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

[alsa-devel] [HDA Intel/AD1984] Alsa 1.0.14 not working on ThinkPad T61
by joe+alsa＠nahmias.net 06 Jul '07

06 Jul '07

Hello, I'm having trouble getting sound on my ThinkPad T61 working. According to the changelog for alsa-driver 1.0.14, support was added in that version; however, no sound is coming out. I'm running Debian unstable and I'd be happy to test any patches, provide more info, etc... Any help would be appreciated. Thanks, --Joe PS - Here's some debugging info that may be of use: $ uname -srvmo Linux 2.6.21-2-686 #1 SMP Mon Jun 25 20:53:36 UTC 2007 i686 GNU/Linux $ dpkg -l alsa* | grep ^i ii alsa-base 1.0.14-1 ALSA driver configuration files ii alsa-modules-2.6.21-2-686 1.0.14-1+2.6.21-5 ALSA modules for kernel 2.6.21-2-686 ii alsa-source 1.0.14-1 ALSA driver sources ii alsa-utils 1.0.14-1 ALSA utilities $ aumix -q pcm 90, 90 mic 0, 0 pcm2 100, 100 igain 56, 56 $ lsmod | grep snd snd_hda_intel 244952 0 snd_pcm_oss 39680 0 snd_mixer_oss 15840 1 snd_pcm_oss snd_pcm 72292 2 snd_hda_intel,snd_pcm_oss snd_timer 20964 1 snd_pcm snd 49316 5 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer snd_page_alloc 9544 2 snd_hda_intel,snd_pcm soundcore 7424 1 snd $ sudo lspci -vv -s 00:1b.0 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Lenovo Unknown device 20ac Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 21 Region 0: Memory at fe020000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) $ cat /proc/asound/card0/codec#0 Codec: Analog Devices AD1984 Address: 0 Vendor Id: 0x11d41984 Subsystem Id: 0x17aa20d7 Revision Id: 0x100400 Default PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 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=0x27, nsteps=0x27, stepsize=0x05, mute=0 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 3 0x01* 0x08 0x09 Node 0x03 [Audio Output] wcaps 0x405: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x80 0x80] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x405: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0xa3 0xa3] Power: 0x0 Node 0x05 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Power: 0x0 Connection: 1 0x17 Node 0x06 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Power: 0x0 Connection: 1 0x18 Node 0x07 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x22 0x21 Node 0x08 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x0c Node 0x09 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x0d Node 0x0a [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x04 0x21 Node 0x0b [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x0f 0x21 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x1e 0x1e] Connection: 5 0x14* 0x15 0x16 0x20 0x25 Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 5 0x14* 0x15 0x16 0x20 0x25 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03 0x04* Node 0x0f [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x10 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x00] Node 0x11 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081f: OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x07 Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 1 0x0a Node 0x13 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00] Pincap 0x0810010: OUT EAPD Pin Default 0x511301f0: [N/A] Speaker at Int Rear Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 1 0x1f Node 0x14 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x083727: IN Detect Pin Default 0x02a15021: [Jack] Mic at Ext Front Conn = 1/8, Color = Red Pin-ctls: 0x24: IN Node 0x15 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x083727: IN Detect Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x24: IN Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0837: IN OUT Detect Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x0b Node 0x17 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown Pin-ctls: 0x24: IN Node 0x18 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown Pin-ctls: 0x00: Node 0x19 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 2 0x20* 0x21 Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x91f311f0: [Fixed] Other at Int Rear Conn = ATAPI, Color = Black Pin-ctls: 0x20: IN Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0x27 0x27] Pincap 0x0810: OUT Pin Default 0x214411a0: [Jack] SPDIF Out at Sep Rear Conn = RCA, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x1c [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083737: IN OUT Detect Pin Default 0x21a15020: [Jack] Mic at Sep Rear Conn = 1/8, Color = Red Pin-ctls: 0x24: IN Connection: 1 0x24 Node 0x1d [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 25 0x07* 0x19 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x1a 0x1c 0x11 0x12 0x13 0x14 0x15 0x16 0x1e 0x1f 0x20 0x21 0x22 0x23 0x24 0x25 0x26 Node 0x1e [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x0e 0x21 Node 0x1f [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x1e Node 0x20 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x9c 0x9c] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x14 0x15 0x16 0x1a 0x25 Node 0x21 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 1 0x20 Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03 0x04* Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x24 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x23 0x21 Node 0x25 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 1 0x1c Node 0x26 [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 3 0x14* 0x15 0x1c

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Re: [alsa-devel] [Fixed-Mostly] ALC268
by Steven Zakulec 06 Jul '07

06 Jul '07

I've found another issue: I've lost some mixer options now. I've lost 1 capture source at least. I've attached the result of alsactl -f somefile store from before sound worked and after. --------------------------------------- Original E-mail From: Steven Zakulec [spzakulec(a)mail.widener.edu] Date: 07/06/2007 10:04 AM To: alsa-devel(a)alsa-project.org Subject: [alsa-devel] [Fixed-Mostly] ALC268 I was finally able to get sound working, thanks to the bugreport here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3104 What I did was get the patch_realtek.c from the bugreport, put it in alsa-kernel/pci/hda, and then re-build the driver (plus I re-built the lib, but probably not necessary). I wasn't able to rmmod snd-hda-intel after the build finished, so I restarted the computer, and then it just worked. Unfortunately, the headphone jack is not working. The sound is only slightly, if at all, muted from the speakers when the headphones are plugged in. _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [Fixed-Mostly] ALC268
by Steven Zakulec 06 Jul '07

06 Jul '07

I was finally able to get sound working, thanks to the bugreport here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3104 What I did was get the patch_realtek.c from the bugreport, put it in alsa-kernel/pci/hda, and then re-build the driver (plus I re-built the lib, but probably not necessary). I wasn't able to rmmod snd-hda-intel after the build finished, so I restarted the computer, and then it just worked. Unfortunately, the headphone jack is not working. The sound is only slightly, if at all, muted from the speakers when the headphones are plugged in.

1 0

[alsa-devel] PS3 - Surround sound on SPDIF set-up
by gt bradley 06 Jul '07

06 Jul '07

Crud :( I had asked them and they said that the hypervisor should allow for AC3 pass-through. In my inital review of the snd_ps3.c code, it looked like it was basically a data-pump, handling the DMA & interupts to move data from the passed in stream to the h/w (via the hypervisor). I also assumed that the iec bits (in headers in the SPDIF frames) were being set by higher levels in the software stack. If the bits are being set by the hardware and/or hypervisor then I'll have to fall back to plan "B". I've been programming for 20+ years, and have put kernel mods into production, but the last kernel hacking/device driver I did was in SYSV. I can forward changes/fixes to Geoff, for inclusion rolling things "upstream". I don't have direct access to inside information other than the source-code, but can forward the requests for more detailed info. let me know if I can help GT p.s. thanks for your time and patience. On 7/5/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 04 Jul 2007 18:23:09 -0700, > GT Bradley wrote: > > > > right, > > For my Fedora install, these files are at /etc/alsa/cards, (although > > the iec958.conf) is in /etc/alsa/pcm). > > > > The iec958.conf seems resonable, although I don't fully understand the > > syntax of the file, nor understand how the asoundrc file wires to > > /etc/alsa/... to the actual linux driver. It seems to pass through to > > > the default device (which is what I want) > > > > using the system-config-soundcard utility, I do get surround sound, and > > is displays the card as PS3 whcih matches the output of aplay -l I've > > > tryed createing a /etc/alsa/cards/PS3.conf file as follows: > > It seems that the current PS3 driver doesn't support the passthru > properly yet. For AC3, you'd have to set up the IEC958 status bits, > but the driver doesn't support it. Some receivers are robust that > work even without non-audio bit, but many receivers require the proper > status bits. I'll contact with SCE guys for fixing this issue. > > > Takashi >

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[alsa-devel] linked streams question
by Eliot Blennerhassett 06 Jul '07

06 Jul '07

Greetings, I'm adding linked stream support to my driver (asihpi), and having some trouble with the end of playback. What happens is that the slave streams never transition to SETUP state at the end of playback. (Eg playing 4 channel wav file to 2 linked stereo streams with aplay). The master stream reaches snd_pcm_drain_done() which then causes trigger(stop), resulting in all linked streams being stopped. But only the master state is set to SETUP However, now the slave streams are stopped, period_elapsed() is not called for a final time to (I assme) cause them to transition to the SETUP state. Eventually I get "playback drain error (DMA or IRQ trouble?)\n" My workaround is to set the state of each substream to SETUP in the trigger-stop operator, but no other driver needs to do this so I think it is probably the wrong solution - maybe misses some other cleanup code? Any suggestions on the proper way to handle this? regards Eliot

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[alsa-devel] [Fwd: Re: default setting not fully working with new ALSA device]
by Tobin Davis 05 Jul '07

05 Jul '07

-------- Forwarded Message -------- From: Peter Matthias <PeterMatthias(a)web.de> To: Tobin Davis <tdavis(a)dsl-only.net> Subject: Re: [alsa-devel] default setting not fully working with new ALSA device Date: Thu, 5 Jul 2007 18:34:08 +0200 The Board is a MSI K9AGM2-FIH, Sockel AM2, ATI RS690G, mATX, PCIe The script gave following output: "Your ALSA information is located at http://pastebin.ca/604529" Other drivers are statically linked in my kernel. Thank you for your work! In case of any questions, contact me. Peter Am Mittwoch, 4. Juli 2007 schrieben Sie: > Could you post the output from > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh ? This > script will give us more detailed information on what sound chip it is. > We will also need to know the brand of board, for the driver file. > > Thanks, > > Tobin > > On Wed, 2007-07-04 at 16:48 +0200, Peter Matthias wrote: > > Hello, > > > > I have got a 690G board with following lspic -nv output: > > > > 01:05.2 0403: 1002:7919 > > Subsystem: 1002:7919 > > Flags: bus master, fast devsel, latency 64, IRQ 19 > > Memory at fe9e8000 (64-bit, non-prefetchable) [size=16K] > > Capabilities: [50] Power Management version 2 > > Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 > > Enable- > > > > With the default setting, microphone does not work. With module > > parameter "modprobe -i snd-hda-intel model=6stack-dig" microphone works > > well. Please add to default setting. It was written in Linux Kernel > > documentation to report this issue here. > > > > Thank you, > > Peter > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel -- Tobin Davis <tdavis(a)dsl-only.net>

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[PATCH 1/1] ALSA sound driver for the AT73C213 DAC using Atmel SSC driver
by Hans-Christian Egtvedt 05 Jul '07

05 Jul '07

This patch adds support for the AT73C213 DAC using the misc Atmel SSC driver in I2S mode. The driver also requires a SPI to setup the registers and control volume. It has been tested with an AT32AP7000 on the ATSTK1000 development board. The driver should also work with any Atmel device with an SSC module supported by the Atmel SSC driver (atmel-ssc). The atmel-ssc driver is just submitted to the Linux kernel. Please see mail thread http://lkml.org/lkml/2007/7/16/32 Signed-off-by: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> --- include/linux/spi/at73c213.h | 25 + sound/spi/Kconfig | 25 + sound/spi/Makefile | 3 + sound/spi/at73c213.c | 1121 ++++++++++++++++++++++++++++++++++++++++++ sound/spi/at73c213.h | 119 +++++ 5 files changed, 1293 insertions(+), 0 deletions(-) create mode 100644 include/linux/spi/at73c213.h create mode 100644 sound/spi/at73c213.c create mode 100644 sound/spi/at73c213.h diff --git a/include/linux/spi/at73c213.h b/include/linux/spi/at73c213.h new file mode 100644 index 0000000..0f20a70 --- /dev/null +++ b/include/linux/spi/at73c213.h @@ -0,0 +1,25 @@ +/* + * Board-specific data used to set up AT73c213 audio DAC driver. + */ + +#ifndef __LINUX_SPI_AT73C213_H +#define __LINUX_SPI_AT73C213_H + +/** + * at73c213_board_info - how the external DAC is wired to the device. + * + * @ssc_id: SSC platform_driver id the DAC shall use to stream the audio. + * @dac_clk: the external clock used to provide master clock to the DAC. + * @shortname: a short discription for the DAC, seen by userspace tools. + * + * This struct contains the configuration of the hardware connection to the + * external DAC. The DAC needs a master clock and a I2S audio stream. It also + * provides a name which is used to identify it in userspace tools. + */ +struct at73c213_board_info { + int ssc_id; + struct clk *dac_clk; + char shortname[32]; +}; + +#endif /* __LINUX_SPI_AT73C213_H */ diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig index 60fbb2a..0d08c29 100644 --- a/sound/spi/Kconfig +++ b/sound/spi/Kconfig @@ -3,4 +3,29 @@ menu "SPI devices" depends on SND != n +config SND_AT73C213 + tristate "Atmel AT73C213 DAC driver" + depends on ATMEL_SSC + select SND_PCM + help + Say Y here if you want to use the Atmel AT73C213 external DAC. This + DAC can be found on Atmel development boards. + + This driver requires the Atmel SSC driver for sound sink, a + peripheral found on most AT91 and AVR32 microprocessors. + + To compile this driver as a module, choose M here: the module will be + called snd-at73c213. + +config SND_AT73C213_TARGET_BITRATE + int "Target bitrate for AT73C213" + depends on SND_AT73C213 + default "48000" + range 8000 50000 + help + Sets the target bitrate for the bitrate calculator in the driver. + Limited by hardware to be between 8000 Hz and 50000 Hz. + + Set to 48000 Hz by default. + endmenu diff --git a/sound/spi/Makefile b/sound/spi/Makefile index 30d8ec9..026fb73 100644 --- a/sound/spi/Makefile +++ b/sound/spi/Makefile @@ -1,2 +1,5 @@ # Makefile for SPI drivers +snd-at73c213-objs := at73c213.o + +obj-$(CONFIG_SND_AT73C213) += snd-at73c213.o diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c new file mode 100644 index 0000000..a466929 --- /dev/null +++ b/sound/spi/at73c213.c @@ -0,0 +1,1121 @@ +/* + * Driver for AT73C213 16-bit stereo DAC connected to Atmel SSC + * + * Copyright (C) 2006-2007 Atmel Norway + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation. + */ + +/*#define DEBUG*/ + +#include <linux/clk.h> +#include <linux/err.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <sound/driver.h> +#include <sound/initval.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/pcm.h> + +#include <linux/atmel-ssc.h> + +#include <linux/spi/spi.h> +#include <linux/spi/at73c213.h> + +#include "at73c213.h" + +#define BITRATE_MIN 8000 /* Hardware limit? */ +#define BITRATE_TARGET CONFIG_SND_AT73C213_TARGET_BITRATE +#define BITRATE_MAX 50000 /* Hardware limit. */ + +/* Initial (hardware reset) AT73C213 register values. */ +static u8 snd_at73c213_original_image[18] = +{ + 0x00, /* 00 - CTRL */ + 0x05, /* 01 - LLIG */ + 0x05, /* 02 - RLIG */ + 0x08, /* 03 - LPMG */ + 0x08, /* 04 - RPMG */ + 0x00, /* 05 - LLOG */ + 0x00, /* 06 - RLOG */ + 0x22, /* 07 - OLC */ + 0x09, /* 08 - MC */ + 0x00, /* 09 - CSFC */ + 0x00, /* 0A - MISC */ + 0x00, /* 0B - */ + 0x00, /* 0C - PRECH */ + 0x05, /* 0D - AUXG */ + 0x00, /* 0E - */ + 0x00, /* 0F - */ + 0x00, /* 10 - RST */ + 0x00, /* 11 - PA_CTRL */ +}; + +struct snd_at73c213 { + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *substream; + struct at73c213_board_info *board; + int irq; + int period; + unsigned long bitrate; + struct clk *bitclk; + struct ssc_device *ssc; + struct spi_device *spi; + u8 spi_wbuffer[2]; + u8 spi_rbuffer[2]; + /* Image of the SPI registers in AT73C213. */ + u8 reg_image[18]; + /* Protect registers against concurrent access. */ + spinlock_t lock; +}; + +#define get_chip(card) ((struct snd_at73c213 *)card->private_data) + +static int +snd_at73c213_write_reg(struct snd_at73c213 *chip, u8 reg, u8 val) +{ + struct spi_message msg; + struct spi_transfer msg_xfer = { + .len = 2, + .cs_change = 0, + }; + int retval; + + spi_message_init(&msg); + + chip->spi_wbuffer[0] = reg; + chip->spi_wbuffer[1] = val; + + msg_xfer.tx_buf = chip->spi_wbuffer; + msg_xfer.rx_buf = chip->spi_rbuffer; + spi_message_add_tail(&msg_xfer, &msg); + + retval = spi_sync(chip->spi, &msg); + + if (!retval) + chip->reg_image[reg] = val; + + return retval; +} + +static struct snd_pcm_hardware snd_at73c213_playback_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, /* Replaced by chip->bitrate later. */ + .rate_max = 50000, /* Replaced by chip->bitrate later. */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024 - 1, + .period_bytes_min = 512, + .period_bytes_max = 64 * 1024 - 1, + .periods_min = 4, + .periods_max = 1024, +}; + +/* + * Calculate and set bitrate and divisions. + */ +static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) +{ + unsigned long ssc_rate = clk_get_rate(chip->ssc->clk); + unsigned long dac_rate_new, ssc_div, status; + unsigned long ssc_div_max, ssc_div_min; + int max_tries; + + /* + * We connect two clocks here, picking divisors so the I2S clocks + * out data at the same rate the DAC clocks it in ... and as close + * as practical to the desired target rate. + * + * The DAC master clock (MCLK) is programmable, and is either 256 + * or (not here) 384 times the I2S output clock (BCLK). + */ + + /* SSC clock / (bitrate * stereo * 16-bit). */ + ssc_div = ssc_rate / (BITRATE_TARGET * 2 * 16); + ssc_div_min = ssc_rate / (BITRATE_MAX * 2 * 16); + ssc_div_max = ssc_rate / (BITRATE_MIN * 2 * 16); + max_tries = (ssc_div_max - ssc_div_min) / 2; + + if (max_tries < 1) + max_tries = 1; + + /* ssc_div must be a power of 2. */ + ssc_div = (ssc_div + 1) & ~1UL; + + if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) { + ssc_div -= 2; + if ((ssc_rate / (ssc_div * 2 * 16)) > BITRATE_MAX) + return -ENXIO; + } + + /* Search for a possible bitrate. */ + do { + /* SSC clock / (ssc divider * 16-bit * stereo). */ + if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) + return -ENXIO; + + /* 256 / (2 * 16) = 8 */ + dac_rate_new = 8 * (ssc_rate / ssc_div); + + status = clk_round_rate(chip->board->dac_clk, dac_rate_new); + if (status < 0) + return status; + + /* Ignore difference smaller than 256 Hz. */ + if ((status/256) == (dac_rate_new/256)) + goto set_rate; + + ssc_div += 2; + } while (--max_tries); + + /* Not able to find a valid bitrate. */ + return -ENXIO; + +set_rate: + status = clk_set_rate(chip->board->dac_clk, status); + if (status < 0) + return status; + + /* Set divider in SSC device. */ + ssc_writel(chip->ssc->regs, CMR, ssc_div/2); + + /* SSC clock / (ssc divider * 16-bit * stereo). */ + chip->bitrate = ssc_rate / (ssc_div * 16 * 2); + + dev_info(&chip->spi->dev, + "at73c213: supported bitrate is %lu (%lu divider)\n", + chip->bitrate, ssc_div); + + return 0; +} + +static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_at73c213_playback_hw.rate_min = chip->bitrate; + snd_at73c213_playback_hw.rate_max = chip->bitrate; + runtime->hw = snd_at73c213_playback_hw; + chip->substream = substream; + + return 0; +} + +static int snd_at73c213_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + chip->substream = NULL; + return 0; +} + +static int snd_at73c213_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_at73c213_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_at73c213_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int block_size; + + block_size = frames_to_bytes(runtime, runtime->period_size); + + chip->period = 0; + + ssc_writel(chip->ssc->regs, PDC_TPR, + (long)runtime->dma_addr); + ssc_writel(chip->ssc->regs, PDC_TCR, runtime->period_size * 2); + ssc_writel(chip->ssc->regs, PDC_TNPR, + (long)runtime->dma_addr + block_size); + ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + + return 0; +} + +static int snd_at73c213_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + int retval = 0; + + spin_lock(&chip->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ssc_writel(chip->ssc->regs, IER, SSC_BIT(IER_ENDTX)); + ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTEN)); + break; + case SNDRV_PCM_TRIGGER_STOP: + ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTDIS)); + ssc_writel(chip->ssc->regs, IDR, SSC_BIT(IDR_ENDTX)); + break; + default: + dev_dbg(&chip->spi->dev, "spurious command %x\n", cmd); + retval = -EINVAL; + break; + } + + spin_unlock(&chip->lock); + + return retval; +} + +static snd_pcm_uframes_t +snd_at73c213_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; + unsigned long bytes; + + bytes = ssc_readl(chip->ssc->regs, PDC_TPR) + - (unsigned long)runtime->dma_addr; + + pos = bytes_to_frames(runtime, bytes); + if (pos >= runtime->buffer_size) + pos -= runtime->buffer_size; + + return pos; +} + +static struct snd_pcm_ops at73c213_playback_ops = { + .open = snd_at73c213_pcm_open, + .close = snd_at73c213_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_at73c213_pcm_hw_params, + .hw_free = snd_at73c213_pcm_hw_free, + .prepare = snd_at73c213_pcm_prepare, + .trigger = snd_at73c213_pcm_trigger, + .pointer = snd_at73c213_pcm_pointer, +}; + +static void snd_at73c213_pcm_free(struct snd_pcm *pcm) +{ + struct snd_at73c213 *chip = snd_pcm_chip(pcm); + if (chip->pcm) { + snd_pcm_lib_preallocate_free_for_all(chip->pcm); + chip->pcm = NULL; + } +} + +static int __devinit snd_at73c213_pcm_new(struct snd_at73c213 *chip, int device) +{ + struct snd_pcm *pcm; + int retval; + + retval = snd_pcm_new(chip->card, chip->card->shortname, + device, 1, 0, &pcm); + if (retval < 0) + goto out; + + pcm->private_data = chip; + pcm->private_free = snd_at73c213_pcm_free; + pcm->info_flags = SNDRV_PCM_INFO_BLOCK_TRANSFER; + strcpy(pcm->name, "at73c213"); + chip->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &at73c213_playback_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(chip->pcm, + SNDRV_DMA_TYPE_DEV, &chip->spi->dev, + 64 * 1024, 64 * 1024); +out: + return retval; +} + +static irqreturn_t snd_at73c213_interrupt(int irq, void *dev_id) +{ + struct snd_at73c213 *chip = dev_id; + struct snd_pcm_runtime *runtime = chip->substream->runtime; + u32 status; + int offset; + int block_size; + int next_period; + int retval = IRQ_NONE; + + spin_lock(&chip->lock); + + block_size = frames_to_bytes(runtime, runtime->period_size); + status = ssc_readl(chip->ssc->regs, IMR); + + if (status & SSC_BIT(IMR_ENDTX)) { + chip->period++; + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + ssc_writel(chip->ssc->regs, PDC_TNPR, + (long)runtime->dma_addr + offset); + ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + retval = IRQ_HANDLED; + } + + ssc_readl(chip->ssc->regs, IMR); + spin_unlock(&chip->lock); + + if (status & SSC_BIT(IMR_ENDTX)) + snd_pcm_period_elapsed(chip->substream); + + return retval; +} + +/* + * Mixer functions. + */ +static int snd_at73c213_mono_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + + spin_lock_irq(&chip->lock); + + ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + (mask - ucontrol->value.integer.value[0]); + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_mono_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + int change, retval; + unsigned short val; + + val = (ucontrol->value.integer.value[0] & mask); + if (invert) + val = mask - val; + val <<= shift; + + spin_lock_irq(&chip->lock); + + val = (chip->reg_image[reg] & ~(mask << shift)) | val; + change = val != chip->reg_image[reg]; + retval = snd_at73c213_write_reg(chip, reg, val); + + spin_unlock_irq(&chip->lock); + + if (retval) + return retval; + + return change; +} + +static int snd_at73c213_stereo_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + if (mask == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + + return 0; +} + +static int snd_at73c213_stereo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x07; + int shift_right = (kcontrol->private_value >> 19) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + + spin_lock_irq(&chip->lock); + + ucontrol->value.integer.value[0] = + (chip->reg_image[left_reg] >> shift_left) & mask; + ucontrol->value.integer.value[1] = + (chip->reg_image[right_reg] >> shift_right) & mask; + + if (invert) { + ucontrol->value.integer.value[0] = + (mask - ucontrol->value.integer.value[0]); + ucontrol->value.integer.value[1] = + (mask - ucontrol->value.integer.value[1]); + } + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_stereo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x07; + int shift_right = (kcontrol->private_value >> 19) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + int change, retval; + unsigned short val1, val2; + + val1 = ucontrol->value.integer.value[0] & mask; + val2 = ucontrol->value.integer.value[1] & mask; + if (invert) { + val1 = mask - val1; + val2 = mask - val2; + } + val1 <<= shift_left; + val2 <<= shift_right; + + spin_lock_irq(&chip->lock); + + val1 = (chip->reg_image[left_reg] & ~(mask << shift_left)) | val1; + val2 = (chip->reg_image[right_reg] & ~(mask << shift_right)) | val2; + change = val1 != chip->reg_image[left_reg] + || val2 != chip->reg_image[right_reg]; + retval = snd_at73c213_write_reg(chip, left_reg, val1); + if (retval) { + spin_unlock_irq(&chip->lock); + goto out; + } + retval = snd_at73c213_write_reg(chip, right_reg, val2); + if (retval) { + spin_unlock_irq(&chip->lock); + goto out; + } + + spin_unlock_irq(&chip->lock); + + return change; + +out: + return retval; +} + +static int snd_at73c213_mono_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int snd_at73c213_mono_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + + spin_lock_irq(&chip->lock); + + ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & 0x01; + + if (invert) + ucontrol->value.integer.value[0] = + (0x01 - ucontrol->value.integer.value[0]); + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_mono_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + int change, retval; + unsigned short val; + + if (ucontrol->value.integer.value[0]) + val = mask; + else + val = 0; + + if (invert) + val = mask - val; + val <<= shift; + + spin_lock_irq(&chip->lock); + + val |= (chip->reg_image[reg] & ~(mask << shift)); + change = val != chip->reg_image[reg]; + + retval = snd_at73c213_write_reg(chip, reg, val); + + spin_unlock_irq(&chip->lock); + + if (retval) + return retval; + + return change; +} + +static int snd_at73c213_pa_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = ((kcontrol->private_value >> 16) & 0xff) - 1; + + return 0; +} + +static int snd_at73c213_line_capture_volume_info( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + /* When inverted will give values 0x10001 => 0. */ + uinfo->value.integer.min = 14; + uinfo->value.integer.max = 31; + + return 0; +} + +static int snd_at73c213_aux_capture_volume_info( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + /* When inverted will give values 0x10001 => 0. */ + uinfo->value.integer.min = 14; + uinfo->value.integer.max = 31; + + return 0; +} + +#define AT73C213_MONO_SWITCH(xname, xindex, reg, shift, mask, invert) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_at73c213_mono_switch_info, \ + .get = snd_at73c213_mono_switch_get, \ + .put = snd_at73c213_mono_switch_put, \ + .private_value = (reg | (shift << 8) | (mask << 16) | (invert << 24)) \ +} + +#define AT73C213_STEREO(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_at73c213_stereo_info, \ + .get = snd_at73c213_stereo_get, \ + .put = snd_at73c213_stereo_put, \ + .private_value = (left_reg | (right_reg << 8) \ + | (shift_left << 16) | (shift_right << 19) \ + | (mask << 24) | (invert << 22)) \ +} + +static struct snd_kcontrol_new snd_at73c213_controls[] __devinitdata = { +AT73C213_STEREO("Master Playback Volume", 0, DAC_LMPG, DAC_RMPG, 0, 0, 0x1f, 1), +AT73C213_STEREO("Master Playback Switch", 0, DAC_LMPG, DAC_RMPG, 5, 5, 1, 1), +AT73C213_STEREO("PCM Playback Volume", 0, DAC_LLOG, DAC_RLOG, 0, 0, 0x1f, 1), +AT73C213_STEREO("PCM Playback Switch", 0, DAC_LLOG, DAC_RLOG, 5, 5, 1, 1), +AT73C213_MONO_SWITCH("Mono PA Playback Switch", 0, DAC_CTRL, DAC_CTRL_ONPADRV, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PA Playback Volume", + .index = 0, + .info = snd_at73c213_pa_volume_info, + .get = snd_at73c213_mono_get, + .put = snd_at73c213_mono_put, + .private_value = PA_CTRL | (PA_CTRL_APAGAIN << 8) | (0x0f << 16) | (1 << 24), +}, +AT73C213_MONO_SWITCH("PA High Gain Playback Switch", 0, PA_CTRL, PA_CTRL_APALP, 0x01, 1), +AT73C213_MONO_SWITCH("PA Playback Switch", 0, PA_CTRL, PA_CTRL_APAON, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Capture Volume", + .index = 0, + .info = snd_at73c213_aux_capture_volume_info, + .get = snd_at73c213_mono_get, + .put = snd_at73c213_mono_put, + .private_value = DAC_AUXG | (0 << 8) | (0x1f << 16) | (1 << 24), +}, +AT73C213_MONO_SWITCH("Aux Capture Switch", 0, DAC_CTRL, DAC_CTRL_ONAUXIN, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Volume", + .index = 0, + .info = snd_at73c213_line_capture_volume_info, + .get = snd_at73c213_stereo_get, + .put = snd_at73c213_stereo_put, + .private_value = DAC_LLIG | (DAC_RLIG << 8) | (0 << 16) | (0 << 19) + | (0x1f << 24) | (1 << 22), +}, +AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0), +}; + +static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) +{ + struct snd_card *card; + int errval, idx; + + if (chip == NULL || chip->pcm == NULL) + return -EINVAL; + + card = chip->card; + + strcpy(card->mixername, chip->pcm->name); + + for (idx = 0; idx < ARRAY_SIZE(snd_at73c213_controls); idx++) { + errval = snd_ctl_add(card, + snd_ctl_new1(&snd_at73c213_controls[idx], + chip)); + if (errval < 0) + goto cleanup; + } + + return 0; + +cleanup: + for (idx = 1; idx < ARRAY_SIZE(snd_at73c213_controls) + 1; idx++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_find_numid(card, idx); + if (kctl) + snd_ctl_remove(card, kctl); + } + return errval; +} + +/* + * Device functions + */ +static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) +{ + /* + * Continuous clock output. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Periode is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + /* + * Data length is 16 bit (16 - 1). + * Transmit MSB first. + * Transmit 2 words each transfer. + * Frame sync length is 16 bit (16 - 1). + * Frame starts on negative pulse. + */ + ssc_writel(chip->ssc->regs, TFMR, + SSC_BF(TFMR_DATLEN, 16 - 1) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATNB, 1) + | SSC_BF(TFMR_FSLEN, 16 - 1) + | SSC_BF(TFMR_FSOS, 1)); + + return 0; +} + +static int snd_at73c213_chip_init(struct snd_at73c213 *chip) +{ + int retval; + unsigned char dac_ctrl = 0; + + retval = snd_at73c213_set_bitrate(chip); + if (retval) + goto out; + + /* Enable DAC master clock. */ + clk_enable(chip->board->dac_clk); + + /* Initialize at73c213 on SPI bus. */ + retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); + if (retval) + goto out_clk; + msleep(1); + retval = snd_at73c213_write_reg(chip, DAC_RST, 0x03); + if (retval) + goto out_clk; + + /* Precharge everything. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0xff); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APAPRECH)); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_CTRL, + (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR)); + if (retval) + goto out_clk; + + msleep(50); + + /* Stop precharging PA. */ + retval = snd_at73c213_write_reg(chip, PA_CTRL, + (1<<PA_CTRL_APALP) | 0x0f); + if (retval) + goto out_clk; + + msleep(450); + + /* Stop precharging DAC, turn on master power. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, (1<<DAC_PRECH_ONMSTR)); + if (retval) + goto out_clk; + + msleep(1); + + /* Turn on DAC. */ + dac_ctrl = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) + | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR); + + retval = snd_at73c213_write_reg(chip, DAC_CTRL, dac_ctrl); + if (retval) + goto out_clk; + + /* Mute sound. */ + retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); + if (retval) + goto out_clk; + + /* Enable I2S device, i.e. clock output. */ + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); + + goto out; + +out_clk: + clk_disable(chip->board->dac_clk); +out: + return retval; +} + +static int snd_at73c213_dev_free(struct snd_device *device) +{ + struct snd_at73c213 *chip = device->device_data; + + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + if (chip->irq >= 0) { + free_irq(chip->irq, chip); + chip->irq = -1; + } + + return 0; +} + +static int __devinit snd_at73c213_dev_init(struct snd_card *card, + struct spi_device *spi) +{ + static struct snd_device_ops ops = { + .dev_free = snd_at73c213_dev_free, + }; + struct snd_at73c213 *chip = get_chip(card); + int irq, retval; + + irq = chip->ssc->irq; + if (irq < 0) + return irq; + + spin_lock_init(&chip->lock); + chip->card = card; + chip->irq = -1; + + retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); + if (retval) { + dev_dbg(&chip->spi->dev, "unable to request irq %d\n", irq); + goto out; + } + chip->irq = irq; + + memcpy(&chip->reg_image, &snd_at73c213_original_image, + sizeof(snd_at73c213_original_image)); + + retval = snd_at73c213_ssc_init(chip); + if (retval) + goto out_irq; + + retval = snd_at73c213_chip_init(chip); + if (retval) + goto out_irq; + + retval = snd_at73c213_pcm_new(chip, 0); + if (retval) + goto out_irq; + + retval = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (retval) + goto out_irq; + + retval = snd_at73c213_mixer(chip); + if (retval) + goto out_snd_dev; + + snd_card_set_dev(card, &spi->dev); + + goto out; + +out_snd_dev: + snd_device_free(card, chip); +out_irq: + free_irq(chip->irq, chip); + chip->irq = -1; +out: + return retval; +} + +static int snd_at73c213_probe(struct spi_device *spi) +{ + struct snd_card *card; + struct snd_at73c213 *chip; + struct at73c213_board_info *board; + int retval; + char id[16]; + + board = spi->dev.platform_data; + if (!board) { + dev_dbg(&spi->dev, "no platform_data\n"); + return -ENXIO; + } + + if (!board->dac_clk) { + dev_dbg(&spi->dev, "no DAC clk\n"); + return -ENXIO; + } + + if (IS_ERR(board->dac_clk)) { + dev_dbg(&spi->dev, "no DAC clk\n"); + return PTR_ERR(board->dac_clk); + } + + retval = -ENOMEM; + + /* Allocate "card" using some unused identifiers. */ + snprintf(id, sizeof id, "at73c213_%d", board->ssc_id); + card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct snd_at73c213)); + if (!card) + goto out; + + chip = card->private_data; + chip->spi = spi; + chip->board = board; + + chip->ssc = ssc_request(board->ssc_id); + if (IS_ERR(chip->ssc)) { + dev_dbg(&spi->dev, "could not get ssc%d device\n", + board->ssc_id); + retval = PTR_ERR(chip->ssc); + goto out_card; + } + + retval = snd_at73c213_dev_init(card, spi); + if (retval) + goto out_ssc; + + strcpy(card->driver, "at73c213"); + strcpy(card->shortname, board->shortname); + sprintf(card->longname, "%s on irq %d", card->shortname, chip->irq); + + retval = snd_card_register(card); + if (retval) + goto out_ssc; + + dev_set_drvdata(&spi->dev, card); + + goto out; + +out_ssc: + ssc_free(chip->ssc); +out_card: + snd_card_free(card); +out: + return retval; +} + +static int __devexit snd_at73c213_remove(struct spi_device *spi) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + int retval; + + /* Stop playback. */ + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + + /* Mute sound. */ + retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); + if (retval) + goto out; + + /* Turn off PA. */ + retval = snd_at73c213_write_reg(chip, PA_CTRL, (chip->reg_image[PA_CTRL]|0x0f)); + if (retval) + goto out; + msleep(10); + retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APALP)|0x0f); + if (retval) + goto out; + + /* Turn off external DAC. */ + retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x0c); + if (retval) + goto out; + msleep(2); + retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x00); + if (retval) + goto out; + + /* Turn off master power. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0x00); + if (retval) + goto out; + +out: + /* Stop DAC master clock. */ + clk_disable(chip->board->dac_clk); + + ssc_free(chip->ssc); + snd_card_free(card); + dev_set_drvdata(&spi->dev, NULL); + + return 0; +} + +#ifdef CONFIG_PM +static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + clk_disable(chip->board->dac_clk); + + return 0; +} + +static int snd_at73c213_resume(struct spi_device *spi) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + + clk_enable(chip->board->dac_clk); + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); + + return 0; +} +#else +#define snd_at73c213_suspend NULL +#define snd_at73c213_resume NULL +#endif + +static struct spi_driver at73c213_driver = { + .driver = { + .name = "at73c213", + }, + .probe = snd_at73c213_probe, + .suspend = snd_at73c213_suspend, + .resume = snd_at73c213_resume, + .remove = __devexit_p(snd_at73c213_remove), +}; + +static int __init at73c213_init(void) +{ + return spi_register_driver(&at73c213_driver); +} +module_init(at73c213_init); + +static void __exit at73c213_exit(void) +{ + spi_unregister_driver(&at73c213_driver); +} +module_exit(at73c213_exit); + +MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt(a)atmel.com>"); +MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); +MODULE_LICENSE("GPL"); diff --git a/sound/spi/at73c213.h b/sound/spi/at73c213.h new file mode 100644 index 0000000..fd8b372 --- /dev/null +++ b/sound/spi/at73c213.h @@ -0,0 +1,119 @@ +/* + * Driver for the AT73C213 16-bit stereo DAC on Atmel ATSTK1000 + * + * Copyright (C) 2006 - 2007 Atmel Corporation + * + * 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; either version 2 of the + * License, or (at your option) any later version. + * + * 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. + * + * The full GNU General Public License is included in this + * distribution in the file called COPYING. + */ + +#ifndef _SND_AT73C213_H +#define _SND_AT73C213_H + +/* DAC control register */ +#define DAC_CTRL 0x00 +#define DAC_CTRL_ONPADRV 7 +#define DAC_CTRL_ONAUXIN 6 +#define DAC_CTRL_ONDACR 5 +#define DAC_CTRL_ONDACL 4 +#define DAC_CTRL_ONLNOR 3 +#define DAC_CTRL_ONLNOL 2 +#define DAC_CTRL_ONLNIR 1 +#define DAC_CTRL_ONLNIL 0 + +/* DAC left line in gain register */ +#define DAC_LLIG 0x01 +#define DAC_LLIG_LLIG 0 + +/* DAC right line in gain register */ +#define DAC_RLIG 0x02 +#define DAC_RLIG_RLIG 0 + +/* DAC Left Master Playback Gain Register */ +#define DAC_LMPG 0x03 +#define DAC_LMPG_LMPG 0 + +/* DAC Right Master Playback Gain Register */ +#define DAC_RMPG 0x04 +#define DAC_RMPG_RMPG 0 + +/* DAC Left Line Out Gain Register */ +#define DAC_LLOG 0x05 +#define DAC_LLOG_LLOG 0 + +/* DAC Right Line Out Gain Register */ +#define DAC_RLOG 0x06 +#define DAC_RLOG_RLOG 0 + +/* DAC Output Level Control Register */ +#define DAC_OLC 0x07 +#define DAC_OLC_RSHORT 7 +#define DAC_OLC_ROLC 4 +#define DAC_OLC_LSHORT 3 +#define DAC_OLC_LOLC 0 + +/* DAC Mixer Control Register */ +#define DAC_MC 0x08 +#define DAC_MC_INVR 5 +#define DAC_MC_INVL 4 +#define DAC_MC_RMSMIN2 3 +#define DAC_MC_RMSMIN1 2 +#define DAC_MC_LMSMIN2 1 +#define DAC_MC_LMSMIN1 0 + +/* DAC Clock and Sampling Frequency Control Register */ +#define DAC_CSFC 0x09 +#define DAC_CSFC_OVRSEL 4 + +/* DAC Miscellaneous Register */ +#define DAC_MISC 0x0A +#define DAC_MISC_VCMCAPSEL 7 +#define DAC_MISC_DINTSEL 4 +#define DAC_MISC_DITHEN 3 +#define DAC_MISC_DEEMPEN 2 +#define DAC_MISC_NBITS 0 + +/* DAC Precharge Control Register */ +#define DAC_PRECH 0x0C +#define DAC_PRECH_PRCHGPDRV 7 +#define DAC_PRECH_PRCHGAUX1 6 +#define DAC_PRECH_PRCHGLNOR 5 +#define DAC_PRECH_PRCHGLNOL 4 +#define DAC_PRECH_PRCHGLNIR 3 +#define DAC_PRECH_PRCHGLNIL 2 +#define DAC_PRECH_PRCHG 1 +#define DAC_PRECH_ONMSTR 0 + +/* DAC Auxiliary Input Gain Control Register */ +#define DAC_AUXG 0x0D +#define DAC_AUXG_AUXG 0 + +/* DAC Reset Register */ +#define DAC_RST 0x10 +#define DAC_RST_RESMASK 2 +#define DAC_RST_RESFILZ 1 +#define DAC_RST_RSTZ 0 + +/* Power Amplifier Control Register */ +#define PA_CTRL 0x11 +#define PA_CTRL_APAON 6 +#define PA_CTRL_APAPRECH 5 +#define PA_CTRL_APALP 4 +#define PA_CTRL_APAGAIN 0 + +#endif /* _SND_AT73C213_H */ -- 1.5.2.3 --=-AUyv9PDBqTOzFdJ/BnKL--

1 0

[PATCH 1/1] ALSA sound driver for the AT73C213 DAC using Atmel SSC driver
by Hans-Christian Egtvedt 05 Jul '07

05 Jul '07

This patch adds support for the AT73C213 DAC using the misc Atmel SSC driver in I2S mode. The driver also requires a SPI to setup the registers and control volume. It has been tested with an AT32AP7000 on the ATSTK1000 development board. The driver should also work with any Atmel device with an SSC module supported by the Atmel SSC driver (atmel-ssc). The atmel-ssc driver is just submitted to the Linux kernel. Please see mail thread http://lkml.org/lkml/2007/7/16/32 Signed-off-by: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> --- include/linux/spi/at73c213.h | 25 + sound/spi/Kconfig | 25 + sound/spi/Makefile | 3 + sound/spi/at73c213.c | 1121 ++++++++++++++++++++++++++++++++++++++++++ sound/spi/at73c213.h | 119 +++++ 5 files changed, 1293 insertions(+), 0 deletions(-) create mode 100644 include/linux/spi/at73c213.h create mode 100644 sound/spi/at73c213.c create mode 100644 sound/spi/at73c213.h diff --git a/include/linux/spi/at73c213.h b/include/linux/spi/at73c213.h new file mode 100644 index 0000000..0f20a70 --- /dev/null +++ b/include/linux/spi/at73c213.h @@ -0,0 +1,25 @@ +/* + * Board-specific data used to set up AT73c213 audio DAC driver. + */ + +#ifndef __LINUX_SPI_AT73C213_H +#define __LINUX_SPI_AT73C213_H + +/** + * at73c213_board_info - how the external DAC is wired to the device. + * + * @ssc_id: SSC platform_driver id the DAC shall use to stream the audio. + * @dac_clk: the external clock used to provide master clock to the DAC. + * @shortname: a short discription for the DAC, seen by userspace tools. + * + * This struct contains the configuration of the hardware connection to the + * external DAC. The DAC needs a master clock and a I2S audio stream. It also + * provides a name which is used to identify it in userspace tools. + */ +struct at73c213_board_info { + int ssc_id; + struct clk *dac_clk; + char shortname[32]; +}; + +#endif /* __LINUX_SPI_AT73C213_H */ diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig index 60fbb2a..0d08c29 100644 --- a/sound/spi/Kconfig +++ b/sound/spi/Kconfig @@ -3,4 +3,29 @@ menu "SPI devices" depends on SND != n +config SND_AT73C213 + tristate "Atmel AT73C213 DAC driver" + depends on ATMEL_SSC + select SND_PCM + help + Say Y here if you want to use the Atmel AT73C213 external DAC. This + DAC can be found on Atmel development boards. + + This driver requires the Atmel SSC driver for sound sink, a + peripheral found on most AT91 and AVR32 microprocessors. + + To compile this driver as a module, choose M here: the module will be + called snd-at73c213. + +config SND_AT73C213_TARGET_BITRATE + int "Target bitrate for AT73C213" + depends on SND_AT73C213 + default "48000" + range 8000 50000 + help + Sets the target bitrate for the bitrate calculator in the driver. + Limited by hardware to be between 8000 Hz and 50000 Hz. + + Set to 48000 Hz by default. + endmenu diff --git a/sound/spi/Makefile b/sound/spi/Makefile index 30d8ec9..026fb73 100644 --- a/sound/spi/Makefile +++ b/sound/spi/Makefile @@ -1,2 +1,5 @@ # Makefile for SPI drivers +snd-at73c213-objs := at73c213.o + +obj-$(CONFIG_SND_AT73C213) += snd-at73c213.o diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c new file mode 100644 index 0000000..f514f47 --- /dev/null +++ b/sound/spi/at73c213.c @@ -0,0 +1,1121 @@ +/* + * Driver for AT73C213 16-bit stereo DAC connected to Atmel SSC + * + * Copyright (C) 2006-2007 Atmel Norway + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation. + */ + +/*#define DEBUG*/ + +#include <linux/clk.h> +#include <linux/err.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <sound/driver.h> +#include <sound/initval.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/pcm.h> + +#include <linux/atmel-ssc.h> + +#include <linux/spi/spi.h> +#include <linux/spi/at73c213.h> + +#include "at73c213.h" + +#define BITRATE_MIN 8000 /* Hardware limit? */ +#define BITRATE_TARGET CONFIG_SND_AT73C213_TARGET_BITRATE +#define BITRATE_MAX 50000 /* Hardware limit. */ + +/* Initial (hardware reset) AT73C213 register values. */ +static u8 snd_at73c213_original_image[18] = +{ + 0x00, /* 00 - CTRL */ + 0x05, /* 01 - LLIG */ + 0x05, /* 02 - RLIG */ + 0x08, /* 03 - LPMG */ + 0x08, /* 04 - RPMG */ + 0x00, /* 05 - LLOG */ + 0x00, /* 06 - RLOG */ + 0x22, /* 07 - OLC */ + 0x09, /* 08 - MC */ + 0x00, /* 09 - CSFC */ + 0x00, /* 0A - MISC */ + 0x00, /* 0B - */ + 0x00, /* 0C - PRECH */ + 0x05, /* 0D - AUXG */ + 0x00, /* 0E - */ + 0x00, /* 0F - */ + 0x00, /* 10 - RST */ + 0x00, /* 11 - PA_CTRL */ +}; + +struct snd_at73c213 { + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *substream; + struct at73c213_board_info *board; + int irq; + int period; + unsigned long bitrate; + struct clk *bitclk; + struct ssc_device *ssc; + struct spi_device *spi; + u8 spi_wbuffer[2]; + u8 spi_rbuffer[2]; + /* Image of the SPI registers in AT73C213. */ + u8 reg_image[18]; + /* Protect registers against concurrent access. */ + spinlock_t lock; +}; + +#define get_chip(card) ((struct snd_at73c213 *)card->private_data) + +static int +snd_at73c213_write_reg(struct snd_at73c213 *chip, u8 reg, u8 val) +{ + struct spi_message msg; + struct spi_transfer msg_xfer = { + .len = 2, + .cs_change = 0, + }; + int retval; + + spi_message_init(&msg); + + chip->spi_wbuffer[0] = reg; + chip->spi_wbuffer[1] = val; + + msg_xfer.tx_buf = chip->spi_wbuffer; + msg_xfer.rx_buf = chip->spi_rbuffer; + spi_message_add_tail(&msg_xfer, &msg); + + retval = spi_sync(chip->spi, &msg); + + if (!retval) + chip->reg_image[reg] = val; + + return retval; +} + +static struct snd_pcm_hardware snd_at73c213_playback_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, /* Replaced by chip->bitrate later. */ + .rate_max = 50000, /* Replaced by chip->bitrate later. */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024 - 1, + .period_bytes_min = 512, + .period_bytes_max = 64 * 1024 - 1, + .periods_min = 4, + .periods_max = 1024, +}; + +/* + * Calculate and set bitrate and divisions. + */ +static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) +{ + unsigned long ssc_rate = clk_get_rate(chip->ssc->clk); + unsigned long dac_rate_new, ssc_div, status; + unsigned long ssc_div_max, ssc_div_min; + int max_tries; + + /* + * We connect two clocks here, picking divisors so the I2S clocks + * out data at the same rate the DAC clocks it in ... and as close + * as practical to the desired target rate. + * + * The DAC master clock (MCLK) is programmable, and is either 256 + * or (not here) 384 times the I2S output clock (BCLK). + */ + + /* SSC clock / (bitrate * stereo * 16-bit). */ + ssc_div = ssc_rate / (BITRATE_TARGET * 2 * 16); + ssc_div_min = ssc_rate / (BITRATE_MAX * 2 * 16); + ssc_div_max = ssc_rate / (BITRATE_MIN * 2 * 16); + max_tries = (ssc_div_max - ssc_div_min) / 2; + + if (max_tries < 1) + max_tries = 1; + + /* ssc_div must be a power of 2. */ + ssc_div = (ssc_div + 1) & ~1UL; + + if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) { + ssc_div -= 2; + if ((ssc_rate / (ssc_div * 2 * 16)) > BITRATE_MAX) + return -ENXIO; + } + + /* Search for a possible bitrate. */ + do { + /* SSC clock / (ssc divider * 16-bit * stereo). */ + if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) + return -ENXIO; + + /* 256 / (2 * 16) = 8 */ + dac_rate_new = 8 * (ssc_rate / ssc_div); + + status = clk_round_rate(chip->board->dac_clk, dac_rate_new); + if (status < 0) + return status; + + /* Ignore difference smaller than 256 Hz. */ + if ((status/256) == (dac_rate_new/256)) + goto set_rate; + + ssc_div += 2; + } while (--max_tries); + + /* Not able to find a valid bitrate. */ + return -ENXIO; + +set_rate: + status = clk_set_rate(chip->board->dac_clk, status); + if (status < 0) + return status; + + /* Set divider in SSC device. */ + ssc_writel(chip->ssc->regs, CMR, ssc_div/2); + + /* SSC clock / (ssc divider * 16-bit * stereo). */ + chip->bitrate = ssc_rate / (ssc_div * 16 * 2); + + dev_info(&chip->spi->dev, + "at73c213: supported bitrate is %lu (%lu divider)\n", + chip->bitrate, ssc_div); + + return 0; +} + +static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_at73c213_playback_hw.rate_min = chip->bitrate; + snd_at73c213_playback_hw.rate_max = chip->bitrate; + runtime->hw = snd_at73c213_playback_hw; + chip->substream = substream; + + return 0; +} + +static int snd_at73c213_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + chip->substream = NULL; + return 0; +} + +static int snd_at73c213_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_at73c213_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_at73c213_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int block_size; + + block_size = frames_to_bytes(runtime, runtime->period_size); + + chip->period = 0; + + ssc_writel(chip->ssc->regs, PDC_TPR, + (long)runtime->dma_addr); + ssc_writel(chip->ssc->regs, PDC_TCR, runtime->period_size * 2); + ssc_writel(chip->ssc->regs, PDC_TNPR, + (long)runtime->dma_addr + block_size); + ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + + return 0; +} + +static int snd_at73c213_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + int retval = 0; + + spin_lock(&chip->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ssc_writel(chip->ssc->regs, IER, SSC_BIT(IER_ENDTX)); + ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTEN)); + break; + case SNDRV_PCM_TRIGGER_STOP: + ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTDIS)); + ssc_writel(chip->ssc->regs, IDR, SSC_BIT(IDR_ENDTX)); + break; + default: + dev_dbg(&chip->spi->dev, "spurious command %x\n", cmd); + retval = -EINVAL; + break; + } + + spin_unlock(&chip->lock); + + return retval; +} + +static snd_pcm_uframes_t +snd_at73c213_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; + unsigned long bytes; + + bytes = ssc_readl(chip->ssc->regs, PDC_TPR) + - (unsigned long)runtime->dma_addr; + + pos = bytes_to_frames(runtime, bytes); + if (pos >= runtime->buffer_size) + pos -= runtime->buffer_size; + + return pos; +} + +static struct snd_pcm_ops at73c213_playback_ops = { + .open = snd_at73c213_pcm_open, + .close = snd_at73c213_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_at73c213_pcm_hw_params, + .hw_free = snd_at73c213_pcm_hw_free, + .prepare = snd_at73c213_pcm_prepare, + .trigger = snd_at73c213_pcm_trigger, + .pointer = snd_at73c213_pcm_pointer, +}; + +static void snd_at73c213_pcm_free(struct snd_pcm *pcm) +{ + struct snd_at73c213 *chip = snd_pcm_chip(pcm); + if (chip->pcm) { + snd_pcm_lib_preallocate_free_for_all(chip->pcm); + chip->pcm = NULL; + } +} + +static int __devinit snd_at73c213_pcm_new(struct snd_at73c213 *chip, int device) +{ + struct snd_pcm *pcm; + int retval; + + retval = snd_pcm_new(chip->card, chip->card->shortname, + device, 1, 0, &pcm); + if (retval < 0) + goto out; + + pcm->private_data = chip; + pcm->private_free = snd_at73c213_pcm_free; + pcm->info_flags = SNDRV_PCM_INFO_BLOCK_TRANSFER; + strcpy(pcm->name, "at73c213"); + chip->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &at73c213_playback_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(chip->pcm, + SNDRV_DMA_TYPE_DEV, &chip->ssc->pdev->dev, + 64 * 1024, 64 * 1024); +out: + return retval; +} + +static irqreturn_t snd_at73c213_interrupt(int irq, void *dev_id) +{ + struct snd_at73c213 *chip = dev_id; + struct snd_pcm_runtime *runtime = chip->substream->runtime; + u32 status; + int offset; + int block_size; + int next_period; + int retval = IRQ_NONE; + + spin_lock(&chip->lock); + + block_size = frames_to_bytes(runtime, runtime->period_size); + status = ssc_readl(chip->ssc->regs, IMR); + + if (status & SSC_BIT(IMR_ENDTX)) { + chip->period++; + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + ssc_writel(chip->ssc->regs, PDC_TNPR, + (long)runtime->dma_addr + offset); + ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + retval = IRQ_HANDLED; + } + + ssc_readl(chip->ssc->regs, IMR); + spin_unlock(&chip->lock); + + if (status & SSC_BIT(IMR_ENDTX)) + snd_pcm_period_elapsed(chip->substream); + + return retval; +} + +/* + * Mixer functions. + */ +static int snd_at73c213_mono_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + + spin_lock_irq(&chip->lock); + + ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + (mask - ucontrol->value.integer.value[0]); + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_mono_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + int change, retval; + unsigned short val; + + val = (ucontrol->value.integer.value[0] & mask); + if (invert) + val = mask - val; + val <<= shift; + + spin_lock_irq(&chip->lock); + + val = (chip->reg_image[reg] & ~(mask << shift)) | val; + change = val != chip->reg_image[reg]; + retval = snd_at73c213_write_reg(chip, reg, val); + + spin_unlock_irq(&chip->lock); + + if (retval) + return retval; + + return change; +} + +static int snd_at73c213_stereo_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + if (mask == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + + return 0; +} + +static int snd_at73c213_stereo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x07; + int shift_right = (kcontrol->private_value >> 19) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + + spin_lock_irq(&chip->lock); + + ucontrol->value.integer.value[0] = + (chip->reg_image[left_reg] >> shift_left) & mask; + ucontrol->value.integer.value[1] = + (chip->reg_image[right_reg] >> shift_right) & mask; + + if (invert) { + ucontrol->value.integer.value[0] = + (mask - ucontrol->value.integer.value[0]); + ucontrol->value.integer.value[1] = + (mask - ucontrol->value.integer.value[1]); + } + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_stereo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x07; + int shift_right = (kcontrol->private_value >> 19) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + int change, retval; + unsigned short val1, val2; + + val1 = ucontrol->value.integer.value[0] & mask; + val2 = ucontrol->value.integer.value[1] & mask; + if (invert) { + val1 = mask - val1; + val2 = mask - val2; + } + val1 <<= shift_left; + val2 <<= shift_right; + + spin_lock_irq(&chip->lock); + + val1 = (chip->reg_image[left_reg] & ~(mask << shift_left)) | val1; + val2 = (chip->reg_image[right_reg] & ~(mask << shift_right)) | val2; + change = val1 != chip->reg_image[left_reg] + || val2 != chip->reg_image[right_reg]; + retval = snd_at73c213_write_reg(chip, left_reg, val1); + if (retval) { + spin_unlock_irq(&chip->lock); + goto out; + } + retval = snd_at73c213_write_reg(chip, right_reg, val2); + if (retval) { + spin_unlock_irq(&chip->lock); + goto out; + } + + spin_unlock_irq(&chip->lock); + + return change; + +out: + return retval; +} + +static int snd_at73c213_mono_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int snd_at73c213_mono_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + + spin_lock_irq(&chip->lock); + + ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & 0x01; + + if (invert) + ucontrol->value.integer.value[0] = + (0x01 - ucontrol->value.integer.value[0]); + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_mono_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + int change, retval; + unsigned short val; + + if (ucontrol->value.integer.value[0]) + val = mask; + else + val = 0; + + if (invert) + val = mask - val; + val <<= shift; + + spin_lock_irq(&chip->lock); + + val |= (chip->reg_image[reg] & ~(mask << shift)); + change = val != chip->reg_image[reg]; + + retval = snd_at73c213_write_reg(chip, reg, val); + + spin_unlock_irq(&chip->lock); + + if (retval) + return retval; + + return change; +} + +static int snd_at73c213_pa_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = ((kcontrol->private_value >> 16) & 0xff) - 1; + + return 0; +} + +static int snd_at73c213_line_capture_volume_info( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + /* When inverted will give values 0x10001 => 0. */ + uinfo->value.integer.min = 14; + uinfo->value.integer.max = 31; + + return 0; +} + +static int snd_at73c213_aux_capture_volume_info( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + /* When inverted will give values 0x10001 => 0. */ + uinfo->value.integer.min = 14; + uinfo->value.integer.max = 31; + + return 0; +} + +#define AT73C213_MONO_SWITCH(xname, xindex, reg, shift, mask, invert) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_at73c213_mono_switch_info, \ + .get = snd_at73c213_mono_switch_get, \ + .put = snd_at73c213_mono_switch_put, \ + .private_value = (reg | (shift << 8) | (mask << 16) | (invert << 24)) \ +} + +#define AT73C213_STEREO(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_at73c213_stereo_info, \ + .get = snd_at73c213_stereo_get, \ + .put = snd_at73c213_stereo_put, \ + .private_value = (left_reg | (right_reg << 8) \ + | (shift_left << 16) | (shift_right << 19) \ + | (mask << 24) | (invert << 22)) \ +} + +static struct snd_kcontrol_new snd_at73c213_controls[] __devinitdata = { +AT73C213_STEREO("Master Playback Volume", 0, DAC_LMPG, DAC_RMPG, 0, 0, 0x1f, 1), +AT73C213_STEREO("Master Playback Switch", 0, DAC_LMPG, DAC_RMPG, 5, 5, 1, 1), +AT73C213_STEREO("PCM Playback Volume", 0, DAC_LLOG, DAC_RLOG, 0, 0, 0x1f, 1), +AT73C213_STEREO("PCM Playback Switch", 0, DAC_LLOG, DAC_RLOG, 5, 5, 1, 1), +AT73C213_MONO_SWITCH("Mono PA Playback Switch", 0, DAC_CTRL, DAC_CTRL_ONPADRV, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PA Playback Volume", + .index = 0, + .info = snd_at73c213_pa_volume_info, + .get = snd_at73c213_mono_get, + .put = snd_at73c213_mono_put, + .private_value = PA_CTRL | (PA_CTRL_APAGAIN << 8) | (0x0f << 16) | (1 << 24), +}, +AT73C213_MONO_SWITCH("PA High Gain Playback Switch", 0, PA_CTRL, PA_CTRL_APALP, 0x01, 1), +AT73C213_MONO_SWITCH("PA Playback Switch", 0, PA_CTRL, PA_CTRL_APAON, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Capture Volume", + .index = 0, + .info = snd_at73c213_aux_capture_volume_info, + .get = snd_at73c213_mono_get, + .put = snd_at73c213_mono_put, + .private_value = DAC_AUXG | (0 << 8) | (0x1f << 16) | (1 << 24), +}, +AT73C213_MONO_SWITCH("Aux Capture Switch", 0, DAC_CTRL, DAC_CTRL_ONAUXIN, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Volume", + .index = 0, + .info = snd_at73c213_line_capture_volume_info, + .get = snd_at73c213_stereo_get, + .put = snd_at73c213_stereo_put, + .private_value = DAC_LLIG | (DAC_RLIG << 8) | (0 << 16) | (0 << 19) + | (0x1f << 24) | (1 << 22), +}, +AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0), +}; + +static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) +{ + struct snd_card *card; + int errval, idx; + + if (chip == NULL || chip->pcm == NULL) + return -EINVAL; + + card = chip->card; + + strcpy(card->mixername, chip->pcm->name); + + for (idx = 0; idx < ARRAY_SIZE(snd_at73c213_controls); idx++) { + errval = snd_ctl_add(card, + snd_ctl_new1(&snd_at73c213_controls[idx], + chip)); + if (errval < 0) + goto cleanup; + } + + return 0; + +cleanup: + for (idx = 1; idx < ARRAY_SIZE(snd_at73c213_controls) + 1; idx++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_find_numid(card, idx); + if (kctl) + snd_ctl_remove(card, kctl); + } + return errval; +} + +/* + * Device functions + */ +static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) +{ + /* + * Continuous clock output. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Periode is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + /* + * Data length is 16 bit (16 - 1). + * Transmit MSB first. + * Transmit 2 words each transfer. + * Frame sync length is 16 bit (16 - 1). + * Frame starts on negative pulse. + */ + ssc_writel(chip->ssc->regs, TFMR, + SSC_BF(TFMR_DATLEN, 16 - 1) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATNB, 1) + | SSC_BF(TFMR_FSLEN, 16 - 1) + | SSC_BF(TFMR_FSOS, 1)); + + return 0; +} + +static int snd_at73c213_chip_init(struct snd_at73c213 *chip) +{ + int retval; + unsigned char dac_ctrl = 0; + + retval = snd_at73c213_set_bitrate(chip); + if (retval) + goto out; + + /* Enable DAC master clock. */ + clk_enable(chip->board->dac_clk); + + /* Initialize at73c213 on SPI bus. */ + retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); + if (retval) + goto out_clk; + msleep(1); + retval = snd_at73c213_write_reg(chip, DAC_RST, 0x03); + if (retval) + goto out_clk; + + /* Precharge everything. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0xff); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APAPRECH)); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_CTRL, + (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR)); + if (retval) + goto out_clk; + + msleep(50); + + /* Stop precharging PA. */ + retval = snd_at73c213_write_reg(chip, PA_CTRL, + (1<<PA_CTRL_APALP) | 0x0f); + if (retval) + goto out_clk; + + msleep(450); + + /* Stop precharging DAC, turn on master power. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, (1<<DAC_PRECH_ONMSTR)); + if (retval) + goto out_clk; + + msleep(1); + + /* Turn on DAC. */ + dac_ctrl = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) + | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR); + + retval = snd_at73c213_write_reg(chip, DAC_CTRL, dac_ctrl); + if (retval) + goto out_clk; + + /* Mute sound. */ + retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); + if (retval) + goto out_clk; + + /* Enable I2S device, i.e. clock output. */ + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); + + goto out; + +out_clk: + clk_disable(chip->board->dac_clk); +out: + return retval; +} + +static int snd_at73c213_dev_free(struct snd_device *device) +{ + struct snd_at73c213 *chip = device->device_data; + + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + if (chip->irq >= 0) { + free_irq(chip->irq, chip); + chip->irq = -1; + } + + return 0; +} + +static int __devinit snd_at73c213_dev_init(struct snd_card *card, + struct spi_device *spi) +{ + static struct snd_device_ops ops = { + .dev_free = snd_at73c213_dev_free, + }; + struct snd_at73c213 *chip = get_chip(card); + int irq, retval; + + irq = chip->ssc->irq; + if (irq < 0) + return irq; + + spin_lock_init(&chip->lock); + chip->card = card; + chip->irq = -1; + + retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); + if (retval) { + dev_dbg(&chip->spi->dev, "unable to request irq %d\n", irq); + goto out; + } + chip->irq = irq; + + memcpy(&chip->reg_image, &snd_at73c213_original_image, + sizeof(snd_at73c213_original_image)); + + retval = snd_at73c213_ssc_init(chip); + if (retval) + goto out_irq; + + retval = snd_at73c213_chip_init(chip); + if (retval) + goto out_irq; + + retval = snd_at73c213_pcm_new(chip, 0); + if (retval) + goto out_irq; + + retval = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (retval) + goto out_irq; + + retval = snd_at73c213_mixer(chip); + if (retval) + goto out_snd_dev; + + snd_card_set_dev(card, &spi->dev); + + goto out; + +out_snd_dev: + snd_device_free(card, chip); +out_irq: + free_irq(chip->irq, chip); + chip->irq = -1; +out: + return retval; +} + +static int snd_at73c213_probe(struct spi_device *spi) +{ + struct snd_card *card; + struct snd_at73c213 *chip; + struct at73c213_board_info *board; + int retval; + char id[16]; + + board = spi->dev.platform_data; + if (!board) { + dev_dbg(&spi->dev, "no platform_data\n"); + return -ENXIO; + } + + if (!board->dac_clk) { + dev_dbg(&spi->dev, "no DAC clk\n"); + return -ENXIO; + } + + if (IS_ERR(board->dac_clk)) { + dev_dbg(&spi->dev, "no DAC clk\n"); + return PTR_ERR(board->dac_clk); + } + + retval = -ENOMEM; + + /* Allocate "card" using some unused identifiers. */ + snprintf(id, sizeof id, "at73c213_%d", board->ssc_id); + card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct snd_at73c213)); + if (!card) + goto out; + + chip = card->private_data; + chip->spi = spi; + chip->board = board; + + chip->ssc = ssc_request(board->ssc_id); + if (IS_ERR(chip->ssc)) { + dev_dbg(&spi->dev, "could not get ssc%d device\n", + board->ssc_id); + retval = PTR_ERR(chip->ssc); + goto out_card; + } + + retval = snd_at73c213_dev_init(card, spi); + if (retval) + goto out_ssc; + + strcpy(card->driver, "at73c213"); + strcpy(card->shortname, board->shortname); + sprintf(card->longname, "%s on irq %d", card->shortname, chip->irq); + + retval = snd_card_register(card); + if (retval) + goto out_ssc; + + dev_set_drvdata(&spi->dev, card); + + goto out; + +out_ssc: + ssc_free(chip->ssc); +out_card: + snd_card_free(card); +out: + return retval; +} + +static int __devexit snd_at73c213_remove(struct spi_device *spi) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + int retval; + + /* Stop playback. */ + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + + /* Mute sound. */ + retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); + if (retval) + goto out; + + /* Turn off PA. */ + retval = snd_at73c213_write_reg(chip, PA_CTRL, (chip->reg_image[PA_CTRL]|0x0f)); + if (retval) + goto out; + msleep(10); + retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APALP)|0x0f); + if (retval) + goto out; + + /* Turn off external DAC. */ + retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x0c); + if (retval) + goto out; + msleep(2); + retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x00); + if (retval) + goto out; + + /* Turn off master power. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0x00); + if (retval) + goto out; + +out: + /* Stop DAC master clock. */ + clk_disable(chip->board->dac_clk); + + ssc_free(chip->ssc); + snd_card_free(card); + dev_set_drvdata(&spi->dev, NULL); + + return 0; +} + +#ifdef CONFIG_PM +static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + clk_disable(chip->board->dac_clk); + + return 0; +} + +static int snd_at73c213_resume(struct spi_device *spi) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + + clk_enable(chip->board->dac_clk); + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); + + return 0; +} +#else +#define snd_at73c213_suspend NULL +#define snd_at73c213_resume NULL +#endif + +static struct spi_driver at73c213_driver = { + .driver = { + .name = "at73c213", + }, + .probe = snd_at73c213_probe, + .suspend = snd_at73c213_suspend, + .resume = snd_at73c213_resume, + .remove = __devexit_p(snd_at73c213_remove), +}; + +static int __init at73c213_init(void) +{ + return spi_register_driver(&at73c213_driver); +} +module_init(at73c213_init); + +static void __exit at73c213_exit(void) +{ + spi_unregister_driver(&at73c213_driver); +} +module_exit(at73c213_exit); + +MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt(a)atmel.com>"); +MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); +MODULE_LICENSE("GPL"); diff --git a/sound/spi/at73c213.h b/sound/spi/at73c213.h new file mode 100644 index 0000000..fd8b372 --- /dev/null +++ b/sound/spi/at73c213.h @@ -0,0 +1,119 @@ +/* + * Driver for the AT73C213 16-bit stereo DAC on Atmel ATSTK1000 + * + * Copyright (C) 2006 - 2007 Atmel Corporation + * + * 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; either version 2 of the + * License, or (at your option) any later version. + * + * 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. + * + * The full GNU General Public License is included in this + * distribution in the file called COPYING. + */ + +#ifndef _SND_AT73C213_H +#define _SND_AT73C213_H + +/* DAC control register */ +#define DAC_CTRL 0x00 +#define DAC_CTRL_ONPADRV 7 +#define DAC_CTRL_ONAUXIN 6 +#define DAC_CTRL_ONDACR 5 +#define DAC_CTRL_ONDACL 4 +#define DAC_CTRL_ONLNOR 3 +#define DAC_CTRL_ONLNOL 2 +#define DAC_CTRL_ONLNIR 1 +#define DAC_CTRL_ONLNIL 0 + +/* DAC left line in gain register */ +#define DAC_LLIG 0x01 +#define DAC_LLIG_LLIG 0 + +/* DAC right line in gain register */ +#define DAC_RLIG 0x02 +#define DAC_RLIG_RLIG 0 + +/* DAC Left Master Playback Gain Register */ +#define DAC_LMPG 0x03 +#define DAC_LMPG_LMPG 0 + +/* DAC Right Master Playback Gain Register */ +#define DAC_RMPG 0x04 +#define DAC_RMPG_RMPG 0 + +/* DAC Left Line Out Gain Register */ +#define DAC_LLOG 0x05 +#define DAC_LLOG_LLOG 0 + +/* DAC Right Line Out Gain Register */ +#define DAC_RLOG 0x06 +#define DAC_RLOG_RLOG 0 + +/* DAC Output Level Control Register */ +#define DAC_OLC 0x07 +#define DAC_OLC_RSHORT 7 +#define DAC_OLC_ROLC 4 +#define DAC_OLC_LSHORT 3 +#define DAC_OLC_LOLC 0 + +/* DAC Mixer Control Register */ +#define DAC_MC 0x08 +#define DAC_MC_INVR 5 +#define DAC_MC_INVL 4 +#define DAC_MC_RMSMIN2 3 +#define DAC_MC_RMSMIN1 2 +#define DAC_MC_LMSMIN2 1 +#define DAC_MC_LMSMIN1 0 + +/* DAC Clock and Sampling Frequency Control Register */ +#define DAC_CSFC 0x09 +#define DAC_CSFC_OVRSEL 4 + +/* DAC Miscellaneous Register */ +#define DAC_MISC 0x0A +#define DAC_MISC_VCMCAPSEL 7 +#define DAC_MISC_DINTSEL 4 +#define DAC_MISC_DITHEN 3 +#define DAC_MISC_DEEMPEN 2 +#define DAC_MISC_NBITS 0 + +/* DAC Precharge Control Register */ +#define DAC_PRECH 0x0C +#define DAC_PRECH_PRCHGPDRV 7 +#define DAC_PRECH_PRCHGAUX1 6 +#define DAC_PRECH_PRCHGLNOR 5 +#define DAC_PRECH_PRCHGLNOL 4 +#define DAC_PRECH_PRCHGLNIR 3 +#define DAC_PRECH_PRCHGLNIL 2 +#define DAC_PRECH_PRCHG 1 +#define DAC_PRECH_ONMSTR 0 + +/* DAC Auxiliary Input Gain Control Register */ +#define DAC_AUXG 0x0D +#define DAC_AUXG_AUXG 0 + +/* DAC Reset Register */ +#define DAC_RST 0x10 +#define DAC_RST_RESMASK 2 +#define DAC_RST_RESFILZ 1 +#define DAC_RST_RSTZ 0 + +/* Power Amplifier Control Register */ +#define PA_CTRL 0x11 +#define PA_CTRL_APAON 6 +#define PA_CTRL_APAPRECH 5 +#define PA_CTRL_APALP 4 +#define PA_CTRL_APAGAIN 0 + +#endif /* _SND_AT73C213_H */ -- 1.5.2.3 --=-zhf1GaBSeEtL5hs3rxas--

1 0

[alsa-devel] error in snd_pcm_drain()
by Eliot Blennerhassett 05 Jul '07

05 Jul '07

Maybe an error in snd_pcm_drain - not sure what effect it might have... snd_pcm_group_for_each_entry(s, substream) { runtime = s->runtime; ... } up_read(&snd_pcm_link_rwsem); snd_pcm_stream_lock_irq(substream); /* pcm_native.c around line 1491: */ /* error? runtime has been changed by group_for_each should use substream->runtime instead? */ /* resume pause */ if (runtime->status->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, 0); -- Eliot

2 1

[alsa-devel] [PATCH]fix mmap emulation bug of recording doesn't work
by Roy Huang 05 Jul '07

05 Jul '07

Record doesn't work if enabling mmap emulation and rate conversion needed, this patch fix this bug. diff -uNr alsa-lib-1.0.14a/src/pcm/pcm_hw.c alsa-lib-new/src/pcm/pcm_hw.c --- alsa-lib-1.0.14a/src/pcm/pcm_hw.c 2007-06-11 16:53:13.000000000 +0800 +++ alsa-lib-new/src/pcm/pcm_hw.c 2007-07-04 12:16:31.000000000 +0800 @@ -96,6 +96,7 @@ int shadow_appl_ptr: 1, avail_update_flag: 1, mmap_shm: 1; + snd_pcm_uframes_t hw_ptr; snd_pcm_uframes_t appl_ptr; /* restricted parameters */ snd_pcm_format_t format; @@ -373,7 +374,9 @@ if (pcm->stream == SND_PCM_STREAM_CAPTURE) { if (hw->mmap_shm) { hw->shadow_appl_ptr = 1; + hw->hw_ptr = 0; hw->appl_ptr = 0; + snd_pcm_set_hw_ptr(pcm, &hw->hw_ptr, -1, 0); snd_pcm_set_appl_ptr(pcm, &hw->appl_ptr, -1, 0); } else { hw->shadow_appl_ptr = 0; @@ -949,28 +952,91 @@ return size; } +static inline snd_pcm_uframes_t snd_pcm_hw_capture_avail(snd_pcm_t *pcm) +{ + snd_pcm_sframes_t avail; + snd_pcm_hw_t *hw = pcm->private_data; + + avail = hw->mmap_status->hw_ptr - hw->mmap_control->appl_ptr; + if (avail < 0) + avail += pcm->boundary; + return avail; +} + +static snd_pcm_sframes_t snd_pcm_hw_read_mmap(snd_pcm_t *pcm, snd_pcm_uframes_t size) +{ + snd_pcm_uframes_t xfer = 0; + snd_pcm_sframes_t err = 0; + snd_pcm_hw_t *hw = pcm->private_data; + if (! size) + return 0; + while (xfer < size) { + snd_pcm_uframes_t frames = size - xfer; + snd_pcm_uframes_t appl_offset = hw->mmap_control->appl_ptr % pcm->buffer_size; + snd_pcm_uframes_t cont = pcm->buffer_size - appl_offset; + if (cont < frames) + frames = cont; + switch (pcm->access) { + case SND_PCM_ACCESS_MMAP_INTERLEAVED: + { + const snd_pcm_channel_area_t *a = snd_pcm_mmap_areas(pcm); + char *buf = snd_pcm_channel_area_addr(a, appl_offset); + err = _snd_pcm_readi(pcm, buf, frames); + if (err >= 0) + frames = err; + break; + } + case SND_PCM_ACCESS_MMAP_NONINTERLEAVED: + { + snd_pcm_uframes_t channels = pcm->channels; + unsigned int c; + void *bufs[channels]; + const snd_pcm_channel_area_t *areas = snd_pcm_mmap_areas(pcm); + for (c = 0; c < channels; ++c) { + const snd_pcm_channel_area_t *a = &areas[c]; + bufs[c] = snd_pcm_channel_area_addr(a, appl_offset); + } + err = _snd_pcm_readn(pcm->fast_op_arg, bufs, frames); + if (err >= 0) + frames = err; + } + default: + SNDMSG("invalid access type %d", pcm->access); + return -EINVAL; + } + if (err < 0) + break; + xfer += frames; + } + if (xfer > 0) + return xfer; + return err; +} + static snd_pcm_sframes_t snd_pcm_hw_avail_update(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; - snd_pcm_uframes_t avail; + snd_pcm_uframes_t avail, xfer_avail; sync_ptr(hw, 0); if (pcm->stream == SND_PCM_STREAM_PLAYBACK) { avail = snd_pcm_mmap_playback_avail(pcm); } else { avail = snd_pcm_mmap_capture_avail(pcm); - if (avail > 0 && hw->mmap_shm) { + if (avail < pcm->avail_min && hw->mmap_shm) { snd_pcm_sframes_t err; - snd_pcm_hw_t *hw = pcm->private_data; - hw->avail_update_flag = 1; - err = snd_pcm_read_mmap(pcm, avail); - hw->avail_update_flag = 0; - if (err < 0) - return err; - if ((snd_pcm_uframes_t)err != avail) - SNDMSG("short read %ld for avail %ld", err, avail); - return err; - } + xfer_avail = snd_pcm_hw_capture_avail(pcm); + xfer_avail -= xfer_avail % pcm->xfer_align; + if (xfer_avail > 0) { + hw->avail_update_flag = 1; + err = snd_pcm_hw_read_mmap(pcm, xfer_avail); + hw->avail_update_flag = 0; + if (err < 0) + return err; + hw->hw_ptr += err; + avail = snd_pcm_mmap_capture_avail(pcm); + } + } } switch (FAST_PCM_STATE(hw)) { case SNDRV_PCM_STATE_RUNNING:

2 4

[alsa-devel] default setting not fully working with new ALSA device
by Peter Matthias 05 Jul '07

05 Jul '07

Hello, I have got a 690G board with following lspic -nv output: 01:05.2 0403: 1002:7919 Subsystem: 1002:7919 Flags: bus master, fast devsel, latency 64, IRQ 19 Memory at fe9e8000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- With the default setting, microphone does not work. With module parameter "modprobe -i snd-hda-intel model=6stack-dig" microphone works well. Please add to default setting. It was written in Linux Kernel documentation to report this issue here. Thank you, Peter

2 1

[alsa-devel] Very low headphones volume on HDA + Realtek ALC268
by Maurizio Monge 04 Jul '07

04 Jul '07

Hi, i've got an hda/alc268 card (pci id: [8086:284b], subsytem id: [1025:011f]) on my laptop, and the sound is ok for the loudspeakers but it is very very low for the headphones (the volume is at the top in the mixer, if i reduce the volume in the mixer the sound gets even lower). Under windows vista the headphones work well (ok, worked, before that i removed windows from my computer). I am using alsa hg, the most recent version at the moment i'm writing this e-mail. I there is any experiment that i can do or if i can help in any way (i know C very well) to fix this bug just let me know, and i'll be happy to contribute. More information about the hardware follows. Thanks! -- Ciao Maurizio http://stregatto.wordpress.com "Well we all shine on Like the moon and the stars and the sun" (John Lennon) lspci entry: 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: Acer Incorporated [ALI] Unknown device [1025:011f] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fc300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1025011f Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2f 0x2f] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x02214020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a19830: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281303e: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4017952d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x90 0x90] Connection: 7 0x18 0x19 0x1a 0x1c 0x14 0x15* 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x01 0x01] Connection: 7 0x18 0x19 0x1a 0x1c 0x14* 0x15 0x13 /proc/asound/card0/codec#1: Codec: Conexant ID 2c06 Address: 1 Vendor Id: 0x14f12c06 Subsystem Id: 0x10250093 Revision Id: 0x100000 Modem Function Group: 0x2

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[alsa-devel] Easiest way to test playback on an ASOC driver?
by Timur Tabi 04 Jul '07

04 Jul '07

I'm in the early stages of testing an ASOC driver, i.e. I'm trying to get playback to work. What is the easiest way to test playback? Is there a particular tool I should be using that does the most basic and direct playback of audio data through ALSA? -- Timur Tabi Linux Kernel Developer @ Freescale

2 1

[alsa-devel] left/right bleed or crossover
by Howdy Pierce 04 Jul '07

04 Jul '07

I'm encountering the following problem on Fedora Core 6 running on a Dell Optiplex 745. This is a pretty vanilla computer with an ICH8 chipset and integrated AD1983 sound capabilities. I have wired up a funky line-input cable so that I have completely separate audio sources playing on the left and right inputs. When I either play these through to the speaker output directly (using alsamixer), or record the audio using arecord, the result is that there is a slight bit of bleeding or crossover, so I can faintly hear the left input on the right channel, and vice versa. If I record a wav file and then play it on another computer, the left/right bleeding follows the wav file, so it seems that the problem is in the capture process somehow. When I install Windows XP onto the same machine, the problem disappears, so I can rule out hardware errors. The application I'm working on relies on being able to reliably capture the right and left channels of audio independently. Since the problem can be seen using the standard arecord/aplay tools, though, I don't think it's my code. I've tried all the possible combinations inside alsamixer, and I don't think there's anything I've messed up there. I've put the relevant system details below. Thanks for any help you can provide. --Howdy ============================= Howdy Pierce Managing Partner Cardinal Peak, LLC work: (303) 665-3962 x101 cell: (303) 589-1645 email: howdy(a)cardinalpeak.com web: www.cardinalpeak.com ============================= [root@dundee ~]# cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xdfdfc000 irq 169 [root@dundee ~]# cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 1]: digital audio playback 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0] : control [root@dundee ~]# lspci 00:00.0 Host bridge: Intel Corporation 82Q963/Q965 Memory Controller Hub (rev 02) 00:01.0 PCI bridge: Intel Corporation 82Q963/Q965 PCI Express Root Port (rev 02) 00:02.0 Display controller: Intel Corporation 82Q963/Q965 Integrated Graphics Controller (rev 02) 00:02.1 Display controller: Intel Corporation 82Q963/Q965 Integrated Graphics Controller (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.4 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 5 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Controller (rev 02) 03:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5754 Gigabit Ethernet PCI Express (rev 02) 04:02.0 VGA compatible controller: nVidia Corporation NV18 [GeForce4 MX 4000 AGP 8x] (rev c1) [root@dundee ~]# rpm --query --all | grep alsa alsa-lib-1.0.12-2.fc6 alsa-lib-devel-1.0.12-2.fc6 alsa-utils-1.0.12-3.fc6 [root@dundee Intel]# cat /proc/asound/Intel/codec#0 Codec: Analog Devices AD1983 Address: 0 Vendor Id: 0x11d41983 Subsystem Id: 0x102801da Revision Id: 0x100400 Default PCM: rates 0x07f, bits 0x0e, types 0x1 Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates 0x060, bits 0x02, types 0x5 Connection: 2 0x01* 0x04 Node 0x03 [Audio Output] wcaps 0x441: Stereo Power: 0x0 Node 0x04 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x14 Node 0x05 [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x32 0x32] Pincap 0x0817: OUT Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 2 0x03 0x0e* Node 0x06 [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x25 0x25] Pincap 0x081f: OUT HP Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 2 0x03 0x0e* Node 0x07 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x991301f0: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x08 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a11020: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Node 0x09 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x0181302e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x0a [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x414511f0: [N/A] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x03 0x0c 0x0d 0x0e* Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x09* 0x08 Node 0x0e [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x11 0x12 0x13 Node 0x0f [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0b Node 0x10 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x80] Node 0x11 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x17] Connection: 1 0x03 Node 0x12 [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 0x0c Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x0d Node 0x14 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x0f 0x0f] Connection: 4 0x0c 0x0d* 0x0e 0x0f Node 0x15 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 15 0x05 0x06* 0x07 0x08 0x09 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14

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[alsa-devel] PCMCIA Audigy 2 broken in 2.6.21?
by Chuck Ebbert 04 Jul '07

04 Jul '07

Multiple users report the PCMCIA Audigy 2 adapter hangs the machine in 2.6.21 while it work OK in 2.6.20: https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=242208

3 3

[alsa-devel] [PATCH] Improve behaviour of Dreamcast aica ALSA driver in poor resource conditions
by Adrian McMenamin 04 Jul '07

04 Jul '07

This patch stops the driver from crashing in certain situations (eg if the network fails when NFS mounted), please apply. Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> --- alsa-kernel/sh/aica.c 2007-06-23 15:25:55.000000000 +0100 +++ linux-2.6.21/sound/sh/aica.c 2007-07-03 22:28:16.000000000 +0100 @@ -366,7 +366,9 @@ *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + flush_workqueue(aica_queue); del_timer(&dreamcastcard->timer); + aica_chn_halt(); kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -402,16 +404,11 @@ static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_card_aica *dreamcastcard; switch (cmd) { case SNDRV_PCM_TRIGGER_START: spu_begin_dma(substream); break; case SNDRV_PCM_TRIGGER_STOP: - dreamcastcard = substream->pcm->private_data; - if (dreamcastcard->timer.data) - del_timer(&dreamcastcard->timer); - aica_chn_halt(); break; default: return -EINVAL;

2 2

[alsa-devel] Errors in application building
by Paul Kavan 03 Jul '07

03 Jul '07

Hello all: I am working on my first bit of application code for ALSA and am not sure how to properly compile it against the libs. I have to cross compile the application on my host machine i686 PC for use on a ARM development board. I copied the alsa include directory into my staging include directory (this is where the cross-compiler works from), which allowed the program to find alsa/asoundlib.h However, I am still getting some errors that indicate to me it cannot find everything it needs. I am getting the following errors: grhuser@pcL7:$ arm-linux-gcc -Wall -o grh_audio grh_audio.c grh_audio.c: In function 'grh_capture': grh_audio.c:113: warning: pointer targets in passing argument 3 of 'snd_pcm_hw_params_set_rate_near' differ in signedness grh_audio.c: In function 'grh_playback': grh_audio.c:208: warning: pointer targets in passing argument 3 of 'snd_pcm_hw_params_set_rate_near' differ in signedness /tmp/ccMn48zN.o: In function `grh_capture': grh_audio.c:(.text+0x16c): undefined reference to `snd_pcm_open' grh_audio.c:(.text+0x198): undefined reference to `snd_strerror' grh_audio.c:(.text+0x1c0): undefined reference to `snd_pcm_hw_params_malloc' grh_audio.c:(.text+0x1e4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x214): undefined reference to `snd_pcm_hw_params_any' grh_audio.c:(.text+0x238): undefined reference to `snd_strerror' grh_audio.c:(.text+0x26c): undefined reference to `snd_pcm_hw_params_set_access' grh_audio.c:(.text+0x290): undefined reference to `snd_strerror' grh_audio.c:(.text+0x2c4): undefined reference to `snd_pcm_hw_params_set_format' grh_audio.c:(.text+0x2e8): undefined reference to `snd_strerror' grh_audio.c:(.text+0x324): undefined reference to `snd_pcm_hw_params_set_rate_near' grh_audio.c:(.text+0x348): undefined reference to `snd_strerror' grh_audio.c:(.text+0x380): undefined reference to `snd_pcm_hw_params_set_channels' grh_audio.c:(.text+0x3a4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x3d4): undefined reference to `snd_pcm_hw_params' grh_audio.c:(.text+0x3f8): undefined reference to `snd_strerror' grh_audio.c:(.text+0x420): undefined reference to `snd_pcm_hw_params_free' grh_audio.c:(.text+0x42c): undefined reference to `snd_pcm_prepare' grh_audio.c:(.text+0x450): undefined reference to `snd_strerror' grh_audio.c:(.text+0x494): undefined reference to `snd_pcm_readi' grh_audio.c:(.text+0x4c0): undefined reference to `snd_strerror' grh_audio.c:(.text+0x500): undefined reference to `snd_pcm_close' /tmp/ccMn48zN.o: In function `grh_playback': grh_audio.c:(.text+0x5b8): undefined reference to `snd_pcm_open' grh_audio.c:(.text+0x5e4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x60c): undefined reference to `snd_pcm_hw_params_malloc' grh_audio.c:(.text+0x630): undefined reference to `snd_strerror' grh_audio.c:(.text+0x660): undefined reference to `snd_pcm_hw_params_any' grh_audio.c:(.text+0x684): undefined reference to `snd_strerror' grh_audio.c:(.text+0x6b8): undefined reference to `snd_pcm_hw_params_set_access' grh_audio.c:(.text+0x6dc): undefined reference to `snd_strerror' grh_audio.c:(.text+0x710): undefined reference to `snd_pcm_hw_params_set_format' grh_audio.c:(.text+0x734): undefined reference to `snd_strerror' grh_audio.c:(.text+0x770): undefined reference to `snd_pcm_hw_params_set_rate_near' grh_audio.c:(.text+0x794): undefined reference to `snd_strerror' grh_audio.c:(.text+0x7cc): undefined reference to `snd_pcm_hw_params_set_channels' grh_audio.c:(.text+0x7f0): undefined reference to `snd_strerror' grh_audio.c:(.text+0x820): undefined reference to `snd_pcm_hw_params' grh_audio.c:(.text+0x844): undefined reference to `snd_strerror' grh_audio.c:(.text+0x86c): undefined reference to `snd_pcm_hw_params_free' grh_audio.c:(.text+0x878): undefined reference to `snd_pcm_prepare' grh_audio.c:(.text+0x89c): undefined reference to `snd_strerror' grh_audio.c:(.text+0x8e0): undefined reference to `snd_pcm_writei' grh_audio.c:(.text+0x90c): undefined reference to `snd_strerror' grh_audio.c:(.text+0x94c): undefined reference to `snd_pcm_close' collect2: ld returned 1 exit status My application code is attached. If anyone could give me some pointers as to why these errors are occurring, I would appreciate it greatly. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

2 4

[alsa-devel] namehint problem...
by Fausto Carvalho Marques SIlva 03 Jul '07

03 Jul '07

Compiling alsa-lib-1.0.14rc4 with the above options: ./configure --disable-aload --disable-alisp --disable-instr --disable-seq --disable-hwdep --disable-rawmidi --with-debug=no --with-pcm-plugins=hw --with-ctl-plugins=hw After the build of asound and a linking against a simple program: #include <stdio.h> #include <stdlib.h> #include <alsa/asoundlib.h> main (int argc, char *argv[]) { int i; int err; short buf[128]; snd_pcm_t *playback_handle; snd_pcm_hw_params_t *hw_params; if ((err = snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0)) < 0) { fprintf (stderr, "cannot open audio device %s (%s)\n", argv[1], snd_strerror (err)); exit (1); } snd_pcm_close (playback_handle); exit (0); } gcc -o teste teste.c -I./include src/.libs/libasound.so.2.0.0 -ldl src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_get_name' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_get_name' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_sizeof' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_set_device' src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_set_device' src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_sizeof' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_set_stream' collect2: ld returned 1 exit status After some verification discoverd that the problem is on src/control/namehint.c that references this functions but the configure removed rawmidi and hwdep. How can we resolve this problem? Fausto Carvalho Marques Silva

2 1

[alsa-devel] Application Coding
by Paul Kavan 03 Jul '07

03 Jul '07

Hello all: Beginning to write my application code now that I have alsa SoC working fairly well on my ARM development board. My audio application is fairly simple. I will be doing raw, S8, samples at an 8k sample rate. Essentially I need to do playback of a stream when signaled by an application to do so and capture when signaled by a button press. Capture takes precedent over playback. I am really new to the alsa api and I am planning on using aplay as a template. However, if anyone knows of an easier piece of code that may work as a template, please feel free to point it out to me. I am using the API documentation from the alsa-project website. If anyone knows of another good piece of documentation, please feel free to let me know. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] SALSA-Lib: snd_pcm_hw_params fails with MMAP
by J. Scott Merritt 03 Jul '07

03 Jul '07

Dear List, Using SALSA-Lib 0.0.5 ... After calling snd_pcm_hw_params_set_access with an access mode of SND_PCM_ACCESS_MMAP_INTERLEAVED, the subsequent call to snd_pcm_hw_params returns a snd_strerror error code of "Invalid argument". This same program operates properly in the target environment (ARM PXA270) when compiled and linked with the standard ALSA LIB 1.0.13. Thanks, Scott.

2 2

Re: [alsa-devel] ALSA - snd-hda-intel - Realtek ALC268
by Peter Poulsen 03 Jul '07

03 Jul '07

Citat af Chris Cheney <chris.cheney(a)canonical.com>: <snip> > I already have alc268 working fine on my box for sound, just tested it. > :) I am now working on making sure all the other functionality is > properly implemented. Sounds awesome :) looking forward to trying out your code /peter

1 0

[alsa-devel] roland synth sh-201
by carlo beccaria 03 Jul '07

03 Jul '07

hi all, with this change in usbquirks.h , i can use my synth roland sh-201 { /* Roland SH-201 */ USB_DEVICE(0x0582, 0x00ad), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Roland", .product_name = "SH-201", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x01, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x81, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 2, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0007, .in_cables = 0x0007 } }, { .ifnum = -1 } } } }, my kernel is 2.6.21.5-rt17, (plus preempt patch) and i use debian testing the hardware seem to be very similar to Roland SC-D70 (windows .inf file are very similar) this is the device id Bus 001 Device 003: ID 0582:00ad Roland Corp. in attach you can see the lsusb (verbose) output becks@vaix:~$ amidi -l Dir Device Name IO hw:1,0,0 SH-201 MIDI 1 IO hw:1,0,1 SH-201 MIDI 2 IO hw:1,0,2 SH-201 MIDI 3 muse works fine with jack and midi output becks@vaix:~$ aseqdump -p 20:0 Waiting for data. Press Ctrl+C to end. Source_ Event_________________ Ch _Data__ 20:0 Note on 1 52 17 20:0 Note off 1 52 55 also midi-in works becks@vaix:~$ cat /proc/asound/cards 0 [A5451 ]: ALI5451 - ALI 5451 ALI 5451 at 0x8400, irq 9 1 [SH201 ]: USB-Audio - SH-201 Roland SH-201 at usb-0000:00:0c.0-1, full speed when i use my synth like a sound card, it don't work, i don't know exactly why (if is a patch problem, or rate/format) becks@vaix:~$ aplay -D hw:1,0 /usr/share/sounds/KDE_Window_UnSticky.wav Playing WAVE '/usr/share/sounds/KDE_Window_UnSticky.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono aplay: set_params:904: Sample format non available let me know if you need more info thanks -- carlo beccaria

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Re: [alsa-devel] [HDA-Intel/ALC268] no sound
by Steven Zakulec 03 Jul '07

03 Jul '07

I've attached the file. I tried using aplay and ffplay- both had no trouble opening and playing a file, there was just no sound. --------------------------------------- Original E-mail From: Takashi Iwai [tiwai(a)suse.de] Date: 07/02/2007 01:25 PM To: Steven Zakulec [spzakulec(a)mail.widener.edu] Subject: Re: [alsa-devel] [HDA-Intel/ALC268] no sound At Wed, 27 Jun 2007 17:34:15 -0400 (EDT), Steven Zakulec wrote: > > (Continued from this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-June/001459.html) > > I've included the result of alsactl -f somefile store. I thought it got sent to the list earlier, but apparently not. Below this I also added the alsa things from dmesg. And the contents of /proc/asound/card0/codec#* files, please. If you change the mixer status, these files also changed, too. Also, please inform what you tested more exactly. In which hardware configuration and which application with what parameters... Takashi > > state.Intel { > control.1 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 64' > iface MIXER > name 'Front Playback Volume' > value.0 57 > value.1 57 > } > control.2 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Front Playback Switch' > value.0 true > value.1 true > } > control.3 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 64' > iface MIXER > name 'Headphone Playback Volume' > value.0 63 > value.1 63 > } > control.4 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Headphone Playback Switch' > value.0 true > value.1 true > } > control.5 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 31' > iface MIXER > name 'Capture Volume' > value.0 1 > value.1 1 > } > control.6 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > value.0 false > value.1 false > } > control.7 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 31' > iface MIXER > name 'Capture Volume' > index 1 > value.0 1 > value.1 1 > } > control.8 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > index 1 > value.0 false > value.1 false > } > control.9 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > value Mic > } > control.10 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > index 1 > value Mic > } > control.11 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Off-hook Switch' > value false > } > control.12 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Caller ID Switch' > value false > } > control.13 { > comment.access 'read write user' > comment.type INTEGER > comment.count 2 > comment.range '0 - 255' > comment.tlv '0000000100000008ffffec1400000014' > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > } > } > hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 > [ 19.236000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 > [ 19.244000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) > > > > comment.count 2 > : > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] HDA Intel/ALC268
by Peter Poulsen 02 Jul '07

02 Jul '07

Is there anybody that has had any luck with this, or I'm a fighting a loosing battle? /peter

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[alsa-devel] [HDA-Intel/ALC268] no sound
by Steven Zakulec 02 Jul '07

02 Jul '07

(Continued from this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-June/001459.html) I've included the result of alsactl -f somefile store. I thought it got sent to the list earlier, but apparently not. Below this I also added the alsa things from dmesg. state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Front Playback Volume' value.0 57 value.1 57 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Headphone Playback Volume' value.0 63 value.1 63 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Capture Volume' value.0 1 value.1 1 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 false value.1 false } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Capture Volume' index 1 value.0 1 value.1 1 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.9 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' iface MIXER name 'Input Source' value Mic } control.10 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' iface MIXER name 'Input Source' index 1 value Mic } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Off-hook Switch' value false } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Caller ID Switch' value false } control.13 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } } hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 19.236000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 19.244000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) comment.count 2 :

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