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

[alsa-devel] [PATCH] ASoC: MPC5200: Changed how appl_ptr is initialized
by John Bonesio 29 Jul '09

29 Jul '09

Hello, I've been working in the ALSA SOC area for the MPC5200. I've noticed that in the routine psc_dma_trigger() in sound/soc/fsl/mpc5200_dma.c there are times when the computed value for s->appl_ptr ends up wrapping negative. This occurs when s->runtime->control->appl_ptr is less than runtime->period_size * runtime->periods It seemed like this would occur at then of the data stream or when the data stream is small (smaller than runtime->period_size * runtime->periods). As far as I can tell, s->appl_ptr is supposed to be the previous value of s->runtime->control->appl_ptr, so the driver can send out the right number of bytes (from previous appl_ptr to current appl_ptr) to avoid pops and clicks at the end of the audio. So it appears the assumption in this code is wrong, that the previous s->runtime->control->appl_ptr is always rumtime->period_size * runtime->periods bytes prior to the current s->runtime->control->appl_ptr. I took out this code and instead added code to intialize s->appl_ptr to 0 in psc_dma_open(). Does this fix seem right? - John The computation for appl_ptr in the trigger() routine made a bad assumption when the stream is at the end of the file and there is not (periods * period_size) bytes remaining in the data. This causes playback issues when the total stream size is less than this amount, as the s->appl_ptr wrapps negative and the data is never played. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index b2eba27..cfe0ea4 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -181,8 +181,6 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) * end of stream and not over running it. */ s->runtime = runtime; - s->appl_ptr = s->runtime->control->appl_ptr - - (runtime->period_size * runtime->periods); /* Fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. @@ -287,6 +285,7 @@ static int psc_dma_open(struct snd_pcm_substream *substream) } s->stream = substream; + s->appl_ptr = 0; return 0; }

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[alsa-devel] [PATCH] aedsp16: Buffer overflow
by Roel Kluin 29 Jul '09

29 Jul '09

DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ on line 498: static char DSPVersion[CARDVERLEN + 1] __initdata = {0, }; diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index 3ee9900..35b5912 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -325,8 +325,9 @@ /* * Size of character arrays that store name and version of sound card */ -#define CARDNAMELEN 15 /* Size of the card's name in chars */ -#define CARDVERLEN 2 /* Size of the card's version in chars */ +#define CARDNAMELEN 15 /* Size of the card's name in chars */ +#define CARDVERLEN 10 /* Size of the card's version in chars */ +#define CARDVERDIGITS 2 /* Number of digits in the version */ #if defined(CONFIG_SC6600) /* @@ -410,7 +411,7 @@ static int soft_cfg __initdata = 0; /* bitmapped config */ static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */ -static int ver[CARDVERLEN] __initdata = {0, 0}; /* DSP Ver: +static int ver[CARDVERDIGITS] __initdata = {0, 0}; /* DSP Ver: hi->ver[0] lo->ver[1] */ #if defined(CONFIG_SC6600) @@ -957,7 +958,7 @@ static int __init aedsp16_dsp_version(int port) * string is finished. */ ver[len++] = ret; - } while (len < CARDVERLEN); + } while (len < CARDVERDIGITS); sprintf(DSPVersion, "%d.%d", ver[0], ver[1]); DBG(("success.\n"));

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[alsa-devel] [PATCH] mpu401.c: Buffer overflow
by Roel Kluin 29 Jul '09

29 Jul '09

mpu_synth_info[m].name is a char[30], and the minimum length of the data written by sprintf is 31 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 1b2316f..734b8f9 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -1074,7 +1074,7 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner) sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name); else sprintf(mpu_synth_info[m].name, - "MPU-401 %d.%d%c Midi interface #%d", + "MPU-401 %d.%d%c MIDI #%d", (int) (devc->version & 0xf0) >> 4, devc->version & 0x0f, revision_char,

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[alsa-devel] my udev rules are breaking my dmixer setup why?
by Jelle de Jong 29 Jul '09

29 Jul '09

Hello everybody, I am trying to use udev to get fixed device names depending on the location of the device on the usb bus. The udev rules seems to be working fine, however i cant get dmix to work with the udev rules using only hw dies work. If i remove the udev rules the dmixer works fine. Can somebody look at the attachment, and maybe do some testing, i would highly appreciate it. I am also on chat.freenode.org and #alsa as tuxcrafter Thanks in advance, Jelle ls -hal /dev/snd/ total 0 drwxr-xr-x 2 root root 320 2008-11-09 15:44 . drwxr-xr-x 12 root root 4.8K 2008-11-09 15:50 .. crw-rw---- 1 root user0 116, 0 2008-11-09 15:44 controlC0 crw-rw---- 1 root user1 116, 32 2008-11-09 15:44 controlC1 crw-rw---- 1 root user2 116, 64 2008-11-09 15:44 controlC2 crw-rw---- 1 root user3 116, 96 2008-11-09 15:44 controlC3 crw-rw---- 1 root user0 116, 24 2008-11-09 15:44 pcmC0D0c crw-rw---- 1 root user0 116, 16 2008-11-09 15:49 pcmC0D0p crw-rw---- 1 root user1 116, 56 2008-11-09 15:44 pcmC1D0c crw-rw---- 1 root user1 116, 48 2008-11-09 15:44 pcmC1D0p crw-rw---- 1 root user2 116, 88 2008-11-09 15:44 pcmC2D0c crw-rw---- 1 root user2 116, 80 2008-11-09 15:44 pcmC2D0p crw-rw---- 1 root user3 116, 120 2008-11-09 15:44 pcmC3D0c crw-rw---- 1 root user3 116, 112 2008-11-09 15:44 pcmC3D0p crw-rw---- 1 root audio 116, 1 2008-11-09 15:44 seq crw-rw---- 1 root audio 116, 33 2008-11-09 15:44 timer ------------------------------------------------------------------------ udevinfo info --attribute-walk --name=/dev/snd/controlC0 ------------------------------------------------------------------------ echo 'SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="controlC[0-9]*", NAME="snd/controlC4" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="controlC[0-9]", NAME="snd/controlC5" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="controlC[0-9]", NAME="snd/controlC6" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="controlC[0-9]", NAME="snd/controlC7" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC4D0p" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC5D0p" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC6D0p" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC7D0p"' | tee /etc/udev/rules.d/10-persistent-sound.rules chmod 600 /etc/udev/rules.d/10-persistent-sound.rules ------------------------------------------------------------------------ # without dmixer # this works .... echo 'pcm.softvol { type softvol slave.pcm hw:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.softvol { type softvol slave.pcm dmix:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.!default { type plug slave.pcm dmixer } pcm.dmixer { type dmix ipc_key 2048 slave.pcm hw:1 } ctl.dmixer { type hw card 1 } pcm.dsp { type plug slave.pcm dmixer } ctl.mixer { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------

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[alsa-devel] [RFC] [PATCH v2] ASoC: CX20442: push down machine independent line discipline bits
by Janusz Krzysztofik 29 Jul '09

29 Jul '09

This corrected patch adds machine independent line discipline code, prevoiusly exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442 codec driver. The code can be used as a standalone line discipline, or as a set of codec specific functions called from machine's line discipline callbacks. Anyway, the line discipline itself must be registered by a machine driver. Applies on top of the followup to my initial driver version: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html Suggested by ASoC manintainer Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Corrections: sound/soc/codecs/cx20442.c line 300: -static struct tty_ldisc_ops v253_ops = { +struct tty_ldisc_ops v253_ops = { line 420: - dev_err(&codec->dev, "Failed to register codec: %d\n", ret); + dev_err(codec->dev, "Failed to register codec: %d\n", ret); line 425: - dev_err(&codec->dev, "Failed to register DAI: %d\n", ret); + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); Corrected patch: --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.h.orig 2009-07-22 04:56:45.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.h 2009-07-28 21:23:17.000000000 +0200 @@ -15,5 +15,6 @@ extern struct snd_soc_dai cx20442_dai; extern struct snd_soc_codec_device cx20442_codec_dev; +extern struct tty_ldisc_ops v253_ops; #endif --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.c.orig 2009-07-28 20:26:18.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.c 2009-07-29 02:16:05.000000000 +0200 @@ -13,6 +13,8 @@ * option) any later version. */ +#include <linux/tty.h> + #include <sound/core.h> #include <sound/initval.h> #include <sound/soc-dapm.h> @@ -172,8 +174,7 @@ static int cx20442_write(struct snd_soc_ return -EINVAL; /* hw_write and control_data pointers required for talking to the modem - * are expected to be set by the machine driver's line discipline - * initialization code */ + * are expected to be set by the line discipline initialization code */ if (!codec->hw_write || !codec->control_data) return -EIO; @@ -208,6 +209,111 @@ static int cx20442_write(struct snd_soc_ return 0; } + +/* Moved up here as line discipline referres it during initialization */ +static struct snd_soc_codec *cx20442_codec; + + +/* + * Line discpline related code + * + * Any of the callback functions below can be used in two ways: + * 1) registerd by a machine driver as one of line discipline operations, + * 2) called from a machine's provided line discipline callback function + * in case when extra machine specific code must be run as well. + */ + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *v253_init = "ate0m0q0+fclass=8\r"; + +/* Line discipline .open() */ +static int v253_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = cx20442_codec; + int ret, len = strlen(v253_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass the codec structure address for use by other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, v253_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Line discipline .close() */ +static void v253_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + tty->disc_data = NULL; + + if (!codec) + return; + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; +} + +/* Line discipline .hangup() */ +static int v253_hangup(struct tty_struct *tty) +{ + v253_close(tty); + return 0; +} + +/* Line discipline .receive_buf() */ +static void v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + + if (!codec) + return; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Set up codec driver access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + } +} + +/* Line discipline .write_wakeup() */ +static void v253_wakeup(struct tty_struct *tty) +{ +} + +struct tty_ldisc_ops v253_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx20442", + .owner = THIS_MODULE, + .open = v253_open, + .close = v253_close, + .hangup = v253_hangup, + .receive_buf = v253_receive, + .write_wakeup = v253_wakeup, +}; +EXPORT_SYMBOL_GPL(v253_ops); + + +/* + * Codec DAI + */ + struct snd_soc_dai cx20442_dai = { .name = "CX20442", .playback = { @@ -227,8 +333,6 @@ struct snd_soc_dai cx20442_dai = { }; EXPORT_SYMBOL_GPL(cx20442_dai); -static struct snd_soc_codec *cx20442_codec; - static int cx20442_codec_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -313,13 +417,13 @@ static int cx20442_register(struct cx204 ret = snd_soc_register_codec(codec); if (ret != 0) { - dev_err(&codec->dev, "Failed to register codec: %d\n", ret); + dev_err(codec->dev, "Failed to register codec: %d\n", ret); goto err; } ret = snd_soc_register_dai(&cx20442_dai); if (ret != 0) { - dev_err(&codec->dev, "Failed to register DAI: %d\n", ret); + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); goto err_codec; }

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[alsa-devel] Strange behaviour with varying frame sizes on read
by Metathronius Galabant 29 Jul '09

29 Jul '09

Hi all, please consider the following: Period size is 80 frames, buffer size is 320 frames. For all 3 tests I have 240 frames ready to read. snd_pcm_avail_update() confirms this. Outcome of my tests: 1) I want to read 80 frames, and get 80 frames. - OK, this is really basic. 2) I want to read 81 frames, and get 80 frames. - FAIL. 3) I want to read 80 frames, and get 80 frames. Then I read 1 frame, and get 1 frame. - HUH? I've both tested snd_pcm_readi() and snd_pcm_mmap_readi() - both have the same semantic. Is there any explanation for this kind of illogical behaviour (I have version 1.0.14, which might be a bit old, but still)? I should be able to read rather arbitrary amounts from a software buffer, don't I? Cheers, M.

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[alsa-devel] Using Alsa for AD1981b on 9263
by Shilpa Kedar Walvekar 29 Jul '09

29 Jul '09

Hi, Addition to my earlier post: We are using AT91SAM9263 evaluation kit with on-board AD1981b codec. We are using Linux kernel version 2.6.27 with patches available on http://linux4sam.com. We found from the forum that we need to use ALSA for playing and recording audio using this codec. So we downloaded Alsa-lib-1.0.20 Alsa-oss-1.0.20 Alsa-utils-1.0.20 We installed above using: ./configure && make && make install We then downloaded alsa-driver-1.0.20 and installed it using below: ./configure --with-cards=atmel-ac97c --with-kernel=../../linux-2.6.27 --with-build=../../linux-2.6.27 && make ARCH=arm && make install-modules Driver was compiled and installed successfully. We found following: # cat /proc/asound/cards 0 [AC97 ]: ac97c - Atmel AC97 Atmel AC97 Controller at 0xfffa0000, irq 18 # lsmod Module Size Used by Not tainted snd_at91_ac97 8420 0 snd_ac97_codec 99904 1 snd_at91_ac97 snd_soc_atmel_pcm 4484 0 snd_soc_core 30852 1 snd_soc_atmel_pcm snd_pcm 60772 4 snd_at91_ac97,snd_ac97_codec,snd_soc_atmel_pcm,s nd_soc_core snd_page_alloc 6056 1 snd_pcm snd_timer 19588 1 snd_pcm snd 47700 5 snd_at91_ac97,snd_ac97_codec,snd_soc_core,snd_pc m,snd_timer soundcore 6052 1 snd ac97_bus 1568 1 snd_ac97_codec # cat /home/.asoundrc pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } We have following devices in /dev/snd # cat /dev/snd/ /dev/snd/controlC0 /dev/snd/controlC3 /dev/snd/hwC0D2 /dev/snd/timer /dev/snd/controlC1 /dev/snd/hwC0D0 /dev/snd/hwC0D3 /dev/snd/controlC2 /dev/snd/hwC0D1 /dev/snd/seq # aplay -l **** List of PLAYBACK Hardware Devices **** ALSA lib conf.c:2700:(snd_config_hooks_call) Cannot open shared library (null) ALSA lib conf.c:3079:(snd_config_update_r) hooks failed, removing configuration aplay: device_list:232: control open (0): No such file or directory We are not able to get if module is getting loaded, then why aplay doesn't list the device? Please help us to get the audio codec working. Regards, --Shilpa.

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 29 Jul '09

29 Jul '09

The following changes since commit 4ce2f2fe61002ab35c9333d52c95b5ff9e949f50: Marek Vasut (1): ASoC: Switch palm27x-asoc to jack detection api are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (2): ASoC: blackfin I2S(TDM mode) CPU DAI driver ASoC: board driver to connect bf5xx with ad1938 Janusz Krzysztofik (1): ASoC: CX20442: fix issues pointed out by subsystem maintainer sound/soc/blackfin/Kconfig | 23 +++- sound/soc/blackfin/Makefile | 6 + sound/soc/blackfin/bf5xx-ad1938.c | 142 +++++++++++++++ sound/soc/blackfin/bf5xx-tdm-pcm.c | 330 ++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm-pcm.h | 21 +++ sound/soc/blackfin/bf5xx-tdm.c | 343 ++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm.h | 14 ++ sound/soc/codecs/cx20442.c | 17 +- 8 files changed, 888 insertions(+), 8 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1938.c create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-tdm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm.h

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Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 29 Jul '09

29 Jul '09

[Please don't do top-posting] At Tue, 28 Jul 2009 14:50:10 -0500, Ted T. Logan wrote: > > Just got the latest as of today, > > click > click > click > click > click > > :( > mplayer yesterday.mp3 For testing this kind of problem, use aplay and try different --period-size and --buffer-size options. mplayer isn't suitable for primary testing. If the click period doesn't change regardless of the period size option value, it implies that the noise doesn't come from the buffer set up but rather from other routes. If the noise period changes with different period size options, the click noise is a buffer setup issue. In addition, it'd be better to use aplay with -Dplughw to avoid the automatic routing to PA. Also, your description was unclear. It it only click without other tone, or repeated sound, or the click noise appear on the normal playback, or what? The repeated sound bug with emu20k2 (it didn't appear with emu20k1) was fixed as I mentioned. But, if it's click noise on the normal playback, then I don't know. Better to ask Creative guys... thanks, Takashi

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[alsa-devel] Assert when attempting to use pulseaudio git master with the a52 plugin.
by Luke Yelavich 29 Jul '09

29 Jul '09

Hi all I would like to bring to your attention a bug in the a52 plugin when working with pulseaudio. The full details are found in the ALSA bug tracker: https://bugtrack.alsa-project.org/alsa-bug/bug_view_advanced_page.php?bug_i… I am happy for follow ups to be in the bug or on this list. Thanks Luke

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