August 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: Fix WM8580 CLKSEL mask selection
by Mark Brown 16 Aug '10

16 Aug '10

The RX and TX directions were inverted. Reported-by: Seungwhan Youn <claude.youn(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index af4517e..42a6699 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -703,12 +703,12 @@ static int wm8580_set_sysclk(struct snd_soc_dai *dai, int clk_id, int sel, sel_mask, sel_shift; switch (dai->driver->id) { - case WM8580_DAI_PAIFTX: + case WM8580_DAI_PAIFRX: sel_mask = 0x3; sel_shift = 0; break; - case WM8580_DAI_PAIFRX: + case WM8580_DAI_PAIFTX: sel_mask = 0xc; sel_shift = 2; break; -- 1.7.1

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Re: [alsa-devel] ALC892 support for multiple spdif outputs
by Sander Knopper 16 Aug '10

16 Aug '10

> At Sun, 15 Aug 2010 14:57:51 +0200, > > Sander Knopper wrote: > > Dear developers, > > > > I've opened a bug about this problem which can be viewed here: > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5101 > > > > Raymond mentions in his last comment that I should ask you to, if I > > understand correctly, do the same trick as in the commit he refers to > > also for the ALC892 chip. Which obviously is the one I have... > > > > You can find all the relevant information in the bug entry. > > Could you post the exact problem and attach alsa-info.sh output to ML? > It'll help much for communication on ML. > > > Takashi Ok, I can do that... The problem is that I have an Asus P7H55 M Pro motherboard which uses the ALC892 codec. I'm using this system as a HTPC and thus I'm trying to get sound from the HTPC to my receiver. The motherboard only has an optical output which shouldn't be an problem, be it that my receiver only has a coaxial input. So I bought an spdif bracket made by Asus which has both a coaxial and optical output and connected it to the internal spdif pins on the motherboard. So far so good, but the real problem is that I can't get the bracket to work. When I unmute all the spdif related entries with alsamixer only the optical output on the motherboard itself "gives light" and the one on the bracket doesn't. I've searched the internet a bit and I tried switching the audio output in the bios from HDMI to SPDIF and vice versa, but that doesn't make any difference. alsa-info is attached. Regards, Sander

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[alsa-devel] ALC892 support for multiple spdif outputs
by Sander Knopper 16 Aug '10

16 Aug '10

Dear developers, I've opened a bug about this problem which can be viewed here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5101 Raymond mentions in his last comment that I should ask you to, if I understand correctly, do the same trick as in the commit he refers to also for the ALC892 chip. Which obviously is the one I have... You can find all the relevant information in the bug entry. Please CC me as I'm not subscribed to the list. Best regards, Sander Knopper

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[alsa-devel] possible fix for buffer underruns on msm7k alsa driver
by Denis 'GNUtoo' Carikli 15 Aug '10

15 Aug '10

hi, I'm still rebasing the codeaurora alsa driver for msm soc like the htcdream on the android msm 2.6.32 kernel. I looked the commits for the other socs like msm7kv2 and qsd8k(snapdragon) and I found that commit: commit ac97ef246a754b9a2b4236ca08c6ade389dc206f Author: Pradeep Jilagam <pjilagam(a)qualcomm.com> Date: Wed Apr 7 17:38:20 2010 +0530 alsa: 8k: Limit period size for sample rates greater than 8KHz The DSP interrupts are non-periodic. ALSA expects periodic acknowledgement for the buffers sent. Hence, the alsa driver uses timer as alternate interrupt source. The min resolution is 10ms which results in distorted audio for some sample rates. Hence restricting the minimum period size for sample rates greater than 8KHz to 512 frames. Change-Id: Ic8c4180fbee7242fa3139230ddcf6a9f98e4c7ab CRs-fixed: 231784 Signed-off-by: Pradeep Jilagam <pjilagam(a)qualcomm.com> I tried a big buck bunny at 44100 Hz: lots(too much) of buffer underrun => too bad to be usable. Then I tried it with 8000 Hz and found nearly no buffer underruns => usable. So I've great expectations for that fix. here's the source code involved in the fix: static int hw_rule_periodsize_by_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval *ps = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval ch; if (!ps || !r){ pr_debug("REMOVE_THAT: exiting from !ps||!r\n"); return 0; }else{ pr_debug("REMOVE_THAT: not exiting\n"); } snd_interval_any(&ch); if (r->min > 8000) { ch.min = 512; pr_debug("Minimum period size is adjusted to 512\n"); return snd_interval_refine(ps, &ch); } return 0; } static int msm_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct msm_audio *prtd; int ret = 0; prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL); if (prtd == NULL) { ret = -ENOMEM; return ret; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { msm_vol_ctl.update = 1; /* Update Volume, with Cached value */ runtime->hw = msm_pcm_playback_hardware; prtd->dir = SNDRV_PCM_STREAM_PLAYBACK; prtd->playback_substream = substream; prtd->eos_ack = 0; } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw = msm_pcm_capture_hardware; prtd->dir = SNDRV_PCM_STREAM_CAPTURE; prtd->capture_substream = substream; } ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_sample_rates); if (ret < 0) goto out; /* Ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) goto out; ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, hw_rule_periodsize_by_rate, substream, SNDRV_PCM_HW_PARAM_RATE, -1); if (ret < 0) goto out; prtd->ops = &snd_msm_audio_ops; prtd->out[0].used = BUF_INVALID_LEN; prtd->out_head = 1; /* point to second buffer on startup */ runtime->private_data = prtd; ret = alsa_adsp_configure(prtd); if (ret) goto out; copy_count = 0; return 0; out: kfree(prtd); return ret; } Unfortunately it doesn't seem to print the REMOVE_THAT debug messages in dmesg,after playing something with mplayer, like a 44100 Hz song in mp3 format with mplayer. I wonder why the hw_rule_periodsize_by_rate function is not executed. Background: The htcdream is the first android smartphone and has by default a dsp interface that is not alsa nor oss nor something that was created before. Codeaurora(qualcomm) had some alsa driver,I rebased it but it was buggy,so I tried to debug it and succedded at making mplayer run,but it had many buffer underruns... Denis.

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[alsa-devel] Extern dsp device
by Christoph Kuhr 15 Aug '10

15 Aug '10

Hi everybody, In my bachlor project im planing to realise an outboard fpga dsp device connected by gbit ethernet and driven by alsa. The idea: computer system with linux rt kernel, extra gbit ethernet if and for example rme hdsp 9652 audio if. Outboard fpga with gbit if and DSP routines (PureD ata Engine?) Now the fpga board is connected via gbit eth, the kernel gets it and inserts the gbit ethernet card into the audio signal path. Managed with and alsa driver, cooperating with the snd-rme9652. With a gui sending OSC to the fpga board, controlling the dsp. Goal: zero-latency routing and dsp Do you think this strategy might work? Ck

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[alsa-devel] [PATCH] ASoC: multi-component: Fix reference to moved header file, which was unused anyway.
by Ian Lartey 15 Aug '10

15 Aug '10

Removed #include of pxa2xx-pcm.h Signed-off-by: Ian Lartey <ian(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/pxa2xx-i2s.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3b473b2..d1b2ca6 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -27,7 +27,6 @@ #include <mach/dma.h> #include <mach/audio.h> -#include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" /* -- 1.7.0.4

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[alsa-devel] [PATCH 2/5] ASoC: fix pxa2xx-pcm.h path
by Haojian Zhuang 15 Aug '10

15 Aug '10

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[alsa-devel] [patch] sound/usb/format: silence uninitialized variable warnings
by Dan Carpenter 15 Aug '10

15 Aug '10

Gcc complains that ret might be used uninitialized: sound/usb/format.c: In function ‘snd_usb_parse_audio_format’: sound/usb/format.c:354: warning: ‘ret’ may be used uninitialized in this function sound/usb/format.c:354: note: ‘ret’ was declared here sound/usb/format.c:414: warning: ‘ret’ may be used uninitialized in this function sound/usb/format.c:414: note: ‘ret’ was declared here I suppose it could be if there is ever a UAC_VERSION_3 released. Anyway this patch is worthwhile if only to silence the gcc warning. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/usb/format.c b/sound/usb/format.c index 4387f54..1400174 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -351,7 +351,8 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, { struct usb_interface_descriptor *altsd = get_iface_desc(iface); int protocol = altsd->bInterfaceProtocol; - int pcm_format, ret; + int pcm_format; + int ret = -EINVAL; if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx @@ -411,9 +412,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, int format, void *_fmt, struct usb_host_interface *iface) { - int brate, framesize, ret; + int brate, framesize; struct usb_interface_descriptor *altsd = get_iface_desc(iface); int protocol = altsd->bInterfaceProtocol; + int ret = -EINVAL; switch (format) { case UAC_FORMAT_TYPE_II_AC3:

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[alsa-devel] ASoC: Make ams-delta/cx20442 pair back working for multi-component
by Janusz Krzysztofik 14 Aug '10

14 Aug '10

1. After a static codec pointer removal from the sound/soc/codecs/cx20442.c code, both codec and card specific tty line discipline functions lost their access to the codec structure data. Arrange for passing the pointer back to the codec tty ldisc bits from the card initialization code. 2. Update incosistent codec device naming in the card code. 3. Since a new element, control_data, has been added to the cx20442_priv structure, it looks like the old snd_soc_codec->control_data may be scheduled for removal. To avoid problems in the near future, use the cx20442_priv->control_data instead of the snd_soc_codec->control_data throughout the cx20442 code. In the ams-delta card code, check for the snd_soc_codec->hw_write not NULL instead of the snd_soc_codec->control_data. Created and tested on Amstrad Delta against git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git, commit a212dd5b2e1da045cbf423c70c87f48265531de4. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- git/sound/soc/codecs/cx20442.c.orig 2010-08-10 17:43:17.000000000 +0200 +++ git/sound/soc/codecs/cx20442.c 2010-08-11 01:08:23.000000000 +0200 @@ -165,6 +165,7 @@ static int cx20442_pm_to_v253_vsp(u8 val static int cx20442_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); u8 *reg_cache = codec->reg_cache; int vls, vsp, old, len; char buf[18]; @@ -174,7 +175,7 @@ static int cx20442_write(struct snd_soc_ /* hw_write and control_data pointers required for talking to the modem * are expected to be set by the line discipline initialization code */ - if (!codec->hw_write || !codec->control_data) + if (!codec->hw_write || !cx20442->control_data) return -EIO; old = reg_cache[reg]; @@ -203,17 +204,13 @@ static int cx20442_write(struct snd_soc_ return -ENOMEM; dev_dbg(codec->dev, "%s: %s\n", __func__, buf); - if (codec->hw_write(codec->control_data, buf, len) != len) + if (codec->hw_write(cx20442->control_data, buf, len) != len) return -EIO; return 0; } -/* Moved up here as line discipline referres it during initialization */ -static struct snd_soc_codec *cx20442_codec; - - /* * Line discpline related code * @@ -229,15 +226,15 @@ static const char *v253_init = "ate0m0q0 /* 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; + /* Won't work if no codec pointer has been passed by a card driver */ + if (!tty->disc_data) + return -ENODEV; if (tty->ops->write(tty, v253_init, len) != len) { ret = -EIO; @@ -254,15 +251,18 @@ err: static void v253_close(struct tty_struct *tty) { struct snd_soc_codec *codec = tty->disc_data; + struct cx20442_priv *cx20442; tty->disc_data = NULL; if (!codec) return; + cx20442 = snd_soc_codec_get_drvdata(codec); + /* Prevent the codec driver from further accessing the modem */ codec->hw_write = NULL; - codec->control_data = NULL; + cx20442->control_data = NULL; codec->pop_time = 0; } @@ -278,15 +278,18 @@ static void v253_receive(struct tty_stru const unsigned char *cp, char *fp, int count) { struct snd_soc_codec *codec = tty->disc_data; + struct cx20442_priv *cx20442; if (!codec) return; - if (!codec->control_data) { + cx20442 = snd_soc_codec_get_drvdata(codec); + + if (!cx20442->control_data) { /* First modem response, complete setup procedure */ /* Set up codec driver access to modem controls */ - codec->control_data = tty; + cx20442->control_data = tty; codec->hw_write = (hw_write_t)tty->ops->write; codec->pop_time = 1; } @@ -343,7 +346,7 @@ static int cx20442_codec_probe(struct sn cx20442_add_widgets(codec); - codec->control_data = NULL; + cx20442->control_data = NULL; codec->hw_write = NULL; codec->pop_time = 0; @@ -355,8 +358,8 @@ static int cx20442_codec_remove(struct s { struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); - if (codec->control_data) { - struct tty_struct *tty = codec->control_data; + if (cx20442->control_data) { + struct tty_struct *tty = cx20442->control_data; tty_hangup(tty); } --- git/sound/soc/omap/ams-delta.c.orig 2010-08-10 17:43:17.000000000 +0200 +++ git/sound/soc/omap/ams-delta.c 2010-08-11 00:58:22.000000000 +0200 @@ -99,7 +99,7 @@ static int ams_delta_set_audio_mode(stru int pin, changed = 0; /* Refuse any mode changes if we are not able to control the codec. */ - if (!codec->control_data) + if (!codec->hw_write) return -EUNATCH; if (ucontrol->value.enumerated.item[0] >= control->max) @@ -268,10 +268,32 @@ static void cx81801_timeout(unsigned lon ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0); } +/* + * Used for passing a codec structure pointer + * from the board initialization code to the tty line discipline. + */ +static struct snd_soc_codec *cx20442_codec; + /* Line discipline .open() */ static int cx81801_open(struct tty_struct *tty) { - return v253_ops.open(tty); + int ret; + + if (!cx20442_codec) + return -ENODEV; + + /* + * Pass the codec structure pointer for use by other ldisc callbacks, + * both the card and the codec specific parts. + */ + tty->disc_data = cx20442_codec; + + ret = v253_ops.open(tty); + + if (ret < 0) + tty->disc_data = NULL; + + return ret; } /* Line discipline .close() */ @@ -281,11 +303,14 @@ static void cx81801_close(struct tty_str del_timer_sync(&cx81801_timer); - v253_ops.close(tty); - /* Prevent the hook switch from further changing the DAPM pins */ INIT_LIST_HEAD(&ams_delta_hook_switch.pins); + if (!codec) + return; + + v253_ops.close(tty); + /* Revert back to default audio input/output constellation */ snd_soc_dapm_disable_pin(codec, "Mouthpiece"); snd_soc_dapm_enable_pin(codec, "Earpiece"); @@ -310,7 +335,10 @@ static void cx81801_receive(struct tty_s const unsigned char *c; int apply, ret; - if (!codec->control_data) { + if (!codec) + return; + + if (!codec->hw_write) { /* First modem response, complete setup procedure */ /* Initialize timer used for config pulse generation */ @@ -469,6 +497,9 @@ static int ams_delta_cx20442_init(struct int ret; /* Codec is ready, now add/activate board specific controls */ + /* Store a pointer to the codec structure for tty ldisc use */ + cx20442_codec = codec; + /* Set up digital mute if not provided by the codec */ if (!codec_dai->driver->ops) { codec_dai->driver->ops = &ams_delta_dai_ops; @@ -595,8 +626,8 @@ static int __init ams_delta_module_init( * Codec platform device could be registered from elsewhere (board?), * but I do it here as it makes sense only if used with the card. */ - cx20442_platform_device = platform_device_register_simple("cx20442", - -1, NULL, 0); + cx20442_platform_device = + platform_device_register_simple("cx20442-codec", -1, NULL, 0); return 0; err: platform_device_put(ams_delta_audio_platform_device);

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[alsa-devel] [PATCH 1/2] ASoC: Set up debugfs only once per CODEC
by Mark Brown 14 Aug '10

14 Aug '10

Since the debugfs directory is current per CODEC we should only init it when the CODEC is initialised, otherwise we end up with errors being generated when an attempt is made to add duplicate debugfs entries. Since most of this stuff is actually for the card we should refactor but this can come later. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a004876..ac0fa22 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1323,6 +1323,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) return ret; } } + + soc_init_codec_debugfs(codec); + /* mark codec as probed and add to card codec list */ codec->probed = 1; list_add(&codec->card_list, &card->codec_dev_list); @@ -1400,8 +1403,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) if (ret < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); - soc_init_codec_debugfs(codec); - /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { -- 1.7.1

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