start the stream playback or capture + ALSA IOCTL reused. +SNDRV_SST_MMAP_PLAY - Send the mmap buffer to driver for playback +SNDRV_SST_MMAP_CAPTURE - Send the mmap buffer to driver for capture + +ioctls for /dev/intel_sst_ctrl: (yet to be implemented in the code) +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +SNDRV_SST_FW_INFO - To get or set SST firmware info. +SNDRV_SST_SET_TARGET_DEVICE - To select the target device + (BT/headphones/speakers/etc) + +Common ioctls for /dev/intel_sst and /dev/intel_sst_ctrl: +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +SNDRV_SST_DRIVER_INFO - To obtain version & capabilities of driver + + +SNDRV_SST_STREAM_SET_PARAMS: +~~~~~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control is to allocate a new stream with the firmware for playback or +capture. The parameters will be used to set up the stream for encoding or +decoding. This operation should be invoked to allocate a stream successfully +before using mmap or write or read system call for playback or capture or any +other ioctl. Upon successful completion of this IO Control, the stream id is +filled with a valid number. If the stream is already allocated then driver will +treat this IOCTL as change of stream parameters. Please note that the +parameters can be only changed when SNDRV_SST_STREAM_DROP has been called. + +Note: When the application or middleware has stopped playing or recording and +the stream is idle, it is recommended to close the device and reopen when it +starts again. It will allow other applications to use the stream for playback +or recording as the platform allows only limited number of parallel streams. + +The set of parameters provided in the structure are: + +Codec Parameters which specify the codec specific parameters. +Post/Pre Processing Parameters specify the post/pre processing enabled +for each of the stream. + + Data Structures: + ~~~~~~~~~~~~~~~~ + struct snd_sst_params { + unsigned long result; + unsigned int stream_id; + enum sst_codec_type codec; + enum snd_sst_stream_op ops; + unsigned int stream_type; + struct snd_sst_stream_params sparams; + }; + + /*result: This field contains the completion status of the + requested operation. The value will vary according the stream + control operations. + + Result Codes Description + 0 Success + 1 Stream is not available + 2 Codec is not available + 3 Codec is not supported + 4 Invalid Stream Parameters + 5 Parameter Modified + 6 Invalid stream identifier*/ + + /*stream_id: This is to identify a particular stream + 0 Request to allocate new stream. + Non Zero Valid stream id*/ + /*codec: This field specifies which codec is required for this stream + Please refer to the code for more info*/ + /*ops: This field specifies if this stream is for playback or capture. + + Operation Code Description + STREAM_OPS_PLAYBACK 0x00 For Playback + STREAM_OPS_CAPTURE 0x01 For Recording + STREAM_OPS_PLAYBACK_DRM 0x02 For Protected content playback + STREAM_OPS_PLAYBACK_ALERT 0x03 Alert playback is treated as same as + STREAM_OPS_PLAYBACK. + STREAM_OPS_CAPTURE_VOICE_CALL 0x04 For Voice conversation recording*/ + + /*stream_type: This field will be used in conjunction with stream_op to + determine the specific operation of the stream + + Stream Type Code Description + STREAM_TYPE_MUSIC 0x01 This stream is for Music capture or playback + STREAM_TYPE_VOICE 0x02 This stream is for Voice capture or playback*/ + + /*params: This field contains the stream parameters. Plese refer the code for + the structure*/ + + +SNDRV_SST_STREAM_DROP: +~~~~~~~~~~~~~~~~~~~~~~ +This IO control is to stop the current stream immediately by dropping all +the frames. If the stream is for playback and it is playing currently, +the SST driver shall discard the remaining frames before entering into the +stopped state. If the stream is for capture (recording) and it is capturing +currently, the driver shall return the current captured data and enter into +the stopped state Any data threads which are blocked and waiting will be +unblocked and returned. + + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + -EBADF - Invalid File Descriptor + -ENODEV - Device FW/codec not available or Sound card not available + -EINVAL - Invalid argument + -EBADRQC - Bad/Invalid request + -EBUSY - FW didn't respond within timeout + +SNDRV_SST_STREAM_PAUSE: +~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control pauses the running stream. If the stream is already paused, +the operation will return success. The SST driver and firmware shall maintain +the current pointers so that it can resume the stream when requested. +The previously called write/read/writev/readv thread if blocked will continue +to be blocked. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + Same as given for SNDRV_SST_STREAM_DROP +SNDRV_SST_STREAM_RESUME: +~~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control is to resume the paused stream. If the stream is not paused already +or it is in running state, the operation will return success + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + Same as given for SNDRV_SST_STREAM_DROP + +SNDRV_SST_STREAM_GET_TSTAMP: +~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control returns the rendered stream time in milliseconds + Data Structures: + ~~~~~~~~~~~~~~~~ + unsigned long long time; /*Elapsed time in milli seconds*/ + +The timestamp will be constantly incremented and can be read anytime. In case +of SNDRV_SST_STREAM_PAUSE, the timestamp will also get PAUSED and return the +same value, as long as resume is not called. In the case of +SNDRV_SST_STREAM_DROP, the timestamp will not be incremented or reset. +It will continue to increment from previous value if playback or capture +is invoked again. When a stream is freed, then timestamp will be reset. +If the stream is muted with SNDRV_SST_MUTE the stream is still getting +rendered at zero volume the timestamp will be updated normally + +SNDRV_SST_STREAM_START: +~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control is to start the stream after stream allocation. This call is required +after any read or write, before which the playback or capture will not start. If the +SNDRV_SST_STREAM_DROP is called, then application needs to call SNDRV_SST_STREAM_START +for starting again. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + Same as given for SNDRV_SST_STREAM_DROP + +SNDRV_SST_MMAP_PLAY: +~~~~~~~~~~~~~~~~~~~~ +The application can use this ioctl to send mmapped buffer containing frames to +play audio/speech to the output devices such as speakers. It is mandatory to +invoke SNDRV_SST_STREAM_SET_PARAMS ioctl to set up the parameters for the +streams and mmap() to get mmap buffer before invoking this system call. +Driver will only accept mmap buffer for this ioctl. Application should fill +the size and offset fields properly. The buffer doesnt have any max +frames/byte/word length constraints but needs to be minimal of one frame size. +To optimize this is recommended to be a multiple of 20KB along the frame +boundaries. +This ioctl() will return when +1) SNDRV_SST_STREAM_DROP is invoked using another application thread +2) The SST firmware identifies watermark level is reached +3) The stream encounters an error + Data Structures: + ~~~~~~~~~~~~~~~~ + struct snd_sst_buff_entry { + union { + void *user; + /*user buffer pointer, not valid for this ioctl + Used for write() call*/ + unsigned int offset; + /*mmap offset in mmap memory. Valid for this call*/ + } buffer; + unsigned int size; + /*size of data valid i.e. the size of mmap buffer sent*/ + }; + struct snd_sst_buffs { + unsigned int entries; + /*Number of buffer entries sent, minimum to be one*/ + enum snd_sst_buff_type type; + /*Buffer type sent, required to SST_BUF_MMAP for this ioctl*/ + struct snd_sst_buff_entry *buff; + /*Pointer to the list of buffers*/ + }; + + Possible Return Values: + ~~~~~~~~~~~~~~~~~~~~~~~ + +ve - Number of bytes written successfully + -EBADF - Invalid File Descriptor + -ENODEV - Device is not working + -EINVAL - Invalid argument + -EBADRQC- Bad request + +SNDRV_SST_MMAP_CAPTURE: +~~~~~~~~~~~~~~~~~~~~~~~ +The application can use this ioctl to send mmapped buffer to capture +audio/speech from the input devices such as mic. It is mandatory to +invoke SNDRV_SST_STREAM_SET_PARAMS ioctl to set up the parameters for +the streams and mmap() to get mmap buffer before invoking this system call. +Driver will only accept mmap buffer for this ioctl.Application should fill +the size and offset fields properly. The buffer doesnt have any max +frames/byte/word length constraints but needs to be minimal of one frame size. +To optimize this is recommended to be a multiple of 20KB. +This ioctl() will return when +1) SNDRV_SST_STREAM_DROP is invoked using another application thread +2) The SST firmware identifies the watermark level for the PCM samples is reached +3) The stream encounters an error + Data Structures: + ~~~~~~~~~~~~~~~~ + Same as given in SNDRV_SST_MMAP_PLAY ioctl. + + Possible Return Values: + ~~~~~~~~~~~~~~~~~~~~~~~ + Same as given in SNDRV_SST_MMAP_PLAY ioctl. + +SNDRV_SST_DRIVER_INFO: +~~~~~~~~~~~~~~~~~~~~~~ +This IO Control is to obtain the SST driver static and dynamic information. + Data Structures: + ~~~~~~~~~~~~~~~~ + struct snd_sst_driver_info { + unsigned int version; + /*bit0 - bit7 Minor Version + bit8 - bit15 Major Version + bit16 - bit31 Reserved*/ + unsigned int active_pcm_streams; + /*total number of PCM streams active at the point*/ + unsigned int active_enc_streams; + /*total number of encoded streams active at the point*/ + unsigned int max_pcm_streams; + /*number of PCM streams supported by platform */ + unsigned int max_enc_streams; + /*number of encoded streams supported by platform*/ + unsigned int buf_per_stream; + /*optimal size buffer per stream for mmap call*/ + }; -- 1.5.4.5

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Re: [alsa-devel] Intel HDA using VIA vt1708s
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 7 Jul 2009 15:23:45 +0200, Chrissi k wrote: > > Here's what I do after a fresh reboot: > Change "input source" from "mic" to "front mic" > Turn up all recording mixers to 80%, including "mic boost" and "front mic > boost" > > % arecord -fdat -vv foo.wav > > shouting at the mic...after 10 seconds I do ^C > > % aplay foo.wav > > *silence* OK. Just to be sure, could you try % arecord -fdat -c1 -vv foo.wav ?? Also, attach the alsa-info output during recording with the internal mic. > Is it possible, that the mic is not really connected with the vt1708s, but > rather linked to the internal webcam (which is not working at all btw)? Hm, not impossible, but I don't guess so. The webcam is usually a usb device, and the vendor doesn't do a complicated wiring. > That would explain, why I still get a peak from the external mic-in, even when > I set the input-source to "front mic". > > Thanks for your great help with the unstable-snapshot. Did you actually write > the fix, or was it available all the time? (I thought about using the > unstable-snapshot before but didn't) I wrote the fix shortly. You can see two commits in http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog;h=… BTW, the patches are now in the normal sound tree, so you can use alsa-driver snapshot (no unstable version) from now on again :) thanks, Takashi

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Re: [alsa-devel] Intel HDA using VIA vt1708s
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 7 Jul 2009 14:57:13 +0200, Chrissi k wrote: > > You were so right, arecord + aplay are working fine. Seems to be a problem > with Audacity. > Only thing that's still not showing any sign of life is the internal mic. Is > that part of alsa or should i consider asking the asus-laptop-module guys? > I tried unchecking any "mute" flags using hda-analyzer, but i don't get a peak > using the internal mic. Did you change "Input Source" mixer element to "Front Mic" (or "Mic") properly? The vt170x support code has still no automatic switch via mic jack detection, thus you have to change the recording source manually. Also, there are "Mic Boost" and "Front Mic Boost" volumes that change the input level, too. Takashi

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Re: [alsa-devel] Intel HDA using VIA vt1708s
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 7 Jul 2009 14:38:44 +0200, Chrissi k wrote: > > Thanks for the advice, i tried > ./configure && make && make install && halt > > After a reboot, sound is instantly working, even playing >2 streams at the > same time (mplayer+youtube+totem) works now, too =) OK, I'll merge to the upstream. > But then i started recording something with Audacity, using the external > mic-in. There is the usual peak, so i saved the file and tried re-playing it, > but now the sound is dead again. EAPD is indeed still enabled, but no more > sound from any source. After a reboot, everything works again. Make sure that it's no application problem or mixer setup problem. Try to reproduce with more simpler command sequence, such as, % arecord -fdat -vv foo.wav % aplay foo.wav thanks, Takashi

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[alsa-devel] Intel HDA using VIA vt1708s
by Chrissi k 07 Jul '09

07 Jul '09

Hi, i'd like to report a bug in alsa, up to 1.0.20, in combination with the via vt1708s codec used for example in the asus X5DIJ laptop per default, patch_via.c enables EAPD during initialization. That's not the case especially for the 1708S, so theres no sound at all from internal laptop speakers and line-out. Enabling EAPD using hda_analyzer fixes this issue. I believe there should be some easy fix by initializing the codec with EAPD. I myself can't commit a patch, because my C -kills suck ;) i'd be really glad if someone could take a look at patch_via.c regards didi2002

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Re: [alsa-devel] [LAD] implementing audio device driver on embedded platforms
by Clemens Ladisch 07 Jul '09

07 Jul '09

Jugalkishore Darak, Pankaj wrote: > We are porting Linux 2.6.25.4 to a platform which has ARM11 as its core > processor. My job is to implement audio device driver. Then you'd better ask on alsa-devel (CC'd). > Most of the docs are meant for PC audio which has sound cards on PCI. > And the source code resides in sound folder under the kernel root. > > Since mine is an embedded platform, where will my architecture specific > code resides? Any platform-specific code will be below your arch/ directory. The DMA controller and the codecs are likely to be used on other platforms too, so they'd go into sound/. > Which all folders under sound folder will be used for embedded devices mostly soc/ > and which APIs to look into? linux/Documentation/sound/alsa/soc/*.txt http://alsa-project.org/main/index.php/ASoC http://opensource.wolfsonmicro.com/content/asoc-audio-embedded-linux-systems HTH Clemens

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[alsa-devel] Fwd: Re: No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Andreas Nüßlein 07 Jul '09

07 Jul '09

On Tuesday 07 July 2009 08:00:19 you wrote: > At Mon, 6 Jul 2009 22:14:56 +0200, > > Andreas Nüßlein wrote: > > On Monday 06 July 2009 21:26:18 you wrote: > > > At Mon, 06 Jul 2009 11:16:35 -0600, > > > > > > Sean Burke wrote: > > > > Scríobh Takashi Iwai: > > > > > At Mon, 6 Jul 2009 17:53:46 +0200, > > > > > > > > > > Andreas Nüßlein wrote: > > > > >>> The missing pin configuration initialization was already fixed by > > > > >>> the driver overriding it after checking PCI SSID (which is > > > > >>> different from the codec SSID). So, this should be no problem. > > > > >>> > > > > >>> However, the reason why the analog output doesn't work might be > > > > >>> different from that. There might be something else missing, but > > > > >>> I don't know. > > > > >>> > > > > >>> > > > > >>> Takashi > > > > >> > > > > >> oh =( > > > > >> > > > > >> > > > > >> hmm.. anything i can do? would it help if i tried changing values > > > > >> randomly with hda-analyzer.py? > > > > > > > > > > Well, did the driver without my change work more or less with > > > > > the analog audio, or have you never gotten the analog output? > > > > > You can use the generic parser (i.e. the state without cirrus > > > > > patch) by passing model=generic option to snd-hda-intel. > > > > > > > > For my part, nothing worked with the generic driver. I can't confirm > > > > digital out, but I can confirm that the kfree error is gone. What > > > > options are open for figuring out what remains? > > > > > > Easy things to test are GPIO bits. Run hda-verb like > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > or > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > > > corresponds to each GPIO. In many case, GPIO0 or GPIO1 corresponds to > > > the amplifier (EAPD) bit. > > > Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > > > turn on/off the GPIO bits by SET_GPIO_DATA. Running > > > hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > > > will show the current GPIO data bits. Or you can check it in codec#* > > > proc file. > > > > > > > > > Takashi > > > > w000000000000000000000000000000000t! =) > > > > takashi, thank you _so_ much! > > > > after running all 4 of those: > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > or > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > i suddendly had sound!! :D :D :D :D :D :D :D > > only via speakers though - there is no sound via headphones right now. > > > > mixer channels: > > - Master (with Mutebutton), PCM and Front (also with Mute) all work =) > > - i don't know what surround would do (or it's extra switch) > > - headphones-volumes and mute button don't affect the speakers, which is > > good =) > > > > > > is there a way to reset what i did with hda-verb, so that i can figure > > out which combination it was exactly? > > You can just change the value 0x0f to a different value. > At least, you can try commands like > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > and check the speaker output at each time. > Also, check the GPIO direction, > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > ... > > Regarding the headphone: is the speaker muted when you plug in the > headphone? If not, it's likely an issue of the jack detection. If > the speaker is muted but no headphone output, it's a missing > initialization (or wrong GPIO setup). > > > Takashi so here are my findings so far: hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 is sufficient to turn on the right functionality for the speakers; meaning: - 'front speaker'-channel actually controls the two front speakers within the macbookpro - even controlling left and right channel seperately works =) - 'surround' mute-switch and volume-control (why are those not within on thing btw?) also work and control a third speaker in the mac, which really seems responsibly for surround :) headphones do not yet work, however i made a diff of "/proc/asound/card0/codec#0" with and without something plugged in: $ diff -u without_headphones.txt with_headphones.txt --- without_headphones.txt 2009-07-07 11:41:36.558833270 +0200 +++ with_headphones.txt 2009-07-07 11:41:44.286832992 +0200 @@ -113,7 +113,7 @@ Conn = Unknown, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE - Pin-ctls: 0x40: OUT + Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 1 samples @@ -125,7 +125,7 @@ Conn = Unknown, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE - Pin-ctls: 0x40: OUT + Pin-ctls: 0x00: Delay: 1 samples Connection: 1 0x04 thanks =) nutz

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 07 Jul 2009 00:24:51 -0600, Sean Burke wrote: > > Scríobh Takashi Iwai: > > At Mon, 6 Jul 2009 22:14:56 +0200, > > Andreas Nüßlein wrote: > > > >> On Monday 06 July 2009 21:26:18 you wrote: > >> > >>> At Mon, 06 Jul 2009 11:16:35 -0600, > >>> > >>> Easy things to test are GPIO bits. Run hda-verb like > >>> > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > >>> or > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > >>> > >>> etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > >>> corresponds to each GPIO. In many case, GPIO0 or GPIO1 corresponds to > >>> the amplifier (EAPD) bit. > >>> Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > >>> turn on/off the GPIO bits by SET_GPIO_DATA. Running > >>> hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > >>> will show the current GPIO data bits. Or you can check it in codec#* > >>> proc file. > >>> > >>> > >>> Takashi > >>> > >> w000000000000000000000000000000000t! =) > >> > >> takashi, thank you _so_ much! > >> > >> after running all 4 of those: > >> > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > >>> or > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > >>> > >>> > >> i suddendly had sound!! :D :D :D :D :D :D :D > >> only via speakers though - there is no sound via headphones right now. > >> > >> mixer channels: > >> - Master (with Mutebutton), PCM and Front (also with Mute) all work =) > >> - i don't know what surround would do (or it's extra switch) > >> - headphones-volumes and mute button don't affect the speakers, which is good > >> =) > >> > >> > >> is there a way to reset what i did with hda-verb, so that i can figure out > >> which combination it was exactly? > >> > > > > You can just change the value 0x0f to a different value. > > At least, you can try commands like > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > and check the speaker output at each time. > > Also, check the GPIO direction, > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > ... > > > > Regarding the headphone: is the speaker muted when you plug in the > > headphone? If not, it's likely an issue of the jack detection. If > > the speaker is muted but no headphone output, it's a missing > > initialization (or wrong GPIO setup). > Setting both _DIR and _DATA to 0x08 worked, as did 0x0f. OK, then GPIO2 is EAPD. > Plugging in > headphones does not mute the speaker. I'll check this later. thanks, Takashi

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[alsa-devel] Model quirks and duplicate device IDs
by Ozan Çağlayan 07 Jul '09

07 Jul '09

Hi, A user reports that toshiba model is working fine with its laptop(Subsystem: 152d:0763). I checked the patch_realtek file and saw that the exact ID was mapped against ALC268_ACER 2 years ago for ALSA issue #3343. Both laptops are not-so-popular local brands and checking against 152d points to: 152d QUANTA Computer Inc according to the pci.ids database. So how should we act on those situations seen that we can't match against DMI data(which would possibly be crap on those laptops)? Thanks, Ozan Caglayan

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[alsa-devel] [PATCH] [RFC 3/13] Intel SST driver include headers
by Vinod Koul 07 Jul '09

07 Jul '09

This patch adds the intel_sst.h header file that is places in the include folder for MAD driver (ALSA sound card driver for the platform given in later patches) to use. This file contains the definitions of interfaces exposed by the SST drivers along with the definitions of all the controls for the sound card to be used by MAD driver. intel_sst_ioctl.h - this file exposes the ioctls definitions for player/middleware to use. Many of the definitions are reuse from ALSA Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: include/sound/intel_sst.h new file: include/sound/intel_sst_ioctl.h --- include/sound/intel_sst.h | 259 +++++++++++++++++++++++++++++ include/sound/intel_sst_ioctl.h | 351 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 610 insertions(+), 0 deletions(-) create mode 100644 include/sound/intel_sst.h create mode 100644 include/sound/intel_sst_ioctl.h diff --git a/include/sound/intel_sst.h b/include/sound/intel_sst.h new file mode 100644 index 0000000..c709390 --- /dev/null +++ b/include/sound/intel_sst.h @@ -0,0 +1,259 @@ +#ifndef __INTEL_SST_H__ +#define __INTEL_SST_H__ +/* + * intel_sst.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This driver exposes the audio engine functionalities to the ALSA + * and middleware. + * This file is shared between the SST and MAD drivers + */ + +#ifndef CONFIG_SST_IPC_NOT_INCLUDED +#include <asm/ipc_defs.h> +#endif + +#define SST_CARD_NAMES "intel_mid_card" + +/* +control list Pmic & Lpe +*/ +/*Input controls*/ + +enum sst_controls { + SST_SND_ALLOC = 0x1000, + SST_SND_PAUSE = 0x1001, + SST_SND_RESUME = 0x1002, + SST_SND_DROP = 0x1003, + SST_SND_FREE = 0x1004, + SST_SND_BUFFER_POINTER = 0x1005, + SST_SND_STREAM_INIT = 0x1006, + SST_MAX_CONTROLS = 0x1006, + SST_CONTROL_BASE = 0x1000, +}; + +struct pcm_stream_info { + int str_id; + void *mad_substream; + void (*period_elapsed) (void *mad_substream); + unsigned long buffer_ptr; +}; + +struct sst_control_list { + int control_element; + char *value; + struct sst_control_list *next_elem; +}; + +struct stream_buffer { + unsigned long addr; + int length; +}; + + +/* +SST debug print statement definition +*/ + +static inline long sst_get_time(void) +{ + struct timeval t; + do_gettimeofday(&t); + return t.tv_usec; +} + +#define _sst_printk_(level, format, arg...) \ + printk(level "Time:%8ld::%8ld >> SST: %s %d " format, \ + get_seconds(), sst_get_time(), __func__ , __LINE__, ## arg) + +#define sst_err(format, arg...) _sst_printk_(KERN_ERR, "ERR: " format, ## arg) +#define sst_info(format, arg...) \ + _sst_printk_(KERN_INFO, "INFO: " format, ## arg) + +#ifdef CONFIG_SST_DBG_PRINT +#define sst_dbg(format, arg...) _sst_printk_(KERN_DEBUG, "DBG: " format, ## arg) +#else +#define sst_dbg(format, arg...) do {} while (0); +#endif +struct snd_pmic_ops { + int (*set_input_dev) (int value); + int (*get_input_dev) (int *value); + + int (*set_output_dev) (int value); + int (*get_output_dev) (int *value); + + int (*set_mute) (int dev_id, int value); + int (*get_mute) (int dev_id, int *value); + + int (*set_vol) (int dev_id, int value); + int (*get_vol) (int dev_id, int *value); + + int(*init_card) (void); + void (*set_pcm_params) (int sfreq, int word_size); + + int (*init_card_capture) (void); + +}; + +struct intel_sst_card_ops { + char *module_name; + int vendor_id; + int (*control_set) (int control_element, void *value); + int (*send_buffer) (int str_id, struct stream_buffer *mad_buf); + struct snd_pmic_ops *scard_ops; +}; + +int register_sst_card(struct intel_sst_card_ops *card); +void unregister_sst_card(struct intel_sst_card_ops *card); + +/*periphral interrupt interface*/ +enum lpe_periphral { + LPE_DMA = 1, + LPE_SSP0, + LPE_SSP1 +}; + +/*modified for generic access*/ +struct sc_reg_access { + u16 reg_addr; + u8 value; + u8 mask; +}; +enum sc_reg_access_type { + PMIC_READ = 0, + PMIC_WRITE, + PMIC_READ_MODIFY, +}; + +static inline int sst_sc_reg_access(struct sc_reg_access *sc_access, + int type, int num_val) +{ +#ifndef CONFIG_SST_IPC_NOT_INCLUDED + int i, retval = 0, j = 0, k = 0, count = 0; + struct ipc_pmic_reg_data reg_data; + struct ipc_pmic_mod_reg_data pmic_mod_reg = {0}; + + reg_data.ioc = TRUE; + if (type == PMIC_WRITE) { + do { + int max_retries = 0; + + if (num_val <= 4) + count = num_val; + else + count = 4; +retry_write: + for (i = 0; i < count; i++, j++) { + reg_data.pmic_reg_data[i]. + register_address = sc_access[j].reg_addr; + + reg_data.pmic_reg_data[i].value = + sc_access[j].value; + } + reg_data.num_entries = count; + retval = ipc_pmic_register_write(&reg_data, 0); + if (E_NO_INTERRUPT_ON_IOC == retval && + max_retries < 10) { + sst_err("write communcation needs retry\n"); + max_retries++; + goto retry_write; + } + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic write failed \n"); + return retval; + } + num_val -= count; + } while (num_val > 0); + } else if (type == PMIC_READ) { + do { + int max_retries = 0; + if (num_val <= 4) + count = num_val; + else + count = 4; +retry_read: + for (i = 0; i < count; i++, j++) + reg_data.pmic_reg_data[i].register_address + = sc_access[j].reg_addr; + reg_data.num_entries = count; + retval = ipc_pmic_register_read(&reg_data); + if (E_NO_INTERRUPT_ON_IOC == retval && + max_retries < 10) { + sst_err("read communcation needs retry\n"); + max_retries++; + goto retry_read; + } + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic read failed \n"); + return retval; + } + + for (i = 0; i < count; i++, k++) + sc_access[k].value = + reg_data.pmic_reg_data[i].value; + num_val -= count; + } while (num_val > 0); + } else { + pmic_mod_reg.ioc = TRUE; + do { + int max_retries = 0; + if (num_val <= 4) + count = num_val; + else + count = 4; +retry_readmod: + for (i = 0; i < count; i++, j++) { + pmic_mod_reg.pmic_mod_reg_data[i]. + register_address = sc_access[j].reg_addr; + pmic_mod_reg.pmic_mod_reg_data[i].value = + sc_access[j].value; + pmic_mod_reg.pmic_mod_reg_data[i].bit_map = + sc_access[j].mask; + } + pmic_mod_reg.num_entries = count; + retval = ipc_pmic_register_read_modify(&pmic_mod_reg); + if (E_NO_INTERRUPT_ON_IOC == retval && + max_retries < 10) { + sst_err("read/modify retry\n"); + max_retries++; + goto retry_readmod; + } + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic read_modify failed \n"); + return retval; + } + num_val -= count; + } while (num_val > 0); + } + return retval; +#else + return 0; +#endif +} + +int lpe_mask_periphral_intr(enum lpe_periphral device); +int lpe_unmask_periphral_intr(enum lpe_periphral device); +int lpe_periphral_intr_status(enum lpe_periphral device, int *status); +#endif /*__INTEL_SST_H__*/ diff --git a/include/sound/intel_sst_ioctl.h b/include/sound/intel_sst_ioctl.h new file mode 100644 index 0000000..848ea26 --- /dev/null +++ b/include/sound/intel_sst_ioctl.h @@ -0,0 +1,351 @@ +#ifndef __INTEL_SST_IOCTL_H__ +#define __INTEL_SST_IOCTL_H__ +/* + * intel_sst_ipc.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file defines all sst ioctls + */ + +/*codec and post/pre processing related info*/ + +enum sst_codec_types { +/* AUDIO/MUSIC CODEC Type Definitions */ + SST_CODEC_TYPE_UNKNOWN = 0, + SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ + SST_CODEC_TYPE_MP3, + SST_CODEC_TYPE_MP24, + SST_CODEC_TYPE_AAC, + SST_CODEC_TYPE_AACP, + SST_CODEC_TYPE_eAACP, + SST_CODEC_TYPE_WMA9, + SST_CODEC_TYPE_WMA10, + SST_CODEC_TYPE_WMA10P, + SST_CODEC_TYPE_RA, + SST_CODEC_TYPE_DDAC3, + SST_CODEC_TYPE_STEREO_TRUE_HD, + SST_CODEC_TYPE_STEREO_HD_PLUS, + + /* VOICE CODEC Type Definitions */ + SST_CODEC_TYPE_VOICE_PCM = 0x21, /* Pass through voice codec */ + SRC = 0x64, + MIXER = 0x65, + DOWN_MIXER = 0x66, + VOLUME_CONTROL = 0x67, + OEM1 = 0xC8, + OEM2 = 0xC9, +}; + +enum snd_sst_stream_ops { + STREAM_OPS_PLAYBACK = 0, /* Decode */ + STREAM_OPS_CAPTURE, /* Encode */ + STREAM_OPS_PLAYBACK_DRM, /* Play Audio/Voice */ + STREAM_OPS_PLAYBACK_ALERT, /* Play Audio/Voice */ + STREAM_OPS_CAPTURE_VOICE_CALL, /* CSV Voice recording */ +}; + +enum stream_type { + STREAM_TYPE_MUSIC = 1, + STREAM_TYPE_VOICE +}; + +/* Firmware Version info */ +struct snd_sst_fw_version { + __u8 build; /* build number*/ + __u8 minor; /* minor number*/ + __u8 major; /* major number*/ + __u8 type; /* build type*/ +}; + +/* Port info structure */ +struct snd_sst_port_info { + __u16 port_type; + __u16 reserved; +}; + +/* Mixer info structure */ +struct snd_sst_mix_info { + __u16 max_streams; + __u16 reserved; +}; + +/* PCM Parameters */ +struct snd_pcm_params { + __u16 codec; /* codec type */ + __u8 num_chan; /* 1=Mono, 2=Stereo */ + __u8 pcm_wd_sz; /* 16/24 - bit*/ + __u32 brate; /* Bitrate in bits per second */ + __u32 sfreq; /* Sampling rate in Hz */ + __u16 frame_size; + __u16 samples_per_frame; /* Frame size num samples per frame */ + __u32 period_count; /* period elapsed time count, in samples,*/ +}; + +/* MP3 Music Parameters Message */ +struct snd_mp3_params { + __u16 codec; + __u8 num_chan; /* 1=Mono, 2=Stereo */ + __u8 pcm_wd_sz; /* 16/24 - bit*/ + __u32 brate; /* Use the hard coded value. */ + __u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + __u8 crc_check; /* crc_check - disable (0) or enable (1) */ + __u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB*/ + __u16 reserved; /* Unused */ +}; + +#define AAC_BIT_STREAM_ADTS 0 +#define AAC_BIT_STREAM_ADIF 1 +#define AAC_BIT_STREAM_RAW 2 + +/* AAC Music Parameters Message */ +struct snd_aac_params { + __u16 codec; + __u8 num_chan; /* 1=Mono, 2=Stereo*/ + __u8 pcm_wd_sz; /* 16/24 - bit*/ + __u32 brate; + __u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + __u32 aac_srate; /* Plain AAC decoder operating sample rate */ + __u8 mpg_id; /* 0=MPEG-2, 1=MPEG-4 */ + __u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ + __u8 aac_profile; /* 0=Main Profile, 1=LC profile, 3=SSR profile */ + __u8 ext_chl; /* No.of external channels */ + __u8 aot; /* Audio object type. 1=Main , 2=LC , 3=SSR, 4=SBR*/ + __u8 op_align; /* output alignment 0=16 bit , 1=MSB, 2= LSB align */ + __u8 brate_type; /* 0=CBR, 1=VBR */ + __u8 crc_check; /* crc check 0= disable, 1=enable */ + __s8 bit_stream_format[8]; /* input bit stream format adts/adif/raw */ +}; + +/* WMA Music Parameters Message */ +struct snd_wma_params { + __u16 codec; + __u8 num_chan; /* 1=Mono, 2=Stereo */ + __u8 pcm_wd_sz; /* 16/24 - bit*/ + __u32 brate; /* Use the hard coded value. */ + __u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + __u32 channel_mask; /* Channel Mask */ + __u16 format_tag; /* Format Tag */ + __u16 block_align; /* packet size */ + __u16 wma_encode_opt;/* Encoder option */ + __u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB*/ + __u8 reserved; /* reserved */ +}; + +/* Pre processing param structure */ +struct snd_prp_params { + __u32 reserved; /* No pre-processing defined yet */ +}; + +/* Post processing param structure */ +struct snd_pop_params { + s16 volume; /* Voulme in db */ + u16 ramp_duration; /* Ramp duration in ms */ + u16 mute; /* 0=Un-mute, 1=Mute*/ + u16 reserved; /* */ +}; + +/* Post processing Capability info structure */ +struct snd_sst_postproc_info { + __u32 src_min; /* Supported SRC Min sampling freq */ + __u32 src_max; /* Supported SRC Max sampling freq */ + __u8 src; /* 0=Not supported, 1=Supported */ + __u8 bass_boost; /* 0=Not Supported, 1=Supported */ + __u8 stereo_widening; /* 0=Not Supported, 1=Supported */ + __u8 volume_control; /* 0=Not Supported, 1=Supported */ + __s16 min_vol; /* Minimum value of Volume in dB */ + __s16 max_vol; /* Maximum value of Volume in dB */ +}; + +/* pre processing Capability info structure */ +struct snd_sst_prp_info { + __s16 min_vol; /* Minimum value of Volume in dB */ + __s16 max_vol; /* Maximum value of Volume in dB */ + __u8 volume_control; /* 0=Not Supported, 1=Supported */ + __u8 reserved1; /* for 32 bit alignment */ + __u16 reserved2; /* for 32 bit alignment */ +} __attribute__ ((packed)); + +/* Firmware capabilities info */ +struct snd_sst_fw_info { + struct snd_sst_fw_version fw_version; /* Firmware version */ + __u8 audio_codecs_supported[8]; /* Codecs supported by FW */ + __u32 recommend_min_duration; /* Min duration for Low power Playback*/ + __u8 max_pcm_streams_supported; /*Max number of PCM streams supported */ + __u8 max_enc_streams_supported; /*Max number of Encoded streams */ + __u16 reserved; /* 32 bit alignment*/ + struct snd_sst_postproc_info pop_info; /* Post processing capability*/ + struct snd_sst_prp_info prp_info; /* pre_processing mod cap info */ + struct snd_sst_port_info port_info[2]; /* Port info */ + struct snd_sst_mix_info mix_info; /* Mixer info */ + __u32 min_input_buf; /*minmum i/p buffer for decode*/ +}; + +/* Add the codec parameter structures for new codecs to be supported */ +#define CODEC_PARAM_STRUCTURES \ + struct snd_pcm_params pcm_params; \ + struct snd_mp3_params mp3_params; \ + struct snd_aac_params aac_params; \ + struct snd_wma_params wma_params; + +/* Pre and Post Processing param structures */ +#define PPP_PARAM_STRUCTURES \ + struct snd_prp_params prp_params;\ + struct snd_pop_params pop_params; + +/* Codec params struture */ +union snd_sst_codec_params { + CODEC_PARAM_STRUCTURES; +}; + +/* Pre-processing params struture */ +union snd_sst_ppp_params{ + PPP_PARAM_STRUCTURES; +}; + +struct snd_sst_stream_params { + union snd_sst_codec_params uc; + union snd_sst_ppp_params up; +} __attribute__ ((packed)); + +struct snd_sst_params { + unsigned long result; + unsigned int stream_id; + enum sst_codec_types codec; + enum snd_sst_stream_ops ops; + unsigned int stream_type; + struct snd_sst_stream_params sparams; +}; + +/*ioctl related stuff here*/ +struct snd_sst_pmic_config { + __u32 sfreq; /* Sampling rate in Hz */ + __u16 num_chan; /* Mono =1 or Stereo =2 */ + __u16 pcm_wd_sz; /* Number of bits per sample */ +} __attribute__ ((packed)); + +struct snd_sst_get_stream_params { + struct snd_sst_params codec_params; + struct snd_sst_pmic_config pcm_params; +}; + +enum snd_sst_target_type { + SND_SST_TARGET_PMIC = 1, + SND_SST_TARGET_OTHER, +}; + +enum snd_sst_port_action { + SND_SST_PORT_PREPARE = 1, + SND_SST_PORT_ACTIVATE, +}; + +/* Target selection per device structure */ +struct snd_sst_slot_info { + __u8 mix_enable; /* Mixer enable or disable */ + __u8 device_type; + __u8 device_instance; /* 0, 1, 2 */ + __u16 slot[2]; + struct snd_sst_pmic_config pcm_params; + enum snd_sst_target_type type; + __s8 master; + enum snd_sst_port_action action; +} __attribute__ ((packed)); + +/* Target device list structure */ +struct snd_sst_target_device { + __u32 device_route; + struct snd_sst_slot_info devices[2]; +} __attribute__ ((packed)); + +struct snd_sst_driver_info { + unsigned int version; /* Version of the driver */ + unsigned int active_pcm_streams; + unsigned int active_enc_streams; + unsigned int max_pcm_streams; + unsigned int max_enc_streams; + unsigned int buf_per_stream; +}; + +struct snd_sst_vol { + unsigned int stream_id; + int volume; + unsigned long ramp_duration; + __u32 ramp_type; /* Ramp type, default=0 */ +}; + +struct snd_sst_mute { + unsigned int stream_id; + int mute; +}; + +enum snd_sst_buff_type { + SST_BUF_USER = 1, + SST_BUF_MMAP, + SST_BUF_RAR, +}; + +struct snd_sst_buff_entry { + union { + void *user; + unsigned int offset; + } buffer; + unsigned int size; +}; + +struct snd_sst_buffs { + unsigned int entries; + enum snd_sst_buff_type type; + struct snd_sst_buff_entry *buff; +}; + +struct snd_sst_dbufs { + struct snd_sst_buff_entry *ibufs; + struct snd_sst_buff_entry *obufs; + __s8 force; +}; + +/*IOCTL defined here*/ +/*SST MMF IOCTLS only*/ +#define SNDRV_SST_STREAM_SET_PARAMS _IOR('L', 0x00, \ + struct snd_sst_stream_params *) +#define SNDRV_SST_STREAM_GET_PARAMS _IOWR('L', 0x01, \ + struct snd_sst_get_stream_params *) +#define SNDRV_SST_STREAM_GET_TSTAMP _IOWR('L', 0x02, unsigned long long *) +#define SNDRV_SST_STREAM_DECODE _IOWR('L', 0x03, struct snd_sst_dbufs *) +#define SNDRV_SST_STREAM_START _IO('A', 0x42) +#define SNDRV_SST_STREAM_DROP _IO('A', 0x43) +#define SNDRV_SST_STREAM_DRAIN _IO('A', 0x44) +#define SNDRV_SST_STREAM_PAUSE _IOW('A', 0x45, int) +#define SNDRV_SST_STREAM_RESUME _IO('A', 0x47) +#define SNDRV_SST_MMAP_PLAY _IOW('L', 0x05, struct snd_sst_buffs *) +#define SNDRV_SST_MMAP_CAPTURE _IOW('L', 0x06, struct snd_sst_buffs *) +/*SST common ioctls */ +#define SNDRV_SST_DRIVER_INFO _IOR('L', 0x10, struct snd_sst_driver_info *) +#define SNDRV_SST_SET_VOL _IOW('L', 0x11, struct snd_sst_vol *) +#define SNDRV_SST_GET_VOL _IOW('L', 0x12, struct snd_sst_vol *) +#define SNDRV_SST_MUTE _IOW('L', 0x13, struct snd_sst_mute *) + +/*AM Ioctly only*/ +#define SNDRV_SST_FW_INFO _IOR('L', 0x20, struct snd_sst_fw_info *) +#define SNDRV_SST_SET_TARGET_DEVICE _IOW('L', 0x21, \ + struct snd_sst_target_device *) + +#endif /*__INTEL_SST_IOCTL_H__*/ -- 1.5.4.5

4 9

[alsa-devel] [PATCH] [RFC 5/13] Intel SST driver interface module
by Vinod Koul 07 Jul '09

07 Jul '09

This patch adds the SST driver interface modules. Interface module is the one which talks to other layers of SST drivers. intel_sst_interface.c - This file implements the MAD driver registration and de registration functions. SST driver is also a character driver that allows players/middleware to communicate with SST driver. All char driver routines are implemented here. The ioctls used by middleware to open/close, control and configure stream and transfer the data are implemented here Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: R Dharageswari <dharageswari.r(a)intel.com> new file: sound/pci/sst/intel_sst_interface.c --- sound/pci/sst/intel_sst_interface.c | 1359 +++++++++++++++++++++++++++++++++++ 1 files changed, 1359 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intel_sst_interface.c diff --git a/sound/pci/sst/intel_sst_interface.c b/sound/pci/sst/intel_sst_interface.c new file mode 100644 index 0000000..f8cefa5 --- /dev/null +++ b/sound/pci/sst/intel_sst_interface.c @@ -0,0 +1,1359 @@ +/* + * intel_sst_interface.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * R Dharageswari <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * This driver exposes the audio engine functionalities to the ALSA + * and middleware. + * Upper layer interfaces (MAD driver, MMF) to SST driver + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/fs.h> +#include <linux/file.h> +#include <linux/fcntl.h> +#include <linux/uaccess.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include <linux/workqueue.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <asm/div64.h> +#include <linux/ioctl.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#ifdef CONFIG_SST_OSPM_SUPPORT +#include <linux/intel_mid.h> +#endif + + +int sst_download_fw(void) +{ + int retval = 0; + const struct firmware *fw_sst; + + sst_dbg("SST Downloading FW now...\n"); + retval = request_firmware(&fw_sst, + SST_FW_STD_FILENAME, + &sst_ops->pci->dev); + if (0 != retval) { + sst_err("fw load failed %d \n", retval); + return retval; + } + sst_ops->alloc_block[0].sst_id = 0xFF; + sst_load_fw(fw_sst, NULL); + retval = sst_wait_timeout(sst_ops, &sst_ops->alloc_block[0]); + release_firmware(fw_sst); + sst_ops->alloc_block[0].sst_id = BLOCK_UNINIT; + return retval; +} + +/** +* intel_sst_open - opens a handle to driver +* @i_node: inode structure +* @file_ptr:pointer to file +* +* This function is called by OS when a user space component +* tries to get a driver handle. Only one handle at a time +* will be allowed +*/ +int intel_sst_open(struct inode *i_node, struct file *file_ptr) +{ + dev_t device = i_node->i_rdev; + unsigned int retval = 0; + struct ioctl_pvt_data *data = NULL; + + if (sst_ops->pmic_state != PMIC_SND_INIT_DONE) { + sst_err("Sound card not availble \n"); + return -EIO; + } + + if (SST_UN_INIT == sst_ops->sst_state) { + /*FW is not downloaded*/ + retval = sst_download_fw(); + if (retval != 0) { + sst_err("FW download failed...abort\n"); + return -ENODEV; + } + } + if (device == MKDEV(INTEL_SST_MAJOR, 0)) { + /*app open*/ + if (sst_ops->active_cnt < MAX_ENC_STREAM) { + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (NULL == data) + return -ENOMEM; + sst_ops->active_cnt++; + sst_ops->stream_cnt++; + data->pvt_id = sst_assign_pvt_id(sst_ops); + data->str_id = 0; + file_ptr->private_data = (void *)data; + sst_dbg("sst id allocated = %d!\n", data->pvt_id); + } else + retval = -EACCES; + } else if (device == MKDEV(INTEL_SST_MAJOR, 1)) { + /*audio manager open*/ + if (sst_ops->am_cnt < MAX_AM_HANDLES) { + sst_ops->am_cnt++; + sst_dbg("AM handle opened...\n"); + } else + retval = -EACCES; + } else + retval = -EINVAL; + return retval; +} + +/** +* intel_sst_release - releases a handle to driver +* @i_node: inode structure +* @file_ptr: pointer to file +* +* This function is called by OS when a user space component +* tries to release a driver handle. +*/ +int intel_sst_release(struct inode *i_node, struct file *file_ptr) +{ + dev_t device = i_node->i_rdev; + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)file_ptr->private_data; + + sst_dbg("Release called \n"); + if (device == MKDEV(INTEL_SST_MAJOR, 0)) { + /*app close*/ + sst_dbg("Closing app handle \n"); + sst_ops->active_cnt--; + sst_ops->stream_cnt--; + if (0 != sst_free_stream(data->str_id)) { + if (sst_validate_strid(data->str_id) == 0) + sst_clean_stream( + &sst_ops->streams[data->str_id]); + } + kfree(data); + } else if (device == MKDEV(INTEL_SST_MAJOR, 1)) + /*audio manager close*/ + sst_dbg("AM handle closed \n"); + sst_ops->am_cnt--; + return 0; +} + +int intel_sst_mmap(struct file *file_ptr, struct vm_area_struct *vma) +{ + int retval = 0, length = 0; + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)file_ptr->private_data; + int str_id = data->str_id; + void *mem_area = NULL; + + retval = sst_validate_strid(str_id); + if (retval != 0) + return -EINVAL; + + length = vma->vm_end - vma->vm_start; + sst_dbg("called for stream %d length 0x%x\n", str_id, length); + + if (length > sst_ops->mmap_len) + return -ENOMEM; + if (sst_ops->mmap_mem == NULL) + return -EIO; + + /* round it up to the page bondary */ + mem_area = (void *)((((unsigned long)sst_ops->mmap_mem) + PAGE_SIZE - 1) + & PAGE_MASK); + + /* map the whole physically contiguous area in one piece */ + retval = remap_pfn_range(vma, + vma->vm_start, + virt_to_phys((void *)mem_area) >> PAGE_SHIFT, + length, + vma->vm_page_prot); + if (retval != 0) { + sst_ops->streams[str_id].mmapped = false; + sst_err("mapping failed %d", retval); + } else + sst_ops->streams[str_id].mmapped = true; + + sst_dbg("mmap ret 0x%x \n", retval); + return retval; +} +/** +* intel_sst_write- send data to play out +*/ +int intel_sst_mmap_play_capture(u32 str_id, struct snd_sst_buffs *mmap_buf) +{ + struct sst_stream_bufs *bufs = NULL; + int retval = 0, i; + struct stream_info *stream = NULL; + struct snd_sst_buff_entry *buf_entry = NULL; + + sst_dbg("called for str_id %d \n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) { + sst_err("val failed %d", retval); + return -EINVAL; + } + BUG_ON(!mmap_buf); + + stream = &sst_ops->streams[str_id]; + if (stream->mmapped != true) { + sst_err("stream not mapped!!!"); + return -EIO; + } + + if (stream->status != STREAM_RUNNING && + stream->status != STREAM_INIT) { + sst_err("BAD REQUEST!, stream state is %d\n", stream->status); + return -EBADRQC; + } + + sst_dbg("new buffers count %d status %d\n", + mmap_buf->entries, stream->status); + buf_entry = mmap_buf->buff; + for (i = 0; i < mmap_buf->entries; i++) { + BUG_ON(!buf_entry); + bufs = kzalloc(sizeof(*bufs), GFP_KERNEL); + if (bufs == NULL) + return -ENOMEM; + bufs->size = buf_entry->size; + bufs->offset = buf_entry->buffer.offset; + bufs->addr = sst_ops->mmap_mem; + bufs->in_use = false; + buf_entry++; + /*locking here*/ + mutex_lock(&stream->lock); + list_add_tail(&bufs->node, &stream->bufs); + mutex_unlock(&stream->lock); + } + + mutex_lock(&stream->lock); + stream->data_blk.condition = false; + stream->data_blk.ret_code = 0; + if (stream->status == STREAM_INIT && stream->prev != STREAM_UN_INIT) { + stream->prev = stream->status; + stream->status = STREAM_RUNNING; + if (stream->ops == STREAM_OPS_PLAYBACK) { + sst_dbg("play frames...\n"); + if (sst_play_frame(str_id) < 0) { + sst_err("play frames failed \n"); + mutex_unlock(&stream->lock); + return -EIO; + } + } else if (stream->ops == STREAM_OPS_CAPTURE) { + sst_dbg("capture frames...\n"); + if (sst_capture_frame(str_id) < 0) { + sst_err("capture frames failed \n"); + mutex_unlock(&stream->lock); + return -EIO; + } + } + } + mutex_unlock(&stream->lock); + /*Block the call for reply*/ + if (0 == list_empty(&stream->bufs)) { + sst_dbg("ioctl waiting...\n"); + stream->data_blk.on = true; + retval = sst_wait_interruptible(sst_ops, &stream->data_blk); + } + + sst_dbg("end of play/rec ioctl!!\n"); + return retval; +} + +int sst_play_capture(struct stream_info *stream, int str_id) +{ + int retval; + + stream->data_blk.condition = false; + stream->data_blk.ret_code = 0; + mutex_lock(&stream->lock); + if (stream->status == STREAM_INIT && stream->prev != STREAM_UN_INIT) { + /*stream is started*/ + stream->prev = stream->status; + stream->status = STREAM_RUNNING; + } + + if (stream->status == STREAM_INIT && stream->prev == STREAM_UN_INIT) { + /*stream is not started yet*/ + sst_err("Stream isnt started yet state %d, prev %d \n", + stream->status, stream->prev); + } else if (stream->status == STREAM_RUNNING) { + /*stream is started*/ + if (stream->ops == STREAM_OPS_PLAYBACK) { + if (sst_play_frame(str_id) < 0) { + sst_err("play frames failed \n"); + mutex_unlock(&stream->lock); + return -EIO; + } + } else if (stream->ops == STREAM_OPS_CAPTURE) { + if (sst_capture_frame(str_id) < 0) { + sst_err("capture frames failed \n"); + mutex_unlock(&stream->lock); + return -EIO; + } + } + } else + sst_err("Stream state %d invalid will still block, prev %d\n", + stream->status, stream->prev); + mutex_unlock(&stream->lock); + /*Block the call for reply*/ + sst_dbg("write waiting...\n"); + stream->data_blk.on = true; + retval = sst_wait_interruptible(sst_ops, &stream->data_blk); + if (retval != 0) { + stream->status = STREAM_INIT; + sst_dbg("wait returned error...\n"); + } + return retval; +} + +int snd_sst_fill_kernel_list(struct stream_info *stream, + const struct iovec *iovec, unsigned long nr_segs, + struct list_head *copy_to_list) +{ + struct sst_stream_bufs *stream_bufs = NULL; + unsigned long index, data_not_copied, mmap_len; + unsigned char *bufp; + unsigned long size, copied_size; + int retval = 0, add_to_list = 0; + static int sent_offset; + static unsigned long sent_index; + + stream_bufs = kzalloc(sizeof(*stream_bufs), GFP_KERNEL); + if (stream_bufs == NULL) { + sst_err("memory allocation failed \n"); + return -ENOMEM; + } + + mmap_len = sst_ops->mmap_len; + stream_bufs->addr = sst_ops->mmap_mem; + bufp = stream->cur_ptr; + + sst_dbg("mmap_len - %lx\n", mmap_len); + copied_size = 0; + + if (stream->sg_index == 0) + sent_index = sent_offset = 0; + + for (index = stream->sg_index; index < nr_segs; index++) { + stream->sg_index = index; + sst_dbg("index - %lx, cur_ptr - %p\n", index, stream->cur_ptr); + sst_dbg("base - %p, size - 0x%x\n", iovec[index].iov_base, + iovec[index].iov_len); + sst_dbg("bufp - %p\n", bufp); + if (stream->cur_ptr == NULL) + bufp = iovec[index].iov_base; + + size = ((unsigned long)iovec[index].iov_base + + iovec[index].iov_len) - (unsigned long) bufp; + + sst_dbg("size - %lx\n", size); + if ((copied_size + size) > mmap_len) + size = mmap_len - copied_size; + + sst_dbg("size - %lx\n", size); + + if (stream->ops == STREAM_OPS_PLAYBACK) { + sst_dbg("Playback stream copying now....\n"); + data_not_copied = copy_from_user( + (void *)(stream_bufs->addr + copied_size), + bufp, size); + if (data_not_copied > 0) { + /*Clean up the list and return error code */ + sst_err("copy from user not copied -%ld\n", + data_not_copied); + retval = -EIO; + break; + } + } else { + struct snd_sst_user_cap_list *entry = + kzalloc(sizeof(*entry), GFP_KERNEL); + + if (entry == NULL) { + sst_err("mem alloacation failed \n"); + return -ENOMEM; + } + entry->iov_index = index; + entry->iov_offset = (unsigned long) bufp - + (unsigned long)iovec[index].iov_base; + entry->offset = copied_size; + entry->size = size; + sst_dbg("ENTRY:ioindx %d,iooff %ld,koff %ld,ksz %ld \n", + entry->iov_index, entry->iov_offset, + entry->offset, entry->size); + list_add_tail(&entry->node, copy_to_list); + } + + sst_dbg("cur_ptr - %lx\n", (unsigned long) stream->cur_ptr); + stream->cur_ptr = bufp + size; + + if (((unsigned long)iovec[index].iov_base + + iovec[index].iov_len) == + (unsigned long)stream->cur_ptr) { + stream->cur_ptr = NULL; + stream->sg_index++; + } + + copied_size += size; + sst_dbg("copied_size - %lx\n", copied_size); + if ((copied_size >= mmap_len) || + (stream->sg_index == nr_segs)) { + add_to_list = 1; + } + + if (add_to_list) { + stream_bufs->in_use = false; + stream_bufs->size = copied_size; + /*locking here*/ + mutex_lock(&stream->lock); + list_add_tail(&stream_bufs->node, &stream->bufs); + mutex_unlock(&stream->lock); + break; + } + } + return retval; +} + +int snd_sst_copy_userbuf_capture(struct stream_info *stream, + const struct iovec *iovec, + struct list_head *copy_to_list) +{ + struct snd_sst_user_cap_list *entry, *_entry; + struct sst_stream_bufs *kbufs = NULL, *_kbufs; + int retval = 0; + unsigned long data_not_copied; + + /*copy sent buffers*/ + sst_dbg("capture stream copying to user now...\n"); + list_for_each_entry_safe(kbufs, _kbufs, &stream->bufs, node) { + if (kbufs->in_use == true) { + /*copy to user*/ + list_for_each_entry_safe(entry, _entry, + copy_to_list, node) { + sst_dbg("filling now... \n"); + sst_dbg("iindx %d,ioff %ld,koff %ld,ksz %ld \n", + entry->iov_index, entry->iov_offset, + entry->offset, entry->size); + sst_dbg("Copying at %p size %lx\n", + iovec[entry->iov_index].iov_base + + entry->iov_offset, + entry->size); + data_not_copied = copy_to_user((void *) + iovec[entry->iov_index].iov_base + + entry->iov_offset, + kbufs->addr + entry->offset, + entry->size); + if (data_not_copied > 0) { + /*Clean up the list and return error*/ + sst_err("copy to user err -%ld\n", + data_not_copied); + retval = -EIO; + break; + } + list_del(&entry->node); + kfree(entry); + } + sst_dbg("coming out of loop\n"); + } + } + sst_dbg("end of cap copy\n"); + return retval; +} +/* + * snd_sst_userbufs_play_cap - constructs the list from user buffers + * @iovec: pointer to iovec structure + * @nr_segs: number entries in the iovec structure + * @str_id: stream id + * @stream: pointer to stream_info structure + * This function will traverse the user list and copy the data to the kernel + * space buffers. + */ +int snd_sst_userbufs_play_cap(const struct iovec *iovec, + unsigned long nr_segs, unsigned int str_id, + struct stream_info *stream) +{ + int retval = 0; + LIST_HEAD(copy_to_list); + + + retval = snd_sst_fill_kernel_list(stream, iovec, nr_segs, + &copy_to_list); + + retval = sst_play_capture(stream, str_id); + if (retval < 0) + return retval; + + if (stream->ops == STREAM_OPS_CAPTURE) { + retval = snd_sst_copy_userbuf_capture(stream, iovec, + &copy_to_list); + } + return retval; +} + +int intel_sst_read_write(unsigned int str_id, char __user *buf, size_t count) +{ + int retval = 0; + struct stream_info *stream = NULL; + struct iovec iovec; + unsigned long nr_segs; + + retval = sst_validate_strid(str_id); + if (retval != 0) + return -EINVAL; + stream = &sst_ops->streams[str_id]; + if (stream->mmapped == true) { + sst_err("user write & stream is mapped!!!"); + return -EIO; + } + if (count == 0) { + sst_err("args invalid %d", retval); + return -EINVAL; + } + /*copy user buf details*/ + if (stream->status != STREAM_RUNNING && stream->status != STREAM_INIT) { + sst_err("BAD REQUEST!\n"); + return -EBADRQC; + } + sst_dbg("new buffers %p, copy size %d, status %d\n" , + buf, (int) count, (int) stream->status); + + stream->buf_type = SST_BUF_USER_STATIC; + iovec.iov_base = (void *)buf; + iovec.iov_len = count; + nr_segs = 1; + + do { + retval = snd_sst_userbufs_play_cap(&iovec, nr_segs, + str_id, stream); + + if (retval < 0) + break; + + } while (stream->sg_index < nr_segs); + + stream->sg_index = 0; + stream->cur_ptr = NULL; + return retval; +} + +int intel_sst_write(struct file *file_ptr, const char __user *buf, + size_t count, loff_t *offset) +{ + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)file_ptr->private_data; + int str_id = data->str_id; + + sst_dbg("called for %d\n", str_id); + return intel_sst_read_write(str_id, (char __user *)buf, count); +} + +/* + * + * intel_sst_aio_write- send data to play out + * + */ +ssize_t intel_sst_aio_write(struct kiocb *kiocb, const struct iovec *iov, + unsigned long nr_segs, loff_t offset) +{ + int retval = 0; + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)kiocb->ki_filp->private_data; + int str_id = data->str_id; + struct stream_info *stream = NULL; + + sst_dbg("entry - %ld\n", nr_segs); + + if (is_sync_kiocb(kiocb) == false) { + sst_dbg("aio_read from user space is not allowed\n"); + return -EINVAL; + } + + sst_dbg("called for str_id %d \n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return -EINVAL; + stream = &sst_ops->streams[str_id]; + if (stream->mmapped == true) { + sst_err("user write & stream is mapped!!!"); + return -EIO; + } + if (stream->status != STREAM_RUNNING && stream->status != STREAM_INIT) { + sst_err("BAD REQUEST!\n"); + return -EBADRQC; + } + sst_dbg("new segs %ld, offset %d, status %d\n" , + nr_segs, (int) offset, (int) stream->status); + stream->buf_type = SST_BUF_USER_STATIC; + do { + retval = snd_sst_userbufs_play_cap(iov, nr_segs, + str_id, stream); + if (retval < 0) + break; + + } while (stream->sg_index < nr_segs); + + stream->sg_index = 0; + stream->cur_ptr = NULL; + + return retval; +} + +/* +* intel_sst_read- get captured data +*/ +int intel_sst_read(struct file *file_ptr, char __user *buf, + size_t count, loff_t *offset) +{ + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)file_ptr->private_data; + int str_id = data->str_id; + + sst_dbg("called for %d\n", str_id); + return intel_sst_read_write(str_id, buf, count); +} + +ssize_t intel_sst_aio_read(struct kiocb *kiocb, const struct iovec *iov, + unsigned long nr_segs, loff_t offset) +{ + int retval = 0; + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)kiocb->ki_filp->private_data; + int str_id = data->str_id; + struct stream_info *stream = NULL; + + sst_dbg("entry - %ld\n", nr_segs); + + if (is_sync_kiocb(kiocb) == false) { + sst_dbg("aio_read from user space is not allowed\n"); + return -EINVAL; + } + + sst_dbg("called for str_id %d \n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return -EINVAL; + stream = &sst_ops->streams[str_id]; + if (stream->mmapped == true) { + sst_err("user write & stream is mapped!!!"); + return -EIO; + } + if (stream->status != STREAM_RUNNING && stream->status != STREAM_INIT) { + sst_err("BAD REQUEST!\n"); + return -EBADRQC; + } + sst_dbg("new segs %ld, offset %d, status %d\n" , + nr_segs, (int) offset, (int) stream->status); + stream->buf_type = SST_BUF_USER_STATIC; + do { + retval = snd_sst_userbufs_play_cap(iov, nr_segs, + str_id, stream); + if (retval < 0) + break; + + } while (stream->sg_index < nr_segs); + + stream->sg_index = 0; + stream->cur_ptr = NULL; + + return retval; +} + +int sst_get_stream_allocated(struct snd_sst_params *str_param, + u32 block, u32 pvt_id) +{ + int retval = 0; + + retval = sst_alloc_stream((char *) &str_param->sparams, str_param->ops, + str_param->codec, pvt_id); + if (0 != retval) { + sst_err("sst_alloc_stream failed %d", retval); + goto err; + } + /*Block the call for reply*/ + retval = sst_wait_timeout(sst_ops, &sst_ops->alloc_block[block]); +err: return retval; +} + +void set_port_params(struct snd_sst_params *str_param, + enum snd_sst_stream_ops ops) +{ + int sfreq = str_param->sparams.uc.pcm_params.sfreq; + int word_size = str_param->sparams.uc.pcm_params.pcm_wd_sz; + + sst_dbg("sampling frequency = %d wd_size = %d \n", sfreq, word_size); + + if (ops == STREAM_OPS_PLAYBACK) { + sst_dbg("Setting playback path and port settings...\n"); + sst_ops->scard_ops.init_card(); + sst_ops->scard_ops.set_pcm_params(sfreq, word_size); + } else if (ops == STREAM_OPS_CAPTURE) { + sst_dbg("Setting capture path...\n"); + if (sst_ops->scard_ops.init_card_capture != NULL) + sst_ops->scard_ops.init_card_capture(); + } + return; +} + +int sst_get_stream(struct snd_sst_params *str_param, u32 pvt_id) +{ + int i = 0, retval = 0; + + /*stream is not allocated, we are allocating*/ + i = sst_get_block_stream(sst_ops); + sst_dbg("alloc block allocated = %d\n", i); + if (i < 0) + return -ENOMEM; + sst_ops->alloc_block[i].sst_id = pvt_id; + + retval = sst_get_stream_allocated(str_param, i, pvt_id); + if (retval == -(SST_LIB_ERR_LIB_DNLD_REQUIRED)) { + /*codec download is required*/ + struct snd_sst_alloc_response *response = + sst_ops->alloc_block[i].ops_block.data; + sst_dbg("Codec is required.... trying that\n"); + retval = sst_load_library(&response->lib_dnld, + str_param->ops, pvt_id); + kfree(response); + + if (retval == 0) { + sst_dbg("codec was downloaded sucesfully \n"); + sst_dbg("try alloac again\n"); + sst_ops->alloc_block[i].ops_block.condition = false; + + retval = sst_get_stream_allocated(str_param, i, pvt_id); + + if (retval <= 0) + goto err; + set_port_params(str_param, str_param->ops); + + sst_dbg("Allocation done stream id %d \n", retval); + } else { + sst_dbg("codec download failed \n"); + retval = -EIO; + goto err; + } + } else if (retval <= 0) + goto err; + else + set_port_params(str_param, str_param->ops); + +err: + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + return retval; +} + +/** +* intel_sst_ioctl - recieves the device ioctl's +* @i_node: inode structure +* @file_ptr: pointer to file +* @cmd: Ioctl cmd +* @arg: data +* +* This function is called by OS when a user space component +* sends an Ioctl to SST driver +*/ +int intel_sst_ioctl(struct inode *i_node, struct file *file_ptr, + unsigned int cmd, unsigned long arg) +{ + int retval = 0, i = 0; + struct ioctl_pvt_data *data = (struct ioctl_pvt_data *) + file_ptr->private_data; + int str_id = data->str_id, minor; + + dev_t device = i_node->i_rdev; + + if (device == MKDEV(INTEL_SST_MAJOR, 0)) + minor = 0; + else if (device == MKDEV(INTEL_SST_MAJOR, 1)) + minor = 1; + else + return -EINVAL; + + if (SST_FW_RUNNING != sst_ops->sst_state) { + sst_err("SST Not runng %d\n", sst_ops->sst_state); + return -EBUSY; + } + switch (_IOC_NR(cmd)) { + case _IOC_NR(SNDRV_SST_STREAM_PAUSE): + sst_dbg("SNDRV_SST_IOCTL_PAUSE recieved for %d!\n", str_id); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = sst_pause_stream(str_id); + break; + + case _IOC_NR(SNDRV_SST_STREAM_RESUME): + sst_dbg("SNDRV_SST_IOCTL_RESUME recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = sst_resume_stream(str_id); + break; + + case _IOC_NR(SNDRV_SST_STREAM_SET_PARAMS): { + struct snd_sst_params *str_param = (struct snd_sst_params *)arg; + + sst_dbg("IOCTL_SET_PARAMS recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + sst_print_params(&str_param->sparams); + + if (str_id == 0) { + retval = sst_get_stream(str_param, data->pvt_id); + if (retval > 0) { + data->str_id = retval; + retval = 0; + } else { + if (retval == -SST_ERR_INVALID_PARAMS) + retval = -EINVAL; + } + } else { + sst_dbg("SET_STREAM_PARAMS recieved!\n"); + /*allocated set params only*/ + retval = sst_set_stream_param(str_id, str_param); + /*Block the call for reply*/ + if (retval == 0) { + int sfreq = 0, word_size = 0; + sfreq = str_param->sparams.uc.pcm_params.sfreq; + word_size = str_param->sparams. + uc.pcm_params.pcm_wd_sz; + retval = sst_wait_timeout(sst_ops, + &sst_ops->alloc_block[i]); + sst_dbg("SST sampling frequency= %d\n" , sfreq); + sst_ops->scard_ops.set_pcm_params(sfreq, + word_size); + } + } + break; + } + + case _IOC_NR(SNDRV_SST_MMAP_PLAY): + case _IOC_NR(SNDRV_SST_MMAP_CAPTURE): + sst_dbg("SNDRV_SST_MMAP_PLAY/CAPTURE recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = intel_sst_mmap_play_capture(str_id, + (struct snd_sst_buffs *)arg); + break; + + case _IOC_NR(SNDRV_SST_STREAM_DROP): + sst_dbg("SNDRV_SST_IOCTL_DROP recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = sst_drop_stream(str_id); + break; + + case _IOC_NR(SNDRV_SST_STREAM_GET_TSTAMP): { + unsigned long long *ms = (unsigned long long *)arg; + struct snd_sst_tstamp tstamp = {0}; + unsigned long long time = 0, freq = 0; + unsigned long mod = 0; + + sst_dbg("SNDRV_SST_STREAM_GET_TSTAMP recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + memcpy_fromio(&tstamp, + ((void *)(sst_ops->mailbox + SST_TIME_STAMP) + +(str_id * sizeof(tstamp))), + sizeof(tstamp)); + time = tstamp.samples_rendered; + sst_dbg("samples rendered! = 0x%llx\n", time); + freq = (unsigned long long) tstamp.sampling_frequency; + sst_dbg("freq = %llx\n", freq); + mod = do_div(time, freq); + sst_dbg("mod = 0x%lx\n", mod); + sst_dbg("sec = 0x%llx\n", time); + time = time * 1000; + sst_dbg("msec = 0x%llx\n", time); + retval = copy_to_user(ms, &time, sizeof(*ms)); + if (retval != 0) + sst_err("copy failed = %d\n", retval); + break; + } + + case _IOC_NR(SNDRV_SST_STREAM_START):{ + struct stream_info *stream = NULL; + + sst_dbg("SNDRV_SST_STREAM_START recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = sst_validate_strid(str_id); + if (retval != 0) + break; + stream = &sst_ops->streams[str_id]; + mutex_lock(&stream->lock); + if (stream->status == STREAM_INIT) { + sst_dbg("calling play frames...\n"); + stream->prev = stream->status; + stream->status = STREAM_RUNNING; + if (stream->ops == STREAM_OPS_PLAYBACK) { + retval = sst_play_frame(str_id); +#ifdef CONFIG_SST_OSPM_SUPPORT + sst_ospm_send_event( + OSPM_EVENT_SUBSYS_START_PLAY); +#endif + } else if (stream->ops == STREAM_OPS_CAPTURE) + retval = sst_capture_frame(str_id); + else { + sst_err("Invalid ops 0x%x\n", stream->ops); + retval = -EINVAL; + mutex_unlock(&sst_ops->streams[str_id].lock); + break; + } + if (retval < 0) { + sst_err("play/capture frames failed \n"); + stream->status = STREAM_INIT; + mutex_unlock(&sst_ops->streams[str_id].lock); + break; + } + } else { + sst_err("Invalid start for stream %d state 0x%x\n", + str_id, stream->status); + retval = -EINVAL; + } + mutex_unlock(&sst_ops->streams[str_id].lock); + break; + } + case _IOC_NR(SNDRV_SST_SET_TARGET_DEVICE): { + struct snd_sst_target_device *target_device = NULL; + + sst_info("SNDRV_SST_SET_TARGET_PLAYBACK DEVICE recieved!\n"); + target_device = (struct snd_sst_target_device *)arg; + BUG_ON(!target_device); + if (minor != 1) { + sst_err("called for non AM handle minor %d\n", minor); + retval = -EINVAL; + break; + } + retval = sst_target_device_select(target_device); + if (0 != retval) + return retval; + break; + + } + + case _IOC_NR(SNDRV_SST_DRIVER_INFO): { + struct snd_sst_driver_info *info = + (struct snd_sst_driver_info *)arg; + sst_dbg("SNDRV_SST_DRIVER_INFO recived \n"); + info->version = SST_VERSION_NUM; + /*hard coding, shud get sumhow later*/ + info->active_pcm_streams = sst_ops->stream_cnt - + sst_ops->active_cnt; + info->active_enc_streams = sst_ops->active_cnt; + info->max_pcm_streams = MAX_ACTIVE_STREAM - MAX_ENC_STREAM; + info->max_enc_streams = MAX_ENC_STREAM; + info->buf_per_stream = sst_ops->mmap_len; + break; + } + + default: + sst_dbg("IOCTL not supported yet !\n"); + retval = -ENOTTY; + } + sst_dbg("...complete ret code = %d\n", retval); + + return retval; +} + +/* + MAD Routines +*/ + +/** +* sst_control_set - Set Control params +* @control_list: list of controls to be set +* +* This function is called by MID sound card driver to set +* SST/Sound card controls. This is registered with MID driver +*/ +int sst_control_set(int control_element, void *value) +{ + int retval = 0, str_id = 0; + + if (SST_UN_INIT == sst_ops->sst_state) { + /*FW is not downloaded*/ + sst_dbg("DSP Downloading FW now...\n"); + retval = sst_download_fw(); + if (retval != 0) { + sst_err("FW download failed = 0x%x, abort\n", retval); + return retval; + } + } + + switch (control_element) { + case SST_SND_ALLOC: { + struct snd_sst_params *str_param = NULL; + int pcm_id = sst_assign_pvt_id(sst_ops); + + str_param = (struct snd_sst_params *)value; + BUG_ON(!str_param); + sst_print_params(&str_param->sparams); + retval = sst_get_stream(str_param, pcm_id); + if (retval >= 0) + sst_ops->stream_cnt++; +#ifdef CONFIG_SST_OSPM_SUPPORT + if (str_param->ops == STREAM_OPS_PLAYBACK) + sst_ospm_send_event(OSPM_EVENT_SUBSYS_START_PLAY); +#endif + + break; + } + + case SST_SND_PAUSE: + retval = sst_pause_stream(*(int *)value); + break; + + case SST_SND_RESUME: + retval = sst_resume_stream(*(int *)value); + break; + + case SST_SND_DROP: + retval = sst_drop_stream(*(int *)value); + break; + + case SST_SND_FREE: { + struct stream_info *stream = NULL; + + str_id = *(int *)value; + stream = &sst_ops->streams[str_id]; + retval = sst_free_stream(str_id); + if (retval == 0) { + stream->pcm_substream = NULL; + stream->period_elapsed = NULL; + sst_ops->stream_cnt--; + } + break; + } + + case SST_SND_STREAM_INIT: { + struct pcm_stream_info *str_info = NULL; + struct stream_info *stream = NULL; + + sst_dbg("stream init called\n"); + str_info = (struct pcm_stream_info *)value; + str_id = str_info->str_id; + retval = sst_validate_strid(str_id); + if (retval != 0) + break; + + stream = &sst_ops->streams[str_id]; + sst_dbg("setting the period ptrs\n"); + stream->pcm_substream = str_info->mad_substream; + stream->period_elapsed = str_info->period_elapsed; + break; + } + + case SST_SND_BUFFER_POINTER: { + struct pcm_stream_info *buf_ptr; + struct snd_sst_tstamp fw_tstamp = {0,}; + struct stream_info *stream = NULL; + + sst_dbg("buffer pointer query\n"); + + buf_ptr = (struct pcm_stream_info *)value; + str_id = buf_ptr->str_id; + retval = sst_validate_strid(str_id); + if (retval != 0) + break; + stream = &sst_ops->streams[str_id]; + + if (stream->pcm_substream == NULL) + break; + memcpy_fromio(&fw_tstamp, + ((void *)(sst_ops->mailbox + SST_TIME_STAMP) + +(str_id * sizeof(fw_tstamp))), + sizeof(fw_tstamp)); + + sst_dbg("strid = %d\n", str_id); + + if (stream->ops == STREAM_OPS_PLAYBACK) + buf_ptr->buffer_ptr = fw_tstamp.samples_rendered; + else + buf_ptr->buffer_ptr = fw_tstamp.samples_processed; + sst_dbg("samples played = %ld\n", + buf_ptr->buffer_ptr); + break; + } + + default: + /*Illegal case*/ + sst_err("illegal req\n"); + return -EINVAL; + } + sst_dbg("...complete ret code = %d\n", retval); + + return retval; +} + + +/** +* sst_send_data_to_HW - send data buffers +* @buffer_data: user buffer +* +* This function is called by MID sound card driver to send buffer +* to HW. This is registered with MID driver +*/ +int sst_send_buffer_to_HW(int str_id, struct stream_buffer *mad_buf) +{ + /*recvd a buffer map it to stream*/ + /*this is a PCM stream and playback*/ + int retval = 0; + bool flag_add = false; + struct sst_stream_bufs *sst_buf = NULL; + struct stream_info *stream = NULL; + + if (NULL == mad_buf || + 0 == mad_buf->addr || + 0 == mad_buf->length) { + sst_err("Null Ptr or buf size = 0\n"); + return -EINVAL; + } + + if (SST_FW_RUNNING != sst_ops->sst_state) { + sst_err("SST Not runng: %d\n", sst_ops->sst_state); + return -EBUSY; + } + + retval = sst_validate_strid(str_id); + if (retval < 0) + return -EINVAL; + + stream = &sst_ops->streams[str_id]; + sst_dbg("stream status = %d strid=%d\n", stream->status, str_id); + sst_dbg("stream codec = %d, prevstate=%d\n", + stream->codec, stream->prev); + if (stream->status == STREAM_UN_INIT) { + sst_err("BAD REQUEST!\n"); + return -EBADRQC; + } + sst_dbg("received addr=0x%x size = 0x%x\n", + (unsigned int)mad_buf->addr, mad_buf->length); + /*list is not empty*/ + list_for_each_entry(sst_buf, &stream->bufs, node) { + if (sst_buf->in_use == true) + continue; + else if ((int) mad_buf->addr != + (int)sst_buf->addr + sst_buf->size) + continue; + else { + sst_buf->size += mad_buf->length; + flag_add = true; + sst_dbg("inc addr = 0x%p, base = 0x%x inc_val = 0x%x\n", + sst_buf->addr, sst_buf->size, mad_buf->length); + break; + } + } + + if (flag_add == false) { + sst_buf = kzalloc(sizeof(*sst_buf), GFP_ATOMIC); + if (sst_buf == NULL) + return -ENOMEM; + sst_buf->size = mad_buf->length; + sst_buf->addr = (void *)mad_buf->addr; + sst_buf->offset = 0; + sst_buf->in_use = false; + list_add_tail(&sst_buf->node, &stream->bufs); + flag_add = true; + sst_dbg("entry added addr = 0x%x size = 0x%x\n", + (unsigned int)mad_buf->addr, mad_buf->length); + } + + if (stream->status == STREAM_INIT) { + sst_dbg("play/capt frames...\n"); + stream->prev = stream->status; + stream->status = STREAM_RUNNING; + stream->data_blk.on = false; + if (stream->ops == STREAM_OPS_PLAYBACK) + retval = sst_play_frame(str_id); + else if (stream->ops == STREAM_OPS_CAPTURE) + retval = sst_capture_frame(str_id); + else + return -EINVAL; + if (retval < 0) { + sst_err("play/capture frames failed \n"); + return -EIO; + } + } + + return retval; +} + +struct intel_sst_card_ops sst_pmic_ops = { + .control_set = sst_control_set, + .send_buffer = sst_send_buffer_to_HW, +}; + +/** +* register_sst_card- function for sound card to register +*@card: pointer to structure of operations +* This function is called card driver loads and is ready for registration +*/ +int register_sst_card(struct intel_sst_card_ops *card) +{ + + if (NULL == card || + NULL == card->module_name) { + sst_err(" Null Pointer Passed\n"); + return -EINVAL; + } + + if (PMIC_SND_UN_INIT == sst_ops->pmic_state) { + /*register this driver*/ + if (0 == (strncmp(SST_CARD_NAMES, card->module_name, + strlen(SST_CARD_NAMES)))) { + sst_ops->pmic_vendor = card->vendor_id; + sst_ops->scard_ops = *card->scard_ops; + sst_pmic_ops.module_name = card->module_name; + sst_ops->pmic_state = PMIC_SND_INIT_DONE; + card->control_set = sst_pmic_ops.control_set; + card->send_buffer = sst_pmic_ops.send_buffer; + return 0; + } else { + sst_err("strcmp failed %s \n", card->module_name); + return -EINVAL; + } + + } else { + /*already registered a driver*/ + sst_err("Repeat for register..denied\n"); + return -EBADRQC; + } + return 0; +} +EXPORT_SYMBOL_GPL(register_sst_card); + +/** +* unregister_sst_card- function for sound card to un-register +*@card: pointer to structure of operations +* This function is called when card driver unloads +*/ +void unregister_sst_card(struct intel_sst_card_ops *card) +{ + if (sst_pmic_ops.module_name == card->module_name) { + /*unreg*/ + sst_pmic_ops.module_name = ""; + sst_ops->pmic_state = PMIC_SND_UN_INIT; + sst_dbg("Unregistered %s\n", card->module_name); + } + return; +} +EXPORT_SYMBOL_GPL(unregister_sst_card); + +int lpe_mask_periphral_intr(enum lpe_periphral device) +{ + union sst_pimr_reg pimr = {{0},}; + if (sst_ops == NULL) + return -EIO; + + pimr.full = readl(sst_ops->shim + SST_PIMR); + + switch (device) { + case LPE_DMA: + pimr.part.dmac_sc = 1; + writel(pimr.full, sst_ops->shim + SST_PIMR); + break; + + case LPE_SSP0: + break; + + case LPE_SSP1: + break; + + default: + break; + } + return 0; +} +EXPORT_SYMBOL_GPL(lpe_mask_periphral_intr); + +int lpe_unmask_periphral_intr(enum lpe_periphral device) +{ + union sst_pimr_reg pimr = {{0},}; + if (sst_ops == NULL) + return -EIO; + + pimr.full = readl(sst_ops->shim + SST_PIMR); + + switch (device) { + case LPE_DMA: + pimr.part.dmac_sc = 0; + writel(pimr.full, sst_ops->shim + SST_PIMR); + break; + + case LPE_SSP0: + break; + + case LPE_SSP1: + break; + + default: + break; + } + return 0; + +} +EXPORT_SYMBOL_GPL(lpe_unmask_periphral_intr); + +int lpe_periphral_intr_status(enum lpe_periphral device, int *status) +{ + union sst_pisr_reg pisr = {{0},}; + if (sst_ops == NULL) + return -EIO; + + pisr.full = readl(sst_ops->shim + SST_PISR); + + switch (device) { + case LPE_DMA: + *status = pisr.part.dmac; + break; + + case LPE_SSP0: + break; + + case LPE_SSP1: + break; + + default: + break; + } + return 0; +} +EXPORT_SYMBOL_GPL(lpe_periphral_intr_status); -- 1.5.4.5

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[alsa-devel] [PATCH] [RFC 0/13] Intel SST drivers
by Koul, Vinod 07 Jul '09

07 Jul '09

Following emails are a of series 13 patches for Intel SST drivers (including documentation). These drivers are for the audio engine/sound card in upcoming Intel based Mobile Internet Devices (MID) platform There are 2 drivers involved. One is for the audio DSP engine that can decode and do audio processing and one is for the ALSA sound card driver with support for three different sound cards that are supported on the platform. The dependant drivers for the platform are not yet available in the upstream. So please consider these patches only for RFC and provide your comments/suggestions. The audio firmware binary associated with this platform that is downloaded by the driver at runtime will be shared in a public location and made available in firmware git when we release the drivers for up streaming. The patches are compiled with 0 warnings and checkpatch-ed. The first patch in the series [RFC 1/13] Intel SST driver's documentation gives the documentation about the drivers and the interface for the audio DSP driver. The rest of the patches that follow have a brief description about the contents. Thanks -Vinod Koul & Harsha Priya

4 4

[alsa-devel] [PATCH] [RFC 2/13] Intel SST driver
by Vinod Koul 07 Jul '09

07 Jul '09

This adds the SST driver for the SST DSP engine. This patch adds the main file intel_sst.c which contains the init, exit probe, interrupt routine and PCI suspend/resume implementation. This file also implements the SST Firmware initialization and firmware and codec libraries download steps Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intel_sst.c --- sound/pci/sst/intel_sst.c | 877 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 877 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intel_sst.c diff --git a/sound/pci/sst/intel_sst.c b/sound/pci/sst/intel_sst.c new file mode 100644 index 0000000..bd30833 --- /dev/null +++ b/sound/pci/sst/intel_sst.c @@ -0,0 +1,877 @@ +/* + * intel_sst.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This driver exposes the audio engine functionalities to the ALSA + * and middleware. + * + * This file contains all init functions + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/fs.h> +#include <linux/file.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/workqueue.h> +#include <linux/firmware.h> +#include <linux/mutex.h> +#include <linux/delay.h> +#include <linux/sched.h> +#ifndef CONFIG_SST_IPC_NOT_INCLUDED +#include <asm/ipc_defs.h> +#endif +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#ifdef CONFIG_SST_OSPM_SUPPORT +#include <linux/intel_mid.h> +#endif + + +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_DESCRIPTION("Intel (R) Moorestown Audio Engine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_VERSION(SST_DRIVER_VERSION); + +struct intel_sst_drv *sst_ops; + +void sst_reset(void) +{ + int retval; + + /*Disable Audio Core clock and Audio Fabric clock*/ + retval = sst_scu_ipc_write(0xff11d83c, 0x80008008); + if (retval != 0) + sst_err("scu ipc write1 failed %d", retval); + /*Reset the Audio subsystem*/ + retval = sst_scu_ipc_write(0xff11d118, 0x7ffffcff); + if (retval != 0) + sst_err("scu ipc write2 failed %d", retval); + /*Bring it out of Audio subsystem reset*/ + retval = sst_scu_ipc_write(0xff11d118, 0x7fffffff); + if (retval != 0) + sst_err("scu ipc write3 failed %d", retval); + /*Enable fabric clock at 50MHz*/ + retval = sst_scu_ipc_write(0xff11d83c, 0x80008301); + if (retval != 0) + sst_err("scu ipc write4 failed %d", retval); + /*Write to the Shim register for ratio 1:1*/ + retval = sst_scu_ipc_write(0xffae8000, 0x382); + if (retval != 0) + sst_err("scu ipc write5 failed %d", retval); + /*Enable the core clock at 1:2*/ + retval = sst_scu_ipc_write(0xff11d83c, 0x80000301); + if (retval != 0) + sst_err("scu ipc write6 failed %d", retval); + return; +} + +void sst_start(void) +{ + union config_status_reg csr; + int retval; + + csr.full = readl(sst_ops->shim + SST_CSR); + csr.part.sst_reset = 0; + csr.part.run_stall = 0; + csr.part.bypass = 0; + /*csr.full = 0x1800;*/ + sst_dbg("Setting SST to execute 0x%x \n", csr.full); + + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) + sst_err("scu ipc write start failed %d", retval); + return; +} +int sst_parse_module(struct fw_module_header *module) +{ + struct dma_block_info *block = NULL; + u32 count; + void __iomem *ram; + + sst_dbg("module sign=%s sz=0x%x blks=0x%x typ=0x%x ep=0x%x sz=0x%x\n", + module->signature, module->mod_size, + module->blocks, module->type, + module->entry_point, sizeof(*module)); + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + if (0 >= block->size) { + sst_err("block size invalid\n"); + return -EINVAL; + } + switch (block->type) { + case SST_IRAM: + ram = sst_ops->iram; + break; + case SST_DRAM: + ram = sst_ops->dram; + break; + default: + sst_err("wrong ram type 0x%x in block 0x%x-abort\n", + block->type, count); + return -EINVAL; + } + memcpy_toio(ram + block->ram_offset, + (void *)block + sizeof(*block), block->size); + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +/** +* sst_parse_fw_image - FW parse and load +* @fw: Pointer to loaded FW +* +* This function is called to parse and download the FW image +*/ +int sst_parse_fw_image(const struct firmware *sst_fw) +{ + struct fw_header *header = NULL; + u32 count; + int ret_val = 0; + struct fw_module_header *module = NULL; + + BUG_ON(!sst_fw); + + /*Read the header information from the data pointer*/ + header = (struct fw_header *)sst_fw->data; + + /*verify FW*/ + if ((0 != strncmp(header->signature, SST_FW_SIGN, 4)) || + (sst_fw->size != header->file_size + sizeof(*header))) { + /*Invalid FW signature*/ + sst_err("Invalid FW sign/file size mismatch\n"); + return -EINVAL; + } + sst_dbg("header sign=%s size=0x%x modules=0x%x fmt=0x%x size=0x%x\n", + header->signature, header->file_size, header->modules, + header->file_format, sizeof(*header)); + module = (void *)sst_fw->data + sizeof(*header); + for (count = 0; count < header->modules; count++) { + /*module*/ + ret_val = sst_parse_module(module); + if (0 != ret_val) + return ret_val; + module = (void *)module + sizeof(*module) + module->mod_size ; + } + + sst_dbg("done....\n"); + return 0; +} + +/** +* sst_load_fw - function to reset FW +* @fw: Pointer to loaded FW +* This function is called when the FW is loaded +*/ +void sst_load_fw(const struct firmware *fw, void *context) +{ + int ret; + + sst_dbg("called \n"); + BUG_ON(!fw); + + sst_reset(); + /*putting the sst state to init*/ + sst_ops->sst_state = SST_UN_INIT; + + ret = sst_parse_fw_image(fw); + if (0 != ret) { + sst_err("error in FW dma\n"); + return; + } + sst_ops->sst_state = SST_FW_LOADED; + /* 7. ask scu to reset the bypass bits*/ + /* 8.bring sst out of reset */ + sst_start(); + sst_dbg("...successful!!!\n"); + return; +} + +int sst_download_library(const struct firmware *fw_lib, + struct snd_sst_lib_download_info *lib) +{ + /*send IPC message and wait*/ + unsigned int i, pvt_id; + struct ipc_post *msg = NULL; + union config_status_reg csr; + struct snd_sst_str_type str_type = {0}; + int retval = 0; + + if (sst_create_large_msg(&msg) != 0) + return -ENOMEM; + + pvt_id = sst_assign_pvt_id(sst_ops); + i = sst_get_block_stream(sst_ops); + sst_dbg("alloc block allocated = %d, pvt_id %d\n", i, pvt_id); + if (i < 0) { + kfree(msg); + return -ENOMEM; + } + sst_ops->alloc_block[i].sst_id = pvt_id; + sst_fill_header(&msg->header, IPC_IA_PREP_LIB_DNLD, 1, 0); + msg->header.part.data = sizeof(u32) + sizeof(str_type); + str_type.codec_type = lib->dload_lib.lib_info.lib_type; + str_type.pvt_id = pvt_id; + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), &str_type, sizeof(str_type)); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + retval = sst_wait_timeout(sst_ops, &sst_ops->alloc_block[i]); + if (retval != 0) { + /*error*/ + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("Prep codec downloaded failed %d\n", retval); + return -EIO; + } + sst_dbg("FW responded, ready for download now...\n"); + /*downloading on success*/ + sst_ops->sst_state = SST_FW_LOADED; + csr.full = readl(sst_ops->shim + SST_CSR); + sst_dbg("CSR reg 0x%x \n", csr.full); + csr.part.run_stall = 1; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("HALT CSR reg setting to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) { + /*error*/ + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("IPC failed to Halt SST 0x%x\n", retval); + return -EAGAIN; + } + csr.full = readl(sst_ops->shim + SST_CSR); + csr.part.bypass = 0x7; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Bypass CSR reg setting to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) { + /*error*/ + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("IPC failed to Bypass SST 0x%x\n", retval); + csr.part.bypass = 0x0; + /*bring LPE out of run stall*/ + csr.part.run_stall = 0x0; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Bypass CSR reg setting to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, + csr.full); + if (retval != 0) + sst_ops->sst_state = SST_UN_INIT; + return -EAGAIN; + } + sst_parse_fw_image(fw_lib); + + /*set the FW to running again*/ + csr.full = readl(sst_ops->shim + SST_CSR); + csr.part.bypass = 0x0; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Bypass CSR reg clearing to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) { + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("Bypass CSR reg clear failed 0x%x \n", retval); + /*bring LPE out of run stall*/ + csr.part.run_stall = 0x0; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Bypass CSR reg setting to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, + csr.full); + if (retval != 0) + sst_ops->sst_state = SST_UN_INIT; + return -EAGAIN; + } + + csr.full = readl(sst_ops->shim + SST_CSR); + csr.part.run_stall = 0; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Stalll CSR reg clearing to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) { + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("Stall CSR reg clear failed 0x%x \n", retval); + if (retval != 0) + sst_ops->sst_state = SST_UN_INIT; + return -EAGAIN; + } + /*send download complete and wait*/ + if (sst_create_large_msg(&msg) != 0) { + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + return -ENOMEM; + } + + sst_fill_header(&msg->header, IPC_IA_LIB_DNLD_CMPLT, 1, 0); + msg->header.part.data = sizeof(u32) + sizeof(*lib); + lib->pvt_id = pvt_id; + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), lib, sizeof(*lib)); + + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + sst_dbg("Waiting for FW to respond on Download complete \n"); + sst_ops->alloc_block[i].ops_block.condition = false; + retval = sst_wait_timeout(sst_ops, &sst_ops->alloc_block[i]); + if (retval != 0) { + /*error*/ + sst_ops->sst_state = SST_FW_RUNNING; + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + return -EIO; + } + + sst_dbg("FW responded sucess on Download complete \n"); + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_ops->sst_state = SST_FW_RUNNING; + return 0; + +} + +int sst_validate_library(const struct firmware *fw_lib, + struct lib_slot_info *slot, + u32 *entry_point) +{ + struct fw_header *header = NULL; + struct fw_module_header *module = NULL; + struct dma_block_info *block = NULL; + unsigned int n_blk, isize = 0, dsize = 0; + int err = 0; + + header = (struct fw_header *)fw_lib->data; + if (header->modules != 1) { + sst_err("Module no mismatch found\n"); + err = -EINVAL; + goto exit; + } + module = (void *)fw_lib->data + sizeof(*header); + *entry_point = module->entry_point; + sst_dbg("Module entry point 0x%x \n", *entry_point); + sst_dbg("Module Sign %s, Size 0x%x, Blocks 0x%x Type 0x%x \n", + module->signature, module->mod_size, + module->blocks, module->type); + + block = (void *)module + sizeof(*module); + for (n_blk = 0; n_blk < module->blocks; n_blk++) { + switch (block->type) { + case SST_IRAM: + isize += block->size; + break; + case SST_DRAM: + dsize += block->size; + break; + default: + sst_err("Invalid block type found for 0x%x\n", n_blk); + err = -EINVAL; + goto exit; + } + block = (void *)block + sizeof(*block) + block->size; + } + if (isize > slot->iram_size || dsize > slot->dram_size) { + sst_err("libarary exceeds size allocated \n"); + err = -EINVAL; + goto exit; + } else + sst_dbg("Library is safe for download...\n"); + + sst_dbg("iram 0x%x, dram 0x%x, allowed iram 0x%x, allowed dram 0x%x\n", + isize, dsize, slot->iram_size, slot->dram_size); +exit: + return err; + +} + +int sst_load_library(struct snd_sst_lib_download *lib, u8 ops, u32 pvt_id) +{ + char buf[20]; + int len = 0, error = 0; + u32 entry_point; + const struct firmware *fw_lib; + struct snd_sst_lib_download_info dload_info = {{{0},},}; + + memset(buf, 0, sizeof(buf)); + + sst_dbg("Lib Type 0x%x, Slot 0x%x, ops 0x%x \n", + lib->lib_info.lib_type, lib->slot_info.slot_num, ops); + sst_dbg("Version 0x%x, name %s, caps 0x%x media type 0x%x \n", + lib->lib_info.lib_version, lib->lib_info.lib_name, + lib->lib_info.lib_caps, lib->lib_info.media_type); + + sst_dbg("IRAM Size 0x%x, offset 0x%x, DRAM Size 0x%x, offset 0x%x \n", + lib->slot_info.iram_size, lib->slot_info.iram_offset, + lib->slot_info.dram_size, lib->slot_info.dram_offset); + + switch (lib->lib_info.lib_type) { + case SST_CODEC_TYPE_MP3: + len += snprintf(buf + len, sizeof(buf) - len, "mp3_"); + break; + case SST_CODEC_TYPE_AAC: + len += snprintf(buf + len, sizeof(buf) - len, "aac_"); + break; + case SST_CODEC_TYPE_WMA9: + len += snprintf(buf + len, sizeof(buf) - len, "wma9_"); + break; + default: + sst_err("Invalid codec type \n"); + error = -EINVAL; + goto wake; + } + + if (ops == STREAM_OPS_CAPTURE) + len += snprintf(buf + len, sizeof(buf) - len, "enc_"); + else + len += snprintf(buf + len, sizeof(buf) - len, "dec_"); + + len += snprintf(buf + len, sizeof(buf) - len, "%d", + lib->slot_info.slot_num); + len += snprintf(buf + len, sizeof(buf) - len, ".bin"); + + sst_dbg("Requesting %s \n", buf); + + error = request_firmware(&fw_lib, buf, &sst_ops->pci->dev); + if (0 != error) { + sst_err("library load failed %d \n", error); + goto wake; + } + error = sst_validate_library(fw_lib, &lib->slot_info, &entry_point); + if (0 != error) { + sst_err("validate lib failed %d \n", error); + goto wake_free; + } + lib->mod_entry_pt = entry_point; + memcpy(&dload_info.dload_lib, lib, sizeof(*lib)); + error = sst_download_library(fw_lib, &dload_info); + if (0 != error) { + sst_err("download lib failed %d \n", error); + goto wake_free; + } + /*lib is downloaded and init send alloc again*/ + sst_dbg("Library is downloaded now... \n"); +wake_free: + /*sst_wake_up_alloc_block(sst_ops, pvt_id, error, NULL);*/ + release_firmware(fw_lib); +wake: + return error; +} + +/*fops Routines*/ +const struct file_operations intel_sst_fops = { + .owner = THIS_MODULE, + .open = intel_sst_open, + .release = intel_sst_release, + .read = intel_sst_read, + .write = intel_sst_write, + .ioctl = intel_sst_ioctl, + .mmap = intel_sst_mmap, + .aio_read = intel_sst_aio_read, + .aio_write = intel_sst_aio_write, +}; + +/* + ISR Routines +*/ +/** +* intel_sst_interrupt - Interrupt service routine for SST +* @irq: irq number of interrupt +* @dev_id: pointer to device structre +* +* This function is called by OS when SST device raises +* an interrupt. This will be result of write in IPC register +* Source can be busy or done interrupt +*/ +static irqreturn_t intel_sst_interrupt(int irq, void *context) +{ + union interrupt_reg isr; + union ipc_header header; + union interrupt_reg imr; + struct intel_sst_drv *drv = (struct intel_sst_drv *) context; + unsigned int size = 0; + + /*Interrupt arrived, check src*/ + isr.full = readl(drv->shim + SST_ISRX); + imr.full = readl(drv->shim + SST_IMRX); + if (1 == isr.part.busy_interrupt) { + header.full = readl(drv->shim + SST_IPCD); + if (1 == header.part.large) + size = header.part.data; + if (0 != (header.part.msg_id & REPLY_MSG)) { + sst_ops->ipc_process_msg.header = header; + memcpy_fromio(sst_ops->ipc_process_msg.mailbox, + drv->mailbox + SST_MAILBOX_RCV, size); + schedule_work(&sst_ops->ipc_process_msg.wq); + } else { + sst_ops->ipc_process_reply.header = header; + memcpy_fromio(sst_ops->ipc_process_reply.mailbox, + drv->mailbox + SST_MAILBOX_RCV, size); + schedule_work(&sst_ops->ipc_process_reply.wq); + } + /*mask busy inetrrupt*/ + imr.part.busy_interrupt = 1; + writel(imr.full, drv->shim + SST_IMRX); + + } else if (1 == isr.part.done_interrupt) { + /*Clear done bit*/ + header.full = readl(drv->shim + SST_IPCX); + header.part.done = 0; + writel(header.full, drv->shim + SST_IPCX); + /* write 1 to clear status register*/; + isr.part.done_interrupt = 1; + writel(isr.full, drv->shim + SST_ISRX); + schedule_work(&sst_ops->ipc_post_msg.wq); + } + return IRQ_HANDLED; +} + + +/* + PCI Routines +*/ +static struct pci_device_id intel_sst_ids[] = { + { 0x8086, 0x080A, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, intel_sst_ids); + + +/* +* intel_sst_probe - PCI probe function +* @pci: PCI device structure +* @pci_id: PCI device ID structure +* +* This function is called by OS when a device is found +* This enables the device, interrupt etc +*/ +static int __devinit intel_sst_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + int ret = 0; + u32 bar = 0, size = 0; + + /*Init the device*/ + ret = pci_enable_device(pci); + if (0 != ret) { + sst_err("device cant be enabled\n"); + return ret; + } + sst_ops->pci = pci; + /*map registers*/ + /*SST Shim*/ + bar = pci_resource_start(pci, 1); + size = pci_resource_len(pci, 1); + ret = pci_request_region(pci, 2, SST_DRV_NAME); + if (ret != 0) + goto err_1; + sst_ops->shim_phy_add = bar; + sst_ops->shim = ioremap_nocache(bar, size); + if (sst_ops->shim == NULL) + goto err_2; + sst_dbg("SST Shim 0x%x,Size 0x%x,Ptr %p \n", bar, size, sst_ops->shim); + + /*Shared SRAM*/ + bar = pci_resource_start(pci, 2); + size = pci_resource_len(pci, 2); + ret = pci_request_region(pci, 3, SST_DRV_NAME); + if (ret != 0) + goto err_2; + sst_ops->mailbox = ioremap_nocache(bar, size); + if (NULL == sst_ops->mailbox) + goto err_3; + sst_dbg("SRAM 0x%x, Size 0x%x, Ptr %p \n", bar, size, sst_ops->mailbox); + + /* IRAM*/ + bar = pci_resource_start(pci, 3); + size = pci_resource_len(pci, 3); + ret = pci_request_region(pci, 4, SST_DRV_NAME); + if (ret != 0) + goto err_3; + sst_ops->iram = ioremap_nocache(bar, size); + if (sst_ops->iram == NULL) + goto err_4; + sst_dbg("IRAM 0x%x, Size 0x%x, Ptr %p \n", bar, size, sst_ops->iram); + + /*DRAM*/ + bar = pci_resource_start(pci, 4); + size = pci_resource_len(pci, 4); + ret = pci_request_region(pci, 5, SST_DRV_NAME); + if (ret != 0) + goto err_4; + sst_ops->dram = ioremap_nocache(bar, size); + if (sst_ops->dram == NULL) + goto err_5; + sst_dbg("DRAM 0x%x, Size 0x%x, Ptr %p \n", bar, size, sst_ops->dram); + + /*Register the ISR*/ + ret = request_irq(pci->irq, intel_sst_interrupt, + IRQF_SHARED, SST_DRV_NAME, sst_ops); + if (0 != ret) + goto err_5; + sst_dbg("Registered IRQ 0x%x\n", pci->irq); + sst_ops->sst_state = SST_UN_INIT; + + /* Register with /dev */ + ret = register_chrdev(INTEL_SST_MAJOR, SST_DRV_NAME, &intel_sst_fops); + if (0 != ret) { + sst_err("couldn't register device number\n"); + goto err_6; + } + + sst_dbg("...successfully done!!!\n"); + return 0; + +err_6: free_irq(pci->irq, sst_ops); +err_5: iounmap(sst_ops->dram); +err_4: iounmap(sst_ops->iram); +err_3: iounmap(sst_ops->mailbox); +err_2: iounmap(sst_ops->shim); +err_1: sst_err("Probe failed with 0x%x \n", ret); + return ret; +} + +/** +* intel_sst_remove - PCI remove function +* @pci: PCI device structure +* +* This function is called by OS when a device is unloaded +* This frees the interrupt etc +*/ +static void __devexit intel_sst_remove(struct pci_dev *pci) +{ + sst_ops->sst_state = SST_UN_INIT; + unregister_chrdev(INTEL_SST_MAJOR, SST_DRV_NAME); + free_irq(pci->irq, sst_ops); + iounmap(sst_ops->dram); + iounmap(sst_ops->iram); + iounmap(sst_ops->mailbox); + iounmap(sst_ops->shim); +} + +#ifdef CONFIG_SST_OSPM_SUPPORT +/* + Power Management + +*/ +/** +* intel_sst_suspend - PCI suspend function +* @pci: PCI device structure +* @state: PM message +* +* This function is called by OS when a power event occurs +*/ +static int intel_sst_suspend(struct pci_dev *pci, pm_message_t state) +{ + int i = 0; + sst_dbg("intel_sst_suspend called\n"); + + /*Pause all running streams*/ + for (i = 1; i < MAX_NUM_STREAMS; i++) { + if (sst_ops->streams[i].status == STREAM_RUNNING) { + sst_ops->active_streams[i] = true; + sst_pause_stream(i); + } else + sst_ops->active_streams[i] = false; + } + + pci_set_drvdata(pci, sst_ops); + + /*Disable everything*/ + /*free_irq(pci->irq, sst_ops);*/ + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +/** +* intel_sst_resume - PCI resume function +* @pci: PCI device structure +* @state: PM message +* +* This function is called by OS when a power event occurs +*/ +static int intel_sst_resume(struct pci_dev *pci) +{ + int i = 0; + + sst_dbg("\nintel_sst_resume called\n"); + if (pci != NULL) + sst_dbg("pci is not null \n"); + /*Enable*/ + sst_ops = pci_get_drvdata(pci); + if (sst_ops != NULL) + sst_dbg("sst_ops is not null\n"); + if (pci->irq) + sst_dbg("irq %d \n", pci->irq); + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + /*ret = request_irq(pci->irq, intel_sst_interrupt, + IRQF_SHARED, "intel_sst_engine", sst_ops); + if (0 != ret) { + sst_err("\tIRQ %d reg error %d\n", pci->irq, ret); + return ret; + }*/ + + /*Start all paused streams*/ + for (i = 1; i < MAX_NUM_STREAMS; i++) { + if (sst_ops->active_streams[i] == true) + sst_resume_stream(i); + } + return 0; +} +/** +* sst_ospm_send_event - Send PM events to OSPM +* @event: OSPM Event +* +* This function is called when stream is started/stopped/paused... +*/ +int sst_ospm_send_event(int event) +{ + return ospm_generate_netlink_event(AUDIO_SUBSYTEM_ID, event); + +} +#endif /* CONFIG_SST_OSPM_SUPPORT */ +static struct pci_driver driver = { + .name = SST_DRV_NAME, + .id_table = intel_sst_ids, + .probe = intel_sst_probe, + .remove = __devexit_p(intel_sst_remove), +#ifdef CONFIG_SST_OSPM_SUPPORT + .suspend = intel_sst_suspend, + .resume = intel_sst_resume, +#endif +}; + +/** +* intel_sst_init - Module init function +* +* Registers with PCI +* Registers with /dev +*Init all data strutures +*/ +/*Module init function*/ +static int __init intel_sst_init(void) +{ + /* + Init all variables, data structure etc.... + */ + int ret = 0, i = 0; + struct stream_info *stream = NULL; + + sst_info("******** SST DRIVER loading.. Ver: %s\n", SST_DRIVER_VERSION); + + sst_ops = kzalloc(sizeof(*sst_ops), GFP_KERNEL); + if (NULL == sst_ops) { + sst_err("intel_sst malloc fail\n"); + return -ENOMEM; + } + + sst_ops->pmic_state = PMIC_SND_UN_INIT; + sst_ops->stream_cnt = 0; + sst_ops->active_cnt = 0; + sst_ops->am_cnt = 0; + sst_ops->unique_id = 0; + + INIT_LIST_HEAD(&sst_ops->ipc_dispatch_list); + INIT_WORK(&sst_ops->ipc_post_msg.wq, sst_post_message); + INIT_WORK(&sst_ops->ipc_process_msg.wq, sst_process_message); + INIT_WORK(&sst_ops->ipc_process_reply.wq, sst_process_reply); + init_waitqueue_head(&sst_ops->wait_queue); + + for (i = 0; i < MAX_ACTIVE_STREAM; i++) { + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_ops->alloc_block[i].ops_block.condition = false; + } + mutex_init(&sst_ops->list_lock); + + for (i = 1; i < MAX_NUM_STREAMS; i++) { + stream = &sst_ops->streams[i]; + INIT_LIST_HEAD(&stream->bufs); + mutex_init(&stream->lock); + } + sst_ops->mmap_mem = NULL; + sst_ops->mmap_len = SST_MMAP_PAGES * PAGE_SIZE; + while (sst_ops->mmap_len > 0) { + sst_ops->mmap_mem = kzalloc(sst_ops->mmap_len, GFP_KERNEL); + if (sst_ops->mmap_mem != NULL) { + sst_dbg("Got memory %p size 0x%x \n", sst_ops->mmap_mem, + sst_ops->mmap_len); + break; + } + sst_ops->mmap_len -= (SST_MMAP_STEP * PAGE_SIZE); + if (sst_ops->mmap_len <= 0) { + sst_err("Couldnt get any mem...abort!!\n"); + goto exit_p; + } + sst_dbg("Failed...trying %d\n", sst_ops->mmap_len); + } + /* Register with PCI */ + ret = pci_register_driver(&driver); + if (0 != ret) { + sst_err("PCI register failed, unregister char drv\n"); + goto exit; + } + + + + sst_dbg("...sucessful\n"); + return 0; + +exit: + pci_unregister_driver(&driver); +exit_p: + kfree(sst_ops->mmap_mem); + kfree(sst_ops); + sst_err("Error in init %d\n", ret); + + return ret; +} + +/*Module exit function*/ +static void __exit intel_sst_exit(void) +{ + int i; + struct stream_info *stream = NULL; + + for (i = 0; i < MAX_NUM_STREAMS; i++) { + stream = &sst_ops->streams[i]; + sst_free_stream(i); + } + sst_ops->pmic_state = PMIC_SND_UN_INIT; + pci_unregister_driver(&driver); + + flush_scheduled_work(); + kfree(sst_ops->mmap_mem); + kfree(sst_ops); + + sst_dbg("...unloaded\n"); + return; +} + +module_init(intel_sst_init); +module_exit(intel_sst_exit); -- 1.5.4.5

3 4

[alsa-devel] [PATCH] [RFC 4/13] Intel SST driver common private headers
by Vinod Koul 07 Jul '09

07 Jul '09

This patch adds header files private to the SST driver and contains the common structures like driver context, DSP register definitions, and inline utility functions used by the SST driver and function prototypes of common functions that are implemented in SST driver Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intel_sst_common.h new file: sound/pci/sst/intel_sst_pvt.h --- sound/pci/sst/intel_sst_common.h | 292 +++++++++++++++++++++++++++++ sound/pci/sst/intel_sst_pvt.h | 378 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 670 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intel_sst_common.h create mode 100644 sound/pci/sst/intel_sst_pvt.h diff --git a/sound/pci/sst/intel_sst_common.h b/sound/pci/sst/intel_sst_common.h new file mode 100644 index 0000000..e039489 --- /dev/null +++ b/sound/pci/sst/intel_sst_common.h @@ -0,0 +1,292 @@ +#ifndef __INTEL_SST_COMMON_H__ +#define __INTEL_SST_COMMON_H__ +/* + * intel_sst_common.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * Common private declarations for SST + */ +#include <linux/time.h> + +/*#define SND_LOOP_TEST*/ + +#define SST_DRIVER_VERSION "0.04.002" +#define SST_VERSION_NUM 0x0402 + +/* +driver names +*/ +#define SST_DRV_NAME "intel_sst_driver" +#define SST_FW_STD_FILENAME "fw_sst.bin" + + +/* +PMIC and SST hardware states +*/ +enum sst_pmic_states { + PMIC_SND_UN_INIT = 0, + PMIC_SND_INIT_DONE, +}; + +enum sst_states { + SST_FW_LOADED = 1, + SST_FW_RUNNING, + SST_UN_INIT, + SST_ERROR, +}; + +#define MAX_ACTIVE_STREAM 3 +#define MAX_ENC_STREAM 1 +#define MAX_AM_HANDLES 1 +#define ALLOC_TIMEOUT 5000 +/* +SST numbers +*/ +#define SST_BLOCK_TIMEOUT 5000 + +#define INTEL_SST_MAJOR 255 +#define BLOCK_UNINIT -1 + +/*SST register map*/ +#define SST_CSR 0x00 +#define SST_PISR 0x08 +#define SST_PIMR 0x10 +#define SST_ISRX 0x18 +#define SST_IMRX 0x28 +#define SST_IPCX 0x38 /*IPC IA-SST*/ +#define SST_IPCD 0x40 /*IPC SST-IA*/ +#define SST_SHIM_SIZE 0X44 + +#define FW_SIGNATURE_SIZE 4 + +/* +stream states +*/ +enum sst_stream_states { + STREAM_UN_INIT = 0, /*Freed/Not used stream*/ + STREAM_RUNNING = 1, /*Running*/ + STREAM_PAUSED = 2, /*Paused stream*/ + STREAM_INIT_WT = 4, /*stream is init and waiting for FW*/ + STREAM_INIT = 5, /*stream init, waiting for data*/ +}; + + +enum sst_ram_type{ + SST_IRAM = 1, + SST_DRAM = 2, +}; +/* +SST shim registers to structure mapping +*/ +union config_status_reg { + struct { + u32 rsvd0:1; + u32 sst_reset:1; + u32 hw_rsvd:3; + u32 sst_clk:2; + u32 bypass:3; + u32 run_stall:1; + u32 rsvd:21; + } part; + u32 full; +}; + +union interrupt_reg { + struct { + u32 done_interrupt:1; + u32 busy_interrupt:1; + u32 rsvd:30; + } part; + u32 full; +}; + +union sst_pisr_reg { + struct { + u32 pssp0:1; + u32 pssp1:1; + u32 rsvd0:3; + u32 dmac:1; + u32 rsvd1:26; + } part; + u32 full; +}; + +union sst_pimr_reg { + struct { + u32 ssp0:1; + u32 ssp1:1; + u32 rsvd0:3; + u32 dmac:1; + u32 rsvd1:10; + u32 ssp0_sc:1; + u32 ssp1_sc:1; + u32 rsvd2:3; + u32 dmac_sc:1; + u32 rsvd3:10; + } part; + u32 full; +}; + + +struct sst_stream_bufs { + struct list_head node; + u32 size; + const char *addr; + u32 data_copied; + bool in_use; + u32 offset; +}; + +struct snd_sst_user_cap_list { + unsigned int iov_index; /*index of iov*/ + unsigned long iov_offset; /*offset in iov*/ + unsigned long offset; /*offset in kmem*/ + unsigned long size; /*size copied*/ + struct list_head node; +}; +/* +This structure is used to block a user/fw data call to another +fw/user call +*/ +struct sst_block { + bool condition; /*condition for blocking check*/ + int ret_code; /*ret code when block is released*/ + void *data; /*data to be appsed for block if any*/ + bool on; +}; + +enum snd_sst_buf_type { + SST_BUF_USER_STATIC = 1, + SST_BUF_USER_DYNAMIC, + SST_BUF_MMAP_STATIC, + SST_BUF_MMAP_DYNAMIC, +}; +/* +structure that holds the stream information +*/ +struct stream_info { + unsigned int status; + unsigned int prev; + unsigned int codec; + unsigned int sst_id; + unsigned int ops; + struct list_head bufs; + struct mutex lock; /*mutex*/ + void *get_parameters; + bool mmapped; + unsigned int sg_index; /* current buf Index */ + unsigned char *cur_ptr; /* Current static bufs */ + struct snd_sst_buf_entry *buf_entry; + struct sst_block data_blk; /*stream ops block*/ + struct sst_block ctrl_blk; /*stream control cmd block*/ + enum snd_sst_buf_type buf_type; + void *pcm_substream; + void (*period_elapsed) (void *pcm_substream); +}; + + + +/* +this structure is used for blocking the user's alloc calls to +fw's response to alloc calls +*/ +struct stream_alloc_block { + int sst_id; /*session id of blocked stream*/ + struct sst_block ops_block; /*ops block struture*/ +}; + +#define SST_FW_SIGN "$SST" +#define SST_FW_LIB_SIGN "$LIB" + +/*FW file headers*/ +struct fw_header { + unsigned char signature[FW_SIGNATURE_SIZE]; /*FW signature*/ + u32 file_size; /*size of fw minus this header*/ + u32 modules; /* # of modules*/ + u32 file_format; /*version of header format*/ + u32 reserved[4]; +}; + +struct fw_module_header { + unsigned char signature[FW_SIGNATURE_SIZE]; /*module signature*/ + u32 mod_size; /*size of module*/ + u32 blocks; /*# of blocks*/ + u32 type; /*codec type, pp lib*/ + u32 entry_point; +}; + +struct dma_block_info { + enum sst_ram_type type; /*IRAM/DRAM*/ + u32 size; /*Bytes*/ + u32 ram_offset; /*Offset in I/DRAM*/ + u32 rsvd; /*Reserved field*/ +}; + +struct ioctl_pvt_data { + int str_id; + int pvt_id; +}; + +struct sst_ipc_msg_wq { + union ipc_header header; + char mailbox[SST_MAILBOX_SIZE]; + struct work_struct wq; +}; +#define SST_MMAP_PAGES (640*1024 / PAGE_SIZE) +#define SST_MMAP_STEP (40*1024 / PAGE_SIZE) + +/*driver ops*/ +struct intel_sst_drv{ + bool pmic_state; + int pmic_vendor; + int sst_state; + void __iomem *shim; + void __iomem *mailbox; + void __iomem *iram; + void __iomem *dram; + unsigned int shim_phy_add; + struct list_head ipc_dispatch_list; + struct work_struct ipc_post_msg_wq; + struct sst_ipc_msg_wq ipc_process_msg; + struct sst_ipc_msg_wq ipc_process_reply; + struct sst_ipc_msg_wq ipc_post_msg; + wait_queue_head_t wait_queue; + + struct stream_info streams[MAX_NUM_STREAMS]; + struct stream_alloc_block alloc_block[MAX_ACTIVE_STREAM]; + + struct mutex list_lock; /*mutex for IPC list locking*/ + struct snd_pmic_ops scard_ops; + struct pci_dev *pci; + int active_streams[MAX_NUM_STREAMS]; + struct sst_block block_status; + + void *mmap_mem; + unsigned int mmap_len; + unsigned int unique_id; + unsigned int stream_cnt; + unsigned int active_cnt; + unsigned int am_cnt; +}; + +extern struct intel_sst_drv *sst_ops; +#endif /*__INTEL_SST_COMMON_H */ diff --git a/sound/pci/sst/intel_sst_pvt.h b/sound/pci/sst/intel_sst_pvt.h new file mode 100644 index 0000000..a1e786e --- /dev/null +++ b/sound/pci/sst/intel_sst_pvt.h @@ -0,0 +1,378 @@ +#ifndef __INTEL_SST_PVT_H__ +#define __INTEL_SST_PVT_H__ +/* + * intel_sst_pvt.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file defines all pvt functions + */ +int sst_alloc_stream(char *params, int stream_ops, int codec, int session_id); +int sst_alloc_stream_response(unsigned int str_id, + struct snd_sst_str_type *type); +int sst_pause_stream(int id); +int sst_resume_stream(int id); +int sst_drop_stream(int id); +int sst_free_stream(int id); +int sst_play_frame(int streamID); +int sst_capture_frame(int streamID); +int sst_set_stream_param(int streamID, struct snd_sst_params *str_param); +int sst_route_control(int streamID); +int sst_target_device_select(struct snd_sst_target_device *target_device); +int sst_set_pmic_config(int streamID, struct snd_sst_pmic_config *pc); + +void sst_post_message(struct work_struct *work); +void sst_process_message(struct work_struct *work); +void sst_process_reply(struct work_struct *work); +void sst_send_loop_test(int x); + +int intel_sst_ioctl(struct inode *i_node, struct file *file_ptr, + unsigned int cmd, unsigned long arg); +int intel_sst_open(struct inode *i_node, struct file *file_ptr); +int intel_sst_release(struct inode *i_node, struct file *file_ptr); +int intel_sst_read(struct file *file_ptr, char __user *buf, + size_t count, loff_t *ppos); +int intel_sst_write(struct file *file_ptr, const char __user *buf, + size_t count, loff_t *ppos); +int intel_sst_mmap(struct file *fp, struct vm_area_struct *vma); +ssize_t intel_sst_aio_write(struct kiocb *kiocb, const struct iovec *iov, + unsigned long nr_segs, loff_t offset); +ssize_t intel_sst_aio_read(struct kiocb *kiocb, const struct iovec *iov, + unsigned long nr_segs, loff_t offset); + +int sst_ospm_send_event(int event); +void sst_load_fw(const struct firmware *fw, void *context); +int sst_load_library(struct snd_sst_lib_download *lib, u8 ops, u32 pvt_id); + +static inline int sst_scu_ipc_write(u32 addr, u32 value) +{ +#ifndef CONFIG_SST_IPC_NOT_INCLUDED + int retval = 0, retry = 3; + struct ipc_reg_data ipc_reg = {0}; + + ipc_reg.address = addr; + ipc_reg.data = value; + ipc_reg.ioc = 1; + + while (retry) { + retval = mrst_ipc_write32(&ipc_reg); + if (retval == 0) + break; + retry--; + /*error*/ + sst_err("IPC write failed %x\n", retval); + } + return retval; +#else + return 0; +#endif +} +static inline void sst_fill_header(union ipc_header *header, + int msg, int large, int strID) +{ + header->part.msg_id = msg; + header->part.str_id = strID; + header->part.large = large; + header->part.done = 0; + header->part.busy = 1; + header->part.data = 0; +} + +static inline int sst_get_stream_id(struct intel_sst_drv *sst_ops, int pvt_id) +{ + int i; + + for (i = 1; i < MAX_NUM_STREAMS; i++) { + if (pvt_id == sst_ops->streams[i].sst_id) + return i; + } + return -EINVAL; +} + +static inline unsigned int sst_assign_pvt_id(struct intel_sst_drv *sst_ops) +{ + sst_ops->unique_id++; + if (sst_ops->unique_id >= MAX_NUM_STREAMS) + sst_ops->unique_id = 1; + return sst_ops->unique_id; +} + +static inline int sst_get_block_stream(struct intel_sst_drv *sst_ops) +{ + int i; + + for (i = 0; i < MAX_ACTIVE_STREAM; i++) { + if (BLOCK_UNINIT == sst_ops->alloc_block[i].sst_id) { + sst_ops->alloc_block[i].ops_block.condition = false; + sst_ops->alloc_block[i].ops_block.ret_code = 0; + sst_ops->alloc_block[i].sst_id = 0; + break; + } + } + if (MAX_ACTIVE_STREAM == i) { + sst_err("max alloc_stream reached \n"); + i = -EBUSY; /*active stream limit reached*/ + } + return i; +} + + + +static inline void sst_print_hex(unsigned char *buf, unsigned int size) +{ + unsigned int i; + + for (i = 0; i < size; i++) { + sst_dbg("%02x ", buf[i]); + if ((i != 0) && ((i % 8) == 0)) + sst_dbg("\n"); + } +} + +static inline void sst_init_stream(struct stream_info *stream, + int codec, int sst_id, int ops) +{ + stream->status = STREAM_INIT; + stream->prev = STREAM_UN_INIT; + stream->codec = codec; + stream->sst_id = sst_id; + stream->ops = ops; + stream->data_blk.on = false; + stream->data_blk.condition = false; + stream->data_blk.ret_code = 0; + stream->data_blk.data = NULL; + stream->ctrl_blk.on = false; + stream->ctrl_blk.condition = false; + stream->ctrl_blk.ret_code = 0; + stream->ctrl_blk.data = NULL; + stream->mmapped = false; +} + +static inline void sst_clean_stream(struct stream_info *stream) +{ + struct sst_stream_bufs *bufs = NULL, *_bufs = NULL; + stream->status = STREAM_UN_INIT; + stream->prev = STREAM_UN_INIT; + list_for_each_entry_safe(bufs, _bufs, &stream->bufs, node) { + list_del(&bufs->node); + kfree(bufs); + } +} + +static inline int sst_wait_interruptible(struct intel_sst_drv *sst_ops, + struct sst_block *block) +{ + int retval = 0; + + if (!wait_event_interruptible(sst_ops->wait_queue, block->condition)) { + /*event wake*/ + if (0 != block->ret_code) { + sst_err("stream failed %d\n", block->ret_code); + retval = -EBUSY; + } else { + sst_dbg("event up\n"); + retval = 0; + } + } else { + sst_err("signal interrupted\n"); + retval = -EINTR; + } + return retval; + +} + +static inline int sst_wait_interruptible_timeout(struct intel_sst_drv *sst_ops, + struct sst_block *block, int timeout) +{ + int retval = 0; + + sst_dbg("waiting....\n"); + if (wait_event_interruptible_timeout(sst_ops->wait_queue, + block->condition, + msecs_to_jiffies(timeout))) { + /*event wake*/ + sst_dbg("Event wake ...\n"); + if (0 != block->ret_code) + sst_err("stream failed %d\n", block->ret_code); + else + sst_dbg("event up\n"); + retval = block->ret_code; + } else { + sst_err("timeout occured...\n"); + retval = -EBUSY; + } + return retval; + +} + +static inline int sst_wait_timeout(struct intel_sst_drv *sst_ops, + struct stream_alloc_block *block) +{ + int retval = 0; + + /*NOTE: + Observed that FW processes the alloc msg and replies even + before the alloc thread has finished execution*/ + sst_dbg("waiting for %x, condition %x \n", block->sst_id, + block->ops_block.condition); + if (wait_event_interruptible_timeout(sst_ops->wait_queue, + block->ops_block.condition, + msecs_to_jiffies(SST_BLOCK_TIMEOUT))) { + /*event wake*/ + sst_dbg("Event wake ... %x \n", block->ops_block.condition); + /*check return*/ + if (0 > block->ops_block.ret_code) + sst_err("blk failed %d\n", block->ops_block.ret_code); + else + sst_dbg("blk ret: %d\n", block->ops_block.ret_code); + retval = block->ops_block.ret_code; + } else { + sst_err("Wait timed-out %x\n", block->ops_block.condition); + retval = -EBUSY; + } + return retval; + +} + + +static inline int sst_create_large_msg(struct ipc_post **arg) +{ + struct ipc_post *msg; + + msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); + if (NULL == msg) { + sst_err("kzalloc msg failed \n"); + return -ENOMEM; + } + + msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); + if (NULL == msg->mailbox_data) { + kfree(msg); + sst_err("kzalloc mailbox_data failed \n"); + return -ENOMEM; + }; + *arg = msg; + return 0; +} + +static inline int sst_create_short_msg(struct ipc_post **arg) +{ + struct ipc_post *msg; + + msg = kzalloc(sizeof(*msg), GFP_ATOMIC); + if (NULL == msg) { + sst_err("kzalloc msg failed \n"); + return -ENOMEM; + } + msg->mailbox_data = NULL; + *arg = msg; + return 0; +} + +static inline void sst_print_params(struct snd_sst_stream_params *sparam) +{ + switch (sparam->uc.pcm_params.codec) { + case SST_CODEC_TYPE_PCM: + sst_dbg("pcm \n"); + sst_dbg("chan=%d, sfreq = %d, wd_sz = %d \ + brate = %d framesize = %d \ + samples_per_frame = %d \ + period_cnt = %d\n", + sparam->uc.pcm_params.num_chan, + sparam->uc.pcm_params.sfreq, + sparam->uc.pcm_params.pcm_wd_sz, + sparam->uc.pcm_params.brate, + sparam->uc.pcm_params.frame_size, + sparam->uc.pcm_params.samples_per_frame, + sparam->uc.pcm_params.period_count); + break; + + case SST_CODEC_TYPE_MP3: + sst_dbg("mp3 \n"); + sst_dbg("chan=%d, brate=%d, sfreq = %d, wd_sz = %d\n", + sparam->uc.mp3_params.num_chan, + sparam->uc.mp3_params.brate, + sparam->uc.mp3_params.sfreq, + sparam->uc.mp3_params.pcm_wd_sz); + break; + + case SST_CODEC_TYPE_AAC: + sst_dbg("aac \n"); + sst_dbg("chan=%d, brate=%d, sfreq = %d, wd_sz = %d,asrate=%d\n", + sparam->uc.aac_params.num_chan, + sparam->uc.aac_params.brate, + sparam->uc.aac_params.sfreq, + sparam->uc.aac_params.pcm_wd_sz, + sparam->uc.aac_params.aac_srate); + sst_dbg("mpgid=%d profile=%d, aot = %d\n", + sparam->uc.aac_params.mpg_id, + sparam->uc.aac_params.aac_profile, + sparam->uc.aac_params.aot); + break; + case SST_CODEC_TYPE_WMA9: + sst_dbg("wma type \n"); + sst_dbg("chan=%d, brate=%d, sfreq = %d, wd_sz = %d, tag=%d\n", + sparam->uc.wma_params.num_chan, + sparam->uc.wma_params.brate, + sparam->uc.wma_params.sfreq, + sparam->uc.wma_params.pcm_wd_sz, + sparam->uc.wma_params.format_tag); + sst_dbg("mask=%d, b align=%d, enc opt =%d, op align =%d\n", + sparam->uc.wma_params.channel_mask, + sparam->uc.wma_params.block_align, + sparam->uc.wma_params.wma_encode_opt, + sparam->uc.wma_params.op_align); + break; + default: + sst_dbg("other codec 0x%x\n", sparam->uc.pcm_params.codec); + } +} +static inline int sst_validate_strid(int str_id) +{ + if (str_id <= 0 || str_id >= MAX_NUM_STREAMS) { + sst_err("invalid stream id \n"); + return -EINVAL; + } else + return 0; +} + +/*NOTE: status will +ve for good cases and -ve for error ones*/ +#define MAX_STREAM_FIELD 255 +static inline void sst_wake_up_alloc_block(struct intel_sst_drv *sst_ops, + u8 sst_id, int status, void *data) +{ + int i; + + /*Unblock with retval code*/ + for (i = 0; i < MAX_STREAM_FIELD; i++) { + if (sst_id == sst_ops->alloc_block[i].sst_id) { + sst_ops->alloc_block[i].ops_block.condition = true; + sst_ops->alloc_block[i].ops_block.ret_code = status; + sst_ops->alloc_block[i].ops_block.data = data; + sst_dbg("wake id %d, sst_id %d condition %x\n", i, + sst_ops->alloc_block[i].sst_id, + sst_ops->alloc_block[i].ops_block.condition); + wake_up(&sst_ops->wait_queue); + break; + } + } +} +#endif /*__INTEL_SST_PVT_H__*/ -- 1.5.4.5

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