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

[alsa-devel] [PATCH 1/1] mfd: add codec resource into 88pm860x driver
by Haojian Zhuang 10 Sep '10

10 Sep '10

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[alsa-devel] [PATCH 0/4 v2] Add FSI - HDMI support V2
by Kuninori Morimoto 07 Sep '10

07 Sep '10

Dear Mark, Paul These are FSI - HDMI prototype sound support V2 Kuninori Morimoto (4): fbdev: sh-mobile: Add HDMI sound type selection ASoC: Add sh_mobile_hdmi sound support ASoC: fsi-codec: Add FSI - HDMI support ARM: mach-shmobile: ap4evb: Add HDMI sound support >> Paul 1st patch is depend on [PATCH] fbdev: sh-mobile_hdmi: remove un-necessity settings which I sent to LinuxSH ML in 2010/08/27 >> Mark 2nd, 3rd ASoC patches are based on Mark's "for-2.6.37"

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[alsa-devel] Problem with USB Class 2 Audio Driver
by Demian Martin 06 Sep '10

06 Sep '10

Daniel: I'm testing an early sample of the Wavelength Wavelink, a USB to S/PDIF adapter that supports 44.1-192 KHz sample rates and uses async usb to talk to ALSA. It works and plays all of the sample rates correctly with the git version of Alsa from 7/30/2010 whenthe files are sourced from the network. However if I try to play from a usb source to the usb dac it doesn't work and the whole system gets unstable. The platforms I have tested it on seem to have a single USB host interface but with USB 2 there should be enough bandwidth to pass the data from a USB stick to the cpu and back. If I use an older 96 only usb dac on the same system it works (unless the down conversion isn't a direct divide, which overloads the CPU, but that is a different issue). The problem seems to be sample rate independent and hits the moment I try to access the file. This is using MPD as a player. What additional info do you need to troubleshoot this? Is it an intrinsic limitation to the interface? What additional tests should I do? It's possible it's a hardware issue but how do I divide them so I can go back to the hardware guy if it's his issue? Demian Martin Product Design Services

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[alsa-devel] Turtle Beach Tahiti (snd-msnd-classic)
by Brand, Mark (Mr) (Summerstrand Campus South) 06 Sep '10

06 Sep '10

Dear Takashi Iwai, Krzysztof Helt, Karsten Wiese and all, Following advice given on the bugtracker, and noting the call "testers wanted" in the Turtle Beach section of the ALSA soundcard matrix: I have been unable to get my Tahiti soundcard working under ALSA (see issue # 5118). Loading the ALSA driver for a Turtle Beach Tahiti fails quietly, with the following anomalies: 1. dmesg says "Assuming PnP mode", despite isapnp=0 option 2. dmesg makes no further mention of the driver (resource assignment, firmware upload, DSP reset, attach?). 3. /proc/interrupts, /proc/ioports and /proc/iomem show no resource assignments 4. aplay -l lists no soundcards 5. YET: lsmod shows driver has been loaded. Most importantly, no sound. To reproduce:

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[alsa-devel] [PATCH - Fix Fast Track Ultra series: 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix Homann 03 Sep '10

03 Sep '10

This fixes issues with M-Audio's Fast Track Ultra series: * devices now work at 44100 and 88200 Hz * apparently, fixes some crashes, too This is based on a patch posted by user 'dhubsith' on the M-Audio user forum. Sorry for not sending via 'git send-email': I just don't get it... Kind regards, Felix

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[alsa-devel] [RFC] ASoC: multi-component: Add optional kcontrol prefix name for a DAI link
by Jarkko Nikula 03 Sep '10

03 Sep '10

This optional kcontrol_prefix allows to specify unique prefix for ALSA control names for each DAI link. This makes possible to have a sound card configuration with multiple DAIs and each of them using the same codec driver without name collision. Currently name collision would occur if a codec driver tries to register a kcontrol with a same name for another DAI link. Now it is possible to specify for instance "Front" and "Rear" prefixes and a sound card can have two separate controls like "Front.PCM Playback Volume" and "Rear.PCM Playback Volume". Those controls will then show as "Front.PCM" and "Rear.PCM" in ALSA mixer application. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 13 +++++++++++-- sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++------ 3 files changed, 35 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d31e8b7..9b83f7e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -423,6 +423,7 @@ struct snd_soc_ops { /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; + const char *kcontrol_prefix; int id; struct device *dev; struct snd_soc_codec_driver *driver; @@ -539,6 +540,7 @@ struct snd_soc_dai_link { const char *platform_name; /* for multi-platform */ const char *cpu_dai_name; const char *codec_dai_name; + const char *kcontrol_prefix; /* kcontrol prefix for multi-codec */ /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a004876..f52eb2a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1289,6 +1289,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* config components */ codec_dai->codec = codec; codec->card = card; + codec->kcontrol_prefix = dai_link->kcontrol_prefix; cpu_dai->platform = platform; rtd->card = card; rtd->dev.parent = card->dev; @@ -1903,14 +1904,22 @@ int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; + char prefixed_name[44], *name; int err, i; for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; - err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL)); + if (codec->kcontrol_prefix) { + snprintf(prefixed_name, sizeof(prefixed_name), "%s.%s", + codec->kcontrol_prefix, control->name); + name = prefixed_name; + } else { + name = control->name; + } + err = snd_ctl_add(card, snd_soc_cnew(control, codec, name)); if (err < 0) { dev_err(codec->dev, "%s: Failed to add %s\n", - codec->name, control->name); + codec->name, name); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 035cab8..5751f4f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -327,6 +327,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, int i, ret = 0; size_t name_len; struct snd_soc_dapm_path *path; + char prefix[10]; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -347,6 +348,13 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, name_len = strlen(w->kcontrols[i].name) + 1; if (w->id != snd_soc_dapm_mixer_named_ctl) name_len += 1 + strlen(w->name); + if (codec->kcontrol_prefix) { + name_len += 1 + strlen(codec->kcontrol_prefix); + snprintf(prefix, sizeof(prefix), "%s.", + codec->kcontrol_prefix); + } else { + prefix[0] = '\0'; + } path->long_name = kmalloc(name_len, GFP_KERNEL); @@ -355,12 +363,12 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, switch (w->id) { default: - snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); + snprintf(path->long_name, name_len, "%s%s %s", + prefix, w->name, w->kcontrols[i].name); break; case snd_soc_dapm_mixer_named_ctl: - snprintf(path->long_name, name_len, "%s", - w->kcontrols[i].name); + snprintf(path->long_name, name_len, "%s%s", + prefix, w->kcontrols[i].name); break; } @@ -388,6 +396,7 @@ static int dapm_new_mux(struct snd_soc_codec *codec, { struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; + char prefixed_name[44], *name; int ret = 0; if (!w->num_kcontrols) { @@ -395,7 +404,14 @@ static int dapm_new_mux(struct snd_soc_codec *codec, return -EINVAL; } - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); + if (codec->kcontrol_prefix) { + snprintf(prefixed_name, sizeof(prefixed_name), "%s.%s", + codec->kcontrol_prefix, w->name); + name = prefixed_name; + } else { + name = w->name; + } + kcontrol = snd_soc_cnew(&w->kcontrols[0], w, name); ret = snd_ctl_add(codec->card->snd_card, kcontrol); if (ret < 0) goto err; @@ -406,7 +422,7 @@ static int dapm_new_mux(struct snd_soc_codec *codec, return ret; err: - printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name); + printk(KERN_ERR "asoc: failed to add kcontrol %s\n", name); return ret; } -- 1.7.1

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[alsa-devel] [PATCH] ALSA: HDA: Fix front mic on Dell Precision M6500
by David Henningsson 03 Sep '10

03 Sep '10

The BIOS of Dell Precision M6500 lists both 0x0e and 0x11 as front mics, and the driver only supports one front mic. This commit adds a quirk for preferring node 0x11 (internal mic) to node 0x0e (possibly docking station mic). Alsa-info reference: http://launchpadlibrarian.net/54232083/alsa-info.txt.xyFVpIysHt -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH 2/4] sound/soc/fsl/p1022_ds.c: Add of_node_put to avoid memory leak
by Julia Lawall 02 Sep '10

02 Sep '10

Add a call to of_node_put in the error handling code following a call to of_parse_phandle. This patch also moves the existing call to of_node_put tothe end of the error handling code, to make it possible to jump to of_node_put without doing the other cleanup operations. These appear to be disjoint operations, so the ordering doesn't matter. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r exists@ local idexpression x; expression E,E1,E2; statement S; @@ *x = (of_find_node_by_path |of_find_node_by_name |of_find_node_by_phandle |of_get_parent |of_get_next_parent |of_get_next_child |of_find_compatible_node |of_match_node |of_find_node_by_type |of_find_node_with_property |of_find_matching_node |of_parse_phandle )(...); ... if (x == NULL) S <... when != x = E *if (...) { ... when != of_node_put(x) when != if (...) { ... of_node_put(x); ... } ( return <+...x...+>; | * return ...; ) } ...> ( E2 = x; | of_node_put(x); ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/p1022_ds.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index f8176e8..63b9eaa 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -346,8 +346,10 @@ static int p1022_ds_probe(struct platform_device *pdev) } mdata = kzalloc(sizeof(struct machine_data), GFP_KERNEL); - if (!mdata) - return -ENOMEM; + if (!mdata) { + ret = -ENOMEM; + goto error_put; + } mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); mdata->dai[0].ops = &p1022_ds_ops; @@ -502,13 +504,12 @@ static int p1022_ds_probe(struct platform_device *pdev) return 0; error: - of_node_put(codec_np); - if (sound_device) platform_device_unregister(sound_device); kfree(mdata); - +error_put: + of_node_put(codec_np); return ret; }

4 4

[alsa-devel] defaults.pcm.minperiodtime 5000
by Raymond Yau 02 Sep '10

02 Sep '10

It seem that there is a bug in the patch which remove "if ( err < 0 )" - if (err < 0) return err; + if (pcm->minperiodtime > 0) { http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=09879a4bb58199f6… BTW , it seem this patch change the behaviour of alsa-lib too. all sound cards will now have a minperiodtime 5000

2 3

[alsa-devel] Missing 4-channel support on older NVIDIA MCP HW
by Stephen Warren 02 Sep '10

02 Sep '10

It looks like even though some NVIDIA MCPs have min/max channel of 2/8, not all HW supports the intermediate # channels (4 and 6) over HDMI. Various combinations are supported on various HW: 2, 2/8, 2/6/8, 2/4/6/8. At present, when an application uses an unsupported number of channels, playback appears to operate correctly, but the HW doesn't actually send the audio data over HDMI. Is it possible for patch_nvhdmi.c to simply program codec the HW in 8-channel mode even though the controller is only sending 4-/6-channel data? I'm not sure if this would cause the codec to get out of sync with the controller's data stream. Would the controller end up grabbing 8 channels worth of data from the stream at a time, have no way to synchronize to the start of each sample, and hence end up packing e.g. 2 complete 4 channel samples into a single 8 channel sample? If not, it seems that patch_nvhdmi.c should be modified so that each codec's _open() function returns an error for unsupported rates. I can code that up if we need. Unfortunately, it doesn't look like it's possible for applications to query this information from ALSA, since a hw_params_t exposes just a min/max channel count rather than a mask. -- nvpublic

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