FWIW, alsamixer is the only thing that needs curses, when I did an embedded linux + ALSA project I just passed --disable-alsamixer to work around it, and used amixer.

Lee

On 6/20/07, Paul Kavan pkavan@gmail.com wrote:

...

--with-alsa-inc-prefix=$HOME/Linux_Builds/buildroot/build_arm/root/usr/include

    --with-alsa-prefix=$HOME/Linux_Builds/buildroot/build_arm/root/usr/lib

Takashi

Takashi:

Thank you. That got me a little further. Now, I am hitting this error:

... checking for libasound headers version >= 1.0.12... found. checking for snd_ctl_open in -lasound... yes checking how to run the C preprocessor... arm-linux-gcc -E checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for initscr in -lncurses... no checking for initscr in -lcurses... no configure: error: this packages requires a curses library

...which I could swear I have....will look into this.

Paul

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Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan@gmail.com

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Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

At Wed, 20 Jun 2007 11:18:13 -0500, Paul Kavan wrote:

On 6/20/07, Lee Revell rlrevell@joe-job.com wrote:

FWIW, alsamixer is the only thing that needs curses, when I did an
embedded linux + ALSA project I just passed --disable-alsamixer to
work around it, and used amixer.

Lee

Lee:

Thanks. I do not think I need a mixer at all...doing mono voice on a dumb single channel codec.

BTW, you can try salsa-lib for such a purpose:

ftp://ftp.suse.com/pub/people/tiwai/salsa-lib/

I've not tested with ARM, so it'd be appreciated if someone lets me know whether it works.

Takashi

Paul Kavan wrote:

...

For alsa-libs: CC=arm-linux-gcc ./configure --target=arm-linux --host=i686-pc-linux-gnu --prefix=/usr

...

However, I have been unable to achieve any kind of playback using speaker-test or aplay. The speaker pops like it wants to play, but I get the following when trying speaker test:

/dev/snd $ speaker-test -D default -r 8000 -f 1020 -F S8

speaker-test 1.0.14

Playback device is default Stream parameters are 8000Hz, S8, 1 channels Using 16 octaves of pink noise Rate set to 8000Hz (requested 8000Hz) Buffer size range from 64 to 32768 Period size range from 32 to 8192 Using max buffer size 32768 Periods = 4 was set period_size = 0 was set buffer_size = 32768 Unable to set sw params for playback: Invalid argument Setting of swparams failed: Invalid argument /dev/snd $

Anyone know where I should look for the problem?

Hi Paul,

Try adding the option "--with-versioned=no" to the alsa-libs configure.

../fam

Paul Kavan wrote:

Frank:

Try adding the option "--with-versioned=no" to the alsa-libs configure. 

That did the trick in terms of the sw params problem. With the default alsa.conf file, I can now get the speaker to make some noise. However, doing:

speaker-test -r8000 -f1000 -FS8 -tsine -l5

gives me a slightly distorted sine wave that has a period of half of what I would expect.

Better than no sound :) At least something is getting through ALSA and ASoC to your codec.

You may need to adjust the RCMR, RFRM, TCMR, TFRM register settings in sound/soc/at91/at91-ssc.c to send the data in the mode that your codec expects. I have only tested I2S and DSP_A modes with S16_LE data. It's fairly easy to add new modes.

Frank, this is what I have for the small conf. I hope it is the right one:

pcm.!default { type hw card 0 } pcm.hw { type hw card 0 device 0 subdevice 0 } ctl.!default { type hw card 0 } ctl.hw { type hw card 0 }

ctl.w6811 { type hw card 0 } pcm.w6811 { type hw card 0 device 0 subdevice 0 }

ctl.!default = ctl.w6811 pcm.!default = pcm.w6811

ctl.hw { @args[ CARD ] @args.CARD { type string default "0" } type hw card $CARD }

pcm.hw { @args[ CARD DEV SUBDEV ] @args.CARD { type string default "0" } @args.DEV { type integer default 0 } @args.SUBDEV { type integer default 0 } type hw card $CARD device $DEV subdevice $SUBDEV }

However, when I use this as the conf, I get the following with aplay -l

/usr/share/alsa $ aplay -l **** List of PLAYBACK Hardware Devices **** ALSA lib conf.c:975:(parse_value) card is not a string ALSA lib conf.c:1588:(snd_config_load1) _toplevel_:45:1:Invalid argument ALSA lib conf.c:3066:(snd_config_update_r) /usr/share/alsa/alsa.conf may be old or corrupted: consit aplay: device_list:213: control open (0): Invalid argument

Not sure if I have the alsa.conf correct. So maybe it is a problem there. You mentioned a bare bones .conf, but I do not this is the only one I have in my email.

I don't see anything wrong with this alsa.conf, but I'm no expert. It is virtually the same as the one I am using, but I am using ALSA 1.0.13.

Here is the bare-bones config file that I was using before I added a second sound device:

pcm.!default { type hw card 0 } pcm.hw { type hw card 0 device 0 subdevice 0 } ctl.!default { type hw card 0 } ctl.hw { type hw card 0 }

../fam

You may need to adjust the RCMR, RFRM, TCMR, TFRM register settings in

...

sound/soc/at91/at91- ssc.c to send the data in the mode that your codec expects. I have only tested I2S and DSP_A modes with S16_LE data. It's fairly easy to add new modes.

This is certainly part of the problem. I probed the clocks and the frame sync lines and they are fine for the most part. However, I can tell from the output that I need to change the default on TD0 to '1' for starters. I also think that I need to change the format so that the atmel board reads on the falling edge after sync. I think it already writes on the rising edge, but will have to see. Also, the width of the sync is one period of the bitclock....and I need at least two, so I will need to change that.

Fortunately, I have done a lot of grunt work already in writing my basic ssc driver for that codec. Now I have to dig into at91-ssc.c to see how to modify that.

...

pcm.!default { type hw card 0 } pcm.hw { type hw card 0 device 0 subdevice 0 } ctl.!default { type hw card 0 } ctl.hw { type hw card 0 }

The bare bones conf is not working for me and I am getting:

/usr/share/alsa $ aplay -l **** List of PLAYBACK Hardware Devices **** ALSA lib conf.c:3949:(snd_config_expand) Unknown parameters 0 ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL hw:0 aplay: device_list:213: control open (0): Invalid argument

For whatever reason, the default conf file is letting me use speaker-test, so I can work on debugging the format for now. I am using alsa-lib-1.0.14aand alsa-utils-1.0.14, so I am not sure if that makes a difference. If anyone on the list has a bare bones conf for a simple pcm codec, I would appreciate some tips on why this one is not working.

Thanks Frank. I am gonna dig into at91-ssc.c now.

Paul